From 780e2806986f9cc980808687da95160c65baa78a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Mar 2011 18:00:19 +0000 Subject: ASoC: Treat WM8958 revision A as WM8994 revision D The first WM8958 revision requires similar treatment. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 36 ++++++++++++++++++++++++++---------- 1 file changed, 26 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3dc64c8b6a5..9458289bb56 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3327,14 +3327,23 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, - ARRAY_SIZE(wm8994_lateclk_widgets)); - snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, - ARRAY_SIZE(wm8994_adc_widgets)); - snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, - ARRAY_SIZE(wm8994_dac_widgets)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); + if (wm8994->revision < 1) { + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, + ARRAY_SIZE(wm8994_adc_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets, + ARRAY_SIZE(wm8994_dac_revd_widgets)); + } else { + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); + } break; } @@ -3358,10 +3367,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: - snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, - ARRAY_SIZE(wm8994_lateclk_intercon)); - snd_soc_dapm_add_routes(dapm, wm8958_intercon, - ARRAY_SIZE(wm8958_intercon)); + if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + snd_soc_dapm_add_routes(dapm, wm8958_intercon, + ARRAY_SIZE(wm8958_intercon)); + } break; } -- cgit v1.2.3 From f701a2e594e62b35d895ad5ec1db8d2d0714c158 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Mar 2011 19:31:01 +0000 Subject: ASoC: Factor WM8958 DSP2 handling into separate file DSP2 on the WM8958 has a default ROM which provides a multi-band compressor for enhanced performance on mobile devices but can also support runtime download of alternative firmware. In preparation for more exploiting this functionality refactor the code to split the handling of DSP2 into a separate file. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wm8958-dsp2.c | 289 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.c | 305 +---------------------------------------- sound/soc/codecs/wm8994.h | 71 ++++++++++ 4 files changed, 364 insertions(+), 303 deletions(-) create mode 100644 sound/soc/codecs/wm8958-dsp2.c (limited to 'sound') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 379bc55f072..49121ad0e17 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -69,7 +69,7 @@ snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o -snd-soc-wm8994-objs := wm8994.o wm8994-tables.o +snd-soc-wm8994-objs := wm8994.o wm8994-tables.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9705-objs := wm9705.o diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c new file mode 100644 index 00000000000..4b4c93c2033 --- /dev/null +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -0,0 +1,289 @@ +/* + * wm8958-dsp2.c -- WM8958 DSP2 support + * + * Copyright 2011 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "wm8994.h" + +static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int pwr_reg = snd_soc_read(codec, WM8994_POWER_MANAGEMENT_5); + int ena, reg, aif, i; + + switch (mbc) { + case 0: + pwr_reg &= (WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA); + aif = 0; + break; + case 1: + pwr_reg &= (WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA); + aif = 0; + break; + case 2: + pwr_reg &= (WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA); + aif = 1; + break; + default: + BUG(); + return; + } + + /* We can only enable the MBC if the AIF is enabled and we + * want it to be enabled. */ + ena = pwr_reg && wm8994->mbc_ena[mbc]; + + reg = snd_soc_read(codec, WM8958_DSP2_PROGRAM); + + dev_dbg(codec->dev, "MBC %d startup: %d, power: %x, DSP: %x\n", + mbc, start, pwr_reg, reg); + + if (start && ena) { + /* If the DSP is already running then noop */ + if (reg & WM8958_DSP2_ENA) + return; + + /* Switch the clock over to the appropriate AIF */ + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8958_DSP2CLK_SRC | WM8958_DSP2CLK_ENA, + aif << WM8958_DSP2CLK_SRC_SHIFT | + WM8958_DSP2CLK_ENA); + + snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, + WM8958_DSP2_ENA, WM8958_DSP2_ENA); + + /* If we've got user supplied MBC settings use them */ + if (pdata && pdata->num_mbc_cfgs) { + struct wm8958_mbc_cfg *cfg + = &pdata->mbc_cfgs[wm8994->mbc_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->coeff_regs); i++) + snd_soc_write(codec, i + WM8958_MBC_BAND_1_K_1, + cfg->coeff_regs[i]); + + for (i = 0; i < ARRAY_SIZE(cfg->cutoff_regs); i++) + snd_soc_write(codec, + i + WM8958_MBC_BAND_2_LOWER_CUTOFF_C1_1, + cfg->cutoff_regs[i]); + } + + /* Run the DSP */ + snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, + WM8958_DSP2_RUNR); + + /* And we're off! */ + snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, + WM8958_MBC_ENA | WM8958_MBC_SEL_MASK, + mbc << WM8958_MBC_SEL_SHIFT | + WM8958_MBC_ENA); + } else { + /* If the DSP is already stopped then noop */ + if (!(reg & WM8958_DSP2_ENA)) + return; + + snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, + WM8958_MBC_ENA, 0); + snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, + WM8958_DSP2_ENA, 0); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8958_DSP2CLK_ENA, 0); + } +} + +int wm8958_aif_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + int mbc; + + switch (w->shift) { + case 13: + case 12: + mbc = 2; + break; + case 11: + case 10: + mbc = 1; + break; + case 9: + case 8: + mbc = 0; + break; + default: + BUG(); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + wm8958_mbc_apply(codec, mbc, 1); + break; + case SND_SOC_DAPM_POST_PMD: + wm8958_mbc_apply(codec, mbc, 0); + break; + } + + return 0; +} + +static int wm8958_put_mbc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int value = ucontrol->value.integer.value[0]; + int reg; + + /* Don't allow on the fly reconfiguration */ + reg = snd_soc_read(codec, WM8994_CLOCKING_1); + if (reg < 0 || reg & WM8958_DSP2CLK_ENA) + return -EBUSY; + + if (value >= pdata->num_mbc_cfgs) + return -EINVAL; + + wm8994->mbc_cfg = value; + + return 0; +} + +static int wm8958_get_mbc_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = wm8994->mbc_cfg; + + return 0; +} + +static int wm8958_mbc_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int wm8958_mbc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int mbc = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = wm8994->mbc_ena[mbc]; + + return 0; +} + +static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int mbc = kcontrol->private_value; + int i; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (ucontrol->value.integer.value[0] > 1) + return -EINVAL; + + for (i = 0; i < ARRAY_SIZE(wm8994->mbc_ena); i++) { + if (mbc != i && wm8994->mbc_ena[i]) { + dev_dbg(codec->dev, "MBC %d active already\n", mbc); + return -EBUSY; + } + } + + wm8994->mbc_ena[mbc] = ucontrol->value.integer.value[0]; + + wm8958_mbc_apply(codec, mbc, wm8994->mbc_ena[mbc]); + + return 0; +} + +#define WM8958_MBC_SWITCH(xname, xval) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .info = wm8958_mbc_info, \ + .get = wm8958_mbc_get, .put = wm8958_mbc_put, \ + .private_value = xval } + +static const struct snd_kcontrol_new wm8958_mbc_snd_controls[] = { +WM8958_MBC_SWITCH("AIF1DAC1 MBC Switch", 0), +WM8958_MBC_SWITCH("AIF1DAC2 MBC Switch", 1), +WM8958_MBC_SWITCH("AIF2DAC MBC Switch", 2), +}; + +void wm8958_dsp2_init(struct snd_soc_codec *codec) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int ret, i; + + snd_soc_add_controls(codec, wm8958_mbc_snd_controls, + ARRAY_SIZE(wm8958_mbc_snd_controls)); + + if (!pdata) + return; + + if (pdata->num_mbc_cfgs) { + struct snd_kcontrol_new control[] = { + SOC_ENUM_EXT("MBC Mode", wm8994->mbc_enum, + wm8958_get_mbc_enum, wm8958_put_mbc_enum), + }; + + /* We need an array of texts for the enum API */ + wm8994->mbc_texts = kmalloc(sizeof(char *) + * pdata->num_mbc_cfgs, GFP_KERNEL); + if (!wm8994->mbc_texts) { + dev_err(wm8994->codec->dev, + "Failed to allocate %d MBC config texts\n", + pdata->num_mbc_cfgs); + return; + } + + for (i = 0; i < pdata->num_mbc_cfgs; i++) + wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name; + + wm8994->mbc_enum.max = pdata->num_mbc_cfgs; + wm8994->mbc_enum.texts = wm8994->mbc_texts; + + ret = snd_soc_add_controls(wm8994->codec, control, 1); + if (ret != 0) + dev_err(wm8994->codec->dev, + "Failed to add MBC mode controls: %d\n", ret); + } + + +} diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9458289bb56..96e1379f4fa 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -38,12 +38,6 @@ #include "wm8994.h" #include "wm_hubs.h" -struct fll_config { - int src; - int in; - int out; -}; - #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 @@ -59,61 +53,6 @@ static int wm8994_retune_mobile_base[] = { WM8994_AIF2_EQ_GAINS_1, }; -struct wm8994_micdet { - struct snd_soc_jack *jack; - int det; - int shrt; -}; - -/* codec private data */ -struct wm8994_priv { - struct wm_hubs_data hubs; - enum snd_soc_control_type control_type; - void *control_data; - struct snd_soc_codec *codec; - int sysclk[2]; - int sysclk_rate[2]; - int mclk[2]; - int aifclk[2]; - struct fll_config fll[2], fll_suspend[2]; - - int dac_rates[2]; - int lrclk_shared[2]; - - int mbc_ena[3]; - - /* Platform dependant DRC configuration */ - const char **drc_texts; - int drc_cfg[WM8994_NUM_DRC]; - struct soc_enum drc_enum; - - /* Platform dependant ReTune mobile configuration */ - int num_retune_mobile_texts; - const char **retune_mobile_texts; - int retune_mobile_cfg[WM8994_NUM_EQ]; - struct soc_enum retune_mobile_enum; - - /* Platform dependant MBC configuration */ - int mbc_cfg; - const char **mbc_texts; - struct soc_enum mbc_enum; - - struct wm8994_micdet micdet[2]; - - wm8958_micdet_cb jack_cb; - void *jack_cb_data; - int micdet_irq; - - int revision; - struct wm8994_pdata *pdata; - - unsigned int aif1clk_enable:1; - unsigned int aif2clk_enable:1; - - unsigned int aif1clk_disable:1; - unsigned int aif2clk_disable:1; -}; - static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { @@ -574,215 +513,6 @@ static const struct soc_enum dac_osr = static const struct soc_enum adc_osr = SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text); -static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) -{ - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994_pdata *pdata = wm8994->pdata; - int pwr_reg = snd_soc_read(codec, WM8994_POWER_MANAGEMENT_5); - int ena, reg, aif, i; - - switch (mbc) { - case 0: - pwr_reg &= (WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA); - aif = 0; - break; - case 1: - pwr_reg &= (WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA); - aif = 0; - break; - case 2: - pwr_reg &= (WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA); - aif = 1; - break; - default: - BUG(); - return; - } - - /* We can only enable the MBC if the AIF is enabled and we - * want it to be enabled. */ - ena = pwr_reg && wm8994->mbc_ena[mbc]; - - reg = snd_soc_read(codec, WM8958_DSP2_PROGRAM); - - dev_dbg(codec->dev, "MBC %d startup: %d, power: %x, DSP: %x\n", - mbc, start, pwr_reg, reg); - - if (start && ena) { - /* If the DSP is already running then noop */ - if (reg & WM8958_DSP2_ENA) - return; - - /* Switch the clock over to the appropriate AIF */ - snd_soc_update_bits(codec, WM8994_CLOCKING_1, - WM8958_DSP2CLK_SRC | WM8958_DSP2CLK_ENA, - aif << WM8958_DSP2CLK_SRC_SHIFT | - WM8958_DSP2CLK_ENA); - - snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, - WM8958_DSP2_ENA, WM8958_DSP2_ENA); - - /* If we've got user supplied MBC settings use them */ - if (pdata && pdata->num_mbc_cfgs) { - struct wm8958_mbc_cfg *cfg - = &pdata->mbc_cfgs[wm8994->mbc_cfg]; - - for (i = 0; i < ARRAY_SIZE(cfg->coeff_regs); i++) - snd_soc_write(codec, i + WM8958_MBC_BAND_1_K_1, - cfg->coeff_regs[i]); - - for (i = 0; i < ARRAY_SIZE(cfg->cutoff_regs); i++) - snd_soc_write(codec, - i + WM8958_MBC_BAND_2_LOWER_CUTOFF_C1_1, - cfg->cutoff_regs[i]); - } - - /* Run the DSP */ - snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, - WM8958_DSP2_RUNR); - - /* And we're off! */ - snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, - WM8958_MBC_ENA | WM8958_MBC_SEL_MASK, - mbc << WM8958_MBC_SEL_SHIFT | - WM8958_MBC_ENA); - } else { - /* If the DSP is already stopped then noop */ - if (!(reg & WM8958_DSP2_ENA)) - return; - - snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, - WM8958_MBC_ENA, 0); - snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, - WM8958_DSP2_ENA, 0); - snd_soc_update_bits(codec, WM8994_CLOCKING_1, - WM8958_DSP2CLK_ENA, 0); - } -} - -static int wm8958_aif_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - int mbc; - - switch (w->shift) { - case 13: - case 12: - mbc = 2; - break; - case 11: - case 10: - mbc = 1; - break; - case 9: - case 8: - mbc = 0; - break; - default: - BUG(); - return -EINVAL; - } - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - wm8958_mbc_apply(codec, mbc, 1); - break; - case SND_SOC_DAPM_POST_PMD: - wm8958_mbc_apply(codec, mbc, 0); - break; - } - - return 0; -} - -static int wm8958_put_mbc_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994_pdata *pdata = wm8994->pdata; - int value = ucontrol->value.integer.value[0]; - int reg; - - /* Don't allow on the fly reconfiguration */ - reg = snd_soc_read(codec, WM8994_CLOCKING_1); - if (reg < 0 || reg & WM8958_DSP2CLK_ENA) - return -EBUSY; - - if (value >= pdata->num_mbc_cfgs) - return -EINVAL; - - wm8994->mbc_cfg = value; - - return 0; -} - -static int wm8958_get_mbc_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - - ucontrol->value.enumerated.item[0] = wm8994->mbc_cfg; - - return 0; -} - -static int wm8958_mbc_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} - -static int wm8958_mbc_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int mbc = kcontrol->private_value; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - - ucontrol->value.integer.value[0] = wm8994->mbc_ena[mbc]; - - return 0; -} - -static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int mbc = kcontrol->private_value; - int i; - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - - if (ucontrol->value.integer.value[0] > 1) - return -EINVAL; - - for (i = 0; i < ARRAY_SIZE(wm8994->mbc_ena); i++) { - if (mbc != i && wm8994->mbc_ena[i]) { - dev_dbg(codec->dev, "MBC %d active already\n", mbc); - return -EBUSY; - } - } - - wm8994->mbc_ena[mbc] = ucontrol->value.integer.value[0]; - - wm8958_mbc_apply(codec, mbc, wm8994->mbc_ena[mbc]); - - return 0; -} - -#define WM8958_MBC_SWITCH(xname, xval) {\ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,\ - .info = wm8958_mbc_info, \ - .get = wm8958_mbc_get, .put = wm8958_mbc_put, \ - .private_value = xval } - static const struct snd_kcontrol_new wm8994_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1_ADC1_RIGHT_VOLUME, @@ -924,9 +654,6 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), -WM8958_MBC_SWITCH("AIF1DAC1 MBC Switch", 0), -WM8958_MBC_SWITCH("AIF1DAC2 MBC Switch", 1), -WM8958_MBC_SWITCH("AIF2DAC MBC Switch", 2), }; static int clk_sys_event(struct snd_soc_dapm_widget *w, @@ -2676,7 +2403,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], - sizeof(struct fll_config)); + sizeof(struct wm8994_fll_config)); ret = _wm8994_set_fll(codec, i + 1, 0, 0, 0); if (ret < 0) dev_warn(codec->dev, "Failed to stop FLL%d: %d\n", @@ -2862,34 +2589,6 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", pdata->num_retune_mobile_cfgs); - if (pdata->num_mbc_cfgs) { - struct snd_kcontrol_new control[] = { - SOC_ENUM_EXT("MBC Mode", wm8994->mbc_enum, - wm8958_get_mbc_enum, wm8958_put_mbc_enum), - }; - - /* We need an array of texts for the enum API */ - wm8994->mbc_texts = kmalloc(sizeof(char *) - * pdata->num_mbc_cfgs, GFP_KERNEL); - if (!wm8994->mbc_texts) { - dev_err(wm8994->codec->dev, - "Failed to allocate %d MBC config texts\n", - pdata->num_mbc_cfgs); - return; - } - - for (i = 0; i < pdata->num_mbc_cfgs; i++) - wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name; - - wm8994->mbc_enum.max = pdata->num_mbc_cfgs; - wm8994->mbc_enum.texts = wm8994->mbc_texts; - - ret = snd_soc_add_controls(wm8994->codec, control, 1); - if (ret != 0) - dev_err(wm8994->codec->dev, - "Failed to add MBC mode controls: %d\n", ret); - } - if (pdata->num_retune_mobile_cfgs) wm8994_handle_retune_mobile_pdata(wm8994); else @@ -3378,6 +3077,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8958_intercon, ARRAY_SIZE(wm8958_intercon)); } + + wm8958_dsp2_init(codec); break; } diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 999b8851226..93a6cf1e130 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -11,6 +11,8 @@ #include +#include "wm_hubs.h" + /* Sources for AIF1/2 SYSCLK - use with set_dai_sysclk() */ #define WM8994_SYSCLK_MCLK1 1 #define WM8994_SYSCLK_MCLK2 2 @@ -45,4 +47,73 @@ struct wm8994_access_mask { extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE]; extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; +int wm8958_aif_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); + +void wm8958_dsp2_init(struct snd_soc_codec *codec); + +struct wm8994_micdet { + struct snd_soc_jack *jack; + int det; + int shrt; +}; + +/* codec private data */ +struct wm8994_fll_config { + int src; + int in; + int out; +}; + +#define WM8994_NUM_DRC 3 +#define WM8994_NUM_EQ 3 + +struct wm8994_priv { + struct wm_hubs_data hubs; + enum snd_soc_control_type control_type; + void *control_data; + struct snd_soc_codec *codec; + int sysclk[2]; + int sysclk_rate[2]; + int mclk[2]; + int aifclk[2]; + struct wm8994_fll_config fll[2], fll_suspend[2]; + + int dac_rates[2]; + int lrclk_shared[2]; + + int mbc_ena[3]; + + /* Platform dependant DRC configuration */ + const char **drc_texts; + int drc_cfg[WM8994_NUM_DRC]; + struct soc_enum drc_enum; + + /* Platform dependant ReTune mobile configuration */ + int num_retune_mobile_texts; + const char **retune_mobile_texts; + int retune_mobile_cfg[WM8994_NUM_EQ]; + struct soc_enum retune_mobile_enum; + + /* Platform dependant MBC configuration */ + int mbc_cfg; + const char **mbc_texts; + struct soc_enum mbc_enum; + + struct wm8994_micdet micdet[2]; + + wm8958_micdet_cb jack_cb; + void *jack_cb_data; + int micdet_irq; + + int revision; + struct wm8994_pdata *pdata; + + unsigned int aif1clk_enable:1; + unsigned int aif2clk_enable:1; + + unsigned int aif1clk_disable:1; + unsigned int aif2clk_disable:1; +}; + #endif -- cgit v1.2.3 From c6b7b57012c8307b936025ad41fcbef6b1b6b52f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Mar 2011 18:13:12 +0000 Subject: ASoC: Handle startup sequencing of WM8958 DSP2 with deferred clocking The DSP2 startup requires that the clock be enable so if we've deferred clock startup we need to defer DSP2 startup Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8958-dsp2.c | 35 ++++++++++++++--------------------- sound/soc/codecs/wm8994.c | 3 +++ 2 files changed, 17 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 4b4c93c2033..f07775ebec0 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -69,6 +69,13 @@ static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) if (reg & WM8958_DSP2_ENA) return; + /* If neither AIFnCLK is not yet enabled postpone */ + if (!(snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) + & WM8994_AIF1CLK_ENA_MASK) && + !(snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) + & WM8994_AIF2CLK_ENA_MASK)) + return; + /* Switch the clock over to the appropriate AIF */ snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8958_DSP2CLK_SRC | WM8958_DSP2CLK_ENA, @@ -120,32 +127,18 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - int mbc; - - switch (w->shift) { - case 13: - case 12: - mbc = 2; - break; - case 11: - case 10: - mbc = 1; - break; - case 9: - case 8: - mbc = 0; - break; - default: - BUG(); - return -EINVAL; - } + int i; switch (event) { case SND_SOC_DAPM_POST_PMU: - wm8958_mbc_apply(codec, mbc, 1); + case SND_SOC_DAPM_PRE_PMU: + for (i = 0; i < 3; i++) + wm8958_mbc_apply(codec, i, 1); break; case SND_SOC_DAPM_POST_PMD: - wm8958_mbc_apply(codec, mbc, 0); + case SND_SOC_DAPM_PRE_PMD: + for (i = 0; i < 3; i++) + wm8958_mbc_apply(codec, i, 0); break; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 96e1379f4fa..328f3283194 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -759,6 +759,9 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, break; } + /* We may also have postponed startup of DSP, handle that. */ + wm8958_aif_ev(w, kcontrol, event); + return 0; } -- cgit v1.2.3 From af9af866020ea341aca32123b3109b6a9408dd8c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Mar 2011 21:05:06 +0000 Subject: ASoC: Mark WM8958 DSP2 registers readable Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 328f3283194..bdd1ac75178 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -55,6 +55,9 @@ static int wm8994_retune_mobile_base[] = { static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->control_data; + switch (reg) { case WM8994_GPIO_1: case WM8994_GPIO_2: @@ -71,6 +74,15 @@ static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) case WM8994_INTERRUPT_STATUS_2: case WM8994_INTERRUPT_RAW_STATUS_2: return 1; + + case WM8958_DSP2_PROGRAM: + case WM8958_DSP2_CONFIG: + case WM8958_DSP2_EXECCONTROL: + if (control->type == WM8958) + return 1; + else + return 0; + default: break; } -- cgit v1.2.3 From 375d135818f32bbe7b3f071bd54d977c4ff8d84a Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sat, 19 Mar 2011 16:32:53 +0100 Subject: ALSA: tea575x-tuner: various improvements Improve tea575x-tuner with various good things from radio-maestro: - extend frequency range to 50-150MHz - fix querycap(): card name, CAP_RADIO - improve g_tuner(): CAP_STEREO, stereo and tuned indication - improve g_frequency(): tuner index checking and reading frequency from HW - improve s_frequency(): tuner index and type checking Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/i2c/other/tea575x-tuner.c | 52 ++++++++++++++++++++++++++++++----------- 1 file changed, 38 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index ee538f1ae84..9f35f378a17 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -37,8 +37,8 @@ static int radio_nr = -1; module_param(radio_nr, int, 0); #define RADIO_VERSION KERNEL_VERSION(0, 0, 2) -#define FREQ_LO (87 * 16000) -#define FREQ_HI (108 * 16000) +#define FREQ_LO (50UL * 16000) +#define FREQ_HI (150UL * 16000) /* * definitions @@ -77,15 +77,29 @@ static struct v4l2_queryctrl radio_qctrl[] = { * lowlevel part */ +static void snd_tea575x_get_freq(struct snd_tea575x *tea) +{ + unsigned long freq; + + freq = tea->ops->read(tea) & TEA575X_BIT_FREQ_MASK; + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + if (tea->tea5759) + freq += tea->freq_fixup; + else + freq -= tea->freq_fixup; + + tea->freq = freq * 16; /* from kHz */ +} + static void snd_tea575x_set_freq(struct snd_tea575x *tea) { unsigned long freq; - freq = tea->freq / 16; /* to kHz */ - if (freq > 108000) - freq = 108000; - if (freq < 87000) - freq = 87000; + freq = clamp(tea->freq, FREQ_LO, FREQ_HI); + freq /= 16; /* to kHz */ /* crystal fixup */ if (tea->tea5759) freq -= tea->freq_fixup; @@ -109,29 +123,33 @@ static int vidioc_querycap(struct file *file, void *priv, { struct snd_tea575x *tea = video_drvdata(file); - strcpy(v->card, tea->tea5759 ? "TEA5759" : "TEA5757"); strlcpy(v->driver, "tea575x-tuner", sizeof(v->driver)); - strlcpy(v->card, "Maestro Radio", sizeof(v->card)); + strlcpy(v->card, tea->tea5759 ? "TEA5759" : "TEA5757", sizeof(v->card)); sprintf(v->bus_info, "PCI"); v->version = RADIO_VERSION; - v->capabilities = V4L2_CAP_TUNER; + v->capabilities = V4L2_CAP_TUNER | V4L2_CAP_RADIO; return 0; } static int vidioc_g_tuner(struct file *file, void *priv, struct v4l2_tuner *v) { + struct snd_tea575x *tea = video_drvdata(file); + if (v->index > 0) return -EINVAL; + tea->ops->read(tea); + strcpy(v->name, "FM"); v->type = V4L2_TUNER_RADIO; + v->capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO; v->rangelow = FREQ_LO; v->rangehigh = FREQ_HI; - v->rxsubchans = V4L2_TUNER_SUB_MONO|V4L2_TUNER_SUB_STEREO; - v->capability = V4L2_TUNER_CAP_LOW; - v->audmode = V4L2_TUNER_MODE_MONO; - v->signal = 0xffff; + v->rxsubchans = V4L2_TUNER_SUB_MONO | V4L2_TUNER_SUB_STEREO; + v->audmode = tea->stereo ? V4L2_TUNER_MODE_STEREO : V4L2_TUNER_MODE_MONO; + v->signal = tea->tuned ? 0xffff : 0; + return 0; } @@ -148,7 +166,10 @@ static int vidioc_g_frequency(struct file *file, void *priv, { struct snd_tea575x *tea = video_drvdata(file); + if (f->tuner != 0) + return -EINVAL; f->type = V4L2_TUNER_RADIO; + snd_tea575x_get_freq(tea); f->frequency = tea->freq; return 0; } @@ -158,6 +179,9 @@ static int vidioc_s_frequency(struct file *file, void *priv, { struct snd_tea575x *tea = video_drvdata(file); + if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO) + return -EINVAL; + if (f->frequency < FREQ_LO || f->frequency > FREQ_HI) return -EINVAL; -- cgit v1.2.3 From f8960d61bc8ba945b07a4de1288aac5d52f8607b Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sat, 19 Mar 2011 16:33:01 +0100 Subject: ALSA: tea575x-tuner: remove dev_nr Remove unused dev_nr from struct tea575x_tuner. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index e1baad74ea4..f4dc1c77202 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1453,7 +1453,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, #ifdef TEA575X_RADIO if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { - chip->tea.dev_nr = tea575x_tuner >> 16; chip->tea.card = card; chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; -- cgit v1.2.3 From 1872f589951caee1afd7cd2ea6729ac892de9ddf Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sat, 19 Mar 2011 16:33:14 +0100 Subject: ALSA: es1968: add radio (tea575x tuner) support Add TEA5757 radio tuner support to es1968 driver. This is found at least on MediaForte SF64-PCE2 sound cards. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/i2c/other/Makefile | 2 +- sound/pci/Kconfig | 14 ++++-- sound/pci/es1968.c | 125 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 137 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile index 2dad40f3f62..c95d8f1aae8 100644 --- a/sound/i2c/other/Makefile +++ b/sound/i2c/other/Makefile @@ -14,4 +14,4 @@ snd-tea575x-tuner-objs := tea575x-tuner.o obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o obj-$(CONFIG_SND_ICE1712) += snd-ak4xxx-adda.o obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4113.o snd-ak4xxx-adda.o snd-pt2258.o -obj-$(CONFIG_SND_FM801_TEA575X) += snd-tea575x-tuner.o +obj-$(CONFIG_SND_TEA575X) += snd-tea575x-tuner.o diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 389cd793166..f9f533fcccc 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -534,6 +534,14 @@ config SND_ES1968_INPUT If you say N the buttons will directly control the master volume. It is recommended to say Y. +config SND_ES1968_RADIO + bool "Enable TEA5757 radio tuner support for es1968" + depends on SND_ES1968 + depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_ES1968 + help + Say Y here to include support for TEA5757 radio tuner integrated on + some MediaForte cards (e.g. SF64-PCE2). + config SND_FM801 tristate "ForteMedia FM801" select SND_OPL3_LIB @@ -555,10 +563,10 @@ config SND_FM801_TEA575X_BOOL FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media Forte SF256-PCS-02) into the snd-fm801 driver. -config SND_FM801_TEA575X +config SND_TEA575X tristate - depends on SND_FM801_TEA575X_BOOL - default SND_FM801 + depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO + default SND_FM801 || SND_ES1968 source "sound/pci/hda/Kconfig" diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 7c17f45d876..faf9138970a 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -112,6 +112,10 @@ #include #include +#ifdef CONFIG_SND_ES1968_RADIO +#include +#endif + #define CARD_NAME "ESS Maestro1/2" #define DRIVER_NAME "ES1968" @@ -553,6 +557,10 @@ struct es1968 { spinlock_t ac97_lock; struct tasklet_struct hwvol_tq; #endif + +#ifdef CONFIG_SND_ES1968_RADIO + struct snd_tea575x tea; +#endif }; static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); @@ -2571,6 +2579,111 @@ static int __devinit snd_es1968_input_register(struct es1968 *chip) } #endif /* CONFIG_SND_ES1968_INPUT */ +#ifdef CONFIG_SND_ES1968_RADIO +#define GPIO_DATA 0x60 +#define IO_MASK 4 /* mask register offset from GPIO_DATA + bits 1=unmask write to given bit */ +#define IO_DIR 8 /* direction register offset from GPIO_DATA + bits 0/1=read/write direction */ +/* mask bits for GPIO lines */ +#define STR_DATA 0x0040 /* GPIO6 */ +#define STR_CLK 0x0080 /* GPIO7 */ +#define STR_WREN 0x0100 /* GPIO8 */ +#define STR_MOST 0x0200 /* GPIO9 */ + +static void snd_es1968_tea575x_write(struct snd_tea575x *tea, unsigned int val) +{ + struct es1968 *chip = tea->private_data; + unsigned long io = chip->io_port + GPIO_DATA; + u16 l, bits; + u16 omask, odir; + + omask = inw(io + IO_MASK); + odir = (inw(io + IO_DIR) & ~STR_DATA) | (STR_CLK | STR_WREN); + outw(odir | STR_DATA, io + IO_DIR); + outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); + udelay(16); + + for (l = 25; l; l--) { + bits = ((val >> 18) & STR_DATA) | STR_WREN; + val <<= 1; /* shift data */ + outw(bits, io); /* start strobe */ + udelay(2); + outw(bits | STR_CLK, io); /* HI level */ + udelay(2); + outw(bits, io); /* LO level */ + udelay(4); + } + + if (!tea->mute) + outw(0, io); + + udelay(4); + outw(omask, io + IO_MASK); + outw(odir, io + IO_DIR); + msleep(125); +} + +static unsigned int snd_es1968_tea575x_read(struct snd_tea575x *tea) +{ + struct es1968 *chip = tea->private_data; + unsigned long io = chip->io_port + GPIO_DATA; + u16 l, rdata; + u32 data = 0; + u16 omask; + + omask = inw(io + IO_MASK); + outw(~(STR_CLK | STR_WREN), io + IO_MASK); + outw(0, io); + udelay(16); + + for (l = 24; l--;) { + outw(STR_CLK, io); /* HI state */ + udelay(2); + if (!l) + tea->tuned = inw(io) & STR_MOST ? 0 : 1; + outw(0, io); /* LO state */ + udelay(2); + data <<= 1; /* shift data */ + rdata = inw(io); + if (!l) + tea->stereo = (rdata & STR_MOST) ? 0 : 1; + else if (l && rdata & STR_DATA) + data++; + udelay(2); + } + + if (tea->mute) + outw(STR_WREN, io); + + udelay(4); + outw(omask, io + IO_MASK); + + return data & 0x3ffe; +} + +static void snd_es1968_tea575x_mute(struct snd_tea575x *tea, unsigned int mute) +{ + struct es1968 *chip = tea->private_data; + unsigned long io = chip->io_port + GPIO_DATA; + u16 omask; + + omask = inw(io + IO_MASK); + outw(~STR_WREN, io + IO_MASK); + tea->mute = mute; + outw(tea->mute ? STR_WREN : 0, io); + udelay(4); + outw(omask, io + IO_MASK); + msleep(125); +} + +static struct snd_tea575x_ops snd_es1968_tea_ops = { + .write = snd_es1968_tea575x_write, + .read = snd_es1968_tea575x_read, + .mute = snd_es1968_tea575x_mute, +}; +#endif + static int snd_es1968_free(struct es1968 *chip) { #ifdef CONFIG_SND_ES1968_INPUT @@ -2585,6 +2698,10 @@ static int snd_es1968_free(struct es1968 *chip) outw(0, chip->io_port + ESM_PORT_HOST_IRQ); /* disable IRQ */ } +#ifdef CONFIG_SND_ES1968_RADIO + snd_tea575x_exit(&chip->tea); +#endif + if (chip->irq >= 0) free_irq(chip->irq, chip); snd_es1968_free_gameport(chip); @@ -2723,6 +2840,14 @@ static int __devinit snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); +#ifdef CONFIG_SND_ES1968_RADIO + chip->tea.card = card; + chip->tea.freq_fixup = 10700; + chip->tea.private_data = chip; + chip->tea.ops = &snd_es1968_tea_ops; + snd_tea575x_init(&chip->tea); +#endif + *chip_ret = chip; return 0; -- cgit v1.2.3 From 66b5b9722b8743f83d4c3f11f39665f5f2c40b12 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 16 Mar 2011 12:16:39 +0000 Subject: ALSA: Add snd_ctl_replace() to dynamically replace a control Add a function to dynamically replace a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or replace existing controls. A separate patch needs to be made to allow ALSA Mixer to render the replaced controls on the fly. Signed-off-by: Dimitris Papastamos Signed-off-by: Takashi Iwai --- sound/core/control.c | 64 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 64 insertions(+) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index a08ad57c49b..5d98194bcad 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -365,6 +365,70 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_add); +/** + * snd_ctl_replace - replace the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to replace + * @add_on_replace: add the control if not already added + * + * Replaces the given control. If the given control does not exist + * and the add_on_replace flag is set, the control is added. If the + * control exists, it is destroyed first. + * + * Returns zero if successful, or a negative error code on failure. + * + * It frees automatically the control which cannot be added or replaced. + */ +int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, + bool add_on_replace) +{ + struct snd_ctl_elem_id id; + unsigned int idx; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) { + ret = -EINVAL; + goto error; + } + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + if (add_on_replace) + goto add; + up_write(&card->controls_rwsem); + ret = -EINVAL; + goto error; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) { + up_write(&card->controls_rwsem); + goto error; + } +add: + if (snd_ctl_find_hole(card, kcontrol->count) < 0) { + up_write(&card->controls_rwsem); + ret = -ENOMEM; + goto error; + } + list_add_tail(&kcontrol->list, &card->controls); + card->controls_count += kcontrol->count; + kcontrol->id.numid = card->last_numid + 1; + card->last_numid += kcontrol->count; + up_write(&card->controls_rwsem); + for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + return 0; + +error: + snd_ctl_free_one(kcontrol); + return ret; +} +EXPORT_SYMBOL(snd_ctl_replace); + /** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance -- cgit v1.2.3 From fbbf592002ee46ed14d5bd88f1150c604b34e705 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Mar 2011 18:09:04 +0000 Subject: ASoC: Support download of WM8958 MBC firmware Allow userspace to supply an update to the ROM firmware. The firmware request is non-blocking so userspace can load the firmware at its leisure without delaying startup, the driver will begin using the firmware the next time MBC is started after it has been supplied. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8958-dsp2.c | 223 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.c | 4 + sound/soc/codecs/wm8994.h | 4 + 3 files changed, 231 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index f07775ebec0..58fe4041670 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -30,10 +30,212 @@ #include "wm8994.h" +#define WM_FW_BLOCK_INFO 0xff +#define WM_FW_BLOCK_PM 0x00 +#define WM_FW_BLOCK_X 0x01 +#define WM_FW_BLOCK_Y 0x02 +#define WM_FW_BLOCK_Z 0x03 +#define WM_FW_BLOCK_I 0x06 +#define WM_FW_BLOCK_A 0x08 +#define WM_FW_BLOCK_C 0x0c + +static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, + const struct firmware *fw, bool check) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + u64 data64; + u32 data32; + const u8 *data; + char *str; + size_t block_len, len; + int ret = 0; + + /* Suppress unneeded downloads */ + if (wm8994->cur_fw == fw) + return 0; + + if (fw->size < 32) { + dev_err(codec->dev, "%s: firmware too short\n", name); + goto err; + } + + if (memcmp(fw->data, "WMFW", 4) != 0) { + dev_err(codec->dev, "%s: firmware has bad file magic %08x\n", + name, data32); + goto err; + } + + memcpy(&data32, fw->data + 4, sizeof(data32)); + len = be32_to_cpu(data32); + + memcpy(&data32, fw->data + 8, sizeof(data32)); + data32 = be32_to_cpu(data32); + if ((data32 >> 24) & 0xff) { + dev_err(codec->dev, "%s: unsupported firmware version %d\n", + name, (data32 >> 24) & 0xff); + goto err; + } + if ((data32 & 0xffff) != 8958) { + dev_err(codec->dev, "%s: unsupported target device %d\n", + name, data32 & 0xffff); + goto err; + } + if (((data32 >> 16) & 0xff) != 0xc) { + dev_err(codec->dev, "%s: unsupported target core %d\n", + name, (data32 >> 16) & 0xff); + goto err; + } + + if (check) { + memcpy(&data64, fw->data + 24, sizeof(u64)); + dev_info(codec->dev, "%s timestamp %llx\n", + name, be64_to_cpu(data64)); + } else { + snd_soc_write(codec, 0x102, 0x2); + snd_soc_write(codec, 0x900, 0x2); + } + + data = fw->data + len; + len = fw->size - len; + while (len) { + if (len < 12) { + dev_err(codec->dev, "%s short data block of %d\n", + name, len); + goto err; + } + + memcpy(&data32, data + 4, sizeof(data32)); + block_len = be32_to_cpu(data32); + if (block_len + 8 > len) { + dev_err(codec->dev, "%d byte block longer than file\n", + block_len); + goto err; + } + if (block_len == 0) { + dev_err(codec->dev, "Zero length block\n"); + goto err; + } + + memcpy(&data32, data, sizeof(data32)); + data32 = be32_to_cpu(data32); + + switch ((data32 >> 24) & 0xff) { + case WM_FW_BLOCK_INFO: + /* Informational text */ + if (!check) + break; + + str = kzalloc(block_len + 1, GFP_KERNEL); + if (str) { + memcpy(str, data + 8, block_len); + dev_info(codec->dev, "%s: %s\n", name, str); + kfree(str); + } else { + dev_err(codec->dev, "Out of memory\n"); + } + break; + case WM_FW_BLOCK_PM: + case WM_FW_BLOCK_X: + case WM_FW_BLOCK_Y: + case WM_FW_BLOCK_Z: + case WM_FW_BLOCK_I: + case WM_FW_BLOCK_A: + case WM_FW_BLOCK_C: + dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name, + block_len, (data32 >> 24) & 0xff, + data32 & 0xffffff); + + if (check) + break; + + data32 &= 0xffffff; + + wm8994_bulk_write(codec->control_data, + data32 & 0xffffff, + block_len / 2, + (void *)(data + 8)); + + break; + default: + dev_warn(codec->dev, "%s: unknown block type %d\n", + name, (data32 >> 24) & 0xff); + break; + } + + /* Round up to the next 32 bit word */ + block_len += block_len % 4; + + data += block_len + 8; + len -= block_len + 8; + } + + if (!check) { + dev_dbg(codec->dev, "%s: download done\n", name); + wm8994->cur_fw = fw; + } else { + dev_info(codec->dev, "%s: got firmware\n", name); + } + + goto ok; + +err: + ret = -EINVAL; +ok: + if (!check) { + snd_soc_write(codec, 0x900, 0x0); + snd_soc_write(codec, 0x102, 0x0); + } + + return ret; +} + static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994_pdata *pdata = wm8994->pdata; + int i; + + /* If the DSP is already running then noop */ + if (snd_soc_read(codec, WM8958_DSP2_PROGRAM) & WM8958_DSP2_ENA) + return; + + /* If we have MBC firmware download it */ + if (wm8994->mbc) + wm8958_dsp2_fw(codec, "MBC", wm8994->mbc, false); + + snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, + WM8958_DSP2_ENA, WM8958_DSP2_ENA); + + /* If we've got user supplied MBC settings use them */ + if (pdata && pdata->num_mbc_cfgs) { + struct wm8958_mbc_cfg *cfg + = &pdata->mbc_cfgs[wm8994->mbc_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->coeff_regs); i++) + snd_soc_write(codec, i + WM8958_MBC_BAND_1_K_1, + cfg->coeff_regs[i]); + + for (i = 0; i < ARRAY_SIZE(cfg->cutoff_regs); i++) + snd_soc_write(codec, + i + WM8958_MBC_BAND_2_LOWER_CUTOFF_C1_1, + cfg->cutoff_regs[i]); + } + + /* Run the DSP */ + snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, + WM8958_DSP2_RUNR); + + /* And we're off! */ + snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, + WM8958_MBC_ENA | + WM8958_MBC_SEL_MASK, + path << WM8958_MBC_SEL_SHIFT | + WM8958_MBC_ENA); +} + +static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int pwr_reg = snd_soc_read(codec, WM8994_POWER_MANAGEMENT_5); int ena, reg, aif, i; @@ -76,6 +278,10 @@ static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) & WM8994_AIF2CLK_ENA_MASK)) return; + /* If we have MBC firmware download it */ + if (wm8994->mbc && wm8994->mbc_ena[mbc]) + wm8958_dsp2_fw(codec, "MBC", wm8994->mbc, false); + /* Switch the clock over to the appropriate AIF */ snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8958_DSP2CLK_SRC | WM8958_DSP2CLK_ENA, @@ -238,6 +444,18 @@ WM8958_MBC_SWITCH("AIF1DAC2 MBC Switch", 1), WM8958_MBC_SWITCH("AIF2DAC MBC Switch", 2), }; +static void wm8958_mbc_loaded(const struct firmware *fw, void *context) +{ + struct snd_soc_codec *codec = context; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (fw && wm8958_dsp2_fw(codec, "MBC", fw, true) != 0) { + mutex_lock(&codec->mutex); + wm8994->mbc = fw; + mutex_unlock(&codec->mutex); + } +} + void wm8958_dsp2_init(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -247,6 +465,11 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8958_mbc_snd_controls, ARRAY_SIZE(wm8958_mbc_snd_controls)); + /* We don't require firmware and don't want to delay boot */ + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_mbc.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_mbc_loaded); + if (!pdata) return; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bdd1ac75178..f622ff691b4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1922,6 +1922,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); + wm8994->cur_fw = NULL; + pm_runtime_put(codec->dev); } break; @@ -3136,6 +3138,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) free_irq(wm8994->micdet_irq, wm8994); break; } + if (wm8994->mbc) + release_firmware(wm8994->mbc); kfree(wm8994->retune_mobile_texts); kfree(wm8994->drc_texts); kfree(wm8994); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 93a6cf1e130..1aa365b31b0 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -10,6 +10,7 @@ #define _WM8994_H #include +#include #include "wm_hubs.h" @@ -114,6 +115,9 @@ struct wm8994_priv { unsigned int aif1clk_disable:1; unsigned int aif2clk_disable:1; + + const struct firmware *cur_fw; + const struct firmware *mbc; }; #endif -- cgit v1.2.3 From f20d77ce2663b31c2994462d9ab9143726b67f3e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Mar 2011 20:55:37 +0000 Subject: ASoC: Refactor WM8958 DSP to support additional algorithms In preparation for the addition of additional WM8958 algorithms reorganise the code to make it easier to add such support later. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8958-dsp2.c | 96 +++++++++++++++++++----------------------- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 44 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 58fe4041670..9c1cbe5b61a 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -189,7 +189,7 @@ ok: return ret; } -static void wm8958_mbc_apply(struct snd_soc_codec *codec, int mbc, int start) +static void wm8958_dsp_start_mbc(struct snd_soc_codec *codec, int path) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994_pdata *pdata = wm8994->pdata; @@ -237,9 +237,9 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int pwr_reg = snd_soc_read(codec, WM8994_POWER_MANAGEMENT_5); - int ena, reg, aif, i; + int ena, reg, aif; - switch (mbc) { + switch (path) { case 0: pwr_reg &= (WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA); aif = 0; @@ -257,64 +257,34 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) return; } - /* We can only enable the MBC if the AIF is enabled and we - * want it to be enabled. */ - ena = pwr_reg && wm8994->mbc_ena[mbc]; + /* Do we have both an active AIF and an active algorithm? */ + ena = wm8994->mbc_ena[path]; + if (!pwr_reg) + ena = 0; reg = snd_soc_read(codec, WM8958_DSP2_PROGRAM); - dev_dbg(codec->dev, "MBC %d startup: %d, power: %x, DSP: %x\n", - mbc, start, pwr_reg, reg); + dev_dbg(codec->dev, "DSP path %d %d startup: %d, power: %x, DSP: %x\n", + path, wm8994->dsp_active, start, pwr_reg, reg); if (start && ena) { - /* If the DSP is already running then noop */ - if (reg & WM8958_DSP2_ENA) - return; - - /* If neither AIFnCLK is not yet enabled postpone */ + /* If either AIFnCLK is not yet enabled postpone */ if (!(snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA_MASK) && !(snd_soc_read(codec, WM8994_AIF2_CLOCKING_1) & WM8994_AIF2CLK_ENA_MASK)) return; - /* If we have MBC firmware download it */ - if (wm8994->mbc && wm8994->mbc_ena[mbc]) - wm8958_dsp2_fw(codec, "MBC", wm8994->mbc, false); - /* Switch the clock over to the appropriate AIF */ snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8958_DSP2CLK_SRC | WM8958_DSP2CLK_ENA, aif << WM8958_DSP2CLK_SRC_SHIFT | WM8958_DSP2CLK_ENA); - snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, - WM8958_DSP2_ENA, WM8958_DSP2_ENA); - - /* If we've got user supplied MBC settings use them */ - if (pdata && pdata->num_mbc_cfgs) { - struct wm8958_mbc_cfg *cfg - = &pdata->mbc_cfgs[wm8994->mbc_cfg]; - - for (i = 0; i < ARRAY_SIZE(cfg->coeff_regs); i++) - snd_soc_write(codec, i + WM8958_MBC_BAND_1_K_1, - cfg->coeff_regs[i]); - - for (i = 0; i < ARRAY_SIZE(cfg->cutoff_regs); i++) - snd_soc_write(codec, - i + WM8958_MBC_BAND_2_LOWER_CUTOFF_C1_1, - cfg->cutoff_regs[i]); - } - - /* Run the DSP */ - snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, - WM8958_DSP2_RUNR); + if (wm8994->mbc_ena[path]) + wm8958_dsp_start_mbc(codec, path); - /* And we're off! */ - snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, - WM8958_MBC_ENA | WM8958_MBC_SEL_MASK, - mbc << WM8958_MBC_SEL_SHIFT | - WM8958_MBC_ENA); + dev_dbg(codec->dev, "DSP running\n"); } else { /* If the DSP is already stopped then noop */ if (!(reg & WM8958_DSP2_ENA)) @@ -322,10 +292,16 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, WM8958_MBC_ENA, 0); + snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, + WM8958_DSP2_STOP); snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, WM8958_DSP2_ENA, 0); snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8958_DSP2CLK_ENA, 0); + + wm8994->dsp_active = -1; + + dev_dbg(codec->dev, "DSP stopped\n"); } } @@ -339,18 +315,33 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: for (i = 0; i < 3; i++) - wm8958_mbc_apply(codec, i, 1); + wm8958_dsp_apply(codec, i, 1); break; case SND_SOC_DAPM_POST_PMD: case SND_SOC_DAPM_PRE_PMD: for (i = 0; i < 3; i++) - wm8958_mbc_apply(codec, i, 0); + wm8958_dsp_apply(codec, i, 0); break; } return 0; } +/* Check if DSP2 is in use on another AIF */ +static int wm8958_dsp2_busy(struct wm8994_priv *wm8994, int aif) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994->mbc_ena); i++) { + if (i == aif) + continue; + if (wm8994->mbc_ena[i]) + return 1; + } + + return 0; +} + static int wm8958_put_mbc_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -410,23 +401,20 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int mbc = kcontrol->private_value; - int i; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); if (ucontrol->value.integer.value[0] > 1) return -EINVAL; - for (i = 0; i < ARRAY_SIZE(wm8994->mbc_ena); i++) { - if (mbc != i && wm8994->mbc_ena[i]) { - dev_dbg(codec->dev, "MBC %d active already\n", mbc); - return -EBUSY; - } + if (wm8958_dsp2_busy(wm8994, mbc)) { + dev_dbg(codec->dev, "DSP2 active on %d already\n", mbc); + return -EBUSY; } wm8994->mbc_ena[mbc] = ucontrol->value.integer.value[0]; - wm8958_mbc_apply(codec, mbc, wm8994->mbc_ena[mbc]); + wm8958_dsp_apply(codec, mbc, wm8994->mbc_ena[mbc]); return 0; } @@ -462,10 +450,12 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) struct wm8994_pdata *pdata = wm8994->pdata; int ret, i; + wm8994->dsp_active = -1; + snd_soc_add_controls(codec, wm8958_mbc_snd_controls, ARRAY_SIZE(wm8958_mbc_snd_controls)); - /* We don't require firmware and don't want to delay boot */ + /* We don't *require* firmware and don't want to delay boot */ request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, "wm8958_mbc.wfw", codec->dev, GFP_KERNEL, codec, wm8958_mbc_loaded); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 1aa365b31b0..a4bfde83065 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -116,6 +116,7 @@ struct wm8994_priv { unsigned int aif1clk_disable:1; unsigned int aif2clk_disable:1; + int dsp_active; const struct firmware *cur_fw; const struct firmware *mbc; }; -- cgit v1.2.3 From 09e10d7fe509408d15818db6a0299f563668a7ba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Mar 2011 22:57:47 +0000 Subject: ASoC: Add WM8958 VSS support With appropriate firmware the WM8958 can support Virtual Surround Sound or VSS, widening the stereo audio image for improved user experience. Enable support for this mode of operation when the appropriate firmware can be loaded at runtime. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8958-dsp2.c | 363 ++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8994.c | 2 + sound/soc/codecs/wm8994.h | 14 ++ 3 files changed, 371 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 9c1cbe5b61a..d0e257315d9 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -233,6 +233,68 @@ static void wm8958_dsp_start_mbc(struct snd_soc_codec *codec, int path) WM8958_MBC_ENA); } +static void wm8958_dsp_start_vss(struct snd_soc_codec *codec, int path) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int i, ena; + + if (wm8994->mbc_vss) + wm8958_dsp2_fw(codec, "MBC+VSS", wm8994->mbc_vss, false); + + snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, + WM8958_DSP2_ENA, WM8958_DSP2_ENA); + + /* If we've got user supplied settings use them */ + if (pdata && pdata->num_mbc_cfgs) { + struct wm8958_mbc_cfg *cfg + = &pdata->mbc_cfgs[wm8994->mbc_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->combined_regs); i++) + snd_soc_write(codec, i + 0x2800, + cfg->combined_regs[i]); + } + + if (pdata && pdata->num_vss_cfgs) { + struct wm8958_vss_cfg *cfg + = &pdata->vss_cfgs[wm8994->vss_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->regs); i++) + snd_soc_write(codec, i + 0x2600, cfg->regs[i]); + } + + if (pdata && pdata->num_vss_hpf_cfgs) { + struct wm8958_vss_hpf_cfg *cfg + = &pdata->vss_hpf_cfgs[wm8994->vss_hpf_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->regs); i++) + snd_soc_write(codec, i + 0x2400, cfg->regs[i]); + } + + /* Run the DSP */ + snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, + WM8958_DSP2_RUNR); + + /* Enable the algorithms we've selected */ + ena = 0; + if (wm8994->mbc_ena[path]) + ena |= 0x8; + if (wm8994->hpf2_ena[path]) + ena |= 0x4; + if (wm8994->hpf1_ena[path]) + ena |= 0x2; + if (wm8994->vss_ena[path]) + ena |= 0x1; + + snd_soc_write(codec, 0x2201, ena); + + /* Switch the DSP into the data path */ + snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, + WM8958_MBC_SEL_MASK | WM8958_MBC_ENA, + path << WM8958_MBC_SEL_SHIFT | WM8958_MBC_ENA); +} + + static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -258,7 +320,8 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) } /* Do we have both an active AIF and an active algorithm? */ - ena = wm8994->mbc_ena[path]; + ena = wm8994->mbc_ena[path] || wm8994->vss_ena[path] || + wm8994->hpf1_ena[path] || wm8994->hpf2_ena[path]; if (!pwr_reg) ena = 0; @@ -281,11 +344,18 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) aif << WM8958_DSP2CLK_SRC_SHIFT | WM8958_DSP2CLK_ENA); - if (wm8994->mbc_ena[path]) + if (wm8994->vss_ena[path] || wm8994->hpf1_ena[path] || + wm8994->hpf2_ena[path]) + wm8958_dsp_start_vss(codec, path); + else if (wm8994->mbc_ena[path]) wm8958_dsp_start_mbc(codec, path); - dev_dbg(codec->dev, "DSP running\n"); - } else { + wm8994->dsp_active = path; + + dev_dbg(codec->dev, "DSP running in path %d\n", path); + } + + if (!start && wm8994->dsp_active == path) { /* If the DSP is already stopped then noop */ if (!(reg & WM8958_DSP2_ENA)) return; @@ -335,7 +405,8 @@ static int wm8958_dsp2_busy(struct wm8994_priv *wm8994, int aif) for (i = 0; i < ARRAY_SIZE(wm8994->mbc_ena); i++) { if (i == aif) continue; - if (wm8994->mbc_ena[i]) + if (wm8994->mbc_ena[i] || wm8994->vss_ena[i] || + wm8994->hpf1_ena[i] || wm8994->hpf2_ena[i]) return 1; } @@ -426,22 +497,239 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, .get = wm8958_mbc_get, .put = wm8958_mbc_put, \ .private_value = xval } +static int wm8958_put_vss_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int value = ucontrol->value.integer.value[0]; + int reg; + + /* Don't allow on the fly reconfiguration */ + reg = snd_soc_read(codec, WM8994_CLOCKING_1); + if (reg < 0 || reg & WM8958_DSP2CLK_ENA) + return -EBUSY; + + if (value >= pdata->num_vss_cfgs) + return -EINVAL; + + wm8994->vss_cfg = value; + + return 0; +} + +static int wm8958_get_vss_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = wm8994->vss_cfg; + + return 0; +} + +static int wm8958_put_vss_hpf_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int value = ucontrol->value.integer.value[0]; + int reg; + + /* Don't allow on the fly reconfiguration */ + reg = snd_soc_read(codec, WM8994_CLOCKING_1); + if (reg < 0 || reg & WM8958_DSP2CLK_ENA) + return -EBUSY; + + if (value >= pdata->num_vss_hpf_cfgs) + return -EINVAL; + + wm8994->vss_hpf_cfg = value; + + return 0; +} + +static int wm8958_get_vss_hpf_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = wm8994->vss_hpf_cfg; + + return 0; +} + +static int wm8958_vss_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int wm8958_vss_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int vss = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = wm8994->vss_ena[vss]; + + return 0; +} + +static int wm8958_vss_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int vss = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (ucontrol->value.integer.value[0] > 1) + return -EINVAL; + + if (!wm8994->mbc_vss) + return -ENODEV; + + if (wm8958_dsp2_busy(wm8994, vss)) { + dev_dbg(codec->dev, "DSP2 active on %d already\n", vss); + return -EBUSY; + } + + wm8994->vss_ena[vss] = ucontrol->value.integer.value[0]; + + wm8958_dsp_apply(codec, vss, wm8994->vss_ena[vss]); + + return 0; +} + + +#define WM8958_VSS_SWITCH(xname, xval) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .info = wm8958_vss_info, \ + .get = wm8958_vss_get, .put = wm8958_vss_put, \ + .private_value = xval } + +static int wm8958_hpf_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int wm8958_hpf_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int hpf = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (hpf < 3) + ucontrol->value.integer.value[0] = wm8994->hpf1_ena[hpf % 3]; + else + ucontrol->value.integer.value[0] = wm8994->hpf2_ena[hpf % 3]; + + return 0; +} + +static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int hpf = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (ucontrol->value.integer.value[0] > 1) + return -EINVAL; + + if (!wm8994->mbc_vss) + return -ENODEV; + + if (wm8958_dsp2_busy(wm8994, hpf % 3)) { + dev_dbg(codec->dev, "DSP2 active on %d already\n", hpf); + return -EBUSY; + } + + if (wm8994->eq[hpf % 3]) + return -EBUSY; + + if (hpf < 3) + wm8994->hpf1_ena[hpf % 3] = ucontrol->value.integer.value[0]; + else + wm8994->hpf2_ena[hpf % 3] = ucontrol->value.integer.value[0]; + + wm8958_dsp_apply(codec, hpf % 3, ucontrol->value.integer.value[0]); + + return 0; +} + +#define WM8958_HPF_SWITCH(xname, xval) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .info = wm8958_hpf_info, \ + .get = wm8958_hpf_get, .put = wm8958_hpf_put, \ + .private_value = xval } + static const struct snd_kcontrol_new wm8958_mbc_snd_controls[] = { WM8958_MBC_SWITCH("AIF1DAC1 MBC Switch", 0), WM8958_MBC_SWITCH("AIF1DAC2 MBC Switch", 1), WM8958_MBC_SWITCH("AIF2DAC MBC Switch", 2), }; -static void wm8958_mbc_loaded(const struct firmware *fw, void *context) +static const struct snd_kcontrol_new wm8958_vss_snd_controls[] = { +WM8958_VSS_SWITCH("AIF1DAC1 VSS Switch", 0), +WM8958_VSS_SWITCH("AIF1DAC2 VSS Switch", 1), +WM8958_VSS_SWITCH("AIF2DAC VSS Switch", 2), +WM8958_HPF_SWITCH("AIF1DAC1 HPF1 Switch", 0), +WM8958_HPF_SWITCH("AIF1DAC2 HPF1 Switch", 1), +WM8958_HPF_SWITCH("AIF2DAC HPF1 Switch", 2), +WM8958_HPF_SWITCH("AIF1DAC1 HPF2 Switch", 3), +WM8958_HPF_SWITCH("AIF1DAC2 HPF2 Switch", 4), +WM8958_HPF_SWITCH("AIF2DAC HPF2 Switch", 5), +}; + +static void wm8958_mbc_vss_loaded(const struct firmware *fw, void *context) { struct snd_soc_codec *codec = context; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - if (fw && wm8958_dsp2_fw(codec, "MBC", fw, true) != 0) { + if (fw && (wm8958_dsp2_fw(codec, "MBC+VSS", fw, true) == 0)) { mutex_lock(&codec->mutex); - wm8994->mbc = fw; + wm8994->mbc_vss = fw; mutex_unlock(&codec->mutex); } + +} + +static void wm8958_mbc_loaded(const struct firmware *fw, void *context) +{ + struct snd_soc_codec *codec = context; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (wm8958_dsp2_fw(codec, "MBC", fw, true) != 0) + return; + + mutex_lock(&codec->mutex); + wm8994->mbc = fw; + mutex_unlock(&codec->mutex); + + /* We can't have more than one request outstanding at once so + * we daisy chain. + */ + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_mbc_vss.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_mbc_vss_loaded); } void wm8958_dsp2_init(struct snd_soc_codec *codec) @@ -454,6 +742,9 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8958_mbc_snd_controls, ARRAY_SIZE(wm8958_mbc_snd_controls)); + snd_soc_add_controls(codec, wm8958_vss_snd_controls, + ARRAY_SIZE(wm8958_vss_snd_controls)); + /* We don't *require* firmware and don't want to delay boot */ request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, @@ -491,5 +782,61 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) "Failed to add MBC mode controls: %d\n", ret); } + if (pdata->num_vss_cfgs) { + struct snd_kcontrol_new control[] = { + SOC_ENUM_EXT("VSS Mode", wm8994->vss_enum, + wm8958_get_vss_enum, wm8958_put_vss_enum), + }; + /* We need an array of texts for the enum API */ + wm8994->vss_texts = kmalloc(sizeof(char *) + * pdata->num_vss_cfgs, GFP_KERNEL); + if (!wm8994->vss_texts) { + dev_err(wm8994->codec->dev, + "Failed to allocate %d VSS config texts\n", + pdata->num_vss_cfgs); + return; + } + + for (i = 0; i < pdata->num_vss_cfgs; i++) + wm8994->vss_texts[i] = pdata->vss_cfgs[i].name; + + wm8994->vss_enum.max = pdata->num_vss_cfgs; + wm8994->vss_enum.texts = wm8994->vss_texts; + + ret = snd_soc_add_controls(wm8994->codec, control, 1); + if (ret != 0) + dev_err(wm8994->codec->dev, + "Failed to add VSS mode controls: %d\n", ret); + } + + if (pdata->num_vss_hpf_cfgs) { + struct snd_kcontrol_new control[] = { + SOC_ENUM_EXT("VSS HPF Mode", wm8994->vss_hpf_enum, + wm8958_get_vss_hpf_enum, + wm8958_put_vss_hpf_enum), + }; + + /* We need an array of texts for the enum API */ + wm8994->vss_hpf_texts = kmalloc(sizeof(char *) + * pdata->num_vss_hpf_cfgs, GFP_KERNEL); + if (!wm8994->vss_hpf_texts) { + dev_err(wm8994->codec->dev, + "Failed to allocate %d VSS HPF config texts\n", + pdata->num_vss_hpf_cfgs); + return; + } + + for (i = 0; i < pdata->num_vss_hpf_cfgs; i++) + wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name; + + wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs; + wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts; + + ret = snd_soc_add_controls(wm8994->codec, control, 1); + if (ret != 0) + dev_err(wm8994->codec->dev, + "Failed to add VSS HPFmode controls: %d\n", + ret); + } } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index f622ff691b4..01ef5704091 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3140,6 +3140,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) } if (wm8994->mbc) release_firmware(wm8994->mbc); + if (wm8994->mbc_vss) + release_firmware(wm8994->mbc_vss); kfree(wm8994->retune_mobile_texts); kfree(wm8994->drc_texts); kfree(wm8994); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index a4bfde83065..f337f3d5059 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -84,6 +84,9 @@ struct wm8994_priv { int lrclk_shared[2]; int mbc_ena[3]; + int hpf1_ena[3]; + int hpf2_ena[3]; + int vss_ena[3]; /* Platform dependant DRC configuration */ const char **drc_texts; @@ -101,6 +104,16 @@ struct wm8994_priv { const char **mbc_texts; struct soc_enum mbc_enum; + /* Platform dependant VSS configuration */ + int vss_cfg; + const char **vss_texts; + struct soc_enum vss_enum; + + /* Platform dependant VSS HPF configuration */ + int vss_hpf_cfg; + const char **vss_hpf_texts; + struct soc_enum vss_hpf_enum; + struct wm8994_micdet micdet[2]; wm8958_micdet_cb jack_cb; @@ -119,6 +132,7 @@ struct wm8994_priv { int dsp_active; const struct firmware *cur_fw; const struct firmware *mbc; + const struct firmware *mbc_vss; }; #endif -- cgit v1.2.3 From 312158718fe2056703b2744801165a9574560495 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Mar 2011 20:23:43 +0000 Subject: ASoC: Add WM8958 enhanced EQ support DSP2 in the WM8958 can be used to support an upgraded EQ for use in demanding applications. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8958-dsp2.c | 192 ++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8994.c | 2 + sound/soc/codecs/wm8994.h | 7 ++ 3 files changed, 198 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index d0e257315d9..74983ee2b87 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -294,6 +294,36 @@ static void wm8958_dsp_start_vss(struct snd_soc_codec *codec, int path) path << WM8958_MBC_SEL_SHIFT | WM8958_MBC_ENA); } +static void wm8958_dsp_start_enh_eq(struct snd_soc_codec *codec, int path) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int i; + + wm8958_dsp2_fw(codec, "ENH_EQ", wm8994->enh_eq, false); + + snd_soc_update_bits(codec, WM8958_DSP2_PROGRAM, + WM8958_DSP2_ENA, WM8958_DSP2_ENA); + + /* If we've got user supplied settings use them */ + if (pdata && pdata->num_enh_eq_cfgs) { + struct wm8958_enh_eq_cfg *cfg + = &pdata->enh_eq_cfgs[wm8994->enh_eq_cfg]; + + for (i = 0; i < ARRAY_SIZE(cfg->regs); i++) + snd_soc_write(codec, i + 0x2200, + cfg->regs[i]); + } + + /* Run the DSP */ + snd_soc_write(codec, WM8958_DSP2_EXECCONTROL, + WM8958_DSP2_RUNR); + + /* Switch the DSP into the data path */ + snd_soc_update_bits(codec, WM8958_DSP2_CONFIG, + WM8958_MBC_SEL_MASK | WM8958_MBC_ENA, + path << WM8958_MBC_SEL_SHIFT | WM8958_MBC_ENA); +} static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) { @@ -321,7 +351,8 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) /* Do we have both an active AIF and an active algorithm? */ ena = wm8994->mbc_ena[path] || wm8994->vss_ena[path] || - wm8994->hpf1_ena[path] || wm8994->hpf2_ena[path]; + wm8994->hpf1_ena[path] || wm8994->hpf2_ena[path] || + wm8994->enh_eq_ena[path]; if (!pwr_reg) ena = 0; @@ -344,7 +375,9 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) aif << WM8958_DSP2CLK_SRC_SHIFT | WM8958_DSP2CLK_ENA); - if (wm8994->vss_ena[path] || wm8994->hpf1_ena[path] || + if (wm8994->enh_eq_ena[path]) + wm8958_dsp_start_enh_eq(codec, path); + else if (wm8994->vss_ena[path] || wm8994->hpf1_ena[path] || wm8994->hpf2_ena[path]) wm8958_dsp_start_vss(codec, path); else if (wm8994->mbc_ena[path]) @@ -483,6 +516,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, return -EBUSY; } + if (wm8994->enh_eq_ena[mbc]) + return -EBUSY; + wm8994->mbc_ena[mbc] = ucontrol->value.integer.value[0]; wm8958_dsp_apply(codec, mbc, wm8994->mbc_ena[mbc]); @@ -603,6 +639,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, return -EBUSY; } + if (wm8994->enh_eq_ena[vss]) + return -EBUSY; + wm8994->vss_ena[vss] = ucontrol->value.integer.value[0]; wm8958_dsp_apply(codec, vss, wm8994->vss_ena[vss]); @@ -661,7 +700,7 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, return -EBUSY; } - if (wm8994->eq[hpf % 3]) + if (wm8994->enh_eq_ena[hpf % 3]) return -EBUSY; if (hpf < 3) @@ -681,6 +720,97 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, .get = wm8958_hpf_get, .put = wm8958_hpf_put, \ .private_value = xval } +static int wm8958_put_enh_eq_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994_pdata *pdata = wm8994->pdata; + int value = ucontrol->value.integer.value[0]; + int reg; + + /* Don't allow on the fly reconfiguration */ + reg = snd_soc_read(codec, WM8994_CLOCKING_1); + if (reg < 0 || reg & WM8958_DSP2CLK_ENA) + return -EBUSY; + + if (value >= pdata->num_enh_eq_cfgs) + return -EINVAL; + + wm8994->enh_eq_cfg = value; + + return 0; +} + +static int wm8958_get_enh_eq_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = wm8994->enh_eq_cfg; + + return 0; +} + +static int wm8958_enh_eq_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int wm8958_enh_eq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int eq = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = wm8994->enh_eq_ena[eq]; + + return 0; +} + +static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int eq = kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (ucontrol->value.integer.value[0] > 1) + return -EINVAL; + + if (!wm8994->enh_eq) + return -ENODEV; + + if (wm8958_dsp2_busy(wm8994, eq)) { + dev_dbg(codec->dev, "DSP2 active on %d already\n", eq); + return -EBUSY; + } + + if (wm8994->mbc_ena[eq] || wm8994->vss_ena[eq] || + wm8994->hpf1_ena[eq] || wm8994->hpf2_ena[eq]) + return -EBUSY; + + wm8994->enh_eq_ena[eq] = ucontrol->value.integer.value[0]; + + wm8958_dsp_apply(codec, eq, ucontrol->value.integer.value[0]); + + return 0; +} + +#define WM8958_ENH_EQ_SWITCH(xname, xval) {\ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .info = wm8958_enh_eq_info, \ + .get = wm8958_enh_eq_get, .put = wm8958_enh_eq_put, \ + .private_value = xval } + static const struct snd_kcontrol_new wm8958_mbc_snd_controls[] = { WM8958_MBC_SWITCH("AIF1DAC1 MBC Switch", 0), WM8958_MBC_SWITCH("AIF1DAC2 MBC Switch", 1), @@ -699,6 +829,24 @@ WM8958_HPF_SWITCH("AIF1DAC2 HPF2 Switch", 4), WM8958_HPF_SWITCH("AIF2DAC HPF2 Switch", 5), }; +static const struct snd_kcontrol_new wm8958_enh_eq_snd_controls[] = { +WM8958_ENH_EQ_SWITCH("AIF1DAC1 Enhanced EQ Switch", 0), +WM8958_ENH_EQ_SWITCH("AIF1DAC2 Enhanced EQ Switch", 1), +WM8958_ENH_EQ_SWITCH("AIF2DAC Enhanced EQ Switch", 2), +}; + +static void wm8958_enh_eq_loaded(const struct firmware *fw, void *context) +{ + struct snd_soc_codec *codec = context; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + if (fw && (wm8958_dsp2_fw(codec, "ENH_EQ", fw, true) == 0)) { + mutex_lock(&codec->mutex); + wm8994->enh_eq = fw; + mutex_unlock(&codec->mutex); + } +} + static void wm8958_mbc_vss_loaded(const struct firmware *fw, void *context) { struct snd_soc_codec *codec = context; @@ -710,6 +858,12 @@ static void wm8958_mbc_vss_loaded(const struct firmware *fw, void *context) mutex_unlock(&codec->mutex); } + /* We can't have more than one request outstanding at once so + * we daisy chain. + */ + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_enh_eq.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_enh_eq_loaded); } static void wm8958_mbc_loaded(const struct firmware *fw, void *context) @@ -744,6 +898,8 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) ARRAY_SIZE(wm8958_mbc_snd_controls)); snd_soc_add_controls(codec, wm8958_vss_snd_controls, ARRAY_SIZE(wm8958_vss_snd_controls)); + snd_soc_add_controls(codec, wm8958_enh_eq_snd_controls, + ARRAY_SIZE(wm8958_enh_eq_snd_controls)); /* We don't *require* firmware and don't want to delay boot */ @@ -839,4 +995,34 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) "Failed to add VSS HPFmode controls: %d\n", ret); } + + if (pdata->num_enh_eq_cfgs) { + struct snd_kcontrol_new control[] = { + SOC_ENUM_EXT("Enhanced EQ Mode", wm8994->enh_eq_enum, + wm8958_get_enh_eq_enum, + wm8958_put_enh_eq_enum), + }; + + /* We need an array of texts for the enum API */ + wm8994->enh_eq_texts = kmalloc(sizeof(char *) + * pdata->num_enh_eq_cfgs, GFP_KERNEL); + if (!wm8994->enh_eq_texts) { + dev_err(wm8994->codec->dev, + "Failed to allocate %d enhanced EQ config texts\n", + pdata->num_enh_eq_cfgs); + return; + } + + for (i = 0; i < pdata->num_enh_eq_cfgs; i++) + wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name; + + wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs; + wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts; + + ret = snd_soc_add_controls(wm8994->codec, control, 1); + if (ret != 0) + dev_err(wm8994->codec->dev, + "Failed to add enhanced EQ controls: %d\n", + ret); + } } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 01ef5704091..03ef62462ba 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3142,6 +3142,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) release_firmware(wm8994->mbc); if (wm8994->mbc_vss) release_firmware(wm8994->mbc_vss); + if (wm8994->enh_eq) + release_firmware(wm8994->enh_eq); kfree(wm8994->retune_mobile_texts); kfree(wm8994->drc_texts); kfree(wm8994); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f337f3d5059..0a1db04b73b 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -87,6 +87,7 @@ struct wm8994_priv { int hpf1_ena[3]; int hpf2_ena[3]; int vss_ena[3]; + int enh_eq_ena[3]; /* Platform dependant DRC configuration */ const char **drc_texts; @@ -114,6 +115,11 @@ struct wm8994_priv { const char **vss_hpf_texts; struct soc_enum vss_hpf_enum; + /* Platform dependant enhanced EQ configuration */ + int enh_eq_cfg; + const char **enh_eq_texts; + struct soc_enum enh_eq_enum; + struct wm8994_micdet micdet[2]; wm8958_micdet_cb jack_cb; @@ -133,6 +139,7 @@ struct wm8994_priv { const struct firmware *cur_fw; const struct firmware *mbc; const struct firmware *mbc_vss; + const struct firmware *enh_eq; }; #endif -- cgit v1.2.3 From 3e95b9aba50deeee69afce87d2bcd94c4cd1d95e Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Wed, 23 Mar 2011 15:13:28 +0800 Subject: ALSA: hda - VIA: Add new power management function. Use set_widgets_power_state() function to seperately control different codecs' power management actions and to replace the original large function. Also fix some wrong widgets power up sequence which caused no sound issue under Smart5.1 mode and Independent HP mode. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 938 ++++++++++++++++++++++------------------------ 1 file changed, 453 insertions(+), 485 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 1371b57c11e..23cb44086b2 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -154,6 +154,9 @@ struct via_spec { struct delayed_work vt1708_hp_work; int vt1708_jack_detectect; int vt1708_hp_present; + + void (*set_widgets_power_state)(struct hda_codec *codec); + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -245,7 +248,6 @@ enum { }; static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); -static void set_jack_power_state(struct hda_codec *codec); static int is_aa_path_mute(struct hda_codec *codec); static void vt1708_start_hp_work(struct via_spec *spec) @@ -271,6 +273,12 @@ static void vt1708_stop_hp_work(struct via_spec *spec) cancel_delayed_work_sync(&spec->vt1708_hp_work); } +static void set_widgets_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + if (spec->set_widgets_power_state) + spec->set_widgets_power_state(codec); +} static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -278,7 +286,7 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol, int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { if (is_aa_path_mute(codec)) @@ -602,482 +610,6 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); } -static void set_jack_power_state(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - - if (spec->codec_type == VT1702) { - imux_is_smixer = snd_hda_codec_read( - codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x16, &parm); - set_pin_power_state(codec, 0x17, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, - parm); - } else if (spec->codec_type == VT1708B_8CH - || spec->codec_type == VT1708B_4CH - || spec->codec_type == VT1708S) { - /* SW0 (17h) = stereo mixer */ - int is_8ch = spec->codec_type != VT1708B_4CH; - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (spec->smart51_enabled) - parm = AC_PWRST_D0; - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1718S) { - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW0/1 (24h/25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - set_pin_power_state(codec, 0x25, &parm); - if (!spec->hp_independent_mode) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, - parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - if (spec->hp_independent_mode) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); - } - } else if (spec->codec_type == VT1716S) { - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, - parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - snd_hda_codec_write( - codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Smart 5.1 PW5(1eh) */ - if (spec->smart51_enabled) - set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->hp_independent_mode && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, - parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, - parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); - } else if (spec->codec_type == VT2002P) { - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, - 0, AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Class-D */ - /* PW0 (24h), MW0(18h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* Mono Out */ - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - present = snd_hda_jack_detect(codec, 0x26); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else { - snd_hda_codec_write( - codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write( - codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } else if (spec->codec_type == VT1812) { - unsigned int present; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = snd_hda_codec_read( - codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* outputs */ - /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); - if (spec->hp_independent_mode) { - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, parm); - } - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, - AC_VERB_SET_POWER_STATE, parm); - - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - else - snd_hda_codec_write( - codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } -} - /* * input MUX handling */ @@ -1120,7 +652,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return ret; } @@ -1225,7 +757,7 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, spec->hp_independent_mode); } /* update jack power state */ - set_jack_power_state(codec); + set_widgets_power_state(codec); return 0; } @@ -1443,7 +975,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, } } spec->smart51_enabled = *ucontrol->value.integer.value; - set_jack_power_state(codec); + set_widgets_power_state(codec); return 1; } @@ -1967,7 +1499,7 @@ static int via_build_controls(struct hda_codec *codec) } /* init power states */ - set_jack_power_state(codec); + set_widgets_power_state(codec); analog_low_current_mode(codec, 1); via_free_kctls(codec); /* no longer needed */ @@ -2195,7 +1727,7 @@ static void via_unsol_event(struct hda_codec *codec, if (res & VIA_GPIO_EVENT) via_gpio_control(codec); if (res & VIA_JACK_EVENT) - set_jack_power_state(codec); + set_widgets_power_state(codec); if (res & VIA_MONO_EVENT) via_mono_automute(codec); if (res & VIA_SPEAKER_EVENT) @@ -3642,6 +3174,74 @@ static struct hda_amp_list vt1708B_loopbacks[] = { { } /* end */ }; #endif + +static void set_widgets_power_state_vt1708B(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + int is_8ch = 0; + if (spec->codec_type != VT1708B_4CH) + is_8ch = 1; + + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0)); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { @@ -3692,6 +3292,8 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -3742,6 +3344,8 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) spec->loopback.amplist = vt1708B_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; + return 0; } @@ -4181,6 +3785,7 @@ static int patch_vt1708S(struct hda_codec *codec) spec->stream_name_analog = "VT1818S Analog"; spec->stream_name_digital = "VT1818S Digital"; } + spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -4438,6 +4043,37 @@ static struct hda_amp_list vt1702_loopbacks[] = { }; #endif +static void set_widgets_power_state_vt1702(struct hda_codec *codec) +{ + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x17, &parm); + set_pin_power_state(codec, 0x16, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); +} + static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -4484,6 +4120,7 @@ static int patch_vt1702(struct hda_codec *codec) spec->loopback.amplist = vt1702_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -4825,6 +4462,72 @@ static struct hda_amp_list vt1718S_loopbacks[] = { }; #endif +static void set_widgets_power_state_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2b, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW0 (24h), AOW0 (8h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + + /* PW1 (25h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x2a, &parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } +} + static int patch_vt1718S(struct hda_codec *codec) { struct via_spec *spec; @@ -4886,6 +4589,8 @@ static int patch_vt1718S(struct hda_codec *codec) spec->loopback.amplist = vt1718S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1718S; + return 0; } @@ -4925,8 +4630,7 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_jack_power_state(codec); - + set_widgets_power_state(codec); return 1; } @@ -5285,6 +4989,99 @@ static struct hda_amp_list vt1716S_loopbacks[] = { }; #endif +static void set_widgets_power_state_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = + (snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_CONNECT_SEL, 0x00) == 5); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write(codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_jack_detect(codec, 0x1c); + + if (present) + mono_out = 0; + else { + present = snd_hda_jack_detect(codec, 0x1d); + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); +} + static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -5339,6 +5136,7 @@ static int patch_vt1716S(struct hda_codec *codec) spec->loopback.amplist = vt1716S_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -5609,6 +5407,83 @@ static struct hda_amp_list vt2002P_loopbacks[] = { }; #endif +static void set_widgets_power_state_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Class-D */ + /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + + /* Mono Out */ + present = snd_hda_jack_detect(codec, 0x26); + + parm = present ? AC_PWRST_D3 : AC_PWRST_D0; + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); +} /* patch for vt2002P */ static int patch_vt2002P(struct hda_codec *codec) @@ -5660,6 +5535,7 @@ static int patch_vt2002P(struct hda_codec *codec) spec->loopback.amplist = vt2002P_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -5931,6 +5807,97 @@ static struct hda_amp_list vt1812_loopbacks[] = { }; #endif +static void set_widgets_power_state_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer = + snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + unsigned int parm; + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = + snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_jack_detect(codec, 0x28); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + +} /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) @@ -5984,6 +5951,7 @@ static int patch_vt1812(struct hda_codec *codec) spec->loopback.amplist = vt1812_loopbacks; #endif + spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; } -- cgit v1.2.3 From bc92df7fe55e49c616a003b0b77e7badf2736429 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Wed, 23 Mar 2011 17:56:05 +0800 Subject: ALSA: hda - VIA: Add support for VT1705 Add support for VT1705 codec, which is similiar with VT1708S except it has 6 channels output. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 100 ++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 87 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 23cb44086b2..b7650a3dea0 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -3181,7 +3181,8 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) int imux_is_smixer; unsigned int parm; int is_8ch = 0; - if (spec->codec_type != VT1708B_4CH) + if ((spec->codec_type != VT1708B_4CH) && + (codec->vendor_id != 0x11064397)) is_8ch = 1; /* SW0 (17h) = stereo mixer */ @@ -3220,6 +3221,16 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_POWER_STATE, parm); + } else if (codec->vendor_id == 0x11064397) { + /* PW7(23h), SW2(27h), AOW2(25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); } /* PW 3/4/7 (1ch/1dh/23h) */ @@ -3239,7 +3250,9 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); - } + } else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); } static int patch_vt1708S(struct hda_codec *codec); @@ -3414,6 +3427,18 @@ static struct hda_verb vt1708S_uniwill_init_verbs[] = { { } }; +static struct hda_verb vt1705_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + static struct hda_pcm_stream vt1708S_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -3427,6 +3452,19 @@ static struct hda_pcm_stream vt1708S_pcm_analog_playback = { }, }; +static struct hda_pcm_stream vt1705_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close + }, +}; + static struct hda_pcm_stream vt1708S_pcm_analog_capture = { .substreams = 2, .channels_min = 2, @@ -3473,7 +3511,10 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: - spec->multiout.dac_nids[i] = 0x24; + if (spec->codec->vendor_id == 0x11064397) + spec->multiout.dac_nids[i] = 0x25; + else + spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: spec->multiout.dac_nids[i] = 0x11; @@ -3489,22 +3530,28 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, if (cfg->line_outs == 1) { spec->multiout.num_dacs = 3; spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; - spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; + if (spec->codec->vendor_id == 0x11064397) + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x25; + else + spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; } return 0; } /* add playback controls from the parsed DAC table */ -static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, +static int vt1708S_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct via_spec *spec = codec->spec; char name[32]; static const char * const chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25}; - hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27}; + hda_nid_t nid_vols[2][4] = { {0x10, 0x11, 0x24, 0x25}, + {0x10, 0x11, 0x25, 0} }; + hda_nid_t nid_mutes[2][4] = { {0x1C, 0x18, 0x26, 0x27}, + {0x1C, 0x18, 0x27, 0} }; hda_nid_t nid, nid_vol, nid_mute; int i, err; @@ -3515,8 +3562,15 @@ static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, if (!nid && i > AUTO_SEQ_CENLFE) continue; - nid_vol = nid_vols[i]; - nid_mute = nid_mutes[i]; + if (codec->vendor_id == 0x11064397) { + nid_vol = nid_vols[1][i]; + nid_mute = nid_mutes[1][i]; + } else { + nid_vol = nid_vols[0][i]; + nid_mute = nid_mutes[0][i]; + } + if (!nid_vol && !nid_mute) + continue; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -3670,7 +3724,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) return 0; /* can't find valid BIOS pin config */ - err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg); + err = vt1708S_auto_create_multi_out_ctls(codec, &spec->autocfg); if (err < 0) return err; err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); @@ -3737,17 +3791,29 @@ static int patch_vt1708S(struct hda_codec *codec) } spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs; - spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs; + if (codec->vendor_id == 0x11064397) + spec->init_verbs[spec->num_iverbs++] = + vt1705_uniwill_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt1708S_uniwill_init_verbs; if (codec->vendor_id == 0x11060440) spec->stream_name_analog = "VT1818S Analog"; + else if (codec->vendor_id == 0x11064397) + spec->stream_name_analog = "VT1705 Analog"; else spec->stream_name_analog = "VT1708S Analog"; - spec->stream_analog_playback = &vt1708S_pcm_analog_playback; + if (codec->vendor_id == 0x11064397) + spec->stream_analog_playback = &vt1705_pcm_analog_playback; + else + spec->stream_analog_playback = &vt1708S_pcm_analog_playback; spec->stream_analog_capture = &vt1708S_pcm_analog_capture; if (codec->vendor_id == 0x11060440) spec->stream_name_digital = "VT1818S Digital"; + else if (codec->vendor_id == 0x11064397) + spec->stream_name_digital = "VT1705 Digital"; else spec->stream_name_digital = "VT1708S Digital"; spec->stream_digital_playback = &vt1708S_pcm_digital_playback; @@ -3785,6 +3851,14 @@ static int patch_vt1708S(struct hda_codec *codec) spec->stream_name_analog = "VT1818S Analog"; spec->stream_name_digital = "VT1818S Digital"; } + /* correct names for VT1705 */ + if (codec->vendor_id == 0x11064397) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1705", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + } spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -6003,7 +6077,7 @@ static struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1708S}, { .id = 0x11063397, .name = "VT1708S", .patch = patch_vt1708S}, - { .id = 0x11064397, .name = "VT1708S", + { .id = 0x11064397, .name = "VT1705", .patch = patch_vt1708S}, { .id = 0x11065397, .name = "VT1708S", .patch = patch_vt1708S}, -- cgit v1.2.3 From 11890956e948e6ed1b3e4acc11b6879db6ace01b Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Wed, 23 Mar 2011 17:57:34 +0800 Subject: ALSA: hda - VIA: Add support for VT1802 Add support for VT1802 codec, which is similiar with VT2002P except VT1802 has no Class-D and has some different pin widget id. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 195 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 163 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b7650a3dea0..74e2a248281 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -98,9 +98,15 @@ enum VIA_HDA_CODEC { VT1716S, VT2002P, VT1812, + VT1802, CODEC_TYPES, }; +#define VT2002P_COMPATIBLE(spec) \ + ((spec)->codec_type == VT2002P ||\ + (spec)->codec_type == VT1812 ||\ + (spec)->codec_type == VT1802) + struct via_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[6]; @@ -221,6 +227,8 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) codec_type = VT1812; else if (dev_id == 0x0440) codec_type = VT1708S; + else if ((dev_id & 0xfff) == 0x446) + codec_type = VT1802; else codec_type = UNKNOWN; return codec_type; @@ -749,8 +757,7 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, || spec->codec_type == VT1702 || spec->codec_type == VT1718S || spec->codec_type == VT1716S - || spec->codec_type == VT2002P - || spec->codec_type == VT1812) { + || VT2002P_COMPATIBLE(spec)) { activate_ctl(codec, "Headphone Playback Volume", spec->hp_independent_mode); activate_ctl(codec, "Headphone Playback Switch", @@ -788,6 +795,7 @@ static int via_hp_build(struct hda_codec *codec) nid = 0x34; break; case VT2002P: + case VT1802: nid = 0x35; break; case VT1812: @@ -1071,6 +1079,7 @@ static int is_aa_path_mute(struct hda_codec *codec) break; case VT2002P: case VT1812: + case VT1802: nid_mixer = 0x21; start_idx = 0; end_idx = 2; @@ -1135,6 +1144,7 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) break; case VT2002P: case VT1812: + case VT1802: verb = 0xf93; parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ break; @@ -2151,7 +2161,8 @@ static int via_auto_init(struct hda_codec *codec) via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); - if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { + + if (VT2002P_COMPATIBLE(spec)) { via_hp_bind_automute(codec); } else { via_hp_automute(codec); @@ -5291,6 +5302,57 @@ static struct hda_verb vt2002P_volume_init_verbs[] = { {0x1, 0xfb8, 0x88}, { } }; +static struct hda_verb vt1802_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; static struct hda_verb vt2002P_uniwill_init_verbs[] = { @@ -5303,6 +5365,16 @@ static struct hda_verb vt2002P_uniwill_init_verbs[] = { {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; +static struct hda_verb vt1802_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; static struct hda_pcm_stream vt2002P_pcm_analog_playback = { .substreams = 2, @@ -5359,10 +5431,15 @@ static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { int err; + hda_nid_t sw_nid; if (!cfg->line_out_pins[0]) return -1; + if (spec->codec_type == VT1802) + sw_nid = 0x28; + else + sw_nid = 0x26; /* Line-Out: PortE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -5372,7 +5449,7 @@ static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec, return err; err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(sw_nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -5507,21 +5584,41 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) /* AOW0 (8h)*/ snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) { + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, 0, + AC_VERB_SET_POWER_STATE, parm); + } - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) { + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + } if (spec->hp_independent_mode) snd_hda_codec_write(codec, 0x9, 0, @@ -5534,22 +5631,35 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); + if (spec->codec_type == VT1802) + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, parm); + else + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); /* Mono Out */ present = snd_hda_jack_detect(codec, 0x26); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - snd_hda_codec_write(codec, 0x31, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, parm); - + if (spec->codec_type == VT1802) { + /* PW15 (33h), MW8(1ch), MUX8(3ch) */ + snd_hda_codec_write(codec, 0x33, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, parm); + } else { + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + snd_hda_codec_write(codec, 0x31, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, parm); + } /* MW9 (21h) */ if (imux_is_smixer || !is_aa_path_mute(codec)) snd_hda_codec_write(codec, 0x21, 0, @@ -5580,14 +5690,31 @@ static int patch_vt2002P(struct hda_codec *codec) "from BIOS. Using genenic mode...\n"); } - spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; - spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + if (spec->codec_type == VT1802) + spec->init_verbs[spec->num_iverbs++] = + vt1802_volume_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt2002P_volume_init_verbs; - spec->stream_name_analog = "VT2002P Analog"; + if (spec->codec_type == VT1802) + spec->init_verbs[spec->num_iverbs++] = + vt1802_uniwill_init_verbs; + else + spec->init_verbs[spec->num_iverbs++] = + vt2002P_uniwill_init_verbs; + + if (spec->codec_type == VT1802) + spec->stream_name_analog = "VT1802 Analog"; + else + spec->stream_name_analog = "VT2002P Analog"; spec->stream_analog_playback = &vt2002P_pcm_analog_playback; spec->stream_analog_capture = &vt2002P_pcm_analog_capture; - spec->stream_name_digital = "VT2002P Digital"; + if (spec->codec_type == VT1802) + spec->stream_name_digital = "VT1802 Digital"; + else + spec->stream_name_digital = "VT2002P Digital"; spec->stream_digital_playback = &vt2002P_pcm_digital_playback; if (!spec->adc_nids && spec->input_mux) { @@ -6118,6 +6245,10 @@ static struct hda_codec_preset snd_hda_preset_via[] = { { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, { .id = 0x11060440, .name = "VT1818S", .patch = patch_vt1708S}, + { .id = 0x11060446, .name = "VT1802", + .patch = patch_vt2002P}, + { .id = 0x11068446, .name = "VT1802", + .patch = patch_vt2002P}, {} /* terminator */ }; -- cgit v1.2.3 From e87885fea58ef49e08f2b4218587397586dc155d Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Thu, 24 Mar 2011 12:39:05 +0800 Subject: ALSA: hda - VIA: Fix side channel mute invalid issue Modify side_mute_channel() and update_side_mute_status() functions to fix invalid side channel mute issue of VT2002P, VT1812 and VT1802 codecs. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 74e2a248281..c90b23b5cf7 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -708,6 +708,9 @@ static hda_nid_t side_mute_channel(struct via_spec *spec) case VT1709_10CH: return 0x29; case VT1708B_8CH: /* fall thru */ case VT1708S: return 0x27; + case VT2002P: return 0x19; + case VT1802: return 0x15; + case VT1812: return 0x15; default: return 0; } } @@ -716,13 +719,22 @@ static int update_side_mute_status(struct hda_codec *codec) { /* mute side channel */ struct via_spec *spec = codec->spec; - unsigned int parm = spec->hp_independent_mode - ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + unsigned int parm; hda_nid_t sw3 = side_mute_channel(spec); - if (sw3) - snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, - parm); + if (sw3) { + if (VT2002P_COMPATIBLE(spec)) + parm = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + else + parm = spec->hp_independent_mode ? + AMP_OUT_MUTE : AMP_OUT_UNMUTE; + snd_hda_codec_write(codec, sw3, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + } return 0; } -- cgit v1.2.3 From 27439ce717fea649ed77d256bb6efee7d1eb8e56 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Thu, 24 Mar 2011 12:40:10 +0800 Subject: ALSA: hda - VIA: Add VT1802 check in via_speaker_automute function Add VT1802 check in via_speaker_automute() function. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c90b23b5cf7..ec6f32929ce 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1685,7 +1685,7 @@ static void via_speaker_automute(struct hda_codec *codec) unsigned int hp_present; struct via_spec *spec = codec->spec; - if (spec->codec_type != VT2002P && spec->codec_type != VT1812) + if (!VT2002P_COMPATIBLE(spec)) return; hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); -- cgit v1.2.3 From aa266fccf569c6ed675f15b1654a5b2c08d9dde5 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Thu, 24 Mar 2011 12:41:01 +0800 Subject: ALSA: hda - VIA: Update VT1708 initial verbs Add a verb of power down jack detect in VT1708 initial verbs. This verb is used to avoid noise caused by hardware issue. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index ec6f32929ce..1d0afdbb52b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1200,6 +1200,8 @@ static struct hda_verb vt1708_volume_init_verbs[] = { {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* power down jack detect function */ + {0x1, 0xf81, 0x1}, { } }; -- cgit v1.2.3 From 4ab2d53a99b6dcee86837d2a9739bfb9f468db45 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Thu, 24 Mar 2011 12:42:03 +0800 Subject: ALSA: hda - VIA: Update VT1718S initial verbs Add a verb to enable control amplifier of stereo mixer in VT1718S initial verbs. Set stereo mixer default amplifier value as un-mute. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 1d0afdbb52b..ac8e8c757f6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4255,7 +4255,8 @@ static struct hda_verb vt1718S_volume_init_verbs[] = { {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Enable MW0 adjust Gain 5 */ + {0x1, 0xfb2, 0x10}, /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget */ @@ -4264,7 +4265,7 @@ static struct hda_verb vt1718S_volume_init_verbs[] = { {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, /* Setup default input of Front HP to MW9 */ {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, -- cgit v1.2.3 From eadb9a804de223ef899ebb64aa037fa0da7bdee9 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Thu, 24 Mar 2011 12:43:02 +0800 Subject: ALSA: hda - VIA: Update VT2002P initial verbs Add some hardware related verbs in VT2002P initial verbs. These verbs are used to fix Class-D speaker no sound issue. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index ac8e8c757f6..ae9fdaa955f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -5267,6 +5267,10 @@ static struct snd_kcontrol_new vt2002P_capture_mixer[] = { }; static struct hda_verb vt2002P_volume_init_verbs[] = { + /* Class-D speaker related verbs */ + {0x1, 0xfe0, 0x4}, + {0x1, 0xfe9, 0x80}, + {0x1, 0xfe2, 0x22}, /* * Unmute ADC0-1 and set the default input to mic-in */ -- cgit v1.2.3 From ec7e7e42da0b33c77f1baafeac93e5128c4eea7a Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Thu, 24 Mar 2011 12:43:44 +0800 Subject: ALSA: hda - VIA: Update unsolicited event function Update unsolicited event process function via_unsol_event() to make it can process more unsolicited events. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index ae9fdaa955f..15b57a101aa 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -234,12 +234,12 @@ static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) return codec_type; }; +#define VIA_JACK_EVENT 0x20 #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 -#define VIA_JACK_EVENT 0x04 -#define VIA_MONO_EVENT 0x08 -#define VIA_SPEAKER_EVENT 0x10 -#define VIA_BIND_HP_EVENT 0x20 +#define VIA_MONO_EVENT 0x03 +#define VIA_SPEAKER_EVENT 0x04 +#define VIA_BIND_HP_EVENT 0x05 enum { VIA_CTL_WIDGET_VOL, @@ -1746,17 +1746,21 @@ static void via_unsol_event(struct hda_codec *codec, unsigned int res) { res >>= 26; - if (res & VIA_HP_EVENT) - via_hp_automute(codec); - if (res & VIA_GPIO_EVENT) - via_gpio_control(codec); + if (res & VIA_JACK_EVENT) set_widgets_power_state(codec); - if (res & VIA_MONO_EVENT) + + res &= ~VIA_JACK_EVENT; + + if (res == VIA_HP_EVENT) + via_hp_automute(codec); + else if (res == VIA_GPIO_EVENT) + via_gpio_control(codec); + else if (res == VIA_MONO_EVENT) via_mono_automute(codec); - if (res & VIA_SPEAKER_EVENT) + else if (res == VIA_SPEAKER_EVENT) via_speaker_automute(codec); - if (res & VIA_BIND_HP_EVENT) + else if (res == VIA_BIND_HP_EVENT) via_hp_bind_automute(codec); } -- cgit v1.2.3 From 4b8ffdb959c35f5e271fb7e08635dbdb2593018f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 24 Mar 2011 09:11:49 +0200 Subject: ASoC: tlv320dac33: Move codec power up to DAPM Move the codec power on (in reg 0x01, bit 4) from set_bias_level:SND_SOC_BIAS_ON to a DAPM supply. In this way we can be sure, that all the things within the codec is powered before the external amp is going to be enabled. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 00b6d87e7bd..4857f2ae066 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -583,6 +583,9 @@ static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Right DAC Power", DAC33_RDAC_PWR_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Codec Power", + DAC33_PWR_CTRL, 4, 0, NULL, 0), + SND_SOC_DAPM_PRE("Pre Playback", dac33_playback_event), SND_SOC_DAPM_POST("Post Playback", dac33_playback_event), }; @@ -615,6 +618,9 @@ static const struct snd_soc_dapm_route audio_map[] = { /* output */ {"LEFT_LO", NULL, "Output Left Amplifier"}, {"RIGHT_LO", NULL, "Output Right Amplifier"}, + + {"LEFT_LO", NULL, "Codec Power"}, + {"RIGHT_LO", NULL, "Codec Power"}, }; static int dac33_add_widgets(struct snd_soc_codec *codec) @@ -632,13 +638,10 @@ static int dac33_add_widgets(struct snd_soc_codec *codec) static int dac33_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { case SND_SOC_BIAS_ON: - if (!dac33->substream) - dac33_soft_power(codec, 1); break; case SND_SOC_BIAS_PREPARE: break; -- cgit v1.2.3 From 8b573c95d7bd1cf28c69ab4e0255cd272d214482 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Fri, 25 Mar 2011 16:51:46 +0100 Subject: ASoC: imx: remove superfluous code in imx-ssi.c Checking if IMX_SSI_DMA is set and then set it again is useless. Signed-off-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index bc92ec62000..c331d65587d 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -667,12 +667,6 @@ static int imx_ssi_probe(struct platform_device *pdev) if (res) ssi->dma_params_rx.dma = res->start; - if ((cpu_is_mx27() || cpu_is_mx21()) && - !(ssi->flags & IMX_SSI_USE_AC97) && - (ssi->flags & IMX_SSI_DMA)) { - ssi->flags |= IMX_SSI_DMA; - } - platform_set_drvdata(pdev, ssi); ret = snd_soc_register_dai(&pdev->dev, dai); -- cgit v1.2.3 From f3594f5c5c489d159f6d487a889d9d68ca4c0123 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 22 Mar 2011 10:37:01 +0000 Subject: ASoC: soc-cache: Factor-out the I2C read code The handling of all snd_soc_x_y_read_i2c() functions is similar. Make a generic I2C read function and update all callers to use it. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 100 ++++++++++++++++---------------------------------- 1 file changed, 32 insertions(+), 68 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 5d76da43b14..9f6737413a6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -351,33 +351,48 @@ static int snd_soc_8_16_spi_write(void *control_data, const char *data, #endif #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) +static unsigned int do_i2c_read(struct snd_soc_codec *codec, + void *reg, int reglen, + void *data, int datalen) { struct i2c_msg xfer[2]; - u8 reg = r; - u8 data; int ret; struct i2c_client *client = codec->control_data; /* Write register */ xfer[0].addr = client->addr; xfer[0].flags = 0; - xfer[0].len = 1; - xfer[0].buf = ® + xfer[0].len = reglen; + xfer[0].buf = reg; /* Read data */ xfer[1].addr = client->addr; xfer[1].flags = I2C_M_RD; - xfer[1].len = 1; - xfer[1].buf = &data; + xfer[1].len = datalen; + xfer[1].buf = data; ret = i2c_transfer(client->adapter, xfer, 2); - if (ret != 2) { - dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + if (ret == 2) return 0; - } + else if (ret < 0) + return ret; + else + return -EIO; +} +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + u8 reg = r; + u8 data; + int ret; + ret = do_i2c_read(codec, ®, 1, &data, 1); + if (ret < 0) + return 0; return data; } #else @@ -388,30 +403,13 @@ static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, unsigned int r) { - struct i2c_msg xfer[2]; u8 reg = r; u16 data; int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = 1; - xfer[0].buf = ® - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 2; - xfer[1].buf = (u8 *)&data; - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret != 2) { - dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + ret = do_i2c_read(codec, ®, 1, &data, 2); + if (ret < 0) return 0; - } - return (data >> 8) | ((data & 0xff) << 8); } #else @@ -422,30 +420,13 @@ static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, unsigned int r) { - struct i2c_msg xfer[2]; u16 reg = r; u8 data; int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = 2; - xfer[0].buf = (u8 *)® - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 1; - xfer[1].buf = &data; - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret != 2) { - dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + ret = do_i2c_read(codec, ®, 2, &data, 1); + if (ret < 0) return 0; - } - return data; } #else @@ -543,30 +524,13 @@ static int snd_soc_16_8_spi_write(void *control_data, const char *data, static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, unsigned int r) { - struct i2c_msg xfer[2]; u16 reg = cpu_to_be16(r); u16 data; int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = 2; - xfer[0].buf = (u8 *)® - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 2; - xfer[1].buf = (u8 *)&data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret != 2) { - dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + ret = do_i2c_read(codec, ®, 2, &data, 2); + if (ret < 0) return 0; - } - return be16_to_cpu(data); } #else -- cgit v1.2.3 From 5fb609d435f0679ed322ddeb1fdafe6142463fdf Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 22 Mar 2011 10:37:03 +0000 Subject: ASoC: soc-cache: Introduce raw bulk write support As it has become more common to have to write firmware or similar large chunks of data to the hardware, add a function to perform raw bulk writes that bypass the cache. This only handles volatile registers as we should avoid getting out of sync with the actual cache. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 7 +++++++ 2 files changed, 46 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9f6737413a6..8bcd4e1cee9 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -625,6 +625,44 @@ static int snd_soc_16_16_spi_write(void *control_data, const char *data, #define snd_soc_16_16_spi_write NULL #endif +/* Primitive bulk write support for soc-cache. The data pointed to by `data' needs + * to already be in the form the hardware expects including any leading register specific + * data. Any data written through this function will not go through the cache as it + * only handles writing to volatile or out of bounds registers. + */ +static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, + const void *data, size_t len) +{ + int ret; + + /* Ensure that the base register is volatile. Subsequently + * any other register that is touched by this routine should be + * volatile as well to ensure that we don't get out of sync with + * the cache. + */ + if (!snd_soc_codec_volatile_register(codec, reg) + && reg < codec->driver->reg_cache_size) + return -EINVAL; + + switch (codec->control_type) { + case SND_SOC_I2C: + ret = i2c_master_send(codec->control_data, data, len); + break; + case SND_SOC_SPI: + ret = do_spi_write(codec->control_data, data, len); + break; + default: + BUG(); + } + + if (ret == len) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + static struct { int addr_bits; int data_bits; @@ -708,6 +746,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, codec->write = io_types[i].write; codec->read = io_types[i].read; + codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; switch (control) { case SND_SOC_CUSTOM: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4dda58926bc..636328e868e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2228,6 +2228,13 @@ unsigned int snd_soc_write(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_write); +unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec, + unsigned int reg, const void *data, size_t len) +{ + return codec->bulk_write_raw(codec, reg, data, len); +} +EXPORT_SYMBOL_GPL(snd_soc_bulk_write_raw); + /** * snd_soc_update_bits - update codec register bits * @codec: audio codec -- cgit v1.2.3 From 26e9984cbcdde100e5af15382f2297fef1ce7804 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 22 Mar 2011 10:36:58 +0000 Subject: ASoC: soc-cache: Factor-out the hw_write() specific code The handling of all snd_soc_x_y_write() functions is similar. Factor it out into a separate function and update all functions to use it. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 153 +++++++++++--------------------------------------- 1 file changed, 33 insertions(+), 120 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8bcd4e1cee9..4ee473d6057 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,6 +20,33 @@ #include +static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value, const void *data, int len) +{ + int ret; + + if (!snd_soc_codec_volatile_register(codec, reg) && + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return -1; + } + + if (codec->cache_only) { + codec->cache_sync = 1; + return 0; + } + + ret = codec->hw_write(codec->control_data, data, len); + if (ret == len) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -46,31 +73,11 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[2]; - int ret; data[0] = (reg << 4) | ((value >> 8) & 0x000f); data[1] = value & 0x00ff; - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - - ret = codec->hw_write(codec->control_data, data, 2); - if (ret == 2) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return do_hw_write(codec, reg, value, data, 2); } #if defined(CONFIG_SPI_MASTER) @@ -129,31 +136,11 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[2]; - int ret; data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - - ret = codec->hw_write(codec->control_data, data, 2); - if (ret == 2) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return do_hw_write(codec, reg, value, data, 2); } #if defined(CONFIG_SPI_MASTER) @@ -190,29 +177,12 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[2]; - int ret; reg &= 0xff; data[0] = reg; data[1] = value & 0xff; - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; + return do_hw_write(codec, reg, value, data, 2); } static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, @@ -272,29 +242,12 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[3]; - int ret; data[0] = reg; data[1] = (value >> 8) & 0xff; data[2] = value & 0xff; - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - - if (codec->hw_write(codec->control_data, data, 3) == 3) - return 0; - else - return -EIO; + return do_hw_write(codec, reg, value, data, 3); } static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, @@ -460,33 +413,13 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[3]; - int ret; data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; - reg &= 0xff; - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - ret = codec->hw_write(codec->control_data, data, 3); - if (ret == 3) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return do_hw_write(codec, reg, value, data, 3); } #if defined(CONFIG_SPI_MASTER) @@ -564,33 +497,13 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u8 data[4]; - int ret; data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = (value >> 8) & 0xff; data[3] = value & 0xff; - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - - ret = codec->hw_write(codec->control_data, data, 4); - if (ret == 4) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return do_hw_write(codec, reg, value, data, 4); } #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From b8cbc195202d05efcda6af81c669577e3cb793e5 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 22 Mar 2011 10:36:59 +0000 Subject: ASoC: soc-cache: Factor-out the hw_read() specific code The handling of all snd_soc_x_y_read() functions is similar. Factor it out into a separate function and update all callers. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 114 ++++++++------------------------------------------ 1 file changed, 18 insertions(+), 96 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 4ee473d6057..abb0243f3ad 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -47,20 +47,19 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, return -EIO; } -static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, - unsigned int reg) +static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) { int ret; unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { + if (codec->cache_only) + return -1; - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); + BUG_ON(!codec->hw_read); + return codec->hw_read(codec, reg); } ret = snd_soc_cache_read(codec, reg, &val); @@ -69,6 +68,12 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, return val; } +static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + return do_hw_read(codec, reg); +} + static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -113,23 +118,7 @@ static int snd_soc_4_12_spi_write(void *control_data, const char *data, static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { - int ret; - unsigned int val; - - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) - return -1; - return val; + return do_hw_read(codec, reg); } static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, @@ -188,24 +177,7 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, unsigned int reg) { - int ret; - unsigned int val; - - reg &= 0xff; - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) - return -1; - return val; + return do_hw_read(codec, reg); } #if defined(CONFIG_SPI_MASTER) @@ -253,23 +225,7 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, unsigned int reg) { - int ret; - unsigned int val; - - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) - return -1; - return val; + return do_hw_read(codec, reg); } #if defined(CONFIG_SPI_MASTER) @@ -389,24 +345,7 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, unsigned int reg) { - int ret; - unsigned int val; - - reg &= 0xff; - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) - return -1; - return val; + return do_hw_read(codec, reg); } static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, @@ -473,24 +412,7 @@ static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, unsigned int reg) { - int ret; - unsigned int val; - - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) - return -1; - - return val; + return do_hw_read(codec, reg); } static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, -- cgit v1.2.3 From 30539a18d366cff6b21f66a81e4d9dccc4a90c89 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 22 Mar 2011 10:37:00 +0000 Subject: ASoC: soc-cache: Factor-out the SPI write code The handling of all snd_soc_x_y_spi_write() functions is similar. Create a separate function and update all callers to use it. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 126 ++++++++++++-------------------------------------- 1 file changed, 30 insertions(+), 96 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index abb0243f3ad..d7bffdd033b 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,6 +20,30 @@ #include +#if defined(CONFIG_SPI_MASTER) +static int do_spi_write(void *control_data, const void *msg, + int len) +{ + struct spi_device *spi = control_data; + struct spi_transfer t; + struct spi_message m; + + if (len <= 0) + return 0; + + spi_message_init(&m); + memset(&t, 0, sizeof t); + + t.tx_buf = msg; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif + static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value, const void *data, int len) { @@ -89,27 +113,12 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_4_12_spi_write(void *control_data, const char *data, int len) { - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; u8 msg[2]; - if (len <= 0) - return 0; - msg[0] = data[1]; msg[1] = data[0]; - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; + return do_spi_write(control_data, msg, len); } #else #define snd_soc_4_12_spi_write NULL @@ -136,27 +145,12 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_7_9_spi_write(void *control_data, const char *data, int len) { - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; u8 msg[2]; - if (len <= 0) - return 0; - msg[0] = data[0]; msg[1] = data[1]; - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; + return do_spi_write(control_data, msg, len); } #else #define snd_soc_7_9_spi_write NULL @@ -184,27 +178,12 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, static int snd_soc_8_8_spi_write(void *control_data, const char *data, int len) { - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; u8 msg[2]; - if (len <= 0) - return 0; - msg[0] = data[0]; msg[1] = data[1]; - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; + return do_spi_write(control_data, msg, len); } #else #define snd_soc_8_8_spi_write NULL @@ -232,28 +211,13 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, static int snd_soc_8_16_spi_write(void *control_data, const char *data, int len) { - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; u8 msg[3]; - if (len <= 0) - return 0; - msg[0] = data[0]; msg[1] = data[1]; msg[2] = data[2]; - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; + return do_spi_write(control_data, msg, len); } #else #define snd_soc_8_16_spi_write NULL @@ -365,28 +329,13 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_16_8_spi_write(void *control_data, const char *data, int len) { - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; u8 msg[3]; - if (len <= 0) - return 0; - msg[0] = data[0]; msg[1] = data[1]; msg[2] = data[2]; - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; + return do_spi_write(control_data, msg, len); } #else #define snd_soc_16_8_spi_write NULL @@ -432,29 +381,14 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_16_16_spi_write(void *control_data, const char *data, int len) { - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; u8 msg[4]; - if (len <= 0) - return 0; - msg[0] = data[0]; msg[1] = data[1]; msg[2] = data[2]; msg[3] = data[3]; - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; + return do_spi_write(control_data, msg, len); } #else #define snd_soc_16_16_spi_write NULL -- cgit v1.2.3 From acd61451e55ea5848a6ab50d39a103e146fcf7ba Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 22 Mar 2011 10:48:49 +0000 Subject: ASoC: soc-cache: Return -ENOSYS instead of -EINVAL These functions fail with -EINVAL if the corresponding callbacks are not implemented. Change them to return -ENOSYS as it is more appropriate for unimplemented callbacks. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d7bffdd033b..258c3b2098b 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1295,7 +1295,7 @@ int snd_soc_cache_init(struct snd_soc_codec *codec) codec->cache_ops->name, codec->name); return codec->cache_ops->init(codec); } - return -EINVAL; + return -ENOSYS; } /* @@ -1310,7 +1310,7 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) codec->cache_ops->name, codec->name); return codec->cache_ops->exit(codec); } - return -EINVAL; + return -ENOSYS; } /** @@ -1334,7 +1334,7 @@ int snd_soc_cache_read(struct snd_soc_codec *codec, } mutex_unlock(&codec->cache_rw_mutex); - return -EINVAL; + return -ENOSYS; } EXPORT_SYMBOL_GPL(snd_soc_cache_read); @@ -1359,7 +1359,7 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, } mutex_unlock(&codec->cache_rw_mutex); - return -EINVAL; + return -ENOSYS; } EXPORT_SYMBOL_GPL(snd_soc_cache_write); @@ -1382,7 +1382,7 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) } if (!codec->cache_ops || !codec->cache_ops->sync) - return -EINVAL; + return -ENOSYS; if (codec->cache_ops->name) name = codec->cache_ops->name; -- cgit v1.2.3 From 8020454c9a1ec5ac5801805896b5f69d0c573e17 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 24 Mar 2011 13:45:17 +0000 Subject: ASoC: Add default snd_soc_default_writable_register() callback By using struct snd_soc_reg_access for the read/write/vol attributes of the registers, we provide callbacks that automatically determine whether a given register is readable/writable or volatile. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 14 ++++++++++++++ sound/soc/soc-core.c | 3 +++ 2 files changed, 17 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 258c3b2098b..1210a6f70a9 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1449,3 +1449,17 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec, return codec->driver->reg_access_default[index].read; } EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); + +int snd_soc_default_writable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].write; +} +EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 636328e868e..074a0c6e99f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3673,6 +3673,7 @@ int snd_soc_register_codec(struct device *dev, codec->read = codec_drv->read; codec->volatile_register = codec_drv->volatile_register; codec->readable_register = codec_drv->readable_register; + codec->writable_register = codec_drv->writable_register; codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; @@ -3707,6 +3708,8 @@ int snd_soc_register_codec(struct device *dev, codec->volatile_register = snd_soc_default_volatile_register; if (!codec->readable_register) codec->readable_register = snd_soc_default_readable_register; + if (!codec->writable_register) + codec->writable_register = snd_soc_default_writable_register; } for (i = 0; i < num_dai; i++) { -- cgit v1.2.3 From 239c970626b9d9c7449de751d91f9a9da1018b85 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 24 Mar 2011 13:45:18 +0000 Subject: ASoC: Add snd_soc_codec_{readable,writable}_register() Provide the top level ASoC core functions for indicating whether a given register is readable or writable. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 074a0c6e99f..5ae70a107f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2146,6 +2146,42 @@ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); +/** + * snd_soc_codec_readable_register: Report if a register is readable. + * + * @codec: CODEC to query. + * @reg: Register to query. + * + * Boolean function indicating if a CODEC register is readable. + */ +int snd_soc_codec_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + if (codec->readable_register) + return codec->readable_register(codec, reg); + else + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register); + +/** + * snd_soc_codec_writable_register: Report if a register is writable. + * + * @codec: CODEC to query. + * @reg: Register to query. + * + * Boolean function indicating if a CODEC register is writable. + */ +int snd_soc_codec_writable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + if (codec->writable_register) + return codec->writable_register(codec, reg); + else + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec -- cgit v1.2.3 From 4b2ffc205cb9964e7270abc7c8bcbd6127dfcf5e Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Sat, 26 Mar 2011 03:05:08 -0400 Subject: ASoC: Blackfin I2S: add 8-bit sample support Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-i2s.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index d453b1e9d60..37d9d3c2f2b 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -140,6 +140,10 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, bf5xx_i2s.tcr2 &= ~0x1f; bf5xx_i2s.rcr2 &= ~0x1f; switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bf5xx_i2s->tcr2 |= 7; + bf5xx_i2s->rcr2 |= 7; + sport_handle->wdsize = 1; case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; @@ -266,8 +270,11 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ SNDRV_PCM_RATE_96000) -#define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) +#define BF5XX_I2S_FORMATS \ + (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { .shutdown = bf5xx_i2s_shutdown, -- cgit v1.2.3 From 119bfef2f954bfca74a960bff7be7d49f97e5daf Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Sat, 26 Mar 2011 03:52:20 -0400 Subject: ASoC: ad193x: tweak style to match other codecs Rename the snd_soc_control_type field from "bus_type" to "control_type", and drop the now unused "control_data" field. Signed-off-by: Scott Jiang Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index da46479bfcf..824529d5c77 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,8 +23,7 @@ /* codec private data */ struct ad193x_priv { - enum snd_soc_control_type bus_type; - void *control_data; + enum snd_soc_control_type control_type; int sysclk; }; @@ -354,14 +353,12 @@ static int ad193x_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - codec->control_data = ad193x->control_data; - if (ad193x->bus_type == SND_SOC_I2C) - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->bus_type); + if (ad193x->control_type == SND_SOC_I2C) + ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->control_type); else - ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->bus_type); + ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->control_type); if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; } @@ -408,8 +405,7 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, ad193x); - ad193x->control_data = spi; - ad193x->bus_type = SND_SOC_SPI; + ad193x->control_type = SND_SOC_SPI; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); @@ -454,8 +450,7 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, ad193x); - ad193x->control_data = client; - ad193x->bus_type = SND_SOC_I2C; + ad193x->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, &ad193x_dai, 1); -- cgit v1.2.3 From 5683dc7ae86987e392d5616731eec636abd6c7ad Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Sat, 26 Mar 2011 04:38:30 -0400 Subject: ASoC: ad73311: drop I2C requirement The AD73311 codec does not use I2C, so don't require it in Kconfig. Signed-off-by: Scott Jiang Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d63c1754e05..a9defd53445 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,7 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_ADS117X - select SND_SOC_AD73311 if I2C + select SND_SOC_AD73311 select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C -- cgit v1.2.3 From ecd015127813c7893f6900cf54c6362b26d8ff90 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Mar 2011 13:43:45 +0100 Subject: ASoC: Convert WM8903 to table based DAPM setup Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ae1cadfae84..a739e09a830 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -926,7 +926,7 @@ SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8903_intercon[] = { { "CLK_DSP", NULL, "CLK_SYS" }, { "Mic Bias", NULL, "CLK_SYS" }, @@ -1079,17 +1079,6 @@ static const struct snd_soc_dapm_route intercon[] = { { "Right Line Output PGA", NULL, "Charge Pump" }, }; -static int wm8903_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8903_dapm_widgets, - ARRAY_SIZE(wm8903_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static int wm8903_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2020,7 +2009,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8903_snd_controls, ARRAY_SIZE(wm8903_snd_controls)); - wm8903_add_widgets(codec); wm8903_init_gpio(codec); @@ -2046,6 +2034,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .reg_cache_default = wm8903_reg_defaults, .volatile_register = wm8903_volatile_register, .seq_notifier = wm8903_seq_notifier, + .dapm_widgets = wm8903_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8903_dapm_widgets), + .dapm_routes = wm8903_intercon, + .num_dapm_routes = ARRAY_SIZE(wm8903_intercon), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.3 From 5e251aecbd4a4bf3ff0b0ff04653afa4d01eba54 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Mar 2011 14:33:03 +0100 Subject: ASoC: Convert WM8731 to table based DAPM setup Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8731.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0a67c31b266..44a591e8255 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -201,7 +201,7 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, return wm8731->sysclk_type == WM8731_SYSCLK_MCLK; } -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8731_intercon[] = { {"DAC", NULL, "OSC", wm8731_check_osc}, {"ADC", NULL, "OSC", wm8731_check_osc}, @@ -227,17 +227,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"Mic Bias", NULL, "MICIN"}, }; -static int wm8731_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8731_dapm_widgets, - ARRAY_SIZE(wm8731_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - struct _coeff_div { u32 mclk; u32 rate; @@ -599,7 +588,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8731_snd_controls, ARRAY_SIZE(wm8731_snd_controls)); - wm8731_add_widgets(codec); /* Regulators will have been enabled by bias management */ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -636,6 +624,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .reg_cache_size = ARRAY_SIZE(wm8731_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8731_reg, + .dapm_widgets = wm8731_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), + .dapm_routes = wm8731_intercon, + .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From 99b59f3ccb72c15f13169236eda16682930e47f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Mar 2011 14:35:15 +0100 Subject: ASoC: Remove -codec from WM8731 driver name Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/poodle.c | 2 +- 5 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index af3c73053ee..28afbbf69ce 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -184,7 +184,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .codec_dai_name = "wm8731-hifi", .init = at91sam9g20ek_wm8731_init, .platform_name = "atmel-pcm-audio", - .codec_name = "wm8731-codec.0-001b", + .codec_name = "wm8731.0-001b", .ops = &at91sam9g20ek_ops, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index cb99f04abe8..1d3e258c9ea 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -77,7 +77,7 @@ static struct snd_soc_dai_link db1200_i2s_dai = { .codec_dai_name = "wm8731-hifi", .cpu_dai_name = "au1xpsc_i2s.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "wm8731-codec.0-001b", + .codec_name = "wm8731.0-001b", .ops = &db1200_i2s_wm8731_ops, }; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 44a591e8255..6dec7cee2cb 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -659,7 +659,7 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) static struct spi_driver wm8731_spi_driver = { .driver = { - .name = "wm8731-codec", + .name = "wm8731", .owner = THIS_MODULE, }, .probe = wm8731_spi_probe, @@ -703,7 +703,7 @@ MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); static struct i2c_driver wm8731_i2c_driver = { .driver = { - .name = "wm8731-codec", + .name = "wm8731", .owner = THIS_MODULE, }, .probe = wm8731_i2c_probe, diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 9027da466ca..28757fb9df3 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -310,7 +310,7 @@ static struct snd_soc_dai_link corgi_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec.0-001b", + .codec_name = "wm8731.0-001b", .init = corgi_wm8731_init, .ops = &corgi_ops, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index a7d4999f9b2..da3ae4316cf 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = { .cpu_dai_name = "pxa2xx-i2s", .codec_dai_name = "wm8731-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm8731-codec.0-001b", + .codec_name = "wm8731.0-001b", .init = poodle_wm8731_init, .ops = &poodle_ops, }; -- cgit v1.2.3 From 754dec6b7f01efadd400f3291591627093a17988 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Sun, 27 Mar 2011 00:43:59 -0400 Subject: ASoC: fix sorting order of codecs in kconfig Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a9defd53445..3e84b528a88 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,8 +16,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS - select SND_SOC_ADS117X select SND_SOC_AD73311 + select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C -- cgit v1.2.3 From 2a161018914d7f12ed73991d1aedb609fdba3fcf Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Sun, 27 Mar 2011 00:44:10 -0400 Subject: ASoC: SSM2602: fix codec name The codec name should not have a "-codec" suffix since this is not part of a MFD. This was incorrectly changed during the multi-component updated. Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2727befd158..5a5a6b16911 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -651,7 +651,7 @@ MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); /* corgi i2c codec control layer */ static struct i2c_driver ssm2602_i2c_driver = { .driver = { - .name = "ssm2602-codec", + .name = "ssm2602", .owner = THIS_MODULE, }, .probe = ssm2602_i2c_probe, -- cgit v1.2.3 From 370fd17dea6056cdb2fbbca47812cf96d2a02726 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Sun, 27 Mar 2011 00:44:11 -0400 Subject: ASoC: ad73311: fix codec name The codec name should not have a "-codec" suffix since this is not part of a MFD. This was incorrectly changed during the multi-component updated. Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad73311.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index de799cd1ba7..8d793e993e9 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -55,7 +55,7 @@ static int __devexit ad73311_remove(struct platform_device *pdev) static struct platform_driver ad73311_codec_driver = { .driver = { - .name = "ad73311-codec", + .name = "ad73311", .owner = THIS_MODULE, }, -- cgit v1.2.3 From e43a7d4116c868676eca329c4211bc5cc5a93a9b Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Sun, 27 Mar 2011 00:44:12 -0400 Subject: ASoC: ad193x: fix codec name The codec name should not have a "-codec" suffix since this is not part of a MFD. This was incorrectly changed during the multi-component updated. Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 824529d5c77..2374ca5ffe6 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -423,7 +423,7 @@ static int __devexit ad193x_spi_remove(struct spi_device *spi) static struct spi_driver ad193x_spi_driver = { .driver = { - .name = "ad193x-codec", + .name = "ad193x", .owner = THIS_MODULE, }, .probe = ad193x_spi_probe, @@ -468,7 +468,7 @@ static int __devexit ad193x_i2c_remove(struct i2c_client *client) static struct i2c_driver ad193x_i2c_driver = { .driver = { - .name = "ad193x-codec", + .name = "ad193x", }, .probe = ad193x_i2c_probe, .remove = __devexit_p(ad193x_i2c_remove), -- cgit v1.2.3 From 1afa98b8074136c81d23e56836a59b2c8f1529c8 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Sun, 27 Mar 2011 00:44:13 -0400 Subject: ASoC: ad1980: fix codec name The codec name should not have a "-codec" suffix since this is not part of a MFD. This was incorrectly changed during the multi-component updated. Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 34cb51ef215..923b364a3e4 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -266,7 +266,7 @@ static int __devexit ad1980_remove(struct platform_device *pdev) static struct platform_driver ad1980_codec_driver = { .driver = { - .name = "ad1980-codec", + .name = "ad1980", .owner = THIS_MODULE, }, -- cgit v1.2.3 From 93547e89b6d2adeec627d2a8d8fc4d4b40b0e2c4 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Sun, 27 Mar 2011 17:22:57 -0400 Subject: ASoC: SSM2602: convert to soc-cache Signed-off-by: Cliff Cai Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 138 ++++++++++++++------------------------------- 1 file changed, 43 insertions(+), 95 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 5a5a6b16911..8a2b52fa4a7 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -48,7 +48,6 @@ struct ssm2602_priv { unsigned int sysclk; enum snd_soc_control_type control_type; - void *control_data; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; }; @@ -65,55 +64,7 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { 0x0000, 0x0000 }; -/* - * read ssm2602 register cache - */ -static inline unsigned int ssm2602_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == SSM2602_RESET) - return 0; - if (reg >= SSM2602_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write ssm2602 register cache - */ -static inline void ssm2602_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= SSM2602_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the ssm2602 register space - */ -static int ssm2602_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 ssm2602 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - ssm2602_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define ssm2602_reset(c) ssm2602_write(c, SSM2602_RESET, 0) +#define ssm2602_reset(c) snd_soc_write(c, SSM2602_RESET, 0) /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { @@ -278,12 +229,11 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; - u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; + u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); if (substream == ssm2602->slave_substream) { - dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); return 0; } @@ -294,8 +244,8 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | coeff_div[i].usb; - ssm2602_write(codec, SSM2602_ACTIVE, 0); - ssm2602_write(codec, SSM2602_SRATE, srate); + snd_soc_write(codec, SSM2602_ACTIVE, 0); + snd_soc_write(codec, SSM2602_SRATE, srate); /* bit size */ switch (params_format(params)) { @@ -311,8 +261,8 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, iface |= 0x000c; break; } - ssm2602_write(codec, SSM2602_IFACE, iface); - ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); + snd_soc_write(codec, SSM2602_IFACE, iface); + snd_soc_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); return 0; } @@ -360,7 +310,7 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; /* set active */ - ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); + snd_soc_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); return 0; } @@ -374,7 +324,7 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) - ssm2602_write(codec, SSM2602_ACTIVE, 0); + snd_soc_write(codec, SSM2602_ACTIVE, 0); if (ssm2602->master_substream == substream) ssm2602->master_substream = ssm2602->slave_substream; @@ -385,12 +335,12 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = ssm2602_read_reg_cache(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + u16 mute_reg = snd_soc_read(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; if (mute) - ssm2602_write(codec, SSM2602_APDIGI, + snd_soc_write(codec, SSM2602_APDIGI, mute_reg | APDIGI_ENABLE_DAC_MUTE); else - ssm2602_write(codec, SSM2602_APDIGI, mute_reg); + snd_soc_write(codec, SSM2602_APDIGI, mute_reg); return 0; } @@ -466,30 +416,30 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - ssm2602_write(codec, SSM2602_IFACE, iface); + snd_soc_write(codec, SSM2602_IFACE, iface); return 0; } static int ssm2602_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = ssm2602_read_reg_cache(codec, SSM2602_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, SSM2602_PWR) & 0xff7f; switch (level) { case SND_SOC_BIAS_ON: /* vref/mid, osc on, dac unmute */ - ssm2602_write(codec, SSM2602_PWR, reg); + snd_soc_write(codec, SSM2602_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* everything off except vref/vmid, */ - ssm2602_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN); + snd_soc_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN); break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - ssm2602_write(codec, SSM2602_ACTIVE, 0); - ssm2602_write(codec, SSM2602_PWR, 0xffff); + snd_soc_write(codec, SSM2602_ACTIVE, 0); + snd_soc_write(codec, SSM2602_PWR, 0xffff); break; } @@ -539,17 +489,10 @@ static int ssm2602_suspend(struct snd_soc_codec *codec, pm_message_t state) static int ssm2602_resume(struct snd_soc_codec *codec) { - int i; - u8 data[2]; - u16 *cache = codec->reg_cache; - - /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(ssm2602_reg); i++) { - data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); - data[1] = cache[i] & 0x00ff; - codec->hw_write(codec->control_data, data, 2); - } + snd_soc_cache_sync(codec); + ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; } @@ -560,31 +503,39 @@ static int ssm2602_probe(struct snd_soc_codec *codec) pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); - codec->control_data = ssm2602->control_data; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, ssm2602->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } - ssm2602_reset(codec); + ret = ssm2602_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + return ret; + } /*power on device*/ - ssm2602_write(codec, SSM2602_ACTIVE, 0); + snd_soc_write(codec, SSM2602_ACTIVE, 0); /* set the update bits */ - reg = ssm2602_read_reg_cache(codec, SSM2602_LINVOL); - ssm2602_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); - reg = ssm2602_read_reg_cache(codec, SSM2602_RINVOL); - ssm2602_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); - reg = ssm2602_read_reg_cache(codec, SSM2602_LOUT1V); - ssm2602_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); - reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V); - ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); + reg = snd_soc_read(codec, SSM2602_LINVOL); + snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); + reg = snd_soc_read(codec, SSM2602_RINVOL); + snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); + reg = snd_soc_read(codec, SSM2602_LOUT1V); + snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); + reg = snd_soc_read(codec, SSM2602_ROUT1V); + snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); /*select Line in as default input*/ - ssm2602_write(codec, SSM2602_APANA, APANA_SELECT_DAC | + snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); - ssm2602_write(codec, SSM2602_PWR, 0); + snd_soc_write(codec, SSM2602_PWR, 0); snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); - return ret; + return 0; } /* remove everything here */ @@ -599,8 +550,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .remove = ssm2602_remove, .suspend = ssm2602_suspend, .resume = ssm2602_resume, - .read = ssm2602_read_reg_cache, - .write = ssm2602_write, .set_bias_level = ssm2602_set_bias_level, .reg_cache_size = sizeof(ssm2602_reg), .reg_word_size = sizeof(u16), @@ -625,7 +574,6 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, ssm2602); - ssm2602->control_data = i2c; ssm2602->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3 From f04cd9cb11469a6a9cbfdad824254143250d78d2 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Sun, 27 Mar 2011 05:33:04 -0400 Subject: ASoC: Blackfin: add ad193x sysclk configuration Signed-off-by: Scott Jiang Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad193x.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d3ccb926b5e..33678828d57 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -60,8 +60,16 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int clk = 0; unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; + + switch (params_rate(params)) { + case 48000: + clk = 12288000; + break; + } + /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); @@ -74,6 +82,12 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + /* set codec DAI slots, 8 channels, all channels are enabled */ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); if (ret < 0) -- cgit v1.2.3 From c8ad38b8b26ad04d5125606065c87315308d6df5 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Mon, 28 Mar 2011 01:45:08 -0400 Subject: ASoC: Blackfin: drop "-codec" from codec names The recent multi-component patch incorrectly added "-codec" suffixes to parts which are not MFD. Drop the suffix from the machine drivers too. Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad1836.c | 2 +- sound/soc/blackfin/bf5xx-ad193x.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 +- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 83012da9dfc..826bae696d3 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -85,7 +85,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai = { .cpu_dai_name = "bf5xx-tdm", .codec_dai_name = "ad1836-hifi", .platform_name = "bf5xx-tdm-pcm-audio", - .codec_name = "ad1836-codec.0", + .codec_name = "ad1836.0", .ops = &bf5xx_ad1836_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 33678828d57..5794d094d00 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -113,7 +113,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai = { .cpu_dai_name = "bf5xx-tdm", .codec_dai_name ="ad193x-hifi", .platform_name = "bf5xx-tdm-pcm-audio", - .codec_name = "ad193x-codec.5", + .codec_name = "ad193x.5", .ops = &bf5xx_ad193x_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index d57c9c9c988..c147130a2e7 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -73,7 +73,7 @@ static struct snd_soc_dai_link bf5xx_board_dai = { .cpu_dai_name = "bfin-ac97", .codec_dai_name = "ad1980-hifi", .platform_name = "bfin-pcm-audio", - .codec_name = "ad1980-codec", + .codec_name = "ad1980", .ops = &bf5xx_board_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732fb8bad07..1655f950dfb 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -186,7 +186,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-pcm-audio", - .codec_name = "ad73311-codec", + .codec_name = "ad73311", .ops = &bf5xx_ad73311_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index ad28663f5bb..732a02ffe68 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-001b", + .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, }; -- cgit v1.2.3 From bfe4ee0a935dccf5980ecb5605c66fe50feb9056 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Mon, 28 Mar 2011 01:45:09 -0400 Subject: ASoC: Blackfin: standardize machine driver names Some machine drivers were using "bf5xx-", others were using "bf5xx_", while others were using "bfin-". Further, some were using the same name in the transport layer which makes it hard to use different codecs at the same time. So standardize all of them to "bfin-" and make sure they are name spaced according to their driver name. Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-ad1836.c | 6 +++--- sound/soc/blackfin/bf5xx-ad193x.c | 6 +++--- sound/soc/blackfin/bf5xx-ad1980.c | 4 ++-- sound/soc/blackfin/bf5xx-ad73311.c | 6 +++--- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 6 +++--- sound/soc/blackfin/bf5xx-tdm-pcm.c | 2 +- 9 files changed, 18 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 5a2fd8abaef..e940d26048d 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -458,7 +458,7 @@ static int __devexit bf5xx_soc_platform_remove(struct platform_device *pdev) static struct platform_driver bf5xx_pcm_driver = { .driver = { - .name = "bf5xx-pcm-audio", + .name = "bfin-ac97-pcm-audio", .owner = THIS_MODULE, }, diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 826bae696d3..5d9d9e2805b 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -82,15 +82,15 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { static struct snd_soc_dai_link bf5xx_ad1836_dai = { .name = "ad1836", .stream_name = "AD1836", - .cpu_dai_name = "bf5xx-tdm", + .cpu_dai_name = "bfin-tdm", .codec_dai_name = "ad1836-hifi", - .platform_name = "bf5xx-tdm-pcm-audio", + .platform_name = "bfin-tdm-pcm-audio", .codec_name = "ad1836.0", .ops = &bf5xx_ad1836_ops, }; static struct snd_soc_card bf5xx_ad1836 = { - .name = "bf5xx_ad1836", + .name = "bfin-ad1836", .dai_link = &bf5xx_ad1836_dai, .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 5794d094d00..355094fad18 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -110,15 +110,15 @@ static struct snd_soc_ops bf5xx_ad193x_ops = { static struct snd_soc_dai_link bf5xx_ad193x_dai = { .name = "ad193x", .stream_name = "AD193X", - .cpu_dai_name = "bf5xx-tdm", + .cpu_dai_name = "bfin-tdm", .codec_dai_name ="ad193x-hifi", - .platform_name = "bf5xx-tdm-pcm-audio", + .platform_name = "bfin-tdm-pcm-audio", .codec_name = "ad193x.5", .ops = &bf5xx_ad193x_ops, }; static struct snd_soc_card bf5xx_ad193x = { - .name = "bf5xx_ad193x", + .name = "bfin-ad193x", .dai_link = &bf5xx_ad193x_dai, .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index c147130a2e7..a3812408c6a 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -72,13 +72,13 @@ static struct snd_soc_dai_link bf5xx_board_dai = { .stream_name = "AC97 HiFi", .cpu_dai_name = "bfin-ac97", .codec_dai_name = "ad1980-hifi", - .platform_name = "bfin-pcm-audio", + .platform_name = "bfin-ac97-pcm-audio", .codec_name = "ad1980", .ops = &bf5xx_board_ops, }; static struct snd_soc_card bf5xx_board = { - .name = "bf5xx-board", + .name = "bfin-ad1980", .dai_link = &bf5xx_board_dai, .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 1655f950dfb..9f0d4f3c590 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -183,15 +183,15 @@ static struct snd_soc_ops bf5xx_ad73311_ops = { static struct snd_soc_dai_link bf5xx_ad73311_dai = { .name = "ad73311", .stream_name = "AD73311", - .cpu_dai_name = "bf5xx-i2s", + .cpu_dai_name = "bfin-i2s", .codec_dai_name = "ad73311-hifi", - .platform_name = "bfin-pcm-audio", + .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", .ops = &bf5xx_ad73311_ops, }; static struct snd_soc_card bf5xx_ad73311 = { - .name = "bf5xx_ad73311", + .name = "bfin-ad73311", .probe = bf5xx_probe, .dai_link = &bf5xx_ad73311_dai, .num_links = 1, diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 890a0dccf90..50b1df8c141 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -292,7 +292,7 @@ static int __devexit bfin_i2s_soc_platform_remove(struct platform_device *pdev) static struct platform_driver bfin_i2s_pcm_driver = { .driver = { - .name = "bfin-pcm-audio", + .name = "bfin-i2s-pcm-audio", .owner = THIS_MODULE, }, diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 37d9d3c2f2b..dacd86c2ead 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -316,7 +316,7 @@ static struct platform_driver bfin_i2s_driver = { .remove = __devexit_p(bfin_i2s_drv_remove), .driver = { - .name = "bf5xx-i2s", + .name = "bfin-i2s", .owner = THIS_MODULE, }, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 732a02ffe68..8a7b589a6fb 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -116,15 +116,15 @@ static struct snd_soc_ops bf5xx_ssm2602_ops = { static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .name = "ssm2602", .stream_name = "SSM2602", - .cpu_dai_name = "bf5xx-i2s", + .cpu_dai_name = "bfin-i2s", .codec_dai_name = "ssm2602-hifi", - .platform_name = "bf5xx-pcm-audio", + .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, }; static struct snd_soc_card bf5xx_ssm2602 = { - .name = "bf5xx_ssm2602", + .name = "bfin-ssm2602", .dai_link = &bf5xx_ssm2602_dai, .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 74cf759b78a..d1bd745d72c 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -326,7 +326,7 @@ static int __devexit bf5xx_soc_platform_remove(struct platform_device *pdev) static struct platform_driver bfin_tdm_driver = { .driver = { - .name = "bf5xx-tdm-pcm-audio", + .name = "bfin-tdm-pcm-audio", .owner = THIS_MODULE, }, -- cgit v1.2.3 From 2c66cb99d134d787827ed1cd93cc59351ab66a95 Mon Sep 17 00:00:00 2001 From: Barry Song Date: Mon, 28 Mar 2011 01:45:10 -0400 Subject: ASoC: Blackfin: push down SPORT settings from global variables Now that we have multi-component support, take the time to unify the SPORT implementations a bit and make the setup dynamic. This kills off the global sport_handle which was shared across all the Blackfin machine drivers. The pin management aspect is off loaded to platform resources, and now multiple SPORTs can be instantiated simultaneously. Signed-off-by: Barry Song Signed-off-by: Scott Jiang Signed-off-by: Mike Frysinger Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 11 ++- sound/soc/blackfin/bf5xx-ac97.c | 166 ++++++++++++------------------------ sound/soc/blackfin/bf5xx-ad1836.c | 40 ++++----- sound/soc/blackfin/bf5xx-ad193x.c | 40 ++++----- sound/soc/blackfin/bf5xx-ad1980.c | 43 +++++----- sound/soc/blackfin/bf5xx-ad73311.c | 40 ++++----- sound/soc/blackfin/bf5xx-i2s-pcm.c | 21 +++-- sound/soc/blackfin/bf5xx-i2s.c | 159 ++++++++++++++-------------------- sound/soc/blackfin/bf5xx-sport.c | 159 +++++++++++++++++++++++++--------- sound/soc/blackfin/bf5xx-sport.h | 16 ++-- sound/soc/blackfin/bf5xx-ssm2602.c | 40 ++++----- sound/soc/blackfin/bf5xx-tdm-pcm.c | 21 +++-- sound/soc/blackfin/bf5xx-tdm.c | 110 +++++++++--------------- 13 files changed, 416 insertions(+), 450 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index e940d26048d..98b44b316e7 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -243,6 +243,9 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) static int bf5xx_pcm_open(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); struct snd_pcm_runtime *runtime = substream->runtime; int ret; @@ -314,6 +317,9 @@ static struct snd_pcm_ops bf5xx_pcm_ac97_ops = { static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = bf5xx_pcm_hardware.buffer_bytes_max @@ -377,6 +383,9 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) struct snd_dma_buffer *buf; int stream; #if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); size_t size = bf5xx_pcm_hardware.buffer_bytes_max * sizeof(struct ac97_frame) / 4; #endif @@ -405,8 +414,6 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } #endif } - if (sport_handle) - sport_done(sport_handle); } static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index ffbac26b9bc..6d216259088 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -41,48 +41,7 @@ * anomaly does not affect blackfin sound drivers. */ -static int *cmd_count; -static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; - -#define SPORT_REQ(x) \ - [x] = {P_SPORT##x##_TFS, P_SPORT##x##_DTPRI, P_SPORT##x##_TSCLK, \ - P_SPORT##x##_RFS, P_SPORT##x##_DRPRI, P_SPORT##x##_RSCLK, 0} -static u16 sport_req[][7] = { -#ifdef SPORT0_TCR1 - SPORT_REQ(0), -#endif -#ifdef SPORT1_TCR1 - SPORT_REQ(1), -#endif -#ifdef SPORT2_TCR1 - SPORT_REQ(2), -#endif -#ifdef SPORT3_TCR1 - SPORT_REQ(3), -#endif -}; - -#define SPORT_PARAMS(x) \ - [x] = { \ - .dma_rx_chan = CH_SPORT##x##_RX, \ - .dma_tx_chan = CH_SPORT##x##_TX, \ - .err_irq = IRQ_SPORT##x##_ERROR, \ - .regs = (struct sport_register *)SPORT##x##_TCR1, \ - } -static struct sport_param sport_params[4] = { -#ifdef SPORT0_TCR1 - SPORT_PARAMS(0), -#endif -#ifdef SPORT1_TCR1 - SPORT_PARAMS(1), -#endif -#ifdef SPORT2_TCR1 - SPORT_PARAMS(2), -#endif -#ifdef SPORT3_TCR1 - SPORT_PARAMS(3), -#endif -}; +static struct sport_device *ac97_sport_handle; void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, size_t count, unsigned int chan_mask) @@ -140,7 +99,8 @@ static unsigned int sport_tx_curr_frag(struct sport_device *sport) static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) { - struct sport_device *sport = sport_handle; + struct sport_device *sport = ac97_sport_handle; + int *cmd_count = sport->private_data; int nextfrag = sport_tx_curr_frag(sport); struct ac97_frame *nextwrite; @@ -161,6 +121,7 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) static unsigned short bf5xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { + struct sport_device *sport_handle = ac97_sport_handle; struct ac97_frame out_frame[2], in_frame[2]; pr_debug("%s enter 0x%x\n", __func__, reg); @@ -185,6 +146,8 @@ static unsigned short bf5xx_ac97_read(struct snd_ac97 *ac97, void bf5xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { + struct sport_device *sport_handle = ac97_sport_handle; + pr_debug("%s enter 0x%x:0x%04x\n", __func__, reg, val); if (sport_handle->tx_run) { @@ -203,28 +166,19 @@ void bf5xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, static void bf5xx_ac97_warm_reset(struct snd_ac97 *ac97) { -#if defined(CONFIG_BF54x) || defined(CONFIG_BF561) || \ - (defined(BF537_FAMILY) && (CONFIG_SND_BF5XX_SPORT_NUM == 1)) - -#define CONCAT(a, b, c) a ## b ## c -#define BFIN_SPORT_RFS(x) CONCAT(P_SPORT, x, _RFS) - - u16 per = BFIN_SPORT_RFS(CONFIG_SND_BF5XX_SPORT_NUM); - u16 gpio = P_IDENT(BFIN_SPORT_RFS(CONFIG_SND_BF5XX_SPORT_NUM)); + struct sport_device *sport_handle = ac97_sport_handle; + u16 gpio = P_IDENT(sport_handle->pin_req[3]); pr_debug("%s enter\n", __func__); - peripheral_free(per); + peripheral_free_list(sport_handle->pin_req); gpio_request(gpio, "bf5xx-ac97"); gpio_direction_output(gpio, 1); udelay(2); gpio_set_value(gpio, 0); udelay(1); gpio_free(gpio); - peripheral_request(per, "soc-audio"); -#else - pr_info("%s: Not implemented\n", __func__); -#endif + peripheral_request_list(sport_handle->pin_req, "soc-audio"); } static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97) @@ -306,18 +260,32 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) #define bf5xx_ac97_resume NULL #endif -static int bf5xx_ac97_probe(struct snd_soc_dai *dai) +static struct snd_soc_dai_driver bfin_ac97_dai = { + .ac97_control = 1, + .suspend = bf5xx_ac97_suspend, + .resume = bf5xx_ac97_resume, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + .channels_max = 6, +#else + .channels_max = 2, +#endif + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; + +static int __devinit asoc_bfin_ac97_probe(struct platform_device *pdev) { - int ret = 0; - cmd_count = (int *)get_zeroed_page(GFP_KERNEL); - if (cmd_count == NULL) - return -ENOMEM; - - if (peripheral_request_list(sport_req[sport_num], "soc-audio")) { - pr_err("Requesting Peripherals failed\n"); - ret = -EFAULT; - goto peripheral_err; - } + struct sport_device *sport_handle; + int ret; #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET /* Request PB3 as reset pin */ @@ -329,12 +297,14 @@ static int bf5xx_ac97_probe(struct snd_soc_dai *dai) } gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); #endif - sport_handle = sport_init(&sport_params[sport_num], 2, \ - sizeof(struct ac97_frame), NULL); + + sport_handle = sport_init(pdev, 2, sizeof(struct ac97_frame), + PAGE_SIZE); if (!sport_handle) { ret = -ENODEV; goto sport_err; } + /*SPORT works in TDM mode to simulate AC97 transfers*/ #if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 1); @@ -361,67 +331,37 @@ static int bf5xx_ac97_probe(struct snd_soc_dai *dai) goto sport_config_err; } + ret = snd_soc_register_dai(&pdev->dev, &bfin_ac97_dai); + if (ret) { + pr_err("Failed to register DAI: %d\n", ret); + goto sport_config_err; + } + + ac97_sport_handle = sport_handle; + return 0; sport_config_err: - kfree(sport_handle); + sport_done(sport_handle); sport_err: #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); gpio_err: #endif - peripheral_free_list(sport_req[sport_num]); -peripheral_err: - free_page((unsigned long)cmd_count); - cmd_count = NULL; return ret; } -static int bf5xx_ac97_remove(struct snd_soc_dai *dai) +static int __devexit asoc_bfin_ac97_remove(struct platform_device *pdev) { - free_page((unsigned long)cmd_count); - cmd_count = NULL; - peripheral_free_list(sport_req[sport_num]); + struct sport_device *sport_handle = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&pdev->dev); + sport_done(sport_handle); #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); #endif - return 0; -} - -struct snd_soc_dai_driver bfin_ac97_dai = { - .ac97_control = 1, - .probe = bf5xx_ac97_probe, - .remove = bf5xx_ac97_remove, - .suspend = bf5xx_ac97_suspend, - .resume = bf5xx_ac97_resume, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - .channels_max = 6, -#else - .channels_max = 2, -#endif - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, -}; -EXPORT_SYMBOL_GPL(bfin_ac97_dai); - -static __devinit int asoc_bfin_ac97_probe(struct platform_device *pdev) -{ - return snd_soc_register_dai(&pdev->dev, &bfin_ac97_dai); -} -static int __devexit asoc_bfin_ac97_remove(struct platform_device *pdev) -{ - snd_soc_unregister_dai(&pdev->dev); return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 5d9d9e2805b..ea4951cf552 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -29,22 +29,12 @@ #include #include "../codecs/ad1836.h" -#include "bf5xx-sport.h" #include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad1836; -static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - snd_soc_dai_set_drvdata(cpu_dai, sport_handle); - return 0; -} - static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -75,23 +65,33 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops bf5xx_ad1836_ops = { - .startup = bf5xx_ad1836_startup, .hw_params = bf5xx_ad1836_hw_params, }; -static struct snd_soc_dai_link bf5xx_ad1836_dai = { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", - .ops = &bf5xx_ad1836_ops, +static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { + { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai_name = "bfin-tdm.0", + .codec_dai_name = "ad1836-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .codec_name = "ad1836.0", + .ops = &bf5xx_ad1836_ops, + }, + { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai_name = "bfin-tdm.1", + .codec_dai_name = "ad1836-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .codec_name = "ad1836.0", + .ops = &bf5xx_ad1836_ops, + }, }; static struct snd_soc_card bf5xx_ad1836 = { .name = "bfin-ad1836", - .dai_link = &bf5xx_ad1836_dai, + .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 355094fad18..d6651c033cb 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -38,22 +38,12 @@ #include #include "../codecs/ad193x.h" -#include "bf5xx-sport.h" #include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" static struct snd_soc_card bf5xx_ad193x; -static int bf5xx_ad193x_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - snd_soc_dai_set_drvdata(cpu_dai, sport_handle); - return 0; -} - static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -103,23 +93,33 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops bf5xx_ad193x_ops = { - .startup = bf5xx_ad193x_startup, .hw_params = bf5xx_ad193x_hw_params, }; -static struct snd_soc_dai_link bf5xx_ad193x_dai = { - .name = "ad193x", - .stream_name = "AD193X", - .cpu_dai_name = "bfin-tdm", - .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", - .ops = &bf5xx_ad193x_ops, +static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { + { + .name = "ad193x", + .stream_name = "AD193X", + .cpu_dai_name = "bfin-tdm.0", + .codec_dai_name ="ad193x-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .codec_name = "ad193x.5", + .ops = &bf5xx_ad193x_ops, + }, + { + .name = "ad193x", + .stream_name = "AD193X", + .cpu_dai_name = "bfin-tdm.1", + .codec_dai_name ="ad193x-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .codec_name = "ad193x.5", + .ops = &bf5xx_ad193x_ops, + }, }; static struct snd_soc_card bf5xx_ad193x = { .name = "bfin-ad193x", - .dai_link = &bf5xx_ad193x_dai, + .dai_link = &bf5xx_ad193x_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index a3812408c6a..06a84b211b5 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -47,39 +47,34 @@ #include #include "../codecs/ad1980.h" -#include "bf5xx-sport.h" + #include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" static struct snd_soc_card bf5xx_board; -static int bf5xx_board_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - pr_debug("%s enter\n", __func__); - snd_soc_dai_set_drvdata(cpu_dai, sport_handle); - return 0; -} - -static struct snd_soc_ops bf5xx_board_ops = { - .startup = bf5xx_board_startup, -}; - -static struct snd_soc_dai_link bf5xx_board_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "bfin-ac97", - .codec_dai_name = "ad1980-hifi", - .platform_name = "bfin-ac97-pcm-audio", - .codec_name = "ad1980", - .ops = &bf5xx_board_ops, +static struct snd_soc_dai_link bf5xx_board_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai_name = "bfin-ac97.0", + .codec_dai_name = "ad1980-hifi", + .platform_name = "bfin-ac97-pcm-audio", + .codec_name = "ad1980", + }, + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai_name = "bfin-ac97.1", + .codec_dai_name = "ad1980-hifi", + .platform_name = "bfin-ac97-pcm-audio", + .codec_name = "ad1980", + }, }; static struct snd_soc_card bf5xx_board = { .name = "bfin-ad1980", - .dai_link = &bf5xx_board_dai, + .dai_link = &bf5xx_board_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 9f0d4f3c590..732a247f252 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -145,16 +145,6 @@ static int bf5xx_probe(struct platform_device *pdev) return 0; } -static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - pr_debug("%s enter\n", __func__); - snd_soc_dai_set_drvdata(cpu_dai, sport_handle); - return 0; -} - static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -176,24 +166,34 @@ static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, static struct snd_soc_ops bf5xx_ad73311_ops = { - .startup = bf5xx_ad73311_startup, .hw_params = bf5xx_ad73311_hw_params, }; -static struct snd_soc_dai_link bf5xx_ad73311_dai = { - .name = "ad73311", - .stream_name = "AD73311", - .cpu_dai_name = "bfin-i2s", - .codec_dai_name = "ad73311-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, +static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { + { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "ad73311-hifi", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "ad73311", + .ops = &bf5xx_ad73311_ops, + }, + { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai_name = "bfin-i2s.1", + .codec_dai_name = "ad73311-hifi", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "ad73311", + .ops = &bf5xx_ad73311_ops, + }, }; static struct snd_soc_card bf5xx_ad73311 = { .name = "bfin-ad73311", .probe = bf5xx_probe, - .dai_link = &bf5xx_ad73311_dai, + .dai_link = &bf5xx_ad73311_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 50b1df8c141..b5101efd1c8 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -148,10 +148,15 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) static int bf5xx_pcm_open(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = &substream->dma_buffer; int ret; pr_debug("%s enter\n", __func__); + snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); ret = snd_pcm_hw_constraint_integer(runtime, \ @@ -159,9 +164,14 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) goto out; - if (sport_handle != NULL) + if (sport_handle != NULL) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_handle->tx_buf = buf->area; + else + sport_handle->rx_buf = buf->area; + runtime->private_data = sport_handle; - else { + } else { pr_err("sport_handle is NULL\n"); return -1; } @@ -214,11 +224,6 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) pr_debug("%s, area:%p, size:0x%08lx\n", __func__, buf->area, buf->bytes); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - return 0; } @@ -239,8 +244,6 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) dma_free_coherent(NULL, buf->bytes, buf->area, 0); buf->area = NULL; } - if (sport_handle) - sport_done(sport_handle); } static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index dacd86c2ead..00cc3e00b2f 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -51,59 +51,24 @@ struct bf5xx_i2s_port { int configured; }; -static struct bf5xx_i2s_port bf5xx_i2s; -static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; - -static struct sport_param sport_params[2] = { - { - .dma_rx_chan = CH_SPORT0_RX, - .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERROR, - .regs = (struct sport_register *)SPORT0_TCR1, - }, - { - .dma_rx_chan = CH_SPORT1_RX, - .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERROR, - .regs = (struct sport_register *)SPORT1_TCR1, - } -}; - -/* - * Setting the TFS pin selector for SPORT 0 based on whether the selected - * port id F or G. If the port is F then no conflict should exist for the - * TFS. When Port G is selected and EMAC then there is a conflict between - * the PHY interrupt line and TFS. Current settings prevent the conflict - * by ignoring the TFS pin when Port G is selected. This allows both - * codecs and EMAC using Port G concurrently. - */ -#ifdef CONFIG_BF527_SPORT0_PORTG -#define LOCAL_SPORT0_TFS (0) -#else -#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) -#endif - -static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, - {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, - P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; - static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; int ret = 0; /* interface format:support I2S,slave mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - bf5xx_i2s.tcr1 |= TFSR | TCKFE; - bf5xx_i2s.rcr1 |= RFSR | RCKFE; - bf5xx_i2s.tcr2 |= TSFSE; - bf5xx_i2s.rcr2 |= RSFSE; + bf5xx_i2s->tcr1 |= TFSR | TCKFE; + bf5xx_i2s->rcr1 |= RFSR | RCKFE; + bf5xx_i2s->tcr2 |= TSFSE; + bf5xx_i2s->rcr2 |= RSFSE; break; case SND_SOC_DAIFMT_DSP_A: - bf5xx_i2s.tcr1 |= TFSR; - bf5xx_i2s.rcr1 |= RFSR; + bf5xx_i2s->tcr1 |= TFSR; + bf5xx_i2s->rcr1 |= RFSR; break; case SND_SOC_DAIFMT_LEFT_J: ret = -EINVAL; @@ -135,33 +100,35 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; int ret = 0; - bf5xx_i2s.tcr2 &= ~0x1f; - bf5xx_i2s.rcr2 &= ~0x1f; + bf5xx_i2s->tcr2 &= ~0x1f; + bf5xx_i2s->rcr2 &= ~0x1f; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: bf5xx_i2s->tcr2 |= 7; bf5xx_i2s->rcr2 |= 7; sport_handle->wdsize = 1; case SNDRV_PCM_FORMAT_S16_LE: - bf5xx_i2s.tcr2 |= 15; - bf5xx_i2s.rcr2 |= 15; + bf5xx_i2s->tcr2 |= 15; + bf5xx_i2s->rcr2 |= 15; sport_handle->wdsize = 2; break; case SNDRV_PCM_FORMAT_S24_LE: - bf5xx_i2s.tcr2 |= 23; - bf5xx_i2s.rcr2 |= 23; + bf5xx_i2s->tcr2 |= 23; + bf5xx_i2s->rcr2 |= 23; sport_handle->wdsize = 3; break; case SNDRV_PCM_FORMAT_S32_LE: - bf5xx_i2s.tcr2 |= 31; - bf5xx_i2s.rcr2 |= 31; + bf5xx_i2s->tcr2 |= 31; + bf5xx_i2s->rcr2 |= 31; sport_handle->wdsize = 4; break; } - if (!bf5xx_i2s.configured) { + if (!bf5xx_i2s->configured) { /* * TX and RX are not independent,they are enabled at the * same time, even if only one side is running. So, we @@ -170,16 +137,16 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, * * CPU DAI:slave mode. */ - bf5xx_i2s.configured = 1; - ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, - bf5xx_i2s.rcr2, 0, 0); + bf5xx_i2s->configured = 1; + ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, + bf5xx_i2s->rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, - bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, + bf5xx_i2s->tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -192,41 +159,19 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; + pr_debug("%s enter\n", __func__); /* No active stream, SPORT is allowed to be configured again. */ if (!dai->active) - bf5xx_i2s.configured = 0; -} - -static int bf5xx_i2s_probe(struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { - pr_err("Requesting Peripherals failed\n"); - return -EFAULT; - } - - /* request DMA for SPORT */ - sport_handle = sport_init(&sport_params[sport_num], 4, \ - 2 * sizeof(u32), NULL); - if (!sport_handle) { - peripheral_free_list(&sport_req[sport_num][0]); - return -ENODEV; - } - - return 0; -} - -static int bf5xx_i2s_remove(struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - peripheral_free_list(&sport_req[sport_num][0]); - return 0; + bf5xx_i2s->configured = 0; } #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); pr_debug("%s : sport %d\n", __func__, dai->id); @@ -239,19 +184,21 @@ static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) static int bf5xx_i2s_resume(struct snd_soc_dai *dai) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; int ret; pr_debug("%s : sport %d\n", __func__, dai->id); - ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, - bf5xx_i2s.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, + bf5xx_i2s->rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, - bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, + bf5xx_i2s->tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -283,8 +230,6 @@ static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { }; static struct snd_soc_dai_driver bf5xx_i2s_dai = { - .probe = bf5xx_i2s_probe, - .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { @@ -300,21 +245,43 @@ static struct snd_soc_dai_driver bf5xx_i2s_dai = { .ops = &bf5xx_i2s_dai_ops, }; -static int bfin_i2s_drv_probe(struct platform_device *pdev) +static int __devinit bf5xx_i2s_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &bf5xx_i2s_dai); + struct sport_device *sport_handle; + int ret; + + /* configure SPORT for I2S */ + sport_handle = sport_init(pdev, 4, 2 * sizeof(u32), + sizeof(struct bf5xx_i2s_port)); + if (!sport_handle) + return -ENODEV; + + /* register with the ASoC layers */ + ret = snd_soc_register_dai(&pdev->dev, &bf5xx_i2s_dai); + if (ret) { + pr_err("Failed to register DAI: %d\n", ret); + sport_done(sport_handle); + return ret; + } + + return 0; } -static int __devexit bfin_i2s_drv_remove(struct platform_device *pdev) +static int __devexit bf5xx_i2s_remove(struct platform_device *pdev) { + struct sport_device *sport_handle = platform_get_drvdata(pdev); + + pr_debug("%s enter\n", __func__); + snd_soc_unregister_dai(&pdev->dev); + sport_done(sport_handle); + return 0; } static struct platform_driver bfin_i2s_driver = { - .probe = bfin_i2s_drv_probe, - .remove = __devexit_p(bfin_i2s_drv_remove), - + .probe = bf5xx_i2s_probe, + .remove = __devexit_p(bf5xx_i2s_remove), .driver = { .name = "bfin-i2s", .owner = THIS_MODULE, diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 99051ff0954..a2d40349fcc 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -42,8 +42,6 @@ /* delay between frame sync pulse and first data bit in multichannel mode */ #define FRAME_DELAY (1<<12) -struct sport_device *sport_handle; -EXPORT_SYMBOL(sport_handle); /* note: multichannel is in units of 8 channels, * tdm_count is # channels NOT / 8 ! */ int sport_set_multichannel(struct sport_device *sport, @@ -798,86 +796,164 @@ int sport_set_err_callback(struct sport_device *sport, } EXPORT_SYMBOL(sport_set_err_callback); -struct sport_device *sport_init(struct sport_param *param, unsigned wdsize, - unsigned dummy_count, void *private_data) +static int sport_config_pdev(struct platform_device *pdev, struct sport_param *param) { - int ret; + /* Extract settings from platform data */ + struct device *dev = &pdev->dev; + struct bfin_snd_platform_data *pdata = dev->platform_data; + struct resource *res; + + param->num = pdev->id; + + if (!pdata) { + dev_err(dev, "no platform_data\n"); + return -ENODEV; + } + param->pin_req = pdata->pin_req; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "no MEM resource\n"); + return -ENODEV; + } + param->regs = (struct sport_register *)res->start; + + /* first RX, then TX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dev, "no rx DMA resource\n"); + return -ENODEV; + } + param->dma_rx_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(dev, "no tx DMA resource\n"); + return -ENODEV; + } + param->dma_tx_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(dev, "no irq resource\n"); + return -ENODEV; + } + param->err_irq = res->start; + + return 0; +} + +struct sport_device *sport_init(struct platform_device *pdev, + unsigned int wdsize, unsigned int dummy_count, size_t priv_size) +{ + struct device *dev = &pdev->dev; + struct sport_param param; struct sport_device *sport; - pr_debug("%s enter\n", __func__); - BUG_ON(param == NULL); - BUG_ON(wdsize == 0 || dummy_count == 0); - sport = kmalloc(sizeof(struct sport_device), GFP_KERNEL); - if (!sport) { - pr_err("Failed to allocate for sport device\n"); + int ret; + + dev_dbg(dev, "%s enter\n", __func__); + + param.wdsize = wdsize; + param.dummy_count = dummy_count; + BUG_ON(param.wdsize == 0 || param.dummy_count == 0); + + ret = sport_config_pdev(pdev, ¶m); + if (ret) + return NULL; + + if (peripheral_request_list(param.pin_req, "soc-audio")) { + dev_err(dev, "requesting Peripherals failed\n"); return NULL; } - memset(sport, 0, sizeof(struct sport_device)); - sport->dma_rx_chan = param->dma_rx_chan; - sport->dma_tx_chan = param->dma_tx_chan; - sport->err_irq = param->err_irq; - sport->regs = param->regs; - sport->private_data = private_data; + sport = kzalloc(sizeof(*sport), GFP_KERNEL); + if (!sport) { + dev_err(dev, "failed to allocate for sport device\n"); + goto __init_err0; + } + + sport->num = param.num; + sport->dma_rx_chan = param.dma_rx_chan; + sport->dma_tx_chan = param.dma_tx_chan; + sport->err_irq = param.err_irq; + sport->regs = param.regs; + sport->pin_req = param.pin_req; if (request_dma(sport->dma_rx_chan, "SPORT RX Data") == -EBUSY) { - pr_err("Failed to request RX dma %d\n", \ - sport->dma_rx_chan); + dev_err(dev, "failed to request RX dma %d\n", sport->dma_rx_chan); goto __init_err1; } if (set_dma_callback(sport->dma_rx_chan, rx_handler, sport) != 0) { - pr_err("Failed to request RX irq %d\n", \ - sport->dma_rx_chan); + dev_err(dev, "failed to request RX irq %d\n", sport->dma_rx_chan); goto __init_err2; } if (request_dma(sport->dma_tx_chan, "SPORT TX Data") == -EBUSY) { - pr_err("Failed to request TX dma %d\n", \ - sport->dma_tx_chan); + dev_err(dev, "failed to request TX dma %d\n", sport->dma_tx_chan); goto __init_err2; } if (set_dma_callback(sport->dma_tx_chan, tx_handler, sport) != 0) { - pr_err("Failed to request TX irq %d\n", \ - sport->dma_tx_chan); + dev_err(dev, "failed to request TX irq %d\n", sport->dma_tx_chan); goto __init_err3; } if (request_irq(sport->err_irq, err_handler, IRQF_SHARED, "SPORT err", sport) < 0) { - pr_err("Failed to request err irq:%d\n", \ - sport->err_irq); + dev_err(dev, "failed to request err irq %d\n", sport->err_irq); goto __init_err3; } - pr_err("dma rx:%d tx:%d, err irq:%d, regs:%p\n", + dev_info(dev, "dma rx:%d tx:%d, err irq:%d, regs:%p\n", sport->dma_rx_chan, sport->dma_tx_chan, sport->err_irq, sport->regs); - sport->wdsize = wdsize; - sport->dummy_count = dummy_count; + sport->wdsize = param.wdsize; + sport->dummy_count = param.dummy_count; + + sport->private_data = kzalloc(priv_size, GFP_KERNEL); + if (!sport->private_data) { + dev_err(dev, "could not alloc priv data %zu bytes\n", priv_size); + goto __init_err4; + } if (L1_DATA_A_LENGTH) - sport->dummy_buf = l1_data_sram_zalloc(dummy_count * 2); + sport->dummy_buf = l1_data_sram_zalloc(param.dummy_count * 2); else - sport->dummy_buf = kzalloc(dummy_count * 2, GFP_KERNEL); + sport->dummy_buf = kzalloc(param.dummy_count * 2, GFP_KERNEL); if (sport->dummy_buf == NULL) { - pr_err("Failed to allocate dummy buffer\n"); - goto __error; + dev_err(dev, "failed to allocate dummy buffer\n"); + goto __error1; } ret = sport_config_rx_dummy(sport); if (ret) { - pr_err("Failed to config rx dummy ring\n"); - goto __error; + dev_err(dev, "failed to config rx dummy ring\n"); + goto __error2; } ret = sport_config_tx_dummy(sport); if (ret) { - pr_err("Failed to config tx dummy ring\n"); - goto __error; + dev_err(dev, "failed to config tx dummy ring\n"); + goto __error3; } + platform_set_drvdata(pdev, sport); + return sport; -__error: +__error3: + if (L1_DATA_A_LENGTH) + l1_data_sram_free(sport->dummy_rx_desc); + else + dma_free_coherent(NULL, 2*sizeof(struct dmasg), + sport->dummy_rx_desc, 0); +__error2: + if (L1_DATA_A_LENGTH) + l1_data_sram_free(sport->dummy_buf); + else + kfree(sport->dummy_buf); +__error1: + kfree(sport->private_data); +__init_err4: free_irq(sport->err_irq, sport); __init_err3: free_dma(sport->dma_tx_chan); @@ -885,6 +961,8 @@ __init_err2: free_dma(sport->dma_rx_chan); __init_err1: kfree(sport); +__init_err0: + peripheral_free_list(param.pin_req); return NULL; } EXPORT_SYMBOL(sport_init); @@ -917,8 +995,9 @@ void sport_done(struct sport_device *sport) free_dma(sport->dma_tx_chan); free_irq(sport->err_irq, sport); + kfree(sport->private_data); + peripheral_free_list(sport->pin_req); kfree(sport); - sport = NULL; } EXPORT_SYMBOL(sport_done); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index a86e8cc0b2d..5ab60bd613e 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -1,5 +1,5 @@ /* - * File: bf5xx_ac97_sport.h + * File: bf5xx_sport.h * Based on: * Author: Roy Huang * @@ -33,15 +33,18 @@ #include #include #include +#include #include #include #define DESC_ELEMENT_COUNT 9 struct sport_device { + int num; int dma_rx_chan; int dma_tx_chan; int err_irq; + const unsigned short *pin_req; struct sport_register *regs; unsigned char *rx_buf; @@ -103,17 +106,20 @@ struct sport_device { void *private_data; }; -extern struct sport_device *sport_handle; - struct sport_param { + int num; int dma_rx_chan; int dma_tx_chan; int err_irq; + const unsigned short *pin_req; struct sport_register *regs; + unsigned int wdsize; + unsigned int dummy_count; + void *private_data; }; -struct sport_device *sport_init(struct sport_param *param, unsigned wdsize, - unsigned dummy_count, void *private_data); +struct sport_device *sport_init(struct platform_device *pdev, + unsigned int wdsize, unsigned int dummy_count, size_t priv_size); void sport_done(struct sport_device *sport); diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 8a7b589a6fb..767e772a815 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -44,16 +44,6 @@ static struct snd_soc_card bf5xx_ssm2602; -static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - pr_debug("%s enter\n", __func__); - snd_soc_dai_set_drvdata(cpu_dai, sport_handle); - return 0; -} - static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -109,23 +99,33 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops bf5xx_ssm2602_ops = { - .startup = bf5xx_ssm2602_startup, .hw_params = bf5xx_ssm2602_hw_params, }; -static struct snd_soc_dai_link bf5xx_ssm2602_dai = { - .name = "ssm2602", - .stream_name = "SSM2602", - .cpu_dai_name = "bfin-i2s", - .codec_dai_name = "ssm2602-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ssm2602.0-001b", - .ops = &bf5xx_ssm2602_ops, +static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { + { + .name = "ssm2602", + .stream_name = "SSM2602", + .cpu_dai_name = "bfin-i2s.0", + .codec_dai_name = "ssm2602-hifi", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "ssm2602.0-001b", + .ops = &bf5xx_ssm2602_ops, + }, + { + .name = "ssm2602", + .stream_name = "SSM2602", + .cpu_dai_name = "bfin-i2s.1", + .codec_dai_name = "ssm2602-hifi", + .platform_name = "bfin-i2s-pcm-audio", + .codec_name = "ssm2602.0-001b", + .ops = &bf5xx_ssm2602_ops, + }, }; static struct snd_soc_card bf5xx_ssm2602 = { .name = "bfin-ssm2602", - .dai_link = &bf5xx_ssm2602_dai, + .dai_link = &bf5xx_ssm2602_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, }; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index d1bd745d72c..07cfc7a9e49 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -154,7 +154,12 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) static int bf5xx_pcm_open(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = &substream->dma_buffer; + int ret = 0; snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); @@ -164,9 +169,14 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) goto out; - if (sport_handle != NULL) + if (sport_handle != NULL) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_handle->tx_buf = buf->area; + else + sport_handle->rx_buf = buf->area; + runtime->private_data = sport_handle; - else { + } else { pr_err("sport_handle is NULL\n"); ret = -ENODEV; } @@ -249,11 +259,6 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) } buf->bytes = size; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - return 0; } @@ -274,8 +279,6 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) dma_free_coherent(NULL, buf->bytes, buf->area, 0); buf->area = NULL; } - if (sport_handle) - sport_done(sport_handle); } static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 5515ac9e05c..a822d1ee138 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,43 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -static struct bf5xx_tdm_port bf5xx_tdm; -static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; - -static struct sport_param sport_params[2] = { - { - .dma_rx_chan = CH_SPORT0_RX, - .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERROR, - .regs = (struct sport_register *)SPORT0_TCR1, - }, - { - .dma_rx_chan = CH_SPORT1_RX, - .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERROR, - .regs = (struct sport_register *)SPORT1_TCR1, - } -}; - -/* - * Setting the TFS pin selector for SPORT 0 based on whether the selected - * port id F or G. If the port is F then no conflict should exist for the - * TFS. When Port G is selected and EMAC then there is a conflict between - * the PHY interrupt line and TFS. Current settings prevent the conflict - * by ignoring the TFS pin when Port G is selected. This allows both - * codecs and EMAC using Port G concurrently. - */ -#ifdef CONFIG_BF527_SPORT0_PORTG -#define LOCAL_SPORT0_TFS (0) -#else -#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) -#endif - -static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, - {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, - P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; - static int bf5xx_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -119,14 +82,16 @@ static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; int ret = 0; - bf5xx_tdm.tcr2 &= ~0x1f; - bf5xx_tdm.rcr2 &= ~0x1f; + bf5xx_tdm->tcr2 &= ~0x1f; + bf5xx_tdm->rcr2 &= ~0x1f; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S32_LE: - bf5xx_tdm.tcr2 |= 31; - bf5xx_tdm.rcr2 |= 31; + bf5xx_tdm->tcr2 |= 31; + bf5xx_tdm->rcr2 |= 31; sport_handle->wdsize = 4; break; /* at present, we only support 32bit transfer */ @@ -136,7 +101,7 @@ static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, break; } - if (!bf5xx_tdm.configured) { + if (!bf5xx_tdm->configured) { /* * TX and RX are not independent,they are enabled at the * same time, even if only one side is running. So, we @@ -145,21 +110,21 @@ static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, * * CPU DAI:slave mode. */ - ret = sport_config_rx(sport_handle, bf5xx_tdm.rcr1, - bf5xx_tdm.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_tdm->rcr1, + bf5xx_tdm->rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, bf5xx_tdm.tcr1, - bf5xx_tdm.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_tdm->tcr1, + bf5xx_tdm->tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - bf5xx_tdm.configured = 1; + bf5xx_tdm->configured = 1; } return 0; @@ -168,15 +133,20 @@ static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; + /* No active stream, SPORT is allowed to be configured again. */ if (!dai->active) - bf5xx_tdm.configured = 0; + bf5xx_tdm->configured = 0; } static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot) { + struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); + struct bf5xx_tdm_port *bf5xx_tdm = sport_handle->private_data; int i; unsigned int slot; unsigned int tx_mapped = 0, rx_mapped = 0; @@ -189,7 +159,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, slot = tx_slot[i]; if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && (!(tx_mapped & (1 << slot)))) { - bf5xx_tdm.tx_map[i] = slot; + bf5xx_tdm->tx_map[i] = slot; tx_mapped |= 1 << slot; } else return -EINVAL; @@ -198,7 +168,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, slot = rx_slot[i]; if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && (!(rx_mapped & (1 << slot)))) { - bf5xx_tdm.rx_map[i] = slot; + bf5xx_tdm->rx_map[i] = slot; rx_mapped |= 1 << slot; } else return -EINVAL; @@ -212,12 +182,14 @@ static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - if (!dai->active) - return 0; - if (dai->capture_active) - sport_rx_stop(sport); if (dai->playback_active) sport_tx_stop(sport); + if (dai->capture_active) + sport_rx_stop(sport); + + /* isolate sync/clock pins from codec while sports resume */ + peripheral_free_list(sport->pin_req); + return 0; } @@ -226,9 +198,6 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai) int ret; struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - if (!dai->active) - return 0; - ret = sport_set_multichannel(sport, 8, 0xFF, 1); if (ret) { pr_err("SPORT is busy!\n"); @@ -247,6 +216,8 @@ static int bf5xx_tdm_resume(struct snd_soc_dai *dai) ret = -EBUSY; } + peripheral_request_list(sport->pin_req, "soc-audio"); + return 0; } @@ -280,20 +251,14 @@ static struct snd_soc_dai_driver bf5xx_tdm_dai = { static int __devinit bfin_tdm_probe(struct platform_device *pdev) { - int ret = 0; - - if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { - pr_err("Requesting Peripherals failed\n"); - return -EFAULT; - } + struct sport_device *sport_handle; + int ret; - /* request DMA for SPORT */ - sport_handle = sport_init(&sport_params[sport_num], 4, \ - 8 * sizeof(u32), NULL); - if (!sport_handle) { - peripheral_free_list(&sport_req[sport_num][0]); + /* configure SPORT for TDM */ + sport_handle = sport_init(pdev, 4, 8 * sizeof(u32), + sizeof(struct bf5xx_tdm_port)); + if (!sport_handle) return -ENODEV; - } /* SPORT works in TDM mode */ ret = sport_set_multichannel(sport_handle, 8, 0xFF, 1); @@ -323,18 +288,19 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) goto sport_config_err; } - sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: - peripheral_free_list(&sport_req[sport_num][0]); + sport_done(sport_handle); return ret; } static int __devexit bfin_tdm_remove(struct platform_device *pdev) { - peripheral_free_list(&sport_req[sport_num][0]); + struct sport_device *sport_handle = platform_get_drvdata(pdev); + snd_soc_unregister_dai(&pdev->dev); + sport_done(sport_handle); return 0; } -- cgit v1.2.3 From fbda18245beff52ec1edf096e7a77d4cdf3effb9 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 28 Mar 2011 11:39:14 +0100 Subject: ASoC: soc-cache: Fix indentation issues Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1210a6f70a9..22c0b95fb00 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -93,13 +93,13 @@ static unsigned int do_hw_read(struct snd_soc_codec *codec, unsigned int reg) } static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[2]; @@ -111,7 +111,7 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_4_12_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[2]; @@ -209,7 +209,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, #if defined(CONFIG_SPI_MASTER) static int snd_soc_8_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -257,7 +257,7 @@ static unsigned int do_i2c_read(struct snd_soc_codec *codec, #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) + unsigned int r) { u8 reg = r; u8 data; @@ -307,13 +307,13 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #endif static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, - unsigned int reg) + unsigned int reg) { return do_hw_read(codec, reg); } static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) + unsigned int value) { u8 data[3]; @@ -327,7 +327,7 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_8_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[3]; @@ -379,7 +379,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, #if defined(CONFIG_SPI_MASTER) static int snd_soc_16_16_spi_write(void *control_data, const char *data, - int len) + int len) { u8 msg[4]; -- cgit v1.2.3 From f20eda5d8fafe35ca3c63ec8149a700a7db228a5 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 28 Mar 2011 11:39:15 +0100 Subject: ASoC: soc-cache: Warn on syncing any non-writable registers When syncing the cache, if the driver has given us a writable_register() callback, use it to check if we are syncing a non-writable register and if so warn the user. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 22c0b95fb00..22b09907622 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -661,6 +661,8 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); if (rbnode->value == rbnode->defval) continue; + WARN_ON(codec->writable_register && + codec->writable_register(codec, rbnode->reg)); ret = snd_soc_cache_read(codec, rbnode->reg, &val); if (ret) return ret; @@ -921,6 +923,8 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) lzo_blocks = codec->reg_cache; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { + WARN_ON(codec->writable_register && + codec->writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; @@ -1179,6 +1183,8 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) codec_drv = codec->driver; for (i = 0; i < codec_drv->reg_cache_size; ++i) { + WARN_ON(codec->writable_register && + codec->writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; -- cgit v1.2.3 From fc3b9230cc2cdbcbef37953fe8cf5af0977e7a95 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:08 +0800 Subject: ASoC: Use data based init for ak4671 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 2ec75abfa3e..88b29f8c748 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -352,7 +352,7 @@ static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route ak4671_intercon[] = { {"DAC Left", "NULL", "PMPLL"}, {"DAC Right", "NULL", "PMPLL"}, {"ADC Left", "NULL", "PMPLL"}, @@ -433,17 +433,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, }; -static int ak4671_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, ak4671_dapm_widgets, - ARRAY_SIZE(ak4671_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static int ak4671_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -650,7 +639,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, ak4671_snd_controls, ARRAY_SIZE(ak4671_snd_controls)); - ak4671_add_widgets(codec); ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -670,6 +658,10 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { .reg_cache_size = AK4671_CACHEREGNUM, .reg_word_size = sizeof(u8), .reg_cache_default = ak4671_reg, + .dapm_widgets = ak4671_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4671_dapm_widgets), + .dapm_routes = ak4671_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4671_intercon), }; static int __devinit ak4671_i2c_probe(struct i2c_client *client, -- cgit v1.2.3 From 51fb1a8704f6bccfcf5248f9edc2751c272fbdf2 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:09 +0800 Subject: ASoC: Use data based init for cx20442 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cx20442.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 0bb424af956..d68ea532cc7 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -86,18 +86,6 @@ static const struct snd_soc_dapm_route cx20442_audio_map[] = { {"ADC", NULL, "Input Mixer"}, }; -static int cx20442_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, cx20442_dapm_widgets, - ARRAY_SIZE(cx20442_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, cx20442_audio_map, - ARRAY_SIZE(cx20442_audio_map)); - - return 0; -} - static unsigned int cx20442_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { @@ -344,8 +332,6 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, cx20442); - cx20442_add_widgets(codec); - cx20442->control_data = NULL; codec->hw_write = NULL; codec->card->pop_time = 0; @@ -377,6 +363,10 @@ static struct snd_soc_codec_driver cx20442_codec_dev = { .reg_word_size = sizeof(u8), .read = cx20442_read_reg_cache, .write = cx20442_write, + .dapm_widgets = cx20442_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cx20442_dapm_widgets), + .dapm_routes = cx20442_audio_map, + .num_dapm_routes = ARRAY_SIZE(cx20442_audio_map), }; static int cx20442_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From dc6fc49b141ed14256e1319ccb9797e9825f1734 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:10 +0800 Subject: ASoC: Use data based init for max98088 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 25 +++++++------------------ 1 file changed, 7 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index bd0517cb798..bb58bdb4985 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1112,7 +1112,7 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = { SND_SOC_DAPM_INPUT("INB2"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route max98088_audio_map[] = { /* Left headphone output mixer */ {"Left HP Mixer", "Left DAC1 Switch", "DACL1"}, {"Left HP Mixer", "Left DAC2 Switch", "DACL2"}, @@ -1226,22 +1226,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MIC2 Input", NULL, "MIC2"}, }; -static int max98088_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, max98088_dapm_widgets, - ARRAY_SIZE(max98088_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - snd_soc_add_controls(codec, max98088_snd_controls, - ARRAY_SIZE(max98088_snd_controls)); - - snd_soc_dapm_new_widgets(dapm); - return 0; -} - /* codec mclk clock divider coefficients */ static const struct { u32 rate; @@ -2010,7 +1994,8 @@ static int max98088_probe(struct snd_soc_codec *codec) max98088_handle_pdata(codec); - max98088_add_widgets(codec); + snd_soc_add_controls(codec, max98088_snd_controls, + ARRAY_SIZE(max98088_snd_controls)); err_access: return ret; @@ -2036,6 +2021,10 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { .reg_word_size = sizeof(u8), .reg_cache_default = max98088_reg, .volatile_register = max98088_volatile_register, + .dapm_widgets = max98088_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), + .dapm_routes = max98088_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98088_audio_map), }; static int max98088_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3 From 7ad7dd125ae936719205672565ef48df018731c4 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:11 +0800 Subject: ASoC: Use data based init for wm9713 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 38ed9855871..7167cb6787d 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -487,7 +487,7 @@ SND_SOC_DAPM_INPUT("MIC2B"), SND_SOC_DAPM_VMID("VMID"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm9713_audio_map[] = { /* left HP mixer */ {"Left HP Mixer", "Beep Playback Switch", "PCBEEP"}, {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, @@ -644,18 +644,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Capture Mono Mux", "Right", "Right Capture Source"}, }; -static int wm9713_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm9713_dapm_widgets, - ARRAY_SIZE(wm9713_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -1231,7 +1219,6 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm9713_snd_ac97_controls, ARRAY_SIZE(wm9713_snd_ac97_controls)); - wm9713_add_widgets(codec); return 0; @@ -1262,6 +1249,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9713 = { .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9713_reg, + .dapm_widgets = wm9713_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm9713_dapm_widgets), + .dapm_routes = wm9713_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm9713_audio_map), }; static __devinit int wm9713_probe(struct platform_device *pdev) -- cgit v1.2.3 From 983347785bb3807c125e7c00f31790ed11c231fc Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:12 +0800 Subject: ASoC: Use data based init for wm9712 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index bf5d4ef1a2a..90117f8156e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -332,7 +332,7 @@ SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm9712_audio_map[] = { /* virtual mixer - mixes left & right channels for spk and mono */ {"AC97 Mixer", NULL, "Left DAC"}, {"AC97 Mixer", NULL, "Right DAC"}, @@ -429,17 +429,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"ROUT2", NULL, "Speaker PGA"}, }; -static int wm9712_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm9712_dapm_widgets, - ARRAY_SIZE(wm9712_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -651,7 +640,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_add_controls(codec, wm9712_snd_ac97_controls, ARRAY_SIZE(wm9712_snd_ac97_controls)); - wm9712_add_widgets(codec); return 0; @@ -678,6 +666,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9712_reg, + .dapm_widgets = wm9712_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm9712_dapm_widgets), + .dapm_routes = wm9712_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm9712_audio_map), }; static __devinit int wm9712_probe(struct platform_device *pdev) -- cgit v1.2.3 From 97a58d6e1ba74c6c3663b50960a521c54c2f3c90 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:13 +0800 Subject: ASoC: Use data based init for wm9705 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 47b357adabd..646b58dda84 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -142,7 +142,7 @@ static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = { * constantly enabled, we use the mutes on those inputs to simulate such * controls. */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm9705_audio_map[] = { /* HP mixer */ {"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"}, {"HP Mixer", "CD Playback Switch", "CD PGA"}, @@ -200,17 +200,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right ADC", NULL, "ADC PGA"}, }; -static int wm9705_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm9705_dapm_widgets, - ARRAY_SIZE(wm9705_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* We use a register cache to enhance read performance. */ static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -364,7 +353,6 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm9705_snd_ac97_controls, ARRAY_SIZE(wm9705_snd_ac97_controls)); - wm9705_add_widgets(codec); return 0; @@ -390,6 +378,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9705 = { .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9705_reg, + .dapm_widgets = wm9705_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm9705_dapm_widgets), + .dapm_routes = wm9705_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm9705_audio_map), }; static __devinit int wm9705_probe(struct platform_device *pdev) -- cgit v1.2.3 From 8428edf96ebb9e8c8a5bec257568101b6ba01dc3 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:14 +0800 Subject: ASoC: Use data based init for wm8728 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8728.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 736b0352d0a..86d4718d3a7 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -65,22 +65,11 @@ SND_SOC_DAPM_OUTPUT("VOUTL"), SND_SOC_DAPM_OUTPUT("VOUTR"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8728_intercon[] = { {"VOUTL", NULL, "DAC"}, {"VOUTR", NULL, "DAC"}, }; -static int wm8728_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8728_dapm_widgets, - ARRAY_SIZE(wm8728_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static int wm8728_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -255,7 +244,6 @@ static int wm8728_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8728_snd_controls, ARRAY_SIZE(wm8728_snd_controls)); - wm8728_add_widgets(codec); return ret; } @@ -275,6 +263,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = { .reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm8728_reg_defaults, + .dapm_widgets = wm8728_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8728_dapm_widgets), + .dapm_routes = wm8728_intercon, + .num_dapm_routes = ARRAY_SIZE(wm8728_intercon), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From 203140006173ec1df42a05d8eb7051d82eb61c25 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:15 +0800 Subject: ASoC: Use data based init for wm8711 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8711.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 97c30382d3f..a537e4af6ae 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -77,7 +77,7 @@ SND_SOC_DAPM_OUTPUT("ROUT"), SND_SOC_DAPM_OUTPUT("RHPOUT"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route wm8711_intercon[] = { /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, @@ -89,17 +89,6 @@ static const struct snd_soc_dapm_route intercon[] = { {"LOUT", NULL, "Output Mixer"}, }; -static int wm8711_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8711_dapm_widgets, - ARRAY_SIZE(wm8711_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - struct _coeff_div { u32 mclk; u32 rate; @@ -398,7 +387,6 @@ static int wm8711_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm8711_snd_controls, ARRAY_SIZE(wm8711_snd_controls)); - wm8711_add_widgets(codec); return ret; @@ -420,6 +408,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = { .reg_cache_size = ARRAY_SIZE(wm8711_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8711_reg, + .dapm_widgets = wm8711_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8711_dapm_widgets), + .dapm_routes = wm8711_intercon, + .num_dapm_routes = ARRAY_SIZE(wm8711_intercon), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From a7dca707a73ca700124e04372fc91334d898e6a6 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:16 +0800 Subject: ASoC: Use data based init for tlv320aic23 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 54a30ef0ec8..33bb52f3f68 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -212,7 +212,7 @@ static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICIN"), }; -static const struct snd_soc_dapm_route intercon[] = { +static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* Output Mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "Playback Switch", "DAC"}, @@ -388,18 +388,6 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, return 0; } -static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - /* set up audio path interconnects */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - return 0; -} - static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -676,7 +664,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); - tlv320aic23_add_widgets(codec); return 0; } @@ -698,6 +685,10 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .read = tlv320aic23_read_reg_cache, .write = tlv320aic23_write, .set_bias_level = tlv320aic23_set_bias_level, + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = tlv320aic23_intercon, + .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.3 From b63c963da21d05f40f20de0a586143922bfa19d9 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:07 +0800 Subject: ASoC: Use data based init for ak4535 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 8b38739c88f..e1a214ee757 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -230,7 +230,7 @@ static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { SND_SOC_DAPM_INPUT("AIN"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route ak4535_audio_map[] = { /*stereo mixer */ {"Stereo Mixer", "Playback Switch", "DAC"}, {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, @@ -287,17 +287,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Input Mixer", "Aux Capture Switch", "Aux In"}, }; -static int ak4535_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, ak4535_dapm_widgets, - ARRAY_SIZE(ak4535_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -457,8 +446,6 @@ static int ak4535_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); - ak4535_add_widgets(codec); - return 0; } @@ -480,6 +467,10 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .reg_cache_size = ARRAY_SIZE(ak4535_reg), .reg_word_size = sizeof(u8), .reg_cache_default = ak4535_reg, + .dapm_widgets = ak4535_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), + .dapm_routes = ak4535_audio_map, + .num_dapm_routes = ARRAY_SIZE(ak4535_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.3 From e27808df97ff7b43b4927aadf410705f33313523 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 30 Mar 2011 21:53:17 +0800 Subject: ASoC: Use data based init for sn95031 DAPM Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 2a30eae1881..569555f4a92 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -825,8 +825,6 @@ EXPORT_SYMBOL_GPL(sn95031_jack_detection); /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { - int ret; - pr_debug("codec_probe called\n"); codec->dapm.bias_level = SND_SOC_BIAS_OFF; @@ -877,16 +875,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, sn95031_snd_controls, ARRAY_SIZE(sn95031_snd_controls)); - ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, - ARRAY_SIZE(sn95031_dapm_widgets)); - if (ret) - pr_err("soc_dapm_new_control failed %d", ret); - ret = snd_soc_dapm_add_routes(&codec->dapm, sn95031_audio_map, - ARRAY_SIZE(sn95031_audio_map)); - if (ret) - pr_err("soc_dapm_add_routes failed %d", ret); - - return ret; + return 0; } static int sn95031_codec_remove(struct snd_soc_codec *codec) @@ -903,6 +892,10 @@ struct snd_soc_codec_driver sn95031_codec = { .read = sn95031_read, .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, + .dapm_widgets = sn95031_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), + .dapm_routes = sn95031_audio_map, + .num_dapm_routes = ARRAY_SIZE(sn95031_audio_map), }; static int __devinit sn95031_device_probe(struct platform_device *pdev) -- cgit v1.2.3 From 363618f013f2f11a1089b610748beb5de93b8630 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 1 Apr 2011 14:50:43 -0600 Subject: ASoC: Name jack GPIOs based on jack not codec snd_soc_jack_gpio has a name field. Use that name when registering the IRQ, since this is far more informative than the codec driver name. This shows up in /proc/interrupts. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcab80b36a3..6203a72d57a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -325,7 +325,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, - jack->codec->dev->driver->name, + gpios[i].name, &gpios[i]); if (ret) goto err; -- cgit v1.2.3 From 04368d051a6aa90c14a3ca5a48f60aca2f6ecdd1 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Sat, 2 Apr 2011 03:43:15 +0200 Subject: ASoC: Properly handle spitz MIC GPIO This patch firstly restructurizes the code a bit by getting rid of continuous checking for machine type in spitz_mic_bias(). Then the patch properly requests the MIC GPIO in the spitz_init() and frees it in spitz_exit(). Signed-off-by: Marek Vasut Signed-off-by: Mark Brown --- sound/soc/pxa/spitz.c | 41 +++++++++++++++++++++++++++++------------ 1 file changed, 29 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 8e157135063..b253d864868 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -42,6 +42,7 @@ static int spitz_jack_func; static int spitz_spk_func; +static int spitz_mic_gpio; static void spitz_ext_control(struct snd_soc_codec *codec) { @@ -217,14 +218,7 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (machine_is_borzoi() || machine_is_spitz()) - gpio_set_value(SPITZ_GPIO_MIC_BIAS, - SND_SOC_DAPM_EVENT_ON(event)); - - if (machine_is_akita()) - gpio_set_value(AKITA_GPIO_MIC_BIAS, - SND_SOC_DAPM_EVENT_ON(event)); - + gpio_set_value_cansleep(spitz_mic_gpio, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -339,22 +333,45 @@ static int __init spitz_init(void) if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) return -ENODEV; + if (machine_is_borzoi() || machine_is_spitz()) + spitz_mic_gpio = SPITZ_GPIO_MIC_BIAS; + else + spitz_mic_gpio = AKITA_GPIO_MIC_BIAS; + + ret = gpio_request(spitz_mic_gpio, "MIC GPIO"); + if (ret) + goto err1; + + ret = gpio_direction_output(spitz_mic_gpio, 0); + if (ret) + goto err2; + spitz_snd_device = platform_device_alloc("soc-audio", -1); - if (!spitz_snd_device) - return -ENOMEM; + if (!spitz_snd_device) { + ret = -ENOMEM; + goto err2; + } platform_set_drvdata(spitz_snd_device, &snd_soc_spitz); - ret = platform_device_add(spitz_snd_device); + ret = platform_device_add(spitz_snd_device); if (ret) - platform_device_put(spitz_snd_device); + goto err3; + + return 0; +err3: + platform_device_put(spitz_snd_device); +err2: + gpio_free(spitz_mic_gpio); +err1: return ret; } static void __exit spitz_exit(void) { platform_device_unregister(spitz_snd_device); + gpio_free(spitz_mic_gpio); } module_init(spitz_init); -- cgit v1.2.3 From 78caf66cb568f2b0c63bf8f87cff2fe1583dd9d0 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Mon, 4 Apr 2011 11:45:28 +0200 Subject: ALSA: 6fire - Update kernel configuration Kernel configuration updated: - experimental dependency removed - description updated Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/Kconfig | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 97724d8fa9f..cc47a9caec9 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -100,7 +100,6 @@ config SND_USB_US122L config SND_USB_6FIRE tristate "TerraTec DMX 6Fire USB" - depends on EXPERIMENTAL select FW_LOADER select SND_RAWMIDI select SND_PCM @@ -108,11 +107,9 @@ config SND_USB_6FIRE Say Y here to include support for TerraTec 6fire DMX USB interface. You will need firmware files in order to be able to use the device - after it has been coldstarted. This driver currently does not support - firmware loading for all devices. If you own such a device, - you could start windows and let the windows driver upload - the firmware. As long as you do not unplug your device from power, - it should be usable. + after it has been coldstarted. An install script for the firmware + and further help can be found at + http://sixfireusb.sourceforge.net endif # SND_USB -- cgit v1.2.3 From e220fa3bf503d63039fa8e0398a1c252d24663f9 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Mon, 4 Apr 2011 11:47:50 +0200 Subject: ALSA: 6fire - Fix pcm rate assignment Completion of signedness bug for pcm_runtime.rate: variable will never get assigned a negative value now. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index ba62c7468ba..2110cbf3547 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -456,7 +456,7 @@ static int usb6fire_pcm_close(struct snd_pcm_substream *alsa_sub) /* all substreams closed? if so, stop streaming */ if (!rt->playback.instance && !rt->capture.instance) { usb6fire_pcm_stream_stop(rt); - rt->rate = -1; + rt->rate = ARRAY_SIZE(rates); } } mutex_unlock(&rt->stream_mutex); @@ -480,7 +480,6 @@ static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub) struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); struct pcm_substream *sub = usb6fire_pcm_get_substream(alsa_sub); struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; - int i; int ret; if (rt->panic) @@ -493,12 +492,10 @@ static int usb6fire_pcm_prepare(struct snd_pcm_substream *alsa_sub) sub->period_off = 0; if (rt->stream_state == STREAM_DISABLED) { - for (i = 0; i < ARRAY_SIZE(rates); i++) - if (alsa_rt->rate == rates[i]) { - rt->rate = i; + for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + if (alsa_rt->rate == rates[rt->rate]) break; - } - if (i == ARRAY_SIZE(rates)) { + if (rt->rate == ARRAY_SIZE(rates)) { mutex_unlock(&rt->stream_mutex); snd_printk("invalid rate %d in prepare.\n", alsa_rt->rate); @@ -613,7 +610,7 @@ int __devinit usb6fire_pcm_init(struct sfire_chip *chip) rt->chip = chip; rt->stream_state = STREAM_DISABLED; - rt->rate = -1; + rt->rate = ARRAY_SIZE(rates); init_waitqueue_head(&rt->stream_wait_queue); mutex_init(&rt->stream_mutex); -- cgit v1.2.3 From 58c54fa47f5de976959767fa8d9bb857eee4c4e5 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Mon, 4 Apr 2011 11:49:00 +0200 Subject: ALSA: 6fire - Add support for S32_LE format Added support for sample format s32_le. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 22 ++++++++++++++++++---- 1 file changed, 18 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 2110cbf3547..7ea698793d4 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -64,7 +64,7 @@ static const struct snd_pcm_hardware pcm_hw = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH, - .formats = SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | @@ -228,7 +228,7 @@ static void usb6fire_pcm_capture(struct pcm_substream *sub, struct pcm_urb *urb) unsigned int total_length = 0; struct pcm_runtime *rt = snd_pcm_substream_chip(sub->instance); struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; - u32 *src = (u32 *) urb->buffer; + u32 *src = NULL; u32 *dest = (u32 *) (alsa_rt->dma_area + sub->dma_off * (alsa_rt->frame_bits >> 3)); u32 *dest_end = (u32 *) (alsa_rt->dma_area + alsa_rt->buffer_size @@ -244,7 +244,12 @@ static void usb6fire_pcm_capture(struct pcm_substream *sub, struct pcm_urb *urb) else frame_count = 0; - src = (u32 *) (urb->buffer + total_length); + if (alsa_rt->format == SNDRV_PCM_FORMAT_S24_LE) + src = (u32 *) (urb->buffer + total_length); + else if (alsa_rt->format == SNDRV_PCM_FORMAT_S32_LE) + src = (u32 *) (urb->buffer - 1 + total_length); + else + return; src++; /* skip leading 4 bytes of every packet */ total_length += urb->packets[i].length; for (frame = 0; frame < frame_count; frame++) { @@ -274,9 +279,18 @@ static void usb6fire_pcm_playback(struct pcm_substream *sub, * (alsa_rt->frame_bits >> 3)); u32 *src_end = (u32 *) (alsa_rt->dma_area + alsa_rt->buffer_size * (alsa_rt->frame_bits >> 3)); - u32 *dest = (u32 *) urb->buffer; + u32 *dest; int bytes_per_frame = alsa_rt->channels << 2; + if (alsa_rt->format == SNDRV_PCM_FORMAT_S32_LE) + dest = (u32 *) (urb->buffer - 1); + else if (alsa_rt->format == SNDRV_PCM_FORMAT_S24_LE) + dest = (u32 *) (urb->buffer); + else { + snd_printk(KERN_ERR PREFIX "Unknown sample format."); + return; + } + for (i = 0; i < PCM_N_PACKETS_PER_URB; i++) { /* at least 4 header bytes for valid packet. * after that: 32 bits per sample for analog channels */ -- cgit v1.2.3 From b84610b95f7e7bcd1cd9ecf3e8506a59e9f557fd Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Mon, 4 Apr 2011 11:49:57 +0200 Subject: ALSA: 6fire - Improve firmware loader Firmware loader: magical device bytes check updated (accepts all device versions now and accepts possibly loaded firmware, if it is knowing to be working) Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/firmware.c | 44 +++++++++++++++++++++++++++++++------------- 1 file changed, 31 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 9081a54a9c6..e720035fbd1 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -3,12 +3,6 @@ * * Firmware loader * - * Currently not working for all devices. To be able to use the device - * in linux, it is also possible to let the windows driver upload the firmware. - * For that, start the computer in windows and reboot. - * As long as the device is connected to the power supply, no firmware reload - * needs to be performed. - * * Author: Torsten Schenk * Created: Jan 01, 2011 * Version: 0.3.0 @@ -72,6 +66,10 @@ static const u8 ep_w_max_packet_size[] = { 0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */ }; +static const u8 known_fw_versions[][4] = { + { 0x03, 0x01, 0x0b, 0x00 } +}; + struct ihex_record { u16 address; u8 len; @@ -363,6 +361,25 @@ static int usb6fire_fw_fpga_upload( return 0; } +/* check, if the firmware version the devices has currently loaded + * is known by this driver. 'version' needs to have 4 bytes version + * info data. */ +static int usb6fire_fw_check(u8 *version) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++) + if (!memcmp(version, known_fw_versions + i, 4)) + return 0; + + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: " + "%02x %02x %02x %02x. " + "please reconnect to power. if this failure " + "still happens, check your firmware installation.", + version[0], version[1], version[2], version[3]); + return -EINVAL; +} + int usb6fire_fw_init(struct usb_interface *intf) { int i; @@ -378,9 +395,7 @@ int usb6fire_fw_init(struct usb_interface *intf) "firmware state.\n"); return ret; } - if (buffer[0] != 0xeb || buffer[1] != 0xaa || buffer[2] != 0x55 - || buffer[4] != 0x03 || buffer[5] != 0x01 || buffer[7] - != 0x00) { + if (buffer[0] != 0xeb || buffer[1] != 0xaa || buffer[2] != 0x55) { snd_printk(KERN_ERR PREFIX "unknown device firmware state " "received from device: "); for (i = 0; i < 8; i++) @@ -389,7 +404,7 @@ int usb6fire_fw_init(struct usb_interface *intf) return -EIO; } /* do we need fpga loader ezusb firmware? */ - if (buffer[3] == 0x01 && buffer[6] == 0x19) { + if (buffer[3] == 0x01) { ret = usb6fire_fw_ezusb_upload(intf, "6fire/dmx6firel2.ihx", 0, NULL, 0); if (ret < 0) @@ -397,7 +412,10 @@ int usb6fire_fw_init(struct usb_interface *intf) return FW_NOT_READY; } /* do we need fpga firmware and application ezusb firmware? */ - else if (buffer[3] == 0x02 && buffer[6] == 0x0b) { + else if (buffer[3] == 0x02) { + ret = usb6fire_fw_check(buffer + 4); + if (ret < 0) + return ret; ret = usb6fire_fw_fpga_upload(intf, "6fire/dmx6firecf.bin"); if (ret < 0) return ret; @@ -410,8 +428,8 @@ int usb6fire_fw_init(struct usb_interface *intf) return FW_NOT_READY; } /* all fw loaded? */ - else if (buffer[3] == 0x03 && buffer[6] == 0x0b) - return 0; + else if (buffer[3] == 0x03) + return usb6fire_fw_check(buffer + 4); /* unknown data? */ else { snd_printk(KERN_ERR PREFIX "unknown device firmware state " -- cgit v1.2.3 From 2475b0d407614ea5a41b8325d45c614d94087088 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Mon, 4 Apr 2011 11:50:53 +0200 Subject: ALSA: 6fire - Add support of digital-thru mixer Digital Thru mixer element added (device can act as converter optical<->coax) Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/control.c | 105 ++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/6fire/control.h | 17 ++++++++ sound/usb/6fire/pcm.c | 62 +++++++-------------------- 3 files changed, 137 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index 24846351118..ac828eff1a6 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -65,6 +65,15 @@ init_data[] = { { 0 } /* TERMINATING ENTRY */ }; +static const int rates_altsetting[] = { 1, 1, 2, 2, 3, 3 }; +/* values to write to soundcard register for all samplerates */ +static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01}; +static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; + +enum { + DIGITAL_THRU_ONLY_SAMPLERATE = 3 +}; + static void usb6fire_control_master_vol_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -95,6 +104,67 @@ static void usb6fire_control_opt_coax_update(struct control_runtime *rt) } } +static int usb6fire_control_set_rate(struct control_runtime *rt, int rate) +{ + int ret; + struct usb_device *device = rt->chip->dev; + struct comm_runtime *comm_rt = rt->chip->comm; + + if (rate < 0 || rate >= CONTROL_N_RATES) + return -EINVAL; + + ret = usb_set_interface(device, 1, rates_altsetting[rate]); + if (ret < 0) + return ret; + + /* set soundcard clock */ + ret = comm_rt->write16(comm_rt, 0x02, 0x01, rates_6fire_vl[rate], + rates_6fire_vh[rate]); + if (ret < 0) + return ret; + + return 0; +} + +static int usb6fire_control_set_channels( + struct control_runtime *rt, int n_analog_out, + int n_analog_in, bool spdif_out, bool spdif_in) +{ + int ret; + struct comm_runtime *comm_rt = rt->chip->comm; + + /* enable analog inputs and outputs + * (one bit per stereo-channel) */ + ret = comm_rt->write16(comm_rt, 0x02, 0x02, + (1 << (n_analog_out / 2)) - 1, + (1 << (n_analog_in / 2)) - 1); + if (ret < 0) + return ret; + + /* disable digital inputs and outputs */ + /* TODO: use spdif_x to enable/disable digital channels */ + ret = comm_rt->write16(comm_rt, 0x02, 0x03, 0x00, 0x00); + if (ret < 0) + return ret; + + return 0; +} + +static int usb6fire_control_streaming_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + + if (comm_rt) { + if (!rt->usb_streaming && rt->digital_thru_switch) + usb6fire_control_set_rate(rt, + DIGITAL_THRU_ONLY_SAMPLERATE); + return comm_rt->write16(comm_rt, 0x02, 0x00, 0x00, + (rt->usb_streaming ? 0x01 : 0x00) | + (rt->digital_thru_switch ? 0x08 : 0x00)); + } + return -EINVAL; +} + static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -195,6 +265,28 @@ static int usb6fire_control_opt_coax_get(struct snd_kcontrol *kcontrol, return 0; } +static int usb6fire_control_digital_thru_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + + if (rt->digital_thru_switch != ucontrol->value.integer.value[0]) { + rt->digital_thru_switch = ucontrol->value.integer.value[0]; + usb6fire_control_streaming_update(rt); + changed = 1; + } + return changed; +} + +static int usb6fire_control_digital_thru_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = rt->digital_thru_switch; + return 0; +} + static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -223,6 +315,15 @@ static struct __devinitdata snd_kcontrol_new elements[] = { .get = usb6fire_control_opt_coax_get, .put = usb6fire_control_opt_coax_put }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Thru Playback Route", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = usb6fire_control_digital_thru_get, + .put = usb6fire_control_digital_thru_put + }, {} }; @@ -238,6 +339,9 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip) return -ENOMEM; rt->chip = chip; + rt->update_streaming = usb6fire_control_streaming_update; + rt->set_rate = usb6fire_control_set_rate; + rt->set_channels = usb6fire_control_set_channels; i = 0; while (init_data[i].type) { @@ -249,6 +353,7 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip) usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); usb6fire_control_master_vol_update(rt); + usb6fire_control_streaming_update(rt); i = 0; while (elements[i].name) { diff --git a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h index b534c777ab0..8f5aeead2e3 100644 --- a/sound/usb/6fire/control.h +++ b/sound/usb/6fire/control.h @@ -21,12 +21,29 @@ enum { CONTROL_MAX_ELEMENTS = 32 }; +enum { + CONTROL_RATE_44KHZ, + CONTROL_RATE_48KHZ, + CONTROL_RATE_88KHZ, + CONTROL_RATE_96KHZ, + CONTROL_RATE_176KHZ, + CONTROL_RATE_192KHZ, + CONTROL_N_RATES +}; + struct control_runtime { + int (*update_streaming)(struct control_runtime *rt); + int (*set_rate)(struct control_runtime *rt, int rate); + int (*set_channels)(struct control_runtime *rt, int n_analog_out, + int n_analog_in, bool spdif_out, bool spdif_in); + struct sfire_chip *chip; struct snd_kcontrol *element[CONTROL_MAX_ELEMENTS]; bool opt_coax_switch; bool line_phono_switch; + bool digital_thru_switch; + bool usb_streaming; u8 master_vol; }; diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 7ea698793d4..b137b25865c 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -17,26 +17,23 @@ #include "pcm.h" #include "chip.h" #include "comm.h" +#include "control.h" enum { OUT_N_CHANNELS = 6, IN_N_CHANNELS = 4 }; /* keep next two synced with - * FW_EP_W_MAX_PACKET_SIZE[] and RATES_MAX_PACKET_SIZE */ + * FW_EP_W_MAX_PACKET_SIZE[] and RATES_MAX_PACKET_SIZE + * and CONTROL_RATE_XXX in control.h */ static const int rates_in_packet_size[] = { 228, 228, 420, 420, 404, 404 }; static const int rates_out_packet_size[] = { 228, 228, 420, 420, 604, 604 }; static const int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000 }; -static const int rates_altsetting[] = { 1, 1, 2, 2, 3, 3 }; static const int rates_alsaid[] = { SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_48000, SNDRV_PCM_RATE_88200, SNDRV_PCM_RATE_96000, SNDRV_PCM_RATE_176400, SNDRV_PCM_RATE_192000 }; -/* values to write to soundcard register for all samplerates */ -static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01}; -static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; - enum { /* settings for pcm */ OUT_EP = 6, IN_EP = 2, MAX_BUFSIZE = 128 * 1024 }; @@ -48,15 +45,6 @@ enum { /* pcm streaming states */ STREAM_STOPPING }; -enum { /* pcm sample rates (also index into RATES_XXX[]) */ - RATE_44KHZ, - RATE_48KHZ, - RATE_88KHZ, - RATE_96KHZ, - RATE_176KHZ, - RATE_192KHZ -}; - static const struct snd_pcm_hardware pcm_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | @@ -87,57 +75,34 @@ static const struct snd_pcm_hardware pcm_hw = { static int usb6fire_pcm_set_rate(struct pcm_runtime *rt) { int ret; - struct usb_device *device = rt->chip->dev; - struct comm_runtime *comm_rt = rt->chip->comm; + struct control_runtime *ctrl_rt = rt->chip->control; - if (rt->rate >= ARRAY_SIZE(rates)) - return -EINVAL; - /* disable streaming */ - ret = comm_rt->write16(comm_rt, 0x02, 0x00, 0x00, 0x00); + ctrl_rt->usb_streaming = false; + ret = ctrl_rt->update_streaming(ctrl_rt); if (ret < 0) { snd_printk(KERN_ERR PREFIX "error stopping streaming while " "setting samplerate %d.\n", rates[rt->rate]); return ret; } - ret = usb_set_interface(device, 1, rates_altsetting[rt->rate]); - if (ret < 0) { - snd_printk(KERN_ERR PREFIX "error setting interface " - "altsetting %d for samplerate %d.\n", - rates_altsetting[rt->rate], rates[rt->rate]); - return ret; - } - - /* set soundcard clock */ - ret = comm_rt->write16(comm_rt, 0x02, 0x01, rates_6fire_vl[rt->rate], - rates_6fire_vh[rt->rate]); + ret = ctrl_rt->set_rate(ctrl_rt, rt->rate); if (ret < 0) { snd_printk(KERN_ERR PREFIX "error setting samplerate %d.\n", rates[rt->rate]); return ret; } - /* enable analog inputs and outputs - * (one bit per stereo-channel) */ - ret = comm_rt->write16(comm_rt, 0x02, 0x02, - (1 << (OUT_N_CHANNELS / 2)) - 1, - (1 << (IN_N_CHANNELS / 2)) - 1); + ret = ctrl_rt->set_channels(ctrl_rt, OUT_N_CHANNELS, IN_N_CHANNELS, + false, false); if (ret < 0) { - snd_printk(KERN_ERR PREFIX "error initializing analog channels " + snd_printk(KERN_ERR PREFIX "error initializing channels " "while setting samplerate %d.\n", rates[rt->rate]); return ret; } - /* disable digital inputs and outputs */ - ret = comm_rt->write16(comm_rt, 0x02, 0x03, 0x00, 0x00); - if (ret < 0) { - snd_printk(KERN_ERR PREFIX "error initializing digital " - "channels while setting samplerate %d.\n", - rates[rt->rate]); - return ret; - } - ret = comm_rt->write16(comm_rt, 0x02, 0x00, 0x00, 0x01); + ctrl_rt->usb_streaming = true; + ret = ctrl_rt->update_streaming(ctrl_rt); if (ret < 0) { snd_printk(KERN_ERR PREFIX "error starting streaming while " "setting samplerate %d.\n", rates[rt->rate]); @@ -168,12 +133,15 @@ static struct pcm_substream *usb6fire_pcm_get_substream( static void usb6fire_pcm_stream_stop(struct pcm_runtime *rt) { int i; + struct control_runtime *ctrl_rt = rt->chip->control; if (rt->stream_state != STREAM_DISABLED) { for (i = 0; i < PCM_N_URBS; i++) { usb_kill_urb(&rt->in_urbs[i].instance); usb_kill_urb(&rt->out_urbs[i].instance); } + ctrl_rt->usb_streaming = false; + ctrl_rt->update_streaming(ctrl_rt); rt->stream_state = STREAM_DISABLED; } } -- cgit v1.2.3 From d8e4f9aed82c68b60f81b2e7977c46868d51062f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 4 Apr 2011 12:43:23 +0200 Subject: ALSA: core - Don't use "default' for default The card-id parser assigns the string "default" when no appropriate word is found in the card name. But this string may confuse the alsa-lib, so better to avoid. Use "Default" now instead. Signed-off-by: Takashi Iwai --- sound/core/init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index a0080aa45ae..30ecad41403 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -514,7 +514,7 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid) id = card->id; if (*id == '\0') - strcpy(id, "default"); + strcpy(id, "Default"); while (1) { if (loops-- == 0) { -- cgit v1.2.3 From 898f8b0b65715843f7afd86f3867270dc9ed8b74 Mon Sep 17 00:00:00 2001 From: Seungwhan Youn Date: Mon, 4 Apr 2011 13:43:42 +0900 Subject: ASoC: Fix to avoid compile error This patch fixes to avoid compile error when ASoC codec doesn't use I2C nor SPI on snd_soc_hw_bulk_write_raw(). Signed-off-by: Seungwhan Youn Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 22b09907622..8418b1fa197 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -414,12 +414,16 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r return -EINVAL; switch (codec->control_type) { +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) case SND_SOC_I2C: ret = i2c_master_send(codec->control_data, data, len); break; +#endif +#if defined(CONFIG_SPI_MASTER) case SND_SOC_SPI: ret = do_spi_write(codec->control_data, data, len); break; +#endif default: BUG(); } -- cgit v1.2.3 From d420d40e9c044e8fae7957ed82cd63d2afe4147f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Apr 2011 14:55:59 +0900 Subject: ASoC: Remove excessively verbose logging on I2C write We don't need to log every I2C transfer, and certainly not at error level. Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 8418b1fa197..ed67c160724 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -245,7 +245,6 @@ static unsigned int do_i2c_read(struct snd_soc_codec *codec, xfer[1].buf = data; ret = i2c_transfer(client->adapter, xfer, 2); - dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); if (ret == 2) return 0; else if (ret < 0) -- cgit v1.2.3 From 34bad69cf63efc761b05f603d99e121b83635c08 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Apr 2011 17:55:42 +0900 Subject: ASoC: Fix comment width in soc-cache.c Lines should be less than 80 columns. Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index ed67c160724..f46a198a48c 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -393,10 +393,11 @@ static int snd_soc_16_16_spi_write(void *control_data, const char *data, #define snd_soc_16_16_spi_write NULL #endif -/* Primitive bulk write support for soc-cache. The data pointed to by `data' needs - * to already be in the form the hardware expects including any leading register specific - * data. Any data written through this function will not go through the cache as it - * only handles writing to volatile or out of bounds registers. +/* Primitive bulk write support for soc-cache. The data pointed to by + * `data' needs to already be in the form the hardware expects + * including any leading register specific data. Any data written + * through this function will not go through the cache as it only + * handles writing to volatile or out of bounds registers. */ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, const void *data, size_t len) -- cgit v1.2.3 From ef49e4fae3f364af8da041dcc1bb4931c749b3da Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Apr 2011 20:48:13 +0900 Subject: ASoC: Add bias level data to DAPM context debugfs This is also in the old sysfs diagnostics but it's nice to have everything in one place. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 81c4052c127..05da8a8f0ae 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1304,6 +1304,47 @@ static const struct file_operations dapm_widget_power_fops = { .llseek = default_llseek, }; +static int dapm_bias_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t dapm_bias_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_dapm_context *dapm = file->private_data; + char *level; + + switch (dapm->bias_level) { + case SND_SOC_BIAS_ON: + level = "On\n"; + break; + case SND_SOC_BIAS_PREPARE: + level = "Prepare\n"; + break; + case SND_SOC_BIAS_STANDBY: + level = "Standby\n"; + break; + case SND_SOC_BIAS_OFF: + level = "Off\n"; + break; + default: + BUG(); + level = "Unknown\n"; + break; + } + + return simple_read_from_buffer(user_buf, count, ppos, level, + strlen(level)); +} + +static const struct file_operations dapm_bias_fops = { + .open = dapm_bias_open_file, + .read = dapm_bias_read_file, + .llseek = default_llseek, +}; + void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; @@ -1312,6 +1353,13 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm) if (!dapm->debugfs_dapm) return; + d = debugfs_create_file("bias_level", 0444, + dapm->debugfs_dapm, dapm, + &dapm_bias_fops); + if (!d) + dev_warn(dapm->dev, + "ASoC: Failed to create bias level debugfs file\n"); + list_for_each_entry(w, &dapm->card->widgets, list) { if (!w->name || w->dapm != dapm) continue; -- cgit v1.2.3 From 550ac6ba4ef0e57f6edbadf2b018d5125d2f7e98 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:41 +1200 Subject: ALSA: snd-asihpi: Control naming Clock source is neither capture nor playback, so change 'Capture Clock' to 'Clock'. Add spaces to control name string for consistency, always 'PCM 0' , never 'PCM0' Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index f53a31e939c..edcbe39d603 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1413,14 +1413,16 @@ static void asihpi_ctl_init(struct snd_kcontrol_new *snd_control, struct hpi_control *hpi_ctl, char *name) { - char *dir = ""; + char *dir; memset(snd_control, 0, sizeof(*snd_control)); snd_control->name = hpi_ctl->name; snd_control->private_value = hpi_ctl->h_control; snd_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER; snd_control->index = 0; - if (hpi_ctl->dst_node_type + HPI_DESTNODE_NONE == HPI_DESTNODE_ISTREAM) + if (hpi_ctl->src_node_type + HPI_SOURCENODE_NONE == HPI_SOURCENODE_CLOCK_SOURCE) + dir = ""; /* clock is neither capture nor playback */ + else if (hpi_ctl->dst_node_type + HPI_DESTNODE_NONE == HPI_DESTNODE_ISTREAM) dir = "Capture "; /* On or towards a PCM capture destination*/ else if ((hpi_ctl->src_node_type + HPI_SOURCENODE_NONE != HPI_SOURCENODE_OSTREAM) && (!hpi_ctl->dst_node_type)) @@ -1433,7 +1435,7 @@ static void asihpi_ctl_init(struct snd_kcontrol_new *snd_control, dir = "Playback "; /* PCM Playback source, or output node */ if (hpi_ctl->src_node_type && hpi_ctl->dst_node_type) - sprintf(hpi_ctl->name, "%s%d %s%d %s%s", + sprintf(hpi_ctl->name, "%s %d %s %d %s%s", asihpi_src_names[hpi_ctl->src_node_type], hpi_ctl->src_node_index, asihpi_dst_names[hpi_ctl->dst_node_type], -- cgit v1.2.3 From a6477134db119a22aa30911ff18e440b8db9df65 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:42 +1200 Subject: ALSA: asihpi: Update debug printing Debug print full substream ID. Other minor debug print updates. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 102 +++++++++++++++++++++++++--------------------- 1 file changed, 56 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index edcbe39d603..434342f874f 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -57,8 +57,25 @@ MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); */ #define snd_printddd(format, args...) \ __snd_printk(3, __FILE__, __LINE__, format, ##args) + +/* copied from pcm_lib.c, hope later patch will make that version public +and this copy can be removed */ +static void pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len) +{ + snprintf(name, len, "pcmC%dD%d%c:%d", + substream->pcm->card->number, + substream->pcm->device, + substream->stream ? 'c' : 'p', + substream->number); +} +#define DEBUG_NAME(substream, name) char name[16]; pcm_debug_name(substream, name, sizeof(name)) + #else -#define snd_printddd(format, args...) do { } while (0) +#define snd_printddd(format, args...) do { } while (0) +#define pcm_debug_name(s, n, l) do { } while (0) +#define DEBUG_NAME(name, substream) do { } while (0) + #endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */ @@ -101,13 +118,6 @@ static int adapter_fs = DEFAULT_SAMPLERATE; #define PERIOD_BYTES_MIN 2048 #define BUFFER_BYTES_MAX (512 * 1024) -/* convert stream to character */ -#define SCHR(s) ((s == SNDRV_PCM_STREAM_PLAYBACK) ? 'P' : 'C') - -/*#define TIMER_MILLISECONDS 20 -#define FORCE_TIMER_JIFFIES ((TIMER_MILLISECONDS * HZ + 999)/1000) -*/ - #define MAX_CLOCKSOURCES (HPI_SAMPLECLOCK_SOURCE_LAST + 1 + 7) struct clk_source { @@ -288,19 +298,26 @@ static u16 handle_error(u16 err, int line, char *filename) #define hpi_handle_error(x) handle_error(x, __LINE__, __FILE__) /***************************** GENERAL PCM ****************/ -static void print_hwparams(struct snd_pcm_hw_params *p) + +static void print_hwparams(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *p) { - snd_printd("HWPARAMS \n"); - snd_printd("samplerate %d \n", params_rate(p)); - snd_printd("Channels %d \n", params_channels(p)); - snd_printd("Format %d \n", params_format(p)); - snd_printd("subformat %d \n", params_subformat(p)); - snd_printd("Buffer bytes %d \n", params_buffer_bytes(p)); - snd_printd("Period bytes %d \n", params_period_bytes(p)); - snd_printd("access %d \n", params_access(p)); - snd_printd("period_size %d \n", params_period_size(p)); - snd_printd("periods %d \n", params_periods(p)); - snd_printd("buffer_size %d \n", params_buffer_size(p)); + DEBUG_NAME(substream, name); + snd_printd("%s HWPARAMS\n", name); + snd_printd(" samplerate %d Hz\n", params_rate(p)); + snd_printd(" channels %d\n", params_channels(p)); + snd_printd(" format %d\n", params_format(p)); + snd_printd(" subformat %d\n", params_subformat(p)); + snd_printd(" buffer %d B\n", params_buffer_bytes(p)); + snd_printd(" period %d B\n", params_period_bytes(p)); + snd_printd(" access %d\n", params_access(p)); + snd_printd(" period_size %d\n", params_period_size(p)); + snd_printd(" periods %d\n", params_periods(p)); + snd_printd(" buffer_size %d\n", params_buffer_size(p)); + snd_printd(" %d B/s\n", params_rate(p) * + params_channels(p) * + snd_pcm_format_width(params_format(p)) / 8); + } static snd_pcm_format_t hpi_to_alsa_formats[] = { @@ -451,7 +468,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, int width; unsigned int bytes_per_sec; - print_hwparams(params); + print_hwparams(substream, params); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); if (err < 0) return err; @@ -459,10 +476,6 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, if (err) return err; - snd_printdd("format %d, %d chans, %d_hz\n", - format, params_channels(params), - params_rate(params)); - hpi_handle_error(hpi_format_create(&dpcm->format, params_channels(params), format, params_rate(params), 0, 0)); @@ -509,8 +522,6 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, dpcm->bytes_per_sec = bytes_per_sec; dpcm->buffer_bytes = params_buffer_bytes(params); dpcm->period_bytes = params_period_bytes(params); - snd_printdd("buffer_bytes=%d, period_bytes=%d, bps=%d\n", - dpcm->buffer_bytes, dpcm->period_bytes, bytes_per_sec); return 0; } @@ -564,9 +575,10 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, struct snd_card_asihpi *card = snd_pcm_substream_chip(substream); struct snd_pcm_substream *s; u16 e; + DEBUG_NAME(substream, name); + + snd_printdd("%s trigger\n", name); - snd_printdd("%c%d trigger\n", - SCHR(substream->stream), substream->number); switch (cmd) { case SNDRV_PCM_TRIGGER_START: snd_pcm_group_for_each_entry(s, substream) { @@ -599,9 +611,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } if (card->support_grouping) { - snd_printdd("\t%c%d group\n", - SCHR(s->stream), - s->number); + snd_printdd("%d group\n", s->number); e = hpi_stream_group_add( dpcm->h_stream, ds->h_stream); @@ -636,9 +646,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, s->runtime->status->state = SNDRV_PCM_STATE_SETUP; if (card->support_grouping) { - snd_printdd("\t%c%d group\n", - SCHR(s->stream), - s->number); + snd_printdd("%d group\n", s->number); snd_pcm_trigger_done(s, substream); } else break; @@ -732,9 +740,9 @@ static void snd_card_asihpi_timer_function(unsigned long data) int loops = 0; u16 state; u32 buffer_size, bytes_avail, samples_played, on_card_bytes; + DEBUG_NAME(substream, name); - snd_printdd("%c%d snd_card_asihpi_timer_function\n", - SCHR(substream->stream), substream->number); + snd_printdd("%s snd_card_asihpi_timer_function\n", name); /* find minimum newdata and buffer pos in group */ snd_pcm_group_for_each_entry(s, substream) { @@ -786,16 +794,18 @@ static void snd_card_asihpi_timer_function(unsigned long data) newdata); } - snd_printdd("hw_ptr x%04lX, appl_ptr x%04lX\n", + snd_printdd("hw_ptr 0x%04lX, appl_ptr 0x%04lX\n", (unsigned long)frames_to_bytes(runtime, runtime->status->hw_ptr), (unsigned long)frames_to_bytes(runtime, runtime->control->appl_ptr)); - snd_printdd("%d %c%d S=%d, rw=%04X, dma=x%04X, left=x%04X," - " aux=x%04X space=x%04X\n", - loops, SCHR(s->stream), s->number, - state, ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs, (int)bytes_avail, + snd_printdd("%d S=%d, " + "rw=0x%04X, dma=0x%04X, left=0x%04X, " + "aux=0x%04X space=0x%04X\n", + s->number, state, + ds->pcm_buf_host_rw_ofs, pcm_buf_dma_ofs, + (int)bytes_avail, (int)on_card_bytes, buffer_size-bytes_avail); loops++; } @@ -814,7 +824,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) next_jiffies = max(next_jiffies, 1U); dpcm->timer.expires = jiffies + next_jiffies; - snd_printdd("jif %d buf pos x%04X newdata x%04X xfer x%04X\n", + snd_printdd("jif %d buf pos 0x%04X newdata 0x%04X xfer 0x%04X\n", next_jiffies, pcm_buf_dma_ofs, newdata, xfercount); snd_pcm_group_for_each_entry(s, substream) { @@ -863,7 +873,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) static int snd_card_asihpi_playback_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - snd_printdd(KERN_INFO "Playback ioctl %d\n", cmd); + snd_printddd(KERN_INFO "P%d ioctl %d\n", substream->number, cmd); return snd_pcm_lib_ioctl(substream, cmd, arg); } @@ -873,7 +883,7 @@ static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream * struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - snd_printdd("playback prepare %d\n", substream->number); + snd_printdd("P%d prepare\n", substream->number); hpi_handle_error(hpi_outstream_reset(dpcm->h_stream)); dpcm->pcm_buf_host_rw_ofs = 0; @@ -890,7 +900,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t ptr; ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs % dpcm->buffer_bytes); - snd_printddd("playback_pointer=x%04lx\n", (unsigned long)ptr); + snd_printddd("P%d pointer = 0x%04lx\n", substream->number, (unsigned long)ptr); return ptr; } -- cgit v1.2.3 From 0b7ce9e2bd2d9dbc8f4797b0cd5e0d138cb529e1 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:43 +1200 Subject: ALSA: asihpi: Handle playback drained status better Use the card drained status reporting for playback, but allow it to persist for a few timer cycles before signalling XRUN, to allow card to recover by itself. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 434342f874f..a5226e3af3d 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -165,6 +165,7 @@ struct snd_card_asihpi_pcm { unsigned int pcm_buf_host_rw_ofs; /* Host R/W pos */ unsigned int pcm_buf_dma_ofs; /* DMA R/W offset in buffer */ unsigned int pcm_buf_elapsed_dma_ofs; /* DMA R/W offset in buffer */ + unsigned int drained_count; struct snd_pcm_substream *substream; u32 h_stream; struct hpi_format format; @@ -592,6 +593,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, if (substream->stream != s->stream) continue; + ds->drained_count = 0; if ((s->stream == SNDRV_PCM_STREAM_PLAYBACK) && (card->support_mmap)) { /* How do I know how much valid data is present @@ -771,12 +773,18 @@ static void snd_card_asihpi_timer_function(unsigned long data) (on_card_bytes < ds->pcm_buf_host_rw_ofs)) { hpi_handle_error(hpi_stream_start(ds->h_stream)); snd_printdd("P%d start\n", s->number); + ds->drained_count = 0; } } else if (state == HPI_STATE_DRAINED) { snd_printd(KERN_WARNING "P%d drained\n", s->number); - /*snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); - continue; */ + ds->drained_count++; + if (ds->drained_count > 2) { + snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); + continue; + } + } else { + ds->drained_count = 0; } } else pcm_buf_dma_ofs = bytes_avail + ds->pcm_buf_host_rw_ofs; -- cgit v1.2.3 From f3d145aac913b318e96e5c2763d8908805a5e30a Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:44 +1200 Subject: ALSA: asihpi: MMAP for non-busmaster cards Allow older non DMA capable cards to use MMAP by emulating the DMA using read and write functions, and getting rid of copy & silence callbacks that were used only by older cards. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 205 ++++++++++++++++------------------------------ 1 file changed, 70 insertions(+), 135 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index a5226e3af3d..d5258aa738a 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -42,6 +42,7 @@ #include #include + MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. "); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); @@ -146,7 +147,7 @@ struct snd_card_asihpi { u32 h_mixer; struct clk_cache cc; - u16 support_mmap; + u16 can_dma; u16 support_grouping; u16 support_mrx; u16 update_interval_frames; @@ -491,8 +492,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, } dpcm->hpi_buffer_attached = 0; - if (card->support_mmap) { - + if (card->can_dma) { err = hpi_stream_host_buffer_attach(dpcm->h_stream, params_buffer_bytes(params), runtime->dma_addr); if (err == 0) { @@ -594,8 +594,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, continue; ds->drained_count = 0; - if ((s->stream == SNDRV_PCM_STREAM_PLAYBACK) && - (card->support_mmap)) { + if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* How do I know how much valid data is present * in buffer? Must be at least one period! * Guessing 2 periods, but if @@ -630,7 +629,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, /* start the master stream */ snd_card_asihpi_pcm_timer_start(substream); if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) || - !card->support_mmap) + !card->can_dma) hpi_handle_error(hpi_stream_start(dpcm->h_stream)); break; @@ -766,6 +765,9 @@ static void snd_card_asihpi_timer_function(unsigned long data) /* number of bytes in on-card buffer */ runtime->delay = on_card_bytes; + if (!card->can_dma) + on_card_bytes = bytes_avail; + if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { pcm_buf_dma_ofs = ds->pcm_buf_host_rw_ofs - bytes_avail; if (state == HPI_STATE_STOPPED) { @@ -844,30 +846,63 @@ static void snd_card_asihpi_timer_function(unsigned long data) ds->pcm_buf_dma_ofs = pcm_buf_dma_ofs; - if (xfercount && (on_card_bytes <= ds->period_bytes)) { - if (card->support_mmap) { - if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { - snd_printddd("P%d write x%04x\n", + if (xfercount && + /* Limit use of on card fifo for playback */ + ((on_card_bytes <= ds->period_bytes) || + (s->stream == SNDRV_PCM_STREAM_CAPTURE))) + + { + + unsigned int buf_ofs = ds->pcm_buf_host_rw_ofs % ds->buffer_bytes; + unsigned int xfer1, xfer2; + char *pd = &s->runtime->dma_area[buf_ofs]; + + if (card->can_dma) { + xfer1 = xfercount; + xfer2 = 0; + } else { + xfer1 = min(xfercount, ds->buffer_bytes - buf_ofs); + xfer2 = xfercount - xfer1; + } + + if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_printddd("P%d write1 0x%04X 0x%04X\n", + s->number, xfer1, buf_ofs); + hpi_handle_error( + hpi_outstream_write_buf( + ds->h_stream, pd, xfer1, + &ds->format)); + + if (xfer2) { + pd = s->runtime->dma_area; + + snd_printddd("P%d write2 0x%04X 0x%04X\n", s->number, - ds->period_bytes); + xfercount - xfer1, buf_ofs); hpi_handle_error( hpi_outstream_write_buf( - ds->h_stream, - &s->runtime-> - dma_area[0], - xfercount, + ds->h_stream, pd, + xfercount - xfer1, &ds->format)); - } else { - snd_printddd("C%d read x%04x\n", - s->number, - xfercount); + } + } else { + snd_printddd("C%d read1 0x%04x\n", + s->number, xfer1); + hpi_handle_error( + hpi_instream_read_buf( + ds->h_stream, + pd, xfer1)); + if (xfer2) { + pd = s->runtime->dma_area; + snd_printddd("C%d read2 0x%04x\n", + s->number, xfer2); hpi_handle_error( hpi_instream_read_buf( ds->h_stream, - NULL, xfercount)); + pd, xfer2)); } - ds->pcm_buf_host_rw_ofs = ds->pcm_buf_host_rw_ofs + xfercount; - } /* else R/W will be handled by read/write callbacks */ + } + ds->pcm_buf_host_rw_ofs = ds->pcm_buf_host_rw_ofs + xfercount; ds->pcm_buf_elapsed_dma_ofs = pcm_buf_dma_ofs; snd_pcm_period_elapsed(s); } @@ -1004,11 +1039,9 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) SNDRV_PCM_INFO_DOUBLE | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE; - - if (card->support_mmap) - snd_card_asihpi_playback.info |= SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID; if (card->support_grouping) snd_card_asihpi_playback.info |= SNDRV_PCM_INFO_SYNC_START; @@ -1016,7 +1049,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) /* struct is copied, so can create initializer dynamically */ runtime->hw = snd_card_asihpi_playback; - if (card->support_mmap) + if (card->can_dma) err = snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES); if (err < 0) @@ -1046,58 +1079,6 @@ static int snd_card_asihpi_playback_close(struct snd_pcm_substream *substream) return 0; } -static int snd_card_asihpi_playback_copy(struct snd_pcm_substream *substream, - int channel, - snd_pcm_uframes_t pos, - void __user *src, - snd_pcm_uframes_t count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - unsigned int len; - - len = frames_to_bytes(runtime, count); - - if (copy_from_user(runtime->dma_area, src, len)) - return -EFAULT; - - snd_printddd("playback copy%d %u bytes\n", - substream->number, len); - - hpi_handle_error(hpi_outstream_write_buf(dpcm->h_stream, - runtime->dma_area, len, &dpcm->format)); - - dpcm->pcm_buf_host_rw_ofs += len; - - return 0; -} - -static int snd_card_asihpi_playback_silence(struct snd_pcm_substream * - substream, int channel, - snd_pcm_uframes_t pos, - snd_pcm_uframes_t count) -{ - /* Usually writes silence to DMA buffer, which should be overwritten - by real audio later. Our fifos cannot be overwritten, and are not - free-running DMAs. Silence is output on fifo underflow. - This callback is still required to allow the copy callback to be used. - */ - return 0; -} - -static struct snd_pcm_ops snd_card_asihpi_playback_ops = { - .open = snd_card_asihpi_playback_open, - .close = snd_card_asihpi_playback_close, - .ioctl = snd_card_asihpi_playback_ioctl, - .hw_params = snd_card_asihpi_pcm_hw_params, - .hw_free = snd_card_asihpi_hw_free, - .prepare = snd_card_asihpi_playback_prepare, - .trigger = snd_card_asihpi_trigger, - .pointer = snd_card_asihpi_playback_pointer, - .copy = snd_card_asihpi_playback_copy, - .silence = snd_card_asihpi_playback_silence, -}; - static struct snd_pcm_ops snd_card_asihpi_playback_mmap_ops = { .open = snd_card_asihpi_playback_open, .close = snd_card_asihpi_playback_close, @@ -1229,18 +1210,16 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) snd_card_asihpi_capture_format(card, dpcm->h_stream, &snd_card_asihpi_capture); snd_card_asihpi_pcm_samplerates(card, &snd_card_asihpi_capture); - snd_card_asihpi_capture.info = SNDRV_PCM_INFO_INTERLEAVED; - - if (card->support_mmap) - snd_card_asihpi_capture.info |= SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; + snd_card_asihpi_capture.info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID; if (card->support_grouping) snd_card_asihpi_capture.info |= SNDRV_PCM_INFO_SYNC_START; runtime->hw = snd_card_asihpi_capture; - if (card->support_mmap) + if (card->can_dma) err = snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES); if (err < 0) @@ -1264,28 +1243,6 @@ static int snd_card_asihpi_capture_close(struct snd_pcm_substream *substream) return 0; } -static int snd_card_asihpi_capture_copy(struct snd_pcm_substream *substream, - int channel, snd_pcm_uframes_t pos, - void __user *dst, snd_pcm_uframes_t count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - u32 len; - - len = frames_to_bytes(runtime, count); - - snd_printddd("capture copy%d %d bytes\n", substream->number, len); - hpi_handle_error(hpi_instream_read_buf(dpcm->h_stream, - runtime->dma_area, len)); - - dpcm->pcm_buf_host_rw_ofs = dpcm->pcm_buf_host_rw_ofs + len; - - if (copy_to_user(dst, runtime->dma_area, len)) - return -EFAULT; - - return 0; -} - static struct snd_pcm_ops snd_card_asihpi_capture_mmap_ops = { .open = snd_card_asihpi_capture_open, .close = snd_card_asihpi_capture_close, @@ -1297,18 +1254,6 @@ static struct snd_pcm_ops snd_card_asihpi_capture_mmap_ops = { .pointer = snd_card_asihpi_capture_pointer, }; -static struct snd_pcm_ops snd_card_asihpi_capture_ops = { - .open = snd_card_asihpi_capture_open, - .close = snd_card_asihpi_capture_close, - .ioctl = snd_card_asihpi_capture_ioctl, - .hw_params = snd_card_asihpi_pcm_hw_params, - .hw_free = snd_card_asihpi_hw_free, - .prepare = snd_card_asihpi_capture_prepare, - .trigger = snd_card_asihpi_trigger, - .pointer = snd_card_asihpi_capture_pointer, - .copy = snd_card_asihpi_capture_copy -}; - static int __devinit snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi, int device, int substreams) { @@ -1321,17 +1266,10 @@ static int __devinit snd_card_asihpi_pcm_new(struct snd_card_asihpi *asihpi, if (err < 0) return err; /* pointer to ops struct is stored, dont change ops afterwards! */ - if (asihpi->support_mmap) { snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_asihpi_playback_mmap_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_asihpi_capture_mmap_ops); - } else { - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_card_asihpi_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_card_asihpi_capture_ops); - } pcm->private_data = asihpi; pcm->info_flags = 0; @@ -2895,14 +2833,14 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, if (err) asihpi->update_interval_frames = 512; - if (!asihpi->support_mmap) + if (!asihpi->can_dma) asihpi->update_interval_frames *= 2; hpi_handle_error(hpi_instream_open(asihpi->adapter_index, 0, &h_stream)); err = hpi_instream_host_buffer_free(h_stream); - asihpi->support_mmap = (!err); + asihpi->can_dma = (!err); hpi_handle_error(hpi_instream_close(h_stream)); @@ -2914,8 +2852,8 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->out_max_chans = 2; } - snd_printk(KERN_INFO "supports mmap:%d grouping:%d mrx:%d\n", - asihpi->support_mmap, + snd_printk(KERN_INFO "has dma:%d, grouping:%d, mrx:%d\n", + asihpi->can_dma, asihpi->support_grouping, asihpi->support_mrx ); @@ -2945,10 +2883,7 @@ static int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, by enable_hwdep module param*/ snd_asihpi_hpi_new(asihpi, 0, NULL); - if (asihpi->support_mmap) - strcpy(card->driver, "ASIHPI-MMAP"); - else - strcpy(card->driver, "ASIHPI"); + strcpy(card->driver, "ASIHPI"); sprintf(card->shortname, "AudioScience ASI%4X", asihpi->type); sprintf(card->longname, "%s %i", -- cgit v1.2.3 From 6027dfa46ea477b1223d83e74a3b3dcc572285b5 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:45 +1200 Subject: ALSA: asihpi: Remove 2 unused functions Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpifunc.c | 27 --------------------------- 1 file changed, 27 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index c38fc948756..7397b169b89 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -105,33 +105,6 @@ u16 hpi_subsys_get_version_ex(u32 *pversion_ex) return hr.error; } -u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource, - u16 *pw_adapter_index) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_CREATE_ADAPTER); - hm.u.s.resource = *p_resource; - - hpi_send_recv(&hm, &hr); - - *pw_adapter_index = hr.u.s.adapter_index; - return hr.error; -} - -u16 hpi_subsys_delete_adapter(u16 adapter_index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_DELETE_ADAPTER); - hm.obj_index = adapter_index; - hpi_send_recv(&hm, &hr); - return hr.error; -} - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters) { struct hpi_message hm; -- cgit v1.2.3 From b0096a65677fa8d7e50975dc7282ce313610d9e8 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:46 +1200 Subject: ALSA: asihpi: Standardise substream name generation Define and use pcm_debug_name if CONFIG_SND_DEBUG Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index d5258aa738a..eac7b08a161 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -47,18 +47,7 @@ MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. "); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); -#if defined CONFIG_SND_DEBUG_VERBOSE -/** - * snd_printddd - very verbose debug printk - * @format: format string - * - * Works like snd_printk() for debugging purposes. - * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. - * Must set snd module debug parameter to 3 to enable at runtime. - */ -#define snd_printddd(format, args...) \ - __snd_printk(3, __FILE__, __LINE__, format, ##args) - +#if defined CONFIG_SND_DEBUG /* copied from pcm_lib.c, hope later patch will make that version public and this copy can be removed */ static void pcm_debug_name(struct snd_pcm_substream *substream, @@ -71,12 +60,24 @@ static void pcm_debug_name(struct snd_pcm_substream *substream, substream->number); } #define DEBUG_NAME(substream, name) char name[16]; pcm_debug_name(substream, name, sizeof(name)) - #else -#define snd_printddd(format, args...) do { } while (0) #define pcm_debug_name(s, n, l) do { } while (0) #define DEBUG_NAME(name, substream) do { } while (0) +#endif +#if defined CONFIG_SND_DEBUG_VERBOSE +/** + * snd_printddd - very verbose debug printk + * @format: format string + * + * Works like snd_printk() for debugging purposes. + * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. + * Must set snd module debug parameter to 3 to enable at runtime. + */ +#define snd_printddd(format, args...) \ + __snd_printk(3, __FILE__, __LINE__, format, ##args) +#else +#define snd_printddd(format, args...) do { } while (0) #endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */ @@ -857,7 +858,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) unsigned int xfer1, xfer2; char *pd = &s->runtime->dma_area[buf_ofs]; - if (card->can_dma) { + if (card->can_dma) { /* buffer wrap is handled at lower level */ xfer1 = xfercount; xfer2 = 0; } else { -- cgit v1.2.3 From 6d0b898e9c402d6b7d0d07adacdbee2ebedafdcd Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:47 +1200 Subject: ALSA: asihpi: Simplify driver unload cleanup Replacing subsys_delete_adapter with adapter_delete allows some special-case adapter lookup code to be removed. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6000.c | 39 +++++++---------- sound/pci/asihpi/hpi6205.c | 97 +++++++++++++++++++++-------------------- sound/pci/asihpi/hpi_internal.h | 17 ++++---- sound/pci/asihpi/hpicmn.c | 1 - sound/pci/asihpi/hpimsgx.c | 31 ++++++------- sound/pci/asihpi/hpioctl.c | 54 ++++++++++++----------- 6 files changed, 113 insertions(+), 126 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index 3e3c2ef6efd..09befdafe09 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -200,8 +200,8 @@ static void hpi_read_block(struct dsp_obj *pdo, u32 address, u32 *pdata, static void subsys_create_adapter(struct hpi_message *phm, struct hpi_response *phr); -static void subsys_delete_adapter(struct hpi_message *phm, - struct hpi_response *phr); +static void adapter_delete(struct hpi_adapter_obj *pao, + struct hpi_message *phm, struct hpi_response *phr); static void adapter_get_asserts(struct hpi_adapter_obj *pao, struct hpi_message *phm, struct hpi_response *phr); @@ -222,9 +222,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr) case HPI_SUBSYS_CREATE_ADAPTER: subsys_create_adapter(phm, phr); break; - case HPI_SUBSYS_DELETE_ADAPTER: - subsys_delete_adapter(phm, phr); - break; default: phr->error = HPI_ERROR_INVALID_FUNC; break; @@ -279,6 +276,10 @@ static void adapter_message(struct hpi_adapter_obj *pao, adapter_get_asserts(pao, phm, phr); break; + case HPI_ADAPTER_DELETE: + adapter_delete(pao, phm, phr); + break; + default: hw_message(pao, phm, phr); break; @@ -333,26 +334,22 @@ void HPI_6000(struct hpi_message *phm, struct hpi_response *phr) { struct hpi_adapter_obj *pao = NULL; - /* subsytem messages get executed by every HPI. */ - /* All other messages are ignored unless the adapter index matches */ - /* an adapter in the HPI */ - /*HPI_DEBUG_LOG(DEBUG, "O %d,F %x\n", phm->wObject, phm->wFunction); */ - - /* if Dsp has crashed then do not communicate with it any more */ if (phm->object != HPI_OBJ_SUBSYSTEM) { pao = hpi_find_adapter(phm->adapter_index); if (!pao) { - HPI_DEBUG_LOG(DEBUG, - " %d,%d refused, for another HPI?\n", - phm->object, phm->function); + hpi_init_response(phr, phm->object, phm->function, + HPI_ERROR_BAD_ADAPTER_NUMBER); + HPI_DEBUG_LOG(DEBUG, "invalid adapter index: %d \n", + phm->adapter_index); return; } + /* Don't even try to communicate with crashed DSP */ if (pao->dsp_crashed >= 10) { hpi_init_response(phr, phm->object, phm->function, HPI_ERROR_DSP_HARDWARE); - HPI_DEBUG_LOG(DEBUG, " %d,%d dsp crashed.\n", - phm->object, phm->function); + HPI_DEBUG_LOG(DEBUG, "adapter %d dsp crashed\n", + phm->adapter_index); return; } } @@ -463,15 +460,9 @@ static void subsys_create_adapter(struct hpi_message *phm, phr->error = 0; } -static void subsys_delete_adapter(struct hpi_message *phm, - struct hpi_response *phr) +static void adapter_delete(struct hpi_adapter_obj *pao, + struct hpi_message *phm, struct hpi_response *phr) { - struct hpi_adapter_obj *pao = NULL; - - pao = hpi_find_adapter(phm->obj_index); - if (!pao) - return; - delete_adapter_obj(pao); hpi_delete_adapter(pao); phr->error = 0; diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 620525bdac5..53037342cd2 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -152,8 +152,8 @@ static void hw_message(struct hpi_adapter_obj *pao, struct hpi_message *phm, static void subsys_create_adapter(struct hpi_message *phm, struct hpi_response *phr); -static void subsys_delete_adapter(struct hpi_message *phm, - struct hpi_response *phr); +static void adapter_delete(struct hpi_adapter_obj *pao, + struct hpi_message *phm, struct hpi_response *phr); static u16 create_adapter_obj(struct hpi_adapter_obj *pao, u32 *pos_error_code); @@ -223,15 +223,13 @@ static u16 boot_loader_test_pld(struct hpi_adapter_obj *pao, int dsp_index); /*****************************************************************************/ -static void subsys_message(struct hpi_message *phm, struct hpi_response *phr) +static void subsys_message(struct hpi_adapter_obj *pao, + struct hpi_message *phm, struct hpi_response *phr) { switch (phm->function) { case HPI_SUBSYS_CREATE_ADAPTER: subsys_create_adapter(phm, phr); break; - case HPI_SUBSYS_DELETE_ADAPTER: - subsys_delete_adapter(phm, phr); - break; default: phr->error = HPI_ERROR_INVALID_FUNC; break; @@ -279,6 +277,10 @@ static void adapter_message(struct hpi_adapter_obj *pao, struct hpi_message *phm, struct hpi_response *phr) { switch (phm->function) { + case HPI_ADAPTER_DELETE: + adapter_delete(pao, phm, phr); + break; + default: hw_message(pao, phm, phr); break; @@ -371,36 +373,17 @@ static void instream_message(struct hpi_adapter_obj *pao, /** Entry point to this HPI backend * All calls to the HPI start here */ -void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) +void _HPI_6205(struct hpi_adapter_obj *pao, struct hpi_message *phm, + struct hpi_response *phr) { - struct hpi_adapter_obj *pao = NULL; - - /* subsytem messages are processed by every HPI. - * All other messages are ignored unless the adapter index matches - * an adapter in the HPI - */ - /* HPI_DEBUG_LOG(DEBUG, "HPI Obj=%d, Func=%d\n", phm->wObject, - phm->wFunction); */ - - /* if Dsp has crashed then do not communicate with it any more */ - if (phm->object != HPI_OBJ_SUBSYSTEM) { - pao = hpi_find_adapter(phm->adapter_index); - if (!pao) { - HPI_DEBUG_LOG(DEBUG, - " %d,%d refused, for another HPI?\n", - phm->object, phm->function); - return; - } - - if ((pao->dsp_crashed >= 10) - && (phm->function != HPI_ADAPTER_DEBUG_READ)) { - /* allow last resort debug read even after crash */ - hpi_init_response(phr, phm->object, phm->function, - HPI_ERROR_DSP_HARDWARE); - HPI_DEBUG_LOG(WARNING, " %d,%d dsp crashed.\n", - phm->object, phm->function); - return; - } + if (pao && (pao->dsp_crashed >= 10) + && (phm->function != HPI_ADAPTER_DEBUG_READ)) { + /* allow last resort debug read even after crash */ + hpi_init_response(phr, phm->object, phm->function, + HPI_ERROR_DSP_HARDWARE); + HPI_DEBUG_LOG(WARNING, " %d,%d dsp crashed.\n", phm->object, + phm->function); + return; } /* Init default response */ @@ -412,7 +395,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) case HPI_TYPE_MESSAGE: switch (phm->object) { case HPI_OBJ_SUBSYSTEM: - subsys_message(phm, phr); + subsys_message(pao, phm, phr); break; case HPI_OBJ_ADAPTER: @@ -444,6 +427,26 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) } } +void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) +{ + struct hpi_adapter_obj *pao = NULL; + + if (phm->object != HPI_OBJ_SUBSYSTEM) { + /* normal messages must have valid adapter index */ + pao = hpi_find_adapter(phm->adapter_index); + } else { + /* subsys messages don't address an adapter */ + _HPI_6205(NULL, phm, phr); + return; + } + + if (pao) + _HPI_6205(pao, phm, phr); + else + hpi_init_response(phr, phm->object, phm->function, + HPI_ERROR_BAD_ADAPTER_NUMBER); +} + /*****************************************************************************/ /* SUBSYSTEM */ @@ -491,13 +494,11 @@ static void subsys_create_adapter(struct hpi_message *phm, } /** delete an adapter - required by WDM driver */ -static void subsys_delete_adapter(struct hpi_message *phm, - struct hpi_response *phr) +static void adapter_delete(struct hpi_adapter_obj *pao, + struct hpi_message *phm, struct hpi_response *phr) { - struct hpi_adapter_obj *pao; struct hpi_hw_obj *phw; - pao = hpi_find_adapter(phm->obj_index); if (!pao) { phr->error = HPI_ERROR_INVALID_OBJ_INDEX; return; @@ -563,11 +564,12 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, } err = adapter_boot_load_dsp(pao, pos_error_code); - if (err) + if (err) { + HPI_DEBUG_LOG(ERROR, "DSP code load failed\n"); /* no need to clean up as SubSysCreateAdapter */ /* calls DeleteAdapter on error. */ return err; - + } HPI_DEBUG_LOG(INFO, "load DSP code OK\n"); /* allow boot load even if mem alloc wont work */ @@ -604,6 +606,7 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, control_cache.number_of_controls, interface->control_cache.size_in_bytes, p_control_cache_virtual); + if (!phw->p_cache) err = HPI_ERROR_MEMORY_ALLOC; } @@ -675,16 +678,14 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, } /** Free memory areas allocated by adapter - * this routine is called from SubSysDeleteAdapter, + * this routine is called from AdapterDelete, * and SubSysCreateAdapter if duplicate index */ static void delete_adapter_obj(struct hpi_adapter_obj *pao) { - struct hpi_hw_obj *phw; + struct hpi_hw_obj *phw = pao->priv; int i; - phw = pao->priv; - if (hpios_locked_mem_valid(&phw->h_control_cache)) { hpios_locked_mem_free(&phw->h_control_cache); hpi_free_control_cache(phw->p_cache); @@ -1275,6 +1276,7 @@ static u16 adapter_boot_load_dsp(struct hpi_adapter_obj *pao, case HPI_ADAPTER_FAMILY_ASI(0x6300): boot_code_id[1] = HPI_ADAPTER_FAMILY_ASI(0x6400); break; + case HPI_ADAPTER_FAMILY_ASI(0x5500): case HPI_ADAPTER_FAMILY_ASI(0x5600): case HPI_ADAPTER_FAMILY_ASI(0x6500): boot_code_id[1] = HPI_ADAPTER_FAMILY_ASI(0x6600); @@ -2059,7 +2061,6 @@ static int wait_dsp_ack(struct hpi_hw_obj *phw, int state, int timeout_us) static void send_dsp_command(struct hpi_hw_obj *phw, int cmd) { struct bus_master_interface *interface = phw->p_interface_buffer; - u32 r; interface->host_cmd = cmd; @@ -2088,7 +2089,7 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao, phr->specific_error = sizeof(interface->u); phr->size = sizeof(struct hpi_response_header); HPI_DEBUG_LOG(ERROR, - "message len %d too big for buffer %zd \n", phm->size, + "message len %d too big for buffer %ld \n", phm->size, sizeof(interface->u)); return 0; } diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index af678be0aa1..d829a65f47d 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -294,7 +294,7 @@ enum HPI_CONTROL_ATTRIBUTES { /* These defines are used to fill in protocol information for an Ethernet packet sent using HMI on CS18102 */ -/** ID supplied by Cirrius for ASI packets. */ +/** ID supplied by Cirrus for ASI packets. */ #define HPI_ETHERNET_PACKET_ID 0x85 /** Simple packet - no special routing required */ #define HPI_ETHERNET_PACKET_V1 0x01 @@ -307,7 +307,7 @@ enum HPI_CONTROL_ATTRIBUTES { /** This packet must make its way to the host across the HPI interface */ #define HPI_ETHERNET_PACKET_HOSTED_VIA_HPI_V1 0x41 -#define HPI_ETHERNET_UDP_PORT (44600) /*!< UDP messaging port */ +#define HPI_ETHERNET_UDP_PORT 44600 /**< HPI UDP service */ /** Default network timeout in milli-seconds. */ #define HPI_ETHERNET_TIMEOUT_MS 500 @@ -397,14 +397,14 @@ enum HPI_FUNCTION_IDS { HPI_SUBSYS_OPEN = HPI_FUNC_ID(SUBSYSTEM, 1), HPI_SUBSYS_GET_VERSION = HPI_FUNC_ID(SUBSYSTEM, 2), HPI_SUBSYS_GET_INFO = HPI_FUNC_ID(SUBSYSTEM, 3), - HPI_SUBSYS_FIND_ADAPTERS = HPI_FUNC_ID(SUBSYSTEM, 4), + /* HPI_SUBSYS_FIND_ADAPTERS = HPI_FUNC_ID(SUBSYSTEM, 4), */ HPI_SUBSYS_CREATE_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 5), HPI_SUBSYS_CLOSE = HPI_FUNC_ID(SUBSYSTEM, 6), - HPI_SUBSYS_DELETE_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 7), + /* HPI_SUBSYS_DELETE_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 7), */ HPI_SUBSYS_DRIVER_LOAD = HPI_FUNC_ID(SUBSYSTEM, 8), HPI_SUBSYS_DRIVER_UNLOAD = HPI_FUNC_ID(SUBSYSTEM, 9), - HPI_SUBSYS_READ_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 10), - HPI_SUBSYS_WRITE_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 11), + /* HPI_SUBSYS_READ_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 10), */ + /* HPI_SUBSYS_WRITE_PORT_8 = HPI_FUNC_ID(SUBSYSTEM, 11), */ HPI_SUBSYS_GET_NUM_ADAPTERS = HPI_FUNC_ID(SUBSYSTEM, 12), HPI_SUBSYS_GET_ADAPTER = HPI_FUNC_ID(SUBSYSTEM, 13), HPI_SUBSYS_SET_NETWORK_INTERFACE = HPI_FUNC_ID(SUBSYSTEM, 14), @@ -433,7 +433,8 @@ enum HPI_FUNCTION_IDS { HPI_ADAPTER_DEBUG_READ = HPI_FUNC_ID(ADAPTER, 18), HPI_ADAPTER_IRQ_QUERY_AND_CLEAR = HPI_FUNC_ID(ADAPTER, 19), HPI_ADAPTER_IRQ_CALLBACK = HPI_FUNC_ID(ADAPTER, 20), -#define HPI_ADAPTER_FUNCTION_COUNT 20 + HPI_ADAPTER_DELETE = HPI_FUNC_ID(ADAPTER, 21), +#define HPI_ADAPTER_FUNCTION_COUNT 21 HPI_OSTREAM_OPEN = HPI_FUNC_ID(OSTREAM, 1), HPI_OSTREAM_CLOSE = HPI_FUNC_ID(OSTREAM, 2), @@ -1561,8 +1562,6 @@ void hpi_send_recv(struct hpi_message *phm, struct hpi_response *phr); u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource, u16 *pw_adapter_index); -u16 hpi_subsys_delete_adapter(u16 adapter_index); - u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, struct hpi_hostbuffer_status **pp_status); diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index 3e9c5c28976..0428f285725 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -667,7 +667,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr) phr->u.s.num_adapters = adapters.gw_num_adapters; break; case HPI_SUBSYS_CREATE_ADAPTER: - case HPI_SUBSYS_DELETE_ADAPTER: break; default: phr->error = HPI_ERROR_INVALID_FUNC; diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index bcbdf30a6aa..888e37966c3 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -211,24 +211,6 @@ static void subsys_message(struct hpi_message *phm, struct hpi_response *phr, HPIMSGX__init(phm, phr); break; - case HPI_SUBSYS_DELETE_ADAPTER: - HPIMSGX__cleanup(phm->obj_index, h_owner); - { - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_CLOSE); - hm.adapter_index = phm->obj_index; - hw_entry_point(&hm, &hr); - } - if ((phm->obj_index < HPI_MAX_ADAPTERS) - && hpi_entry_points[phm->obj_index]) { - hpi_entry_points[phm->obj_index] (phm, phr); - hpi_entry_points[phm->obj_index] = NULL; - } else - phr->error = HPI_ERROR_INVALID_OBJ_INDEX; - - break; default: /* Must explicitly handle every subsys message in this switch */ hpi_init_response(phr, HPI_OBJ_SUBSYSTEM, phm->function, @@ -247,6 +229,19 @@ static void adapter_message(struct hpi_message *phm, struct hpi_response *phr, case HPI_ADAPTER_CLOSE: adapter_close(phm, phr); break; + case HPI_ADAPTER_DELETE: + HPIMSGX__cleanup(phm->adapter_index, h_owner); + { + struct hpi_message hm; + struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, + HPI_ADAPTER_CLOSE); + hm.adapter_index = phm->adapter_index; + hw_entry_point(&hm, &hr); + } + hw_entry_point(phm, phr); + break; + default: hw_entry_point(phm, phr); break; diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index cd624f13ff8..484f4118986 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -25,6 +25,7 @@ Common Linux HPI ioctl and module probe/remove functions #include "hpidebug.h" #include "hpimsgx.h" #include "hpioctl.h" +#include "hpicmn.h" #include #include @@ -161,26 +162,24 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } - pa = &adapters[hm->h.adapter_index]; + switch (hm->h.function) { + case HPI_SUBSYS_CREATE_ADAPTER: + case HPI_ADAPTER_DELETE: + /* Application must not use these functions! */ + hr->h.size = sizeof(hr->h); + hr->h.error = HPI_ERROR_INVALID_OPERATION; + hr->h.function = hm->h.function; + uncopied_bytes = copy_to_user(puhr, hr, hr->h.size); + if (uncopied_bytes) + err = -EFAULT; + else + err = 0; + goto out; + } + hr->h.size = res_max_size; if (hm->h.object == HPI_OBJ_SUBSYSTEM) { - switch (hm->h.function) { - case HPI_SUBSYS_CREATE_ADAPTER: - case HPI_SUBSYS_DELETE_ADAPTER: - /* Application must not use these functions! */ - hr->h.size = sizeof(hr->h); - hr->h.error = HPI_ERROR_INVALID_OPERATION; - hr->h.function = hm->h.function; - uncopied_bytes = copy_to_user(puhr, hr, hr->h.size); - if (uncopied_bytes) - err = -EFAULT; - else - err = 0; - goto out; - - default: - hpi_send_recv_f(&hm->m0, &hr->r0, file); - } + hpi_send_recv_f(&hm->m0, &hr->r0, file); } else { u16 __user *ptr = NULL; u32 size = 0; @@ -188,8 +187,9 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; u32 adapter = hm->h.adapter_index; + pa = &adapters[adapter]; - if ((hm->h.adapter_index > HPI_MAX_ADAPTERS) || (!pa->type)) { + if ((adapter > HPI_MAX_ADAPTERS) || (!pa->type)) { hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, HPI_ADAPTER_OPEN, HPI_ERROR_BAD_ADAPTER_NUMBER); @@ -395,17 +395,20 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, adapter.index = hr.u.s.adapter_index; adapter.type = hr.u.s.adapter_type; + + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, + HPI_ADAPTER_OPEN); hm.adapter_index = adapter.index; + hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); - err = hpi_adapter_open(adapter.index); - if (err) + if (hr.error) goto err; adapter.snd_card_asihpi = NULL; /* WARNING can't init mutex in 'adapter' * and then copy it to adapters[] ?!?! */ - adapters[hr.u.s.adapter_index] = adapter; + adapters[adapter.index] = adapter; mutex_init(&adapters[adapter.index].mutex); pci_set_drvdata(pci_dev, &adapters[adapter.index]); @@ -440,10 +443,9 @@ void __devexit asihpi_adapter_remove(struct pci_dev *pci_dev) struct hpi_adapter *pa; pa = pci_get_drvdata(pci_dev); - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_DELETE_ADAPTER); - hm.obj_index = pa->index; - hm.adapter_index = HPI_ADAPTER_INDEX_INVALID; + hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, + HPI_ADAPTER_DELETE); + hm.adapter_index = pa->index; hpi_send_recv_ex(&hm, &hr, HOWNER_KERNEL); /* unmap PCI memory space, mapped during device init. */ -- cgit v1.2.3 From 42258daba22897f9859b0f3cb42700322b7c16bc Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Tue, 5 Apr 2011 20:55:48 +1200 Subject: ALSA: asihpi: Minor cleanups Remove some unneeded defintions Use %pR to print resources Make some data const Consistent braces for else Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi_internal.h | 2 -- sound/pci/asihpi/hpicmn.c | 9 +++++++-- sound/pci/asihpi/hpicmn.h | 2 ++ sound/pci/asihpi/hpioctl.c | 9 +++------ 4 files changed, 12 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index d829a65f47d..4fab1595a6a 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1583,9 +1583,7 @@ void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR); /*////////////////////////////////////////////////////////////////////////// */ /* declarations for individual HPI entry points */ -hpi_handler_func HPI_1000; hpi_handler_func HPI_6000; hpi_handler_func HPI_6205; -hpi_handler_func HPI_COMMON; #endif /* _HPI_INTERNAL_H_ */ diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index 0428f285725..b15a02e91f8 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -227,8 +227,9 @@ static unsigned int control_cache_alloc_check(struct hpi_control_cache *pC) if (info->control_type) { pC->p_info[info->control_index] = info; cached++; - } else /* dummy cache entry */ + } else { /* dummy cache entry */ pC->p_info[info->control_index] = NULL; + } byte_count += info->size_in32bit_words * 4; @@ -298,7 +299,7 @@ struct pad_ofs_size { unsigned int field_size; }; -static struct pad_ofs_size pad_desc[] = { +static const struct pad_ofs_size pad_desc[] = { HPICMN_PAD_OFS_AND_SIZE(c_channel), /* HPI_PAD_CHANNEL_NAME */ HPICMN_PAD_OFS_AND_SIZE(c_artist), /* HPI_PAD_ARTIST */ HPICMN_PAD_OFS_AND_SIZE(c_title), /* HPI_PAD_TITLE */ @@ -617,6 +618,10 @@ void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *p_cache, } } +/** Allocate control cache. + +\return Cache pointer, or NULL if allocation fails. +*/ struct hpi_control_cache *hpi_alloc_control_cache(const u32 control_count, const u32 size_in_bytes, u8 *p_dsp_control_buffer) { diff --git a/sound/pci/asihpi/hpicmn.h b/sound/pci/asihpi/hpicmn.h index 590f0b69e65..d53cdf6e535 100644 --- a/sound/pci/asihpi/hpicmn.h +++ b/sound/pci/asihpi/hpicmn.h @@ -60,3 +60,5 @@ void hpi_cmn_control_cache_sync_to_msg(struct hpi_control_cache *pC, struct hpi_message *phm, struct hpi_response *phr); u16 hpi_validate_response(struct hpi_message *phm, struct hpi_response *phr); + +hpi_handler_func HPI_COMMON; diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 484f4118986..d8e7047512f 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -317,7 +317,7 @@ out: int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, const struct pci_device_id *pci_id) { - int err, idx, nm; + int idx, nm; unsigned int memlen; struct hpi_message hm; struct hpi_response hr; @@ -351,11 +351,8 @@ int __devinit asihpi_adapter_probe(struct pci_dev *pci_dev, nm = HPI_MAX_ADAPTER_MEM_SPACES; for (idx = 0; idx < nm; idx++) { - HPI_DEBUG_LOG(INFO, "resource %d %s %08llx-%08llx %04llx\n", - idx, pci_dev->resource[idx].name, - (unsigned long long)pci_resource_start(pci_dev, idx), - (unsigned long long)pci_resource_end(pci_dev, idx), - (unsigned long long)pci_resource_flags(pci_dev, idx)); + HPI_DEBUG_LOG(INFO, "resource %d %pR\n", idx, + &pci_dev->resource[idx]); if (pci_resource_flags(pci_dev, idx) & IORESOURCE_MEM) { memlen = pci_resource_len(pci_dev, idx); -- cgit v1.2.3 From 4bf4a6c5b1b2c73cd61d46d77e7da814effc6058 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Tue, 5 Apr 2011 22:47:15 +0800 Subject: ALSA: hda - Fix alc662_dac_nid and change "6stack-dig" to "5stack-dig" alc662 series only have 3 DAC, so it can only support 5stack-dig instead of 6stack-dig. [updated HD-Audio-Models.txt as well by tiwai] Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 12c6f4508c5..a0262306508 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17432,8 +17432,8 @@ static int patch_alc861vd(struct hda_codec *codec) #define ALC662_DIGOUT_NID 0x06 #define ALC662_DIGIN_NID 0x0a -static hda_nid_t alc662_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ +static hda_nid_t alc662_dac_nids[3] = { + /* front, rear, clfe */ 0x02, 0x03, 0x04 }; @@ -18691,7 +18691,7 @@ static const char * const alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_2ch_DIG] = "3stack-dig", [ALC662_3ST_6ch_DIG] = "3stack-6ch-dig", [ALC662_3ST_6ch] = "3stack-6ch", - [ALC662_5ST_DIG] = "6stack-dig", + [ALC662_5ST_DIG] = "5stack-dig", [ALC662_LENOVO_101E] = "lenovo-101e", [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20", -- cgit v1.2.3 From e217b960e4f82610946fcad764b8af017a4811c0 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Wed, 30 Mar 2011 20:00:39 +0800 Subject: ALSA: emu10k1 - Remove CLFE-related controls for SB Live! Platinum CT4760P SB Live! Platinum CT4760P is just a 4 channels sound card with STAC9721 and Philips UDA1334 DAC. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emumixer.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 05afe06e353..1b50a23232b 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1913,6 +1913,12 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, for (; *c; c += 2) rename_ctl(card, c[0], c[1]); + if (emu->card_capabilities->subsystem == 0x80401102) { /* SB Live! Platinum CT4760P */ + remove_ctl(card, "Center Playback Volume"); + remove_ctl(card, "LFE Playback Volume"); + remove_ctl(card, "Wave Center Playback Volume"); + remove_ctl(card, "Wave LFE Playback Volume"); + } if (emu->card_capabilities->subsystem == 0x20071102) { /* Audigy 4 Pro */ rename_ctl(card, "Line2 Capture Volume", "Line1/Mic Capture Volume"); rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume"); -- cgit v1.2.3 From 1bc7cf99a90b4cfbeed362f3f238bcc9fc2f2c28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Apr 2011 09:42:29 +0200 Subject: ALSA: hda - Correct initial dac_nids for some ALC272-quirks Some ALC272-quirks use alc662_dac_nids instead of alc272_dac_nids. This patch fixes these entries. No functional change since the first two elements are identical in both arrays. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a0262306508..e62fe7f2aef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19059,7 +19059,7 @@ static struct alc_config_preset alc662_presets[] = { .cap_mixer = alc272_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc272_dell_init_verbs }, .num_dacs = ARRAY_SIZE(alc272_dac_nids), - .dac_nids = alc662_dac_nids, + .dac_nids = alc272_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .adc_nids = alc272_adc_nids, .num_adc_nids = ARRAY_SIZE(alc272_adc_nids), @@ -19074,7 +19074,7 @@ static struct alc_config_preset alc662_presets[] = { .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, alc272_dell_zm1_init_verbs }, .num_dacs = ARRAY_SIZE(alc272_dac_nids), - .dac_nids = alc662_dac_nids, + .dac_nids = alc272_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .adc_nids = alc662_adc_nids, .num_adc_nids = 1, -- cgit v1.2.3 From 1304ac8993e32c0530bc82bf1d3f953548a20971 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Apr 2011 15:16:21 +0200 Subject: ALSA: hda - Fix mix->DAC deduction for ALC892 The current alc662 parser doesn't set the DAC for the mixer 0x0f properly for ALC892, which has 4 DACs while ALC662 has 3. Fixed by implementing alc662_mix_to_dac() more genericly with the dynamic widget list. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e62fe7f2aef..d566eac08d6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19105,16 +19105,17 @@ static struct alc_config_preset alc662_presets[] = { */ /* convert from MIX nid to DAC */ -static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid) -{ - if (nid == 0x0f) - return 0x02; - else if (nid >= 0x0c && nid <= 0x0e) - return nid - 0x0c + 0x02; - else if (nid == 0x26) /* ALC887-VD has this DAC too */ - return 0x25; - else - return 0; +static hda_nid_t alc662_mix_to_dac(struct hda_codec *codec, hda_nid_t nid) +{ + hda_nid_t list[4]; + int i, num; + + num = snd_hda_get_connections(codec, nid, list, ARRAY_SIZE(list)); + for (i = 0; i < num; i++) { + if (get_wcaps_type(get_wcaps(codec, list[i])) == AC_WID_AUD_OUT) + return list[i]; + } + return 0; } /* get MIX nid connected to the given pin targeted to DAC */ @@ -19126,7 +19127,7 @@ static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin, num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix)); for (i = 0; i < num; i++) { - if (alc662_mix_to_dac(mix[i]) == dac) + if (alc662_mix_to_dac(codec, mix[i]) == dac) return mix[i]; } return 0; @@ -19143,7 +19144,7 @@ static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin) if (num < 0) return 0; for (i = 0; i < num; i++) { - hda_nid_t nid = alc662_mix_to_dac(srcs[i]); + hda_nid_t nid = alc662_mix_to_dac(codec, srcs[i]); if (!nid) continue; for (j = 0; j < spec->multiout.num_dacs; j++) @@ -19297,7 +19298,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, if (num <= 1) return; for (i = 0; i < num; i++) { - if (alc662_mix_to_dac(srcs[i]) != dac) + if (alc662_mix_to_dac(codec, srcs[i]) != dac) continue; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i); return; -- cgit v1.2.3 From 82a5a936f6dea13849d93a2899a9b7294a8db336 Mon Sep 17 00:00:00 2001 From: Peter Hsiang Date: Mon, 4 Apr 2011 19:35:30 -0700 Subject: ASoC: Add max98095 CODEC driver This patch adds the MAX98095 CODEC driver. Signed-off-by: Peter Hsiang Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98095.c | 2009 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98095.h | 284 ++++++ 4 files changed, 2299 insertions(+) create mode 100644 sound/soc/codecs/max98095.c create mode 100644 sound/soc/codecs/max98095.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b814ed0a163..78da05b5e5e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -187,6 +188,9 @@ config SND_SOC_DMIC config SND_SOC_MAX98088 tristate +config SND_SOC_MAX98095 + tristate + config SND_SOC_MAX9850 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49121ad0e17..f030c182674 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o +snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -108,6 +109,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c new file mode 100644 index 00000000000..9c77f17a6af --- /dev/null +++ b/sound/soc/codecs/max98095.c @@ -0,0 +1,2009 @@ +/* + * max98095.c -- MAX98095 ALSA SoC Audio driver + * + * Copyright 2011 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "max98095.h" + +enum max98095_type { + MAX98095, +}; + +struct max98095_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98095_priv { + enum max98095_type devtype; + void *control_data; + struct max98095_pdata *pdata; + unsigned int sysclk; + struct max98095_cdata dai[3]; + u8 lin_state; + unsigned int mic1pre; + unsigned int mic2pre; +}; + +static const u8 max98095_reg_def[M98095_REG_CNT] = { + 0x00, /* 00 */ + 0x00, /* 01 */ + 0x00, /* 02 */ + 0x00, /* 03 */ + 0x00, /* 04 */ + 0x00, /* 05 */ + 0x00, /* 06 */ + 0x00, /* 07 */ + 0x00, /* 08 */ + 0x00, /* 09 */ + 0x00, /* 0A */ + 0x00, /* 0B */ + 0x00, /* 0C */ + 0x00, /* 0D */ + 0x00, /* 0E */ + 0x00, /* 0F */ + 0x00, /* 10 */ + 0x00, /* 11 */ + 0x00, /* 12 */ + 0x00, /* 13 */ + 0x00, /* 14 */ + 0x00, /* 15 */ + 0x00, /* 16 */ + 0x00, /* 17 */ + 0x00, /* 18 */ + 0x00, /* 19 */ + 0x00, /* 1A */ + 0x00, /* 1B */ + 0x00, /* 1C */ + 0x00, /* 1D */ + 0x00, /* 1E */ + 0x00, /* 1F */ + 0x00, /* 20 */ + 0x00, /* 21 */ + 0x00, /* 22 */ + 0x00, /* 23 */ + 0x00, /* 24 */ + 0x00, /* 25 */ + 0x00, /* 26 */ + 0x00, /* 27 */ + 0x00, /* 28 */ + 0x00, /* 29 */ + 0x00, /* 2A */ + 0x00, /* 2B */ + 0x00, /* 2C */ + 0x00, /* 2D */ + 0x00, /* 2E */ + 0x00, /* 2F */ + 0x00, /* 30 */ + 0x00, /* 31 */ + 0x00, /* 32 */ + 0x00, /* 33 */ + 0x00, /* 34 */ + 0x00, /* 35 */ + 0x00, /* 36 */ + 0x00, /* 37 */ + 0x00, /* 38 */ + 0x00, /* 39 */ + 0x00, /* 3A */ + 0x00, /* 3B */ + 0x00, /* 3C */ + 0x00, /* 3D */ + 0x00, /* 3E */ + 0x00, /* 3F */ + 0x00, /* 40 */ + 0x00, /* 41 */ + 0x00, /* 42 */ + 0x00, /* 43 */ + 0x00, /* 44 */ + 0x00, /* 45 */ + 0x00, /* 46 */ + 0x00, /* 47 */ + 0x00, /* 48 */ + 0x00, /* 49 */ + 0x00, /* 4A */ + 0x00, /* 4B */ + 0x00, /* 4C */ + 0x00, /* 4D */ + 0x00, /* 4E */ + 0x00, /* 4F */ + 0x00, /* 50 */ + 0x00, /* 51 */ + 0x00, /* 52 */ + 0x00, /* 53 */ + 0x00, /* 54 */ + 0x00, /* 55 */ + 0x00, /* 56 */ + 0x00, /* 57 */ + 0x00, /* 58 */ + 0x00, /* 59 */ + 0x00, /* 5A */ + 0x00, /* 5B */ + 0x00, /* 5C */ + 0x00, /* 5D */ + 0x00, /* 5E */ + 0x00, /* 5F */ + 0x00, /* 60 */ + 0x00, /* 61 */ + 0x00, /* 62 */ + 0x00, /* 63 */ + 0x00, /* 64 */ + 0x00, /* 65 */ + 0x00, /* 66 */ + 0x00, /* 67 */ + 0x00, /* 68 */ + 0x00, /* 69 */ + 0x00, /* 6A */ + 0x00, /* 6B */ + 0x00, /* 6C */ + 0x00, /* 6D */ + 0x00, /* 6E */ + 0x00, /* 6F */ + 0x00, /* 70 */ + 0x00, /* 71 */ + 0x00, /* 72 */ + 0x00, /* 73 */ + 0x00, /* 74 */ + 0x00, /* 75 */ + 0x00, /* 76 */ + 0x00, /* 77 */ + 0x00, /* 78 */ + 0x00, /* 79 */ + 0x00, /* 7A */ + 0x00, /* 7B */ + 0x00, /* 7C */ + 0x00, /* 7D */ + 0x00, /* 7E */ + 0x00, /* 7F */ + 0x00, /* 80 */ + 0x00, /* 81 */ + 0x00, /* 82 */ + 0x00, /* 83 */ + 0x00, /* 84 */ + 0x00, /* 85 */ + 0x00, /* 86 */ + 0x00, /* 87 */ + 0x00, /* 88 */ + 0x00, /* 89 */ + 0x00, /* 8A */ + 0x00, /* 8B */ + 0x00, /* 8C */ + 0x00, /* 8D */ + 0x00, /* 8E */ + 0x00, /* 8F */ + 0x00, /* 90 */ + 0x00, /* 91 */ + 0x30, /* 92 */ + 0xF0, /* 93 */ + 0x00, /* 94 */ + 0x00, /* 95 */ + 0x3F, /* 96 */ + 0x00, /* 97 */ + 0x00, /* 98 */ + 0x00, /* 99 */ + 0x00, /* 9A */ + 0x00, /* 9B */ + 0x00, /* 9C */ + 0x00, /* 9D */ + 0x00, /* 9E */ + 0x00, /* 9F */ + 0x00, /* A0 */ + 0x00, /* A1 */ + 0x00, /* A2 */ + 0x00, /* A3 */ + 0x00, /* A4 */ + 0x00, /* A5 */ + 0x00, /* A6 */ + 0x00, /* A7 */ + 0x00, /* A8 */ + 0x00, /* A9 */ + 0x00, /* AA */ + 0x00, /* AB */ + 0x00, /* AC */ + 0x00, /* AD */ + 0x00, /* AE */ + 0x00, /* AF */ + 0x00, /* B0 */ + 0x00, /* B1 */ + 0x00, /* B2 */ + 0x00, /* B3 */ + 0x00, /* B4 */ + 0x00, /* B5 */ + 0x00, /* B6 */ + 0x00, /* B7 */ + 0x00, /* B8 */ + 0x00, /* B9 */ + 0x00, /* BA */ + 0x00, /* BB */ + 0x00, /* BC */ + 0x00, /* BD */ + 0x00, /* BE */ + 0x00, /* BF */ + 0x00, /* C0 */ + 0x00, /* C1 */ + 0x00, /* C2 */ + 0x00, /* C3 */ + 0x00, /* C4 */ + 0x00, /* C5 */ + 0x00, /* C6 */ + 0x00, /* C7 */ + 0x00, /* C8 */ + 0x00, /* C9 */ + 0x00, /* CA */ + 0x00, /* CB */ + 0x00, /* CC */ + 0x00, /* CD */ + 0x00, /* CE */ + 0x00, /* CF */ + 0x00, /* D0 */ + 0x00, /* D1 */ + 0x00, /* D2 */ + 0x00, /* D3 */ + 0x00, /* D4 */ + 0x00, /* D5 */ + 0x00, /* D6 */ + 0x00, /* D7 */ + 0x00, /* D8 */ + 0x00, /* D9 */ + 0x00, /* DA */ + 0x00, /* DB */ + 0x00, /* DC */ + 0x00, /* DD */ + 0x00, /* DE */ + 0x00, /* DF */ + 0x00, /* E0 */ + 0x00, /* E1 */ + 0x00, /* E2 */ + 0x00, /* E3 */ + 0x00, /* E4 */ + 0x00, /* E5 */ + 0x00, /* E6 */ + 0x00, /* E7 */ + 0x00, /* E8 */ + 0x00, /* E9 */ + 0x00, /* EA */ + 0x00, /* EB */ + 0x00, /* EC */ + 0x00, /* ED */ + 0x00, /* EE */ + 0x00, /* EF */ + 0x00, /* F0 */ + 0x00, /* F1 */ + 0x00, /* F2 */ + 0x00, /* F3 */ + 0x00, /* F4 */ + 0x00, /* F5 */ + 0x00, /* F6 */ + 0x00, /* F7 */ + 0x00, /* F8 */ + 0x00, /* F9 */ + 0x00, /* FA */ + 0x00, /* FB */ + 0x00, /* FC */ + 0x00, /* FD */ + 0x00, /* FE */ + 0x00, /* FF */ +}; + +static struct { + int readable; + int writable; +} max98095_access[M98095_REG_CNT] = { + { 0x00, 0x00 }, /* 00 */ + { 0xFF, 0x00 }, /* 01 */ + { 0xFF, 0x00 }, /* 02 */ + { 0xFF, 0x00 }, /* 03 */ + { 0xFF, 0x00 }, /* 04 */ + { 0xFF, 0x00 }, /* 05 */ + { 0xFF, 0x00 }, /* 06 */ + { 0xFF, 0x00 }, /* 07 */ + { 0xFF, 0x00 }, /* 08 */ + { 0xFF, 0x00 }, /* 09 */ + { 0xFF, 0x00 }, /* 0A */ + { 0xFF, 0x00 }, /* 0B */ + { 0xFF, 0x00 }, /* 0C */ + { 0xFF, 0x00 }, /* 0D */ + { 0xFF, 0x00 }, /* 0E */ + { 0xFF, 0x9F }, /* 0F */ + { 0xFF, 0xFF }, /* 10 */ + { 0xFF, 0xFF }, /* 11 */ + { 0xFF, 0xFF }, /* 12 */ + { 0xFF, 0xFF }, /* 13 */ + { 0xFF, 0xFF }, /* 14 */ + { 0xFF, 0xFF }, /* 15 */ + { 0xFF, 0xFF }, /* 16 */ + { 0xFF, 0xFF }, /* 17 */ + { 0xFF, 0xFF }, /* 18 */ + { 0xFF, 0xFF }, /* 19 */ + { 0xFF, 0xFF }, /* 1A */ + { 0xFF, 0xFF }, /* 1B */ + { 0xFF, 0xFF }, /* 1C */ + { 0xFF, 0xFF }, /* 1D */ + { 0xFF, 0x77 }, /* 1E */ + { 0xFF, 0x77 }, /* 1F */ + { 0xFF, 0x77 }, /* 20 */ + { 0xFF, 0x77 }, /* 21 */ + { 0xFF, 0x77 }, /* 22 */ + { 0xFF, 0x77 }, /* 23 */ + { 0xFF, 0xFF }, /* 24 */ + { 0xFF, 0x7F }, /* 25 */ + { 0xFF, 0x31 }, /* 26 */ + { 0xFF, 0xFF }, /* 27 */ + { 0xFF, 0xFF }, /* 28 */ + { 0xFF, 0xFF }, /* 29 */ + { 0xFF, 0xF7 }, /* 2A */ + { 0xFF, 0x2F }, /* 2B */ + { 0xFF, 0xEF }, /* 2C */ + { 0xFF, 0xFF }, /* 2D */ + { 0xFF, 0xFF }, /* 2E */ + { 0xFF, 0xFF }, /* 2F */ + { 0xFF, 0xFF }, /* 30 */ + { 0xFF, 0xFF }, /* 31 */ + { 0xFF, 0xFF }, /* 32 */ + { 0xFF, 0xFF }, /* 33 */ + { 0xFF, 0xF7 }, /* 34 */ + { 0xFF, 0x2F }, /* 35 */ + { 0xFF, 0xCF }, /* 36 */ + { 0xFF, 0xFF }, /* 37 */ + { 0xFF, 0xFF }, /* 38 */ + { 0xFF, 0xFF }, /* 39 */ + { 0xFF, 0xFF }, /* 3A */ + { 0xFF, 0xFF }, /* 3B */ + { 0xFF, 0xFF }, /* 3C */ + { 0xFF, 0xFF }, /* 3D */ + { 0xFF, 0xF7 }, /* 3E */ + { 0xFF, 0x2F }, /* 3F */ + { 0xFF, 0xCF }, /* 40 */ + { 0xFF, 0xFF }, /* 41 */ + { 0xFF, 0x77 }, /* 42 */ + { 0xFF, 0xFF }, /* 43 */ + { 0xFF, 0xFF }, /* 44 */ + { 0xFF, 0xFF }, /* 45 */ + { 0xFF, 0xFF }, /* 46 */ + { 0xFF, 0xFF }, /* 47 */ + { 0xFF, 0xFF }, /* 48 */ + { 0xFF, 0x0F }, /* 49 */ + { 0xFF, 0xFF }, /* 4A */ + { 0xFF, 0xFF }, /* 4B */ + { 0xFF, 0x3F }, /* 4C */ + { 0xFF, 0x3F }, /* 4D */ + { 0xFF, 0x3F }, /* 4E */ + { 0xFF, 0xFF }, /* 4F */ + { 0xFF, 0x7F }, /* 50 */ + { 0xFF, 0x7F }, /* 51 */ + { 0xFF, 0x0F }, /* 52 */ + { 0xFF, 0x3F }, /* 53 */ + { 0xFF, 0x3F }, /* 54 */ + { 0xFF, 0x3F }, /* 55 */ + { 0xFF, 0xFF }, /* 56 */ + { 0xFF, 0xFF }, /* 57 */ + { 0xFF, 0xBF }, /* 58 */ + { 0xFF, 0x1F }, /* 59 */ + { 0xFF, 0xBF }, /* 5A */ + { 0xFF, 0x1F }, /* 5B */ + { 0xFF, 0xBF }, /* 5C */ + { 0xFF, 0x3F }, /* 5D */ + { 0xFF, 0x3F }, /* 5E */ + { 0xFF, 0x7F }, /* 5F */ + { 0xFF, 0x7F }, /* 60 */ + { 0xFF, 0x47 }, /* 61 */ + { 0xFF, 0x9F }, /* 62 */ + { 0xFF, 0x9F }, /* 63 */ + { 0xFF, 0x9F }, /* 64 */ + { 0xFF, 0x9F }, /* 65 */ + { 0xFF, 0x9F }, /* 66 */ + { 0xFF, 0xBF }, /* 67 */ + { 0xFF, 0xBF }, /* 68 */ + { 0xFF, 0xFF }, /* 69 */ + { 0xFF, 0xFF }, /* 6A */ + { 0xFF, 0x7F }, /* 6B */ + { 0xFF, 0xF7 }, /* 6C */ + { 0xFF, 0xFF }, /* 6D */ + { 0xFF, 0xFF }, /* 6E */ + { 0xFF, 0x1F }, /* 6F */ + { 0xFF, 0xF7 }, /* 70 */ + { 0xFF, 0xFF }, /* 71 */ + { 0xFF, 0xFF }, /* 72 */ + { 0xFF, 0x1F }, /* 73 */ + { 0xFF, 0xF7 }, /* 74 */ + { 0xFF, 0xFF }, /* 75 */ + { 0xFF, 0xFF }, /* 76 */ + { 0xFF, 0x1F }, /* 77 */ + { 0xFF, 0xF7 }, /* 78 */ + { 0xFF, 0xFF }, /* 79 */ + { 0xFF, 0xFF }, /* 7A */ + { 0xFF, 0x1F }, /* 7B */ + { 0xFF, 0xF7 }, /* 7C */ + { 0xFF, 0xFF }, /* 7D */ + { 0xFF, 0xFF }, /* 7E */ + { 0xFF, 0x1F }, /* 7F */ + { 0xFF, 0xF7 }, /* 80 */ + { 0xFF, 0xFF }, /* 81 */ + { 0xFF, 0xFF }, /* 82 */ + { 0xFF, 0x1F }, /* 83 */ + { 0xFF, 0x7F }, /* 84 */ + { 0xFF, 0x0F }, /* 85 */ + { 0xFF, 0xD8 }, /* 86 */ + { 0xFF, 0xFF }, /* 87 */ + { 0xFF, 0xEF }, /* 88 */ + { 0xFF, 0xFE }, /* 89 */ + { 0xFF, 0xFE }, /* 8A */ + { 0xFF, 0xFF }, /* 8B */ + { 0xFF, 0xFF }, /* 8C */ + { 0xFF, 0x3F }, /* 8D */ + { 0xFF, 0xFF }, /* 8E */ + { 0xFF, 0x3F }, /* 8F */ + { 0xFF, 0x8F }, /* 90 */ + { 0xFF, 0xFF }, /* 91 */ + { 0xFF, 0x3F }, /* 92 */ + { 0xFF, 0xFF }, /* 93 */ + { 0xFF, 0xFF }, /* 94 */ + { 0xFF, 0x0F }, /* 95 */ + { 0xFF, 0x3F }, /* 96 */ + { 0xFF, 0x8C }, /* 97 */ + { 0x00, 0x00 }, /* 98 */ + { 0x00, 0x00 }, /* 99 */ + { 0x00, 0x00 }, /* 9A */ + { 0x00, 0x00 }, /* 9B */ + { 0x00, 0x00 }, /* 9C */ + { 0x00, 0x00 }, /* 9D */ + { 0x00, 0x00 }, /* 9E */ + { 0x00, 0x00 }, /* 9F */ + { 0x00, 0x00 }, /* A0 */ + { 0x00, 0x00 }, /* A1 */ + { 0x00, 0x00 }, /* A2 */ + { 0x00, 0x00 }, /* A3 */ + { 0x00, 0x00 }, /* A4 */ + { 0x00, 0x00 }, /* A5 */ + { 0x00, 0x00 }, /* A6 */ + { 0x00, 0x00 }, /* A7 */ + { 0x00, 0x00 }, /* A8 */ + { 0x00, 0x00 }, /* A9 */ + { 0x00, 0x00 }, /* AA */ + { 0x00, 0x00 }, /* AB */ + { 0x00, 0x00 }, /* AC */ + { 0x00, 0x00 }, /* AD */ + { 0x00, 0x00 }, /* AE */ + { 0x00, 0x00 }, /* AF */ + { 0x00, 0x00 }, /* B0 */ + { 0x00, 0x00 }, /* B1 */ + { 0x00, 0x00 }, /* B2 */ + { 0x00, 0x00 }, /* B3 */ + { 0x00, 0x00 }, /* B4 */ + { 0x00, 0x00 }, /* B5 */ + { 0x00, 0x00 }, /* B6 */ + { 0x00, 0x00 }, /* B7 */ + { 0x00, 0x00 }, /* B8 */ + { 0x00, 0x00 }, /* B9 */ + { 0x00, 0x00 }, /* BA */ + { 0x00, 0x00 }, /* BB */ + { 0x00, 0x00 }, /* BC */ + { 0x00, 0x00 }, /* BD */ + { 0x00, 0x00 }, /* BE */ + { 0x00, 0x00 }, /* BF */ + { 0x00, 0x00 }, /* C0 */ + { 0x00, 0x00 }, /* C1 */ + { 0x00, 0x00 }, /* C2 */ + { 0x00, 0x00 }, /* C3 */ + { 0x00, 0x00 }, /* C4 */ + { 0x00, 0x00 }, /* C5 */ + { 0x00, 0x00 }, /* C6 */ + { 0x00, 0x00 }, /* C7 */ + { 0x00, 0x00 }, /* C8 */ + { 0x00, 0x00 }, /* C9 */ + { 0x00, 0x00 }, /* CA */ + { 0x00, 0x00 }, /* CB */ + { 0x00, 0x00 }, /* CC */ + { 0x00, 0x00 }, /* CD */ + { 0x00, 0x00 }, /* CE */ + { 0x00, 0x00 }, /* CF */ + { 0x00, 0x00 }, /* D0 */ + { 0x00, 0x00 }, /* D1 */ + { 0x00, 0x00 }, /* D2 */ + { 0x00, 0x00 }, /* D3 */ + { 0x00, 0x00 }, /* D4 */ + { 0x00, 0x00 }, /* D5 */ + { 0x00, 0x00 }, /* D6 */ + { 0x00, 0x00 }, /* D7 */ + { 0x00, 0x00 }, /* D8 */ + { 0x00, 0x00 }, /* D9 */ + { 0x00, 0x00 }, /* DA */ + { 0x00, 0x00 }, /* DB */ + { 0x00, 0x00 }, /* DC */ + { 0x00, 0x00 }, /* DD */ + { 0x00, 0x00 }, /* DE */ + { 0x00, 0x00 }, /* DF */ + { 0x00, 0x00 }, /* E0 */ + { 0x00, 0x00 }, /* E1 */ + { 0x00, 0x00 }, /* E2 */ + { 0x00, 0x00 }, /* E3 */ + { 0x00, 0x00 }, /* E4 */ + { 0x00, 0x00 }, /* E5 */ + { 0x00, 0x00 }, /* E6 */ + { 0x00, 0x00 }, /* E7 */ + { 0x00, 0x00 }, /* E8 */ + { 0x00, 0x00 }, /* E9 */ + { 0x00, 0x00 }, /* EA */ + { 0x00, 0x00 }, /* EB */ + { 0x00, 0x00 }, /* EC */ + { 0x00, 0x00 }, /* ED */ + { 0x00, 0x00 }, /* EE */ + { 0x00, 0x00 }, /* EF */ + { 0x00, 0x00 }, /* F0 */ + { 0x00, 0x00 }, /* F1 */ + { 0x00, 0x00 }, /* F2 */ + { 0x00, 0x00 }, /* F3 */ + { 0x00, 0x00 }, /* F4 */ + { 0x00, 0x00 }, /* F5 */ + { 0x00, 0x00 }, /* F6 */ + { 0x00, 0x00 }, /* F7 */ + { 0x00, 0x00 }, /* F8 */ + { 0x00, 0x00 }, /* F9 */ + { 0x00, 0x00 }, /* FA */ + { 0x00, 0x00 }, /* FB */ + { 0x00, 0x00 }, /* FC */ + { 0x00, 0x00 }, /* FD */ + { 0x00, 0x00 }, /* FE */ + { 0xFF, 0x00 }, /* FF */ +}; + +static int max98095_readable(struct snd_soc_codec *codec, unsigned int reg) +{ + if (reg >= M98095_REG_CNT) + return 0; + return max98095_access[reg].readable != 0; +} + +static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) +{ + if (reg > M98095_REG_MAX_CACHED) + return 1; + + switch (reg) { + case M98095_000_HOST_DATA: + case M98095_001_HOST_INT_STS: + case M98095_002_HOST_RSP_STS: + case M98095_003_HOST_CMD_STS: + case M98095_004_CODEC_STS: + case M98095_005_DAI1_ALC_STS: + case M98095_006_DAI2_ALC_STS: + case M98095_007_JACK_AUTO_STS: + case M98095_008_JACK_MANUAL_STS: + case M98095_009_JACK_VBAT_STS: + case M98095_00A_ACC_ADC_STS: + case M98095_00B_MIC_NG_AGC_STS: + case M98095_00C_SPK_L_VOLT_STS: + case M98095_00D_SPK_R_VOLT_STS: + case M98095_00E_TEMP_SENSOR_STS: + return 1; + } + + return 0; +} + +static const char * const max98095_fltr_mode[] = { "Voice", "Music" }; +static const struct soc_enum max98095_dai1_filter_mode_enum[] = { + SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode), +}; +static const struct soc_enum max98095_dai2_filter_mode_enum[] = { + SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 7, 2, max98095_fltr_mode), +}; + +static const char * const max98095_extmic_text[] = { "None", "MIC1", "MIC2" }; + +static const struct soc_enum max98095_extmic_enum = + SOC_ENUM_SINGLE(M98095_087_CFG_MIC, 0, 3, max98095_extmic_text); + +static const struct snd_kcontrol_new max98095_extmic_mux = + SOC_DAPM_ENUM("External MIC Mux", max98095_extmic_enum); + +static const char * const max98095_linein_text[] = { "INA", "INB" }; + +static const struct soc_enum max98095_linein_enum = + SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 6, 2, max98095_linein_text); + +static const struct snd_kcontrol_new max98095_linein_mux = + SOC_DAPM_ENUM("Linein Input Mux", max98095_linein_enum); + +static const char * const max98095_line_mode_text[] = { + "Stereo", "Differential"}; + +static const struct soc_enum max98095_linein_mode_enum = + SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 7, 2, max98095_line_mode_text); + +static const struct soc_enum max98095_lineout_mode_enum = + SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 4, 2, max98095_line_mode_text); + +static const char * const max98095_dai_fltr[] = { + "Off", "Elliptical-HPF-16k", "Butterworth-HPF-16k", + "Elliptical-HPF-8k", "Butterworth-HPF-8k", "Butterworth-HPF-Fs/240"}; +static const struct soc_enum max98095_dai1_dac_filter_enum[] = { + SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 0, 6, max98095_dai_fltr), +}; +static const struct soc_enum max98095_dai2_dac_filter_enum[] = { + SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 0, 6, max98095_dai_fltr), +}; +static const struct soc_enum max98095_dai3_dac_filter_enum[] = { + SOC_ENUM_SINGLE(M98095_042_DAI3_FILTERS, 0, 6, max98095_dai_fltr), +}; + +static int max98095_mic1pre_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + max98095->mic1pre = sel; + snd_soc_update_bits(codec, M98095_05F_LVL_MIC1, M98095_MICPRE_MASK, + (1+sel)<value.integer.value[0] = max98095->mic1pre; + return 0; +} + +static int max98095_mic2pre_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + max98095->mic2pre = sel; + snd_soc_update_bits(codec, M98095_060_LVL_MIC2, M98095_MICPRE_MASK, + (1+sel)<value.integer.value[0] = max98095->mic2pre; + return 0; +} + +static const unsigned int max98095_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98095_mic_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98095_adc_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98095_adcboost_tlv, 0, 600, 0); + +static const unsigned int max98095_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98095_spk_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 10, TLV_DB_SCALE_ITEM(-5900, 400, 0), + 11, 18, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 19, 27, TLV_DB_SCALE_ITEM(-200, 100, 0), + 28, 39, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static const unsigned int max98095_rcv_lout_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static const unsigned int max98095_lin_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 2, TLV_DB_SCALE_ITEM(-600, 300, 0), + 3, 3, TLV_DB_SCALE_ITEM(300, 1100, 0), + 4, 5, TLV_DB_SCALE_ITEM(1400, 600, 0), +}; + +static const struct snd_kcontrol_new max98095_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Headphone Volume", M98095_064_LVL_HP_L, + M98095_065_LVL_HP_R, 0, 31, 0, max98095_hp_tlv), + + SOC_DOUBLE_R_TLV("Speaker Volume", M98095_067_LVL_SPK_L, + M98095_068_LVL_SPK_R, 0, 39, 0, max98095_spk_tlv), + + SOC_SINGLE_TLV("Receiver Volume", M98095_066_LVL_RCV, + 0, 31, 0, max98095_rcv_lout_tlv), + + SOC_DOUBLE_R_TLV("Lineout Volume", M98095_062_LVL_LINEOUT1, + M98095_063_LVL_LINEOUT2, 0, 31, 0, max98095_rcv_lout_tlv), + + SOC_DOUBLE_R("Headphone Switch", M98095_064_LVL_HP_L, + M98095_065_LVL_HP_R, 7, 1, 1), + + SOC_DOUBLE_R("Speaker Switch", M98095_067_LVL_SPK_L, + M98095_068_LVL_SPK_R, 7, 1, 1), + + SOC_SINGLE("Receiver Switch", M98095_066_LVL_RCV, 7, 1, 1), + + SOC_DOUBLE_R("Lineout Switch", M98095_062_LVL_LINEOUT1, + M98095_063_LVL_LINEOUT2, 7, 1, 1), + + SOC_SINGLE_TLV("MIC1 Volume", M98095_05F_LVL_MIC1, 0, 20, 1, + max98095_mic_tlv), + + SOC_SINGLE_TLV("MIC2 Volume", M98095_060_LVL_MIC2, 0, 20, 1, + max98095_mic_tlv), + + SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", + M98095_05F_LVL_MIC1, 5, 2, 0, + max98095_mic1pre_get, max98095_mic1pre_set, + max98095_micboost_tlv), + SOC_SINGLE_EXT_TLV("MIC2 Boost Volume", + M98095_060_LVL_MIC2, 5, 2, 0, + max98095_mic2pre_get, max98095_mic2pre_set, + max98095_micboost_tlv), + + SOC_SINGLE_TLV("Linein Volume", M98095_061_LVL_LINEIN, 0, 5, 1, + max98095_lin_tlv), + + SOC_SINGLE_TLV("ADCL Volume", M98095_05D_LVL_ADC_L, 0, 15, 1, + max98095_adc_tlv), + SOC_SINGLE_TLV("ADCR Volume", M98095_05E_LVL_ADC_R, 0, 15, 1, + max98095_adc_tlv), + + SOC_SINGLE_TLV("ADCL Boost Volume", M98095_05D_LVL_ADC_L, 4, 3, 0, + max98095_adcboost_tlv), + SOC_SINGLE_TLV("ADCR Boost Volume", M98095_05E_LVL_ADC_R, 4, 3, 0, + max98095_adcboost_tlv), + + SOC_ENUM("DAI1 Filter Mode", max98095_dai1_filter_mode_enum), + SOC_ENUM("DAI2 Filter Mode", max98095_dai2_filter_mode_enum), + SOC_ENUM("DAI1 DAC Filter", max98095_dai1_dac_filter_enum), + SOC_ENUM("DAI2 DAC Filter", max98095_dai2_dac_filter_enum), + SOC_ENUM("DAI3 DAC Filter", max98095_dai3_dac_filter_enum), + + SOC_ENUM("Linein Mode", max98095_linein_mode_enum), + SOC_ENUM("Lineout Mode", max98095_lineout_mode_enum), +}; + +/* Left speaker mixer switch */ +static const struct snd_kcontrol_new max98095_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98095_050_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98095_050_MIX_SPK_LEFT, 6, 1, 0), + SOC_DAPM_SINGLE("Mono DAC2 Switch", M98095_050_MIX_SPK_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("Mono DAC3 Switch", M98095_050_MIX_SPK_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98095_050_MIX_SPK_LEFT, 4, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_050_MIX_SPK_LEFT, 5, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_050_MIX_SPK_LEFT, 1, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_050_MIX_SPK_LEFT, 2, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct snd_kcontrol_new max98095_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98095_051_MIX_SPK_RIGHT, 6, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98095_051_MIX_SPK_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("Mono DAC2 Switch", M98095_051_MIX_SPK_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("Mono DAC3 Switch", M98095_051_MIX_SPK_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98095_051_MIX_SPK_RIGHT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_051_MIX_SPK_RIGHT, 4, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_051_MIX_SPK_RIGHT, 1, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_051_MIX_SPK_RIGHT, 2, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98095_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98095_04C_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98095_04C_MIX_HP_LEFT, 5, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98095_04C_MIX_HP_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_04C_MIX_HP_LEFT, 4, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_04C_MIX_HP_LEFT, 1, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_04C_MIX_HP_LEFT, 2, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98095_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98095_04D_MIX_HP_RIGHT, 5, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98095_04D_MIX_HP_RIGHT, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98095_04D_MIX_HP_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_04D_MIX_HP_RIGHT, 4, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_04D_MIX_HP_RIGHT, 1, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_04D_MIX_HP_RIGHT, 2, 1, 0), +}; + +/* Receiver earpiece mixer switch */ +static const struct snd_kcontrol_new max98095_mono_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98095_04F_MIX_RCV, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98095_04F_MIX_RCV, 5, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98095_04F_MIX_RCV, 3, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_04F_MIX_RCV, 4, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_04F_MIX_RCV, 1, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_04F_MIX_RCV, 2, 1, 0), +}; + +/* Left lineout mixer switch */ +static const struct snd_kcontrol_new max98095_left_lineout_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98095_053_MIX_LINEOUT1, 5, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98095_053_MIX_LINEOUT1, 0, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98095_053_MIX_LINEOUT1, 3, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_053_MIX_LINEOUT1, 4, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_053_MIX_LINEOUT1, 1, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_053_MIX_LINEOUT1, 2, 1, 0), +}; + +/* Right lineout mixer switch */ +static const struct snd_kcontrol_new max98095_right_lineout_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", M98095_054_MIX_LINEOUT2, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98095_054_MIX_LINEOUT2, 5, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98095_054_MIX_LINEOUT2, 3, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_054_MIX_LINEOUT2, 4, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_054_MIX_LINEOUT2, 1, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_054_MIX_LINEOUT2, 2, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98095_left_ADC_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1 Switch", M98095_04A_MIX_ADC_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_04A_MIX_ADC_LEFT, 6, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_04A_MIX_ADC_LEFT, 3, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_04A_MIX_ADC_LEFT, 2, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98095_right_ADC_mixer_controls[] = { + SOC_DAPM_SINGLE("MIC1 Switch", M98095_04B_MIX_ADC_RIGHT, 7, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98095_04B_MIX_ADC_RIGHT, 6, 1, 0), + SOC_DAPM_SINGLE("IN1 Switch", M98095_04B_MIX_ADC_RIGHT, 3, 1, 0), + SOC_DAPM_SINGLE("IN2 Switch", M98095_04B_MIX_ADC_RIGHT, 2, 1, 0), +}; + +static int max98095_mic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (w->reg == M98095_05F_LVL_MIC1) { + snd_soc_update_bits(codec, w->reg, M98095_MICPRE_MASK, + (1+max98095->mic1pre)<reg, M98095_MICPRE_MASK, + (1+max98095->mic2pre)<reg, M98095_MICPRE_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * The line inputs are stereo inputs with the left and right + * channels sharing a common PGA power control signal. + */ +static int max98095_line_pga(struct snd_soc_dapm_widget *w, + int event, u8 channel) +{ + struct snd_soc_codec *codec = w->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + u8 *state; + + BUG_ON(!((channel == 1) || (channel == 2))); + + state = &max98095->lin_state; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + *state |= channel; + snd_soc_update_bits(codec, w->reg, + (1 << w->shift), (1 << w->shift)); + break; + case SND_SOC_DAPM_POST_PMD: + *state &= ~channel; + if (*state == 0) { + snd_soc_update_bits(codec, w->reg, + (1 << w->shift), 0); + } + break; + default: + return -EINVAL; + } + + return 0; +} + +static int max98095_pga_in1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return max98095_line_pga(w, event, 1); +} + +static int max98095_pga_in2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return max98095_line_pga(w, event, 2); +} + +/* + * The stereo line out mixer outputs to two stereo line outs. + * The 2nd pair has a separate set of enables. + */ +static int max98095_lineout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, w->reg, + (1 << (w->shift+2)), (1 << (w->shift+2))); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, + (1 << (w->shift+2)), 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget max98095_dapm_widgets[] = { + + SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", M98095_090_PWR_EN_IN, 0, 0), + SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", M98095_090_PWR_EN_IN, 1, 0), + + SND_SOC_DAPM_DAC("DACL1", "HiFi Playback", + M98095_091_PWR_EN_OUT, 0, 0), + SND_SOC_DAPM_DAC("DACR1", "HiFi Playback", + M98095_091_PWR_EN_OUT, 1, 0), + SND_SOC_DAPM_DAC("DACM2", "Aux Playback", + M98095_091_PWR_EN_OUT, 2, 0), + SND_SOC_DAPM_DAC("DACM3", "Voice Playback", + M98095_091_PWR_EN_OUT, 2, 0), + + SND_SOC_DAPM_PGA("HP Left Out", M98095_091_PWR_EN_OUT, + 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Right Out", M98095_091_PWR_EN_OUT, + 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Left Out", M98095_091_PWR_EN_OUT, + 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98095_091_PWR_EN_OUT, + 5, 0, NULL, 0), + + SND_SOC_DAPM_PGA("RCV Mono Out", M98095_091_PWR_EN_OUT, + 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA_E("LINE Left Out", M98095_092_PWR_EN_OUT, + 0, 0, NULL, 0, max98095_lineout_event, SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LINE Right Out", M98095_092_PWR_EN_OUT, + 1, 0, NULL, 0, max98095_lineout_event, SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("External MIC", SND_SOC_NOPM, 0, 0, + &max98095_extmic_mux), + + SND_SOC_DAPM_MUX("Linein Mux", SND_SOC_NOPM, 0, 0, + &max98095_linein_mux), + + SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98095_left_hp_mixer_controls[0], + ARRAY_SIZE(max98095_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98095_right_hp_mixer_controls[0], + ARRAY_SIZE(max98095_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98095_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98095_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98095_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98095_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98095_mono_rcv_mixer_controls[0], + ARRAY_SIZE(max98095_mono_rcv_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Lineout Mixer", SND_SOC_NOPM, 0, 0, + &max98095_left_lineout_mixer_controls[0], + ARRAY_SIZE(max98095_left_lineout_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Lineout Mixer", SND_SOC_NOPM, 0, 0, + &max98095_right_lineout_mixer_controls[0], + ARRAY_SIZE(max98095_right_lineout_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98095_left_ADC_mixer_controls[0], + ARRAY_SIZE(max98095_left_ADC_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98095_right_ADC_mixer_controls[0], + ARRAY_SIZE(max98095_right_ADC_mixer_controls)), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98095_05F_LVL_MIC1, + 5, 0, NULL, 0, max98095_mic_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98095_060_LVL_MIC2, + 5, 0, NULL, 0, max98095_mic_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("IN1 Input", M98095_090_PWR_EN_IN, + 7, 0, NULL, 0, max98095_pga_in1_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("IN2 Input", M98095_090_PWR_EN_IN, + 7, 0, NULL, 0, max98095_pga_in2_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MICBIAS("MICBIAS1", M98095_090_PWR_EN_IN, 2, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS2", M98095_090_PWR_EN_IN, 3, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RCV"), + SND_SOC_DAPM_OUTPUT("OUT1"), + SND_SOC_DAPM_OUTPUT("OUT2"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("OUT4"), + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("INA1"), + SND_SOC_DAPM_INPUT("INA2"), + SND_SOC_DAPM_INPUT("INB1"), + SND_SOC_DAPM_INPUT("INB2"), +}; + +static const struct snd_soc_dapm_route max98095_audio_map[] = { + /* Left headphone output mixer */ + {"Left Headphone Mixer", "Left DAC1 Switch", "DACL1"}, + {"Left Headphone Mixer", "Right DAC1 Switch", "DACR1"}, + {"Left Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Headphone Mixer", "IN1 Switch", "IN1 Input"}, + {"Left Headphone Mixer", "IN2 Switch", "IN2 Input"}, + + /* Right headphone output mixer */ + {"Right Headphone Mixer", "Left DAC1 Switch", "DACL1"}, + {"Right Headphone Mixer", "Right DAC1 Switch", "DACR1"}, + {"Right Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Headphone Mixer", "IN1 Switch", "IN1 Input"}, + {"Right Headphone Mixer", "IN2 Switch", "IN2 Input"}, + + /* Left speaker output mixer */ + {"Left Speaker Mixer", "Left DAC1 Switch", "DACL1"}, + {"Left Speaker Mixer", "Right DAC1 Switch", "DACR1"}, + {"Left Speaker Mixer", "Mono DAC2 Switch", "DACM2"}, + {"Left Speaker Mixer", "Mono DAC3 Switch", "DACM3"}, + {"Left Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Speaker Mixer", "IN1 Switch", "IN1 Input"}, + {"Left Speaker Mixer", "IN2 Switch", "IN2 Input"}, + + /* Right speaker output mixer */ + {"Right Speaker Mixer", "Left DAC1 Switch", "DACL1"}, + {"Right Speaker Mixer", "Right DAC1 Switch", "DACR1"}, + {"Right Speaker Mixer", "Mono DAC2 Switch", "DACM2"}, + {"Right Speaker Mixer", "Mono DAC3 Switch", "DACM3"}, + {"Right Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Speaker Mixer", "IN1 Switch", "IN1 Input"}, + {"Right Speaker Mixer", "IN2 Switch", "IN2 Input"}, + + /* Earpiece/Receiver output mixer */ + {"Receiver Mixer", "Left DAC1 Switch", "DACL1"}, + {"Receiver Mixer", "Right DAC1 Switch", "DACR1"}, + {"Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Receiver Mixer", "IN1 Switch", "IN1 Input"}, + {"Receiver Mixer", "IN2 Switch", "IN2 Input"}, + + /* Left Lineout output mixer */ + {"Left Lineout Mixer", "Left DAC1 Switch", "DACL1"}, + {"Left Lineout Mixer", "Right DAC1 Switch", "DACR1"}, + {"Left Lineout Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Lineout Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Lineout Mixer", "IN1 Switch", "IN1 Input"}, + {"Left Lineout Mixer", "IN2 Switch", "IN2 Input"}, + + /* Right lineout output mixer */ + {"Right Lineout Mixer", "Left DAC1 Switch", "DACL1"}, + {"Right Lineout Mixer", "Right DAC1 Switch", "DACR1"}, + {"Right Lineout Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Lineout Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Lineout Mixer", "IN1 Switch", "IN1 Input"}, + {"Right Lineout Mixer", "IN2 Switch", "IN2 Input"}, + + {"HP Left Out", NULL, "Left Headphone Mixer"}, + {"HP Right Out", NULL, "Right Headphone Mixer"}, + {"SPK Left Out", NULL, "Left Speaker Mixer"}, + {"SPK Right Out", NULL, "Right Speaker Mixer"}, + {"RCV Mono Out", NULL, "Receiver Mixer"}, + {"LINE Left Out", NULL, "Left Lineout Mixer"}, + {"LINE Right Out", NULL, "Right Lineout Mixer"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RCV", NULL, "RCV Mono Out"}, + {"OUT1", NULL, "LINE Left Out"}, + {"OUT2", NULL, "LINE Right Out"}, + {"OUT3", NULL, "LINE Left Out"}, + {"OUT4", NULL, "LINE Right Out"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left ADC Mixer", "IN1 Switch", "IN1 Input"}, + {"Left ADC Mixer", "IN2 Switch", "IN2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right ADC Mixer", "IN1 Switch", "IN1 Input"}, + {"Right ADC Mixer", "IN2 Switch", "IN2 Input"}, + + /* Inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + + {"IN1 Input", NULL, "INA1"}, + {"IN2 Input", NULL, "INA2"}, + + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, +}; + +static int max98095_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_add_controls(codec, max98095_snd_controls, + ARRAY_SIZE(max98095_snd_controls)); + + return 0; +} + +/* codec mclk clock divider coefficients */ +static const struct { + u32 rate; + u8 sr; +} rate_table[] = { + {8000, 0x01}, + {11025, 0x02}, + {16000, 0x03}, + {22050, 0x04}, + {24000, 0x05}, + {32000, 0x06}, + {44100, 0x07}, + {48000, 0x08}, + {88200, 0x09}, + {96000, 0x0A}, +}; + +static int rate_value(int rate, u8 *value) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + *value = rate_table[i].sr; + return 0; + } + } + *value = rate_table[0].sr; + return -EINVAL; +} + +static int max98095_dai1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_cdata *cdata; + unsigned long long ni; + unsigned int rate; + u8 regval; + + cdata = &max98095->dai[0]; + + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98095_02A_DAI1_FORMAT, + M98095_DAI_WS, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_update_bits(codec, M98095_02A_DAI1_FORMAT, + M98095_DAI_WS, M98095_DAI_WS); + break; + default: + return -EINVAL; + } + + if (rate_value(rate, ®val)) + return -EINVAL; + + snd_soc_update_bits(codec, M98095_027_DAI1_CLKMODE, + M98095_CLKMODE_MASK, regval); + cdata->rate = rate; + + /* Configure NI when operating as master */ + if (snd_soc_read(codec, M98095_02A_DAI1_FORMAT) & M98095_DAI_MAS) { + if (max98095->sysclk == 0) { + dev_err(codec->dev, "Invalid system clock frequency\n"); + return -EINVAL; + } + ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)rate; + do_div(ni, (unsigned long long int)max98095->sysclk); + snd_soc_write(codec, M98095_028_DAI1_CLKCFG_HI, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98095_029_DAI1_CLKCFG_LO, + ni & 0xFF); + } + + /* Update sample rate mode */ + if (rate < 50000) + snd_soc_update_bits(codec, M98095_02E_DAI1_FILTERS, + M98095_DAI_DHF, 0); + else + snd_soc_update_bits(codec, M98095_02E_DAI1_FILTERS, + M98095_DAI_DHF, M98095_DAI_DHF); + + return 0; +} + +static int max98095_dai2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_cdata *cdata; + unsigned long long ni; + unsigned int rate; + u8 regval; + + cdata = &max98095->dai[1]; + + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT, + M98095_DAI_WS, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT, + M98095_DAI_WS, M98095_DAI_WS); + break; + default: + return -EINVAL; + } + + if (rate_value(rate, ®val)) + return -EINVAL; + + snd_soc_update_bits(codec, M98095_031_DAI2_CLKMODE, + M98095_CLKMODE_MASK, regval); + cdata->rate = rate; + + /* Configure NI when operating as master */ + if (snd_soc_read(codec, M98095_034_DAI2_FORMAT) & M98095_DAI_MAS) { + if (max98095->sysclk == 0) { + dev_err(codec->dev, "Invalid system clock frequency\n"); + return -EINVAL; + } + ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)rate; + do_div(ni, (unsigned long long int)max98095->sysclk); + snd_soc_write(codec, M98095_032_DAI2_CLKCFG_HI, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98095_033_DAI2_CLKCFG_LO, + ni & 0xFF); + } + + /* Update sample rate mode */ + if (rate < 50000) + snd_soc_update_bits(codec, M98095_038_DAI2_FILTERS, + M98095_DAI_DHF, 0); + else + snd_soc_update_bits(codec, M98095_038_DAI2_FILTERS, + M98095_DAI_DHF, M98095_DAI_DHF); + + return 0; +} + +static int max98095_dai3_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_cdata *cdata; + unsigned long long ni; + unsigned int rate; + u8 regval; + + cdata = &max98095->dai[2]; + + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT, + M98095_DAI_WS, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT, + M98095_DAI_WS, M98095_DAI_WS); + break; + default: + return -EINVAL; + } + + if (rate_value(rate, ®val)) + return -EINVAL; + + snd_soc_update_bits(codec, M98095_03B_DAI3_CLKMODE, + M98095_CLKMODE_MASK, regval); + cdata->rate = rate; + + /* Configure NI when operating as master */ + if (snd_soc_read(codec, M98095_03E_DAI3_FORMAT) & M98095_DAI_MAS) { + if (max98095->sysclk == 0) { + dev_err(codec->dev, "Invalid system clock frequency\n"); + return -EINVAL; + } + ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)rate; + do_div(ni, (unsigned long long int)max98095->sysclk); + snd_soc_write(codec, M98095_03C_DAI3_CLKCFG_HI, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98095_03D_DAI3_CLKCFG_LO, + ni & 0xFF); + } + + /* Update sample rate mode */ + if (rate < 50000) + snd_soc_update_bits(codec, M98095_042_DAI3_FILTERS, + M98095_DAI_DHF, 0); + else + snd_soc_update_bits(codec, M98095_042_DAI3_FILTERS, + M98095_DAI_DHF, M98095_DAI_DHF); + + return 0; +} + +static int max98095_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + + /* Requested clock frequency is already setup */ + if (freq == max98095->sysclk) + return 0; + + max98095->sysclk = freq; /* remember current sysclk */ + + /* Setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 40MHz).. + * 0x03 (when master clk is 40MHz to 60MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98095_026_SYS_CLK, 0x10); + } else if ((freq >= 20000000) && (freq < 40000000)) { + snd_soc_write(codec, M98095_026_SYS_CLK, 0x20); + } else if ((freq >= 40000000) && (freq < 60000000)) { + snd_soc_write(codec, M98095_026_SYS_CLK, 0x30); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq); + + max98095->sysclk = freq; + return 0; +} + +static int max98095_dai1_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_cdata *cdata; + u8 regval = 0; + + cdata = &max98095->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Slave mode PLL */ + snd_soc_write(codec, M98095_028_DAI1_CLKCFG_HI, + 0x80); + snd_soc_write(codec, M98095_029_DAI1_CLKCFG_LO, + 0x00); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + regval |= M98095_DAI_MAS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "Clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98095_DAI_DLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98095_DAI_WCI; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98095_DAI_BCI; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98095_DAI_BCI|M98095_DAI_WCI; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98095_02A_DAI1_FORMAT, + M98095_DAI_MAS | M98095_DAI_DLY | M98095_DAI_BCI | + M98095_DAI_WCI, regval); + + snd_soc_write(codec, M98095_02B_DAI1_CLOCK, M98095_DAI_BSEL64); + } + + return 0; +} + +static int max98095_dai2_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_cdata *cdata; + u8 regval = 0; + + cdata = &max98095->dai[1]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Slave mode PLL */ + snd_soc_write(codec, M98095_032_DAI2_CLKCFG_HI, + 0x80); + snd_soc_write(codec, M98095_033_DAI2_CLKCFG_LO, + 0x00); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + regval |= M98095_DAI_MAS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "Clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98095_DAI_DLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98095_DAI_WCI; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98095_DAI_BCI; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98095_DAI_BCI|M98095_DAI_WCI; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT, + M98095_DAI_MAS | M98095_DAI_DLY | M98095_DAI_BCI | + M98095_DAI_WCI, regval); + + snd_soc_write(codec, M98095_035_DAI2_CLOCK, + M98095_DAI_BSEL64); + } + + return 0; +} + +static int max98095_dai3_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_cdata *cdata; + u8 regval = 0; + + cdata = &max98095->dai[2]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Slave mode PLL */ + snd_soc_write(codec, M98095_03C_DAI3_CLKCFG_HI, + 0x80); + snd_soc_write(codec, M98095_03D_DAI3_CLKCFG_LO, + 0x00); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + regval |= M98095_DAI_MAS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "Clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98095_DAI_DLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98095_DAI_WCI; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98095_DAI_BCI; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98095_DAI_BCI|M98095_DAI_WCI; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT, + M98095_DAI_MAS | M98095_DAI_DLY | M98095_DAI_BCI | + M98095_DAI_WCI, regval); + + snd_soc_write(codec, M98095_03F_DAI3_CLOCK, + M98095_DAI_BSEL64); + } + + return 0; +} + +static int max98095_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + + if (ret != 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + } + + snd_soc_update_bits(codec, M98095_090_PWR_EN_IN, + M98095_MBEN, M98095_MBEN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, M98095_090_PWR_EN_IN, + M98095_MBEN, 0); + codec->cache_sync = 1; + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define MAX98095_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98095_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98095_dai1_ops = { + .set_sysclk = max98095_dai_set_sysclk, + .set_fmt = max98095_dai1_set_fmt, + .hw_params = max98095_dai1_hw_params, +}; + +static struct snd_soc_dai_ops max98095_dai2_ops = { + .set_sysclk = max98095_dai_set_sysclk, + .set_fmt = max98095_dai2_set_fmt, + .hw_params = max98095_dai2_hw_params, +}; + +static struct snd_soc_dai_ops max98095_dai3_ops = { + .set_sysclk = max98095_dai_set_sysclk, + .set_fmt = max98095_dai3_set_fmt, + .hw_params = max98095_dai3_hw_params, +}; + +static struct snd_soc_dai_driver max98095_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98095_RATES, + .formats = MAX98095_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98095_RATES, + .formats = MAX98095_FORMATS, + }, + .ops = &max98095_dai1_ops, +}, +{ + .name = "Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = MAX98095_RATES, + .formats = MAX98095_FORMATS, + }, + .ops = &max98095_dai2_ops, +}, +{ + .name = "Voice", + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = MAX98095_RATES, + .formats = MAX98095_FORMATS, + }, + .ops = &max98095_dai3_ops, +} + +}; + +static void max98095_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_pdata *pdata = max98095->pdata; + u8 regval = 0; + + if (!pdata) { + dev_dbg(codec->dev, "No platform data\n"); + return; + } + + /* Configure mic for analog/digital mic mode */ + if (pdata->digmic_left_mode) + regval |= M98095_DIGMIC_L; + + if (pdata->digmic_right_mode) + regval |= M98095_DIGMIC_R; + + snd_soc_write(codec, M98095_087_CFG_MIC, regval); +} + +#ifdef CONFIG_PM +static int max98095_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int max98095_resume(struct snd_soc_codec *codec) +{ + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define max98095_suspend NULL +#define max98095_resume NULL +#endif + +static int max98095_reset(struct snd_soc_codec *codec) +{ + int i, ret; + + /* Gracefully reset the DSP core and the codec hardware + * in a proper sequence */ + ret = snd_soc_write(codec, M98095_00F_HOST_CFG, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to reset DSP: %d\n", ret); + return ret; + } + + ret = snd_soc_write(codec, M98095_097_PWR_SYS, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to reset codec: %d\n", ret); + return ret; + } + + /* Reset to hardware default for registers, as there is not + * a soft reset hardware control register */ + for (i = M98095_010_HOST_INT_CFG; i < M98095_REG_MAX_CACHED; i++) { + ret = snd_soc_write(codec, i, max98095_reg_def[i]); + if (ret < 0) { + dev_err(codec->dev, "Failed to reset: %d\n", ret); + return ret; + } + } + + return ret; +} + +static int max98095_probe(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_cdata *cdata; + int ret = 0; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* reset the codec, the DSP core, and disable all interrupts */ + max98095_reset(codec); + + /* initialize private data */ + + max98095->sysclk = (unsigned)-1; + + cdata = &max98095->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + cdata = &max98095->dai[1]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + cdata = &max98095->dai[2]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + max98095->lin_state = 0; + max98095->mic1pre = 0; + max98095->mic2pre = 0; + + ret = snd_soc_read(codec, M98095_0FF_REV_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + dev_info(codec->dev, "revision %c\n", ret + 'A'); + + snd_soc_write(codec, M98095_097_PWR_SYS, M98095_PWRSV); + + /* initialize registers cache to hardware default */ + max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + snd_soc_write(codec, M98095_048_MIX_DAC_LR, + M98095_DAI1L_TO_DACL|M98095_DAI1R_TO_DACR); + + snd_soc_write(codec, M98095_049_MIX_DAC_M, + M98095_DAI2M_TO_DACM|M98095_DAI3M_TO_DACM); + + snd_soc_write(codec, M98095_092_PWR_EN_OUT, M98095_SPK_SPREADSPECTRUM); + snd_soc_write(codec, M98095_045_CFG_DSP, M98095_DSPNORMAL); + snd_soc_write(codec, M98095_04E_CFG_HP, M98095_HPNORMAL); + + snd_soc_write(codec, M98095_02C_DAI1_IOCFG, + M98095_S1NORMAL|M98095_SDATA); + + snd_soc_write(codec, M98095_036_DAI2_IOCFG, + M98095_S2NORMAL|M98095_SDATA); + + snd_soc_write(codec, M98095_040_DAI3_IOCFG, + M98095_S3NORMAL|M98095_SDATA); + + max98095_handle_pdata(codec); + + /* take the codec out of the shut down */ + snd_soc_update_bits(codec, M98095_097_PWR_SYS, M98095_SHDNRUN, + M98095_SHDNRUN); + + max98095_add_widgets(codec); + +err_access: + return ret; +} + +static int max98095_remove(struct snd_soc_codec *codec) +{ + max98095_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98095 = { + .probe = max98095_probe, + .remove = max98095_remove, + .suspend = max98095_suspend, + .resume = max98095_resume, + .set_bias_level = max98095_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max98095_reg_def), + .reg_word_size = sizeof(u8), + .reg_cache_default = max98095_reg_def, + .readable_register = max98095_readable, + .volatile_register = max98095_volatile, + .dapm_widgets = max98095_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98095_dapm_widgets), + .dapm_routes = max98095_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98095_audio_map), +}; + +static int max98095_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98095_priv *max98095; + int ret; + + max98095 = kzalloc(sizeof(struct max98095_priv), GFP_KERNEL); + if (max98095 == NULL) + return -ENOMEM; + + max98095->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98095); + max98095->control_data = i2c; + max98095->pdata = i2c->dev.platform_data; + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98095, &max98095_dai[0], 3); + if (ret < 0) + kfree(max98095); + return ret; +} + +static int __devexit max98095_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + + return 0; +} + +static const struct i2c_device_id max98095_i2c_id[] = { + { "max98095", MAX98095 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98095_i2c_id); + +static struct i2c_driver max98095_i2c_driver = { + .driver = { + .name = "max98095", + .owner = THIS_MODULE, + }, + .probe = max98095_i2c_probe, + .remove = __devexit_p(max98095_i2c_remove), + .id_table = max98095_i2c_id, +}; + +static int __init max98095_init(void) +{ + int ret; + + ret = i2c_add_driver(&max98095_i2c_driver); + if (ret) + pr_err("Failed to register max98095 I2C driver: %d\n", ret); + + return ret; +} +module_init(max98095_init); + +static void __exit max98095_exit(void) +{ + i2c_del_driver(&max98095_i2c_driver); +} +module_exit(max98095_exit); + +MODULE_DESCRIPTION("ALSA SoC MAX98095 driver"); +MODULE_AUTHOR("Peter Hsiang"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h new file mode 100644 index 00000000000..5b22bc8dbed --- /dev/null +++ b/sound/soc/codecs/max98095.h @@ -0,0 +1,284 @@ +/* + * max98095.h -- MAX98095 ALSA SoC Audio driver + * + * Copyright 2011 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98095_H +#define _MAX98095_H + +/* + * MAX98095 Registers Definition + */ + +#define M98095_000_HOST_DATA 0x00 +#define M98095_001_HOST_INT_STS 0x01 +#define M98095_002_HOST_RSP_STS 0x02 +#define M98095_003_HOST_CMD_STS 0x03 +#define M98095_004_CODEC_STS 0x04 +#define M98095_005_DAI1_ALC_STS 0x05 +#define M98095_006_DAI2_ALC_STS 0x06 +#define M98095_007_JACK_AUTO_STS 0x07 +#define M98095_008_JACK_MANUAL_STS 0x08 +#define M98095_009_JACK_VBAT_STS 0x09 +#define M98095_00A_ACC_ADC_STS 0x0A +#define M98095_00B_MIC_NG_AGC_STS 0x0B +#define M98095_00C_SPK_L_VOLT_STS 0x0C +#define M98095_00D_SPK_R_VOLT_STS 0x0D +#define M98095_00E_TEMP_SENSOR_STS 0x0E +#define M98095_00F_HOST_CFG 0x0F +#define M98095_010_HOST_INT_CFG 0x10 +#define M98095_011_HOST_INT_EN 0x11 +#define M98095_012_CODEC_INT_EN 0x12 +#define M98095_013_JACK_INT_EN 0x13 +#define M98095_014_JACK_INT_EN 0x14 +#define M98095_015_DEC 0x15 +#define M98095_016_RESERVED 0x16 +#define M98095_017_RESERVED 0x17 +#define M98095_018_KEYCODE3 0x18 +#define M98095_019_KEYCODE2 0x19 +#define M98095_01A_KEYCODE1 0x1A +#define M98095_01B_KEYCODE0 0x1B +#define M98095_01C_OEMCODE1 0x1C +#define M98095_01D_OEMCODE0 0x1D +#define M98095_01E_XCFG1 0x1E +#define M98095_01F_XCFG2 0x1F +#define M98095_020_XCFG3 0x20 +#define M98095_021_XCFG4 0x21 +#define M98095_022_XCFG5 0x22 +#define M98095_023_XCFG6 0x23 +#define M98095_024_XGPIO 0x24 +#define M98095_025_XCLKCFG 0x25 +#define M98095_026_SYS_CLK 0x26 +#define M98095_027_DAI1_CLKMODE 0x27 +#define M98095_028_DAI1_CLKCFG_HI 0x28 +#define M98095_029_DAI1_CLKCFG_LO 0x29 +#define M98095_02A_DAI1_FORMAT 0x2A +#define M98095_02B_DAI1_CLOCK 0x2B +#define M98095_02C_DAI1_IOCFG 0x2C +#define M98095_02D_DAI1_TDM 0x2D +#define M98095_02E_DAI1_FILTERS 0x2E +#define M98095_02F_DAI1_LVL1 0x2F +#define M98095_030_DAI1_LVL2 0x30 +#define M98095_031_DAI2_CLKMODE 0x31 +#define M98095_032_DAI2_CLKCFG_HI 0x32 +#define M98095_033_DAI2_CLKCFG_LO 0x33 +#define M98095_034_DAI2_FORMAT 0x34 +#define M98095_035_DAI2_CLOCK 0x35 +#define M98095_036_DAI2_IOCFG 0x36 +#define M98095_037_DAI2_TDM 0x37 +#define M98095_038_DAI2_FILTERS 0x38 +#define M98095_039_DAI2_LVL1 0x39 +#define M98095_03A_DAI2_LVL2 0x3A +#define M98095_03B_DAI3_CLKMODE 0x3B +#define M98095_03C_DAI3_CLKCFG_HI 0x3C +#define M98095_03D_DAI3_CLKCFG_LO 0x3D +#define M98095_03E_DAI3_FORMAT 0x3E +#define M98095_03F_DAI3_CLOCK 0x3F +#define M98095_040_DAI3_IOCFG 0x40 +#define M98095_041_DAI3_TDM 0x41 +#define M98095_042_DAI3_FILTERS 0x42 +#define M98095_043_DAI3_LVL1 0x43 +#define M98095_044_DAI3_LVL2 0x44 +#define M98095_045_CFG_DSP 0x45 +#define M98095_046_DAC_CTRL1 0x46 +#define M98095_047_DAC_CTRL2 0x47 +#define M98095_048_MIX_DAC_LR 0x48 +#define M98095_049_MIX_DAC_M 0x49 +#define M98095_04A_MIX_ADC_LEFT 0x4A +#define M98095_04B_MIX_ADC_RIGHT 0x4B +#define M98095_04C_MIX_HP_LEFT 0x4C +#define M98095_04D_MIX_HP_RIGHT 0x4D +#define M98095_04E_CFG_HP 0x4E +#define M98095_04F_MIX_RCV 0x4F +#define M98095_050_MIX_SPK_LEFT 0x50 +#define M98095_051_MIX_SPK_RIGHT 0x51 +#define M98095_052_MIX_SPK_CFG 0x52 +#define M98095_053_MIX_LINEOUT1 0x53 +#define M98095_054_MIX_LINEOUT2 0x54 +#define M98095_055_MIX_LINEOUT_CFG 0x55 +#define M98095_056_LVL_SIDETONE_DAI12 0x56 +#define M98095_057_LVL_SIDETONE_DAI3 0x57 +#define M98095_058_LVL_DAI1_PLAY 0x58 +#define M98095_059_LVL_DAI1_EQ 0x59 +#define M98095_05A_LVL_DAI2_PLAY 0x5A +#define M98095_05B_LVL_DAI2_EQ 0x5B +#define M98095_05C_LVL_DAI3_PLAY 0x5C +#define M98095_05D_LVL_ADC_L 0x5D +#define M98095_05E_LVL_ADC_R 0x5E +#define M98095_05F_LVL_MIC1 0x5F +#define M98095_060_LVL_MIC2 0x60 +#define M98095_061_LVL_LINEIN 0x61 +#define M98095_062_LVL_LINEOUT1 0x62 +#define M98095_063_LVL_LINEOUT2 0x63 +#define M98095_064_LVL_HP_L 0x64 +#define M98095_065_LVL_HP_R 0x65 +#define M98095_066_LVL_RCV 0x66 +#define M98095_067_LVL_SPK_L 0x67 +#define M98095_068_LVL_SPK_R 0x68 +#define M98095_069_MICAGC_CFG 0x69 +#define M98095_06A_MICAGC_THRESH 0x6A +#define M98095_06B_SPK_NOISEGATE 0x6B +#define M98095_06C_DAI1_ALC1_TIME 0x6C +#define M98095_06D_DAI1_ALC1_COMP 0x6D +#define M98095_06E_DAI1_ALC1_EXPN 0x6E +#define M98095_06F_DAI1_ALC1_GAIN 0x6F +#define M98095_070_DAI1_ALC2_TIME 0x70 +#define M98095_071_DAI1_ALC2_COMP 0x71 +#define M98095_072_DAI1_ALC2_EXPN 0x72 +#define M98095_073_DAI1_ALC2_GAIN 0x73 +#define M98095_074_DAI1_ALC3_TIME 0x74 +#define M98095_075_DAI1_ALC3_COMP 0x75 +#define M98095_076_DAI1_ALC3_EXPN 0x76 +#define M98095_077_DAI1_ALC3_GAIN 0x77 +#define M98095_078_DAI2_ALC1_TIME 0x78 +#define M98095_079_DAI2_ALC1_COMP 0x79 +#define M98095_07A_DAI2_ALC1_EXPN 0x7A +#define M98095_07B_DAI2_ALC1_GAIN 0x7B +#define M98095_07C_DAI2_ALC2_TIME 0x7C +#define M98095_07D_DAI2_ALC2_COMP 0x7D +#define M98095_07E_DAI2_ALC2_EXPN 0x7E +#define M98095_07F_DAI2_ALC2_GAIN 0x7F +#define M98095_080_DAI2_ALC3_TIME 0x80 +#define M98095_081_DAI2_ALC3_COMP 0x81 +#define M98095_082_DAI2_ALC3_EXPN 0x82 +#define M98095_083_DAI2_ALC3_GAIN 0x83 +#define M98095_084_HP_NOISE_GATE 0x84 +#define M98095_085_AUX_ADC 0x85 +#define M98095_086_CFG_LINE 0x86 +#define M98095_087_CFG_MIC 0x87 +#define M98095_088_CFG_LEVEL 0x88 +#define M98095_089_JACK_DET_AUTO 0x89 +#define M98095_08A_JACK_DET_MANUAL 0x8A +#define M98095_08B_JACK_KEYSCAN_DBC 0x8B +#define M98095_08C_JACK_KEYSCAN_DLY 0x8C +#define M98095_08D_JACK_KEY_THRESH 0x8D +#define M98095_08E_JACK_DC_SLEW 0x8E +#define M98095_08F_JACK_TEST_CFG 0x8F +#define M98095_090_PWR_EN_IN 0x90 +#define M98095_091_PWR_EN_OUT 0x91 +#define M98095_092_PWR_EN_OUT 0x92 +#define M98095_093_BIAS_CTRL 0x93 +#define M98095_094_PWR_DAC_21 0x94 +#define M98095_095_PWR_DAC_03 0x95 +#define M98095_096_PWR_DAC_CK 0x96 +#define M98095_097_PWR_SYS 0x97 + +#define M98095_0FF_REV_ID 0xFF + +#define M98095_REG_CNT (0xFF+1) +#define M98095_REG_MAX_CACHED 0X97 + +/* MAX98095 Registers Bit Fields */ + +/* M98095_00F_HOST_CFG */ + #define M98095_SEG (1<<0) + #define M98095_XTEN (1<<1) + #define M98095_MDLLEN (1<<2) + +/* M98095_027_DAI1_CLKMODE, M98095_031_DAI2_CLKMODE, M98095_03B_DAI3_CLKMODE */ + #define M98095_CLKMODE_MASK 0xFF + +/* M98095_02A_DAI1_FORMAT, M98095_034_DAI2_FORMAT, M98095_03E_DAI3_FORMAT */ + #define M98095_DAI_MAS (1<<7) + #define M98095_DAI_WCI (1<<6) + #define M98095_DAI_BCI (1<<5) + #define M98095_DAI_DLY (1<<4) + #define M98095_DAI_TDM (1<<2) + #define M98095_DAI_FSW (1<<1) + #define M98095_DAI_WS (1<<0) + +/* M98095_02B_DAI1_CLOCK, M98095_035_DAI2_CLOCK, M98095_03F_DAI3_CLOCK */ + #define M98095_DAI_BSEL64 (1<<0) + #define M98095_DAI_DOSR_DIV2 (0<<5) + #define M98095_DAI_DOSR_DIV4 (1<<5) + +/* M98095_02C_DAI1_IOCFG, M98095_036_DAI2_IOCFG, M98095_040_DAI3_IOCFG */ + #define M98095_S1NORMAL (1<<6) + #define M98095_S2NORMAL (2<<6) + #define M98095_S3NORMAL (3<<6) + #define M98095_SDATA (3<<0) + +/* M98095_02E_DAI1_FILTERS, M98095_038_DAI2_FILTERS, M98095_042_DAI3_FILTERS */ + #define M98095_DAI_DHF (1<<3) + +/* M98095_045_DSP_CFG */ + #define M98095_DSPNORMAL (5<<4) + +/* M98095_048_MIX_DAC_LR */ + #define M98095_DAI1L_TO_DACR (1<<7) + #define M98095_DAI1R_TO_DACR (1<<6) + #define M98095_DAI2M_TO_DACR (1<<5) + #define M98095_DAI1L_TO_DACL (1<<3) + #define M98095_DAI1R_TO_DACL (1<<2) + #define M98095_DAI2M_TO_DACL (1<<1) + #define M98095_DAI3M_TO_DACL (1<<0) + +/* M98095_049_MIX_DAC_M */ + #define M98095_DAI1L_TO_DACM (1<<3) + #define M98095_DAI1R_TO_DACM (1<<2) + #define M98095_DAI2M_TO_DACM (1<<1) + #define M98095_DAI3M_TO_DACM (1<<0) + +/* M98095_04E_MIX_HP_CFG */ + #define M98095_HPNORMAL (3<<4) + +/* M98095_05F_LVL_MIC1, M98095_060_LVL_MIC2 */ + #define M98095_MICPRE_MASK (3<<5) + #define M98095_MICPRE_SHIFT 5 + +/* M98095_064_LVL_HP_L, M98095_065_LVL_HP_R */ + #define M98095_HP_MUTE (1<<7) + +/* M98095_066_LVL_RCV */ + #define M98095_REC_MUTE (1<<7) + +/* M98095_067_LVL_SPK_L, M98095_068_LVL_SPK_R */ + #define M98095_SP_MUTE (1<<7) + +/* M98095_087_CFG_MIC */ + #define M98095_MICSEL_MASK (3<<0) + #define M98095_DIGMIC_L (1<<2) + #define M98095_DIGMIC_R (1<<3) + #define M98095_DIGMIC2L (1<<4) + #define M98095_DIGMIC2R (1<<5) + +/* M98095_088_CFG_LEVEL */ + #define M98095_VSEN (1<<6) + #define M98095_ZDEN (1<<5) + #define M98095_EQ2EN (1<<1) + #define M98095_EQ1EN (1<<0) + +/* M98095_090_PWR_EN_IN */ + #define M98095_INEN (1<<7) + #define M98095_MB2EN (1<<3) + #define M98095_MB1EN (1<<2) + #define M98095_MBEN (3<<2) + #define M98095_ADREN (1<<1) + #define M98095_ADLEN (1<<0) + +/* M98095_091_PWR_EN_OUT */ + #define M98095_HPLEN (1<<7) + #define M98095_HPREN (1<<6) + #define M98095_SPLEN (1<<5) + #define M98095_SPREN (1<<4) + #define M98095_RECEN (1<<3) + #define M98095_DALEN (1<<1) + #define M98095_DAREN (1<<0) + +/* M98095_092_PWR_EN_OUT */ + #define M98095_SPK_FIXEDSPECTRUM (0<<4) + #define M98095_SPK_SPREADSPECTRUM (1<<4) + +/* M98095_097_PWR_SYS */ + #define M98095_SHDNRUN (1<<7) + #define M98095_PERFMODE (1<<3) + #define M98095_HPPLYBACK (1<<2) + #define M98095_PWRSV8K (1<<1) + #define M98095_PWRSV (1<<0) + +#endif -- cgit v1.2.3 From 0d1d7ce95156e0b040f1a4029613716aafd791b1 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 6 Apr 2011 10:20:26 +0800 Subject: ASoC: sst_platform: initialize module_name properly module_name will be checked in register_sst_card. It will fail to register sst card if it's not initialized. Signed-off-by: Lu Guanqun Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index ee2c22475a7..bd9d928dc76 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -247,6 +247,7 @@ static int sst_platform_open(struct snd_pcm_substream *substream) return -ENOMEM; } stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; + stream->sstdrv_ops->module_name = SST_CARD_NAMES; /* registering with SST driver to get access to SST APIs to use */ ret_val = register_sst_card(stream->sstdrv_ops); if (ret_val) { -- cgit v1.2.3 From 83a3fd3cf0bfdadfdfc633f6437f9121e28252b9 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 6 Apr 2011 10:20:32 +0800 Subject: ASoC: sst_platform: free the resources on fail path Signed-off-by: Lu Guanqun Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index bd9d928dc76..848ad3ce732 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -252,6 +252,8 @@ static int sst_platform_open(struct snd_pcm_substream *substream) ret_val = register_sst_card(stream->sstdrv_ops); if (ret_val) { pr_err("sst: sst card registration failed\n"); + kfree(stream->sstdrv_ops); + kfree(stream); return ret_val; } runtime->private_data = stream; -- cgit v1.2.3 From fb631eae1f2171033327e1b9ab427d4a113dc179 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 6 Apr 2011 10:20:37 +0800 Subject: ASoC: sst_platform: unregister sst card when being closed Signed-off-by: Lu Guanqun Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 848ad3ce732..9ba941447d4 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -271,6 +271,7 @@ static int sst_platform_close(struct snd_pcm_substream *substream) str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->sstdrv_ops->pcm_control->close(str_id); + unregister_sst_card(stream->sstdrv_ops); kfree(stream->sstdrv_ops); kfree(stream); return ret_val; -- cgit v1.2.3 From a7f2371f9e9730ccdb70d6d5803da2a732c97cf3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 10:24:23 +0200 Subject: ALSA: hda - Split EAPD init to a separate array from alc662_init_verbs So far, alc662_init_verbs[] is used for all ALC662-compatible chips, but the EAPD controls for 0x15 in there is invalid for ALC892. Also, since EAPDs should be set up in alc_auto_init_amp(), these static elements aren't needed for auto-parser, too. In this patch, the EAPD init verbs are split from alc662_init_verbs, and applied only to static quirks. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 55 +++++++++++++++++++++++++++++++++---------- 1 file changed, 43 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d566eac08d6..4971d777d9c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1268,6 +1268,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: + case 0x10ec0892: set_eapd(codec, 0x14, 1); set_eapd(codec, 0x15, 1); break; @@ -4244,6 +4245,7 @@ static void alc_power_eapd(struct hda_codec *codec) case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: + case 0x10ec0892: set_eapd(codec, 0x14, 0); set_eapd(codec, 0x15, 0); break; @@ -17922,10 +17924,13 @@ static struct hda_verb alc662_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + +static struct hda_verb alc662_eapd_init_verbs[] = { /* always trun on EAPD */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } }; @@ -18797,7 +18802,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { static struct alc_config_preset alc662_presets[] = { [ALC662_3ST_2ch_DIG] = { .mixers = { alc662_3ST_2ch_mixer }, - .init_verbs = { alc662_init_verbs }, + .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, @@ -18808,7 +18813,7 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC662_3ST_6ch_DIG] = { .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, - .init_verbs = { alc662_init_verbs }, + .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, @@ -18820,7 +18825,7 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC662_3ST_6ch] = { .mixers = { alc662_3ST_6ch_mixer, alc662_chmode_mixer }, - .init_verbs = { alc662_init_verbs }, + .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), @@ -18830,7 +18835,7 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC662_5ST_DIG] = { .mixers = { alc662_base_mixer, alc662_chmode_mixer }, - .init_verbs = { alc662_init_verbs }, + .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, @@ -18841,7 +18846,9 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC662_LENOVO_101E] = { .mixers = { alc662_lenovo_101e_mixer }, - .init_verbs = { alc662_init_verbs, alc662_sue_init_verbs }, + .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, + alc662_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), @@ -18853,6 +18860,7 @@ static struct alc_config_preset alc662_presets[] = { [ALC662_ASUS_EEEPC_P701] = { .mixers = { alc662_eeepc_p701_mixer }, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc662_eeepc_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -18866,6 +18874,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc662_eeepc_ep20_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc662_eeepc_ep20_sue_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -18879,6 +18888,7 @@ static struct alc_config_preset alc662_presets[] = { [ALC662_ECS] = { .mixers = { alc662_ecs_mixer }, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc662_ecs_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -18890,7 +18900,9 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC663_ASUS_M51VA] = { .mixers = { alc663_m51va_mixer }, - .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, + alc663_m51va_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, @@ -18902,7 +18914,9 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC663_ASUS_G71V] = { .mixers = { alc663_g71v_mixer }, - .init_verbs = { alc662_init_verbs, alc663_g71v_init_verbs }, + .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, + alc663_g71v_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, @@ -18914,7 +18928,9 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC663_ASUS_H13] = { .mixers = { alc663_m51va_mixer }, - .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, + .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, + alc663_m51va_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), @@ -18924,7 +18940,9 @@ static struct alc_config_preset alc662_presets[] = { }, [ALC663_ASUS_G50V] = { .mixers = { alc663_g50v_mixer }, - .init_verbs = { alc662_init_verbs, alc663_g50v_init_verbs }, + .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, + alc663_g50v_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, .dig_out_nid = ALC662_DIGOUT_NID, @@ -18939,6 +18957,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc663_m51va_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_21jd_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .hp_nid = 0x03, @@ -18954,6 +18973,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc662_1bjd_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc662_1bjd_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, @@ -18968,6 +18988,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc663_two_hp_m1_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_two_hp_amic_m1_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .hp_nid = 0x03, @@ -18983,6 +19004,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc663_asus_21jd_clfe_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_21jd_amic_init_verbs}, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .hp_nid = 0x03, @@ -18998,6 +19020,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc663_asus_15jd_clfe_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_15jd_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .hp_nid = 0x03, @@ -19013,6 +19036,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc663_two_hp_m2_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_two_hp_amic_m2_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .hp_nid = 0x03, @@ -19028,6 +19052,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc663_mode7_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_mode7_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .hp_nid = 0x03, @@ -19043,6 +19068,7 @@ static struct alc_config_preset alc662_presets[] = { .mixers = { alc663_mode8_mixer }, .cap_mixer = alc662_auto_capture_mixer, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_mode8_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .hp_nid = 0x03, @@ -19057,7 +19083,9 @@ static struct alc_config_preset alc662_presets[] = { [ALC272_DELL] = { .mixers = { alc663_m51va_mixer }, .cap_mixer = alc272_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, alc272_dell_init_verbs }, + .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, + alc272_dell_init_verbs }, .num_dacs = ARRAY_SIZE(alc272_dac_nids), .dac_nids = alc272_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), @@ -19072,7 +19100,9 @@ static struct alc_config_preset alc662_presets[] = { [ALC272_DELL_ZM1] = { .mixers = { alc663_m51va_mixer }, .cap_mixer = alc662_auto_capture_mixer, - .init_verbs = { alc662_init_verbs, alc272_dell_zm1_init_verbs }, + .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, + alc272_dell_zm1_init_verbs }, .num_dacs = ARRAY_SIZE(alc272_dac_nids), .dac_nids = alc272_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), @@ -19087,6 +19117,7 @@ static struct alc_config_preset alc662_presets[] = { [ALC272_SAMSUNG_NC10] = { .mixers = { alc272_nc10_mixer }, .init_verbs = { alc662_init_verbs, + alc662_eapd_init_verbs, alc663_21jd_amic_init_verbs }, .num_dacs = ARRAY_SIZE(alc272_dac_nids), .dac_nids = alc272_dac_nids, -- cgit v1.2.3 From 691f1fccf72e3092e3a7a79fd672940afc9305ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 10:31:43 +0200 Subject: ALSA: hda - Refactoring EAPD controls Reduced the duplicated codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 75 +++++++++++++++++-------------------------- 1 file changed, 30 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4971d777d9c..c798e18d453 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1236,6 +1236,34 @@ static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on) on ? 2 : 0); } +/* turn on/off EAPD controls of the codec */ +static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) +{ + /* We currently only handle front, HP */ + switch (codec->vendor_id) { + case 0x10ec0260: + set_eapd(codec, 0x0f, on); + set_eapd(codec, 0x10, on); + break; + case 0x10ec0262: + case 0x10ec0267: + case 0x10ec0268: + case 0x10ec0269: + case 0x10ec0270: + case 0x10ec0272: + case 0x10ec0660: + case 0x10ec0662: + case 0x10ec0663: + case 0x10ec0665: + case 0x10ec0862: + case 0x10ec0889: + case 0x10ec0892: + set_eapd(codec, 0x14, on); + set_eapd(codec, 0x15, on); + break; + } +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -1251,28 +1279,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) snd_hda_sequence_write(codec, alc_gpio3_init_verbs); break; case ALC_INIT_DEFAULT: - switch (codec->vendor_id) { - case 0x10ec0260: - set_eapd(codec, 0x0f, 1); - set_eapd(codec, 0x10, 1); - break; - case 0x10ec0262: - case 0x10ec0267: - case 0x10ec0268: - case 0x10ec0269: - case 0x10ec0270: - case 0x10ec0272: - case 0x10ec0660: - case 0x10ec0662: - case 0x10ec0663: - case 0x10ec0665: - case 0x10ec0862: - case 0x10ec0889: - case 0x10ec0892: - set_eapd(codec, 0x14, 1); - set_eapd(codec, 0x15, 1); - break; - } + alc_auto_setup_eapd(codec, true); switch (codec->vendor_id) { case 0x10ec0260: snd_hda_codec_write(codec, 0x1a, 0, @@ -4227,29 +4234,7 @@ static void alc_free(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE static void alc_power_eapd(struct hda_codec *codec) { - /* We currently only handle front, HP */ - switch (codec->vendor_id) { - case 0x10ec0260: - set_eapd(codec, 0x0f, 0); - set_eapd(codec, 0x10, 0); - break; - case 0x10ec0262: - case 0x10ec0267: - case 0x10ec0268: - case 0x10ec0269: - case 0x10ec0270: - case 0x10ec0272: - case 0x10ec0660: - case 0x10ec0662: - case 0x10ec0663: - case 0x10ec0665: - case 0x10ec0862: - case 0x10ec0889: - case 0x10ec0892: - set_eapd(codec, 0x14, 0); - set_eapd(codec, 0x15, 0); - break; - } + alc_auto_setup_eapd(codec, false); } static int alc_suspend(struct hda_codec *codec, pm_message_t state) -- cgit v1.2.3 From 1c716153a87ae75c8bd63f1b6f7916300d87df7e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 10:37:16 +0200 Subject: ALSA: hda - Introduce shutup callback to Realtek spec struct Add shutup callback to be called codec-specifically for avoiding pop noises at suspend or shutdown. As a generic callback, just turn EAPD off. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c798e18d453..3761fba5bce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -375,6 +375,7 @@ struct alc_spec { #ifdef CONFIG_SND_HDA_POWER_SAVE void (*power_hook)(struct hda_codec *codec); #endif + void (*shutup)(struct hda_codec *codec); /* for pin sensing */ unsigned int sense_updated: 1; @@ -1264,6 +1265,15 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) } } +/* generic shutup callback; + * just turning off EPAD and a little pause for avoiding pop-noise + */ +static void alc_eapd_shutup(struct hda_codec *codec) +{ + alc_auto_setup_eapd(codec, false); + msleep(200); +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -4201,6 +4211,10 @@ static int alc_build_pcms(struct hda_codec *codec) static inline void alc_shutup(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + + if (spec && spec->shutup) + spec->shutup(codec); snd_hda_shutup_pins(codec); } @@ -4250,6 +4264,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { + msleep(150); /* to avoid pop noise */ codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -7370,6 +7385,7 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC260_AUTO) spec->init_hook = alc260_auto_init; + spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc260_loopbacks; @@ -13005,6 +13021,7 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) spec->init_hook = alc262_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -14079,6 +14096,7 @@ static int patch_alc268(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); @@ -17397,6 +17415,7 @@ static int patch_alc861vd(struct hda_codec *codec) if (board_config == ALC861VD_AUTO) spec->init_hook = alc861vd_auto_init; + spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc861vd_loopbacks; @@ -19628,6 +19647,7 @@ static int patch_alc662(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC662_AUTO) spec->init_hook = alc662_auto_init; + spec->shutup = alc_eapd_shutup; alc_init_jacks(codec); -- cgit v1.2.3 From 5402e4cb80dc2cb407ca07e31cb7668ba45e5320 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 10:39:25 +0200 Subject: ALSA: hda - Rewrite alc269_suspend to alc269_shutup alc269_suspend is just calling the shut-up, so we can use the new shutup callback for the purpose. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3761fba5bce..8406248236f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14808,7 +14808,6 @@ static void alc269_auto_init(struct hda_codec *codec) alc_inithook(codec); } -#ifdef SND_HDA_NEEDS_RESUME static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); @@ -14819,8 +14818,7 @@ static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) alc_write_coef_idx(codec, 0x04, val); } -#ifdef CONFIG_SND_HDA_POWER_SAVE -static int alc269_suspend(struct hda_codec *codec, pm_message_t state) +static void alc269_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -14830,14 +14828,9 @@ static int alc269_suspend(struct hda_codec *codec, pm_message_t state) alc269_toggle_power_output(codec, 0); msleep(150); } - - alc_shutup(codec); - if (spec && spec->power_hook) - spec->power_hook(codec); - return 0; } -#endif /* CONFIG_SND_HDA_POWER_SAVE */ +#ifdef SND_HDA_NEEDS_RESUME static int alc269_resume(struct hda_codec *codec) { if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { @@ -15302,14 +15295,12 @@ static int patch_alc269(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; -#ifdef CONFIG_SND_HDA_POWER_SAVE - codec->patch_ops.suspend = alc269_suspend; -#endif #ifdef SND_HDA_NEEDS_RESUME codec->patch_ops.resume = alc269_resume; #endif if (board_config == ALC269_AUTO) spec->init_hook = alc269_auto_init; + spec->shutup = alc269_shutup; alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE -- cgit v1.2.3 From 0e53f3440925aa36fe3bd2307e5fa0238a66f8bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 12:27:32 +0200 Subject: ALSA: hda - Unmute mixer dynamically in alc662 auto-parser Instead of static init array, better to determine the connection and the mute status of the pin/mixer/DAC route dynamically. This fixes the uninitialized mixer 0x0f on ALC892. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8406248236f..e3756a7ebee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19319,14 +19319,21 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t srcs[HDA_MAX_CONNECTIONS]; alc_set_pin_output(codec, nid, pin_type); - /* need the manual connection? */ num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs)); - if (num <= 1) - return; for (i = 0; i < num; i++) { if (alc662_mix_to_dac(codec, srcs[i]) != dac) continue; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, i); + /* need the manual connection? */ + if (num > 1) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, i); + /* unmute mixer widget inputs */ + snd_hda_codec_write(codec, srcs[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + snd_hda_codec_write(codec, srcs[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); return; } } -- cgit v1.2.3 From 10696aa0e56b29c6f3ce27081092785c564423e1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 12:46:45 +0200 Subject: ALSA: hda - Mute ADC as default in ALC882 and other auto-parsers Mute the ADC as default in the auto-parser dynamically instead of relying on the static init verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e3756a7ebee..e5dfed35ca6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10852,6 +10852,11 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) const struct hda_input_mux *imux; int conns, mute, idx, item; + /* mute ADC */ + snd_hda_codec_write(codec, spec->adc_nids[c], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + conns = snd_hda_get_connections(codec, nid, conn_list, ARRAY_SIZE(conn_list)); if (conns < 0) -- cgit v1.2.3 From 35ffe11587fb6e93725d420aa17cf1e7a6bf427f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 12:34:26 +0200 Subject: ALSA: hda - Remove superfluous inits for ALC662 auto-parser Since we now set up the connections and mutes dynamically in the auto-parser, all static initializations via alc662_init_verbs & co are no longer needed. Let's drop them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 30 ------------------------------ 1 file changed, 30 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5dfed35ca6..c22a7ca0199 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17934,28 +17934,6 @@ static struct hda_verb alc662_eapd_init_verbs[] = { { } }; -static struct hda_verb alc663_init_verbs[] = { - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } -}; - -static struct hda_verb alc272_init_verbs[] = { - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } -}; - static struct hda_verb alc662_sue_init_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, @@ -19441,14 +19419,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - add_verb(spec, alc662_init_verbs); - if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) - add_verb(spec, alc663_init_verbs); - - if (codec->vendor_id == 0x10ec0272) - add_verb(spec, alc272_init_verbs); - err = alc_auto_add_mic_boost(codec); if (err < 0) return err; -- cgit v1.2.3 From ad93ffe6e4fc02993987008e4a5dcdcf4c926cd5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 12:49:26 +0200 Subject: ALSA: hda - Fix unused variable warning in patch_realtek.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c22a7ca0199..b1e5eb17f1b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14825,8 +14825,6 @@ static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) static void alc269_shutup(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) alc269_toggle_power_output(codec, 0); if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { -- cgit v1.2.3 From a12d3e1e1cb67bab2411966b345151c316633847 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Apr 2011 15:55:15 +0200 Subject: ALSA: hda - Remember connection lists The connection lists are static and we can reuse the previous results instead of querying via verb at each time. This will reduce the I/O in the runtime especially for some codec auto-parsers. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 69 +++++++++++++++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_codec.h | 2 ++ 2 files changed, 69 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c79e96d032..11ead159abd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -307,6 +307,12 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes); +static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t *conn_list, int max_conns); +static bool add_conn_list(struct snd_array *array, hda_nid_t nid); +static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst, + hda_nid_t *src, int len); + /** * snd_hda_get_connections - get connection list * @codec: the HDA codec @@ -320,7 +326,44 @@ EXPORT_SYMBOL_HDA(snd_hda_get_sub_nodes); * Returns the number of connections, or a negative error code. */ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *conn_list, int max_conns) + hda_nid_t *conn_list, int max_conns) +{ + struct snd_array *array = &codec->conn_lists; + int i, j, len, old_used; + hda_nid_t list[HDA_MAX_CONNECTIONS]; + + /* look up the cached results */ + for (i = 0; i < array->used; ) { + hda_nid_t *p = snd_array_elem(array, i); + len = p[1]; + if (nid == *p) + return copy_conn_list(nid, conn_list, max_conns, + p + 2, len); + i += len + 2; + } + + len = _hda_get_connections(codec, nid, list, HDA_MAX_CONNECTIONS); + if (len < 0) + return len; + + /* add to the cache */ + old_used = array->used; + if (!add_conn_list(array, nid) || !add_conn_list(array, len)) + goto error_add; + for (i = 0; i < len; i++) + if (!add_conn_list(array, list[i])) + goto error_add; + + return copy_conn_list(nid, conn_list, max_conns, list, len); + + error_add: + array->used = old_used; + return -ENOMEM; +} +EXPORT_SYMBOL_HDA(snd_hda_get_connections); + +static int _hda_get_connections(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t *conn_list, int max_conns) { unsigned int parm; int i, conn_len, conns; @@ -417,8 +460,28 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, } return conns; } -EXPORT_SYMBOL_HDA(snd_hda_get_connections); +static bool add_conn_list(struct snd_array *array, hda_nid_t nid) +{ + hda_nid_t *p = snd_array_new(array); + if (!p) + return false; + *p = nid; + return true; +} + +static int copy_conn_list(hda_nid_t nid, hda_nid_t *dst, int max_dst, + hda_nid_t *src, int len) +{ + if (len > max_dst) { + snd_printk(KERN_ERR "hda_codec: " + "Too many connections %d for NID 0x%x\n", + len, nid); + return -EINVAL; + } + memcpy(dst, src, len * sizeof(hda_nid_t)); + return len; +} /** * snd_hda_queue_unsol_event - add an unsolicited event to queue @@ -1017,6 +1080,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) list_del(&codec->list); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); + snd_array_free(&codec->conn_lists); codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); @@ -1077,6 +1141,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->conn_lists, sizeof(hda_nid_t), 64); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e46d5420a9f..7d57a66c1b3 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -825,6 +825,8 @@ struct hda_codec { struct hda_cache_rec amp_cache; /* cache for amp access */ struct hda_cache_rec cmd_cache; /* cache for other commands */ + struct snd_array conn_lists; /* connection-list array */ + struct mutex spdif_mutex; struct mutex control_mutex; unsigned int spdif_status; /* IEC958 status bits */ -- cgit v1.2.3 From b7af1dafdfaf8419065399d07fb7cbae14b286ef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 7 Apr 2011 19:18:44 +0900 Subject: ASoC: Add data based control initialisation for CODECs and cards Allow CODEC and card drivers to point to an array of controls from their driver structure rather than explicitly calling snd_soc_add_controls(). Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f31afe9d4ed..f75f1392604 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1493,6 +1493,9 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + if (driver->controls) + snd_soc_add_controls(codec, driver->controls, + driver->num_controls); if (driver->dapm_widgets) snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); @@ -1890,6 +1893,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* We should have a non-codec control add function but we don't */ + if (card->controls) + snd_soc_add_controls(list_first_entry(&card->codec_dev_list, + struct snd_soc_codec, + card_list), + card->controls, + card->num_controls); + if (card->dapm_widgets) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); -- cgit v1.2.3 From b39e285545a2bd7f331a53b32c8c40748fdd348e Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Thu, 7 Apr 2011 02:05:11 -0400 Subject: ASoC: SSM2602: add SPI support The ssm2602 codec has a SPI interface as well as I2C, so add the simple bit of glue to make it usable. Signed-off-by: Mike Frysinger Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/ssm2602.c | 56 ++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 53 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 78da05b5e5e..ee7374e8e97 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF - select SND_SOC_SSM2602 if I2C + select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 8a2b52fa4a7..7e219497536 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -32,6 +32,7 @@ #include #include #include +#include #include #include #include @@ -556,6 +557,43 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .reg_cache_default = ssm2602_reg, }; +#if defined(CONFIG_SPI_MASTER) +static int __devinit ssm2602_spi_probe(struct spi_device *spi) +{ + struct ssm2602_priv *ssm2602; + int ret; + + ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + if (ssm2602 == NULL) + return -ENOMEM; + + spi_set_drvdata(spi, ssm2602); + ssm2602->control_type = SND_SOC_SPI; + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_ssm2602, &ssm2602_dai, 1); + if (ret < 0) + kfree(ssm2602); + return ret; +} + +static int __devexit ssm2602_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver ssm2602_spi_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_spi_probe, + .remove = __devexit_p(ssm2602_spi_remove), +}; +#endif + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * ssm2602 2 wire address is determined by GPIO5 @@ -612,19 +650,29 @@ static struct i2c_driver ssm2602_i2c_driver = { static int __init ssm2602_modinit(void) { int ret = 0; + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&ssm2602_spi_driver); + if (ret) + return ret; +#endif + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ssm2602_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register SSM2602 I2C driver: %d\n", - ret); - } + if (ret) + return ret; #endif + return ret; } module_init(ssm2602_modinit); static void __exit ssm2602_exit(void) { +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&ssm2602_spi_driver); +#endif + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&ssm2602_i2c_driver); #endif -- cgit v1.2.3 From d25b7c1ec7da4636587ad1a22b324bcd7b89b6bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 2 Apr 2011 16:39:39 +0900 Subject: ASoC: Remove special casing for registerless widgets Since we recently explicitly set the register for registerless widgets to no register there is no longer any need to special case power updates for them, we can allow them to be handled with the register compression code as other widgets are. As this is the only remaining user of dapm_generic_apply_power() and dapm_update_bits() also remove those functions. Noticed-by: Lu Guanqun Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 100 --------------------------------------------------- 1 file changed, 100 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 05da8a8f0ae..567645c0308 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -322,45 +322,6 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, return -ENODEV; } -/* update dapm codec register bits */ -static int dapm_update_bits(struct snd_soc_dapm_widget *widget) -{ - int change, power; - unsigned int old, new; - struct snd_soc_codec *codec = widget->codec; - struct snd_soc_dapm_context *dapm = widget->dapm; - struct snd_soc_card *card = dapm->card; - - /* check for valid widgets */ - if (widget->reg < 0 || widget->id == snd_soc_dapm_input || - widget->id == snd_soc_dapm_output || - widget->id == snd_soc_dapm_hp || - widget->id == snd_soc_dapm_mic || - widget->id == snd_soc_dapm_line || - widget->id == snd_soc_dapm_spk) - return 0; - - power = widget->power; - if (widget->invert) - power = (power ? 0:1); - - old = snd_soc_read(codec, widget->reg); - new = (old & ~(0x1 << widget->shift)) | (power << widget->shift); - - change = old != new; - if (change) { - pop_dbg(dapm->dev, card->pop_time, - "pop test %s : %s in %d ms\n", - widget->name, widget->power ? "on" : "off", - card->pop_time); - pop_wait(card->pop_time); - snd_soc_write(codec, widget->reg, new); - } - dev_dbg(dapm->dev, "reg %x old %x new %x change %d\n", widget->reg, - old, new, change); - return change; -} - /* create new dapm mixer control */ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) @@ -644,57 +605,6 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); -/* Standard power change method, used to apply power changes to most - * widgets. - */ -static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) -{ - int ret; - - /* call any power change event handlers */ - if (w->event) - dev_dbg(w->dapm->dev, "power %s event for %s flags %x\n", - w->power ? "on" : "off", - w->name, w->event_flags); - - /* power up pre event */ - if (w->power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - - /* power down pre event */ - if (!w->power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - - dapm_update_bits(w); - - /* power up post event */ - if (w->power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMU)) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - - /* power down post event */ - if (!w->power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - - return 0; -} - /* Generic check to see if a widget should be powered. */ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) @@ -981,16 +891,6 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, NULL, SND_SOC_DAPM_POST_PMD); break; - case snd_soc_dapm_input: - case snd_soc_dapm_output: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_line: - case snd_soc_dapm_spk: - /* No register support currently */ - ret = dapm_generic_apply_power(w); - break; - default: /* Queue it up for application */ cur_sort = sort[w->id]; -- cgit v1.2.3 From 52ba67bf85889828b3766207fa43ce7159c84c78 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Apr 2011 21:05:11 +0900 Subject: ASoC: Force all DAPM contexts into the same bias state Currently we allow all DAPM contexts to determine their own bias level. While this should in general work in most situations and will deliver the lowest possible power it causes problems for our integration with the card bias level as we're calling the card bias level functions for each DAPM context even though they're card wide but don't say which CODEC we're calling them for. Mitigate against this by forcing everything to be in the same state. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 567645c0308..68879209b31 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1101,6 +1101,15 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } + /* Force all contexts in the card to the same bias state */ + power = 0; + list_for_each_entry(d, &card->dapm_list, list) + if (d->dev_power) + power = 1; + list_for_each_entry(d, &card->dapm_list, list) + d->dev_power = power; + + /* Run all the bias changes in parallel */ list_for_each_entry(d, &dapm->card->dapm_list, list) async_schedule_domain(dapm_pre_sequence_async, d, -- cgit v1.2.3 From 0d86733cce776ca0262b850ee8eb46bc52dc8244 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Apr 2011 11:38:14 +0900 Subject: ASoC: Allow DAPM pin operations to match any context The DAPM pin operations currently require that the specific DAPM context that the pin being operated in is contained in be specified. With multi component and especially with the addition of a per-card DAPM context this isn't ideal as it means that things like disabling unused pins on CODECs require looking up the CODEC DAPM context. Fix this by falling back to matching a widget in any context if there isn't a match in the current context. The code isn't ideal currently but will do the job. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 68879209b31..2ee738c08ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1477,6 +1477,19 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, } } + /* Try again in other contexts */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strcmp(w->name, pin)) { + dev_dbg(w->dapm->dev, "dapm: pin %s = %d\n", + pin, status); + w->connected = status; + /* Allow disabling of forced pins */ + if (status == 0) + w->force = 0; + return 0; + } + } + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } @@ -2317,6 +2330,17 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, } } + /* Try again with other contexts */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strcmp(w->name, pin)) { + dev_dbg(w->dapm->dev, + "dapm: force enable pin %s\n", pin); + w->connected = 1; + w->force = 1; + return 0; + } + } + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } -- cgit v1.2.3 From b8eeee68dc81f08993ed5dc18dc6d574ba146674 Mon Sep 17 00:00:00 2001 From: Sangbeom Kim Date: Sat, 9 Apr 2011 10:57:59 +0900 Subject: ASoC: SAMSUNG: Add WM8580 PCM Machine driver This patch add WM8580 PCM machine driver to support PCM audio on SMDKC110, SMDKV210, SMDK6450, SMDK6440 boards. Playback and Capture supports 8kHz sampling rates. and It is tested on SMDKC110, SMDKV210, SMDK6450 Signed-off-by: Sangbeom Kim Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 8 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/smdk_wm8580pcm.c | 206 +++++++++++++++++++++++++++++++++++++ 3 files changed, 216 insertions(+) create mode 100644 sound/soc/samsung/smdk_wm8580pcm.c (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a3fdfb63146..b8c7a2e0d21 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -162,3 +162,11 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF select SND_SAMSUNG_SPDIF help Say Y if you want to add support for SoC S/PDIF audio on the SMDK. + +config SND_SOC_SMDK_WM8580_PCM + tristate "SoC PCM Audio support for WM8580 on SMDK" + depends on SND_SOC_SAMSUNG && (MACH_SMDK6450 || MACH_SMDKV210 || MACH_SMDKC110) + select SND_SOC_WM8580 + select SND_SAMSUNG_PCM + help + Say Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 294dec05c26..6c598fc51a0 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -34,6 +34,7 @@ snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o +snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -51,3 +52,4 @@ obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o +obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c new file mode 100644 index 00000000000..0d12092df16 --- /dev/null +++ b/sound/soc/samsung/smdk_wm8580pcm.c @@ -0,0 +1,206 @@ +/* + * sound/soc/samsung/smdk_wm8580pcm.c + * + * Copyright (c) 2011 Samsung Electronics Co. Ltd + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include +#include +#include + +#include + +#include "../codecs/wm8580.h" +#include "dma.h" +#include "pcm.h" + +/* + * Board Settings: + * o '1' means 'ON' + * o '0' means 'OFF' + * o 'X' means 'Don't care' + * + * SMDK6410, SMDK6440, SMDK6450 Base B/D: CFG1-0000, CFG2-1111 + * SMDKC110, SMDKV210: CFGB11-100100, CFGB12-0000 + */ + +#define SMDK_WM8580_EXT_OSC 12000000 +#define SMDK_WM8580_EXT_MCLK 4096000 +#define SMDK_WM8580_EXT_VOICE 2048000 + +static unsigned long mclk_freq; +static unsigned long xtal_freq; + +/* + * If MCLK clock directly gets from XTAL, we don't have to use PLL + * to make MCLK, but if XTAL clock source connects with other codec + * pin (like XTI), we should have to set codec's PLL to make MCLK. + * Because Samsung SoC does not support pcmcdclk output like I2S. + */ + +static int smdk_wm8580_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int rfs, ret; + + switch (params_rate(params)) { + case 8000: + break; + default: + printk(KERN_ERR "%s:%d Sampling Rate %u not supported!\n", + __func__, __LINE__, params_rate(params)); + return -EINVAL; + } + + rfs = mclk_freq / params_rate(params) / 2; + + /* Set the codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* Set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + if (mclk_freq == xtal_freq) { + ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_MCLK, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + } else { + ret = snd_soc_dai_set_sysclk(codec_dai, WM8580_CLKSRC_PLLA, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, + xtal_freq, mclk_freq); + if (ret < 0) + return ret; + } + + /* Set PCM source clock on CPU */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set SCLK_DIV for making bclk */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops smdk_wm8580_pcm_ops = { + .hw_params = smdk_wm8580_pcm_hw_params, +}; + +static struct snd_soc_dai_link smdk_dai[] = { + { + .name = "WM8580 PAIF PCM RX", + .stream_name = "Playback", + .cpu_dai_name = "samsung-pcm.0", + .codec_dai_name = "wm8580-hifi-playback", + .platform_name = "samsung-audio", + .codec_name = "wm8580-codec.0-001b", + .ops = &smdk_wm8580_pcm_ops, + }, { + .name = "WM8580 PAIF PCM TX", + .stream_name = "Capture", + .cpu_dai_name = "samsung-pcm.0", + .codec_dai_name = "wm8580-hifi-capture", + .platform_name = "samsung-audio", + .codec_name = "wm8580-codec.0-001b", + .ops = &smdk_wm8580_pcm_ops, + }, +}; + +static struct snd_soc_card smdk_pcm = { + .name = "SMDK-PCM", + .dai_link = smdk_dai, + .num_links = 2, +}; + +/* + * After SMDKC110 Base Board's Rev is '0.1', 12MHz External OSC(X1) + * is absent (or not connected), so we connect EXT_VOICE_CLK(OSC4), + * 2.0484Mhz, directly with MCLK both Codec and SoC. + */ +static int __devinit snd_smdk_probe(struct platform_device *pdev) +{ + int ret = 0; + + xtal_freq = SMDK_WM8580_EXT_OSC; + mclk_freq = SMDK_WM8580_EXT_MCLK; + + if (machine_is_smdkc110() || machine_is_smdkv210()) + xtal_freq = mclk_freq = SMDK_WM8580_EXT_VOICE; + + smdk_pcm.dev = &pdev->dev; + ret = snd_soc_register_card(&smdk_pcm); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + return 0; +} + +static int __devexit snd_smdk_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&smdk_pcm); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_smdk_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "samsung-smdk-pcm", + }, + .probe = snd_smdk_probe, + .remove = __devexit_p(snd_smdk_remove), +}; + +static int __init smdk_audio_init(void) +{ + return platform_driver_register(&snd_smdk_driver); +} + +module_init(smdk_audio_init); + +static void __exit smdk_audio_exit(void) +{ + platform_driver_unregister(&snd_smdk_driver); +} + +module_exit(smdk_audio_exit); + +MODULE_AUTHOR("Sangbeom Kim, "); +MODULE_DESCRIPTION("ALSA SoC SMDK WM8580 for PCM"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 0671fd8ef4b32200e75396cd299f0853002fc11e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 8 Apr 2011 14:50:44 +0900 Subject: ASoC: Add soc_remove_dai_links card->num_rtd should be 0 after soc_romve_dai_link Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f75f1392604..1f114673e95 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1453,6 +1453,16 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } } +static void soc_remove_dai_links(struct snd_soc_card *card) +{ + int i; + + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i); + + card->num_rtd = 0; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_codec *codec) { @@ -1960,8 +1970,7 @@ probe_aux_dev_err: soc_remove_aux_dev(card, i); probe_dai_err: - for (i = 0; i < card->num_links; i++) - soc_remove_dai_link(card, i); + soc_remove_dai_links(card); card_probe_error: if (card->remove) @@ -2023,8 +2032,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) soc_remove_aux_dev(card, i); /* remove and free each DAI */ - for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + soc_remove_dai_links(card); soc_cleanup_card_debugfs(card); -- cgit v1.2.3 From c93993aca45a223452d2a95383b655c85878c6e8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Feb 2011 14:09:41 +0000 Subject: ASoC: Add WM8915 CODEC driver The WM8915 is an ultra low power mobile CODEC designed for smartphones, featuring a mixture of digital and analogue I/O with flexible mixing options and advanced low power accessory detection functionality in a compact package. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8915.c | 2925 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8915.h | 3717 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 6648 insertions(+) create mode 100644 sound/soc/codecs/wm8915.c create mode 100644 sound/soc/codecs/wm8915.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ee7374e8e97..54d5dd6c63a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -73,6 +73,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C + select SND_SOC_WM8915 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8955 if I2C select SND_SOC_WM8960 if I2C @@ -302,6 +303,9 @@ config SND_SOC_WM8903 config SND_SOC_WM8904 tristate +config SND_SOC_WM8915 + tristate + config SND_SOC_WM8940 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f030c182674..02425e6a87e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -57,6 +57,7 @@ snd-soc-wm8804-objs := wm8804.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o +snd-soc-wm8915-objs := wm8915.o snd-soc-wm8940-objs := wm8940.o snd-soc-wm8955-objs := wm8955.o snd-soc-wm8960-objs := wm8960.o @@ -146,6 +147,7 @@ obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o +obj-$(CONFIG_SND_SOC_WM8915) += snd-soc-wm8915.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8955) += snd-soc-wm8955.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c new file mode 100644 index 00000000000..3adad2872c7 --- /dev/null +++ b/sound/soc/codecs/wm8915.c @@ -0,0 +1,2925 @@ +/* + * wm8915.c - WM8915 audio codec interface + * + * Copyright 2011 Wolfson Microelectronics PLC. + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "wm8915.h" + +#define WM8915_AIFS 2 + +#define HPOUT1L 1 +#define HPOUT1R 2 +#define HPOUT2L 4 +#define HPOUT2R 8 + +#define WM8915_NUM_SUPPLIES 6 +static const char *wm8915_supply_names[WM8915_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD", + "AVDD1", + "AVDD2", + "CPVDD", + "MICVDD", +}; + +struct wm8915_priv { + struct snd_soc_codec *codec; + + int ldo1ena; + + int sysclk; + + int fll_src; + int fll_fref; + int fll_fout; + + struct completion fll_lock; + + u16 dcs_pending; + struct completion dcs_done; + + u16 hpout_ena; + u16 hpout_pending; + + struct regulator_bulk_data supplies[WM8915_NUM_SUPPLIES]; + struct notifier_block disable_nb[WM8915_NUM_SUPPLIES]; + + struct wm8915_pdata pdata; + + int rx_rate[WM8915_AIFS]; + + /* Platform dependant ReTune mobile configuration */ + int num_retune_mobile_texts; + const char **retune_mobile_texts; + int retune_mobile_cfg[2]; + struct soc_enum retune_mobile_enum; + + struct snd_soc_jack *jack; + bool detecting; + bool jack_mic; + wm8915_polarity_fn polarity_cb; + +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif +}; + +/* We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define WM8915_REGULATOR_EVENT(n) \ +static int wm8915_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct wm8915_priv *wm8915 = container_of(nb, struct wm8915_priv, \ + disable_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + wm8915->codec->cache_sync = 1; \ + } \ + return 0; \ +} + +WM8915_REGULATOR_EVENT(0) +WM8915_REGULATOR_EVENT(1) +WM8915_REGULATOR_EVENT(2) +WM8915_REGULATOR_EVENT(3) +WM8915_REGULATOR_EVENT(4) +WM8915_REGULATOR_EVENT(5) + +static const u16 wm8915_reg[WM8915_MAX_REGISTER] = { + [WM8915_SOFTWARE_RESET] = 0x8915, + [WM8915_POWER_MANAGEMENT_7] = 0x10, + [WM8915_DAC1_HPOUT1_VOLUME] = 0x88, + [WM8915_DAC2_HPOUT2_VOLUME] = 0x88, + [WM8915_DAC1_LEFT_VOLUME] = 0x2c0, + [WM8915_DAC1_RIGHT_VOLUME] = 0x2c0, + [WM8915_DAC2_LEFT_VOLUME] = 0x2c0, + [WM8915_DAC2_RIGHT_VOLUME] = 0x2c0, + [WM8915_OUTPUT1_LEFT_VOLUME] = 0x80, + [WM8915_OUTPUT1_RIGHT_VOLUME] = 0x80, + [WM8915_OUTPUT2_LEFT_VOLUME] = 0x80, + [WM8915_OUTPUT2_RIGHT_VOLUME] = 0x80, + [WM8915_MICBIAS_1] = 0x39, + [WM8915_MICBIAS_2] = 0x39, + [WM8915_LDO_1] = 0x3, + [WM8915_LDO_2] = 0x13, + [WM8915_ACCESSORY_DETECT_MODE_1] = 0x4, + [WM8915_HEADPHONE_DETECT_1] = 0x20, + [WM8915_MIC_DETECT_1] = 0x7600, + [WM8915_MIC_DETECT_2] = 0xbf, + [WM8915_CHARGE_PUMP_1] = 0x1f25, + [WM8915_CHARGE_PUMP_2] = 0xab19, + [WM8915_DC_SERVO_5] = 0x2a2a, + [WM8915_CONTROL_INTERFACE_1] = 0x8004, + [WM8915_CLOCKING_1] = 0x10, + [WM8915_AIF_RATE] = 0x83, + [WM8915_FLL_CONTROL_4] = 0x5dc0, + [WM8915_FLL_CONTROL_5] = 0xc84, + [WM8915_FLL_EFS_2] = 0x2, + [WM8915_AIF1_TX_LRCLK_1] = 0x80, + [WM8915_AIF1_TX_LRCLK_2] = 0x8, + [WM8915_AIF1_RX_LRCLK_1] = 0x80, + [WM8915_AIF1TX_DATA_CONFIGURATION_1] = 0x1818, + [WM8915_AIF1RX_DATA_CONFIGURATION] = 0x1818, + [WM8915_AIF1TX_TEST] = 0x7, + [WM8915_AIF2_TX_LRCLK_1] = 0x80, + [WM8915_AIF2_TX_LRCLK_2] = 0x8, + [WM8915_AIF2_RX_LRCLK_1] = 0x80, + [WM8915_AIF2TX_DATA_CONFIGURATION_1] = 0x1818, + [WM8915_AIF2RX_DATA_CONFIGURATION] = 0x1818, + [WM8915_AIF2TX_TEST] = 0x1, + [WM8915_DSP1_TX_LEFT_VOLUME] = 0xc0, + [WM8915_DSP1_TX_RIGHT_VOLUME] = 0xc0, + [WM8915_DSP1_RX_LEFT_VOLUME] = 0xc0, + [WM8915_DSP1_RX_RIGHT_VOLUME] = 0xc0, + [WM8915_DSP1_TX_FILTERS] = 0x2000, + [WM8915_DSP1_RX_FILTERS_1] = 0x200, + [WM8915_DSP1_RX_FILTERS_2] = 0x10, + [WM8915_DSP1_DRC_1] = 0x98, + [WM8915_DSP1_DRC_2] = 0x845, + [WM8915_DSP1_RX_EQ_GAINS_1] = 0x6318, + [WM8915_DSP1_RX_EQ_GAINS_2] = 0x6300, + [WM8915_DSP1_RX_EQ_BAND_1_A] = 0xfca, + [WM8915_DSP1_RX_EQ_BAND_1_B] = 0x400, + [WM8915_DSP1_RX_EQ_BAND_1_PG] = 0xd8, + [WM8915_DSP1_RX_EQ_BAND_2_A] = 0x1eb5, + [WM8915_DSP1_RX_EQ_BAND_2_B] = 0xf145, + [WM8915_DSP1_RX_EQ_BAND_2_C] = 0xb75, + [WM8915_DSP1_RX_EQ_BAND_2_PG] = 0x1c5, + [WM8915_DSP1_RX_EQ_BAND_3_A] = 0x1c58, + [WM8915_DSP1_RX_EQ_BAND_3_B] = 0xf373, + [WM8915_DSP1_RX_EQ_BAND_3_C] = 0xa54, + [WM8915_DSP1_RX_EQ_BAND_3_PG] = 0x558, + [WM8915_DSP1_RX_EQ_BAND_4_A] = 0x168e, + [WM8915_DSP1_RX_EQ_BAND_4_B] = 0xf829, + [WM8915_DSP1_RX_EQ_BAND_4_C] = 0x7ad, + [WM8915_DSP1_RX_EQ_BAND_4_PG] = 0x1103, + [WM8915_DSP1_RX_EQ_BAND_5_A] = 0x564, + [WM8915_DSP1_RX_EQ_BAND_5_B] = 0x559, + [WM8915_DSP1_RX_EQ_BAND_5_PG] = 0x4000, + [WM8915_DSP2_TX_LEFT_VOLUME] = 0xc0, + [WM8915_DSP2_TX_RIGHT_VOLUME] = 0xc0, + [WM8915_DSP2_RX_LEFT_VOLUME] = 0xc0, + [WM8915_DSP2_RX_RIGHT_VOLUME] = 0xc0, + [WM8915_DSP2_TX_FILTERS] = 0x2000, + [WM8915_DSP2_RX_FILTERS_1] = 0x200, + [WM8915_DSP2_RX_FILTERS_2] = 0x10, + [WM8915_DSP2_DRC_1] = 0x98, + [WM8915_DSP2_DRC_2] = 0x845, + [WM8915_DSP2_RX_EQ_GAINS_1] = 0x6318, + [WM8915_DSP2_RX_EQ_GAINS_2] = 0x6300, + [WM8915_DSP2_RX_EQ_BAND_1_A] = 0xfca, + [WM8915_DSP2_RX_EQ_BAND_1_B] = 0x400, + [WM8915_DSP2_RX_EQ_BAND_1_PG] = 0xd8, + [WM8915_DSP2_RX_EQ_BAND_2_A] = 0x1eb5, + [WM8915_DSP2_RX_EQ_BAND_2_B] = 0xf145, + [WM8915_DSP2_RX_EQ_BAND_2_C] = 0xb75, + [WM8915_DSP2_RX_EQ_BAND_2_PG] = 0x1c5, + [WM8915_DSP2_RX_EQ_BAND_3_A] = 0x1c58, + [WM8915_DSP2_RX_EQ_BAND_3_B] = 0xf373, + [WM8915_DSP2_RX_EQ_BAND_3_C] = 0xa54, + [WM8915_DSP2_RX_EQ_BAND_3_PG] = 0x558, + [WM8915_DSP2_RX_EQ_BAND_4_A] = 0x168e, + [WM8915_DSP2_RX_EQ_BAND_4_B] = 0xf829, + [WM8915_DSP2_RX_EQ_BAND_4_C] = 0x7ad, + [WM8915_DSP2_RX_EQ_BAND_4_PG] = 0x1103, + [WM8915_DSP2_RX_EQ_BAND_5_A] = 0x564, + [WM8915_DSP2_RX_EQ_BAND_5_B] = 0x559, + [WM8915_DSP2_RX_EQ_BAND_5_PG] = 0x4000, + [WM8915_OVERSAMPLING] = 0xd, + [WM8915_SIDETONE] = 0x1040, + [WM8915_GPIO_1] = 0xa101, + [WM8915_GPIO_2] = 0xa101, + [WM8915_GPIO_3] = 0xa101, + [WM8915_GPIO_4] = 0xa101, + [WM8915_GPIO_5] = 0xa101, + [WM8915_PULL_CONTROL_2] = 0x140, + [WM8915_INTERRUPT_STATUS_1_MASK] = 0x1f, + [WM8915_INTERRUPT_STATUS_2_MASK] = 0x1ecf, + [WM8915_RIGHT_PDM_SPEAKER] = 0x1, + [WM8915_PDM_SPEAKER_MUTE_SEQUENCE] = 0x69, + [WM8915_PDM_SPEAKER_VOLUME] = 0x66, + [WM8915_WRITE_SEQUENCER_0] = 0x1, + [WM8915_WRITE_SEQUENCER_1] = 0x1, + [WM8915_WRITE_SEQUENCER_3] = 0x6, + [WM8915_WRITE_SEQUENCER_4] = 0x40, + [WM8915_WRITE_SEQUENCER_5] = 0x1, + [WM8915_WRITE_SEQUENCER_6] = 0xf, + [WM8915_WRITE_SEQUENCER_7] = 0x6, + [WM8915_WRITE_SEQUENCER_8] = 0x1, + [WM8915_WRITE_SEQUENCER_9] = 0x3, + [WM8915_WRITE_SEQUENCER_10] = 0x104, + [WM8915_WRITE_SEQUENCER_12] = 0x60, + [WM8915_WRITE_SEQUENCER_13] = 0x11, + [WM8915_WRITE_SEQUENCER_14] = 0x401, + [WM8915_WRITE_SEQUENCER_16] = 0x50, + [WM8915_WRITE_SEQUENCER_17] = 0x3, + [WM8915_WRITE_SEQUENCER_18] = 0x100, + [WM8915_WRITE_SEQUENCER_20] = 0x51, + [WM8915_WRITE_SEQUENCER_21] = 0x3, + [WM8915_WRITE_SEQUENCER_22] = 0x104, + [WM8915_WRITE_SEQUENCER_23] = 0xa, + [WM8915_WRITE_SEQUENCER_24] = 0x60, + [WM8915_WRITE_SEQUENCER_25] = 0x3b, + [WM8915_WRITE_SEQUENCER_26] = 0x502, + [WM8915_WRITE_SEQUENCER_27] = 0x100, + [WM8915_WRITE_SEQUENCER_28] = 0x2fff, + [WM8915_WRITE_SEQUENCER_32] = 0x2fff, + [WM8915_WRITE_SEQUENCER_36] = 0x2fff, + [WM8915_WRITE_SEQUENCER_40] = 0x2fff, + [WM8915_WRITE_SEQUENCER_44] = 0x2fff, + [WM8915_WRITE_SEQUENCER_48] = 0x2fff, + [WM8915_WRITE_SEQUENCER_52] = 0x2fff, + [WM8915_WRITE_SEQUENCER_56] = 0x2fff, + [WM8915_WRITE_SEQUENCER_60] = 0x2fff, + [WM8915_WRITE_SEQUENCER_64] = 0x1, + [WM8915_WRITE_SEQUENCER_65] = 0x1, + [WM8915_WRITE_SEQUENCER_67] = 0x6, + [WM8915_WRITE_SEQUENCER_68] = 0x40, + [WM8915_WRITE_SEQUENCER_69] = 0x1, + [WM8915_WRITE_SEQUENCER_70] = 0xf, + [WM8915_WRITE_SEQUENCER_71] = 0x6, + [WM8915_WRITE_SEQUENCER_72] = 0x1, + [WM8915_WRITE_SEQUENCER_73] = 0x3, + [WM8915_WRITE_SEQUENCER_74] = 0x104, + [WM8915_WRITE_SEQUENCER_76] = 0x60, + [WM8915_WRITE_SEQUENCER_77] = 0x11, + [WM8915_WRITE_SEQUENCER_78] = 0x401, + [WM8915_WRITE_SEQUENCER_80] = 0x50, + [WM8915_WRITE_SEQUENCER_81] = 0x3, + [WM8915_WRITE_SEQUENCER_82] = 0x100, + [WM8915_WRITE_SEQUENCER_84] = 0x60, + [WM8915_WRITE_SEQUENCER_85] = 0x3b, + [WM8915_WRITE_SEQUENCER_86] = 0x502, + [WM8915_WRITE_SEQUENCER_87] = 0x100, + [WM8915_WRITE_SEQUENCER_88] = 0x2fff, + [WM8915_WRITE_SEQUENCER_92] = 0x2fff, + [WM8915_WRITE_SEQUENCER_96] = 0x2fff, + [WM8915_WRITE_SEQUENCER_100] = 0x2fff, + [WM8915_WRITE_SEQUENCER_104] = 0x2fff, + [WM8915_WRITE_SEQUENCER_108] = 0x2fff, + [WM8915_WRITE_SEQUENCER_112] = 0x2fff, + [WM8915_WRITE_SEQUENCER_116] = 0x2fff, + [WM8915_WRITE_SEQUENCER_120] = 0x2fff, + [WM8915_WRITE_SEQUENCER_124] = 0x2fff, + [WM8915_WRITE_SEQUENCER_128] = 0x1, + [WM8915_WRITE_SEQUENCER_129] = 0x1, + [WM8915_WRITE_SEQUENCER_131] = 0x6, + [WM8915_WRITE_SEQUENCER_132] = 0x40, + [WM8915_WRITE_SEQUENCER_133] = 0x1, + [WM8915_WRITE_SEQUENCER_134] = 0xf, + [WM8915_WRITE_SEQUENCER_135] = 0x6, + [WM8915_WRITE_SEQUENCER_136] = 0x1, + [WM8915_WRITE_SEQUENCER_137] = 0x3, + [WM8915_WRITE_SEQUENCER_138] = 0x106, + [WM8915_WRITE_SEQUENCER_140] = 0x61, + [WM8915_WRITE_SEQUENCER_141] = 0x11, + [WM8915_WRITE_SEQUENCER_142] = 0x401, + [WM8915_WRITE_SEQUENCER_144] = 0x50, + [WM8915_WRITE_SEQUENCER_145] = 0x3, + [WM8915_WRITE_SEQUENCER_146] = 0x102, + [WM8915_WRITE_SEQUENCER_148] = 0x51, + [WM8915_WRITE_SEQUENCER_149] = 0x3, + [WM8915_WRITE_SEQUENCER_150] = 0x106, + [WM8915_WRITE_SEQUENCER_151] = 0xa, + [WM8915_WRITE_SEQUENCER_152] = 0x61, + [WM8915_WRITE_SEQUENCER_153] = 0x3b, + [WM8915_WRITE_SEQUENCER_154] = 0x502, + [WM8915_WRITE_SEQUENCER_155] = 0x100, + [WM8915_WRITE_SEQUENCER_156] = 0x2fff, + [WM8915_WRITE_SEQUENCER_160] = 0x2fff, + [WM8915_WRITE_SEQUENCER_164] = 0x2fff, + [WM8915_WRITE_SEQUENCER_168] = 0x2fff, + [WM8915_WRITE_SEQUENCER_172] = 0x2fff, + [WM8915_WRITE_SEQUENCER_176] = 0x2fff, + [WM8915_WRITE_SEQUENCER_180] = 0x2fff, + [WM8915_WRITE_SEQUENCER_184] = 0x2fff, + [WM8915_WRITE_SEQUENCER_188] = 0x2fff, + [WM8915_WRITE_SEQUENCER_192] = 0x1, + [WM8915_WRITE_SEQUENCER_193] = 0x1, + [WM8915_WRITE_SEQUENCER_195] = 0x6, + [WM8915_WRITE_SEQUENCER_196] = 0x40, + [WM8915_WRITE_SEQUENCER_197] = 0x1, + [WM8915_WRITE_SEQUENCER_198] = 0xf, + [WM8915_WRITE_SEQUENCER_199] = 0x6, + [WM8915_WRITE_SEQUENCER_200] = 0x1, + [WM8915_WRITE_SEQUENCER_201] = 0x3, + [WM8915_WRITE_SEQUENCER_202] = 0x106, + [WM8915_WRITE_SEQUENCER_204] = 0x61, + [WM8915_WRITE_SEQUENCER_205] = 0x11, + [WM8915_WRITE_SEQUENCER_206] = 0x401, + [WM8915_WRITE_SEQUENCER_208] = 0x50, + [WM8915_WRITE_SEQUENCER_209] = 0x3, + [WM8915_WRITE_SEQUENCER_210] = 0x102, + [WM8915_WRITE_SEQUENCER_212] = 0x61, + [WM8915_WRITE_SEQUENCER_213] = 0x3b, + [WM8915_WRITE_SEQUENCER_214] = 0x502, + [WM8915_WRITE_SEQUENCER_215] = 0x100, + [WM8915_WRITE_SEQUENCER_216] = 0x2fff, + [WM8915_WRITE_SEQUENCER_220] = 0x2fff, + [WM8915_WRITE_SEQUENCER_224] = 0x2fff, + [WM8915_WRITE_SEQUENCER_228] = 0x2fff, + [WM8915_WRITE_SEQUENCER_232] = 0x2fff, + [WM8915_WRITE_SEQUENCER_236] = 0x2fff, + [WM8915_WRITE_SEQUENCER_240] = 0x2fff, + [WM8915_WRITE_SEQUENCER_244] = 0x2fff, + [WM8915_WRITE_SEQUENCER_248] = 0x2fff, + [WM8915_WRITE_SEQUENCER_252] = 0x2fff, + [WM8915_WRITE_SEQUENCER_256] = 0x60, + [WM8915_WRITE_SEQUENCER_258] = 0x601, + [WM8915_WRITE_SEQUENCER_260] = 0x50, + [WM8915_WRITE_SEQUENCER_262] = 0x100, + [WM8915_WRITE_SEQUENCER_264] = 0x1, + [WM8915_WRITE_SEQUENCER_266] = 0x104, + [WM8915_WRITE_SEQUENCER_267] = 0x100, + [WM8915_WRITE_SEQUENCER_268] = 0x2fff, + [WM8915_WRITE_SEQUENCER_272] = 0x2fff, + [WM8915_WRITE_SEQUENCER_276] = 0x2fff, + [WM8915_WRITE_SEQUENCER_280] = 0x2fff, + [WM8915_WRITE_SEQUENCER_284] = 0x2fff, + [WM8915_WRITE_SEQUENCER_288] = 0x2fff, + [WM8915_WRITE_SEQUENCER_292] = 0x2fff, + [WM8915_WRITE_SEQUENCER_296] = 0x2fff, + [WM8915_WRITE_SEQUENCER_300] = 0x2fff, + [WM8915_WRITE_SEQUENCER_304] = 0x2fff, + [WM8915_WRITE_SEQUENCER_308] = 0x2fff, + [WM8915_WRITE_SEQUENCER_312] = 0x2fff, + [WM8915_WRITE_SEQUENCER_316] = 0x2fff, + [WM8915_WRITE_SEQUENCER_320] = 0x61, + [WM8915_WRITE_SEQUENCER_322] = 0x601, + [WM8915_WRITE_SEQUENCER_324] = 0x50, + [WM8915_WRITE_SEQUENCER_326] = 0x102, + [WM8915_WRITE_SEQUENCER_328] = 0x1, + [WM8915_WRITE_SEQUENCER_330] = 0x106, + [WM8915_WRITE_SEQUENCER_331] = 0x100, + [WM8915_WRITE_SEQUENCER_332] = 0x2fff, + [WM8915_WRITE_SEQUENCER_336] = 0x2fff, + [WM8915_WRITE_SEQUENCER_340] = 0x2fff, + [WM8915_WRITE_SEQUENCER_344] = 0x2fff, + [WM8915_WRITE_SEQUENCER_348] = 0x2fff, + [WM8915_WRITE_SEQUENCER_352] = 0x2fff, + [WM8915_WRITE_SEQUENCER_356] = 0x2fff, + [WM8915_WRITE_SEQUENCER_360] = 0x2fff, + [WM8915_WRITE_SEQUENCER_364] = 0x2fff, + [WM8915_WRITE_SEQUENCER_368] = 0x2fff, + [WM8915_WRITE_SEQUENCER_372] = 0x2fff, + [WM8915_WRITE_SEQUENCER_376] = 0x2fff, + [WM8915_WRITE_SEQUENCER_380] = 0x2fff, + [WM8915_WRITE_SEQUENCER_384] = 0x60, + [WM8915_WRITE_SEQUENCER_386] = 0x601, + [WM8915_WRITE_SEQUENCER_388] = 0x61, + [WM8915_WRITE_SEQUENCER_390] = 0x601, + [WM8915_WRITE_SEQUENCER_392] = 0x50, + [WM8915_WRITE_SEQUENCER_394] = 0x300, + [WM8915_WRITE_SEQUENCER_396] = 0x1, + [WM8915_WRITE_SEQUENCER_398] = 0x304, + [WM8915_WRITE_SEQUENCER_400] = 0x40, + [WM8915_WRITE_SEQUENCER_402] = 0xf, + [WM8915_WRITE_SEQUENCER_404] = 0x1, + [WM8915_WRITE_SEQUENCER_407] = 0x100, +}; + +static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 150, 0); +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(out_digital_tlv, -1200, 150, 0); +static const DECLARE_TLV_DB_SCALE(out_tlv, -900, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -900, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); + +static const char *sidetone_hpf_text[] = { + "2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz" +}; + +static const struct soc_enum sidetone_hpf = + SOC_ENUM_SINGLE(WM8915_SIDETONE, 7, 6, sidetone_hpf_text); + +static const char *hpf_mode_text[] = { + "HiFi", "Custom", "Voice" +}; + +static const struct soc_enum dsp1tx_hpf_mode = + SOC_ENUM_SINGLE(WM8915_DSP1_TX_FILTERS, 3, 3, hpf_mode_text); + +static const struct soc_enum dsp2tx_hpf_mode = + SOC_ENUM_SINGLE(WM8915_DSP2_TX_FILTERS, 3, 3, hpf_mode_text); + +static const char *hpf_cutoff_text[] = { + "50Hz", "75Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum dsp1tx_hpf_cutoff = + SOC_ENUM_SINGLE(WM8915_DSP1_TX_FILTERS, 0, 7, hpf_cutoff_text); + +static const struct soc_enum dsp2tx_hpf_cutoff = + SOC_ENUM_SINGLE(WM8915_DSP2_TX_FILTERS, 0, 7, hpf_cutoff_text); + +static void wm8915_set_retune_mobile(struct snd_soc_codec *codec, int block) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + struct wm8915_pdata *pdata = &wm8915->pdata; + int base, best, best_val, save, i, cfg, iface; + + if (!wm8915->num_retune_mobile_texts) + return; + + switch (block) { + case 0: + base = WM8915_DSP1_RX_EQ_GAINS_1; + if (snd_soc_read(codec, WM8915_POWER_MANAGEMENT_8) & + WM8915_DSP1RX_SRC) + iface = 1; + else + iface = 0; + break; + case 1: + base = WM8915_DSP1_RX_EQ_GAINS_2; + if (snd_soc_read(codec, WM8915_POWER_MANAGEMENT_8) & + WM8915_DSP2RX_SRC) + iface = 1; + else + iface = 0; + break; + default: + return; + } + + /* Find the version of the currently selected configuration + * with the nearest sample rate. */ + cfg = wm8915->retune_mobile_cfg[block]; + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8915->retune_mobile_texts[cfg]) == 0 && + abs(pdata->retune_mobile_cfgs[i].rate + - wm8915->rx_rate[iface]) < best_val) { + best = i; + best_val = abs(pdata->retune_mobile_cfgs[i].rate + - wm8915->rx_rate[iface]); + } + } + + dev_dbg(codec->dev, "ReTune Mobile %d %s/%dHz for %dHz sample rate\n", + block, + pdata->retune_mobile_cfgs[best].name, + pdata->retune_mobile_cfgs[best].rate, + wm8915->rx_rate[iface]); + + /* The EQ will be disabled while reconfiguring it, remember the + * current configuration. + */ + save = snd_soc_read(codec, base); + save &= WM8915_DSP1RX_EQ_ENA; + + for (i = 0; i < ARRAY_SIZE(pdata->retune_mobile_cfgs[best].regs); i++) + snd_soc_update_bits(codec, base + i, 0xffff, + pdata->retune_mobile_cfgs[best].regs[i]); + + snd_soc_update_bits(codec, base, WM8915_DSP1RX_EQ_ENA, save); +} + +/* Icky as hell but saves code duplication */ +static int wm8915_get_retune_mobile_block(const char *name) +{ + if (strcmp(name, "DSP1 EQ Mode") == 0) + return 0; + if (strcmp(name, "DSP2 EQ Mode") == 0) + return 1; + return -EINVAL; +} + +static int wm8915_put_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + struct wm8915_pdata *pdata = &wm8915->pdata; + int block = wm8915_get_retune_mobile_block(kcontrol->id.name); + int value = ucontrol->value.integer.value[0]; + + if (block < 0) + return block; + + if (value >= pdata->num_retune_mobile_cfgs) + return -EINVAL; + + wm8915->retune_mobile_cfg[block] = value; + + wm8915_set_retune_mobile(codec, block); + + return 0; +} + +static int wm8915_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int block = wm8915_get_retune_mobile_block(kcontrol->id.name); + + ucontrol->value.enumerated.item[0] = wm8915->retune_mobile_cfg[block]; + + return 0; +} + +static const struct snd_kcontrol_new wm8915_snd_controls[] = { +SOC_DOUBLE_R_TLV("Capture Volume", WM8915_LEFT_LINE_INPUT_VOLUME, + WM8915_RIGHT_LINE_INPUT_VOLUME, 0, 31, 0, inpga_tlv), +SOC_DOUBLE_R("Capture ZC Switch", WM8915_LEFT_LINE_INPUT_VOLUME, + WM8915_RIGHT_LINE_INPUT_VOLUME, 5, 1, 0), + +SOC_DOUBLE_TLV("DAC1 Sidetone Volume", WM8915_DAC1_MIXER_VOLUMES, + 0, 5, 24, 0, sidetone_tlv), +SOC_DOUBLE_TLV("DAC2 Sidetone Volume", WM8915_DAC2_MIXER_VOLUMES, + 0, 5, 24, 0, sidetone_tlv), +SOC_SINGLE("Sidetone LPF Switch", WM8915_SIDETONE, 12, 1, 0), +SOC_ENUM("Sidetone HPF Cut-off", sidetone_hpf), +SOC_SINGLE("Sidetone HPF Switch", WM8915_SIDETONE, 6, 1, 0), + +SOC_DOUBLE_R_TLV("DSP1 Capture Volume", WM8915_DSP1_TX_LEFT_VOLUME, + WM8915_DSP1_TX_RIGHT_VOLUME, 1, 96, 0, digital_tlv), +SOC_DOUBLE_R_TLV("DSP2 Capture Volume", WM8915_DSP2_TX_LEFT_VOLUME, + WM8915_DSP2_TX_RIGHT_VOLUME, 1, 96, 0, digital_tlv), + +SOC_SINGLE("DSP1 Capture Notch Filter Switch", WM8915_DSP1_TX_FILTERS, + 13, 1, 0), +SOC_DOUBLE("DSP1 Capture HPF Switch", WM8915_DSP1_TX_FILTERS, 12, 11, 1, 0), +SOC_ENUM("DSP1 Capture HPF Mode", dsp1tx_hpf_mode), +SOC_ENUM("DSP1 Capture HPF Cutoff", dsp1tx_hpf_cutoff), + +SOC_SINGLE("DSP2 Capture Notch Filter Switch", WM8915_DSP2_TX_FILTERS, + 13, 1, 0), +SOC_DOUBLE("DSP2 Capture HPF Switch", WM8915_DSP2_TX_FILTERS, 12, 11, 1, 0), +SOC_ENUM("DSP2 Capture HPF Mode", dsp2tx_hpf_mode), +SOC_ENUM("DSP2 Capture HPF Cutoff", dsp2tx_hpf_cutoff), + +SOC_DOUBLE_R_TLV("DSP1 Playback Volume", WM8915_DSP1_RX_LEFT_VOLUME, + WM8915_DSP1_RX_RIGHT_VOLUME, 1, 112, 0, digital_tlv), +SOC_SINGLE("DSP1 Playback Switch", WM8915_DSP1_RX_FILTERS_1, 9, 1, 1), + +SOC_DOUBLE_R_TLV("DSP2 Playback Volume", WM8915_DSP2_RX_LEFT_VOLUME, + WM8915_DSP2_RX_RIGHT_VOLUME, 1, 112, 0, digital_tlv), +SOC_SINGLE("DSP2 Playback Switch", WM8915_DSP2_RX_FILTERS_1, 9, 1, 1), + +SOC_DOUBLE_R_TLV("DAC1 Volume", WM8915_DAC1_LEFT_VOLUME, + WM8915_DAC1_RIGHT_VOLUME, 1, 112, 0, digital_tlv), +SOC_DOUBLE_R("DAC1 Switch", WM8915_DAC1_LEFT_VOLUME, + WM8915_DAC1_RIGHT_VOLUME, 9, 1, 1), + +SOC_DOUBLE_R_TLV("DAC2 Volume", WM8915_DAC2_LEFT_VOLUME, + WM8915_DAC2_RIGHT_VOLUME, 1, 112, 0, digital_tlv), +SOC_DOUBLE_R("DAC2 Switch", WM8915_DAC2_LEFT_VOLUME, + WM8915_DAC2_RIGHT_VOLUME, 9, 1, 1), + +SOC_SINGLE("Speaker High Performance Switch", WM8915_OVERSAMPLING, 3, 1, 0), +SOC_SINGLE("DMIC High Performance Switch", WM8915_OVERSAMPLING, 2, 1, 0), +SOC_SINGLE("ADC High Performance Switch", WM8915_OVERSAMPLING, 1, 1, 0), +SOC_SINGLE("DAC High Performance Switch", WM8915_OVERSAMPLING, 0, 1, 0), + +SOC_SINGLE("DAC Soft Mute Switch", WM8915_DAC_SOFTMUTE, 1, 1, 0), +SOC_SINGLE("DAC Slow Soft Mute Switch", WM8915_DAC_SOFTMUTE, 0, 1, 0), + +SOC_DOUBLE_TLV("Digital Output 1 Volume", WM8915_DAC1_HPOUT1_VOLUME, 0, 4, + 8, 0, out_digital_tlv), +SOC_DOUBLE_TLV("Digital Output 2 Volume", WM8915_DAC2_HPOUT2_VOLUME, 0, 4, + 8, 0, out_digital_tlv), + +SOC_DOUBLE_R_TLV("Output 1 Volume", WM8915_OUTPUT1_LEFT_VOLUME, + WM8915_OUTPUT1_RIGHT_VOLUME, 0, 12, 0, out_tlv), +SOC_DOUBLE_R("Output 1 ZC Switch", WM8915_OUTPUT1_LEFT_VOLUME, + WM8915_OUTPUT1_RIGHT_VOLUME, 7, 1, 0), + +SOC_DOUBLE_R_TLV("Output 2 Volume", WM8915_OUTPUT2_LEFT_VOLUME, + WM8915_OUTPUT2_RIGHT_VOLUME, 0, 12, 0, out_tlv), +SOC_DOUBLE_R("Output 2 ZC Switch", WM8915_OUTPUT2_LEFT_VOLUME, + WM8915_OUTPUT2_RIGHT_VOLUME, 7, 1, 0), + +SOC_DOUBLE_TLV("Speaker Volume", WM8915_PDM_SPEAKER_VOLUME, 0, 4, 8, 0, + spk_tlv), +SOC_DOUBLE_R("Speaker Switch", WM8915_LEFT_PDM_SPEAKER, + WM8915_RIGHT_PDM_SPEAKER, 3, 1, 1), +SOC_DOUBLE_R("Speaker ZC Switch", WM8915_LEFT_PDM_SPEAKER, + WM8915_RIGHT_PDM_SPEAKER, 2, 1, 0), + +SOC_SINGLE("DSP1 EQ Switch", WM8915_DSP1_RX_EQ_GAINS_1, 0, 1, 0), +SOC_SINGLE("DSP2 EQ Switch", WM8915_DSP2_RX_EQ_GAINS_1, 0, 1, 0), +}; + +static const struct snd_kcontrol_new wm8915_eq_controls[] = { +SOC_SINGLE_TLV("DSP1 EQ B1 Volume", WM8915_DSP1_RX_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP1 EQ B2 Volume", WM8915_DSP1_RX_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP1 EQ B3 Volume", WM8915_DSP1_RX_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP1 EQ B4 Volume", WM8915_DSP1_RX_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP1 EQ B5 Volume", WM8915_DSP1_RX_EQ_GAINS_2, 6, 31, 0, + eq_tlv), + +SOC_SINGLE_TLV("DSP2 EQ B1 Volume", WM8915_DSP2_RX_EQ_GAINS_1, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP2 EQ B2 Volume", WM8915_DSP2_RX_EQ_GAINS_1, 6, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP2 EQ B3 Volume", WM8915_DSP2_RX_EQ_GAINS_1, 1, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP2 EQ B4 Volume", WM8915_DSP2_RX_EQ_GAINS_2, 11, 31, 0, + eq_tlv), +SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8915_DSP2_RX_EQ_GAINS_2, 6, 31, 0, + eq_tlv), +}; + +static int cp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(5); + break; + default: + BUG(); + return -EINVAL; + } + + return 0; +} + +static int rmv_short_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(w->codec); + + /* Record which outputs we enabled */ + switch (event) { + case SND_SOC_DAPM_PRE_PMD: + wm8915->hpout_pending &= ~w->shift; + break; + case SND_SOC_DAPM_PRE_PMU: + wm8915->hpout_pending |= w->shift; + break; + default: + BUG(); + return -EINVAL; + } + + return 0; +} + +static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) +{ + struct i2c_client *i2c = to_i2c_client(codec->dev); + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int i, ret; + unsigned long timeout = 200; + + snd_soc_write(codec, WM8915_DC_SERVO_2, mask); + + /* Use the interrupt if possible */ + do { + if (i2c->irq) { + timeout = wait_for_completion_timeout(&wm8915->dcs_done, + msecs_to_jiffies(200)); + if (timeout == 0) + dev_err(codec->dev, "DC servo timed out\n"); + + } else { + msleep(1); + if (--i) { + timeout = 0; + break; + } + } + + ret = snd_soc_read(codec, WM8915_DC_SERVO_2); + dev_dbg(codec->dev, "DC servo state: %x\n", ret); + } while (ret & mask); + + if (timeout == 0) + dev_err(codec->dev, "DC servo timed out for %x\n", mask); + else + dev_dbg(codec->dev, "DC servo complete for %x\n", mask); +} + +static void wm8915_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = container_of(dapm, + struct snd_soc_codec, dapm); + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + u16 val, mask; + + /* Complete any pending DC servo starts */ + if (wm8915->dcs_pending) { + dev_dbg(codec->dev, "Starting DC servo for %x\n", + wm8915->dcs_pending); + + /* Trigger a startup sequence */ + wait_for_dc_servo(codec, wm8915->dcs_pending + << WM8915_DCS_TRIG_STARTUP_0_SHIFT); + + wm8915->dcs_pending = 0; + } + + if (wm8915->hpout_pending != wm8915->hpout_ena) { + dev_dbg(codec->dev, "Applying RMV_SHORTs %x->%x\n", + wm8915->hpout_ena, wm8915->hpout_pending); + + val = 0; + mask = 0; + if (wm8915->hpout_pending & HPOUT1L) { + val |= WM8915_HPOUT1L_RMV_SHORT; + mask |= WM8915_HPOUT1L_RMV_SHORT; + } else { + mask |= WM8915_HPOUT1L_RMV_SHORT | + WM8915_HPOUT1L_OUTP | + WM8915_HPOUT1L_DLY; + } + + if (wm8915->hpout_pending & HPOUT1R) { + val |= WM8915_HPOUT1R_RMV_SHORT; + mask |= WM8915_HPOUT1R_RMV_SHORT; + } else { + mask |= WM8915_HPOUT1R_RMV_SHORT | + WM8915_HPOUT1R_OUTP | + WM8915_HPOUT1R_DLY; + } + + snd_soc_update_bits(codec, WM8915_ANALOGUE_HP_1, mask, val); + + val = 0; + mask = 0; + if (wm8915->hpout_pending & HPOUT2L) { + val |= WM8915_HPOUT2L_RMV_SHORT; + mask |= WM8915_HPOUT2L_RMV_SHORT; + } else { + mask |= WM8915_HPOUT2L_RMV_SHORT | + WM8915_HPOUT2L_OUTP | + WM8915_HPOUT2L_DLY; + } + + if (wm8915->hpout_pending & HPOUT2R) { + val |= WM8915_HPOUT2R_RMV_SHORT; + mask |= WM8915_HPOUT2R_RMV_SHORT; + } else { + mask |= WM8915_HPOUT2R_RMV_SHORT | + WM8915_HPOUT2R_OUTP | + WM8915_HPOUT2R_DLY; + } + + snd_soc_update_bits(codec, WM8915_ANALOGUE_HP_2, mask, val); + + wm8915->hpout_ena = wm8915->hpout_pending; + } +} + +static int dcs_start(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(w->codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + wm8915->dcs_pending |= 1 << w->shift; + break; + default: + BUG(); + return -EINVAL; + } + + return 0; +} + +static const char *sidetone_text[] = { + "IN1", "IN2", +}; + +static const struct soc_enum left_sidetone_enum = + SOC_ENUM_SINGLE(WM8915_SIDETONE, 0, 2, sidetone_text); + +static const struct snd_kcontrol_new left_sidetone = + SOC_DAPM_ENUM("Left Sidetone", left_sidetone_enum); + +static const struct soc_enum right_sidetone_enum = + SOC_ENUM_SINGLE(WM8915_SIDETONE, 1, 2, sidetone_text); + +static const struct snd_kcontrol_new right_sidetone = + SOC_DAPM_ENUM("Right Sidetone", right_sidetone_enum); + +static const char *spk_text[] = { + "DAC1L", "DAC1R", "DAC2L", "DAC2R" +}; + +static const struct soc_enum spkl_enum = + SOC_ENUM_SINGLE(WM8915_LEFT_PDM_SPEAKER, 0, 4, spk_text); + +static const struct snd_kcontrol_new spkl_mux = + SOC_DAPM_ENUM("SPKL", spkl_enum); + +static const struct soc_enum spkr_enum = + SOC_ENUM_SINGLE(WM8915_RIGHT_PDM_SPEAKER, 0, 4, spk_text); + +static const struct snd_kcontrol_new spkr_mux = + SOC_DAPM_ENUM("SPKR", spkr_enum); + +static const char *dsp1rx_text[] = { + "AIF1", "AIF2" +}; + +static const struct soc_enum dsp1rx_enum = + SOC_ENUM_SINGLE(WM8915_POWER_MANAGEMENT_8, 0, 2, dsp1rx_text); + +static const struct snd_kcontrol_new dsp1rx = + SOC_DAPM_ENUM("DSP1RX", dsp1rx_enum); + +static const char *dsp2rx_text[] = { + "AIF2", "AIF1" +}; + +static const struct soc_enum dsp2rx_enum = + SOC_ENUM_SINGLE(WM8915_POWER_MANAGEMENT_8, 4, 2, dsp2rx_text); + +static const struct snd_kcontrol_new dsp2rx = + SOC_DAPM_ENUM("DSP2RX", dsp2rx_enum); + +static const char *aif2tx_text[] = { + "DSP2", "DSP1", "AIF1" +}; + +static const struct soc_enum aif2tx_enum = + SOC_ENUM_SINGLE(WM8915_POWER_MANAGEMENT_8, 6, 3, aif2tx_text); + +static const struct snd_kcontrol_new aif2tx = + SOC_DAPM_ENUM("AIF2TX", aif2tx_enum); + +static const char *inmux_text[] = { + "ADC", "DMIC1", "DMIC2" +}; + +static const struct soc_enum in1_enum = + SOC_ENUM_SINGLE(WM8915_POWER_MANAGEMENT_7, 0, 3, inmux_text); + +static const struct snd_kcontrol_new in1_mux = + SOC_DAPM_ENUM("IN1 Mux", in1_enum); + +static const struct soc_enum in2_enum = + SOC_ENUM_SINGLE(WM8915_POWER_MANAGEMENT_7, 4, 3, inmux_text); + +static const struct snd_kcontrol_new in2_mux = + SOC_DAPM_ENUM("IN2 Mux", in2_enum); + +static const struct snd_kcontrol_new dac2r_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8915_DAC2_RIGHT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8915_DAC2_RIGHT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("DSP2 Switch", WM8915_DAC2_RIGHT_MIXER_ROUTING, 1, 1, 0), +SOC_DAPM_SINGLE("DSP1 Switch", WM8915_DAC2_RIGHT_MIXER_ROUTING, 0, 1, 0), +}; + +static const struct snd_kcontrol_new dac2l_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8915_DAC2_LEFT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8915_DAC2_LEFT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("DSP2 Switch", WM8915_DAC2_LEFT_MIXER_ROUTING, 1, 1, 0), +SOC_DAPM_SINGLE("DSP1 Switch", WM8915_DAC2_LEFT_MIXER_ROUTING, 0, 1, 0), +}; + +static const struct snd_kcontrol_new dac1r_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8915_DAC1_RIGHT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8915_DAC1_RIGHT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("DSP2 Switch", WM8915_DAC1_RIGHT_MIXER_ROUTING, 1, 1, 0), +SOC_DAPM_SINGLE("DSP1 Switch", WM8915_DAC1_RIGHT_MIXER_ROUTING, 0, 1, 0), +}; + +static const struct snd_kcontrol_new dac1l_mix[] = { +SOC_DAPM_SINGLE("Right Sidetone Switch", WM8915_DAC1_LEFT_MIXER_ROUTING, + 5, 1, 0), +SOC_DAPM_SINGLE("Left Sidetone Switch", WM8915_DAC1_LEFT_MIXER_ROUTING, + 4, 1, 0), +SOC_DAPM_SINGLE("DSP2 Switch", WM8915_DAC1_LEFT_MIXER_ROUTING, 1, 1, 0), +SOC_DAPM_SINGLE("DSP1 Switch", WM8915_DAC1_LEFT_MIXER_ROUTING, 0, 1, 0), +}; + +static const struct snd_kcontrol_new dsp1txl[] = { +SOC_DAPM_SINGLE("IN1 Switch", WM8915_DSP1_TX_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8915_DSP1_TX_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new dsp1txr[] = { +SOC_DAPM_SINGLE("IN1 Switch", WM8915_DSP1_TX_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8915_DSP1_TX_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new dsp2txl[] = { +SOC_DAPM_SINGLE("IN1 Switch", WM8915_DSP2_TX_LEFT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8915_DSP2_TX_LEFT_MIXER_ROUTING, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new dsp2txr[] = { +SOC_DAPM_SINGLE("IN1 Switch", WM8915_DSP2_TX_RIGHT_MIXER_ROUTING, + 1, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8915_DSP2_TX_RIGHT_MIXER_ROUTING, + 0, 1, 0), +}; + + +static const struct snd_soc_dapm_widget wm8915_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN1LN"), +SND_SOC_DAPM_INPUT("IN1LP"), +SND_SOC_DAPM_INPUT("IN1RN"), +SND_SOC_DAPM_INPUT("IN1RP"), + +SND_SOC_DAPM_INPUT("IN2LN"), +SND_SOC_DAPM_INPUT("IN2LP"), +SND_SOC_DAPM_INPUT("IN2RN"), +SND_SOC_DAPM_INPUT("IN2RP"), + +SND_SOC_DAPM_INPUT("DMIC1DAT"), +SND_SOC_DAPM_INPUT("DMIC2DAT"), + +SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8915_AIF_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8915_CLOCKING_1, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8915_CLOCKING_1, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8915_CHARGE_PUMP_1, 15, 0, cp_event, + SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("LDO2", WM8915_POWER_MANAGEMENT_2, 1, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("MICB2", WM8915_POWER_MANAGEMENT_1, 9, 0), +SND_SOC_DAPM_MICBIAS("MICB1", WM8915_POWER_MANAGEMENT_1, 8, 0), + +SND_SOC_DAPM_PGA("IN1L PGA", WM8915_POWER_MANAGEMENT_2, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("IN1R PGA", WM8915_POWER_MANAGEMENT_2, 4, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + +SND_SOC_DAPM_MUX("IN1 Mux", SND_SOC_NOPM, 0, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN2 Mux", SND_SOC_NOPM, 0, 0, &in2_mux), + +SND_SOC_DAPM_PGA("IN1L", WM8915_POWER_MANAGEMENT_7, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("IN1R", WM8915_POWER_MANAGEMENT_7, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("IN2L", WM8915_POWER_MANAGEMENT_7, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("IN2R", WM8915_POWER_MANAGEMENT_7, 7, 0, NULL, 0), + +/* FIXME - these need to be concentrator widgets */ +SND_SOC_DAPM_SUPPLY("DMIC2", WM8915_POWER_MANAGEMENT_7, 9, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DMIC1", WM8915_POWER_MANAGEMENT_7, 8, 0, NULL, 0), + +SND_SOC_DAPM_ADC("DMIC2L", NULL, WM8915_POWER_MANAGEMENT_3, 5, 0), +SND_SOC_DAPM_ADC("DMIC2R", NULL, WM8915_POWER_MANAGEMENT_3, 4, 0), +SND_SOC_DAPM_ADC("DMIC1L", NULL, WM8915_POWER_MANAGEMENT_3, 3, 0), +SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8915_POWER_MANAGEMENT_3, 2, 0), + +SND_SOC_DAPM_ADC("ADCL", NULL, WM8915_POWER_MANAGEMENT_3, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, WM8915_POWER_MANAGEMENT_3, 0, 0), + +SND_SOC_DAPM_MUX("Left Sidetone", SND_SOC_NOPM, 0, 0, &left_sidetone), +SND_SOC_DAPM_MUX("Right Sidetone", SND_SOC_NOPM, 0, 0, &right_sidetone), + +SND_SOC_DAPM_AIF_IN("DSP2RXL", NULL, 0, WM8915_POWER_MANAGEMENT_3, 11, 0), +SND_SOC_DAPM_AIF_IN("DSP2RXR", NULL, 1, WM8915_POWER_MANAGEMENT_3, 10, 0), +SND_SOC_DAPM_AIF_IN("DSP1RXL", NULL, 0, WM8915_POWER_MANAGEMENT_3, 9, 0), +SND_SOC_DAPM_AIF_IN("DSP1RXR", NULL, 1, WM8915_POWER_MANAGEMENT_3, 8, 0), + +SND_SOC_DAPM_MIXER("DSP2TXL", WM8915_POWER_MANAGEMENT_5, 11, 0, + dsp2txl, ARRAY_SIZE(dsp2txl)), +SND_SOC_DAPM_MIXER("DSP2TXR", WM8915_POWER_MANAGEMENT_5, 10, 0, + dsp2txr, ARRAY_SIZE(dsp2txr)), +SND_SOC_DAPM_MIXER("DSP1TXL", WM8915_POWER_MANAGEMENT_5, 9, 0, + dsp1txl, ARRAY_SIZE(dsp1txl)), +SND_SOC_DAPM_MIXER("DSP1TXR", WM8915_POWER_MANAGEMENT_5, 8, 0, + dsp1txr, ARRAY_SIZE(dsp1txr)), + +SND_SOC_DAPM_MIXER("DAC2L Mixer", SND_SOC_NOPM, 0, 0, + dac2l_mix, ARRAY_SIZE(dac2l_mix)), +SND_SOC_DAPM_MIXER("DAC2R Mixer", SND_SOC_NOPM, 0, 0, + dac2r_mix, ARRAY_SIZE(dac2r_mix)), +SND_SOC_DAPM_MIXER("DAC1L Mixer", SND_SOC_NOPM, 0, 0, + dac1l_mix, ARRAY_SIZE(dac1l_mix)), +SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, + dac1r_mix, ARRAY_SIZE(dac1r_mix)), + +SND_SOC_DAPM_DAC("DAC2L", NULL, WM8915_POWER_MANAGEMENT_5, 3, 0), +SND_SOC_DAPM_DAC("DAC2R", NULL, WM8915_POWER_MANAGEMENT_5, 2, 0), +SND_SOC_DAPM_DAC("DAC1L", NULL, WM8915_POWER_MANAGEMENT_5, 1, 0), +SND_SOC_DAPM_DAC("DAC1R", NULL, WM8915_POWER_MANAGEMENT_5, 0, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 1, + WM8915_POWER_MANAGEMENT_4, 9, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 2, + WM8915_POWER_MANAGEMENT_4, 8, 0), + +SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 1, + WM8915_POWER_MANAGEMENT_6, 9, 0), +SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 2, + WM8915_POWER_MANAGEMENT_6, 8, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5, + WM8915_POWER_MANAGEMENT_4, 5, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", "AIF1 Playback", 4, + WM8915_POWER_MANAGEMENT_4, 4, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", "AIF1 Playback", 3, + WM8915_POWER_MANAGEMENT_4, 3, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", "AIF1 Playback", 2, + WM8915_POWER_MANAGEMENT_4, 2, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX1", "AIF1 Playback", 1, + WM8915_POWER_MANAGEMENT_4, 1, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX0", "AIF1 Playback", 0, + WM8915_POWER_MANAGEMENT_4, 0, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX5", "AIF1 Capture", 5, + WM8915_POWER_MANAGEMENT_6, 5, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", "AIF1 Capture", 4, + WM8915_POWER_MANAGEMENT_6, 4, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", "AIF1 Capture", 3, + WM8915_POWER_MANAGEMENT_6, 3, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", "AIF1 Capture", 2, + WM8915_POWER_MANAGEMENT_6, 2, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX1", "AIF1 Capture", 1, + WM8915_POWER_MANAGEMENT_6, 1, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX0", "AIF1 Capture", 0, + WM8915_POWER_MANAGEMENT_6, 0, 0), + +/* We route as stereo pairs so define some dummy widgets to squash + * things down for now. RXA = 0,1, RXB = 2,3 and so on */ +SND_SOC_DAPM_PGA("AIF1RXA", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("AIF1RXB", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("AIF1RXC", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("AIF2RX", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("DSP2TX", SND_SOC_NOPM, 0, 0, NULL, 0), + +SND_SOC_DAPM_MUX("DSP1RX", SND_SOC_NOPM, 0, 0, &dsp1rx), +SND_SOC_DAPM_MUX("DSP2RX", SND_SOC_NOPM, 0, 0, &dsp2rx), +SND_SOC_DAPM_MUX("AIF2TX", SND_SOC_NOPM, 0, 0, &aif2tx), + +SND_SOC_DAPM_MUX("SPKL", SND_SOC_NOPM, 0, 0, &spkl_mux), +SND_SOC_DAPM_MUX("SPKR", SND_SOC_NOPM, 0, 0, &spkr_mux), +SND_SOC_DAPM_PGA("SPKL PGA", WM8915_LEFT_PDM_SPEAKER, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("SPKR PGA", WM8915_RIGHT_PDM_SPEAKER, 4, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("HPOUT2L PGA", 0, WM8915_POWER_MANAGEMENT_1, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT2L_DLY", 1, WM8915_ANALOGUE_HP_2, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT2L_DCS", 2, WM8915_DC_SERVO_1, 2, 0, dcs_start, + SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_S("HPOUT2L_OUTP", 3, WM8915_ANALOGUE_HP_2, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT2L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2L, 0, + rmv_short_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_PGA_S("HPOUT2R PGA", 0, WM8915_POWER_MANAGEMENT_1, 6, 0,NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT2R_DLY", 1, WM8915_ANALOGUE_HP_2, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT2R_DCS", 2, WM8915_DC_SERVO_1, 3, 0, dcs_start, + SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_S("HPOUT2R_OUTP", 3, WM8915_ANALOGUE_HP_2, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT2R_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2R, 0, + rmv_short_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_PGA_S("HPOUT1L PGA", 0, WM8915_POWER_MANAGEMENT_1, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT1L_DLY", 1, WM8915_ANALOGUE_HP_1, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT1L_DCS", 2, WM8915_DC_SERVO_1, 0, 0, dcs_start, + SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_S("HPOUT1L_OUTP", 3, WM8915_ANALOGUE_HP_1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT1L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT1L, 0, + rmv_short_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_PGA_S("HPOUT1R PGA", 0, WM8915_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT1R_DLY", 1, WM8915_ANALOGUE_HP_1, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT1R_DCS", 2, WM8915_DC_SERVO_1, 1, 0, dcs_start, + SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_S("HPOUT1R_OUTP", 3, WM8915_ANALOGUE_HP_1, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPOUT1R_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT1R, 0, + rmv_short_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("SPKDAT"), +}; + +static const struct snd_soc_dapm_route wm8915_dapm_routes[] = { + { "AIFCLK", NULL, "SYSCLK" }, + { "SYSDSPCLK", NULL, "SYSCLK" }, + { "Charge Pump", NULL, "SYSCLK" }, + + { "MICB1", NULL, "LDO2" }, + { "MICB2", NULL, "LDO2" }, + + { "IN1L PGA", NULL, "IN2LN" }, + { "IN1L PGA", NULL, "IN2LP" }, + { "IN1L PGA", NULL, "IN1LN" }, + { "IN1L PGA", NULL, "IN1LP" }, + + { "IN1R PGA", NULL, "IN2RN" }, + { "IN1R PGA", NULL, "IN2RP" }, + { "IN1R PGA", NULL, "IN1RN" }, + { "IN1R PGA", NULL, "IN1RP" }, + + { "ADCL", NULL, "IN1L PGA" }, + + { "ADCR", NULL, "IN1R PGA" }, + + { "DMIC1L", NULL, "DMIC1DAT" }, + { "DMIC1R", NULL, "DMIC1DAT" }, + { "DMIC2L", NULL, "DMIC2DAT" }, + { "DMIC2R", NULL, "DMIC2DAT" }, + + { "DMIC2L", NULL, "DMIC2" }, + { "DMIC2R", NULL, "DMIC2" }, + { "DMIC1L", NULL, "DMIC1" }, + { "DMIC1R", NULL, "DMIC1" }, + + { "ADC", NULL, "ADCL" }, + { "ADC", NULL, "ADCR" }, + + { "IN1 Mux", "ADC", "ADC" }, + { "IN1 Mux", "DMIC1", "DMIC1" }, + { "IN1 Mux", "DMIC2", "DMIC2" }, + + { "IN2 Mux", "ADC", "ADC" }, + { "IN2 Mux", "DMIC1", "DMIC1" }, + { "IN2 Mux", "DMIC2", "DMIC2" }, + + { "Left Sidetone", "IN1", "IN1 Mux" }, + { "Left Sidetone", "IN2", "IN2 Mux" }, + + { "Right Sidetone", "IN1", "IN1 Mux" }, + { "Right Sidetone", "IN2", "IN2 Mux" }, + + { "DSP1TXL", "IN1 Switch", "IN1 Mux" }, + { "DSP1TXR", "IN1 Switch", "IN1 Mux" }, + + { "DSP2TXL", "IN1 Switch", "IN2 Mux" }, + { "DSP2TXR", "IN1 Switch", "IN2 Mux" }, + + { "AIF1TX0", NULL, "DSP1TXL" }, + { "AIF1TX1", NULL, "DSP1TXR" }, + { "AIF1TX2", NULL, "DSP2TXL" }, + { "AIF1TX3", NULL, "DSP2TXR" }, + { "AIF1TX4", NULL, "AIF2RX0" }, + { "AIF1TX5", NULL, "AIF2RX1" }, + + { "AIF1RX0", NULL, "AIFCLK" }, + { "AIF1RX1", NULL, "AIFCLK" }, + { "AIF1RX2", NULL, "AIFCLK" }, + { "AIF1RX3", NULL, "AIFCLK" }, + { "AIF1RX4", NULL, "AIFCLK" }, + { "AIF1RX5", NULL, "AIFCLK" }, + + { "AIF2RX0", NULL, "AIFCLK" }, + { "AIF2RX1", NULL, "AIFCLK" }, + + { "DSP1RXL", NULL, "SYSDSPCLK" }, + { "DSP1RXR", NULL, "SYSDSPCLK" }, + { "DSP2RXL", NULL, "SYSDSPCLK" }, + { "DSP2RXR", NULL, "SYSDSPCLK" }, + { "DSP1TXL", NULL, "SYSDSPCLK" }, + { "DSP1TXR", NULL, "SYSDSPCLK" }, + { "DSP2TXL", NULL, "SYSDSPCLK" }, + { "DSP2TXR", NULL, "SYSDSPCLK" }, + + { "AIF1RXA", NULL, "AIF1RX0" }, + { "AIF1RXA", NULL, "AIF1RX1" }, + { "AIF1RXB", NULL, "AIF1RX2" }, + { "AIF1RXB", NULL, "AIF1RX3" }, + { "AIF1RXC", NULL, "AIF1RX4" }, + { "AIF1RXC", NULL, "AIF1RX5" }, + + { "AIF2RX", NULL, "AIF2RX0" }, + { "AIF2RX", NULL, "AIF2RX1" }, + + { "AIF2TX", "DSP2", "DSP2TX" }, + { "AIF2TX", "DSP1", "DSP1RX" }, + { "AIF2TX", "AIF1", "AIF1RXC" }, + + { "DSP1RXL", NULL, "DSP1RX" }, + { "DSP1RXR", NULL, "DSP1RX" }, + { "DSP2RXL", NULL, "DSP2RX" }, + { "DSP2RXR", NULL, "DSP2RX" }, + + { "DSP2TX", NULL, "DSP2TXL" }, + { "DSP2TX", NULL, "DSP2TXR" }, + + { "DSP1RX", "AIF1", "AIF1RXA" }, + { "DSP1RX", "AIF2", "AIF2RX" }, + + { "DSP2RX", "AIF1", "AIF1RXB" }, + { "DSP2RX", "AIF2", "AIF2RX" }, + + { "DAC2L Mixer", "DSP2 Switch", "DSP2RXL" }, + { "DAC2L Mixer", "DSP1 Switch", "DSP1RXL" }, + { "DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "DAC2L Mixer", "Left Sidetone Switch", "Left Sidetone" }, + + { "DAC2R Mixer", "DSP2 Switch", "DSP2RXR" }, + { "DAC2R Mixer", "DSP1 Switch", "DSP1RXR" }, + { "DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, + + { "DAC1L Mixer", "DSP2 Switch", "DSP2RXL" }, + { "DAC1L Mixer", "DSP1 Switch", "DSP1RXL" }, + { "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" }, + + { "DAC1R Mixer", "DSP2 Switch", "DSP2RXR" }, + { "DAC1R Mixer", "DSP1 Switch", "DSP1RXR" }, + { "DAC1R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "DAC1R Mixer", "Left Sidetone Switch", "Left Sidetone" }, + + { "DAC1L", NULL, "DAC1L Mixer" }, + { "DAC1R", NULL, "DAC1R Mixer" }, + { "DAC2L", NULL, "DAC2L Mixer" }, + { "DAC2R", NULL, "DAC2R Mixer" }, + + { "HPOUT2L PGA", NULL, "Charge Pump" }, + { "HPOUT2L PGA", NULL, "DAC2L" }, + { "HPOUT2L_DLY", NULL, "HPOUT2L PGA" }, + { "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" }, + { "HPOUT2L_OUTP", NULL, "HPOUT2L_DCS" }, + { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" }, + + { "HPOUT2R PGA", NULL, "Charge Pump" }, + { "HPOUT2R PGA", NULL, "DAC2R" }, + { "HPOUT2R_DLY", NULL, "HPOUT2R PGA" }, + { "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" }, + { "HPOUT2R_OUTP", NULL, "HPOUT2R_DCS" }, + { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" }, + + { "HPOUT1L PGA", NULL, "Charge Pump" }, + { "HPOUT1L PGA", NULL, "DAC1L" }, + { "HPOUT1L_DLY", NULL, "HPOUT1L PGA" }, + { "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" }, + { "HPOUT1L_OUTP", NULL, "HPOUT1L_DCS" }, + { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" }, + + { "HPOUT1R PGA", NULL, "Charge Pump" }, + { "HPOUT1R PGA", NULL, "DAC1R" }, + { "HPOUT1R_DLY", NULL, "HPOUT1R PGA" }, + { "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" }, + { "HPOUT1R_OUTP", NULL, "HPOUT1R_DCS" }, + { "HPOUT1R_RMV_SHORT", NULL, "HPOUT1R_OUTP" }, + + { "HPOUT2L", NULL, "HPOUT2L_RMV_SHORT" }, + { "HPOUT2R", NULL, "HPOUT2R_RMV_SHORT" }, + { "HPOUT1L", NULL, "HPOUT1L_RMV_SHORT" }, + { "HPOUT1R", NULL, "HPOUT1R_RMV_SHORT" }, + + { "SPKL", "DAC1L", "DAC1L" }, + { "SPKL", "DAC1R", "DAC1R" }, + { "SPKL", "DAC2L", "DAC2L" }, + { "SPKL", "DAC2R", "DAC2R" }, + + { "SPKR", "DAC1L", "DAC1L" }, + { "SPKR", "DAC1R", "DAC1R" }, + { "SPKR", "DAC2L", "DAC2L" }, + { "SPKR", "DAC2R", "DAC2R" }, + + { "SPKL PGA", NULL, "SPKL" }, + { "SPKR PGA", NULL, "SPKR" }, + + { "SPKDAT", NULL, "SPKL PGA" }, + { "SPKDAT", NULL, "SPKR PGA" }, +}; + +static int wm8915_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + /* Due to the sparseness of the register map the compiler + * output from an explicit switch statement ends up being much + * more efficient than a table. + */ + switch (reg) { + case WM8915_SOFTWARE_RESET: + case WM8915_POWER_MANAGEMENT_1: + case WM8915_POWER_MANAGEMENT_2: + case WM8915_POWER_MANAGEMENT_3: + case WM8915_POWER_MANAGEMENT_4: + case WM8915_POWER_MANAGEMENT_5: + case WM8915_POWER_MANAGEMENT_6: + case WM8915_POWER_MANAGEMENT_7: + case WM8915_POWER_MANAGEMENT_8: + case WM8915_LEFT_LINE_INPUT_VOLUME: + case WM8915_RIGHT_LINE_INPUT_VOLUME: + case WM8915_LINE_INPUT_CONTROL: + case WM8915_DAC1_HPOUT1_VOLUME: + case WM8915_DAC2_HPOUT2_VOLUME: + case WM8915_DAC1_LEFT_VOLUME: + case WM8915_DAC1_RIGHT_VOLUME: + case WM8915_DAC2_LEFT_VOLUME: + case WM8915_DAC2_RIGHT_VOLUME: + case WM8915_OUTPUT1_LEFT_VOLUME: + case WM8915_OUTPUT1_RIGHT_VOLUME: + case WM8915_OUTPUT2_LEFT_VOLUME: + case WM8915_OUTPUT2_RIGHT_VOLUME: + case WM8915_MICBIAS_1: + case WM8915_MICBIAS_2: + case WM8915_LDO_1: + case WM8915_LDO_2: + case WM8915_ACCESSORY_DETECT_MODE_1: + case WM8915_ACCESSORY_DETECT_MODE_2: + case WM8915_HEADPHONE_DETECT_1: + case WM8915_HEADPHONE_DETECT_2: + case WM8915_MIC_DETECT_1: + case WM8915_MIC_DETECT_2: + case WM8915_MIC_DETECT_3: + case WM8915_CHARGE_PUMP_1: + case WM8915_CHARGE_PUMP_2: + case WM8915_DC_SERVO_1: + case WM8915_DC_SERVO_2: + case WM8915_DC_SERVO_3: + case WM8915_DC_SERVO_5: + case WM8915_DC_SERVO_6: + case WM8915_DC_SERVO_7: + case WM8915_DC_SERVO_READBACK_0: + case WM8915_ANALOGUE_HP_1: + case WM8915_ANALOGUE_HP_2: + case WM8915_CHIP_REVISION: + case WM8915_CONTROL_INTERFACE_1: + case WM8915_WRITE_SEQUENCER_CTRL_1: + case WM8915_WRITE_SEQUENCER_CTRL_2: + case WM8915_AIF_CLOCKING_1: + case WM8915_AIF_CLOCKING_2: + case WM8915_CLOCKING_1: + case WM8915_CLOCKING_2: + case WM8915_AIF_RATE: + case WM8915_FLL_CONTROL_1: + case WM8915_FLL_CONTROL_2: + case WM8915_FLL_CONTROL_3: + case WM8915_FLL_CONTROL_4: + case WM8915_FLL_CONTROL_5: + case WM8915_FLL_CONTROL_6: + case WM8915_FLL_EFS_1: + case WM8915_FLL_EFS_2: + case WM8915_AIF1_CONTROL: + case WM8915_AIF1_BCLK: + case WM8915_AIF1_TX_LRCLK_1: + case WM8915_AIF1_TX_LRCLK_2: + case WM8915_AIF1_RX_LRCLK_1: + case WM8915_AIF1_RX_LRCLK_2: + case WM8915_AIF1TX_DATA_CONFIGURATION_1: + case WM8915_AIF1TX_DATA_CONFIGURATION_2: + case WM8915_AIF1RX_DATA_CONFIGURATION: + case WM8915_AIF1TX_CHANNEL_0_CONFIGURATION: + case WM8915_AIF1TX_CHANNEL_1_CONFIGURATION: + case WM8915_AIF1TX_CHANNEL_2_CONFIGURATION: + case WM8915_AIF1TX_CHANNEL_3_CONFIGURATION: + case WM8915_AIF1TX_CHANNEL_4_CONFIGURATION: + case WM8915_AIF1TX_CHANNEL_5_CONFIGURATION: + case WM8915_AIF1RX_CHANNEL_0_CONFIGURATION: + case WM8915_AIF1RX_CHANNEL_1_CONFIGURATION: + case WM8915_AIF1RX_CHANNEL_2_CONFIGURATION: + case WM8915_AIF1RX_CHANNEL_3_CONFIGURATION: + case WM8915_AIF1RX_CHANNEL_4_CONFIGURATION: + case WM8915_AIF1RX_CHANNEL_5_CONFIGURATION: + case WM8915_AIF1RX_MONO_CONFIGURATION: + case WM8915_AIF1TX_TEST: + case WM8915_AIF2_CONTROL: + case WM8915_AIF2_BCLK: + case WM8915_AIF2_TX_LRCLK_1: + case WM8915_AIF2_TX_LRCLK_2: + case WM8915_AIF2_RX_LRCLK_1: + case WM8915_AIF2_RX_LRCLK_2: + case WM8915_AIF2TX_DATA_CONFIGURATION_1: + case WM8915_AIF2TX_DATA_CONFIGURATION_2: + case WM8915_AIF2RX_DATA_CONFIGURATION: + case WM8915_AIF2TX_CHANNEL_0_CONFIGURATION: + case WM8915_AIF2TX_CHANNEL_1_CONFIGURATION: + case WM8915_AIF2RX_CHANNEL_0_CONFIGURATION: + case WM8915_AIF2RX_CHANNEL_1_CONFIGURATION: + case WM8915_AIF2RX_MONO_CONFIGURATION: + case WM8915_AIF2TX_TEST: + case WM8915_DSP1_TX_LEFT_VOLUME: + case WM8915_DSP1_TX_RIGHT_VOLUME: + case WM8915_DSP1_RX_LEFT_VOLUME: + case WM8915_DSP1_RX_RIGHT_VOLUME: + case WM8915_DSP1_TX_FILTERS: + case WM8915_DSP1_RX_FILTERS_1: + case WM8915_DSP1_RX_FILTERS_2: + case WM8915_DSP1_DRC_1: + case WM8915_DSP1_DRC_2: + case WM8915_DSP1_DRC_3: + case WM8915_DSP1_DRC_4: + case WM8915_DSP1_DRC_5: + case WM8915_DSP1_RX_EQ_GAINS_1: + case WM8915_DSP1_RX_EQ_GAINS_2: + case WM8915_DSP1_RX_EQ_BAND_1_A: + case WM8915_DSP1_RX_EQ_BAND_1_B: + case WM8915_DSP1_RX_EQ_BAND_1_PG: + case WM8915_DSP1_RX_EQ_BAND_2_A: + case WM8915_DSP1_RX_EQ_BAND_2_B: + case WM8915_DSP1_RX_EQ_BAND_2_C: + case WM8915_DSP1_RX_EQ_BAND_2_PG: + case WM8915_DSP1_RX_EQ_BAND_3_A: + case WM8915_DSP1_RX_EQ_BAND_3_B: + case WM8915_DSP1_RX_EQ_BAND_3_C: + case WM8915_DSP1_RX_EQ_BAND_3_PG: + case WM8915_DSP1_RX_EQ_BAND_4_A: + case WM8915_DSP1_RX_EQ_BAND_4_B: + case WM8915_DSP1_RX_EQ_BAND_4_C: + case WM8915_DSP1_RX_EQ_BAND_4_PG: + case WM8915_DSP1_RX_EQ_BAND_5_A: + case WM8915_DSP1_RX_EQ_BAND_5_B: + case WM8915_DSP1_RX_EQ_BAND_5_PG: + case WM8915_DSP2_TX_LEFT_VOLUME: + case WM8915_DSP2_TX_RIGHT_VOLUME: + case WM8915_DSP2_RX_LEFT_VOLUME: + case WM8915_DSP2_RX_RIGHT_VOLUME: + case WM8915_DSP2_TX_FILTERS: + case WM8915_DSP2_RX_FILTERS_1: + case WM8915_DSP2_RX_FILTERS_2: + case WM8915_DSP2_DRC_1: + case WM8915_DSP2_DRC_2: + case WM8915_DSP2_DRC_3: + case WM8915_DSP2_DRC_4: + case WM8915_DSP2_DRC_5: + case WM8915_DSP2_RX_EQ_GAINS_1: + case WM8915_DSP2_RX_EQ_GAINS_2: + case WM8915_DSP2_RX_EQ_BAND_1_A: + case WM8915_DSP2_RX_EQ_BAND_1_B: + case WM8915_DSP2_RX_EQ_BAND_1_PG: + case WM8915_DSP2_RX_EQ_BAND_2_A: + case WM8915_DSP2_RX_EQ_BAND_2_B: + case WM8915_DSP2_RX_EQ_BAND_2_C: + case WM8915_DSP2_RX_EQ_BAND_2_PG: + case WM8915_DSP2_RX_EQ_BAND_3_A: + case WM8915_DSP2_RX_EQ_BAND_3_B: + case WM8915_DSP2_RX_EQ_BAND_3_C: + case WM8915_DSP2_RX_EQ_BAND_3_PG: + case WM8915_DSP2_RX_EQ_BAND_4_A: + case WM8915_DSP2_RX_EQ_BAND_4_B: + case WM8915_DSP2_RX_EQ_BAND_4_C: + case WM8915_DSP2_RX_EQ_BAND_4_PG: + case WM8915_DSP2_RX_EQ_BAND_5_A: + case WM8915_DSP2_RX_EQ_BAND_5_B: + case WM8915_DSP2_RX_EQ_BAND_5_PG: + case WM8915_DAC1_MIXER_VOLUMES: + case WM8915_DAC1_LEFT_MIXER_ROUTING: + case WM8915_DAC1_RIGHT_MIXER_ROUTING: + case WM8915_DAC2_MIXER_VOLUMES: + case WM8915_DAC2_LEFT_MIXER_ROUTING: + case WM8915_DAC2_RIGHT_MIXER_ROUTING: + case WM8915_DSP1_TX_LEFT_MIXER_ROUTING: + case WM8915_DSP1_TX_RIGHT_MIXER_ROUTING: + case WM8915_DSP2_TX_LEFT_MIXER_ROUTING: + case WM8915_DSP2_TX_RIGHT_MIXER_ROUTING: + case WM8915_DSP_TX_MIXER_SELECT: + case WM8915_DAC_SOFTMUTE: + case WM8915_OVERSAMPLING: + case WM8915_SIDETONE: + case WM8915_GPIO_1: + case WM8915_GPIO_2: + case WM8915_GPIO_3: + case WM8915_GPIO_4: + case WM8915_GPIO_5: + case WM8915_PULL_CONTROL_1: + case WM8915_PULL_CONTROL_2: + case WM8915_INTERRUPT_STATUS_1: + case WM8915_INTERRUPT_STATUS_2: + case WM8915_INTERRUPT_RAW_STATUS_2: + case WM8915_INTERRUPT_STATUS_1_MASK: + case WM8915_INTERRUPT_STATUS_2_MASK: + case WM8915_INTERRUPT_CONTROL: + case WM8915_LEFT_PDM_SPEAKER: + case WM8915_RIGHT_PDM_SPEAKER: + case WM8915_PDM_SPEAKER_MUTE_SEQUENCE: + case WM8915_PDM_SPEAKER_VOLUME: + return 1; + default: + return 0; + } +} + +static int wm8915_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case WM8915_SOFTWARE_RESET: + case WM8915_CHIP_REVISION: + case WM8915_LDO_1: + case WM8915_LDO_2: + case WM8915_INTERRUPT_STATUS_1: + case WM8915_INTERRUPT_STATUS_2: + case WM8915_INTERRUPT_RAW_STATUS_2: + case WM8915_DC_SERVO_READBACK_0: + case WM8915_DC_SERVO_2: + case WM8915_DC_SERVO_6: + case WM8915_DC_SERVO_7: + case WM8915_FLL_CONTROL_6: + case WM8915_MIC_DETECT_3: + case WM8915_HEADPHONE_DETECT_1: + case WM8915_HEADPHONE_DETECT_2: + return 1; + default: + return 0; + } +} + +static int wm8915_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8915_SOFTWARE_RESET, 0x8915); +} + +static int wm8915_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + snd_soc_update_bits(codec, WM8915_POWER_MANAGEMENT_1, + WM8915_BG_ENA, WM8915_BG_ENA); + msleep(2); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8915->supplies), + wm8915->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (wm8915->pdata.ldo_ena >= 0) { + gpio_set_value_cansleep(wm8915->pdata.ldo_ena, + 1); + msleep(5); + } + + codec->cache_only = false; + snd_soc_cache_sync(codec); + } + + snd_soc_update_bits(codec, WM8915_POWER_MANAGEMENT_1, + WM8915_BG_ENA, 0); + break; + + case SND_SOC_BIAS_OFF: + codec->cache_only = true; + if (wm8915->pdata.ldo_ena >= 0) + gpio_set_value_cansleep(wm8915->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm8915->supplies), + wm8915->supplies); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int wm8915_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int aifctrl = 0; + int bclk = 0; + int lrclk_tx = 0; + int lrclk_rx = 0; + int aifctrl_reg, bclk_reg, lrclk_tx_reg, lrclk_rx_reg; + + switch (dai->id) { + case 0: + aifctrl_reg = WM8915_AIF1_CONTROL; + bclk_reg = WM8915_AIF1_BCLK; + lrclk_tx_reg = WM8915_AIF1_TX_LRCLK_2; + lrclk_rx_reg = WM8915_AIF1_RX_LRCLK_2; + break; + case 1: + aifctrl_reg = WM8915_AIF2_CONTROL; + bclk_reg = WM8915_AIF2_BCLK; + lrclk_tx_reg = WM8915_AIF2_TX_LRCLK_2; + lrclk_rx_reg = WM8915_AIF2_RX_LRCLK_2; + break; + default: + BUG(); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= WM8915_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk_tx |= WM8915_AIF1TX_LRCLK_INV; + lrclk_rx |= WM8915_AIF1RX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= WM8915_AIF1_BCLK_INV; + lrclk_tx |= WM8915_AIF1TX_LRCLK_INV; + lrclk_rx |= WM8915_AIF1RX_LRCLK_INV; + break; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk_tx |= WM8915_AIF1TX_LRCLK_MSTR; + lrclk_rx |= WM8915_AIF1RX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= WM8915_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + bclk |= WM8915_AIF1_BCLK_MSTR; + lrclk_tx |= WM8915_AIF1TX_LRCLK_MSTR; + lrclk_rx |= WM8915_AIF1RX_LRCLK_MSTR; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + case SND_SOC_DAIFMT_DSP_B: + aifctrl |= 1; + break; + case SND_SOC_DAIFMT_I2S: + aifctrl |= 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + aifctrl |= 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, aifctrl_reg, WM8915_AIF1_FMT_MASK, aifctrl); + snd_soc_update_bits(codec, bclk_reg, + WM8915_AIF1_BCLK_INV | WM8915_AIF1_BCLK_MSTR, + bclk); + snd_soc_update_bits(codec, lrclk_tx_reg, + WM8915_AIF1TX_LRCLK_INV | + WM8915_AIF1TX_LRCLK_MSTR, + lrclk_tx); + snd_soc_update_bits(codec, lrclk_rx_reg, + WM8915_AIF1RX_LRCLK_INV | + WM8915_AIF1RX_LRCLK_MSTR, + lrclk_rx); + + return 0; +} + +static const int bclk_divs[] = { + 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 +}; + +static const int dsp_divs[] = { + 48000, 32000, 16000, 8000 +}; + +static int wm8915_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int bits, i, bclk_rate, best, cur_val; + int aifdata = 0; + int bclk = 0; + int lrclk = 0; + int dsp = 0; + int aifdata_reg, bclk_reg, lrclk_reg, dsp_shift; + + if (!wm8915->sysclk) { + dev_err(codec->dev, "SYSCLK not configured\n"); + return -EINVAL; + } + + switch (dai->id) { + case 0: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + (snd_soc_read(codec, WM8915_GPIO_1)) & WM8915_GP1_FN_MASK) { + aifdata_reg = WM8915_AIF1RX_DATA_CONFIGURATION; + lrclk_reg = WM8915_AIF1_RX_LRCLK_1; + } else { + aifdata_reg = WM8915_AIF1TX_DATA_CONFIGURATION_1; + lrclk_reg = WM8915_AIF1_TX_LRCLK_1; + } + bclk_reg = WM8915_AIF1_BCLK; + dsp_shift = 0; + break; + case 1: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + (snd_soc_read(codec, WM8915_GPIO_2)) & WM8915_GP2_FN_MASK) { + aifdata_reg = WM8915_AIF2RX_DATA_CONFIGURATION; + lrclk_reg = WM8915_AIF2_RX_LRCLK_1; + } else { + aifdata_reg = WM8915_AIF2TX_DATA_CONFIGURATION_1; + lrclk_reg = WM8915_AIF2_TX_LRCLK_1; + } + bclk_reg = WM8915_AIF2_BCLK; + dsp_shift = WM8915_DSP2_DIV_SHIFT; + break; + default: + BUG(); + return -EINVAL; + } + + bclk_rate = snd_soc_params_to_bclk(params); + if (bclk_rate < 0) { + dev_err(codec->dev, "Unsupported BCLK rate: %d\n", bclk_rate); + return bclk_rate; + } + + /* Needs looking at for TDM */ + bits = snd_pcm_format_width(params_format(params)); + if (bits < 0) + return bits; + aifdata |= (bits << WM8915_AIF1TX_WL_SHIFT) | bits; + + for (i = 0; i < ARRAY_SIZE(dsp_divs); i++) { + if (dsp_divs[i] == params_rate(params)) + break; + } + if (i == ARRAY_SIZE(dsp_divs)) { + dev_err(codec->dev, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + dsp |= i << dsp_shift; + + /* Pick a divisor for BCLK as close as we can get to ideal */ + best = 0; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = (wm8915->sysclk / bclk_divs[i]) - bclk_rate; + if (cur_val < 0) /* BCLK table is sorted */ + break; + best = i; + } + bclk_rate = wm8915->sysclk / bclk_divs[best]; + dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n", + bclk_divs[best], bclk_rate); + bclk |= best; + + lrclk = bclk_rate / params_rate(params); + dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", + lrclk, bclk_rate / lrclk); + + snd_soc_update_bits(codec, aifdata_reg, + WM8915_AIF1TX_WL_MASK | + WM8915_AIF1TX_SLOT_LEN_MASK, + aifdata); + snd_soc_update_bits(codec, bclk_reg, WM8915_AIF1_BCLK_DIV_MASK, bclk); + snd_soc_update_bits(codec, lrclk_reg, WM8915_AIF1RX_RATE_MASK, + lrclk); + snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_2, + WM8915_DSP1_DIV_SHIFT << dsp_shift, dsp); + + wm8915->rx_rate[dai->id] = params_rate(params); + + return 0; +} + +static int wm8915_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int lfclk = 0; + int src; + int old; + + /* Disable SYSCLK while we reconfigure */ + old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1); + snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, + WM8915_SYSCLK_ENA, 0); + + switch (clk_id) { + case WM8915_SYSCLK_MCLK1: + wm8915->sysclk = freq; + src = 0; + break; + case WM8915_SYSCLK_MCLK2: + wm8915->sysclk = freq; + src = 1; + break; + case WM8915_SYSCLK_FLL: + wm8915->sysclk = freq; + src = 2; + break; + default: + dev_err(codec->dev, "Unsupported clock source %d\n", clk_id); + return -EINVAL; + } + + switch (wm8915->sysclk) { + case 6144000: + snd_soc_update_bits(codec, WM8915_AIF_RATE, + WM8915_SYSCLK_RATE, 0); + break; + case 12288000: + snd_soc_update_bits(codec, WM8915_AIF_RATE, + WM8915_SYSCLK_RATE, WM8915_SYSCLK_RATE); + break; + case 32000: + case 32768: + lfclk = WM8915_LFCLK_ENA; + break; + default: + dev_warn(codec->dev, "Unsupported clock rate %dHz\n", + wm8915->sysclk); + return -EINVAL; + } + + snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, + WM8915_SYSCLK_SRC_MASK, + src << WM8915_SYSCLK_SRC_SHIFT); + snd_soc_update_bits(codec, WM8915_CLOCKING_1, WM8915_LFCLK_ENA, lfclk); + snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, + WM8915_SYSCLK_ENA, old); + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_refclk_div; + u16 fll_loop_gain; + u16 fll_ref_freq; + u16 n; + u16 theta; + u16 lambda; +}; + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + unsigned int target; + unsigned int div; + unsigned int fratio, gcd_fll; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + fll_div->fll_refclk_div = 0; + while ((Fref / div) > 13500000) { + div *= 2; + fll_div->fll_refclk_div++; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + pr_debug("FLL Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + if (Fref >= 3000000) + fll_div->fll_loop_gain = 5; + else + fll_div->fll_loop_gain = 0; + + if (Fref >= 48000) + fll_div->fll_ref_freq = 0; + else + fll_div->fll_ref_freq = 1; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 2; + while (Fout * div < 90000000) { + div++; + if (div > 64) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + fll_div->fll_outdiv = div - 1; + + pr_debug("FLL Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + fratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + fll_div->n = target / (fratio * Fref); + + if (target % Fref == 0) { + fll_div->theta = 0; + fll_div->lambda = 0; + } else { + gcd_fll = gcd(target, fratio * Fref); + + fll_div->theta = (target - (fll_div->n * fratio * Fref)) + / gcd_fll; + fll_div->lambda = (fratio * Fref) / gcd_fll; + } + + pr_debug("FLL N=%x THETA=%x LAMBDA=%x\n", + fll_div->n, fll_div->theta, fll_div->lambda); + pr_debug("FLL_FRATIO=%x FLL_OUTDIV=%x FLL_REFCLK_DIV=%x\n", + fll_div->fll_fratio, fll_div->fll_outdiv, + fll_div->fll_refclk_div); + + return 0; +} + +static int wm8915_set_fll(struct snd_soc_dai *dai, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + struct _fll_div fll_div; + unsigned long timeout; + int ret, reg; + + /* Any change? */ + if (source == wm8915->fll_src && Fref == wm8915->fll_fref && + Fout == wm8915->fll_fout) + return 0; + + if (Fout == 0) { + dev_dbg(codec->dev, "FLL disabled\n"); + + wm8915->fll_fref = 0; + wm8915->fll_fout = 0; + + snd_soc_update_bits(codec, WM8915_FLL_CONTROL_1, + WM8915_FLL_ENA, 0); + + return 0; + } + + ret = fll_factors(&fll_div, Fref, Fout); + if (ret != 0) + return ret; + + switch (source) { + case WM8915_FLL_MCLK1: + reg = 0; + break; + case WM8915_FLL_MCLK2: + reg = 1; + case WM8915_FLL_DACLRCLK1: + reg = 2; + break; + case WM8915_FLL_BCLK1: + reg = 3; + break; + default: + dev_err(codec->dev, "Unknown FLL source %d\n", ret); + return -EINVAL; + } + + reg |= fll_div.fll_refclk_div << WM8915_FLL_REFCLK_DIV_SHIFT; + reg |= fll_div.fll_ref_freq << WM8915_FLL_REF_FREQ_SHIFT; + + snd_soc_update_bits(codec, WM8915_FLL_CONTROL_5, + WM8915_FLL_REFCLK_DIV_MASK | WM8915_FLL_REF_FREQ | + WM8915_FLL_REFCLK_SRC_MASK, reg); + + reg = 0; + if (fll_div.theta || fll_div.lambda) + reg |= WM8915_FLL_EFS_ENA | (3 << WM8915_FLL_LFSR_SEL_SHIFT); + else + reg |= 1 << WM8915_FLL_LFSR_SEL_SHIFT; + snd_soc_write(codec, WM8915_FLL_EFS_2, reg); + + snd_soc_update_bits(codec, WM8915_FLL_CONTROL_2, + WM8915_FLL_OUTDIV_MASK | + WM8915_FLL_FRATIO_MASK, + (fll_div.fll_outdiv << WM8915_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio)); + + snd_soc_write(codec, WM8915_FLL_CONTROL_3, fll_div.theta); + + snd_soc_update_bits(codec, WM8915_FLL_CONTROL_4, + WM8915_FLL_N_MASK | WM8915_FLL_LOOP_GAIN_MASK, + (fll_div.n << WM8915_FLL_N_SHIFT) | + fll_div.fll_loop_gain); + + snd_soc_write(codec, WM8915_FLL_EFS_1, fll_div.lambda); + + snd_soc_update_bits(codec, WM8915_FLL_CONTROL_1, + WM8915_FLL_ENA, WM8915_FLL_ENA); + + /* The FLL supports live reconfiguration - kick that in case we were + * already enabled. + */ + snd_soc_write(codec, WM8915_FLL_CONTROL_6, WM8915_FLL_SWITCH_CLK); + + /* Wait for the FLL to lock, using the interrupt if possible */ + if (Fref > 1000000) + timeout = usecs_to_jiffies(300); + else + timeout = msecs_to_jiffies(2); + + wait_for_completion_timeout(&wm8915->fll_lock, timeout); + + dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); + + wm8915->fll_fref = Fref; + wm8915->fll_fout = Fout; + wm8915->fll_src = source; + + return 0; +} + +#ifdef CONFIG_GPIOLIB +static inline struct wm8915_priv *gpio_to_wm8915(struct gpio_chip *chip) +{ + return container_of(chip, struct wm8915_priv, gpio_chip); +} + +static void wm8915_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct wm8915_priv *wm8915 = gpio_to_wm8915(chip); + struct snd_soc_codec *codec = wm8915->codec; + + snd_soc_update_bits(codec, WM8915_GPIO_1 + offset, + WM8915_GP1_LVL, !!value << WM8915_GP1_LVL_SHIFT); +} + +static int wm8915_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct wm8915_priv *wm8915 = gpio_to_wm8915(chip); + struct snd_soc_codec *codec = wm8915->codec; + int val; + + val = (1 << WM8915_GP1_FN_SHIFT) | (!!value << WM8915_GP1_LVL_SHIFT); + + return snd_soc_update_bits(codec, WM8915_GPIO_1 + offset, + WM8915_GP1_FN_MASK | WM8915_GP1_DIR | + WM8915_GP1_LVL, val); +} + +static int wm8915_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct wm8915_priv *wm8915 = gpio_to_wm8915(chip); + struct snd_soc_codec *codec = wm8915->codec; + int ret; + + ret = snd_soc_read(codec, WM8915_GPIO_1 + offset); + if (ret < 0) + return ret; + + return (ret & WM8915_GP1_LVL) != 0; +} + +static int wm8915_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct wm8915_priv *wm8915 = gpio_to_wm8915(chip); + struct snd_soc_codec *codec = wm8915->codec; + + return snd_soc_update_bits(codec, WM8915_GPIO_1 + offset, + WM8915_GP1_FN_MASK | WM8915_GP1_DIR, + (1 << WM8915_GP1_FN_SHIFT) | + (1 << WM8915_GP1_DIR_SHIFT)); +} + +static struct gpio_chip wm8915_template_chip = { + .label = "wm8915", + .owner = THIS_MODULE, + .direction_output = wm8915_gpio_direction_out, + .set = wm8915_gpio_set, + .direction_input = wm8915_gpio_direction_in, + .get = wm8915_gpio_get, + .can_sleep = 1, +}; + +static void wm8915_init_gpio(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int ret; + + wm8915->gpio_chip = wm8915_template_chip; + wm8915->gpio_chip.ngpio = 5; + wm8915->gpio_chip.dev = codec->dev; + + if (wm8915->pdata.gpio_base) + wm8915->gpio_chip.base = wm8915->pdata.gpio_base; + else + wm8915->gpio_chip.base = -1; + + ret = gpiochip_add(&wm8915->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void wm8915_free_gpio(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = gpiochip_remove(&wm8915->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void wm8915_init_gpio(struct snd_soc_codec *codec) +{ +} + +static void wm8915_free_gpio(struct snd_soc_codec *codec) +{ +} +#endif + +/** + * wm8915_detect - Enable default WM8915 jack detection + * + * The WM8915 has advanced accessory detection support for headsets. + * This function provides a default implementation which integrates + * the majority of this functionality with minimal user configuration. + * + * This will detect headset, headphone and short circuit button and + * will also detect inverted microphone ground connections and update + * the polarity of the connections. + */ +int wm8915_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, + wm8915_polarity_fn polarity_cb) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + + wm8915->jack = jack; + wm8915->detecting = true; + wm8915->polarity_cb = polarity_cb; + + if (wm8915->polarity_cb) + wm8915->polarity_cb(codec, 0); + + /* Clear discarge to avoid noise during detection */ + snd_soc_update_bits(codec, WM8915_MICBIAS_1, + WM8915_MICB1_DISCH, 0); + snd_soc_update_bits(codec, WM8915_MICBIAS_2, + WM8915_MICB2_DISCH, 0); + + /* LDO2 powers the microphones, SYSCLK clocks detection */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + + /* We start off just enabling microphone detection - even a + * plain headphone will trigger detection. + */ + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_ENA, WM8915_MICD_ENA); + + /* Slowest detection rate, gives debounce for initial detection */ + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_RATE_MASK, + WM8915_MICD_RATE_MASK); + + /* Enable interrupts and we're off */ + snd_soc_update_bits(codec, WM8915_INTERRUPT_STATUS_2_MASK, + WM8915_IM_MICD_EINT, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(wm8915_detect); + +static void wm8915_micd(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int val, reg; + + val = snd_soc_read(codec, WM8915_MIC_DETECT_3); + + dev_dbg(codec->dev, "Microphone event: %x\n", val); + + if (!(val & WM8915_MICD_VALID)) { + dev_warn(codec->dev, "Microphone detection state invalid\n"); + return; + } + + /* No accessory, reset everything and report removal */ + if (!(val & WM8915_MICD_STS)) { + dev_dbg(codec->dev, "Jack removal detected\n"); + wm8915->jack_mic = false; + wm8915->detecting = true; + snd_soc_jack_report(wm8915->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_RATE_MASK, + WM8915_MICD_RATE_MASK); + return; + } + + /* If the measurement is very high we've got a microphone but + * do a little debounce to account for mechanical issues. + */ + if (val & 0x400) { + dev_dbg(codec->dev, "Microphone detected\n"); + snd_soc_jack_report(wm8915->jack, SND_JACK_HEADSET, + SND_JACK_HEADSET | SND_JACK_BTN_0); + wm8915->jack_mic = true; + wm8915->detecting = false; + } + + /* If we detected a lower impedence during initial startup + * then we probably have the wrong polarity, flip it. Don't + * do this for the lowest impedences to speed up detection of + * plain headphones. + */ + if (wm8915->detecting && (val & 0x3f0)) { + reg = snd_soc_read(codec, WM8915_ACCESSORY_DETECT_MODE_2); + reg ^= WM8915_HPOUT1FB_SRC | WM8915_MICD_SRC | + WM8915_MICD_BIAS_SRC; + snd_soc_update_bits(codec, WM8915_ACCESSORY_DETECT_MODE_2, + WM8915_HPOUT1FB_SRC | WM8915_MICD_SRC | + WM8915_MICD_BIAS_SRC, reg); + + if (wm8915->polarity_cb) + wm8915->polarity_cb(codec, + (reg & WM8915_MICD_SRC) != 0); + + dev_dbg(codec->dev, "Set microphone polarity to %d\n", + (reg & WM8915_MICD_SRC) != 0); + + return; + } + + /* Don't distinguish between buttons, just report any low + * impedence as BTN_0. + */ + if (val & 0x3fc) { + if (wm8915->jack_mic) { + dev_dbg(codec->dev, "Mic button detected\n"); + snd_soc_jack_report(wm8915->jack, + SND_JACK_HEADSET | SND_JACK_BTN_0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + } else { + dev_dbg(codec->dev, "Headphone detected\n"); + snd_soc_jack_report(wm8915->jack, + SND_JACK_HEADPHONE, + SND_JACK_HEADSET | + SND_JACK_BTN_0); + wm8915->detecting = false; + } + } + + /* Increase poll rate to give better responsiveness for buttons */ + if (!wm8915->detecting) + snd_soc_update_bits(codec, WM8915_MIC_DETECT_1, + WM8915_MICD_RATE_MASK, + 5 << WM8915_MICD_RATE_SHIFT); +} + +static irqreturn_t wm8915_irq(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + int irq_val; + + irq_val = snd_soc_read(codec, WM8915_INTERRUPT_STATUS_2); + if (irq_val < 0) { + dev_err(codec->dev, "Failed to read IRQ status: %d\n", + irq_val); + return IRQ_NONE; + } + irq_val &= ~snd_soc_read(codec, WM8915_INTERRUPT_STATUS_2_MASK); + + if (irq_val & (WM8915_DCS_DONE_01_EINT | WM8915_DCS_DONE_23_EINT)) { + dev_dbg(codec->dev, "DC servo IRQ\n"); + complete(&wm8915->dcs_done); + } + + if (irq_val & WM8915_FIFOS_ERR_EINT) + dev_err(codec->dev, "Digital core FIFO error\n"); + + if (irq_val & WM8915_FLL_LOCK_EINT) { + dev_dbg(codec->dev, "FLL locked\n"); + complete(&wm8915->fll_lock); + } + + if (irq_val & WM8915_MICD_EINT) + wm8915_micd(codec); + + if (irq_val) { + snd_soc_write(codec, WM8915_INTERRUPT_STATUS_2, irq_val); + + return IRQ_HANDLED; + } else { + return IRQ_NONE; + } +} + +static void wm8915_retune_mobile_pdata(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + struct wm8915_pdata *pdata = &wm8915->pdata; + + struct snd_kcontrol_new controls[] = { + SOC_ENUM_EXT("DSP1 EQ Mode", + wm8915->retune_mobile_enum, + wm8915_get_retune_mobile_enum, + wm8915_put_retune_mobile_enum), + SOC_ENUM_EXT("DSP2 EQ Mode", + wm8915->retune_mobile_enum, + wm8915_get_retune_mobile_enum, + wm8915_put_retune_mobile_enum), + }; + int ret, i, j; + const char **t; + + /* We need an array of texts for the enum API but the number + * of texts is likely to be less than the number of + * configurations due to the sample rate dependency of the + * configurations. */ + wm8915->num_retune_mobile_texts = 0; + wm8915->retune_mobile_texts = NULL; + for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) { + for (j = 0; j < wm8915->num_retune_mobile_texts; j++) { + if (strcmp(pdata->retune_mobile_cfgs[i].name, + wm8915->retune_mobile_texts[j]) == 0) + break; + } + + if (j != wm8915->num_retune_mobile_texts) + continue; + + /* Expand the array... */ + t = krealloc(wm8915->retune_mobile_texts, + sizeof(char *) * + (wm8915->num_retune_mobile_texts + 1), + GFP_KERNEL); + if (t == NULL) + continue; + + /* ...store the new entry... */ + t[wm8915->num_retune_mobile_texts] = + pdata->retune_mobile_cfgs[i].name; + + /* ...and remember the new version. */ + wm8915->num_retune_mobile_texts++; + wm8915->retune_mobile_texts = t; + } + + dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", + wm8915->num_retune_mobile_texts); + + wm8915->retune_mobile_enum.max = wm8915->num_retune_mobile_texts; + wm8915->retune_mobile_enum.texts = wm8915->retune_mobile_texts; + + ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + if (ret != 0) + dev_err(codec->dev, + "Failed to add ReTune Mobile controls: %d\n", ret); +} + +static int wm8915_probe(struct snd_soc_codec *codec) +{ + int ret; + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = to_i2c_client(codec->dev); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int i, irq_flags; + + wm8915->codec = codec; + + init_completion(&wm8915->dcs_done); + init_completion(&wm8915->fll_lock); + + dapm->idle_bias_off = true; + dapm->bias_level = SND_SOC_BIAS_OFF; + + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) + wm8915->supplies[i].supply = wm8915_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8915->supplies), + wm8915->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + wm8915->disable_nb[0].notifier_call = wm8915_regulator_event_0; + wm8915->disable_nb[1].notifier_call = wm8915_regulator_event_1; + wm8915->disable_nb[2].notifier_call = wm8915_regulator_event_2; + wm8915->disable_nb[3].notifier_call = wm8915_regulator_event_3; + wm8915->disable_nb[4].notifier_call = wm8915_regulator_event_4; + wm8915->disable_nb[5].notifier_call = wm8915_regulator_event_5; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) { + ret = regulator_register_notifier(wm8915->supplies[i].consumer, + &wm8915->disable_nb[i]); + if (ret != 0) { + dev_err(codec->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8915->supplies), + wm8915->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + if (wm8915->pdata.ldo_ena >= 0) { + gpio_set_value_cansleep(wm8915->pdata.ldo_ena, 1); + msleep(5); + } + + ret = snd_soc_read(codec, WM8915_SOFTWARE_RESET); + if (ret < 0) { + dev_err(codec->dev, "Failed to read ID register: %d\n", ret); + goto err_enable; + } + if (ret != 0x8915) { + dev_err(codec->dev, "Device is not a WM8915, ID %x\n", ret); + ret = -EINVAL; + goto err_enable; + } + + ret = snd_soc_read(codec, WM8915_CHIP_REVISION); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_enable; + } + + dev_info(codec->dev, "revision %c\n", + (ret & WM8915_CHIP_REV_MASK) + 'A'); + + if (wm8915->pdata.ldo_ena >= 0) { + gpio_set_value_cansleep(wm8915->pdata.ldo_ena, 0); + } else { + ret = wm8915_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_enable; + } + } + + codec->cache_only = true; + + /* Apply platform data settings */ + snd_soc_update_bits(codec, WM8915_LINE_INPUT_CONTROL, + WM8915_INL_MODE_MASK | WM8915_INR_MODE_MASK, + wm8915->pdata.inl_mode << WM8915_INL_MODE_SHIFT | + wm8915->pdata.inr_mode); + + for (i = 0; i < ARRAY_SIZE(wm8915->pdata.gpio_default); i++) { + if (!wm8915->pdata.gpio_default[i]) + continue; + + snd_soc_write(codec, WM8915_GPIO_1 + i, + wm8915->pdata.gpio_default[i] & 0xffff); + } + + if (wm8915->pdata.spkmute_seq) + snd_soc_update_bits(codec, WM8915_PDM_SPEAKER_MUTE_SEQUENCE, + WM8915_SPK_MUTE_ENDIAN | + WM8915_SPK_MUTE_SEQ1_MASK, + wm8915->pdata.spkmute_seq); + + snd_soc_update_bits(codec, WM8915_ACCESSORY_DETECT_MODE_2, + WM8915_MICD_BIAS_SRC | WM8915_HPOUT1FB_SRC | + WM8915_MICD_SRC, wm8915->pdata.micdet_def); + + /* Latch volume update bits */ + snd_soc_update_bits(codec, WM8915_LEFT_LINE_INPUT_VOLUME, + WM8915_IN1_VU, WM8915_IN1_VU); + snd_soc_update_bits(codec, WM8915_RIGHT_LINE_INPUT_VOLUME, + WM8915_IN1_VU, WM8915_IN1_VU); + + snd_soc_update_bits(codec, WM8915_DAC1_LEFT_VOLUME, + WM8915_DAC1_VU, WM8915_DAC1_VU); + snd_soc_update_bits(codec, WM8915_DAC1_RIGHT_VOLUME, + WM8915_DAC1_VU, WM8915_DAC1_VU); + snd_soc_update_bits(codec, WM8915_DAC2_LEFT_VOLUME, + WM8915_DAC2_VU, WM8915_DAC2_VU); + snd_soc_update_bits(codec, WM8915_DAC2_RIGHT_VOLUME, + WM8915_DAC2_VU, WM8915_DAC2_VU); + + snd_soc_update_bits(codec, WM8915_OUTPUT1_LEFT_VOLUME, + WM8915_DAC1_VU, WM8915_DAC1_VU); + snd_soc_update_bits(codec, WM8915_OUTPUT1_RIGHT_VOLUME, + WM8915_DAC1_VU, WM8915_DAC1_VU); + snd_soc_update_bits(codec, WM8915_OUTPUT2_LEFT_VOLUME, + WM8915_DAC2_VU, WM8915_DAC2_VU); + snd_soc_update_bits(codec, WM8915_OUTPUT2_RIGHT_VOLUME, + WM8915_DAC2_VU, WM8915_DAC2_VU); + + snd_soc_update_bits(codec, WM8915_DSP1_TX_LEFT_VOLUME, + WM8915_DSP1TX_VU, WM8915_DSP1TX_VU); + snd_soc_update_bits(codec, WM8915_DSP1_TX_RIGHT_VOLUME, + WM8915_DSP1TX_VU, WM8915_DSP1TX_VU); + snd_soc_update_bits(codec, WM8915_DSP2_TX_LEFT_VOLUME, + WM8915_DSP2TX_VU, WM8915_DSP2TX_VU); + snd_soc_update_bits(codec, WM8915_DSP2_TX_RIGHT_VOLUME, + WM8915_DSP2TX_VU, WM8915_DSP2TX_VU); + + snd_soc_update_bits(codec, WM8915_DSP1_RX_LEFT_VOLUME, + WM8915_DSP1RX_VU, WM8915_DSP1RX_VU); + snd_soc_update_bits(codec, WM8915_DSP1_RX_RIGHT_VOLUME, + WM8915_DSP1RX_VU, WM8915_DSP1RX_VU); + snd_soc_update_bits(codec, WM8915_DSP2_RX_LEFT_VOLUME, + WM8915_DSP2RX_VU, WM8915_DSP2RX_VU); + snd_soc_update_bits(codec, WM8915_DSP2_RX_RIGHT_VOLUME, + WM8915_DSP2RX_VU, WM8915_DSP2RX_VU); + + /* No support currently for the underclocked TDM modes and + * pick a default TDM layout with each channel pair working with + * slots 0 and 1. */ + snd_soc_update_bits(codec, WM8915_AIF1RX_CHANNEL_0_CONFIGURATION, + WM8915_AIF1RX_CHAN0_SLOTS_MASK | + WM8915_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1RX_CHAN0_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF1RX_CHANNEL_1_CONFIGURATION, + WM8915_AIF1RX_CHAN1_SLOTS_MASK | + WM8915_AIF1RX_CHAN1_START_SLOT_MASK, + 1 << WM8915_AIF1RX_CHAN1_SLOTS_SHIFT | 1); + snd_soc_update_bits(codec, WM8915_AIF1RX_CHANNEL_2_CONFIGURATION, + WM8915_AIF1RX_CHAN2_SLOTS_MASK | + WM8915_AIF1RX_CHAN2_START_SLOT_MASK, + 1 << WM8915_AIF1RX_CHAN2_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF1RX_CHANNEL_3_CONFIGURATION, + WM8915_AIF1RX_CHAN3_SLOTS_MASK | + WM8915_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1RX_CHAN3_SLOTS_SHIFT | 1); + snd_soc_update_bits(codec, WM8915_AIF1RX_CHANNEL_4_CONFIGURATION, + WM8915_AIF1RX_CHAN4_SLOTS_MASK | + WM8915_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1RX_CHAN4_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF1RX_CHANNEL_5_CONFIGURATION, + WM8915_AIF1RX_CHAN5_SLOTS_MASK | + WM8915_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1RX_CHAN5_SLOTS_SHIFT | 1); + + snd_soc_update_bits(codec, WM8915_AIF2RX_CHANNEL_0_CONFIGURATION, + WM8915_AIF2RX_CHAN0_SLOTS_MASK | + WM8915_AIF2RX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF2RX_CHAN0_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF2RX_CHANNEL_1_CONFIGURATION, + WM8915_AIF2RX_CHAN1_SLOTS_MASK | + WM8915_AIF2RX_CHAN1_START_SLOT_MASK, + 1 << WM8915_AIF2RX_CHAN1_SLOTS_SHIFT | 1); + + snd_soc_update_bits(codec, WM8915_AIF1TX_CHANNEL_0_CONFIGURATION, + WM8915_AIF1TX_CHAN0_SLOTS_MASK | + WM8915_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1TX_CHAN0_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8915_AIF1TX_CHAN1_SLOTS_MASK | + WM8915_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + snd_soc_update_bits(codec, WM8915_AIF1TX_CHANNEL_2_CONFIGURATION, + WM8915_AIF1TX_CHAN2_SLOTS_MASK | + WM8915_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1TX_CHAN2_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF1TX_CHANNEL_3_CONFIGURATION, + WM8915_AIF1TX_CHAN3_SLOTS_MASK | + WM8915_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1TX_CHAN3_SLOTS_SHIFT | 1); + snd_soc_update_bits(codec, WM8915_AIF1TX_CHANNEL_4_CONFIGURATION, + WM8915_AIF1TX_CHAN4_SLOTS_MASK | + WM8915_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1TX_CHAN4_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF1TX_CHANNEL_5_CONFIGURATION, + WM8915_AIF1TX_CHAN5_SLOTS_MASK | + WM8915_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF1TX_CHAN5_SLOTS_SHIFT | 1); + + snd_soc_update_bits(codec, WM8915_AIF2TX_CHANNEL_0_CONFIGURATION, + WM8915_AIF2TX_CHAN0_SLOTS_MASK | + WM8915_AIF2TX_CHAN0_START_SLOT_MASK, + 1 << WM8915_AIF2TX_CHAN0_SLOTS_SHIFT | 0); + snd_soc_update_bits(codec, WM8915_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8915_AIF2TX_CHAN1_SLOTS_MASK | + WM8915_AIF2TX_CHAN1_START_SLOT_MASK, + 1 << WM8915_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + + if (wm8915->pdata.num_retune_mobile_cfgs) + wm8915_retune_mobile_pdata(codec); + else + snd_soc_add_controls(codec, wm8915_eq_controls, + ARRAY_SIZE(wm8915_eq_controls)); + + /* If the TX LRCLK pins are not in LRCLK mode configure the + * AIFs to source their clocks from the RX LRCLKs. + */ + if ((snd_soc_read(codec, WM8915_GPIO_1))) + snd_soc_update_bits(codec, WM8915_AIF1_TX_LRCLK_2, + WM8915_AIF1TX_LRCLK_MODE, + WM8915_AIF1TX_LRCLK_MODE); + + if ((snd_soc_read(codec, WM8915_GPIO_2))) + snd_soc_update_bits(codec, WM8915_AIF2_TX_LRCLK_2, + WM8915_AIF2TX_LRCLK_MODE, + WM8915_AIF2TX_LRCLK_MODE); + + regulator_bulk_disable(ARRAY_SIZE(wm8915->supplies), wm8915->supplies); + + wm8915_init_gpio(codec); + + if (i2c->irq) { + if (wm8915->pdata.irq_flags) + irq_flags = wm8915->pdata.irq_flags; + else + irq_flags = IRQF_TRIGGER_LOW; + + irq_flags |= IRQF_ONESHOT; + + ret = request_threaded_irq(i2c->irq, NULL, wm8915_irq, + irq_flags, "wm8915", codec); + if (ret == 0) { + /* Unmask the interrupt */ + snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL, + WM8915_IM_IRQ, 0); + + /* Enable error reporting and DC servo status */ + snd_soc_update_bits(codec, + WM8915_INTERRUPT_STATUS_2_MASK, + WM8915_IM_DCS_DONE_23_EINT | + WM8915_IM_DCS_DONE_01_EINT | + WM8915_IM_FLL_LOCK_EINT | + WM8915_IM_FIFOS_ERR_EINT, + 0); + } else { + dev_err(codec->dev, "Failed to request IRQ: %d\n", + ret); + } + } + + return 0; + +err_enable: + if (wm8915->pdata.ldo_ena >= 0) + gpio_set_value_cansleep(wm8915->pdata.ldo_ena, 0); + + regulator_bulk_disable(ARRAY_SIZE(wm8915->supplies), wm8915->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8915->supplies), wm8915->supplies); +err: + return ret; +} + +static int wm8915_remove(struct snd_soc_codec *codec) +{ + struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = to_i2c_client(codec->dev); + int i; + + snd_soc_update_bits(codec, WM8915_INTERRUPT_CONTROL, + WM8915_IM_IRQ, WM8915_IM_IRQ); + + if (i2c->irq) + free_irq(i2c->irq, codec); + + wm8915_free_gpio(codec); + + for (i = 0; i < ARRAY_SIZE(wm8915->supplies); i++) + regulator_unregister_notifier(wm8915->supplies[i].consumer, + &wm8915->disable_nb[i]); + regulator_bulk_free(ARRAY_SIZE(wm8915->supplies), wm8915->supplies); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_wm8915 = { + .probe = wm8915_probe, + .remove = wm8915_remove, + .set_bias_level = wm8915_set_bias_level, + .seq_notifier = wm8915_seq_notifier, + .reg_cache_size = WM8915_MAX_REGISTER + 1, + .reg_word_size = sizeof(u16), + .reg_cache_default = wm8915_reg, + .volatile_register = wm8915_volatile_register, + .readable_register = wm8915_readable_register, + .compress_type = SND_SOC_RBTREE_COMPRESSION, + .controls = wm8915_snd_controls, + .num_controls = ARRAY_SIZE(wm8915_snd_controls), + .dapm_widgets = wm8915_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8915_dapm_widgets), + .dapm_routes = wm8915_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8915_dapm_routes), +}; + +#define WM8915_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) +#define WM8915_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8915_dai_ops = { + .set_fmt = wm8915_set_fmt, + .hw_params = wm8915_hw_params, + .set_sysclk = wm8915_set_sysclk, + .set_pll = wm8915_set_fll, +}; + +static struct snd_soc_dai_driver wm8915_dai[] = { + { + .name = "wm8915-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 6, + .rates = WM8915_RATES, + .formats = WM8915_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 6, + .rates = WM8915_RATES, + .formats = WM8915_FORMATS, + }, + .ops = &wm8915_dai_ops, + }, + { + .name = "wm8915-aif2", + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8915_RATES, + .formats = WM8915_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8915_RATES, + .formats = WM8915_FORMATS, + }, + .ops = &wm8915_dai_ops, + }, +}; + +static __devinit int wm8915_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8915_priv *wm8915; + int ret; + + wm8915 = kzalloc(sizeof(struct wm8915_priv), GFP_KERNEL); + if (wm8915 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, wm8915); + + if (dev_get_platdata(&i2c->dev)) + memcpy(&wm8915->pdata, dev_get_platdata(&i2c->dev), + sizeof(wm8915->pdata)); + + if (wm8915->pdata.ldo_ena > 0) { + ret = gpio_request_one(wm8915->pdata.ldo_ena, + GPIOF_OUT_INIT_LOW, "WM8915 ENA"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request GPIO %d: %d\n", + wm8915->pdata.ldo_ena, ret); + goto err; + } + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8915, wm8915_dai, + ARRAY_SIZE(wm8915_dai)); + if (ret < 0) + goto err_gpio; + + return ret; + +err_gpio: + if (wm8915->pdata.ldo_ena > 0) + gpio_free(wm8915->pdata.ldo_ena); +err: + kfree(wm8915); + + return ret; +} + +static __devexit int wm8915_i2c_remove(struct i2c_client *client) +{ + struct wm8915_priv *wm8915 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + if (wm8915->pdata.ldo_ena > 0) + gpio_free(wm8915->pdata.ldo_ena); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id wm8915_i2c_id[] = { + { "wm8915", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8915_i2c_id); + +static struct i2c_driver wm8915_i2c_driver = { + .driver = { + .name = "wm8915", + .owner = THIS_MODULE, + }, + .probe = wm8915_i2c_probe, + .remove = __devexit_p(wm8915_i2c_remove), + .id_table = wm8915_i2c_id, +}; + +static int __init wm8915_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8915_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8915 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm8915_modinit); + +static void __exit wm8915_exit(void) +{ + i2c_del_driver(&wm8915_i2c_driver); +} +module_exit(wm8915_exit); + +MODULE_DESCRIPTION("ASoC WM8915 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8915.h b/sound/soc/codecs/wm8915.h new file mode 100644 index 00000000000..200ffd7bf95 --- /dev/null +++ b/sound/soc/codecs/wm8915.h @@ -0,0 +1,3717 @@ +/* + * wm8915.h - WM8915 audio codec interface + * + * Copyright 2011 Wolfson Microelectronics PLC. + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _WM8915_H +#define _WM8915_H + +#define WM8915_SYSCLK_MCLK1 1 +#define WM8915_SYSCLK_MCLK2 2 +#define WM8915_SYSCLK_FLL 3 + +#define WM8915_FLL_MCLK1 1 +#define WM8915_FLL_MCLK2 2 +#define WM8915_FLL_DACLRCLK1 3 +#define WM8915_FLL_BCLK1 4 + +typedef void (*wm8915_polarity_fn)(struct snd_soc_codec *codec, int polarity); + +int wm8915_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, + wm8915_polarity_fn polarity_cb); + +/* + * Register values. + */ +#define WM8915_SOFTWARE_RESET 0x00 +#define WM8915_POWER_MANAGEMENT_1 0x01 +#define WM8915_POWER_MANAGEMENT_2 0x02 +#define WM8915_POWER_MANAGEMENT_3 0x03 +#define WM8915_POWER_MANAGEMENT_4 0x04 +#define WM8915_POWER_MANAGEMENT_5 0x05 +#define WM8915_POWER_MANAGEMENT_6 0x06 +#define WM8915_POWER_MANAGEMENT_7 0x07 +#define WM8915_POWER_MANAGEMENT_8 0x08 +#define WM8915_LEFT_LINE_INPUT_VOLUME 0x10 +#define WM8915_RIGHT_LINE_INPUT_VOLUME 0x11 +#define WM8915_LINE_INPUT_CONTROL 0x12 +#define WM8915_DAC1_HPOUT1_VOLUME 0x15 +#define WM8915_DAC2_HPOUT2_VOLUME 0x16 +#define WM8915_DAC1_LEFT_VOLUME 0x18 +#define WM8915_DAC1_RIGHT_VOLUME 0x19 +#define WM8915_DAC2_LEFT_VOLUME 0x1A +#define WM8915_DAC2_RIGHT_VOLUME 0x1B +#define WM8915_OUTPUT1_LEFT_VOLUME 0x1C +#define WM8915_OUTPUT1_RIGHT_VOLUME 0x1D +#define WM8915_OUTPUT2_LEFT_VOLUME 0x1E +#define WM8915_OUTPUT2_RIGHT_VOLUME 0x1F +#define WM8915_MICBIAS_1 0x20 +#define WM8915_MICBIAS_2 0x21 +#define WM8915_LDO_1 0x28 +#define WM8915_LDO_2 0x29 +#define WM8915_ACCESSORY_DETECT_MODE_1 0x30 +#define WM8915_ACCESSORY_DETECT_MODE_2 0x31 +#define WM8915_HEADPHONE_DETECT_1 0x34 +#define WM8915_HEADPHONE_DETECT_2 0x35 +#define WM8915_MIC_DETECT_1 0x38 +#define WM8915_MIC_DETECT_2 0x39 +#define WM8915_MIC_DETECT_3 0x3A +#define WM8915_CHARGE_PUMP_1 0x40 +#define WM8915_CHARGE_PUMP_2 0x41 +#define WM8915_DC_SERVO_1 0x50 +#define WM8915_DC_SERVO_2 0x51 +#define WM8915_DC_SERVO_3 0x52 +#define WM8915_DC_SERVO_5 0x54 +#define WM8915_DC_SERVO_6 0x55 +#define WM8915_DC_SERVO_7 0x56 +#define WM8915_DC_SERVO_READBACK_0 0x57 +#define WM8915_ANALOGUE_HP_1 0x60 +#define WM8915_ANALOGUE_HP_2 0x61 +#define WM8915_CHIP_REVISION 0x100 +#define WM8915_CONTROL_INTERFACE_1 0x101 +#define WM8915_WRITE_SEQUENCER_CTRL_1 0x110 +#define WM8915_WRITE_SEQUENCER_CTRL_2 0x111 +#define WM8915_AIF_CLOCKING_1 0x200 +#define WM8915_AIF_CLOCKING_2 0x201 +#define WM8915_CLOCKING_1 0x208 +#define WM8915_CLOCKING_2 0x209 +#define WM8915_AIF_RATE 0x210 +#define WM8915_FLL_CONTROL_1 0x220 +#define WM8915_FLL_CONTROL_2 0x221 +#define WM8915_FLL_CONTROL_3 0x222 +#define WM8915_FLL_CONTROL_4 0x223 +#define WM8915_FLL_CONTROL_5 0x224 +#define WM8915_FLL_CONTROL_6 0x225 +#define WM8915_FLL_EFS_1 0x226 +#define WM8915_FLL_EFS_2 0x227 +#define WM8915_AIF1_CONTROL 0x300 +#define WM8915_AIF1_BCLK 0x301 +#define WM8915_AIF1_TX_LRCLK_1 0x302 +#define WM8915_AIF1_TX_LRCLK_2 0x303 +#define WM8915_AIF1_RX_LRCLK_1 0x304 +#define WM8915_AIF1_RX_LRCLK_2 0x305 +#define WM8915_AIF1TX_DATA_CONFIGURATION_1 0x306 +#define WM8915_AIF1TX_DATA_CONFIGURATION_2 0x307 +#define WM8915_AIF1RX_DATA_CONFIGURATION 0x308 +#define WM8915_AIF1TX_CHANNEL_0_CONFIGURATION 0x309 +#define WM8915_AIF1TX_CHANNEL_1_CONFIGURATION 0x30A +#define WM8915_AIF1TX_CHANNEL_2_CONFIGURATION 0x30B +#define WM8915_AIF1TX_CHANNEL_3_CONFIGURATION 0x30C +#define WM8915_AIF1TX_CHANNEL_4_CONFIGURATION 0x30D +#define WM8915_AIF1TX_CHANNEL_5_CONFIGURATION 0x30E +#define WM8915_AIF1RX_CHANNEL_0_CONFIGURATION 0x30F +#define WM8915_AIF1RX_CHANNEL_1_CONFIGURATION 0x310 +#define WM8915_AIF1RX_CHANNEL_2_CONFIGURATION 0x311 +#define WM8915_AIF1RX_CHANNEL_3_CONFIGURATION 0x312 +#define WM8915_AIF1RX_CHANNEL_4_CONFIGURATION 0x313 +#define WM8915_AIF1RX_CHANNEL_5_CONFIGURATION 0x314 +#define WM8915_AIF1RX_MONO_CONFIGURATION 0x315 +#define WM8915_AIF1TX_TEST 0x31A +#define WM8915_AIF2_CONTROL 0x320 +#define WM8915_AIF2_BCLK 0x321 +#define WM8915_AIF2_TX_LRCLK_1 0x322 +#define WM8915_AIF2_TX_LRCLK_2 0x323 +#define WM8915_AIF2_RX_LRCLK_1 0x324 +#define WM8915_AIF2_RX_LRCLK_2 0x325 +#define WM8915_AIF2TX_DATA_CONFIGURATION_1 0x326 +#define WM8915_AIF2TX_DATA_CONFIGURATION_2 0x327 +#define WM8915_AIF2RX_DATA_CONFIGURATION 0x328 +#define WM8915_AIF2TX_CHANNEL_0_CONFIGURATION 0x329 +#define WM8915_AIF2TX_CHANNEL_1_CONFIGURATION 0x32A +#define WM8915_AIF2RX_CHANNEL_0_CONFIGURATION 0x32B +#define WM8915_AIF2RX_CHANNEL_1_CONFIGURATION 0x32C +#define WM8915_AIF2RX_MONO_CONFIGURATION 0x32D +#define WM8915_AIF2TX_TEST 0x32F +#define WM8915_DSP1_TX_LEFT_VOLUME 0x400 +#define WM8915_DSP1_TX_RIGHT_VOLUME 0x401 +#define WM8915_DSP1_RX_LEFT_VOLUME 0x402 +#define WM8915_DSP1_RX_RIGHT_VOLUME 0x403 +#define WM8915_DSP1_TX_FILTERS 0x410 +#define WM8915_DSP1_RX_FILTERS_1 0x420 +#define WM8915_DSP1_RX_FILTERS_2 0x421 +#define WM8915_DSP1_DRC_1 0x440 +#define WM8915_DSP1_DRC_2 0x441 +#define WM8915_DSP1_DRC_3 0x442 +#define WM8915_DSP1_DRC_4 0x443 +#define WM8915_DSP1_DRC_5 0x444 +#define WM8915_DSP1_RX_EQ_GAINS_1 0x480 +#define WM8915_DSP1_RX_EQ_GAINS_2 0x481 +#define WM8915_DSP1_RX_EQ_BAND_1_A 0x482 +#define WM8915_DSP1_RX_EQ_BAND_1_B 0x483 +#define WM8915_DSP1_RX_EQ_BAND_1_PG 0x484 +#define WM8915_DSP1_RX_EQ_BAND_2_A 0x485 +#define WM8915_DSP1_RX_EQ_BAND_2_B 0x486 +#define WM8915_DSP1_RX_EQ_BAND_2_C 0x487 +#define WM8915_DSP1_RX_EQ_BAND_2_PG 0x488 +#define WM8915_DSP1_RX_EQ_BAND_3_A 0x489 +#define WM8915_DSP1_RX_EQ_BAND_3_B 0x48A +#define WM8915_DSP1_RX_EQ_BAND_3_C 0x48B +#define WM8915_DSP1_RX_EQ_BAND_3_PG 0x48C +#define WM8915_DSP1_RX_EQ_BAND_4_A 0x48D +#define WM8915_DSP1_RX_EQ_BAND_4_B 0x48E +#define WM8915_DSP1_RX_EQ_BAND_4_C 0x48F +#define WM8915_DSP1_RX_EQ_BAND_4_PG 0x490 +#define WM8915_DSP1_RX_EQ_BAND_5_A 0x491 +#define WM8915_DSP1_RX_EQ_BAND_5_B 0x492 +#define WM8915_DSP1_RX_EQ_BAND_5_PG 0x493 +#define WM8915_DSP2_TX_LEFT_VOLUME 0x500 +#define WM8915_DSP2_TX_RIGHT_VOLUME 0x501 +#define WM8915_DSP2_RX_LEFT_VOLUME 0x502 +#define WM8915_DSP2_RX_RIGHT_VOLUME 0x503 +#define WM8915_DSP2_TX_FILTERS 0x510 +#define WM8915_DSP2_RX_FILTERS_1 0x520 +#define WM8915_DSP2_RX_FILTERS_2 0x521 +#define WM8915_DSP2_DRC_1 0x540 +#define WM8915_DSP2_DRC_2 0x541 +#define WM8915_DSP2_DRC_3 0x542 +#define WM8915_DSP2_DRC_4 0x543 +#define WM8915_DSP2_DRC_5 0x544 +#define WM8915_DSP2_RX_EQ_GAINS_1 0x580 +#define WM8915_DSP2_RX_EQ_GAINS_2 0x581 +#define WM8915_DSP2_RX_EQ_BAND_1_A 0x582 +#define WM8915_DSP2_RX_EQ_BAND_1_B 0x583 +#define WM8915_DSP2_RX_EQ_BAND_1_PG 0x584 +#define WM8915_DSP2_RX_EQ_BAND_2_A 0x585 +#define WM8915_DSP2_RX_EQ_BAND_2_B 0x586 +#define WM8915_DSP2_RX_EQ_BAND_2_C 0x587 +#define WM8915_DSP2_RX_EQ_BAND_2_PG 0x588 +#define WM8915_DSP2_RX_EQ_BAND_3_A 0x589 +#define WM8915_DSP2_RX_EQ_BAND_3_B 0x58A +#define WM8915_DSP2_RX_EQ_BAND_3_C 0x58B +#define WM8915_DSP2_RX_EQ_BAND_3_PG 0x58C +#define WM8915_DSP2_RX_EQ_BAND_4_A 0x58D +#define WM8915_DSP2_RX_EQ_BAND_4_B 0x58E +#define WM8915_DSP2_RX_EQ_BAND_4_C 0x58F +#define WM8915_DSP2_RX_EQ_BAND_4_PG 0x590 +#define WM8915_DSP2_RX_EQ_BAND_5_A 0x591 +#define WM8915_DSP2_RX_EQ_BAND_5_B 0x592 +#define WM8915_DSP2_RX_EQ_BAND_5_PG 0x593 +#define WM8915_DAC1_MIXER_VOLUMES 0x600 +#define WM8915_DAC1_LEFT_MIXER_ROUTING 0x601 +#define WM8915_DAC1_RIGHT_MIXER_ROUTING 0x602 +#define WM8915_DAC2_MIXER_VOLUMES 0x603 +#define WM8915_DAC2_LEFT_MIXER_ROUTING 0x604 +#define WM8915_DAC2_RIGHT_MIXER_ROUTING 0x605 +#define WM8915_DSP1_TX_LEFT_MIXER_ROUTING 0x606 +#define WM8915_DSP1_TX_RIGHT_MIXER_ROUTING 0x607 +#define WM8915_DSP2_TX_LEFT_MIXER_ROUTING 0x608 +#define WM8915_DSP2_TX_RIGHT_MIXER_ROUTING 0x609 +#define WM8915_DSP_TX_MIXER_SELECT 0x60A +#define WM8915_DAC_SOFTMUTE 0x610 +#define WM8915_OVERSAMPLING 0x620 +#define WM8915_SIDETONE 0x621 +#define WM8915_GPIO_1 0x700 +#define WM8915_GPIO_2 0x701 +#define WM8915_GPIO_3 0x702 +#define WM8915_GPIO_4 0x703 +#define WM8915_GPIO_5 0x704 +#define WM8915_PULL_CONTROL_1 0x720 +#define WM8915_PULL_CONTROL_2 0x721 +#define WM8915_INTERRUPT_STATUS_1 0x730 +#define WM8915_INTERRUPT_STATUS_2 0x731 +#define WM8915_INTERRUPT_RAW_STATUS_2 0x732 +#define WM8915_INTERRUPT_STATUS_1_MASK 0x738 +#define WM8915_INTERRUPT_STATUS_2_MASK 0x739 +#define WM8915_INTERRUPT_CONTROL 0x740 +#define WM8915_LEFT_PDM_SPEAKER 0x800 +#define WM8915_RIGHT_PDM_SPEAKER 0x801 +#define WM8915_PDM_SPEAKER_MUTE_SEQUENCE 0x802 +#define WM8915_PDM_SPEAKER_VOLUME 0x803 +#define WM8915_WRITE_SEQUENCER_0 0x3000 +#define WM8915_WRITE_SEQUENCER_1 0x3001 +#define WM8915_WRITE_SEQUENCER_2 0x3002 +#define WM8915_WRITE_SEQUENCER_3 0x3003 +#define WM8915_WRITE_SEQUENCER_4 0x3004 +#define WM8915_WRITE_SEQUENCER_5 0x3005 +#define WM8915_WRITE_SEQUENCER_6 0x3006 +#define WM8915_WRITE_SEQUENCER_7 0x3007 +#define WM8915_WRITE_SEQUENCER_8 0x3008 +#define WM8915_WRITE_SEQUENCER_9 0x3009 +#define WM8915_WRITE_SEQUENCER_10 0x300A +#define WM8915_WRITE_SEQUENCER_11 0x300B +#define WM8915_WRITE_SEQUENCER_12 0x300C +#define WM8915_WRITE_SEQUENCER_13 0x300D +#define WM8915_WRITE_SEQUENCER_14 0x300E +#define WM8915_WRITE_SEQUENCER_15 0x300F +#define WM8915_WRITE_SEQUENCER_16 0x3010 +#define WM8915_WRITE_SEQUENCER_17 0x3011 +#define WM8915_WRITE_SEQUENCER_18 0x3012 +#define WM8915_WRITE_SEQUENCER_19 0x3013 +#define WM8915_WRITE_SEQUENCER_20 0x3014 +#define WM8915_WRITE_SEQUENCER_21 0x3015 +#define WM8915_WRITE_SEQUENCER_22 0x3016 +#define WM8915_WRITE_SEQUENCER_23 0x3017 +#define WM8915_WRITE_SEQUENCER_24 0x3018 +#define WM8915_WRITE_SEQUENCER_25 0x3019 +#define WM8915_WRITE_SEQUENCER_26 0x301A +#define WM8915_WRITE_SEQUENCER_27 0x301B +#define WM8915_WRITE_SEQUENCER_28 0x301C +#define WM8915_WRITE_SEQUENCER_29 0x301D +#define WM8915_WRITE_SEQUENCER_30 0x301E +#define WM8915_WRITE_SEQUENCER_31 0x301F +#define WM8915_WRITE_SEQUENCER_32 0x3020 +#define WM8915_WRITE_SEQUENCER_33 0x3021 +#define WM8915_WRITE_SEQUENCER_34 0x3022 +#define WM8915_WRITE_SEQUENCER_35 0x3023 +#define WM8915_WRITE_SEQUENCER_36 0x3024 +#define WM8915_WRITE_SEQUENCER_37 0x3025 +#define WM8915_WRITE_SEQUENCER_38 0x3026 +#define WM8915_WRITE_SEQUENCER_39 0x3027 +#define WM8915_WRITE_SEQUENCER_40 0x3028 +#define WM8915_WRITE_SEQUENCER_41 0x3029 +#define WM8915_WRITE_SEQUENCER_42 0x302A +#define WM8915_WRITE_SEQUENCER_43 0x302B +#define WM8915_WRITE_SEQUENCER_44 0x302C +#define WM8915_WRITE_SEQUENCER_45 0x302D +#define WM8915_WRITE_SEQUENCER_46 0x302E +#define WM8915_WRITE_SEQUENCER_47 0x302F +#define WM8915_WRITE_SEQUENCER_48 0x3030 +#define WM8915_WRITE_SEQUENCER_49 0x3031 +#define WM8915_WRITE_SEQUENCER_50 0x3032 +#define WM8915_WRITE_SEQUENCER_51 0x3033 +#define WM8915_WRITE_SEQUENCER_52 0x3034 +#define WM8915_WRITE_SEQUENCER_53 0x3035 +#define WM8915_WRITE_SEQUENCER_54 0x3036 +#define WM8915_WRITE_SEQUENCER_55 0x3037 +#define WM8915_WRITE_SEQUENCER_56 0x3038 +#define WM8915_WRITE_SEQUENCER_57 0x3039 +#define WM8915_WRITE_SEQUENCER_58 0x303A +#define WM8915_WRITE_SEQUENCER_59 0x303B +#define WM8915_WRITE_SEQUENCER_60 0x303C +#define WM8915_WRITE_SEQUENCER_61 0x303D +#define WM8915_WRITE_SEQUENCER_62 0x303E +#define WM8915_WRITE_SEQUENCER_63 0x303F +#define WM8915_WRITE_SEQUENCER_64 0x3040 +#define WM8915_WRITE_SEQUENCER_65 0x3041 +#define WM8915_WRITE_SEQUENCER_66 0x3042 +#define WM8915_WRITE_SEQUENCER_67 0x3043 +#define WM8915_WRITE_SEQUENCER_68 0x3044 +#define WM8915_WRITE_SEQUENCER_69 0x3045 +#define WM8915_WRITE_SEQUENCER_70 0x3046 +#define WM8915_WRITE_SEQUENCER_71 0x3047 +#define WM8915_WRITE_SEQUENCER_72 0x3048 +#define WM8915_WRITE_SEQUENCER_73 0x3049 +#define WM8915_WRITE_SEQUENCER_74 0x304A +#define WM8915_WRITE_SEQUENCER_75 0x304B +#define WM8915_WRITE_SEQUENCER_76 0x304C +#define WM8915_WRITE_SEQUENCER_77 0x304D +#define WM8915_WRITE_SEQUENCER_78 0x304E +#define WM8915_WRITE_SEQUENCER_79 0x304F +#define WM8915_WRITE_SEQUENCER_80 0x3050 +#define WM8915_WRITE_SEQUENCER_81 0x3051 +#define WM8915_WRITE_SEQUENCER_82 0x3052 +#define WM8915_WRITE_SEQUENCER_83 0x3053 +#define WM8915_WRITE_SEQUENCER_84 0x3054 +#define WM8915_WRITE_SEQUENCER_85 0x3055 +#define WM8915_WRITE_SEQUENCER_86 0x3056 +#define WM8915_WRITE_SEQUENCER_87 0x3057 +#define WM8915_WRITE_SEQUENCER_88 0x3058 +#define WM8915_WRITE_SEQUENCER_89 0x3059 +#define WM8915_WRITE_SEQUENCER_90 0x305A +#define WM8915_WRITE_SEQUENCER_91 0x305B +#define WM8915_WRITE_SEQUENCER_92 0x305C +#define WM8915_WRITE_SEQUENCER_93 0x305D +#define WM8915_WRITE_SEQUENCER_94 0x305E +#define WM8915_WRITE_SEQUENCER_95 0x305F +#define WM8915_WRITE_SEQUENCER_96 0x3060 +#define WM8915_WRITE_SEQUENCER_97 0x3061 +#define WM8915_WRITE_SEQUENCER_98 0x3062 +#define WM8915_WRITE_SEQUENCER_99 0x3063 +#define WM8915_WRITE_SEQUENCER_100 0x3064 +#define WM8915_WRITE_SEQUENCER_101 0x3065 +#define WM8915_WRITE_SEQUENCER_102 0x3066 +#define WM8915_WRITE_SEQUENCER_103 0x3067 +#define WM8915_WRITE_SEQUENCER_104 0x3068 +#define WM8915_WRITE_SEQUENCER_105 0x3069 +#define WM8915_WRITE_SEQUENCER_106 0x306A +#define WM8915_WRITE_SEQUENCER_107 0x306B +#define WM8915_WRITE_SEQUENCER_108 0x306C +#define WM8915_WRITE_SEQUENCER_109 0x306D +#define WM8915_WRITE_SEQUENCER_110 0x306E +#define WM8915_WRITE_SEQUENCER_111 0x306F +#define WM8915_WRITE_SEQUENCER_112 0x3070 +#define WM8915_WRITE_SEQUENCER_113 0x3071 +#define WM8915_WRITE_SEQUENCER_114 0x3072 +#define WM8915_WRITE_SEQUENCER_115 0x3073 +#define WM8915_WRITE_SEQUENCER_116 0x3074 +#define WM8915_WRITE_SEQUENCER_117 0x3075 +#define WM8915_WRITE_SEQUENCER_118 0x3076 +#define WM8915_WRITE_SEQUENCER_119 0x3077 +#define WM8915_WRITE_SEQUENCER_120 0x3078 +#define WM8915_WRITE_SEQUENCER_121 0x3079 +#define WM8915_WRITE_SEQUENCER_122 0x307A +#define WM8915_WRITE_SEQUENCER_123 0x307B +#define WM8915_WRITE_SEQUENCER_124 0x307C +#define WM8915_WRITE_SEQUENCER_125 0x307D +#define WM8915_WRITE_SEQUENCER_126 0x307E +#define WM8915_WRITE_SEQUENCER_127 0x307F +#define WM8915_WRITE_SEQUENCER_128 0x3080 +#define WM8915_WRITE_SEQUENCER_129 0x3081 +#define WM8915_WRITE_SEQUENCER_130 0x3082 +#define WM8915_WRITE_SEQUENCER_131 0x3083 +#define WM8915_WRITE_SEQUENCER_132 0x3084 +#define WM8915_WRITE_SEQUENCER_133 0x3085 +#define WM8915_WRITE_SEQUENCER_134 0x3086 +#define WM8915_WRITE_SEQUENCER_135 0x3087 +#define WM8915_WRITE_SEQUENCER_136 0x3088 +#define WM8915_WRITE_SEQUENCER_137 0x3089 +#define WM8915_WRITE_SEQUENCER_138 0x308A +#define WM8915_WRITE_SEQUENCER_139 0x308B +#define WM8915_WRITE_SEQUENCER_140 0x308C +#define WM8915_WRITE_SEQUENCER_141 0x308D +#define WM8915_WRITE_SEQUENCER_142 0x308E +#define WM8915_WRITE_SEQUENCER_143 0x308F +#define WM8915_WRITE_SEQUENCER_144 0x3090 +#define WM8915_WRITE_SEQUENCER_145 0x3091 +#define WM8915_WRITE_SEQUENCER_146 0x3092 +#define WM8915_WRITE_SEQUENCER_147 0x3093 +#define WM8915_WRITE_SEQUENCER_148 0x3094 +#define WM8915_WRITE_SEQUENCER_149 0x3095 +#define WM8915_WRITE_SEQUENCER_150 0x3096 +#define WM8915_WRITE_SEQUENCER_151 0x3097 +#define WM8915_WRITE_SEQUENCER_152 0x3098 +#define WM8915_WRITE_SEQUENCER_153 0x3099 +#define WM8915_WRITE_SEQUENCER_154 0x309A +#define WM8915_WRITE_SEQUENCER_155 0x309B +#define WM8915_WRITE_SEQUENCER_156 0x309C +#define WM8915_WRITE_SEQUENCER_157 0x309D +#define WM8915_WRITE_SEQUENCER_158 0x309E +#define WM8915_WRITE_SEQUENCER_159 0x309F +#define WM8915_WRITE_SEQUENCER_160 0x30A0 +#define WM8915_WRITE_SEQUENCER_161 0x30A1 +#define WM8915_WRITE_SEQUENCER_162 0x30A2 +#define WM8915_WRITE_SEQUENCER_163 0x30A3 +#define WM8915_WRITE_SEQUENCER_164 0x30A4 +#define WM8915_WRITE_SEQUENCER_165 0x30A5 +#define WM8915_WRITE_SEQUENCER_166 0x30A6 +#define WM8915_WRITE_SEQUENCER_167 0x30A7 +#define WM8915_WRITE_SEQUENCER_168 0x30A8 +#define WM8915_WRITE_SEQUENCER_169 0x30A9 +#define WM8915_WRITE_SEQUENCER_170 0x30AA +#define WM8915_WRITE_SEQUENCER_171 0x30AB +#define WM8915_WRITE_SEQUENCER_172 0x30AC +#define WM8915_WRITE_SEQUENCER_173 0x30AD +#define WM8915_WRITE_SEQUENCER_174 0x30AE +#define WM8915_WRITE_SEQUENCER_175 0x30AF +#define WM8915_WRITE_SEQUENCER_176 0x30B0 +#define WM8915_WRITE_SEQUENCER_177 0x30B1 +#define WM8915_WRITE_SEQUENCER_178 0x30B2 +#define WM8915_WRITE_SEQUENCER_179 0x30B3 +#define WM8915_WRITE_SEQUENCER_180 0x30B4 +#define WM8915_WRITE_SEQUENCER_181 0x30B5 +#define WM8915_WRITE_SEQUENCER_182 0x30B6 +#define WM8915_WRITE_SEQUENCER_183 0x30B7 +#define WM8915_WRITE_SEQUENCER_184 0x30B8 +#define WM8915_WRITE_SEQUENCER_185 0x30B9 +#define WM8915_WRITE_SEQUENCER_186 0x30BA +#define WM8915_WRITE_SEQUENCER_187 0x30BB +#define WM8915_WRITE_SEQUENCER_188 0x30BC +#define WM8915_WRITE_SEQUENCER_189 0x30BD +#define WM8915_WRITE_SEQUENCER_190 0x30BE +#define WM8915_WRITE_SEQUENCER_191 0x30BF +#define WM8915_WRITE_SEQUENCER_192 0x30C0 +#define WM8915_WRITE_SEQUENCER_193 0x30C1 +#define WM8915_WRITE_SEQUENCER_194 0x30C2 +#define WM8915_WRITE_SEQUENCER_195 0x30C3 +#define WM8915_WRITE_SEQUENCER_196 0x30C4 +#define WM8915_WRITE_SEQUENCER_197 0x30C5 +#define WM8915_WRITE_SEQUENCER_198 0x30C6 +#define WM8915_WRITE_SEQUENCER_199 0x30C7 +#define WM8915_WRITE_SEQUENCER_200 0x30C8 +#define WM8915_WRITE_SEQUENCER_201 0x30C9 +#define WM8915_WRITE_SEQUENCER_202 0x30CA +#define WM8915_WRITE_SEQUENCER_203 0x30CB +#define WM8915_WRITE_SEQUENCER_204 0x30CC +#define WM8915_WRITE_SEQUENCER_205 0x30CD +#define WM8915_WRITE_SEQUENCER_206 0x30CE +#define WM8915_WRITE_SEQUENCER_207 0x30CF +#define WM8915_WRITE_SEQUENCER_208 0x30D0 +#define WM8915_WRITE_SEQUENCER_209 0x30D1 +#define WM8915_WRITE_SEQUENCER_210 0x30D2 +#define WM8915_WRITE_SEQUENCER_211 0x30D3 +#define WM8915_WRITE_SEQUENCER_212 0x30D4 +#define WM8915_WRITE_SEQUENCER_213 0x30D5 +#define WM8915_WRITE_SEQUENCER_214 0x30D6 +#define WM8915_WRITE_SEQUENCER_215 0x30D7 +#define WM8915_WRITE_SEQUENCER_216 0x30D8 +#define WM8915_WRITE_SEQUENCER_217 0x30D9 +#define WM8915_WRITE_SEQUENCER_218 0x30DA +#define WM8915_WRITE_SEQUENCER_219 0x30DB +#define WM8915_WRITE_SEQUENCER_220 0x30DC +#define WM8915_WRITE_SEQUENCER_221 0x30DD +#define WM8915_WRITE_SEQUENCER_222 0x30DE +#define WM8915_WRITE_SEQUENCER_223 0x30DF +#define WM8915_WRITE_SEQUENCER_224 0x30E0 +#define WM8915_WRITE_SEQUENCER_225 0x30E1 +#define WM8915_WRITE_SEQUENCER_226 0x30E2 +#define WM8915_WRITE_SEQUENCER_227 0x30E3 +#define WM8915_WRITE_SEQUENCER_228 0x30E4 +#define WM8915_WRITE_SEQUENCER_229 0x30E5 +#define WM8915_WRITE_SEQUENCER_230 0x30E6 +#define WM8915_WRITE_SEQUENCER_231 0x30E7 +#define WM8915_WRITE_SEQUENCER_232 0x30E8 +#define WM8915_WRITE_SEQUENCER_233 0x30E9 +#define WM8915_WRITE_SEQUENCER_234 0x30EA +#define WM8915_WRITE_SEQUENCER_235 0x30EB +#define WM8915_WRITE_SEQUENCER_236 0x30EC +#define WM8915_WRITE_SEQUENCER_237 0x30ED +#define WM8915_WRITE_SEQUENCER_238 0x30EE +#define WM8915_WRITE_SEQUENCER_239 0x30EF +#define WM8915_WRITE_SEQUENCER_240 0x30F0 +#define WM8915_WRITE_SEQUENCER_241 0x30F1 +#define WM8915_WRITE_SEQUENCER_242 0x30F2 +#define WM8915_WRITE_SEQUENCER_243 0x30F3 +#define WM8915_WRITE_SEQUENCER_244 0x30F4 +#define WM8915_WRITE_SEQUENCER_245 0x30F5 +#define WM8915_WRITE_SEQUENCER_246 0x30F6 +#define WM8915_WRITE_SEQUENCER_247 0x30F7 +#define WM8915_WRITE_SEQUENCER_248 0x30F8 +#define WM8915_WRITE_SEQUENCER_249 0x30F9 +#define WM8915_WRITE_SEQUENCER_250 0x30FA +#define WM8915_WRITE_SEQUENCER_251 0x30FB +#define WM8915_WRITE_SEQUENCER_252 0x30FC +#define WM8915_WRITE_SEQUENCER_253 0x30FD +#define WM8915_WRITE_SEQUENCER_254 0x30FE +#define WM8915_WRITE_SEQUENCER_255 0x30FF +#define WM8915_WRITE_SEQUENCER_256 0x3100 +#define WM8915_WRITE_SEQUENCER_257 0x3101 +#define WM8915_WRITE_SEQUENCER_258 0x3102 +#define WM8915_WRITE_SEQUENCER_259 0x3103 +#define WM8915_WRITE_SEQUENCER_260 0x3104 +#define WM8915_WRITE_SEQUENCER_261 0x3105 +#define WM8915_WRITE_SEQUENCER_262 0x3106 +#define WM8915_WRITE_SEQUENCER_263 0x3107 +#define WM8915_WRITE_SEQUENCER_264 0x3108 +#define WM8915_WRITE_SEQUENCER_265 0x3109 +#define WM8915_WRITE_SEQUENCER_266 0x310A +#define WM8915_WRITE_SEQUENCER_267 0x310B +#define WM8915_WRITE_SEQUENCER_268 0x310C +#define WM8915_WRITE_SEQUENCER_269 0x310D +#define WM8915_WRITE_SEQUENCER_270 0x310E +#define WM8915_WRITE_SEQUENCER_271 0x310F +#define WM8915_WRITE_SEQUENCER_272 0x3110 +#define WM8915_WRITE_SEQUENCER_273 0x3111 +#define WM8915_WRITE_SEQUENCER_274 0x3112 +#define WM8915_WRITE_SEQUENCER_275 0x3113 +#define WM8915_WRITE_SEQUENCER_276 0x3114 +#define WM8915_WRITE_SEQUENCER_277 0x3115 +#define WM8915_WRITE_SEQUENCER_278 0x3116 +#define WM8915_WRITE_SEQUENCER_279 0x3117 +#define WM8915_WRITE_SEQUENCER_280 0x3118 +#define WM8915_WRITE_SEQUENCER_281 0x3119 +#define WM8915_WRITE_SEQUENCER_282 0x311A +#define WM8915_WRITE_SEQUENCER_283 0x311B +#define WM8915_WRITE_SEQUENCER_284 0x311C +#define WM8915_WRITE_SEQUENCER_285 0x311D +#define WM8915_WRITE_SEQUENCER_286 0x311E +#define WM8915_WRITE_SEQUENCER_287 0x311F +#define WM8915_WRITE_SEQUENCER_288 0x3120 +#define WM8915_WRITE_SEQUENCER_289 0x3121 +#define WM8915_WRITE_SEQUENCER_290 0x3122 +#define WM8915_WRITE_SEQUENCER_291 0x3123 +#define WM8915_WRITE_SEQUENCER_292 0x3124 +#define WM8915_WRITE_SEQUENCER_293 0x3125 +#define WM8915_WRITE_SEQUENCER_294 0x3126 +#define WM8915_WRITE_SEQUENCER_295 0x3127 +#define WM8915_WRITE_SEQUENCER_296 0x3128 +#define WM8915_WRITE_SEQUENCER_297 0x3129 +#define WM8915_WRITE_SEQUENCER_298 0x312A +#define WM8915_WRITE_SEQUENCER_299 0x312B +#define WM8915_WRITE_SEQUENCER_300 0x312C +#define WM8915_WRITE_SEQUENCER_301 0x312D +#define WM8915_WRITE_SEQUENCER_302 0x312E +#define WM8915_WRITE_SEQUENCER_303 0x312F +#define WM8915_WRITE_SEQUENCER_304 0x3130 +#define WM8915_WRITE_SEQUENCER_305 0x3131 +#define WM8915_WRITE_SEQUENCER_306 0x3132 +#define WM8915_WRITE_SEQUENCER_307 0x3133 +#define WM8915_WRITE_SEQUENCER_308 0x3134 +#define WM8915_WRITE_SEQUENCER_309 0x3135 +#define WM8915_WRITE_SEQUENCER_310 0x3136 +#define WM8915_WRITE_SEQUENCER_311 0x3137 +#define WM8915_WRITE_SEQUENCER_312 0x3138 +#define WM8915_WRITE_SEQUENCER_313 0x3139 +#define WM8915_WRITE_SEQUENCER_314 0x313A +#define WM8915_WRITE_SEQUENCER_315 0x313B +#define WM8915_WRITE_SEQUENCER_316 0x313C +#define WM8915_WRITE_SEQUENCER_317 0x313D +#define WM8915_WRITE_SEQUENCER_318 0x313E +#define WM8915_WRITE_SEQUENCER_319 0x313F +#define WM8915_WRITE_SEQUENCER_320 0x3140 +#define WM8915_WRITE_SEQUENCER_321 0x3141 +#define WM8915_WRITE_SEQUENCER_322 0x3142 +#define WM8915_WRITE_SEQUENCER_323 0x3143 +#define WM8915_WRITE_SEQUENCER_324 0x3144 +#define WM8915_WRITE_SEQUENCER_325 0x3145 +#define WM8915_WRITE_SEQUENCER_326 0x3146 +#define WM8915_WRITE_SEQUENCER_327 0x3147 +#define WM8915_WRITE_SEQUENCER_328 0x3148 +#define WM8915_WRITE_SEQUENCER_329 0x3149 +#define WM8915_WRITE_SEQUENCER_330 0x314A +#define WM8915_WRITE_SEQUENCER_331 0x314B +#define WM8915_WRITE_SEQUENCER_332 0x314C +#define WM8915_WRITE_SEQUENCER_333 0x314D +#define WM8915_WRITE_SEQUENCER_334 0x314E +#define WM8915_WRITE_SEQUENCER_335 0x314F +#define WM8915_WRITE_SEQUENCER_336 0x3150 +#define WM8915_WRITE_SEQUENCER_337 0x3151 +#define WM8915_WRITE_SEQUENCER_338 0x3152 +#define WM8915_WRITE_SEQUENCER_339 0x3153 +#define WM8915_WRITE_SEQUENCER_340 0x3154 +#define WM8915_WRITE_SEQUENCER_341 0x3155 +#define WM8915_WRITE_SEQUENCER_342 0x3156 +#define WM8915_WRITE_SEQUENCER_343 0x3157 +#define WM8915_WRITE_SEQUENCER_344 0x3158 +#define WM8915_WRITE_SEQUENCER_345 0x3159 +#define WM8915_WRITE_SEQUENCER_346 0x315A +#define WM8915_WRITE_SEQUENCER_347 0x315B +#define WM8915_WRITE_SEQUENCER_348 0x315C +#define WM8915_WRITE_SEQUENCER_349 0x315D +#define WM8915_WRITE_SEQUENCER_350 0x315E +#define WM8915_WRITE_SEQUENCER_351 0x315F +#define WM8915_WRITE_SEQUENCER_352 0x3160 +#define WM8915_WRITE_SEQUENCER_353 0x3161 +#define WM8915_WRITE_SEQUENCER_354 0x3162 +#define WM8915_WRITE_SEQUENCER_355 0x3163 +#define WM8915_WRITE_SEQUENCER_356 0x3164 +#define WM8915_WRITE_SEQUENCER_357 0x3165 +#define WM8915_WRITE_SEQUENCER_358 0x3166 +#define WM8915_WRITE_SEQUENCER_359 0x3167 +#define WM8915_WRITE_SEQUENCER_360 0x3168 +#define WM8915_WRITE_SEQUENCER_361 0x3169 +#define WM8915_WRITE_SEQUENCER_362 0x316A +#define WM8915_WRITE_SEQUENCER_363 0x316B +#define WM8915_WRITE_SEQUENCER_364 0x316C +#define WM8915_WRITE_SEQUENCER_365 0x316D +#define WM8915_WRITE_SEQUENCER_366 0x316E +#define WM8915_WRITE_SEQUENCER_367 0x316F +#define WM8915_WRITE_SEQUENCER_368 0x3170 +#define WM8915_WRITE_SEQUENCER_369 0x3171 +#define WM8915_WRITE_SEQUENCER_370 0x3172 +#define WM8915_WRITE_SEQUENCER_371 0x3173 +#define WM8915_WRITE_SEQUENCER_372 0x3174 +#define WM8915_WRITE_SEQUENCER_373 0x3175 +#define WM8915_WRITE_SEQUENCER_374 0x3176 +#define WM8915_WRITE_SEQUENCER_375 0x3177 +#define WM8915_WRITE_SEQUENCER_376 0x3178 +#define WM8915_WRITE_SEQUENCER_377 0x3179 +#define WM8915_WRITE_SEQUENCER_378 0x317A +#define WM8915_WRITE_SEQUENCER_379 0x317B +#define WM8915_WRITE_SEQUENCER_380 0x317C +#define WM8915_WRITE_SEQUENCER_381 0x317D +#define WM8915_WRITE_SEQUENCER_382 0x317E +#define WM8915_WRITE_SEQUENCER_383 0x317F +#define WM8915_WRITE_SEQUENCER_384 0x3180 +#define WM8915_WRITE_SEQUENCER_385 0x3181 +#define WM8915_WRITE_SEQUENCER_386 0x3182 +#define WM8915_WRITE_SEQUENCER_387 0x3183 +#define WM8915_WRITE_SEQUENCER_388 0x3184 +#define WM8915_WRITE_SEQUENCER_389 0x3185 +#define WM8915_WRITE_SEQUENCER_390 0x3186 +#define WM8915_WRITE_SEQUENCER_391 0x3187 +#define WM8915_WRITE_SEQUENCER_392 0x3188 +#define WM8915_WRITE_SEQUENCER_393 0x3189 +#define WM8915_WRITE_SEQUENCER_394 0x318A +#define WM8915_WRITE_SEQUENCER_395 0x318B +#define WM8915_WRITE_SEQUENCER_396 0x318C +#define WM8915_WRITE_SEQUENCER_397 0x318D +#define WM8915_WRITE_SEQUENCER_398 0x318E +#define WM8915_WRITE_SEQUENCER_399 0x318F +#define WM8915_WRITE_SEQUENCER_400 0x3190 +#define WM8915_WRITE_SEQUENCER_401 0x3191 +#define WM8915_WRITE_SEQUENCER_402 0x3192 +#define WM8915_WRITE_SEQUENCER_403 0x3193 +#define WM8915_WRITE_SEQUENCER_404 0x3194 +#define WM8915_WRITE_SEQUENCER_405 0x3195 +#define WM8915_WRITE_SEQUENCER_406 0x3196 +#define WM8915_WRITE_SEQUENCER_407 0x3197 +#define WM8915_WRITE_SEQUENCER_408 0x3198 +#define WM8915_WRITE_SEQUENCER_409 0x3199 +#define WM8915_WRITE_SEQUENCER_410 0x319A +#define WM8915_WRITE_SEQUENCER_411 0x319B +#define WM8915_WRITE_SEQUENCER_412 0x319C +#define WM8915_WRITE_SEQUENCER_413 0x319D +#define WM8915_WRITE_SEQUENCER_414 0x319E +#define WM8915_WRITE_SEQUENCER_415 0x319F +#define WM8915_WRITE_SEQUENCER_416 0x31A0 +#define WM8915_WRITE_SEQUENCER_417 0x31A1 +#define WM8915_WRITE_SEQUENCER_418 0x31A2 +#define WM8915_WRITE_SEQUENCER_419 0x31A3 +#define WM8915_WRITE_SEQUENCER_420 0x31A4 +#define WM8915_WRITE_SEQUENCER_421 0x31A5 +#define WM8915_WRITE_SEQUENCER_422 0x31A6 +#define WM8915_WRITE_SEQUENCER_423 0x31A7 +#define WM8915_WRITE_SEQUENCER_424 0x31A8 +#define WM8915_WRITE_SEQUENCER_425 0x31A9 +#define WM8915_WRITE_SEQUENCER_426 0x31AA +#define WM8915_WRITE_SEQUENCER_427 0x31AB +#define WM8915_WRITE_SEQUENCER_428 0x31AC +#define WM8915_WRITE_SEQUENCER_429 0x31AD +#define WM8915_WRITE_SEQUENCER_430 0x31AE +#define WM8915_WRITE_SEQUENCER_431 0x31AF +#define WM8915_WRITE_SEQUENCER_432 0x31B0 +#define WM8915_WRITE_SEQUENCER_433 0x31B1 +#define WM8915_WRITE_SEQUENCER_434 0x31B2 +#define WM8915_WRITE_SEQUENCER_435 0x31B3 +#define WM8915_WRITE_SEQUENCER_436 0x31B4 +#define WM8915_WRITE_SEQUENCER_437 0x31B5 +#define WM8915_WRITE_SEQUENCER_438 0x31B6 +#define WM8915_WRITE_SEQUENCER_439 0x31B7 +#define WM8915_WRITE_SEQUENCER_440 0x31B8 +#define WM8915_WRITE_SEQUENCER_441 0x31B9 +#define WM8915_WRITE_SEQUENCER_442 0x31BA +#define WM8915_WRITE_SEQUENCER_443 0x31BB +#define WM8915_WRITE_SEQUENCER_444 0x31BC +#define WM8915_WRITE_SEQUENCER_445 0x31BD +#define WM8915_WRITE_SEQUENCER_446 0x31BE +#define WM8915_WRITE_SEQUENCER_447 0x31BF +#define WM8915_WRITE_SEQUENCER_448 0x31C0 +#define WM8915_WRITE_SEQUENCER_449 0x31C1 +#define WM8915_WRITE_SEQUENCER_450 0x31C2 +#define WM8915_WRITE_SEQUENCER_451 0x31C3 +#define WM8915_WRITE_SEQUENCER_452 0x31C4 +#define WM8915_WRITE_SEQUENCER_453 0x31C5 +#define WM8915_WRITE_SEQUENCER_454 0x31C6 +#define WM8915_WRITE_SEQUENCER_455 0x31C7 +#define WM8915_WRITE_SEQUENCER_456 0x31C8 +#define WM8915_WRITE_SEQUENCER_457 0x31C9 +#define WM8915_WRITE_SEQUENCER_458 0x31CA +#define WM8915_WRITE_SEQUENCER_459 0x31CB +#define WM8915_WRITE_SEQUENCER_460 0x31CC +#define WM8915_WRITE_SEQUENCER_461 0x31CD +#define WM8915_WRITE_SEQUENCER_462 0x31CE +#define WM8915_WRITE_SEQUENCER_463 0x31CF +#define WM8915_WRITE_SEQUENCER_464 0x31D0 +#define WM8915_WRITE_SEQUENCER_465 0x31D1 +#define WM8915_WRITE_SEQUENCER_466 0x31D2 +#define WM8915_WRITE_SEQUENCER_467 0x31D3 +#define WM8915_WRITE_SEQUENCER_468 0x31D4 +#define WM8915_WRITE_SEQUENCER_469 0x31D5 +#define WM8915_WRITE_SEQUENCER_470 0x31D6 +#define WM8915_WRITE_SEQUENCER_471 0x31D7 +#define WM8915_WRITE_SEQUENCER_472 0x31D8 +#define WM8915_WRITE_SEQUENCER_473 0x31D9 +#define WM8915_WRITE_SEQUENCER_474 0x31DA +#define WM8915_WRITE_SEQUENCER_475 0x31DB +#define WM8915_WRITE_SEQUENCER_476 0x31DC +#define WM8915_WRITE_SEQUENCER_477 0x31DD +#define WM8915_WRITE_SEQUENCER_478 0x31DE +#define WM8915_WRITE_SEQUENCER_479 0x31DF +#define WM8915_WRITE_SEQUENCER_480 0x31E0 +#define WM8915_WRITE_SEQUENCER_481 0x31E1 +#define WM8915_WRITE_SEQUENCER_482 0x31E2 +#define WM8915_WRITE_SEQUENCER_483 0x31E3 +#define WM8915_WRITE_SEQUENCER_484 0x31E4 +#define WM8915_WRITE_SEQUENCER_485 0x31E5 +#define WM8915_WRITE_SEQUENCER_486 0x31E6 +#define WM8915_WRITE_SEQUENCER_487 0x31E7 +#define WM8915_WRITE_SEQUENCER_488 0x31E8 +#define WM8915_WRITE_SEQUENCER_489 0x31E9 +#define WM8915_WRITE_SEQUENCER_490 0x31EA +#define WM8915_WRITE_SEQUENCER_491 0x31EB +#define WM8915_WRITE_SEQUENCER_492 0x31EC +#define WM8915_WRITE_SEQUENCER_493 0x31ED +#define WM8915_WRITE_SEQUENCER_494 0x31EE +#define WM8915_WRITE_SEQUENCER_495 0x31EF +#define WM8915_WRITE_SEQUENCER_496 0x31F0 +#define WM8915_WRITE_SEQUENCER_497 0x31F1 +#define WM8915_WRITE_SEQUENCER_498 0x31F2 +#define WM8915_WRITE_SEQUENCER_499 0x31F3 +#define WM8915_WRITE_SEQUENCER_500 0x31F4 +#define WM8915_WRITE_SEQUENCER_501 0x31F5 +#define WM8915_WRITE_SEQUENCER_502 0x31F6 +#define WM8915_WRITE_SEQUENCER_503 0x31F7 +#define WM8915_WRITE_SEQUENCER_504 0x31F8 +#define WM8915_WRITE_SEQUENCER_505 0x31F9 +#define WM8915_WRITE_SEQUENCER_506 0x31FA +#define WM8915_WRITE_SEQUENCER_507 0x31FB +#define WM8915_WRITE_SEQUENCER_508 0x31FC +#define WM8915_WRITE_SEQUENCER_509 0x31FD +#define WM8915_WRITE_SEQUENCER_510 0x31FE +#define WM8915_WRITE_SEQUENCER_511 0x31FF + +#define WM8915_REGISTER_COUNT 706 +#define WM8915_MAX_REGISTER 0x31FF + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Software Reset + */ +#define WM8915_SW_RESET_MASK 0xFFFF /* SW_RESET - [15:0] */ +#define WM8915_SW_RESET_SHIFT 0 /* SW_RESET - [15:0] */ +#define WM8915_SW_RESET_WIDTH 16 /* SW_RESET - [15:0] */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8915_MICB2_ENA 0x0200 /* MICB2_ENA */ +#define WM8915_MICB2_ENA_MASK 0x0200 /* MICB2_ENA */ +#define WM8915_MICB2_ENA_SHIFT 9 /* MICB2_ENA */ +#define WM8915_MICB2_ENA_WIDTH 1 /* MICB2_ENA */ +#define WM8915_MICB1_ENA 0x0100 /* MICB1_ENA */ +#define WM8915_MICB1_ENA_MASK 0x0100 /* MICB1_ENA */ +#define WM8915_MICB1_ENA_SHIFT 8 /* MICB1_ENA */ +#define WM8915_MICB1_ENA_WIDTH 1 /* MICB1_ENA */ +#define WM8915_HPOUT2L_ENA 0x0080 /* HPOUT2L_ENA */ +#define WM8915_HPOUT2L_ENA_MASK 0x0080 /* HPOUT2L_ENA */ +#define WM8915_HPOUT2L_ENA_SHIFT 7 /* HPOUT2L_ENA */ +#define WM8915_HPOUT2L_ENA_WIDTH 1 /* HPOUT2L_ENA */ +#define WM8915_HPOUT2R_ENA 0x0040 /* HPOUT2R_ENA */ +#define WM8915_HPOUT2R_ENA_MASK 0x0040 /* HPOUT2R_ENA */ +#define WM8915_HPOUT2R_ENA_SHIFT 6 /* HPOUT2R_ENA */ +#define WM8915_HPOUT2R_ENA_WIDTH 1 /* HPOUT2R_ENA */ +#define WM8915_HPOUT1L_ENA 0x0020 /* HPOUT1L_ENA */ +#define WM8915_HPOUT1L_ENA_MASK 0x0020 /* HPOUT1L_ENA */ +#define WM8915_HPOUT1L_ENA_SHIFT 5 /* HPOUT1L_ENA */ +#define WM8915_HPOUT1L_ENA_WIDTH 1 /* HPOUT1L_ENA */ +#define WM8915_HPOUT1R_ENA 0x0010 /* HPOUT1R_ENA */ +#define WM8915_HPOUT1R_ENA_MASK 0x0010 /* HPOUT1R_ENA */ +#define WM8915_HPOUT1R_ENA_SHIFT 4 /* HPOUT1R_ENA */ +#define WM8915_HPOUT1R_ENA_WIDTH 1 /* HPOUT1R_ENA */ +#define WM8915_BG_ENA 0x0001 /* BG_ENA */ +#define WM8915_BG_ENA_MASK 0x0001 /* BG_ENA */ +#define WM8915_BG_ENA_SHIFT 0 /* BG_ENA */ +#define WM8915_BG_ENA_WIDTH 1 /* BG_ENA */ + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8915_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8915_OPCLK_ENA_MASK 0x0800 /* OPCLK_ENA */ +#define WM8915_OPCLK_ENA_SHIFT 11 /* OPCLK_ENA */ +#define WM8915_OPCLK_ENA_WIDTH 1 /* OPCLK_ENA */ +#define WM8915_INL_ENA 0x0020 /* INL_ENA */ +#define WM8915_INL_ENA_MASK 0x0020 /* INL_ENA */ +#define WM8915_INL_ENA_SHIFT 5 /* INL_ENA */ +#define WM8915_INL_ENA_WIDTH 1 /* INL_ENA */ +#define WM8915_INR_ENA 0x0010 /* INR_ENA */ +#define WM8915_INR_ENA_MASK 0x0010 /* INR_ENA */ +#define WM8915_INR_ENA_SHIFT 4 /* INR_ENA */ +#define WM8915_INR_ENA_WIDTH 1 /* INR_ENA */ +#define WM8915_LDO2_ENA 0x0002 /* LDO2_ENA */ +#define WM8915_LDO2_ENA_MASK 0x0002 /* LDO2_ENA */ +#define WM8915_LDO2_ENA_SHIFT 1 /* LDO2_ENA */ +#define WM8915_LDO2_ENA_WIDTH 1 /* LDO2_ENA */ + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8915_DSP2RXL_ENA 0x0800 /* DSP2RXL_ENA */ +#define WM8915_DSP2RXL_ENA_MASK 0x0800 /* DSP2RXL_ENA */ +#define WM8915_DSP2RXL_ENA_SHIFT 11 /* DSP2RXL_ENA */ +#define WM8915_DSP2RXL_ENA_WIDTH 1 /* DSP2RXL_ENA */ +#define WM8915_DSP2RXR_ENA 0x0400 /* DSP2RXR_ENA */ +#define WM8915_DSP2RXR_ENA_MASK 0x0400 /* DSP2RXR_ENA */ +#define WM8915_DSP2RXR_ENA_SHIFT 10 /* DSP2RXR_ENA */ +#define WM8915_DSP2RXR_ENA_WIDTH 1 /* DSP2RXR_ENA */ +#define WM8915_DSP1RXL_ENA 0x0200 /* DSP1RXL_ENA */ +#define WM8915_DSP1RXL_ENA_MASK 0x0200 /* DSP1RXL_ENA */ +#define WM8915_DSP1RXL_ENA_SHIFT 9 /* DSP1RXL_ENA */ +#define WM8915_DSP1RXL_ENA_WIDTH 1 /* DSP1RXL_ENA */ +#define WM8915_DSP1RXR_ENA 0x0100 /* DSP1RXR_ENA */ +#define WM8915_DSP1RXR_ENA_MASK 0x0100 /* DSP1RXR_ENA */ +#define WM8915_DSP1RXR_ENA_SHIFT 8 /* DSP1RXR_ENA */ +#define WM8915_DSP1RXR_ENA_WIDTH 1 /* DSP1RXR_ENA */ +#define WM8915_DMIC2L_ENA 0x0020 /* DMIC2L_ENA */ +#define WM8915_DMIC2L_ENA_MASK 0x0020 /* DMIC2L_ENA */ +#define WM8915_DMIC2L_ENA_SHIFT 5 /* DMIC2L_ENA */ +#define WM8915_DMIC2L_ENA_WIDTH 1 /* DMIC2L_ENA */ +#define WM8915_DMIC2R_ENA 0x0010 /* DMIC2R_ENA */ +#define WM8915_DMIC2R_ENA_MASK 0x0010 /* DMIC2R_ENA */ +#define WM8915_DMIC2R_ENA_SHIFT 4 /* DMIC2R_ENA */ +#define WM8915_DMIC2R_ENA_WIDTH 1 /* DMIC2R_ENA */ +#define WM8915_DMIC1L_ENA 0x0008 /* DMIC1L_ENA */ +#define WM8915_DMIC1L_ENA_MASK 0x0008 /* DMIC1L_ENA */ +#define WM8915_DMIC1L_ENA_SHIFT 3 /* DMIC1L_ENA */ +#define WM8915_DMIC1L_ENA_WIDTH 1 /* DMIC1L_ENA */ +#define WM8915_DMIC1R_ENA 0x0004 /* DMIC1R_ENA */ +#define WM8915_DMIC1R_ENA_MASK 0x0004 /* DMIC1R_ENA */ +#define WM8915_DMIC1R_ENA_SHIFT 2 /* DMIC1R_ENA */ +#define WM8915_DMIC1R_ENA_WIDTH 1 /* DMIC1R_ENA */ +#define WM8915_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8915_ADCL_ENA_MASK 0x0002 /* ADCL_ENA */ +#define WM8915_ADCL_ENA_SHIFT 1 /* ADCL_ENA */ +#define WM8915_ADCL_ENA_WIDTH 1 /* ADCL_ENA */ +#define WM8915_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8915_ADCR_ENA_MASK 0x0001 /* ADCR_ENA */ +#define WM8915_ADCR_ENA_SHIFT 0 /* ADCR_ENA */ +#define WM8915_ADCR_ENA_WIDTH 1 /* ADCR_ENA */ + +/* + * R4 (0x04) - Power Management (4) + */ +#define WM8915_AIF2RX_CHAN1_ENA 0x0200 /* AIF2RX_CHAN1_ENA */ +#define WM8915_AIF2RX_CHAN1_ENA_MASK 0x0200 /* AIF2RX_CHAN1_ENA */ +#define WM8915_AIF2RX_CHAN1_ENA_SHIFT 9 /* AIF2RX_CHAN1_ENA */ +#define WM8915_AIF2RX_CHAN1_ENA_WIDTH 1 /* AIF2RX_CHAN1_ENA */ +#define WM8915_AIF2RX_CHAN0_ENA 0x0100 /* AIF2RX_CHAN0_ENA */ +#define WM8915_AIF2RX_CHAN0_ENA_MASK 0x0100 /* AIF2RX_CHAN0_ENA */ +#define WM8915_AIF2RX_CHAN0_ENA_SHIFT 8 /* AIF2RX_CHAN0_ENA */ +#define WM8915_AIF2RX_CHAN0_ENA_WIDTH 1 /* AIF2RX_CHAN0_ENA */ +#define WM8915_AIF1RX_CHAN5_ENA 0x0020 /* AIF1RX_CHAN5_ENA */ +#define WM8915_AIF1RX_CHAN5_ENA_MASK 0x0020 /* AIF1RX_CHAN5_ENA */ +#define WM8915_AIF1RX_CHAN5_ENA_SHIFT 5 /* AIF1RX_CHAN5_ENA */ +#define WM8915_AIF1RX_CHAN5_ENA_WIDTH 1 /* AIF1RX_CHAN5_ENA */ +#define WM8915_AIF1RX_CHAN4_ENA 0x0010 /* AIF1RX_CHAN4_ENA */ +#define WM8915_AIF1RX_CHAN4_ENA_MASK 0x0010 /* AIF1RX_CHAN4_ENA */ +#define WM8915_AIF1RX_CHAN4_ENA_SHIFT 4 /* AIF1RX_CHAN4_ENA */ +#define WM8915_AIF1RX_CHAN4_ENA_WIDTH 1 /* AIF1RX_CHAN4_ENA */ +#define WM8915_AIF1RX_CHAN3_ENA 0x0008 /* AIF1RX_CHAN3_ENA */ +#define WM8915_AIF1RX_CHAN3_ENA_MASK 0x0008 /* AIF1RX_CHAN3_ENA */ +#define WM8915_AIF1RX_CHAN3_ENA_SHIFT 3 /* AIF1RX_CHAN3_ENA */ +#define WM8915_AIF1RX_CHAN3_ENA_WIDTH 1 /* AIF1RX_CHAN3_ENA */ +#define WM8915_AIF1RX_CHAN2_ENA 0x0004 /* AIF1RX_CHAN2_ENA */ +#define WM8915_AIF1RX_CHAN2_ENA_MASK 0x0004 /* AIF1RX_CHAN2_ENA */ +#define WM8915_AIF1RX_CHAN2_ENA_SHIFT 2 /* AIF1RX_CHAN2_ENA */ +#define WM8915_AIF1RX_CHAN2_ENA_WIDTH 1 /* AIF1RX_CHAN2_ENA */ +#define WM8915_AIF1RX_CHAN1_ENA 0x0002 /* AIF1RX_CHAN1_ENA */ +#define WM8915_AIF1RX_CHAN1_ENA_MASK 0x0002 /* AIF1RX_CHAN1_ENA */ +#define WM8915_AIF1RX_CHAN1_ENA_SHIFT 1 /* AIF1RX_CHAN1_ENA */ +#define WM8915_AIF1RX_CHAN1_ENA_WIDTH 1 /* AIF1RX_CHAN1_ENA */ +#define WM8915_AIF1RX_CHAN0_ENA 0x0001 /* AIF1RX_CHAN0_ENA */ +#define WM8915_AIF1RX_CHAN0_ENA_MASK 0x0001 /* AIF1RX_CHAN0_ENA */ +#define WM8915_AIF1RX_CHAN0_ENA_SHIFT 0 /* AIF1RX_CHAN0_ENA */ +#define WM8915_AIF1RX_CHAN0_ENA_WIDTH 1 /* AIF1RX_CHAN0_ENA */ + +/* + * R5 (0x05) - Power Management (5) + */ +#define WM8915_DSP2TXL_ENA 0x0800 /* DSP2TXL_ENA */ +#define WM8915_DSP2TXL_ENA_MASK 0x0800 /* DSP2TXL_ENA */ +#define WM8915_DSP2TXL_ENA_SHIFT 11 /* DSP2TXL_ENA */ +#define WM8915_DSP2TXL_ENA_WIDTH 1 /* DSP2TXL_ENA */ +#define WM8915_DSP2TXR_ENA 0x0400 /* DSP2TXR_ENA */ +#define WM8915_DSP2TXR_ENA_MASK 0x0400 /* DSP2TXR_ENA */ +#define WM8915_DSP2TXR_ENA_SHIFT 10 /* DSP2TXR_ENA */ +#define WM8915_DSP2TXR_ENA_WIDTH 1 /* DSP2TXR_ENA */ +#define WM8915_DSP1TXL_ENA 0x0200 /* DSP1TXL_ENA */ +#define WM8915_DSP1TXL_ENA_MASK 0x0200 /* DSP1TXL_ENA */ +#define WM8915_DSP1TXL_ENA_SHIFT 9 /* DSP1TXL_ENA */ +#define WM8915_DSP1TXL_ENA_WIDTH 1 /* DSP1TXL_ENA */ +#define WM8915_DSP1TXR_ENA 0x0100 /* DSP1TXR_ENA */ +#define WM8915_DSP1TXR_ENA_MASK 0x0100 /* DSP1TXR_ENA */ +#define WM8915_DSP1TXR_ENA_SHIFT 8 /* DSP1TXR_ENA */ +#define WM8915_DSP1TXR_ENA_WIDTH 1 /* DSP1TXR_ENA */ +#define WM8915_DAC2L_ENA 0x0008 /* DAC2L_ENA */ +#define WM8915_DAC2L_ENA_MASK 0x0008 /* DAC2L_ENA */ +#define WM8915_DAC2L_ENA_SHIFT 3 /* DAC2L_ENA */ +#define WM8915_DAC2L_ENA_WIDTH 1 /* DAC2L_ENA */ +#define WM8915_DAC2R_ENA 0x0004 /* DAC2R_ENA */ +#define WM8915_DAC2R_ENA_MASK 0x0004 /* DAC2R_ENA */ +#define WM8915_DAC2R_ENA_SHIFT 2 /* DAC2R_ENA */ +#define WM8915_DAC2R_ENA_WIDTH 1 /* DAC2R_ENA */ +#define WM8915_DAC1L_ENA 0x0002 /* DAC1L_ENA */ +#define WM8915_DAC1L_ENA_MASK 0x0002 /* DAC1L_ENA */ +#define WM8915_DAC1L_ENA_SHIFT 1 /* DAC1L_ENA */ +#define WM8915_DAC1L_ENA_WIDTH 1 /* DAC1L_ENA */ +#define WM8915_DAC1R_ENA 0x0001 /* DAC1R_ENA */ +#define WM8915_DAC1R_ENA_MASK 0x0001 /* DAC1R_ENA */ +#define WM8915_DAC1R_ENA_SHIFT 0 /* DAC1R_ENA */ +#define WM8915_DAC1R_ENA_WIDTH 1 /* DAC1R_ENA */ + +/* + * R6 (0x06) - Power Management (6) + */ +#define WM8915_AIF2TX_CHAN1_ENA 0x0200 /* AIF2TX_CHAN1_ENA */ +#define WM8915_AIF2TX_CHAN1_ENA_MASK 0x0200 /* AIF2TX_CHAN1_ENA */ +#define WM8915_AIF2TX_CHAN1_ENA_SHIFT 9 /* AIF2TX_CHAN1_ENA */ +#define WM8915_AIF2TX_CHAN1_ENA_WIDTH 1 /* AIF2TX_CHAN1_ENA */ +#define WM8915_AIF2TX_CHAN0_ENA 0x0100 /* AIF2TX_CHAN0_ENA */ +#define WM8915_AIF2TX_CHAN0_ENA_MASK 0x0100 /* AIF2TX_CHAN0_ENA */ +#define WM8915_AIF2TX_CHAN0_ENA_SHIFT 8 /* AIF2TX_CHAN0_ENA */ +#define WM8915_AIF2TX_CHAN0_ENA_WIDTH 1 /* AIF2TX_CHAN0_ENA */ +#define WM8915_AIF1TX_CHAN5_ENA 0x0020 /* AIF1TX_CHAN5_ENA */ +#define WM8915_AIF1TX_CHAN5_ENA_MASK 0x0020 /* AIF1TX_CHAN5_ENA */ +#define WM8915_AIF1TX_CHAN5_ENA_SHIFT 5 /* AIF1TX_CHAN5_ENA */ +#define WM8915_AIF1TX_CHAN5_ENA_WIDTH 1 /* AIF1TX_CHAN5_ENA */ +#define WM8915_AIF1TX_CHAN4_ENA 0x0010 /* AIF1TX_CHAN4_ENA */ +#define WM8915_AIF1TX_CHAN4_ENA_MASK 0x0010 /* AIF1TX_CHAN4_ENA */ +#define WM8915_AIF1TX_CHAN4_ENA_SHIFT 4 /* AIF1TX_CHAN4_ENA */ +#define WM8915_AIF1TX_CHAN4_ENA_WIDTH 1 /* AIF1TX_CHAN4_ENA */ +#define WM8915_AIF1TX_CHAN3_ENA 0x0008 /* AIF1TX_CHAN3_ENA */ +#define WM8915_AIF1TX_CHAN3_ENA_MASK 0x0008 /* AIF1TX_CHAN3_ENA */ +#define WM8915_AIF1TX_CHAN3_ENA_SHIFT 3 /* AIF1TX_CHAN3_ENA */ +#define WM8915_AIF1TX_CHAN3_ENA_WIDTH 1 /* AIF1TX_CHAN3_ENA */ +#define WM8915_AIF1TX_CHAN2_ENA 0x0004 /* AIF1TX_CHAN2_ENA */ +#define WM8915_AIF1TX_CHAN2_ENA_MASK 0x0004 /* AIF1TX_CHAN2_ENA */ +#define WM8915_AIF1TX_CHAN2_ENA_SHIFT 2 /* AIF1TX_CHAN2_ENA */ +#define WM8915_AIF1TX_CHAN2_ENA_WIDTH 1 /* AIF1TX_CHAN2_ENA */ +#define WM8915_AIF1TX_CHAN1_ENA 0x0002 /* AIF1TX_CHAN1_ENA */ +#define WM8915_AIF1TX_CHAN1_ENA_MASK 0x0002 /* AIF1TX_CHAN1_ENA */ +#define WM8915_AIF1TX_CHAN1_ENA_SHIFT 1 /* AIF1TX_CHAN1_ENA */ +#define WM8915_AIF1TX_CHAN1_ENA_WIDTH 1 /* AIF1TX_CHAN1_ENA */ +#define WM8915_AIF1TX_CHAN0_ENA 0x0001 /* AIF1TX_CHAN0_ENA */ +#define WM8915_AIF1TX_CHAN0_ENA_MASK 0x0001 /* AIF1TX_CHAN0_ENA */ +#define WM8915_AIF1TX_CHAN0_ENA_SHIFT 0 /* AIF1TX_CHAN0_ENA */ +#define WM8915_AIF1TX_CHAN0_ENA_WIDTH 1 /* AIF1TX_CHAN0_ENA */ + +/* + * R7 (0x07) - Power Management (7) + */ +#define WM8915_DMIC2_FN 0x0200 /* DMIC2_FN */ +#define WM8915_DMIC2_FN_MASK 0x0200 /* DMIC2_FN */ +#define WM8915_DMIC2_FN_SHIFT 9 /* DMIC2_FN */ +#define WM8915_DMIC2_FN_WIDTH 1 /* DMIC2_FN */ +#define WM8915_DMIC1_FN 0x0100 /* DMIC1_FN */ +#define WM8915_DMIC1_FN_MASK 0x0100 /* DMIC1_FN */ +#define WM8915_DMIC1_FN_SHIFT 8 /* DMIC1_FN */ +#define WM8915_DMIC1_FN_WIDTH 1 /* DMIC1_FN */ +#define WM8915_ADC_DMIC_DSP2R_ENA 0x0080 /* ADC_DMIC_DSP2R_ENA */ +#define WM8915_ADC_DMIC_DSP2R_ENA_MASK 0x0080 /* ADC_DMIC_DSP2R_ENA */ +#define WM8915_ADC_DMIC_DSP2R_ENA_SHIFT 7 /* ADC_DMIC_DSP2R_ENA */ +#define WM8915_ADC_DMIC_DSP2R_ENA_WIDTH 1 /* ADC_DMIC_DSP2R_ENA */ +#define WM8915_ADC_DMIC_DSP2L_ENA 0x0040 /* ADC_DMIC_DSP2L_ENA */ +#define WM8915_ADC_DMIC_DSP2L_ENA_MASK 0x0040 /* ADC_DMIC_DSP2L_ENA */ +#define WM8915_ADC_DMIC_DSP2L_ENA_SHIFT 6 /* ADC_DMIC_DSP2L_ENA */ +#define WM8915_ADC_DMIC_DSP2L_ENA_WIDTH 1 /* ADC_DMIC_DSP2L_ENA */ +#define WM8915_ADC_DMIC_SRC2_MASK 0x0030 /* ADC_DMIC_SRC2 - [5:4] */ +#define WM8915_ADC_DMIC_SRC2_SHIFT 4 /* ADC_DMIC_SRC2 - [5:4] */ +#define WM8915_ADC_DMIC_SRC2_WIDTH 2 /* ADC_DMIC_SRC2 - [5:4] */ +#define WM8915_ADC_DMIC_DSP1R_ENA 0x0008 /* ADC_DMIC_DSP1R_ENA */ +#define WM8915_ADC_DMIC_DSP1R_ENA_MASK 0x0008 /* ADC_DMIC_DSP1R_ENA */ +#define WM8915_ADC_DMIC_DSP1R_ENA_SHIFT 3 /* ADC_DMIC_DSP1R_ENA */ +#define WM8915_ADC_DMIC_DSP1R_ENA_WIDTH 1 /* ADC_DMIC_DSP1R_ENA */ +#define WM8915_ADC_DMIC_DSP1L_ENA 0x0004 /* ADC_DMIC_DSP1L_ENA */ +#define WM8915_ADC_DMIC_DSP1L_ENA_MASK 0x0004 /* ADC_DMIC_DSP1L_ENA */ +#define WM8915_ADC_DMIC_DSP1L_ENA_SHIFT 2 /* ADC_DMIC_DSP1L_ENA */ +#define WM8915_ADC_DMIC_DSP1L_ENA_WIDTH 1 /* ADC_DMIC_DSP1L_ENA */ +#define WM8915_ADC_DMIC_SRC1_MASK 0x0003 /* ADC_DMIC_SRC1 - [1:0] */ +#define WM8915_ADC_DMIC_SRC1_SHIFT 0 /* ADC_DMIC_SRC1 - [1:0] */ +#define WM8915_ADC_DMIC_SRC1_WIDTH 2 /* ADC_DMIC_SRC1 - [1:0] */ + +/* + * R8 (0x08) - Power Management (8) + */ +#define WM8915_AIF2TX_SRC_MASK 0x00C0 /* AIF2TX_SRC - [7:6] */ +#define WM8915_AIF2TX_SRC_SHIFT 6 /* AIF2TX_SRC - [7:6] */ +#define WM8915_AIF2TX_SRC_WIDTH 2 /* AIF2TX_SRC - [7:6] */ +#define WM8915_DSP2RX_SRC 0x0010 /* DSP2RX_SRC */ +#define WM8915_DSP2RX_SRC_MASK 0x0010 /* DSP2RX_SRC */ +#define WM8915_DSP2RX_SRC_SHIFT 4 /* DSP2RX_SRC */ +#define WM8915_DSP2RX_SRC_WIDTH 1 /* DSP2RX_SRC */ +#define WM8915_DSP1RX_SRC 0x0001 /* DSP1RX_SRC */ +#define WM8915_DSP1RX_SRC_MASK 0x0001 /* DSP1RX_SRC */ +#define WM8915_DSP1RX_SRC_SHIFT 0 /* DSP1RX_SRC */ +#define WM8915_DSP1RX_SRC_WIDTH 1 /* DSP1RX_SRC */ + +/* + * R16 (0x10) - Left Line Input Volume + */ +#define WM8915_IN1_VU 0x0080 /* IN1_VU */ +#define WM8915_IN1_VU_MASK 0x0080 /* IN1_VU */ +#define WM8915_IN1_VU_SHIFT 7 /* IN1_VU */ +#define WM8915_IN1_VU_WIDTH 1 /* IN1_VU */ +#define WM8915_IN1L_ZC 0x0020 /* IN1L_ZC */ +#define WM8915_IN1L_ZC_MASK 0x0020 /* IN1L_ZC */ +#define WM8915_IN1L_ZC_SHIFT 5 /* IN1L_ZC */ +#define WM8915_IN1L_ZC_WIDTH 1 /* IN1L_ZC */ +#define WM8915_IN1L_VOL_MASK 0x001F /* IN1L_VOL - [4:0] */ +#define WM8915_IN1L_VOL_SHIFT 0 /* IN1L_VOL - [4:0] */ +#define WM8915_IN1L_VOL_WIDTH 5 /* IN1L_VOL - [4:0] */ + +/* + * R17 (0x11) - Right Line Input Volume + */ +#define WM8915_IN1_VU 0x0080 /* IN1_VU */ +#define WM8915_IN1_VU_MASK 0x0080 /* IN1_VU */ +#define WM8915_IN1_VU_SHIFT 7 /* IN1_VU */ +#define WM8915_IN1_VU_WIDTH 1 /* IN1_VU */ +#define WM8915_IN1R_ZC 0x0020 /* IN1R_ZC */ +#define WM8915_IN1R_ZC_MASK 0x0020 /* IN1R_ZC */ +#define WM8915_IN1R_ZC_SHIFT 5 /* IN1R_ZC */ +#define WM8915_IN1R_ZC_WIDTH 1 /* IN1R_ZC */ +#define WM8915_IN1R_VOL_MASK 0x001F /* IN1R_VOL - [4:0] */ +#define WM8915_IN1R_VOL_SHIFT 0 /* IN1R_VOL - [4:0] */ +#define WM8915_IN1R_VOL_WIDTH 5 /* IN1R_VOL - [4:0] */ + +/* + * R18 (0x12) - Line Input Control + */ +#define WM8915_INL_MODE_MASK 0x000C /* INL_MODE - [3:2] */ +#define WM8915_INL_MODE_SHIFT 2 /* INL_MODE - [3:2] */ +#define WM8915_INL_MODE_WIDTH 2 /* INL_MODE - [3:2] */ +#define WM8915_INR_MODE_MASK 0x0003 /* INR_MODE - [1:0] */ +#define WM8915_INR_MODE_SHIFT 0 /* INR_MODE - [1:0] */ +#define WM8915_INR_MODE_WIDTH 2 /* INR_MODE - [1:0] */ + +/* + * R21 (0x15) - DAC1 HPOUT1 Volume + */ +#define WM8915_DAC1R_HPOUT1R_VOL_MASK 0x00F0 /* DAC1R_HPOUT1R_VOL - [7:4] */ +#define WM8915_DAC1R_HPOUT1R_VOL_SHIFT 4 /* DAC1R_HPOUT1R_VOL - [7:4] */ +#define WM8915_DAC1R_HPOUT1R_VOL_WIDTH 4 /* DAC1R_HPOUT1R_VOL - [7:4] */ +#define WM8915_DAC1L_HPOUT1L_VOL_MASK 0x000F /* DAC1L_HPOUT1L_VOL - [3:0] */ +#define WM8915_DAC1L_HPOUT1L_VOL_SHIFT 0 /* DAC1L_HPOUT1L_VOL - [3:0] */ +#define WM8915_DAC1L_HPOUT1L_VOL_WIDTH 4 /* DAC1L_HPOUT1L_VOL - [3:0] */ + +/* + * R22 (0x16) - DAC2 HPOUT2 Volume + */ +#define WM8915_DAC2R_HPOUT2R_VOL_MASK 0x00F0 /* DAC2R_HPOUT2R_VOL - [7:4] */ +#define WM8915_DAC2R_HPOUT2R_VOL_SHIFT 4 /* DAC2R_HPOUT2R_VOL - [7:4] */ +#define WM8915_DAC2R_HPOUT2R_VOL_WIDTH 4 /* DAC2R_HPOUT2R_VOL - [7:4] */ +#define WM8915_DAC2L_HPOUT2L_VOL_MASK 0x000F /* DAC2L_HPOUT2L_VOL - [3:0] */ +#define WM8915_DAC2L_HPOUT2L_VOL_SHIFT 0 /* DAC2L_HPOUT2L_VOL - [3:0] */ +#define WM8915_DAC2L_HPOUT2L_VOL_WIDTH 4 /* DAC2L_HPOUT2L_VOL - [3:0] */ + +/* + * R24 (0x18) - DAC1 Left Volume + */ +#define WM8915_DAC1L_MUTE 0x0200 /* DAC1L_MUTE */ +#define WM8915_DAC1L_MUTE_MASK 0x0200 /* DAC1L_MUTE */ +#define WM8915_DAC1L_MUTE_SHIFT 9 /* DAC1L_MUTE */ +#define WM8915_DAC1L_MUTE_WIDTH 1 /* DAC1L_MUTE */ +#define WM8915_DAC1_VU 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_MASK 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_SHIFT 8 /* DAC1_VU */ +#define WM8915_DAC1_VU_WIDTH 1 /* DAC1_VU */ +#define WM8915_DAC1L_VOL_MASK 0x00FF /* DAC1L_VOL - [7:0] */ +#define WM8915_DAC1L_VOL_SHIFT 0 /* DAC1L_VOL - [7:0] */ +#define WM8915_DAC1L_VOL_WIDTH 8 /* DAC1L_VOL - [7:0] */ + +/* + * R25 (0x19) - DAC1 Right Volume + */ +#define WM8915_DAC1R_MUTE 0x0200 /* DAC1R_MUTE */ +#define WM8915_DAC1R_MUTE_MASK 0x0200 /* DAC1R_MUTE */ +#define WM8915_DAC1R_MUTE_SHIFT 9 /* DAC1R_MUTE */ +#define WM8915_DAC1R_MUTE_WIDTH 1 /* DAC1R_MUTE */ +#define WM8915_DAC1_VU 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_MASK 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_SHIFT 8 /* DAC1_VU */ +#define WM8915_DAC1_VU_WIDTH 1 /* DAC1_VU */ +#define WM8915_DAC1R_VOL_MASK 0x00FF /* DAC1R_VOL - [7:0] */ +#define WM8915_DAC1R_VOL_SHIFT 0 /* DAC1R_VOL - [7:0] */ +#define WM8915_DAC1R_VOL_WIDTH 8 /* DAC1R_VOL - [7:0] */ + +/* + * R26 (0x1A) - DAC2 Left Volume + */ +#define WM8915_DAC2L_MUTE 0x0200 /* DAC2L_MUTE */ +#define WM8915_DAC2L_MUTE_MASK 0x0200 /* DAC2L_MUTE */ +#define WM8915_DAC2L_MUTE_SHIFT 9 /* DAC2L_MUTE */ +#define WM8915_DAC2L_MUTE_WIDTH 1 /* DAC2L_MUTE */ +#define WM8915_DAC2_VU 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_MASK 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_SHIFT 8 /* DAC2_VU */ +#define WM8915_DAC2_VU_WIDTH 1 /* DAC2_VU */ +#define WM8915_DAC2L_VOL_MASK 0x00FF /* DAC2L_VOL - [7:0] */ +#define WM8915_DAC2L_VOL_SHIFT 0 /* DAC2L_VOL - [7:0] */ +#define WM8915_DAC2L_VOL_WIDTH 8 /* DAC2L_VOL - [7:0] */ + +/* + * R27 (0x1B) - DAC2 Right Volume + */ +#define WM8915_DAC2R_MUTE 0x0200 /* DAC2R_MUTE */ +#define WM8915_DAC2R_MUTE_MASK 0x0200 /* DAC2R_MUTE */ +#define WM8915_DAC2R_MUTE_SHIFT 9 /* DAC2R_MUTE */ +#define WM8915_DAC2R_MUTE_WIDTH 1 /* DAC2R_MUTE */ +#define WM8915_DAC2_VU 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_MASK 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_SHIFT 8 /* DAC2_VU */ +#define WM8915_DAC2_VU_WIDTH 1 /* DAC2_VU */ +#define WM8915_DAC2R_VOL_MASK 0x00FF /* DAC2R_VOL - [7:0] */ +#define WM8915_DAC2R_VOL_SHIFT 0 /* DAC2R_VOL - [7:0] */ +#define WM8915_DAC2R_VOL_WIDTH 8 /* DAC2R_VOL - [7:0] */ + +/* + * R28 (0x1C) - Output1 Left Volume + */ +#define WM8915_DAC1_VU 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_MASK 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_SHIFT 8 /* DAC1_VU */ +#define WM8915_DAC1_VU_WIDTH 1 /* DAC1_VU */ +#define WM8915_HPOUT1L_ZC 0x0080 /* HPOUT1L_ZC */ +#define WM8915_HPOUT1L_ZC_MASK 0x0080 /* HPOUT1L_ZC */ +#define WM8915_HPOUT1L_ZC_SHIFT 7 /* HPOUT1L_ZC */ +#define WM8915_HPOUT1L_ZC_WIDTH 1 /* HPOUT1L_ZC */ +#define WM8915_HPOUT1L_VOL_MASK 0x000F /* HPOUT1L_VOL - [3:0] */ +#define WM8915_HPOUT1L_VOL_SHIFT 0 /* HPOUT1L_VOL - [3:0] */ +#define WM8915_HPOUT1L_VOL_WIDTH 4 /* HPOUT1L_VOL - [3:0] */ + +/* + * R29 (0x1D) - Output1 Right Volume + */ +#define WM8915_DAC1_VU 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_MASK 0x0100 /* DAC1_VU */ +#define WM8915_DAC1_VU_SHIFT 8 /* DAC1_VU */ +#define WM8915_DAC1_VU_WIDTH 1 /* DAC1_VU */ +#define WM8915_HPOUT1R_ZC 0x0080 /* HPOUT1R_ZC */ +#define WM8915_HPOUT1R_ZC_MASK 0x0080 /* HPOUT1R_ZC */ +#define WM8915_HPOUT1R_ZC_SHIFT 7 /* HPOUT1R_ZC */ +#define WM8915_HPOUT1R_ZC_WIDTH 1 /* HPOUT1R_ZC */ +#define WM8915_HPOUT1R_VOL_MASK 0x000F /* HPOUT1R_VOL - [3:0] */ +#define WM8915_HPOUT1R_VOL_SHIFT 0 /* HPOUT1R_VOL - [3:0] */ +#define WM8915_HPOUT1R_VOL_WIDTH 4 /* HPOUT1R_VOL - [3:0] */ + +/* + * R30 (0x1E) - Output2 Left Volume + */ +#define WM8915_DAC2_VU 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_MASK 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_SHIFT 8 /* DAC2_VU */ +#define WM8915_DAC2_VU_WIDTH 1 /* DAC2_VU */ +#define WM8915_HPOUT2L_ZC 0x0080 /* HPOUT2L_ZC */ +#define WM8915_HPOUT2L_ZC_MASK 0x0080 /* HPOUT2L_ZC */ +#define WM8915_HPOUT2L_ZC_SHIFT 7 /* HPOUT2L_ZC */ +#define WM8915_HPOUT2L_ZC_WIDTH 1 /* HPOUT2L_ZC */ +#define WM8915_HPOUT2L_VOL_MASK 0x000F /* HPOUT2L_VOL - [3:0] */ +#define WM8915_HPOUT2L_VOL_SHIFT 0 /* HPOUT2L_VOL - [3:0] */ +#define WM8915_HPOUT2L_VOL_WIDTH 4 /* HPOUT2L_VOL - [3:0] */ + +/* + * R31 (0x1F) - Output2 Right Volume + */ +#define WM8915_DAC2_VU 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_MASK 0x0100 /* DAC2_VU */ +#define WM8915_DAC2_VU_SHIFT 8 /* DAC2_VU */ +#define WM8915_DAC2_VU_WIDTH 1 /* DAC2_VU */ +#define WM8915_HPOUT2R_ZC 0x0080 /* HPOUT2R_ZC */ +#define WM8915_HPOUT2R_ZC_MASK 0x0080 /* HPOUT2R_ZC */ +#define WM8915_HPOUT2R_ZC_SHIFT 7 /* HPOUT2R_ZC */ +#define WM8915_HPOUT2R_ZC_WIDTH 1 /* HPOUT2R_ZC */ +#define WM8915_HPOUT2R_VOL_MASK 0x000F /* HPOUT2R_VOL - [3:0] */ +#define WM8915_HPOUT2R_VOL_SHIFT 0 /* HPOUT2R_VOL - [3:0] */ +#define WM8915_HPOUT2R_VOL_WIDTH 4 /* HPOUT2R_VOL - [3:0] */ + +/* + * R32 (0x20) - MICBIAS (1) + */ +#define WM8915_MICB1_RATE 0x0020 /* MICB1_RATE */ +#define WM8915_MICB1_RATE_MASK 0x0020 /* MICB1_RATE */ +#define WM8915_MICB1_RATE_SHIFT 5 /* MICB1_RATE */ +#define WM8915_MICB1_RATE_WIDTH 1 /* MICB1_RATE */ +#define WM8915_MICB1_MODE 0x0010 /* MICB1_MODE */ +#define WM8915_MICB1_MODE_MASK 0x0010 /* MICB1_MODE */ +#define WM8915_MICB1_MODE_SHIFT 4 /* MICB1_MODE */ +#define WM8915_MICB1_MODE_WIDTH 1 /* MICB1_MODE */ +#define WM8915_MICB1_LVL_MASK 0x000E /* MICB1_LVL - [3:1] */ +#define WM8915_MICB1_LVL_SHIFT 1 /* MICB1_LVL - [3:1] */ +#define WM8915_MICB1_LVL_WIDTH 3 /* MICB1_LVL - [3:1] */ +#define WM8915_MICB1_DISCH 0x0001 /* MICB1_DISCH */ +#define WM8915_MICB1_DISCH_MASK 0x0001 /* MICB1_DISCH */ +#define WM8915_MICB1_DISCH_SHIFT 0 /* MICB1_DISCH */ +#define WM8915_MICB1_DISCH_WIDTH 1 /* MICB1_DISCH */ + +/* + * R33 (0x21) - MICBIAS (2) + */ +#define WM8915_MICB2_RATE 0x0020 /* MICB2_RATE */ +#define WM8915_MICB2_RATE_MASK 0x0020 /* MICB2_RATE */ +#define WM8915_MICB2_RATE_SHIFT 5 /* MICB2_RATE */ +#define WM8915_MICB2_RATE_WIDTH 1 /* MICB2_RATE */ +#define WM8915_MICB2_MODE 0x0010 /* MICB2_MODE */ +#define WM8915_MICB2_MODE_MASK 0x0010 /* MICB2_MODE */ +#define WM8915_MICB2_MODE_SHIFT 4 /* MICB2_MODE */ +#define WM8915_MICB2_MODE_WIDTH 1 /* MICB2_MODE */ +#define WM8915_MICB2_LVL_MASK 0x000E /* MICB2_LVL - [3:1] */ +#define WM8915_MICB2_LVL_SHIFT 1 /* MICB2_LVL - [3:1] */ +#define WM8915_MICB2_LVL_WIDTH 3 /* MICB2_LVL - [3:1] */ +#define WM8915_MICB2_DISCH 0x0001 /* MICB2_DISCH */ +#define WM8915_MICB2_DISCH_MASK 0x0001 /* MICB2_DISCH */ +#define WM8915_MICB2_DISCH_SHIFT 0 /* MICB2_DISCH */ +#define WM8915_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ + +/* + * R40 (0x28) - LDO 1 + */ +#define WM8915_LDO1_MODE 0x0020 /* LDO1_MODE */ +#define WM8915_LDO1_MODE_MASK 0x0020 /* LDO1_MODE */ +#define WM8915_LDO1_MODE_SHIFT 5 /* LDO1_MODE */ +#define WM8915_LDO1_MODE_WIDTH 1 /* LDO1_MODE */ +#define WM8915_LDO1_VSEL_MASK 0x0006 /* LDO1_VSEL - [2:1] */ +#define WM8915_LDO1_VSEL_SHIFT 1 /* LDO1_VSEL - [2:1] */ +#define WM8915_LDO1_VSEL_WIDTH 2 /* LDO1_VSEL - [2:1] */ +#define WM8915_LDO1_DISCH 0x0001 /* LDO1_DISCH */ +#define WM8915_LDO1_DISCH_MASK 0x0001 /* LDO1_DISCH */ +#define WM8915_LDO1_DISCH_SHIFT 0 /* LDO1_DISCH */ +#define WM8915_LDO1_DISCH_WIDTH 1 /* LDO1_DISCH */ + +/* + * R41 (0x29) - LDO 2 + */ +#define WM8915_LDO2_MODE 0x0020 /* LDO2_MODE */ +#define WM8915_LDO2_MODE_MASK 0x0020 /* LDO2_MODE */ +#define WM8915_LDO2_MODE_SHIFT 5 /* LDO2_MODE */ +#define WM8915_LDO2_MODE_WIDTH 1 /* LDO2_MODE */ +#define WM8915_LDO2_VSEL_MASK 0x001E /* LDO2_VSEL - [4:1] */ +#define WM8915_LDO2_VSEL_SHIFT 1 /* LDO2_VSEL - [4:1] */ +#define WM8915_LDO2_VSEL_WIDTH 4 /* LDO2_VSEL - [4:1] */ +#define WM8915_LDO2_DISCH 0x0001 /* LDO2_DISCH */ +#define WM8915_LDO2_DISCH_MASK 0x0001 /* LDO2_DISCH */ +#define WM8915_LDO2_DISCH_SHIFT 0 /* LDO2_DISCH */ +#define WM8915_LDO2_DISCH_WIDTH 1 /* LDO2_DISCH */ + +/* + * R48 (0x30) - Accessory Detect Mode 1 + */ +#define WM8915_JD_MODE_MASK 0x0003 /* JD_MODE - [1:0] */ +#define WM8915_JD_MODE_SHIFT 0 /* JD_MODE - [1:0] */ +#define WM8915_JD_MODE_WIDTH 2 /* JD_MODE - [1:0] */ + +/* + * R49 (0x31) - Accessory Detect Mode 2 + */ +#define WM8915_HPOUT1FB_SRC 0x0004 /* HPOUT1FB_SRC */ +#define WM8915_HPOUT1FB_SRC_MASK 0x0004 /* HPOUT1FB_SRC */ +#define WM8915_HPOUT1FB_SRC_SHIFT 2 /* HPOUT1FB_SRC */ +#define WM8915_HPOUT1FB_SRC_WIDTH 1 /* HPOUT1FB_SRC */ +#define WM8915_MICD_SRC 0x0002 /* MICD_SRC */ +#define WM8915_MICD_SRC_MASK 0x0002 /* MICD_SRC */ +#define WM8915_MICD_SRC_SHIFT 1 /* MICD_SRC */ +#define WM8915_MICD_SRC_WIDTH 1 /* MICD_SRC */ +#define WM8915_MICD_BIAS_SRC 0x0001 /* MICD_BIAS_SRC */ +#define WM8915_MICD_BIAS_SRC_MASK 0x0001 /* MICD_BIAS_SRC */ +#define WM8915_MICD_BIAS_SRC_SHIFT 0 /* MICD_BIAS_SRC */ +#define WM8915_MICD_BIAS_SRC_WIDTH 1 /* MICD_BIAS_SRC */ + +/* + * R52 (0x34) - Headphone Detect 1 + */ +#define WM8915_HP_HOLDTIME_MASK 0x00E0 /* HP_HOLDTIME - [7:5] */ +#define WM8915_HP_HOLDTIME_SHIFT 5 /* HP_HOLDTIME - [7:5] */ +#define WM8915_HP_HOLDTIME_WIDTH 3 /* HP_HOLDTIME - [7:5] */ +#define WM8915_HP_CLK_DIV_MASK 0x0018 /* HP_CLK_DIV - [4:3] */ +#define WM8915_HP_CLK_DIV_SHIFT 3 /* HP_CLK_DIV - [4:3] */ +#define WM8915_HP_CLK_DIV_WIDTH 2 /* HP_CLK_DIV - [4:3] */ +#define WM8915_HP_STEP_SIZE 0x0002 /* HP_STEP_SIZE */ +#define WM8915_HP_STEP_SIZE_MASK 0x0002 /* HP_STEP_SIZE */ +#define WM8915_HP_STEP_SIZE_SHIFT 1 /* HP_STEP_SIZE */ +#define WM8915_HP_STEP_SIZE_WIDTH 1 /* HP_STEP_SIZE */ +#define WM8915_HP_POLL 0x0001 /* HP_POLL */ +#define WM8915_HP_POLL_MASK 0x0001 /* HP_POLL */ +#define WM8915_HP_POLL_SHIFT 0 /* HP_POLL */ +#define WM8915_HP_POLL_WIDTH 1 /* HP_POLL */ + +/* + * R53 (0x35) - Headphone Detect 2 + */ +#define WM8915_HP_DONE 0x0080 /* HP_DONE */ +#define WM8915_HP_DONE_MASK 0x0080 /* HP_DONE */ +#define WM8915_HP_DONE_SHIFT 7 /* HP_DONE */ +#define WM8915_HP_DONE_WIDTH 1 /* HP_DONE */ +#define WM8915_HP_LVL_MASK 0x007F /* HP_LVL - [6:0] */ +#define WM8915_HP_LVL_SHIFT 0 /* HP_LVL - [6:0] */ +#define WM8915_HP_LVL_WIDTH 7 /* HP_LVL - [6:0] */ + +/* + * R56 (0x38) - Mic Detect 1 + */ +#define WM8915_MICD_BIAS_STARTTIME_MASK 0xF000 /* MICD_BIAS_STARTTIME - [15:12] */ +#define WM8915_MICD_BIAS_STARTTIME_SHIFT 12 /* MICD_BIAS_STARTTIME - [15:12] */ +#define WM8915_MICD_BIAS_STARTTIME_WIDTH 4 /* MICD_BIAS_STARTTIME - [15:12] */ +#define WM8915_MICD_RATE_MASK 0x0F00 /* MICD_RATE - [11:8] */ +#define WM8915_MICD_RATE_SHIFT 8 /* MICD_RATE - [11:8] */ +#define WM8915_MICD_RATE_WIDTH 4 /* MICD_RATE - [11:8] */ +#define WM8915_MICD_DBTIME 0x0002 /* MICD_DBTIME */ +#define WM8915_MICD_DBTIME_MASK 0x0002 /* MICD_DBTIME */ +#define WM8915_MICD_DBTIME_SHIFT 1 /* MICD_DBTIME */ +#define WM8915_MICD_DBTIME_WIDTH 1 /* MICD_DBTIME */ +#define WM8915_MICD_ENA 0x0001 /* MICD_ENA */ +#define WM8915_MICD_ENA_MASK 0x0001 /* MICD_ENA */ +#define WM8915_MICD_ENA_SHIFT 0 /* MICD_ENA */ +#define WM8915_MICD_ENA_WIDTH 1 /* MICD_ENA */ + +/* + * R57 (0x39) - Mic Detect 2 + */ +#define WM8915_MICD_LVL_SEL_MASK 0x00FF /* MICD_LVL_SEL - [7:0] */ +#define WM8915_MICD_LVL_SEL_SHIFT 0 /* MICD_LVL_SEL - [7:0] */ +#define WM8915_MICD_LVL_SEL_WIDTH 8 /* MICD_LVL_SEL - [7:0] */ + +/* + * R58 (0x3A) - Mic Detect 3 + */ +#define WM8915_MICD_LVL_MASK 0x07FC /* MICD_LVL - [10:2] */ +#define WM8915_MICD_LVL_SHIFT 2 /* MICD_LVL - [10:2] */ +#define WM8915_MICD_LVL_WIDTH 9 /* MICD_LVL - [10:2] */ +#define WM8915_MICD_VALID 0x0002 /* MICD_VALID */ +#define WM8915_MICD_VALID_MASK 0x0002 /* MICD_VALID */ +#define WM8915_MICD_VALID_SHIFT 1 /* MICD_VALID */ +#define WM8915_MICD_VALID_WIDTH 1 /* MICD_VALID */ +#define WM8915_MICD_STS 0x0001 /* MICD_STS */ +#define WM8915_MICD_STS_MASK 0x0001 /* MICD_STS */ +#define WM8915_MICD_STS_SHIFT 0 /* MICD_STS */ +#define WM8915_MICD_STS_WIDTH 1 /* MICD_STS */ + +/* + * R64 (0x40) - Charge Pump (1) + */ +#define WM8915_CP_ENA 0x8000 /* CP_ENA */ +#define WM8915_CP_ENA_MASK 0x8000 /* CP_ENA */ +#define WM8915_CP_ENA_SHIFT 15 /* CP_ENA */ +#define WM8915_CP_ENA_WIDTH 1 /* CP_ENA */ + +/* + * R65 (0x41) - Charge Pump (2) + */ +#define WM8915_CP_DISCH 0x8000 /* CP_DISCH */ +#define WM8915_CP_DISCH_MASK 0x8000 /* CP_DISCH */ +#define WM8915_CP_DISCH_SHIFT 15 /* CP_DISCH */ +#define WM8915_CP_DISCH_WIDTH 1 /* CP_DISCH */ + +/* + * R80 (0x50) - DC Servo (1) + */ +#define WM8915_DCS_ENA_CHAN_3 0x0008 /* DCS_ENA_CHAN_3 */ +#define WM8915_DCS_ENA_CHAN_3_MASK 0x0008 /* DCS_ENA_CHAN_3 */ +#define WM8915_DCS_ENA_CHAN_3_SHIFT 3 /* DCS_ENA_CHAN_3 */ +#define WM8915_DCS_ENA_CHAN_3_WIDTH 1 /* DCS_ENA_CHAN_3 */ +#define WM8915_DCS_ENA_CHAN_2 0x0004 /* DCS_ENA_CHAN_2 */ +#define WM8915_DCS_ENA_CHAN_2_MASK 0x0004 /* DCS_ENA_CHAN_2 */ +#define WM8915_DCS_ENA_CHAN_2_SHIFT 2 /* DCS_ENA_CHAN_2 */ +#define WM8915_DCS_ENA_CHAN_2_WIDTH 1 /* DCS_ENA_CHAN_2 */ +#define WM8915_DCS_ENA_CHAN_1 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8915_DCS_ENA_CHAN_1_MASK 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8915_DCS_ENA_CHAN_1_SHIFT 1 /* DCS_ENA_CHAN_1 */ +#define WM8915_DCS_ENA_CHAN_1_WIDTH 1 /* DCS_ENA_CHAN_1 */ +#define WM8915_DCS_ENA_CHAN_0 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8915_DCS_ENA_CHAN_0_MASK 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8915_DCS_ENA_CHAN_0_SHIFT 0 /* DCS_ENA_CHAN_0 */ +#define WM8915_DCS_ENA_CHAN_0_WIDTH 1 /* DCS_ENA_CHAN_0 */ + +/* + * R81 (0x51) - DC Servo (2) + */ +#define WM8915_DCS_TRIG_SINGLE_3 0x8000 /* DCS_TRIG_SINGLE_3 */ +#define WM8915_DCS_TRIG_SINGLE_3_MASK 0x8000 /* DCS_TRIG_SINGLE_3 */ +#define WM8915_DCS_TRIG_SINGLE_3_SHIFT 15 /* DCS_TRIG_SINGLE_3 */ +#define WM8915_DCS_TRIG_SINGLE_3_WIDTH 1 /* DCS_TRIG_SINGLE_3 */ +#define WM8915_DCS_TRIG_SINGLE_2 0x4000 /* DCS_TRIG_SINGLE_2 */ +#define WM8915_DCS_TRIG_SINGLE_2_MASK 0x4000 /* DCS_TRIG_SINGLE_2 */ +#define WM8915_DCS_TRIG_SINGLE_2_SHIFT 14 /* DCS_TRIG_SINGLE_2 */ +#define WM8915_DCS_TRIG_SINGLE_2_WIDTH 1 /* DCS_TRIG_SINGLE_2 */ +#define WM8915_DCS_TRIG_SINGLE_1 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8915_DCS_TRIG_SINGLE_1_MASK 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8915_DCS_TRIG_SINGLE_1_SHIFT 13 /* DCS_TRIG_SINGLE_1 */ +#define WM8915_DCS_TRIG_SINGLE_1_WIDTH 1 /* DCS_TRIG_SINGLE_1 */ +#define WM8915_DCS_TRIG_SINGLE_0 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8915_DCS_TRIG_SINGLE_0_MASK 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8915_DCS_TRIG_SINGLE_0_SHIFT 12 /* DCS_TRIG_SINGLE_0 */ +#define WM8915_DCS_TRIG_SINGLE_0_WIDTH 1 /* DCS_TRIG_SINGLE_0 */ +#define WM8915_DCS_TRIG_SERIES_3 0x0800 /* DCS_TRIG_SERIES_3 */ +#define WM8915_DCS_TRIG_SERIES_3_MASK 0x0800 /* DCS_TRIG_SERIES_3 */ +#define WM8915_DCS_TRIG_SERIES_3_SHIFT 11 /* DCS_TRIG_SERIES_3 */ +#define WM8915_DCS_TRIG_SERIES_3_WIDTH 1 /* DCS_TRIG_SERIES_3 */ +#define WM8915_DCS_TRIG_SERIES_2 0x0400 /* DCS_TRIG_SERIES_2 */ +#define WM8915_DCS_TRIG_SERIES_2_MASK 0x0400 /* DCS_TRIG_SERIES_2 */ +#define WM8915_DCS_TRIG_SERIES_2_SHIFT 10 /* DCS_TRIG_SERIES_2 */ +#define WM8915_DCS_TRIG_SERIES_2_WIDTH 1 /* DCS_TRIG_SERIES_2 */ +#define WM8915_DCS_TRIG_SERIES_1 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8915_DCS_TRIG_SERIES_1_MASK 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8915_DCS_TRIG_SERIES_1_SHIFT 9 /* DCS_TRIG_SERIES_1 */ +#define WM8915_DCS_TRIG_SERIES_1_WIDTH 1 /* DCS_TRIG_SERIES_1 */ +#define WM8915_DCS_TRIG_SERIES_0 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8915_DCS_TRIG_SERIES_0_MASK 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8915_DCS_TRIG_SERIES_0_SHIFT 8 /* DCS_TRIG_SERIES_0 */ +#define WM8915_DCS_TRIG_SERIES_0_WIDTH 1 /* DCS_TRIG_SERIES_0 */ +#define WM8915_DCS_TRIG_STARTUP_3 0x0080 /* DCS_TRIG_STARTUP_3 */ +#define WM8915_DCS_TRIG_STARTUP_3_MASK 0x0080 /* DCS_TRIG_STARTUP_3 */ +#define WM8915_DCS_TRIG_STARTUP_3_SHIFT 7 /* DCS_TRIG_STARTUP_3 */ +#define WM8915_DCS_TRIG_STARTUP_3_WIDTH 1 /* DCS_TRIG_STARTUP_3 */ +#define WM8915_DCS_TRIG_STARTUP_2 0x0040 /* DCS_TRIG_STARTUP_2 */ +#define WM8915_DCS_TRIG_STARTUP_2_MASK 0x0040 /* DCS_TRIG_STARTUP_2 */ +#define WM8915_DCS_TRIG_STARTUP_2_SHIFT 6 /* DCS_TRIG_STARTUP_2 */ +#define WM8915_DCS_TRIG_STARTUP_2_WIDTH 1 /* DCS_TRIG_STARTUP_2 */ +#define WM8915_DCS_TRIG_STARTUP_1 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8915_DCS_TRIG_STARTUP_1_MASK 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8915_DCS_TRIG_STARTUP_1_SHIFT 5 /* DCS_TRIG_STARTUP_1 */ +#define WM8915_DCS_TRIG_STARTUP_1_WIDTH 1 /* DCS_TRIG_STARTUP_1 */ +#define WM8915_DCS_TRIG_STARTUP_0 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8915_DCS_TRIG_STARTUP_0_MASK 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8915_DCS_TRIG_STARTUP_0_SHIFT 4 /* DCS_TRIG_STARTUP_0 */ +#define WM8915_DCS_TRIG_STARTUP_0_WIDTH 1 /* DCS_TRIG_STARTUP_0 */ +#define WM8915_DCS_TRIG_DAC_WR_3 0x0008 /* DCS_TRIG_DAC_WR_3 */ +#define WM8915_DCS_TRIG_DAC_WR_3_MASK 0x0008 /* DCS_TRIG_DAC_WR_3 */ +#define WM8915_DCS_TRIG_DAC_WR_3_SHIFT 3 /* DCS_TRIG_DAC_WR_3 */ +#define WM8915_DCS_TRIG_DAC_WR_3_WIDTH 1 /* DCS_TRIG_DAC_WR_3 */ +#define WM8915_DCS_TRIG_DAC_WR_2 0x0004 /* DCS_TRIG_DAC_WR_2 */ +#define WM8915_DCS_TRIG_DAC_WR_2_MASK 0x0004 /* DCS_TRIG_DAC_WR_2 */ +#define WM8915_DCS_TRIG_DAC_WR_2_SHIFT 2 /* DCS_TRIG_DAC_WR_2 */ +#define WM8915_DCS_TRIG_DAC_WR_2_WIDTH 1 /* DCS_TRIG_DAC_WR_2 */ +#define WM8915_DCS_TRIG_DAC_WR_1 0x0002 /* DCS_TRIG_DAC_WR_1 */ +#define WM8915_DCS_TRIG_DAC_WR_1_MASK 0x0002 /* DCS_TRIG_DAC_WR_1 */ +#define WM8915_DCS_TRIG_DAC_WR_1_SHIFT 1 /* DCS_TRIG_DAC_WR_1 */ +#define WM8915_DCS_TRIG_DAC_WR_1_WIDTH 1 /* DCS_TRIG_DAC_WR_1 */ +#define WM8915_DCS_TRIG_DAC_WR_0 0x0001 /* DCS_TRIG_DAC_WR_0 */ +#define WM8915_DCS_TRIG_DAC_WR_0_MASK 0x0001 /* DCS_TRIG_DAC_WR_0 */ +#define WM8915_DCS_TRIG_DAC_WR_0_SHIFT 0 /* DCS_TRIG_DAC_WR_0 */ +#define WM8915_DCS_TRIG_DAC_WR_0_WIDTH 1 /* DCS_TRIG_DAC_WR_0 */ + +/* + * R82 (0x52) - DC Servo (3) + */ +#define WM8915_DCS_TIMER_PERIOD_23_MASK 0x0F00 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8915_DCS_TIMER_PERIOD_23_SHIFT 8 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8915_DCS_TIMER_PERIOD_23_WIDTH 4 /* DCS_TIMER_PERIOD_23 - [11:8] */ +#define WM8915_DCS_TIMER_PERIOD_01_MASK 0x000F /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8915_DCS_TIMER_PERIOD_01_SHIFT 0 /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8915_DCS_TIMER_PERIOD_01_WIDTH 4 /* DCS_TIMER_PERIOD_01 - [3:0] */ + +/* + * R84 (0x54) - DC Servo (5) + */ +#define WM8915_DCS_SERIES_NO_23_MASK 0x7F00 /* DCS_SERIES_NO_23 - [14:8] */ +#define WM8915_DCS_SERIES_NO_23_SHIFT 8 /* DCS_SERIES_NO_23 - [14:8] */ +#define WM8915_DCS_SERIES_NO_23_WIDTH 7 /* DCS_SERIES_NO_23 - [14:8] */ +#define WM8915_DCS_SERIES_NO_01_MASK 0x007F /* DCS_SERIES_NO_01 - [6:0] */ +#define WM8915_DCS_SERIES_NO_01_SHIFT 0 /* DCS_SERIES_NO_01 - [6:0] */ +#define WM8915_DCS_SERIES_NO_01_WIDTH 7 /* DCS_SERIES_NO_01 - [6:0] */ + +/* + * R85 (0x55) - DC Servo (6) + */ +#define WM8915_DCS_DAC_WR_VAL_3_MASK 0xFF00 /* DCS_DAC_WR_VAL_3 - [15:8] */ +#define WM8915_DCS_DAC_WR_VAL_3_SHIFT 8 /* DCS_DAC_WR_VAL_3 - [15:8] */ +#define WM8915_DCS_DAC_WR_VAL_3_WIDTH 8 /* DCS_DAC_WR_VAL_3 - [15:8] */ +#define WM8915_DCS_DAC_WR_VAL_2_MASK 0x00FF /* DCS_DAC_WR_VAL_2 - [7:0] */ +#define WM8915_DCS_DAC_WR_VAL_2_SHIFT 0 /* DCS_DAC_WR_VAL_2 - [7:0] */ +#define WM8915_DCS_DAC_WR_VAL_2_WIDTH 8 /* DCS_DAC_WR_VAL_2 - [7:0] */ + +/* + * R86 (0x56) - DC Servo (7) + */ +#define WM8915_DCS_DAC_WR_VAL_1_MASK 0xFF00 /* DCS_DAC_WR_VAL_1 - [15:8] */ +#define WM8915_DCS_DAC_WR_VAL_1_SHIFT 8 /* DCS_DAC_WR_VAL_1 - [15:8] */ +#define WM8915_DCS_DAC_WR_VAL_1_WIDTH 8 /* DCS_DAC_WR_VAL_1 - [15:8] */ +#define WM8915_DCS_DAC_WR_VAL_0_MASK 0x00FF /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8915_DCS_DAC_WR_VAL_0_SHIFT 0 /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8915_DCS_DAC_WR_VAL_0_WIDTH 8 /* DCS_DAC_WR_VAL_0 - [7:0] */ + +/* + * R87 (0x57) - DC Servo Readback 0 + */ +#define WM8915_DCS_CAL_COMPLETE_MASK 0x0F00 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8915_DCS_CAL_COMPLETE_SHIFT 8 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8915_DCS_CAL_COMPLETE_WIDTH 4 /* DCS_CAL_COMPLETE - [11:8] */ +#define WM8915_DCS_DAC_WR_COMPLETE_MASK 0x00F0 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8915_DCS_DAC_WR_COMPLETE_SHIFT 4 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8915_DCS_DAC_WR_COMPLETE_WIDTH 4 /* DCS_DAC_WR_COMPLETE - [7:4] */ +#define WM8915_DCS_STARTUP_COMPLETE_MASK 0x000F /* DCS_STARTUP_COMPLETE - [3:0] */ +#define WM8915_DCS_STARTUP_COMPLETE_SHIFT 0 /* DCS_STARTUP_COMPLETE - [3:0] */ +#define WM8915_DCS_STARTUP_COMPLETE_WIDTH 4 /* DCS_STARTUP_COMPLETE - [3:0] */ + +/* + * R96 (0x60) - Analogue HP (1) + */ +#define WM8915_HPOUT1L_RMV_SHORT 0x0080 /* HPOUT1L_RMV_SHORT */ +#define WM8915_HPOUT1L_RMV_SHORT_MASK 0x0080 /* HPOUT1L_RMV_SHORT */ +#define WM8915_HPOUT1L_RMV_SHORT_SHIFT 7 /* HPOUT1L_RMV_SHORT */ +#define WM8915_HPOUT1L_RMV_SHORT_WIDTH 1 /* HPOUT1L_RMV_SHORT */ +#define WM8915_HPOUT1L_OUTP 0x0040 /* HPOUT1L_OUTP */ +#define WM8915_HPOUT1L_OUTP_MASK 0x0040 /* HPOUT1L_OUTP */ +#define WM8915_HPOUT1L_OUTP_SHIFT 6 /* HPOUT1L_OUTP */ +#define WM8915_HPOUT1L_OUTP_WIDTH 1 /* HPOUT1L_OUTP */ +#define WM8915_HPOUT1L_DLY 0x0020 /* HPOUT1L_DLY */ +#define WM8915_HPOUT1L_DLY_MASK 0x0020 /* HPOUT1L_DLY */ +#define WM8915_HPOUT1L_DLY_SHIFT 5 /* HPOUT1L_DLY */ +#define WM8915_HPOUT1L_DLY_WIDTH 1 /* HPOUT1L_DLY */ +#define WM8915_HPOUT1R_RMV_SHORT 0x0008 /* HPOUT1R_RMV_SHORT */ +#define WM8915_HPOUT1R_RMV_SHORT_MASK 0x0008 /* HPOUT1R_RMV_SHORT */ +#define WM8915_HPOUT1R_RMV_SHORT_SHIFT 3 /* HPOUT1R_RMV_SHORT */ +#define WM8915_HPOUT1R_RMV_SHORT_WIDTH 1 /* HPOUT1R_RMV_SHORT */ +#define WM8915_HPOUT1R_OUTP 0x0004 /* HPOUT1R_OUTP */ +#define WM8915_HPOUT1R_OUTP_MASK 0x0004 /* HPOUT1R_OUTP */ +#define WM8915_HPOUT1R_OUTP_SHIFT 2 /* HPOUT1R_OUTP */ +#define WM8915_HPOUT1R_OUTP_WIDTH 1 /* HPOUT1R_OUTP */ +#define WM8915_HPOUT1R_DLY 0x0002 /* HPOUT1R_DLY */ +#define WM8915_HPOUT1R_DLY_MASK 0x0002 /* HPOUT1R_DLY */ +#define WM8915_HPOUT1R_DLY_SHIFT 1 /* HPOUT1R_DLY */ +#define WM8915_HPOUT1R_DLY_WIDTH 1 /* HPOUT1R_DLY */ + +/* + * R97 (0x61) - Analogue HP (2) + */ +#define WM8915_HPOUT2L_RMV_SHORT 0x0080 /* HPOUT2L_RMV_SHORT */ +#define WM8915_HPOUT2L_RMV_SHORT_MASK 0x0080 /* HPOUT2L_RMV_SHORT */ +#define WM8915_HPOUT2L_RMV_SHORT_SHIFT 7 /* HPOUT2L_RMV_SHORT */ +#define WM8915_HPOUT2L_RMV_SHORT_WIDTH 1 /* HPOUT2L_RMV_SHORT */ +#define WM8915_HPOUT2L_OUTP 0x0040 /* HPOUT2L_OUTP */ +#define WM8915_HPOUT2L_OUTP_MASK 0x0040 /* HPOUT2L_OUTP */ +#define WM8915_HPOUT2L_OUTP_SHIFT 6 /* HPOUT2L_OUTP */ +#define WM8915_HPOUT2L_OUTP_WIDTH 1 /* HPOUT2L_OUTP */ +#define WM8915_HPOUT2L_DLY 0x0020 /* HPOUT2L_DLY */ +#define WM8915_HPOUT2L_DLY_MASK 0x0020 /* HPOUT2L_DLY */ +#define WM8915_HPOUT2L_DLY_SHIFT 5 /* HPOUT2L_DLY */ +#define WM8915_HPOUT2L_DLY_WIDTH 1 /* HPOUT2L_DLY */ +#define WM8915_HPOUT2R_RMV_SHORT 0x0008 /* HPOUT2R_RMV_SHORT */ +#define WM8915_HPOUT2R_RMV_SHORT_MASK 0x0008 /* HPOUT2R_RMV_SHORT */ +#define WM8915_HPOUT2R_RMV_SHORT_SHIFT 3 /* HPOUT2R_RMV_SHORT */ +#define WM8915_HPOUT2R_RMV_SHORT_WIDTH 1 /* HPOUT2R_RMV_SHORT */ +#define WM8915_HPOUT2R_OUTP 0x0004 /* HPOUT2R_OUTP */ +#define WM8915_HPOUT2R_OUTP_MASK 0x0004 /* HPOUT2R_OUTP */ +#define WM8915_HPOUT2R_OUTP_SHIFT 2 /* HPOUT2R_OUTP */ +#define WM8915_HPOUT2R_OUTP_WIDTH 1 /* HPOUT2R_OUTP */ +#define WM8915_HPOUT2R_DLY 0x0002 /* HPOUT2R_DLY */ +#define WM8915_HPOUT2R_DLY_MASK 0x0002 /* HPOUT2R_DLY */ +#define WM8915_HPOUT2R_DLY_SHIFT 1 /* HPOUT2R_DLY */ +#define WM8915_HPOUT2R_DLY_WIDTH 1 /* HPOUT2R_DLY */ + +/* + * R256 (0x100) - Chip Revision + */ +#define WM8915_CHIP_REV_MASK 0x000F /* CHIP_REV - [3:0] */ +#define WM8915_CHIP_REV_SHIFT 0 /* CHIP_REV - [3:0] */ +#define WM8915_CHIP_REV_WIDTH 4 /* CHIP_REV - [3:0] */ + +/* + * R257 (0x101) - Control Interface (1) + */ +#define WM8915_AUTO_INC 0x0004 /* AUTO_INC */ +#define WM8915_AUTO_INC_MASK 0x0004 /* AUTO_INC */ +#define WM8915_AUTO_INC_SHIFT 2 /* AUTO_INC */ +#define WM8915_AUTO_INC_WIDTH 1 /* AUTO_INC */ + +/* + * R272 (0x110) - Write Sequencer Ctrl (1) + */ +#define WM8915_WSEQ_ENA 0x8000 /* WSEQ_ENA */ +#define WM8915_WSEQ_ENA_MASK 0x8000 /* WSEQ_ENA */ +#define WM8915_WSEQ_ENA_SHIFT 15 /* WSEQ_ENA */ +#define WM8915_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM8915_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */ +#define WM8915_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */ +#define WM8915_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */ +#define WM8915_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM8915_WSEQ_START 0x0100 /* WSEQ_START */ +#define WM8915_WSEQ_START_MASK 0x0100 /* WSEQ_START */ +#define WM8915_WSEQ_START_SHIFT 8 /* WSEQ_START */ +#define WM8915_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM8915_WSEQ_START_INDEX_MASK 0x007F /* WSEQ_START_INDEX - [6:0] */ +#define WM8915_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [6:0] */ +#define WM8915_WSEQ_START_INDEX_WIDTH 7 /* WSEQ_START_INDEX - [6:0] */ + +/* + * R273 (0x111) - Write Sequencer Ctrl (2) + */ +#define WM8915_WSEQ_BUSY 0x0100 /* WSEQ_BUSY */ +#define WM8915_WSEQ_BUSY_MASK 0x0100 /* WSEQ_BUSY */ +#define WM8915_WSEQ_BUSY_SHIFT 8 /* WSEQ_BUSY */ +#define WM8915_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ +#define WM8915_WSEQ_CURRENT_INDEX_MASK 0x007F /* WSEQ_CURRENT_INDEX - [6:0] */ +#define WM8915_WSEQ_CURRENT_INDEX_SHIFT 0 /* WSEQ_CURRENT_INDEX - [6:0] */ +#define WM8915_WSEQ_CURRENT_INDEX_WIDTH 7 /* WSEQ_CURRENT_INDEX - [6:0] */ + +/* + * R512 (0x200) - AIF Clocking (1) + */ +#define WM8915_SYSCLK_SRC_MASK 0x0018 /* SYSCLK_SRC - [4:3] */ +#define WM8915_SYSCLK_SRC_SHIFT 3 /* SYSCLK_SRC - [4:3] */ +#define WM8915_SYSCLK_SRC_WIDTH 2 /* SYSCLK_SRC - [4:3] */ +#define WM8915_SYSCLK_INV 0x0004 /* SYSCLK_INV */ +#define WM8915_SYSCLK_INV_MASK 0x0004 /* SYSCLK_INV */ +#define WM8915_SYSCLK_INV_SHIFT 2 /* SYSCLK_INV */ +#define WM8915_SYSCLK_INV_WIDTH 1 /* SYSCLK_INV */ +#define WM8915_SYSCLK_DIV 0x0002 /* SYSCLK_DIV */ +#define WM8915_SYSCLK_DIV_MASK 0x0002 /* SYSCLK_DIV */ +#define WM8915_SYSCLK_DIV_SHIFT 1 /* SYSCLK_DIV */ +#define WM8915_SYSCLK_DIV_WIDTH 1 /* SYSCLK_DIV */ +#define WM8915_SYSCLK_ENA 0x0001 /* SYSCLK_ENA */ +#define WM8915_SYSCLK_ENA_MASK 0x0001 /* SYSCLK_ENA */ +#define WM8915_SYSCLK_ENA_SHIFT 0 /* SYSCLK_ENA */ +#define WM8915_SYSCLK_ENA_WIDTH 1 /* SYSCLK_ENA */ + +/* + * R513 (0x201) - AIF Clocking (2) + */ +#define WM8915_DSP2_DIV_MASK 0x0018 /* DSP2_DIV - [4:3] */ +#define WM8915_DSP2_DIV_SHIFT 3 /* DSP2_DIV - [4:3] */ +#define WM8915_DSP2_DIV_WIDTH 2 /* DSP2_DIV - [4:3] */ +#define WM8915_DSP1_DIV_MASK 0x0003 /* DSP1_DIV - [1:0] */ +#define WM8915_DSP1_DIV_SHIFT 0 /* DSP1_DIV - [1:0] */ +#define WM8915_DSP1_DIV_WIDTH 2 /* DSP1_DIV - [1:0] */ + +/* + * R520 (0x208) - Clocking (1) + */ +#define WM8915_LFCLK_ENA 0x0020 /* LFCLK_ENA */ +#define WM8915_LFCLK_ENA_MASK 0x0020 /* LFCLK_ENA */ +#define WM8915_LFCLK_ENA_SHIFT 5 /* LFCLK_ENA */ +#define WM8915_LFCLK_ENA_WIDTH 1 /* LFCLK_ENA */ +#define WM8915_TOCLK_ENA 0x0010 /* TOCLK_ENA */ +#define WM8915_TOCLK_ENA_MASK 0x0010 /* TOCLK_ENA */ +#define WM8915_TOCLK_ENA_SHIFT 4 /* TOCLK_ENA */ +#define WM8915_TOCLK_ENA_WIDTH 1 /* TOCLK_ENA */ +#define WM8915_AIFCLK_ENA 0x0004 /* AIFCLK_ENA */ +#define WM8915_AIFCLK_ENA_MASK 0x0004 /* AIFCLK_ENA */ +#define WM8915_AIFCLK_ENA_SHIFT 2 /* AIFCLK_ENA */ +#define WM8915_AIFCLK_ENA_WIDTH 1 /* AIFCLK_ENA */ +#define WM8915_SYSDSPCLK_ENA 0x0002 /* SYSDSPCLK_ENA */ +#define WM8915_SYSDSPCLK_ENA_MASK 0x0002 /* SYSDSPCLK_ENA */ +#define WM8915_SYSDSPCLK_ENA_SHIFT 1 /* SYSDSPCLK_ENA */ +#define WM8915_SYSDSPCLK_ENA_WIDTH 1 /* SYSDSPCLK_ENA */ + +/* + * R521 (0x209) - Clocking (2) + */ +#define WM8915_TOCLK_DIV_MASK 0x0700 /* TOCLK_DIV - [10:8] */ +#define WM8915_TOCLK_DIV_SHIFT 8 /* TOCLK_DIV - [10:8] */ +#define WM8915_TOCLK_DIV_WIDTH 3 /* TOCLK_DIV - [10:8] */ +#define WM8915_DBCLK_DIV_MASK 0x00F0 /* DBCLK_DIV - [7:4] */ +#define WM8915_DBCLK_DIV_SHIFT 4 /* DBCLK_DIV - [7:4] */ +#define WM8915_DBCLK_DIV_WIDTH 4 /* DBCLK_DIV - [7:4] */ +#define WM8915_OPCLK_DIV_MASK 0x0007 /* OPCLK_DIV - [2:0] */ +#define WM8915_OPCLK_DIV_SHIFT 0 /* OPCLK_DIV - [2:0] */ +#define WM8915_OPCLK_DIV_WIDTH 3 /* OPCLK_DIV - [2:0] */ + +/* + * R528 (0x210) - AIF Rate + */ +#define WM8915_SYSCLK_RATE 0x0001 /* SYSCLK_RATE */ +#define WM8915_SYSCLK_RATE_MASK 0x0001 /* SYSCLK_RATE */ +#define WM8915_SYSCLK_RATE_SHIFT 0 /* SYSCLK_RATE */ +#define WM8915_SYSCLK_RATE_WIDTH 1 /* SYSCLK_RATE */ + +/* + * R544 (0x220) - FLL Control (1) + */ +#define WM8915_FLL_OSC_ENA 0x0002 /* FLL_OSC_ENA */ +#define WM8915_FLL_OSC_ENA_MASK 0x0002 /* FLL_OSC_ENA */ +#define WM8915_FLL_OSC_ENA_SHIFT 1 /* FLL_OSC_ENA */ +#define WM8915_FLL_OSC_ENA_WIDTH 1 /* FLL_OSC_ENA */ +#define WM8915_FLL_ENA 0x0001 /* FLL_ENA */ +#define WM8915_FLL_ENA_MASK 0x0001 /* FLL_ENA */ +#define WM8915_FLL_ENA_SHIFT 0 /* FLL_ENA */ +#define WM8915_FLL_ENA_WIDTH 1 /* FLL_ENA */ + +/* + * R545 (0x221) - FLL Control (2) + */ +#define WM8915_FLL_OUTDIV_MASK 0x3F00 /* FLL_OUTDIV - [13:8] */ +#define WM8915_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [13:8] */ +#define WM8915_FLL_OUTDIV_WIDTH 6 /* FLL_OUTDIV - [13:8] */ +#define WM8915_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */ +#define WM8915_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */ +#define WM8915_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */ + +/* + * R546 (0x222) - FLL Control (3) + */ +#define WM8915_FLL_THETA_MASK 0xFFFF /* FLL_THETA - [15:0] */ +#define WM8915_FLL_THETA_SHIFT 0 /* FLL_THETA - [15:0] */ +#define WM8915_FLL_THETA_WIDTH 16 /* FLL_THETA - [15:0] */ + +/* + * R547 (0x223) - FLL Control (4) + */ +#define WM8915_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */ +#define WM8915_FLL_N_SHIFT 5 /* FLL_N - [14:5] */ +#define WM8915_FLL_N_WIDTH 10 /* FLL_N - [14:5] */ +#define WM8915_FLL_LOOP_GAIN_MASK 0x000F /* FLL_LOOP_GAIN - [3:0] */ +#define WM8915_FLL_LOOP_GAIN_SHIFT 0 /* FLL_LOOP_GAIN - [3:0] */ +#define WM8915_FLL_LOOP_GAIN_WIDTH 4 /* FLL_LOOP_GAIN - [3:0] */ + +/* + * R548 (0x224) - FLL Control (5) + */ +#define WM8915_FLL_FRC_NCO_VAL_MASK 0x1F80 /* FLL_FRC_NCO_VAL - [12:7] */ +#define WM8915_FLL_FRC_NCO_VAL_SHIFT 7 /* FLL_FRC_NCO_VAL - [12:7] */ +#define WM8915_FLL_FRC_NCO_VAL_WIDTH 6 /* FLL_FRC_NCO_VAL - [12:7] */ +#define WM8915_FLL_FRC_NCO 0x0040 /* FLL_FRC_NCO */ +#define WM8915_FLL_FRC_NCO_MASK 0x0040 /* FLL_FRC_NCO */ +#define WM8915_FLL_FRC_NCO_SHIFT 6 /* FLL_FRC_NCO */ +#define WM8915_FLL_FRC_NCO_WIDTH 1 /* FLL_FRC_NCO */ +#define WM8915_FLL_REFCLK_DIV_MASK 0x0018 /* FLL_REFCLK_DIV - [4:3] */ +#define WM8915_FLL_REFCLK_DIV_SHIFT 3 /* FLL_REFCLK_DIV - [4:3] */ +#define WM8915_FLL_REFCLK_DIV_WIDTH 2 /* FLL_REFCLK_DIV - [4:3] */ +#define WM8915_FLL_REF_FREQ 0x0004 /* FLL_REF_FREQ */ +#define WM8915_FLL_REF_FREQ_MASK 0x0004 /* FLL_REF_FREQ */ +#define WM8915_FLL_REF_FREQ_SHIFT 2 /* FLL_REF_FREQ */ +#define WM8915_FLL_REF_FREQ_WIDTH 1 /* FLL_REF_FREQ */ +#define WM8915_FLL_REFCLK_SRC_MASK 0x0003 /* FLL_REFCLK_SRC - [1:0] */ +#define WM8915_FLL_REFCLK_SRC_SHIFT 0 /* FLL_REFCLK_SRC - [1:0] */ +#define WM8915_FLL_REFCLK_SRC_WIDTH 2 /* FLL_REFCLK_SRC - [1:0] */ + +/* + * R549 (0x225) - FLL Control (6) + */ +#define WM8915_FLL_REFCLK_SRC_STS_MASK 0x000C /* FLL_REFCLK_SRC_STS - [3:2] */ +#define WM8915_FLL_REFCLK_SRC_STS_SHIFT 2 /* FLL_REFCLK_SRC_STS - [3:2] */ +#define WM8915_FLL_REFCLK_SRC_STS_WIDTH 2 /* FLL_REFCLK_SRC_STS - [3:2] */ +#define WM8915_FLL_SWITCH_CLK 0x0001 /* FLL_SWITCH_CLK */ +#define WM8915_FLL_SWITCH_CLK_MASK 0x0001 /* FLL_SWITCH_CLK */ +#define WM8915_FLL_SWITCH_CLK_SHIFT 0 /* FLL_SWITCH_CLK */ +#define WM8915_FLL_SWITCH_CLK_WIDTH 1 /* FLL_SWITCH_CLK */ + +/* + * R550 (0x226) - FLL EFS 1 + */ +#define WM8915_FLL_LAMBDA_MASK 0xFFFF /* FLL_LAMBDA - [15:0] */ +#define WM8915_FLL_LAMBDA_SHIFT 0 /* FLL_LAMBDA - [15:0] */ +#define WM8915_FLL_LAMBDA_WIDTH 16 /* FLL_LAMBDA - [15:0] */ + +/* + * R551 (0x227) - FLL EFS 2 + */ +#define WM8915_FLL_LFSR_SEL_MASK 0x0006 /* FLL_LFSR_SEL - [2:1] */ +#define WM8915_FLL_LFSR_SEL_SHIFT 1 /* FLL_LFSR_SEL - [2:1] */ +#define WM8915_FLL_LFSR_SEL_WIDTH 2 /* FLL_LFSR_SEL - [2:1] */ +#define WM8915_FLL_EFS_ENA 0x0001 /* FLL_EFS_ENA */ +#define WM8915_FLL_EFS_ENA_MASK 0x0001 /* FLL_EFS_ENA */ +#define WM8915_FLL_EFS_ENA_SHIFT 0 /* FLL_EFS_ENA */ +#define WM8915_FLL_EFS_ENA_WIDTH 1 /* FLL_EFS_ENA */ + +/* + * R768 (0x300) - AIF1 Control + */ +#define WM8915_AIF1_TRI 0x0004 /* AIF1_TRI */ +#define WM8915_AIF1_TRI_MASK 0x0004 /* AIF1_TRI */ +#define WM8915_AIF1_TRI_SHIFT 2 /* AIF1_TRI */ +#define WM8915_AIF1_TRI_WIDTH 1 /* AIF1_TRI */ +#define WM8915_AIF1_FMT_MASK 0x0003 /* AIF1_FMT - [1:0] */ +#define WM8915_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [1:0] */ +#define WM8915_AIF1_FMT_WIDTH 2 /* AIF1_FMT - [1:0] */ + +/* + * R769 (0x301) - AIF1 BCLK + */ +#define WM8915_AIF1_BCLK_INV 0x0400 /* AIF1_BCLK_INV */ +#define WM8915_AIF1_BCLK_INV_MASK 0x0400 /* AIF1_BCLK_INV */ +#define WM8915_AIF1_BCLK_INV_SHIFT 10 /* AIF1_BCLK_INV */ +#define WM8915_AIF1_BCLK_INV_WIDTH 1 /* AIF1_BCLK_INV */ +#define WM8915_AIF1_BCLK_FRC 0x0200 /* AIF1_BCLK_FRC */ +#define WM8915_AIF1_BCLK_FRC_MASK 0x0200 /* AIF1_BCLK_FRC */ +#define WM8915_AIF1_BCLK_FRC_SHIFT 9 /* AIF1_BCLK_FRC */ +#define WM8915_AIF1_BCLK_FRC_WIDTH 1 /* AIF1_BCLK_FRC */ +#define WM8915_AIF1_BCLK_MSTR 0x0100 /* AIF1_BCLK_MSTR */ +#define WM8915_AIF1_BCLK_MSTR_MASK 0x0100 /* AIF1_BCLK_MSTR */ +#define WM8915_AIF1_BCLK_MSTR_SHIFT 8 /* AIF1_BCLK_MSTR */ +#define WM8915_AIF1_BCLK_MSTR_WIDTH 1 /* AIF1_BCLK_MSTR */ +#define WM8915_AIF1_BCLK_DIV_MASK 0x000F /* AIF1_BCLK_DIV - [3:0] */ +#define WM8915_AIF1_BCLK_DIV_SHIFT 0 /* AIF1_BCLK_DIV - [3:0] */ +#define WM8915_AIF1_BCLK_DIV_WIDTH 4 /* AIF1_BCLK_DIV - [3:0] */ + +/* + * R770 (0x302) - AIF1 TX LRCLK(1) + */ +#define WM8915_AIF1TX_RATE_MASK 0x07FF /* AIF1TX_RATE - [10:0] */ +#define WM8915_AIF1TX_RATE_SHIFT 0 /* AIF1TX_RATE - [10:0] */ +#define WM8915_AIF1TX_RATE_WIDTH 11 /* AIF1TX_RATE - [10:0] */ + +/* + * R771 (0x303) - AIF1 TX LRCLK(2) + */ +#define WM8915_AIF1TX_LRCLK_MODE 0x0008 /* AIF1TX_LRCLK_MODE */ +#define WM8915_AIF1TX_LRCLK_MODE_MASK 0x0008 /* AIF1TX_LRCLK_MODE */ +#define WM8915_AIF1TX_LRCLK_MODE_SHIFT 3 /* AIF1TX_LRCLK_MODE */ +#define WM8915_AIF1TX_LRCLK_MODE_WIDTH 1 /* AIF1TX_LRCLK_MODE */ +#define WM8915_AIF1TX_LRCLK_INV 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM8915_AIF1TX_LRCLK_INV_MASK 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM8915_AIF1TX_LRCLK_INV_SHIFT 2 /* AIF1TX_LRCLK_INV */ +#define WM8915_AIF1TX_LRCLK_INV_WIDTH 1 /* AIF1TX_LRCLK_INV */ +#define WM8915_AIF1TX_LRCLK_FRC 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM8915_AIF1TX_LRCLK_FRC_MASK 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM8915_AIF1TX_LRCLK_FRC_SHIFT 1 /* AIF1TX_LRCLK_FRC */ +#define WM8915_AIF1TX_LRCLK_FRC_WIDTH 1 /* AIF1TX_LRCLK_FRC */ +#define WM8915_AIF1TX_LRCLK_MSTR 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM8915_AIF1TX_LRCLK_MSTR_MASK 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM8915_AIF1TX_LRCLK_MSTR_SHIFT 0 /* AIF1TX_LRCLK_MSTR */ +#define WM8915_AIF1TX_LRCLK_MSTR_WIDTH 1 /* AIF1TX_LRCLK_MSTR */ + +/* + * R772 (0x304) - AIF1 RX LRCLK(1) + */ +#define WM8915_AIF1RX_RATE_MASK 0x07FF /* AIF1RX_RATE - [10:0] */ +#define WM8915_AIF1RX_RATE_SHIFT 0 /* AIF1RX_RATE - [10:0] */ +#define WM8915_AIF1RX_RATE_WIDTH 11 /* AIF1RX_RATE - [10:0] */ + +/* + * R773 (0x305) - AIF1 RX LRCLK(2) + */ +#define WM8915_AIF1RX_LRCLK_INV 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM8915_AIF1RX_LRCLK_INV_MASK 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM8915_AIF1RX_LRCLK_INV_SHIFT 2 /* AIF1RX_LRCLK_INV */ +#define WM8915_AIF1RX_LRCLK_INV_WIDTH 1 /* AIF1RX_LRCLK_INV */ +#define WM8915_AIF1RX_LRCLK_FRC 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM8915_AIF1RX_LRCLK_FRC_MASK 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM8915_AIF1RX_LRCLK_FRC_SHIFT 1 /* AIF1RX_LRCLK_FRC */ +#define WM8915_AIF1RX_LRCLK_FRC_WIDTH 1 /* AIF1RX_LRCLK_FRC */ +#define WM8915_AIF1RX_LRCLK_MSTR 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM8915_AIF1RX_LRCLK_MSTR_MASK 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM8915_AIF1RX_LRCLK_MSTR_SHIFT 0 /* AIF1RX_LRCLK_MSTR */ +#define WM8915_AIF1RX_LRCLK_MSTR_WIDTH 1 /* AIF1RX_LRCLK_MSTR */ + +/* + * R774 (0x306) - AIF1TX Data Configuration (1) + */ +#define WM8915_AIF1TX_WL_MASK 0xFF00 /* AIF1TX_WL - [15:8] */ +#define WM8915_AIF1TX_WL_SHIFT 8 /* AIF1TX_WL - [15:8] */ +#define WM8915_AIF1TX_WL_WIDTH 8 /* AIF1TX_WL - [15:8] */ +#define WM8915_AIF1TX_SLOT_LEN_MASK 0x00FF /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM8915_AIF1TX_SLOT_LEN_SHIFT 0 /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM8915_AIF1TX_SLOT_LEN_WIDTH 8 /* AIF1TX_SLOT_LEN - [7:0] */ + +/* + * R775 (0x307) - AIF1TX Data Configuration (2) + */ +#define WM8915_AIF1TX_DAT_TRI 0x0001 /* AIF1TX_DAT_TRI */ +#define WM8915_AIF1TX_DAT_TRI_MASK 0x0001 /* AIF1TX_DAT_TRI */ +#define WM8915_AIF1TX_DAT_TRI_SHIFT 0 /* AIF1TX_DAT_TRI */ +#define WM8915_AIF1TX_DAT_TRI_WIDTH 1 /* AIF1TX_DAT_TRI */ + +/* + * R776 (0x308) - AIF1RX Data Configuration + */ +#define WM8915_AIF1RX_WL_MASK 0xFF00 /* AIF1RX_WL - [15:8] */ +#define WM8915_AIF1RX_WL_SHIFT 8 /* AIF1RX_WL - [15:8] */ +#define WM8915_AIF1RX_WL_WIDTH 8 /* AIF1RX_WL - [15:8] */ +#define WM8915_AIF1RX_SLOT_LEN_MASK 0x00FF /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM8915_AIF1RX_SLOT_LEN_SHIFT 0 /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM8915_AIF1RX_SLOT_LEN_WIDTH 8 /* AIF1RX_SLOT_LEN - [7:0] */ + +/* + * R777 (0x309) - AIF1TX Channel 0 Configuration + */ +#define WM8915_AIF1TX_CHAN0_DAT_INV 0x8000 /* AIF1TX_CHAN0_DAT_INV */ +#define WM8915_AIF1TX_CHAN0_DAT_INV_MASK 0x8000 /* AIF1TX_CHAN0_DAT_INV */ +#define WM8915_AIF1TX_CHAN0_DAT_INV_SHIFT 15 /* AIF1TX_CHAN0_DAT_INV */ +#define WM8915_AIF1TX_CHAN0_DAT_INV_WIDTH 1 /* AIF1TX_CHAN0_DAT_INV */ +#define WM8915_AIF1TX_CHAN0_SPACING_MASK 0x7E00 /* AIF1TX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN0_SPACING_SHIFT 9 /* AIF1TX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN0_SPACING_WIDTH 6 /* AIF1TX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN0_SLOTS_MASK 0x01C0 /* AIF1TX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN0_SLOTS_SHIFT 6 /* AIF1TX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN0_SLOTS_WIDTH 3 /* AIF1TX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN0_START_SLOT_MASK 0x003F /* AIF1TX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN0_START_SLOT_SHIFT 0 /* AIF1TX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN0_START_SLOT_WIDTH 6 /* AIF1TX_CHAN0_START_SLOT - [5:0] */ + +/* + * R778 (0x30A) - AIF1TX Channel 1 Configuration + */ +#define WM8915_AIF1TX_CHAN1_DAT_INV 0x8000 /* AIF1TX_CHAN1_DAT_INV */ +#define WM8915_AIF1TX_CHAN1_DAT_INV_MASK 0x8000 /* AIF1TX_CHAN1_DAT_INV */ +#define WM8915_AIF1TX_CHAN1_DAT_INV_SHIFT 15 /* AIF1TX_CHAN1_DAT_INV */ +#define WM8915_AIF1TX_CHAN1_DAT_INV_WIDTH 1 /* AIF1TX_CHAN1_DAT_INV */ +#define WM8915_AIF1TX_CHAN1_SPACING_MASK 0x7E00 /* AIF1TX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN1_SPACING_SHIFT 9 /* AIF1TX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN1_SPACING_WIDTH 6 /* AIF1TX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN1_SLOTS_MASK 0x01C0 /* AIF1TX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN1_SLOTS_SHIFT 6 /* AIF1TX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN1_SLOTS_WIDTH 3 /* AIF1TX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN1_START_SLOT_MASK 0x003F /* AIF1TX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN1_START_SLOT_SHIFT 0 /* AIF1TX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN1_START_SLOT_WIDTH 6 /* AIF1TX_CHAN1_START_SLOT - [5:0] */ + +/* + * R779 (0x30B) - AIF1TX Channel 2 Configuration + */ +#define WM8915_AIF1TX_CHAN2_DAT_INV 0x8000 /* AIF1TX_CHAN2_DAT_INV */ +#define WM8915_AIF1TX_CHAN2_DAT_INV_MASK 0x8000 /* AIF1TX_CHAN2_DAT_INV */ +#define WM8915_AIF1TX_CHAN2_DAT_INV_SHIFT 15 /* AIF1TX_CHAN2_DAT_INV */ +#define WM8915_AIF1TX_CHAN2_DAT_INV_WIDTH 1 /* AIF1TX_CHAN2_DAT_INV */ +#define WM8915_AIF1TX_CHAN2_SPACING_MASK 0x7E00 /* AIF1TX_CHAN2_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN2_SPACING_SHIFT 9 /* AIF1TX_CHAN2_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN2_SPACING_WIDTH 6 /* AIF1TX_CHAN2_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN2_SLOTS_MASK 0x01C0 /* AIF1TX_CHAN2_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN2_SLOTS_SHIFT 6 /* AIF1TX_CHAN2_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN2_SLOTS_WIDTH 3 /* AIF1TX_CHAN2_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN2_START_SLOT_MASK 0x003F /* AIF1TX_CHAN2_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN2_START_SLOT_SHIFT 0 /* AIF1TX_CHAN2_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN2_START_SLOT_WIDTH 6 /* AIF1TX_CHAN2_START_SLOT - [5:0] */ + +/* + * R780 (0x30C) - AIF1TX Channel 3 Configuration + */ +#define WM8915_AIF1TX_CHAN3_DAT_INV 0x8000 /* AIF1TX_CHAN3_DAT_INV */ +#define WM8915_AIF1TX_CHAN3_DAT_INV_MASK 0x8000 /* AIF1TX_CHAN3_DAT_INV */ +#define WM8915_AIF1TX_CHAN3_DAT_INV_SHIFT 15 /* AIF1TX_CHAN3_DAT_INV */ +#define WM8915_AIF1TX_CHAN3_DAT_INV_WIDTH 1 /* AIF1TX_CHAN3_DAT_INV */ +#define WM8915_AIF1TX_CHAN3_SPACING_MASK 0x7E00 /* AIF1TX_CHAN3_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN3_SPACING_SHIFT 9 /* AIF1TX_CHAN3_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN3_SPACING_WIDTH 6 /* AIF1TX_CHAN3_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN3_SLOTS_MASK 0x01C0 /* AIF1TX_CHAN3_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN3_SLOTS_SHIFT 6 /* AIF1TX_CHAN3_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN3_SLOTS_WIDTH 3 /* AIF1TX_CHAN3_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN3_START_SLOT_MASK 0x003F /* AIF1TX_CHAN3_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN3_START_SLOT_SHIFT 0 /* AIF1TX_CHAN3_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN3_START_SLOT_WIDTH 6 /* AIF1TX_CHAN3_START_SLOT - [5:0] */ + +/* + * R781 (0x30D) - AIF1TX Channel 4 Configuration + */ +#define WM8915_AIF1TX_CHAN4_DAT_INV 0x8000 /* AIF1TX_CHAN4_DAT_INV */ +#define WM8915_AIF1TX_CHAN4_DAT_INV_MASK 0x8000 /* AIF1TX_CHAN4_DAT_INV */ +#define WM8915_AIF1TX_CHAN4_DAT_INV_SHIFT 15 /* AIF1TX_CHAN4_DAT_INV */ +#define WM8915_AIF1TX_CHAN4_DAT_INV_WIDTH 1 /* AIF1TX_CHAN4_DAT_INV */ +#define WM8915_AIF1TX_CHAN4_SPACING_MASK 0x7E00 /* AIF1TX_CHAN4_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN4_SPACING_SHIFT 9 /* AIF1TX_CHAN4_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN4_SPACING_WIDTH 6 /* AIF1TX_CHAN4_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN4_SLOTS_MASK 0x01C0 /* AIF1TX_CHAN4_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN4_SLOTS_SHIFT 6 /* AIF1TX_CHAN4_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN4_SLOTS_WIDTH 3 /* AIF1TX_CHAN4_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN4_START_SLOT_MASK 0x003F /* AIF1TX_CHAN4_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN4_START_SLOT_SHIFT 0 /* AIF1TX_CHAN4_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN4_START_SLOT_WIDTH 6 /* AIF1TX_CHAN4_START_SLOT - [5:0] */ + +/* + * R782 (0x30E) - AIF1TX Channel 5 Configuration + */ +#define WM8915_AIF1TX_CHAN5_DAT_INV 0x8000 /* AIF1TX_CHAN5_DAT_INV */ +#define WM8915_AIF1TX_CHAN5_DAT_INV_MASK 0x8000 /* AIF1TX_CHAN5_DAT_INV */ +#define WM8915_AIF1TX_CHAN5_DAT_INV_SHIFT 15 /* AIF1TX_CHAN5_DAT_INV */ +#define WM8915_AIF1TX_CHAN5_DAT_INV_WIDTH 1 /* AIF1TX_CHAN5_DAT_INV */ +#define WM8915_AIF1TX_CHAN5_SPACING_MASK 0x7E00 /* AIF1TX_CHAN5_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN5_SPACING_SHIFT 9 /* AIF1TX_CHAN5_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN5_SPACING_WIDTH 6 /* AIF1TX_CHAN5_SPACING - [14:9] */ +#define WM8915_AIF1TX_CHAN5_SLOTS_MASK 0x01C0 /* AIF1TX_CHAN5_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN5_SLOTS_SHIFT 6 /* AIF1TX_CHAN5_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN5_SLOTS_WIDTH 3 /* AIF1TX_CHAN5_SLOTS - [8:6] */ +#define WM8915_AIF1TX_CHAN5_START_SLOT_MASK 0x003F /* AIF1TX_CHAN5_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN5_START_SLOT_SHIFT 0 /* AIF1TX_CHAN5_START_SLOT - [5:0] */ +#define WM8915_AIF1TX_CHAN5_START_SLOT_WIDTH 6 /* AIF1TX_CHAN5_START_SLOT - [5:0] */ + +/* + * R783 (0x30F) - AIF1RX Channel 0 Configuration + */ +#define WM8915_AIF1RX_CHAN0_DAT_INV 0x8000 /* AIF1RX_CHAN0_DAT_INV */ +#define WM8915_AIF1RX_CHAN0_DAT_INV_MASK 0x8000 /* AIF1RX_CHAN0_DAT_INV */ +#define WM8915_AIF1RX_CHAN0_DAT_INV_SHIFT 15 /* AIF1RX_CHAN0_DAT_INV */ +#define WM8915_AIF1RX_CHAN0_DAT_INV_WIDTH 1 /* AIF1RX_CHAN0_DAT_INV */ +#define WM8915_AIF1RX_CHAN0_SPACING_MASK 0x7E00 /* AIF1RX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN0_SPACING_SHIFT 9 /* AIF1RX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN0_SPACING_WIDTH 6 /* AIF1RX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN0_SLOTS_MASK 0x01C0 /* AIF1RX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN0_SLOTS_SHIFT 6 /* AIF1RX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN0_SLOTS_WIDTH 3 /* AIF1RX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN0_START_SLOT_MASK 0x003F /* AIF1RX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN0_START_SLOT_SHIFT 0 /* AIF1RX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN0_START_SLOT_WIDTH 6 /* AIF1RX_CHAN0_START_SLOT - [5:0] */ + +/* + * R784 (0x310) - AIF1RX Channel 1 Configuration + */ +#define WM8915_AIF1RX_CHAN1_DAT_INV 0x8000 /* AIF1RX_CHAN1_DAT_INV */ +#define WM8915_AIF1RX_CHAN1_DAT_INV_MASK 0x8000 /* AIF1RX_CHAN1_DAT_INV */ +#define WM8915_AIF1RX_CHAN1_DAT_INV_SHIFT 15 /* AIF1RX_CHAN1_DAT_INV */ +#define WM8915_AIF1RX_CHAN1_DAT_INV_WIDTH 1 /* AIF1RX_CHAN1_DAT_INV */ +#define WM8915_AIF1RX_CHAN1_SPACING_MASK 0x7E00 /* AIF1RX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN1_SPACING_SHIFT 9 /* AIF1RX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN1_SPACING_WIDTH 6 /* AIF1RX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN1_SLOTS_MASK 0x01C0 /* AIF1RX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN1_SLOTS_SHIFT 6 /* AIF1RX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN1_SLOTS_WIDTH 3 /* AIF1RX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN1_START_SLOT_MASK 0x003F /* AIF1RX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN1_START_SLOT_SHIFT 0 /* AIF1RX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN1_START_SLOT_WIDTH 6 /* AIF1RX_CHAN1_START_SLOT - [5:0] */ + +/* + * R785 (0x311) - AIF1RX Channel 2 Configuration + */ +#define WM8915_AIF1RX_CHAN2_DAT_INV 0x8000 /* AIF1RX_CHAN2_DAT_INV */ +#define WM8915_AIF1RX_CHAN2_DAT_INV_MASK 0x8000 /* AIF1RX_CHAN2_DAT_INV */ +#define WM8915_AIF1RX_CHAN2_DAT_INV_SHIFT 15 /* AIF1RX_CHAN2_DAT_INV */ +#define WM8915_AIF1RX_CHAN2_DAT_INV_WIDTH 1 /* AIF1RX_CHAN2_DAT_INV */ +#define WM8915_AIF1RX_CHAN2_SPACING_MASK 0x7E00 /* AIF1RX_CHAN2_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN2_SPACING_SHIFT 9 /* AIF1RX_CHAN2_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN2_SPACING_WIDTH 6 /* AIF1RX_CHAN2_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN2_SLOTS_MASK 0x01C0 /* AIF1RX_CHAN2_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN2_SLOTS_SHIFT 6 /* AIF1RX_CHAN2_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN2_SLOTS_WIDTH 3 /* AIF1RX_CHAN2_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN2_START_SLOT_MASK 0x003F /* AIF1RX_CHAN2_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN2_START_SLOT_SHIFT 0 /* AIF1RX_CHAN2_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN2_START_SLOT_WIDTH 6 /* AIF1RX_CHAN2_START_SLOT - [5:0] */ + +/* + * R786 (0x312) - AIF1RX Channel 3 Configuration + */ +#define WM8915_AIF1RX_CHAN3_DAT_INV 0x8000 /* AIF1RX_CHAN3_DAT_INV */ +#define WM8915_AIF1RX_CHAN3_DAT_INV_MASK 0x8000 /* AIF1RX_CHAN3_DAT_INV */ +#define WM8915_AIF1RX_CHAN3_DAT_INV_SHIFT 15 /* AIF1RX_CHAN3_DAT_INV */ +#define WM8915_AIF1RX_CHAN3_DAT_INV_WIDTH 1 /* AIF1RX_CHAN3_DAT_INV */ +#define WM8915_AIF1RX_CHAN3_SPACING_MASK 0x7E00 /* AIF1RX_CHAN3_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN3_SPACING_SHIFT 9 /* AIF1RX_CHAN3_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN3_SPACING_WIDTH 6 /* AIF1RX_CHAN3_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN3_SLOTS_MASK 0x01C0 /* AIF1RX_CHAN3_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN3_SLOTS_SHIFT 6 /* AIF1RX_CHAN3_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN3_SLOTS_WIDTH 3 /* AIF1RX_CHAN3_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN3_START_SLOT_MASK 0x003F /* AIF1RX_CHAN3_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN3_START_SLOT_SHIFT 0 /* AIF1RX_CHAN3_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN3_START_SLOT_WIDTH 6 /* AIF1RX_CHAN3_START_SLOT - [5:0] */ + +/* + * R787 (0x313) - AIF1RX Channel 4 Configuration + */ +#define WM8915_AIF1RX_CHAN4_DAT_INV 0x8000 /* AIF1RX_CHAN4_DAT_INV */ +#define WM8915_AIF1RX_CHAN4_DAT_INV_MASK 0x8000 /* AIF1RX_CHAN4_DAT_INV */ +#define WM8915_AIF1RX_CHAN4_DAT_INV_SHIFT 15 /* AIF1RX_CHAN4_DAT_INV */ +#define WM8915_AIF1RX_CHAN4_DAT_INV_WIDTH 1 /* AIF1RX_CHAN4_DAT_INV */ +#define WM8915_AIF1RX_CHAN4_SPACING_MASK 0x7E00 /* AIF1RX_CHAN4_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN4_SPACING_SHIFT 9 /* AIF1RX_CHAN4_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN4_SPACING_WIDTH 6 /* AIF1RX_CHAN4_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN4_SLOTS_MASK 0x01C0 /* AIF1RX_CHAN4_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN4_SLOTS_SHIFT 6 /* AIF1RX_CHAN4_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN4_SLOTS_WIDTH 3 /* AIF1RX_CHAN4_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN4_START_SLOT_MASK 0x003F /* AIF1RX_CHAN4_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN4_START_SLOT_SHIFT 0 /* AIF1RX_CHAN4_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN4_START_SLOT_WIDTH 6 /* AIF1RX_CHAN4_START_SLOT - [5:0] */ + +/* + * R788 (0x314) - AIF1RX Channel 5 Configuration + */ +#define WM8915_AIF1RX_CHAN5_DAT_INV 0x8000 /* AIF1RX_CHAN5_DAT_INV */ +#define WM8915_AIF1RX_CHAN5_DAT_INV_MASK 0x8000 /* AIF1RX_CHAN5_DAT_INV */ +#define WM8915_AIF1RX_CHAN5_DAT_INV_SHIFT 15 /* AIF1RX_CHAN5_DAT_INV */ +#define WM8915_AIF1RX_CHAN5_DAT_INV_WIDTH 1 /* AIF1RX_CHAN5_DAT_INV */ +#define WM8915_AIF1RX_CHAN5_SPACING_MASK 0x7E00 /* AIF1RX_CHAN5_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN5_SPACING_SHIFT 9 /* AIF1RX_CHAN5_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN5_SPACING_WIDTH 6 /* AIF1RX_CHAN5_SPACING - [14:9] */ +#define WM8915_AIF1RX_CHAN5_SLOTS_MASK 0x01C0 /* AIF1RX_CHAN5_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN5_SLOTS_SHIFT 6 /* AIF1RX_CHAN5_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN5_SLOTS_WIDTH 3 /* AIF1RX_CHAN5_SLOTS - [8:6] */ +#define WM8915_AIF1RX_CHAN5_START_SLOT_MASK 0x003F /* AIF1RX_CHAN5_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN5_START_SLOT_SHIFT 0 /* AIF1RX_CHAN5_START_SLOT - [5:0] */ +#define WM8915_AIF1RX_CHAN5_START_SLOT_WIDTH 6 /* AIF1RX_CHAN5_START_SLOT - [5:0] */ + +/* + * R789 (0x315) - AIF1RX Mono Configuration + */ +#define WM8915_AIF1RX_CHAN4_MONO_MODE 0x0004 /* AIF1RX_CHAN4_MONO_MODE */ +#define WM8915_AIF1RX_CHAN4_MONO_MODE_MASK 0x0004 /* AIF1RX_CHAN4_MONO_MODE */ +#define WM8915_AIF1RX_CHAN4_MONO_MODE_SHIFT 2 /* AIF1RX_CHAN4_MONO_MODE */ +#define WM8915_AIF1RX_CHAN4_MONO_MODE_WIDTH 1 /* AIF1RX_CHAN4_MONO_MODE */ +#define WM8915_AIF1RX_CHAN2_MONO_MODE 0x0002 /* AIF1RX_CHAN2_MONO_MODE */ +#define WM8915_AIF1RX_CHAN2_MONO_MODE_MASK 0x0002 /* AIF1RX_CHAN2_MONO_MODE */ +#define WM8915_AIF1RX_CHAN2_MONO_MODE_SHIFT 1 /* AIF1RX_CHAN2_MONO_MODE */ +#define WM8915_AIF1RX_CHAN2_MONO_MODE_WIDTH 1 /* AIF1RX_CHAN2_MONO_MODE */ +#define WM8915_AIF1RX_CHAN0_MONO_MODE 0x0001 /* AIF1RX_CHAN0_MONO_MODE */ +#define WM8915_AIF1RX_CHAN0_MONO_MODE_MASK 0x0001 /* AIF1RX_CHAN0_MONO_MODE */ +#define WM8915_AIF1RX_CHAN0_MONO_MODE_SHIFT 0 /* AIF1RX_CHAN0_MONO_MODE */ +#define WM8915_AIF1RX_CHAN0_MONO_MODE_WIDTH 1 /* AIF1RX_CHAN0_MONO_MODE */ + +/* + * R794 (0x31A) - AIF1TX Test + */ +#define WM8915_AIF1TX45_DITHER_ENA 0x0004 /* AIF1TX45_DITHER_ENA */ +#define WM8915_AIF1TX45_DITHER_ENA_MASK 0x0004 /* AIF1TX45_DITHER_ENA */ +#define WM8915_AIF1TX45_DITHER_ENA_SHIFT 2 /* AIF1TX45_DITHER_ENA */ +#define WM8915_AIF1TX45_DITHER_ENA_WIDTH 1 /* AIF1TX45_DITHER_ENA */ +#define WM8915_AIF1TX23_DITHER_ENA 0x0002 /* AIF1TX23_DITHER_ENA */ +#define WM8915_AIF1TX23_DITHER_ENA_MASK 0x0002 /* AIF1TX23_DITHER_ENA */ +#define WM8915_AIF1TX23_DITHER_ENA_SHIFT 1 /* AIF1TX23_DITHER_ENA */ +#define WM8915_AIF1TX23_DITHER_ENA_WIDTH 1 /* AIF1TX23_DITHER_ENA */ +#define WM8915_AIF1TX01_DITHER_ENA 0x0001 /* AIF1TX01_DITHER_ENA */ +#define WM8915_AIF1TX01_DITHER_ENA_MASK 0x0001 /* AIF1TX01_DITHER_ENA */ +#define WM8915_AIF1TX01_DITHER_ENA_SHIFT 0 /* AIF1TX01_DITHER_ENA */ +#define WM8915_AIF1TX01_DITHER_ENA_WIDTH 1 /* AIF1TX01_DITHER_ENA */ + +/* + * R800 (0x320) - AIF2 Control + */ +#define WM8915_AIF2_TRI 0x0004 /* AIF2_TRI */ +#define WM8915_AIF2_TRI_MASK 0x0004 /* AIF2_TRI */ +#define WM8915_AIF2_TRI_SHIFT 2 /* AIF2_TRI */ +#define WM8915_AIF2_TRI_WIDTH 1 /* AIF2_TRI */ +#define WM8915_AIF2_FMT_MASK 0x0003 /* AIF2_FMT - [1:0] */ +#define WM8915_AIF2_FMT_SHIFT 0 /* AIF2_FMT - [1:0] */ +#define WM8915_AIF2_FMT_WIDTH 2 /* AIF2_FMT - [1:0] */ + +/* + * R801 (0x321) - AIF2 BCLK + */ +#define WM8915_AIF2_BCLK_INV 0x0400 /* AIF2_BCLK_INV */ +#define WM8915_AIF2_BCLK_INV_MASK 0x0400 /* AIF2_BCLK_INV */ +#define WM8915_AIF2_BCLK_INV_SHIFT 10 /* AIF2_BCLK_INV */ +#define WM8915_AIF2_BCLK_INV_WIDTH 1 /* AIF2_BCLK_INV */ +#define WM8915_AIF2_BCLK_FRC 0x0200 /* AIF2_BCLK_FRC */ +#define WM8915_AIF2_BCLK_FRC_MASK 0x0200 /* AIF2_BCLK_FRC */ +#define WM8915_AIF2_BCLK_FRC_SHIFT 9 /* AIF2_BCLK_FRC */ +#define WM8915_AIF2_BCLK_FRC_WIDTH 1 /* AIF2_BCLK_FRC */ +#define WM8915_AIF2_BCLK_MSTR 0x0100 /* AIF2_BCLK_MSTR */ +#define WM8915_AIF2_BCLK_MSTR_MASK 0x0100 /* AIF2_BCLK_MSTR */ +#define WM8915_AIF2_BCLK_MSTR_SHIFT 8 /* AIF2_BCLK_MSTR */ +#define WM8915_AIF2_BCLK_MSTR_WIDTH 1 /* AIF2_BCLK_MSTR */ +#define WM8915_AIF2_BCLK_DIV_MASK 0x000F /* AIF2_BCLK_DIV - [3:0] */ +#define WM8915_AIF2_BCLK_DIV_SHIFT 0 /* AIF2_BCLK_DIV - [3:0] */ +#define WM8915_AIF2_BCLK_DIV_WIDTH 4 /* AIF2_BCLK_DIV - [3:0] */ + +/* + * R802 (0x322) - AIF2 TX LRCLK(1) + */ +#define WM8915_AIF2TX_RATE_MASK 0x07FF /* AIF2TX_RATE - [10:0] */ +#define WM8915_AIF2TX_RATE_SHIFT 0 /* AIF2TX_RATE - [10:0] */ +#define WM8915_AIF2TX_RATE_WIDTH 11 /* AIF2TX_RATE - [10:0] */ + +/* + * R803 (0x323) - AIF2 TX LRCLK(2) + */ +#define WM8915_AIF2TX_LRCLK_MODE 0x0008 /* AIF2TX_LRCLK_MODE */ +#define WM8915_AIF2TX_LRCLK_MODE_MASK 0x0008 /* AIF2TX_LRCLK_MODE */ +#define WM8915_AIF2TX_LRCLK_MODE_SHIFT 3 /* AIF2TX_LRCLK_MODE */ +#define WM8915_AIF2TX_LRCLK_MODE_WIDTH 1 /* AIF2TX_LRCLK_MODE */ +#define WM8915_AIF2TX_LRCLK_INV 0x0004 /* AIF2TX_LRCLK_INV */ +#define WM8915_AIF2TX_LRCLK_INV_MASK 0x0004 /* AIF2TX_LRCLK_INV */ +#define WM8915_AIF2TX_LRCLK_INV_SHIFT 2 /* AIF2TX_LRCLK_INV */ +#define WM8915_AIF2TX_LRCLK_INV_WIDTH 1 /* AIF2TX_LRCLK_INV */ +#define WM8915_AIF2TX_LRCLK_FRC 0x0002 /* AIF2TX_LRCLK_FRC */ +#define WM8915_AIF2TX_LRCLK_FRC_MASK 0x0002 /* AIF2TX_LRCLK_FRC */ +#define WM8915_AIF2TX_LRCLK_FRC_SHIFT 1 /* AIF2TX_LRCLK_FRC */ +#define WM8915_AIF2TX_LRCLK_FRC_WIDTH 1 /* AIF2TX_LRCLK_FRC */ +#define WM8915_AIF2TX_LRCLK_MSTR 0x0001 /* AIF2TX_LRCLK_MSTR */ +#define WM8915_AIF2TX_LRCLK_MSTR_MASK 0x0001 /* AIF2TX_LRCLK_MSTR */ +#define WM8915_AIF2TX_LRCLK_MSTR_SHIFT 0 /* AIF2TX_LRCLK_MSTR */ +#define WM8915_AIF2TX_LRCLK_MSTR_WIDTH 1 /* AIF2TX_LRCLK_MSTR */ + +/* + * R804 (0x324) - AIF2 RX LRCLK(1) + */ +#define WM8915_AIF2RX_RATE_MASK 0x07FF /* AIF2RX_RATE - [10:0] */ +#define WM8915_AIF2RX_RATE_SHIFT 0 /* AIF2RX_RATE - [10:0] */ +#define WM8915_AIF2RX_RATE_WIDTH 11 /* AIF2RX_RATE - [10:0] */ + +/* + * R805 (0x325) - AIF2 RX LRCLK(2) + */ +#define WM8915_AIF2RX_LRCLK_INV 0x0004 /* AIF2RX_LRCLK_INV */ +#define WM8915_AIF2RX_LRCLK_INV_MASK 0x0004 /* AIF2RX_LRCLK_INV */ +#define WM8915_AIF2RX_LRCLK_INV_SHIFT 2 /* AIF2RX_LRCLK_INV */ +#define WM8915_AIF2RX_LRCLK_INV_WIDTH 1 /* AIF2RX_LRCLK_INV */ +#define WM8915_AIF2RX_LRCLK_FRC 0x0002 /* AIF2RX_LRCLK_FRC */ +#define WM8915_AIF2RX_LRCLK_FRC_MASK 0x0002 /* AIF2RX_LRCLK_FRC */ +#define WM8915_AIF2RX_LRCLK_FRC_SHIFT 1 /* AIF2RX_LRCLK_FRC */ +#define WM8915_AIF2RX_LRCLK_FRC_WIDTH 1 /* AIF2RX_LRCLK_FRC */ +#define WM8915_AIF2RX_LRCLK_MSTR 0x0001 /* AIF2RX_LRCLK_MSTR */ +#define WM8915_AIF2RX_LRCLK_MSTR_MASK 0x0001 /* AIF2RX_LRCLK_MSTR */ +#define WM8915_AIF2RX_LRCLK_MSTR_SHIFT 0 /* AIF2RX_LRCLK_MSTR */ +#define WM8915_AIF2RX_LRCLK_MSTR_WIDTH 1 /* AIF2RX_LRCLK_MSTR */ + +/* + * R806 (0x326) - AIF2TX Data Configuration (1) + */ +#define WM8915_AIF2TX_WL_MASK 0xFF00 /* AIF2TX_WL - [15:8] */ +#define WM8915_AIF2TX_WL_SHIFT 8 /* AIF2TX_WL - [15:8] */ +#define WM8915_AIF2TX_WL_WIDTH 8 /* AIF2TX_WL - [15:8] */ +#define WM8915_AIF2TX_SLOT_LEN_MASK 0x00FF /* AIF2TX_SLOT_LEN - [7:0] */ +#define WM8915_AIF2TX_SLOT_LEN_SHIFT 0 /* AIF2TX_SLOT_LEN - [7:0] */ +#define WM8915_AIF2TX_SLOT_LEN_WIDTH 8 /* AIF2TX_SLOT_LEN - [7:0] */ + +/* + * R807 (0x327) - AIF2TX Data Configuration (2) + */ +#define WM8915_AIF2TX_DAT_TRI 0x0001 /* AIF2TX_DAT_TRI */ +#define WM8915_AIF2TX_DAT_TRI_MASK 0x0001 /* AIF2TX_DAT_TRI */ +#define WM8915_AIF2TX_DAT_TRI_SHIFT 0 /* AIF2TX_DAT_TRI */ +#define WM8915_AIF2TX_DAT_TRI_WIDTH 1 /* AIF2TX_DAT_TRI */ + +/* + * R808 (0x328) - AIF2RX Data Configuration + */ +#define WM8915_AIF2RX_WL_MASK 0xFF00 /* AIF2RX_WL - [15:8] */ +#define WM8915_AIF2RX_WL_SHIFT 8 /* AIF2RX_WL - [15:8] */ +#define WM8915_AIF2RX_WL_WIDTH 8 /* AIF2RX_WL - [15:8] */ +#define WM8915_AIF2RX_SLOT_LEN_MASK 0x00FF /* AIF2RX_SLOT_LEN - [7:0] */ +#define WM8915_AIF2RX_SLOT_LEN_SHIFT 0 /* AIF2RX_SLOT_LEN - [7:0] */ +#define WM8915_AIF2RX_SLOT_LEN_WIDTH 8 /* AIF2RX_SLOT_LEN - [7:0] */ + +/* + * R809 (0x329) - AIF2TX Channel 0 Configuration + */ +#define WM8915_AIF2TX_CHAN0_DAT_INV 0x8000 /* AIF2TX_CHAN0_DAT_INV */ +#define WM8915_AIF2TX_CHAN0_DAT_INV_MASK 0x8000 /* AIF2TX_CHAN0_DAT_INV */ +#define WM8915_AIF2TX_CHAN0_DAT_INV_SHIFT 15 /* AIF2TX_CHAN0_DAT_INV */ +#define WM8915_AIF2TX_CHAN0_DAT_INV_WIDTH 1 /* AIF2TX_CHAN0_DAT_INV */ +#define WM8915_AIF2TX_CHAN0_SPACING_MASK 0x7E00 /* AIF2TX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF2TX_CHAN0_SPACING_SHIFT 9 /* AIF2TX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF2TX_CHAN0_SPACING_WIDTH 6 /* AIF2TX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF2TX_CHAN0_SLOTS_MASK 0x01C0 /* AIF2TX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF2TX_CHAN0_SLOTS_SHIFT 6 /* AIF2TX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF2TX_CHAN0_SLOTS_WIDTH 3 /* AIF2TX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF2TX_CHAN0_START_SLOT_MASK 0x003F /* AIF2TX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF2TX_CHAN0_START_SLOT_SHIFT 0 /* AIF2TX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF2TX_CHAN0_START_SLOT_WIDTH 6 /* AIF2TX_CHAN0_START_SLOT - [5:0] */ + +/* + * R810 (0x32A) - AIF2TX Channel 1 Configuration + */ +#define WM8915_AIF2TX_CHAN1_DAT_INV 0x8000 /* AIF2TX_CHAN1_DAT_INV */ +#define WM8915_AIF2TX_CHAN1_DAT_INV_MASK 0x8000 /* AIF2TX_CHAN1_DAT_INV */ +#define WM8915_AIF2TX_CHAN1_DAT_INV_SHIFT 15 /* AIF2TX_CHAN1_DAT_INV */ +#define WM8915_AIF2TX_CHAN1_DAT_INV_WIDTH 1 /* AIF2TX_CHAN1_DAT_INV */ +#define WM8915_AIF2TX_CHAN1_SPACING_MASK 0x7E00 /* AIF2TX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF2TX_CHAN1_SPACING_SHIFT 9 /* AIF2TX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF2TX_CHAN1_SPACING_WIDTH 6 /* AIF2TX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF2TX_CHAN1_SLOTS_MASK 0x01C0 /* AIF2TX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF2TX_CHAN1_SLOTS_SHIFT 6 /* AIF2TX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF2TX_CHAN1_SLOTS_WIDTH 3 /* AIF2TX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF2TX_CHAN1_START_SLOT_MASK 0x003F /* AIF2TX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF2TX_CHAN1_START_SLOT_SHIFT 0 /* AIF2TX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF2TX_CHAN1_START_SLOT_WIDTH 6 /* AIF2TX_CHAN1_START_SLOT - [5:0] */ + +/* + * R811 (0x32B) - AIF2RX Channel 0 Configuration + */ +#define WM8915_AIF2RX_CHAN0_DAT_INV 0x8000 /* AIF2RX_CHAN0_DAT_INV */ +#define WM8915_AIF2RX_CHAN0_DAT_INV_MASK 0x8000 /* AIF2RX_CHAN0_DAT_INV */ +#define WM8915_AIF2RX_CHAN0_DAT_INV_SHIFT 15 /* AIF2RX_CHAN0_DAT_INV */ +#define WM8915_AIF2RX_CHAN0_DAT_INV_WIDTH 1 /* AIF2RX_CHAN0_DAT_INV */ +#define WM8915_AIF2RX_CHAN0_SPACING_MASK 0x7E00 /* AIF2RX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF2RX_CHAN0_SPACING_SHIFT 9 /* AIF2RX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF2RX_CHAN0_SPACING_WIDTH 6 /* AIF2RX_CHAN0_SPACING - [14:9] */ +#define WM8915_AIF2RX_CHAN0_SLOTS_MASK 0x01C0 /* AIF2RX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF2RX_CHAN0_SLOTS_SHIFT 6 /* AIF2RX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF2RX_CHAN0_SLOTS_WIDTH 3 /* AIF2RX_CHAN0_SLOTS - [8:6] */ +#define WM8915_AIF2RX_CHAN0_START_SLOT_MASK 0x003F /* AIF2RX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF2RX_CHAN0_START_SLOT_SHIFT 0 /* AIF2RX_CHAN0_START_SLOT - [5:0] */ +#define WM8915_AIF2RX_CHAN0_START_SLOT_WIDTH 6 /* AIF2RX_CHAN0_START_SLOT - [5:0] */ + +/* + * R812 (0x32C) - AIF2RX Channel 1 Configuration + */ +#define WM8915_AIF2RX_CHAN1_DAT_INV 0x8000 /* AIF2RX_CHAN1_DAT_INV */ +#define WM8915_AIF2RX_CHAN1_DAT_INV_MASK 0x8000 /* AIF2RX_CHAN1_DAT_INV */ +#define WM8915_AIF2RX_CHAN1_DAT_INV_SHIFT 15 /* AIF2RX_CHAN1_DAT_INV */ +#define WM8915_AIF2RX_CHAN1_DAT_INV_WIDTH 1 /* AIF2RX_CHAN1_DAT_INV */ +#define WM8915_AIF2RX_CHAN1_SPACING_MASK 0x7E00 /* AIF2RX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF2RX_CHAN1_SPACING_SHIFT 9 /* AIF2RX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF2RX_CHAN1_SPACING_WIDTH 6 /* AIF2RX_CHAN1_SPACING - [14:9] */ +#define WM8915_AIF2RX_CHAN1_SLOTS_MASK 0x01C0 /* AIF2RX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF2RX_CHAN1_SLOTS_SHIFT 6 /* AIF2RX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF2RX_CHAN1_SLOTS_WIDTH 3 /* AIF2RX_CHAN1_SLOTS - [8:6] */ +#define WM8915_AIF2RX_CHAN1_START_SLOT_MASK 0x003F /* AIF2RX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF2RX_CHAN1_START_SLOT_SHIFT 0 /* AIF2RX_CHAN1_START_SLOT - [5:0] */ +#define WM8915_AIF2RX_CHAN1_START_SLOT_WIDTH 6 /* AIF2RX_CHAN1_START_SLOT - [5:0] */ + +/* + * R813 (0x32D) - AIF2RX Mono Configuration + */ +#define WM8915_AIF2RX_CHAN0_MONO_MODE 0x0001 /* AIF2RX_CHAN0_MONO_MODE */ +#define WM8915_AIF2RX_CHAN0_MONO_MODE_MASK 0x0001 /* AIF2RX_CHAN0_MONO_MODE */ +#define WM8915_AIF2RX_CHAN0_MONO_MODE_SHIFT 0 /* AIF2RX_CHAN0_MONO_MODE */ +#define WM8915_AIF2RX_CHAN0_MONO_MODE_WIDTH 1 /* AIF2RX_CHAN0_MONO_MODE */ + +/* + * R815 (0x32F) - AIF2TX Test + */ +#define WM8915_AIF2TX_DITHER_ENA 0x0001 /* AIF2TX_DITHER_ENA */ +#define WM8915_AIF2TX_DITHER_ENA_MASK 0x0001 /* AIF2TX_DITHER_ENA */ +#define WM8915_AIF2TX_DITHER_ENA_SHIFT 0 /* AIF2TX_DITHER_ENA */ +#define WM8915_AIF2TX_DITHER_ENA_WIDTH 1 /* AIF2TX_DITHER_ENA */ + +/* + * R1024 (0x400) - DSP1 TX Left Volume + */ +#define WM8915_DSP1TX_VU 0x0100 /* DSP1TX_VU */ +#define WM8915_DSP1TX_VU_MASK 0x0100 /* DSP1TX_VU */ +#define WM8915_DSP1TX_VU_SHIFT 8 /* DSP1TX_VU */ +#define WM8915_DSP1TX_VU_WIDTH 1 /* DSP1TX_VU */ +#define WM8915_DSP1TXL_VOL_MASK 0x00FF /* DSP1TXL_VOL - [7:0] */ +#define WM8915_DSP1TXL_VOL_SHIFT 0 /* DSP1TXL_VOL - [7:0] */ +#define WM8915_DSP1TXL_VOL_WIDTH 8 /* DSP1TXL_VOL - [7:0] */ + +/* + * R1025 (0x401) - DSP1 TX Right Volume + */ +#define WM8915_DSP1TX_VU 0x0100 /* DSP1TX_VU */ +#define WM8915_DSP1TX_VU_MASK 0x0100 /* DSP1TX_VU */ +#define WM8915_DSP1TX_VU_SHIFT 8 /* DSP1TX_VU */ +#define WM8915_DSP1TX_VU_WIDTH 1 /* DSP1TX_VU */ +#define WM8915_DSP1TXR_VOL_MASK 0x00FF /* DSP1TXR_VOL - [7:0] */ +#define WM8915_DSP1TXR_VOL_SHIFT 0 /* DSP1TXR_VOL - [7:0] */ +#define WM8915_DSP1TXR_VOL_WIDTH 8 /* DSP1TXR_VOL - [7:0] */ + +/* + * R1026 (0x402) - DSP1 RX Left Volume + */ +#define WM8915_DSP1RX_VU 0x0100 /* DSP1RX_VU */ +#define WM8915_DSP1RX_VU_MASK 0x0100 /* DSP1RX_VU */ +#define WM8915_DSP1RX_VU_SHIFT 8 /* DSP1RX_VU */ +#define WM8915_DSP1RX_VU_WIDTH 1 /* DSP1RX_VU */ +#define WM8915_DSP1RXL_VOL_MASK 0x00FF /* DSP1RXL_VOL - [7:0] */ +#define WM8915_DSP1RXL_VOL_SHIFT 0 /* DSP1RXL_VOL - [7:0] */ +#define WM8915_DSP1RXL_VOL_WIDTH 8 /* DSP1RXL_VOL - [7:0] */ + +/* + * R1027 (0x403) - DSP1 RX Right Volume + */ +#define WM8915_DSP1RX_VU 0x0100 /* DSP1RX_VU */ +#define WM8915_DSP1RX_VU_MASK 0x0100 /* DSP1RX_VU */ +#define WM8915_DSP1RX_VU_SHIFT 8 /* DSP1RX_VU */ +#define WM8915_DSP1RX_VU_WIDTH 1 /* DSP1RX_VU */ +#define WM8915_DSP1RXR_VOL_MASK 0x00FF /* DSP1RXR_VOL - [7:0] */ +#define WM8915_DSP1RXR_VOL_SHIFT 0 /* DSP1RXR_VOL - [7:0] */ +#define WM8915_DSP1RXR_VOL_WIDTH 8 /* DSP1RXR_VOL - [7:0] */ + +/* + * R1040 (0x410) - DSP1 TX Filters + */ +#define WM8915_DSP1TX_NF 0x2000 /* DSP1TX_NF */ +#define WM8915_DSP1TX_NF_MASK 0x2000 /* DSP1TX_NF */ +#define WM8915_DSP1TX_NF_SHIFT 13 /* DSP1TX_NF */ +#define WM8915_DSP1TX_NF_WIDTH 1 /* DSP1TX_NF */ +#define WM8915_DSP1TXL_HPF 0x1000 /* DSP1TXL_HPF */ +#define WM8915_DSP1TXL_HPF_MASK 0x1000 /* DSP1TXL_HPF */ +#define WM8915_DSP1TXL_HPF_SHIFT 12 /* DSP1TXL_HPF */ +#define WM8915_DSP1TXL_HPF_WIDTH 1 /* DSP1TXL_HPF */ +#define WM8915_DSP1TXR_HPF 0x0800 /* DSP1TXR_HPF */ +#define WM8915_DSP1TXR_HPF_MASK 0x0800 /* DSP1TXR_HPF */ +#define WM8915_DSP1TXR_HPF_SHIFT 11 /* DSP1TXR_HPF */ +#define WM8915_DSP1TXR_HPF_WIDTH 1 /* DSP1TXR_HPF */ +#define WM8915_DSP1TX_HPF_MODE_MASK 0x0018 /* DSP1TX_HPF_MODE - [4:3] */ +#define WM8915_DSP1TX_HPF_MODE_SHIFT 3 /* DSP1TX_HPF_MODE - [4:3] */ +#define WM8915_DSP1TX_HPF_MODE_WIDTH 2 /* DSP1TX_HPF_MODE - [4:3] */ +#define WM8915_DSP1TX_HPF_CUT_MASK 0x0007 /* DSP1TX_HPF_CUT - [2:0] */ +#define WM8915_DSP1TX_HPF_CUT_SHIFT 0 /* DSP1TX_HPF_CUT - [2:0] */ +#define WM8915_DSP1TX_HPF_CUT_WIDTH 3 /* DSP1TX_HPF_CUT - [2:0] */ + +/* + * R1056 (0x420) - DSP1 RX Filters (1) + */ +#define WM8915_DSP1RX_MUTE 0x0200 /* DSP1RX_MUTE */ +#define WM8915_DSP1RX_MUTE_MASK 0x0200 /* DSP1RX_MUTE */ +#define WM8915_DSP1RX_MUTE_SHIFT 9 /* DSP1RX_MUTE */ +#define WM8915_DSP1RX_MUTE_WIDTH 1 /* DSP1RX_MUTE */ +#define WM8915_DSP1RX_MONO 0x0080 /* DSP1RX_MONO */ +#define WM8915_DSP1RX_MONO_MASK 0x0080 /* DSP1RX_MONO */ +#define WM8915_DSP1RX_MONO_SHIFT 7 /* DSP1RX_MONO */ +#define WM8915_DSP1RX_MONO_WIDTH 1 /* DSP1RX_MONO */ +#define WM8915_DSP1RX_MUTERATE 0x0020 /* DSP1RX_MUTERATE */ +#define WM8915_DSP1RX_MUTERATE_MASK 0x0020 /* DSP1RX_MUTERATE */ +#define WM8915_DSP1RX_MUTERATE_SHIFT 5 /* DSP1RX_MUTERATE */ +#define WM8915_DSP1RX_MUTERATE_WIDTH 1 /* DSP1RX_MUTERATE */ +#define WM8915_DSP1RX_UNMUTE_RAMP 0x0010 /* DSP1RX_UNMUTE_RAMP */ +#define WM8915_DSP1RX_UNMUTE_RAMP_MASK 0x0010 /* DSP1RX_UNMUTE_RAMP */ +#define WM8915_DSP1RX_UNMUTE_RAMP_SHIFT 4 /* DSP1RX_UNMUTE_RAMP */ +#define WM8915_DSP1RX_UNMUTE_RAMP_WIDTH 1 /* DSP1RX_UNMUTE_RAMP */ + +/* + * R1057 (0x421) - DSP1 RX Filters (2) + */ +#define WM8915_DSP1RX_3D_GAIN_MASK 0x3E00 /* DSP1RX_3D_GAIN - [13:9] */ +#define WM8915_DSP1RX_3D_GAIN_SHIFT 9 /* DSP1RX_3D_GAIN - [13:9] */ +#define WM8915_DSP1RX_3D_GAIN_WIDTH 5 /* DSP1RX_3D_GAIN - [13:9] */ +#define WM8915_DSP1RX_3D_ENA 0x0100 /* DSP1RX_3D_ENA */ +#define WM8915_DSP1RX_3D_ENA_MASK 0x0100 /* DSP1RX_3D_ENA */ +#define WM8915_DSP1RX_3D_ENA_SHIFT 8 /* DSP1RX_3D_ENA */ +#define WM8915_DSP1RX_3D_ENA_WIDTH 1 /* DSP1RX_3D_ENA */ + +/* + * R1088 (0x440) - DSP1 DRC (1) + */ +#define WM8915_DSP1DRC_SIG_DET_RMS_MASK 0xF800 /* DSP1DRC_SIG_DET_RMS - [15:11] */ +#define WM8915_DSP1DRC_SIG_DET_RMS_SHIFT 11 /* DSP1DRC_SIG_DET_RMS - [15:11] */ +#define WM8915_DSP1DRC_SIG_DET_RMS_WIDTH 5 /* DSP1DRC_SIG_DET_RMS - [15:11] */ +#define WM8915_DSP1DRC_SIG_DET_PK_MASK 0x0600 /* DSP1DRC_SIG_DET_PK - [10:9] */ +#define WM8915_DSP1DRC_SIG_DET_PK_SHIFT 9 /* DSP1DRC_SIG_DET_PK - [10:9] */ +#define WM8915_DSP1DRC_SIG_DET_PK_WIDTH 2 /* DSP1DRC_SIG_DET_PK - [10:9] */ +#define WM8915_DSP1DRC_NG_ENA 0x0100 /* DSP1DRC_NG_ENA */ +#define WM8915_DSP1DRC_NG_ENA_MASK 0x0100 /* DSP1DRC_NG_ENA */ +#define WM8915_DSP1DRC_NG_ENA_SHIFT 8 /* DSP1DRC_NG_ENA */ +#define WM8915_DSP1DRC_NG_ENA_WIDTH 1 /* DSP1DRC_NG_ENA */ +#define WM8915_DSP1DRC_SIG_DET_MODE 0x0080 /* DSP1DRC_SIG_DET_MODE */ +#define WM8915_DSP1DRC_SIG_DET_MODE_MASK 0x0080 /* DSP1DRC_SIG_DET_MODE */ +#define WM8915_DSP1DRC_SIG_DET_MODE_SHIFT 7 /* DSP1DRC_SIG_DET_MODE */ +#define WM8915_DSP1DRC_SIG_DET_MODE_WIDTH 1 /* DSP1DRC_SIG_DET_MODE */ +#define WM8915_DSP1DRC_SIG_DET 0x0040 /* DSP1DRC_SIG_DET */ +#define WM8915_DSP1DRC_SIG_DET_MASK 0x0040 /* DSP1DRC_SIG_DET */ +#define WM8915_DSP1DRC_SIG_DET_SHIFT 6 /* DSP1DRC_SIG_DET */ +#define WM8915_DSP1DRC_SIG_DET_WIDTH 1 /* DSP1DRC_SIG_DET */ +#define WM8915_DSP1DRC_KNEE2_OP_ENA 0x0020 /* DSP1DRC_KNEE2_OP_ENA */ +#define WM8915_DSP1DRC_KNEE2_OP_ENA_MASK 0x0020 /* DSP1DRC_KNEE2_OP_ENA */ +#define WM8915_DSP1DRC_KNEE2_OP_ENA_SHIFT 5 /* DSP1DRC_KNEE2_OP_ENA */ +#define WM8915_DSP1DRC_KNEE2_OP_ENA_WIDTH 1 /* DSP1DRC_KNEE2_OP_ENA */ +#define WM8915_DSP1DRC_QR 0x0010 /* DSP1DRC_QR */ +#define WM8915_DSP1DRC_QR_MASK 0x0010 /* DSP1DRC_QR */ +#define WM8915_DSP1DRC_QR_SHIFT 4 /* DSP1DRC_QR */ +#define WM8915_DSP1DRC_QR_WIDTH 1 /* DSP1DRC_QR */ +#define WM8915_DSP1DRC_ANTICLIP 0x0008 /* DSP1DRC_ANTICLIP */ +#define WM8915_DSP1DRC_ANTICLIP_MASK 0x0008 /* DSP1DRC_ANTICLIP */ +#define WM8915_DSP1DRC_ANTICLIP_SHIFT 3 /* DSP1DRC_ANTICLIP */ +#define WM8915_DSP1DRC_ANTICLIP_WIDTH 1 /* DSP1DRC_ANTICLIP */ +#define WM8915_DSP1RX_DRC_ENA 0x0004 /* DSP1RX_DRC_ENA */ +#define WM8915_DSP1RX_DRC_ENA_MASK 0x0004 /* DSP1RX_DRC_ENA */ +#define WM8915_DSP1RX_DRC_ENA_SHIFT 2 /* DSP1RX_DRC_ENA */ +#define WM8915_DSP1RX_DRC_ENA_WIDTH 1 /* DSP1RX_DRC_ENA */ +#define WM8915_DSP1TXL_DRC_ENA 0x0002 /* DSP1TXL_DRC_ENA */ +#define WM8915_DSP1TXL_DRC_ENA_MASK 0x0002 /* DSP1TXL_DRC_ENA */ +#define WM8915_DSP1TXL_DRC_ENA_SHIFT 1 /* DSP1TXL_DRC_ENA */ +#define WM8915_DSP1TXL_DRC_ENA_WIDTH 1 /* DSP1TXL_DRC_ENA */ +#define WM8915_DSP1TXR_DRC_ENA 0x0001 /* DSP1TXR_DRC_ENA */ +#define WM8915_DSP1TXR_DRC_ENA_MASK 0x0001 /* DSP1TXR_DRC_ENA */ +#define WM8915_DSP1TXR_DRC_ENA_SHIFT 0 /* DSP1TXR_DRC_ENA */ +#define WM8915_DSP1TXR_DRC_ENA_WIDTH 1 /* DSP1TXR_DRC_ENA */ + +/* + * R1089 (0x441) - DSP1 DRC (2) + */ +#define WM8915_DSP1DRC_ATK_MASK 0x1E00 /* DSP1DRC_ATK - [12:9] */ +#define WM8915_DSP1DRC_ATK_SHIFT 9 /* DSP1DRC_ATK - [12:9] */ +#define WM8915_DSP1DRC_ATK_WIDTH 4 /* DSP1DRC_ATK - [12:9] */ +#define WM8915_DSP1DRC_DCY_MASK 0x01E0 /* DSP1DRC_DCY - [8:5] */ +#define WM8915_DSP1DRC_DCY_SHIFT 5 /* DSP1DRC_DCY - [8:5] */ +#define WM8915_DSP1DRC_DCY_WIDTH 4 /* DSP1DRC_DCY - [8:5] */ +#define WM8915_DSP1DRC_MINGAIN_MASK 0x001C /* DSP1DRC_MINGAIN - [4:2] */ +#define WM8915_DSP1DRC_MINGAIN_SHIFT 2 /* DSP1DRC_MINGAIN - [4:2] */ +#define WM8915_DSP1DRC_MINGAIN_WIDTH 3 /* DSP1DRC_MINGAIN - [4:2] */ +#define WM8915_DSP1DRC_MAXGAIN_MASK 0x0003 /* DSP1DRC_MAXGAIN - [1:0] */ +#define WM8915_DSP1DRC_MAXGAIN_SHIFT 0 /* DSP1DRC_MAXGAIN - [1:0] */ +#define WM8915_DSP1DRC_MAXGAIN_WIDTH 2 /* DSP1DRC_MAXGAIN - [1:0] */ + +/* + * R1090 (0x442) - DSP1 DRC (3) + */ +#define WM8915_DSP1DRC_NG_MINGAIN_MASK 0xF000 /* DSP1DRC_NG_MINGAIN - [15:12] */ +#define WM8915_DSP1DRC_NG_MINGAIN_SHIFT 12 /* DSP1DRC_NG_MINGAIN - [15:12] */ +#define WM8915_DSP1DRC_NG_MINGAIN_WIDTH 4 /* DSP1DRC_NG_MINGAIN - [15:12] */ +#define WM8915_DSP1DRC_NG_EXP_MASK 0x0C00 /* DSP1DRC_NG_EXP - [11:10] */ +#define WM8915_DSP1DRC_NG_EXP_SHIFT 10 /* DSP1DRC_NG_EXP - [11:10] */ +#define WM8915_DSP1DRC_NG_EXP_WIDTH 2 /* DSP1DRC_NG_EXP - [11:10] */ +#define WM8915_DSP1DRC_QR_THR_MASK 0x0300 /* DSP1DRC_QR_THR - [9:8] */ +#define WM8915_DSP1DRC_QR_THR_SHIFT 8 /* DSP1DRC_QR_THR - [9:8] */ +#define WM8915_DSP1DRC_QR_THR_WIDTH 2 /* DSP1DRC_QR_THR - [9:8] */ +#define WM8915_DSP1DRC_QR_DCY_MASK 0x00C0 /* DSP1DRC_QR_DCY - [7:6] */ +#define WM8915_DSP1DRC_QR_DCY_SHIFT 6 /* DSP1DRC_QR_DCY - [7:6] */ +#define WM8915_DSP1DRC_QR_DCY_WIDTH 2 /* DSP1DRC_QR_DCY - [7:6] */ +#define WM8915_DSP1DRC_HI_COMP_MASK 0x0038 /* DSP1DRC_HI_COMP - [5:3] */ +#define WM8915_DSP1DRC_HI_COMP_SHIFT 3 /* DSP1DRC_HI_COMP - [5:3] */ +#define WM8915_DSP1DRC_HI_COMP_WIDTH 3 /* DSP1DRC_HI_COMP - [5:3] */ +#define WM8915_DSP1DRC_LO_COMP_MASK 0x0007 /* DSP1DRC_LO_COMP - [2:0] */ +#define WM8915_DSP1DRC_LO_COMP_SHIFT 0 /* DSP1DRC_LO_COMP - [2:0] */ +#define WM8915_DSP1DRC_LO_COMP_WIDTH 3 /* DSP1DRC_LO_COMP - [2:0] */ + +/* + * R1091 (0x443) - DSP1 DRC (4) + */ +#define WM8915_DSP1DRC_KNEE_IP_MASK 0x07E0 /* DSP1DRC_KNEE_IP - [10:5] */ +#define WM8915_DSP1DRC_KNEE_IP_SHIFT 5 /* DSP1DRC_KNEE_IP - [10:5] */ +#define WM8915_DSP1DRC_KNEE_IP_WIDTH 6 /* DSP1DRC_KNEE_IP - [10:5] */ +#define WM8915_DSP1DRC_KNEE_OP_MASK 0x001F /* DSP1DRC_KNEE_OP - [4:0] */ +#define WM8915_DSP1DRC_KNEE_OP_SHIFT 0 /* DSP1DRC_KNEE_OP - [4:0] */ +#define WM8915_DSP1DRC_KNEE_OP_WIDTH 5 /* DSP1DRC_KNEE_OP - [4:0] */ + +/* + * R1092 (0x444) - DSP1 DRC (5) + */ +#define WM8915_DSP1DRC_KNEE2_IP_MASK 0x03E0 /* DSP1DRC_KNEE2_IP - [9:5] */ +#define WM8915_DSP1DRC_KNEE2_IP_SHIFT 5 /* DSP1DRC_KNEE2_IP - [9:5] */ +#define WM8915_DSP1DRC_KNEE2_IP_WIDTH 5 /* DSP1DRC_KNEE2_IP - [9:5] */ +#define WM8915_DSP1DRC_KNEE2_OP_MASK 0x001F /* DSP1DRC_KNEE2_OP - [4:0] */ +#define WM8915_DSP1DRC_KNEE2_OP_SHIFT 0 /* DSP1DRC_KNEE2_OP - [4:0] */ +#define WM8915_DSP1DRC_KNEE2_OP_WIDTH 5 /* DSP1DRC_KNEE2_OP - [4:0] */ + +/* + * R1152 (0x480) - DSP1 RX EQ Gains (1) + */ +#define WM8915_DSP1RX_EQ_B1_GAIN_MASK 0xF800 /* DSP1RX_EQ_B1_GAIN - [15:11] */ +#define WM8915_DSP1RX_EQ_B1_GAIN_SHIFT 11 /* DSP1RX_EQ_B1_GAIN - [15:11] */ +#define WM8915_DSP1RX_EQ_B1_GAIN_WIDTH 5 /* DSP1RX_EQ_B1_GAIN - [15:11] */ +#define WM8915_DSP1RX_EQ_B2_GAIN_MASK 0x07C0 /* DSP1RX_EQ_B2_GAIN - [10:6] */ +#define WM8915_DSP1RX_EQ_B2_GAIN_SHIFT 6 /* DSP1RX_EQ_B2_GAIN - [10:6] */ +#define WM8915_DSP1RX_EQ_B2_GAIN_WIDTH 5 /* DSP1RX_EQ_B2_GAIN - [10:6] */ +#define WM8915_DSP1RX_EQ_B3_GAIN_MASK 0x003E /* DSP1RX_EQ_B3_GAIN - [5:1] */ +#define WM8915_DSP1RX_EQ_B3_GAIN_SHIFT 1 /* DSP1RX_EQ_B3_GAIN - [5:1] */ +#define WM8915_DSP1RX_EQ_B3_GAIN_WIDTH 5 /* DSP1RX_EQ_B3_GAIN - [5:1] */ +#define WM8915_DSP1RX_EQ_ENA 0x0001 /* DSP1RX_EQ_ENA */ +#define WM8915_DSP1RX_EQ_ENA_MASK 0x0001 /* DSP1RX_EQ_ENA */ +#define WM8915_DSP1RX_EQ_ENA_SHIFT 0 /* DSP1RX_EQ_ENA */ +#define WM8915_DSP1RX_EQ_ENA_WIDTH 1 /* DSP1RX_EQ_ENA */ + +/* + * R1153 (0x481) - DSP1 RX EQ Gains (2) + */ +#define WM8915_DSP1RX_EQ_B4_GAIN_MASK 0xF800 /* DSP1RX_EQ_B4_GAIN - [15:11] */ +#define WM8915_DSP1RX_EQ_B4_GAIN_SHIFT 11 /* DSP1RX_EQ_B4_GAIN - [15:11] */ +#define WM8915_DSP1RX_EQ_B4_GAIN_WIDTH 5 /* DSP1RX_EQ_B4_GAIN - [15:11] */ +#define WM8915_DSP1RX_EQ_B5_GAIN_MASK 0x07C0 /* DSP1RX_EQ_B5_GAIN - [10:6] */ +#define WM8915_DSP1RX_EQ_B5_GAIN_SHIFT 6 /* DSP1RX_EQ_B5_GAIN - [10:6] */ +#define WM8915_DSP1RX_EQ_B5_GAIN_WIDTH 5 /* DSP1RX_EQ_B5_GAIN - [10:6] */ + +/* + * R1154 (0x482) - DSP1 RX EQ Band 1 A + */ +#define WM8915_DSP1RX_EQ_B1_A_MASK 0xFFFF /* DSP1RX_EQ_B1_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B1_A_SHIFT 0 /* DSP1RX_EQ_B1_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B1_A_WIDTH 16 /* DSP1RX_EQ_B1_A - [15:0] */ + +/* + * R1155 (0x483) - DSP1 RX EQ Band 1 B + */ +#define WM8915_DSP1RX_EQ_B1_B_MASK 0xFFFF /* DSP1RX_EQ_B1_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B1_B_SHIFT 0 /* DSP1RX_EQ_B1_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B1_B_WIDTH 16 /* DSP1RX_EQ_B1_B - [15:0] */ + +/* + * R1156 (0x484) - DSP1 RX EQ Band 1 PG + */ +#define WM8915_DSP1RX_EQ_B1_PG_MASK 0xFFFF /* DSP1RX_EQ_B1_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B1_PG_SHIFT 0 /* DSP1RX_EQ_B1_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B1_PG_WIDTH 16 /* DSP1RX_EQ_B1_PG - [15:0] */ + +/* + * R1157 (0x485) - DSP1 RX EQ Band 2 A + */ +#define WM8915_DSP1RX_EQ_B2_A_MASK 0xFFFF /* DSP1RX_EQ_B2_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_A_SHIFT 0 /* DSP1RX_EQ_B2_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_A_WIDTH 16 /* DSP1RX_EQ_B2_A - [15:0] */ + +/* + * R1158 (0x486) - DSP1 RX EQ Band 2 B + */ +#define WM8915_DSP1RX_EQ_B2_B_MASK 0xFFFF /* DSP1RX_EQ_B2_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_B_SHIFT 0 /* DSP1RX_EQ_B2_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_B_WIDTH 16 /* DSP1RX_EQ_B2_B - [15:0] */ + +/* + * R1159 (0x487) - DSP1 RX EQ Band 2 C + */ +#define WM8915_DSP1RX_EQ_B2_C_MASK 0xFFFF /* DSP1RX_EQ_B2_C - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_C_SHIFT 0 /* DSP1RX_EQ_B2_C - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_C_WIDTH 16 /* DSP1RX_EQ_B2_C - [15:0] */ + +/* + * R1160 (0x488) - DSP1 RX EQ Band 2 PG + */ +#define WM8915_DSP1RX_EQ_B2_PG_MASK 0xFFFF /* DSP1RX_EQ_B2_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_PG_SHIFT 0 /* DSP1RX_EQ_B2_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B2_PG_WIDTH 16 /* DSP1RX_EQ_B2_PG - [15:0] */ + +/* + * R1161 (0x489) - DSP1 RX EQ Band 3 A + */ +#define WM8915_DSP1RX_EQ_B3_A_MASK 0xFFFF /* DSP1RX_EQ_B3_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_A_SHIFT 0 /* DSP1RX_EQ_B3_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_A_WIDTH 16 /* DSP1RX_EQ_B3_A - [15:0] */ + +/* + * R1162 (0x48A) - DSP1 RX EQ Band 3 B + */ +#define WM8915_DSP1RX_EQ_B3_B_MASK 0xFFFF /* DSP1RX_EQ_B3_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_B_SHIFT 0 /* DSP1RX_EQ_B3_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_B_WIDTH 16 /* DSP1RX_EQ_B3_B - [15:0] */ + +/* + * R1163 (0x48B) - DSP1 RX EQ Band 3 C + */ +#define WM8915_DSP1RX_EQ_B3_C_MASK 0xFFFF /* DSP1RX_EQ_B3_C - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_C_SHIFT 0 /* DSP1RX_EQ_B3_C - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_C_WIDTH 16 /* DSP1RX_EQ_B3_C - [15:0] */ + +/* + * R1164 (0x48C) - DSP1 RX EQ Band 3 PG + */ +#define WM8915_DSP1RX_EQ_B3_PG_MASK 0xFFFF /* DSP1RX_EQ_B3_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_PG_SHIFT 0 /* DSP1RX_EQ_B3_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B3_PG_WIDTH 16 /* DSP1RX_EQ_B3_PG - [15:0] */ + +/* + * R1165 (0x48D) - DSP1 RX EQ Band 4 A + */ +#define WM8915_DSP1RX_EQ_B4_A_MASK 0xFFFF /* DSP1RX_EQ_B4_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_A_SHIFT 0 /* DSP1RX_EQ_B4_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_A_WIDTH 16 /* DSP1RX_EQ_B4_A - [15:0] */ + +/* + * R1166 (0x48E) - DSP1 RX EQ Band 4 B + */ +#define WM8915_DSP1RX_EQ_B4_B_MASK 0xFFFF /* DSP1RX_EQ_B4_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_B_SHIFT 0 /* DSP1RX_EQ_B4_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_B_WIDTH 16 /* DSP1RX_EQ_B4_B - [15:0] */ + +/* + * R1167 (0x48F) - DSP1 RX EQ Band 4 C + */ +#define WM8915_DSP1RX_EQ_B4_C_MASK 0xFFFF /* DSP1RX_EQ_B4_C - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_C_SHIFT 0 /* DSP1RX_EQ_B4_C - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_C_WIDTH 16 /* DSP1RX_EQ_B4_C - [15:0] */ + +/* + * R1168 (0x490) - DSP1 RX EQ Band 4 PG + */ +#define WM8915_DSP1RX_EQ_B4_PG_MASK 0xFFFF /* DSP1RX_EQ_B4_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_PG_SHIFT 0 /* DSP1RX_EQ_B4_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B4_PG_WIDTH 16 /* DSP1RX_EQ_B4_PG - [15:0] */ + +/* + * R1169 (0x491) - DSP1 RX EQ Band 5 A + */ +#define WM8915_DSP1RX_EQ_B5_A_MASK 0xFFFF /* DSP1RX_EQ_B5_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B5_A_SHIFT 0 /* DSP1RX_EQ_B5_A - [15:0] */ +#define WM8915_DSP1RX_EQ_B5_A_WIDTH 16 /* DSP1RX_EQ_B5_A - [15:0] */ + +/* + * R1170 (0x492) - DSP1 RX EQ Band 5 B + */ +#define WM8915_DSP1RX_EQ_B5_B_MASK 0xFFFF /* DSP1RX_EQ_B5_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B5_B_SHIFT 0 /* DSP1RX_EQ_B5_B - [15:0] */ +#define WM8915_DSP1RX_EQ_B5_B_WIDTH 16 /* DSP1RX_EQ_B5_B - [15:0] */ + +/* + * R1171 (0x493) - DSP1 RX EQ Band 5 PG + */ +#define WM8915_DSP1RX_EQ_B5_PG_MASK 0xFFFF /* DSP1RX_EQ_B5_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B5_PG_SHIFT 0 /* DSP1RX_EQ_B5_PG - [15:0] */ +#define WM8915_DSP1RX_EQ_B5_PG_WIDTH 16 /* DSP1RX_EQ_B5_PG - [15:0] */ + +/* + * R1280 (0x500) - DSP2 TX Left Volume + */ +#define WM8915_DSP2TX_VU 0x0100 /* DSP2TX_VU */ +#define WM8915_DSP2TX_VU_MASK 0x0100 /* DSP2TX_VU */ +#define WM8915_DSP2TX_VU_SHIFT 8 /* DSP2TX_VU */ +#define WM8915_DSP2TX_VU_WIDTH 1 /* DSP2TX_VU */ +#define WM8915_DSP2TXL_VOL_MASK 0x00FF /* DSP2TXL_VOL - [7:0] */ +#define WM8915_DSP2TXL_VOL_SHIFT 0 /* DSP2TXL_VOL - [7:0] */ +#define WM8915_DSP2TXL_VOL_WIDTH 8 /* DSP2TXL_VOL - [7:0] */ + +/* + * R1281 (0x501) - DSP2 TX Right Volume + */ +#define WM8915_DSP2TX_VU 0x0100 /* DSP2TX_VU */ +#define WM8915_DSP2TX_VU_MASK 0x0100 /* DSP2TX_VU */ +#define WM8915_DSP2TX_VU_SHIFT 8 /* DSP2TX_VU */ +#define WM8915_DSP2TX_VU_WIDTH 1 /* DSP2TX_VU */ +#define WM8915_DSP2TXR_VOL_MASK 0x00FF /* DSP2TXR_VOL - [7:0] */ +#define WM8915_DSP2TXR_VOL_SHIFT 0 /* DSP2TXR_VOL - [7:0] */ +#define WM8915_DSP2TXR_VOL_WIDTH 8 /* DSP2TXR_VOL - [7:0] */ + +/* + * R1282 (0x502) - DSP2 RX Left Volume + */ +#define WM8915_DSP2RX_VU 0x0100 /* DSP2RX_VU */ +#define WM8915_DSP2RX_VU_MASK 0x0100 /* DSP2RX_VU */ +#define WM8915_DSP2RX_VU_SHIFT 8 /* DSP2RX_VU */ +#define WM8915_DSP2RX_VU_WIDTH 1 /* DSP2RX_VU */ +#define WM8915_DSP2RXL_VOL_MASK 0x00FF /* DSP2RXL_VOL - [7:0] */ +#define WM8915_DSP2RXL_VOL_SHIFT 0 /* DSP2RXL_VOL - [7:0] */ +#define WM8915_DSP2RXL_VOL_WIDTH 8 /* DSP2RXL_VOL - [7:0] */ + +/* + * R1283 (0x503) - DSP2 RX Right Volume + */ +#define WM8915_DSP2RX_VU 0x0100 /* DSP2RX_VU */ +#define WM8915_DSP2RX_VU_MASK 0x0100 /* DSP2RX_VU */ +#define WM8915_DSP2RX_VU_SHIFT 8 /* DSP2RX_VU */ +#define WM8915_DSP2RX_VU_WIDTH 1 /* DSP2RX_VU */ +#define WM8915_DSP2RXR_VOL_MASK 0x00FF /* DSP2RXR_VOL - [7:0] */ +#define WM8915_DSP2RXR_VOL_SHIFT 0 /* DSP2RXR_VOL - [7:0] */ +#define WM8915_DSP2RXR_VOL_WIDTH 8 /* DSP2RXR_VOL - [7:0] */ + +/* + * R1296 (0x510) - DSP2 TX Filters + */ +#define WM8915_DSP2TX_NF 0x2000 /* DSP2TX_NF */ +#define WM8915_DSP2TX_NF_MASK 0x2000 /* DSP2TX_NF */ +#define WM8915_DSP2TX_NF_SHIFT 13 /* DSP2TX_NF */ +#define WM8915_DSP2TX_NF_WIDTH 1 /* DSP2TX_NF */ +#define WM8915_DSP2TXL_HPF 0x1000 /* DSP2TXL_HPF */ +#define WM8915_DSP2TXL_HPF_MASK 0x1000 /* DSP2TXL_HPF */ +#define WM8915_DSP2TXL_HPF_SHIFT 12 /* DSP2TXL_HPF */ +#define WM8915_DSP2TXL_HPF_WIDTH 1 /* DSP2TXL_HPF */ +#define WM8915_DSP2TXR_HPF 0x0800 /* DSP2TXR_HPF */ +#define WM8915_DSP2TXR_HPF_MASK 0x0800 /* DSP2TXR_HPF */ +#define WM8915_DSP2TXR_HPF_SHIFT 11 /* DSP2TXR_HPF */ +#define WM8915_DSP2TXR_HPF_WIDTH 1 /* DSP2TXR_HPF */ +#define WM8915_DSP2TX_HPF_MODE_MASK 0x0018 /* DSP2TX_HPF_MODE - [4:3] */ +#define WM8915_DSP2TX_HPF_MODE_SHIFT 3 /* DSP2TX_HPF_MODE - [4:3] */ +#define WM8915_DSP2TX_HPF_MODE_WIDTH 2 /* DSP2TX_HPF_MODE - [4:3] */ +#define WM8915_DSP2TX_HPF_CUT_MASK 0x0007 /* DSP2TX_HPF_CUT - [2:0] */ +#define WM8915_DSP2TX_HPF_CUT_SHIFT 0 /* DSP2TX_HPF_CUT - [2:0] */ +#define WM8915_DSP2TX_HPF_CUT_WIDTH 3 /* DSP2TX_HPF_CUT - [2:0] */ + +/* + * R1312 (0x520) - DSP2 RX Filters (1) + */ +#define WM8915_DSP2RX_MUTE 0x0200 /* DSP2RX_MUTE */ +#define WM8915_DSP2RX_MUTE_MASK 0x0200 /* DSP2RX_MUTE */ +#define WM8915_DSP2RX_MUTE_SHIFT 9 /* DSP2RX_MUTE */ +#define WM8915_DSP2RX_MUTE_WIDTH 1 /* DSP2RX_MUTE */ +#define WM8915_DSP2RX_MONO 0x0080 /* DSP2RX_MONO */ +#define WM8915_DSP2RX_MONO_MASK 0x0080 /* DSP2RX_MONO */ +#define WM8915_DSP2RX_MONO_SHIFT 7 /* DSP2RX_MONO */ +#define WM8915_DSP2RX_MONO_WIDTH 1 /* DSP2RX_MONO */ +#define WM8915_DSP2RX_MUTERATE 0x0020 /* DSP2RX_MUTERATE */ +#define WM8915_DSP2RX_MUTERATE_MASK 0x0020 /* DSP2RX_MUTERATE */ +#define WM8915_DSP2RX_MUTERATE_SHIFT 5 /* DSP2RX_MUTERATE */ +#define WM8915_DSP2RX_MUTERATE_WIDTH 1 /* DSP2RX_MUTERATE */ +#define WM8915_DSP2RX_UNMUTE_RAMP 0x0010 /* DSP2RX_UNMUTE_RAMP */ +#define WM8915_DSP2RX_UNMUTE_RAMP_MASK 0x0010 /* DSP2RX_UNMUTE_RAMP */ +#define WM8915_DSP2RX_UNMUTE_RAMP_SHIFT 4 /* DSP2RX_UNMUTE_RAMP */ +#define WM8915_DSP2RX_UNMUTE_RAMP_WIDTH 1 /* DSP2RX_UNMUTE_RAMP */ + +/* + * R1313 (0x521) - DSP2 RX Filters (2) + */ +#define WM8915_DSP2RX_3D_GAIN_MASK 0x3E00 /* DSP2RX_3D_GAIN - [13:9] */ +#define WM8915_DSP2RX_3D_GAIN_SHIFT 9 /* DSP2RX_3D_GAIN - [13:9] */ +#define WM8915_DSP2RX_3D_GAIN_WIDTH 5 /* DSP2RX_3D_GAIN - [13:9] */ +#define WM8915_DSP2RX_3D_ENA 0x0100 /* DSP2RX_3D_ENA */ +#define WM8915_DSP2RX_3D_ENA_MASK 0x0100 /* DSP2RX_3D_ENA */ +#define WM8915_DSP2RX_3D_ENA_SHIFT 8 /* DSP2RX_3D_ENA */ +#define WM8915_DSP2RX_3D_ENA_WIDTH 1 /* DSP2RX_3D_ENA */ + +/* + * R1344 (0x540) - DSP2 DRC (1) + */ +#define WM8915_DSP2DRC_SIG_DET_RMS_MASK 0xF800 /* DSP2DRC_SIG_DET_RMS - [15:11] */ +#define WM8915_DSP2DRC_SIG_DET_RMS_SHIFT 11 /* DSP2DRC_SIG_DET_RMS - [15:11] */ +#define WM8915_DSP2DRC_SIG_DET_RMS_WIDTH 5 /* DSP2DRC_SIG_DET_RMS - [15:11] */ +#define WM8915_DSP2DRC_SIG_DET_PK_MASK 0x0600 /* DSP2DRC_SIG_DET_PK - [10:9] */ +#define WM8915_DSP2DRC_SIG_DET_PK_SHIFT 9 /* DSP2DRC_SIG_DET_PK - [10:9] */ +#define WM8915_DSP2DRC_SIG_DET_PK_WIDTH 2 /* DSP2DRC_SIG_DET_PK - [10:9] */ +#define WM8915_DSP2DRC_NG_ENA 0x0100 /* DSP2DRC_NG_ENA */ +#define WM8915_DSP2DRC_NG_ENA_MASK 0x0100 /* DSP2DRC_NG_ENA */ +#define WM8915_DSP2DRC_NG_ENA_SHIFT 8 /* DSP2DRC_NG_ENA */ +#define WM8915_DSP2DRC_NG_ENA_WIDTH 1 /* DSP2DRC_NG_ENA */ +#define WM8915_DSP2DRC_SIG_DET_MODE 0x0080 /* DSP2DRC_SIG_DET_MODE */ +#define WM8915_DSP2DRC_SIG_DET_MODE_MASK 0x0080 /* DSP2DRC_SIG_DET_MODE */ +#define WM8915_DSP2DRC_SIG_DET_MODE_SHIFT 7 /* DSP2DRC_SIG_DET_MODE */ +#define WM8915_DSP2DRC_SIG_DET_MODE_WIDTH 1 /* DSP2DRC_SIG_DET_MODE */ +#define WM8915_DSP2DRC_SIG_DET 0x0040 /* DSP2DRC_SIG_DET */ +#define WM8915_DSP2DRC_SIG_DET_MASK 0x0040 /* DSP2DRC_SIG_DET */ +#define WM8915_DSP2DRC_SIG_DET_SHIFT 6 /* DSP2DRC_SIG_DET */ +#define WM8915_DSP2DRC_SIG_DET_WIDTH 1 /* DSP2DRC_SIG_DET */ +#define WM8915_DSP2DRC_KNEE2_OP_ENA 0x0020 /* DSP2DRC_KNEE2_OP_ENA */ +#define WM8915_DSP2DRC_KNEE2_OP_ENA_MASK 0x0020 /* DSP2DRC_KNEE2_OP_ENA */ +#define WM8915_DSP2DRC_KNEE2_OP_ENA_SHIFT 5 /* DSP2DRC_KNEE2_OP_ENA */ +#define WM8915_DSP2DRC_KNEE2_OP_ENA_WIDTH 1 /* DSP2DRC_KNEE2_OP_ENA */ +#define WM8915_DSP2DRC_QR 0x0010 /* DSP2DRC_QR */ +#define WM8915_DSP2DRC_QR_MASK 0x0010 /* DSP2DRC_QR */ +#define WM8915_DSP2DRC_QR_SHIFT 4 /* DSP2DRC_QR */ +#define WM8915_DSP2DRC_QR_WIDTH 1 /* DSP2DRC_QR */ +#define WM8915_DSP2DRC_ANTICLIP 0x0008 /* DSP2DRC_ANTICLIP */ +#define WM8915_DSP2DRC_ANTICLIP_MASK 0x0008 /* DSP2DRC_ANTICLIP */ +#define WM8915_DSP2DRC_ANTICLIP_SHIFT 3 /* DSP2DRC_ANTICLIP */ +#define WM8915_DSP2DRC_ANTICLIP_WIDTH 1 /* DSP2DRC_ANTICLIP */ +#define WM8915_DSP2RX_DRC_ENA 0x0004 /* DSP2RX_DRC_ENA */ +#define WM8915_DSP2RX_DRC_ENA_MASK 0x0004 /* DSP2RX_DRC_ENA */ +#define WM8915_DSP2RX_DRC_ENA_SHIFT 2 /* DSP2RX_DRC_ENA */ +#define WM8915_DSP2RX_DRC_ENA_WIDTH 1 /* DSP2RX_DRC_ENA */ +#define WM8915_DSP2TXL_DRC_ENA 0x0002 /* DSP2TXL_DRC_ENA */ +#define WM8915_DSP2TXL_DRC_ENA_MASK 0x0002 /* DSP2TXL_DRC_ENA */ +#define WM8915_DSP2TXL_DRC_ENA_SHIFT 1 /* DSP2TXL_DRC_ENA */ +#define WM8915_DSP2TXL_DRC_ENA_WIDTH 1 /* DSP2TXL_DRC_ENA */ +#define WM8915_DSP2TXR_DRC_ENA 0x0001 /* DSP2TXR_DRC_ENA */ +#define WM8915_DSP2TXR_DRC_ENA_MASK 0x0001 /* DSP2TXR_DRC_ENA */ +#define WM8915_DSP2TXR_DRC_ENA_SHIFT 0 /* DSP2TXR_DRC_ENA */ +#define WM8915_DSP2TXR_DRC_ENA_WIDTH 1 /* DSP2TXR_DRC_ENA */ + +/* + * R1345 (0x541) - DSP2 DRC (2) + */ +#define WM8915_DSP2DRC_ATK_MASK 0x1E00 /* DSP2DRC_ATK - [12:9] */ +#define WM8915_DSP2DRC_ATK_SHIFT 9 /* DSP2DRC_ATK - [12:9] */ +#define WM8915_DSP2DRC_ATK_WIDTH 4 /* DSP2DRC_ATK - [12:9] */ +#define WM8915_DSP2DRC_DCY_MASK 0x01E0 /* DSP2DRC_DCY - [8:5] */ +#define WM8915_DSP2DRC_DCY_SHIFT 5 /* DSP2DRC_DCY - [8:5] */ +#define WM8915_DSP2DRC_DCY_WIDTH 4 /* DSP2DRC_DCY - [8:5] */ +#define WM8915_DSP2DRC_MINGAIN_MASK 0x001C /* DSP2DRC_MINGAIN - [4:2] */ +#define WM8915_DSP2DRC_MINGAIN_SHIFT 2 /* DSP2DRC_MINGAIN - [4:2] */ +#define WM8915_DSP2DRC_MINGAIN_WIDTH 3 /* DSP2DRC_MINGAIN - [4:2] */ +#define WM8915_DSP2DRC_MAXGAIN_MASK 0x0003 /* DSP2DRC_MAXGAIN - [1:0] */ +#define WM8915_DSP2DRC_MAXGAIN_SHIFT 0 /* DSP2DRC_MAXGAIN - [1:0] */ +#define WM8915_DSP2DRC_MAXGAIN_WIDTH 2 /* DSP2DRC_MAXGAIN - [1:0] */ + +/* + * R1346 (0x542) - DSP2 DRC (3) + */ +#define WM8915_DSP2DRC_NG_MINGAIN_MASK 0xF000 /* DSP2DRC_NG_MINGAIN - [15:12] */ +#define WM8915_DSP2DRC_NG_MINGAIN_SHIFT 12 /* DSP2DRC_NG_MINGAIN - [15:12] */ +#define WM8915_DSP2DRC_NG_MINGAIN_WIDTH 4 /* DSP2DRC_NG_MINGAIN - [15:12] */ +#define WM8915_DSP2DRC_NG_EXP_MASK 0x0C00 /* DSP2DRC_NG_EXP - [11:10] */ +#define WM8915_DSP2DRC_NG_EXP_SHIFT 10 /* DSP2DRC_NG_EXP - [11:10] */ +#define WM8915_DSP2DRC_NG_EXP_WIDTH 2 /* DSP2DRC_NG_EXP - [11:10] */ +#define WM8915_DSP2DRC_QR_THR_MASK 0x0300 /* DSP2DRC_QR_THR - [9:8] */ +#define WM8915_DSP2DRC_QR_THR_SHIFT 8 /* DSP2DRC_QR_THR - [9:8] */ +#define WM8915_DSP2DRC_QR_THR_WIDTH 2 /* DSP2DRC_QR_THR - [9:8] */ +#define WM8915_DSP2DRC_QR_DCY_MASK 0x00C0 /* DSP2DRC_QR_DCY - [7:6] */ +#define WM8915_DSP2DRC_QR_DCY_SHIFT 6 /* DSP2DRC_QR_DCY - [7:6] */ +#define WM8915_DSP2DRC_QR_DCY_WIDTH 2 /* DSP2DRC_QR_DCY - [7:6] */ +#define WM8915_DSP2DRC_HI_COMP_MASK 0x0038 /* DSP2DRC_HI_COMP - [5:3] */ +#define WM8915_DSP2DRC_HI_COMP_SHIFT 3 /* DSP2DRC_HI_COMP - [5:3] */ +#define WM8915_DSP2DRC_HI_COMP_WIDTH 3 /* DSP2DRC_HI_COMP - [5:3] */ +#define WM8915_DSP2DRC_LO_COMP_MASK 0x0007 /* DSP2DRC_LO_COMP - [2:0] */ +#define WM8915_DSP2DRC_LO_COMP_SHIFT 0 /* DSP2DRC_LO_COMP - [2:0] */ +#define WM8915_DSP2DRC_LO_COMP_WIDTH 3 /* DSP2DRC_LO_COMP - [2:0] */ + +/* + * R1347 (0x543) - DSP2 DRC (4) + */ +#define WM8915_DSP2DRC_KNEE_IP_MASK 0x07E0 /* DSP2DRC_KNEE_IP - [10:5] */ +#define WM8915_DSP2DRC_KNEE_IP_SHIFT 5 /* DSP2DRC_KNEE_IP - [10:5] */ +#define WM8915_DSP2DRC_KNEE_IP_WIDTH 6 /* DSP2DRC_KNEE_IP - [10:5] */ +#define WM8915_DSP2DRC_KNEE_OP_MASK 0x001F /* DSP2DRC_KNEE_OP - [4:0] */ +#define WM8915_DSP2DRC_KNEE_OP_SHIFT 0 /* DSP2DRC_KNEE_OP - [4:0] */ +#define WM8915_DSP2DRC_KNEE_OP_WIDTH 5 /* DSP2DRC_KNEE_OP - [4:0] */ + +/* + * R1348 (0x544) - DSP2 DRC (5) + */ +#define WM8915_DSP2DRC_KNEE2_IP_MASK 0x03E0 /* DSP2DRC_KNEE2_IP - [9:5] */ +#define WM8915_DSP2DRC_KNEE2_IP_SHIFT 5 /* DSP2DRC_KNEE2_IP - [9:5] */ +#define WM8915_DSP2DRC_KNEE2_IP_WIDTH 5 /* DSP2DRC_KNEE2_IP - [9:5] */ +#define WM8915_DSP2DRC_KNEE2_OP_MASK 0x001F /* DSP2DRC_KNEE2_OP - [4:0] */ +#define WM8915_DSP2DRC_KNEE2_OP_SHIFT 0 /* DSP2DRC_KNEE2_OP - [4:0] */ +#define WM8915_DSP2DRC_KNEE2_OP_WIDTH 5 /* DSP2DRC_KNEE2_OP - [4:0] */ + +/* + * R1408 (0x580) - DSP2 RX EQ Gains (1) + */ +#define WM8915_DSP2RX_EQ_B1_GAIN_MASK 0xF800 /* DSP2RX_EQ_B1_GAIN - [15:11] */ +#define WM8915_DSP2RX_EQ_B1_GAIN_SHIFT 11 /* DSP2RX_EQ_B1_GAIN - [15:11] */ +#define WM8915_DSP2RX_EQ_B1_GAIN_WIDTH 5 /* DSP2RX_EQ_B1_GAIN - [15:11] */ +#define WM8915_DSP2RX_EQ_B2_GAIN_MASK 0x07C0 /* DSP2RX_EQ_B2_GAIN - [10:6] */ +#define WM8915_DSP2RX_EQ_B2_GAIN_SHIFT 6 /* DSP2RX_EQ_B2_GAIN - [10:6] */ +#define WM8915_DSP2RX_EQ_B2_GAIN_WIDTH 5 /* DSP2RX_EQ_B2_GAIN - [10:6] */ +#define WM8915_DSP2RX_EQ_B3_GAIN_MASK 0x003E /* DSP2RX_EQ_B3_GAIN - [5:1] */ +#define WM8915_DSP2RX_EQ_B3_GAIN_SHIFT 1 /* DSP2RX_EQ_B3_GAIN - [5:1] */ +#define WM8915_DSP2RX_EQ_B3_GAIN_WIDTH 5 /* DSP2RX_EQ_B3_GAIN - [5:1] */ +#define WM8915_DSP2RX_EQ_ENA 0x0001 /* DSP2RX_EQ_ENA */ +#define WM8915_DSP2RX_EQ_ENA_MASK 0x0001 /* DSP2RX_EQ_ENA */ +#define WM8915_DSP2RX_EQ_ENA_SHIFT 0 /* DSP2RX_EQ_ENA */ +#define WM8915_DSP2RX_EQ_ENA_WIDTH 1 /* DSP2RX_EQ_ENA */ + +/* + * R1409 (0x581) - DSP2 RX EQ Gains (2) + */ +#define WM8915_DSP2RX_EQ_B4_GAIN_MASK 0xF800 /* DSP2RX_EQ_B4_GAIN - [15:11] */ +#define WM8915_DSP2RX_EQ_B4_GAIN_SHIFT 11 /* DSP2RX_EQ_B4_GAIN - [15:11] */ +#define WM8915_DSP2RX_EQ_B4_GAIN_WIDTH 5 /* DSP2RX_EQ_B4_GAIN - [15:11] */ +#define WM8915_DSP2RX_EQ_B5_GAIN_MASK 0x07C0 /* DSP2RX_EQ_B5_GAIN - [10:6] */ +#define WM8915_DSP2RX_EQ_B5_GAIN_SHIFT 6 /* DSP2RX_EQ_B5_GAIN - [10:6] */ +#define WM8915_DSP2RX_EQ_B5_GAIN_WIDTH 5 /* DSP2RX_EQ_B5_GAIN - [10:6] */ + +/* + * R1410 (0x582) - DSP2 RX EQ Band 1 A + */ +#define WM8915_DSP2RX_EQ_B1_A_MASK 0xFFFF /* DSP2RX_EQ_B1_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B1_A_SHIFT 0 /* DSP2RX_EQ_B1_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B1_A_WIDTH 16 /* DSP2RX_EQ_B1_A - [15:0] */ + +/* + * R1411 (0x583) - DSP2 RX EQ Band 1 B + */ +#define WM8915_DSP2RX_EQ_B1_B_MASK 0xFFFF /* DSP2RX_EQ_B1_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B1_B_SHIFT 0 /* DSP2RX_EQ_B1_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B1_B_WIDTH 16 /* DSP2RX_EQ_B1_B - [15:0] */ + +/* + * R1412 (0x584) - DSP2 RX EQ Band 1 PG + */ +#define WM8915_DSP2RX_EQ_B1_PG_MASK 0xFFFF /* DSP2RX_EQ_B1_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B1_PG_SHIFT 0 /* DSP2RX_EQ_B1_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B1_PG_WIDTH 16 /* DSP2RX_EQ_B1_PG - [15:0] */ + +/* + * R1413 (0x585) - DSP2 RX EQ Band 2 A + */ +#define WM8915_DSP2RX_EQ_B2_A_MASK 0xFFFF /* DSP2RX_EQ_B2_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_A_SHIFT 0 /* DSP2RX_EQ_B2_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_A_WIDTH 16 /* DSP2RX_EQ_B2_A - [15:0] */ + +/* + * R1414 (0x586) - DSP2 RX EQ Band 2 B + */ +#define WM8915_DSP2RX_EQ_B2_B_MASK 0xFFFF /* DSP2RX_EQ_B2_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_B_SHIFT 0 /* DSP2RX_EQ_B2_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_B_WIDTH 16 /* DSP2RX_EQ_B2_B - [15:0] */ + +/* + * R1415 (0x587) - DSP2 RX EQ Band 2 C + */ +#define WM8915_DSP2RX_EQ_B2_C_MASK 0xFFFF /* DSP2RX_EQ_B2_C - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_C_SHIFT 0 /* DSP2RX_EQ_B2_C - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_C_WIDTH 16 /* DSP2RX_EQ_B2_C - [15:0] */ + +/* + * R1416 (0x588) - DSP2 RX EQ Band 2 PG + */ +#define WM8915_DSP2RX_EQ_B2_PG_MASK 0xFFFF /* DSP2RX_EQ_B2_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_PG_SHIFT 0 /* DSP2RX_EQ_B2_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B2_PG_WIDTH 16 /* DSP2RX_EQ_B2_PG - [15:0] */ + +/* + * R1417 (0x589) - DSP2 RX EQ Band 3 A + */ +#define WM8915_DSP2RX_EQ_B3_A_MASK 0xFFFF /* DSP2RX_EQ_B3_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_A_SHIFT 0 /* DSP2RX_EQ_B3_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_A_WIDTH 16 /* DSP2RX_EQ_B3_A - [15:0] */ + +/* + * R1418 (0x58A) - DSP2 RX EQ Band 3 B + */ +#define WM8915_DSP2RX_EQ_B3_B_MASK 0xFFFF /* DSP2RX_EQ_B3_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_B_SHIFT 0 /* DSP2RX_EQ_B3_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_B_WIDTH 16 /* DSP2RX_EQ_B3_B - [15:0] */ + +/* + * R1419 (0x58B) - DSP2 RX EQ Band 3 C + */ +#define WM8915_DSP2RX_EQ_B3_C_MASK 0xFFFF /* DSP2RX_EQ_B3_C - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_C_SHIFT 0 /* DSP2RX_EQ_B3_C - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_C_WIDTH 16 /* DSP2RX_EQ_B3_C - [15:0] */ + +/* + * R1420 (0x58C) - DSP2 RX EQ Band 3 PG + */ +#define WM8915_DSP2RX_EQ_B3_PG_MASK 0xFFFF /* DSP2RX_EQ_B3_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_PG_SHIFT 0 /* DSP2RX_EQ_B3_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B3_PG_WIDTH 16 /* DSP2RX_EQ_B3_PG - [15:0] */ + +/* + * R1421 (0x58D) - DSP2 RX EQ Band 4 A + */ +#define WM8915_DSP2RX_EQ_B4_A_MASK 0xFFFF /* DSP2RX_EQ_B4_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_A_SHIFT 0 /* DSP2RX_EQ_B4_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_A_WIDTH 16 /* DSP2RX_EQ_B4_A - [15:0] */ + +/* + * R1422 (0x58E) - DSP2 RX EQ Band 4 B + */ +#define WM8915_DSP2RX_EQ_B4_B_MASK 0xFFFF /* DSP2RX_EQ_B4_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_B_SHIFT 0 /* DSP2RX_EQ_B4_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_B_WIDTH 16 /* DSP2RX_EQ_B4_B - [15:0] */ + +/* + * R1423 (0x58F) - DSP2 RX EQ Band 4 C + */ +#define WM8915_DSP2RX_EQ_B4_C_MASK 0xFFFF /* DSP2RX_EQ_B4_C - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_C_SHIFT 0 /* DSP2RX_EQ_B4_C - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_C_WIDTH 16 /* DSP2RX_EQ_B4_C - [15:0] */ + +/* + * R1424 (0x590) - DSP2 RX EQ Band 4 PG + */ +#define WM8915_DSP2RX_EQ_B4_PG_MASK 0xFFFF /* DSP2RX_EQ_B4_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_PG_SHIFT 0 /* DSP2RX_EQ_B4_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B4_PG_WIDTH 16 /* DSP2RX_EQ_B4_PG - [15:0] */ + +/* + * R1425 (0x591) - DSP2 RX EQ Band 5 A + */ +#define WM8915_DSP2RX_EQ_B5_A_MASK 0xFFFF /* DSP2RX_EQ_B5_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B5_A_SHIFT 0 /* DSP2RX_EQ_B5_A - [15:0] */ +#define WM8915_DSP2RX_EQ_B5_A_WIDTH 16 /* DSP2RX_EQ_B5_A - [15:0] */ + +/* + * R1426 (0x592) - DSP2 RX EQ Band 5 B + */ +#define WM8915_DSP2RX_EQ_B5_B_MASK 0xFFFF /* DSP2RX_EQ_B5_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B5_B_SHIFT 0 /* DSP2RX_EQ_B5_B - [15:0] */ +#define WM8915_DSP2RX_EQ_B5_B_WIDTH 16 /* DSP2RX_EQ_B5_B - [15:0] */ + +/* + * R1427 (0x593) - DSP2 RX EQ Band 5 PG + */ +#define WM8915_DSP2RX_EQ_B5_PG_MASK 0xFFFF /* DSP2RX_EQ_B5_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B5_PG_SHIFT 0 /* DSP2RX_EQ_B5_PG - [15:0] */ +#define WM8915_DSP2RX_EQ_B5_PG_WIDTH 16 /* DSP2RX_EQ_B5_PG - [15:0] */ + +/* + * R1536 (0x600) - DAC1 Mixer Volumes + */ +#define WM8915_ADCR_DAC1_VOL_MASK 0x03E0 /* ADCR_DAC1_VOL - [9:5] */ +#define WM8915_ADCR_DAC1_VOL_SHIFT 5 /* ADCR_DAC1_VOL - [9:5] */ +#define WM8915_ADCR_DAC1_VOL_WIDTH 5 /* ADCR_DAC1_VOL - [9:5] */ +#define WM8915_ADCL_DAC1_VOL_MASK 0x001F /* ADCL_DAC1_VOL - [4:0] */ +#define WM8915_ADCL_DAC1_VOL_SHIFT 0 /* ADCL_DAC1_VOL - [4:0] */ +#define WM8915_ADCL_DAC1_VOL_WIDTH 5 /* ADCL_DAC1_VOL - [4:0] */ + +/* + * R1537 (0x601) - DAC1 Left Mixer Routing + */ +#define WM8915_ADCR_TO_DAC1L 0x0020 /* ADCR_TO_DAC1L */ +#define WM8915_ADCR_TO_DAC1L_MASK 0x0020 /* ADCR_TO_DAC1L */ +#define WM8915_ADCR_TO_DAC1L_SHIFT 5 /* ADCR_TO_DAC1L */ +#define WM8915_ADCR_TO_DAC1L_WIDTH 1 /* ADCR_TO_DAC1L */ +#define WM8915_ADCL_TO_DAC1L 0x0010 /* ADCL_TO_DAC1L */ +#define WM8915_ADCL_TO_DAC1L_MASK 0x0010 /* ADCL_TO_DAC1L */ +#define WM8915_ADCL_TO_DAC1L_SHIFT 4 /* ADCL_TO_DAC1L */ +#define WM8915_ADCL_TO_DAC1L_WIDTH 1 /* ADCL_TO_DAC1L */ +#define WM8915_DSP2RXL_TO_DAC1L 0x0002 /* DSP2RXL_TO_DAC1L */ +#define WM8915_DSP2RXL_TO_DAC1L_MASK 0x0002 /* DSP2RXL_TO_DAC1L */ +#define WM8915_DSP2RXL_TO_DAC1L_SHIFT 1 /* DSP2RXL_TO_DAC1L */ +#define WM8915_DSP2RXL_TO_DAC1L_WIDTH 1 /* DSP2RXL_TO_DAC1L */ +#define WM8915_DSP1RXL_TO_DAC1L 0x0001 /* DSP1RXL_TO_DAC1L */ +#define WM8915_DSP1RXL_TO_DAC1L_MASK 0x0001 /* DSP1RXL_TO_DAC1L */ +#define WM8915_DSP1RXL_TO_DAC1L_SHIFT 0 /* DSP1RXL_TO_DAC1L */ +#define WM8915_DSP1RXL_TO_DAC1L_WIDTH 1 /* DSP1RXL_TO_DAC1L */ + +/* + * R1538 (0x602) - DAC1 Right Mixer Routing + */ +#define WM8915_ADCR_TO_DAC1R 0x0020 /* ADCR_TO_DAC1R */ +#define WM8915_ADCR_TO_DAC1R_MASK 0x0020 /* ADCR_TO_DAC1R */ +#define WM8915_ADCR_TO_DAC1R_SHIFT 5 /* ADCR_TO_DAC1R */ +#define WM8915_ADCR_TO_DAC1R_WIDTH 1 /* ADCR_TO_DAC1R */ +#define WM8915_ADCL_TO_DAC1R 0x0010 /* ADCL_TO_DAC1R */ +#define WM8915_ADCL_TO_DAC1R_MASK 0x0010 /* ADCL_TO_DAC1R */ +#define WM8915_ADCL_TO_DAC1R_SHIFT 4 /* ADCL_TO_DAC1R */ +#define WM8915_ADCL_TO_DAC1R_WIDTH 1 /* ADCL_TO_DAC1R */ +#define WM8915_DSP2RXR_TO_DAC1R 0x0002 /* DSP2RXR_TO_DAC1R */ +#define WM8915_DSP2RXR_TO_DAC1R_MASK 0x0002 /* DSP2RXR_TO_DAC1R */ +#define WM8915_DSP2RXR_TO_DAC1R_SHIFT 1 /* DSP2RXR_TO_DAC1R */ +#define WM8915_DSP2RXR_TO_DAC1R_WIDTH 1 /* DSP2RXR_TO_DAC1R */ +#define WM8915_DSP1RXR_TO_DAC1R 0x0001 /* DSP1RXR_TO_DAC1R */ +#define WM8915_DSP1RXR_TO_DAC1R_MASK 0x0001 /* DSP1RXR_TO_DAC1R */ +#define WM8915_DSP1RXR_TO_DAC1R_SHIFT 0 /* DSP1RXR_TO_DAC1R */ +#define WM8915_DSP1RXR_TO_DAC1R_WIDTH 1 /* DSP1RXR_TO_DAC1R */ + +/* + * R1539 (0x603) - DAC2 Mixer Volumes + */ +#define WM8915_ADCR_DAC2_VOL_MASK 0x03E0 /* ADCR_DAC2_VOL - [9:5] */ +#define WM8915_ADCR_DAC2_VOL_SHIFT 5 /* ADCR_DAC2_VOL - [9:5] */ +#define WM8915_ADCR_DAC2_VOL_WIDTH 5 /* ADCR_DAC2_VOL - [9:5] */ +#define WM8915_ADCL_DAC2_VOL_MASK 0x001F /* ADCL_DAC2_VOL - [4:0] */ +#define WM8915_ADCL_DAC2_VOL_SHIFT 0 /* ADCL_DAC2_VOL - [4:0] */ +#define WM8915_ADCL_DAC2_VOL_WIDTH 5 /* ADCL_DAC2_VOL - [4:0] */ + +/* + * R1540 (0x604) - DAC2 Left Mixer Routing + */ +#define WM8915_ADCR_TO_DAC2L 0x0020 /* ADCR_TO_DAC2L */ +#define WM8915_ADCR_TO_DAC2L_MASK 0x0020 /* ADCR_TO_DAC2L */ +#define WM8915_ADCR_TO_DAC2L_SHIFT 5 /* ADCR_TO_DAC2L */ +#define WM8915_ADCR_TO_DAC2L_WIDTH 1 /* ADCR_TO_DAC2L */ +#define WM8915_ADCL_TO_DAC2L 0x0010 /* ADCL_TO_DAC2L */ +#define WM8915_ADCL_TO_DAC2L_MASK 0x0010 /* ADCL_TO_DAC2L */ +#define WM8915_ADCL_TO_DAC2L_SHIFT 4 /* ADCL_TO_DAC2L */ +#define WM8915_ADCL_TO_DAC2L_WIDTH 1 /* ADCL_TO_DAC2L */ +#define WM8915_DSP2RXL_TO_DAC2L 0x0002 /* DSP2RXL_TO_DAC2L */ +#define WM8915_DSP2RXL_TO_DAC2L_MASK 0x0002 /* DSP2RXL_TO_DAC2L */ +#define WM8915_DSP2RXL_TO_DAC2L_SHIFT 1 /* DSP2RXL_TO_DAC2L */ +#define WM8915_DSP2RXL_TO_DAC2L_WIDTH 1 /* DSP2RXL_TO_DAC2L */ +#define WM8915_DSP1RXL_TO_DAC2L 0x0001 /* DSP1RXL_TO_DAC2L */ +#define WM8915_DSP1RXL_TO_DAC2L_MASK 0x0001 /* DSP1RXL_TO_DAC2L */ +#define WM8915_DSP1RXL_TO_DAC2L_SHIFT 0 /* DSP1RXL_TO_DAC2L */ +#define WM8915_DSP1RXL_TO_DAC2L_WIDTH 1 /* DSP1RXL_TO_DAC2L */ + +/* + * R1541 (0x605) - DAC2 Right Mixer Routing + */ +#define WM8915_ADCR_TO_DAC2R 0x0020 /* ADCR_TO_DAC2R */ +#define WM8915_ADCR_TO_DAC2R_MASK 0x0020 /* ADCR_TO_DAC2R */ +#define WM8915_ADCR_TO_DAC2R_SHIFT 5 /* ADCR_TO_DAC2R */ +#define WM8915_ADCR_TO_DAC2R_WIDTH 1 /* ADCR_TO_DAC2R */ +#define WM8915_ADCL_TO_DAC2R 0x0010 /* ADCL_TO_DAC2R */ +#define WM8915_ADCL_TO_DAC2R_MASK 0x0010 /* ADCL_TO_DAC2R */ +#define WM8915_ADCL_TO_DAC2R_SHIFT 4 /* ADCL_TO_DAC2R */ +#define WM8915_ADCL_TO_DAC2R_WIDTH 1 /* ADCL_TO_DAC2R */ +#define WM8915_DSP2RXR_TO_DAC2R 0x0002 /* DSP2RXR_TO_DAC2R */ +#define WM8915_DSP2RXR_TO_DAC2R_MASK 0x0002 /* DSP2RXR_TO_DAC2R */ +#define WM8915_DSP2RXR_TO_DAC2R_SHIFT 1 /* DSP2RXR_TO_DAC2R */ +#define WM8915_DSP2RXR_TO_DAC2R_WIDTH 1 /* DSP2RXR_TO_DAC2R */ +#define WM8915_DSP1RXR_TO_DAC2R 0x0001 /* DSP1RXR_TO_DAC2R */ +#define WM8915_DSP1RXR_TO_DAC2R_MASK 0x0001 /* DSP1RXR_TO_DAC2R */ +#define WM8915_DSP1RXR_TO_DAC2R_SHIFT 0 /* DSP1RXR_TO_DAC2R */ +#define WM8915_DSP1RXR_TO_DAC2R_WIDTH 1 /* DSP1RXR_TO_DAC2R */ + +/* + * R1542 (0x606) - DSP1 TX Left Mixer Routing + */ +#define WM8915_ADC1L_TO_DSP1TXL 0x0002 /* ADC1L_TO_DSP1TXL */ +#define WM8915_ADC1L_TO_DSP1TXL_MASK 0x0002 /* ADC1L_TO_DSP1TXL */ +#define WM8915_ADC1L_TO_DSP1TXL_SHIFT 1 /* ADC1L_TO_DSP1TXL */ +#define WM8915_ADC1L_TO_DSP1TXL_WIDTH 1 /* ADC1L_TO_DSP1TXL */ +#define WM8915_DACL_TO_DSP1TXL 0x0001 /* DACL_TO_DSP1TXL */ +#define WM8915_DACL_TO_DSP1TXL_MASK 0x0001 /* DACL_TO_DSP1TXL */ +#define WM8915_DACL_TO_DSP1TXL_SHIFT 0 /* DACL_TO_DSP1TXL */ +#define WM8915_DACL_TO_DSP1TXL_WIDTH 1 /* DACL_TO_DSP1TXL */ + +/* + * R1543 (0x607) - DSP1 TX Right Mixer Routing + */ +#define WM8915_ADC1R_TO_DSP1TXR 0x0002 /* ADC1R_TO_DSP1TXR */ +#define WM8915_ADC1R_TO_DSP1TXR_MASK 0x0002 /* ADC1R_TO_DSP1TXR */ +#define WM8915_ADC1R_TO_DSP1TXR_SHIFT 1 /* ADC1R_TO_DSP1TXR */ +#define WM8915_ADC1R_TO_DSP1TXR_WIDTH 1 /* ADC1R_TO_DSP1TXR */ +#define WM8915_DACR_TO_DSP1TXR 0x0001 /* DACR_TO_DSP1TXR */ +#define WM8915_DACR_TO_DSP1TXR_MASK 0x0001 /* DACR_TO_DSP1TXR */ +#define WM8915_DACR_TO_DSP1TXR_SHIFT 0 /* DACR_TO_DSP1TXR */ +#define WM8915_DACR_TO_DSP1TXR_WIDTH 1 /* DACR_TO_DSP1TXR */ + +/* + * R1544 (0x608) - DSP2 TX Left Mixer Routing + */ +#define WM8915_ADC2L_TO_DSP2TXL 0x0002 /* ADC2L_TO_DSP2TXL */ +#define WM8915_ADC2L_TO_DSP2TXL_MASK 0x0002 /* ADC2L_TO_DSP2TXL */ +#define WM8915_ADC2L_TO_DSP2TXL_SHIFT 1 /* ADC2L_TO_DSP2TXL */ +#define WM8915_ADC2L_TO_DSP2TXL_WIDTH 1 /* ADC2L_TO_DSP2TXL */ +#define WM8915_DACL_TO_DSP2TXL 0x0001 /* DACL_TO_DSP2TXL */ +#define WM8915_DACL_TO_DSP2TXL_MASK 0x0001 /* DACL_TO_DSP2TXL */ +#define WM8915_DACL_TO_DSP2TXL_SHIFT 0 /* DACL_TO_DSP2TXL */ +#define WM8915_DACL_TO_DSP2TXL_WIDTH 1 /* DACL_TO_DSP2TXL */ + +/* + * R1545 (0x609) - DSP2 TX Right Mixer Routing + */ +#define WM8915_ADC2R_TO_DSP2TXR 0x0002 /* ADC2R_TO_DSP2TXR */ +#define WM8915_ADC2R_TO_DSP2TXR_MASK 0x0002 /* ADC2R_TO_DSP2TXR */ +#define WM8915_ADC2R_TO_DSP2TXR_SHIFT 1 /* ADC2R_TO_DSP2TXR */ +#define WM8915_ADC2R_TO_DSP2TXR_WIDTH 1 /* ADC2R_TO_DSP2TXR */ +#define WM8915_DACR_TO_DSP2TXR 0x0001 /* DACR_TO_DSP2TXR */ +#define WM8915_DACR_TO_DSP2TXR_MASK 0x0001 /* DACR_TO_DSP2TXR */ +#define WM8915_DACR_TO_DSP2TXR_SHIFT 0 /* DACR_TO_DSP2TXR */ +#define WM8915_DACR_TO_DSP2TXR_WIDTH 1 /* DACR_TO_DSP2TXR */ + +/* + * R1546 (0x60A) - DSP TX Mixer Select + */ +#define WM8915_DAC_TO_DSPTX_SRC 0x0001 /* DAC_TO_DSPTX_SRC */ +#define WM8915_DAC_TO_DSPTX_SRC_MASK 0x0001 /* DAC_TO_DSPTX_SRC */ +#define WM8915_DAC_TO_DSPTX_SRC_SHIFT 0 /* DAC_TO_DSPTX_SRC */ +#define WM8915_DAC_TO_DSPTX_SRC_WIDTH 1 /* DAC_TO_DSPTX_SRC */ + +/* + * R1552 (0x610) - DAC Softmute + */ +#define WM8915_DAC_SOFTMUTEMODE 0x0002 /* DAC_SOFTMUTEMODE */ +#define WM8915_DAC_SOFTMUTEMODE_MASK 0x0002 /* DAC_SOFTMUTEMODE */ +#define WM8915_DAC_SOFTMUTEMODE_SHIFT 1 /* DAC_SOFTMUTEMODE */ +#define WM8915_DAC_SOFTMUTEMODE_WIDTH 1 /* DAC_SOFTMUTEMODE */ +#define WM8915_DAC_MUTERATE 0x0001 /* DAC_MUTERATE */ +#define WM8915_DAC_MUTERATE_MASK 0x0001 /* DAC_MUTERATE */ +#define WM8915_DAC_MUTERATE_SHIFT 0 /* DAC_MUTERATE */ +#define WM8915_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */ + +/* + * R1568 (0x620) - Oversampling + */ +#define WM8915_SPK_OSR128 0x0008 /* SPK_OSR128 */ +#define WM8915_SPK_OSR128_MASK 0x0008 /* SPK_OSR128 */ +#define WM8915_SPK_OSR128_SHIFT 3 /* SPK_OSR128 */ +#define WM8915_SPK_OSR128_WIDTH 1 /* SPK_OSR128 */ +#define WM8915_DMIC_OSR64 0x0004 /* DMIC_OSR64 */ +#define WM8915_DMIC_OSR64_MASK 0x0004 /* DMIC_OSR64 */ +#define WM8915_DMIC_OSR64_SHIFT 2 /* DMIC_OSR64 */ +#define WM8915_DMIC_OSR64_WIDTH 1 /* DMIC_OSR64 */ +#define WM8915_ADC_OSR128 0x0002 /* ADC_OSR128 */ +#define WM8915_ADC_OSR128_MASK 0x0002 /* ADC_OSR128 */ +#define WM8915_ADC_OSR128_SHIFT 1 /* ADC_OSR128 */ +#define WM8915_ADC_OSR128_WIDTH 1 /* ADC_OSR128 */ +#define WM8915_DAC_OSR128 0x0001 /* DAC_OSR128 */ +#define WM8915_DAC_OSR128_MASK 0x0001 /* DAC_OSR128 */ +#define WM8915_DAC_OSR128_SHIFT 0 /* DAC_OSR128 */ +#define WM8915_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */ + +/* + * R1569 (0x621) - Sidetone + */ +#define WM8915_ST_LPF 0x1000 /* ST_LPF */ +#define WM8915_ST_LPF_MASK 0x1000 /* ST_LPF */ +#define WM8915_ST_LPF_SHIFT 12 /* ST_LPF */ +#define WM8915_ST_LPF_WIDTH 1 /* ST_LPF */ +#define WM8915_ST_HPF_CUT_MASK 0x0380 /* ST_HPF_CUT - [9:7] */ +#define WM8915_ST_HPF_CUT_SHIFT 7 /* ST_HPF_CUT - [9:7] */ +#define WM8915_ST_HPF_CUT_WIDTH 3 /* ST_HPF_CUT - [9:7] */ +#define WM8915_ST_HPF 0x0040 /* ST_HPF */ +#define WM8915_ST_HPF_MASK 0x0040 /* ST_HPF */ +#define WM8915_ST_HPF_SHIFT 6 /* ST_HPF */ +#define WM8915_ST_HPF_WIDTH 1 /* ST_HPF */ +#define WM8915_STR_SEL 0x0002 /* STR_SEL */ +#define WM8915_STR_SEL_MASK 0x0002 /* STR_SEL */ +#define WM8915_STR_SEL_SHIFT 1 /* STR_SEL */ +#define WM8915_STR_SEL_WIDTH 1 /* STR_SEL */ +#define WM8915_STL_SEL 0x0001 /* STL_SEL */ +#define WM8915_STL_SEL_MASK 0x0001 /* STL_SEL */ +#define WM8915_STL_SEL_SHIFT 0 /* STL_SEL */ +#define WM8915_STL_SEL_WIDTH 1 /* STL_SEL */ + +/* + * R1792 (0x700) - GPIO 1 + */ +#define WM8915_GP1_DIR 0x8000 /* GP1_DIR */ +#define WM8915_GP1_DIR_MASK 0x8000 /* GP1_DIR */ +#define WM8915_GP1_DIR_SHIFT 15 /* GP1_DIR */ +#define WM8915_GP1_DIR_WIDTH 1 /* GP1_DIR */ +#define WM8915_GP1_PU 0x4000 /* GP1_PU */ +#define WM8915_GP1_PU_MASK 0x4000 /* GP1_PU */ +#define WM8915_GP1_PU_SHIFT 14 /* GP1_PU */ +#define WM8915_GP1_PU_WIDTH 1 /* GP1_PU */ +#define WM8915_GP1_PD 0x2000 /* GP1_PD */ +#define WM8915_GP1_PD_MASK 0x2000 /* GP1_PD */ +#define WM8915_GP1_PD_SHIFT 13 /* GP1_PD */ +#define WM8915_GP1_PD_WIDTH 1 /* GP1_PD */ +#define WM8915_GP1_POL 0x0400 /* GP1_POL */ +#define WM8915_GP1_POL_MASK 0x0400 /* GP1_POL */ +#define WM8915_GP1_POL_SHIFT 10 /* GP1_POL */ +#define WM8915_GP1_POL_WIDTH 1 /* GP1_POL */ +#define WM8915_GP1_OP_CFG 0x0200 /* GP1_OP_CFG */ +#define WM8915_GP1_OP_CFG_MASK 0x0200 /* GP1_OP_CFG */ +#define WM8915_GP1_OP_CFG_SHIFT 9 /* GP1_OP_CFG */ +#define WM8915_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */ +#define WM8915_GP1_DB 0x0100 /* GP1_DB */ +#define WM8915_GP1_DB_MASK 0x0100 /* GP1_DB */ +#define WM8915_GP1_DB_SHIFT 8 /* GP1_DB */ +#define WM8915_GP1_DB_WIDTH 1 /* GP1_DB */ +#define WM8915_GP1_LVL 0x0040 /* GP1_LVL */ +#define WM8915_GP1_LVL_MASK 0x0040 /* GP1_LVL */ +#define WM8915_GP1_LVL_SHIFT 6 /* GP1_LVL */ +#define WM8915_GP1_LVL_WIDTH 1 /* GP1_LVL */ +#define WM8915_GP1_FN_MASK 0x000F /* GP1_FN - [3:0] */ +#define WM8915_GP1_FN_SHIFT 0 /* GP1_FN - [3:0] */ +#define WM8915_GP1_FN_WIDTH 4 /* GP1_FN - [3:0] */ + +/* + * R1793 (0x701) - GPIO 2 + */ +#define WM8915_GP2_DIR 0x8000 /* GP2_DIR */ +#define WM8915_GP2_DIR_MASK 0x8000 /* GP2_DIR */ +#define WM8915_GP2_DIR_SHIFT 15 /* GP2_DIR */ +#define WM8915_GP2_DIR_WIDTH 1 /* GP2_DIR */ +#define WM8915_GP2_PU 0x4000 /* GP2_PU */ +#define WM8915_GP2_PU_MASK 0x4000 /* GP2_PU */ +#define WM8915_GP2_PU_SHIFT 14 /* GP2_PU */ +#define WM8915_GP2_PU_WIDTH 1 /* GP2_PU */ +#define WM8915_GP2_PD 0x2000 /* GP2_PD */ +#define WM8915_GP2_PD_MASK 0x2000 /* GP2_PD */ +#define WM8915_GP2_PD_SHIFT 13 /* GP2_PD */ +#define WM8915_GP2_PD_WIDTH 1 /* GP2_PD */ +#define WM8915_GP2_POL 0x0400 /* GP2_POL */ +#define WM8915_GP2_POL_MASK 0x0400 /* GP2_POL */ +#define WM8915_GP2_POL_SHIFT 10 /* GP2_POL */ +#define WM8915_GP2_POL_WIDTH 1 /* GP2_POL */ +#define WM8915_GP2_OP_CFG 0x0200 /* GP2_OP_CFG */ +#define WM8915_GP2_OP_CFG_MASK 0x0200 /* GP2_OP_CFG */ +#define WM8915_GP2_OP_CFG_SHIFT 9 /* GP2_OP_CFG */ +#define WM8915_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */ +#define WM8915_GP2_DB 0x0100 /* GP2_DB */ +#define WM8915_GP2_DB_MASK 0x0100 /* GP2_DB */ +#define WM8915_GP2_DB_SHIFT 8 /* GP2_DB */ +#define WM8915_GP2_DB_WIDTH 1 /* GP2_DB */ +#define WM8915_GP2_LVL 0x0040 /* GP2_LVL */ +#define WM8915_GP2_LVL_MASK 0x0040 /* GP2_LVL */ +#define WM8915_GP2_LVL_SHIFT 6 /* GP2_LVL */ +#define WM8915_GP2_LVL_WIDTH 1 /* GP2_LVL */ +#define WM8915_GP2_FN_MASK 0x000F /* GP2_FN - [3:0] */ +#define WM8915_GP2_FN_SHIFT 0 /* GP2_FN - [3:0] */ +#define WM8915_GP2_FN_WIDTH 4 /* GP2_FN - [3:0] */ + +/* + * R1794 (0x702) - GPIO 3 + */ +#define WM8915_GP3_DIR 0x8000 /* GP3_DIR */ +#define WM8915_GP3_DIR_MASK 0x8000 /* GP3_DIR */ +#define WM8915_GP3_DIR_SHIFT 15 /* GP3_DIR */ +#define WM8915_GP3_DIR_WIDTH 1 /* GP3_DIR */ +#define WM8915_GP3_PU 0x4000 /* GP3_PU */ +#define WM8915_GP3_PU_MASK 0x4000 /* GP3_PU */ +#define WM8915_GP3_PU_SHIFT 14 /* GP3_PU */ +#define WM8915_GP3_PU_WIDTH 1 /* GP3_PU */ +#define WM8915_GP3_PD 0x2000 /* GP3_PD */ +#define WM8915_GP3_PD_MASK 0x2000 /* GP3_PD */ +#define WM8915_GP3_PD_SHIFT 13 /* GP3_PD */ +#define WM8915_GP3_PD_WIDTH 1 /* GP3_PD */ +#define WM8915_GP3_POL 0x0400 /* GP3_POL */ +#define WM8915_GP3_POL_MASK 0x0400 /* GP3_POL */ +#define WM8915_GP3_POL_SHIFT 10 /* GP3_POL */ +#define WM8915_GP3_POL_WIDTH 1 /* GP3_POL */ +#define WM8915_GP3_OP_CFG 0x0200 /* GP3_OP_CFG */ +#define WM8915_GP3_OP_CFG_MASK 0x0200 /* GP3_OP_CFG */ +#define WM8915_GP3_OP_CFG_SHIFT 9 /* GP3_OP_CFG */ +#define WM8915_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */ +#define WM8915_GP3_DB 0x0100 /* GP3_DB */ +#define WM8915_GP3_DB_MASK 0x0100 /* GP3_DB */ +#define WM8915_GP3_DB_SHIFT 8 /* GP3_DB */ +#define WM8915_GP3_DB_WIDTH 1 /* GP3_DB */ +#define WM8915_GP3_LVL 0x0040 /* GP3_LVL */ +#define WM8915_GP3_LVL_MASK 0x0040 /* GP3_LVL */ +#define WM8915_GP3_LVL_SHIFT 6 /* GP3_LVL */ +#define WM8915_GP3_LVL_WIDTH 1 /* GP3_LVL */ +#define WM8915_GP3_FN_MASK 0x000F /* GP3_FN - [3:0] */ +#define WM8915_GP3_FN_SHIFT 0 /* GP3_FN - [3:0] */ +#define WM8915_GP3_FN_WIDTH 4 /* GP3_FN - [3:0] */ + +/* + * R1795 (0x703) - GPIO 4 + */ +#define WM8915_GP4_DIR 0x8000 /* GP4_DIR */ +#define WM8915_GP4_DIR_MASK 0x8000 /* GP4_DIR */ +#define WM8915_GP4_DIR_SHIFT 15 /* GP4_DIR */ +#define WM8915_GP4_DIR_WIDTH 1 /* GP4_DIR */ +#define WM8915_GP4_PU 0x4000 /* GP4_PU */ +#define WM8915_GP4_PU_MASK 0x4000 /* GP4_PU */ +#define WM8915_GP4_PU_SHIFT 14 /* GP4_PU */ +#define WM8915_GP4_PU_WIDTH 1 /* GP4_PU */ +#define WM8915_GP4_PD 0x2000 /* GP4_PD */ +#define WM8915_GP4_PD_MASK 0x2000 /* GP4_PD */ +#define WM8915_GP4_PD_SHIFT 13 /* GP4_PD */ +#define WM8915_GP4_PD_WIDTH 1 /* GP4_PD */ +#define WM8915_GP4_POL 0x0400 /* GP4_POL */ +#define WM8915_GP4_POL_MASK 0x0400 /* GP4_POL */ +#define WM8915_GP4_POL_SHIFT 10 /* GP4_POL */ +#define WM8915_GP4_POL_WIDTH 1 /* GP4_POL */ +#define WM8915_GP4_OP_CFG 0x0200 /* GP4_OP_CFG */ +#define WM8915_GP4_OP_CFG_MASK 0x0200 /* GP4_OP_CFG */ +#define WM8915_GP4_OP_CFG_SHIFT 9 /* GP4_OP_CFG */ +#define WM8915_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */ +#define WM8915_GP4_DB 0x0100 /* GP4_DB */ +#define WM8915_GP4_DB_MASK 0x0100 /* GP4_DB */ +#define WM8915_GP4_DB_SHIFT 8 /* GP4_DB */ +#define WM8915_GP4_DB_WIDTH 1 /* GP4_DB */ +#define WM8915_GP4_LVL 0x0040 /* GP4_LVL */ +#define WM8915_GP4_LVL_MASK 0x0040 /* GP4_LVL */ +#define WM8915_GP4_LVL_SHIFT 6 /* GP4_LVL */ +#define WM8915_GP4_LVL_WIDTH 1 /* GP4_LVL */ +#define WM8915_GP4_FN_MASK 0x000F /* GP4_FN - [3:0] */ +#define WM8915_GP4_FN_SHIFT 0 /* GP4_FN - [3:0] */ +#define WM8915_GP4_FN_WIDTH 4 /* GP4_FN - [3:0] */ + +/* + * R1796 (0x704) - GPIO 5 + */ +#define WM8915_GP5_DIR 0x8000 /* GP5_DIR */ +#define WM8915_GP5_DIR_MASK 0x8000 /* GP5_DIR */ +#define WM8915_GP5_DIR_SHIFT 15 /* GP5_DIR */ +#define WM8915_GP5_DIR_WIDTH 1 /* GP5_DIR */ +#define WM8915_GP5_PU 0x4000 /* GP5_PU */ +#define WM8915_GP5_PU_MASK 0x4000 /* GP5_PU */ +#define WM8915_GP5_PU_SHIFT 14 /* GP5_PU */ +#define WM8915_GP5_PU_WIDTH 1 /* GP5_PU */ +#define WM8915_GP5_PD 0x2000 /* GP5_PD */ +#define WM8915_GP5_PD_MASK 0x2000 /* GP5_PD */ +#define WM8915_GP5_PD_SHIFT 13 /* GP5_PD */ +#define WM8915_GP5_PD_WIDTH 1 /* GP5_PD */ +#define WM8915_GP5_POL 0x0400 /* GP5_POL */ +#define WM8915_GP5_POL_MASK 0x0400 /* GP5_POL */ +#define WM8915_GP5_POL_SHIFT 10 /* GP5_POL */ +#define WM8915_GP5_POL_WIDTH 1 /* GP5_POL */ +#define WM8915_GP5_OP_CFG 0x0200 /* GP5_OP_CFG */ +#define WM8915_GP5_OP_CFG_MASK 0x0200 /* GP5_OP_CFG */ +#define WM8915_GP5_OP_CFG_SHIFT 9 /* GP5_OP_CFG */ +#define WM8915_GP5_OP_CFG_WIDTH 1 /* GP5_OP_CFG */ +#define WM8915_GP5_DB 0x0100 /* GP5_DB */ +#define WM8915_GP5_DB_MASK 0x0100 /* GP5_DB */ +#define WM8915_GP5_DB_SHIFT 8 /* GP5_DB */ +#define WM8915_GP5_DB_WIDTH 1 /* GP5_DB */ +#define WM8915_GP5_LVL 0x0040 /* GP5_LVL */ +#define WM8915_GP5_LVL_MASK 0x0040 /* GP5_LVL */ +#define WM8915_GP5_LVL_SHIFT 6 /* GP5_LVL */ +#define WM8915_GP5_LVL_WIDTH 1 /* GP5_LVL */ +#define WM8915_GP5_FN_MASK 0x000F /* GP5_FN - [3:0] */ +#define WM8915_GP5_FN_SHIFT 0 /* GP5_FN - [3:0] */ +#define WM8915_GP5_FN_WIDTH 4 /* GP5_FN - [3:0] */ + +/* + * R1824 (0x720) - Pull Control (1) + */ +#define WM8915_DMICDAT2_PD 0x1000 /* DMICDAT2_PD */ +#define WM8915_DMICDAT2_PD_MASK 0x1000 /* DMICDAT2_PD */ +#define WM8915_DMICDAT2_PD_SHIFT 12 /* DMICDAT2_PD */ +#define WM8915_DMICDAT2_PD_WIDTH 1 /* DMICDAT2_PD */ +#define WM8915_DMICDAT1_PD 0x0400 /* DMICDAT1_PD */ +#define WM8915_DMICDAT1_PD_MASK 0x0400 /* DMICDAT1_PD */ +#define WM8915_DMICDAT1_PD_SHIFT 10 /* DMICDAT1_PD */ +#define WM8915_DMICDAT1_PD_WIDTH 1 /* DMICDAT1_PD */ +#define WM8915_MCLK2_PU 0x0200 /* MCLK2_PU */ +#define WM8915_MCLK2_PU_MASK 0x0200 /* MCLK2_PU */ +#define WM8915_MCLK2_PU_SHIFT 9 /* MCLK2_PU */ +#define WM8915_MCLK2_PU_WIDTH 1 /* MCLK2_PU */ +#define WM8915_MCLK2_PD 0x0100 /* MCLK2_PD */ +#define WM8915_MCLK2_PD_MASK 0x0100 /* MCLK2_PD */ +#define WM8915_MCLK2_PD_SHIFT 8 /* MCLK2_PD */ +#define WM8915_MCLK2_PD_WIDTH 1 /* MCLK2_PD */ +#define WM8915_MCLK1_PU 0x0080 /* MCLK1_PU */ +#define WM8915_MCLK1_PU_MASK 0x0080 /* MCLK1_PU */ +#define WM8915_MCLK1_PU_SHIFT 7 /* MCLK1_PU */ +#define WM8915_MCLK1_PU_WIDTH 1 /* MCLK1_PU */ +#define WM8915_MCLK1_PD 0x0040 /* MCLK1_PD */ +#define WM8915_MCLK1_PD_MASK 0x0040 /* MCLK1_PD */ +#define WM8915_MCLK1_PD_SHIFT 6 /* MCLK1_PD */ +#define WM8915_MCLK1_PD_WIDTH 1 /* MCLK1_PD */ +#define WM8915_DACDAT1_PU 0x0020 /* DACDAT1_PU */ +#define WM8915_DACDAT1_PU_MASK 0x0020 /* DACDAT1_PU */ +#define WM8915_DACDAT1_PU_SHIFT 5 /* DACDAT1_PU */ +#define WM8915_DACDAT1_PU_WIDTH 1 /* DACDAT1_PU */ +#define WM8915_DACDAT1_PD 0x0010 /* DACDAT1_PD */ +#define WM8915_DACDAT1_PD_MASK 0x0010 /* DACDAT1_PD */ +#define WM8915_DACDAT1_PD_SHIFT 4 /* DACDAT1_PD */ +#define WM8915_DACDAT1_PD_WIDTH 1 /* DACDAT1_PD */ +#define WM8915_DACLRCLK1_PU 0x0008 /* DACLRCLK1_PU */ +#define WM8915_DACLRCLK1_PU_MASK 0x0008 /* DACLRCLK1_PU */ +#define WM8915_DACLRCLK1_PU_SHIFT 3 /* DACLRCLK1_PU */ +#define WM8915_DACLRCLK1_PU_WIDTH 1 /* DACLRCLK1_PU */ +#define WM8915_DACLRCLK1_PD 0x0004 /* DACLRCLK1_PD */ +#define WM8915_DACLRCLK1_PD_MASK 0x0004 /* DACLRCLK1_PD */ +#define WM8915_DACLRCLK1_PD_SHIFT 2 /* DACLRCLK1_PD */ +#define WM8915_DACLRCLK1_PD_WIDTH 1 /* DACLRCLK1_PD */ +#define WM8915_BCLK1_PU 0x0002 /* BCLK1_PU */ +#define WM8915_BCLK1_PU_MASK 0x0002 /* BCLK1_PU */ +#define WM8915_BCLK1_PU_SHIFT 1 /* BCLK1_PU */ +#define WM8915_BCLK1_PU_WIDTH 1 /* BCLK1_PU */ +#define WM8915_BCLK1_PD 0x0001 /* BCLK1_PD */ +#define WM8915_BCLK1_PD_MASK 0x0001 /* BCLK1_PD */ +#define WM8915_BCLK1_PD_SHIFT 0 /* BCLK1_PD */ +#define WM8915_BCLK1_PD_WIDTH 1 /* BCLK1_PD */ + +/* + * R1825 (0x721) - Pull Control (2) + */ +#define WM8915_LDO1ENA_PD 0x0100 /* LDO1ENA_PD */ +#define WM8915_LDO1ENA_PD_MASK 0x0100 /* LDO1ENA_PD */ +#define WM8915_LDO1ENA_PD_SHIFT 8 /* LDO1ENA_PD */ +#define WM8915_LDO1ENA_PD_WIDTH 1 /* LDO1ENA_PD */ +#define WM8915_ADDR_PD 0x0040 /* ADDR_PD */ +#define WM8915_ADDR_PD_MASK 0x0040 /* ADDR_PD */ +#define WM8915_ADDR_PD_SHIFT 6 /* ADDR_PD */ +#define WM8915_ADDR_PD_WIDTH 1 /* ADDR_PD */ +#define WM8915_DACDAT2_PU 0x0020 /* DACDAT2_PU */ +#define WM8915_DACDAT2_PU_MASK 0x0020 /* DACDAT2_PU */ +#define WM8915_DACDAT2_PU_SHIFT 5 /* DACDAT2_PU */ +#define WM8915_DACDAT2_PU_WIDTH 1 /* DACDAT2_PU */ +#define WM8915_DACDAT2_PD 0x0010 /* DACDAT2_PD */ +#define WM8915_DACDAT2_PD_MASK 0x0010 /* DACDAT2_PD */ +#define WM8915_DACDAT2_PD_SHIFT 4 /* DACDAT2_PD */ +#define WM8915_DACDAT2_PD_WIDTH 1 /* DACDAT2_PD */ +#define WM8915_DACLRCLK2_PU 0x0008 /* DACLRCLK2_PU */ +#define WM8915_DACLRCLK2_PU_MASK 0x0008 /* DACLRCLK2_PU */ +#define WM8915_DACLRCLK2_PU_SHIFT 3 /* DACLRCLK2_PU */ +#define WM8915_DACLRCLK2_PU_WIDTH 1 /* DACLRCLK2_PU */ +#define WM8915_DACLRCLK2_PD 0x0004 /* DACLRCLK2_PD */ +#define WM8915_DACLRCLK2_PD_MASK 0x0004 /* DACLRCLK2_PD */ +#define WM8915_DACLRCLK2_PD_SHIFT 2 /* DACLRCLK2_PD */ +#define WM8915_DACLRCLK2_PD_WIDTH 1 /* DACLRCLK2_PD */ +#define WM8915_BCLK2_PU 0x0002 /* BCLK2_PU */ +#define WM8915_BCLK2_PU_MASK 0x0002 /* BCLK2_PU */ +#define WM8915_BCLK2_PU_SHIFT 1 /* BCLK2_PU */ +#define WM8915_BCLK2_PU_WIDTH 1 /* BCLK2_PU */ +#define WM8915_BCLK2_PD 0x0001 /* BCLK2_PD */ +#define WM8915_BCLK2_PD_MASK 0x0001 /* BCLK2_PD */ +#define WM8915_BCLK2_PD_SHIFT 0 /* BCLK2_PD */ +#define WM8915_BCLK2_PD_WIDTH 1 /* BCLK2_PD */ + +/* + * R1840 (0x730) - Interrupt Status 1 + */ +#define WM8915_GP5_EINT 0x0010 /* GP5_EINT */ +#define WM8915_GP5_EINT_MASK 0x0010 /* GP5_EINT */ +#define WM8915_GP5_EINT_SHIFT 4 /* GP5_EINT */ +#define WM8915_GP5_EINT_WIDTH 1 /* GP5_EINT */ +#define WM8915_GP4_EINT 0x0008 /* GP4_EINT */ +#define WM8915_GP4_EINT_MASK 0x0008 /* GP4_EINT */ +#define WM8915_GP4_EINT_SHIFT 3 /* GP4_EINT */ +#define WM8915_GP4_EINT_WIDTH 1 /* GP4_EINT */ +#define WM8915_GP3_EINT 0x0004 /* GP3_EINT */ +#define WM8915_GP3_EINT_MASK 0x0004 /* GP3_EINT */ +#define WM8915_GP3_EINT_SHIFT 2 /* GP3_EINT */ +#define WM8915_GP3_EINT_WIDTH 1 /* GP3_EINT */ +#define WM8915_GP2_EINT 0x0002 /* GP2_EINT */ +#define WM8915_GP2_EINT_MASK 0x0002 /* GP2_EINT */ +#define WM8915_GP2_EINT_SHIFT 1 /* GP2_EINT */ +#define WM8915_GP2_EINT_WIDTH 1 /* GP2_EINT */ +#define WM8915_GP1_EINT 0x0001 /* GP1_EINT */ +#define WM8915_GP1_EINT_MASK 0x0001 /* GP1_EINT */ +#define WM8915_GP1_EINT_SHIFT 0 /* GP1_EINT */ +#define WM8915_GP1_EINT_WIDTH 1 /* GP1_EINT */ + +/* + * R1841 (0x731) - Interrupt Status 2 + */ +#define WM8915_DCS_DONE_23_EINT 0x1000 /* DCS_DONE_23_EINT */ +#define WM8915_DCS_DONE_23_EINT_MASK 0x1000 /* DCS_DONE_23_EINT */ +#define WM8915_DCS_DONE_23_EINT_SHIFT 12 /* DCS_DONE_23_EINT */ +#define WM8915_DCS_DONE_23_EINT_WIDTH 1 /* DCS_DONE_23_EINT */ +#define WM8915_DCS_DONE_01_EINT 0x0800 /* DCS_DONE_01_EINT */ +#define WM8915_DCS_DONE_01_EINT_MASK 0x0800 /* DCS_DONE_01_EINT */ +#define WM8915_DCS_DONE_01_EINT_SHIFT 11 /* DCS_DONE_01_EINT */ +#define WM8915_DCS_DONE_01_EINT_WIDTH 1 /* DCS_DONE_01_EINT */ +#define WM8915_WSEQ_DONE_EINT 0x0400 /* WSEQ_DONE_EINT */ +#define WM8915_WSEQ_DONE_EINT_MASK 0x0400 /* WSEQ_DONE_EINT */ +#define WM8915_WSEQ_DONE_EINT_SHIFT 10 /* WSEQ_DONE_EINT */ +#define WM8915_WSEQ_DONE_EINT_WIDTH 1 /* WSEQ_DONE_EINT */ +#define WM8915_FIFOS_ERR_EINT 0x0200 /* FIFOS_ERR_EINT */ +#define WM8915_FIFOS_ERR_EINT_MASK 0x0200 /* FIFOS_ERR_EINT */ +#define WM8915_FIFOS_ERR_EINT_SHIFT 9 /* FIFOS_ERR_EINT */ +#define WM8915_FIFOS_ERR_EINT_WIDTH 1 /* FIFOS_ERR_EINT */ +#define WM8915_DSP2DRC_SIG_DET_EINT 0x0080 /* DSP2DRC_SIG_DET_EINT */ +#define WM8915_DSP2DRC_SIG_DET_EINT_MASK 0x0080 /* DSP2DRC_SIG_DET_EINT */ +#define WM8915_DSP2DRC_SIG_DET_EINT_SHIFT 7 /* DSP2DRC_SIG_DET_EINT */ +#define WM8915_DSP2DRC_SIG_DET_EINT_WIDTH 1 /* DSP2DRC_SIG_DET_EINT */ +#define WM8915_DSP1DRC_SIG_DET_EINT 0x0040 /* DSP1DRC_SIG_DET_EINT */ +#define WM8915_DSP1DRC_SIG_DET_EINT_MASK 0x0040 /* DSP1DRC_SIG_DET_EINT */ +#define WM8915_DSP1DRC_SIG_DET_EINT_SHIFT 6 /* DSP1DRC_SIG_DET_EINT */ +#define WM8915_DSP1DRC_SIG_DET_EINT_WIDTH 1 /* DSP1DRC_SIG_DET_EINT */ +#define WM8915_FLL_SW_CLK_DONE_EINT 0x0008 /* FLL_SW_CLK_DONE_EINT */ +#define WM8915_FLL_SW_CLK_DONE_EINT_MASK 0x0008 /* FLL_SW_CLK_DONE_EINT */ +#define WM8915_FLL_SW_CLK_DONE_EINT_SHIFT 3 /* FLL_SW_CLK_DONE_EINT */ +#define WM8915_FLL_SW_CLK_DONE_EINT_WIDTH 1 /* FLL_SW_CLK_DONE_EINT */ +#define WM8915_FLL_LOCK_EINT 0x0004 /* FLL_LOCK_EINT */ +#define WM8915_FLL_LOCK_EINT_MASK 0x0004 /* FLL_LOCK_EINT */ +#define WM8915_FLL_LOCK_EINT_SHIFT 2 /* FLL_LOCK_EINT */ +#define WM8915_FLL_LOCK_EINT_WIDTH 1 /* FLL_LOCK_EINT */ +#define WM8915_HP_DONE_EINT 0x0002 /* HP_DONE_EINT */ +#define WM8915_HP_DONE_EINT_MASK 0x0002 /* HP_DONE_EINT */ +#define WM8915_HP_DONE_EINT_SHIFT 1 /* HP_DONE_EINT */ +#define WM8915_HP_DONE_EINT_WIDTH 1 /* HP_DONE_EINT */ +#define WM8915_MICD_EINT 0x0001 /* MICD_EINT */ +#define WM8915_MICD_EINT_MASK 0x0001 /* MICD_EINT */ +#define WM8915_MICD_EINT_SHIFT 0 /* MICD_EINT */ +#define WM8915_MICD_EINT_WIDTH 1 /* MICD_EINT */ + +/* + * R1842 (0x732) - Interrupt Raw Status 2 + */ +#define WM8915_DCS_DONE_23_STS 0x1000 /* DCS_DONE_23_STS */ +#define WM8915_DCS_DONE_23_STS_MASK 0x1000 /* DCS_DONE_23_STS */ +#define WM8915_DCS_DONE_23_STS_SHIFT 12 /* DCS_DONE_23_STS */ +#define WM8915_DCS_DONE_23_STS_WIDTH 1 /* DCS_DONE_23_STS */ +#define WM8915_DCS_DONE_01_STS 0x0800 /* DCS_DONE_01_STS */ +#define WM8915_DCS_DONE_01_STS_MASK 0x0800 /* DCS_DONE_01_STS */ +#define WM8915_DCS_DONE_01_STS_SHIFT 11 /* DCS_DONE_01_STS */ +#define WM8915_DCS_DONE_01_STS_WIDTH 1 /* DCS_DONE_01_STS */ +#define WM8915_WSEQ_DONE_STS 0x0400 /* WSEQ_DONE_STS */ +#define WM8915_WSEQ_DONE_STS_MASK 0x0400 /* WSEQ_DONE_STS */ +#define WM8915_WSEQ_DONE_STS_SHIFT 10 /* WSEQ_DONE_STS */ +#define WM8915_WSEQ_DONE_STS_WIDTH 1 /* WSEQ_DONE_STS */ +#define WM8915_FIFOS_ERR_STS 0x0200 /* FIFOS_ERR_STS */ +#define WM8915_FIFOS_ERR_STS_MASK 0x0200 /* FIFOS_ERR_STS */ +#define WM8915_FIFOS_ERR_STS_SHIFT 9 /* FIFOS_ERR_STS */ +#define WM8915_FIFOS_ERR_STS_WIDTH 1 /* FIFOS_ERR_STS */ +#define WM8915_DSP2DRC_SIG_DET_STS 0x0080 /* DSP2DRC_SIG_DET_STS */ +#define WM8915_DSP2DRC_SIG_DET_STS_MASK 0x0080 /* DSP2DRC_SIG_DET_STS */ +#define WM8915_DSP2DRC_SIG_DET_STS_SHIFT 7 /* DSP2DRC_SIG_DET_STS */ +#define WM8915_DSP2DRC_SIG_DET_STS_WIDTH 1 /* DSP2DRC_SIG_DET_STS */ +#define WM8915_DSP1DRC_SIG_DET_STS 0x0040 /* DSP1DRC_SIG_DET_STS */ +#define WM8915_DSP1DRC_SIG_DET_STS_MASK 0x0040 /* DSP1DRC_SIG_DET_STS */ +#define WM8915_DSP1DRC_SIG_DET_STS_SHIFT 6 /* DSP1DRC_SIG_DET_STS */ +#define WM8915_DSP1DRC_SIG_DET_STS_WIDTH 1 /* DSP1DRC_SIG_DET_STS */ +#define WM8915_FLL_LOCK_STS 0x0004 /* FLL_LOCK_STS */ +#define WM8915_FLL_LOCK_STS_MASK 0x0004 /* FLL_LOCK_STS */ +#define WM8915_FLL_LOCK_STS_SHIFT 2 /* FLL_LOCK_STS */ +#define WM8915_FLL_LOCK_STS_WIDTH 1 /* FLL_LOCK_STS */ + +/* + * R1848 (0x738) - Interrupt Status 1 Mask + */ +#define WM8915_IM_GP5_EINT 0x0010 /* IM_GP5_EINT */ +#define WM8915_IM_GP5_EINT_MASK 0x0010 /* IM_GP5_EINT */ +#define WM8915_IM_GP5_EINT_SHIFT 4 /* IM_GP5_EINT */ +#define WM8915_IM_GP5_EINT_WIDTH 1 /* IM_GP5_EINT */ +#define WM8915_IM_GP4_EINT 0x0008 /* IM_GP4_EINT */ +#define WM8915_IM_GP4_EINT_MASK 0x0008 /* IM_GP4_EINT */ +#define WM8915_IM_GP4_EINT_SHIFT 3 /* IM_GP4_EINT */ +#define WM8915_IM_GP4_EINT_WIDTH 1 /* IM_GP4_EINT */ +#define WM8915_IM_GP3_EINT 0x0004 /* IM_GP3_EINT */ +#define WM8915_IM_GP3_EINT_MASK 0x0004 /* IM_GP3_EINT */ +#define WM8915_IM_GP3_EINT_SHIFT 2 /* IM_GP3_EINT */ +#define WM8915_IM_GP3_EINT_WIDTH 1 /* IM_GP3_EINT */ +#define WM8915_IM_GP2_EINT 0x0002 /* IM_GP2_EINT */ +#define WM8915_IM_GP2_EINT_MASK 0x0002 /* IM_GP2_EINT */ +#define WM8915_IM_GP2_EINT_SHIFT 1 /* IM_GP2_EINT */ +#define WM8915_IM_GP2_EINT_WIDTH 1 /* IM_GP2_EINT */ +#define WM8915_IM_GP1_EINT 0x0001 /* IM_GP1_EINT */ +#define WM8915_IM_GP1_EINT_MASK 0x0001 /* IM_GP1_EINT */ +#define WM8915_IM_GP1_EINT_SHIFT 0 /* IM_GP1_EINT */ +#define WM8915_IM_GP1_EINT_WIDTH 1 /* IM_GP1_EINT */ + +/* + * R1849 (0x739) - Interrupt Status 2 Mask + */ +#define WM8915_IM_DCS_DONE_23_EINT 0x1000 /* IM_DCS_DONE_23_EINT */ +#define WM8915_IM_DCS_DONE_23_EINT_MASK 0x1000 /* IM_DCS_DONE_23_EINT */ +#define WM8915_IM_DCS_DONE_23_EINT_SHIFT 12 /* IM_DCS_DONE_23_EINT */ +#define WM8915_IM_DCS_DONE_23_EINT_WIDTH 1 /* IM_DCS_DONE_23_EINT */ +#define WM8915_IM_DCS_DONE_01_EINT 0x0800 /* IM_DCS_DONE_01_EINT */ +#define WM8915_IM_DCS_DONE_01_EINT_MASK 0x0800 /* IM_DCS_DONE_01_EINT */ +#define WM8915_IM_DCS_DONE_01_EINT_SHIFT 11 /* IM_DCS_DONE_01_EINT */ +#define WM8915_IM_DCS_DONE_01_EINT_WIDTH 1 /* IM_DCS_DONE_01_EINT */ +#define WM8915_IM_WSEQ_DONE_EINT 0x0400 /* IM_WSEQ_DONE_EINT */ +#define WM8915_IM_WSEQ_DONE_EINT_MASK 0x0400 /* IM_WSEQ_DONE_EINT */ +#define WM8915_IM_WSEQ_DONE_EINT_SHIFT 10 /* IM_WSEQ_DONE_EINT */ +#define WM8915_IM_WSEQ_DONE_EINT_WIDTH 1 /* IM_WSEQ_DONE_EINT */ +#define WM8915_IM_FIFOS_ERR_EINT 0x0200 /* IM_FIFOS_ERR_EINT */ +#define WM8915_IM_FIFOS_ERR_EINT_MASK 0x0200 /* IM_FIFOS_ERR_EINT */ +#define WM8915_IM_FIFOS_ERR_EINT_SHIFT 9 /* IM_FIFOS_ERR_EINT */ +#define WM8915_IM_FIFOS_ERR_EINT_WIDTH 1 /* IM_FIFOS_ERR_EINT */ +#define WM8915_IM_DSP2DRC_SIG_DET_EINT 0x0080 /* IM_DSP2DRC_SIG_DET_EINT */ +#define WM8915_IM_DSP2DRC_SIG_DET_EINT_MASK 0x0080 /* IM_DSP2DRC_SIG_DET_EINT */ +#define WM8915_IM_DSP2DRC_SIG_DET_EINT_SHIFT 7 /* IM_DSP2DRC_SIG_DET_EINT */ +#define WM8915_IM_DSP2DRC_SIG_DET_EINT_WIDTH 1 /* IM_DSP2DRC_SIG_DET_EINT */ +#define WM8915_IM_DSP1DRC_SIG_DET_EINT 0x0040 /* IM_DSP1DRC_SIG_DET_EINT */ +#define WM8915_IM_DSP1DRC_SIG_DET_EINT_MASK 0x0040 /* IM_DSP1DRC_SIG_DET_EINT */ +#define WM8915_IM_DSP1DRC_SIG_DET_EINT_SHIFT 6 /* IM_DSP1DRC_SIG_DET_EINT */ +#define WM8915_IM_DSP1DRC_SIG_DET_EINT_WIDTH 1 /* IM_DSP1DRC_SIG_DET_EINT */ +#define WM8915_IM_FLL_SW_CLK_DONE_EINT 0x0008 /* IM_FLL_SW_CLK_DONE_EINT */ +#define WM8915_IM_FLL_SW_CLK_DONE_EINT_MASK 0x0008 /* IM_FLL_SW_CLK_DONE_EINT */ +#define WM8915_IM_FLL_SW_CLK_DONE_EINT_SHIFT 3 /* IM_FLL_SW_CLK_DONE_EINT */ +#define WM8915_IM_FLL_SW_CLK_DONE_EINT_WIDTH 1 /* IM_FLL_SW_CLK_DONE_EINT */ +#define WM8915_IM_FLL_LOCK_EINT 0x0004 /* IM_FLL_LOCK_EINT */ +#define WM8915_IM_FLL_LOCK_EINT_MASK 0x0004 /* IM_FLL_LOCK_EINT */ +#define WM8915_IM_FLL_LOCK_EINT_SHIFT 2 /* IM_FLL_LOCK_EINT */ +#define WM8915_IM_FLL_LOCK_EINT_WIDTH 1 /* IM_FLL_LOCK_EINT */ +#define WM8915_IM_HP_DONE_EINT 0x0002 /* IM_HP_DONE_EINT */ +#define WM8915_IM_HP_DONE_EINT_MASK 0x0002 /* IM_HP_DONE_EINT */ +#define WM8915_IM_HP_DONE_EINT_SHIFT 1 /* IM_HP_DONE_EINT */ +#define WM8915_IM_HP_DONE_EINT_WIDTH 1 /* IM_HP_DONE_EINT */ +#define WM8915_IM_MICD_EINT 0x0001 /* IM_MICD_EINT */ +#define WM8915_IM_MICD_EINT_MASK 0x0001 /* IM_MICD_EINT */ +#define WM8915_IM_MICD_EINT_SHIFT 0 /* IM_MICD_EINT */ +#define WM8915_IM_MICD_EINT_WIDTH 1 /* IM_MICD_EINT */ + +/* + * R1856 (0x740) - Interrupt Control + */ +#define WM8915_IM_IRQ 0x0001 /* IM_IRQ */ +#define WM8915_IM_IRQ_MASK 0x0001 /* IM_IRQ */ +#define WM8915_IM_IRQ_SHIFT 0 /* IM_IRQ */ +#define WM8915_IM_IRQ_WIDTH 1 /* IM_IRQ */ + +/* + * R2048 (0x800) - Left PDM Speaker + */ +#define WM8915_SPKL_ENA 0x0010 /* SPKL_ENA */ +#define WM8915_SPKL_ENA_MASK 0x0010 /* SPKL_ENA */ +#define WM8915_SPKL_ENA_SHIFT 4 /* SPKL_ENA */ +#define WM8915_SPKL_ENA_WIDTH 1 /* SPKL_ENA */ +#define WM8915_SPKL_MUTE 0x0008 /* SPKL_MUTE */ +#define WM8915_SPKL_MUTE_MASK 0x0008 /* SPKL_MUTE */ +#define WM8915_SPKL_MUTE_SHIFT 3 /* SPKL_MUTE */ +#define WM8915_SPKL_MUTE_WIDTH 1 /* SPKL_MUTE */ +#define WM8915_SPKL_MUTE_ZC 0x0004 /* SPKL_MUTE_ZC */ +#define WM8915_SPKL_MUTE_ZC_MASK 0x0004 /* SPKL_MUTE_ZC */ +#define WM8915_SPKL_MUTE_ZC_SHIFT 2 /* SPKL_MUTE_ZC */ +#define WM8915_SPKL_MUTE_ZC_WIDTH 1 /* SPKL_MUTE_ZC */ +#define WM8915_SPKL_SRC_MASK 0x0003 /* SPKL_SRC - [1:0] */ +#define WM8915_SPKL_SRC_SHIFT 0 /* SPKL_SRC - [1:0] */ +#define WM8915_SPKL_SRC_WIDTH 2 /* SPKL_SRC - [1:0] */ + +/* + * R2049 (0x801) - Right PDM Speaker + */ +#define WM8915_SPKR_ENA 0x0010 /* SPKR_ENA */ +#define WM8915_SPKR_ENA_MASK 0x0010 /* SPKR_ENA */ +#define WM8915_SPKR_ENA_SHIFT 4 /* SPKR_ENA */ +#define WM8915_SPKR_ENA_WIDTH 1 /* SPKR_ENA */ +#define WM8915_SPKR_MUTE 0x0008 /* SPKR_MUTE */ +#define WM8915_SPKR_MUTE_MASK 0x0008 /* SPKR_MUTE */ +#define WM8915_SPKR_MUTE_SHIFT 3 /* SPKR_MUTE */ +#define WM8915_SPKR_MUTE_WIDTH 1 /* SPKR_MUTE */ +#define WM8915_SPKR_MUTE_ZC 0x0004 /* SPKR_MUTE_ZC */ +#define WM8915_SPKR_MUTE_ZC_MASK 0x0004 /* SPKR_MUTE_ZC */ +#define WM8915_SPKR_MUTE_ZC_SHIFT 2 /* SPKR_MUTE_ZC */ +#define WM8915_SPKR_MUTE_ZC_WIDTH 1 /* SPKR_MUTE_ZC */ +#define WM8915_SPKR_SRC_MASK 0x0003 /* SPKR_SRC - [1:0] */ +#define WM8915_SPKR_SRC_SHIFT 0 /* SPKR_SRC - [1:0] */ +#define WM8915_SPKR_SRC_WIDTH 2 /* SPKR_SRC - [1:0] */ + +/* + * R2050 (0x802) - PDM Speaker Mute Sequence + */ +#define WM8915_SPK_MUTE_ENDIAN 0x0100 /* SPK_MUTE_ENDIAN */ +#define WM8915_SPK_MUTE_ENDIAN_MASK 0x0100 /* SPK_MUTE_ENDIAN */ +#define WM8915_SPK_MUTE_ENDIAN_SHIFT 8 /* SPK_MUTE_ENDIAN */ +#define WM8915_SPK_MUTE_ENDIAN_WIDTH 1 /* SPK_MUTE_ENDIAN */ +#define WM8915_SPK_MUTE_SEQ1_MASK 0x00FF /* SPK_MUTE_SEQ1 - [7:0] */ +#define WM8915_SPK_MUTE_SEQ1_SHIFT 0 /* SPK_MUTE_SEQ1 - [7:0] */ +#define WM8915_SPK_MUTE_SEQ1_WIDTH 8 /* SPK_MUTE_SEQ1 - [7:0] */ + +/* + * R2051 (0x803) - PDM Speaker Volume + */ +#define WM8915_SPKR_VOL_MASK 0x00F0 /* SPKR_VOL - [7:4] */ +#define WM8915_SPKR_VOL_SHIFT 4 /* SPKR_VOL - [7:4] */ +#define WM8915_SPKR_VOL_WIDTH 4 /* SPKR_VOL - [7:4] */ +#define WM8915_SPKL_VOL_MASK 0x000F /* SPKL_VOL - [3:0] */ +#define WM8915_SPKL_VOL_SHIFT 0 /* SPKL_VOL - [3:0] */ +#define WM8915_SPKL_VOL_WIDTH 4 /* SPKL_VOL - [3:0] */ + +#endif -- cgit v1.2.3 From 4bb3f43c6e3f3f6897f60ee1bab423ce23539d2c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Apr 2011 16:49:42 +0900 Subject: ASoC: Add initial WM1250-EV1 Springbank audio I/O module driver The WM1250-EV1 Springbank audio I/O module for the Wolfson Glenfarclas reference platform provides a simple audio I/O with an independant clock domain, intended to simulate cellular modem and bluetooth subsystems within the platform. The card supports some limited GPIO based control but this is currently not implemented. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm1250-ev1.c | 108 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 114 insertions(+) create mode 100644 sound/soc/codecs/wm1250-ev1.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 54d5dd6c63a..2a6971891d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,6 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE + select SND_SOC_WM1250_EV1 if I2C select SND_SOC_WM2000 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 @@ -246,6 +247,9 @@ config SND_SOC_UDA1380 config SND_SOC_WL1273 tristate +config SND_SOC_WM1250_EV1 + tristate + config SND_SOC_WM8350 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 02425e6a87e..4cb2f42dbff 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -38,6 +38,7 @@ snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o +snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o @@ -128,6 +129,7 @@ obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o +obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c new file mode 100644 index 00000000000..9cf699e165e --- /dev/null +++ b/sound/soc/codecs/wm1250-ev1.c @@ -0,0 +1,108 @@ +/* + * Driver for the 1250-EV1 audio I/O module + * + * Copyright 2011 Wolfson Microelectronics plc + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include + +#include +#include + +static const struct snd_soc_dapm_widget wm1250_ev1_dapm_widgets[] = { +SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DAC", "wm1250-ev1 Playback", SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_INPUT("WM1250 Input"), +SND_SOC_DAPM_INPUT("WM1250 Output"), +}; + +static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = { + { "ADC", NULL, "WM1250 Input" }, + { "WM1250 Output", NULL, "DAC" }, +}; + +static struct snd_soc_dai_driver wm1250_ev1_dai = { + .name = "wm1250-ev1", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { + .dapm_widgets = wm1250_ev1_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm1250_ev1_dapm_widgets), + .dapm_routes = wm1250_ev1_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm1250_ev1_dapm_routes), +}; + +static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, + &wm1250_ev1_dai, 1); +} + +static int __devexit wm1250_ev1_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static const struct i2c_device_id wm1250_ev1_i2c_id[] = { + { "wm1250-ev1", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm1250_ev1__id); + +static struct i2c_driver wm1250_ev1_i2c_driver = { + .driver = { + .name = "wm1250-ev1", + .owner = THIS_MODULE, + }, + .probe = wm1250_ev1_probe, + .remove = __devexit_p(wm1250_ev1_remove), + .id_table = wm1250_ev1_i2c_id, +}; + +static int __init wm1250_ev1_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&wm1250_ev1_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM1250-EV1 I2C driver: %d\n", ret); + + return ret; +} +module_init(wm1250_ev1_modinit); + +static void __exit wm1250_ev1_exit(void) +{ + i2c_del_driver(&wm1250_ev1_i2c_driver); +} +module_exit(wm1250_ev1_exit); + +MODULE_AUTHOR("Mark Brown "); +MODULE_DESCRIPTION("WM1250-EV1 audio I/O module driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 420dd718ad0bc4a4e07ae0ac7f8eac7545eb253a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Apr 2011 21:40:29 -0700 Subject: ASoC: Fix mis cherry-pick of wm1250-ev1 driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 9cf699e165e..14d0716bf00 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -73,7 +73,7 @@ static const struct i2c_device_id wm1250_ev1_i2c_id[] = { { "wm1250-ev1", 0 }, { } }; -MODULE_DEVICE_TABLE(i2c, wm1250_ev1__id); +MODULE_DEVICE_TABLE(i2c, wm1250_ev1_i2c_id); static struct i2c_driver wm1250_ev1_i2c_driver = { .driver = { -- cgit v1.2.3 From 82a58a8b7f293e5bab3dd41ee160867bcad41f37 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 12 Apr 2011 09:09:17 +0300 Subject: ASoC: tlv320dac33: Lower the OSC calibration time To get correct calibration, we can decrease the time needed for the OSC to calibrate itself. With this change we can save ~15ms in the OSC calibration phase. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4857f2ae066..869c1a9766b 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -939,8 +939,8 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) /* Write registers 0x08 and 0x09 (MSB, LSB) */ dac33_write16(codec, DAC33_INT_OSC_FREQ_RAT_A, oscset); - /* calib time: 128 is a nice number ;) */ - dac33_write(codec, DAC33_CALIB_TIME, 128); + /* OSC calibration time */ + dac33_write(codec, DAC33_CALIB_TIME, 96); /* adjustment treshold & step */ dac33_write(codec, DAC33_INT_OSC_CTRL_B, DAC33_ADJTHRSHLD(2) | -- cgit v1.2.3 From 01b07e2d84c887b432353ead846f4497a33b5f5d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Apr 2011 23:39:10 -0700 Subject: ASoC: Move WM8915 FLL operations onto the CODEC Since the WM8915 FLL is not tied to a particular audio interface move it to a CODEC wide operation. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 3adad2872c7..083609418bf 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -1997,10 +1997,9 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, return 0; } -static int wm8915_set_fll(struct snd_soc_dai *dai, int fll_id, int source, +static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source, unsigned int Fref, unsigned int Fout) { - struct snd_soc_codec *codec = dai->codec; struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); struct _fll_div fll_div; unsigned long timeout; @@ -2776,6 +2775,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8915 = { .num_dapm_widgets = ARRAY_SIZE(wm8915_dapm_widgets), .dapm_routes = wm8915_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8915_dapm_routes), + .set_pll = wm8915_set_fll, }; #define WM8915_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ @@ -2788,7 +2788,6 @@ static struct snd_soc_dai_ops wm8915_dai_ops = { .set_fmt = wm8915_set_fmt, .hw_params = wm8915_hw_params, .set_sysclk = wm8915_set_sysclk, - .set_pll = wm8915_set_fll, }; static struct snd_soc_dai_driver wm8915_dai[] = { -- cgit v1.2.3 From 9a841ebb9cac3f1b7253bb01c304f89b1af25aba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Apr 2011 17:51:37 -0700 Subject: ASoC: Create card DAPM widgets early so they can be used in callbacks This helps with things like setting up the initial state. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1f114673e95..3b3a377d087 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1878,6 +1878,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); #endif + if (card->dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, + card->num_dapm_widgets); + /* initialise the sound card only once */ if (card->probe) { ret = card->probe(card); @@ -1911,9 +1915,6 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->controls, card->num_controls); - if (card->dapm_widgets) - snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, - card->num_dapm_widgets); if (card->dapm_routes) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); -- cgit v1.2.3 From 9b8dc66fbae53def2547df8c0a5cbe8924ea2b1f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Apr 2011 17:24:39 -0700 Subject: ASoC: Initial audio support for Speyside on Cragganmore 6410 This is minimal code required to get audio out of the Speyside audio subsystem on the Wolfson Cragganmore 6410 reference platform. It sets up the link between the CPU and AIF1 of the WM8915 on the system, enabling audio playback via the headphone and speaker outputs of the device (which require no further configuration except runtime). It allows verification of basic functionality of the system. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/Kconfig | 6 +++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/speyside.c | 123 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 131 insertions(+) create mode 100644 sound/soc/samsung/speyside.c (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b8c7a2e0d21..726af2ede79 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -170,3 +170,9 @@ config SND_SOC_SMDK_WM8580_PCM select SND_SAMSUNG_PCM help Say Y if you want to add support for SoC audio on the SMDK. + +config SND_SOC_SPEYSIDE + tristate "Audio support for Wolfson Speyside" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM8915 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 6c598fc51a0..683843a2744 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -35,6 +35,7 @@ snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o +snd-soc-speyside-objs := speyside.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -53,3 +54,4 @@ obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o +obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c new file mode 100644 index 00000000000..f6dcee7b218 --- /dev/null +++ b/sound/soc/samsung/speyside.c @@ -0,0 +1,123 @@ +/* + * Speyside audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include + +#include "../codecs/wm8915.h" + +static int speyside_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK1, + 32768, 256 * 48000); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_FLL, + 256 * 48000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops speyside_ops = { + .hw_params = speyside_hw_params, +}; + +static struct snd_soc_dai_link speyside_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm8915-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm8915.1-001a", + .ops = &speyside_ops, + }, +}; + +static struct snd_soc_card speyside = { + .name = "Speyside", + .dai_link = speyside_dai, + .num_links = ARRAY_SIZE(speyside_dai), +}; + +static __devinit int speyside_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &speyside; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit speyside_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver speyside_driver = { + .driver = { + .name = "speyside", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = speyside_probe, + .remove = __devexit_p(speyside_remove), +}; + +static int __init speyside_audio_init(void) +{ + return platform_driver_register(&speyside_driver); +} +module_init(speyside_audio_init); + +static void __exit speyside_audio_exit(void) +{ + platform_driver_unregister(&speyside_driver); +} +module_exit(speyside_audio_exit); + +MODULE_DESCRIPTION("Speyside audio support"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:speyside"); -- cgit v1.2.3 From ecfb1adf5f037eaff0b678918d84a8febd9f1c3e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Apr 2011 23:09:15 -0700 Subject: ASoC: Add basic widgets for WM8915 Speyside Provide widgets for the basic widgets connected directly to the WM8915 on Speyside - the headphones, speaker, digital and analogue microphones. For the outputs this is just documentation, for the inputs this ensures that the relevant microphone biases are enabled when they are in use. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index f6dcee7b218..6ec44a354a6 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -63,10 +63,38 @@ static struct snd_soc_dai_link speyside_dai[] = { }, }; +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + + SND_SOC_DAPM_SPK("Main Speaker", NULL), + + SND_SOC_DAPM_MIC("Main AMIC", NULL), + SND_SOC_DAPM_MIC("Main DMIC", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "IN1LP", NULL, "MICB2" }, + { "MICB2", NULL, "Main AMIC" }, + + { "DMIC1DAT", NULL, "MICB1" }, + { "DMIC2DAT", NULL, "MICB1" }, + { "MICB1", NULL, "Main DMIC" }, + + { "Headphone", NULL, "HPOUT1L" }, + { "Headphone", NULL, "HPOUT1R" }, + + { "Main Speaker", NULL, "SPKDAT" }, +}; + static struct snd_soc_card speyside = { .name = "Speyside", .dai_link = speyside_dai, .num_links = ARRAY_SIZE(speyside_dai), + + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), }; static __devinit int speyside_probe(struct platform_device *pdev) -- cgit v1.2.3 From ea0a591a28b9249ea585f2cf8045e43f57f48fbb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Apr 2011 23:32:03 -0700 Subject: ASoC: Optimise clock management for WM8915 Speyside Dynamically enable and disable the FLL on the WM8915, configuring the system clock to 256fs for 48kHz when the device is active but reverting to using the input 32.768kHz clock directly at other times to support features such as jack detection with minimal power consumption. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 6ec44a354a6..1e51750fd3a 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -14,6 +14,33 @@ #include "../codecs/wm8915.h" +static int speyside_set_bias_level(struct snd_soc_card *card, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + int ret; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK1, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8915_FLL_MCLK1, + 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL\n"); + return ret; + } + + default: + break; + } + + return 0; +} + static int speyside_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -51,6 +78,13 @@ static struct snd_soc_ops speyside_ops = { .hw_params = speyside_hw_params, }; +static int speyside_wm8915_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *dai = rtd->codec_dai; + + return snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0); +} + static struct snd_soc_dai_link speyside_dai[] = { { .name = "CPU", @@ -59,6 +93,7 @@ static struct snd_soc_dai_link speyside_dai[] = { .codec_dai_name = "wm8915-aif1", .platform_name = "samsung-audio", .codec_name = "wm8915.1-001a", + .init = speyside_wm8915_init, .ops = &speyside_ops, }, }; @@ -91,6 +126,8 @@ static struct snd_soc_card speyside = { .dai_link = speyside_dai, .num_links = ARRAY_SIZE(speyside_dai), + .set_bias_level = speyside_set_bias_level, + .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, -- cgit v1.2.3 From ea3e98e75a6f38522450b66e22e34267977915ef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Apr 2011 23:42:25 -0700 Subject: ASoC: Support the sub speaker driver on Speyside Speyside includes a WM9081 configured as an external speaker driver taking an analogue input from HPOUT2 on the WM8915 on the system. Add support for this to the driver, using a prefix of "Sub" for the WM9081 controls to ensure we avoid collisions with controls on the WM8915. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/speyside.c | 34 ++++++++++++++++++++++++++++++++++ 2 files changed, 35 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 726af2ede79..459566bfcd3 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -176,3 +176,4 @@ config SND_SOC_SPEYSIDE depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S select SND_SOC_WM8915 + select SND_SOC_WM9081 diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 1e51750fd3a..0adaf7fe681 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -13,6 +13,7 @@ #include #include "../codecs/wm8915.h" +#include "../codecs/wm9081.h" static int speyside_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) @@ -98,6 +99,30 @@ static struct snd_soc_dai_link speyside_dai[] = { }, }; +static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) +{ + snd_soc_dapm_nc_pin(dapm, "LINEOUT"); + + /* At any time the WM9081 is active it will have this clock */ + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, + 48000 * 256, 0); +} + +static struct snd_soc_aux_dev speyside_aux_dev[] = { + { + .name = "wm9081", + .codec_name = "wm9081.1-006c", + .init = speyside_wm9081_init, + }, +}; + +static struct snd_soc_codec_conf speyside_codec_conf[] = { + { + .dev_name = "wm9081.1-006c", + .name_prefix = "Sub", + }, +}; + static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), @@ -118,6 +143,11 @@ static struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUT1L" }, { "Headphone", NULL, "HPOUT1R" }, + { "Sub IN1", NULL, "HPOUT2L" }, + { "Sub IN2", NULL, "HPOUT2R" }, + + { "Main Speaker", NULL, "Sub SPKN" }, + { "Main Speaker", NULL, "Sub SPKP" }, { "Main Speaker", NULL, "SPKDAT" }, }; @@ -125,6 +155,10 @@ static struct snd_soc_card speyside = { .name = "Speyside", .dai_link = speyside_dai, .num_links = ARRAY_SIZE(speyside_dai), + .aux_dev = speyside_aux_dev, + .num_aux_devs = ARRAY_SIZE(speyside_aux_dev), + .codec_conf = speyside_codec_conf, + .num_configs = ARRAY_SIZE(speyside_codec_conf), .set_bias_level = speyside_set_bias_level, -- cgit v1.2.3 From 68688e78ed4bc8d1a811ca29e2f8681561706db3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Apr 2011 00:00:36 -0700 Subject: ASoC: Add Speyside headset jack detection support Speyside makes use of support the WM8915 has for detecting the polarity of the microphone and ground connections on headsets, using a GPIO to control the polarity of the ground connection and switching between the two microphone bias supplies available on the device in order to do so. As a result of this the detection support is more involved than for most other CODECs, using a callback to configure the current polarity of the jack and translate this into the board-specific connections required for the current scenario. On Android some additional work is required to hook this up to the application layer as the Android HeadsetObserver monitors a custom drivers/switch API rather than the standard Linux APIs. This can be done by either updating HeadsetObserver or modifying the ALSA core to report via drivers/switch as well. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 78 +++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 77 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 0adaf7fe681..612a39e0e10 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -11,10 +11,14 @@ #include #include +#include +#include #include "../codecs/wm8915.h" #include "../codecs/wm9081.h" +#define WM8915_HPSEL_GPIO 214 + static int speyside_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { @@ -79,11 +83,74 @@ static struct snd_soc_ops speyside_ops = { .hw_params = speyside_hw_params, }; +static struct snd_soc_jack speyside_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin speyside_headset_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +/* Default the headphone selection to active high */ +static int speyside_jack_polarity; + +static int speyside_get_micbias(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + if (speyside_jack_polarity && (strcmp(source->name, "MICB1") == 0)) + return 1; + if (!speyside_jack_polarity && (strcmp(source->name, "MICB2") == 0)) + return 1; + + return 0; +} + +static void speyside_set_polarity(struct snd_soc_codec *codec, + int polarity) +{ + speyside_jack_polarity = !polarity; + gpio_direction_output(WM8915_HPSEL_GPIO, speyside_jack_polarity); + + /* Re-run DAPM to make sure we're using the correct mic bias */ + snd_soc_dapm_sync(&codec->dapm); +} + static int speyside_wm8915_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0); + if (ret < 0) + return ret; + + ret = gpio_request(WM8915_HPSEL_GPIO, "HP_SEL"); + if (ret != 0) + pr_err("Failed to request HP_SEL GPIO: %d\n", ret); + gpio_direction_output(WM8915_HPSEL_GPIO, speyside_jack_polarity); - return snd_soc_dai_set_sysclk(dai, WM8915_SYSCLK_MCLK1, 32768, 0); + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &speyside_headset); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&speyside_headset, + ARRAY_SIZE(speyside_headset_pins), + speyside_headset_pins); + if (ret) + return ret; + + wm8915_detect(codec, &speyside_headset, speyside_set_polarity); + + return 0; } static struct snd_soc_dai_link speyside_dai[] = { @@ -125,6 +192,7 @@ static struct snd_soc_codec_conf speyside_codec_conf[] = { static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Main Speaker", NULL), @@ -133,7 +201,15 @@ static struct snd_soc_dapm_widget widgets[] = { }; static struct snd_soc_dapm_route audio_paths[] = { + { "IN1RN", NULL, "MICB1" }, + { "IN1RP", NULL, "MICB1" }, + { "IN1RN", NULL, "MICB2" }, + { "IN1RP", NULL, "MICB2" }, + { "MICB1", NULL, "Headset Mic", speyside_get_micbias }, + { "MICB2", NULL, "Headset Mic", speyside_get_micbias }, + { "IN1LP", NULL, "MICB2" }, + { "IN1RN", NULL, "MICB1" }, { "MICB2", NULL, "Main AMIC" }, { "DMIC1DAT", NULL, "MICB1" }, -- cgit v1.2.3 From ea0e60de385a984ab54d6694b03119d1706f548e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Apr 2011 00:09:53 -0700 Subject: ASoC: Add pin switches for fixed analogue inputs and outputs on Speyside Pin switches enable direct control of the DAPM state from userspace, enabling simple enabling and disabling of the path. This is especially useful for outputs such as the speaker which are composed of several physical devices as it allows them to be controlled as a group. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 612a39e0e10..e171a321889 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -190,6 +190,12 @@ static struct snd_soc_codec_conf speyside_codec_conf[] = { }, }; +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("Main DMIC"), + SOC_DAPM_PIN_SWITCH("Main AMIC"), +}; + static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -238,6 +244,8 @@ static struct snd_soc_card speyside = { .set_bias_level = speyside_set_bias_level, + .controls = controls, + .num_controls = ARRAY_SIZE(controls), .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, -- cgit v1.2.3 From 556e4fb1d801b0f977b1522e14fb06e2f3059f4c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Apr 2011 14:15:10 -0700 Subject: ASoC: Add stub baseband link on Speyside Demonstrate the connection of a baseband to the system. We add a DAI for the link to the baseband. This will become visible to the application layer - audio should be started from the application layer using an application such as this: http://opensource.wolfsonmicro.com/~gg/bluetooth-pcm/bluetooth_pcm.c which starts up audio as for CPU based playback and record up to the point where data is streamed. Due to non-availability of baseband simulation hardware we reuse the configuration for the CPU link with the CODEC acting as clock master, allowing signals to be observed with a scope. A more standard system would have separate configuration for the baseband with its own ops structure and operations. Normally the baseband would be clock master as the baseband audio will be synchronised to the external telephony network. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index e171a321889..a0c14a87e23 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -164,6 +164,15 @@ static struct snd_soc_dai_link speyside_dai[] = { .init = speyside_wm8915_init, .ops = &speyside_ops, }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm8915-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .platform_name = "samsung-audio", + .ops = &speyside_ops, + }, }; static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) @@ -194,6 +203,8 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main Speaker"), SOC_DAPM_PIN_SWITCH("Main DMIC"), SOC_DAPM_PIN_SWITCH("Main AMIC"), + SOC_DAPM_PIN_SWITCH("WM1250 Input"), + SOC_DAPM_PIN_SWITCH("WM1250 Output"), }; static struct snd_soc_dapm_widget widgets[] = { -- cgit v1.2.3 From b7a5d14c607093de1a030e9933bc8c95ab2afd3f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Apr 2011 17:37:52 -0700 Subject: ASoC: Mark Speyside widgets as ignoring suspend Allow audio paths through the Speyside system to be kept active while the system is suspended (for example, when on a voice call) by marking all the external widgets and the DAI link to the WM1250-EV1 baseband module as ignoring suspend. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index a0c14a87e23..337855a3f2f 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -153,6 +153,19 @@ static int speyside_wm8915_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int speyside_late_probe(struct snd_soc_card *card) +{ + snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output"); + snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input"); + + return 0; +} + static struct snd_soc_dai_link speyside_dai[] = { { .name = "CPU", @@ -172,6 +185,7 @@ static struct snd_soc_dai_link speyside_dai[] = { .codec_name = "wm1250-ev1.1-0027", .platform_name = "samsung-audio", .ops = &speyside_ops, + .ignore_suspend = 1, }, }; @@ -261,6 +275,8 @@ static struct snd_soc_card speyside = { .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, .num_dapm_routes = ARRAY_SIZE(audio_paths), + + .late_probe = speyside_late_probe, }; static __devinit int speyside_probe(struct platform_device *pdev) -- cgit v1.2.3 From 13319699113a78e90625ece124666d3ee53033f6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 12 Apr 2011 19:31:03 +0200 Subject: ASoC: JZ4740: Convert qi_lb60 codec to table based DAPM setup Use the newly introduced dapm_widgets, dpam_routes and fields of the snd_soc_card struct to setup DAPM. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/jz4740/qi_lb60.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 49723e3e7e3..875abc94a00 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -70,12 +70,6 @@ static int qi_lb60_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_soc_dapm_new_controls(dapm, qi_lb60_widgets, - ARRAY_SIZE(qi_lb60_widgets)); - snd_soc_dapm_add_routes(dapm, qi_lb60_routes, - ARRAY_SIZE(qi_lb60_routes)); - snd_soc_dapm_sync(dapm); - return 0; } @@ -93,6 +87,11 @@ static struct snd_soc_card qi_lb60 = { .name = "QI LB60", .dai_link = &qi_lb60_dai, .num_links = 1, + + .dapm_widgets = qi_lb60_widgets, + .num_dapm_widgets = ARRAY_SIZE(qi_lb60_widgets), + .dapm_routes = qi_lb60_routes, + .num_dapm_routes = ARRAY_SIZE(qi_lb60_routes), }; static struct platform_device *qi_lb60_snd_device; -- cgit v1.2.3 From c6f0ede7c563497f6d7fef847a965d7fd63d86f9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 12 Apr 2011 19:31:04 +0200 Subject: ASoC: JZ4740: qi_lb60: Use gpio_request_array to request and setup gpios This patch changes the qi_lb60 setup code to use gpio_request_array instead of manually calling gpio_request and gpio_direction_output for each gpio. Doing so makes the code a bit more compact. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/jz4740/qi_lb60.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 875abc94a00..8c4e84bcc32 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -96,6 +96,11 @@ static struct snd_soc_card qi_lb60 = { static struct platform_device *qi_lb60_snd_device; +static const struct gpio qi_lb60_gpios[] = { + { QI_LB60_SND_GPIO, GPIOF_OUT_INIT_LOW, "SND" }, + { QI_LB60_AMP_GPIO, GPIOF_OUT_INIT_LOW, "AMP" }, +}; + static int __init qi_lb60_init(void) { int ret; @@ -105,23 +110,12 @@ static int __init qi_lb60_init(void) if (!qi_lb60_snd_device) return -ENOMEM; - ret = gpio_request(QI_LB60_SND_GPIO, "SND"); + ret = gpio_request_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); if (ret) { - pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n", - QI_LB60_SND_GPIO, ret); + pr_err("qi_lb60 snd: Failed to request gpios: %d\n", ret); goto err_device_put; } - ret = gpio_request(QI_LB60_AMP_GPIO, "AMP"); - if (ret) { - pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n", - QI_LB60_AMP_GPIO, ret); - goto err_gpio_free_snd; - } - - gpio_direction_output(QI_LB60_SND_GPIO, 0); - gpio_direction_output(QI_LB60_AMP_GPIO, 0); - platform_set_drvdata(qi_lb60_snd_device, &qi_lb60); ret = platform_device_add(qi_lb60_snd_device); @@ -134,10 +128,8 @@ static int __init qi_lb60_init(void) err_unset_pdata: platform_set_drvdata(qi_lb60_snd_device, NULL); -/*err_gpio_free_amp:*/ - gpio_free(QI_LB60_AMP_GPIO); -err_gpio_free_snd: - gpio_free(QI_LB60_SND_GPIO); +/*err_gpio_free_array:*/ + gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); err_device_put: platform_device_put(qi_lb60_snd_device); @@ -147,9 +139,8 @@ module_init(qi_lb60_init); static void __exit qi_lb60_exit(void) { - gpio_free(QI_LB60_AMP_GPIO); - gpio_free(QI_LB60_SND_GPIO); platform_device_unregister(qi_lb60_snd_device); + gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); } module_exit(qi_lb60_exit); -- cgit v1.2.3 From 621206b76825f24ccb650b9f59c18c17f6760a4c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 12 Apr 2011 19:31:05 +0200 Subject: ASoC: JZ4740: qi_lb60: Use the SND_SOC_DAPM_EVENT_OFF for the speakers status Use SND_SOC_DAPM_EVENT_OFF for determining whether the speaker should be turned on or off instead of open coding it. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/jz4740/qi_lb60.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 8c4e84bcc32..c5fc339f68f 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -27,11 +27,7 @@ static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *ctrl, int event) { - int on = 0; - if (event & SND_SOC_DAPM_POST_PMU) - on = 1; - else if (event & SND_SOC_DAPM_PRE_PMD) - on = 0; + int on = !SND_SOC_DAPM_EVENT_OFF(event); gpio_set_value(QI_LB60_SND_GPIO, on); gpio_set_value(QI_LB60_AMP_GPIO, on); -- cgit v1.2.3 From 674479124f8cbc4b5507d914ccd63ad08fd5326e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 12 Apr 2011 19:33:29 +0200 Subject: ASoC: codecs: JZ4740: Convert to table based controls and DAPM setup Use the newly introduced dapm_widgets, dpam_routes and controls fields of the snd_soc_dai_driver struct to setup controls and DAPM. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/jz4740.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index f5ccdbf7ebc..e373f8f0690 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -294,20 +294,9 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) { - struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); - snd_soc_add_controls(codec, jz4740_codec_controls, - ARRAY_SIZE(jz4740_codec_controls)); - - snd_soc_dapm_new_controls(dapm, jz4740_codec_dapm_widgets, - ARRAY_SIZE(jz4740_codec_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes, - ARRAY_SIZE(jz4740_codec_dapm_routes)); - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -348,6 +337,13 @@ static struct snd_soc_codec_driver soc_codec_dev_jz4740_codec = { .reg_cache_default = jz4740_codec_regs, .reg_word_size = sizeof(u32), .reg_cache_size = 2, + + .controls = jz4740_codec_controls, + .num_controls = ARRAY_SIZE(jz4740_codec_controls), + .dapm_widgets = jz4740_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(jz4740_codec_dapm_widgets), + .dapm_routes = jz4740_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(jz4740_codec_dapm_routes), }; static int __devinit jz4740_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From 9cdc352936311eea55624cbabafda296b99ff137 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 11 Apr 2011 17:56:32 +0200 Subject: ALSA: usb-audio: Add quirks for Audio Kontrol 6 This new device by Native Instruments is also compliant to the USB standard v2.0, but hides this detail at when connected. It needs the same boot quirks than other models, and also has two non-class-compliant mixer controls. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 17 +++++++++++++++++ sound/usb/quirks-table.h | 6 ++++++ sound/usb/quirks.c | 1 + 3 files changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 73dcc8256bc..4a7ad7ed62f 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -398,6 +398,17 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, return 0; } +static struct snd_kcontrol_new snd_nativeinstruments_ak6_mixers[] = { + { + .name = "Direct Monitor Channel 1+2", + .private_value = _MAKE_NI_CONTROL(0x03, 0x03), + }, + { + .name = "Direct Monitor Channel 3+4", + .private_value = _MAKE_NI_CONTROL(0x03, 0x05), + }, +}; + static struct snd_kcontrol_new snd_nativeinstruments_ta6_mixers[] = { { .name = "Direct Thru Channel A", @@ -526,6 +537,12 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_xonar_u1_controls_create(mixer); break; + case USB_ID(0x17cc, 0x1001): /* Audio Kontrol 6 */ + err = snd_nativeinstruments_create_mixer(mixer, + snd_nativeinstruments_ak6_mixers, + ARRAY_SIZE(snd_nativeinstruments_ak6_mixers)); + break; + case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c66d3f64dcf..54e18c181a1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2331,6 +2331,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* Native Instruments MK2 series */ +{ + /* Audio Kontrol 6 */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x17cc, + .idProduct = 0x1000, +}, { /* Traktor Audio 6 */ .match_flags = USB_DEVICE_ID_MATCH_DEVICE, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 355759bad58..2452edd2f14 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -539,6 +539,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Access Music VirusTI Desktop */ return snd_usb_accessmusic_boot_quirk(dev); + case USB_ID(0x17cc, 0x1000): /* Audio Kontrol 6 */ case USB_ID(0x17cc, 0x1010): /* Traktor Audio 6 */ case USB_ID(0x17cc, 0x1020): /* Traktor Audio 10 */ return snd_usb_nativeinstruments_boot_quirk(dev); -- cgit v1.2.3 From 5b17b077eb069365ac9508dd0be6e09e0b604bd2 Mon Sep 17 00:00:00 2001 From: Andrew Morton Date: Thu, 14 Apr 2011 15:06:13 -0700 Subject: ALSA: hda - sound/pci/hda/hda_codec.c: fix warning sound/pci/hda/hda_codec.c: In function 'snd_hda_get_connections': sound/pci/hda/hda_codec.c:332: warning: unused variable 'j' Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 11ead159abd..2b601939048 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -329,7 +329,7 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns) { struct snd_array *array = &codec->conn_lists; - int i, j, len, old_used; + int i, len, old_used; hda_nid_t list[HDA_MAX_CONNECTIONS]; /* look up the cached results */ -- cgit v1.2.3 From 4651d5566840e911b14a5052f18ed39558677937 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 12 Apr 2011 11:28:59 -0600 Subject: ARM: Tegra: Rename harmony_audio.h -> tegra_wm8903_pdata.h The audio driver will soon support more than just the Tegra Harmony board. Rename the platform data header file and data type to reflect this. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 8585957477e..c3096c3b9e2 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -35,7 +35,7 @@ #include #include -#include +#include #include #include @@ -58,7 +58,7 @@ struct tegra_harmony { struct tegra_asoc_utils_data util_data; - struct harmony_audio_platform_data *pdata; + struct tegra_wm8903_platform_data *pdata; int gpio_requested; }; @@ -163,7 +163,7 @@ static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct snd_soc_card *card = codec->card; struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); - struct harmony_audio_platform_data *pdata = harmony->pdata; + struct tegra_wm8903_platform_data *pdata = harmony->pdata; gpio_set_value_cansleep(pdata->gpio_spkr_en, SND_SOC_DAPM_EVENT_ON(event)); @@ -198,7 +198,7 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = codec->card; struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); - struct harmony_audio_platform_data *pdata = harmony->pdata; + struct tegra_wm8903_platform_data *pdata = harmony->pdata; int ret; ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); @@ -291,7 +291,7 @@ static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev) { struct snd_soc_card *card = &snd_soc_harmony; struct tegra_harmony *harmony; - struct harmony_audio_platform_data *pdata; + struct tegra_wm8903_platform_data *pdata; int ret; if (!machine_is_harmony()) { @@ -344,7 +344,7 @@ static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); - struct harmony_audio_platform_data *pdata = harmony->pdata; + struct tegra_wm8903_platform_data *pdata = harmony->pdata; snd_soc_unregister_card(card); -- cgit v1.2.3 From 7b33af252fbbf3beb694448da3ba6687022fd602 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 12 Apr 2011 11:29:00 -0600 Subject: ASoC: Tegra: Rename pdev tegra-snd-harmony to tegra-snd-wm8903 Soon, this machine driver will be updated to handle a number of Tegra boards using the WM8903 codec. Rename the platform device in advance to reflect this. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index c3096c3b9e2..ae168f2446a 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -50,7 +50,7 @@ #include "tegra_pcm.h" #include "tegra_asoc_utils.h" -#define DRV_NAME "tegra-snd-harmony" +#define DRV_NAME "tegra-snd-wm8903" #define GPIO_SPKR_EN BIT(0) #define GPIO_INT_MIC_EN BIT(1) -- cgit v1.2.3 From dc0a50afa67c3dbd51211881b7568917dbbc6861 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 12 Apr 2011 11:40:36 -0600 Subject: ASoC: Tegra: Rename harmony.c to tegra_wm8903.c Soon, this machine driver will be updated to handle a number of Tegra boards using the WM8903 codec. Rename the file in advance to reflect this. Fix the content of tegra_wm8903.c to match the rename; replace references to Harmony board with something more generic. * s/struct tegra_harmony/struct tegra_wm8903/ * s/harmony/machine/ # variable name * Similar rename for some functions * Similar comment fix * Similar MODULE_DESCRIPTION fix Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 9 +- sound/soc/tegra/Makefile | 4 +- sound/soc/tegra/harmony.c | 394 ---------------------------------------- sound/soc/tegra/tegra_wm8903.c | 395 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 402 insertions(+), 400 deletions(-) delete mode 100644 sound/soc/tegra/harmony.c create mode 100644 sound/soc/tegra/tegra_wm8903.c (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 66b504f06c2..9e53fec52d6 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -14,13 +14,14 @@ config SND_TEGRA_SOC_I2S Tegra I2S interface. You will also need to select the individual machine drivers to support below. -config SND_TEGRA_SOC_HARMONY - tristate "SoC Audio support for Tegra Harmony reference board" +config SND_TEGRA_SOC_WM8903 + tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_TEGRA_SOC && MACH_HARMONY && I2C default m select SND_TEGRA_SOC_I2S select SND_SOC_WM8903 help - Say Y or M here if you want to add support for SoC audio on the - Tegra Harmony reference board. + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the WM8093 codec. Currently, the only supported board + is Harmony. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index fd183d3ab4f..c9f7ea43037 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -10,6 +10,6 @@ obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o # Tegra machine Support -snd-soc-tegra-harmony-objs := harmony.o +snd-soc-tegra-wm8903-objs := tegra_wm8903.o -obj-$(CONFIG_SND_TEGRA_SOC_HARMONY) += snd-soc-tegra-harmony.o +obj-$(CONFIG_SND_TEGRA_SOC_WM8903) += snd-soc-tegra-wm8903.o diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c deleted file mode 100644 index 6bd1e42d538..00000000000 --- a/sound/soc/tegra/harmony.c +++ /dev/null @@ -1,394 +0,0 @@ -/* - * harmony.c - Harmony machine ASoC driver - * - * Author: Stephen Warren - * Copyright (C) 2010-2011 - NVIDIA, Inc. - * - * Based on code copyright/by: - * - * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include - -#include -#include -#include -#include - -#include - -#include -#include -#include -#include -#include - -#include "../codecs/wm8903.h" - -#include "tegra_das.h" -#include "tegra_i2s.h" -#include "tegra_pcm.h" -#include "tegra_asoc_utils.h" - -#define DRV_NAME "tegra-snd-wm8903" - -#define GPIO_SPKR_EN BIT(0) -#define GPIO_INT_MIC_EN BIT(1) -#define GPIO_EXT_MIC_EN BIT(2) - -struct tegra_harmony { - struct tegra_asoc_utils_data util_data; - struct tegra_wm8903_platform_data *pdata; - int gpio_requested; -}; - -static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_card *card = codec->card; - struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); - int srate, mclk, mclk_change; - int err; - - srate = params_rate(params); - switch (srate) { - case 64000: - case 88200: - case 96000: - mclk = 128 * srate; - break; - default: - mclk = 256 * srate; - break; - } - /* FIXME: Codec only requires >= 3MHz if OSR==0 */ - while (mclk < 6000000) - mclk *= 2; - - err = tegra_asoc_utils_set_rate(&harmony->util_data, srate, mclk, - &mclk_change); - if (err < 0) { - dev_err(card->dev, "Can't configure clocks\n"); - return err; - } - - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "codec_dai fmt not set\n"); - return err; - } - - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "cpu_dai fmt not set\n"); - return err; - } - - if (mclk_change) { - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (err < 0) { - dev_err(card->dev, "codec_dai clock not set\n"); - return err; - } - } - - return 0; -} - -static struct snd_soc_ops harmony_asoc_ops = { - .hw_params = harmony_asoc_hw_params, -}; - -static struct snd_soc_jack harmony_hp_jack; - -static struct snd_soc_jack_pin harmony_hp_jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static struct snd_soc_jack_gpio harmony_hp_jack_gpios[] = { - { - .name = "headphone detect", - .report = SND_JACK_HEADPHONE, - .debounce_time = 150, - .invert = 1, - } -}; - -static struct snd_soc_jack harmony_mic_jack; - -static struct snd_soc_jack_pin harmony_mic_jack_pins[] = { - { - .pin = "Mic Jack", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct snd_soc_card *card = codec->card; - struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = harmony->pdata; - - gpio_set_value_cansleep(pdata->gpio_spkr_en, - SND_SOC_DAPM_EVENT_ON(event)); - - return 0; -} - -static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Int Spk", harmony_event_int_spk), - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route harmony_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "ROP"}, - {"Int Spk", NULL, "RON"}, - {"Int Spk", NULL, "LOP"}, - {"Int Spk", NULL, "LON"}, - {"Mic Bias", NULL, "Mic Jack"}, - {"IN1L", NULL, "Mic Bias"}, -}; - -static const struct snd_kcontrol_new harmony_controls[] = { - SOC_DAPM_PIN_SWITCH("Int Spk"), -}; - -static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct snd_soc_card *card = codec->card; - struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = harmony->pdata; - int ret; - - ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); - if (ret) { - dev_err(card->dev, "cannot get spkr_en gpio\n"); - return ret; - } - harmony->gpio_requested |= GPIO_SPKR_EN; - - gpio_direction_output(pdata->gpio_spkr_en, 0); - - ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get int_mic_en gpio\n"); - return ret; - } - harmony->gpio_requested |= GPIO_INT_MIC_EN; - - /* Disable int mic; enable signal is active-high */ - gpio_direction_output(pdata->gpio_int_mic_en, 0); - - ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get ext_mic_en gpio\n"); - return ret; - } - harmony->gpio_requested |= GPIO_EXT_MIC_EN; - - /* Enable ext mic; enable signal is active-low */ - gpio_direction_output(pdata->gpio_ext_mic_en, 0); - - ret = snd_soc_add_controls(codec, harmony_controls, - ARRAY_SIZE(harmony_controls)); - if (ret < 0) - return ret; - - snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets, - ARRAY_SIZE(harmony_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, harmony_audio_map, - ARRAY_SIZE(harmony_audio_map)); - - harmony_hp_jack_gpios[0].gpio = pdata->gpio_hp_det; - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &harmony_hp_jack); - snd_soc_jack_add_pins(&harmony_hp_jack, - ARRAY_SIZE(harmony_hp_jack_pins), - harmony_hp_jack_pins); - snd_soc_jack_add_gpios(&harmony_hp_jack, - ARRAY_SIZE(harmony_hp_jack_gpios), - harmony_hp_jack_gpios); - - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &harmony_mic_jack); - snd_soc_jack_add_pins(&harmony_mic_jack, - ARRAY_SIZE(harmony_mic_jack_pins), - harmony_mic_jack_pins); - wm8903_mic_detect(codec, &harmony_mic_jack, SND_JACK_MICROPHONE, 0); - - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); - - snd_soc_dapm_nc_pin(dapm, "IN3L"); - snd_soc_dapm_nc_pin(dapm, "IN3R"); - snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); - snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); - - snd_soc_dapm_sync(dapm); - - return 0; -} - -static struct snd_soc_dai_link harmony_wm8903_dai = { - .name = "WM8903", - .stream_name = "WM8903 PCM", - .codec_name = "wm8903.0-001a", - .platform_name = "tegra-pcm-audio", - .cpu_dai_name = "tegra-i2s.0", - .codec_dai_name = "wm8903-hifi", - .init = harmony_asoc_init, - .ops = &harmony_asoc_ops, -}; - -static struct snd_soc_card snd_soc_harmony = { - .name = "tegra-harmony", - .dai_link = &harmony_wm8903_dai, - .num_links = 1, -}; - -static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &snd_soc_harmony; - struct tegra_harmony *harmony; - struct tegra_wm8903_platform_data *pdata; - int ret; - - if (!machine_is_harmony()) { - dev_err(&pdev->dev, "Not running on Tegra Harmony!\n"); - return -ENODEV; - } - - pdata = pdev->dev.platform_data; - if (!pdata) { - dev_err(&pdev->dev, "no platform data supplied\n"); - return -EINVAL; - } - - harmony = kzalloc(sizeof(struct tegra_harmony), GFP_KERNEL); - if (!harmony) { - dev_err(&pdev->dev, "Can't allocate tegra_harmony\n"); - return -ENOMEM; - } - - harmony->pdata = pdata; - - ret = tegra_asoc_utils_init(&harmony->util_data, &pdev->dev); - if (ret) - goto err_free_harmony; - - card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); - snd_soc_card_set_drvdata(card, harmony); - - ret = snd_soc_register_card(card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - goto err_clear_drvdata; - } - - return 0; - -err_clear_drvdata: - snd_soc_card_set_drvdata(card, NULL); - platform_set_drvdata(pdev, NULL); - card->dev = NULL; - tegra_asoc_utils_fini(&harmony->util_data); -err_free_harmony: - kfree(harmony); - return ret; -} - -static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = harmony->pdata; - - snd_soc_unregister_card(card); - - snd_soc_card_set_drvdata(card, NULL); - platform_set_drvdata(pdev, NULL); - card->dev = NULL; - - tegra_asoc_utils_fini(&harmony->util_data); - - if (harmony->gpio_requested & GPIO_EXT_MIC_EN) - gpio_free(pdata->gpio_ext_mic_en); - if (harmony->gpio_requested & GPIO_INT_MIC_EN) - gpio_free(pdata->gpio_int_mic_en); - if (harmony->gpio_requested & GPIO_SPKR_EN) - gpio_free(pdata->gpio_spkr_en); - - kfree(harmony); - - return 0; -} - -static struct platform_driver tegra_snd_harmony_driver = { - .driver = { - .name = DRV_NAME, - .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, - }, - .probe = tegra_snd_harmony_probe, - .remove = __devexit_p(tegra_snd_harmony_remove), -}; - -static int __init snd_tegra_harmony_init(void) -{ - return platform_driver_register(&tegra_snd_harmony_driver); -} -module_init(snd_tegra_harmony_init); - -static void __exit snd_tegra_harmony_exit(void) -{ - platform_driver_unregister(&tegra_snd_harmony_driver); -} -module_exit(snd_tegra_harmony_exit); - -MODULE_AUTHOR("Stephen Warren "); -MODULE_DESCRIPTION("Harmony machine ASoC driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c new file mode 100644 index 00000000000..0e31925b9d5 --- /dev/null +++ b/sound/soc/tegra/tegra_wm8903.c @@ -0,0 +1,395 @@ +/* + * tegra_wm8903.c - Tegra machine ASoC driver for boards using WM8903 codec. + * + * Author: Stephen Warren + * Copyright (C) 2010-2011 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include + +#include +#include +#include +#include + +#include + +#include +#include +#include +#include +#include + +#include "../codecs/wm8903.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-wm8903" + +#define GPIO_SPKR_EN BIT(0) +#define GPIO_INT_MIC_EN BIT(1) +#define GPIO_EXT_MIC_EN BIT(2) + +struct tegra_wm8903 { + struct tegra_asoc_utils_data util_data; + struct tegra_wm8903_platform_data *pdata; + int gpio_requested; +}; + +static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, mclk_change; + int err; + + srate = params_rate(params); + switch (srate) { + case 64000: + case 88200: + case 96000: + mclk = 128 * srate; + break; + default: + mclk = 256 * srate; + break; + } + /* FIXME: Codec only requires >= 3MHz if OSR==0 */ + while (mclk < 6000000) + mclk *= 2; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk, + &mclk_change); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "codec_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "cpu_dai fmt not set\n"); + return err; + } + + if (mclk_change) { + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + } + + return 0; +} + +static struct snd_soc_ops tegra_wm8903_ops = { + .hw_params = tegra_wm8903_hw_params, +}; + +static struct snd_soc_jack tegra_wm8903_hp_jack; + +static struct snd_soc_jack_pin tegra_wm8903_hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_wm8903_hp_jack_gpios[] = { + { + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, + } +}; + +static struct snd_soc_jack tegra_wm8903_mic_jack; + +static struct snd_soc_jack_pin tegra_wm8903_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct snd_soc_card *card = codec->card; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + struct tegra_wm8903_platform_data *pdata = machine->pdata; + + gpio_set_value_cansleep(pdata->gpio_spkr_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", tegra_wm8903_event_int_spk), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route harmony_audio_map[] = { + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "HPOUTL"}, + {"Int Spk", NULL, "ROP"}, + {"Int Spk", NULL, "RON"}, + {"Int Spk", NULL, "LOP"}, + {"Int Spk", NULL, "LON"}, + {"Mic Bias", NULL, "Mic Jack"}, + {"IN1L", NULL, "Mic Bias"}, +}; + +static const struct snd_kcontrol_new tegra_wm8903_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + +static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = codec->card; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + struct tegra_wm8903_platform_data *pdata = machine->pdata; + int ret; + + ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); + if (ret) { + dev_err(card->dev, "cannot get spkr_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_SPKR_EN; + + gpio_direction_output(pdata->gpio_spkr_en, 0); + + ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get int_mic_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_INT_MIC_EN; + + /* Disable int mic; enable signal is active-high */ + gpio_direction_output(pdata->gpio_int_mic_en, 0); + + ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get ext_mic_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_EXT_MIC_EN; + + /* Enable ext mic; enable signal is active-low */ + gpio_direction_output(pdata->gpio_ext_mic_en, 0); + + ret = snd_soc_add_controls(codec, tegra_wm8903_controls, + ARRAY_SIZE(tegra_wm8903_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_new_controls(dapm, tegra_wm8903_dapm_widgets, + ARRAY_SIZE(tegra_wm8903_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, harmony_audio_map, + ARRAY_SIZE(harmony_audio_map)); + + tegra_wm8903_hp_jack_gpios[0].gpio = pdata->gpio_hp_det; + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_wm8903_hp_jack); + snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, + ARRAY_SIZE(tegra_wm8903_hp_jack_pins), + tegra_wm8903_hp_jack_pins); + snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, + ARRAY_SIZE(tegra_wm8903_hp_jack_gpios), + tegra_wm8903_hp_jack_gpios); + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_wm8903_mic_jack); + snd_soc_jack_add_pins(&tegra_wm8903_mic_jack, + ARRAY_SIZE(tegra_wm8903_mic_jack_pins), + tegra_wm8903_mic_jack_pins); + wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, + 0); + + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + + snd_soc_dapm_nc_pin(dapm, "IN3L"); + snd_soc_dapm_nc_pin(dapm, "IN3R"); + snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); + snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link tegra_wm8903_dai = { + .name = "WM8903", + .stream_name = "WM8903 PCM", + .codec_name = "wm8903.0-001a", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "wm8903-hifi", + .init = tegra_wm8903_init, + .ops = &tegra_wm8903_ops, +}; + +static struct snd_soc_card snd_soc_tegra_wm8903 = { + .name = "tegra-wm8903", + .dai_link = &tegra_wm8903_dai, + .num_links = 1, +}; + +static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_tegra_wm8903; + struct tegra_wm8903 *machine; + struct tegra_wm8903_platform_data *pdata; + int ret; + + if (!machine_is_harmony()) { + dev_err(&pdev->dev, "Not running on Tegra Harmony!\n"); + return -ENODEV; + } + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + machine = kzalloc(sizeof(struct tegra_wm8903), GFP_KERNEL); + if (!machine) { + dev_err(&pdev->dev, "Can't allocate tegra_wm8903 struct\n"); + return -ENOMEM; + } + + machine->pdata = pdata; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err_free_machine; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_clear_drvdata; + } + + return 0; + +err_clear_drvdata: + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + tegra_asoc_utils_fini(&machine->util_data); +err_free_machine: + kfree(machine); + return ret; +} + +static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + struct tegra_wm8903_platform_data *pdata = machine->pdata; + + snd_soc_unregister_card(card); + + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + + tegra_asoc_utils_fini(&machine->util_data); + + if (machine->gpio_requested & GPIO_EXT_MIC_EN) + gpio_free(pdata->gpio_ext_mic_en); + if (machine->gpio_requested & GPIO_INT_MIC_EN) + gpio_free(pdata->gpio_int_mic_en); + if (machine->gpio_requested & GPIO_SPKR_EN) + gpio_free(pdata->gpio_spkr_en); + + kfree(machine); + + return 0; +} + +static struct platform_driver tegra_wm8903_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = tegra_wm8903_driver_probe, + .remove = __devexit_p(tegra_wm8903_driver_remove), +}; + +static int __init tegra_wm8903_modinit(void) +{ + return platform_driver_register(&tegra_wm8903_driver); +} +module_init(tegra_wm8903_modinit); + +static void __exit tegra_wm8903_modexit(void) +{ + platform_driver_unregister(&tegra_wm8903_driver); +} +module_exit(tegra_wm8903_modexit); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra+WM8903 machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.3 From 2ba9471b34f48eab9f6e097ef305746b33e12f85 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 12 Apr 2011 11:40:37 -0600 Subject: ASoC: Tegra: Rename Kconfig SND_TEGRA_SOC_* to SND_SOC_TEGRA_* The previous commit renames SND_TEGRA_SOC_HARMONY to SND_TEGRA_SOC_WM8903. While we're breaking people's .config files, rename all Tegra/SOC-related Kconfig variables to be more consistent with at least the core codec variables. Note that there exist machines that name their variables both ways. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 12 ++++++------ sound/soc/tegra/Makefile | 10 +++++----- 2 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 9e53fec52d6..21125b30263 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,24 +1,24 @@ -config SND_TEGRA_SOC +config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA default m help Say Y or M here if you want support for SoC audio on Tegra. -config SND_TEGRA_SOC_I2S +config SND_SOC_TEGRA_I2S tristate - depends on SND_TEGRA_SOC + depends on SND_SOC_TEGRA default m help Say Y or M if you want to add support for codecs attached to the Tegra I2S interface. You will also need to select the individual machine drivers to support below. -config SND_TEGRA_SOC_WM8903 +config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_TEGRA_SOC && MACH_HARMONY && I2C + depends on SND_SOC_TEGRA && MACH_HARMONY && I2C default m - select SND_TEGRA_SOC_I2S + select SND_SOC_TEGRA_I2S select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index c9f7ea43037..13bef8d572c 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -4,12 +4,12 @@ snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-i2s-objs := tegra_i2s.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o -obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-utils.o -obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-das.o -obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o -obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o +obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o +obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o +obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o -obj-$(CONFIG_SND_TEGRA_SOC_WM8903) += snd-soc-tegra-wm8903.o +obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o -- cgit v1.2.3 From 3eb25f998d3aede5f0011ba236e7586351e450bf Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 12 Apr 2011 11:40:38 -0600 Subject: ASoC: Tegra: Don't store snd_soc_jack_gpio in an array Storing the struct in an array makes the assignments to the GPIO member a little non-obvious, and is pointless when there's only a single GPIO. (I thought I fixed this during the review cycle when first submitting this driver, but I guess I overlooked that) Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0e31925b9d5..2b3844322c4 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -139,13 +139,11 @@ static struct snd_soc_jack_pin tegra_wm8903_hp_jack_pins[] = { }, }; -static struct snd_soc_jack_gpio tegra_wm8903_hp_jack_gpios[] = { - { - .name = "headphone detect", - .report = SND_JACK_HEADPHONE, - .debounce_time = 150, - .invert = 1, - } +static struct snd_soc_jack_gpio tegra_wm8903_hp_jack_gpio = { + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, }; static struct snd_soc_jack tegra_wm8903_mic_jack; @@ -241,15 +239,15 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, harmony_audio_map, ARRAY_SIZE(harmony_audio_map)); - tegra_wm8903_hp_jack_gpios[0].gpio = pdata->gpio_hp_det; + tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack); snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, ARRAY_SIZE(tegra_wm8903_hp_jack_pins), tegra_wm8903_hp_jack_pins); snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, - ARRAY_SIZE(tegra_wm8903_hp_jack_gpios), - tegra_wm8903_hp_jack_gpios); + 1, + &tegra_wm8903_hp_jack_gpio); snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, &tegra_wm8903_mic_jack); -- cgit v1.2.3 From 773b1d3d31bbf7257c48f6257b4ab06bcf4f5dfa Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 12 Apr 2011 11:40:39 -0600 Subject: ASoC: Tegra: Support more boards * Ventana is identical to Harmony. * Seaboard, Kaen, and Aebl are all pretty similar, mainly with slightly different sets of GPIOs, and slightly different WM8903 pin connectivity. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 7 +- sound/soc/tegra/tegra_wm8903.c | 180 ++++++++++++++++++++++++++++++++--------- 2 files changed, 145 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 21125b30263..14f711977ae 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -16,12 +16,13 @@ config SND_SOC_TEGRA_I2S config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_SOC_TEGRA && MACH_HARMONY && I2C + depends on SND_SOC_TEGRA && I2C + depends on MACH_HARMONY || MACH_VENTANA || MACH_SEABOARD || MACH_KAEN || MACH_AEBL default m select SND_SOC_TEGRA_I2S select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra - boards using the WM8093 codec. Currently, the only supported board - is Harmony. + boards using the WM8093 codec. Currently, the supported boards are + Harmony, Ventana, Seaboard, Kaen, and Aebl. diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2b3844322c4..37f6010a910 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -53,8 +53,9 @@ #define DRV_NAME "tegra-snd-wm8903" #define GPIO_SPKR_EN BIT(0) -#define GPIO_INT_MIC_EN BIT(1) -#define GPIO_EXT_MIC_EN BIT(2) +#define GPIO_HP_MUTE BIT(1) +#define GPIO_INT_MIC_EN BIT(2) +#define GPIO_EXT_MIC_EN BIT(3) struct tegra_wm8903 { struct tegra_asoc_utils_data util_data; @@ -163,15 +164,35 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = machine->pdata; + if (!(machine->gpio_requested & GPIO_SPKR_EN)) + return 0; + gpio_set_value_cansleep(pdata->gpio_spkr_en, SND_SOC_DAPM_EVENT_ON(event)); return 0; } +static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct snd_soc_card *card = codec->card; + struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + struct tegra_wm8903_platform_data *pdata = machine->pdata; + + if (!(machine->gpio_requested & GPIO_HP_MUTE)) + return 0; + + gpio_set_value_cansleep(pdata->gpio_hp_mute, + !SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", tegra_wm8903_event_int_spk), - SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", tegra_wm8903_event_hp), SND_SOC_DAPM_MIC("Mic Jack", NULL), }; @@ -186,6 +207,37 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = { {"IN1L", NULL, "Mic Bias"}, }; +static const struct snd_soc_dapm_route seaboard_audio_map[] = { + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "HPOUTL"}, + {"Int Spk", NULL, "ROP"}, + {"Int Spk", NULL, "RON"}, + {"Int Spk", NULL, "LOP"}, + {"Int Spk", NULL, "LON"}, + {"Mic Bias", NULL, "Mic Jack"}, + {"IN1R", NULL, "Mic Bias"}, +}; + +static const struct snd_soc_dapm_route kaen_audio_map[] = { + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "HPOUTL"}, + {"Int Spk", NULL, "ROP"}, + {"Int Spk", NULL, "RON"}, + {"Int Spk", NULL, "LOP"}, + {"Int Spk", NULL, "LON"}, + {"Mic Bias", NULL, "Mic Jack"}, + {"IN2R", NULL, "Mic Bias"}, +}; + +static const struct snd_soc_dapm_route aebl_audio_map[] = { + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "HPOUTL"}, + {"Int Spk", NULL, "LINEOUTR"}, + {"Int Spk", NULL, "LINEOUTL"}, + {"Mic Bias", NULL, "Mic Jack"}, + {"IN1R", NULL, "Mic Bias"}, +}; + static const struct snd_kcontrol_new tegra_wm8903_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), }; @@ -199,34 +251,51 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct tegra_wm8903_platform_data *pdata = machine->pdata; int ret; - ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); - if (ret) { - dev_err(card->dev, "cannot get spkr_en gpio\n"); - return ret; + if (gpio_is_valid(pdata->gpio_spkr_en)) { + ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); + if (ret) { + dev_err(card->dev, "cannot get spkr_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_SPKR_EN; + + gpio_direction_output(pdata->gpio_spkr_en, 0); } - machine->gpio_requested |= GPIO_SPKR_EN; - gpio_direction_output(pdata->gpio_spkr_en, 0); + if (gpio_is_valid(pdata->gpio_hp_mute)) { + ret = gpio_request(pdata->gpio_hp_mute, "hp_mute"); + if (ret) { + dev_err(card->dev, "cannot get hp_mute gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_HP_MUTE; - ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get int_mic_en gpio\n"); - return ret; + gpio_direction_output(pdata->gpio_hp_mute, 0); } - machine->gpio_requested |= GPIO_INT_MIC_EN; - /* Disable int mic; enable signal is active-high */ - gpio_direction_output(pdata->gpio_int_mic_en, 0); + if (gpio_is_valid(pdata->gpio_int_mic_en)) { + ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get int_mic_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_INT_MIC_EN; - ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get ext_mic_en gpio\n"); - return ret; + /* Disable int mic; enable signal is active-high */ + gpio_direction_output(pdata->gpio_int_mic_en, 0); } - machine->gpio_requested |= GPIO_EXT_MIC_EN; - /* Enable ext mic; enable signal is active-low */ - gpio_direction_output(pdata->gpio_ext_mic_en, 0); + if (gpio_is_valid(pdata->gpio_ext_mic_en)) { + ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get ext_mic_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_EXT_MIC_EN; + + /* Enable ext mic; enable signal is active-low */ + gpio_direction_output(pdata->gpio_ext_mic_en, 0); + } ret = snd_soc_add_controls(codec, tegra_wm8903_controls, ARRAY_SIZE(tegra_wm8903_controls)); @@ -236,18 +305,31 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_new_controls(dapm, tegra_wm8903_dapm_widgets, ARRAY_SIZE(tegra_wm8903_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, harmony_audio_map, - ARRAY_SIZE(harmony_audio_map)); + if (machine_is_harmony() || machine_is_ventana()) { + snd_soc_dapm_add_routes(dapm, harmony_audio_map, + ARRAY_SIZE(harmony_audio_map)); + } else if (machine_is_seaboard()) { + snd_soc_dapm_add_routes(dapm, seaboard_audio_map, + ARRAY_SIZE(seaboard_audio_map)); + } else if (machine_is_kaen()) { + snd_soc_dapm_add_routes(dapm, kaen_audio_map, + ARRAY_SIZE(kaen_audio_map)); + } else { + snd_soc_dapm_add_routes(dapm, aebl_audio_map, + ARRAY_SIZE(aebl_audio_map)); + } - tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_wm8903_hp_jack); - snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, - ARRAY_SIZE(tegra_wm8903_hp_jack_pins), - tegra_wm8903_hp_jack_pins); - snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, - 1, - &tegra_wm8903_hp_jack_gpio); + if (gpio_is_valid(pdata->gpio_hp_det)) { + tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_wm8903_hp_jack); + snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, + ARRAY_SIZE(tegra_wm8903_hp_jack_pins), + tegra_wm8903_hp_jack_pins); + snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, + 1, + &tegra_wm8903_hp_jack_gpio); + } snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, &tegra_wm8903_mic_jack); @@ -259,10 +341,26 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + /* FIXME: Calculate automatically based on DAPM routes? */ + if (!machine_is_harmony() && !machine_is_ventana()) + snd_soc_dapm_nc_pin(dapm, "IN1L"); + if (!machine_is_seaboard() && !machine_is_aebl()) + snd_soc_dapm_nc_pin(dapm, "IN1R"); + snd_soc_dapm_nc_pin(dapm, "IN2L"); + if (!machine_is_kaen()) + snd_soc_dapm_nc_pin(dapm, "IN2R"); snd_soc_dapm_nc_pin(dapm, "IN3L"); snd_soc_dapm_nc_pin(dapm, "IN3R"); - snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); - snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); + + if (machine_is_aebl()) { + snd_soc_dapm_nc_pin(dapm, "LON"); + snd_soc_dapm_nc_pin(dapm, "RON"); + snd_soc_dapm_nc_pin(dapm, "ROP"); + snd_soc_dapm_nc_pin(dapm, "LOP"); + } else { + snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); + snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); + } snd_soc_dapm_sync(dapm); @@ -293,14 +391,16 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) struct tegra_wm8903_platform_data *pdata; int ret; - if (!machine_is_harmony()) { - dev_err(&pdev->dev, "Not running on Tegra Harmony!\n"); + if (!machine_is_harmony() && !machine_is_ventana() && + !machine_is_seaboard() && !machine_is_kaen() && + !machine_is_aebl()) { + dev_err(&pdev->dev, "Not running on a supported board!\n"); return -ENODEV; } pdata = pdev->dev.platform_data; if (!pdata) { - dev_err(&pdev->dev, "no platform data supplied\n"); + dev_err(&pdev->dev, "No platform data supplied\n"); return -EINVAL; } @@ -357,6 +457,8 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) gpio_free(pdata->gpio_ext_mic_en); if (machine->gpio_requested & GPIO_INT_MIC_EN) gpio_free(pdata->gpio_int_mic_en); + if (machine->gpio_requested & GPIO_HP_MUTE) + gpio_free(pdata->gpio_hp_mute); if (machine->gpio_requested & GPIO_SPKR_EN) gpio_free(pdata->gpio_spkr_en); -- cgit v1.2.3 From b6a48404088d91514b9e78c3237976a5aa87cb14 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Fri, 15 Apr 2011 11:42:42 +0800 Subject: ALSA: emu10k1 - Remove "Front" controls only for STAC9758/59 Remove "Front Playback Volume" and "Front Playback Switch" from emu10k1 only for STAC9758/59 Since commit 7eae36fbd5ea9db3d3fe0d671199121be782a5b3 "Fix the confliction of 'Front' control", the "Front Playback Volume" control created by commit edf8e4565c44bffbb4d09e8984df941d0ae9e6e8 "emu10k1: Front channels via fxbus 8 and 9" was removed "Front Playback Volume" and "Surround Playback Volume" have same dB range since I2S DAC of SB Live! and SB Live! Platinum does not has any hardware volume control. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emumixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 1b50a23232b..9d890a5aec5 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1729,8 +1729,6 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Master Mono Playback Volume", "PCM Out Path & Mute", "Mono Output Select", - "Front Playback Switch", - "Front Playback Volume", "Surround Playback Switch", "Surround Playback Volume", "Center Playback Switch", @@ -1879,6 +1877,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, emu->rear_ac97 = 1; snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE|AC97SLOT_REAR_LEFT|AC97SLOT_REAR_RIGHT); snd_ac97_write_cache(emu->ac97, AC97_HEADPHONE, 0x0202); + remove_ctl(card,"Front Playback Volume"); + remove_ctl(card,"Front Playback Switch"); } /* remove unused AC97 controls */ snd_ac97_write_cache(emu->ac97, AC97_SURROUND_MASTER, 0x0202); -- cgit v1.2.3 From d9e3c4cc6801b03bce6dd31ba0e1a6acff8f8a1a Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 19 Apr 2011 15:25:07 -0600 Subject: ASoC: Tegra: wm8903 probe: Don't call machine_is_*() This machine driver is a platform driver, and hence will only be instantiated on the correct machines. Hence, there is no need to check the current machine during probe. (Applying Mark's TrimSlice review comments to the existing driver) Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 37f6010a910..1eb0632db48 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -391,13 +391,6 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) struct tegra_wm8903_platform_data *pdata; int ret; - if (!machine_is_harmony() && !machine_is_ventana() && - !machine_is_seaboard() && !machine_is_kaen() && - !machine_is_aebl()) { - dev_err(&pdev->dev, "Not running on a supported board!\n"); - return -ENODEV; - } - pdata = pdev->dev.platform_data; if (!pdata) { dev_err(&pdev->dev, "No platform data supplied\n"); -- cgit v1.2.3 From acb8303f15768c72796f3f95bb32a955333e0fbc Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 19 Apr 2011 15:25:08 -0600 Subject: ASoC: Tegra: wm8903: Remove redundant drvdata clears When the driver is not initialized/registered, nothing should be touching these fields anyway, so there's no point clearing them out. (Applying Mark's TrimSlice review comments to the existing driver) Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 1eb0632db48..b12b1fd1cb6 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -417,15 +417,12 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto err_clear_drvdata; + goto err_fini_utils; } return 0; -err_clear_drvdata: - snd_soc_card_set_drvdata(card, NULL); - platform_set_drvdata(pdev, NULL); - card->dev = NULL; +err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); err_free_machine: kfree(machine); @@ -440,10 +437,6 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) snd_soc_unregister_card(card); - snd_soc_card_set_drvdata(card, NULL); - platform_set_drvdata(pdev, NULL); - card->dev = NULL; - tegra_asoc_utils_fini(&machine->util_data); if (machine->gpio_requested & GPIO_EXT_MIC_EN) -- cgit v1.2.3 From 075413966a6ea389f78f4cc2e957708c1d6db8c5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 19 Apr 2011 15:25:09 -0600 Subject: ASoC: Tegra: Don't return mclk_changed from utils_set_rate Only the clock programming code needs to know whether the clocks changed, and that is encapsulated within tegra_asoc_utils_set_rate(). The machine driver's call to snd_soc_dai_set_sysclk(codec_dai, ...) is safe irrespective of whether the clocks changed. (Applying Mark's TrimSlice review comments to the existing driver) Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 9 +++++---- sound/soc/tegra/tegra_asoc_utils.h | 2 +- sound/soc/tegra/tegra_wm8903.c | 17 +++++++---------- 3 files changed, 13 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 52f0a3f9ce4..dfa85cbb05c 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -28,9 +28,10 @@ #include "tegra_asoc_utils.h" int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, - int mclk, int *mclk_change) + int mclk) { int new_baseclock; + bool clk_change; int err; switch (srate) { @@ -52,10 +53,10 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return -EINVAL; } - *mclk_change = ((new_baseclock != data->set_baseclock) || + clk_change = ((new_baseclock != data->set_baseclock) || (mclk != data->set_mclk)); - if (!*mclk_change) - return 0; + if (!clk_change) + return 0; data->set_baseclock = 0; data->set_mclk = 0; diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index bbba7afdfc2..4818195da25 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -36,7 +36,7 @@ struct tegra_asoc_utils_data { }; int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, - int mclk, int *mclk_change); + int mclk); int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev); void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index b12b1fd1cb6..988ff504dd5 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -72,7 +72,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - int srate, mclk, mclk_change; + int srate, mclk; int err; srate = params_rate(params); @@ -90,8 +90,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, while (mclk < 6000000) mclk *= 2; - err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk, - &mclk_change); + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); if (err < 0) { dev_err(card->dev, "Can't configure clocks\n"); return err; @@ -115,13 +114,11 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, return err; } - if (mclk_change) { - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (err < 0) { - dev_err(card->dev, "codec_dai clock not set\n"); - return err; - } + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; } return 0; -- cgit v1.2.3 From a32955dba2e2629bddacf7c1b9de3bf1b7c56acf Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 19 Apr 2011 15:25:10 -0600 Subject: ASoC: Tegra: Retrieve card from DAPM context not codec Card widgets are created in the card's DAPM context, not any codec's DAPM context. Hence, w->codec==NULL. Instead, find the card from the widget through the DAPM context of the widget, not the codec of the widget. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 988ff504dd5..9a439fb1244 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -156,8 +156,8 @@ static struct snd_soc_jack_pin tegra_wm8903_mic_jack_pins[] = { static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - struct snd_soc_codec *codec = w->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = machine->pdata; @@ -173,8 +173,8 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - struct snd_soc_codec *codec = w->codec; - struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = machine->pdata; -- cgit v1.2.3 From a68b38ada57bc9aeeec574d3e76e67ad89fea303 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 19 Apr 2011 15:25:11 -0600 Subject: ASoC: snd_soc_dapm_get_pin_status: Match other contexts too Not all widgets on a card are within the codec's DAPM context. Fix snd_soc_dapm_get_pin_status to search all contexts when looking for a widget. This change is required when modifying tegra_wm8903 to use snd_soc_card.widgets rather than calling snd_soc_dapm_new_controls; the former adds the widgets to the card's DAPM context, whereas tegra_wm8903 uses the codec's DAPM context when calling snd_soc_dapm_new_controls. By code inspection, I suspect this also applies to Samsung Speyside. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2ee738c08ca..4c868f9087f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2403,6 +2403,12 @@ int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm, return w->connected; } + /* Try again in other contexts */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strcmp(w->name, pin)) + return w->connected; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); -- cgit v1.2.3 From dc2bea616a4026860a8ba9eae778cfd4fda061b1 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 20 Apr 2011 16:00:36 +0800 Subject: ASoC: fix a simple coding style issue Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4c868f9087f..856471703bd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1597,7 +1597,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, } /* connect dynamic paths */ - switch(wsink->id) { + switch (wsink->id) { case snd_soc_dapm_adc: case snd_soc_dapm_dac: case snd_soc_dapm_pga: -- cgit v1.2.3 From f9861e17bd078f0a8c234157ddade03572415f8f Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 20 Apr 2011 16:00:42 +0800 Subject: ASoC: remove unused comment `type` parameter is not longer used in `snd_soc_codec_set_cache_io`, so remove this line. Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f46a198a48c..a217db25670 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -484,7 +484,6 @@ static struct { * snd_soc_codec_set_cache_io: Set up standard I/O functions. * * @codec: CODEC to configure. - * @type: Type of cache. * @addr_bits: Number of bits of register address data. * @data_bits: Number of bits of data per register. * @control: Control bus used. -- cgit v1.2.3 From a739362362982f3d8177df0621b68cb9156b1b60 Mon Sep 17 00:00:00 2001 From: Lu Guanqun Date: Wed, 20 Apr 2011 16:00:51 +0800 Subject: ASoC: fix two ident style problems Signed-off-by: Lu Guanqun Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3b3a377d087..a6f37d45422 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3475,7 +3475,7 @@ int snd_soc_register_dai(struct device *dev, dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); if (dai == NULL) - return -ENOMEM; + return -ENOMEM; /* create DAI component name */ dai->name = fmt_single_name(dev, &dai->id); @@ -3614,7 +3614,7 @@ int snd_soc_register_platform(struct device *dev, platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL); if (platform == NULL) - return -ENOMEM; + return -ENOMEM; /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); -- cgit v1.2.3 From dea8b6eef03afdec475b981fca8622c41f8de7e2 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 19 Apr 2011 15:25:12 -0600 Subject: ASoC: Tegra: wm8903: s/code/data/ for control/widget/maps Replace calls to a variety of registration functions by updating struct snd_soc_card snd_soc_tegra_wm8903 to directly point at the various control/widget/map tables instead. The ASoC core now performs any required registration based on these data fields. (Applying Mark's TrimSlice review comments to the existing driver) Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 41 +++++++++++++++++++---------------------- 1 file changed, 19 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 9a439fb1244..0d6738a8b29 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -294,28 +294,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) gpio_direction_output(pdata->gpio_ext_mic_en, 0); } - ret = snd_soc_add_controls(codec, tegra_wm8903_controls, - ARRAY_SIZE(tegra_wm8903_controls)); - if (ret < 0) - return ret; - - snd_soc_dapm_new_controls(dapm, tegra_wm8903_dapm_widgets, - ARRAY_SIZE(tegra_wm8903_dapm_widgets)); - - if (machine_is_harmony() || machine_is_ventana()) { - snd_soc_dapm_add_routes(dapm, harmony_audio_map, - ARRAY_SIZE(harmony_audio_map)); - } else if (machine_is_seaboard()) { - snd_soc_dapm_add_routes(dapm, seaboard_audio_map, - ARRAY_SIZE(seaboard_audio_map)); - } else if (machine_is_kaen()) { - snd_soc_dapm_add_routes(dapm, kaen_audio_map, - ARRAY_SIZE(kaen_audio_map)); - } else { - snd_soc_dapm_add_routes(dapm, aebl_audio_map, - ARRAY_SIZE(aebl_audio_map)); - } - if (gpio_is_valid(pdata->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, @@ -379,6 +357,11 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .name = "tegra-wm8903", .dai_link = &tegra_wm8903_dai, .num_links = 1, + + .controls = tegra_wm8903_controls, + .num_controls = ARRAY_SIZE(tegra_wm8903_controls), + .dapm_widgets = tegra_wm8903_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_wm8903_dapm_widgets), }; static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) @@ -410,6 +393,20 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); + if (machine_is_harmony() || machine_is_ventana()) { + card->dapm_routes = harmony_audio_map; + card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); + } else if (machine_is_seaboard()) { + card->dapm_routes = seaboard_audio_map; + card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map); + } else if (machine_is_kaen()) { + card->dapm_routes = kaen_audio_map; + card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map); + } else { + card->dapm_routes = aebl_audio_map; + card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map); + } + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", -- cgit v1.2.3 From dad31ec133adb20c8fd10bfd9379da3f08b8721e Mon Sep 17 00:00:00 2001 From: Peter Hsiang Date: Tue, 19 Apr 2011 18:20:40 -0700 Subject: ASoC: Add EQ and filter to max98095 CODEC driver This patch adds the equalizer and biquad filter controls. Signed-off-by: Peter Hsiang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 391 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98095.h | 15 ++ 2 files changed, 406 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 9c77f17a6af..a6cc94e1750 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -34,6 +34,8 @@ enum max98095_type { struct max98095_cdata { unsigned int rate; unsigned int fmt; + int eq_sel; + int bq_sel; }; struct max98095_priv { @@ -42,6 +44,12 @@ struct max98095_priv { struct max98095_pdata *pdata; unsigned int sysclk; struct max98095_cdata dai[3]; + const char **eq_texts; + const char **bq_texts; + struct soc_enum eq_enum; + struct soc_enum bq_enum; + int eq_textcnt; + int bq_textcnt; u8 lin_state; unsigned int mic1pre; unsigned int mic2pre; @@ -602,6 +610,74 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) return 0; } +/* + * Filter coefficients are in a separate register segment + * and they share the address space of the normal registers. + * The coefficient registers do not need or share the cache. + */ +static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + data[0] = reg; + data[1] = value; + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +/* + * Load equalizer DSP coefficient configurations registers + */ +static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, + unsigned int band, u16 *coefs) +{ + unsigned int eq_reg; + unsigned int i; + + BUG_ON(band > 4); + BUG_ON(dai > 1); + + /* Load the base register address */ + eq_reg = dai ? M98095_142_DAI2_EQ_BASE : M98095_110_DAI1_EQ_BASE; + + /* Add the band address offset, note adjustment for word address */ + eq_reg += band * (M98095_COEFS_PER_BAND << 1); + + /* Step through the registers and coefs */ + for (i = 0; i < M98095_COEFS_PER_BAND; i++) { + max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + } +} + +/* + * Load biquad filter coefficient configurations registers + */ +static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, + unsigned int band, u16 *coefs) +{ + unsigned int bq_reg; + unsigned int i; + + BUG_ON(band > 1); + BUG_ON(dai > 1); + + /* Load the base register address */ + bq_reg = dai ? M98095_17E_DAI2_BQ_BASE : M98095_174_DAI1_BQ_BASE; + + /* Add the band address offset, note adjustment for word address */ + bq_reg += band * (M98095_COEFS_PER_BAND << 1); + + /* Step through the registers and coefs */ + for (i = 0; i < M98095_COEFS_PER_BAND; i++) { + max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + } +} + static const char * const max98095_fltr_mode[] = { "Voice", "Music" }; static const struct soc_enum max98095_dai1_filter_mode_enum[] = { SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode), @@ -792,6 +868,12 @@ static const struct snd_kcontrol_new max98095_snd_controls[] = { SOC_SINGLE_TLV("ADCR Boost Volume", M98095_05E_LVL_ADC_R, 4, 3, 0, max98095_adcboost_tlv), + SOC_SINGLE("EQ1 Switch", M98095_088_CFG_LEVEL, 0, 1, 0), + SOC_SINGLE("EQ2 Switch", M98095_088_CFG_LEVEL, 1, 1, 0), + + SOC_SINGLE("Biquad1 Switch", M98095_088_CFG_LEVEL, 2, 1, 0), + SOC_SINGLE("Biquad2 Switch", M98095_088_CFG_LEVEL, 3, 1, 0), + SOC_ENUM("DAI1 Filter Mode", max98095_dai1_filter_mode_enum), SOC_ENUM("DAI2 Filter Mode", max98095_dai2_filter_mode_enum), SOC_ENUM("DAI1 DAC Filter", max98095_dai1_dac_filter_enum), @@ -1766,6 +1848,299 @@ static struct snd_soc_dai_driver max98095_dai[] = { }; +static int max98095_get_eq_channel(const char *name) +{ + if (strcmp(name, "EQ1 Mode") == 0) + return 0; + if (strcmp(name, "EQ2 Mode") == 0) + return 1; + return -EINVAL; +} + +static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_pdata *pdata = max98095->pdata; + int channel = max98095_get_eq_channel(kcontrol->id.name); + struct max98095_cdata *cdata; + int sel = ucontrol->value.integer.value[0]; + struct max98095_eq_cfg *coef_set; + int fs, best, best_val, i; + int regmask, regsave; + + BUG_ON(channel > 1); + + cdata = &max98095->dai[channel]; + + if (sel >= pdata->eq_cfgcnt) + return -EINVAL; + + cdata->eq_sel = sel; + + if (!pdata || !max98095->eq_textcnt) + return 0; + + fs = cdata->rate; + + /* Find the selected configuration with nearest sample rate */ + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->eq_cfgcnt; i++) { + if (strcmp(pdata->eq_cfg[i].name, max98095->eq_texts[sel]) == 0 && + abs(pdata->eq_cfg[i].rate - fs) < best_val) { + best = i; + best_val = abs(pdata->eq_cfg[i].rate - fs); + } + } + + dev_dbg(codec->dev, "Selected %s/%dHz for %dHz sample rate\n", + pdata->eq_cfg[best].name, + pdata->eq_cfg[best].rate, fs); + + coef_set = &pdata->eq_cfg[best]; + + regmask = (channel == 0) ? M98095_EQ1EN : M98095_EQ2EN; + + /* Disable filter while configuring, and save current on/off state */ + regsave = snd_soc_read(codec, M98095_088_CFG_LEVEL); + snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, 0); + + mutex_lock(&codec->mutex); + snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, M98095_SEG); + m98095_eq_band(codec, channel, 0, coef_set->band1); + m98095_eq_band(codec, channel, 1, coef_set->band2); + m98095_eq_band(codec, channel, 2, coef_set->band3); + m98095_eq_band(codec, channel, 3, coef_set->band4); + m98095_eq_band(codec, channel, 4, coef_set->band5); + snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, 0); + mutex_unlock(&codec->mutex); + + /* Restore the original on/off state */ + snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, regsave); + return 0; +} + +static int max98095_get_eq_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int channel = max98095_get_eq_channel(kcontrol->id.name); + struct max98095_cdata *cdata; + + cdata = &max98095->dai[channel]; + ucontrol->value.enumerated.item[0] = cdata->eq_sel; + + return 0; +} + +static void max98095_handle_eq_pdata(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_pdata *pdata = max98095->pdata; + struct max98095_eq_cfg *cfg; + unsigned int cfgcnt; + int i, j; + const char **t; + int ret; + + struct snd_kcontrol_new controls[] = { + SOC_ENUM_EXT("EQ1 Mode", + max98095->eq_enum, + max98095_get_eq_enum, + max98095_put_eq_enum), + SOC_ENUM_EXT("EQ2 Mode", + max98095->eq_enum, + max98095_get_eq_enum, + max98095_put_eq_enum), + }; + + cfg = pdata->eq_cfg; + cfgcnt = pdata->eq_cfgcnt; + + /* Setup an array of texts for the equalizer enum. + * This is based on Mark Brown's equalizer driver code. + */ + max98095->eq_textcnt = 0; + max98095->eq_texts = NULL; + for (i = 0; i < cfgcnt; i++) { + for (j = 0; j < max98095->eq_textcnt; j++) { + if (strcmp(cfg[i].name, max98095->eq_texts[j]) == 0) + break; + } + + if (j != max98095->eq_textcnt) + continue; + + /* Expand the array */ + t = krealloc(max98095->eq_texts, + sizeof(char *) * (max98095->eq_textcnt + 1), + GFP_KERNEL); + if (t == NULL) + continue; + + /* Store the new entry */ + t[max98095->eq_textcnt] = cfg[i].name; + max98095->eq_textcnt++; + max98095->eq_texts = t; + } + + /* Now point the soc_enum to .texts array items */ + max98095->eq_enum.texts = max98095->eq_texts; + max98095->eq_enum.max = max98095->eq_textcnt; + + ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + if (ret != 0) + dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); +} + +static int max98095_get_bq_channel(const char *name) +{ + if (strcmp(name, "Biquad1 Mode") == 0) + return 0; + if (strcmp(name, "Biquad2 Mode") == 0) + return 1; + return -EINVAL; +} + +static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_pdata *pdata = max98095->pdata; + int channel = max98095_get_bq_channel(kcontrol->id.name); + struct max98095_cdata *cdata; + int sel = ucontrol->value.integer.value[0]; + struct max98095_biquad_cfg *coef_set; + int fs, best, best_val, i; + int regmask, regsave; + + BUG_ON(channel > 1); + + cdata = &max98095->dai[channel]; + + if (sel >= pdata->bq_cfgcnt) + return -EINVAL; + + cdata->bq_sel = sel; + + if (!pdata || !max98095->bq_textcnt) + return 0; + + fs = cdata->rate; + + /* Find the selected configuration with nearest sample rate */ + best = 0; + best_val = INT_MAX; + for (i = 0; i < pdata->bq_cfgcnt; i++) { + if (strcmp(pdata->bq_cfg[i].name, max98095->bq_texts[sel]) == 0 && + abs(pdata->bq_cfg[i].rate - fs) < best_val) { + best = i; + best_val = abs(pdata->bq_cfg[i].rate - fs); + } + } + + dev_dbg(codec->dev, "Selected %s/%dHz for %dHz sample rate\n", + pdata->bq_cfg[best].name, + pdata->bq_cfg[best].rate, fs); + + coef_set = &pdata->bq_cfg[best]; + + regmask = (channel == 0) ? M98095_BQ1EN : M98095_BQ2EN; + + /* Disable filter while configuring, and save current on/off state */ + regsave = snd_soc_read(codec, M98095_088_CFG_LEVEL); + snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, 0); + + mutex_lock(&codec->mutex); + snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, M98095_SEG); + m98095_biquad_band(codec, channel, 0, coef_set->band1); + m98095_biquad_band(codec, channel, 1, coef_set->band2); + snd_soc_update_bits(codec, M98095_00F_HOST_CFG, M98095_SEG, 0); + mutex_unlock(&codec->mutex); + + /* Restore the original on/off state */ + snd_soc_update_bits(codec, M98095_088_CFG_LEVEL, regmask, regsave); + return 0; +} + +static int max98095_get_bq_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + int channel = max98095_get_bq_channel(kcontrol->id.name); + struct max98095_cdata *cdata; + + cdata = &max98095->dai[channel]; + ucontrol->value.enumerated.item[0] = cdata->bq_sel; + + return 0; +} + +static void max98095_handle_bq_pdata(struct snd_soc_codec *codec) +{ + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); + struct max98095_pdata *pdata = max98095->pdata; + struct max98095_biquad_cfg *cfg; + unsigned int cfgcnt; + int i, j; + const char **t; + int ret; + + struct snd_kcontrol_new controls[] = { + SOC_ENUM_EXT("Biquad1 Mode", + max98095->bq_enum, + max98095_get_bq_enum, + max98095_put_bq_enum), + SOC_ENUM_EXT("Biquad2 Mode", + max98095->bq_enum, + max98095_get_bq_enum, + max98095_put_bq_enum), + }; + + cfg = pdata->bq_cfg; + cfgcnt = pdata->bq_cfgcnt; + + /* Setup an array of texts for the biquad enum. + * This is based on Mark Brown's equalizer driver code. + */ + max98095->bq_textcnt = 0; + max98095->bq_texts = NULL; + for (i = 0; i < cfgcnt; i++) { + for (j = 0; j < max98095->bq_textcnt; j++) { + if (strcmp(cfg[i].name, max98095->bq_texts[j]) == 0) + break; + } + + if (j != max98095->bq_textcnt) + continue; + + /* Expand the array */ + t = krealloc(max98095->bq_texts, + sizeof(char *) * (max98095->bq_textcnt + 1), + GFP_KERNEL); + if (t == NULL) + continue; + + /* Store the new entry */ + t[max98095->bq_textcnt] = cfg[i].name; + max98095->bq_textcnt++; + max98095->bq_texts = t; + } + + /* Now point the soc_enum to .texts array items */ + max98095->bq_enum.texts = max98095->bq_texts; + max98095->bq_enum.max = max98095->bq_textcnt; + + ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + if (ret != 0) + dev_err(codec->dev, "Failed to add Biquad control: %d\n", ret); +} + static void max98095_handle_pdata(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); @@ -1785,6 +2160,14 @@ static void max98095_handle_pdata(struct snd_soc_codec *codec) regval |= M98095_DIGMIC_R; snd_soc_write(codec, M98095_087_CFG_MIC, regval); + + /* Configure equalizers */ + if (pdata->eq_cfgcnt) + max98095_handle_eq_pdata(codec); + + /* Configure bi-quad filters */ + if (pdata->bq_cfgcnt) + max98095_handle_bq_pdata(codec); } #ifdef CONFIG_PM @@ -1855,18 +2238,26 @@ static int max98095_probe(struct snd_soc_codec *codec) /* initialize private data */ max98095->sysclk = (unsigned)-1; + max98095->eq_textcnt = 0; + max98095->bq_textcnt = 0; cdata = &max98095->dai[0]; cdata->rate = (unsigned)-1; cdata->fmt = (unsigned)-1; + cdata->eq_sel = 0; + cdata->bq_sel = 0; cdata = &max98095->dai[1]; cdata->rate = (unsigned)-1; cdata->fmt = (unsigned)-1; + cdata->eq_sel = 0; + cdata->bq_sel = 0; cdata = &max98095->dai[2]; cdata->rate = (unsigned)-1; cdata->fmt = (unsigned)-1; + cdata->eq_sel = 0; + cdata->bq_sel = 0; max98095->lin_state = 0; max98095->mic1pre = 0; diff --git a/sound/soc/codecs/max98095.h b/sound/soc/codecs/max98095.h index 5b22bc8dbed..891584a0eb0 100644 --- a/sound/soc/codecs/max98095.h +++ b/sound/soc/codecs/max98095.h @@ -250,6 +250,8 @@ /* M98095_088_CFG_LEVEL */ #define M98095_VSEN (1<<6) #define M98095_ZDEN (1<<5) + #define M98095_BQ2EN (1<<3) + #define M98095_BQ1EN (1<<2) #define M98095_EQ2EN (1<<1) #define M98095_EQ1EN (1<<0) @@ -281,4 +283,17 @@ #define M98095_PWRSV8K (1<<1) #define M98095_PWRSV (1<<0) +#define M98095_COEFS_PER_BAND 5 + +#define M98095_BYTE1(w) ((w >> 8) & 0xff) +#define M98095_BYTE0(w) (w & 0xff) + +/* Equalizer filter coefficients */ +#define M98095_110_DAI1_EQ_BASE 0x10 +#define M98095_142_DAI2_EQ_BASE 0x42 + +/* Biquad filter coefficients */ +#define M98095_174_DAI1_BQ_BASE 0x74 +#define M98095_17E_DAI2_BQ_BASE 0x7E + #endif -- cgit v1.2.3 From 97945c46a23de5f9dfedf1b4a33e51d074df9a9c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 18 Apr 2011 20:58:11 -0600 Subject: ASoC: WM8903: Implement DMIC support In addition to the currently supported analog capture path, the WM8903 also supports digital mics. The analog and digital capture paths are exclusive; a mux is present to select the capture source. Logically, the mux exists to select the decimator's input, from either the ADC or DMIC block outputs. However, the ADC power domain also includes the DMIC interface. Consequently, this change represents the mux as existing immediately before the ADC, and selecting between the Input PGA and DMIC block outputs. An alternative might be to represent the mux in its correct location, and associate the ADC power enable controls with both the real ADC, and a fake ADC for the DMIC? Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d53f206d641..f3cab849143 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -634,6 +634,13 @@ static const struct soc_enum lsidetone_enum = static const struct soc_enum rsidetone_enum = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static const char *adcinput_text[] = { + "ADC", "DMIC" +}; + +static const struct soc_enum adcinput_enum = + SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text); + static const char *aif_text[] = { "Left", "Right" }; @@ -767,6 +774,9 @@ static const struct snd_kcontrol_new lsidetone_mux = static const struct snd_kcontrol_new rsidetone_mux = SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); +static const struct snd_kcontrol_new adcinput_mux = + SOC_DAPM_ENUM("ADC Input", adcinput_enum); + static const struct snd_kcontrol_new lcapture_mux = SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum); @@ -817,6 +827,7 @@ SND_SOC_DAPM_INPUT("IN2L"), SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), @@ -842,6 +853,9 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux), SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), +SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), +SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), + SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0), @@ -979,6 +993,11 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Left Input PGA", NULL, "Left Input Mode Mux" }, { "Right Input PGA", NULL, "Right Input Mode Mux" }, + { "Left ADC Input", "ADC", "Left Input PGA" }, + { "Left ADC Input", "DMIC", "DMICDAT" }, + { "Right ADC Input", "ADC", "Right Input PGA" }, + { "Right ADC Input", "DMIC", "DMICDAT" }, + { "Left Capture Mux", "Left", "ADCL" }, { "Left Capture Mux", "Right", "ADCR" }, @@ -988,9 +1007,9 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "AIFTXL", NULL, "Left Capture Mux" }, { "AIFTXR", NULL, "Right Capture Mux" }, - { "ADCL", NULL, "Left Input PGA" }, + { "ADCL", NULL, "Left ADC Input" }, { "ADCL", NULL, "CLK_DSP" }, - { "ADCR", NULL, "Right Input PGA" }, + { "ADCR", NULL, "Right ADC Input" }, { "ADCR", NULL, "CLK_DSP" }, { "Left Playback Mux", "Left", "AIFRXL" }, -- cgit v1.2.3 From e66d74ced12699ed9b7ac62f68b44d842a817ecd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 Mar 2011 18:20:46 +0100 Subject: ALSA: asihpi - Use %zd for size_t argument in error message (again) This was reverted mistakenly in the recent update patch. Fixed again. Reported-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 53037342cd2..3b5562acb55 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -2089,7 +2089,7 @@ static u16 message_response_sequence(struct hpi_adapter_obj *pao, phr->specific_error = sizeof(interface->u); phr->size = sizeof(struct hpi_response_header); HPI_DEBUG_LOG(ERROR, - "message len %d too big for buffer %ld \n", phm->size, + "message len %d too big for buffer %zd \n", phm->size, sizeof(interface->u)); return 0; } -- cgit v1.2.3 From d2edeb7c6f1dada8ca7d5c23e42d604e92ae0c76 Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Wed, 20 Apr 2011 10:59:57 -0700 Subject: ALSA: hda - ALSA HD Audio patch for Intel Panther Point DeviceIDs This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 70a9d32f0e9..6f891ee82a7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -126,6 +126,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH10}," "{Intel, PCH}," "{Intel, CPT}," + "{Intel, PPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -2759,6 +2760,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, + /* Panther Point */ + { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* Generic Intel */ -- cgit v1.2.3 From 47912a657ec2aa52c7af5f5e2ecc4efe41094d44 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 19 Apr 2011 10:18:01 -0600 Subject: ARM: Tegra: select MACH_HAS_SND_SOC_TEGRA_WM8903 CONFIG_SND_SOC_TEGRA_WM8903 is useful for many Tegra boards. To avoid the ASoC tegra/Kconfig enumerating them all, instead have the Tegra machine Kconfig select MACH_HAS_SND_SOC_TEGRA_WM8903 where appropriate, and have SND_SOC_TEGRA_WM8903 depend on this. [Redid ASoC diff so it applies. -- broonie] Signed-off-by: Stephen Warren Acked-by: Olof Johansson Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 66b504f06c2..0f103a101e3 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -14,6 +14,13 @@ config SND_TEGRA_SOC_I2S Tegra I2S interface. You will also need to select the individual machine drivers to support below. +config MACH_HAS_SND_SOC_TEGRA_WM8903 + bool + help + Machines that use the SND_SOC_TEGRA_WM8903 driver should select + this config option, in order to allow the user to enable + SND_SOC_TEGRA_WM8903. + config SND_TEGRA_SOC_HARMONY tristate "SoC Audio support for Tegra Harmony reference board" depends on SND_TEGRA_SOC && MACH_HARMONY && I2C -- cgit v1.2.3 From 885f42e1f466c36e3663d912a831e940f01a112b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Apr 2011 15:27:58 +0200 Subject: ALSA: hda - Enable sync_write for AMD chipset with IDT 92HD8x codecs The AMD chipset seems unstable in the normal operation mode, and it seems requring more sensible access for each verb. Enabling sync_write mode and allowing bus-reset is a sort of workaround for these chipset stability issues. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 05fcd60cc46..c391bfb95e0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5446,6 +5446,13 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->multiout.dac_nids = spec->dac_nids; spec->init = stac92hd83xxx_core_init; + if (codec->bus->pci && codec->bus->pci->vendor == PCI_VENDOR_ID_AMD) { + snd_printk(KERN_INFO "idt92hd83xxx: " + "Enable sync_write for AMD chipset\n"); + codec->bus->sync_write = 1; + codec->bus->allow_bus_reset = 1; + } + spec->board_config = snd_hda_check_board_config(codec, STAC_92HD83XXX_MODELS, stac92hd83xxx_models, -- cgit v1.2.3 From 30282f96d1eef33be774d4ecf4bddba30a6152ec Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Thu, 21 Apr 2011 21:54:09 +0300 Subject: ALSA: powermac - Correct lineout detection on PowerMac G4 DA Correct lineout (Pro Speaker) detection on PowerMac G4 Digital Audio (Tumbler). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/tumbler.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 961d9829769..9cea84c3e0c 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -1000,7 +1000,7 @@ static void device_change_handler(struct work_struct *work) chip->lineout_sw_ctl); if (mix->anded_reset) msleep(10); - check_mute(chip, &mix->amp_mute, 1, mix->auto_mute_notify, + check_mute(chip, &mix->amp_mute, !IS_G4DA, mix->auto_mute_notify, chip->speaker_sw_ctl); } else { /* unmute speaker, mute others */ -- cgit v1.2.3 From 8ae9572b5b08f1d2a2ea6613f59d00f741b38b2d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 23 Apr 2011 20:56:43 +0200 Subject: ALSA: 6fire: use the kernel's built-in bit reverse table Signed-off-by: Daniel Mack Cc: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/firmware.c | 29 ++--------------------------- sound/usb/Kconfig | 1 + 2 files changed, 3 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index e720035fbd1..45ffa12f0da 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -15,6 +15,7 @@ */ #include +#include #include "firmware.h" #include "chip.h" @@ -27,32 +28,6 @@ enum { FPGA_BUFSIZE = 512, FPGA_EP = 2 }; -static const u8 BIT_REVERSE_TABLE[256] = { - 0x00, 0x80, 0x40, 0xc0, 0x20, 0xa0, 0x60, 0xe0, 0x10, 0x90, 0x50, - 0xd0, 0x30, 0xb0, 0x70, 0xf0, 0x08, 0x88, 0x48, 0xc8, 0x28, 0xa8, - 0x68, 0xe8, 0x18, 0x98, 0x58, 0xd8, 0x38, 0xb8, 0x78, 0xf8, 0x04, - 0x84, 0x44, 0xc4, 0x24, 0xa4, 0x64, 0xe4, 0x14, 0x94, 0x54, 0xd4, - 0x34, 0xb4, 0x74, 0xf4, 0x0c, 0x8c, 0x4c, 0xcc, 0x2c, 0xac, 0x6c, - 0xec, 0x1c, 0x9c, 0x5c, 0xdc, 0x3c, 0xbc, 0x7c, 0xfc, 0x02, 0x82, - 0x42, 0xc2, 0x22, 0xa2, 0x62, 0xe2, 0x12, 0x92, 0x52, 0xd2, 0x32, - 0xb2, 0x72, 0xf2, 0x0a, 0x8a, 0x4a, 0xca, 0x2a, 0xaa, 0x6a, 0xea, - 0x1a, 0x9a, 0x5a, 0xda, 0x3a, 0xba, 0x7a, 0xfa, 0x06, 0x86, 0x46, - 0xc6, 0x26, 0xa6, 0x66, 0xe6, 0x16, 0x96, 0x56, 0xd6, 0x36, 0xb6, - 0x76, 0xf6, 0x0e, 0x8e, 0x4e, 0xce, 0x2e, 0xae, 0x6e, 0xee, 0x1e, - 0x9e, 0x5e, 0xde, 0x3e, 0xbe, 0x7e, 0xfe, 0x01, 0x81, 0x41, 0xc1, - 0x21, 0xa1, 0x61, 0xe1, 0x11, 0x91, 0x51, 0xd1, 0x31, 0xb1, 0x71, - 0xf1, 0x09, 0x89, 0x49, 0xc9, 0x29, 0xa9, 0x69, 0xe9, 0x19, 0x99, - 0x59, 0xd9, 0x39, 0xb9, 0x79, 0xf9, 0x05, 0x85, 0x45, 0xc5, 0x25, - 0xa5, 0x65, 0xe5, 0x15, 0x95, 0x55, 0xd5, 0x35, 0xb5, 0x75, 0xf5, - 0x0d, 0x8d, 0x4d, 0xcd, 0x2d, 0xad, 0x6d, 0xed, 0x1d, 0x9d, 0x5d, - 0xdd, 0x3d, 0xbd, 0x7d, 0xfd, 0x03, 0x83, 0x43, 0xc3, 0x23, 0xa3, - 0x63, 0xe3, 0x13, 0x93, 0x53, 0xd3, 0x33, 0xb3, 0x73, 0xf3, 0x0b, - 0x8b, 0x4b, 0xcb, 0x2b, 0xab, 0x6b, 0xeb, 0x1b, 0x9b, 0x5b, 0xdb, - 0x3b, 0xbb, 0x7b, 0xfb, 0x07, 0x87, 0x47, 0xc7, 0x27, 0xa7, 0x67, - 0xe7, 0x17, 0x97, 0x57, 0xd7, 0x37, 0xb7, 0x77, 0xf7, 0x0f, 0x8f, - 0x4f, 0xcf, 0x2f, 0xaf, 0x6f, 0xef, 0x1f, 0x9f, 0x5f, 0xdf, 0x3f, - 0xbf, 0x7f, 0xff }; - /* * wMaxPacketSize of pcm endpoints. * keep synced with rates_in_packet_size and rates_out_packet_size in pcm.c @@ -338,7 +313,7 @@ static int usb6fire_fw_fpga_upload( while (c != end) { for (i = 0; c != end && i < FPGA_BUFSIZE; i++, c++) - buffer[i] = BIT_REVERSE_TABLE[(u8) *c]; + buffer[i] = byte_rev_table[(u8) *c]; ret = usb6fire_fw_fpga_write(device, buffer, i); if (ret < 0) { diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index cc47a9caec9..8beb77563da 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -101,6 +101,7 @@ config SND_USB_US122L config SND_USB_6FIRE tristate "TerraTec DMX 6Fire USB" select FW_LOADER + select BITREVERSE select SND_RAWMIDI select SND_PCM help -- cgit v1.2.3 From 5debd6c14c302614764d7fcc8fe958c310c3d3b7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 21 Apr 2011 12:01:49 +0100 Subject: ASoC: Remove default settings from Tegra Kconfig There needs to be a strong reason for overriding the Kconfig default. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/tegra/Kconfig | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index a759fbe1263..0f5bd8298a0 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,14 +1,12 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA - default m help Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA_I2S tristate depends on SND_SOC_TEGRA - default m help Say Y or M if you want to add support for codecs attached to the Tegra I2S interface. You will also need to select the individual @@ -25,7 +23,6 @@ config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C depends on MACH_HAS_SND_SOC_TEGRA_WM8903 - default m select SND_SOC_TEGRA_I2S select SND_SOC_WM8903 help -- cgit v1.2.3 From 13eb4ab8ca719c852ae5fbd6e803afa333ad569a Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Tue, 26 Apr 2011 12:15:23 +0800 Subject: ALSA: au88x0 - Use a better name for pcm devices of au88x0 Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au8810.h | 2 +- sound/pci/au88x0/au8820.h | 2 +- sound/pci/au88x0/au8830.h | 2 +- sound/pci/au88x0/au88x0_pcm.c | 13 +++++++------ 4 files changed, 10 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au8810.h b/sound/pci/au88x0/au8810.h index 5d69c31fe3f..79fbee3845e 100644 --- a/sound/pci/au88x0/au8810.h +++ b/sound/pci/au88x0/au8810.h @@ -4,7 +4,7 @@ #define CHIP_AU8810 -#define CARD_NAME "Aureal Advantage 3D Sound Processor" +#define CARD_NAME "Aureal Advantage" #define CARD_NAME_SHORT "au8810" #define NR_ADB 0x10 diff --git a/sound/pci/au88x0/au8820.h b/sound/pci/au88x0/au8820.h index abbe85e4f7a..cafdb9668a3 100644 --- a/sound/pci/au88x0/au8820.h +++ b/sound/pci/au88x0/au8820.h @@ -11,7 +11,7 @@ #define CHIP_AU8820 -#define CARD_NAME "Aureal Vortex 3D Sound Processor" +#define CARD_NAME "Aureal Vortex" #define CARD_NAME_SHORT "au8820" /* Number of ADB and WT channels */ diff --git a/sound/pci/au88x0/au8830.h b/sound/pci/au88x0/au8830.h index 04ece1b1c21..999b29ab34a 100644 --- a/sound/pci/au88x0/au8830.h +++ b/sound/pci/au88x0/au8830.h @@ -11,7 +11,7 @@ #define CHIP_AU8830 -#define CARD_NAME "Aureal Vortex 2 3D Sound Processor" +#define CARD_NAME "Aureal Vortex 2" #define CARD_NAME_SHORT "au8830" #define NR_ADB 0x20 diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5439d662d10..04b10fdb0d7 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -423,11 +423,11 @@ static struct snd_pcm_ops snd_vortex_playback_ops = { */ static char *vortex_pcm_prettyname[VORTEX_PCM_LAST] = { - "AU88x0 ADB", - "AU88x0 SPDIF", - "AU88x0 A3D", - "AU88x0 WT", - "AU88x0 I2S", + CARD_NAME " ADB", + CARD_NAME " SPDIF", + CARD_NAME " A3D", + CARD_NAME " WT", + CARD_NAME " I2S", }; static char *vortex_pcm_name[VORTEX_PCM_LAST] = { "adb", @@ -524,7 +524,8 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) nr_capt, &pcm); if (err < 0) return err; - strcpy(pcm->name, vortex_pcm_name[idx]); + snprintf(pcm->name, sizeof(pcm->name), + "%s %s", CARD_NAME_SHORT, vortex_pcm_name[idx]); chip->pcm[idx] = pcm; // This is an evil hack, but it saves a lot of duplicated code. VORTEX_PCM_TYPE(pcm) = idx; -- cgit v1.2.3 From 6a9ebad8214bba404255d1b209a038dc739c37b7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 21 Apr 2011 10:33:36 +0900 Subject: ASoC: sh: fsi: add fsi_is_clk_master function If FSI port is clock master, it use set_rate function which is callback from platform, and it is not necessary to call it if FSI port is clock slave. Current FSI driver called this callback if platform provide it. This patch modify it. Signed-off-by: Kuninori Morimoto Reviewed-by: Simon Horman Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 33 +++++++++++++++++++++++++-------- 1 file changed, 25 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 23c0e83d4c1..5a2fdf3b84b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -146,10 +146,12 @@ struct fsi_priv { void __iomem *base; struct fsi_master *master; - int chan_num; struct fsi_stream playback; struct fsi_stream capture; + int chan_num:16; + int clk_master:1; + long rate; }; @@ -244,6 +246,11 @@ static struct fsi_master *fsi_get_master(struct fsi_priv *fsi) return fsi->master; } +static int fsi_is_clk_master(struct fsi_priv *fsi) +{ + return fsi->clk_master; +} + static int fsi_is_port_a(struct fsi_priv *fsi) { return fsi->master->base == fsi->base; @@ -793,14 +800,15 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct fsi_priv *fsi = fsi_get_priv(substream); int is_play = fsi_is_play(substream); struct fsi_master *master = fsi_get_master(fsi); - set_rate_func set_rate; + set_rate_func set_rate = fsi_get_info_set_rate(master); fsi_irq_disable(fsi, is_play); - fsi_clk_ctrl(fsi, 0); - set_rate = fsi_get_info_set_rate(master); - if (set_rate && fsi->rate) + if (fsi_is_clk_master(fsi)) { + fsi_clk_ctrl(fsi, 0); set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0); + } + fsi->rate = 0; pm_runtime_put_sync(dai->dev); @@ -876,6 +884,8 @@ static int fsi_set_fmt_spdif(struct fsi_priv *fsi) static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); + struct fsi_master *master = fsi_get_master(fsi); + set_rate_func set_rate = fsi_get_info_set_rate(master); u32 flags = fsi_get_info_flags(fsi); u32 data = 0; int ret; @@ -886,6 +896,7 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: data = DIMD | DOMD; + fsi->clk_master = 1; break; case SND_SOC_DAIFMT_CBS_CFS: break; @@ -893,6 +904,13 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = -EINVAL; goto set_fmt_exit; } + + if (fsi_is_clk_master(fsi) && !set_rate) { + dev_err(dai->dev, "platform doesn't have set_rate\n"); + ret = -EINVAL; + goto set_fmt_exit; + } + fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data); /* set format */ @@ -919,13 +937,12 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_master *master = fsi_get_master(fsi); - set_rate_func set_rate; + set_rate_func set_rate = fsi_get_info_set_rate(master); int fsi_ver = master->core->ver; long rate = params_rate(params); int ret; - set_rate = fsi_get_info_set_rate(master); - if (!set_rate) + if (!fsi_is_clk_master(fsi)) return 0; ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1); -- cgit v1.2.3 From 106c79ecf2db141fcd6073de55ebeb3f041e0509 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 21 Apr 2011 10:33:47 +0900 Subject: ASoC: sh: fsi: add dev_pm_ops :: suspend/resume Current FSI driver sets important settings when probing. And it are not set again as long as driver is not bind again. This mean FSI driver will lost it from register if suspend/resume are happen. This patch save important settings for suspend/resume. Signed-off-by: Kuninori Morimoto Reviewed-by: Simon Horman Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 86 ++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 86 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5a2fdf3b84b..1029a03c618 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -153,6 +153,13 @@ struct fsi_priv { int clk_master:1; long rate; + + /* for suspend/resume */ + u32 saved_do_fmt; + u32 saved_di_fmt; + u32 saved_ckg1; + u32 saved_ckg2; + u32 saved_out_sel; }; struct fsi_core { @@ -173,6 +180,13 @@ struct fsi_master { struct fsi_core *core; struct sh_fsi_platform_info *info; spinlock_t lock; + + /* for suspend/resume */ + u32 saved_a_mclk; + u32 saved_b_mclk; + u32 saved_iemsk; + u32 saved_imsk; + u32 saved_clk_rst; }; /* @@ -1277,6 +1291,76 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static void __fsi_suspend(struct fsi_priv *fsi, + struct device *dev, + set_rate_func set_rate) +{ + fsi->saved_do_fmt = fsi_reg_read(fsi, DO_FMT); + fsi->saved_di_fmt = fsi_reg_read(fsi, DI_FMT); + fsi->saved_ckg1 = fsi_reg_read(fsi, CKG1); + fsi->saved_ckg2 = fsi_reg_read(fsi, CKG2); + fsi->saved_out_sel = fsi_reg_read(fsi, OUT_SEL); + + if (fsi_is_clk_master(fsi)) + set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 0); +} + +static void __fsi_resume(struct fsi_priv *fsi, + struct device *dev, + set_rate_func set_rate) +{ + fsi_reg_write(fsi, DO_FMT, fsi->saved_do_fmt); + fsi_reg_write(fsi, DI_FMT, fsi->saved_di_fmt); + fsi_reg_write(fsi, CKG1, fsi->saved_ckg1); + fsi_reg_write(fsi, CKG2, fsi->saved_ckg2); + fsi_reg_write(fsi, OUT_SEL, fsi->saved_out_sel); + + if (fsi_is_clk_master(fsi)) + set_rate(dev, fsi_is_port_a(fsi), fsi->rate, 1); +} + +static int fsi_suspend(struct device *dev) +{ + struct fsi_master *master = dev_get_drvdata(dev); + set_rate_func set_rate = fsi_get_info_set_rate(master); + + pm_runtime_get_sync(dev); + + __fsi_suspend(&master->fsia, dev, set_rate); + __fsi_suspend(&master->fsib, dev, set_rate); + + master->saved_a_mclk = fsi_core_read(master, a_mclk); + master->saved_b_mclk = fsi_core_read(master, b_mclk); + master->saved_iemsk = fsi_core_read(master, iemsk); + master->saved_imsk = fsi_core_read(master, imsk); + master->saved_clk_rst = fsi_master_read(master, CLK_RST); + + pm_runtime_put_sync(dev); + + return 0; +} + +static int fsi_resume(struct device *dev) +{ + struct fsi_master *master = dev_get_drvdata(dev); + set_rate_func set_rate = fsi_get_info_set_rate(master); + + pm_runtime_get_sync(dev); + + __fsi_resume(&master->fsia, dev, set_rate); + __fsi_resume(&master->fsib, dev, set_rate); + + fsi_core_mask_set(master, a_mclk, 0xffff, master->saved_a_mclk); + fsi_core_mask_set(master, b_mclk, 0xffff, master->saved_b_mclk); + fsi_core_mask_set(master, iemsk, 0xffff, master->saved_iemsk); + fsi_core_mask_set(master, imsk, 0xffff, master->saved_imsk); + fsi_master_mask_set(master, CLK_RST, 0xffff, master->saved_clk_rst); + + pm_runtime_put_sync(dev); + + return 0; +} + static int fsi_runtime_nop(struct device *dev) { /* Runtime PM callback shared between ->runtime_suspend() @@ -1290,6 +1374,8 @@ static int fsi_runtime_nop(struct device *dev) } static struct dev_pm_ops fsi_pm_ops = { + .suspend = fsi_suspend, + .resume = fsi_resume, .runtime_suspend = fsi_runtime_nop, .runtime_resume = fsi_runtime_nop, }; -- cgit v1.2.3 From 1f5e2a319d2ba80bfea5c3b5cbafea09d5164a51 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 21 Apr 2011 10:33:52 +0900 Subject: ASoC: sh: fsi: Add module/port clock control function The FIFO of each port were always working though it was not used in current FSI driver. This patch add module/port clock control function for fixing it. This patch is also caring suspend/resume. Signed-off-by: Kuninori Morimoto Reviewed-by: Simon Horman Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 81 +++++++++++++++++++++++++++++++++--------------------- 1 file changed, 50 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1029a03c618..4a9da6b5f4e 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -86,8 +86,8 @@ #define SE (1 << 0) /* Fix the master clock */ /* CLK_RST */ -#define B_CLK 0x00000010 -#define A_CLK 0x00000001 +#define CRB (1 << 4) +#define CRA (1 << 0) /* IO SHIFT / MACRO */ #define BI_SHIFT 12 @@ -187,6 +187,7 @@ struct fsi_master { u32 saved_iemsk; u32 saved_imsk; u32 saved_clk_rst; + u32 saved_soft_rst; }; /* @@ -556,20 +557,45 @@ static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable) } /* - * ctrl function + * clock function */ +#define fsi_module_init(m, d) __fsi_module_clk_ctrl(m, d, 1) +#define fsi_module_kill(m, d) __fsi_module_clk_ctrl(m, d, 0) +static void __fsi_module_clk_ctrl(struct fsi_master *master, + struct device *dev, + int enable) +{ + pm_runtime_get_sync(dev); + + if (enable) { + /* enable only SR */ + fsi_master_mask_set(master, SOFT_RST, FSISR, FSISR); + fsi_master_mask_set(master, SOFT_RST, PASR | PBSR, 0); + } else { + /* clear all registers */ + fsi_master_mask_set(master, SOFT_RST, FSISR, 0); + } + + pm_runtime_put_sync(dev); +} -static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) +#define fsi_port_start(f) __fsi_port_clk_ctrl(f, 1) +#define fsi_port_stop(f) __fsi_port_clk_ctrl(f, 0) +static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int enable) { - u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); struct fsi_master *master = fsi_get_master(fsi); + u32 soft = fsi_is_port_a(fsi) ? PASR : PBSR; + u32 clk = fsi_is_port_a(fsi) ? CRA : CRB; + int is_master = fsi_is_clk_master(fsi); - if (enable) - fsi_master_mask_set(master, CLK_RST, val, val); - else - fsi_master_mask_set(master, CLK_RST, val, 0); + fsi_master_mask_set(master, SOFT_RST, soft, (enable) ? soft : 0); + if (is_master) + fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } +/* + * ctrl function + */ static void fsi_fifo_init(struct fsi_priv *fsi, int is_play, struct snd_soc_dai *dai) @@ -622,18 +648,6 @@ static void fsi_fifo_init(struct fsi_priv *fsi, } } -static void fsi_soft_all_reset(struct fsi_master *master) -{ - /* port AB reset */ - fsi_master_mask_set(master, SOFT_RST, PASR | PBSR, 0); - mdelay(10); - - /* soft reset */ - fsi_master_mask_set(master, SOFT_RST, FSISR, 0); - fsi_master_mask_set(master, SOFT_RST, FSISR, FSISR); - mdelay(10); -} - static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) { struct snd_pcm_runtime *runtime; @@ -818,10 +832,8 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); - if (fsi_is_clk_master(fsi)) { - fsi_clk_ctrl(fsi, 0); + if (fsi_is_clk_master(fsi)) set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0); - } fsi->rate = 0; @@ -843,8 +855,10 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, frames_to_bytes(runtime, runtime->period_size)); ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); fsi_irq_enable(fsi, is_play); + fsi_port_start(fsi); break; case SNDRV_PCM_TRIGGER_STOP: + fsi_port_stop(fsi); fsi_irq_disable(fsi, is_play); fsi_stream_pop(fsi, is_play); break; @@ -1018,7 +1032,6 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data); udelay(10); - fsi_clk_ctrl(fsi, 1); ret = 0; } @@ -1233,9 +1246,7 @@ static int fsi_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); - pm_runtime_get_sync(&pdev->dev); - fsi_soft_all_reset(master); - pm_runtime_put_sync(&pdev->dev); + fsi_module_init(master, &pdev->dev); ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, id_entry->name, master); @@ -1279,6 +1290,8 @@ static int fsi_remove(struct platform_device *pdev) master = dev_get_drvdata(&pdev->dev); + fsi_module_kill(master, &pdev->dev); + free_irq(master->irq, master); pm_runtime_disable(&pdev->dev); @@ -1334,6 +1347,9 @@ static int fsi_suspend(struct device *dev) master->saved_iemsk = fsi_core_read(master, iemsk); master->saved_imsk = fsi_core_read(master, imsk); master->saved_clk_rst = fsi_master_read(master, CLK_RST); + master->saved_soft_rst = fsi_master_read(master, SOFT_RST); + + fsi_module_kill(master, dev); pm_runtime_put_sync(dev); @@ -1347,14 +1363,17 @@ static int fsi_resume(struct device *dev) pm_runtime_get_sync(dev); - __fsi_resume(&master->fsia, dev, set_rate); - __fsi_resume(&master->fsib, dev, set_rate); + fsi_module_init(master, dev); + fsi_master_mask_set(master, SOFT_RST, 0xffff, master->saved_soft_rst); + fsi_master_mask_set(master, CLK_RST, 0xffff, master->saved_clk_rst); fsi_core_mask_set(master, a_mclk, 0xffff, master->saved_a_mclk); fsi_core_mask_set(master, b_mclk, 0xffff, master->saved_b_mclk); fsi_core_mask_set(master, iemsk, 0xffff, master->saved_iemsk); fsi_core_mask_set(master, imsk, 0xffff, master->saved_imsk); - fsi_master_mask_set(master, CLK_RST, 0xffff, master->saved_clk_rst); + + __fsi_resume(&master->fsia, dev, set_rate); + __fsi_resume(&master->fsib, dev, set_rate); pm_runtime_put_sync(dev); -- cgit v1.2.3 From c5f336cc003dfa071aaa7a01ed1c16232b227aa4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 21 Apr 2011 14:16:14 +0100 Subject: ASoC: Support 24.576MHz MCLKs in WM8915 We can safely divide these down to within the supported SYSCLK range. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 083609418bf..4a3c5cc77e5 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -1831,6 +1831,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, struct snd_soc_codec *codec = dai->codec; struct wm8915_priv *wm8915 = snd_soc_codec_get_drvdata(codec); int lfclk = 0; + int ratediv = 0; int src; int old; @@ -1862,6 +1863,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8915_AIF_RATE, WM8915_SYSCLK_RATE, 0); break; + case 24576000: + ratediv = WM8915_SYSCLK_DIV; case 12288000: snd_soc_update_bits(codec, WM8915_AIF_RATE, WM8915_SYSCLK_RATE, WM8915_SYSCLK_RATE); @@ -1877,8 +1880,8 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, } snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, - WM8915_SYSCLK_SRC_MASK, - src << WM8915_SYSCLK_SRC_SHIFT); + WM8915_SYSCLK_SRC_MASK | WM8915_SYSCLK_DIV_MASK, + src << WM8915_SYSCLK_SRC_SHIFT | ratediv); snd_soc_update_bits(codec, WM8915_CLOCKING_1, WM8915_LFCLK_ENA, lfclk); snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_ENA, old); -- cgit v1.2.3 From 3b8a6d80e5d02a827d882935b96c1c3c3b56c977 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 25 Apr 2011 17:53:43 +0100 Subject: ASoC: Support FLL lock interrupt on WM8962 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 500011eb8b2..d6b78b2a10d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -58,6 +58,7 @@ struct wm8962_priv { int bclk; /* Desired BCLK */ int lrclk; + struct completion fll_lock; int fll_src; int fll_fref; int fll_fout; @@ -3178,6 +3179,7 @@ static int wm8962_set_fll(struct snd_soc_dai *dai, int fll_id, int source, struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct _fll_div fll_div; + unsigned long timeout; int ret; int fll1 = snd_soc_read(codec, WM8962_FLL_CONTROL_1) & WM8962_FLL_ENA; @@ -3244,6 +3246,11 @@ static int wm8962_set_fll(struct snd_soc_dai *dai, int fll_id, int source, dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); + /* This should be a massive overestimate */ + timeout = msecs_to_jiffies(1); + + wait_for_completion_timeout(&wm8962->fll_lock, timeout); + wm8962->fll_fref = Fref; wm8962->fll_fout = Fout; wm8962->fll_src = source; @@ -3340,6 +3347,11 @@ static irqreturn_t wm8962_irq(int irq, void *data) active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + if (active & WM8962_FLL_LOCK_EINT) { + dev_dbg(codec->dev, "FLL locked\n"); + complete(&wm8962->fll_lock); + } + if (active & WM8962_FIFOS_ERR_EINT) dev_err(codec->dev, "FIFO error\n"); @@ -3712,6 +3724,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962->codec = codec; INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work); + init_completion(&wm8962->fll_lock); codec->cache_sync = 1; codec->dapm.idle_bias_off = 1; @@ -3868,9 +3881,10 @@ static int wm8962_probe(struct snd_soc_codec *codec) i2c->irq, ret); /* Non-fatal */ } else { - /* Enable error reporting IRQs by default */ + /* Enable some IRQs by default */ snd_soc_update_bits(codec, WM8962_INTERRUPT_STATUS_2_MASK, + WM8962_FLL_LOCK_EINT | WM8962_TEMP_SHUT_EINT | WM8962_FIFOS_ERR_EINT, 0); } -- cgit v1.2.3 From 92a4352cdb53443ea5cb8bafd018e69933edb0a5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 25 Apr 2011 18:44:01 +0100 Subject: ASoC: Move WM8962 FLL configuration to CODEC There's only one DAI anyway. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d6b78b2a10d..549018e4623 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3173,10 +3173,9 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, return 0; } -static int wm8962_set_fll(struct snd_soc_dai *dai, int fll_id, int source, +static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, unsigned int Fref, unsigned int Fout) { - struct snd_soc_codec *codec = dai->codec; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); struct _fll_div fll_div; unsigned long timeout; @@ -3281,7 +3280,6 @@ static struct snd_soc_dai_ops wm8962_dai_ops = { .hw_params = wm8962_hw_params, .set_sysclk = wm8962_set_dai_sysclk, .set_fmt = wm8962_set_dai_fmt, - .set_pll = wm8962_set_fll, .digital_mute = wm8962_mute, }; @@ -3932,6 +3930,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .reg_cache_default = wm8962_reg, .volatile_register = wm8962_volatile_register, .readable_register = wm8962_readable_register, + .set_pll = wm8962_set_fll, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.3 From e47ac37c01bd9bd840dbbbc57a6dc1ba1e49ccc0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 25 Apr 2011 20:14:21 +0100 Subject: ASoC: Implement WM8962 DMIC support DMIC support is automatically disabled when none of the GPIOs are set up to bring out the DMICCLK and DMICDAT pins at startup. Note that there's no support for controlling DMIC routing except the power control so the board DAPM configuration will need to manage DMIC enable and disable if analogue mics (eg, a headset) also exist. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 549018e4623..ec4417a5e66 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2468,6 +2468,7 @@ SND_SOC_DAPM_INPUT("IN3R"), SND_SOC_DAPM_INPUT("IN4L"), SND_SOC_DAPM_INPUT("IN4R"), SND_SOC_DAPM_INPUT("Beep"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_MICBIAS("MICBIAS", WM8962_PWR_MGMT_1, 1, 0), @@ -2487,6 +2488,8 @@ SND_SOC_DAPM_MIXER("MIXINL", WM8962_PWR_MGMT_1, 5, 0, SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0, mixinr, ARRAY_SIZE(mixinr)), +SND_SOC_DAPM_AIF_IN("DMIC", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), + SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0), @@ -2564,13 +2567,17 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "MICBIAS", NULL, "SYSCLK" }, + { "DMIC", NULL, "DMICDAT" }, + { "ADCL", NULL, "SYSCLK" }, { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, + { "ADCL", NULL, "DMIC" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, + { "ADCR", NULL, "DMIC" }, { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, @@ -3719,6 +3726,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) dev); u16 *reg_cache = codec->reg_cache; int i, trigger, irq_pol; + bool dmicclk, dmicdat; wm8962->codec = codec; INIT_DELAYED_WORK(&wm8962->mic_work, wm8962_mic_work); @@ -3856,6 +3864,29 @@ static int wm8962_probe(struct snd_soc_codec *codec) wm8962_add_widgets(codec); + /* Save boards having to disable DMIC when not in use */ + dmicclk = false; + dmicdat = false; + for (i = 0; i < WM8962_MAX_GPIO; i++) { + switch (snd_soc_read(codec, WM8962_GPIO_BASE + i) + & WM8962_GP2_FN_MASK) { + case WM8962_GPIO_FN_DMICCLK: + dmicclk = true; + break; + case WM8962_GPIO_FN_DMICDAT: + dmicdat = true; + break; + default: + break; + } + } + if (!dmicclk || !dmicdat) { + dev_dbg(codec->dev, "DMIC not in use, disabling\n"); + snd_soc_dapm_nc_pin(&codec->dapm, "DMICDAT"); + } + if (dmicclk != dmicdat) + dev_warn(codec->dev, "DMIC GPIOs partially configured\n"); + wm8962_init_beep(codec); wm8962_init_gpio(codec); -- cgit v1.2.3 From 5357e8f505d058b7419eb6a91b6e42b8f1fc02d5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 25 Apr 2011 18:27:35 +0100 Subject: ASoC: Don't warn if the WM8962 SYSCLK FLL setting doesn't match reality When bringing up audio low power modes boards may configure SYSCLK before they actually start the FLL as we do much of the clocking setup prior to the power up sequence. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ec4417a5e66..7949f89a8f8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2998,7 +2998,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, case WM8962_SYSCLK_FLL: wm8962->sysclk = WM8962_SYSCLK_FLL; src = 1 << WM8962_SYSCLK_SRC_SHIFT; - WARN_ON(freq != wm8962->fll_fout); break; default: return -EINVAL; -- cgit v1.2.3 From 0da2692256ed65bec588f7797c77f9c84ef4274e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Apr 2011 15:18:33 +0200 Subject: ALSA: hda - Move EAPD power-down into shutup callback for AD codecs EAPD power-down should be called also for normal shutup cases. Let's move to there. This also fixes the compile warnings when CONFIG_PM isn't set automatically. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2942d2a9ea1..12317488cd9 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -489,11 +489,6 @@ static int ad198x_build_pcms(struct hda_codec *codec) return 0; } -static inline void ad198x_shutup(struct hda_codec *codec) -{ - snd_hda_shutup_pins(codec); -} - static void ad198x_free_kctls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -547,6 +542,12 @@ static void ad198x_power_eapd(struct hda_codec *codec) } } +static void ad198x_shutup(struct hda_codec *codec) +{ + snd_hda_shutup_pins(codec); + ad198x_power_eapd(codec); +} + static void ad198x_free(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -564,7 +565,6 @@ static void ad198x_free(struct hda_codec *codec) static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) { ad198x_shutup(codec); - ad198x_power_eapd(codec); return 0; } #endif -- cgit v1.2.3 From d507cd668a3f6d07b31e914722b453c454b03204 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Apr 2011 15:25:02 +0200 Subject: ALSA: hda - Enable sync_write workaround for AMD generically The workaround for AMD chipset via sync_write flag seems needed for machines with Realtek codecs. So, it's better to activate it generically in hda_intel.c from the beginning. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 11 +++++++++++ sound/pci/hda/patch_sigmatel.c | 16 ---------------- 2 files changed, 11 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6f891ee82a7..f95ff6e029a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1447,6 +1447,17 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) } } + /* AMD chipsets often cause the communication stalls upon certain + * sequence like the pin-detection. It seems that forcing the synced + * access works around the stall. Grrr... + */ + if (chip->pci->vendor == PCI_VENDOR_ID_AMD || + chip->pci->vendor == PCI_VENDOR_ID_ATI) { + snd_printk(KERN_INFO SFX "Enable sync_write for AMD chipset\n"); + chip->bus->sync_write = 1; + chip->bus->allow_bus_reset = 1; + } + /* Then create codec instances */ for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 75b7155b164..6c5af3e77d8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5449,13 +5449,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->multiout.dac_nids = spec->dac_nids; spec->init = stac92hd83xxx_core_init; - if (codec->bus->pci && codec->bus->pci->vendor == PCI_VENDOR_ID_AMD) { - snd_printk(KERN_INFO "idt92hd83xxx: " - "Enable sync_write for AMD chipset\n"); - codec->bus->sync_write = 1; - codec->bus->allow_bus_reset = 1; - } - spec->board_config = snd_hda_check_board_config(codec, STAC_92HD83XXX_MODELS, stac92hd83xxx_models, @@ -5736,15 +5729,6 @@ again: if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); - /* Some HP machines seem to have unstable codec communications - * especially with ATI fglrx driver. For recovering from the - * CORB/RIRB stall, allow the BUS reset and keep always sync - */ - if (spec->board_config == STAC_HP_DV5) { - codec->bus->sync_write = 1; - codec->bus->allow_bus_reset = 1; - } - spec->aloopback_ctl = stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- cgit v1.2.3 From cb34c207af4944e9c93e2b462e351430f15daad6 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Wed, 27 Apr 2011 17:44:16 +0800 Subject: ALSA: hda - VIA: Fix Smart5.1 isn't useful for 6 audio jacks motherboard. For some motherboards with 5 or 6 audio jacks which had six or eight multiple channels output, smart5.1 item is no useful and should be removed. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 15b57a101aa..62fe64c419d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1021,6 +1021,11 @@ static int via_smart51_build(struct via_spec *spec) hda_nid_t nid; int i; + if (!cfg) + return 0; + if (cfg->line_outs > 2) + return 0; + knew = via_clone_control(spec, &via_smart51_mixer[0]); if (knew == NULL) return -ENOMEM; -- cgit v1.2.3 From 604401a92cb3b93047c199f68a068ccbc5fb70fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Apr 2011 15:14:23 +0200 Subject: ALSA: hda - Minor update for alc662-parser functions Allow alc662_dac_to_mix() and alc662_look_for_dac() to parse down the selector widget that is found in ALC880-type codecs, and rename them to alc_auto_*() accordingly. This is for the next coming multi-io extensions. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 38 +++++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e631874a3e3..0ed95dd9d64 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19135,9 +19135,9 @@ static struct alc_config_preset alc662_presets[] = { */ /* convert from MIX nid to DAC */ -static hda_nid_t alc662_mix_to_dac(struct hda_codec *codec, hda_nid_t nid) +static hda_nid_t alc_auto_mix_to_dac(struct hda_codec *codec, hda_nid_t nid) { - hda_nid_t list[4]; + hda_nid_t list[5]; int i, num; num = snd_hda_get_connections(codec, nid, list, ARRAY_SIZE(list)); @@ -19148,33 +19148,45 @@ static hda_nid_t alc662_mix_to_dac(struct hda_codec *codec, hda_nid_t nid) return 0; } +/* go down to the selector widget before the mixer */ +static hda_nid_t alc_go_down_to_selector(struct hda_codec *codec, hda_nid_t pin) +{ + hda_nid_t srcs[5]; + int num = snd_hda_get_connections(codec, pin, srcs, + ARRAY_SIZE(srcs)); + if (num != 1 || + get_wcaps_type(get_wcaps(codec, srcs[0])) != AC_WID_AUD_SEL) + return pin; + return srcs[0]; +} + /* get MIX nid connected to the given pin targeted to DAC */ -static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin, +static hda_nid_t alc_auto_dac_to_mix(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac) { hda_nid_t mix[5]; int i, num; + pin = alc_go_down_to_selector(codec, pin); num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix)); for (i = 0; i < num; i++) { - if (alc662_mix_to_dac(codec, mix[i]) == dac) + if (alc_auto_mix_to_dac(codec, mix[i]) == dac) return mix[i]; } return 0; } /* look for an empty DAC slot */ -static hda_nid_t alc662_look_for_dac(struct hda_codec *codec, hda_nid_t pin) +static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) { struct alc_spec *spec = codec->spec; hda_nid_t srcs[5]; int i, j, num; + pin = alc_go_down_to_selector(codec, pin); num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs)); - if (num < 0) - return 0; for (i = 0; i < num; i++) { - hda_nid_t nid = alc662_mix_to_dac(codec, srcs[i]); + hda_nid_t nid = alc_auto_mix_to_dac(codec, srcs[i]); if (!nid) continue; for (j = 0; j < spec->multiout.num_dacs; j++) @@ -19196,7 +19208,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec, spec->multiout.dac_nids = spec->private_dac_nids; for (i = 0; i < cfg->line_outs; i++) { - dac = alc662_look_for_dac(codec, cfg->line_out_pins[i]); + dac = alc_auto_look_for_dac(codec, cfg->line_out_pins[i]); if (!dac) continue; spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; @@ -19243,7 +19255,7 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, nid = spec->multiout.dac_nids[i]; if (!nid) continue; - mix = alc662_dac_to_mix(codec, cfg->line_out_pins[i], nid); + mix = alc_auto_dac_to_mix(codec, cfg->line_out_pins[i], nid); if (!mix) continue; if (!pfx && i == 2) { @@ -19289,7 +19301,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, if (!pin) return 0; - nid = alc662_look_for_dac(codec, pin); + nid = alc_auto_look_for_dac(codec, pin); if (!nid) { /* the corresponding DAC is already occupied */ if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)) @@ -19299,7 +19311,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); } - mix = alc662_dac_to_mix(codec, pin, nid); + mix = alc_auto_dac_to_mix(codec, pin, nid); if (!mix) return 0; err = alc662_add_vol_ctl(spec, pfx, nid, 3); @@ -19325,7 +19337,7 @@ static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, alc_set_pin_output(codec, nid, pin_type); num = snd_hda_get_connections(codec, nid, srcs, ARRAY_SIZE(srcs)); for (i = 0; i < num; i++) { - if (alc662_mix_to_dac(codec, srcs[i]) != dac) + if (alc_auto_mix_to_dac(codec, srcs[i]) != dac) continue; /* need the manual connection? */ if (num > 1) -- cgit v1.2.3 From ce764ab22e40a046065c9417ee5f04ad2a816ac1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Apr 2011 16:35:23 +0200 Subject: ALSA: hda - Add channel-mode support to Realtek auto-parser This patch adds the support of "Channel Mode" enum control to Realtek auto-parser. When line-in or mic-in jacks are capable to output and free DACs are available, the driver allows to switch to multi-channel mode via "Channel Mode" enum switch, as already implemented in some preset cases. Not implemented in all Realtek codecs. Currently, ALC880, 882, 861, 662 and the compatible codecs are supported. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 269 ++++++++++++++++++++++++++++++++++++++---- 1 file changed, 249 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ed95dd9d64..4c4d51a220b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -299,6 +299,12 @@ struct alc_customize_define { struct alc_fixup; +struct alc_multi_io { + hda_nid_t pin; /* multi-io widget pin NID */ + hda_nid_t dac; /* DAC to be connected */ + unsigned int ctl_in; /* cached input-pin control value */ +}; + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -404,6 +410,10 @@ struct alc_spec { int fixup_id; const struct alc_fixup *fixup_list; const char *fixup_name; + + /* multi-io */ + int multi_ios; + struct alc_multi_io multi_io[4]; }; /* @@ -4992,14 +5002,19 @@ static struct snd_kcontrol_new alc880_control_templates[] = { HDA_BIND_MUTE(NULL, 0, 0, 0), }; +static struct snd_kcontrol_new *alc_kcontrol_new(struct alc_spec *spec) +{ + snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32); + return snd_array_new(&spec->kctls); +} + /* add dynamic controls */ static int add_control(struct alc_spec *spec, int type, const char *name, int cidx, unsigned long val) { struct snd_kcontrol_new *knew; - snd_array_init(&spec->kctls, sizeof(*knew), 32); - knew = snd_array_new(&spec->kctls); + knew = alc_kcontrol_new(spec); if (!knew) return -ENOMEM; *knew = alc880_control_templates[type]; @@ -5080,10 +5095,13 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, return 0; } -static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg, +static const char *alc_get_line_out_pfx(struct alc_spec *spec, bool can_be_master) { - if (!cfg->hp_outs && !cfg->speaker_outs && can_be_master) + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (cfg->line_outs == 1 && !spec->multi_ios && + !cfg->hp_outs && !cfg->speaker_outs && can_be_master) return "Master"; switch (cfg->line_out_type) { @@ -5094,7 +5112,7 @@ static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg, case AUTO_PIN_HP_OUT: return "Headphone"; default: - if (cfg->line_outs == 1) + if (cfg->line_outs == 1 && !spec->multi_ios) return "PCM"; break; } @@ -5108,11 +5126,15 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - const char *pfx = alc_get_line_out_pfx(cfg, false); + const char *pfx = alc_get_line_out_pfx(spec, false); hda_nid_t nid; - int i, err; + int i, err, noutputs; - for (i = 0; i < cfg->line_outs; i++) { + noutputs = cfg->line_outs; + if (spec->multi_ios > 0) + noutputs += spec->multi_ios; + + for (i = 0; i < noutputs; i++) { if (!spec->multiout.dac_nids[i]) continue; nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i])); @@ -5378,6 +5400,8 @@ static void alc880_auto_init_input_src(struct hda_codec *codec) } } +static int alc_auto_add_multi_channel_mode(struct hda_codec *codec); + /* parse the BIOS configuration and set up the alc_spec */ /* return 1 if successful, 0 if the proper config is not found, * or a negative error code @@ -5396,6 +5420,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec) return 0; /* can't find valid BIOS pin config */ err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc_auto_add_multi_channel_mode(codec); if (err < 0) return err; err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); @@ -10947,6 +10974,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec) return 0; /* can't find valid BIOS pin config */ err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc_auto_add_multi_channel_mode(codec); if (err < 0) return err; err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); @@ -12165,7 +12195,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, spec->multiout.dac_nids = spec->private_dac_nids; spec->multiout.dac_nids[0] = 2; - pfx = alc_get_line_out_pfx(cfg, true); + pfx = alc_get_line_out_pfx(spec, true); if (!pfx) pfx = "Front"; for (i = 0; i < 2; i++) { @@ -16020,11 +16050,15 @@ static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - const char *pfx = alc_get_line_out_pfx(cfg, true); + const char *pfx = alc_get_line_out_pfx(spec, true); hda_nid_t nid; - int i, err; + int i, err, noutputs; - for (i = 0; i < cfg->line_outs; i++) { + noutputs = cfg->line_outs; + if (spec->multi_ios > 0) + noutputs += spec->multi_ios; + + for (i = 0; i < noutputs; i++) { nid = spec->multiout.dac_nids[i]; if (!nid) continue; @@ -16167,6 +16201,9 @@ static int alc861_parse_auto_config(struct hda_codec *codec) return 0; /* can't find valid BIOS pin config */ err = alc861_auto_fill_dac_nids(codec, &spec->autocfg); + if (err < 0) + return err; + err = alc_auto_add_multi_channel_mode(codec); if (err < 0) return err; err = alc861_auto_create_multi_out_ctls(codec, &spec->autocfg); @@ -17152,11 +17189,15 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, static const char * const chname[4] = { "Front", "Surround", "CLFE", "Side" }; - const char *pfx = alc_get_line_out_pfx(cfg, true); + const char *pfx = alc_get_line_out_pfx(spec, true); hda_nid_t nid_v, nid_s; - int i, err; + int i, err, noutputs; - for (i = 0; i < cfg->line_outs; i++) { + noutputs = cfg->line_outs; + if (spec->multi_ios > 0) + noutputs += spec->multi_ios; + + for (i = 0; i < noutputs; i++) { if (!spec->multiout.dac_nids[i]) continue; nid_v = alc861vd_idx_to_mixer_vol( @@ -17279,6 +17320,9 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) return 0; /* can't find valid BIOS pin config */ err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc_auto_add_multi_channel_mode(codec); if (err < 0) return err; err = alc861vd_auto_create_multi_out_ctls(spec, &spec->autocfg); @@ -19176,6 +19220,27 @@ static hda_nid_t alc_auto_dac_to_mix(struct hda_codec *codec, hda_nid_t pin, return 0; } +/* select the connection from pin to DAC if needed */ +static int alc_auto_select_dac(struct hda_codec *codec, hda_nid_t pin, + hda_nid_t dac) +{ + hda_nid_t mix[5]; + int i, num; + + pin = alc_go_down_to_selector(codec, pin); + num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix)); + if (num < 2) + return 0; + for (i = 0; i < num; i++) { + if (alc_auto_mix_to_dac(codec, mix[i]) == dac) { + snd_hda_codec_update_cache(codec, pin, 0, + AC_VERB_SET_CONNECT_SEL, i); + return 0; + } + } + return 0; +} + /* look for an empty DAC slot */ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) { @@ -19247,15 +19312,23 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, static const char * const chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - const char *pfx = alc_get_line_out_pfx(cfg, true); - hda_nid_t nid, mix; - int i, err; + const char *pfx = alc_get_line_out_pfx(spec, true); + hda_nid_t nid, mix, pin; + int i, err, noutputs; - for (i = 0; i < cfg->line_outs; i++) { + noutputs = cfg->line_outs; + if (spec->multi_ios > 0) + noutputs += spec->multi_ios; + + for (i = 0; i < noutputs; i++) { nid = spec->multiout.dac_nids[i]; if (!nid) continue; - mix = alc_auto_dac_to_mix(codec, cfg->line_out_pins[i], nid); + if (i >= cfg->line_outs) + pin = spec->multi_io[i - 1].pin; + else + pin = cfg->line_out_pins[i]; + mix = alc_auto_dac_to_mix(codec, pin, nid); if (!mix) continue; if (!pfx && i == 2) { @@ -19406,6 +19479,159 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) #define alc662_auto_init_input_src alc882_auto_init_input_src +/* + * multi-io helper + */ +static int alc_auto_fill_multi_ios(struct hda_codec *codec, + unsigned int location) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int type, i, num_pins = 0; + + for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + hda_nid_t dac; + unsigned int defcfg, caps; + if (cfg->inputs[i].type != type) + continue; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX) + continue; + if (location && get_defcfg_location(defcfg) != location) + continue; + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_OUT)) + continue; + dac = alc_auto_look_for_dac(codec, nid); + if (!dac) + continue; + spec->multi_io[num_pins].pin = nid; + spec->multi_io[num_pins].dac = dac; + num_pins++; + spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + } + } + spec->multiout.num_dacs = 1; + if (num_pins < 2) + return 0; + return num_pins; +} + +static int alc_auto_ch_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = spec->multi_ios + 1; + if (uinfo->value.enumerated.item > spec->multi_ios) + uinfo->value.enumerated.item = spec->multi_ios; + sprintf(uinfo->value.enumerated.name, "%dch", + (uinfo->value.enumerated.item + 1) * 2); + return 0; +} + +static int alc_auto_ch_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = (spec->ext_channel_count - 1) / 2; + return 0; +} + +static int alc_set_multi_io(struct hda_codec *codec, int idx, bool output) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid = spec->multi_io[idx].pin; + + if (!spec->multi_io[idx].ctl_in) + spec->multi_io[idx].ctl_in = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (output) { + snd_hda_codec_update_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_OUT); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, 0); + alc_auto_select_dac(codec, nid, spec->multi_io[idx].dac); + } else { + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_update_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->multi_io[idx].ctl_in); + } + return 0; +} + +static int alc_auto_ch_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + int i, ch; + + ch = ucontrol->value.enumerated.item[0]; + if (ch < 0 || ch > spec->multi_ios) + return -EINVAL; + if (ch == (spec->ext_channel_count - 1) / 2) + return 0; + spec->ext_channel_count = (ch + 1) * 2; + for (i = 0; i < spec->multi_ios; i++) + alc_set_multi_io(codec, i, i < ch); + spec->multiout.max_channels = spec->ext_channel_count; + return 1; +} + +static struct snd_kcontrol_new alc_auto_channel_mode_enum = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_auto_ch_mode_info, + .get = alc_auto_ch_mode_get, + .put = alc_auto_ch_mode_put, +}; + +static int alc_auto_add_multi_channel_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int location, defcfg; + int num_pins; + + if (cfg->line_outs != 1 || + cfg->line_out_type != AUTO_PIN_LINE_OUT) + return 0; + + defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]); + location = get_defcfg_location(defcfg); + + num_pins = alc_auto_fill_multi_ios(codec, location); + if (num_pins > 0) { + struct snd_kcontrol_new *knew; + + knew = alc_kcontrol_new(spec); + if (!knew) + return -ENOMEM; + *knew = alc_auto_channel_mode_enum; + knew->name = kstrdup("Channel Mode", GFP_KERNEL); + if (!knew->name) + return -ENOMEM; + + spec->multi_ios = num_pins; + spec->ext_channel_count = 2; + spec->multiout.num_dacs = num_pins + 1; + } + return 0; +} + static int alc662_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -19420,6 +19646,9 @@ static int alc662_parse_auto_config(struct hda_codec *codec) return 0; /* can't find valid BIOS pin config */ err = alc662_auto_fill_dac_nids(codec, &spec->autocfg); + if (err < 0) + return err; + err = alc_auto_add_multi_channel_mode(codec); if (err < 0) return err; err = alc662_auto_create_multi_out_ctls(codec, &spec->autocfg); -- cgit v1.2.3 From 8842c72afe9f954d9462da577a25d4a32bfe0a2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Apr 2011 18:58:17 +0100 Subject: ASoC: Allow platform drivers to have no ops structure Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 43 ++++++++++++++++++++++++------------------- 1 file changed, 24 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a6f37d45422..9dc13113fa2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -555,7 +555,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (platform->driver->ops->open) { + if (platform->driver->ops && platform->driver->ops->open) { ret = platform->driver->ops->open(substream); if (ret < 0) { printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); @@ -685,7 +685,7 @@ machine_err: codec_dai->driver->ops->shutdown(substream, codec_dai); codec_dai_err: - if (platform->driver->ops->close) + if (platform->driver->ops && platform->driver->ops->close) platform->driver->ops->close(substream); platform_err: @@ -767,7 +767,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) rtd->dai_link->ops->shutdown(substream); - if (platform->driver->ops->close) + if (platform->driver->ops && platform->driver->ops->close) platform->driver->ops->close(substream); cpu_dai->runtime = NULL; @@ -810,7 +810,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (platform->driver->ops->prepare) { + if (platform->driver->ops && platform->driver->ops->prepare) { ret = platform->driver->ops->prepare(substream); if (ret < 0) { printk(KERN_ERR "asoc: platform prepare error\n"); @@ -899,7 +899,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (platform->driver->ops->hw_params) { + if (platform->driver->ops && platform->driver->ops->hw_params) { ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { printk(KERN_ERR "asoc: platform %s hw params failed\n", @@ -952,7 +952,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) rtd->dai_link->ops->hw_free(substream); /* free any DMA resources */ - if (platform->driver->ops->hw_free) + if (platform->driver->ops && platform->driver->ops->hw_free) platform->driver->ops->hw_free(substream); /* now free hw params for the DAIs */ @@ -980,7 +980,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (platform->driver->ops->trigger) { + if (platform->driver->ops && platform->driver->ops->trigger) { ret = platform->driver->ops->trigger(substream, cmd); if (ret < 0) return ret; @@ -1009,7 +1009,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; - if (platform->driver->ops->pointer) + if (platform->driver->ops && platform->driver->ops->pointer) offset = platform->driver->ops->pointer(substream); if (cpu_dai->driver->ops->delay) @@ -2125,13 +2125,15 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + if (platform->driver->ops) { + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; + } if (playback) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); @@ -2139,10 +2141,13 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - ret = platform->driver->pcm_new(rtd->card->snd_card, codec_dai, pcm); - if (ret < 0) { - printk(KERN_ERR "asoc: platform pcm constructor failed\n"); - return ret; + if (platform->driver->pcm_new) { + ret = platform->driver->pcm_new(rtd->card->snd_card, + codec_dai, pcm); + if (ret < 0) { + pr_err("asoc: platform pcm constructor failed\n"); + return ret; + } } pcm->private_free = platform->driver->pcm_free; -- cgit v1.2.3 From 848dd8beef44df18f2eb61e00981b0692adb801b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Apr 2011 18:16:32 +0100 Subject: ASoC: Add more natural support for no-DMA DAIs Since we can now support multiple platforms allow machines to not specify a platform in a DAI link. Since the rest of the code requires that we have a struct device for all objects we do this by substituting in a dummy device that we register automatically. Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 16 ++++++++++----- sound/soc/soc-utils.c | 55 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 66 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9dc13113fa2..16be3e5cf9e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1299,6 +1299,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) struct snd_soc_codec *codec; struct snd_soc_platform *platform; struct snd_soc_dai *codec_dai, *cpu_dai; + const char *platform_name; if (rtd->complete) return 1; @@ -1351,13 +1352,18 @@ find_codec: dai_link->codec_name); find_platform: - /* do we already have the CODEC DAI for this link ? */ - if (rtd->platform) { + /* do we need a platform? */ + if (rtd->platform) goto out; - } - /* no, then find CPU DAI from registered DAIs*/ + + /* if there's no platform we match on the empty platform */ + platform_name = dai_link->platform_name; + if (!platform_name) + platform_name = "snd-soc-dummy"; + + /* no, then find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { - if (!strcmp(platform->name, dai_link->platform_name)) { + if (!strcmp(platform->name, platform_name)) { rtd->platform = platform; goto out; } diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 3f45e6a439b..28657914088 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -13,6 +13,7 @@ * option) any later version. */ +#include #include #include #include @@ -55,3 +56,57 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) return ret; } EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); + +static struct snd_soc_platform_driver dummy_platform; + +static __devinit int snd_soc_dummy_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &dummy_platform); +} + +static __devexit int snd_soc_dummy_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver soc_dummy_driver = { + .driver = { + .name = "snd-soc-dummy", + .owner = THIS_MODULE, + }, + .probe = snd_soc_dummy_probe, + .remove = __devexit_p(snd_soc_dummy_remove), +}; + +static struct platform_device *soc_dummy_dev; + +static int __init snd_soc_util_init(void) +{ + int ret; + + soc_dummy_dev = platform_device_alloc("snd-soc-dummy", -1); + if (!soc_dummy_dev) + return -ENOMEM; + + ret = platform_device_add(soc_dummy_dev); + if (ret != 0) { + platform_device_put(soc_dummy_dev); + return ret; + } + + ret = platform_driver_register(&soc_dummy_driver); + if (ret != 0) + platform_device_unregister(soc_dummy_dev); + + return ret; +} +module_init(snd_soc_util_init); + +static void __exit snd_soc_util_exit(void) +{ + platform_device_unregister(soc_dummy_dev); + platform_driver_unregister(&soc_dummy_driver); +} +module_exit(snd_soc_util_exit); -- cgit v1.2.3 From b864a8c9dd93f08ccaa706e075810e9398e25680 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Apr 2011 18:17:01 +0100 Subject: ASoC: Don't specify the DMA driver for Speyside baseband link Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/speyside.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 337855a3f2f..360a333cb7c 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -183,7 +183,6 @@ static struct snd_soc_dai_link speyside_dai[] = { .cpu_dai_name = "wm8915-aif2", .codec_dai_name = "wm1250-ev1", .codec_name = "wm1250-ev1.1-0027", - .platform_name = "samsung-audio", .ops = &speyside_ops, .ignore_suspend = 1, }, -- cgit v1.2.3 From 91a5fca4b1987324f829efeff3bc5efb2ce6e752 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 27 Apr 2011 18:34:31 +0200 Subject: ASoC: Add dapm_find_widget helper This patch adds a helper function for searching DAPM widgets by name. This allows to streamline functions which operate on widgets by name. It also allows to get rid of copy'n'pasted code which was added to fallback to widgets from other contexts if the widget was not found in the current context. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 115 ++++++++++++++++++++------------------------------- 1 file changed, 45 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 856471703bd..378f08adee5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1458,40 +1458,43 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) } } -static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, - const char *pin, int status) +static struct snd_soc_dapm_widget *dapm_find_widget( + struct snd_soc_dapm_context *dapm, const char *pin, + bool search_other_contexts) { struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *fallback = NULL; list_for_each_entry(w, &dapm->card->widgets, list) { - if (w->dapm != dapm) - continue; if (!strcmp(w->name, pin)) { - dev_dbg(w->dapm->dev, "dapm: pin %s = %d\n", - pin, status); - w->connected = status; - /* Allow disabling of forced pins */ - if (status == 0) - w->force = 0; - return 0; + if (w->dapm == dapm) + return w; + else + fallback = w; } } - /* Try again in other contexts */ - list_for_each_entry(w, &dapm->card->widgets, list) { - if (!strcmp(w->name, pin)) { - dev_dbg(w->dapm->dev, "dapm: pin %s = %d\n", - pin, status); - w->connected = status; - /* Allow disabling of forced pins */ - if (status == 0) - w->force = 0; - return 0; - } + if (search_other_contexts) + return fallback; + + return NULL; +} + +static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) +{ + struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + + if (!w) { + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); + return -EINVAL; } - dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); - return -EINVAL; + w->connected = status; + if (status == 0) + w->force = 0; + + return 0; } /** @@ -2316,33 +2319,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); - list_for_each_entry(w, &dapm->card->widgets, list) { - if (w->dapm != dapm) - continue; - if (!strcmp(w->name, pin)) { - dev_dbg(w->dapm->dev, - "dapm: force enable pin %s\n", pin); - w->connected = 1; - w->force = 1; - return 0; - } + if (!w) { + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); + return -EINVAL; } - /* Try again with other contexts */ - list_for_each_entry(w, &dapm->card->widgets, list) { - if (!strcmp(w->name, pin)) { - dev_dbg(w->dapm->dev, - "dapm: force enable pin %s\n", pin); - w->connected = 1; - w->force = 1; - return 0; - } - } + dev_dbg(w->dapm->dev, "dapm: force enable pin %s\n", pin); + w->connected = 1; + w->force = 1; - dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); - return -EINVAL; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin); @@ -2394,20 +2382,10 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm, const char *pin) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); - list_for_each_entry(w, &dapm->card->widgets, list) { - if (w->dapm != dapm) - continue; - if (!strcmp(w->name, pin)) - return w->connected; - } - - /* Try again in other contexts */ - list_for_each_entry(w, &dapm->card->widgets, list) { - if (!strcmp(w->name, pin)) - return w->connected; - } + if (w) + return w->connected; return 0; } @@ -2427,19 +2405,16 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, false); - list_for_each_entry(w, &dapm->card->widgets, list) { - if (w->dapm != dapm) - continue; - if (!strcmp(w->name, pin)) { - w->ignore_suspend = 1; - return 0; - } + if (!w) { + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); + return -EINVAL; } - dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); - return -EINVAL; + w->ignore_suspend = 1; + + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); -- cgit v1.2.3 From 6be449e53dc3593ec69308d972a9fab852fb9bf1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Apr 2011 16:04:37 +0100 Subject: ASoC: Implement WM8962 ADC high pass filter configuration Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7949f89a8f8..f90ae427242 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2039,6 +2039,13 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol, return 0; } +static const char *cap_hpf_mode_text[] = { + "Hi-fi", "Application" +}; + +static const struct soc_enum cap_hpf_mode = + SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text); + static const struct snd_kcontrol_new wm8962_snd_controls[] = { SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1), @@ -2064,6 +2071,9 @@ SOC_DOUBLE_R("Capture Switch", WM8962_LEFT_INPUT_VOLUME, WM8962_RIGHT_INPUT_VOLUME, 7, 1, 1), SOC_DOUBLE_R("Capture ZC Switch", WM8962_LEFT_INPUT_VOLUME, WM8962_RIGHT_INPUT_VOLUME, 6, 1, 1), +SOC_SINGLE("Capture HPF Switch", WM8962_ADC_DAC_CONTROL_1, 0, 1, 1), +SOC_ENUM("Capture HPF Mode", cap_hpf_mode), +SOC_SINGLE("Capture HPF Cutoff", WM8962_ADC_DAC_CONTROL_2, 7, 7, 0), SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, WM8962_DAC_DSP_MIXING_2, 4, 12, 0, st_tlv), -- cgit v1.2.3 From 59bb7f0eebe69aa32a5c7917a23a7da1c5667d73 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 09:58:43 +0200 Subject: ALSA: usb-audio - Don't expose broken dB ranges Some crappy USB-audio devices give broken dB ranges, e.g. both min and max are 0dB. This confuses the volume control that prefers dB expression such as alsactl or PulseAudio. In such a case, it's much better not to expose the broken dB information. Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5e477571660..c8c28cd7553 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1097,11 +1097,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, append_ctl_name(kctl, control == UAC_FU_MUTE ? " Switch" : " Volume"); if (control == UAC_FU_VOLUME) { - kctl->tlv.c = mixer_vol_tlv; - kctl->vd[0].access |= - SNDRV_CTL_ELEM_ACCESS_TLV_READ | - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; check_mapped_dB(map, cval); + if (cval->dBmin < cval->dBmax) { + kctl->tlv.c = mixer_vol_tlv; + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } } break; -- cgit v1.2.3 From fb257897bf20c5f0e1df584bb5b874e811651263 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Apr 2011 10:57:54 +0100 Subject: ASoC: Work around allmodconfig failure Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 4 ++++ sound/soc/soc-utils.c | 6 ++---- 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16be3e5cf9e..a823654ef36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3868,12 +3868,16 @@ static int __init snd_soc_init(void) pr_warn("ASoC: Failed to create platform list debugfs file\n"); #endif + snd_soc_util_init(); + return platform_driver_register(&soc_driver); } module_init(snd_soc_init); static void __exit snd_soc_exit(void) { + snd_soc_util_exit(); + #ifdef CONFIG_DEBUG_FS debugfs_remove_recursive(snd_soc_debugfs_root); #endif diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 28657914088..ec921ec99c2 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -82,7 +82,7 @@ static struct platform_driver soc_dummy_driver = { static struct platform_device *soc_dummy_dev; -static int __init snd_soc_util_init(void) +int __init snd_soc_util_init(void) { int ret; @@ -102,11 +102,9 @@ static int __init snd_soc_util_init(void) return ret; } -module_init(snd_soc_util_init); -static void __exit snd_soc_util_exit(void) +void __exit snd_soc_util_exit(void) { platform_device_unregister(soc_dummy_dev); platform_driver_unregister(&soc_dummy_driver); } -module_exit(snd_soc_util_exit); -- cgit v1.2.3 From 3784019af3da88014c7222f2cec2e5afff8a6b4f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Apr 2011 18:17:27 +0100 Subject: ASoC: Don't specify the DMA driver for OpenMoko baseband link Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/neo1973_wm8753.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 45223097563..16152ed0864 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -432,7 +432,6 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .platform_name = "samsung-audio", .cpu_dai_name = "dfbmcs320-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753-codec.0-001a", -- cgit v1.2.3 From 9b1b937c77f94f8c94874351a2d5ba92cd99f1ec Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Apr 2011 18:25:34 +0100 Subject: ASoC: Don't specify the DMA driver for Goni baseband link Signed-off-by: Mark Brown Acked-by: Jassi Brar Acked-by: Liam Girdwood --- sound/soc/samsung/goni_wm8994.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 0e80daee8b6..eb6d72ed55a 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -246,7 +246,6 @@ static struct snd_soc_dai_link goni_dai[] = { .stream_name = "Voice", .cpu_dai_name = "goni-voice-dai", .codec_dai_name = "wm8994-aif2", - .platform_name = "samsung-audio", .codec_name = "wm8994-codec.0-001a", .ops = &goni_voice_ops, }, -- cgit v1.2.3 From d922b51dabbe9b6aeee44e2e3a605323591e3707 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 12:18:53 +0200 Subject: ALSA: hda - Consolidate default automute functions for Realtek There are two entry points for the headphone automute functions for Realtek, alc_automute_amp() and alc_automute_pin(). These call the same function in the end, so we can basically consolidate these with a flag in spec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 260 ++++++++++++++++++++++++++---------------- 1 file changed, 163 insertions(+), 97 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c4d51a220b..5715b6f3e2a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -305,6 +305,11 @@ struct alc_multi_io { unsigned int ctl_in; /* cached input-pin control value */ }; +enum { + ALC_AUTOMUTE_PIN, + ALC_AUTOMUTE_AMP, +}; + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -388,11 +393,16 @@ struct alc_spec { unsigned int jack_present: 1; unsigned int master_sw: 1; unsigned int auto_mic:1; + unsigned int automute:1; /* HP automute enabled */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */ unsigned int single_input_src:1; + + /* auto-mute control */ + int automute_mode; + int init_amp; int codec_variant; /* flag for other variants */ @@ -1062,13 +1072,16 @@ static int alc_init_jacks(struct hda_codec *codec) return 0; } -static void alc_automute_speaker(struct hda_codec *codec, int pinctl) +static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int mute; hda_nid_t nid; int i; + if (!spec->automute) + return; + spec->jack_present = 0; for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { nid = spec->autocfg.hp_pins[i]; @@ -1084,7 +1097,7 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl) nid = spec->autocfg.speaker_pins[i]; if (!nid) break; - if (pinctl) { + if (spec->automute_mode == ALC_AUTOMUTE_PIN) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->jack_present ? 0 : PIN_OUT); @@ -1095,11 +1108,6 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl) } } -static void alc_automute_pin(struct hda_codec *codec) -{ - alc_automute_speaker(codec, 1); -} - static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid) { @@ -1195,7 +1203,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) res >>= 26; switch (res) { case ALC880_HP_EVENT: - alc_automute_pin(codec); + alc_hp_automute(codec); break; case ALC880_MIC_EVENT: alc_mic_automute(codec); @@ -1205,7 +1213,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) static void alc_inithook(struct hda_codec *codec) { - alc_automute_pin(codec); + alc_hp_automute(codec); alc_mic_automute(codec); } @@ -1374,6 +1382,8 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, cfg->hp_pins[i], 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } spec->unsol_event = alc_sku_unsol_event; } @@ -1971,22 +1981,6 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { {} }; -static void alc_automute_amp(struct hda_codec *codec) -{ - alc_automute_speaker(codec, 0); -} - -static void alc_automute_amp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if (codec->vendor_id == 0x10ec0880) - res >>= 28; - else - res >>= 26; - if (res == ALC880_HP_EVENT) - alc_automute_amp(codec); -} - static void alc889_automute_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1997,12 +1991,14 @@ static void alc889_automute_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x17; spec->autocfg.speaker_pins[3] = 0x19; spec->autocfg.speaker_pins[4] = 0x1a; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc889_intel_init_hook(struct hda_codec *codec) { alc889_coef_init(codec); - alc_automute_amp(codec); + alc_hp_automute(codec); } static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) @@ -2013,6 +2009,8 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* @@ -2295,6 +2293,8 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) @@ -2305,6 +2305,8 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) @@ -2315,6 +2317,8 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) @@ -2325,6 +2329,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* @@ -3400,11 +3406,13 @@ static void alc880_uniwill_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc880_uniwill_init_hook(struct hda_codec *codec) { - alc_automute_amp(codec); + alc_hp_automute(codec); alc88x_simple_mic_automute(codec); } @@ -3419,7 +3427,7 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, alc88x_simple_mic_automute(codec); break; default: - alc_automute_amp_unsol_event(codec, res); + alc_sku_unsol_event(codec, res); break; } } @@ -3430,6 +3438,8 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -3454,7 +3464,7 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, if ((res >> 28) == ALC880_DCVOL_EVENT) alc880_uniwill_p53_dcvol_automute(codec); else - alc_automute_amp_unsol_event(codec, res); + alc_sku_unsol_event(codec, res); } /* @@ -3698,6 +3708,8 @@ static void alc880_lg_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* @@ -3782,6 +3794,8 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct snd_kcontrol_new alc880_medion_rim_mixer[] = { @@ -3829,7 +3843,7 @@ static struct hda_verb alc880_medion_rim_init_verbs[] = { static void alc880_medion_rim_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - alc_automute_amp(codec); + alc_hp_automute(codec); /* toggle EAPD */ if (spec->jack_present) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); @@ -3853,6 +3867,8 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -4806,7 +4822,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_f1734_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, .setup = alc880_uniwill_p53_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, @@ -4897,7 +4913,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, .setup = alc880_uniwill_p53_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC880_FUJITSU] = { .mixers = { alc880_fujitsu_mixer }, @@ -4912,7 +4928,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, .setup = alc880_uniwill_p53_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC880_CLEVO] = { .mixers = { alc880_three_stack_mixer }, @@ -4937,9 +4953,9 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_lg_ch_modes, .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc880_lg_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, #ifdef CONFIG_SND_HDA_POWER_SAVE .loopbacks = alc880_lg_loopbacks, #endif @@ -4954,9 +4970,9 @@ static struct alc_config_preset alc880_presets[] = { .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes), .channel_mode = alc880_lg_lw_modes, .input_mux = &alc880_lg_lw_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc880_lg_lw_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC880_MEDION_RIM] = { .mixers = { alc880_medion_rim_mixer }, @@ -8629,6 +8645,8 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } #define alc885_mb5_setup alc885_imac24_setup @@ -8641,6 +8659,8 @@ static void alc885_mba21_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8651,6 +8671,8 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc885_imac91_setup(struct hda_codec *codec) @@ -8660,6 +8682,8 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct hda_verb alc882_targa_verbs[] = { @@ -8681,7 +8705,7 @@ static struct hda_verb alc882_targa_verbs[] = { static void alc882_targa_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - alc_automute_amp(codec); + alc_hp_automute(codec); snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, spec->jack_present ? 1 : 3); } @@ -8692,6 +8716,8 @@ static void alc882_targa_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) @@ -8779,7 +8805,7 @@ static void alc885_macpro_init_hook(struct hda_codec *codec) static void alc885_imac24_init_hook(struct hda_codec *codec) { alc885_macpro_init_hook(codec); - alc_automute_amp(codec); + alc_hp_automute(codec); } /* @@ -9137,6 +9163,8 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1a; spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { @@ -9289,6 +9317,8 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct hda_verb alc883_mitac_verbs[] = { @@ -9453,6 +9483,8 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct hda_verb alc888_3st_hp_verbs[] = { @@ -9541,6 +9573,8 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* toggle speaker-output according to the hp-jack state */ @@ -9553,11 +9587,13 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc883_clevo_m720_init_hook(struct hda_codec *codec) { - alc_automute_amp(codec); + alc_hp_automute(codec); alc88x_simple_mic_automute(codec); } @@ -9569,7 +9605,7 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, alc88x_simple_mic_automute(codec); break; default: - alc_automute_amp_unsol_event(codec, res); + alc_sku_unsol_event(codec, res); break; } } @@ -9581,6 +9617,8 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc883_haier_w66_setup(struct hda_codec *codec) @@ -9589,6 +9627,8 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) @@ -9626,6 +9666,8 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct hda_verb alc883_acer_eapd_verbs[] = { @@ -9655,6 +9697,8 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc888_lenovo_sky_setup(struct hda_codec *codec) @@ -9667,6 +9711,8 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc883_vaiott_setup(struct hda_codec *codec) @@ -9676,6 +9722,8 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct hda_verb alc888_asus_m90v_verbs[] = { @@ -9701,6 +9749,8 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->ext_mic.mux_idx = 0; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct hda_verb alc888_asus_eee1601_verbs[] = { @@ -9722,7 +9772,7 @@ static void alc883_eee1601_inithook(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - alc_automute_pin(codec); + alc_hp_automute(codec); } static struct hda_verb alc889A_mb31_verbs[] = { @@ -10043,9 +10093,9 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc885_mba21_ch_modes, .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), .input_mux = &alc882_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc885_mba21_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC885_MBP3] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, @@ -10059,9 +10109,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc885_mbp3_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC885_MB5] = { .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, @@ -10074,9 +10124,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc885_mb5_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC885_MACMINI3] = { .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, @@ -10089,9 +10139,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &macmini3_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc885_macmini3_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, @@ -10115,7 +10165,7 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc885_imac24_setup, .init_hook = alc885_imac24_init_hook, }, @@ -10130,9 +10180,9 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc889A_imac91_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc885_imac91_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, @@ -10148,7 +10198,7 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_3ST_6ch_modes, .need_dac_fix = 1, .input_mux = &alc882_capture_source, - .unsol_event = alc882_targa_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc882_targa_setup, .init_hook = alc882_targa_automute, }, @@ -10242,8 +10292,8 @@ static struct alc_config_preset alc882_presets[] = { .capsrc_nids = alc889_capsrc_nids, .input_mux = &alc889_capture_source, .setup = alc889_automute_setup, - .init_hook = alc_automute_amp, - .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc_hp_automute, + .unsol_event = alc_sku_unsol_event, .need_dac_fix = 1, }, [ALC889_INTEL] = { @@ -10263,7 +10313,7 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc889_capture_source, .setup = alc889_automute_setup, .init_hook = alc889_intel_init_hook, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .need_dac_fix = 1, }, [ALC883_6ST_DIG] = { @@ -10352,9 +10402,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc883_acer_aspire_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_ACER_ASPIRE_4930G] = { .mixers = { alc888_acer_aspire_4930g_mixer, @@ -10374,9 +10424,9 @@ static struct alc_config_preset alc882_presets[] = { .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc888_acer_aspire_4930g_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_ACER_ASPIRE_6530G] = { .mixers = { alc888_acer_aspire_6530_mixer }, @@ -10393,9 +10443,9 @@ static struct alc_config_preset alc882_presets[] = { .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_acer_aspire_6530_sources, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc888_acer_aspire_6530g_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc889_acer_aspire_8930g_mixer, @@ -10416,9 +10466,9 @@ static struct alc_config_preset alc882_presets[] = { .num_mux_defs = ARRAY_SIZE(alc889_capture_sources), .input_mux = alc889_capture_sources, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc889_acer_aspire_8930g_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, #ifdef CONFIG_SND_HDA_POWER_SAVE .power_hook = alc_power_eapd, #endif @@ -10439,9 +10489,9 @@ static struct alc_config_preset alc882_presets[] = { .need_dac_fix = 1, .const_channel_count = 6, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc888_acer_aspire_7730g_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, @@ -10468,9 +10518,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc883_medion_wim2160_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, @@ -10532,9 +10582,9 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .need_dac_fix = 1, .input_mux = &alc883_lenovo_nb0763_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc883_lenovo_nb0763_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_LENOVO_MS7195_DIG] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -10558,9 +10608,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc883_haier_w66_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_3ST_HP] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -10571,9 +10621,9 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc888_3st_hp_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, @@ -10585,9 +10635,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc888_6st_dell_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC883_MITAC] = { .mixers = { alc883_mitac_mixer }, @@ -10597,9 +10647,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc883_mitac_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC883_FUJITSU_PI2515] = { .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, @@ -10611,9 +10661,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_fujitsu_pi2515_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc883_2ch_fujitsu_pi2515_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_FUJITSU_XA3530] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, @@ -10630,9 +10680,9 @@ static struct alc_config_preset alc882_presets[] = { .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc888_fujitsu_xa3530_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_LENOVO_SKY] = { .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer }, @@ -10644,9 +10694,9 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc883_sixstack_modes, .need_dac_fix = 1, .input_mux = &alc883_lenovo_sky_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc888_lenovo_sky_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC888_ASUS_M90V] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -10714,9 +10764,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc883_vaiott_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, }; @@ -11347,6 +11397,8 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { @@ -11580,6 +11632,8 @@ static void alc262_tyan_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11723,6 +11777,8 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) spec->int_mic.pin = 0x12; spec->int_mic.mux_idx = 9; spec->auto_mic = 1; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } /* @@ -12933,9 +12989,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_tyan_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, }; @@ -13349,6 +13405,8 @@ static void alc268_dell_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { @@ -13379,6 +13437,8 @@ static void alc267_quanta_il1_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } /* @@ -16878,11 +16938,13 @@ static void alc861vd_lenovo_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc861vd_lenovo_init_hook(struct hda_codec *codec) { - alc_automute_amp(codec); + alc_hp_automute(codec); alc88x_simple_mic_automute(codec); } @@ -16894,7 +16956,7 @@ static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, alc88x_simple_mic_automute(codec); break; default: - alc_automute_amp_unsol_event(codec, res); + alc_sku_unsol_event(codec, res); break; } } @@ -16951,6 +17013,8 @@ static void alc861vd_dallas_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -17075,9 +17139,9 @@ static struct alc_config_preset alc861vd_presets[] = { .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_dallas_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc861vd_dallas_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC861VD_HP] = { .mixers = { alc861vd_hp_mixer }, @@ -17088,9 +17152,9 @@ static struct alc_config_preset alc861vd_presets[] = { .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_hp_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc861vd_dallas_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_hp_automute, }, [ALC660VD_ASUS_V1S] = { .mixers = { alc861vd_lenovo_mixer }, @@ -20081,20 +20145,22 @@ static void alc680_base_setup(struct hda_codec *codec) spec->autocfg.inputs[0].type = AUTO_PIN_MIC; spec->autocfg.inputs[1].pin = 0x19; spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc680_unsol_event(struct hda_codec *codec, unsigned int res) { if ((res >> 26) == ALC880_HP_EVENT) - alc_automute_amp(codec); + alc_hp_automute(codec); if ((res >> 26) == ALC880_MIC_EVENT) alc680_rec_autoswitch(codec); } static void alc680_inithook(struct hda_codec *codec) { - alc_automute_amp(codec); + alc_hp_automute(codec); alc680_rec_autoswitch(codec); } -- cgit v1.2.3 From 3b8510ce972c93000ea65122bab3a278074eda98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 14:03:24 +0200 Subject: ALSA: hda - Add common automute support for mxier-amp on/off for Reatek Some models do mute on/off the connected mixer widget for the automatic muting, instead of controlling the pin widget itself. This patch adds the implementation of such type of auto-mute in the common helper function, and reduces the redundant codes for each model preset. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 651 +++++++++++++----------------------------- 1 file changed, 194 insertions(+), 457 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5715b6f3e2a..af9bb96d684 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -306,8 +306,9 @@ struct alc_multi_io { }; enum { - ALC_AUTOMUTE_PIN, - ALC_AUTOMUTE_AMP, + ALC_AUTOMUTE_PIN, /* change the pin control */ + ALC_AUTOMUTE_AMP, /* mute/unmute the pin AMP */ + ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; struct alc_spec { @@ -402,6 +403,7 @@ struct alc_spec { /* auto-mute control */ int automute_mode; + hda_nid_t automute_mixer_nid[AUTO_CFG_MAX_OUTS]; int init_amp; int codec_variant; /* flag for other variants */ @@ -1097,13 +1099,25 @@ static void alc_hp_automute(struct hda_codec *codec) nid = spec->autocfg.speaker_pins[i]; if (!nid) break; - if (spec->automute_mode == ALC_AUTOMUTE_PIN) { + switch (spec->automute_mode) { + case ALC_AUTOMUTE_PIN: snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->jack_present ? 0 : PIN_OUT); - } else { + break; + case ALC_AUTOMUTE_AMP: snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, mute); + break; + case ALC_AUTOMUTE_MIXER: + nid = spec->automute_mixer_nid[i]; + if (!nid) + break; + snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, + HDA_AMP_MUTE, mute); + snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 1, + HDA_AMP_MUTE, mute); + break; } } } @@ -13331,36 +13345,14 @@ static void alc268_acer_init_hook(struct hda_codec *codec) alc268_acer_automute(codec, 1); } -/* toggle speaker-output according to the hp-jack state */ -static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x15); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); -} - -static void alc268_acer_lc_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc268_aspire_one_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - static void alc268_acer_lc_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0f; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; @@ -13368,12 +13360,6 @@ static void alc268_acer_lc_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc268_acer_lc_init_hook(struct hda_codec *codec) -{ - alc268_aspire_one_speaker_automute(codec); - alc_mic_automute(codec); -} - static struct snd_kcontrol_new alc268_dell_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), @@ -14032,9 +14018,9 @@ static struct alc_config_preset alc268_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, - .unsol_event = alc268_acer_lc_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc268_acer_lc_setup, - .init_hook = alc268_acer_lc_init_hook, + .init_hook = alc_inithook, }, [ALC268_DELL] = { .mixers = { alc268_dell_mixer, alc268_beep_mixer, @@ -14374,15 +14360,7 @@ static struct hda_verb alc269_lifebook_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x15); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); + alc_hp_automute(codec); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -14395,34 +14373,8 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) AC_VERB_SET_PROC_COEF, 0x480); } -/* toggle speaker-output according to the hp-jacks state */ -static void alc269_lifebook_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - /* Check laptop headphone socket */ - present = snd_hda_jack_detect(codec, 0x15); - - /* Check port replicator headphone socket */ - present |= snd_hda_jack_detect(codec, 0x1a); - - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); - - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 0x0c); - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_PROC_COEF, 0x680); - - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 0x0c); - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_PROC_COEF, 0x480); -} +#define alc269_lifebook_speaker_automute \ + alc269_quanta_fl1_speaker_automute static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) { @@ -14471,6 +14423,9 @@ static void alc269_quanta_fl1_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x19; @@ -14484,6 +14439,17 @@ static void alc269_quanta_fl1_init_hook(struct hda_codec *codec) alc_mic_automute(codec); } +static void alc269_lifebook_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.hp_pins[1] = 0x1a; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; +} + static void alc269_lifebook_init_hook(struct hda_codec *codec) { alc269_lifebook_speaker_automute(codec); @@ -14547,42 +14513,14 @@ static struct hda_verb alc271_acer_dmic_verbs[] = { { } }; -/* toggle speaker-output according to the hp-jack state */ -static void alc269_speaker_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - unsigned int nid = spec->autocfg.hp_pins[0]; - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, nid); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); - snd_hda_input_jack_report(codec, nid); -} - -/* unsolicited event for HP jack sensing */ -static void alc269_laptop_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc269_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - static void alc269_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x19; @@ -14595,6 +14533,9 @@ static void alc269_laptop_dmic_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; @@ -14607,6 +14548,9 @@ static void alc269vb_laptop_amic_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x19; @@ -14619,6 +14563,9 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; @@ -14626,12 +14573,6 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc269_laptop_inithook(struct hda_codec *codec) -{ - alc269_speaker_automute(codec); - alc_mic_automute(codec); -} - /* * generic initialization of ADC, input mixers and output mixers */ @@ -15168,9 +15109,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_laptop_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc269_laptop_amic_setup, - .init_hook = alc269_laptop_inithook, + .init_hook = alc_inithook, }, [ALC269_DMIC] = { .mixers = { alc269_laptop_mixer }, @@ -15182,9 +15123,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_laptop_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc269_laptop_dmic_setup, - .init_hook = alc269_laptop_inithook, + .init_hook = alc_inithook, }, [ALC269VB_AMIC] = { .mixers = { alc269vb_laptop_mixer }, @@ -15196,9 +15137,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_laptop_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc269vb_laptop_amic_setup, - .init_hook = alc269_laptop_inithook, + .init_hook = alc_inithook, }, [ALC269VB_DMIC] = { .mixers = { alc269vb_laptop_mixer }, @@ -15210,9 +15151,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_laptop_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc269vb_laptop_dmic_setup, - .init_hook = alc269_laptop_inithook, + .init_hook = alc_inithook, }, [ALC269_FUJITSU] = { .mixers = { alc269_fujitsu_mixer }, @@ -15224,9 +15165,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .unsol_event = alc269_laptop_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc269_laptop_dmic_setup, - .init_hook = alc269_laptop_inithook, + .init_hook = alc_inithook, }, [ALC269_LIFEBOOK] = { .mixers = { alc269_lifebook_mixer }, @@ -15238,6 +15179,7 @@ static struct alc_config_preset alc269_presets[] = { .channel_mode = alc269_modes, .input_mux = &alc269_capture_source, .unsol_event = alc269_lifebook_unsol_event, + .setup = alc269_lifebook_setup, .init_hook = alc269_lifebook_init_hook, }, [ALC271_ACER] = { @@ -18348,164 +18290,14 @@ static void alc662_eeepc_ep20_setup(struct hda_codec *codec) #define alc662_eeepc_ep20_inithook alc262_hippo_master_update -static void alc663_m51va_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x21); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); -} - -static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x21); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); -} - -static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x15); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - HDA_AMP_MUTE, bits); -} - -static void alc662_f5z_speaker_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x1b); - bits = present ? 0 : PIN_OUT; - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, bits); -} - -static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec) -{ - unsigned int present1, present2; - - present1 = snd_hda_jack_detect(codec, 0x21); - present2 = snd_hda_jack_detect(codec, 0x15); - - if (present1 || present2) { - snd_hda_codec_write_cache(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } else { - snd_hda_codec_write_cache(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - } -} - -static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) -{ - unsigned int present1, present2; - - present1 = snd_hda_jack_detect(codec, 0x1b); - present2 = snd_hda_jack_detect(codec, 0x15); - - if (present1 || present2) { - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - HDA_AMP_MUTE, 0); - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - HDA_AMP_MUTE, 0); - } -} - -static void alc663_two_hp_m7_speaker_automute(struct hda_codec *codec) -{ - unsigned int present1, present2; - - present1 = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - present2 = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - - if (present1 || present2) { - snd_hda_codec_write_cache(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write_cache(codec, 0x17, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } else { - snd_hda_codec_write_cache(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - snd_hda_codec_write_cache(codec, 0x17, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - } -} - -static void alc663_two_hp_m8_speaker_automute(struct hda_codec *codec) -{ - unsigned int present1, present2; - - present1 = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - present2 = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - - if (present1 || present2) { - snd_hda_codec_write_cache(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write_cache(codec, 0x17, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } else { - snd_hda_codec_write_cache(codec, 0x14, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - snd_hda_codec_write_cache(codec, 0x17, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - } -} - -static void alc663_m51va_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_m51va_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - static void alc663_m51va_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; @@ -18513,18 +18305,15 @@ static void alc663_m51va_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc663_m51va_inithook(struct hda_codec *codec) -{ - alc663_m51va_speaker_automute(codec); - alc_mic_automute(codec); -} - /* ***************** Mode1 ******************************/ -#define alc663_mode1_unsol_event alc663_m51va_unsol_event - static void alc663_mode1_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x19; @@ -18532,156 +18321,122 @@ static void alc663_mode1_setup(struct hda_codec *codec) spec->auto_mic = 1; } -#define alc663_mode1_inithook alc663_m51va_inithook - /* ***************** Mode2 ******************************/ -static void alc662_mode2_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc662_mode2_setup(struct hda_codec *codec) { - switch (res >> 26) { - case ALC880_HP_EVENT: - alc662_f5z_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } -#define alc662_mode2_setup alc663_mode1_setup - -static void alc662_mode2_inithook(struct hda_codec *codec) -{ - alc662_f5z_speaker_automute(codec); - alc_mic_automute(codec); -} /* ***************** Mode3 ******************************/ -static void alc663_mode3_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc663_mode3_setup(struct hda_codec *codec) { - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_two_hp_m1_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } -#define alc663_mode3_setup alc663_mode1_setup - -static void alc663_mode3_inithook(struct hda_codec *codec) -{ - alc663_two_hp_m1_speaker_automute(codec); - alc_mic_automute(codec); -} /* ***************** Mode4 ******************************/ -static void alc663_mode4_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc663_mode4_setup(struct hda_codec *codec) { - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_21jd_two_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute_mixer_nid[1] = 0x0e; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } -#define alc663_mode4_setup alc663_mode1_setup - -static void alc663_mode4_inithook(struct hda_codec *codec) -{ - alc663_21jd_two_speaker_automute(codec); - alc_mic_automute(codec); -} /* ***************** Mode5 ******************************/ -static void alc663_mode5_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc663_mode5_setup(struct hda_codec *codec) { - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_15jd_two_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute_mixer_nid[1] = 0x0e; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } -#define alc663_mode5_setup alc663_mode1_setup - -static void alc663_mode5_inithook(struct hda_codec *codec) -{ - alc663_15jd_two_speaker_automute(codec); - alc_mic_automute(codec); -} /* ***************** Mode6 ******************************/ -static void alc663_mode6_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc663_mode6_setup(struct hda_codec *codec) { - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_two_hp_m2_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - -#define alc663_mode6_setup alc663_mode1_setup - -static void alc663_mode6_inithook(struct hda_codec *codec) -{ - alc663_two_hp_m2_speaker_automute(codec); - alc_mic_automute(codec); + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute_mixer_nid[0] = 0x0c; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_MIXER; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } /* ***************** Mode7 ******************************/ -static void alc663_mode7_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_two_hp_m7_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - -#define alc663_mode7_setup alc663_mode1_setup - -static void alc663_mode7_inithook(struct hda_codec *codec) +static void alc663_mode7_setup(struct hda_codec *codec) { - alc663_two_hp_m7_speaker_automute(codec); - alc_mic_automute(codec); + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } /* ***************** Mode8 ******************************/ -static void alc663_mode8_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_two_hp_m8_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - -#define alc663_mode8_setup alc663_m51va_setup - -static void alc663_mode8_inithook(struct hda_codec *codec) +static void alc663_mode8_setup(struct hda_codec *codec) { - alc663_two_hp_m8_speaker_automute(codec); - alc_mic_automute(codec); + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.hp_pins[1] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x17; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 9; + spec->auto_mic = 1; } static void alc663_g71v_hp_automute(struct hda_codec *codec) @@ -18733,27 +18488,8 @@ static void alc663_g71v_inithook(struct hda_codec *codec) alc_mic_automute(codec); } -static void alc663_g50v_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_m51va_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - #define alc663_g50v_setup alc663_m51va_setup -static void alc663_g50v_inithook(struct hda_codec *codec) -{ - alc663_m51va_speaker_automute(codec); - alc_mic_automute(codec); -} - static struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -19015,9 +18751,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_m51va_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_m51va_setup, - .init_hook = alc663_m51va_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_G71V] = { .mixers = { alc663_g71v_mixer }, @@ -19042,8 +18778,9 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_m51va_unsol_event, - .init_hook = alc663_m51va_inithook, + .setup = alc663_m51va_setup, + .unsol_event = alc_sku_unsol_event, + .init_hook = alc_inithook, }, [ALC663_ASUS_G50V] = { .mixers = { alc663_g50v_mixer }, @@ -19056,9 +18793,9 @@ static struct alc_config_preset alc662_presets[] = { .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc663_capture_source, - .unsol_event = alc663_g50v_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_g50v_setup, - .init_hook = alc663_g50v_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_MODE1] = { .mixers = { alc663_m51va_mixer }, @@ -19072,9 +18809,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_mode1_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode1_setup, - .init_hook = alc663_mode1_inithook, + .init_hook = alc_inithook, }, [ALC662_ASUS_MODE2] = { .mixers = { alc662_1bjd_mixer }, @@ -19087,9 +18824,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc662_mode2_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc662_mode2_setup, - .init_hook = alc662_mode2_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_MODE3] = { .mixers = { alc663_two_hp_m1_mixer }, @@ -19103,9 +18840,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_mode3_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode3_setup, - .init_hook = alc663_mode3_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_MODE4] = { .mixers = { alc663_asus_21jd_clfe_mixer }, @@ -19119,9 +18856,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_mode4_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode4_setup, - .init_hook = alc663_mode4_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_MODE5] = { .mixers = { alc663_asus_15jd_clfe_mixer }, @@ -19135,9 +18872,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_mode5_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode5_setup, - .init_hook = alc663_mode5_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_MODE6] = { .mixers = { alc663_two_hp_m2_mixer }, @@ -19151,9 +18888,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_mode6_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode6_setup, - .init_hook = alc663_mode6_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_MODE7] = { .mixers = { alc663_mode7_mixer }, @@ -19167,9 +18904,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_mode7_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode7_setup, - .init_hook = alc663_mode7_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_MODE8] = { .mixers = { alc663_mode8_mixer }, @@ -19183,9 +18920,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_mode8_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode8_setup, - .init_hook = alc663_mode8_inithook, + .init_hook = alc_inithook, }, [ALC272_DELL] = { .mixers = { alc663_m51va_mixer }, @@ -19200,9 +18937,9 @@ static struct alc_config_preset alc662_presets[] = { .num_adc_nids = ARRAY_SIZE(alc272_adc_nids), .capsrc_nids = alc272_capsrc_nids, .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_m51va_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_m51va_setup, - .init_hook = alc663_m51va_inithook, + .init_hook = alc_inithook, }, [ALC272_DELL_ZM1] = { .mixers = { alc663_m51va_mixer }, @@ -19217,9 +18954,9 @@ static struct alc_config_preset alc662_presets[] = { .num_adc_nids = 1, .capsrc_nids = alc662_capsrc_nids, .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_m51va_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_m51va_setup, - .init_hook = alc663_m51va_inithook, + .init_hook = alc_inithook, }, [ALC272_SAMSUNG_NC10] = { .mixers = { alc272_nc10_mixer }, @@ -19231,9 +18968,9 @@ static struct alc_config_preset alc662_presets[] = { .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, /*.input_mux = &alc272_nc10_capture_source,*/ - .unsol_event = alc663_mode4_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_mode4_setup, - .init_hook = alc663_mode4_inithook, + .init_hook = alc_inithook, }, }; -- cgit v1.2.3 From e6a5e1b7094bdd5cc9ae969aff7f75fbc53517fc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 14:41:52 +0200 Subject: ALSA: hda - Add support of line-out automute for Realtek Add the common helper function and flags to support the auto-mute per line-out jack detection, and also the mute of line-out jacks. A few model-specific implementations are replaced with the common helpers. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 274 ++++++++++++++++++------------------------ 1 file changed, 120 insertions(+), 154 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index af9bb96d684..222abba9816 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -392,9 +392,12 @@ struct alc_spec { /* for pin sensing */ unsigned int sense_updated: 1; unsigned int jack_present: 1; + unsigned int line_jack_present:1; unsigned int master_sw: 1; unsigned int auto_mic:1; unsigned int automute:1; /* HP automute enabled */ + unsigned int detect_line:1; /* Line-out detection enabled */ + unsigned int automute_lines:1; /* automute line-out as well */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -1074,54 +1077,96 @@ static int alc_init_jacks(struct hda_codec *codec) return 0; } -static void alc_hp_automute(struct hda_codec *codec) +static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) { - struct alc_spec *spec = codec->spec; - unsigned int mute; - hda_nid_t nid; - int i; + int i, present = 0; - if (!spec->automute) - return; - - spec->jack_present = 0; - for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { - nid = spec->autocfg.hp_pins[i]; + for (i = 0; i < num_pins; i++) { + hda_nid_t nid = pins[i]; if (!nid) break; snd_hda_input_jack_report(codec, nid); - spec->jack_present |= snd_hda_jack_detect(codec, nid); + present |= snd_hda_jack_detect(codec, nid); } + return present; +} - mute = spec->jack_present ? HDA_AMP_MUTE : 0; - /* Toggle internal speakers muting */ - for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { - nid = spec->autocfg.speaker_pins[i]; +static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, + bool mute) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute_bits = mute ? HDA_AMP_MUTE : 0; + unsigned int pin_bits = mute ? 0 : PIN_OUT; + int i; + + for (i = 0; i < num_pins; i++) { + hda_nid_t nid = pins[i]; if (!nid) break; switch (spec->automute_mode) { case ALC_AUTOMUTE_PIN: snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - spec->jack_present ? 0 : PIN_OUT); + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_bits); break; case ALC_AUTOMUTE_AMP: snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); + HDA_AMP_MUTE, mute_bits); break; case ALC_AUTOMUTE_MIXER: nid = spec->automute_mixer_nid[i]; if (!nid) break; snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, - HDA_AMP_MUTE, mute); + HDA_AMP_MUTE, mute_bits); snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 1, - HDA_AMP_MUTE, mute); + HDA_AMP_MUTE, mute_bits); break; } } } +/* Toggle internal speakers muting */ +static void update_speakers(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), + spec->autocfg.speaker_pins, + spec->jack_present | spec->line_jack_present); + + /* toggle line-out mutes if needed, too */ + if (!spec->automute_lines) + return; + do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), + spec->autocfg.line_out_pins, + spec->jack_present); +} + +static void alc_hp_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->automute) + return; + spec->jack_present = + detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), + spec->autocfg.hp_pins); + update_speakers(codec); +} + +static void alc_line_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->automute || !spec->detect_line) + return; + spec->line_jack_present = + detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), + spec->autocfg.line_out_pins); + update_speakers(codec); +} + static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid) { @@ -1219,6 +1264,9 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) case ALC880_HP_EVENT: alc_hp_automute(codec); break; + case ALC880_FRONT_EVENT: + alc_line_automute(codec); + break; case ALC880_MIC_EVENT: alc_mic_automute(codec); break; @@ -1228,6 +1276,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) static void alc_inithook(struct hda_codec *codec) { alc_hp_automute(codec); + alc_line_automute(codec); alc_mic_automute(codec); } @@ -9551,33 +9600,15 @@ static struct hda_channel_mode alc888_3st_hp_modes[3] = { { 6, alc888_3st_hp_6ch_init }, }; -/* toggle front-jack and RCA according to the hp-jack state */ -static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) -{ - unsigned int present = snd_hda_jack_detect(codec, 0x1b); - - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -/* toggle RCA according to the front-jack state */ -static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec) +static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) { - unsigned int present = snd_hda_jack_detect(codec, 0x14); - - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} + struct alc_spec *spec = codec->spec; -static void alc883_lenovo_ms7195_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc888_lenovo_ms7195_front_automute(codec); - if ((res >> 26) == ALC880_FRONT_EVENT) - alc888_lenovo_ms7195_rca_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.line_out_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* toggle speaker-output according to the hp-jack state */ @@ -9645,31 +9676,17 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) +static void alc883_lenovo_101e_setup(struct hda_codec *codec) { - int bits = snd_hda_jack_detect(codec, 0x14) ? HDA_AMP_MUTE : 0; - - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -static void alc883_lenovo_101e_all_automute(struct hda_codec *codec) -{ - int bits = snd_hda_jack_detect(codec, 0x1b) ? HDA_AMP_MUTE : 0; - - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc883_lenovo_101e_all_automute(codec); - if ((res >> 26) == ALC880_FRONT_EVENT) - alc883_lenovo_101e_ispeaker_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.line_out_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->detect_line = 1; + spec->automute_lines = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* toggle speaker-output according to the hp-jack state */ @@ -10584,8 +10601,9 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_lenovo_101e_capture_source, - .unsol_event = alc883_lenovo_101e_unsol_event, - .init_hook = alc883_lenovo_101e_all_automute, + .setup = alc883_lenovo_101e_setup, + .unsol_event = alc_sku_unsol_event, + .init_hook = alc_inithook, }, [ALC883_LENOVO_NB0763] = { .mixers = { alc883_lenovo_nb0763_mixer }, @@ -10610,8 +10628,9 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, - .unsol_event = alc883_lenovo_ms7195_unsol_event, - .init_hook = alc888_lenovo_ms7195_front_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc888_lenovo_ms7195_setup, + .init_hook = alc_inithook, }, [ALC883_HAIER_W66] = { .mixers = { alc883_targa_2ch_mixer}, @@ -18217,39 +18236,17 @@ static struct snd_kcontrol_new alc272_auto_capture_mixer[] = { { } /* end */ }; -static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) +static void alc662_lenovo_101e_setup(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x14); - bits = present ? HDA_AMP_MUTE : 0; - - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -static void alc662_lenovo_101e_all_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x1b); - bits = present ? HDA_AMP_MUTE : 0; - - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} + struct alc_spec *spec = codec->spec; -static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc662_lenovo_101e_all_automute(codec); - if ((res >> 26) == ALC880_FRONT_EVENT) - alc662_lenovo_101e_ispeaker_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.line_out_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->detect_line = 1; + spec->automute_lines = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* unsolicited event for HP jack sensing */ @@ -18439,53 +18436,21 @@ static void alc663_mode8_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc663_g71v_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x21); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -static void alc663_g71v_front_automute(struct hda_codec *codec) +static void alc663_g71v_setup(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x15); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -static void alc663_g71v_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_g71v_hp_automute(codec); - break; - case ALC880_FRONT_EVENT: - alc663_g71v_front_automute(codec); - break; - case ALC880_MIC_EVENT: - alc_mic_automute(codec); - break; - } -} - -#define alc663_g71v_setup alc663_m51va_setup - -static void alc663_g71v_inithook(struct hda_codec *codec) -{ - alc663_g71v_front_automute(codec); - alc663_g71v_hp_automute(codec); - alc_mic_automute(codec); + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.line_out_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; + spec->detect_line = 1; + spec->automute_lines = 1; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 9; + spec->auto_mic = 1; } #define alc663_g50v_setup alc663_m51va_setup @@ -18697,8 +18662,9 @@ static struct alc_config_preset alc662_presets[] = { .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, - .unsol_event = alc662_lenovo_101e_unsol_event, - .init_hook = alc662_lenovo_101e_all_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc662_lenovo_101e_setup, + .init_hook = alc_inithook, }, [ALC662_ASUS_EEEPC_P701] = { .mixers = { alc662_eeepc_p701_mixer }, @@ -18765,9 +18731,9 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc663_g71v_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc663_g71v_setup, - .init_hook = alc663_g71v_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_H13] = { .mixers = { alc663_m51va_mixer }, -- cgit v1.2.3 From e9427969f560f664d78b2512fd8ebf44863b5072 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 15:46:07 +0200 Subject: ALSA: hda - Consolidate auto-mute with master-switch for Realtek Yet another consolidation of auto-mute functions for the devices controlling the output muts together with the master mixer switch, typically found for ALC262 machines. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 323 +++++++++++++----------------------------- 1 file changed, 99 insertions(+), 224 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 222abba9816..a97e4602741 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -393,7 +393,7 @@ struct alc_spec { unsigned int sense_updated: 1; unsigned int jack_present: 1; unsigned int line_jack_present:1; - unsigned int master_sw: 1; + unsigned int master_mute:1; unsigned int auto_mic:1; unsigned int automute:1; /* HP automute enabled */ unsigned int detect_line:1; /* Line-out detection enabled */ @@ -1092,11 +1092,11 @@ static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) } static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, - bool mute) + bool mute, bool hp_out) { struct alc_spec *spec = codec->spec; unsigned int mute_bits = mute ? HDA_AMP_MUTE : 0; - unsigned int pin_bits = mute ? 0 : PIN_OUT; + unsigned int pin_bits = mute ? 0 : (hp_out ? PIN_HP : PIN_OUT); int i; for (i = 0; i < num_pins; i++) { @@ -1133,14 +1133,15 @@ static void update_speakers(struct hda_codec *codec) do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), spec->autocfg.speaker_pins, - spec->jack_present | spec->line_jack_present); + spec->jack_present | spec->line_jack_present | + spec->master_mute, false); /* toggle line-out mutes if needed, too */ if (!spec->automute_lines) return; do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins, - spec->jack_present); + spec->jack_present | spec->master_mute, false); } static void alc_hp_automute(struct hda_codec *codec) @@ -6010,21 +6011,14 @@ static struct snd_kcontrol_new alc260_input_mixer[] = { }; /* update HP, line and mono out pins according to the master switch */ -static void alc260_hp_master_update(struct hda_codec *codec, - hda_nid_t hp, hda_nid_t line, - hda_nid_t mono) +static void alc260_hp_master_update(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int val = spec->master_sw ? PIN_HP : 0; - /* change HP and line-out pins */ - snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - val); - snd_hda_codec_write(codec, line, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - val); - /* mono (speaker) depending on the HP jack sense */ - val = (val && !spec->jack_present) ? PIN_OUT : 0; - snd_hda_codec_write(codec, mono, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - val); + + /* change HP pins */ + do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), + spec->autocfg.hp_pins, spec->master_mute, true); + update_speakers(codec); } static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol, @@ -6032,7 +6026,7 @@ static int alc260_hp_master_sw_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - *ucontrol->value.integer.value = spec->master_sw; + *ucontrol->value.integer.value = !spec->master_mute; return 0; } @@ -6041,16 +6035,12 @@ static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - int val = !!*ucontrol->value.integer.value; - hda_nid_t hp, line, mono; + int val = !*ucontrol->value.integer.value; - if (val == spec->master_sw) + if (val == spec->master_mute) return 0; - spec->master_sw = val; - hp = (kcontrol->private_value >> 16) & 0xff; - line = (kcontrol->private_value >> 8) & 0xff; - mono = kcontrol->private_value & 0xff; - alc260_hp_master_update(codec, hp, line, mono); + spec->master_mute = val; + alc260_hp_master_update(codec); return 1; } @@ -6062,7 +6052,6 @@ static struct snd_kcontrol_new alc260_hp_output_mixer[] = { .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, - .private_value = (0x0f << 16) | (0x10 << 8) | 0x11 }, HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), @@ -6079,18 +6068,15 @@ static struct hda_verb alc260_hp_unsol_verbs[] = { {}, }; -static void alc260_hp_automute(struct hda_codec *codec) +static void alc260_hp_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->jack_present = snd_hda_jack_detect(codec, 0x10); - alc260_hp_master_update(codec, 0x0f, 0x10, 0x11); -} - -static void alc260_hp_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc260_hp_automute(codec); + spec->autocfg.hp_pins[0] = 0x0f; + spec->autocfg.speaker_pins[0] = 0x10; + spec->autocfg.speaker_pins[1] = 0x11; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { @@ -6101,7 +6087,6 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, - .private_value = (0x15 << 16) | (0x10 << 8) | 0x11 }, HDA_CODEC_VOLUME("Front Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x10, 0x0, HDA_OUTPUT), @@ -6114,6 +6099,17 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { } /* end */ }; +static void alc260_hp_3013_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x10; + spec->autocfg.speaker_pins[1] = 0x11; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; +} + static struct hda_bind_ctls alc260_dc7600_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { @@ -6146,38 +6142,16 @@ static struct hda_verb alc260_hp_3013_unsol_verbs[] = { {}, }; -static void alc260_hp_3013_automute(struct hda_codec *codec) +static void alc260_hp_3012_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->jack_present = snd_hda_jack_detect(codec, 0x15); - alc260_hp_master_update(codec, 0x15, 0x10, 0x11); -} - -static void alc260_hp_3013_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc260_hp_3013_automute(codec); -} - -static void alc260_hp_3012_automute(struct hda_codec *codec) -{ - unsigned int bits = snd_hda_jack_detect(codec, 0x10) ? 0 : PIN_OUT; - - snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); - snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - bits); -} - -static void alc260_hp_3012_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_HP_EVENT) - alc260_hp_3012_automute(codec); + spec->autocfg.hp_pins[0] = 0x10; + spec->autocfg.speaker_pins[0] = 0x0f; + spec->autocfg.speaker_pins[1] = 0x11; + spec->autocfg.speaker_pins[2] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } /* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, @@ -7302,8 +7276,9 @@ static struct alc_config_preset alc260_presets[] = { .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, - .unsol_event = alc260_hp_unsol_event, - .init_hook = alc260_hp_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc260_hp_setup, + .init_hook = alc_inithook, }, [ALC260_HP_DC7600] = { .mixers = { alc260_hp_dc7600_mixer, @@ -7317,8 +7292,9 @@ static struct alc_config_preset alc260_presets[] = { .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, - .unsol_event = alc260_hp_3012_unsol_event, - .init_hook = alc260_hp_3012_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc260_hp_3012_setup, + .init_hook = alc_inithook, }, [ALC260_HP_3013] = { .mixers = { alc260_hp_3013_mixer, @@ -7332,8 +7308,9 @@ static struct alc_config_preset alc260_presets[] = { .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, - .unsol_event = alc260_hp_3013_unsol_event, - .init_hook = alc260_hp_3013_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc260_hp_3013_setup, + .init_hook = alc_inithook, }, [ALC260_FUJITSU_S702X] = { .mixers = { alc260_fujitsu_mixer }, @@ -11291,71 +11268,30 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { }; /* update HP, line and mono-out pins according to the master switch */ -static void alc262_hp_master_update(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int val = spec->master_sw; - - /* HP & line-out */ - snd_hda_codec_write_cache(codec, 0x1b, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - val ? PIN_HP : 0); - snd_hda_codec_write_cache(codec, 0x15, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - val ? PIN_HP : 0); - /* mono (speaker) depending on the HP jack sense */ - val = val && !spec->jack_present; - snd_hda_codec_write_cache(codec, 0x16, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - val ? PIN_OUT : 0); -} +#define alc262_hp_master_update alc260_hp_master_update -static void alc262_hp_bpc_automute(struct hda_codec *codec) +static void alc262_hp_bpc_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->jack_present = snd_hda_jack_detect(codec, 0x1b); - alc262_hp_master_update(codec); -} - -static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hp_bpc_automute(codec); + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x16; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } -static void alc262_hp_wildwest_automute(struct hda_codec *codec) +static void alc262_hp_wildwest_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->jack_present = snd_hda_jack_detect(codec, 0x15); - alc262_hp_master_update(codec); -} - -static void alc262_hp_wildwest_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hp_wildwest_automute(codec); + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x16; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_PIN; } #define alc262_hp_master_sw_get alc260_hp_master_sw_get - -static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - int val = !!*ucontrol->value.integer.value; - - if (val == spec->master_sw) - return 0; - spec->master_sw = val; - alc262_hp_master_update(codec); - return 1; -} +#define alc262_hp_master_sw_put alc260_hp_master_sw_put #define ALC262_HP_MASTER_SWITCH \ { \ @@ -11485,44 +11421,9 @@ static struct hda_input_mux alc262_hp_rp5700_capture_source = { }; /* bind hp and internal speaker mute (with plug check) as master switch */ -static void alc262_hippo_master_update(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; - hda_nid_t line_nid = spec->autocfg.line_out_pins[0]; - hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0]; - unsigned int mute; - - /* HP */ - mute = spec->master_sw ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, hp_nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - /* mute internal speaker per jack sense */ - if (spec->jack_present) - mute = HDA_AMP_MUTE; - if (line_nid) - snd_hda_codec_amp_stereo(codec, line_nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - if (speaker_nid && speaker_nid != line_nid) - snd_hda_codec_amp_stereo(codec, speaker_nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); -} - +#define alc262_hippo_master_update alc262_hp_master_update #define alc262_hippo_master_sw_get alc262_hp_master_sw_get - -static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - int val = !!*ucontrol->value.integer.value; - - if (val == spec->master_sw) - return 0; - spec->master_sw = val; - alc262_hippo_master_update(codec); - return 1; -} +#define alc262_hippo_master_sw_put alc262_hp_master_sw_put #define ALC262_HIPPO_MASTER_SWITCH \ { \ @@ -11573,28 +11474,14 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hippo_automute(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; - - spec->jack_present = snd_hda_jack_detect(codec, hp_nid); - alc262_hippo_master_update(codec); -} - -static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc262_hippo_automute(codec); -} - static void alc262_hippo_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static void alc262_hippo1_setup(struct hda_codec *codec) @@ -11603,6 +11490,8 @@ static void alc262_hippo1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -12816,9 +12705,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hippo_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hippo_setup, - .init_hook = alc262_hippo_automute, + .init_hook = alc_inithook, }, [ALC262_HIPPO_1] = { .mixers = { alc262_hippo1_mixer }, @@ -12830,9 +12719,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hippo_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hippo1_setup, - .init_hook = alc262_hippo_automute, + .init_hook = alc_inithook, }, [ALC262_FUJITSU] = { .mixers = { alc262_fujitsu_mixer }, @@ -12857,8 +12746,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_HP_capture_source, - .unsol_event = alc262_hp_bpc_unsol_event, - .init_hook = alc262_hp_bpc_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc262_hp_bpc_setup, + .init_hook = alc_inithook, }, [ALC262_HP_BPC_D7000_WF] = { .mixers = { alc262_HP_BPC_WildWest_mixer }, @@ -12869,8 +12759,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_HP_D7000_capture_source, - .unsol_event = alc262_hp_wildwest_unsol_event, - .init_hook = alc262_hp_wildwest_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc262_hp_wildwest_setup, + .init_hook = alc_inithook, }, [ALC262_HP_BPC_D7000_WL] = { .mixers = { alc262_HP_BPC_WildWest_mixer, @@ -12882,8 +12773,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_HP_D7000_capture_source, - .unsol_event = alc262_hp_wildwest_unsol_event, - .init_hook = alc262_hp_wildwest_automute, + .unsol_event = alc_sku_unsol_event, + .setup = alc262_hp_wildwest_setup, + .init_hook = alc_inithook, }, [ALC262_HP_TC_T5735] = { .mixers = { alc262_hp_t5735_mixer }, @@ -12926,9 +12818,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hippo_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hippo_setup, - .init_hook = alc262_hippo_automute, + .init_hook = alc_inithook, }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, @@ -12940,9 +12832,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hippo_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hippo_setup, - .init_hook = alc262_hippo_automute, + .init_hook = alc_inithook, }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, @@ -13008,9 +12900,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc262_hippo_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hippo_setup, - .init_hook = alc262_hippo_automute, + .init_hook = alc_inithook, }, [ALC262_TYAN] = { .mixers = { alc262_tyan_mixer }, @@ -13347,9 +13239,7 @@ static struct hda_verb alc268_acer_verbs[] = { }; /* unsolicited event for HP jack sensing */ -#define alc268_toshiba_unsol_event alc262_hippo_unsol_event #define alc268_toshiba_setup alc262_hippo_setup -#define alc268_toshiba_automute alc262_hippo_automute static void alc268_acer_unsol_event(struct hda_codec *codec, unsigned int res) @@ -13985,9 +13875,9 @@ static struct alc_config_preset alc268_presets[] = { .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, - .unsol_event = alc268_toshiba_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc268_toshiba_setup, - .init_hook = alc268_toshiba_automute, + .init_hook = alc_inithook, }, [ALC268_ACER] = { .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, @@ -14073,8 +13963,9 @@ static struct alc_config_preset alc268_presets[] = { .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, + .unsol_event = alc_sku_unsol_event, .setup = alc268_toshiba_setup, - .init_hook = alc268_toshiba_automute, + .init_hook = alc_inithook, }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { @@ -18249,16 +18140,6 @@ static void alc662_lenovo_101e_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -/* unsolicited event for HP jack sensing */ -static void alc662_eeepc_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_MIC_EVENT) - alc_mic_automute(codec); - else - alc262_hippo_unsol_event(codec, res); -} - static void alc662_eeepc_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -18271,22 +18152,16 @@ static void alc662_eeepc_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc662_eeepc_inithook(struct hda_codec *codec) -{ - alc262_hippo_automute(codec); - alc_mic_automute(codec); -} - static void alc662_eeepc_ep20_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } -#define alc662_eeepc_ep20_inithook alc262_hippo_master_update - static void alc663_m51va_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -18675,9 +18550,9 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc662_eeepc_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc662_eeepc_setup, - .init_hook = alc662_eeepc_inithook, + .init_hook = alc_inithook, }, [ALC662_ASUS_EEEPC_EP20] = { .mixers = { alc662_eeepc_ep20_mixer, @@ -18690,9 +18565,9 @@ static struct alc_config_preset alc662_presets[] = { .num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes), .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, - .unsol_event = alc662_eeepc_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc662_eeepc_ep20_setup, - .init_hook = alc662_eeepc_ep20_inithook, + .init_hook = alc_inithook, }, [ALC662_ECS] = { .mixers = { alc662_ecs_mixer }, @@ -18703,9 +18578,9 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .unsol_event = alc662_eeepc_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc662_eeepc_setup, - .init_hook = alc662_eeepc_inithook, + .init_hook = alc_inithook, }, [ALC663_ASUS_M51VA] = { .mixers = { alc663_m51va_mixer }, -- cgit v1.2.3 From 0f0f391c730228d1fd1c3933275ed75ee96e4db2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 16:26:24 +0200 Subject: ALSA: hda - More reduction of redundant automute codes in Realtek parser Removed the redundant codes by replacing with the common helper functions. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 222 ++++++++++-------------------------------- 1 file changed, 49 insertions(+), 173 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a97e4602741..b76d3b3fb63 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -390,7 +390,6 @@ struct alc_spec { void (*shutup)(struct hda_codec *codec); /* for pin sensing */ - unsigned int sense_updated: 1; unsigned int jack_present: 1; unsigned int line_jack_present:1; unsigned int master_mute:1; @@ -11797,40 +11796,15 @@ static struct hda_input_mux alc262_HP_D7000_capture_source = { }, }; -/* mute/unmute internal speaker according to the hp jacks and mute state */ -static void alc262_fujitsu_automute(struct hda_codec *codec, int force) +static void alc262_fujitsu_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int mute; - - if (force || !spec->sense_updated) { - spec->jack_present = snd_hda_jack_detect(codec, 0x14) || - snd_hda_jack_detect(codec, 0x1b); - spec->sense_updated = 1; - } - /* unmute internal speaker only if both HPs are unplugged and - * master switch is on - */ - if (spec->jack_present) - mute = HDA_AMP_MUTE; - else - mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); -} - -/* unsolicited event for HP jack sensing */ -static void alc262_fujitsu_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC_HP_EVENT) - return; - alc262_fujitsu_automute(codec, 1); -} -static void alc262_fujitsu_init_hook(struct hda_codec *codec) -{ - alc262_fujitsu_automute(codec, 1); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.hp_pins[1] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } /* bind volumes of both NID 0x0c and 0x0d */ @@ -11843,78 +11817,15 @@ static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { }, }; -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force) -{ - struct alc_spec *spec = codec->spec; - unsigned int mute; - - if (force || !spec->sense_updated) { - spec->jack_present = snd_hda_jack_detect(codec, 0x1b); - spec->sense_updated = 1; - } - if (spec->jack_present) { - /* mute internal speaker */ - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - } else { - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } -} - -/* unsolicited event for HP jack sensing */ -static void alc262_lenovo_3000_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC_HP_EVENT) - return; - alc262_lenovo_3000_automute(codec, 1); -} - -static int amp_stereo_mute_update(struct hda_codec *codec, hda_nid_t nid, - int dir, int idx, long *valp) -{ - int i, change = 0; - - for (i = 0; i < 2; i++, valp++) - change |= snd_hda_codec_amp_update(codec, nid, i, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); - return change; -} - -/* bind hp and internal speaker mute (with plug check) */ -static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; - - change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); - change |= amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); - if (change) - alc262_fujitsu_automute(codec, 0); - return change; -} - static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc262_fujitsu_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .info = snd_ctl_boolean_mono_info, + .get = alc262_hp_master_sw_get, + .put = alc262_hp_master_sw_put, }, { .iface = NID_MAPPING, @@ -11932,18 +11843,15 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { } /* end */ }; -/* bind hp and internal speaker mute (with plug check) */ -static int alc262_lenovo_3000_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void alc262_lenovo_3000_setup(struct hda_codec *codec) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; + struct alc_spec *spec = codec->spec; - change = amp_stereo_mute_update(codec, 0x1b, HDA_OUTPUT, 0, valp); - if (change) - alc262_lenovo_3000_automute(codec, 0); - return change; + spec->autocfg.hp_pins[0] = 0x1b; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { @@ -11951,11 +11859,10 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = alc262_lenovo_3000_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, + .info = snd_ctl_boolean_mono_info, + .get = alc262_hp_master_sw_get, + .put = alc262_hp_master_sw_put, }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -12734,8 +12641,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_fujitsu_capture_source, - .unsol_event = alc262_fujitsu_unsol_event, - .init_hook = alc262_fujitsu_init_hook, + .unsol_event = alc_sku_unsol_event, + .setup = alc262_fujitsu_setup, + .init_hook = alc_inithook, }, [ALC262_HP_BPC] = { .mixers = { alc262_HP_BPC_mixer }, @@ -12863,7 +12771,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_fujitsu_capture_source, - .unsol_event = alc262_lenovo_3000_unsol_event, + .unsol_event = alc_sku_unsol_event, + .setup = alc262_lenovo_3000_setup, + .init_hook = alc_inithook, }, [ALC262_NEC] = { .mixers = { alc262_nec_mixer }, @@ -13133,38 +13043,18 @@ static struct hda_bind_ctls alc268_acer_bind_master_vol = { }, }; -/* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc268_acer_automute(struct hda_codec *codec, int force) +static void alc268_acer_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int mute; - if (force || !spec->sense_updated) { - spec->jack_present = snd_hda_jack_detect(codec, 0x14); - spec->sense_updated = 1; - } - if (spec->jack_present) - mute = HDA_AMP_MUTE; /* mute internal speaker */ - else /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x15; + spec->automute = 1; + spec->automute_mode = ALC_AUTOMUTE_AMP; } - -/* bind hp and internal speaker mute (with plug check) */ -static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; - - change = amp_stereo_mute_update(codec, 0x14, HDA_OUTPUT, 0, valp); - if (change) - alc268_acer_automute(codec, 0); - return change; -} +#define alc268_acer_master_sw_get alc262_hp_master_sw_get +#define alc268_acer_master_sw_put alc262_hp_master_sw_put static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { /* output mixer control */ @@ -13172,11 +13062,10 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x15, + .info = snd_ctl_boolean_mono_info, + .get = alc268_acer_master_sw_get, .put = alc268_acer_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT), { } @@ -13188,11 +13077,10 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .info = snd_ctl_boolean_mono_info, + .get = alc268_acer_master_sw_get, .put = alc268_acer_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x19, 0, HDA_INPUT), @@ -13206,11 +13094,10 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .info = snd_ctl_boolean_mono_info, + .get = alc268_acer_master_sw_get, .put = alc268_acer_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Line In Boost Volume", 0x1a, 0, HDA_INPUT), @@ -13241,19 +13128,6 @@ static struct hda_verb alc268_acer_verbs[] = { /* unsolicited event for HP jack sensing */ #define alc268_toshiba_setup alc262_hippo_setup -static void alc268_acer_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != ALC880_HP_EVENT) - return; - alc268_acer_automute(codec, 1); -} - -static void alc268_acer_init_hook(struct hda_codec *codec) -{ - alc268_acer_automute(codec, 1); -} - static void alc268_acer_lc_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -13893,8 +13767,9 @@ static struct alc_config_preset alc268_presets[] = { .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, .input_mux = &alc268_acer_capture_source, - .unsol_event = alc268_acer_unsol_event, - .init_hook = alc268_acer_init_hook, + .unsol_event = alc_sku_unsol_event, + .setup = alc268_acer_setup, + .init_hook = alc_inithook, }, [ALC268_ACER_DMIC] = { .mixers = { alc268_acer_dmic_mixer, alc268_capture_alt_mixer, @@ -13910,8 +13785,9 @@ static struct alc_config_preset alc268_presets[] = { .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, .input_mux = &alc268_acer_dmic_capture_source, - .unsol_event = alc268_acer_unsol_event, - .init_hook = alc268_acer_init_hook, + .unsol_event = alc_sku_unsol_event, + .setup = alc268_acer_setup, + .init_hook = alc_inithook, }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, -- cgit v1.2.3 From 1a1455de10d89b9f2107fe5ad1746e7c18838492 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 17:36:18 +0200 Subject: ALSA: hda - Add support for Line-Out automute to Realtek auto-parser By popular demands, I add the functionality to mute / unmute the line-out jacks per the headphone plug / unplug. For achieving this and keeping the compatibility with the old behavior, the new mixer enum "Auto-Mute Mode" is added. With this, user can control the auto-mute behavior either disabled, speaker-only or lineout+speaker. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 146 ++++++++++++++++++++++++++++++++++++++---- 1 file changed, 133 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b76d3b3fb63..9e2594d6db9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1129,18 +1129,28 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, static void update_speakers(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + int on; + if (!spec->automute) + on = 0; + else + on = spec->jack_present | spec->line_jack_present; + on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), - spec->autocfg.speaker_pins, - spec->jack_present | spec->line_jack_present | - spec->master_mute, false); + spec->autocfg.speaker_pins, on, false); /* toggle line-out mutes if needed, too */ - if (!spec->automute_lines) + /* if LO is a copy of either HP or Speaker, don't need to handle it */ + if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || + spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; + if (!spec->automute_lines || !spec->automute) + on = 0; + else + on = spec->jack_present; + on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), - spec->autocfg.line_out_pins, - spec->jack_present | spec->master_mute, false); + spec->autocfg.line_out_pins, on, false); } static void alc_hp_automute(struct hda_codec *codec) @@ -1414,6 +1424,95 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) } } +static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { + "Disabled", "Speaker Only", "Line-Out+Speaker" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + unsigned int val; + if (!spec->automute) + val = 0; + else if (!spec->automute_lines) + val = 1; + else + val = 2; + ucontrol->value.enumerated.item[0] = val; + return 0; +} + +static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + + switch (ucontrol->value.enumerated.item[0]) { + case 0: + if (!spec->automute) + return 0; + spec->automute = 0; + break; + case 1: + if (spec->automute && !spec->automute_lines) + return 0; + spec->automute = 1; + spec->automute_lines = 0; + break; + case 2: + if (spec->automute && spec->automute_lines) + return 0; + spec->automute = 1; + spec->automute_lines = 1; + break; + default: + return -EINVAL; + } + update_speakers(codec); + return 1; +} + +static struct snd_kcontrol_new alc_automute_mode_enum = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Auto-Mute Mode", + .info = alc_automute_mode_info, + .get = alc_automute_mode_get, + .put = alc_automute_mode_put, +}; + +static struct snd_kcontrol_new *alc_kcontrol_new(struct alc_spec *spec); + +static int alc_add_automute_mode_enum(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct snd_kcontrol_new *knew; + + knew = alc_kcontrol_new(spec); + if (!knew) + return -ENOMEM; + *knew = alc_automute_mode_enum; + knew->name = kstrdup("Auto-Mute Mode", GFP_KERNEL); + if (!knew->name) + return -ENOMEM; + return 0; +} + static void alc_init_auto_hp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1440,14 +1539,37 @@ static void alc_init_auto_hp(struct hda_codec *codec) } for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + if (!(snd_hda_query_pin_caps(codec, nid) & + AC_PINCAP_PRES_DETECT)) + continue; snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n", - cfg->hp_pins[i]); - snd_hda_codec_write_cache(codec, cfg->hp_pins[i], 0, + nid); + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } + if (spec->automute && cfg->line_out_pins[0] && + cfg->line_out_pins[0] != cfg->hp_pins[0] && + cfg->line_out_pins[0] != cfg->speaker_pins[0]) { + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t nid = cfg->line_out_pins[i]; + if (!(snd_hda_query_pin_caps(codec, nid) & + AC_PINCAP_PRES_DETECT)) + continue; + snd_printdd("realtek: Enable Line-Out auto-muting " + "on NID 0x%x\n", nid); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_FRONT_EVENT); + spec->detect_line = 1; + } + /* create a control for automute mode */ + alc_add_automute_mode_enum(codec); + spec->automute_lines = 1; + } spec->unsol_event = alc_sku_unsol_event; } @@ -1684,9 +1806,6 @@ do_sku: return 1; spec->autocfg.hp_pins[0] = nid; } - - alc_init_auto_hp(codec); - alc_init_auto_mic(codec); return 1; } @@ -1699,9 +1818,10 @@ static void alc_ssid_check(struct hda_codec *codec, snd_printd("realtek: " "Enable default setup for auto mode as fallback\n"); spec->init_amp = ALC_INIT_DEFAULT; - alc_init_auto_hp(codec); - alc_init_auto_mic(codec); } + + alc_init_auto_hp(codec); + alc_init_auto_mic(codec); } /* -- cgit v1.2.3 From 1daf5f46c62bc69922e1ca8ccebbbff04f48e1f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 17:57:46 +0200 Subject: ALSA: hda - More line-out auto-mute support for Realtek Not only supporting the line-out automute as additional feature to the existing headphone automute, now the headphone jack can mute the line-out alone even without the speaker outs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9e2594d6db9..3e41d563776 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1517,16 +1517,19 @@ static void alc_init_auto_hp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; + int present = 0; int i; - if (!cfg->hp_pins[0]) { - if (cfg->line_out_type != AUTO_PIN_HP_OUT) - return; - } + if (cfg->hp_pins[0]) + present++; + if (cfg->line_out_pins[0]) + present++; + if (cfg->speaker_pins[0]) + present++; + if (present < 2) /* need two different output types */ + return; if (!cfg->speaker_pins[0]) { - if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) - return; memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins)); cfg->speaker_outs = cfg->line_outs; -- cgit v1.2.3 From ae8a60a598ce39e8e42fd4ce1348c0883a23b5d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Apr 2011 18:09:52 +0200 Subject: ALSA: hda - Add Auto-Mute Mode enum for two-output cases The Auto-Mute Mode control is useful even when only two outputs (e.g. HP and speaker) are available. Then user can enable/disable the auto-mute behavior on the fly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 32 ++++++++++++++++++++++++++------ 1 file changed, 26 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3e41d563776..faede0f49cc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -397,6 +397,7 @@ struct alc_spec { unsigned int automute:1; /* HP automute enabled */ unsigned int detect_line:1; /* Line-out detection enabled */ unsigned int automute_lines:1; /* automute line-out as well */ + unsigned int automute_hp_lo:1; /* both HP and LO available */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -1427,15 +1428,27 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static const char * const texts[] = { + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + static const char * const texts2[] = { + "Disabled", "Enabled" + }; + static const char * const texts3[] = { "Disabled", "Speaker Only", "Line-Out+Speaker" }; + const char * const *texts; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item >= 3) - uinfo->value.enumerated.item = 2; + if (spec->automute_hp_lo) { + uinfo->value.enumerated.items = 3; + texts = texts3; + } else { + uinfo->value.enumerated.items = 2; + texts = texts2; + } + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; @@ -1476,6 +1489,8 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, spec->automute_lines = 0; break; case 2: + if (!spec->automute_hp_lo) + return -EINVAL; if (spec->automute && spec->automute_lines) return 0; spec->automute = 1; @@ -1528,6 +1543,8 @@ static void alc_init_auto_hp(struct hda_codec *codec) present++; if (present < 2) /* need two different output types */ return; + if (present == 3) + spec->automute_hp_lo = 1; /* both HP and LO automute */ if (!cfg->speaker_pins[0]) { memcpy(cfg->speaker_pins, cfg->line_out_pins, @@ -1569,11 +1586,14 @@ static void alc_init_auto_hp(struct hda_codec *codec) AC_USRSP_EN | ALC880_FRONT_EVENT); spec->detect_line = 1; } + spec->automute_lines = 1; + } + + if (spec->automute) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); - spec->automute_lines = 1; + spec->unsol_event = alc_sku_unsol_event; } - spec->unsol_event = alc_sku_unsol_event; } static void alc_init_auto_mic(struct hda_codec *codec) -- cgit v1.2.3 From c2de187e5b0f25b572ac1cb6cdf383f16123717d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Apr 2011 13:00:40 +0200 Subject: ALSA: hda - Show the line-out type in snd_hda_parse_pin_def_config() Helpful for debugging. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fb07def1586..10f429f9616 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4697,10 +4697,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, /* * debug prints of the parsed results */ - snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x) type:%s\n", cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], cfg->line_out_pins[2], cfg->line_out_pins[3], - cfg->line_out_pins[4]); + cfg->line_out_pins[4], + cfg->line_out_type == AUTO_PIN_HP_OUT ? "hp" : + (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT ? + "speaker" : "line")); snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", cfg->speaker_outs, cfg->speaker_pins[0], cfg->speaker_pins[1], cfg->speaker_pins[2], -- cgit v1.2.3 From 031024eea8d4b2a11331a7c269216e067a46625e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:29:30 +0200 Subject: ALSA: hda - Constify some API function arguments Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +++-- sound/pci/hda/hda_local.h | 4 ++-- 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c40e415852..d1a2351aabf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3872,7 +3872,8 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config); * * Returns 0 if successful, or a negative error code. */ -int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) +int snd_hda_add_new_ctls(struct hda_codec *codec, + const struct snd_kcontrol_new *knew) { int err; @@ -4015,7 +4016,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, hda_nid_t nid) { - struct hda_amp_list *p; + const struct hda_amp_list *p; int ch, v; if (!check->amplist) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ff5e2ac2239..f36af375abb 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -347,7 +347,7 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, int num_configs, const char * const *models, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, - struct snd_kcontrol_new *knew); + const struct snd_kcontrol_new *knew); /* * unsolicited event handler @@ -567,7 +567,7 @@ struct hda_amp_list { }; struct hda_loopback_check { - struct hda_amp_list *amplist; + const struct hda_amp_list *amplist; int power_on; }; -- cgit v1.2.3 From a9111321f2fb6a23fbed82b8b4cbd77f5580ba75 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:30:18 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_realtek.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1006 ++++++++++++++++++++--------------------- 1 file changed, 503 insertions(+), 503 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 46959159a77..5b79080be76 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -313,9 +313,9 @@ enum { struct alc_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ + const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; - struct snd_kcontrol_new *cap_mixer; /* capture mixer */ + const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[10]; /* initialization verbs @@ -325,14 +325,14 @@ struct alc_spec { unsigned int num_init_verbs; char stream_name_analog[32]; /* analog PCM stream */ - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; - struct hda_pcm_stream *stream_analog_alt_playback; - struct hda_pcm_stream *stream_analog_alt_capture; + const struct hda_pcm_stream *stream_analog_playback; + const struct hda_pcm_stream *stream_analog_capture; + const struct hda_pcm_stream *stream_analog_alt_playback; + const struct hda_pcm_stream *stream_analog_alt_capture; char stream_name_digital[32]; /* digital PCM stream */ - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; + const struct hda_pcm_stream *stream_digital_playback; + const struct hda_pcm_stream *stream_digital_capture; /* playback */ struct hda_multi_out multiout; /* playback set-up @@ -435,10 +435,10 @@ struct alc_spec { * configuration template - to be copied to the spec instance */ struct alc_config_preset { - struct snd_kcontrol_new *mixers[5]; /* should be identical size + const struct snd_kcontrol_new *mixers[5]; /* should be identical size * with spec */ - struct snd_kcontrol_new *cap_mixer; /* capture mixer */ + const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ const struct hda_verb *init_verbs[5]; unsigned int num_dacs; hda_nid_t *dac_nids; @@ -459,7 +459,7 @@ struct alc_config_preset { void (*setup)(struct hda_codec *); void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE - struct hda_amp_list *loopbacks; + const struct hda_amp_list *loopbacks; void (*power_hook)(struct hda_codec *codec); #endif }; @@ -586,11 +586,11 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, * NIDs 0x0f and 0x10 have been observed to have this behaviour as of * March 2006. */ -static char *alc_pin_mode_names[] = { +static const char * const alc_pin_mode_names[] = { "Mic 50pc bias", "Mic 80pc bias", "Line in", "Line out", "Headphone out", }; -static unsigned char alc_pin_mode_values[] = { +static const unsigned char alc_pin_mode_values[] = { PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP, }; /* The control can present all 5 options, or it can limit the options based @@ -609,7 +609,7 @@ static unsigned char alc_pin_mode_values[] = { /* Info about the pin modes supported by the different pin direction modes. * For each direction the minimum and maximum values are given. */ -static signed char alc_pin_mode_dir_info[5][2] = { +static const signed char alc_pin_mode_dir_info[5][2] = { { 0, 2 }, /* ALC_PIN_DIR_IN */ { 3, 4 }, /* ALC_PIN_DIR_OUT */ { 0, 4 }, /* ALC_PIN_DIR_INOUT */ @@ -926,7 +926,7 @@ static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec) /* */ -static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix) +static void add_mixer(struct alc_spec *spec, const struct snd_kcontrol_new *mix) { if (snd_BUG_ON(spec->num_mixers >= ARRAY_SIZE(spec->mixers))) return; @@ -997,21 +997,21 @@ static void setup_preset(struct hda_codec *codec, } /* Enable GPIO mask and set output */ -static struct hda_verb alc_gpio1_init_verbs[] = { +static const struct hda_verb alc_gpio1_init_verbs[] = { {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, { } }; -static struct hda_verb alc_gpio2_init_verbs[] = { +static const struct hda_verb alc_gpio2_init_verbs[] = { {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, { } }; -static struct hda_verb alc_gpio3_init_verbs[] = { +static const struct hda_verb alc_gpio3_init_verbs[] = { {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, @@ -1503,7 +1503,7 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new alc_automute_mode_enum = { +static const struct snd_kcontrol_new alc_automute_mode_enum = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Auto-Mute Mode", .info = alc_automute_mode_info, @@ -2086,7 +2086,7 @@ static void alc_auto_parse_digital(struct hda_codec *codec) /* * 2ch mode */ -static struct hda_verb alc888_4ST_ch2_intel_init[] = { +static const struct hda_verb alc888_4ST_ch2_intel_init[] = { /* Mic-in jack as mic in */ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -2101,7 +2101,7 @@ static struct hda_verb alc888_4ST_ch2_intel_init[] = { /* * 4ch mode */ -static struct hda_verb alc888_4ST_ch4_intel_init[] = { +static const struct hda_verb alc888_4ST_ch4_intel_init[] = { /* Mic-in jack as mic in */ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -2116,7 +2116,7 @@ static struct hda_verb alc888_4ST_ch4_intel_init[] = { /* * 6ch mode */ -static struct hda_verb alc888_4ST_ch6_intel_init[] = { +static const struct hda_verb alc888_4ST_ch6_intel_init[] = { /* Mic-in jack as CLFE */ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, @@ -2131,7 +2131,7 @@ static struct hda_verb alc888_4ST_ch6_intel_init[] = { /* * 8ch mode */ -static struct hda_verb alc888_4ST_ch8_intel_init[] = { +static const struct hda_verb alc888_4ST_ch8_intel_init[] = { /* Mic-in jack as CLFE */ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, @@ -2143,7 +2143,7 @@ static struct hda_verb alc888_4ST_ch8_intel_init[] = { { } /* end */ }; -static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = { +static const struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = { { 2, alc888_4ST_ch2_intel_init }, { 4, alc888_4ST_ch4_intel_init }, { 6, alc888_4ST_ch6_intel_init }, @@ -2154,7 +2154,7 @@ static struct hda_channel_mode alc888_4ST_8ch_intel_modes[4] = { * ALC888 Fujitsu Siemens Amillo xa3530 */ -static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { +static const struct hda_verb alc888_fujitsu_xa3530_verbs[] = { /* Front Mic: set to PIN_IN (empty by default) */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Connect Internal HP to Front */ @@ -2223,7 +2223,7 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) * ALC888 Acer Aspire 4930G model */ -static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { +static const struct hda_verb alc888_acer_aspire_4930g_verbs[] = { /* Front Mic: set to PIN_IN (empty by default) */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Unselect Front Mic by default in input mixer 3 */ @@ -2246,7 +2246,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { * ALC888 Acer Aspire 6530G model */ -static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +static const struct hda_verb alc888_acer_aspire_6530g_verbs[] = { /* Route to built-in subwoofer as well as speakers */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -2276,7 +2276,7 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { *ALC888 Acer Aspire 7730G model */ -static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { +static const struct hda_verb alc888_acer_aspire_7730G_verbs[] = { /* Bias voltage on for external mic port */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, /* Front Mic: set to PIN_IN (empty by default) */ @@ -2306,7 +2306,7 @@ static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { * ALC889 Acer Aspire 8930G model */ -static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { +static const struct hda_verb alc889_acer_aspire_8930g_verbs[] = { /* Front Mic: set to PIN_IN (empty by default) */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Unselect Front Mic by default in input mixer 3 */ @@ -2352,7 +2352,7 @@ static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { { } }; -static struct hda_input_mux alc888_2_capture_sources[2] = { +static const struct hda_input_mux alc888_2_capture_sources[2] = { /* Front mic only available on one ADC */ { .num_items = 4, @@ -2373,7 +2373,7 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { } }; -static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { +static const struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { /* Interal mic only available on one ADC */ { .num_items = 5, @@ -2396,7 +2396,7 @@ static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { } }; -static struct hda_input_mux alc889_capture_sources[3] = { +static const struct hda_input_mux alc889_capture_sources[3] = { /* Digital mic only available on first "ADC" */ { .num_items = 5, @@ -2428,7 +2428,7 @@ static struct hda_input_mux alc889_capture_sources[3] = { } }; -static struct snd_kcontrol_new alc888_base_mixer[] = { +static const struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -2450,7 +2450,7 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { +static const struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -2472,7 +2472,7 @@ static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { +static const struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -2569,7 +2569,7 @@ static hda_nid_t alc880_adc_nids_alt[2] = { #define ALC880_DIGOUT_NID 0x06 #define ALC880_DIGIN_NID 0x0a -static struct hda_input_mux alc880_capture_source = { +static const struct hda_input_mux alc880_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -2581,7 +2581,7 @@ static struct hda_input_mux alc880_capture_source = { /* channel source setting (2/6 channel selection for 3-stack) */ /* 2ch mode */ -static struct hda_verb alc880_threestack_ch2_init[] = { +static const struct hda_verb alc880_threestack_ch2_init[] = { /* set line-in to input, mute it */ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -2592,7 +2592,7 @@ static struct hda_verb alc880_threestack_ch2_init[] = { }; /* 6ch mode */ -static struct hda_verb alc880_threestack_ch6_init[] = { +static const struct hda_verb alc880_threestack_ch6_init[] = { /* set line-in to output, unmute it */ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, @@ -2602,12 +2602,12 @@ static struct hda_verb alc880_threestack_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc880_threestack_modes[2] = { +static const struct hda_channel_mode alc880_threestack_modes[2] = { { 2, alc880_threestack_ch2_init }, { 6, alc880_threestack_ch6_init }, }; -static struct snd_kcontrol_new alc880_three_stack_mixer[] = { +static const struct snd_kcontrol_new alc880_three_stack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), @@ -2752,14 +2752,14 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, } #define DEFINE_CAPMIX(num) \ -static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ +static const struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ _DEFINE_CAPMIX(num), \ _DEFINE_CAPSRC(num), \ { } /* end */ \ } #define DEFINE_CAPMIX_NOSRC(num) \ -static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \ +static const struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \ _DEFINE_CAPMIX(num), \ { } /* end */ \ } @@ -2782,7 +2782,7 @@ DEFINE_CAPMIX_NOSRC(3); */ /* additional mixers to alc880_three_stack_mixer */ -static struct snd_kcontrol_new alc880_five_stack_mixer[] = { +static const struct snd_kcontrol_new alc880_five_stack_mixer[] = { HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT), { } /* end */ @@ -2790,7 +2790,7 @@ static struct snd_kcontrol_new alc880_five_stack_mixer[] = { /* channel source setting (6/8 channel selection for 5-stack) */ /* 6ch mode */ -static struct hda_verb alc880_fivestack_ch6_init[] = { +static const struct hda_verb alc880_fivestack_ch6_init[] = { /* set line-in to input, mute it */ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -2798,14 +2798,14 @@ static struct hda_verb alc880_fivestack_ch6_init[] = { }; /* 8ch mode */ -static struct hda_verb alc880_fivestack_ch8_init[] = { +static const struct hda_verb alc880_fivestack_ch8_init[] = { /* set line-in to output, unmute it */ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { } /* end */ }; -static struct hda_channel_mode alc880_fivestack_modes[2] = { +static const struct hda_channel_mode alc880_fivestack_modes[2] = { { 6, alc880_fivestack_ch6_init }, { 8, alc880_fivestack_ch8_init }, }; @@ -2825,7 +2825,7 @@ static hda_nid_t alc880_6st_dac_nids[4] = { 0x02, 0x03, 0x04, 0x05 }; -static struct hda_input_mux alc880_6stack_capture_source = { +static const struct hda_input_mux alc880_6stack_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -2836,11 +2836,11 @@ static struct hda_input_mux alc880_6stack_capture_source = { }; /* fixed 8-channels */ -static struct hda_channel_mode alc880_sixstack_modes[1] = { +static const struct hda_channel_mode alc880_sixstack_modes[1] = { { 8, NULL }, }; -static struct snd_kcontrol_new alc880_six_stack_mixer[] = { +static const struct snd_kcontrol_new alc880_six_stack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -2901,12 +2901,12 @@ static hda_nid_t alc880_w810_dac_nids[3] = { }; /* fixed 6 channels */ -static struct hda_channel_mode alc880_w810_modes[1] = { +static const struct hda_channel_mode alc880_w810_modes[1] = { { 6, NULL } }; /* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */ -static struct snd_kcontrol_new alc880_w810_base_mixer[] = { +static const struct snd_kcontrol_new alc880_w810_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -2934,11 +2934,11 @@ static hda_nid_t alc880_z71v_dac_nids[1] = { #define ALC880_Z71V_HP_DAC 0x03 /* fixed 2 channels */ -static struct hda_channel_mode alc880_2_jack_modes[1] = { +static const struct hda_channel_mode alc880_2_jack_modes[1] = { { 2, NULL } }; -static struct snd_kcontrol_new alc880_z71v_mixer[] = { +static const struct snd_kcontrol_new alc880_z71v_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -2963,7 +2963,7 @@ static hda_nid_t alc880_f1734_dac_nids[1] = { }; #define ALC880_F1734_HP_DAC 0x02 -static struct snd_kcontrol_new alc880_f1734_mixer[] = { +static const struct snd_kcontrol_new alc880_f1734_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -2975,7 +2975,7 @@ static struct snd_kcontrol_new alc880_f1734_mixer[] = { { } /* end */ }; -static struct hda_input_mux alc880_f1734_capture_source = { +static const struct hda_input_mux alc880_f1734_capture_source = { .num_items = 2, .items = { { "Mic", 0x1 }, @@ -2995,7 +2995,7 @@ static struct hda_input_mux alc880_f1734_capture_source = { #define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */ #define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */ -static struct snd_kcontrol_new alc880_asus_mixer[] = { +static const struct snd_kcontrol_new alc880_asus_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -3029,14 +3029,14 @@ static struct snd_kcontrol_new alc880_asus_mixer[] = { */ /* additional mixers to alc880_asus_mixer */ -static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { +static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT), { } /* end */ }; /* TCL S700 */ -static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { +static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -3050,7 +3050,7 @@ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { }; /* Uniwill */ -static struct snd_kcontrol_new alc880_uniwill_mixer[] = { +static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -3077,7 +3077,7 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc880_fujitsu_mixer[] = { +static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -3091,7 +3091,7 @@ static struct snd_kcontrol_new alc880_fujitsu_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { +static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -3154,7 +3154,7 @@ static void alc_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ -static struct snd_kcontrol_new alc_beep_mixer[] = { +static const struct snd_kcontrol_new alc_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), { } /* end */ @@ -3165,7 +3165,7 @@ static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct snd_kcontrol *kctl = NULL; - struct snd_kcontrol_new *knew; + const struct snd_kcontrol_new *knew; int i, j, err; unsigned int u; hda_nid_t nid; @@ -3202,7 +3202,7 @@ static int alc_build_controls(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_INPUT_BEEP /* create beep controls if needed */ if (spec->beep_amp) { - struct snd_kcontrol_new *knew; + const struct snd_kcontrol_new *knew; for (knew = alc_beep_mixer; knew->name; knew++) { struct snd_kcontrol *kctl; kctl = snd_ctl_new1(knew, codec); @@ -3319,7 +3319,7 @@ static int alc_build_controls(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc880_volume_init_verbs[] = { +static const struct hda_verb alc880_volume_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -3370,7 +3370,7 @@ static struct hda_verb alc880_volume_init_verbs[] = { * 3-stack pin configuration: * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b */ -static struct hda_verb alc880_pin_3stack_init_verbs[] = { +static const struct hda_verb alc880_pin_3stack_init_verbs[] = { /* * preset connection lists of input pins * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround @@ -3408,7 +3408,7 @@ static struct hda_verb alc880_pin_3stack_init_verbs[] = { * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19, * line-in/side = 0x1a, f-mic = 0x1b */ -static struct hda_verb alc880_pin_5stack_init_verbs[] = { +static const struct hda_verb alc880_pin_5stack_init_verbs[] = { /* * preset connection lists of input pins * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround @@ -3452,7 +3452,7 @@ static struct hda_verb alc880_pin_5stack_init_verbs[] = { * W810 pin configuration: * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b */ -static struct hda_verb alc880_pin_w810_init_verbs[] = { +static const struct hda_verb alc880_pin_w810_init_verbs[] = { /* hphone/speaker input selector: front DAC */ {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, @@ -3473,7 +3473,7 @@ static struct hda_verb alc880_pin_w810_init_verbs[] = { * Z71V pin configuration: * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?) */ -static struct hda_verb alc880_pin_z71v_init_verbs[] = { +static const struct hda_verb alc880_pin_z71v_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -3492,7 +3492,7 @@ static struct hda_verb alc880_pin_z71v_init_verbs[] = { * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, * f-mic = 0x19, line = 0x1a, HP = 0x1b */ -static struct hda_verb alc880_pin_6stack_init_verbs[] = { +static const struct hda_verb alc880_pin_6stack_init_verbs[] = { {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -3522,7 +3522,7 @@ static struct hda_verb alc880_pin_6stack_init_verbs[] = { * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19, * line = 0x1a */ -static struct hda_verb alc880_uniwill_init_verbs[] = { +static const struct hda_verb alc880_uniwill_init_verbs[] = { {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -3560,7 +3560,7 @@ static struct hda_verb alc880_uniwill_init_verbs[] = { * Uniwill P53 * HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, */ -static struct hda_verb alc880_uniwill_p53_init_verbs[] = { +static const struct hda_verb alc880_uniwill_p53_init_verbs[] = { {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -3589,7 +3589,7 @@ static struct hda_verb alc880_uniwill_p53_init_verbs[] = { { } }; -static struct hda_verb alc880_beep_init_verbs[] = { +static const struct hda_verb alc880_beep_init_verbs[] = { { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) }, { } }; @@ -3677,7 +3677,7 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, * F1734 pin configuration: * HP = 0x14, speaker-out = 0x15, mic = 0x18 */ -static struct hda_verb alc880_pin_f1734_init_verbs[] = { +static const struct hda_verb alc880_pin_f1734_init_verbs[] = { {0x07, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -3709,7 +3709,7 @@ static struct hda_verb alc880_pin_f1734_init_verbs[] = { * ASUS pin configuration: * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a */ -static struct hda_verb alc880_pin_asus_init_verbs[] = { +static const struct hda_verb alc880_pin_asus_init_verbs[] = { {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -3743,7 +3743,7 @@ static struct hda_verb alc880_pin_asus_init_verbs[] = { #define alc880_gpio3_init_verbs alc_gpio3_init_verbs /* Clevo m520g init */ -static struct hda_verb alc880_pin_clevo_init_verbs[] = { +static const struct hda_verb alc880_pin_clevo_init_verbs[] = { /* headphone output */ {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line-out */ @@ -3771,7 +3771,7 @@ static struct hda_verb alc880_pin_clevo_init_verbs[] = { { } }; -static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { +static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { /* change to EAPD mode */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, @@ -3814,7 +3814,7 @@ static hda_nid_t alc880_lg_dac_nids[3] = { }; /* seems analog CD is not working */ -static struct hda_input_mux alc880_lg_capture_source = { +static const struct hda_input_mux alc880_lg_capture_source = { .num_items = 3, .items = { { "Mic", 0x1 }, @@ -3824,34 +3824,34 @@ static struct hda_input_mux alc880_lg_capture_source = { }; /* 2,4,6 channel modes */ -static struct hda_verb alc880_lg_ch2_init[] = { +static const struct hda_verb alc880_lg_ch2_init[] = { /* set line-in and mic-in to input */ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { } }; -static struct hda_verb alc880_lg_ch4_init[] = { +static const struct hda_verb alc880_lg_ch4_init[] = { /* set line-in to out and mic-in to input */ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { } }; -static struct hda_verb alc880_lg_ch6_init[] = { +static const struct hda_verb alc880_lg_ch6_init[] = { /* set line-in and mic-in to output */ { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, { } }; -static struct hda_channel_mode alc880_lg_ch_modes[3] = { +static const struct hda_channel_mode alc880_lg_ch_modes[3] = { { 2, alc880_lg_ch2_init }, { 4, alc880_lg_ch4_init }, { 6, alc880_lg_ch6_init }, }; -static struct snd_kcontrol_new alc880_lg_mixer[] = { +static const struct snd_kcontrol_new alc880_lg_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3876,7 +3876,7 @@ static struct snd_kcontrol_new alc880_lg_mixer[] = { { } /* end */ }; -static struct hda_verb alc880_lg_init_verbs[] = { +static const struct hda_verb alc880_lg_init_verbs[] = { /* set capture source to mic-in */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -3930,7 +3930,7 @@ static void alc880_lg_setup(struct hda_codec *codec) * SPDIF-Out: 0x1e */ -static struct hda_input_mux alc880_lg_lw_capture_source = { +static const struct hda_input_mux alc880_lg_lw_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -3941,7 +3941,7 @@ static struct hda_input_mux alc880_lg_lw_capture_source = { #define alc880_lg_lw_modes alc880_threestack_modes -static struct snd_kcontrol_new alc880_lg_lw_mixer[] = { +static const struct snd_kcontrol_new alc880_lg_lw_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), @@ -3966,7 +3966,7 @@ static struct snd_kcontrol_new alc880_lg_lw_mixer[] = { { } /* end */ }; -static struct hda_verb alc880_lg_lw_init_verbs[] = { +static const struct hda_verb alc880_lg_lw_init_verbs[] = { {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ @@ -4004,7 +4004,7 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct snd_kcontrol_new alc880_medion_rim_mixer[] = { +static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -4014,7 +4014,7 @@ static struct snd_kcontrol_new alc880_medion_rim_mixer[] = { { } /* end */ }; -static struct hda_input_mux alc880_medion_rim_capture_source = { +static const struct hda_input_mux alc880_medion_rim_capture_source = { .num_items = 2, .items = { { "Mic", 0x0 }, @@ -4022,7 +4022,7 @@ static struct hda_input_mux alc880_medion_rim_capture_source = { }, }; -static struct hda_verb alc880_medion_rim_init_verbs[] = { +static const struct hda_verb alc880_medion_rim_init_verbs[] = { {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -4078,7 +4078,7 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list alc880_loopbacks[] = { +static const struct hda_amp_list alc880_loopbacks[] = { { 0x0b, HDA_INPUT, 0 }, { 0x0b, HDA_INPUT, 1 }, { 0x0b, HDA_INPUT, 2 }, @@ -4087,7 +4087,7 @@ static struct hda_amp_list alc880_loopbacks[] = { { } /* end */ }; -static struct hda_amp_list alc880_lg_loopbacks[] = { +static const struct hda_amp_list alc880_lg_loopbacks[] = { { 0x0b, HDA_INPUT, 1 }, { 0x0b, HDA_INPUT, 6 }, { 0x0b, HDA_INPUT, 7 }, @@ -4259,7 +4259,7 @@ static int dualmic_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream dualmic_pcm_analog_capture = { +static const struct hda_pcm_stream dualmic_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -4272,7 +4272,7 @@ static struct hda_pcm_stream dualmic_pcm_analog_capture = { /* */ -static struct hda_pcm_stream alc880_pcm_analog_playback = { +static const struct hda_pcm_stream alc880_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 8, @@ -4284,21 +4284,21 @@ static struct hda_pcm_stream alc880_pcm_analog_playback = { }, }; -static struct hda_pcm_stream alc880_pcm_analog_capture = { +static const struct hda_pcm_stream alc880_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in alc_build_pcms */ }; -static struct hda_pcm_stream alc880_pcm_analog_alt_playback = { +static const struct hda_pcm_stream alc880_pcm_analog_alt_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in alc_build_pcms */ }; -static struct hda_pcm_stream alc880_pcm_analog_alt_capture = { +static const struct hda_pcm_stream alc880_pcm_analog_alt_capture = { .substreams = 2, /* can be overridden */ .channels_min = 2, .channels_max = 2, @@ -4309,7 +4309,7 @@ static struct hda_pcm_stream alc880_pcm_analog_alt_capture = { }, }; -static struct hda_pcm_stream alc880_pcm_digital_playback = { +static const struct hda_pcm_stream alc880_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -4322,7 +4322,7 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = { }, }; -static struct hda_pcm_stream alc880_pcm_digital_capture = { +static const struct hda_pcm_stream alc880_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -4330,7 +4330,7 @@ static struct hda_pcm_stream alc880_pcm_digital_capture = { }; /* Used by alc_build_pcms to flag that a PCM has no playback stream */ -static struct hda_pcm_stream alc_pcm_null_stream = { +static const struct hda_pcm_stream alc_pcm_null_stream = { .substreams = 0, .channels_min = 0, .channels_max = 0, @@ -4507,7 +4507,7 @@ static int alc_resume(struct hda_codec *codec) /* */ -static struct hda_codec_ops alc_patch_ops = { +static const struct hda_codec_ops alc_patch_ops = { .build_controls = alc_build_controls, .build_pcms = alc_build_pcms, .init = alc_init, @@ -4546,7 +4546,7 @@ static hda_nid_t alc880_test_dac_nids[4] = { 0x02, 0x03, 0x04, 0x05 }; -static struct hda_input_mux alc880_test_capture_source = { +static const struct hda_input_mux alc880_test_capture_source = { .num_items = 7, .items = { { "In-1", 0x0 }, @@ -4559,7 +4559,7 @@ static struct hda_input_mux alc880_test_capture_source = { }, }; -static struct hda_channel_mode alc880_test_modes[4] = { +static const struct hda_channel_mode alc880_test_modes[4] = { { 2, NULL }, { 4, NULL }, { 6, NULL }, @@ -4705,7 +4705,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, .private_value = nid \ } -static struct snd_kcontrol_new alc880_test_mixer[] = { +static const struct snd_kcontrol_new alc880_test_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -4746,7 +4746,7 @@ static struct snd_kcontrol_new alc880_test_mixer[] = { { } /* end */ }; -static struct hda_verb alc880_test_init_verbs[] = { +static const struct hda_verb alc880_test_init_verbs[] = { /* Unmute inputs of 0x0c - 0x0f */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -4830,7 +4830,7 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_AUTO] = "auto", }; -static struct snd_pci_quirk alc880_cfg_tbl[] = { +static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810), SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST), @@ -4910,7 +4910,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { /* * ALC880 codec presets */ -static struct alc_config_preset alc880_presets[] = { +static const struct alc_config_preset alc880_presets[] = { [ALC880_3ST] = { .mixers = { alc880_three_stack_mixer }, .init_verbs = { alc880_volume_init_verbs, @@ -5218,7 +5218,7 @@ enum { ALC_CTL_WIDGET_MUTE, ALC_CTL_BIND_MUTE, }; -static struct snd_kcontrol_new alc880_control_templates[] = { +static const struct snd_kcontrol_new alc880_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), HDA_BIND_MUTE(NULL, 0, 0, 0), @@ -5801,7 +5801,7 @@ static void alc_init_special_input_src(struct hda_codec *codec) static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - static struct snd_kcontrol_new *caps[2][3] = { + static const struct snd_kcontrol_new *caps[2][3] = { { alc_capture_mixer_nosrc1, alc_capture_mixer_nosrc2, alc_capture_mixer_nosrc3 }, @@ -5893,7 +5893,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) -static struct snd_pci_quirk beep_white_list[] = { +static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), @@ -6030,7 +6030,7 @@ static hda_nid_t alc260_dual_adc_nids[2] = { #define ALC260_DIGOUT_NID 0x03 #define ALC260_DIGIN_NID 0x06 -static struct hda_input_mux alc260_capture_source = { +static const struct hda_input_mux alc260_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -6046,7 +6046,7 @@ static struct hda_input_mux alc260_capture_source = { * recording the mixer output on the second ADC (ADC0 doesn't have a * connection to the mixer output). */ -static struct hda_input_mux alc260_fujitsu_capture_sources[2] = { +static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = { { .num_items = 3, .items = { @@ -6070,7 +6070,7 @@ static struct hda_input_mux alc260_fujitsu_capture_sources[2] = { /* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to * the Fujitsu S702x, but jacks are marked differently. */ -static struct hda_input_mux alc260_acer_capture_sources[2] = { +static const struct hda_input_mux alc260_acer_capture_sources[2] = { { .num_items = 4, .items = { @@ -6093,7 +6093,7 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = { }; /* Maxdata Favorit 100XS */ -static struct hda_input_mux alc260_favorit100_capture_sources[2] = { +static const struct hda_input_mux alc260_favorit100_capture_sources[2] = { { .num_items = 2, .items = { @@ -6117,7 +6117,7 @@ static struct hda_input_mux alc260_favorit100_capture_sources[2] = { * element which allows changing the channel mode, so the verb list is * never used. */ -static struct hda_channel_mode alc260_modes[1] = { +static const struct hda_channel_mode alc260_modes[1] = { { 2, NULL }, }; @@ -6131,7 +6131,7 @@ static struct hda_channel_mode alc260_modes[1] = { * acer: acer + capture */ -static struct snd_kcontrol_new alc260_base_output_mixer[] = { +static const struct snd_kcontrol_new alc260_base_output_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), @@ -6141,7 +6141,7 @@ static struct snd_kcontrol_new alc260_base_output_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc260_input_mixer[] = { +static const struct snd_kcontrol_new alc260_input_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), @@ -6187,7 +6187,7 @@ static int alc260_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new alc260_hp_output_mixer[] = { +static const struct snd_kcontrol_new alc260_hp_output_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -6206,7 +6206,7 @@ static struct snd_kcontrol_new alc260_hp_output_mixer[] = { { } /* end */ }; -static struct hda_verb alc260_hp_unsol_verbs[] = { +static const struct hda_verb alc260_hp_unsol_verbs[] = { {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {}, }; @@ -6222,7 +6222,7 @@ static void alc260_hp_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_PIN; } -static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { +static const struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -6253,7 +6253,7 @@ static void alc260_hp_3013_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_PIN; } -static struct hda_bind_ctls alc260_dc7600_bind_master_vol = { +static const struct hda_bind_ctls alc260_dc7600_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT), @@ -6263,7 +6263,7 @@ static struct hda_bind_ctls alc260_dc7600_bind_master_vol = { }, }; -static struct hda_bind_ctls alc260_dc7600_bind_switch = { +static const struct hda_bind_ctls alc260_dc7600_bind_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT), @@ -6272,7 +6272,7 @@ static struct hda_bind_ctls alc260_dc7600_bind_switch = { }, }; -static struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = { +static const struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol), HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch), HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT), @@ -6280,7 +6280,7 @@ static struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = { { } /* end */ }; -static struct hda_verb alc260_hp_3013_unsol_verbs[] = { +static const struct hda_verb alc260_hp_3013_unsol_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {}, }; @@ -6300,7 +6300,7 @@ static void alc260_hp_3012_setup(struct hda_codec *codec) /* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. */ -static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { +static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), @@ -6337,7 +6337,7 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { * controls for such models. On models without a "mono speaker" the control * won't do anything. */ -static struct snd_kcontrol_new alc260_acer_mixer[] = { +static const struct snd_kcontrol_new alc260_acer_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), @@ -6358,7 +6358,7 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { /* Maxdata Favorit 100XS: one output and one input (0x12) jack */ -static struct snd_kcontrol_new alc260_favorit100_mixer[] = { +static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), @@ -6371,7 +6371,7 @@ static struct snd_kcontrol_new alc260_favorit100_mixer[] = { /* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12, * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17. */ -static struct snd_kcontrol_new alc260_will_mixer[] = { +static const struct snd_kcontrol_new alc260_will_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), @@ -6388,7 +6388,7 @@ static struct snd_kcontrol_new alc260_will_mixer[] = { /* Replacer 672V ALC260 pin usage: Mic jack = 0x12, * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f. */ -static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { +static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), @@ -6405,7 +6405,7 @@ static struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { /* * initialization verbs */ -static struct hda_verb alc260_init_verbs[] = { +static const struct hda_verb alc260_init_verbs[] = { /* Line In pin widget for input */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* CD pin widget for input */ @@ -6469,7 +6469,7 @@ static struct hda_verb alc260_init_verbs[] = { }; #if 0 /* should be identical with alc260_init_verbs? */ -static struct hda_verb alc260_hp_init_verbs[] = { +static const struct hda_verb alc260_hp_init_verbs[] = { /* Headphone and output */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, /* mono output */ @@ -6519,7 +6519,7 @@ static struct hda_verb alc260_hp_init_verbs[] = { }; #endif -static struct hda_verb alc260_hp_3013_init_verbs[] = { +static const struct hda_verb alc260_hp_3013_init_verbs[] = { /* Line out and output */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* mono output */ @@ -6572,7 +6572,7 @@ static struct hda_verb alc260_hp_3013_init_verbs[] = { * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD * audio = 0x16, internal speaker = 0x10. */ -static struct hda_verb alc260_fujitsu_init_verbs[] = { +static const struct hda_verb alc260_fujitsu_init_verbs[] = { /* Disable all GPIOs */ {0x01, AC_VERB_SET_GPIO_MASK, 0}, /* Internal speaker is connected to headphone pin */ @@ -6654,7 +6654,7 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { /* Initialisation sequence for ALC260 as configured in Acer TravelMate and * similar laptops (adapted from Fujitsu init verbs). */ -static struct hda_verb alc260_acer_init_verbs[] = { +static const struct hda_verb alc260_acer_init_verbs[] = { /* On TravelMate laptops, GPIO 0 enables the internal speaker and * the headphone jack. Turn this on and rely on the standard mute * methods whenever the user wants to turn these outputs off. @@ -6742,7 +6742,7 @@ static struct hda_verb alc260_acer_init_verbs[] = { /* Initialisation sequence for Maxdata Favorit 100XS * (adapted from Acer init verbs). */ -static struct hda_verb alc260_favorit100_init_verbs[] = { +static const struct hda_verb alc260_favorit100_init_verbs[] = { /* GPIO 0 enables the output jack. * Turn this on and rely on the standard mute * methods whenever the user wants to turn these outputs off. @@ -6822,7 +6822,7 @@ static struct hda_verb alc260_favorit100_init_verbs[] = { { } }; -static struct hda_verb alc260_will_verbs[] = { +static const struct hda_verb alc260_will_verbs[] = { {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -6832,7 +6832,7 @@ static struct hda_verb alc260_will_verbs[] = { {} }; -static struct hda_verb alc260_replacer_672v_verbs[] = { +static const struct hda_verb alc260_replacer_672v_verbs[] = { {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, {0x1a, AC_VERB_SET_PROC_COEF, 0x3050}, @@ -6874,7 +6874,7 @@ static void alc260_replacer_672v_unsol_event(struct hda_codec *codec, alc260_replacer_672v_automute(codec); } -static struct hda_verb alc260_hp_dc7600_verbs[] = { +static const struct hda_verb alc260_hp_dc7600_verbs[] = { {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -6902,7 +6902,7 @@ static hda_nid_t alc260_test_adc_nids[2] = { * the signal assignments are different. This assumes that the first ADC * is NID 0x04. */ -static struct hda_input_mux alc260_test_capture_sources[2] = { +static const struct hda_input_mux alc260_test_capture_sources[2] = { { .num_items = 7, .items = { @@ -6929,7 +6929,7 @@ static struct hda_input_mux alc260_test_capture_sources[2] = { }, }, }; -static struct snd_kcontrol_new alc260_test_mixer[] = { +static const struct snd_kcontrol_new alc260_test_mixer[] = { /* Output driver widgets */ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), @@ -6993,7 +6993,7 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { { } /* end */ }; -static struct hda_verb alc260_test_init_verbs[] = { +static const struct hda_verb alc260_test_init_verbs[] = { /* Enable all GPIOs as outputs with an initial value of 0 */ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f}, {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, @@ -7229,7 +7229,7 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc260_volume_init_verbs[] = { +static const struct hda_verb alc260_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -7319,7 +7319,7 @@ static void alc260_auto_init(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list alc260_loopbacks[] = { +static const struct hda_amp_list alc260_loopbacks[] = { { 0x07, HDA_INPUT, 0 }, { 0x07, HDA_INPUT, 1 }, { 0x07, HDA_INPUT, 2 }, @@ -7346,7 +7346,7 @@ static const struct alc_fixup alc260_fixups[] = { }, }; -static struct snd_pci_quirk alc260_fixup_tbl[] = { +static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750), {} }; @@ -7370,7 +7370,7 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_AUTO] = "auto", }; -static struct snd_pci_quirk alc260_cfg_tbl[] = { +static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), @@ -7394,7 +7394,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { {} }; -static struct alc_config_preset alc260_presets[] = { +static const struct alc_config_preset alc260_presets[] = { [ALC260_BASIC] = { .mixers = { alc260_base_output_mixer, alc260_input_mixer }, @@ -7639,7 +7639,7 @@ static int patch_alc260(struct hda_codec *codec) #define ALC1200_DIGOUT_NID 0x10 -static struct hda_channel_mode alc882_ch_modes[1] = { +static const struct hda_channel_mode alc882_ch_modes[1] = { { 8, NULL } }; @@ -7667,7 +7667,7 @@ static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; /* input MUX */ /* FIXME: should be a matrix-type input source selection */ -static struct hda_input_mux alc882_capture_source = { +static const struct hda_input_mux alc882_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -7679,7 +7679,7 @@ static struct hda_input_mux alc882_capture_source = { #define alc883_capture_source alc882_capture_source -static struct hda_input_mux alc889_capture_source = { +static const struct hda_input_mux alc889_capture_source = { .num_items = 3, .items = { { "Front Mic", 0x0 }, @@ -7688,7 +7688,7 @@ static struct hda_input_mux alc889_capture_source = { }, }; -static struct hda_input_mux mb5_capture_source = { +static const struct hda_input_mux mb5_capture_source = { .num_items = 3, .items = { { "Mic", 0x1 }, @@ -7697,7 +7697,7 @@ static struct hda_input_mux mb5_capture_source = { }, }; -static struct hda_input_mux macmini3_capture_source = { +static const struct hda_input_mux macmini3_capture_source = { .num_items = 2, .items = { { "Line", 0x2 }, @@ -7705,7 +7705,7 @@ static struct hda_input_mux macmini3_capture_source = { }, }; -static struct hda_input_mux alc883_3stack_6ch_intel = { +static const struct hda_input_mux alc883_3stack_6ch_intel = { .num_items = 4, .items = { { "Mic", 0x1 }, @@ -7715,7 +7715,7 @@ static struct hda_input_mux alc883_3stack_6ch_intel = { }, }; -static struct hda_input_mux alc883_lenovo_101e_capture_source = { +static const struct hda_input_mux alc883_lenovo_101e_capture_source = { .num_items = 2, .items = { { "Mic", 0x1 }, @@ -7723,7 +7723,7 @@ static struct hda_input_mux alc883_lenovo_101e_capture_source = { }, }; -static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { +static const struct hda_input_mux alc883_lenovo_nb0763_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -7733,7 +7733,7 @@ static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { }, }; -static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { +static const struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { .num_items = 2, .items = { { "Mic", 0x0 }, @@ -7741,7 +7741,7 @@ static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { }, }; -static struct hda_input_mux alc883_lenovo_sky_capture_source = { +static const struct hda_input_mux alc883_lenovo_sky_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -7750,7 +7750,7 @@ static struct hda_input_mux alc883_lenovo_sky_capture_source = { }, }; -static struct hda_input_mux alc883_asus_eee1601_capture_source = { +static const struct hda_input_mux alc883_asus_eee1601_capture_source = { .num_items = 2, .items = { { "Mic", 0x0 }, @@ -7758,7 +7758,7 @@ static struct hda_input_mux alc883_asus_eee1601_capture_source = { }, }; -static struct hda_input_mux alc889A_mb31_capture_source = { +static const struct hda_input_mux alc889A_mb31_capture_source = { .num_items = 2, .items = { { "Mic", 0x0 }, @@ -7769,7 +7769,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = { }, }; -static struct hda_input_mux alc889A_imac91_capture_source = { +static const struct hda_input_mux alc889A_imac91_capture_source = { .num_items = 2, .items = { { "Mic", 0x01 }, @@ -7780,14 +7780,14 @@ static struct hda_input_mux alc889A_imac91_capture_source = { /* * 2ch mode */ -static struct hda_channel_mode alc883_3ST_2ch_modes[1] = { +static const struct hda_channel_mode alc883_3ST_2ch_modes[1] = { { 2, NULL } }; /* * 2ch mode */ -static struct hda_verb alc882_3ST_ch2_init[] = { +static const struct hda_verb alc882_3ST_ch2_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -7798,7 +7798,7 @@ static struct hda_verb alc882_3ST_ch2_init[] = { /* * 4ch mode */ -static struct hda_verb alc882_3ST_ch4_init[] = { +static const struct hda_verb alc882_3ST_ch4_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -7810,7 +7810,7 @@ static struct hda_verb alc882_3ST_ch4_init[] = { /* * 6ch mode */ -static struct hda_verb alc882_3ST_ch6_init[] = { +static const struct hda_verb alc882_3ST_ch6_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, @@ -7820,7 +7820,7 @@ static struct hda_verb alc882_3ST_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc882_3ST_6ch_modes[3] = { +static const struct hda_channel_mode alc882_3ST_6ch_modes[3] = { { 2, alc882_3ST_ch2_init }, { 4, alc882_3ST_ch4_init }, { 6, alc882_3ST_ch6_init }, @@ -7831,7 +7831,7 @@ static struct hda_channel_mode alc882_3ST_6ch_modes[3] = { /* * 2ch mode */ -static struct hda_verb alc883_3ST_ch2_clevo_init[] = { +static const struct hda_verb alc883_3ST_ch2_clevo_init[] = { { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -7843,7 +7843,7 @@ static struct hda_verb alc883_3ST_ch2_clevo_init[] = { /* * 4ch mode */ -static struct hda_verb alc883_3ST_ch4_clevo_init[] = { +static const struct hda_verb alc883_3ST_ch4_clevo_init[] = { { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, @@ -7856,7 +7856,7 @@ static struct hda_verb alc883_3ST_ch4_clevo_init[] = { /* * 6ch mode */ -static struct hda_verb alc883_3ST_ch6_clevo_init[] = { +static const struct hda_verb alc883_3ST_ch6_clevo_init[] = { { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, @@ -7867,7 +7867,7 @@ static struct hda_verb alc883_3ST_ch6_clevo_init[] = { { } /* end */ }; -static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = { +static const struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = { { 2, alc883_3ST_ch2_clevo_init }, { 4, alc883_3ST_ch4_clevo_init }, { 6, alc883_3ST_ch6_clevo_init }, @@ -7877,7 +7877,7 @@ static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = { /* * 6ch mode */ -static struct hda_verb alc882_sixstack_ch6_init[] = { +static const struct hda_verb alc882_sixstack_ch6_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -7888,7 +7888,7 @@ static struct hda_verb alc882_sixstack_ch6_init[] = { /* * 8ch mode */ -static struct hda_verb alc882_sixstack_ch8_init[] = { +static const struct hda_verb alc882_sixstack_ch8_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -7896,7 +7896,7 @@ static struct hda_verb alc882_sixstack_ch8_init[] = { { } /* end */ }; -static struct hda_channel_mode alc882_sixstack_modes[2] = { +static const struct hda_channel_mode alc882_sixstack_modes[2] = { { 6, alc882_sixstack_ch6_init }, { 8, alc882_sixstack_ch8_init }, }; @@ -7904,7 +7904,7 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = { /* Macbook Air 2,1 */ -static struct hda_channel_mode alc885_mba21_ch_modes[1] = { +static const struct hda_channel_mode alc885_mba21_ch_modes[1] = { { 2, NULL }, }; @@ -7915,7 +7915,7 @@ static struct hda_channel_mode alc885_mba21_ch_modes[1] = { /* * 2ch mode */ -static struct hda_verb alc885_mbp_ch2_init[] = { +static const struct hda_verb alc885_mbp_ch2_init[] = { { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -7925,7 +7925,7 @@ static struct hda_verb alc885_mbp_ch2_init[] = { /* * 4ch mode */ -static struct hda_verb alc885_mbp_ch4_init[] = { +static const struct hda_verb alc885_mbp_ch4_init[] = { { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, @@ -7934,7 +7934,7 @@ static struct hda_verb alc885_mbp_ch4_init[] = { { } /* end */ }; -static struct hda_channel_mode alc885_mbp_4ch_modes[2] = { +static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = { { 2, alc885_mbp_ch2_init }, { 4, alc885_mbp_ch4_init }, }; @@ -7944,7 +7944,7 @@ static struct hda_channel_mode alc885_mbp_4ch_modes[2] = { * Speakers/Woofer/HP = Front * LineIn = Input */ -static struct hda_verb alc885_mb5_ch2_init[] = { +static const struct hda_verb alc885_mb5_ch2_init[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, { } /* end */ @@ -7956,14 +7956,14 @@ static struct hda_verb alc885_mb5_ch2_init[] = { * Woofer = LFE * LineIn = Surround */ -static struct hda_verb alc885_mb5_ch6_init[] = { +static const struct hda_verb alc885_mb5_ch6_init[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, { } /* end */ }; -static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { +static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 2, alc885_mb5_ch2_init }, { 6, alc885_mb5_ch6_init }, }; @@ -7973,7 +7973,7 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { /* * 2ch mode */ -static struct hda_verb alc883_4ST_ch2_init[] = { +static const struct hda_verb alc883_4ST_ch2_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, @@ -7986,7 +7986,7 @@ static struct hda_verb alc883_4ST_ch2_init[] = { /* * 4ch mode */ -static struct hda_verb alc883_4ST_ch4_init[] = { +static const struct hda_verb alc883_4ST_ch4_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, @@ -8000,7 +8000,7 @@ static struct hda_verb alc883_4ST_ch4_init[] = { /* * 6ch mode */ -static struct hda_verb alc883_4ST_ch6_init[] = { +static const struct hda_verb alc883_4ST_ch6_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -8015,7 +8015,7 @@ static struct hda_verb alc883_4ST_ch6_init[] = { /* * 8ch mode */ -static struct hda_verb alc883_4ST_ch8_init[] = { +static const struct hda_verb alc883_4ST_ch8_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, @@ -8028,7 +8028,7 @@ static struct hda_verb alc883_4ST_ch8_init[] = { { } /* end */ }; -static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { +static const struct hda_channel_mode alc883_4ST_8ch_modes[4] = { { 2, alc883_4ST_ch2_init }, { 4, alc883_4ST_ch4_init }, { 6, alc883_4ST_ch6_init }, @@ -8039,7 +8039,7 @@ static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { /* * 2ch mode */ -static struct hda_verb alc883_3ST_ch2_intel_init[] = { +static const struct hda_verb alc883_3ST_ch2_intel_init[] = { { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -8050,7 +8050,7 @@ static struct hda_verb alc883_3ST_ch2_intel_init[] = { /* * 4ch mode */ -static struct hda_verb alc883_3ST_ch4_intel_init[] = { +static const struct hda_verb alc883_3ST_ch4_intel_init[] = { { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -8062,7 +8062,7 @@ static struct hda_verb alc883_3ST_ch4_intel_init[] = { /* * 6ch mode */ -static struct hda_verb alc883_3ST_ch6_intel_init[] = { +static const struct hda_verb alc883_3ST_ch6_intel_init[] = { { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, @@ -8072,7 +8072,7 @@ static struct hda_verb alc883_3ST_ch6_intel_init[] = { { } /* end */ }; -static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { +static const struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { { 2, alc883_3ST_ch2_intel_init }, { 4, alc883_3ST_ch4_intel_init }, { 6, alc883_3ST_ch6_intel_init }, @@ -8081,7 +8081,7 @@ static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { /* * 2ch mode */ -static struct hda_verb alc889_ch2_intel_init[] = { +static const struct hda_verb alc889_ch2_intel_init[] = { { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 }, { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 }, @@ -8094,7 +8094,7 @@ static struct hda_verb alc889_ch2_intel_init[] = { /* * 6ch mode */ -static struct hda_verb alc889_ch6_intel_init[] = { +static const struct hda_verb alc889_ch6_intel_init[] = { { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, @@ -8107,7 +8107,7 @@ static struct hda_verb alc889_ch6_intel_init[] = { /* * 8ch mode */ -static struct hda_verb alc889_ch8_intel_init[] = { +static const struct hda_verb alc889_ch8_intel_init[] = { { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, @@ -8118,7 +8118,7 @@ static struct hda_verb alc889_ch8_intel_init[] = { { } /* end */ }; -static struct hda_channel_mode alc889_8ch_intel_modes[3] = { +static const struct hda_channel_mode alc889_8ch_intel_modes[3] = { { 2, alc889_ch2_intel_init }, { 6, alc889_ch6_intel_init }, { 8, alc889_ch8_intel_init }, @@ -8127,7 +8127,7 @@ static struct hda_channel_mode alc889_8ch_intel_modes[3] = { /* * 6ch mode */ -static struct hda_verb alc883_sixstack_ch6_init[] = { +static const struct hda_verb alc883_sixstack_ch6_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -8138,7 +8138,7 @@ static struct hda_verb alc883_sixstack_ch6_init[] = { /* * 8ch mode */ -static struct hda_verb alc883_sixstack_ch8_init[] = { +static const struct hda_verb alc883_sixstack_ch8_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -8146,7 +8146,7 @@ static struct hda_verb alc883_sixstack_ch8_init[] = { { } /* end */ }; -static struct hda_channel_mode alc883_sixstack_modes[2] = { +static const struct hda_channel_mode alc883_sixstack_modes[2] = { { 6, alc883_sixstack_ch6_init }, { 8, alc883_sixstack_ch8_init }, }; @@ -8155,7 +8155,7 @@ static struct hda_channel_mode alc883_sixstack_modes[2] = { /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ -static struct snd_kcontrol_new alc882_base_mixer[] = { +static const struct snd_kcontrol_new alc882_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -8182,14 +8182,14 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { /* Macbook Air 2,1 same control for HP and internal Speaker */ -static struct snd_kcontrol_new alc885_mba21_mixer[] = { +static const struct snd_kcontrol_new alc885_mba21_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), { } }; -static struct snd_kcontrol_new alc885_mbp3_mixer[] = { +static const struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT), @@ -8204,7 +8204,7 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc885_mb5_mixer[] = { +static const struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), @@ -8222,7 +8222,7 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc885_macmini3_mixer[] = { +static const struct snd_kcontrol_new alc885_macmini3_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), @@ -8237,14 +8237,14 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc885_imac91_mixer[] = { +static const struct snd_kcontrol_new alc885_imac91_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), { } /* end */ }; -static struct snd_kcontrol_new alc882_w2jc_mixer[] = { +static const struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), @@ -8257,7 +8257,7 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc882_targa_mixer[] = { +static const struct snd_kcontrol_new alc882_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -8277,7 +8277,7 @@ static struct snd_kcontrol_new alc882_targa_mixer[] = { /* Pin assignment: Front=0x14, HP = 0x15, Front = 0x16, ??? * Front Mic=0x18, Line In = 0x1a, Line In = 0x1b, CD = 0x1c */ -static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = { +static const struct snd_kcontrol_new alc882_asus_a7j_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -8294,7 +8294,7 @@ static struct snd_kcontrol_new alc882_asus_a7j_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { +static const struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -8308,7 +8308,7 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc882_chmode_mixer[] = { +static const struct snd_kcontrol_new alc882_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -8319,7 +8319,7 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = { { } /* end */ }; -static struct hda_verb alc882_base_init_verbs[] = { +static const struct hda_verb alc882_base_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -8381,7 +8381,7 @@ static struct hda_verb alc882_base_init_verbs[] = { { } }; -static struct hda_verb alc882_adc1_init_verbs[] = { +static const struct hda_verb alc882_adc1_init_verbs[] = { /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, @@ -8393,26 +8393,26 @@ static struct hda_verb alc882_adc1_init_verbs[] = { { } }; -static struct hda_verb alc882_eapd_verbs[] = { +static const struct hda_verb alc882_eapd_verbs[] = { /* change to EAPD mode */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, { } }; -static struct hda_verb alc889_eapd_verbs[] = { +static const struct hda_verb alc889_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; -static struct hda_verb alc_hp15_unsol_verbs[] = { +static const struct hda_verb alc_hp15_unsol_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {} }; -static struct hda_verb alc885_init_verbs[] = { +static const struct hda_verb alc885_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -8471,7 +8471,7 @@ static struct hda_verb alc885_init_verbs[] = { { } }; -static struct hda_verb alc885_init_input_verbs[] = { +static const struct hda_verb alc885_init_input_verbs[] = { {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, @@ -8480,7 +8480,7 @@ static struct hda_verb alc885_init_input_verbs[] = { /* Unmute Selector 24h and set the default input to front mic */ -static struct hda_verb alc889_init_input_verbs[] = { +static const struct hda_verb alc889_init_input_verbs[] = { {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, { } @@ -8490,7 +8490,7 @@ static struct hda_verb alc889_init_input_verbs[] = { #define alc883_init_verbs alc882_base_init_verbs /* Mac Pro test */ -static struct snd_kcontrol_new alc882_macpro_mixer[] = { +static const struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT), @@ -8503,7 +8503,7 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { { } /* end */ }; -static struct hda_verb alc882_macpro_init_verbs[] = { +static const struct hda_verb alc882_macpro_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -8555,7 +8555,7 @@ static struct hda_verb alc882_macpro_init_verbs[] = { }; /* Macbook 5,1 */ -static struct hda_verb alc885_mb5_init_verbs[] = { +static const struct hda_verb alc885_mb5_init_verbs[] = { /* DACs */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -8604,7 +8604,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { }; /* Macmini 3,1 */ -static struct hda_verb alc885_macmini3_init_verbs[] = { +static const struct hda_verb alc885_macmini3_init_verbs[] = { /* DACs */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -8651,7 +8651,7 @@ static struct hda_verb alc885_macmini3_init_verbs[] = { }; -static struct hda_verb alc885_mba21_init_verbs[] = { +static const struct hda_verb alc885_mba21_init_verbs[] = { /*Internal and HP Speaker Mixer*/ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -8674,7 +8674,7 @@ static struct hda_verb alc885_mba21_init_verbs[] = { /* Macbook Pro rev3 */ -static struct hda_verb alc885_mbp3_init_verbs[] = { +static const struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -8738,7 +8738,7 @@ static struct hda_verb alc885_mbp3_init_verbs[] = { }; /* iMac 9,1 */ -static struct hda_verb alc885_imac91_init_verbs[] = { +static const struct hda_verb alc885_imac91_init_verbs[] = { /* Internal Speaker Pin (0x0c) */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -8793,14 +8793,14 @@ static struct hda_verb alc885_imac91_init_verbs[] = { }; /* iMac 24 mixer. */ -static struct snd_kcontrol_new alc885_imac24_mixer[] = { +static const struct snd_kcontrol_new alc885_imac24_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT), { } /* end */ }; /* iMac 24 init verbs. */ -static struct hda_verb alc885_imac24_init_verbs[] = { +static const struct hda_verb alc885_imac24_init_verbs[] = { /* Internal speakers: output 0 (0x0c) */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -8869,7 +8869,7 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct hda_verb alc882_targa_verbs[] = { +static const struct hda_verb alc882_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -8909,7 +8909,7 @@ static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) alc882_targa_automute(codec); } -static struct hda_verb alc882_asus_a7j_verbs[] = { +static const struct hda_verb alc882_asus_a7j_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -8927,7 +8927,7 @@ static struct hda_verb alc882_asus_a7j_verbs[] = { { } /* end */ }; -static struct hda_verb alc882_asus_a7m_verbs[] = { +static const struct hda_verb alc882_asus_a7m_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -8994,7 +8994,7 @@ static void alc885_imac24_init_hook(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc883_auto_init_verbs[] = { +static const struct hda_verb alc883_auto_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -9034,7 +9034,7 @@ static struct hda_verb alc883_auto_init_verbs[] = { }; /* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ -static struct hda_verb alc889A_mb31_ch2_init[] = { +static const struct hda_verb alc889A_mb31_ch2_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ @@ -9043,7 +9043,7 @@ static struct hda_verb alc889A_mb31_ch2_init[] = { }; /* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */ -static struct hda_verb alc889A_mb31_ch4_init[] = { +static const struct hda_verb alc889A_mb31_ch4_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ @@ -9052,7 +9052,7 @@ static struct hda_verb alc889A_mb31_ch4_init[] = { }; /* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */ -static struct hda_verb alc889A_mb31_ch5_init[] = { +static const struct hda_verb alc889A_mb31_ch5_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ @@ -9061,7 +9061,7 @@ static struct hda_verb alc889A_mb31_ch5_init[] = { }; /* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */ -static struct hda_verb alc889A_mb31_ch6_init[] = { +static const struct hda_verb alc889A_mb31_ch6_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ @@ -9069,14 +9069,14 @@ static struct hda_verb alc889A_mb31_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { +static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { { 2, alc889A_mb31_ch2_init }, { 4, alc889A_mb31_ch4_init }, { 5, alc889A_mb31_ch5_init }, { 6, alc889A_mb31_ch6_init }, }; -static struct hda_verb alc883_medion_eapd_verbs[] = { +static const struct hda_verb alc883_medion_eapd_verbs[] = { /* eanable EAPD on medion laptop */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, @@ -9085,7 +9085,7 @@ static struct hda_verb alc883_medion_eapd_verbs[] = { #define alc883_base_mixer alc882_base_mixer -static struct snd_kcontrol_new alc883_mitac_mixer[] = { +static const struct snd_kcontrol_new alc883_mitac_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), @@ -9102,7 +9102,7 @@ static struct snd_kcontrol_new alc883_mitac_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { +static const struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9116,7 +9116,7 @@ static struct snd_kcontrol_new alc883_clevo_m720_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { +static const struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9130,7 +9130,7 @@ static struct snd_kcontrol_new alc883_2ch_fujitsu_pi2515_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { +static const struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -9147,7 +9147,7 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { +static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9170,7 +9170,7 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { +static const struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9194,7 +9194,7 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc885_8ch_intel_mixer[] = { +static const struct snd_kcontrol_new alc885_8ch_intel_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9218,7 +9218,7 @@ static struct snd_kcontrol_new alc885_8ch_intel_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_fivestack_mixer[] = { +static const struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9241,7 +9241,7 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_targa_mixer[] = { +static const struct snd_kcontrol_new alc883_targa_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -9262,7 +9262,7 @@ static struct snd_kcontrol_new alc883_targa_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { +static const struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -9278,7 +9278,7 @@ static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { +static const struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), @@ -9287,7 +9287,7 @@ static struct snd_kcontrol_new alc883_targa_8ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { +static const struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -9299,7 +9299,7 @@ static struct snd_kcontrol_new alc883_lenovo_101e_2ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { +static const struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -9312,7 +9312,7 @@ static struct snd_kcontrol_new alc883_lenovo_nb0763_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { +static const struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -9322,7 +9322,7 @@ static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { { } /* end */ }; -static struct hda_verb alc883_medion_wim2160_verbs[] = { +static const struct hda_verb alc883_medion_wim2160_verbs[] = { /* Unmute front mixer */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -9350,7 +9350,7 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { +static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -9362,7 +9362,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { +static const struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), @@ -9375,7 +9375,7 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { +static const struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -9400,7 +9400,7 @@ static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc889A_mb31_mixer[] = { +static const struct snd_kcontrol_new alc889A_mb31_mixer[] = { /* Output mixers */ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -9426,7 +9426,7 @@ static struct snd_kcontrol_new alc889A_mb31_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_vaiott_mixer[] = { +static const struct snd_kcontrol_new alc883_vaiott_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -9436,7 +9436,7 @@ static struct snd_kcontrol_new alc883_vaiott_mixer[] = { { } /* end */ }; -static struct hda_bind_ctls alc883_bind_cap_vol = { +static const struct hda_bind_ctls alc883_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), @@ -9445,7 +9445,7 @@ static struct hda_bind_ctls alc883_bind_cap_vol = { }, }; -static struct hda_bind_ctls alc883_bind_cap_switch = { +static const struct hda_bind_ctls alc883_bind_cap_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), @@ -9454,7 +9454,7 @@ static struct hda_bind_ctls alc883_bind_cap_switch = { }, }; -static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { +static const struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -9466,7 +9466,7 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = { +static const struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = { HDA_BIND_VOL("Capture Volume", &alc883_bind_cap_vol), HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch), { @@ -9481,7 +9481,7 @@ static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc883_chmode_mixer[] = { +static const struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -9504,7 +9504,7 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct hda_verb alc883_mitac_verbs[] = { +static const struct hda_verb alc883_mitac_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -9519,7 +9519,7 @@ static struct hda_verb alc883_mitac_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_clevo_m540r_verbs[] = { +static const struct hda_verb alc883_clevo_m540r_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -9535,7 +9535,7 @@ static struct hda_verb alc883_clevo_m540r_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_clevo_m720_verbs[] = { +static const struct hda_verb alc883_clevo_m720_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -9550,7 +9550,7 @@ static struct hda_verb alc883_clevo_m720_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { +static const struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { /* HP */ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -9564,7 +9564,7 @@ static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_targa_verbs[] = { +static const struct hda_verb alc883_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -9593,14 +9593,14 @@ static struct hda_verb alc883_targa_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_lenovo_101e_verbs[] = { +static const struct hda_verb alc883_lenovo_101e_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_FRONT_EVENT|AC_USRSP_EN}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT|AC_USRSP_EN}, { } /* end */ }; -static struct hda_verb alc883_lenovo_nb0763_verbs[] = { +static const struct hda_verb alc883_lenovo_nb0763_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, @@ -9608,7 +9608,7 @@ static struct hda_verb alc883_lenovo_nb0763_verbs[] = { { } /* end */ }; -static struct hda_verb alc888_lenovo_ms7195_verbs[] = { +static const struct hda_verb alc888_lenovo_ms7195_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -9617,7 +9617,7 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = { { } /* end */ }; -static struct hda_verb alc883_haier_w66_verbs[] = { +static const struct hda_verb alc883_haier_w66_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -9630,7 +9630,7 @@ static struct hda_verb alc883_haier_w66_verbs[] = { { } /* end */ }; -static struct hda_verb alc888_lenovo_sky_verbs[] = { +static const struct hda_verb alc888_lenovo_sky_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -9642,12 +9642,12 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = { { } /* end */ }; -static struct hda_verb alc888_6st_dell_verbs[] = { +static const struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; -static struct hda_verb alc883_vaiott_verbs[] = { +static const struct hda_verb alc883_vaiott_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -9670,7 +9670,7 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct hda_verb alc888_3st_hp_verbs[] = { +static const struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ @@ -9681,7 +9681,7 @@ static struct hda_verb alc888_3st_hp_verbs[] = { /* * 2ch mode */ -static struct hda_verb alc888_3st_hp_2ch_init[] = { +static const struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -9692,7 +9692,7 @@ static struct hda_verb alc888_3st_hp_2ch_init[] = { /* * 4ch mode */ -static struct hda_verb alc888_3st_hp_4ch_init[] = { +static const struct hda_verb alc888_3st_hp_4ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -9704,7 +9704,7 @@ static struct hda_verb alc888_3st_hp_4ch_init[] = { /* * 6ch mode */ -static struct hda_verb alc888_3st_hp_6ch_init[] = { +static const struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, @@ -9714,7 +9714,7 @@ static struct hda_verb alc888_3st_hp_6ch_init[] = { { } /* end */ }; -static struct hda_channel_mode alc888_3st_hp_modes[3] = { +static const struct hda_channel_mode alc888_3st_hp_modes[3] = { { 2, alc888_3st_hp_2ch_init }, { 4, alc888_3st_hp_4ch_init }, { 6, alc888_3st_hp_6ch_init }, @@ -9821,7 +9821,7 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct hda_verb alc883_acer_eapd_verbs[] = { +static const struct hda_verb alc883_acer_eapd_verbs[] = { /* HP Pin: output 0 (0x0c) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -9877,7 +9877,7 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct hda_verb alc888_asus_m90v_verbs[] = { +static const struct hda_verb alc888_asus_m90v_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -9904,7 +9904,7 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct hda_verb alc888_asus_eee1601_verbs[] = { +static const struct hda_verb alc888_asus_eee1601_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -9926,7 +9926,7 @@ static void alc883_eee1601_inithook(struct hda_codec *codec) alc_hp_automute(codec); } -static struct hda_verb alc889A_mb31_verbs[] = { +static const struct hda_verb alc889A_mb31_verbs[] = { /* Init rear pin (used as headphone output) */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */ @@ -10035,7 +10035,7 @@ static const char * const alc882_models[ALC882_MODEL_LAST] = { [ALC882_AUTO] = "auto", }; -static struct snd_pci_quirk alc882_cfg_tbl[] = { +static const struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), @@ -10162,7 +10162,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { }; /* codec SSID table for Intel Mac */ -static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { +static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3), @@ -10189,7 +10189,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { {} /* terminator */ }; -static struct alc_config_preset alc882_presets[] = { +static const struct alc_config_preset alc882_presets[] = { [ALC882_3ST_DIG] = { .mixers = { alc882_base_mixer }, .init_verbs = { alc882_base_init_verbs, @@ -10966,7 +10966,7 @@ static const struct alc_fixup alc882_fixups[] = { }, }; -static struct snd_pci_quirk alc882_fixup_tbl[] = { +static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), @@ -11382,7 +11382,7 @@ static hda_nid_t alc262_dmic_adc_nids[1] = { static hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 }; -static struct snd_kcontrol_new alc262_base_mixer[] = { +static const struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), @@ -11443,7 +11443,7 @@ static void alc262_hp_wildwest_setup(struct hda_codec *codec) } -static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { +static const struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -11467,7 +11467,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { +static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -11487,7 +11487,7 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { +static const struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { HDA_CODEC_VOLUME("Rear Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Rear Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Rear Mic Boost Volume", 0x18, 0, HDA_INPUT), @@ -11505,7 +11505,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_PIN; } -static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { +static const struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), @@ -11516,7 +11516,7 @@ static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { { } /* end */ }; -static struct hda_verb alc262_hp_t5735_verbs[] = { +static const struct hda_verb alc262_hp_t5735_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -11524,7 +11524,7 @@ static struct hda_verb alc262_hp_t5735_verbs[] = { { } }; -static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = { +static const struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0e, 0x0, HDA_OUTPUT), @@ -11534,7 +11534,7 @@ static struct snd_kcontrol_new alc262_hp_rp5700_mixer[] = { { } /* end */ }; -static struct hda_verb alc262_hp_rp5700_verbs[] = { +static const struct hda_verb alc262_hp_rp5700_verbs[] = { {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -11548,7 +11548,7 @@ static struct hda_verb alc262_hp_rp5700_verbs[] = { {} }; -static struct hda_input_mux alc262_hp_rp5700_capture_source = { +static const struct hda_input_mux alc262_hp_rp5700_capture_source = { .num_items = 1, .items = { { "Line", 0x1 }, @@ -11575,7 +11575,7 @@ static struct hda_input_mux alc262_hp_rp5700_capture_source = { (SUBDEV_SPEAKER(0) << 16), \ } -static struct snd_kcontrol_new alc262_hippo_mixer[] = { +static const struct snd_kcontrol_new alc262_hippo_mixer[] = { ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), @@ -11592,7 +11592,7 @@ static struct snd_kcontrol_new alc262_hippo_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_hippo1_mixer[] = { +static const struct snd_kcontrol_new alc262_hippo1_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), @@ -11630,7 +11630,7 @@ static void alc262_hippo1_setup(struct hda_codec *codec) } -static struct snd_kcontrol_new alc262_sony_mixer[] = { +static const struct snd_kcontrol_new alc262_sony_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -11640,7 +11640,7 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { +static const struct snd_kcontrol_new alc262_benq_t31_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -11651,7 +11651,7 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_tyan_mixer[] = { +static const struct snd_kcontrol_new alc262_tyan_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), @@ -11667,7 +11667,7 @@ static struct snd_kcontrol_new alc262_tyan_mixer[] = { { } /* end */ }; -static struct hda_verb alc262_tyan_verbs[] = { +static const struct hda_verb alc262_tyan_verbs[] = { /* Headphone automute */ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -11700,7 +11700,7 @@ static void alc262_tyan_setup(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc262_init_verbs[] = { +static const struct hda_verb alc262_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -11776,13 +11776,13 @@ static struct hda_verb alc262_init_verbs[] = { { } }; -static struct hda_verb alc262_eapd_verbs[] = { +static const struct hda_verb alc262_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; -static struct hda_verb alc262_hippo1_unsol_verbs[] = { +static const struct hda_verb alc262_hippo1_unsol_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, @@ -11792,7 +11792,7 @@ static struct hda_verb alc262_hippo1_unsol_verbs[] = { {} }; -static struct hda_verb alc262_sony_unsol_verbs[] = { +static const struct hda_verb alc262_sony_unsol_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, // Front Mic @@ -11802,7 +11802,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { {} }; -static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { +static const struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -11811,7 +11811,7 @@ static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { { } /* end */ }; -static struct hda_verb alc262_toshiba_s06_verbs[] = { +static const struct hda_verb alc262_toshiba_s06_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -11845,7 +11845,7 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) * 0x18 = external mic */ -static struct snd_kcontrol_new alc262_nec_mixer[] = { +static const struct snd_kcontrol_new alc262_nec_mixer[] = { HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 0x0, HDA_OUTPUT), @@ -11858,7 +11858,7 @@ static struct snd_kcontrol_new alc262_nec_mixer[] = { { } /* end */ }; -static struct hda_verb alc262_nec_verbs[] = { +static const struct hda_verb alc262_nec_verbs[] = { /* Unmute Speaker */ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -11881,7 +11881,7 @@ static struct hda_verb alc262_nec_verbs[] = { #define ALC_HP_EVENT 0x37 -static struct hda_verb alc262_fujitsu_unsol_verbs[] = { +static const struct hda_verb alc262_fujitsu_unsol_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, @@ -11889,20 +11889,20 @@ static struct hda_verb alc262_fujitsu_unsol_verbs[] = { {} }; -static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = { +static const struct hda_verb alc262_lenovo_3000_unsol_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {} }; -static struct hda_verb alc262_lenovo_3000_init_verbs[] = { +static const struct hda_verb alc262_lenovo_3000_init_verbs[] = { /* Front Mic pin: input vref at 50% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {} }; -static struct hda_input_mux alc262_fujitsu_capture_source = { +static const struct hda_input_mux alc262_fujitsu_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -11911,7 +11911,7 @@ static struct hda_input_mux alc262_fujitsu_capture_source = { }, }; -static struct hda_input_mux alc262_HP_capture_source = { +static const struct hda_input_mux alc262_HP_capture_source = { .num_items = 5, .items = { { "Mic", 0x0 }, @@ -11922,7 +11922,7 @@ static struct hda_input_mux alc262_HP_capture_source = { }, }; -static struct hda_input_mux alc262_HP_D7000_capture_source = { +static const struct hda_input_mux alc262_HP_D7000_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -11944,7 +11944,7 @@ static void alc262_fujitsu_setup(struct hda_codec *codec) } /* bind volumes of both NID 0x0c and 0x0d */ -static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { +static const struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), @@ -11953,7 +11953,7 @@ static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = { }, }; -static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { +static const struct snd_kcontrol_new alc262_fujitsu_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -11990,7 +11990,7 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_AMP; } -static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { +static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -12011,7 +12011,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { +static const struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -12024,13 +12024,13 @@ static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { }; /* additional init verbs for Benq laptops */ -static struct hda_verb alc262_EAPD_verbs[] = { +static const struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, {} }; -static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { +static const struct hda_verb alc262_benq_t31_EAPD_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, @@ -12040,7 +12040,7 @@ static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { }; /* Samsung Q1 Ultra Vista model setup */ -static struct snd_kcontrol_new alc262_ultra_mixer[] = { +static const struct snd_kcontrol_new alc262_ultra_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -12050,7 +12050,7 @@ static struct snd_kcontrol_new alc262_ultra_mixer[] = { { } /* end */ }; -static struct hda_verb alc262_ultra_verbs[] = { +static const struct hda_verb alc262_ultra_verbs[] = { /* output mixer */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -12113,7 +12113,7 @@ static void alc262_ultra_unsol_event(struct hda_codec *codec, alc262_ultra_automute(codec); } -static struct hda_input_mux alc262_ultra_capture_source = { +static const struct hda_input_mux alc262_ultra_capture_source = { .num_items = 2, .items = { { "Mic", 0x1 }, @@ -12139,7 +12139,7 @@ static int alc262_ultra_mux_enum_put(struct snd_kcontrol *kcontrol, return ret; } -static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { +static const struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), { @@ -12270,7 +12270,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc262_volume_init_verbs[] = { +static const struct hda_verb alc262_volume_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -12331,7 +12331,7 @@ static struct hda_verb alc262_volume_init_verbs[] = { { } }; -static struct hda_verb alc262_HP_BPC_init_verbs[] = { +static const struct hda_verb alc262_HP_BPC_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -12435,7 +12435,7 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { { } }; -static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { +static const struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -12531,7 +12531,7 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { { } }; -static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { +static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, @@ -12567,7 +12567,7 @@ static const struct alc_fixup alc262_fixups[] = { }, }; -static struct snd_pci_quirk alc262_fixup_tbl[] = { +static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270), {} }; @@ -12675,7 +12675,7 @@ static const char * const alc262_models[ALC262_MODEL_LAST] = { [ALC262_AUTO] = "auto", }; -static struct snd_pci_quirk alc262_cfg_tbl[] = { +static const struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series", @@ -12727,7 +12727,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { {} }; -static struct alc_config_preset alc262_presets[] = { +static const struct alc_config_preset alc262_presets[] = { [ALC262_BASIC] = { .mixers = { alc262_base_mixer }, .init_verbs = { alc262_init_verbs }, @@ -13117,7 +13117,7 @@ static hda_nid_t alc268_adc_nids_alt[1] = { static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 }; -static struct snd_kcontrol_new alc268_base_mixer[] = { +static const struct snd_kcontrol_new alc268_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -13129,7 +13129,7 @@ static struct snd_kcontrol_new alc268_base_mixer[] = { { } }; -static struct snd_kcontrol_new alc268_toshiba_mixer[] = { +static const struct snd_kcontrol_new alc268_toshiba_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), @@ -13141,7 +13141,7 @@ static struct snd_kcontrol_new alc268_toshiba_mixer[] = { }; /* bind Beep switches of both NID 0x0f and 0x10 */ -static struct hda_bind_ctls alc268_bind_beep_sw = { +static const struct hda_bind_ctls alc268_bind_beep_sw = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x0f, 3, 1, HDA_INPUT), @@ -13150,27 +13150,27 @@ static struct hda_bind_ctls alc268_bind_beep_sw = { }, }; -static struct snd_kcontrol_new alc268_beep_mixer[] = { +static const struct snd_kcontrol_new alc268_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0x1d, 0x0, HDA_INPUT), HDA_BIND_SW("Beep Playback Switch", &alc268_bind_beep_sw), { } }; -static struct hda_verb alc268_eapd_verbs[] = { +static const struct hda_verb alc268_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; /* Toshiba specific */ -static struct hda_verb alc268_toshiba_verbs[] = { +static const struct hda_verb alc268_toshiba_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } /* end */ }; /* Acer specific */ /* bind volumes of both NID 0x02 and 0x03 */ -static struct hda_bind_ctls alc268_acer_bind_master_vol = { +static const struct hda_bind_ctls alc268_acer_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), @@ -13192,7 +13192,7 @@ static void alc268_acer_setup(struct hda_codec *codec) #define alc268_acer_master_sw_get alc262_hp_master_sw_get #define alc268_acer_master_sw_put alc262_hp_master_sw_put -static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { +static const struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { /* output mixer control */ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), { @@ -13207,7 +13207,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { } }; -static struct snd_kcontrol_new alc268_acer_mixer[] = { +static const struct snd_kcontrol_new alc268_acer_mixer[] = { /* output mixer control */ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), { @@ -13224,7 +13224,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { } }; -static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { +static const struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { /* output mixer control */ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), { @@ -13240,7 +13240,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { } }; -static struct hda_verb alc268_acer_aspire_one_verbs[] = { +static const struct hda_verb alc268_acer_aspire_one_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, @@ -13250,7 +13250,7 @@ static struct hda_verb alc268_acer_aspire_one_verbs[] = { { } }; -static struct hda_verb alc268_acer_verbs[] = { +static const struct hda_verb alc268_acer_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -13279,7 +13279,7 @@ static void alc268_acer_lc_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static struct snd_kcontrol_new alc268_dell_mixer[] = { +static const struct snd_kcontrol_new alc268_dell_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -13290,7 +13290,7 @@ static struct snd_kcontrol_new alc268_dell_mixer[] = { { } }; -static struct hda_verb alc268_dell_verbs[] = { +static const struct hda_verb alc268_dell_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, @@ -13314,7 +13314,7 @@ static void alc268_dell_setup(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_PIN; } -static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { +static const struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), @@ -13326,7 +13326,7 @@ static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { { } }; -static struct hda_verb alc267_quanta_il1_verbs[] = { +static const struct hda_verb alc267_quanta_il1_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, { } @@ -13349,7 +13349,7 @@ static void alc267_quanta_il1_setup(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc268_base_init_verbs[] = { +static const struct hda_verb alc268_base_init_verbs[] = { /* Unmute DAC0-1 and set vol = 0 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -13397,7 +13397,7 @@ static struct hda_verb alc268_base_init_verbs[] = { /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc268_volume_init_verbs[] = { +static const struct hda_verb alc268_volume_init_verbs[] = { /* set output DAC */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -13423,20 +13423,20 @@ static struct hda_verb alc268_volume_init_verbs[] = { { } }; -static struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = { +static const struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), { } /* end */ }; -static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { +static const struct snd_kcontrol_new alc268_capture_alt_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), _DEFINE_CAPSRC(1), { } /* end */ }; -static struct snd_kcontrol_new alc268_capture_mixer[] = { +static const struct snd_kcontrol_new alc268_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT), @@ -13445,7 +13445,7 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = { { } /* end */ }; -static struct hda_input_mux alc268_capture_source = { +static const struct hda_input_mux alc268_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -13455,7 +13455,7 @@ static struct hda_input_mux alc268_capture_source = { }, }; -static struct hda_input_mux alc268_acer_capture_source = { +static const struct hda_input_mux alc268_acer_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -13464,7 +13464,7 @@ static struct hda_input_mux alc268_acer_capture_source = { }, }; -static struct hda_input_mux alc268_acer_dmic_capture_source = { +static const struct hda_input_mux alc268_acer_dmic_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -13474,7 +13474,7 @@ static struct hda_input_mux alc268_acer_dmic_capture_source = { }; #ifdef CONFIG_SND_DEBUG -static struct snd_kcontrol_new alc268_test_mixer[] = { +static const struct snd_kcontrol_new alc268_test_mixer[] = { /* Volume widgets */ HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -13806,7 +13806,7 @@ static const char * const alc268_models[ALC268_MODEL_LAST] = { [ALC268_AUTO] = "auto", }; -static struct snd_pci_quirk alc268_cfg_tbl[] = { +static const struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x011e, "Acer Aspire 5720z", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER), @@ -13831,7 +13831,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { }; /* Toshiba laptops have no unique PCI SSID but only codec SSID */ -static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { +static const struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO), SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO), SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", @@ -13839,7 +13839,7 @@ static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { {} }; -static struct alc_config_preset alc268_presets[] = { +static const struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer, alc268_capture_nosrc_mixer }, @@ -14138,7 +14138,7 @@ static hda_nid_t alc269_adc_candidates[] = { #define alc269_modes alc260_modes #define alc269_capture_source alc880_lg_lw_capture_source -static struct snd_kcontrol_new alc269_base_mixer[] = { +static const struct snd_kcontrol_new alc269_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), @@ -14154,7 +14154,7 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { +static const struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { /* output mixer control */ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), { @@ -14175,7 +14175,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { } }; -static struct snd_kcontrol_new alc269_lifebook_mixer[] = { +static const struct snd_kcontrol_new alc269_lifebook_mixer[] = { /* output mixer control */ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), { @@ -14199,7 +14199,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -static struct snd_kcontrol_new alc269_laptop_mixer[] = { +static const struct snd_kcontrol_new alc269_laptop_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), @@ -14207,7 +14207,7 @@ static struct snd_kcontrol_new alc269_laptop_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc269vb_laptop_mixer[] = { +static const struct snd_kcontrol_new alc269vb_laptop_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -14215,14 +14215,14 @@ static struct snd_kcontrol_new alc269vb_laptop_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc269_asus_mixer[] = { +static const struct snd_kcontrol_new alc269_asus_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x0, HDA_INPUT), { } /* end */ }; /* capture mixer elements */ -static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { +static const struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), @@ -14230,14 +14230,14 @@ static struct snd_kcontrol_new alc269_laptop_analog_capture_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = { +static const struct snd_kcontrol_new alc269_laptop_digital_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), { } /* end */ }; -static struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = { +static const struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), @@ -14245,7 +14245,7 @@ static struct snd_kcontrol_new alc269vb_laptop_analog_capture_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = { +static const struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), @@ -14255,7 +14255,7 @@ static struct snd_kcontrol_new alc269vb_laptop_digital_capture_mixer[] = { /* FSC amilo */ #define alc269_fujitsu_mixer alc269_laptop_mixer -static struct hda_verb alc269_quanta_fl1_verbs[] = { +static const struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -14265,7 +14265,7 @@ static struct hda_verb alc269_quanta_fl1_verbs[] = { { } }; -static struct hda_verb alc269_lifebook_verbs[] = { +static const struct hda_verb alc269_lifebook_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -14378,7 +14378,7 @@ static void alc269_lifebook_init_hook(struct hda_codec *codec) alc269_lifebook_mic_autoswitch(codec); } -static struct hda_verb alc269_laptop_dmic_init_verbs[] = { +static const struct hda_verb alc269_laptop_dmic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -14389,7 +14389,7 @@ static struct hda_verb alc269_laptop_dmic_init_verbs[] = { {} }; -static struct hda_verb alc269_laptop_amic_init_verbs[] = { +static const struct hda_verb alc269_laptop_amic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -14399,7 +14399,7 @@ static struct hda_verb alc269_laptop_amic_init_verbs[] = { {} }; -static struct hda_verb alc269vb_laptop_dmic_init_verbs[] = { +static const struct hda_verb alc269vb_laptop_dmic_init_verbs[] = { {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x22, AC_VERB_SET_CONNECT_SEL, 0x06}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -14410,7 +14410,7 @@ static struct hda_verb alc269vb_laptop_dmic_init_verbs[] = { {} }; -static struct hda_verb alc269vb_laptop_amic_init_verbs[] = { +static const struct hda_verb alc269vb_laptop_amic_init_verbs[] = { {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x22, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x02, AC_VERB_SET_AMP_GAIN_MUTE, 0xb026 }, @@ -14421,7 +14421,7 @@ static struct hda_verb alc269vb_laptop_amic_init_verbs[] = { {} }; -static struct hda_verb alc271_acer_dmic_verbs[] = { +static const struct hda_verb alc271_acer_dmic_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x0d}, {0x20, AC_VERB_SET_PROC_COEF, 0x4000}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -14498,7 +14498,7 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc269_init_verbs[] = { +static const struct hda_verb alc269_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -14541,7 +14541,7 @@ static struct hda_verb alc269_init_verbs[] = { { } }; -static struct hda_verb alc269vb_init_verbs[] = { +static const struct hda_verb alc269vb_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -14599,7 +14599,7 @@ static struct hda_verb alc269vb_init_verbs[] = { #define alc269_pcm_digital_playback alc880_pcm_digital_playback #define alc269_pcm_digital_capture alc880_pcm_digital_capture -static struct hda_pcm_stream alc269_44k_pcm_analog_playback = { +static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 8, @@ -14612,7 +14612,7 @@ static struct hda_pcm_stream alc269_44k_pcm_analog_playback = { }, }; -static struct hda_pcm_stream alc269_44k_pcm_analog_capture = { +static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -14825,7 +14825,7 @@ static void alc269_fixup_hweq(struct hda_codec *codec, static void alc271_fixup_dmic(struct hda_codec *codec, const struct alc_fixup *fix, int action) { - static struct hda_verb verbs[] = { + static const struct hda_verb verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x0d}, {0x20, AC_VERB_SET_PROC_COEF, 0x4000}, {} @@ -14908,7 +14908,7 @@ static const struct alc_fixup alc269_fixups[] = { }, }; -static struct snd_pci_quirk alc269_fixup_tbl[] = { +static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), @@ -14939,7 +14939,7 @@ static const char * const alc269_models[ALC269_MODEL_LAST] = { [ALC269_AUTO] = "auto", }; -static struct snd_pci_quirk alc269_cfg_tbl[] = { +static const struct snd_pci_quirk alc269_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1), SND_PCI_QUIRK(0x1025, 0x047c, "ACER ZGA", ALC271_ACER), SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", @@ -14997,7 +14997,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { {} }; -static struct alc_config_preset alc269_presets[] = { +static const struct alc_config_preset alc269_presets[] = { [ALC269_BASIC] = { .mixers = { alc269_base_mixer }, .init_verbs = { alc269_init_verbs }, @@ -15301,7 +15301,7 @@ static int patch_alc269(struct hda_codec *codec) * set the path ways for 2 channel output * need to set the codec line out and mic 1 pin widgets to inputs */ -static struct hda_verb alc861_threestack_ch2_init[] = { +static const struct hda_verb alc861_threestack_ch2_init[] = { /* set pin widget 1Ah (line in) for input */ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, /* set pin widget 18h (mic1/2) for input, for mic also enable @@ -15320,7 +15320,7 @@ static struct hda_verb alc861_threestack_ch2_init[] = { * 6ch mode * need to set the codec line out and mic 1 pin widgets to outputs */ -static struct hda_verb alc861_threestack_ch6_init[] = { +static const struct hda_verb alc861_threestack_ch6_init[] = { /* set pin widget 1Ah (line in) for output (Back Surround)*/ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, /* set pin widget 18h (mic1) for output (CLFE)*/ @@ -15337,30 +15337,30 @@ static struct hda_verb alc861_threestack_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc861_threestack_modes[2] = { +static const struct hda_channel_mode alc861_threestack_modes[2] = { { 2, alc861_threestack_ch2_init }, { 6, alc861_threestack_ch6_init }, }; /* Set mic1 as input and unmute the mixer */ -static struct hda_verb alc861_uniwill_m31_ch2_init[] = { +static const struct hda_verb alc861_uniwill_m31_ch2_init[] = { { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/ { } /* end */ }; /* Set mic1 as output and mute mixer */ -static struct hda_verb alc861_uniwill_m31_ch4_init[] = { +static const struct hda_verb alc861_uniwill_m31_ch4_init[] = { { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/ { } /* end */ }; -static struct hda_channel_mode alc861_uniwill_m31_modes[2] = { +static const struct hda_channel_mode alc861_uniwill_m31_modes[2] = { { 2, alc861_uniwill_m31_ch2_init }, { 4, alc861_uniwill_m31_ch4_init }, }; /* Set mic1 and line-in as input and unmute the mixer */ -static struct hda_verb alc861_asus_ch2_init[] = { +static const struct hda_verb alc861_asus_ch2_init[] = { /* set pin widget 1Ah (line in) for input */ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, /* set pin widget 18h (mic1/2) for input, for mic also enable @@ -15376,7 +15376,7 @@ static struct hda_verb alc861_asus_ch2_init[] = { { } /* end */ }; /* Set mic1 nad line-in as output and mute mixer */ -static struct hda_verb alc861_asus_ch6_init[] = { +static const struct hda_verb alc861_asus_ch6_init[] = { /* set pin widget 1Ah (line in) for output (Back Surround)*/ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, /* { 0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, */ @@ -15394,14 +15394,14 @@ static struct hda_verb alc861_asus_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc861_asus_modes[2] = { +static const struct hda_channel_mode alc861_asus_modes[2] = { { 2, alc861_asus_ch2_init }, { 6, alc861_asus_ch6_init }, }; /* patch-ALC861 */ -static struct snd_kcontrol_new alc861_base_mixer[] = { +static const struct snd_kcontrol_new alc861_base_mixer[] = { /* output mixer control */ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), @@ -15424,7 +15424,7 @@ static struct snd_kcontrol_new alc861_base_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc861_3ST_mixer[] = { +static const struct snd_kcontrol_new alc861_3ST_mixer[] = { /* output mixer control */ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), @@ -15455,7 +15455,7 @@ static struct snd_kcontrol_new alc861_3ST_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc861_toshiba_mixer[] = { +static const struct snd_kcontrol_new alc861_toshiba_mixer[] = { /* output mixer control */ HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), @@ -15464,7 +15464,7 @@ static struct snd_kcontrol_new alc861_toshiba_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = { +static const struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = { /* output mixer control */ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), @@ -15495,7 +15495,7 @@ static struct snd_kcontrol_new alc861_uniwill_m31_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc861_asus_mixer[] = { +static const struct snd_kcontrol_new alc861_asus_mixer[] = { /* output mixer control */ HDA_CODEC_MUTE("Front Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Surround Playback Switch", 0x06, 0x0, HDA_OUTPUT), @@ -15527,7 +15527,7 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = { }; /* additional mixer */ -static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { +static const struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), { } @@ -15536,7 +15536,7 @@ static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc861_base_init_verbs[] = { +static const struct hda_verb alc861_base_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -15602,7 +15602,7 @@ static struct hda_verb alc861_base_init_verbs[] = { { } }; -static struct hda_verb alc861_threestack_init_verbs[] = { +static const struct hda_verb alc861_threestack_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -15663,7 +15663,7 @@ static struct hda_verb alc861_threestack_init_verbs[] = { { } }; -static struct hda_verb alc861_uniwill_m31_init_verbs[] = { +static const struct hda_verb alc861_uniwill_m31_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -15725,7 +15725,7 @@ static struct hda_verb alc861_uniwill_m31_init_verbs[] = { { } }; -static struct hda_verb alc861_asus_init_verbs[] = { +static const struct hda_verb alc861_asus_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -15791,7 +15791,7 @@ static struct hda_verb alc861_asus_init_verbs[] = { }; /* additional init verbs for ASUS laptops */ -static struct hda_verb alc861_asus_laptop_init_verbs[] = { +static const struct hda_verb alc861_asus_laptop_init_verbs[] = { { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x45 }, /* HP-out */ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2) }, /* mute line-in */ { } @@ -15800,7 +15800,7 @@ static struct hda_verb alc861_asus_laptop_init_verbs[] = { /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc861_auto_init_verbs[] = { +static const struct hda_verb alc861_auto_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -15849,7 +15849,7 @@ static struct hda_verb alc861_auto_init_verbs[] = { { } }; -static struct hda_verb alc861_toshiba_init_verbs[] = { +static const struct hda_verb alc861_toshiba_init_verbs[] = { {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, { } @@ -15882,7 +15882,7 @@ static void alc861_toshiba_unsol_event(struct hda_codec *codec, #define ALC861_DIGOUT_NID 0x07 -static struct hda_channel_mode alc861_8ch_modes[1] = { +static const struct hda_channel_mode alc861_8ch_modes[1] = { { 8, NULL } }; @@ -15901,7 +15901,7 @@ static hda_nid_t alc861_adc_nids[1] = { 0x08, }; -static struct hda_input_mux alc861_capture_source = { +static const struct hda_input_mux alc861_capture_source = { .num_items = 5, .items = { { "Mic", 0x0 }, @@ -16174,7 +16174,7 @@ static void alc861_auto_init(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list alc861_loopbacks[] = { +static const struct hda_amp_list alc861_loopbacks[] = { { 0x15, HDA_INPUT, 0 }, { 0x15, HDA_INPUT, 1 }, { 0x15, HDA_INPUT, 2 }, @@ -16199,7 +16199,7 @@ static const char * const alc861_models[ALC861_MODEL_LAST] = { [ALC861_AUTO] = "auto", }; -static struct snd_pci_quirk alc861_cfg_tbl[] = { +static const struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST), SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), @@ -16223,7 +16223,7 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { {} }; -static struct alc_config_preset alc861_presets[] = { +static const struct alc_config_preset alc861_presets[] = { [ALC861_3ST] = { .mixers = { alc861_3ST_mixer }, .init_verbs = { alc861_threestack_init_verbs }, @@ -16346,7 +16346,7 @@ static const struct alc_fixup alc861_fixups[] = { }, }; -static struct snd_pci_quirk alc861_fixup_tbl[] = { +static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} }; @@ -16465,7 +16465,7 @@ static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 }; /* input MUX */ /* FIXME: should be a matrix-type input source selection */ -static struct hda_input_mux alc861vd_capture_source = { +static const struct hda_input_mux alc861vd_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -16475,7 +16475,7 @@ static struct hda_input_mux alc861vd_capture_source = { }, }; -static struct hda_input_mux alc861vd_dallas_capture_source = { +static const struct hda_input_mux alc861vd_dallas_capture_source = { .num_items = 2, .items = { { "Mic", 0x0 }, @@ -16483,7 +16483,7 @@ static struct hda_input_mux alc861vd_dallas_capture_source = { }, }; -static struct hda_input_mux alc861vd_hp_capture_source = { +static const struct hda_input_mux alc861vd_hp_capture_source = { .num_items = 2, .items = { { "Front Mic", 0x0 }, @@ -16494,14 +16494,14 @@ static struct hda_input_mux alc861vd_hp_capture_source = { /* * 2ch mode */ -static struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = { +static const struct hda_channel_mode alc861vd_3stack_2ch_modes[1] = { { 2, NULL } }; /* * 6ch mode */ -static struct hda_verb alc861vd_6stack_ch6_init[] = { +static const struct hda_verb alc861vd_6stack_ch6_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -16512,7 +16512,7 @@ static struct hda_verb alc861vd_6stack_ch6_init[] = { /* * 8ch mode */ -static struct hda_verb alc861vd_6stack_ch8_init[] = { +static const struct hda_verb alc861vd_6stack_ch8_init[] = { { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -16520,12 +16520,12 @@ static struct hda_verb alc861vd_6stack_ch8_init[] = { { } /* end */ }; -static struct hda_channel_mode alc861vd_6stack_modes[2] = { +static const struct hda_channel_mode alc861vd_6stack_modes[2] = { { 6, alc861vd_6stack_ch6_init }, { 8, alc861vd_6stack_ch8_init }, }; -static struct snd_kcontrol_new alc861vd_chmode_mixer[] = { +static const struct snd_kcontrol_new alc861vd_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -16539,7 +16539,7 @@ static struct snd_kcontrol_new alc861vd_chmode_mixer[] = { /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ -static struct snd_kcontrol_new alc861vd_6st_mixer[] = { +static const struct snd_kcontrol_new alc861vd_6st_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -16575,7 +16575,7 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc861vd_3st_mixer[] = { +static const struct snd_kcontrol_new alc861vd_3st_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -16598,7 +16598,7 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { +static const struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), /*HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),*/ HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -16622,7 +16622,7 @@ static struct snd_kcontrol_new alc861vd_lenovo_mixer[] = { /* Pin assignment: Speaker=0x14, HP = 0x15, * Mic=0x18, Internal Mic = 0x19, CD = 0x1c, PC Beep = 0x1d */ -static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { +static const struct snd_kcontrol_new alc861vd_dallas_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -16639,7 +16639,7 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { /* Pin assignment: Speaker=0x14, Line-out = 0x15, * Front Mic=0x18, ATAPI Mic = 0x19, */ -static struct snd_kcontrol_new alc861vd_hp_mixer[] = { +static const struct snd_kcontrol_new alc861vd_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -16655,7 +16655,7 @@ static struct snd_kcontrol_new alc861vd_hp_mixer[] = { /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc861vd_volume_init_verbs[] = { +static const struct hda_verb alc861vd_volume_init_verbs[] = { /* * Unmute ADC0 and set the default input to mic-in */ @@ -16705,7 +16705,7 @@ static struct hda_verb alc861vd_volume_init_verbs[] = { * 3-stack pin configuration: * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b */ -static struct hda_verb alc861vd_3stack_init_verbs[] = { +static const struct hda_verb alc861vd_3stack_init_verbs[] = { /* * Set pin mode and muting */ @@ -16736,7 +16736,7 @@ static struct hda_verb alc861vd_3stack_init_verbs[] = { /* * 6-stack pin configuration: */ -static struct hda_verb alc861vd_6stack_init_verbs[] = { +static const struct hda_verb alc861vd_6stack_init_verbs[] = { /* * Set pin mode and muting */ @@ -16777,18 +16777,18 @@ static struct hda_verb alc861vd_6stack_init_verbs[] = { { } }; -static struct hda_verb alc861vd_eapd_verbs[] = { +static const struct hda_verb alc861vd_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; -static struct hda_verb alc660vd_eapd_verbs[] = { +static const struct hda_verb alc660vd_eapd_verbs[] = { {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; -static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { +static const struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, @@ -16825,7 +16825,7 @@ static void alc861vd_lenovo_unsol_event(struct hda_codec *codec, } } -static struct hda_verb alc861vd_dallas_verbs[] = { +static const struct hda_verb alc861vd_dallas_verbs[] = { {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -16907,7 +16907,7 @@ static const char * const alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC861VD_AUTO] = "auto", }; -static struct snd_pci_quirk alc861vd_cfg_tbl[] = { +static const struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), @@ -16926,7 +16926,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { {} }; -static struct alc_config_preset alc861vd_presets[] = { +static const struct alc_config_preset alc861vd_presets[] = { [ALC660VD_3ST] = { .mixers = { alc861vd_3st_mixer }, .init_verbs = { alc861vd_volume_init_verbs, @@ -17321,7 +17321,7 @@ static const struct alc_fixup alc861vd_fixups[] = { }, }; -static struct snd_pci_quirk alc861vd_fixup_tbl[] = { +static const struct snd_pci_quirk alc861vd_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1), {} }; @@ -17452,7 +17452,7 @@ static hda_nid_t alc272_capsrc_nids[1] = { 0x23 }; /* input MUX */ /* FIXME: should be a matrix-type input source selection */ -static struct hda_input_mux alc662_capture_source = { +static const struct hda_input_mux alc662_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -17462,7 +17462,7 @@ static struct hda_input_mux alc662_capture_source = { }, }; -static struct hda_input_mux alc662_lenovo_101e_capture_source = { +static const struct hda_input_mux alc662_lenovo_101e_capture_source = { .num_items = 2, .items = { { "Mic", 0x1 }, @@ -17470,7 +17470,7 @@ static struct hda_input_mux alc662_lenovo_101e_capture_source = { }, }; -static struct hda_input_mux alc663_capture_source = { +static const struct hda_input_mux alc663_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -17480,7 +17480,7 @@ static struct hda_input_mux alc663_capture_source = { }; #if 0 /* set to 1 for testing other input sources below */ -static struct hda_input_mux alc272_nc10_capture_source = { +static const struct hda_input_mux alc272_nc10_capture_source = { .num_items = 16, .items = { { "Autoselect Mic", 0x0 }, @@ -17506,14 +17506,14 @@ static struct hda_input_mux alc272_nc10_capture_source = { /* * 2ch mode */ -static struct hda_channel_mode alc662_3ST_2ch_modes[1] = { +static const struct hda_channel_mode alc662_3ST_2ch_modes[1] = { { 2, NULL } }; /* * 2ch mode */ -static struct hda_verb alc662_3ST_ch2_init[] = { +static const struct hda_verb alc662_3ST_ch2_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -17524,7 +17524,7 @@ static struct hda_verb alc662_3ST_ch2_init[] = { /* * 6ch mode */ -static struct hda_verb alc662_3ST_ch6_init[] = { +static const struct hda_verb alc662_3ST_ch6_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, @@ -17534,7 +17534,7 @@ static struct hda_verb alc662_3ST_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc662_3ST_6ch_modes[2] = { +static const struct hda_channel_mode alc662_3ST_6ch_modes[2] = { { 2, alc662_3ST_ch2_init }, { 6, alc662_3ST_ch6_init }, }; @@ -17542,7 +17542,7 @@ static struct hda_channel_mode alc662_3ST_6ch_modes[2] = { /* * 2ch mode */ -static struct hda_verb alc662_sixstack_ch6_init[] = { +static const struct hda_verb alc662_sixstack_ch6_init[] = { { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -17552,14 +17552,14 @@ static struct hda_verb alc662_sixstack_ch6_init[] = { /* * 6ch mode */ -static struct hda_verb alc662_sixstack_ch8_init[] = { +static const struct hda_verb alc662_sixstack_ch8_init[] = { { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { } /* end */ }; -static struct hda_channel_mode alc662_5stack_modes[2] = { +static const struct hda_channel_mode alc662_5stack_modes[2] = { { 2, alc662_sixstack_ch6_init }, { 6, alc662_sixstack_ch8_init }, }; @@ -17568,7 +17568,7 @@ static struct hda_channel_mode alc662_5stack_modes[2] = { * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ -static struct snd_kcontrol_new alc662_base_mixer[] = { +static const struct snd_kcontrol_new alc662_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), @@ -17592,7 +17592,7 @@ static struct snd_kcontrol_new alc662_base_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { +static const struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -17607,7 +17607,7 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { +static const struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x0c, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -17628,7 +17628,7 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { +static const struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x02, 2, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -17641,7 +17641,7 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { +static const struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -17655,7 +17655,7 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { +static const struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { ALC262_HIPPO_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -17669,7 +17669,7 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { { } /* end */ }; -static struct hda_bind_ctls alc663_asus_bind_master_vol = { +static const struct hda_bind_ctls alc663_asus_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), @@ -17678,7 +17678,7 @@ static struct hda_bind_ctls alc663_asus_bind_master_vol = { }, }; -static struct hda_bind_ctls alc663_asus_one_bind_switch = { +static const struct hda_bind_ctls alc663_asus_one_bind_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), @@ -17687,7 +17687,7 @@ static struct hda_bind_ctls alc663_asus_one_bind_switch = { }, }; -static struct snd_kcontrol_new alc663_m51va_mixer[] = { +static const struct snd_kcontrol_new alc663_m51va_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -17695,7 +17695,7 @@ static struct snd_kcontrol_new alc663_m51va_mixer[] = { { } /* end */ }; -static struct hda_bind_ctls alc663_asus_tree_bind_switch = { +static const struct hda_bind_ctls alc663_asus_tree_bind_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), @@ -17705,7 +17705,7 @@ static struct hda_bind_ctls alc663_asus_tree_bind_switch = { }, }; -static struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = { +static const struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -17716,7 +17716,7 @@ static struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = { { } /* end */ }; -static struct hda_bind_ctls alc663_asus_four_bind_switch = { +static const struct hda_bind_ctls alc663_asus_four_bind_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), @@ -17726,7 +17726,7 @@ static struct hda_bind_ctls alc663_asus_four_bind_switch = { }, }; -static struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = { +static const struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), @@ -17736,7 +17736,7 @@ static struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc662_1bjd_mixer[] = { +static const struct snd_kcontrol_new alc662_1bjd_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -17747,7 +17747,7 @@ static struct snd_kcontrol_new alc662_1bjd_mixer[] = { { } /* end */ }; -static struct hda_bind_ctls alc663_asus_two_bind_master_vol = { +static const struct hda_bind_ctls alc663_asus_two_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), @@ -17756,7 +17756,7 @@ static struct hda_bind_ctls alc663_asus_two_bind_master_vol = { }, }; -static struct hda_bind_ctls alc663_asus_two_bind_switch = { +static const struct hda_bind_ctls alc663_asus_two_bind_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), @@ -17765,7 +17765,7 @@ static struct hda_bind_ctls alc663_asus_two_bind_switch = { }, }; -static struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = { +static const struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc663_asus_two_bind_master_vol), HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch), @@ -17776,7 +17776,7 @@ static struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = { +static const struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = { HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -17786,7 +17786,7 @@ static struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc663_g71v_mixer[] = { +static const struct snd_kcontrol_new alc663_g71v_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Playback Volume", 0x03, 0x0, HDA_OUTPUT), @@ -17800,7 +17800,7 @@ static struct snd_kcontrol_new alc663_g71v_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc663_g50v_mixer[] = { +static const struct snd_kcontrol_new alc663_g50v_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -17814,7 +17814,7 @@ static struct snd_kcontrol_new alc663_g50v_mixer[] = { { } /* end */ }; -static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = { +static const struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), @@ -17826,7 +17826,7 @@ static struct hda_bind_ctls alc663_asus_mode7_8_all_bind_switch = { }, }; -static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = { +static const struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), @@ -17835,7 +17835,7 @@ static struct hda_bind_ctls alc663_asus_mode7_8_sp_bind_switch = { }, }; -static struct snd_kcontrol_new alc663_mode7_mixer[] = { +static const struct snd_kcontrol_new alc663_mode7_mixer[] = { HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch), HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol), HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch), @@ -17848,7 +17848,7 @@ static struct snd_kcontrol_new alc663_mode7_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc663_mode8_mixer[] = { +static const struct snd_kcontrol_new alc663_mode8_mixer[] = { HDA_BIND_SW("Master Playback Switch", &alc663_asus_mode7_8_all_bind_switch), HDA_BIND_VOL("Speaker Playback Volume", &alc663_asus_bind_master_vol), HDA_BIND_SW("Speaker Playback Switch", &alc663_asus_mode7_8_sp_bind_switch), @@ -17860,7 +17860,7 @@ static struct snd_kcontrol_new alc663_mode8_mixer[] = { }; -static struct snd_kcontrol_new alc662_chmode_mixer[] = { +static const struct snd_kcontrol_new alc662_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -17871,7 +17871,7 @@ static struct snd_kcontrol_new alc662_chmode_mixer[] = { { } /* end */ }; -static struct hda_verb alc662_init_verbs[] = { +static const struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -17920,33 +17920,33 @@ static struct hda_verb alc662_init_verbs[] = { { } }; -static struct hda_verb alc662_eapd_init_verbs[] = { +static const struct hda_verb alc662_eapd_init_verbs[] = { /* always trun on EAPD */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; -static struct hda_verb alc662_sue_init_verbs[] = { +static const struct hda_verb alc662_sue_init_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_FRONT_EVENT}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC880_HP_EVENT}, {} }; -static struct hda_verb alc662_eeepc_sue_init_verbs[] = { +static const struct hda_verb alc662_eeepc_sue_init_verbs[] = { {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {} }; /* Set Unsolicited Event*/ -static struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = { +static const struct hda_verb alc662_eeepc_ep20_sue_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {} }; -static struct hda_verb alc663_m51va_init_verbs[] = { +static const struct hda_verb alc663_m51va_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -17959,7 +17959,7 @@ static struct hda_verb alc663_m51va_init_verbs[] = { {} }; -static struct hda_verb alc663_21jd_amic_init_verbs[] = { +static const struct hda_verb alc663_21jd_amic_init_verbs[] = { {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ @@ -17970,7 +17970,7 @@ static struct hda_verb alc663_21jd_amic_init_verbs[] = { {} }; -static struct hda_verb alc662_1bjd_amic_init_verbs[] = { +static const struct hda_verb alc662_1bjd_amic_init_verbs[] = { {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -17982,7 +17982,7 @@ static struct hda_verb alc662_1bjd_amic_init_verbs[] = { {} }; -static struct hda_verb alc663_15jd_amic_init_verbs[] = { +static const struct hda_verb alc663_15jd_amic_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ @@ -17993,7 +17993,7 @@ static struct hda_verb alc663_15jd_amic_init_verbs[] = { {} }; -static struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = { +static const struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = { {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -18009,7 +18009,7 @@ static struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = { {} }; -static struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = { +static const struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = { {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -18025,7 +18025,7 @@ static struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = { {} }; -static struct hda_verb alc663_g71v_init_verbs[] = { +static const struct hda_verb alc663_g71v_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ /* {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, */ /* Headphone */ @@ -18040,7 +18040,7 @@ static struct hda_verb alc663_g71v_init_verbs[] = { {} }; -static struct hda_verb alc663_g50v_init_verbs[] = { +static const struct hda_verb alc663_g50v_init_verbs[] = { {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ @@ -18050,7 +18050,7 @@ static struct hda_verb alc663_g50v_init_verbs[] = { {} }; -static struct hda_verb alc662_ecs_init_verbs[] = { +static const struct hda_verb alc662_ecs_init_verbs[] = { {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, @@ -18058,7 +18058,7 @@ static struct hda_verb alc662_ecs_init_verbs[] = { {} }; -static struct hda_verb alc272_dell_zm1_init_verbs[] = { +static const struct hda_verb alc272_dell_zm1_init_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -18073,7 +18073,7 @@ static struct hda_verb alc272_dell_zm1_init_verbs[] = { {} }; -static struct hda_verb alc272_dell_init_verbs[] = { +static const struct hda_verb alc272_dell_init_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -18088,7 +18088,7 @@ static struct hda_verb alc272_dell_init_verbs[] = { {} }; -static struct hda_verb alc663_mode7_init_verbs[] = { +static const struct hda_verb alc663_mode7_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -18107,7 +18107,7 @@ static struct hda_verb alc663_mode7_init_verbs[] = { {} }; -static struct hda_verb alc663_mode8_init_verbs[] = { +static const struct hda_verb alc663_mode8_init_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -18127,13 +18127,13 @@ static struct hda_verb alc663_mode8_init_verbs[] = { {} }; -static struct snd_kcontrol_new alc662_auto_capture_mixer[] = { +static const struct snd_kcontrol_new alc662_auto_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), { } /* end */ }; -static struct snd_kcontrol_new alc272_auto_capture_mixer[] = { +static const struct snd_kcontrol_new alc272_auto_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), { } /* end */ @@ -18342,7 +18342,7 @@ static void alc663_g71v_setup(struct hda_codec *codec) #define alc663_g50v_setup alc663_m51va_setup -static struct snd_kcontrol_new alc662_ecs_mixer[] = { +static const struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -18356,7 +18356,7 @@ static struct snd_kcontrol_new alc662_ecs_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc272_nc10_mixer[] = { +static const struct snd_kcontrol_new alc272_nc10_mixer[] = { /* Master Playback automatically created from Speaker and Headphone */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -18414,7 +18414,7 @@ static const char * const alc662_models[ALC662_MODEL_LAST] = { [ALC662_AUTO] = "auto", }; -static struct snd_pci_quirk alc662_cfg_tbl[] = { +static const struct snd_pci_quirk alc662_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x1028, 0x02d6, "DELL", ALC272_DELL), SND_PCI_QUIRK(0x1028, 0x02f4, "DELL ZM1", ALC272_DELL_ZM1), @@ -18496,7 +18496,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { {} }; -static struct alc_config_preset alc662_presets[] = { +static const struct alc_config_preset alc662_presets[] = { [ALC662_3ST_2ch_DIG] = { .mixers = { alc662_3ST_2ch_mixer }, .init_verbs = { alc662_init_verbs, alc662_eapd_init_verbs }, @@ -19247,7 +19247,7 @@ static int alc_auto_ch_mode_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new alc_auto_channel_mode_enum = { +static const struct snd_kcontrol_new alc_auto_channel_mode_enum = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", .info = alc_auto_ch_mode_info, @@ -19417,7 +19417,7 @@ static const struct alc_fixup alc662_fixups[] = { }, }; -static struct snd_pci_quirk alc662_fixup_tbl[] = { +static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), @@ -19644,7 +19644,7 @@ static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream alc680_pcm_analog_auto_capture = { +static const struct hda_pcm_stream alc680_pcm_analog_auto_capture = { .substreams = 1, /* can be overridden */ .channels_min = 2, .channels_max = 2, @@ -19655,7 +19655,7 @@ static struct hda_pcm_stream alc680_pcm_analog_auto_capture = { }, }; -static struct snd_kcontrol_new alc680_base_mixer[] = { +static const struct snd_kcontrol_new alc680_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -19667,7 +19667,7 @@ static struct snd_kcontrol_new alc680_base_mixer[] = { { } }; -static struct hda_bind_ctls alc680_bind_cap_vol = { +static const struct hda_bind_ctls alc680_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), @@ -19677,7 +19677,7 @@ static struct hda_bind_ctls alc680_bind_cap_vol = { }, }; -static struct hda_bind_ctls alc680_bind_cap_switch = { +static const struct hda_bind_ctls alc680_bind_cap_switch = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), @@ -19687,7 +19687,7 @@ static struct hda_bind_ctls alc680_bind_cap_switch = { }, }; -static struct snd_kcontrol_new alc680_master_capture_mixer[] = { +static const struct snd_kcontrol_new alc680_master_capture_mixer[] = { HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol), HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch), { } /* end */ @@ -19696,7 +19696,7 @@ static struct snd_kcontrol_new alc680_master_capture_mixer[] = { /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc680_init_verbs[] = { +static const struct hda_verb alc680_init_verbs[] = { {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -19930,12 +19930,12 @@ static const char * const alc680_models[ALC680_MODEL_LAST] = { [ALC680_AUTO] = "auto", }; -static struct snd_pci_quirk alc680_cfg_tbl[] = { +static const struct snd_pci_quirk alc680_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12f3, "ASUS NX90", ALC680_BASE), {} }; -static struct alc_config_preset alc680_presets[] = { +static const struct alc_config_preset alc680_presets[] = { [ALC680_BASE] = { .mixers = { alc680_base_mixer }, .cap_mixer = alc680_master_capture_mixer, @@ -20016,7 +20016,7 @@ static int patch_alc680(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_realtek[] = { +static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, -- cgit v1.2.3 From dda144103c4a47a504fcaa8cddd08a4440c87060 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:29:30 +0200 Subject: ALSA: hda - Constify some API function arguments Also fixed the assignment of multiout.dac_nids to satisfy const. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 14 ++++---- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_local.h | 6 ++-- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_realtek.c | 18 +++++----- sound/pci/hda/patch_sigmatel.c | 3 +- sound/pci/hda/patch_via.c | 74 +++++++++++++++++++++--------------------- 7 files changed, 60 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d1a2351aabf..c63f376ba9a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2621,7 +2621,7 @@ static unsigned int convert_to_spdif_status(unsigned short val) static void set_dig_out(struct hda_codec *codec, hda_nid_t nid, int verb, int val) { - hda_nid_t *d; + const hda_nid_t *d; snd_hda_codec_write_cache(codec, nid, 0, verb, val); d = codec->slave_dig_outs; @@ -4184,7 +4184,7 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, -1); snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); if (codec->slave_dig_outs) { - hda_nid_t *d; + const hda_nid_t *d; for (d = codec->slave_dig_outs; *d; d++) snd_hda_codec_setup_stream(codec, *d, stream_tag, 0, format); @@ -4199,7 +4199,7 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) { snd_hda_codec_cleanup_stream(codec, nid); if (codec->slave_dig_outs) { - hda_nid_t *d; + const hda_nid_t *d; for (d = codec->slave_dig_outs; *d; d++) snd_hda_codec_cleanup_stream(codec, *d); } @@ -4346,7 +4346,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, unsigned int format, struct snd_pcm_substream *substream) { - hda_nid_t *nids = mout->dac_nids; + const hda_nid_t *nids = mout->dac_nids; int chs = substream->runtime->channels; int i; @@ -4401,7 +4401,7 @@ EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) { - hda_nid_t *nids = mout->dac_nids; + const hda_nid_t *nids = mout->dac_nids; int i; for (i = 0; i < mout->num_dacs; i++) @@ -4426,7 +4426,7 @@ EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_cleanup); * Helper for automatic pin configuration */ -static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) +static int is_in_nid_list(hda_nid_t nid, const hda_nid_t *list) { for (; *list; list++) if (*list == nid) @@ -4507,7 +4507,7 @@ static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) */ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg, - hda_nid_t *ignore_nids) + const hda_nid_t *ignore_nids) { hda_nid_t nid, end_nid; short seq, assoc_line_out, assoc_speaker; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7d57a66c1b3..59c97306c1d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -832,7 +832,7 @@ struct hda_codec { unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ - hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ + const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_array init_pins; /* initial (BIOS) pin configurations */ struct snd_array driver_pins; /* pin configs set by codec parser */ struct snd_array cvt_setups; /* audio convert setups */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index f36af375abb..1ed6ee5a1e7 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -267,11 +267,11 @@ enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */ struct hda_multi_out { int num_dacs; /* # of DACs, must be more than 1 */ - hda_nid_t *dac_nids; /* DAC list */ + const hda_nid_t *dac_nids; /* DAC list */ hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */ hda_nid_t dig_out_nid; /* digital out audio widget */ - hda_nid_t *slave_dig_outs; + const hda_nid_t *slave_dig_outs; int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ @@ -443,7 +443,7 @@ struct auto_pin_cfg { int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg, - hda_nid_t *ignore_nids); + const hda_nid_t *ignore_nids); /* amp values */ #define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8)) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 12317488cd9..6df9943bbf6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2845,7 +2845,7 @@ static int ad1988_auto_fill_dac_nids(struct hda_codec *codec, /* check the pins hardwired to audio widget */ for (i = 0; i < cfg->line_outs; i++) { idx = ad1988_pin_idx(cfg->line_out_pins[i]); - spec->multiout.dac_nids[i] = ad1988_idx_to_dac(codec, idx); + spec->private_dac_nids[i] = ad1988_idx_to_dac(codec, idx); } spec->multiout.num_dacs = cfg->line_outs; return 0; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5b79080be76..ac0400929dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5294,7 +5294,7 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, nid = cfg->line_out_pins[i]; if (alc880_is_fixed_pin(nid)) { int idx = alc880_fixed_pin_idx(nid); - spec->multiout.dac_nids[i] = alc880_idx_to_dac(idx); + spec->private_dac_nids[i] = alc880_idx_to_dac(idx); assigned[idx] = 1; } } @@ -5306,7 +5306,7 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, /* search for an empty channel */ for (j = 0; j < cfg->line_outs; j++) { if (!assigned[j]) { - spec->multiout.dac_nids[i] = + spec->private_dac_nids[i] = alc880_idx_to_dac(j); assigned[j] = 1; break; @@ -7131,7 +7131,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, spec->multiout.num_dacs = 1; spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.dac_nids[0] = 0x02; + spec->private_dac_nids[0] = 0x02; nid = cfg->line_out_pins[0]; if (nid) { @@ -12214,7 +12214,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, spec->multiout.num_dacs = 1; /* only use one dac */ spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.dac_nids[0] = 2; + spec->private_dac_nids[0] = 2; pfx = alc_get_line_out_pfx(spec, true); if (!pfx) @@ -13553,7 +13553,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, HDA_OUTPUT)); if (err < 0) return err; - spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } if (nid != 0x16) @@ -15951,7 +15951,7 @@ static int alc861_auto_fill_dac_nids(struct hda_codec *codec, dac = alc861_look_for_dac(codec, nid); if (!dac) continue; - spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } return 0; } @@ -18932,7 +18932,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec, dac = alc_auto_look_for_dac(codec, cfg->line_out_pins[i]); if (!dac) continue; - spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } return 0; } @@ -19166,7 +19166,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, spec->multi_io[num_pins].pin = nid; spec->multi_io[num_pins].dac = dac; num_pins++; - spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } } spec->multiout.num_dacs = 1; @@ -19786,7 +19786,7 @@ static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid, if (err < 0) return err; - spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + spec->private_dac_nids[spec->multiout.num_dacs++] = dac; } return 0; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2b590d92ff9..1ce65d4e05a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3072,7 +3072,8 @@ static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); return 1; } else { - spec->multiout.dac_nids[spec->multiout.num_dacs] = nid; + snd_BUG_ON(spec->multiout.dac_nids != spec->dac_nids); + spec->dac_nids[spec->multiout.num_dacs] = nid; spec->multiout.num_dacs++; } return 0; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 6a51ffb266f..0bfbacb4e34 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1233,7 +1233,7 @@ static void playback_multi_pcm_prep_0(struct hda_codec *codec, { struct via_spec *spec = codec->spec; struct hda_multi_out *mout = &spec->multiout; - hda_nid_t *nids = mout->dac_nids; + const hda_nid_t *nids = mout->dac_nids; int chs = substream->runtime->channels; int i; @@ -1302,7 +1302,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo, { struct via_spec *spec = codec->spec; struct hda_multi_out *mout = &spec->multiout; - hda_nid_t *nids = mout->dac_nids; + const hda_nid_t *nids = mout->dac_nids; if (substream->number == 0) playback_multi_pcm_prep_0(codec, stream_tag, format, @@ -1323,7 +1323,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct via_spec *spec = codec->spec; struct hda_multi_out *mout = &spec->multiout; - hda_nid_t *nids = mout->dac_nids; + const hda_nid_t *nids = mout->dac_nids; int i; if (substream->number == 0) { @@ -1849,16 +1849,16 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec, /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: - spec->multiout.dac_nids[i] = 0x10; + spec->private_dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: - spec->multiout.dac_nids[i] = 0x12; + spec->private_dac_nids[i] = 0x12; break; case AUTO_SEQ_SURROUND: - spec->multiout.dac_nids[i] = 0x11; + spec->private_dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: - spec->multiout.dac_nids[i] = 0x13; + spec->private_dac_nids[i] = 0x13; break; } } @@ -2437,26 +2437,26 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec, switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ - spec->multiout.dac_nids[i] = 0x10; + spec->private_dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: /* AOW2 */ - spec->multiout.dac_nids[i] = 0x12; + spec->private_dac_nids[i] = 0x12; break; case AUTO_SEQ_SURROUND: /* AOW3 */ - spec->multiout.dac_nids[i] = 0x11; + spec->private_dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: /* AOW1 */ - spec->multiout.dac_nids[i] = 0x27; + spec->private_dac_nids[i] = 0x27; break; default: break; } } } - spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ + spec->private_dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ for (i = 0; i < cfg->line_outs; i++) { @@ -2466,15 +2466,15 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec, switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ - spec->multiout.dac_nids[i] = 0x10; + spec->private_dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: /* AOW2 */ - spec->multiout.dac_nids[i] = 0x12; + spec->private_dac_nids[i] = 0x12; break; case AUTO_SEQ_SURROUND: /* AOW1 */ - spec->multiout.dac_nids[i] = 0x11; + spec->private_dac_nids[i] = 0x11; break; default: break; @@ -3013,16 +3013,16 @@ static int vt1708B_auto_fill_dac_nids(struct via_spec *spec, /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: - spec->multiout.dac_nids[i] = 0x10; + spec->private_dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: - spec->multiout.dac_nids[i] = 0x24; + spec->private_dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: - spec->multiout.dac_nids[i] = 0x11; + spec->private_dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: - spec->multiout.dac_nids[i] = 0x25; + spec->private_dac_nids[i] = 0x25; break; } } @@ -3546,19 +3546,19 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: - spec->multiout.dac_nids[i] = 0x10; + spec->private_dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: if (spec->codec->vendor_id == 0x11064397) - spec->multiout.dac_nids[i] = 0x25; + spec->private_dac_nids[i] = 0x25; else - spec->multiout.dac_nids[i] = 0x24; + spec->private_dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: - spec->multiout.dac_nids[i] = 0x11; + spec->private_dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: - spec->multiout.dac_nids[i] = 0x25; + spec->private_dac_nids[i] = 0x25; break; } } @@ -3567,11 +3567,11 @@ static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, /* for Smart 5.1, line/mic inputs double as output pins */ if (cfg->line_outs == 1) { spec->multiout.num_dacs = 3; - spec->multiout.dac_nids[AUTO_SEQ_SURROUND] = 0x11; + spec->private_dac_nids[AUTO_SEQ_SURROUND] = 0x11; if (spec->codec->vendor_id == 0x11064397) - spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x25; + spec->private_dac_nids[AUTO_SEQ_CENLFE] = 0x25; else - spec->multiout.dac_nids[AUTO_SEQ_CENLFE] = 0x24; + spec->private_dac_nids[AUTO_SEQ_CENLFE] = 0x24; } return 0; @@ -4017,7 +4017,7 @@ static int vt1702_auto_fill_dac_nids(struct via_spec *spec, if (cfg->line_out_pins[0]) { /* config dac list */ - spec->multiout.dac_nids[0] = 0x10; + spec->private_dac_nids[0] = 0x10; } return 0; @@ -4384,16 +4384,16 @@ static int vt1718S_auto_fill_dac_nids(struct via_spec *spec, /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: - spec->multiout.dac_nids[i] = 0x8; + spec->private_dac_nids[i] = 0x8; break; case AUTO_SEQ_CENLFE: - spec->multiout.dac_nids[i] = 0xa; + spec->private_dac_nids[i] = 0xa; break; case AUTO_SEQ_SURROUND: - spec->multiout.dac_nids[i] = 0x9; + spec->private_dac_nids[i] = 0x9; break; case AUTO_SEQ_SIDE: - spec->multiout.dac_nids[i] = 0xb; + spec->private_dac_nids[i] = 0xb; break; } } @@ -4905,13 +4905,13 @@ static int vt1716S_auto_fill_dac_nids(struct via_spec *spec, /* config dac list */ switch (i) { case AUTO_SEQ_FRONT: - spec->multiout.dac_nids[i] = 0x10; + spec->private_dac_nids[i] = 0x10; break; case AUTO_SEQ_CENLFE: - spec->multiout.dac_nids[i] = 0x25; + spec->private_dac_nids[i] = 0x25; break; case AUTO_SEQ_SURROUND: - spec->multiout.dac_nids[i] = 0x11; + spec->private_dac_nids[i] = 0x11; break; } } @@ -5454,7 +5454,7 @@ static int vt2002P_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.num_dacs = 1; spec->multiout.dac_nids = spec->private_dac_nids; if (cfg->line_out_pins[0]) - spec->multiout.dac_nids[0] = 0x8; + spec->private_dac_nids[0] = 0x8; return 0; } @@ -5907,7 +5907,7 @@ static int vt1812_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.num_dacs = 1; spec->multiout.dac_nids = spec->private_dac_nids; if (cfg->line_out_pins[0]) - spec->multiout.dac_nids[0] = 0x8; + spec->private_dac_nids[0] = 0x8; return 0; } -- cgit v1.2.3 From 4c6d72d1380f2f4056635592c07bc50f5d08296c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:30:18 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_realtek.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 140 +++++++++++++++++++++--------------------- 1 file changed, 70 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac0400929dc..0f2b52aeb40 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -345,8 +345,8 @@ struct alc_spec { /* capture */ unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t *capsrc_nids; + const hda_nid_t *adc_nids; + const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ /* capture setup for dynamic dual-adc switch */ @@ -441,13 +441,13 @@ struct alc_config_preset { const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ const struct hda_verb *init_verbs[5]; unsigned int num_dacs; - hda_nid_t *dac_nids; + const hda_nid_t *dac_nids; hda_nid_t dig_out_nid; /* optional */ hda_nid_t hp_nid; /* optional */ - hda_nid_t *slave_dig_outs; + const hda_nid_t *slave_dig_outs; unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t *capsrc_nids; + const hda_nid_t *adc_nids; + const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; @@ -2547,12 +2547,12 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) * F-Mic = 0x1b, HP = 0x19 */ -static hda_nid_t alc880_dac_nids[4] = { +static const hda_nid_t alc880_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x05, 0x04, 0x03 }; -static hda_nid_t alc880_adc_nids[3] = { +static const hda_nid_t alc880_adc_nids[3] = { /* ADC0-2 */ 0x07, 0x08, 0x09, }; @@ -2561,7 +2561,7 @@ static hda_nid_t alc880_adc_nids[3] = { * but it shows zero connection in the real implementation on some devices. * Note: this is a 915GAV bug, fixed on 915GLV */ -static hda_nid_t alc880_adc_nids_alt[2] = { +static const hda_nid_t alc880_adc_nids_alt[2] = { /* ADC1-2 */ 0x08, 0x09, }; @@ -2820,7 +2820,7 @@ static const struct hda_channel_mode alc880_fivestack_modes[2] = { * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b */ -static hda_nid_t alc880_6st_dac_nids[4] = { +static const hda_nid_t alc880_6st_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x03, 0x04, 0x05 }; @@ -2895,7 +2895,7 @@ static const struct snd_kcontrol_new alc880_six_stack_mixer[] = { * haven't setup any initialization verbs for these yet... */ -static hda_nid_t alc880_w810_dac_nids[3] = { +static const hda_nid_t alc880_w810_dac_nids[3] = { /* front, rear/surround, clfe */ 0x02, 0x03, 0x04 }; @@ -2928,7 +2928,7 @@ static const struct snd_kcontrol_new alc880_w810_base_mixer[] = { * Line = 0x1a */ -static hda_nid_t alc880_z71v_dac_nids[1] = { +static const hda_nid_t alc880_z71v_dac_nids[1] = { 0x02 }; #define ALC880_Z71V_HP_DAC 0x03 @@ -2958,7 +2958,7 @@ static const struct snd_kcontrol_new alc880_z71v_mixer[] = { * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18 */ -static hda_nid_t alc880_f1734_dac_nids[1] = { +static const hda_nid_t alc880_f1734_dac_nids[1] = { 0x03 }; #define ALC880_F1734_HP_DAC 0x02 @@ -3241,7 +3241,7 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - hda_nid_t *nids = spec->capsrc_nids; + const hda_nid_t *nids = spec->capsrc_nids; if (!nids) nids = spec->adc_nids; err = snd_hda_add_nid(codec, kctl, i, nids[i]); @@ -3809,7 +3809,7 @@ static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { */ /* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */ -static hda_nid_t alc880_lg_dac_nids[3] = { +static const hda_nid_t alc880_lg_dac_nids[3] = { 0x05, 0x02, 0x03 }; @@ -4542,7 +4542,7 @@ static int alc_codec_rename(struct hda_codec *codec, const char *name) * enum controls. */ #ifdef CONFIG_SND_DEBUG -static hda_nid_t alc880_test_dac_nids[4] = { +static const hda_nid_t alc880_test_dac_nids[4] = { 0x02, 0x03, 0x04, 0x05 }; @@ -4569,7 +4569,7 @@ static const struct hda_channel_mode alc880_test_modes[4] = { static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { + static const char * const texts[] = { "N/A", "Line Out", "HP Out", "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%" }; @@ -4614,7 +4614,7 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - static unsigned int ctls[] = { + static const unsigned int ctls[] = { 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN, AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ, AC_PINCTL_IN_EN | AC_PINCTL_VREF_50, @@ -4644,7 +4644,7 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { + static const char * const texts[] = { "Front", "Surround", "CLFE", "Side" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -5632,7 +5632,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; + static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc880_ignore); @@ -5827,7 +5827,7 @@ static void set_capture_mixer(struct hda_codec *codec) } /* fill adc_nids (and capsrc_nids) containing all active input pins */ -static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, +static void fillup_priv_adc_nids(struct hda_codec *codec, const hda_nid_t *nids, int num_nids) { struct alc_spec *spec = codec->spec; @@ -6004,17 +6004,17 @@ static int patch_alc880(struct hda_codec *codec) * ALC260 support */ -static hda_nid_t alc260_dac_nids[1] = { +static const hda_nid_t alc260_dac_nids[1] = { /* front */ 0x02, }; -static hda_nid_t alc260_adc_nids[1] = { +static const hda_nid_t alc260_adc_nids[1] = { /* ADC0 */ 0x04, }; -static hda_nid_t alc260_adc_nids_alt[1] = { +static const hda_nid_t alc260_adc_nids_alt[1] = { /* ADC1 */ 0x05, }; @@ -6022,7 +6022,7 @@ static hda_nid_t alc260_adc_nids_alt[1] = { /* NIDs used when simultaneous access to both ADCs makes sense. Note that * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. */ -static hda_nid_t alc260_dual_adc_nids[2] = { +static const hda_nid_t alc260_dual_adc_nids[2] = { /* ADC0, ADC1 */ 0x04, 0x05 }; @@ -6892,10 +6892,10 @@ static const struct hda_verb alc260_hp_dc7600_verbs[] = { * configuration. */ #ifdef CONFIG_SND_DEBUG -static hda_nid_t alc260_test_dac_nids[1] = { +static const hda_nid_t alc260_test_dac_nids[1] = { 0x02, }; -static hda_nid_t alc260_test_adc_nids[2] = { +static const hda_nid_t alc260_test_adc_nids[2] = { 0x04, 0x05, }; /* For testing the ALC260, each input MUX needs its own definition since @@ -7274,7 +7274,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc260_ignore[] = { 0x17, 0 }; + static const hda_nid_t alc260_ignore[] = { 0x17, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc260_ignore); @@ -7644,7 +7644,7 @@ static const struct hda_channel_mode alc882_ch_modes[1] = { }; /* DACs */ -static hda_nid_t alc882_dac_nids[4] = { +static const hda_nid_t alc882_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x03, 0x04, 0x05 }; @@ -7654,14 +7654,14 @@ static hda_nid_t alc882_dac_nids[4] = { #define alc882_adc_nids alc880_adc_nids #define alc882_adc_nids_alt alc880_adc_nids_alt #define alc883_adc_nids alc882_adc_nids_alt -static hda_nid_t alc883_adc_nids_alt[1] = { 0x08 }; -static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +static const hda_nid_t alc883_adc_nids_alt[1] = { 0x08 }; +static const hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; #define alc889_adc_nids alc880_adc_nids -static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 }; -static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; +static const hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 }; +static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; #define alc883_capsrc_nids alc882_capsrc_nids_alt -static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +static const hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; #define alc889_capsrc_nids alc882_capsrc_nids /* input MUX */ @@ -9972,11 +9972,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) #define alc882_pcm_digital_playback alc880_pcm_digital_playback #define alc882_pcm_digital_capture alc880_pcm_digital_capture -static hda_nid_t alc883_slave_dig_outs[] = { +static const hda_nid_t alc883_slave_dig_outs[] = { ALC1200_DIGOUT_NID, 0, }; -static hda_nid_t alc1200_slave_dig_outs[] = { +static const hda_nid_t alc1200_slave_dig_outs[] = { ALC883_DIGOUT_NID, 0, }; @@ -11158,7 +11158,7 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) static int alc882_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - static hda_nid_t alc882_ignore[] = { 0x1d, 0 }; + static const hda_nid_t alc882_ignore[] = { 0x1d, 0 }; int err; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -11375,12 +11375,12 @@ static int patch_alc882(struct hda_codec *codec) #define alc262_modes alc260_modes #define alc262_capture_source alc882_capture_source -static hda_nid_t alc262_dmic_adc_nids[1] = { +static const hda_nid_t alc262_dmic_adc_nids[1] = { /* ADC0 */ 0x09 }; -static hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 }; +static const hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 }; static const struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -12590,7 +12590,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc262_ignore[] = { 0x1d, 0 }; + static const hda_nid_t alc262_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc262_ignore); @@ -13100,22 +13100,22 @@ static int patch_alc262(struct hda_codec *codec) #define ALC268_DIGOUT_NID ALC880_DIGOUT_NID #define alc268_modes alc260_modes -static hda_nid_t alc268_dac_nids[2] = { +static const hda_nid_t alc268_dac_nids[2] = { /* front, hp */ 0x02, 0x03 }; -static hda_nid_t alc268_adc_nids[2] = { +static const hda_nid_t alc268_adc_nids[2] = { /* ADC0-1 */ 0x08, 0x07 }; -static hda_nid_t alc268_adc_nids_alt[1] = { +static const hda_nid_t alc268_adc_nids_alt[1] = { /* ADC0 */ 0x08 }; -static hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 }; +static const hda_nid_t alc268_capsrc_nids[2] = { 0x23, 0x24 }; static const struct snd_kcontrol_new alc268_base_mixer[] = { /* output mixer control */ @@ -13726,7 +13726,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc268_ignore[] = { 0 }; + static const hda_nid_t alc268_ignore[] = { 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc268_ignore); @@ -14113,25 +14113,25 @@ static int patch_alc268(struct hda_codec *codec) #define alc269_dac_nids alc260_dac_nids -static hda_nid_t alc269_adc_nids[1] = { +static const hda_nid_t alc269_adc_nids[1] = { /* ADC1 */ 0x08, }; -static hda_nid_t alc269_capsrc_nids[1] = { +static const hda_nid_t alc269_capsrc_nids[1] = { 0x23, }; -static hda_nid_t alc269vb_adc_nids[1] = { +static const hda_nid_t alc269vb_adc_nids[1] = { /* ADC1 */ 0x09, }; -static hda_nid_t alc269vb_capsrc_nids[1] = { +static const hda_nid_t alc269vb_capsrc_nids[1] = { 0x22, }; -static hda_nid_t alc269_adc_candidates[] = { +static const hda_nid_t alc269_adc_candidates[] = { 0x08, 0x09, 0x07, 0x11, }; @@ -14696,7 +14696,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; + static const hda_nid_t alc269_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc269_ignore); @@ -15886,17 +15886,17 @@ static const struct hda_channel_mode alc861_8ch_modes[1] = { { 8, NULL } }; -static hda_nid_t alc861_dac_nids[4] = { +static const hda_nid_t alc861_dac_nids[4] = { /* front, surround, clfe, side */ 0x03, 0x06, 0x05, 0x04 }; -static hda_nid_t alc660_dac_nids[3] = { +static const hda_nid_t alc660_dac_nids[3] = { /* front, clfe, surround */ 0x03, 0x05, 0x06 }; -static hda_nid_t alc861_adc_nids[1] = { +static const hda_nid_t alc861_adc_nids[1] = { /* ADC0-2 */ 0x08, }; @@ -16115,7 +16115,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc861_ignore[] = { 0x1d, 0 }; + static const hda_nid_t alc861_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc861_ignore); @@ -16439,7 +16439,7 @@ static int patch_alc861(struct hda_codec *codec) */ #define ALC861VD_DIGOUT_NID 0x06 -static hda_nid_t alc861vd_dac_nids[4] = { +static const hda_nid_t alc861vd_dac_nids[4] = { /* front, surr, clfe, side surr */ 0x02, 0x03, 0x04, 0x05 }; @@ -16451,17 +16451,17 @@ static hda_nid_t alc861vd_dac_nids[4] = { * - and it is the same as in 861vd. * adc_nids in ALC660vd are (is) the same as in 861vd */ -static hda_nid_t alc660vd_dac_nids[3] = { +static const hda_nid_t alc660vd_dac_nids[3] = { /* front, rear, clfe, rear_surr */ 0x02, 0x04, 0x03 }; -static hda_nid_t alc861vd_adc_nids[1] = { +static const hda_nid_t alc861vd_adc_nids[1] = { /* ADC0 */ 0x09, }; -static hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 }; +static const hda_nid_t alc861vd_capsrc_nids[1] = { 0x22 }; /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -17238,7 +17238,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc861vd_ignore[] = { 0x1d, 0 }; + static const hda_nid_t alc861vd_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc861vd_ignore); @@ -17427,27 +17427,27 @@ static int patch_alc861vd(struct hda_codec *codec) #define ALC662_DIGOUT_NID 0x06 #define ALC662_DIGIN_NID 0x0a -static hda_nid_t alc662_dac_nids[3] = { +static const hda_nid_t alc662_dac_nids[3] = { /* front, rear, clfe */ 0x02, 0x03, 0x04 }; -static hda_nid_t alc272_dac_nids[2] = { +static const hda_nid_t alc272_dac_nids[2] = { 0x02, 0x03 }; -static hda_nid_t alc662_adc_nids[2] = { +static const hda_nid_t alc662_adc_nids[2] = { /* ADC1-2 */ 0x09, 0x08 }; -static hda_nid_t alc272_adc_nids[1] = { +static const hda_nid_t alc272_adc_nids[1] = { /* ADC1-2 */ 0x08, }; -static hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 }; -static hda_nid_t alc272_capsrc_nids[1] = { 0x23 }; +static const hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 }; +static const hda_nid_t alc272_capsrc_nids[1] = { 0x23 }; /* input MUX */ @@ -19292,7 +19292,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc662_ignore[] = { 0x1d, 0 }; + static const hda_nid_t alc662_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc662_ignore); @@ -19571,12 +19571,12 @@ static int patch_alc888(struct hda_codec *codec) #define ALC680_DIGOUT_NID ALC880_DIGOUT_NID #define alc680_modes alc260_modes -static hda_nid_t alc680_dac_nids[3] = { +static const hda_nid_t alc680_dac_nids[3] = { /* Lout1, Lout2, hp */ 0x02, 0x03, 0x04 }; -static hda_nid_t alc680_adc_nids[3] = { +static const hda_nid_t alc680_adc_nids[3] = { /* ADC0-2 */ /* DMIC, MIC, Line-in*/ 0x07, 0x08, 0x09 @@ -19872,7 +19872,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err; - static hda_nid_t alc680_ignore[] = { 0 }; + static const hda_nid_t alc680_ignore[] = { 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc680_ignore); -- cgit v1.2.3 From 498f5b175b90597608e48390183933d3875d5429 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:33:15 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_analog.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 328 +++++++++++++++++++++---------------------- 1 file changed, 164 insertions(+), 164 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6df9943bbf6..981c631624d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -30,7 +30,7 @@ #include "hda_beep.h" struct ad198x_spec { - struct snd_kcontrol_new *mixers[6]; + const struct snd_kcontrol_new *mixers[6]; int num_mixers; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[6]; /* initialization verbs @@ -46,17 +46,17 @@ struct ad198x_spec { unsigned int cur_eapd; unsigned int need_dac_fix; - hda_nid_t *alt_dac_nid; - struct hda_pcm_stream *stream_analog_alt_playback; + const hda_nid_t *alt_dac_nid; + const struct hda_pcm_stream *stream_analog_alt_playback; /* capture */ unsigned int num_adc_nids; - hda_nid_t *adc_nids; + const hda_nid_t *adc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ /* capture source */ const struct hda_input_mux *input_mux; - hda_nid_t *capsrc_nids; + const hda_nid_t *capsrc_nids; unsigned int cur_mux[3]; /* channel model */ @@ -182,13 +182,13 @@ static void ad198x_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ -static struct snd_kcontrol_new ad_beep_mixer[] = { +static const struct snd_kcontrol_new ad_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_OUTPUT), { } /* end */ }; -static struct snd_kcontrol_new ad_beep2_mixer[] = { +static const struct snd_kcontrol_new ad_beep2_mixer[] = { HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT), HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT), { } /* end */ @@ -231,7 +231,7 @@ static int ad198x_build_controls(struct hda_codec *codec) /* create beep controls if needed */ #ifdef CONFIG_SND_HDA_INPUT_BEEP if (spec->beep_amp) { - struct snd_kcontrol_new *knew; + const struct snd_kcontrol_new *knew; knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer; for ( ; knew->name; knew++) { struct snd_kcontrol *kctl; @@ -331,7 +331,7 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } -static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { +static const struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -403,7 +403,7 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static struct hda_pcm_stream ad198x_pcm_analog_playback = { +static const struct hda_pcm_stream ad198x_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 6, /* changed later */ @@ -415,7 +415,7 @@ static struct hda_pcm_stream ad198x_pcm_analog_playback = { }, }; -static struct hda_pcm_stream ad198x_pcm_analog_capture = { +static const struct hda_pcm_stream ad198x_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -426,7 +426,7 @@ static struct hda_pcm_stream ad198x_pcm_analog_capture = { }, }; -static struct hda_pcm_stream ad198x_pcm_digital_playback = { +static const struct hda_pcm_stream ad198x_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -439,7 +439,7 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = { }, }; -static struct hda_pcm_stream ad198x_pcm_digital_capture = { +static const struct hda_pcm_stream ad198x_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -569,7 +569,7 @@ static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) } #endif -static struct hda_codec_ops ad198x_patch_ops = { +static const struct hda_codec_ops ad198x_patch_ops = { .build_controls = ad198x_build_controls, .build_pcms = ad198x_build_pcms, .init = ad198x_init, @@ -639,13 +639,13 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, #define AD1986A_CLFE_DAC 0x05 #define AD1986A_ADC 0x06 -static hda_nid_t ad1986a_dac_nids[3] = { +static const hda_nid_t ad1986a_dac_nids[3] = { AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC }; -static hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC }; -static hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 }; +static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC }; +static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 }; -static struct hda_input_mux ad1986a_capture_source = { +static const struct hda_input_mux ad1986a_capture_source = { .num_items = 7, .items = { { "Mic", 0x0 }, @@ -659,7 +659,7 @@ static struct hda_input_mux ad1986a_capture_source = { }; -static struct hda_bind_ctls ad1986a_bind_pcm_vol = { +static const struct hda_bind_ctls ad1986a_bind_pcm_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), @@ -669,7 +669,7 @@ static struct hda_bind_ctls ad1986a_bind_pcm_vol = { }, }; -static struct hda_bind_ctls ad1986a_bind_pcm_sw = { +static const struct hda_bind_ctls ad1986a_bind_pcm_sw = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), @@ -682,7 +682,7 @@ static struct hda_bind_ctls ad1986a_bind_pcm_sw = { /* * mixers */ -static struct snd_kcontrol_new ad1986a_mixers[] = { +static const struct snd_kcontrol_new ad1986a_mixers[] = { /* * bind volumes/mutes of 3 DACs as a single PCM control for simplicity */ @@ -723,7 +723,7 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { }; /* additional mixers for 3stack mode */ -static struct snd_kcontrol_new ad1986a_3st_mixers[] = { +static const struct snd_kcontrol_new ad1986a_3st_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -735,10 +735,10 @@ static struct snd_kcontrol_new ad1986a_3st_mixers[] = { }; /* laptop model - 2ch only */ -static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC }; +static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC }; /* master controls both pins 0x1a and 0x1b */ -static struct hda_bind_ctls ad1986a_laptop_master_vol = { +static const struct hda_bind_ctls ad1986a_laptop_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), @@ -747,7 +747,7 @@ static struct hda_bind_ctls ad1986a_laptop_master_vol = { }, }; -static struct hda_bind_ctls ad1986a_laptop_master_sw = { +static const struct hda_bind_ctls ad1986a_laptop_master_sw = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), @@ -756,7 +756,7 @@ static struct hda_bind_ctls ad1986a_laptop_master_sw = { }, }; -static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { +static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), @@ -787,7 +787,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { /* laptop-eapd model - 2ch only */ -static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { +static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -796,7 +796,7 @@ static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { }, }; -static struct hda_input_mux ad1986a_automic_capture_source = { +static const struct hda_input_mux ad1986a_automic_capture_source = { .num_items = 2, .items = { { "Mic", 0x0 }, @@ -804,13 +804,13 @@ static struct hda_input_mux ad1986a_automic_capture_source = { }, }; -static struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = { +static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), { } /* end */ }; -static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { +static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), @@ -837,7 +837,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { +static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT), { } /* end */ @@ -931,7 +931,7 @@ static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { +static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -949,7 +949,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { /* * initialization verbs */ -static struct hda_verb ad1986a_init_verbs[] = { +static const struct hda_verb ad1986a_init_verbs[] = { /* Front, Surround, CLFE DAC; mute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, @@ -1004,7 +1004,7 @@ static struct hda_verb ad1986a_init_verbs[] = { { } /* end */ }; -static struct hda_verb ad1986a_ch2_init[] = { +static const struct hda_verb ad1986a_ch2_init[] = { /* Surround out -> Line In */ { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* Line-in selectors */ @@ -1016,7 +1016,7 @@ static struct hda_verb ad1986a_ch2_init[] = { { } /* end */ }; -static struct hda_verb ad1986a_ch4_init[] = { +static const struct hda_verb ad1986a_ch4_init[] = { /* Surround out -> Surround */ { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, @@ -1026,7 +1026,7 @@ static struct hda_verb ad1986a_ch4_init[] = { { } /* end */ }; -static struct hda_verb ad1986a_ch6_init[] = { +static const struct hda_verb ad1986a_ch6_init[] = { /* Surround out -> Surround out */ { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, @@ -1036,19 +1036,19 @@ static struct hda_verb ad1986a_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode ad1986a_modes[3] = { +static const struct hda_channel_mode ad1986a_modes[3] = { { 2, ad1986a_ch2_init }, { 4, ad1986a_ch4_init }, { 6, ad1986a_ch6_init }, }; /* eapd initialization */ -static struct hda_verb ad1986a_eapd_init_verbs[] = { +static const struct hda_verb ad1986a_eapd_init_verbs[] = { {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, {} }; -static struct hda_verb ad1986a_automic_verbs[] = { +static const struct hda_verb ad1986a_automic_verbs[] = { {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/ @@ -1058,7 +1058,7 @@ static struct hda_verb ad1986a_automic_verbs[] = { }; /* Ultra initialization */ -static struct hda_verb ad1986a_ultra_init[] = { +static const struct hda_verb ad1986a_ultra_init[] = { /* eapd initialization */ { 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* CLFE -> Mic in */ @@ -1069,7 +1069,7 @@ static struct hda_verb ad1986a_ultra_init[] = { }; /* pin sensing on HP jack */ -static struct hda_verb ad1986a_hp_init_verbs[] = { +static const struct hda_verb ad1986a_hp_init_verbs[] = { {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT}, {} }; @@ -1120,7 +1120,7 @@ static const char * const ad1986a_models[AD1986A_MODELS] = { [AD1986A_SAMSUNG_P50] = "samsung-p50", }; -static struct snd_pci_quirk ad1986a_cfg_tbl[] = { +static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD), @@ -1152,7 +1152,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { }; #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list ad1986a_loopbacks[] = { +static const struct hda_amp_list ad1986a_loopbacks[] = { { 0x13, HDA_OUTPUT, 0 }, /* Mic */ { 0x14, HDA_OUTPUT, 0 }, /* Phone */ { 0x15, HDA_OUTPUT, 0 }, /* CD */ @@ -1329,11 +1329,11 @@ static int patch_ad1986a(struct hda_codec *codec) #define AD1983_DAC 0x03 #define AD1983_ADC 0x04 -static hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC }; -static hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC }; -static hda_nid_t ad1983_capsrc_nids[1] = { 0x15 }; +static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC }; +static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC }; +static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 }; -static struct hda_input_mux ad1983_capture_source = { +static const struct hda_input_mux ad1983_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -1348,7 +1348,7 @@ static struct hda_input_mux ad1983_capture_source = { */ static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "PCM", "ADC" }; + static const char * const texts[] = { "PCM", "ADC" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -1385,7 +1385,7 @@ static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ return 0; } -static struct snd_kcontrol_new ad1983_mixers[] = { +static const struct snd_kcontrol_new ad1983_mixers[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), @@ -1418,7 +1418,7 @@ static struct snd_kcontrol_new ad1983_mixers[] = { { } /* end */ }; -static struct hda_verb ad1983_init_verbs[] = { +static const struct hda_verb ad1983_init_verbs[] = { /* Front, HP, Mono; mute as default */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, @@ -1458,7 +1458,7 @@ static struct hda_verb ad1983_init_verbs[] = { }; #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list ad1983_loopbacks[] = { +static const struct hda_amp_list ad1983_loopbacks[] = { { 0x12, HDA_OUTPUT, 0 }, /* Mic */ { 0x13, HDA_OUTPUT, 0 }, /* Line */ { } /* end */ @@ -1518,12 +1518,12 @@ static int patch_ad1983(struct hda_codec *codec) #define AD1981_DAC 0x03 #define AD1981_ADC 0x04 -static hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC }; -static hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC }; -static hda_nid_t ad1981_capsrc_nids[1] = { 0x15 }; +static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC }; +static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC }; +static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 }; /* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */ -static struct hda_input_mux ad1981_capture_source = { +static const struct hda_input_mux ad1981_capture_source = { .num_items = 7, .items = { { "Front Mic", 0x0 }, @@ -1536,7 +1536,7 @@ static struct hda_input_mux ad1981_capture_source = { }, }; -static struct snd_kcontrol_new ad1981_mixers[] = { +static const struct snd_kcontrol_new ad1981_mixers[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), @@ -1577,7 +1577,7 @@ static struct snd_kcontrol_new ad1981_mixers[] = { { } /* end */ }; -static struct hda_verb ad1981_init_verbs[] = { +static const struct hda_verb ad1981_init_verbs[] = { /* Front, HP, Mono; mute as default */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, @@ -1625,7 +1625,7 @@ static struct hda_verb ad1981_init_verbs[] = { }; #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list ad1981_loopbacks[] = { +static const struct hda_amp_list ad1981_loopbacks[] = { { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */ { 0x13, HDA_OUTPUT, 0 }, /* Line */ { 0x1b, HDA_OUTPUT, 0 }, /* Aux */ @@ -1645,7 +1645,7 @@ static struct hda_amp_list ad1981_loopbacks[] = { #define AD1981_HP_EVENT 0x37 #define AD1981_MIC_EVENT 0x38 -static struct hda_verb ad1981_hp_init_verbs[] = { +static const struct hda_verb ad1981_hp_init_verbs[] = { {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */ /* pin sensing on HP and Mic jacks */ {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, @@ -1674,7 +1674,7 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, } /* bind volumes of both NID 0x05 and 0x06 */ -static struct hda_bind_ctls ad1981_hp_bind_master_vol = { +static const struct hda_bind_ctls ad1981_hp_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), @@ -1696,12 +1696,12 @@ static void ad1981_hp_automute(struct hda_codec *codec) /* toggle input of built-in and mic jack appropriately */ static void ad1981_hp_automic(struct hda_codec *codec) { - static struct hda_verb mic_jack_on[] = { + static const struct hda_verb mic_jack_on[] = { {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {} }; - static struct hda_verb mic_jack_off[] = { + static const struct hda_verb mic_jack_off[] = { {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {} @@ -1730,7 +1730,7 @@ static void ad1981_hp_unsol_event(struct hda_codec *codec, } } -static struct hda_input_mux ad1981_hp_capture_source = { +static const struct hda_input_mux ad1981_hp_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -1739,7 +1739,7 @@ static struct hda_input_mux ad1981_hp_capture_source = { }, }; -static struct snd_kcontrol_new ad1981_hp_mixers[] = { +static const struct snd_kcontrol_new ad1981_hp_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1790,7 +1790,7 @@ static int ad1981_hp_init(struct hda_codec *codec) } /* configuration for Toshiba Laptops */ -static struct hda_verb ad1981_toshiba_init_verbs[] = { +static const struct hda_verb ad1981_toshiba_init_verbs[] = { {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */ /* pin sensing on HP and Mic jacks */ {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, @@ -1798,14 +1798,14 @@ static struct hda_verb ad1981_toshiba_init_verbs[] = { {} }; -static struct snd_kcontrol_new ad1981_toshiba_mixers[] = { +static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = { HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT), { } }; /* configuration for Lenovo Thinkpad T60 */ -static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { +static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), @@ -1835,7 +1835,7 @@ static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { { } /* end */ }; -static struct hda_input_mux ad1981_thinkpad_capture_source = { +static const struct hda_input_mux ad1981_thinkpad_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -1860,7 +1860,7 @@ static const char * const ad1981_models[AD1981_MODELS] = { [AD1981_TOSHIBA] = "toshiba" }; -static struct snd_pci_quirk ad1981_cfg_tbl[] = { +static const struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ @@ -2075,32 +2075,32 @@ enum { * mixers */ -static hda_nid_t ad1988_6stack_dac_nids[4] = { +static const hda_nid_t ad1988_6stack_dac_nids[4] = { 0x04, 0x06, 0x05, 0x0a }; -static hda_nid_t ad1988_3stack_dac_nids[3] = { +static const hda_nid_t ad1988_3stack_dac_nids[3] = { 0x04, 0x05, 0x0a }; /* for AD1988A revision-2, DAC2-4 are swapped */ -static hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { +static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { 0x04, 0x05, 0x0a, 0x06 }; -static hda_nid_t ad1988_alt_dac_nid[1] = { +static const hda_nid_t ad1988_alt_dac_nid[1] = { 0x03 }; -static hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { +static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { 0x04, 0x0a, 0x06 }; -static hda_nid_t ad1988_adc_nids[3] = { +static const hda_nid_t ad1988_adc_nids[3] = { 0x08, 0x09, 0x0f }; -static hda_nid_t ad1988_capsrc_nids[3] = { +static const hda_nid_t ad1988_capsrc_nids[3] = { 0x0c, 0x0d, 0x0e }; @@ -2108,11 +2108,11 @@ static hda_nid_t ad1988_capsrc_nids[3] = { #define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 -static hda_nid_t ad1989b_slave_dig_outs[] = { +static const hda_nid_t ad1989b_slave_dig_outs[] = { AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0 }; -static struct hda_input_mux ad1988_6stack_capture_source = { +static const struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x1 }, /* port-B */ @@ -2123,7 +2123,7 @@ static struct hda_input_mux ad1988_6stack_capture_source = { }, }; -static struct hda_input_mux ad1988_laptop_capture_source = { +static const struct hda_input_mux ad1988_laptop_capture_source = { .num_items = 3, .items = { { "Mic/Line", 0x1 }, /* port-B */ @@ -2166,7 +2166,7 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, } /* 6-stack mode */ -static struct snd_kcontrol_new ad1988_6stack_mixers1[] = { +static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), @@ -2175,7 +2175,7 @@ static struct snd_kcontrol_new ad1988_6stack_mixers1[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = { +static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), @@ -2184,7 +2184,7 @@ static struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { +static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), @@ -2211,14 +2211,14 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { +static const struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; /* 3-stack mode */ -static struct snd_kcontrol_new ad1988_3stack_mixers1[] = { +static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), @@ -2226,7 +2226,7 @@ static struct snd_kcontrol_new ad1988_3stack_mixers1[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { +static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT), @@ -2234,7 +2234,7 @@ static struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { +static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = { HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT), HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT), @@ -2268,7 +2268,7 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { }; /* laptop mode */ -static struct snd_kcontrol_new ad1988_laptop_mixers[] = { +static const struct snd_kcontrol_new ad1988_laptop_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT), HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), @@ -2299,7 +2299,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { }; /* capture */ -static struct snd_kcontrol_new ad1988_capture_mixers[] = { +static const struct snd_kcontrol_new ad1988_capture_mixers[] = { HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -2324,7 +2324,7 @@ static struct snd_kcontrol_new ad1988_capture_mixers[] = { static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { + static const char * const texts[] = { "PCM", "ADC1", "ADC2", "ADC3" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -2405,7 +2405,7 @@ static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { +static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2418,12 +2418,12 @@ static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { +static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT), { } /* end */ }; -static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { +static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT), { } /* end */ @@ -2436,7 +2436,7 @@ static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { /* * for 6-stack (+dig) */ -static struct hda_verb ad1988_6stack_init_verbs[] = { +static const struct hda_verb ad1988_6stack_init_verbs[] = { /* Front, Surround, CLFE, side DAC; unmute as default */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -2496,7 +2496,7 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { { } }; -static struct hda_verb ad1988_6stack_fp_init_verbs[] = { +static const struct hda_verb ad1988_6stack_fp_init_verbs[] = { /* Headphone; unmute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ @@ -2509,7 +2509,7 @@ static struct hda_verb ad1988_6stack_fp_init_verbs[] = { { } }; -static struct hda_verb ad1988_capture_init_verbs[] = { +static const struct hda_verb ad1988_capture_init_verbs[] = { /* mute analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -2527,7 +2527,7 @@ static struct hda_verb ad1988_capture_init_verbs[] = { { } }; -static struct hda_verb ad1988_spdif_init_verbs[] = { +static const struct hda_verb ad1988_spdif_init_verbs[] = { /* SPDIF out sel */ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ @@ -2539,14 +2539,14 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { { } }; -static struct hda_verb ad1988_spdif_in_init_verbs[] = { +static const struct hda_verb ad1988_spdif_in_init_verbs[] = { /* unmute SPDIF input pin */ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } }; /* AD1989 has no ADC -> SPDIF route */ -static struct hda_verb ad1989_spdif_init_verbs[] = { +static const struct hda_verb ad1989_spdif_init_verbs[] = { /* SPDIF-1 out pin */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ @@ -2559,7 +2559,7 @@ static struct hda_verb ad1989_spdif_init_verbs[] = { /* * verbs for 3stack (+dig) */ -static struct hda_verb ad1988_3stack_ch2_init[] = { +static const struct hda_verb ad1988_3stack_ch2_init[] = { /* set port-C to line-in */ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, @@ -2569,7 +2569,7 @@ static struct hda_verb ad1988_3stack_ch2_init[] = { { } /* end */ }; -static struct hda_verb ad1988_3stack_ch6_init[] = { +static const struct hda_verb ad1988_3stack_ch6_init[] = { /* set port-C to surround out */ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, @@ -2579,12 +2579,12 @@ static struct hda_verb ad1988_3stack_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode ad1988_3stack_modes[2] = { +static const struct hda_channel_mode ad1988_3stack_modes[2] = { { 2, ad1988_3stack_ch2_init }, { 6, ad1988_3stack_ch6_init }, }; -static struct hda_verb ad1988_3stack_init_verbs[] = { +static const struct hda_verb ad1988_3stack_init_verbs[] = { /* Front, Surround, CLFE, side DAC; unmute as default */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -2644,13 +2644,13 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { /* * verbs for laptop mode (+dig) */ -static struct hda_verb ad1988_laptop_hp_on[] = { +static const struct hda_verb ad1988_laptop_hp_on[] = { /* unmute port-A and mute port-D */ { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { } /* end */ }; -static struct hda_verb ad1988_laptop_hp_off[] = { +static const struct hda_verb ad1988_laptop_hp_off[] = { /* mute port-A and unmute port-D */ { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, @@ -2659,7 +2659,7 @@ static struct hda_verb ad1988_laptop_hp_off[] = { #define AD1988_HP_EVENT 0x01 -static struct hda_verb ad1988_laptop_init_verbs[] = { +static const struct hda_verb ad1988_laptop_init_verbs[] = { /* Front, Surround, CLFE, side DAC; unmute as default */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -2723,7 +2723,7 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list ad1988_loopbacks[] = { +static const struct hda_amp_list ad1988_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Line */ { 0x20, HDA_INPUT, 4 }, /* Mic */ @@ -2741,7 +2741,7 @@ enum { AD_CTL_WIDGET_MUTE, AD_CTL_BIND_MUTE, }; -static struct snd_kcontrol_new ad1988_control_templates[] = { +static const struct snd_kcontrol_new ad1988_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), HDA_BIND_MUTE(NULL, 0, 0, 0), @@ -2770,18 +2770,18 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, #define AD1988_PIN_CD_NID 0x18 #define AD1988_PIN_BEEP_NID 0x10 -static hda_nid_t ad1988_mixer_nids[8] = { +static const hda_nid_t ad1988_mixer_nids[8] = { /* A B C D E F G H */ 0x22, 0x2b, 0x2c, 0x29, 0x26, 0x2a, 0x27, 0x28 }; static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx) { - static hda_nid_t idx_to_dac[8] = { + static const hda_nid_t idx_to_dac[8] = { /* A B C D E F G H */ 0x04, 0x06, 0x05, 0x04, 0x0a, 0x06, 0x05, 0x0a }; - static hda_nid_t idx_to_dac_rev2[8] = { + static const hda_nid_t idx_to_dac_rev2[8] = { /* A B C D E F G H */ 0x04, 0x05, 0x0a, 0x04, 0x06, 0x05, 0x0a, 0x06 }; @@ -2791,13 +2791,13 @@ static inline hda_nid_t ad1988_idx_to_dac(struct hda_codec *codec, int idx) return idx_to_dac[idx]; } -static hda_nid_t ad1988_boost_nids[8] = { +static const hda_nid_t ad1988_boost_nids[8] = { 0x38, 0x39, 0x3a, 0x3d, 0x3c, 0x3b, 0, 0 }; static int ad1988_pin_idx(hda_nid_t nid) { - static hda_nid_t ad1988_io_pins[8] = { + static const hda_nid_t ad1988_io_pins[8] = { 0x11, 0x14, 0x15, 0x12, 0x17, 0x16, 0x24, 0x25 }; int i; @@ -2809,7 +2809,7 @@ static int ad1988_pin_idx(hda_nid_t nid) static int ad1988_pin_to_loopback_idx(hda_nid_t nid) { - static int loopback_idx[8] = { + static const int loopback_idx[8] = { 2, 0, 1, 3, 4, 5, 1, 4 }; switch (nid) { @@ -2822,7 +2822,7 @@ static int ad1988_pin_to_loopback_idx(hda_nid_t nid) static int ad1988_pin_to_adc_idx(hda_nid_t nid) { - static int adc_idx[8] = { + static const int adc_idx[8] = { 0, 1, 2, 8, 4, 3, 6, 7 }; switch (nid) { @@ -3154,7 +3154,7 @@ static const char * const ad1988_models[AD1988_MODEL_LAST] = { [AD1988_AUTO] = "auto", }; -static struct snd_pci_quirk ad1988_cfg_tbl[] = { +static const struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), @@ -3342,21 +3342,21 @@ static int patch_ad1988(struct hda_codec *codec) * but no build-up framework is given, so far. */ -static hda_nid_t ad1884_dac_nids[1] = { +static const hda_nid_t ad1884_dac_nids[1] = { 0x04, }; -static hda_nid_t ad1884_adc_nids[2] = { +static const hda_nid_t ad1884_adc_nids[2] = { 0x08, 0x09, }; -static hda_nid_t ad1884_capsrc_nids[2] = { +static const hda_nid_t ad1884_capsrc_nids[2] = { 0x0c, 0x0d, }; #define AD1884_SPDIF_OUT 0x02 -static struct hda_input_mux ad1884_capture_source = { +static const struct hda_input_mux ad1884_capture_source = { .num_items = 4, .items = { { "Front Mic", 0x0 }, @@ -3366,7 +3366,7 @@ static struct hda_input_mux ad1884_capture_source = { }, }; -static struct snd_kcontrol_new ad1884_base_mixers[] = { +static const struct snd_kcontrol_new ad1884_base_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), @@ -3410,7 +3410,7 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1984_dmic_mixers[] = { +static const struct snd_kcontrol_new ad1984_dmic_mixers[] = { HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, @@ -3423,7 +3423,7 @@ static struct snd_kcontrol_new ad1984_dmic_mixers[] = { /* * initialization verbs */ -static struct hda_verb ad1884_init_verbs[] = { +static const struct hda_verb ad1884_init_verbs[] = { /* DACs; mute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -3469,7 +3469,7 @@ static struct hda_verb ad1884_init_verbs[] = { }; #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list ad1884_loopbacks[] = { +static const struct hda_amp_list ad1884_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Mic */ { 0x20, HDA_INPUT, 2 }, /* CD */ @@ -3541,7 +3541,7 @@ static int patch_ad1884(struct hda_codec *codec) /* * Lenovo Thinkpad T61/X61 */ -static struct hda_input_mux ad1984_thinkpad_capture_source = { +static const struct hda_input_mux ad1984_thinkpad_capture_source = { .num_items = 4, .items = { { "Mic", 0x0 }, @@ -3555,7 +3555,7 @@ static struct hda_input_mux ad1984_thinkpad_capture_source = { /* * Dell Precision T3400 */ -static struct hda_input_mux ad1984_dell_desktop_capture_source = { +static const struct hda_input_mux ad1984_dell_desktop_capture_source = { .num_items = 3, .items = { { "Front Mic", 0x0 }, @@ -3565,7 +3565,7 @@ static struct hda_input_mux ad1984_dell_desktop_capture_source = { }; -static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { +static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), @@ -3611,7 +3611,7 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { }; /* additional verbs */ -static struct hda_verb ad1984_thinkpad_init_verbs[] = { +static const struct hda_verb ad1984_thinkpad_init_verbs[] = { /* Port-E (docking station mic) pin */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -3629,7 +3629,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { /* * Dell Precision T3400 */ -static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { +static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), @@ -3680,7 +3680,7 @@ static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream ad1984_pcm_dmic_capture = { +static const struct hda_pcm_stream ad1984_pcm_dmic_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -3722,7 +3722,7 @@ static const char * const ad1984_models[AD1984_MODELS] = { [AD1984_DELL_DESKTOP] = "dell_desktop", }; -static struct snd_pci_quirk ad1984_cfg_tbl[] = { +static const struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), @@ -3787,7 +3787,7 @@ static int patch_ad1984(struct hda_codec *codec) * We share the single DAC for both HP and line-outs (see AD1884/1984). */ -static hda_nid_t ad1884a_dac_nids[1] = { +static const hda_nid_t ad1884a_dac_nids[1] = { 0x03, }; @@ -3796,7 +3796,7 @@ static hda_nid_t ad1884a_dac_nids[1] = { #define AD1884A_SPDIF_OUT 0x02 -static struct hda_input_mux ad1884a_capture_source = { +static const struct hda_input_mux ad1884a_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x0 }, @@ -3807,7 +3807,7 @@ static struct hda_input_mux ad1884a_capture_source = { }, }; -static struct snd_kcontrol_new ad1884a_base_mixers[] = { +static const struct snd_kcontrol_new ad1884a_base_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), @@ -3859,7 +3859,7 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = { /* * initialization verbs */ -static struct hda_verb ad1884a_init_verbs[] = { +static const struct hda_verb ad1884a_init_verbs[] = { /* DACs; unmute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ @@ -3914,7 +3914,7 @@ static struct hda_verb ad1884a_init_verbs[] = { }; #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list ad1884a_loopbacks[] = { +static const struct hda_amp_list ad1884a_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Mic */ { 0x20, HDA_INPUT, 2 }, /* CD */ @@ -3947,7 +3947,7 @@ static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, return ret; } -static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { +static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -3975,7 +3975,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { +static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { @@ -4095,7 +4095,7 @@ static int ad1884a_laptop_init(struct hda_codec *codec) } /* additional verbs for laptop model */ -static struct hda_verb ad1884a_laptop_verbs[] = { +static const struct hda_verb ad1884a_laptop_verbs[] = { /* Port-A (HP) pin - always unmuted */ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-F (int speaker) mixer - route only from analog mixer */ @@ -4126,7 +4126,7 @@ static struct hda_verb ad1884a_laptop_verbs[] = { { } /* end */ }; -static struct hda_verb ad1884a_mobile_verbs[] = { +static const struct hda_verb ad1884a_mobile_verbs[] = { /* DACs; unmute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ @@ -4181,7 +4181,7 @@ static struct hda_verb ad1884a_mobile_verbs[] = { * 0x17 - built-in mic */ -static struct hda_verb ad1984a_thinkpad_verbs[] = { +static const struct hda_verb ad1984a_thinkpad_verbs[] = { /* HP unmute */ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* analog mix */ @@ -4198,7 +4198,7 @@ static struct hda_verb ad1984a_thinkpad_verbs[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { +static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), @@ -4219,7 +4219,7 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { { } /* end */ }; -static struct hda_input_mux ad1984a_thinkpad_capture_source = { +static const struct hda_input_mux ad1984a_thinkpad_capture_source = { .num_items = 3, .items = { { "Mic", 0x0 }, @@ -4262,7 +4262,7 @@ static int ad1984a_thinkpad_init(struct hda_codec *codec) * 0x15 - mic-in */ -static struct hda_verb ad1984a_precision_verbs[] = { +static const struct hda_verb ad1984a_precision_verbs[] = { /* Unmute main output path */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */ @@ -4288,7 +4288,7 @@ static struct hda_verb ad1984a_precision_verbs[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1984a_precision_mixers[] = { +static const struct snd_kcontrol_new ad1984a_precision_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), @@ -4344,7 +4344,7 @@ static int ad1984a_precision_init(struct hda_codec *codec) * digital-mic (0x17) - Internal mic */ -static struct hda_verb ad1984a_touchsmart_verbs[] = { +static const struct hda_verb ad1984a_touchsmart_verbs[] = { /* DACs; unmute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ @@ -4396,7 +4396,7 @@ static struct hda_verb ad1984a_touchsmart_verbs[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { +static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { @@ -4475,7 +4475,7 @@ static const char * const ad1884a_models[AD1884A_MODELS] = { [AD1984A_PRECISION] = "precision", }; -static struct snd_pci_quirk ad1884a_cfg_tbl[] = { +static const struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION), SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), @@ -4614,22 +4614,22 @@ static int patch_ad1884a(struct hda_codec *codec) * port-G - rear clfe-out (6stack) */ -static hda_nid_t ad1882_dac_nids[3] = { +static const hda_nid_t ad1882_dac_nids[3] = { 0x04, 0x03, 0x05 }; -static hda_nid_t ad1882_adc_nids[2] = { +static const hda_nid_t ad1882_adc_nids[2] = { 0x08, 0x09, }; -static hda_nid_t ad1882_capsrc_nids[2] = { +static const hda_nid_t ad1882_capsrc_nids[2] = { 0x0c, 0x0d, }; #define AD1882_SPDIF_OUT 0x02 /* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */ -static struct hda_input_mux ad1882_capture_source = { +static const struct hda_input_mux ad1882_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x1 }, @@ -4641,7 +4641,7 @@ static struct hda_input_mux ad1882_capture_source = { }; /* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */ -static struct hda_input_mux ad1882a_capture_source = { +static const struct hda_input_mux ad1882a_capture_source = { .num_items = 5, .items = { { "Front Mic", 0x1 }, @@ -4652,7 +4652,7 @@ static struct hda_input_mux ad1882a_capture_source = { }, }; -static struct snd_kcontrol_new ad1882_base_mixers[] = { +static const struct snd_kcontrol_new ad1882_base_mixers[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), @@ -4694,7 +4694,7 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1882_loopback_mixers[] = { +static const struct snd_kcontrol_new ad1882_loopback_mixers[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), @@ -4706,7 +4706,7 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { +static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT), @@ -4719,7 +4719,7 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1882_3stack_mixers[] = { +static const struct snd_kcontrol_new ad1882_3stack_mixers[] = { HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT), @@ -4733,14 +4733,14 @@ static struct snd_kcontrol_new ad1882_3stack_mixers[] = { { } /* end */ }; -static struct snd_kcontrol_new ad1882_6stack_mixers[] = { +static const struct snd_kcontrol_new ad1882_6stack_mixers[] = { HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT), { } /* end */ }; -static struct hda_verb ad1882_ch2_init[] = { +static const struct hda_verb ad1882_ch2_init[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -4750,7 +4750,7 @@ static struct hda_verb ad1882_ch2_init[] = { { } /* end */ }; -static struct hda_verb ad1882_ch4_init[] = { +static const struct hda_verb ad1882_ch4_init[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -4760,7 +4760,7 @@ static struct hda_verb ad1882_ch4_init[] = { { } /* end */ }; -static struct hda_verb ad1882_ch6_init[] = { +static const struct hda_verb ad1882_ch6_init[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -4770,7 +4770,7 @@ static struct hda_verb ad1882_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode ad1882_modes[3] = { +static const struct hda_channel_mode ad1882_modes[3] = { { 2, ad1882_ch2_init }, { 4, ad1882_ch4_init }, { 6, ad1882_ch6_init }, @@ -4779,7 +4779,7 @@ static struct hda_channel_mode ad1882_modes[3] = { /* * initialization verbs */ -static struct hda_verb ad1882_init_verbs[] = { +static const struct hda_verb ad1882_init_verbs[] = { /* DACs; mute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -4848,7 +4848,7 @@ static struct hda_verb ad1882_init_verbs[] = { }; #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list ad1882_loopbacks[] = { +static const struct hda_amp_list ad1882_loopbacks[] = { { 0x20, HDA_INPUT, 0 }, /* Front Mic */ { 0x20, HDA_INPUT, 1 }, /* Mic */ { 0x20, HDA_INPUT, 4 }, /* Line */ @@ -4945,7 +4945,7 @@ static int patch_ad1882(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_analog[] = { +static const struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a }, { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a }, -- cgit v1.2.3 From 779d065983f30b952d66e34d230b244f44a4e032 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:34:20 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_cmedia.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cmedia.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 1f8bbcd0f80..ab3308daa96 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -53,7 +53,7 @@ struct cmi_spec { int num_dacs; /* capture */ - hda_nid_t *adc_nids; + const hda_nid_t *adc_nids; hda_nid_t dig_in_nid; /* capture source */ @@ -110,7 +110,7 @@ static int cmi_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v */ /* 3-stack / 2 channel */ -static struct hda_verb cmi9880_ch2_init[] = { +static const struct hda_verb cmi9880_ch2_init[] = { /* set line-in PIN for input */ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* set mic PIN for input, also enable vref */ @@ -121,7 +121,7 @@ static struct hda_verb cmi9880_ch2_init[] = { }; /* 3-stack / 6 channel */ -static struct hda_verb cmi9880_ch6_init[] = { +static const struct hda_verb cmi9880_ch6_init[] = { /* set line-in PIN for output */ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* set mic PIN for output */ @@ -132,7 +132,7 @@ static struct hda_verb cmi9880_ch6_init[] = { }; /* 3-stack+front / 8 channel */ -static struct hda_verb cmi9880_ch8_init[] = { +static const struct hda_verb cmi9880_ch8_init[] = { /* set line-in PIN for output */ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* set mic PIN for output */ @@ -142,7 +142,7 @@ static struct hda_verb cmi9880_ch8_init[] = { {} }; -static struct hda_channel_mode cmi9880_channel_modes[3] = { +static const struct hda_channel_mode cmi9880_channel_modes[3] = { { 2, cmi9880_ch2_init }, { 6, cmi9880_ch6_init }, { 8, cmi9880_ch8_init }, @@ -174,7 +174,7 @@ static int cmi_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va /* */ -static struct snd_kcontrol_new cmi9880_basic_mixer[] = { +static const struct snd_kcontrol_new cmi9880_basic_mixer[] = { /* CMI9880 has no playback volumes! */ HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */ HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT), @@ -205,7 +205,7 @@ static struct snd_kcontrol_new cmi9880_basic_mixer[] = { /* * shared I/O pins */ -static struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = { +static const struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -219,7 +219,7 @@ static struct snd_kcontrol_new cmi9880_ch_mode_mixer[] = { /* AUD-in selections: * 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x1f 0x20 */ -static struct hda_input_mux cmi9880_basic_mux = { +static const struct hda_input_mux cmi9880_basic_mux = { .num_items = 4, .items = { { "Front Mic", 0x5 }, @@ -229,7 +229,7 @@ static struct hda_input_mux cmi9880_basic_mux = { } }; -static struct hda_input_mux cmi9880_no_line_mux = { +static const struct hda_input_mux cmi9880_no_line_mux = { .num_items = 3, .items = { { "Front Mic", 0x5 }, @@ -239,11 +239,11 @@ static struct hda_input_mux cmi9880_no_line_mux = { }; /* front, rear, clfe, rear_surr */ -static hda_nid_t cmi9880_dac_nids[4] = { +static const hda_nid_t cmi9880_dac_nids[4] = { 0x03, 0x04, 0x05, 0x06 }; /* ADC0, ADC1 */ -static hda_nid_t cmi9880_adc_nids[2] = { +static const hda_nid_t cmi9880_adc_nids[2] = { 0x08, 0x09 }; @@ -252,7 +252,7 @@ static hda_nid_t cmi9880_adc_nids[2] = { /* */ -static struct hda_verb cmi9880_basic_init[] = { +static const struct hda_verb cmi9880_basic_init[] = { /* port-D for line out (rear panel) */ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* port-E for HP out (front panel) */ @@ -281,7 +281,7 @@ static struct hda_verb cmi9880_basic_init[] = { {} /* terminator */ }; -static struct hda_verb cmi9880_allout_init[] = { +static const struct hda_verb cmi9880_allout_init[] = { /* port-D for line out (rear panel) */ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* port-E for HP out (front panel) */ @@ -528,7 +528,7 @@ static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static struct hda_pcm_stream cmi9880_pcm_analog_playback = { +static const struct hda_pcm_stream cmi9880_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 8, @@ -540,7 +540,7 @@ static struct hda_pcm_stream cmi9880_pcm_analog_playback = { }, }; -static struct hda_pcm_stream cmi9880_pcm_analog_capture = { +static const struct hda_pcm_stream cmi9880_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -551,7 +551,7 @@ static struct hda_pcm_stream cmi9880_pcm_analog_capture = { }, }; -static struct hda_pcm_stream cmi9880_pcm_digital_playback = { +static const struct hda_pcm_stream cmi9880_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -563,7 +563,7 @@ static struct hda_pcm_stream cmi9880_pcm_digital_playback = { }, }; -static struct hda_pcm_stream cmi9880_pcm_digital_capture = { +static const struct hda_pcm_stream cmi9880_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -617,14 +617,14 @@ static const char * const cmi9880_models[CMI_MODELS] = { [CMI_AUTO] = "auto", }; -static struct snd_pci_quirk cmi9880_cfg_tbl[] = { +static const struct snd_pci_quirk cmi9880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG), SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL), SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG), {} /* terminator */ }; -static struct hda_codec_ops cmi9880_patch_ops = { +static const struct hda_codec_ops cmi9880_patch_ops = { .build_controls = cmi9880_build_controls, .build_pcms = cmi9880_build_pcms, .init = cmi9880_init, @@ -745,7 +745,7 @@ static int patch_cmi9880(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_cmedia[] = { +static const struct hda_codec_preset snd_hda_preset_cmedia[] = { { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 }, { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 }, {} /* terminator */ -- cgit v1.2.3 From 728850a7f25f4f4b41f5685945b02579f6b19276 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:36:00 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_ca0110.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 46c8bf48c31..61b92634b16 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -134,7 +134,7 @@ static int ca0110_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static char *dirstr[2] = { "Playback", "Capture" }; +static const char * const dirstr[2] = { "Playback", "Capture" }; static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int chan, int dir) @@ -171,7 +171,7 @@ static int ca0110_build_controls(struct hda_codec *codec) { struct ca0110_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - static char *prefix[AUTO_CFG_MAX_OUTS] = { + static const char * const prefix[AUTO_CFG_MAX_OUTS] = { "Front", "Surround", NULL, "Side", "Multi" }; hda_nid_t mutenid; @@ -259,7 +259,7 @@ static int ca0110_build_controls(struct hda_codec *codec) /* */ -static struct hda_pcm_stream ca0110_pcm_analog_playback = { +static const struct hda_pcm_stream ca0110_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 8, @@ -270,7 +270,7 @@ static struct hda_pcm_stream ca0110_pcm_analog_playback = { }, }; -static struct hda_pcm_stream ca0110_pcm_analog_capture = { +static const struct hda_pcm_stream ca0110_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -280,7 +280,7 @@ static struct hda_pcm_stream ca0110_pcm_analog_capture = { }, }; -static struct hda_pcm_stream ca0110_pcm_digital_playback = { +static const struct hda_pcm_stream ca0110_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -291,7 +291,7 @@ static struct hda_pcm_stream ca0110_pcm_digital_playback = { }, }; -static struct hda_pcm_stream ca0110_pcm_digital_capture = { +static const struct hda_pcm_stream ca0110_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -389,7 +389,7 @@ static void ca0110_free(struct hda_codec *codec) kfree(codec->spec); } -static struct hda_codec_ops ca0110_patch_ops = { +static const struct hda_codec_ops ca0110_patch_ops = { .build_controls = ca0110_build_controls, .build_pcms = ca0110_build_pcms, .init = ca0110_init, @@ -539,7 +539,7 @@ static int patch_ca0110(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_ca0110[] = { +static const struct hda_codec_preset snd_hda_preset_ca0110[] = { { .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 }, { .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 }, { .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 }, -- cgit v1.2.3 From c42d47829a8e62bd00b551782760d836d65d8888 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:36:09 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_cirrus.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 067982f4f18..8b735052bc1 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -51,7 +51,7 @@ struct cs_spec { unsigned int cur_adc_format; hda_nid_t dig_in; - struct hda_bind_ctls *capture_bind[2]; + const struct hda_bind_ctls *capture_bind[2]; unsigned int gpio_mask; unsigned int gpio_dir; @@ -231,7 +231,7 @@ static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static struct hda_pcm_stream cs_pcm_analog_playback = { +static const struct hda_pcm_stream cs_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -242,7 +242,7 @@ static struct hda_pcm_stream cs_pcm_analog_playback = { }, }; -static struct hda_pcm_stream cs_pcm_analog_capture = { +static const struct hda_pcm_stream cs_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -252,7 +252,7 @@ static struct hda_pcm_stream cs_pcm_analog_capture = { }, }; -static struct hda_pcm_stream cs_pcm_digital_playback = { +static const struct hda_pcm_stream cs_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -264,7 +264,7 @@ static struct hda_pcm_stream cs_pcm_digital_playback = { }, }; -static struct hda_pcm_stream cs_pcm_digital_capture = { +static const struct hda_pcm_stream cs_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -559,10 +559,10 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, const char *name; int err, index; struct snd_kcontrol *kctl; - static char *speakers[] = { + static const char * const speakers[] = { "Front Speaker", "Surround Speaker", "Bass Speaker" }; - static char *line_outs[] = { + static const char * const line_outs[] = { "Front Line-Out", "Surround Line-Out", "Bass Line-Out" }; @@ -642,7 +642,7 @@ static int build_output(struct hda_codec *codec) /* */ -static struct snd_kcontrol_new cs_capture_ctls[] = { +static const struct snd_kcontrol_new cs_capture_ctls[] = { HDA_BIND_SW("Capture Switch", 0), HDA_BIND_VOL("Capture Volume", 0), }; @@ -710,7 +710,7 @@ static int cs_capture_source_put(struct snd_kcontrol *kcontrol, return change_cur_input(codec, idx, 0); } -static struct snd_kcontrol_new cs_capture_source = { +static const struct snd_kcontrol_new cs_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -719,7 +719,7 @@ static struct snd_kcontrol_new cs_capture_source = { .put = cs_capture_source_put, }; -static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, +static const struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, struct hda_ctl_ops *ops) { struct cs_spec *spec = codec->spec; @@ -983,7 +983,7 @@ static void init_input(struct hda_codec *codec) cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); } -static struct hda_verb cs_coef_init_verbs[] = { +static const struct hda_verb cs_coef_init_verbs[] = { {0x11, AC_VERB_SET_PROC_STATE, 1}, {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, {0x11, AC_VERB_SET_PROC_COEF, @@ -1017,7 +1017,7 @@ static struct hda_verb cs_coef_init_verbs[] = { * blocks, which will alleviate the issue. */ -static struct hda_verb cs_errata_init_verbs[] = { +static const struct hda_verb cs_errata_init_verbs[] = { {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ @@ -1126,7 +1126,7 @@ static void cs_unsol_event(struct hda_codec *codec, unsigned int res) } } -static struct hda_codec_ops cs_patch_ops = { +static const struct hda_codec_ops cs_patch_ops = { .build_controls = cs_build_controls, .build_pcms = cs_build_pcms, .init = cs_init, @@ -1166,7 +1166,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { }; -static struct snd_pci_quirk cs420x_cfg_tbl[] = { +static const struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), @@ -1180,7 +1180,7 @@ struct cs_pincfg { u32 val; }; -static struct cs_pincfg mbp53_pincfgs[] = { +static const struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1194,7 +1194,7 @@ static struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static struct cs_pincfg mbp55_pincfgs[] = { +static const struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1208,7 +1208,7 @@ static struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static struct cs_pincfg imac27_pincfgs[] = { +static const struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1222,7 +1222,7 @@ static struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, @@ -1283,7 +1283,7 @@ static int patch_cs420x(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_cirrus[] = { +static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, {} /* terminator */ -- cgit v1.2.3 From 34cbe3a6faa8715c4f9e07484248aa32c1c7577e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 11:38:21 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_conexant.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 190 ++++++++++++++++++++--------------------- 1 file changed, 95 insertions(+), 95 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ad97d937d3a..35bebe59908 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -57,7 +57,7 @@ struct pin_dac_pair { struct conexant_spec { - struct snd_kcontrol_new *mixers[5]; + const struct snd_kcontrol_new *mixers[5]; int num_mixers; hda_nid_t vmaster_nid; @@ -81,7 +81,7 @@ struct conexant_spec { /* capture */ unsigned int num_adc_nids; - hda_nid_t *adc_nids; + const hda_nid_t *adc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ unsigned int cur_adc_idx; @@ -91,7 +91,7 @@ struct conexant_spec { /* capture source */ const struct hda_input_mux *input_mux; - hda_nid_t *capsrc_nids; + const hda_nid_t *capsrc_nids; unsigned int cur_mux[3]; /* channel model */ @@ -227,7 +227,7 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, -static struct hda_pcm_stream conexant_pcm_analog_playback = { +static const struct hda_pcm_stream conexant_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -239,7 +239,7 @@ static struct hda_pcm_stream conexant_pcm_analog_playback = { }, }; -static struct hda_pcm_stream conexant_pcm_analog_capture = { +static const struct hda_pcm_stream conexant_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -251,7 +251,7 @@ static struct hda_pcm_stream conexant_pcm_analog_capture = { }; -static struct hda_pcm_stream conexant_pcm_digital_playback = { +static const struct hda_pcm_stream conexant_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -263,7 +263,7 @@ static struct hda_pcm_stream conexant_pcm_digital_playback = { }, }; -static struct hda_pcm_stream conexant_pcm_digital_capture = { +static const struct hda_pcm_stream conexant_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -294,7 +294,7 @@ static int cx5051_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream cx5051_pcm_analog_capture = { +static const struct hda_pcm_stream cx5051_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -433,7 +433,7 @@ static void conexant_free(struct hda_codec *codec) kfree(codec->spec); } -static struct snd_kcontrol_new cxt_capture_mixers[] = { +static const struct snd_kcontrol_new cxt_capture_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -446,7 +446,7 @@ static struct snd_kcontrol_new cxt_capture_mixers[] = { #ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ -static struct snd_kcontrol_new cxt_beep_mixer[] = { +static const struct snd_kcontrol_new cxt_beep_mixer[] = { HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0, 1, 0, HDA_OUTPUT), HDA_CODEC_MUTE_BEEP_MONO("Beep Playback Switch", 0, 1, 0, HDA_OUTPUT), { } /* end */ @@ -521,7 +521,7 @@ static int conexant_build_controls(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_INPUT_BEEP /* create beep controls if needed */ if (spec->beep_amp) { - struct snd_kcontrol_new *knew; + const struct snd_kcontrol_new *knew; for (knew = cxt_beep_mixer; knew->name; knew++) { struct snd_kcontrol *kctl; kctl = snd_ctl_new1(knew, codec); @@ -546,7 +546,7 @@ static int conexant_suspend(struct hda_codec *codec, pm_message_t state) } #endif -static struct hda_codec_ops conexant_patch_ops = { +static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, @@ -662,16 +662,16 @@ static int conexant_ch_mode_put(struct snd_kcontrol *kcontrol, /* Conexant 5045 specific */ -static hda_nid_t cxt5045_dac_nids[1] = { 0x19 }; -static hda_nid_t cxt5045_adc_nids[1] = { 0x1a }; -static hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a }; +static const hda_nid_t cxt5045_dac_nids[1] = { 0x19 }; +static const hda_nid_t cxt5045_adc_nids[1] = { 0x1a }; +static const hda_nid_t cxt5045_capsrc_nids[1] = { 0x1a }; #define CXT5045_SPDIF_OUT 0x18 -static struct hda_channel_mode cxt5045_modes[1] = { +static const struct hda_channel_mode cxt5045_modes[1] = { { 2, NULL }, }; -static struct hda_input_mux cxt5045_capture_source = { +static const struct hda_input_mux cxt5045_capture_source = { .num_items = 2, .items = { { "IntMic", 0x1 }, @@ -679,7 +679,7 @@ static struct hda_input_mux cxt5045_capture_source = { } }; -static struct hda_input_mux cxt5045_capture_source_benq = { +static const struct hda_input_mux cxt5045_capture_source_benq = { .num_items = 5, .items = { { "IntMic", 0x1 }, @@ -690,7 +690,7 @@ static struct hda_input_mux cxt5045_capture_source_benq = { } }; -static struct hda_input_mux cxt5045_capture_source_hp530 = { +static const struct hda_input_mux cxt5045_capture_source_hp530 = { .num_items = 2, .items = { { "ExtMic", 0x1 }, @@ -723,7 +723,7 @@ static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol, } /* bind volumes of both NID 0x10 and 0x11 */ -static struct hda_bind_ctls cxt5045_hp_bind_master_vol = { +static const struct hda_bind_ctls cxt5045_hp_bind_master_vol = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT), @@ -735,12 +735,12 @@ static struct hda_bind_ctls cxt5045_hp_bind_master_vol = { /* toggle input of built-in and mic jack appropriately */ static void cxt5045_hp_automic(struct hda_codec *codec) { - static struct hda_verb mic_jack_on[] = { + static const struct hda_verb mic_jack_on[] = { {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {} }; - static struct hda_verb mic_jack_off[] = { + static const struct hda_verb mic_jack_off[] = { {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {} @@ -784,7 +784,7 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } } -static struct snd_kcontrol_new cxt5045_mixers[] = { +static const struct snd_kcontrol_new cxt5045_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), @@ -808,7 +808,7 @@ static struct snd_kcontrol_new cxt5045_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5045_benq_mixers[] = { +static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), @@ -825,7 +825,7 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { +static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = { HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), @@ -849,7 +849,7 @@ static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { {} }; -static struct hda_verb cxt5045_init_verbs[] = { +static const struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, @@ -875,7 +875,7 @@ static struct hda_verb cxt5045_init_verbs[] = { { } /* end */ }; -static struct hda_verb cxt5045_benq_init_verbs[] = { +static const struct hda_verb cxt5045_benq_init_verbs[] = { /* Internal Mic, Mic */ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, @@ -901,13 +901,13 @@ static struct hda_verb cxt5045_benq_init_verbs[] = { { } /* end */ }; -static struct hda_verb cxt5045_hp_sense_init_verbs[] = { +static const struct hda_verb cxt5045_hp_sense_init_verbs[] = { /* pin sensing on HP jack */ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, { } /* end */ }; -static struct hda_verb cxt5045_mic_sense_init_verbs[] = { +static const struct hda_verb cxt5045_mic_sense_init_verbs[] = { /* pin sensing on HP jack */ {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, { } /* end */ @@ -917,7 +917,7 @@ static struct hda_verb cxt5045_mic_sense_init_verbs[] = { /* Test configuration for debugging, modelled after the ALC260 test * configuration. */ -static struct hda_input_mux cxt5045_test_capture_source = { +static const struct hda_input_mux cxt5045_test_capture_source = { .num_items = 5, .items = { { "MIXER", 0x0 }, @@ -928,7 +928,7 @@ static struct hda_input_mux cxt5045_test_capture_source = { }, }; -static struct snd_kcontrol_new cxt5045_test_mixer[] = { +static const struct snd_kcontrol_new cxt5045_test_mixer[] = { /* Output controls */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), @@ -978,7 +978,7 @@ static struct snd_kcontrol_new cxt5045_test_mixer[] = { { } /* end */ }; -static struct hda_verb cxt5045_test_init_verbs[] = { +static const struct hda_verb cxt5045_test_init_verbs[] = { /* Set connections */ { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, { 0x11, AC_VERB_SET_CONNECT_SEL, 0x0 }, @@ -1061,7 +1061,7 @@ static const char * const cxt5045_models[CXT5045_MODELS] = { #endif }; -static struct snd_pci_quirk cxt5045_cfg_tbl[] = { +static const struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", CXT5045_LAPTOP_HPSENSE), @@ -1196,15 +1196,15 @@ static int patch_cxt5045(struct hda_codec *codec) /* Conexant 5047 specific */ #define CXT5047_SPDIF_OUT 0x11 -static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ -static hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; -static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; +static const hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ +static const hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; +static const hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; -static struct hda_channel_mode cxt5047_modes[1] = { +static const struct hda_channel_mode cxt5047_modes[1] = { { 2, NULL }, }; -static struct hda_input_mux cxt5047_toshiba_capture_source = { +static const struct hda_input_mux cxt5047_toshiba_capture_source = { .num_items = 2, .items = { { "ExtMic", 0x2 }, @@ -1256,12 +1256,12 @@ static void cxt5047_hp_automute(struct hda_codec *codec) /* toggle input of built-in and mic jack appropriately */ static void cxt5047_hp_automic(struct hda_codec *codec) { - static struct hda_verb mic_jack_on[] = { + static const struct hda_verb mic_jack_on[] = { {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {} }; - static struct hda_verb mic_jack_off[] = { + static const struct hda_verb mic_jack_off[] = { {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {} @@ -1289,7 +1289,7 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec, } } -static struct snd_kcontrol_new cxt5047_base_mixers[] = { +static const struct snd_kcontrol_new cxt5047_base_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), @@ -1309,19 +1309,19 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { +static const struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { /* See the note in cxt5047_hp_master_sw_put */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), {} }; -static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { +static const struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), { } /* end */ }; -static struct hda_verb cxt5047_init_verbs[] = { +static const struct hda_verb cxt5047_init_verbs[] = { /* Line in, Mic, Built-in Mic */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, @@ -1348,7 +1348,7 @@ static struct hda_verb cxt5047_init_verbs[] = { }; /* configuration for Toshiba Laptops */ -static struct hda_verb cxt5047_toshiba_init_verbs[] = { +static const struct hda_verb cxt5047_toshiba_init_verbs[] = { {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ {} }; @@ -1357,7 +1357,7 @@ static struct hda_verb cxt5047_toshiba_init_verbs[] = { * configuration. */ #ifdef CONFIG_SND_DEBUG -static struct hda_input_mux cxt5047_test_capture_source = { +static const struct hda_input_mux cxt5047_test_capture_source = { .num_items = 4, .items = { { "LINE1 pin", 0x0 }, @@ -1367,7 +1367,7 @@ static struct hda_input_mux cxt5047_test_capture_source = { }, }; -static struct snd_kcontrol_new cxt5047_test_mixer[] = { +static const struct snd_kcontrol_new cxt5047_test_mixer[] = { /* Output only controls */ HDA_CODEC_VOLUME("OutAmp-1 Volume", 0x10, 0x0, HDA_OUTPUT), @@ -1420,7 +1420,7 @@ static struct snd_kcontrol_new cxt5047_test_mixer[] = { { } /* end */ }; -static struct hda_verb cxt5047_test_init_verbs[] = { +static const struct hda_verb cxt5047_test_init_verbs[] = { /* Enable retasking pins as output, initially without power amp */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -1504,7 +1504,7 @@ static const char * const cxt5047_models[CXT5047_MODELS] = { #endif }; -static struct snd_pci_quirk cxt5047_cfg_tbl[] = { +static const struct snd_pci_quirk cxt5047_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", CXT5047_LAPTOP), @@ -1591,10 +1591,10 @@ static int patch_cxt5047(struct hda_codec *codec) } /* Conexant 5051 specific */ -static hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; -static hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; +static const hda_nid_t cxt5051_dac_nids[1] = { 0x10 }; +static const hda_nid_t cxt5051_adc_nids[2] = { 0x14, 0x15 }; -static struct hda_channel_mode cxt5051_modes[1] = { +static const struct hda_channel_mode cxt5051_modes[1] = { { 2, NULL }, }; @@ -1696,7 +1696,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec, snd_hda_input_jack_report(codec, nid); } -static struct snd_kcontrol_new cxt5051_playback_mixers[] = { +static const struct snd_kcontrol_new cxt5051_playback_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1709,7 +1709,7 @@ static struct snd_kcontrol_new cxt5051_playback_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5051_capture_mixers[] = { +static const struct snd_kcontrol_new cxt5051_capture_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), @@ -1719,7 +1719,7 @@ static struct snd_kcontrol_new cxt5051_capture_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5051_hp_mixers[] = { +static const struct snd_kcontrol_new cxt5051_hp_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Volume", 0x15, 0x00, HDA_INPUT), @@ -1727,19 +1727,19 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { +static const struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0x00, HDA_INPUT), {} }; -static struct snd_kcontrol_new cxt5051_f700_mixers[] = { +static const struct snd_kcontrol_new cxt5051_f700_mixers[] = { HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0x01, HDA_INPUT), {} }; -static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { +static const struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { HDA_CODEC_VOLUME("Internal Mic Volume", 0x14, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Switch", 0x14, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), @@ -1747,7 +1747,7 @@ static struct snd_kcontrol_new cxt5051_toshiba_mixers[] = { {} }; -static struct hda_verb cxt5051_init_verbs[] = { +static const struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -1776,7 +1776,7 @@ static struct hda_verb cxt5051_init_verbs[] = { { } /* end */ }; -static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { +static const struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -1801,7 +1801,7 @@ static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { { } /* end */ }; -static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { +static const struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -1834,7 +1834,7 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { { } /* end */ }; -static struct hda_verb cxt5051_f700_init_verbs[] = { +static const struct hda_verb cxt5051_f700_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -1869,7 +1869,7 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, snd_hda_input_jack_report(codec, nid); } -static struct hda_verb cxt5051_ideapad_init_verbs[] = { +static const struct hda_verb cxt5051_ideapad_init_verbs[] = { /* Subwoofer */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1919,7 +1919,7 @@ static const char *const cxt5051_models[CXT5051_MODELS] = { [CXT5051_IDEAPAD] = "ideapad", }; -static struct snd_pci_quirk cxt5051_cfg_tbl[] = { +static const struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), @@ -2011,17 +2011,17 @@ static int patch_cxt5051(struct hda_codec *codec) /* Conexant 5066 specific */ -static hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; -static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; -static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; -static hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 }; +static const hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; +static const hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; +static const hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; +static const hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 }; /* OLPC's microphone port is DC coupled for use with external sensors, * therefore we use a 50% mic bias in order to center the input signal with * the DC input range of the codec. */ #define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50 -static struct hda_channel_mode cxt5066_modes[1] = { +static const struct hda_channel_mode cxt5066_modes[1] = { { 2, NULL }, }; @@ -2176,7 +2176,7 @@ static void cxt5066_vostro_automic(struct hda_codec *codec) {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {} }; - static struct hda_verb ext_mic_absent[] = { + static const struct hda_verb ext_mic_absent[] = { /* enable internal mic, port C */ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -2209,7 +2209,7 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {} }; - static struct hda_verb ext_mic_absent[] = { + static const struct hda_verb ext_mic_absent[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 2}, {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, @@ -2257,7 +2257,7 @@ static void cxt5066_thinkpad_automic(struct hda_codec *codec) { unsigned int ext_present, dock_present; - static struct hda_verb ext_mic_present[] = { + static const struct hda_verb ext_mic_present[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0}, {0x17, AC_VERB_SET_CONNECT_SEL, 1}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -2265,7 +2265,7 @@ static void cxt5066_thinkpad_automic(struct hda_codec *codec) {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {} }; - static struct hda_verb dock_mic_present[] = { + static const struct hda_verb dock_mic_present[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0}, {0x17, AC_VERB_SET_CONNECT_SEL, 0}, {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -2273,7 +2273,7 @@ static void cxt5066_thinkpad_automic(struct hda_codec *codec) {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {} }; - static struct hda_verb ext_mic_absent[] = { + static const struct hda_verb ext_mic_absent[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 2}, {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, @@ -2537,7 +2537,7 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) } static void conexant_check_dig_outs(struct hda_codec *codec, - hda_nid_t *dig_pins, + const hda_nid_t *dig_pins, int num_pins) { struct conexant_spec *spec = codec->spec; @@ -2557,7 +2557,7 @@ static void conexant_check_dig_outs(struct hda_codec *codec, } } -static struct hda_input_mux cxt5066_capture_source = { +static const struct hda_input_mux cxt5066_capture_source = { .num_items = 4, .items = { { "Mic B", 0 }, @@ -2567,7 +2567,7 @@ static struct hda_input_mux cxt5066_capture_source = { }, }; -static struct hda_bind_ctls cxt5066_bind_capture_vol_others = { +static const struct hda_bind_ctls cxt5066_bind_capture_vol_others = { .ops = &snd_hda_bind_vol, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), @@ -2576,7 +2576,7 @@ static struct hda_bind_ctls cxt5066_bind_capture_vol_others = { }, }; -static struct hda_bind_ctls cxt5066_bind_capture_sw_others = { +static const struct hda_bind_ctls cxt5066_bind_capture_sw_others = { .ops = &snd_hda_bind_sw, .values = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), @@ -2585,12 +2585,12 @@ static struct hda_bind_ctls cxt5066_bind_capture_sw_others = { }, }; -static struct snd_kcontrol_new cxt5066_mixer_master[] = { +static const struct snd_kcontrol_new cxt5066_mixer_master[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), {} }; -static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { +static const struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", @@ -2609,7 +2609,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { {} }; -static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = { +static const struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "DC Mode Enable Switch", @@ -2627,7 +2627,7 @@ static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = { {} }; -static struct snd_kcontrol_new cxt5066_mixers[] = { +static const struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -2650,7 +2650,7 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { +static const struct snd_kcontrol_new cxt5066_vostro_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Internal Mic Boost Capture Enum", @@ -2662,7 +2662,7 @@ static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { {} }; -static struct hda_verb cxt5066_init_verbs[] = { +static const struct hda_verb cxt5066_init_verbs[] = { {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ @@ -2717,7 +2717,7 @@ static struct hda_verb cxt5066_init_verbs[] = { { } /* end */ }; -static struct hda_verb cxt5066_init_verbs_olpc[] = { +static const struct hda_verb cxt5066_init_verbs_olpc[] = { /* Port A: headphones */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ @@ -2778,7 +2778,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { { } /* end */ }; -static struct hda_verb cxt5066_init_verbs_vostro[] = { +static const struct hda_verb cxt5066_init_verbs_vostro[] = { /* Port A: headphones */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ @@ -2839,7 +2839,7 @@ static struct hda_verb cxt5066_init_verbs_vostro[] = { { } /* end */ }; -static struct hda_verb cxt5066_init_verbs_ideapad[] = { +static const struct hda_verb cxt5066_init_verbs_ideapad[] = { {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ @@ -2889,7 +2889,7 @@ static struct hda_verb cxt5066_init_verbs_ideapad[] = { { } /* end */ }; -static struct hda_verb cxt5066_init_verbs_thinkpad[] = { +static const struct hda_verb cxt5066_init_verbs_thinkpad[] = { {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ @@ -2947,13 +2947,13 @@ static struct hda_verb cxt5066_init_verbs_thinkpad[] = { { } /* end */ }; -static struct hda_verb cxt5066_init_verbs_portd_lo[] = { +static const struct hda_verb cxt5066_init_verbs_portd_lo[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { } /* end */ }; -static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { +static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, @@ -3011,7 +3011,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_HP_LAPTOP] = "hp-laptop", }; -static struct snd_pci_quirk cxt5066_cfg_tbl[] = { +static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), @@ -3195,7 +3195,7 @@ static int patch_cxt5066(struct hda_codec *codec) * Automatic parser for CX20641 & co */ -static hda_nid_t cx_auto_adc_nids[] = { 0x14 }; +static const hda_nid_t cx_auto_adc_nids[] = { 0x14 }; /* get the connection index of @nid in the widget @mux */ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, @@ -3646,7 +3646,7 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, HDA_CODEC_VOLUME(name, 0, 0, 0), HDA_CODEC_MUTE(name, 0, 0, 0), }; - static char *sfx[2] = { "Volume", "Switch" }; + static const char * const sfx[2] = { "Volume", "Switch" }; int i, err; for (i = 0; i < 2; i++) { @@ -3791,7 +3791,7 @@ static int cx_auto_build_controls(struct hda_codec *codec) return conexant_build_controls(codec); } -static struct hda_codec_ops cx_auto_patch_ops = { +static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = conexant_build_pcms, .init = cx_auto_init, @@ -3830,7 +3830,7 @@ static int patch_conexant_auto(struct hda_codec *codec) /* */ -static struct hda_codec_preset snd_hda_preset_conexant[] = { +static const struct hda_codec_preset snd_hda_preset_conexant[] = { { .id = 0x14f15045, .name = "CX20549 (Venice)", .patch = patch_cxt5045 }, { .id = 0x14f15047, .name = "CX20551 (Waikiki)", -- cgit v1.2.3 From fb79e1e0a2eef3859f9bcc0df67fa3f1352735d3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 12:17:41 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_hdmi.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 715615a88a8..6eb209d4b5b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -76,7 +76,7 @@ struct hdmi_spec { * ati/nvhdmi specific */ struct hda_multi_out multiout; - struct hda_pcm_stream *pcm_playback; + const struct hda_pcm_stream *pcm_playback; /* misc flags */ /* PD bit indicates only the update, not the current state */ @@ -1044,7 +1044,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } -static struct hda_pcm_stream generic_hdmi_pcm_playback = { +static const struct hda_pcm_stream generic_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, .ops = { @@ -1124,7 +1124,7 @@ static void generic_hdmi_free(struct hda_codec *codec) kfree(spec); } -static struct hda_codec_ops generic_hdmi_patch_ops = { +static const struct hda_codec_ops generic_hdmi_patch_ops = { .init = generic_hdmi_init, .free = generic_hdmi_free, .build_pcms = generic_hdmi_build_pcms, @@ -1169,12 +1169,12 @@ static int patch_generic_hdmi(struct hda_codec *codec) #define nvhdmi_master_con_nid_7x 0x04 #define nvhdmi_master_pin_nid_7x 0x05 -static hda_nid_t nvhdmi_con_nids_7x[4] = { +static const hda_nid_t nvhdmi_con_nids_7x[4] = { /*front, rear, clfe, rear_surr */ 0x6, 0x8, 0xa, 0xc, }; -static struct hda_verb nvhdmi_basic_init_7x[] = { +static const struct hda_verb nvhdmi_basic_init_7x[] = { /* set audio protect on */ { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, /* enable digital output on pin widget */ @@ -1435,7 +1435,7 @@ static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = { +static const struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = { .substreams = 1, .channels_min = 2, .channels_max = 8, @@ -1450,7 +1450,7 @@ static struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = { }, }; -static struct hda_pcm_stream nvhdmi_pcm_playback_2ch = { +static const struct hda_pcm_stream nvhdmi_pcm_playback_2ch = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -1465,14 +1465,14 @@ static struct hda_pcm_stream nvhdmi_pcm_playback_2ch = { }, }; -static struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { +static const struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { .build_controls = generic_hdmi_build_controls, .build_pcms = generic_hdmi_build_pcms, .init = nvhdmi_7x_init, .free = generic_hdmi_free, }; -static struct hda_codec_ops nvhdmi_patch_ops_2ch = { +static const struct hda_codec_ops nvhdmi_patch_ops_2ch = { .build_controls = generic_hdmi_build_controls, .build_pcms = generic_hdmi_build_pcms, .init = nvhdmi_7x_init, @@ -1568,7 +1568,7 @@ static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream atihdmi_pcm_digital_playback = { +static const struct hda_pcm_stream atihdmi_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -1580,7 +1580,7 @@ static struct hda_pcm_stream atihdmi_pcm_digital_playback = { }, }; -static struct hda_verb atihdmi_basic_init[] = { +static const struct hda_verb atihdmi_basic_init[] = { /* enable digital output on pin widget */ { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {} /* terminator */ @@ -1599,7 +1599,7 @@ static int atihdmi_init(struct hda_codec *codec) return 0; } -static struct hda_codec_ops atihdmi_patch_ops = { +static const struct hda_codec_ops atihdmi_patch_ops = { .build_controls = generic_hdmi_build_controls, .build_pcms = generic_hdmi_build_pcms, .init = atihdmi_init, @@ -1634,7 +1634,7 @@ static int patch_atihdmi(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_hdmi[] = { +static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, -- cgit v1.2.3 From 9cf0aa9ebae703c117421d94f20b2ece8112218e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 12:22:39 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_si3054.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_si3054.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index f419ee8d75f..2f55f32876f 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -130,7 +130,7 @@ static int si3054_switch_put(struct snd_kcontrol *kcontrol, } -static struct snd_kcontrol_new si3054_modem_mixer[] = { +static const struct snd_kcontrol_new si3054_modem_mixer[] = { SI3054_KCONTROL("Off-hook Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_OH), SI3054_KCONTROL("Caller ID Switch", SI3054_GPIO_CONTROL, SI3054_GPIO_CID), {} @@ -181,7 +181,7 @@ static int si3054_pcm_open(struct hda_pcm_stream *hinfo, } -static struct hda_pcm_stream si3054_pcm = { +static const struct hda_pcm_stream si3054_pcm = { .substreams = 1, .channels_min = 1, .channels_max = 1, @@ -200,12 +200,13 @@ static int si3054_build_pcms(struct hda_codec *codec) { struct si3054_spec *spec = codec->spec; struct hda_pcm *info = &spec->pcm; - si3054_pcm.nid = codec->mfg; codec->num_pcms = 1; codec->pcm_info = info; info->name = "Si3054 Modem"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = codec->mfg; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = codec->mfg; info->pcm_type = HDA_PCM_TYPE_MODEM; return 0; } @@ -263,7 +264,7 @@ static void si3054_free(struct hda_codec *codec) /* */ -static struct hda_codec_ops si3054_patch_ops = { +static const struct hda_codec_ops si3054_patch_ops = { .build_controls = si3054_build_controls, .build_pcms = si3054_build_pcms, .init = si3054_init, @@ -283,7 +284,7 @@ static int patch_si3054(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_si3054[] = { +static const struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 }, -- cgit v1.2.3 From 2b63536f0cc20c30619a90dc2d36cc64a9d75c28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 12:33:43 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_sigmatel.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 394 +++++++++++++++++++++-------------------- 1 file changed, 199 insertions(+), 195 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1ce65d4e05a..a2884bb8803 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -217,15 +217,15 @@ struct sigmatel_spec { unsigned int stream_delay; /* analog loopback */ - struct snd_kcontrol_new *aloopback_ctl; + const struct snd_kcontrol_new *aloopback_ctl; unsigned char aloopback_mask; unsigned char aloopback_shift; /* power management */ unsigned int num_pwrs; - unsigned int *pwr_mapping; - hda_nid_t *pwr_nids; - hda_nid_t *dac_list; + const unsigned int *pwr_mapping; + const hda_nid_t *pwr_nids; + const hda_nid_t *dac_list; /* events */ struct snd_array events; @@ -241,20 +241,20 @@ struct sigmatel_spec { int volume_offset; /* capture */ - hda_nid_t *adc_nids; + const hda_nid_t *adc_nids; unsigned int num_adcs; - hda_nid_t *mux_nids; + const hda_nid_t *mux_nids; unsigned int num_muxes; - hda_nid_t *dmic_nids; + const hda_nid_t *dmic_nids; unsigned int num_dmics; - hda_nid_t *dmux_nids; + const hda_nid_t *dmux_nids; unsigned int num_dmuxes; - hda_nid_t *smux_nids; + const hda_nid_t *smux_nids; unsigned int num_smuxes; unsigned int num_analog_muxes; - unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */ - unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */ + const unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */ + const unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */ unsigned int num_caps; /* number of capture volume/switch elements */ struct sigmatel_mic_route ext_mic; @@ -269,12 +269,12 @@ struct sigmatel_spec { hda_nid_t digbeep_nid; /* pin widgets */ - hda_nid_t *pin_nids; + const hda_nid_t *pin_nids; unsigned int num_pins; /* codec specific stuff */ - struct hda_verb *init; - struct snd_kcontrol_new *mixer; + const struct hda_verb *init; + const struct snd_kcontrol_new *mixer; /* capture source */ struct hda_input_mux *dinput_mux; @@ -317,52 +317,52 @@ struct sigmatel_spec { hda_nid_t auto_dmic_nids[MAX_DMICS_NUM]; }; -static hda_nid_t stac9200_adc_nids[1] = { +static const hda_nid_t stac9200_adc_nids[1] = { 0x03, }; -static hda_nid_t stac9200_mux_nids[1] = { +static const hda_nid_t stac9200_mux_nids[1] = { 0x0c, }; -static hda_nid_t stac9200_dac_nids[1] = { +static const hda_nid_t stac9200_dac_nids[1] = { 0x02, }; -static hda_nid_t stac92hd73xx_pwr_nids[8] = { +static const hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0a, 0x0b, 0x0c, 0xd, 0x0e, 0x0f, 0x10, 0x11 }; -static hda_nid_t stac92hd73xx_slave_dig_outs[2] = { +static const hda_nid_t stac92hd73xx_slave_dig_outs[2] = { 0x26, 0, }; -static hda_nid_t stac92hd73xx_adc_nids[2] = { +static const hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; #define STAC92HD73XX_NUM_DMICS 2 -static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { +static const hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { 0x13, 0x14, 0 }; #define STAC92HD73_DAC_COUNT 5 -static hda_nid_t stac92hd73xx_mux_nids[2] = { +static const hda_nid_t stac92hd73xx_mux_nids[2] = { 0x20, 0x21, }; -static hda_nid_t stac92hd73xx_dmux_nids[2] = { +static const hda_nid_t stac92hd73xx_dmux_nids[2] = { 0x20, 0x21, }; -static hda_nid_t stac92hd73xx_smux_nids[2] = { +static const hda_nid_t stac92hd73xx_smux_nids[2] = { 0x22, 0x23, }; #define STAC92HD73XX_NUM_CAPS 2 -static unsigned long stac92hd73xx_capvols[] = { +static const unsigned long stac92hd73xx_capvols[] = { HDA_COMPOSE_AMP_VAL(0x20, 3, 0, HDA_OUTPUT), HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), }; @@ -370,137 +370,141 @@ static unsigned long stac92hd73xx_capvols[] = { #define STAC92HD83_DAC_COUNT 3 -static hda_nid_t stac92hd83xxx_pwr_nids[4] = { +static const hda_nid_t stac92hd83xxx_pwr_nids[4] = { 0xa, 0xb, 0xd, 0xe, }; -static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { +static const hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { 0x1e, 0, }; -static unsigned int stac92hd83xxx_pwr_mapping[4] = { +static const unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x20, 0x40, }; -static hda_nid_t stac92hd83xxx_dmic_nids[] = { +static const hda_nid_t stac92hd83xxx_dmic_nids[] = { 0x11, 0x20, }; -static hda_nid_t stac92hd71bxx_pwr_nids[3] = { +static const hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f }; -static hda_nid_t stac92hd71bxx_adc_nids[2] = { +static const hda_nid_t stac92hd71bxx_adc_nids[2] = { 0x12, 0x13, }; -static hda_nid_t stac92hd71bxx_mux_nids[2] = { +static const hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd71bxx_dmux_nids[2] = { +static const hda_nid_t stac92hd71bxx_dmux_nids[2] = { 0x1c, 0x1d, }; -static hda_nid_t stac92hd71bxx_smux_nids[2] = { +static const hda_nid_t stac92hd71bxx_smux_nids[2] = { 0x24, 0x25, }; #define STAC92HD71BXX_NUM_DMICS 2 -static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { +static const hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 }; -static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { +static const hda_nid_t stac92hd71bxx_dmic_5port_nids[STAC92HD71BXX_NUM_DMICS] = { + 0x18, 0 +}; + +static const hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { 0x22, 0 }; #define STAC92HD71BXX_NUM_CAPS 2 -static unsigned long stac92hd71bxx_capvols[] = { +static const unsigned long stac92hd71bxx_capvols[] = { HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), }; #define stac92hd71bxx_capsws stac92hd71bxx_capvols -static hda_nid_t stac925x_adc_nids[1] = { +static const hda_nid_t stac925x_adc_nids[1] = { 0x03, }; -static hda_nid_t stac925x_mux_nids[1] = { +static const hda_nid_t stac925x_mux_nids[1] = { 0x0f, }; -static hda_nid_t stac925x_dac_nids[1] = { +static const hda_nid_t stac925x_dac_nids[1] = { 0x02, }; #define STAC925X_NUM_DMICS 1 -static hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = { +static const hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = { 0x15, 0 }; -static hda_nid_t stac925x_dmux_nids[1] = { +static const hda_nid_t stac925x_dmux_nids[1] = { 0x14, }; -static unsigned long stac925x_capvols[] = { +static const unsigned long stac925x_capvols[] = { HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT), }; -static unsigned long stac925x_capsws[] = { +static const unsigned long stac925x_capsws[] = { HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }; -static hda_nid_t stac922x_adc_nids[2] = { +static const hda_nid_t stac922x_adc_nids[2] = { 0x06, 0x07, }; -static hda_nid_t stac922x_mux_nids[2] = { +static const hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; #define STAC922X_NUM_CAPS 2 -static unsigned long stac922x_capvols[] = { +static const unsigned long stac922x_capvols[] = { HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT), HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), }; #define stac922x_capsws stac922x_capvols -static hda_nid_t stac927x_slave_dig_outs[2] = { +static const hda_nid_t stac927x_slave_dig_outs[2] = { 0x1f, 0, }; -static hda_nid_t stac927x_adc_nids[3] = { +static const hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; -static hda_nid_t stac927x_mux_nids[3] = { +static const hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; -static hda_nid_t stac927x_smux_nids[1] = { +static const hda_nid_t stac927x_smux_nids[1] = { 0x21, }; -static hda_nid_t stac927x_dac_nids[6] = { +static const hda_nid_t stac927x_dac_nids[6] = { 0x02, 0x03, 0x04, 0x05, 0x06, 0 }; -static hda_nid_t stac927x_dmux_nids[1] = { +static const hda_nid_t stac927x_dmux_nids[1] = { 0x1b, }; #define STAC927X_NUM_DMICS 2 -static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { +static const hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 }; #define STAC927X_NUM_CAPS 3 -static unsigned long stac927x_capvols[] = { +static const unsigned long stac927x_capvols[] = { HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), HDA_COMPOSE_AMP_VAL(0x19, 3, 0, HDA_INPUT), HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_INPUT), }; -static unsigned long stac927x_capsws[] = { +static const unsigned long stac927x_capsws[] = { HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), @@ -511,77 +515,77 @@ static const char * const stac927x_spdif_labels[5] = { "Analog Mux 2", "Analog Mux 3" }; -static hda_nid_t stac9205_adc_nids[2] = { +static const hda_nid_t stac9205_adc_nids[2] = { 0x12, 0x13 }; -static hda_nid_t stac9205_mux_nids[2] = { +static const hda_nid_t stac9205_mux_nids[2] = { 0x19, 0x1a }; -static hda_nid_t stac9205_dmux_nids[1] = { +static const hda_nid_t stac9205_dmux_nids[1] = { 0x1d, }; -static hda_nid_t stac9205_smux_nids[1] = { +static const hda_nid_t stac9205_smux_nids[1] = { 0x21, }; #define STAC9205_NUM_DMICS 2 -static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { +static const hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 }; #define STAC9205_NUM_CAPS 2 -static unsigned long stac9205_capvols[] = { +static const unsigned long stac9205_capvols[] = { HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_INPUT), HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_INPUT), }; -static unsigned long stac9205_capsws[] = { +static const unsigned long stac9205_capsws[] = { HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), HDA_COMPOSE_AMP_VAL(0x1e, 3, 0, HDA_OUTPUT), }; -static hda_nid_t stac9200_pin_nids[8] = { +static const hda_nid_t stac9200_pin_nids[8] = { 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, }; -static hda_nid_t stac925x_pin_nids[8] = { +static const hda_nid_t stac925x_pin_nids[8] = { 0x07, 0x08, 0x0a, 0x0b, 0x0c, 0x0d, 0x10, 0x11, }; -static hda_nid_t stac922x_pin_nids[10] = { +static const hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, }; -static hda_nid_t stac92hd73xx_pin_nids[13] = { +static const hda_nid_t stac92hd73xx_pin_nids[13] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, 0x14, 0x22, 0x23 }; #define STAC92HD71BXX_NUM_PINS 13 -static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { +static const hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x00, 0x00, 0x14, 0x18, 0x19, 0x1e, 0x1f, 0x20, 0x27 }; -static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { +static const hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, 0x1f, 0x20, 0x27 }; -static hda_nid_t stac927x_pin_nids[14] = { +static const hda_nid_t stac927x_pin_nids[14] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, 0x14, 0x21, 0x22, 0x23, }; -static hda_nid_t stac9205_pin_nids[12] = { +static const hda_nid_t stac9205_pin_nids[12] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x16, 0x17, 0x18, 0x21, 0x22, @@ -841,45 +845,45 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, return 1; } -static struct hda_verb stac9200_core_init[] = { +static const struct hda_verb stac9200_core_init[] = { /* set dac0mux for dac converter */ { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {} }; -static struct hda_verb stac9200_eapd_init[] = { +static const struct hda_verb stac9200_eapd_init[] = { /* set dac0mux for dac converter */ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {} }; -static struct hda_verb dell_eq_core_init[] = { +static const struct hda_verb dell_eq_core_init[] = { /* set master volume to max value without distortion * and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, {} }; -static struct hda_verb stac92hd73xx_core_init[] = { +static const struct hda_verb stac92hd73xx_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; -static struct hda_verb stac92hd83xxx_core_init[] = { +static const struct hda_verb stac92hd83xxx_core_init[] = { /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, {} }; -static struct hda_verb stac92hd71bxx_core_init[] = { +static const struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; -static struct hda_verb stac92hd71bxx_unmute_core_init[] = { +static const struct hda_verb stac92hd71bxx_unmute_core_init[] = { /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -887,7 +891,7 @@ static struct hda_verb stac92hd71bxx_unmute_core_init[] = { {} }; -static struct hda_verb stac925x_core_init[] = { +static const struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, /* mute the master volume */ @@ -895,13 +899,13 @@ static struct hda_verb stac925x_core_init[] = { {} }; -static struct hda_verb stac922x_core_init[] = { +static const struct hda_verb stac922x_core_init[] = { /* set master volume and direct control */ { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; -static struct hda_verb d965_core_init[] = { +static const struct hda_verb d965_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* unmute node 0x1b */ @@ -911,7 +915,7 @@ static struct hda_verb d965_core_init[] = { {} }; -static struct hda_verb dell_3st_core_init[] = { +static const struct hda_verb dell_3st_core_init[] = { /* don't set delta bit */ {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, /* unmute node 0x1b */ @@ -921,7 +925,7 @@ static struct hda_verb dell_3st_core_init[] = { {} }; -static struct hda_verb stac927x_core_init[] = { +static const struct hda_verb stac927x_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* enable analog pc beep path */ @@ -929,7 +933,7 @@ static struct hda_verb stac927x_core_init[] = { {} }; -static struct hda_verb stac927x_volknob_core_init[] = { +static const struct hda_verb stac927x_volknob_core_init[] = { /* don't set delta bit */ {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, /* enable analog pc beep path */ @@ -937,7 +941,7 @@ static struct hda_verb stac927x_volknob_core_init[] = { {} }; -static struct hda_verb stac9205_core_init[] = { +static const struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* enable analog pc beep path */ @@ -977,7 +981,7 @@ static struct hda_verb stac9205_core_init[] = { .private_value = nid, \ } -static struct snd_kcontrol_new stac9200_mixer[] = { +static const struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), @@ -985,38 +989,38 @@ static struct snd_kcontrol_new stac9200_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { +static const struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), {} }; -static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { +static const struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), {} }; -static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { +static const struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), {} }; -static struct snd_kcontrol_new stac92hd71bxx_loopback[] = { +static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2) }; -static struct snd_kcontrol_new stac925x_mixer[] = { +static const struct snd_kcontrol_new stac925x_mixer[] = { HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), { } /* end */ }; -static struct snd_kcontrol_new stac9205_loopback[] = { +static const struct snd_kcontrol_new stac9205_loopback[] = { STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), {} }; -static struct snd_kcontrol_new stac927x_loopback[] = { +static const struct snd_kcontrol_new stac927x_loopback[] = { STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), {} }; @@ -1182,16 +1186,16 @@ static int stac92xx_build_controls(struct hda_codec *codec) return 0; } -static unsigned int ref9200_pin_configs[8] = { +static const unsigned int ref9200_pin_configs[8] = { 0x01c47010, 0x01447010, 0x0221401f, 0x01114010, 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; -static unsigned int gateway9200_m4_pin_configs[8] = { +static const unsigned int gateway9200_m4_pin_configs[8] = { 0x400000fe, 0x404500f4, 0x400100f0, 0x90110010, 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, }; -static unsigned int gateway9200_m4_2_pin_configs[8] = { +static const unsigned int gateway9200_m4_2_pin_configs[8] = { 0x400000fe, 0x404500f4, 0x400100f0, 0x90110010, 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, }; @@ -1202,7 +1206,7 @@ static unsigned int gateway9200_m4_2_pin_configs[8] = { 102801DE 102801E8 */ -static unsigned int dell9200_d21_pin_configs[8] = { +static const unsigned int dell9200_d21_pin_configs[8] = { 0x400001f0, 0x400001f1, 0x02214030, 0x01014010, 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; @@ -1212,7 +1216,7 @@ static unsigned int dell9200_d21_pin_configs[8] = { 102801C0 102801C1 */ -static unsigned int dell9200_d22_pin_configs[8] = { +static const unsigned int dell9200_d22_pin_configs[8] = { 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, 0x01813020, 0x02a19021, 0x90100140, 0x400001f2, }; @@ -1226,7 +1230,7 @@ static unsigned int dell9200_d22_pin_configs[8] = { 102801DA 102801E3 */ -static unsigned int dell9200_d23_pin_configs[8] = { +static const unsigned int dell9200_d23_pin_configs[8] = { 0x400001f0, 0x400001f1, 0x0221401f, 0x01014010, 0x01813020, 0x01a19021, 0x90100140, 0x400001f2, }; @@ -1237,7 +1241,7 @@ static unsigned int dell9200_d23_pin_configs[8] = { 102801B5 (Dell Inspiron 630m) 102801D8 (Dell Inspiron 640m) */ -static unsigned int dell9200_m21_pin_configs[8] = { +static const unsigned int dell9200_m21_pin_configs[8] = { 0x40c003fa, 0x03441340, 0x0321121f, 0x90170310, 0x408003fb, 0x03a11020, 0x401003fc, 0x403003fd, }; @@ -1250,7 +1254,7 @@ static unsigned int dell9200_m21_pin_configs[8] = { 102801D4 102801D6 */ -static unsigned int dell9200_m22_pin_configs[8] = { +static const unsigned int dell9200_m22_pin_configs[8] = { 0x40c003fa, 0x0144131f, 0x0321121f, 0x90170310, 0x90a70321, 0x03a11020, 0x401003fb, 0x40f000fc, }; @@ -1260,7 +1264,7 @@ static unsigned int dell9200_m22_pin_configs[8] = { 102801CE (Dell XPS M1710) 102801CF (Dell Precision M90) */ -static unsigned int dell9200_m23_pin_configs[8] = { +static const unsigned int dell9200_m23_pin_configs[8] = { 0x40c003fa, 0x01441340, 0x0421421f, 0x90170310, 0x408003fb, 0x04a1102e, 0x90170311, 0x403003fc, }; @@ -1272,7 +1276,7 @@ static unsigned int dell9200_m23_pin_configs[8] = { 102801CB (Dell Latitude 120L) 102801D3 */ -static unsigned int dell9200_m24_pin_configs[8] = { +static const unsigned int dell9200_m24_pin_configs[8] = { 0x40c003fa, 0x404003fb, 0x0321121f, 0x90170310, 0x408003fc, 0x03a11020, 0x401003fd, 0x403003fe, }; @@ -1283,7 +1287,7 @@ static unsigned int dell9200_m24_pin_configs[8] = { 102801EE 102801EF */ -static unsigned int dell9200_m25_pin_configs[8] = { +static const unsigned int dell9200_m25_pin_configs[8] = { 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310, 0x408003fb, 0x04a11020, 0x401003fc, 0x403003fd, }; @@ -1293,7 +1297,7 @@ static unsigned int dell9200_m25_pin_configs[8] = { 102801F5 (Dell Inspiron 1501) 102801F6 */ -static unsigned int dell9200_m26_pin_configs[8] = { +static const unsigned int dell9200_m26_pin_configs[8] = { 0x40c003fa, 0x404003fb, 0x0421121f, 0x90170310, 0x408003fc, 0x04a11020, 0x401003fd, 0x403003fe, }; @@ -1302,18 +1306,18 @@ static unsigned int dell9200_m26_pin_configs[8] = { STAC 9200-32 102801CD (Dell Inspiron E1705/9400) */ -static unsigned int dell9200_m27_pin_configs[8] = { +static const unsigned int dell9200_m27_pin_configs[8] = { 0x40c003fa, 0x01441340, 0x0421121f, 0x90170310, 0x90170310, 0x04a11020, 0x90170310, 0x40f003fc, }; -static unsigned int oqo9200_pin_configs[8] = { +static const unsigned int oqo9200_pin_configs[8] = { 0x40c000f0, 0x404000f1, 0x0221121f, 0x02211210, 0x90170111, 0x90a70120, 0x400000f2, 0x400000f3, }; -static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { +static const unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_REF] = ref9200_pin_configs, [STAC_9200_OQO] = oqo9200_pin_configs, [STAC_9200_DELL_D21] = dell9200_d21_pin_configs, @@ -1350,7 +1354,7 @@ static const char * const stac9200_models[STAC_9200_MODELS] = { [STAC_9200_PANASONIC] = "panasonic", }; -static struct snd_pci_quirk stac9200_cfg_tbl[] = { +static const struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), @@ -1426,47 +1430,47 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref925x_pin_configs[8] = { +static const unsigned int ref925x_pin_configs[8] = { 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, 0x90a70320, 0x02214210, 0x01019020, 0x9033032e, }; -static unsigned int stac925xM1_pin_configs[8] = { +static const unsigned int stac925xM1_pin_configs[8] = { 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; -static unsigned int stac925xM1_2_pin_configs[8] = { +static const unsigned int stac925xM1_2_pin_configs[8] = { 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; -static unsigned int stac925xM2_pin_configs[8] = { +static const unsigned int stac925xM2_pin_configs[8] = { 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; -static unsigned int stac925xM2_2_pin_configs[8] = { +static const unsigned int stac925xM2_2_pin_configs[8] = { 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; -static unsigned int stac925xM3_pin_configs[8] = { +static const unsigned int stac925xM3_pin_configs[8] = { 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, 0x40a000f0, 0x90100210, 0x400003f1, 0x503303f3, }; -static unsigned int stac925xM5_pin_configs[8] = { +static const unsigned int stac925xM5_pin_configs[8] = { 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; -static unsigned int stac925xM6_pin_configs[8] = { +static const unsigned int stac925xM6_pin_configs[8] = { 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, 0x40a000f0, 0x90100210, 0x400003f1, 0x90330320, }; -static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { +static const unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { [STAC_REF] = ref925x_pin_configs, [STAC_M1] = stac925xM1_pin_configs, [STAC_M1_2] = stac925xM1_2_pin_configs, @@ -1489,7 +1493,7 @@ static const char * const stac925x_models[STAC_925x_MODELS] = { [STAC_M6] = "m6", }; -static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { +static const struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { SND_PCI_QUIRK(0x107b, 0x0316, "Gateway M255", STAC_M2), SND_PCI_QUIRK(0x107b, 0x0366, "Gateway MP6954", STAC_M5), SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_M1), @@ -1503,7 +1507,7 @@ static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { {} /* terminator */ }; -static struct snd_pci_quirk stac925x_cfg_tbl[] = { +static const struct snd_pci_quirk stac925x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), @@ -1515,33 +1519,33 @@ static struct snd_pci_quirk stac925x_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd73xx_pin_configs[13] = { +static const unsigned int ref92hd73xx_pin_configs[13] = { 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, 0x0181302e, 0x01014010, 0x01014020, 0x01014030, 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, 0x01452050, }; -static unsigned int dell_m6_pin_configs[13] = { +static const unsigned int dell_m6_pin_configs[13] = { 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, 0x03a11020, 0x0321101f, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, }; -static unsigned int alienware_m17x_pin_configs[13] = { +static const unsigned int alienware_m17x_pin_configs[13] = { 0x0321101f, 0x0321101f, 0x03a11020, 0x03014020, 0x90170110, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, 0x904601b0, }; -static unsigned int intel_dg45id_pin_configs[13] = { +static const unsigned int intel_dg45id_pin_configs[13] = { 0x02214230, 0x02A19240, 0x01013214, 0x01014210, 0x01A19250, 0x01011212, 0x01016211 }; -static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { +static const unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, [STAC_DELL_M6_AMIC] = dell_m6_pin_configs, [STAC_DELL_M6_DMIC] = dell_m6_pin_configs, @@ -1563,7 +1567,7 @@ static const char * const stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_ALIENWARE_M17X] = "alienware", }; -static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { +static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD73XX_REF), @@ -1604,7 +1608,7 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; -static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { +static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1, "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a, @@ -1612,25 +1616,25 @@ static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd83xxx_pin_configs[10] = { +static const unsigned int ref92hd83xxx_pin_configs[10] = { 0x02214030, 0x02211010, 0x02a19020, 0x02170130, 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, 0x01451160, 0x98560170, }; -static unsigned int dell_s14_pin_configs[10] = { +static const unsigned int dell_s14_pin_configs[10] = { 0x0221403f, 0x0221101f, 0x02a19020, 0x90170110, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a60160, 0x40f000f0, 0x40f000f0, }; -static unsigned int hp_dv7_4000_pin_configs[10] = { +static const unsigned int hp_dv7_4000_pin_configs[10] = { 0x03a12050, 0x0321201f, 0x40f000f0, 0x90170110, 0x40f000f0, 0x40f000f0, 0x90170110, 0xd5a30140, 0x40f000f0, 0x40f000f0, }; -static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { +static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, [STAC_DELL_S14] = dell_s14_pin_configs, @@ -1646,7 +1650,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_HP_DV7_4000] = "hp-dv7-4000", }; -static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { +static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD83XXX_REF), @@ -1659,35 +1663,35 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = { +static const unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, 0x01452050, 0x00000000, 0x00000000 }; -static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = { +static const unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000, 0x00000000 }; -static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = { +static const unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, 0x00000000 }; -static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = { +static const unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, 0x00000000 }; -static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { +static const unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, [STAC_DELL_M4_1] = dell_m4_1_pin_configs, [STAC_DELL_M4_2] = dell_m4_2_pin_configs, @@ -1712,7 +1716,7 @@ static const char * const stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr", }; -static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { +static const struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), @@ -1769,7 +1773,7 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref922x_pin_configs[10] = { +static const unsigned int ref922x_pin_configs[10] = { 0x01014010, 0x01016011, 0x01012012, 0x0221401f, 0x01813122, 0x01011014, 0x01441030, 0x01c41030, 0x40000100, 0x40000100, @@ -1783,7 +1787,7 @@ static unsigned int ref922x_pin_configs[10] = { 102801D1 102801D2 */ -static unsigned int dell_922x_d81_pin_configs[10] = { +static const unsigned int dell_922x_d81_pin_configs[10] = { 0x02214030, 0x01a19021, 0x01111012, 0x01114010, 0x02a19020, 0x01117011, 0x400001f0, 0x400001f1, 0x01813122, 0x400001f2, @@ -1794,7 +1798,7 @@ static unsigned int dell_922x_d81_pin_configs[10] = { 102801AC 102801D0 */ -static unsigned int dell_922x_d82_pin_configs[10] = { +static const unsigned int dell_922x_d82_pin_configs[10] = { 0x02214030, 0x01a19021, 0x01111012, 0x01114010, 0x02a19020, 0x01117011, 0x01451140, 0x400001f0, 0x01813122, 0x400001f1, @@ -1804,7 +1808,7 @@ static unsigned int dell_922x_d82_pin_configs[10] = { STAC 922X pin configs for 102801BF */ -static unsigned int dell_922x_m81_pin_configs[10] = { +static const unsigned int dell_922x_m81_pin_configs[10] = { 0x0321101f, 0x01112024, 0x01111222, 0x91174220, 0x03a11050, 0x01116221, 0x90a70330, 0x01452340, 0x40C003f1, 0x405003f0, @@ -1814,61 +1818,61 @@ static unsigned int dell_922x_m81_pin_configs[10] = { STAC 9221 A1 pin configs for 102801D7 (Dell XPS M1210) */ -static unsigned int dell_922x_m82_pin_configs[10] = { +static const unsigned int dell_922x_m82_pin_configs[10] = { 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, 0x508003f3, 0x405003f4, }; -static unsigned int d945gtp3_pin_configs[10] = { +static const unsigned int d945gtp3_pin_configs[10] = { 0x0221401f, 0x01a19022, 0x01813021, 0x01014010, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x02a19120, 0x40000100, }; -static unsigned int d945gtp5_pin_configs[10] = { +static const unsigned int d945gtp5_pin_configs[10] = { 0x0221401f, 0x01011012, 0x01813024, 0x01014010, 0x01a19021, 0x01016011, 0x01452130, 0x40000100, 0x02a19320, 0x40000100, }; -static unsigned int intel_mac_v1_pin_configs[10] = { +static const unsigned int intel_mac_v1_pin_configs[10] = { 0x0121e21f, 0x400000ff, 0x9017e110, 0x400000fd, 0x400000fe, 0x0181e020, 0x1145e030, 0x11c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int intel_mac_v2_pin_configs[10] = { +static const unsigned int intel_mac_v2_pin_configs[10] = { 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, 0x400000fe, 0x0181e020, 0x1145e230, 0x500000fa, 0x400000fc, 0x400000fb, }; -static unsigned int intel_mac_v3_pin_configs[10] = { +static const unsigned int intel_mac_v3_pin_configs[10] = { 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, 0x400000fe, 0x0181e020, 0x1145e230, 0x11c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int intel_mac_v4_pin_configs[10] = { +static const unsigned int intel_mac_v4_pin_configs[10] = { 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int intel_mac_v5_pin_configs[10] = { +static const unsigned int intel_mac_v5_pin_configs[10] = { 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int ecs202_pin_configs[10] = { +static const unsigned int ecs202_pin_configs[10] = { 0x0221401f, 0x02a19020, 0x01a19020, 0x01114010, 0x408000f0, 0x01813022, 0x074510a0, 0x40c400f1, 0x9037012e, 0x40e000f2, }; -static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { +static const unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, [STAC_D945GTP5] = d945gtp5_pin_configs, @@ -1917,7 +1921,7 @@ static const char * const stac922x_models[STAC_922X_MODELS] = { [STAC_922X_DELL_M82] = "dell-m82", }; -static struct snd_pci_quirk stac922x_cfg_tbl[] = { +static const struct snd_pci_quirk stac922x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D945_REF), @@ -2008,42 +2012,42 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref927x_pin_configs[14] = { +static const unsigned int ref927x_pin_configs[14] = { 0x02214020, 0x02a19080, 0x0181304e, 0x01014010, 0x01a19040, 0x01011012, 0x01016011, 0x0101201f, 0x183301f0, 0x18a001f0, 0x18a001f0, 0x01442070, 0x01c42190, 0x40000100, }; -static unsigned int d965_3st_pin_configs[14] = { +static const unsigned int d965_3st_pin_configs[14] = { 0x0221401f, 0x02a19120, 0x40000100, 0x01014011, 0x01a19021, 0x01813024, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x40000100 }; -static unsigned int d965_5st_pin_configs[14] = { +static const unsigned int d965_5st_pin_configs[14] = { 0x02214020, 0x02a19080, 0x0181304e, 0x01014010, 0x01a19040, 0x01011012, 0x01016011, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x01442070, 0x40000100, 0x40000100 }; -static unsigned int d965_5st_no_fp_pin_configs[14] = { +static const unsigned int d965_5st_no_fp_pin_configs[14] = { 0x40000100, 0x40000100, 0x0181304e, 0x01014010, 0x01a19040, 0x01011012, 0x01016011, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x01442070, 0x40000100, 0x40000100 }; -static unsigned int dell_3st_pin_configs[14] = { +static const unsigned int dell_3st_pin_configs[14] = { 0x02211230, 0x02a11220, 0x01a19040, 0x01114210, 0x01111212, 0x01116211, 0x01813050, 0x01112214, 0x403003fa, 0x90a60040, 0x90a60040, 0x404003fb, 0x40c003fc, 0x40000100 }; -static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { +static const unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { [STAC_D965_REF_NO_JD] = ref927x_pin_configs, [STAC_D965_REF] = ref927x_pin_configs, [STAC_D965_3ST] = d965_3st_pin_configs, @@ -2066,7 +2070,7 @@ static const char * const stac927x_models[STAC_927X_MODELS] = { [STAC_927X_VOLKNOB] = "volknob", }; -static struct snd_pci_quirk stac927x_cfg_tbl[] = { +static const struct snd_pci_quirk stac927x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D965_REF), @@ -2104,7 +2108,7 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref9205_pin_configs[12] = { +static const unsigned int ref9205_pin_configs[12] = { 0x40000100, 0x40000100, 0x01016011, 0x01014010, 0x01813122, 0x01a19021, 0x01019020, 0x40000100, 0x90a000f0, 0x90a000f0, 0x01441030, 0x01c41030 @@ -2121,7 +2125,7 @@ static unsigned int ref9205_pin_configs[12] = { 10280228 (Dell Vostro 1500) 10280229 (Dell Vostro 1700) */ -static unsigned int dell_9205_m42_pin_configs[12] = { +static const unsigned int dell_9205_m42_pin_configs[12] = { 0x0321101F, 0x03A11020, 0x400003FA, 0x90170310, 0x400003FB, 0x400003FC, 0x400003FD, 0x40F000F9, 0x90A60330, 0x400003FF, 0x0144131F, 0x40C003FE, @@ -2137,19 +2141,19 @@ static unsigned int dell_9205_m42_pin_configs[12] = { 10280200 10280201 */ -static unsigned int dell_9205_m43_pin_configs[12] = { +static const unsigned int dell_9205_m43_pin_configs[12] = { 0x0321101f, 0x03a11020, 0x90a70330, 0x90170310, 0x400000fe, 0x400000ff, 0x400000fd, 0x40f000f9, 0x400000fa, 0x400000fc, 0x0144131f, 0x40c003f8, }; -static unsigned int dell_9205_m44_pin_configs[12] = { +static const unsigned int dell_9205_m44_pin_configs[12] = { 0x0421101f, 0x04a11020, 0x400003fa, 0x90170310, 0x400003fb, 0x400003fc, 0x400003fd, 0x400003f9, 0x90a60330, 0x400003ff, 0x01441340, 0x40c003fe, }; -static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { +static const unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { [STAC_9205_REF] = ref9205_pin_configs, [STAC_9205_DELL_M42] = dell_9205_m42_pin_configs, [STAC_9205_DELL_M43] = dell_9205_m43_pin_configs, @@ -2166,7 +2170,7 @@ static const char * const stac9205_models[STAC_9205_MODELS] = { [STAC_9205_EAPD] = "eapd", }; -static struct snd_pci_quirk stac9205_cfg_tbl[] = { +static const struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), @@ -2214,7 +2218,7 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { }; static void stac92xx_set_config_regs(struct hda_codec *codec, - unsigned int *pincfgs) + const unsigned int *pincfgs) { int i; struct sigmatel_spec *spec = codec->spec; @@ -2334,7 +2338,7 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream stac92xx_pcm_digital_playback = { +static const struct hda_pcm_stream stac92xx_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -2347,14 +2351,14 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = { }, }; -static struct hda_pcm_stream stac92xx_pcm_digital_capture = { +static const struct hda_pcm_stream stac92xx_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in stac92xx_build_pcms */ }; -static struct hda_pcm_stream stac92xx_pcm_analog_playback = { +static const struct hda_pcm_stream stac92xx_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 8, @@ -2366,7 +2370,7 @@ static struct hda_pcm_stream stac92xx_pcm_analog_playback = { }, }; -static struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = { +static const struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -2378,7 +2382,7 @@ static struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = { }, }; -static struct hda_pcm_stream stac92xx_pcm_analog_capture = { +static const struct hda_pcm_stream stac92xx_pcm_analog_capture = { .channels_min = 2, .channels_max = 2, /* NID + .substreams is set in stac92xx_build_pcms */ @@ -2487,7 +2491,7 @@ static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int i; - static char *texts[] = { + static const char * const texts[] = { "Mic In", "Line In", "Line Out" }; @@ -2556,7 +2560,7 @@ static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[2]; + char *texts[2]; struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -2687,7 +2691,7 @@ enum { STAC_CTL_WIDGET_DC_BIAS }; -static struct snd_kcontrol_new stac92xx_control_templates[] = { +static const struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0), @@ -2701,7 +2705,7 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { /* add dynamic controls */ static struct snd_kcontrol_new * stac_control_new(struct sigmatel_spec *spec, - struct snd_kcontrol_new *ktemp, + const struct snd_kcontrol_new *ktemp, const char *name, unsigned int subdev) { @@ -2724,7 +2728,7 @@ stac_control_new(struct sigmatel_spec *spec, } static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - struct snd_kcontrol_new *ktemp, + const struct snd_kcontrol_new *ktemp, int idx, const char *name, unsigned long val) { @@ -2754,7 +2758,7 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, return stac92xx_add_control_idx(spec, type, 0, name, val); } -static struct snd_kcontrol_new stac_input_src_temp = { +static const struct snd_kcontrol_new stac_input_src_temp = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", .info = stac92xx_mux_enum_info, @@ -3310,7 +3314,7 @@ static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol, return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]); } -static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { +static const struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = stac92xx_dig_beep_switch_info, .get = stac92xx_dig_beep_switch_get, @@ -4172,7 +4176,7 @@ static void stac92xx_power_down(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; /* power down inactive DACs */ - hda_nid_t *dac; + const hda_nid_t *dac; for (dac = spec->dac_list; *dac; dac++) if (!check_all_dac_nids(spec, *dac)) snd_hda_codec_write(codec, *dac, 0, @@ -4645,7 +4649,7 @@ static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) } static int stac92xx_connected_ports(struct hda_codec *codec, - hda_nid_t *nids, int num_nids) + const hda_nid_t *nids, int num_nids) { struct sigmatel_spec *spec = codec->spec; int idx, num; @@ -4969,7 +4973,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) } #endif -static struct hda_codec_ops stac92xx_patch_ops = { +static const struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, .build_pcms = stac92xx_build_pcms, .init = stac92xx_init, @@ -5589,7 +5593,7 @@ static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { +static const struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = stac_hp_bass_gpio_info, .get = stac_hp_bass_gpio_get, @@ -5613,7 +5617,7 @@ static int stac_add_hp_bass_switch(struct hda_codec *codec) static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; - struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; + const struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; unsigned int pin_cfg; int err = 0; @@ -5706,9 +5710,9 @@ again: unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); - stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; + spec->dmic_nids = stac92hd71bxx_dmic_5port_nids; spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd71bxx_dmic_nids, + stac92hd71bxx_dmic_5port_nids, STAC92HD71BXX_NUM_DMICS - 1); break; case 0x111d7603: /* 6 Port with Analog Mixer */ @@ -6215,31 +6219,31 @@ static int patch_stac9205(struct hda_codec *codec) * STAC9872 hack */ -static struct hda_verb stac9872_core_init[] = { +static const struct hda_verb stac9872_core_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; -static hda_nid_t stac9872_pin_nids[] = { +static const hda_nid_t stac9872_pin_nids[] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x11, 0x13, 0x14, }; -static hda_nid_t stac9872_adc_nids[] = { +static const hda_nid_t stac9872_adc_nids[] = { 0x8 /*,0x6*/ }; -static hda_nid_t stac9872_mux_nids[] = { +static const hda_nid_t stac9872_mux_nids[] = { 0x15 }; -static unsigned long stac9872_capvols[] = { +static const unsigned long stac9872_capvols[] = { HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), }; #define stac9872_capsws stac9872_capvols -static unsigned int stac9872_vaio_pin_configs[9] = { +static const unsigned int stac9872_vaio_pin_configs[9] = { 0x03211020, 0x411111f0, 0x411111f0, 0x03a15030, 0x411111f0, 0x90170110, 0x411111f0, 0x411111f0, 0x90a7013e @@ -6250,11 +6254,11 @@ static const char * const stac9872_models[STAC_9872_MODELS] = { [STAC_9872_VAIO] = "vaio", }; -static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = { +static const unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = { [STAC_9872_VAIO] = stac9872_vaio_pin_configs, }; -static struct snd_pci_quirk stac9872_cfg_tbl[] = { +static const struct snd_pci_quirk stac9872_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x104d, 0xfff0, 0x81e0, "Sony VAIO F/S", STAC_9872_VAIO), {} /* terminator */ @@ -6308,7 +6312,7 @@ static int patch_stac9872(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_sigmatel[] = { +static const struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 }, { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x }, { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x }, -- cgit v1.2.3 From 90dd48a1a941d0efd413ce011895e3b6edd49b9a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 May 2011 12:38:19 +0200 Subject: ALSA: hda - Constify fixup and other array data in patch_via.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 223 +++++++++++++++++++++++----------------------- 1 file changed, 112 insertions(+), 111 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 0bfbacb4e34..605c99e1e52 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -109,19 +109,19 @@ enum VIA_HDA_CODEC { struct via_spec { /* codec parameterization */ - struct snd_kcontrol_new *mixers[6]; + const struct snd_kcontrol_new *mixers[6]; unsigned int num_mixers; - struct hda_verb *init_verbs[5]; + const struct hda_verb *init_verbs[5]; unsigned int num_iverbs; char *stream_name_analog; - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; + const struct hda_pcm_stream *stream_analog_playback; + const struct hda_pcm_stream *stream_analog_capture; char *stream_name_digital; - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; + const struct hda_pcm_stream *stream_digital_playback; + const struct hda_pcm_stream *stream_digital_capture; /* playback */ struct hda_multi_out multiout; @@ -129,7 +129,7 @@ struct via_spec { /* capture */ unsigned int num_adc_nids; - hda_nid_t *adc_nids; + const hda_nid_t *adc_nids; hda_nid_t mux_nids[3]; hda_nid_t dig_in_nid; hda_nid_t dig_in_pin; @@ -410,54 +410,54 @@ static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, .put = bind_pin_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } -static struct snd_kcontrol_new via_control_templates[] = { +static const struct snd_kcontrol_new via_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), ANALOG_INPUT_MUTE, BIND_PIN_MUTE, }; -static hda_nid_t vt1708_adc_nids[2] = { +static const hda_nid_t vt1708_adc_nids[2] = { /* ADC1-2 */ 0x15, 0x27 }; -static hda_nid_t vt1709_adc_nids[3] = { +static const hda_nid_t vt1709_adc_nids[3] = { /* ADC1-2 */ 0x14, 0x15, 0x16 }; -static hda_nid_t vt1708B_adc_nids[2] = { +static const hda_nid_t vt1708B_adc_nids[2] = { /* ADC1-2 */ 0x13, 0x14 }; -static hda_nid_t vt1708S_adc_nids[2] = { +static const hda_nid_t vt1708S_adc_nids[2] = { /* ADC1-2 */ 0x13, 0x14 }; -static hda_nid_t vt1702_adc_nids[3] = { +static const hda_nid_t vt1702_adc_nids[3] = { /* ADC1-2 */ 0x12, 0x20, 0x1F }; -static hda_nid_t vt1718S_adc_nids[2] = { +static const hda_nid_t vt1718S_adc_nids[2] = { /* ADC1-2 */ 0x10, 0x11 }; -static hda_nid_t vt1716S_adc_nids[2] = { +static const hda_nid_t vt1716S_adc_nids[2] = { /* ADC1-2 */ 0x13, 0x14 }; -static hda_nid_t vt2002P_adc_nids[2] = { +static const hda_nid_t vt2002P_adc_nids[2] = { /* ADC1-2 */ 0x10, 0x11 }; -static hda_nid_t vt1812_adc_nids[2] = { +static const hda_nid_t vt1812_adc_nids[2] = { /* ADC1-2 */ 0x10, 0x11 }; @@ -487,7 +487,7 @@ static int __via_add_control(struct via_spec *spec, int type, const char *name, __via_add_control(spec, type, name, 0, val) static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, - struct snd_kcontrol_new *tmpl) + const struct snd_kcontrol_new *tmpl) { struct snd_kcontrol_new *knew; @@ -780,7 +780,7 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new via_hp_mixer[2] = { +static const struct snd_kcontrol_new via_hp_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Independent HP", @@ -1003,7 +1003,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new via_smart51_mixer[2] = { +static const struct snd_kcontrol_new via_smart51_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Smart 5.1", @@ -1049,7 +1049,7 @@ static int via_smart51_build(struct via_spec *spec) } /* capture mixer elements */ -static struct snd_kcontrol_new vt1708_capture_mixer[] = { +static const struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x27, 0x0, HDA_INPUT), @@ -1179,7 +1179,7 @@ static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb vt1708_volume_init_verbs[] = { +static const struct hda_verb vt1708_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -1421,7 +1421,7 @@ static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream vt1708_pcm_analog_playback = { +static const struct hda_pcm_stream vt1708_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 8, @@ -1433,7 +1433,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { +static const struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { .substreams = 2, .channels_min = 2, .channels_max = 8, @@ -1450,7 +1450,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { }, }; -static struct hda_pcm_stream vt1708_pcm_analog_capture = { +static const struct hda_pcm_stream vt1708_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -1461,7 +1461,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1708_pcm_digital_playback = { +static const struct hda_pcm_stream vt1708_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -1474,7 +1474,7 @@ static struct hda_pcm_stream vt1708_pcm_digital_playback = { }, }; -static struct hda_pcm_stream vt1708_pcm_digital_capture = { +static const struct hda_pcm_stream vt1708_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -1484,7 +1484,7 @@ static int via_build_controls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; struct snd_kcontrol *kctl; - struct snd_kcontrol_new *knew; + const struct snd_kcontrol_new *knew; int err, i; for (i = 0; i < spec->num_mixers; i++) { @@ -1819,7 +1819,7 @@ static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) /* */ -static struct hda_codec_ops via_patch_ops = { +static const struct hda_codec_ops via_patch_ops = { .build_controls = via_build_controls, .build_pcms = via_build_pcms, .init = via_init, @@ -2002,7 +2002,8 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg, hda_nid_t cap_nid, - hda_nid_t pin_idxs[], int num_idxs) + const hda_nid_t pin_idxs[], + int num_idxs) { struct via_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; @@ -2048,13 +2049,13 @@ static int vt_auto_create_analog_input_ctls(struct hda_codec *codec, static int vt1708_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - static hda_nid_t pin_idxs[] = { 0xff, 0x24, 0x1d, 0x1e, 0x21 }; + static const hda_nid_t pin_idxs[] = { 0xff, 0x24, 0x1d, 0x1e, 0x21 }; return vt_auto_create_analog_input_ctls(codec, cfg, 0x17, pin_idxs, ARRAY_SIZE(pin_idxs)); } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1708_loopbacks[] = { +static const struct hda_amp_list vt1708_loopbacks[] = { { 0x17, HDA_INPUT, 1 }, { 0x17, HDA_INPUT, 2 }, { 0x17, HDA_INPUT, 3 }, @@ -2113,7 +2114,7 @@ static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new vt1708_jack_detectect[] = { +static const struct snd_kcontrol_new vt1708_jack_detectect[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Jack Detect", @@ -2293,7 +2294,7 @@ static int patch_vt1708(struct hda_codec *codec) } /* capture mixer elements */ -static struct snd_kcontrol_new vt1709_capture_mixer[] = { +static const struct snd_kcontrol_new vt1709_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x15, 0x0, HDA_INPUT), @@ -2315,7 +2316,7 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = { { } /* end */ }; -static struct hda_verb vt1709_uniwill_init_verbs[] = { +static const struct hda_verb vt1709_uniwill_init_verbs[] = { {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, { } @@ -2324,7 +2325,7 @@ static struct hda_verb vt1709_uniwill_init_verbs[] = { /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb vt1709_10ch_volume_init_verbs[] = { +static const struct hda_verb vt1709_10ch_volume_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -2364,7 +2365,7 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = { { } }; -static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = { +static const struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 10, @@ -2376,7 +2377,7 @@ static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = { +static const struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 6, @@ -2388,7 +2389,7 @@ static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1709_pcm_analog_capture = { +static const struct hda_pcm_stream vt1709_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -2399,7 +2400,7 @@ static struct hda_pcm_stream vt1709_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1709_pcm_digital_playback = { +static const struct hda_pcm_stream vt1709_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -2410,7 +2411,7 @@ static struct hda_pcm_stream vt1709_pcm_digital_playback = { }, }; -static struct hda_pcm_stream vt1709_pcm_digital_capture = { +static const struct hda_pcm_stream vt1709_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -2622,7 +2623,7 @@ static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1709_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - static hda_nid_t pin_idxs[] = { 0xff, 0x23, 0x1d, 0x1e, 0x21 }; + static const hda_nid_t pin_idxs[] = { 0xff, 0x23, 0x1d, 0x1e, 0x21 }; return vt_auto_create_analog_input_ctls(codec, cfg, 0x18, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -2672,7 +2673,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1709_loopbacks[] = { +static const struct hda_amp_list vt1709_loopbacks[] = { { 0x18, HDA_INPUT, 1 }, { 0x18, HDA_INPUT, 2 }, { 0x18, HDA_INPUT, 3 }, @@ -2733,7 +2734,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb vt1709_6ch_volume_init_verbs[] = { +static const struct hda_verb vt1709_6ch_volume_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ @@ -2823,7 +2824,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) } /* capture mixer elements */ -static struct snd_kcontrol_new vt1708B_capture_mixer[] = { +static const struct snd_kcontrol_new vt1708B_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), @@ -2845,7 +2846,7 @@ static struct snd_kcontrol_new vt1708B_capture_mixer[] = { /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb vt1708B_8ch_volume_init_verbs[] = { +static const struct hda_verb vt1708B_8ch_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -2880,7 +2881,7 @@ static struct hda_verb vt1708B_8ch_volume_init_verbs[] = { { } }; -static struct hda_verb vt1708B_4ch_volume_init_verbs[] = { +static const struct hda_verb vt1708B_4ch_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -2915,7 +2916,7 @@ static struct hda_verb vt1708B_4ch_volume_init_verbs[] = { { } }; -static struct hda_verb vt1708B_uniwill_init_verbs[] = { +static const struct hda_verb vt1708B_uniwill_init_verbs[] = { {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, @@ -2939,7 +2940,7 @@ static int via_pcm_open_close(struct hda_pcm_stream *hinfo, return 0; } -static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { +static const struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 8, @@ -2952,7 +2953,7 @@ static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = { +static const struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 4, @@ -2964,7 +2965,7 @@ static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1708B_pcm_analog_capture = { +static const struct hda_pcm_stream vt1708B_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -2977,7 +2978,7 @@ static struct hda_pcm_stream vt1708B_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1708B_pcm_digital_playback = { +static const struct hda_pcm_stream vt1708B_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -2990,7 +2991,7 @@ static struct hda_pcm_stream vt1708B_pcm_digital_playback = { }, }; -static struct hda_pcm_stream vt1708B_pcm_digital_capture = { +static const struct hda_pcm_stream vt1708B_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -3154,7 +3155,7 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1708B_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - static hda_nid_t pin_idxs[] = { 0xff, 0x1f, 0x1a, 0x1b, 0x1e }; + static const hda_nid_t pin_idxs[] = { 0xff, 0x1f, 0x1a, 0x1b, 0x1e }; return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -3204,7 +3205,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1708B_loopbacks[] = { +static const struct hda_amp_list vt1708B_loopbacks[] = { { 0x16, HDA_INPUT, 1 }, { 0x16, HDA_INPUT, 2 }, { 0x16, HDA_INPUT, 3 }, @@ -3403,7 +3404,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) /* Patch for VT1708S */ /* capture mixer elements */ -static struct snd_kcontrol_new vt1708S_capture_mixer[] = { +static const struct snd_kcontrol_new vt1708S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), @@ -3426,7 +3427,7 @@ static struct snd_kcontrol_new vt1708S_capture_mixer[] = { { } /* end */ }; -static struct hda_verb vt1708S_volume_init_verbs[] = { +static const struct hda_verb vt1708S_volume_init_verbs[] = { /* Unmute ADC0-1 and set the default input to mic-in */ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -3452,7 +3453,7 @@ static struct hda_verb vt1708S_volume_init_verbs[] = { { } }; -static struct hda_verb vt1708S_uniwill_init_verbs[] = { +static const struct hda_verb vt1708S_uniwill_init_verbs[] = { {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, @@ -3465,7 +3466,7 @@ static struct hda_verb vt1708S_uniwill_init_verbs[] = { { } }; -static struct hda_verb vt1705_uniwill_init_verbs[] = { +static const struct hda_verb vt1705_uniwill_init_verbs[] = { {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, @@ -3477,7 +3478,7 @@ static struct hda_verb vt1705_uniwill_init_verbs[] = { { } }; -static struct hda_pcm_stream vt1708S_pcm_analog_playback = { +static const struct hda_pcm_stream vt1708S_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 8, @@ -3490,7 +3491,7 @@ static struct hda_pcm_stream vt1708S_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1705_pcm_analog_playback = { +static const struct hda_pcm_stream vt1705_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 6, @@ -3503,7 +3504,7 @@ static struct hda_pcm_stream vt1705_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1708S_pcm_analog_capture = { +static const struct hda_pcm_stream vt1708S_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -3516,7 +3517,7 @@ static struct hda_pcm_stream vt1708S_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1708S_pcm_digital_playback = { +static const struct hda_pcm_stream vt1708S_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -3718,7 +3719,7 @@ static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1708S_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; + static const hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -3789,7 +3790,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1708S_loopbacks[] = { +static const struct hda_amp_list vt1708S_loopbacks[] = { { 0x16, HDA_INPUT, 1 }, { 0x16, HDA_INPUT, 2 }, { 0x16, HDA_INPUT, 3 }, @@ -3904,7 +3905,7 @@ static int patch_vt1708S(struct hda_codec *codec) /* Patch for VT1702 */ /* capture mixer elements */ -static struct snd_kcontrol_new vt1702_capture_mixer[] = { +static const struct snd_kcontrol_new vt1702_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x20, 0x0, HDA_INPUT), @@ -3928,7 +3929,7 @@ static struct snd_kcontrol_new vt1702_capture_mixer[] = { { } /* end */ }; -static struct hda_verb vt1702_volume_init_verbs[] = { +static const struct hda_verb vt1702_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -3959,7 +3960,7 @@ static struct hda_verb vt1702_volume_init_verbs[] = { { } }; -static struct hda_verb vt1702_uniwill_init_verbs[] = { +static const struct hda_verb vt1702_uniwill_init_verbs[] = { {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, @@ -3969,7 +3970,7 @@ static struct hda_verb vt1702_uniwill_init_verbs[] = { { } }; -static struct hda_pcm_stream vt1702_pcm_analog_playback = { +static const struct hda_pcm_stream vt1702_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -3982,7 +3983,7 @@ static struct hda_pcm_stream vt1702_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1702_pcm_analog_capture = { +static const struct hda_pcm_stream vt1702_pcm_analog_capture = { .substreams = 3, .channels_min = 2, .channels_max = 2, @@ -3995,7 +3996,7 @@ static struct hda_pcm_stream vt1702_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1702_pcm_digital_playback = { +static const struct hda_pcm_stream vt1702_pcm_digital_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -4095,7 +4096,7 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1702_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - static hda_nid_t pin_idxs[] = { 0x14, 0x15, 0x18, 0xff }; + static const hda_nid_t pin_idxs[] = { 0x14, 0x15, 0x18, 0xff }; return vt_auto_create_analog_input_ctls(codec, cfg, 0x1a, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -4146,7 +4147,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1702_loopbacks[] = { +static const struct hda_amp_list vt1702_loopbacks[] = { { 0x1A, HDA_INPUT, 1 }, { 0x1A, HDA_INPUT, 2 }, { 0x1A, HDA_INPUT, 3 }, @@ -4239,7 +4240,7 @@ static int patch_vt1702(struct hda_codec *codec) /* Patch for VT1718S */ /* capture mixer elements */ -static struct snd_kcontrol_new vt1718S_capture_mixer[] = { +static const struct snd_kcontrol_new vt1718S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), @@ -4261,7 +4262,7 @@ static struct snd_kcontrol_new vt1718S_capture_mixer[] = { { } /* end */ }; -static struct hda_verb vt1718S_volume_init_verbs[] = { +static const struct hda_verb vt1718S_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -4309,7 +4310,7 @@ static struct hda_verb vt1718S_volume_init_verbs[] = { }; -static struct hda_verb vt1718S_uniwill_init_verbs[] = { +static const struct hda_verb vt1718S_uniwill_init_verbs[] = { {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, @@ -4322,7 +4323,7 @@ static struct hda_verb vt1718S_uniwill_init_verbs[] = { { } }; -static struct hda_pcm_stream vt1718S_pcm_analog_playback = { +static const struct hda_pcm_stream vt1718S_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 10, @@ -4335,7 +4336,7 @@ static struct hda_pcm_stream vt1718S_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1718S_pcm_analog_capture = { +static const struct hda_pcm_stream vt1718S_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -4348,7 +4349,7 @@ static struct hda_pcm_stream vt1718S_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1718S_pcm_digital_playback = { +static const struct hda_pcm_stream vt1718S_pcm_digital_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -4361,7 +4362,7 @@ static struct hda_pcm_stream vt1718S_pcm_digital_playback = { }, }; -static struct hda_pcm_stream vt1718S_pcm_digital_capture = { +static const struct hda_pcm_stream vt1718S_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -4515,7 +4516,7 @@ static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1718S_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - static hda_nid_t pin_idxs[] = { 0x2c, 0x2b, 0x2a, 0x29, 0, 0xff }; + static const hda_nid_t pin_idxs[] = { 0x2c, 0x2b, 0x2a, 0x29, 0, 0xff }; return vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -4566,7 +4567,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1718S_loopbacks[] = { +static const struct hda_amp_list vt1718S_loopbacks[] = { { 0x21, HDA_INPUT, 1 }, { 0x21, HDA_INPUT, 2 }, { 0x21, HDA_INPUT, 3 }, @@ -4748,7 +4749,7 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, } /* capture mixer elements */ -static struct snd_kcontrol_new vt1716S_capture_mixer[] = { +static const struct snd_kcontrol_new vt1716S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), @@ -4767,7 +4768,7 @@ static struct snd_kcontrol_new vt1716S_capture_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { +static const struct snd_kcontrol_new vt1716s_dmic_mixer[] = { HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -4783,12 +4784,12 @@ static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { /* mono-out mixer elements */ -static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { +static const struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), { } /* end */ }; -static struct hda_verb vt1716S_volume_init_verbs[] = { +static const struct hda_verb vt1716S_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -4837,7 +4838,7 @@ static struct hda_verb vt1716S_volume_init_verbs[] = { }; -static struct hda_verb vt1716S_uniwill_init_verbs[] = { +static const struct hda_verb vt1716S_uniwill_init_verbs[] = { {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, @@ -4850,7 +4851,7 @@ static struct hda_verb vt1716S_uniwill_init_verbs[] = { { } }; -static struct hda_pcm_stream vt1716S_pcm_analog_playback = { +static const struct hda_pcm_stream vt1716S_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 6, @@ -4863,7 +4864,7 @@ static struct hda_pcm_stream vt1716S_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1716S_pcm_analog_capture = { +static const struct hda_pcm_stream vt1716S_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -4876,7 +4877,7 @@ static struct hda_pcm_stream vt1716S_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1716S_pcm_digital_playback = { +static const struct hda_pcm_stream vt1716S_pcm_digital_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -5046,7 +5047,7 @@ static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1716S_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; + static const hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -5093,7 +5094,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1716S_loopbacks[] = { +static const struct hda_amp_list vt1716S_loopbacks[] = { { 0x16, HDA_INPUT, 1 }, { 0x16, HDA_INPUT, 2 }, { 0x16, HDA_INPUT, 3 }, @@ -5256,7 +5257,7 @@ static int patch_vt1716S(struct hda_codec *codec) /* for vt2002P */ /* capture mixer elements */ -static struct snd_kcontrol_new vt2002P_capture_mixer[] = { +static const struct snd_kcontrol_new vt2002P_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), @@ -5279,7 +5280,7 @@ static struct snd_kcontrol_new vt2002P_capture_mixer[] = { { } /* end */ }; -static struct hda_verb vt2002P_volume_init_verbs[] = { +static const struct hda_verb vt2002P_volume_init_verbs[] = { /* Class-D speaker related verbs */ {0x1, 0xfe0, 0x4}, {0x1, 0xfe9, 0x80}, @@ -5334,7 +5335,7 @@ static struct hda_verb vt2002P_volume_init_verbs[] = { {0x1, 0xfb8, 0x88}, { } }; -static struct hda_verb vt1802_volume_init_verbs[] = { +static const struct hda_verb vt1802_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -5387,7 +5388,7 @@ static struct hda_verb vt1802_volume_init_verbs[] = { }; -static struct hda_verb vt2002P_uniwill_init_verbs[] = { +static const struct hda_verb vt2002P_uniwill_init_verbs[] = { {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, @@ -5397,7 +5398,7 @@ static struct hda_verb vt2002P_uniwill_init_verbs[] = { {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; -static struct hda_verb vt1802_uniwill_init_verbs[] = { +static const struct hda_verb vt1802_uniwill_init_verbs[] = { {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, @@ -5408,7 +5409,7 @@ static struct hda_verb vt1802_uniwill_init_verbs[] = { { } }; -static struct hda_pcm_stream vt2002P_pcm_analog_playback = { +static const struct hda_pcm_stream vt2002P_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -5421,7 +5422,7 @@ static struct hda_pcm_stream vt2002P_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt2002P_pcm_analog_capture = { +static const struct hda_pcm_stream vt2002P_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -5434,7 +5435,7 @@ static struct hda_pcm_stream vt2002P_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt2002P_pcm_digital_playback = { +static const struct hda_pcm_stream vt2002P_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -5521,7 +5522,7 @@ static int vt2002P_auto_create_analog_input_ctls(struct hda_codec *codec, { struct via_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; - static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0xff }; + static const hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0xff }; int err; err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs, @@ -5582,7 +5583,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt2002P_loopbacks[] = { +static const struct hda_amp_list vt2002P_loopbacks[] = { { 0x21, HDA_INPUT, 0 }, { 0x21, HDA_INPUT, 1 }, { 0x21, HDA_INPUT, 2 }, @@ -5775,7 +5776,7 @@ static int patch_vt2002P(struct hda_codec *codec) /* for vt1812 */ /* capture mixer elements */ -static struct snd_kcontrol_new vt1812_capture_mixer[] = { +static const struct snd_kcontrol_new vt1812_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), @@ -5797,7 +5798,7 @@ static struct snd_kcontrol_new vt1812_capture_mixer[] = { { } /* end */ }; -static struct hda_verb vt1812_volume_init_verbs[] = { +static const struct hda_verb vt1812_volume_init_verbs[] = { /* * Unmute ADC0-1 and set the default input to mic-in */ @@ -5850,7 +5851,7 @@ static struct hda_verb vt1812_volume_init_verbs[] = { }; -static struct hda_verb vt1812_uniwill_init_verbs[] = { +static const struct hda_verb vt1812_uniwill_init_verbs[] = { {0x33, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT }, @@ -5862,7 +5863,7 @@ static struct hda_verb vt1812_uniwill_init_verbs[] = { { } }; -static struct hda_pcm_stream vt1812_pcm_analog_playback = { +static const struct hda_pcm_stream vt1812_pcm_analog_playback = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -5875,7 +5876,7 @@ static struct hda_pcm_stream vt1812_pcm_analog_playback = { }, }; -static struct hda_pcm_stream vt1812_pcm_analog_capture = { +static const struct hda_pcm_stream vt1812_pcm_analog_capture = { .substreams = 2, .channels_min = 2, .channels_max = 2, @@ -5888,7 +5889,7 @@ static struct hda_pcm_stream vt1812_pcm_analog_capture = { }, }; -static struct hda_pcm_stream vt1812_pcm_digital_playback = { +static const struct hda_pcm_stream vt1812_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -5970,7 +5971,7 @@ static int vt1812_auto_create_analog_input_ctls(struct hda_codec *codec, { struct via_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; - static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0, 0, 0xff }; + static const hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0, 0, 0xff }; int err; err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs, @@ -6032,7 +6033,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE -static struct hda_amp_list vt1812_loopbacks[] = { +static const struct hda_amp_list vt1812_loopbacks[] = { { 0x21, HDA_INPUT, 0 }, { 0x21, HDA_INPUT, 1 }, { 0x21, HDA_INPUT, 2 }, @@ -6191,7 +6192,7 @@ static int patch_vt1812(struct hda_codec *codec) /* * patch entries */ -static struct hda_codec_preset snd_hda_preset_via[] = { +static const struct hda_codec_preset snd_hda_preset_via[] = { { .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708}, { .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708}, { .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708}, -- cgit v1.2.3 From ce85c9ac8dd53a658d85b4f39f612c23aa704e49 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Tue, 3 May 2011 13:33:53 +0800 Subject: ALSA: hda - fix NULL-dereference in patch_realtek Fix NULL-dereference when try to use alt_playback since those codecs which support multistreaming playback usually have more than 1 adc but the driver should create alt_capture when spec->stream_analog_alt_capture is also defined. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c9f77152e07..4dd0cccc5a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4424,7 +4424,7 @@ static int alc_build_pcms(struct hda_codec *codec) alc_pcm_null_stream; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; } - if (spec->num_adc_nids > 1) { + if (spec->num_adc_nids > 1 && spec->stream_analog_alt_capture) { info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->stream_analog_alt_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = -- cgit v1.2.3 From d43f3010b8fa7530c3780c087fad9b0a8a437ba1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:14:46 +0200 Subject: ALSA: Add the driver for Digigram Lola PCI-e boards Added a new driver for supporting Digigram Lola PCI-e boards. Lola has a similar h/w design like HD-audio but with extended verbs. Thus the driver is written similarly like HD-audio driver in the bus part. The codec part is rather written in a fixed way specific to the Lola board because of the verb incompatibility. The driver provides basic PCM, supporting multi-streams and mixing. Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 9 + sound/pci/Makefile | 1 + sound/pci/lola/Makefile | 3 + sound/pci/lola/lola.c | 731 +++++++++++++++++++++++++++++++++++++++ sound/pci/lola/lola.h | 520 ++++++++++++++++++++++++++++ sound/pci/lola/lola_clock.c | 322 +++++++++++++++++ sound/pci/lola/lola_mixer.c | 826 ++++++++++++++++++++++++++++++++++++++++++++ sound/pci/lola/lola_pcm.c | 607 ++++++++++++++++++++++++++++++++ sound/pci/lola/lola_proc.c | 86 +++++ 9 files changed, 3105 insertions(+) create mode 100644 sound/pci/lola/Makefile create mode 100644 sound/pci/lola/lola.c create mode 100644 sound/pci/lola/lola.h create mode 100644 sound/pci/lola/lola_clock.c create mode 100644 sound/pci/lola/lola_mixer.c create mode 100644 sound/pci/lola/lola_pcm.c create mode 100644 sound/pci/lola/lola_proc.c (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 9823d59d7ad..92ead69cc4b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -652,6 +652,15 @@ config SND_KORG1212 To compile this driver as a module, choose M here: the module will be called snd-korg1212. +config SND_LOLA + tristate "Digigram Lola" + select SND_PCM + help + Say Y to include support for Digigram Lola boards. + + To compile this driver as a module, choose M here: the module + will be called snd-lola. + config SND_LX6464ES tristate "Digigram LX6464ES" select SND_PCM diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 9cf4348ec13..54fe325e3aa 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -64,6 +64,7 @@ obj-$(CONFIG_SND) += \ ca0106/ \ cs46xx/ \ cs5535audio/ \ + lola/ \ lx6464es/ \ echoaudio/ \ emu10k1/ \ diff --git a/sound/pci/lola/Makefile b/sound/pci/lola/Makefile new file mode 100644 index 00000000000..674715a8496 --- /dev/null +++ b/sound/pci/lola/Makefile @@ -0,0 +1,3 @@ +snd-lola-y := lola.o lola_pcm.o lola_clock.o lola_mixer.o +snd-lola-$(CONFIG_SND_DEBUG) += lola_proc.o +obj-m = snd-lola.o diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c new file mode 100644 index 00000000000..f59ce085a1a --- /dev/null +++ b/sound/pci/lola/lola.c @@ -0,0 +1,731 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License along with + * this program; if not, write to the Free Software Foundation, Inc., 59 + * Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "lola.h" + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Digigram Lola driver."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Digigram Lola driver."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Digigram Lola driver."); + +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Digigram, Lola}}"); +MODULE_DESCRIPTION("Digigram Lola driver"); +MODULE_AUTHOR("Takashi Iwai "); + +#ifdef CONFIG_SND_DEBUG_VERBOSE +static int debug; +module_param(debug, int, 0644); +#define verbose_debug(fmt, args...) \ + do { if (debug > 1) printk(KERN_DEBUG SFX fmt, ##args); } while (0) +#else +#define verbose_debug(fmt, args...) +#endif + +/* + * pseudo-codec read/write via CORB/RIRB + */ + +static int corb_send_verb(struct lola *chip, unsigned int nid, + unsigned int verb, unsigned int data, + unsigned int extdata) +{ + unsigned long flags; + int ret = -EIO; + + chip->last_cmd_nid = nid; + chip->last_verb = verb; + chip->last_data = data; + chip->last_extdata = extdata; + data |= (nid << 20) | (verb << 8); + spin_lock_irqsave(&chip->reg_lock, flags); + if (chip->rirb.cmds < LOLA_CORB_ENTRIES - 1) { + unsigned int wp = chip->corb.wp + 1; + wp %= LOLA_CORB_ENTRIES; + chip->corb.wp = wp; + chip->corb.buf[wp * 2] = cpu_to_le32(data); + chip->corb.buf[wp * 2 + 1] = cpu_to_le32(extdata); + lola_writew(chip, BAR0, CORBWP, wp); + chip->rirb.cmds++; + smp_wmb(); + ret = 0; + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + return ret; +} + +static void lola_queue_unsol_event(struct lola *chip, unsigned int res, + unsigned int res_ex) +{ + lola_update_ext_clock_freq(chip, res); +} + +/* retrieve RIRB entry - called from interrupt handler */ +static void lola_update_rirb(struct lola *chip) +{ + unsigned int rp, wp; + u32 res, res_ex; + + wp = lola_readw(chip, BAR0, RIRBWP); + if (wp == chip->rirb.wp) + return; + chip->rirb.wp = wp; + + while (chip->rirb.rp != wp) { + chip->rirb.rp++; + chip->rirb.rp %= LOLA_CORB_ENTRIES; + + rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ + res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); + res = le32_to_cpu(chip->rirb.buf[rp]); + if (res_ex & LOLA_RIRB_EX_UNSOL_EV) + lola_queue_unsol_event(chip, res, res_ex); + else if (chip->rirb.cmds) { + chip->res = res; + chip->res_ex = res_ex; + smp_wmb(); + chip->rirb.cmds--; + } + } +} + +static int rirb_get_response(struct lola *chip, unsigned int *val, + unsigned int *extval) +{ + unsigned long timeout; + + timeout = jiffies + msecs_to_jiffies(1000); + for (;;) { + if (!chip->rirb.cmds) { + *val = chip->res; + if (extval) + *extval = chip->res_ex; + verbose_debug("get_response: %x, %x\n", + chip->res, chip->res_ex); + if (chip->res_ex & LOLA_RIRB_EX_ERROR) { + printk(KERN_WARNING SFX "RIRB ERROR: " + "NID=%x, verb=%x, data=%x, ext=%x\n", + chip->last_cmd_nid, + chip->last_verb, chip->last_data, + chip->last_extdata); + return -EIO; + } + return 0; + } + if (time_after(jiffies, timeout)) + break; + udelay(20); + cond_resched(); + lola_update_rirb(chip); + } + printk(KERN_WARNING SFX "RIRB response error\n"); + return -EIO; +} + +/* aynchronous write of a codec verb with data */ +int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata) +{ + verbose_debug("codec_write NID=%x, verb=%x, data=%x, ext=%x\n", + nid, verb, data, extdata); + return corb_send_verb(chip, nid, verb, data, extdata); +} + +/* write a codec verb with data and read the returned status */ +int lola_codec_read(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata, + unsigned int *val, unsigned int *extval) +{ + int err; + + verbose_debug("codec_read NID=%x, verb=%x, data=%x, ext=%x\n", + nid, verb, data, extdata); + err = corb_send_verb(chip, nid, verb, data, extdata); + if (err < 0) + return err; + err = rirb_get_response(chip, val, extval); + return err; +} + +/* flush all pending codec writes */ +int lola_codec_flush(struct lola *chip) +{ + unsigned int tmp; + return rirb_get_response(chip, &tmp, NULL); +} + +/* + * interrupt handler + */ +static irqreturn_t lola_interrupt(int irq, void *dev_id) +{ + struct lola *chip = dev_id; + unsigned int notify_ins, notify_outs, error_ins, error_outs; + int handled = 0; + int i; + + notify_ins = notify_outs = error_ins = error_outs = 0; + spin_lock(&chip->reg_lock); + for (;;) { + unsigned int status, in_sts, out_sts; + unsigned int reg; + + status = lola_readl(chip, BAR1, DINTSTS); + if (!status || status == -1) + break; + + in_sts = lola_readl(chip, BAR1, DIINTSTS); + out_sts = lola_readl(chip, BAR1, DOINTSTS); + + /* clear Input Interrupts */ + for (i = 0; in_sts && i < chip->pcm[CAPT].num_streams; i++) { + if (!(in_sts & (1 << i))) + continue; + in_sts &= ~(1 << i); + reg = lola_dsd_read(chip, i, STS); + if (reg & LOLA_DSD_STS_DESE) /* error */ + error_ins |= (1 << i); + if (reg & LOLA_DSD_STS_BCIS) /* notify */ + notify_ins |= (1 << i); + /* clear */ + lola_dsd_write(chip, i, STS, reg); + } + + /* clear Output Interrupts */ + for (i = 0; out_sts && i < chip->pcm[PLAY].num_streams; i++) { + if (!(out_sts & (1 << i))) + continue; + out_sts &= ~(1 << i); + reg = lola_dsd_read(chip, i + MAX_STREAM_IN_COUNT, STS); + if (reg & LOLA_DSD_STS_DESE) /* error */ + error_outs |= (1 << i); + if (reg & LOLA_DSD_STS_BCIS) /* notify */ + notify_outs |= (1 << i); + lola_dsd_write(chip, i + MAX_STREAM_IN_COUNT, STS, reg); + } + + if (status & LOLA_DINT_CTRL) { + unsigned char rbsts; /* ring status is byte access */ + rbsts = lola_readb(chip, BAR0, RIRBSTS); + rbsts &= LOLA_RIRB_INT_MASK; + if (rbsts) + lola_writeb(chip, BAR0, RIRBSTS, rbsts); + rbsts = lola_readb(chip, BAR0, CORBSTS); + rbsts &= LOLA_CORB_INT_MASK; + if (rbsts) + lola_writeb(chip, BAR0, CORBSTS, rbsts); + + lola_update_rirb(chip); + } + + if (status & (LOLA_DINT_FIFOERR | LOLA_DINT_MUERR)) { + /* clear global fifo error interrupt */ + lola_writel(chip, BAR1, DINTSTS, + (status & (LOLA_DINT_FIFOERR | LOLA_DINT_MUERR))); + } + handled = 1; + } + spin_unlock(&chip->reg_lock); + + lola_pcm_update(chip, &chip->pcm[CAPT], notify_ins); + lola_pcm_update(chip, &chip->pcm[PLAY], notify_outs); + + return IRQ_RETVAL(handled); +} + + +/* + * controller + */ +static int reset_controller(struct lola *chip) +{ + unsigned int gctl = lola_readl(chip, BAR0, GCTL); + unsigned long end_time; + + if (gctl) { + /* to be sure */ + lola_writel(chip, BAR1, BOARD_MODE, 0); + return 0; + } + + chip->cold_reset = 1; + lola_writel(chip, BAR0, GCTL, LOLA_GCTL_RESET); + end_time = jiffies + msecs_to_jiffies(200); + do { + msleep(1); + gctl = lola_readl(chip, BAR0, GCTL); + if (gctl) + break; + } while (time_before(jiffies, end_time)); + if (!gctl) { + printk(KERN_ERR SFX "cannot reset controller\n"); + return -EIO; + } + return 0; +} + +static void lola_irq_enable(struct lola *chip) +{ + unsigned int val; + + /* enalbe all I/O streams */ + val = (1 << chip->pcm[PLAY].num_streams) - 1; + lola_writel(chip, BAR1, DOINTCTL, val); + val = (1 << chip->pcm[CAPT].num_streams) - 1; + lola_writel(chip, BAR1, DIINTCTL, val); + + /* enable global irqs */ + val = LOLA_DINT_GLOBAL | LOLA_DINT_CTRL | LOLA_DINT_FIFOERR | + LOLA_DINT_MUERR; + lola_writel(chip, BAR1, DINTCTL, val); +} + +static void lola_irq_disable(struct lola *chip) +{ + lola_writel(chip, BAR1, DINTCTL, 0); + lola_writel(chip, BAR1, DIINTCTL, 0); + lola_writel(chip, BAR1, DOINTCTL, 0); +} + +static int setup_corb_rirb(struct lola *chip) +{ + int err; + unsigned char tmp; + unsigned long end_time; + + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + PAGE_SIZE, &chip->rb); + if (err < 0) + return err; + + chip->corb.addr = chip->rb.addr; + chip->corb.buf = (u32 *)chip->rb.area; + chip->rirb.addr = chip->rb.addr + 2048; + chip->rirb.buf = (u32 *)(chip->rb.area + 2048); + lola_writel(chip, BAR0, CORBLBASE, (u32)chip->corb.addr); + lola_writel(chip, BAR0, CORBUBASE, upper_32_bits(chip->corb.addr)); + + /* disable ringbuffer DMAs */ + lola_writeb(chip, BAR0, RIRBCTL, 0); + lola_writeb(chip, BAR0, CORBCTL, 0); + + end_time = jiffies + msecs_to_jiffies(200); + do { + if (!lola_readb(chip, BAR0, RIRBCTL) && + !lola_readb(chip, BAR0, CORBCTL)) + break; + msleep(1); + } while (time_before(jiffies, end_time)); + + /* CORB set up */ + lola_writel(chip, BAR0, CORBLBASE, (u32)chip->corb.addr); + lola_writel(chip, BAR0, CORBUBASE, upper_32_bits(chip->corb.addr)); + /* set the corb size to 256 entries */ + lola_writeb(chip, BAR0, CORBSIZE, 0x02); + /* set the corb write pointer to 0 */ + lola_writew(chip, BAR0, CORBWP, 0); + /* reset the corb hw read pointer */ + lola_writew(chip, BAR0, CORBRP, LOLA_RBRWP_CLR); + /* enable corb dma */ + lola_writeb(chip, BAR0, CORBCTL, LOLA_RBCTL_DMA_EN); + /* clear flags if set */ + tmp = lola_readb(chip, BAR0, CORBSTS) & LOLA_CORB_INT_MASK; + if (tmp) + lola_writeb(chip, BAR0, CORBSTS, tmp); + chip->corb.wp = 0; + + /* RIRB set up */ + lola_writel(chip, BAR0, RIRBLBASE, (u32)chip->rirb.addr); + lola_writel(chip, BAR0, RIRBUBASE, upper_32_bits(chip->rirb.addr)); + /* set the rirb size to 256 entries */ + lola_writeb(chip, BAR0, RIRBSIZE, 0x02); + /* reset the rirb hw write pointer */ + lola_writew(chip, BAR0, RIRBWP, LOLA_RBRWP_CLR); + /* set N=1, get RIRB response interrupt for new entry */ + lola_writew(chip, BAR0, RINTCNT, 1); + /* enable rirb dma and response irq */ + lola_writeb(chip, BAR0, RIRBCTL, LOLA_RBCTL_DMA_EN | LOLA_RBCTL_IRQ_EN); + /* clear flags if set */ + tmp = lola_readb(chip, BAR0, RIRBSTS) & LOLA_RIRB_INT_MASK; + if (tmp) + lola_writeb(chip, BAR0, RIRBSTS, tmp); + chip->rirb.rp = chip->rirb.cmds = 0; + + return 0; +} + +static void stop_corb_rirb(struct lola *chip) +{ + /* disable ringbuffer DMAs */ + lola_writeb(chip, BAR0, RIRBCTL, 0); + lola_writeb(chip, BAR0, CORBCTL, 0); +} + +static void lola_reset_setups(struct lola *chip) +{ + /* update the granularity */ + lola_set_granularity(chip, chip->granularity, true); + /* update the sample clock */ + lola_set_clock_index(chip, chip->clock.cur_index); + /* enable unsolicited events of the clock widget */ + lola_enable_clock_events(chip); + /* update the analog gains */ + lola_setup_all_analog_gains(chip, CAPT, false); /* input, update */ + /* update SRC configuration if applicable */ + lola_set_src_config(chip, chip->input_src_mask, false); + /* update the analog outputs */ + lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */ +} + +static int lola_parse_tree(struct lola *chip) +{ + unsigned int val; + int nid, err; + + err = lola_read_param(chip, 0, LOLA_PAR_VENDOR_ID, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read VENDOR_ID\n"); + return err; + } + val >>= 16; + if (val != 0x1369) { + printk(KERN_ERR SFX "Unknown codec vendor 0x%x\n", val); + return -EINVAL; + } + + err = lola_read_param(chip, 1, LOLA_PAR_FUNCTION_TYPE, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read FUNCTION_TYPE for 0x%x\n", nid); + return err; + } + if (val != 1) { + printk(KERN_ERR SFX "Unknown function type %d\n", val); + return -EINVAL; + } + + err = lola_read_param(chip, 1, LOLA_PAR_SPECIFIC_CAPS, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read SPECCAPS\n"); + return err; + } + chip->lola_caps = val; + chip->pin[CAPT].num_pins = LOLA_AFG_INPUT_PIN_COUNT(chip->lola_caps); + chip->pin[PLAY].num_pins = LOLA_AFG_OUTPUT_PIN_COUNT(chip->lola_caps); + snd_printd(SFX "speccaps=0x%x, pins in=%d, out=%d\n", + chip->lola_caps, + chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins); + + if (chip->pin[CAPT].num_pins > MAX_AUDIO_INOUT_COUNT || + chip->pin[PLAY].num_pins > MAX_AUDIO_INOUT_COUNT) { + printk(KERN_ERR SFX "Invalid Lola-spec caps 0x%x\n", val); + return -EINVAL; + } + + nid = 0x02; + err = lola_init_pcm(chip, CAPT, &nid); + if (err < 0) + return err; + err = lola_init_pcm(chip, PLAY, &nid); + if (err < 0) + return err; + + err = lola_init_pins(chip, CAPT, &nid); + if (err < 0) + return err; + err = lola_init_pins(chip, PLAY, &nid); + if (err < 0) + return err; + + if (LOLA_AFG_CLOCK_WIDGET_PRESENT(chip->lola_caps)) { + err = lola_init_clock_widget(chip, nid); + if (err < 0) + return err; + nid++; + } + if (LOLA_AFG_MIXER_WIDGET_PRESENT(chip->lola_caps)) { + err = lola_init_mixer_widget(chip, nid); + if (err < 0) + return err; + nid++; + } + + /* enable unsolicited events of the clock widget */ + err = lola_enable_clock_events(chip); + if (err < 0) + return err; + + /* if last ResetController was not a ColdReset, we don't know + * the state of the card; initialize here again + */ + if (!chip->cold_reset) { + lola_reset_setups(chip); + chip->cold_reset = 1; + } + + return 0; +} + +static void lola_stop_hw(struct lola *chip) +{ + stop_corb_rirb(chip); + lola_irq_disable(chip); +} + +static void lola_free(struct lola *chip) +{ + if (chip->initialized) + lola_stop_hw(chip); + lola_free_pcm(chip); + lola_free_mixer(chip); + if (chip->irq >= 0) + free_irq(chip->irq, (void *)chip); + if (chip->bar[0].remap_addr) + iounmap(chip->bar[0].remap_addr); + if (chip->bar[1].remap_addr) + iounmap(chip->bar[1].remap_addr); + if (chip->rb.area) + snd_dma_free_pages(&chip->rb); + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); +} + +static int lola_dev_free(struct snd_device *device) +{ + lola_free(device->device_data); + return 0; +} + +static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, + struct lola **rchip) +{ + struct lola *chip; + int err; + unsigned int dever; + static struct snd_device_ops ops = { + .dev_free = lola_dev_free, + }; + + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + snd_printk(KERN_ERR SFX "cannot allocate chip\n"); + pci_disable_device(pci); + return -ENOMEM; + } + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + chip->sample_rate_min = 48000; + chip->granularity = LOLA_GRANULARITY_MIN; + + err = pci_request_regions(pci, DRVNAME); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + + chip->bar[0].addr = pci_resource_start(pci, 0); + chip->bar[0].remap_addr = pci_ioremap_bar(pci, 0); + chip->bar[1].addr = pci_resource_start(pci, 2); + chip->bar[1].remap_addr = pci_ioremap_bar(pci, 2); + if (!chip->bar[0].remap_addr || !chip->bar[1].remap_addr) { + snd_printk(KERN_ERR SFX "ioremap error\n"); + err = -ENXIO; + goto errout; + } + + pci_set_master(pci); + + err = reset_controller(chip); + if (err < 0) + goto errout; + + if (request_irq(pci->irq, lola_interrupt, IRQF_SHARED, + DRVNAME, chip)) { + printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq); + err = -EBUSY; + goto errout; + } + chip->irq = pci->irq; + synchronize_irq(chip->irq); + + dever = lola_readl(chip, BAR1, DEVER); + chip->pcm[CAPT].num_streams = (dever >> 0) & 0x3ff; + chip->pcm[PLAY].num_streams = (dever >> 10) & 0x3ff; + chip->version = (dever >> 24) & 0xff; + snd_printd(SFX "streams in=%d, out=%d, version=0x%x\n", + chip->pcm[CAPT].num_streams, chip->pcm[PLAY].num_streams, + chip->version); + + /* Test LOLA_BAR1_DEVER */ + if (chip->pcm[CAPT].num_streams > MAX_STREAM_IN_COUNT || + chip->pcm[PLAY].num_streams > MAX_STREAM_OUT_COUNT || + (!chip->pcm[CAPT].num_streams && + !chip->pcm[PLAY].num_streams)) { + printk(KERN_ERR SFX "invalid DEVER = %x\n", dever); + err = -EINVAL; + goto errout; + } + + err = setup_corb_rirb(chip); + if (err < 0) + goto errout; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_printk(KERN_ERR SFX "Error creating device [card]!\n"); + goto errout; + } + + strcpy(card->driver, "Lola"); + strlcpy(card->shortname, "Digigram Lola", sizeof(card->shortname)); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->bar[0].addr, chip->irq); + strcpy(card->mixername, card->shortname); + + lola_irq_enable(chip); + + chip->initialized = 1; + *rchip = chip; + return 0; + + errout: + lola_free(chip); + return err; +} + +static int __devinit lola_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct lola *chip; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR SFX "Error creating card!\n"); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + err = lola_create(card, pci, &chip); + if (err < 0) + goto out_free; + card->private_data = chip; + + err = lola_parse_tree(chip); + if (err < 0) + goto out_free; + + err = lola_create_pcm(chip); + if (err < 0) + goto out_free; + + err = lola_create_mixer(chip); + if (err < 0) + goto out_free; + + lola_proc_debug_new(chip); + + err = snd_card_register(card); + if (err < 0) + goto out_free; + + pci_set_drvdata(pci, card); + dev++; + return err; +out_free: + snd_card_free(card); + return err; +} + +static void __devexit lola_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +/* PCI IDs */ +static DEFINE_PCI_DEVICE_TABLE(lola_ids) = { + { PCI_VDEVICE(DIGIGRAM, 0x0001) }, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, lola_ids); + +/* pci_driver definition */ +static struct pci_driver driver = { + .name = DRVNAME, + .id_table = lola_ids, + .probe = lola_probe, + .remove = __devexit_p(lola_remove), +}; + +static int __init alsa_card_lola_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_lola_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_lola_init) +module_exit(alsa_card_lola_exit) diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h new file mode 100644 index 00000000000..4734c7c6f83 --- /dev/null +++ b/sound/pci/lola/lola.h @@ -0,0 +1,520 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License along with + * this program; if not, write to the Free Software Foundation, Inc., 59 + * Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#ifndef _LOLA_H +#define _LOLA_H + +#define DRVNAME "snd-lola" +#define SFX DRVNAME ": " + +/* + * Lola HD Audio Registers BAR0 + */ +#define LOLA_BAR0_GCAP 0x00 +#define LOLA_BAR0_VMIN 0x02 +#define LOLA_BAR0_VMAJ 0x03 +#define LOLA_BAR0_OUTPAY 0x04 +#define LOLA_BAR0_INPAY 0x06 +#define LOLA_BAR0_GCTL 0x08 +#define LOLA_BAR0_WAKEEN 0x0c +#define LOLA_BAR0_STATESTS 0x0e +#define LOLA_BAR0_GSTS 0x10 +#define LOLA_BAR0_OUTSTRMPAY 0x18 +#define LOLA_BAR0_INSTRMPAY 0x1a +#define LOLA_BAR0_INTCTL 0x20 +#define LOLA_BAR0_INTSTS 0x24 +#define LOLA_BAR0_WALCLK 0x30 +#define LOLA_BAR0_SSYNC 0x38 + +#define LOLA_BAR0_CORBLBASE 0x40 +#define LOLA_BAR0_CORBUBASE 0x44 +#define LOLA_BAR0_CORBWP 0x48 /* no ULONG access */ +#define LOLA_BAR0_CORBRP 0x4a /* no ULONG access */ +#define LOLA_BAR0_CORBCTL 0x4c /* no ULONG access */ +#define LOLA_BAR0_CORBSTS 0x4d /* UCHAR access only */ +#define LOLA_BAR0_CORBSIZE 0x4e /* no ULONG access */ + +#define LOLA_BAR0_RIRBLBASE 0x50 +#define LOLA_BAR0_RIRBUBASE 0x54 +#define LOLA_BAR0_RIRBWP 0x58 +#define LOLA_BAR0_RINTCNT 0x5a /* no ULONG access */ +#define LOLA_BAR0_RIRBCTL 0x5c +#define LOLA_BAR0_RIRBSTS 0x5d /* UCHAR access only */ +#define LOLA_BAR0_RIRBSIZE 0x5e /* no ULONG access */ + +#define LOLA_BAR0_ICW 0x60 +#define LOLA_BAR0_IRR 0x64 +#define LOLA_BAR0_ICS 0x68 +#define LOLA_BAR0_DPLBASE 0x70 +#define LOLA_BAR0_DPUBASE 0x74 + +/* stream register offsets from stream base 0x80 */ +#define LOLA_BAR0_SD0_OFFSET 0x80 +#define LOLA_REG0_SD_CTL 0x00 +#define LOLA_REG0_SD_STS 0x03 +#define LOLA_REG0_SD_LPIB 0x04 +#define LOLA_REG0_SD_CBL 0x08 +#define LOLA_REG0_SD_LVI 0x0c +#define LOLA_REG0_SD_FIFOW 0x0e +#define LOLA_REG0_SD_FIFOSIZE 0x10 +#define LOLA_REG0_SD_FORMAT 0x12 +#define LOLA_REG0_SD_BDLPL 0x18 +#define LOLA_REG0_SD_BDLPU 0x1c + +/* + * Lola Digigram Registers BAR1 + */ +#define LOLA_BAR1_FPGAVER 0x00 +#define LOLA_BAR1_DEVER 0x04 +#define LOLA_BAR1_UCBMV 0x08 +#define LOLA_BAR1_JTAG 0x0c +#define LOLA_BAR1_UARTRX 0x10 +#define LOLA_BAR1_UARTTX 0x14 +#define LOLA_BAR1_UARTCR 0x18 +#define LOLA_BAR1_NVRAMVER 0x1c +#define LOLA_BAR1_CTRLSPI 0x20 +#define LOLA_BAR1_DSPI 0x24 +#define LOLA_BAR1_AISPI 0x28 +#define LOLA_BAR1_GRAN 0x2c + +#define LOLA_BAR1_DINTCTL 0x80 +#define LOLA_BAR1_DIINTCTL 0x84 +#define LOLA_BAR1_DOINTCTL 0x88 +#define LOLA_BAR1_LRC 0x90 +#define LOLA_BAR1_DINTSTS 0x94 +#define LOLA_BAR1_DIINTSTS 0x98 +#define LOLA_BAR1_DOINTSTS 0x9c + +#define LOLA_BAR1_DSD0_OFFSET 0xa0 +#define LOLA_BAR1_DSD_SIZE 0x18 + +#define LOLA_BAR1_DSDnSTS 0x00 +#define LOLA_BAR1_DSDnLPIB 0x04 +#define LOLA_BAR1_DSDnCTL 0x08 +#define LOLA_BAR1_DSDnLVI 0x0c +#define LOLA_BAR1_DSDnBDPL 0x10 +#define LOLA_BAR1_DSDnBDPU 0x14 + +#define LOLA_BAR1_SSYNC 0x03e8 + +#define LOLA_BAR1_BOARD_CTRL 0x0f00 +#define LOLA_BAR1_BOARD_MODE 0x0f02 + +#define LOLA_BAR1_SOURCE_GAIN_ENABLE 0x1000 +#define LOLA_BAR1_DEST00_MIX_GAIN_ENABLE 0x1004 +#define LOLA_BAR1_DEST31_MIX_GAIN_ENABLE 0x1080 +#define LOLA_BAR1_SOURCE00_01_GAIN 0x1084 +#define LOLA_BAR1_SOURCE30_31_GAIN 0x10c0 +#define LOLA_BAR1_SOURCE_GAIN(src) \ + (LOLA_BAR1_SOURCE00_01_GAIN + (src) * 2) +#define LOLA_BAR1_DEST00_MIX00_01_GAIN 0x10c4 +#define LOLA_BAR1_DEST00_MIX30_31_GAIN 0x1100 +#define LOLA_BAR1_DEST01_MIX00_01_GAIN 0x1104 +#define LOLA_BAR1_DEST01_MIX30_31_GAIN 0x1140 +#define LOLA_BAR1_DEST31_MIX00_01_GAIN 0x1884 +#define LOLA_BAR1_DEST31_MIX30_31_GAIN 0x18c0 +#define LOLA_BAR1_MIX_GAIN(dest, mix) \ + (LOLA_BAR1_DEST00_MIX00_01_GAIN + (dest) * 0x40 + (mix) * 2) +#define LOLA_BAR1_ANALOG_CLIP_IN 0x18c4 +#define LOLA_BAR1_PEAKMETERS_SOURCE00_01 0x18c8 +#define LOLA_BAR1_PEAKMETERS_SOURCE30_31 0x1904 +#define LOLA_BAR1_PEAKMETERS_SOURCE(src) \ + (LOLA_BAR1_PEAKMETERS_SOURCE00_01 + (src) * 2) +#define LOLA_BAR1_PEAKMETERS_DEST00_01 0x1908 +#define LOLA_BAR1_PEAKMETERS_DEST30_31 0x1944 +#define LOLA_BAR1_PEAKMETERS_DEST(dest) \ + (LOLA_BAR1_PEAKMETERS_DEST00_01 + (dest) * 2) +#define LOLA_BAR1_PEAKMETERS_AGC00_01 0x1948 +#define LOLA_BAR1_PEAKMETERS_AGC14_15 0x1964 +#define LOLA_BAR1_PEAKMETERS_AGC(x) \ + (LOLA_BAR1_PEAKMETERS_AGC00_01 + (x) * 2) + +/* GCTL reset bit */ +#define LOLA_GCTL_RESET (1 << 0) +/* GCTL unsolicited response enable bit */ +#define LOLA_GCTL_UREN (1 << 8) + +/* CORB/RIRB control, read/write pointer */ +#define LOLA_RBCTL_DMA_EN 0x02 /* enable DMA */ +#define LOLA_RBCTL_IRQ_EN 0x01 /* enable IRQ */ +#define LOLA_RBRWP_CLR 0x8000 /* read/write pointer clear */ + +#define LOLA_RIRB_EX_UNSOL_EV 0x40000000 +#define LOLA_RIRB_EX_ERROR 0x80000000 + +/* CORB int mask: CMEI[0] */ +#define LOLA_CORB_INT_CMEI 0x01 +#define LOLA_CORB_INT_MASK LOLA_CORB_INT_CMEI + +/* RIRB int mask: overrun[2], response[0] */ +#define LOLA_RIRB_INT_RESPONSE 0x01 +#define LOLA_RIRB_INT_OVERRUN 0x04 +#define LOLA_RIRB_INT_MASK (LOLA_RIRB_INT_RESPONSE | LOLA_RIRB_INT_OVERRUN) + +/* DINTCTL and DINTSTS */ +#define LOLA_DINT_GLOBAL 0x80000000 /* global interrupt enable bit */ +#define LOLA_DINT_CTRL 0x40000000 /* controller interrupt enable bit */ +#define LOLA_DINT_FIFOERR 0x20000000 /* global fifo error enable bit */ +#define LOLA_DINT_MUERR 0x10000000 /* global microcontroller underrun error */ + +/* DSDnCTL bits */ +#define LOLA_DSD_CTL_SRST 0x01 /* stream reset bit */ +#define LOLA_DSD_CTL_SRUN 0x02 /* stream DMA start bit */ +#define LOLA_DSD_CTL_IOCE 0x04 /* interrupt on completion enable */ +#define LOLA_DSD_CTL_DEIE 0x10 /* descriptor error interrupt enable */ +#define LOLA_DSD_CTL_VLRCV 0x20 /* valid LRCountValue information in bits 8..31 */ +#define LOLA_LRC_MASK 0xffffff00 + +/* DSDnSTS */ +#define LOLA_DSD_STS_BCIS 0x04 /* buffer completion interrupt status */ +#define LOLA_DSD_STS_DESE 0x10 /* descriptor error interrupt */ +#define LOLA_DSD_STS_FIFORDY 0x20 /* fifo ready */ + +#define LOLA_CORB_ENTRIES 256 + +#define MAX_STREAM_IN_COUNT 16 +#define MAX_STREAM_OUT_COUNT 16 +#define MAX_STREAM_COUNT 16 +#define MAX_PINS MAX_STREAM_COUNT +#define MAX_STREAM_BUFFER_COUNT 16 +#define MAX_AUDIO_INOUT_COUNT 16 + +#define LOLA_CLOCK_TYPE_INTERNAL 0 +#define LOLA_CLOCK_TYPE_AES 1 +#define LOLA_CLOCK_TYPE_AES_SYNC 2 +#define LOLA_CLOCK_TYPE_WORDCLOCK 3 +#define LOLA_CLOCK_TYPE_ETHERSOUND 4 +#define LOLA_CLOCK_TYPE_VIDEO 5 + +#define LOLA_CLOCK_FORMAT_NONE 0 +#define LOLA_CLOCK_FORMAT_NTSC 1 +#define LOLA_CLOCK_FORMAT_PAL 2 + +#define MAX_SAMPLE_CLOCK_COUNT 48 + +/* parameters used with mixer widget's mixer capabilities */ +#define LOLA_PEAK_METER_CAN_AGC_MASK 1 +#define LOLA_PEAK_METER_CAN_ANALOG_CLIP_MASK 2 + +struct lola_bar { + unsigned long addr; + void __iomem *remap_addr; +}; + +/* CORB/RIRB */ +struct lola_rb { + u32 *buf; /* CORB/RIRB buffer, 8 byte per each entry */ + dma_addr_t addr; /* physical address of CORB/RIRB buffer */ + unsigned short rp, wp; /* read/write pointers */ + int cmds; /* number of pending requests */ +}; + +/* Pin widget setup */ +struct lola_pin { + unsigned int nid; + bool is_analog; + unsigned int amp_mute; + unsigned int amp_step_size; + unsigned int amp_num_steps; + unsigned int amp_offset; + unsigned int max_level; + unsigned int config_default_reg; + unsigned int fixed_gain_list_len; + unsigned int cur_gain_step; +}; + +struct lola_pin_array { + unsigned int num_pins; + struct lola_pin pins[MAX_PINS]; +}; + +/* Clock widget setup */ +struct lola_sample_clock { + unsigned int type; + unsigned int format; + unsigned int freq; +}; + +struct lola_clock_widget { + unsigned int nid; + unsigned int items; + unsigned int cur_index; + unsigned int cur_freq; + bool cur_valid; + struct lola_sample_clock sample_clock[MAX_SAMPLE_CLOCK_COUNT]; + unsigned int idx_lookup[MAX_SAMPLE_CLOCK_COUNT]; +}; + +#define LOLA_MIXER_DIM 32 +struct lola_mixer_array { + u32 src_gain_enable; + u32 dest_mix_gain_enable[LOLA_MIXER_DIM]; + u16 src_gain[LOLA_MIXER_DIM]; + u16 dest_mix_gain[LOLA_MIXER_DIM][LOLA_MIXER_DIM]; +}; + +/* Mixer widget setup */ +struct lola_mixer_widget { + unsigned int nid; + unsigned int caps; + struct lola_mixer_array __user *array; + struct lola_mixer_array *array_saved; + unsigned int src_stream_outs; + unsigned int src_phys_ins; + unsigned int dest_stream_ins; + unsigned int dest_phys_outs; + unsigned int src_stream_out_ofs; + unsigned int dest_phys_out_ofs; + unsigned int src_mask; + unsigned int dest_mask; +}; + +/* Audio stream */ +struct lola_stream { + unsigned int nid; /* audio widget NID */ + unsigned int index; /* array index */ + unsigned int dsd; /* DSD index */ + bool can_float; + struct snd_pcm_substream *substream; /* assigned PCM substream */ + struct lola_stream *master; /* master stream (for multi-channel) */ + + /* buffer setup */ + unsigned int bufsize; + unsigned int period_bytes; + unsigned int frags; + struct snd_dma_buffer bdl; /* BDL buffer */ + + /* format + channel setup */ + unsigned int format_verb; + + /* flags */ + unsigned int opened:1; + unsigned int running:1; +}; + +#define PLAY SNDRV_PCM_STREAM_PLAYBACK +#define CAPT SNDRV_PCM_STREAM_CAPTURE + +struct lola_pcm { + unsigned int num_streams; + struct lola_stream streams[MAX_STREAM_COUNT]; +}; + +/* card instance */ +struct lola { + struct snd_card *card; + struct pci_dev *pci; + + /* pci resources */ + struct lola_bar bar[2]; + int irq; + + /* locks */ + spinlock_t reg_lock; + struct mutex open_mutex; + + /* CORB/RIRB */ + struct lola_rb corb; + struct lola_rb rirb; + unsigned int res, res_ex; /* last read values */ + /* last command (for debugging) */ + unsigned int last_cmd_nid, last_verb, last_data, last_extdata; + + /* CORB/RIRB buffers */ + struct snd_dma_buffer rb; + + /* unsolicited events */ + unsigned int last_unsol_res; + + /* streams */ + struct lola_pcm pcm[2]; + + /* input src */ + unsigned int input_src_caps_mask; + unsigned int input_src_mask; + + /* pins */ + struct lola_pin_array pin[2]; + + /* clock */ + struct lola_clock_widget clock; + + /* mixer */ + struct lola_mixer_widget mixer; + + /* hw info */ + unsigned int version; + unsigned int lola_caps; + + /* parameters */ + unsigned int granularity; + unsigned int sample_rate_min; + + /* flags */ + unsigned int running :1; + unsigned int initialized :1; + unsigned int cold_reset :1; + + /* for debugging */ + unsigned int debug_res; + unsigned int debug_res_ex; +}; + +#define BAR0 0 +#define BAR1 1 + +/* Helper macros */ +#define lola_readl(chip, idx, name) \ + readl((chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_readw(chip, idx, name) \ + readw((chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_readb(chip, idx, name) \ + readb((chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_writel(chip, idx, name, val) \ + writel((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_writew(chip, idx, name, val) \ + writew((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name) +#define lola_writeb(chip, idx, name, val) \ + writeb((val), (chip)->bar[idx].remap_addr + LOLA_##idx##_##name) + +#define lola_dsd_read(chip, dsd, name) \ + readl((chip)->bar[BAR1].remap_addr + LOLA_BAR1_DSD0_OFFSET + \ + (LOLA_BAR1_DSD_SIZE * (dsd)) + LOLA_BAR1_DSDn##name) +#define lola_dsd_write(chip, dsd, name, val) \ + writel((val), (chip)->bar[BAR1].remap_addr + LOLA_BAR1_DSD0_OFFSET + \ + (LOLA_BAR1_DSD_SIZE * (dsd)) + LOLA_BAR1_DSDn##name) + +/* GET verbs HDAudio */ +#define LOLA_VERB_GET_STREAM_FORMAT 0xa00 +#define LOLA_VERB_GET_AMP_GAIN_MUTE 0xb00 +#define LOLA_VERB_PARAMETERS 0xf00 +#define LOLA_VERB_GET_POWER_STATE 0xf05 +#define LOLA_VERB_GET_CONV 0xf06 +#define LOLA_VERB_GET_UNSOLICITED_RESPONSE 0xf08 +#define LOLA_VERB_GET_DIGI_CONVERT_1 0xf0d +#define LOLA_VERB_GET_CONFIG_DEFAULT 0xf1c +#define LOLA_VERB_GET_SUBSYSTEM_ID 0xf20 +/* GET verbs Digigram */ +#define LOLA_VERB_GET_FIXED_GAIN 0xfc0 +#define LOLA_VERB_GET_GAIN_SELECT 0xfc1 +#define LOLA_VERB_GET_MAX_LEVEL 0xfc2 +#define LOLA_VERB_GET_CLOCK_LIST 0xfc3 +#define LOLA_VERB_GET_CLOCK_SELECT 0xfc4 +#define LOLA_VERB_GET_CLOCK_STATUS 0xfc5 + +/* SET verbs HDAudio */ +#define LOLA_VERB_SET_STREAM_FORMAT 0x200 +#define LOLA_VERB_SET_AMP_GAIN_MUTE 0x300 +#define LOLA_VERB_SET_POWER_STATE 0x705 +#define LOLA_VERB_SET_CHANNEL_STREAMID 0x706 +#define LOLA_VERB_SET_UNSOLICITED_ENABLE 0x708 +#define LOLA_VERB_SET_DIGI_CONVERT_1 0x70d +/* SET verbs Digigram */ +#define LOLA_VERB_SET_GAIN_SELECT 0xf81 +#define LOLA_VERB_SET_CLOCK_SELECT 0xf84 +#define LOLA_VERB_SET_GRANULARITY_STEPS 0xf86 +#define LOLA_VERB_SET_SOURCE_GAIN 0xf87 +#define LOLA_VERB_SET_MIX_GAIN 0xf88 +#define LOLA_VERB_SET_DESTINATION_GAIN 0xf89 +#define LOLA_VERB_SET_SRC 0xf8a + +/* Parameter IDs used with LOLA_VERB_PARAMETERS */ +#define LOLA_PAR_VENDOR_ID 0x00 +#define LOLA_PAR_FUNCTION_TYPE 0x05 +#define LOLA_PAR_AUDIO_WIDGET_CAP 0x09 +#define LOLA_PAR_PCM 0x0a +#define LOLA_PAR_STREAM_FORMATS 0x0b +#define LOLA_PAR_PIN_CAP 0x0c +#define LOLA_PAR_AMP_IN_CAP 0x0d +#define LOLA_PAR_CONNLIST_LEN 0x0e +#define LOLA_PAR_POWER_STATE 0x0f +#define LOLA_PAR_GPIO_CAP 0x11 +#define LOLA_PAR_AMP_OUT_CAP 0x12 +#define LOLA_PAR_SPECIFIC_CAPS 0x80 +#define LOLA_PAR_FIXED_GAIN_LIST 0x81 + +/* extract results of LOLA_PAR_SPECIFIC_CAPS */ +#define LOLA_AFG_MIXER_WIDGET_PRESENT(res) ((res & (1 << 21)) != 0) +#define LOLA_AFG_CLOCK_WIDGET_PRESENT(res) ((res & (1 << 20)) != 0) +#define LOLA_AFG_INPUT_PIN_COUNT(res) ((res >> 10) & 0x2ff) +#define LOLA_AFG_OUTPUT_PIN_COUNT(res) ((res) & 0x2ff) + +/* extract results of LOLA_PAR_AMP_IN_CAP / LOLA_PAR_AMP_OUT_CAP */ +#define LOLA_AMP_MUTE_CAPABLE(res) ((res & (1 << 31)) != 0) +#define LOLA_AMP_STEP_SIZE(res) ((res >> 24) & 0x7f) +#define LOLA_AMP_NUM_STEPS(res) ((res >> 12) & 0x3ff) +#define LOLA_AMP_OFFSET(res) ((res) & 0x3ff) + +#define LOLA_GRANULARITY_MIN 8 +#define LOLA_GRANULARITY_MAX 32 +#define LOLA_GRANULARITY_STEP 8 + +/* parameters used with unsolicited command/response */ +#define LOLA_UNSOLICITED_TAG_MASK 0x3f +#define LOLA_UNSOLICITED_TAG 0x1a +#define LOLA_UNSOLICITED_ENABLE 0x80 +#define LOLA_UNSOL_RESP_TAG_OFFSET 26 + +/* count values in the Vendor Specific Mixer Widget's Audio Widget Capabilities */ +#define LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(res) ((res >> 2) & 0x1f) +#define LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(res) ((res >> 7) & 0x1f) + +int lola_codec_write(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata); +int lola_codec_read(struct lola *chip, unsigned int nid, unsigned int verb, + unsigned int data, unsigned int extdata, + unsigned int *val, unsigned int *extval); +int lola_codec_flush(struct lola *chip); +#define lola_read_param(chip, nid, param, val) \ + lola_codec_read(chip, nid, LOLA_VERB_PARAMETERS, param, 0, val, NULL) + +/* PCM */ +int lola_create_pcm(struct lola *chip); +void lola_free_pcm(struct lola *chip); +int lola_init_pcm(struct lola *chip, int dir, int *nidp); +void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits); + +/* clock */ +int lola_init_clock_widget(struct lola *chip, int nid); +int lola_set_granularity(struct lola *chip, unsigned int val, bool force); +int lola_enable_clock_events(struct lola *chip); +int lola_set_clock_index(struct lola *chip, unsigned int idx); +int lola_set_clock(struct lola *chip, int idx); +int lola_set_sample_rate(struct lola *chip, int rate); +bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val); + +/* mixer */ +int lola_init_pins(struct lola *chip, int dir, int *nidp); +int lola_init_mixer_widget(struct lola *chip, int nid); +void lola_free_mixer(struct lola *chip); +int lola_create_mixer(struct lola *chip); +int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute); +void lola_save_mixer(struct lola *chip); +void lola_restore_mixer(struct lola *chip); +int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update); + +/* proc */ +#ifdef CONFIG_SND_DEBUG +void lola_proc_debug_new(struct lola *chip); +#else +#define lola_proc_debug_new(chip) +#endif + +#endif /* _LOLA_H */ diff --git a/sound/pci/lola/lola_clock.c b/sound/pci/lola/lola_clock.c new file mode 100644 index 00000000000..a364dc67e55 --- /dev/null +++ b/sound/pci/lola/lola_clock.c @@ -0,0 +1,322 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License along with + * this program; if not, write to the Free Software Foundation, Inc., 59 + * Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#include +#include +#include +#include +#include "lola.h" + +static unsigned int sample_rate_convert(unsigned int coded) +{ + unsigned int freq; + + /* base frequency */ + switch (coded & 0x3) { + case 0: freq = 48000; break; + case 1: freq = 44100; break; + case 2: freq = 32000; break; + default: return 0; /* error */ + } + + /* multiplier / devisor */ + switch (coded & 0x1c) { + case (0 << 2): break; + case (4 << 2): break; + case (1 << 2): freq *= 2; break; + case (2 << 2): freq *= 4; break; + case (5 << 2): freq /= 2; break; + case (6 << 2): freq /= 4; break; + default: return 0; /* error */ + } + + /* ajustement */ + switch (coded & 0x60) { + case (0 << 5): break; + case (1 << 5): freq = (freq * 999) / 1000; break; + case (2 << 5): freq = (freq * 1001) / 1000; break; + default: return 0; /* error */ + } + return freq; +} + +/* + * Granualrity + */ + +#define LOLA_MAXFREQ_AT_GRANULARITY_MIN 48000 +#define LOLA_MAXFREQ_AT_GRANULARITY_BELOW_MAX 96000 + +static bool check_gran_clock_compatibility(struct lola *chip, + unsigned int val, + unsigned int freq) +{ + if (!chip->granularity) + return true; + + if (val < LOLA_GRANULARITY_MIN || val > LOLA_GRANULARITY_MAX || + (val % LOLA_GRANULARITY_STEP) != 0) + return false; + + if (val == LOLA_GRANULARITY_MIN) { + if (freq > LOLA_MAXFREQ_AT_GRANULARITY_MIN) + return false; + } else if (val < LOLA_GRANULARITY_MAX) { + if (freq > LOLA_MAXFREQ_AT_GRANULARITY_BELOW_MAX) + return false; + } + return true; +} + +int lola_set_granularity(struct lola *chip, unsigned int val, bool force) +{ + int err; + + if (!force) { + if (val == chip->granularity) + return 0; +#if 0 + /* change Gran only if there are no streams allocated ! */ + if (chip->audio_in_alloc_mask || chip->audio_out_alloc_mask) + return -EBUSY; +#endif + if (!check_gran_clock_compatibility(chip, val, + chip->clock.cur_freq)) + return -EINVAL; + } + + chip->granularity = val; + val /= LOLA_GRANULARITY_STEP; + + /* audio function group */ + err = lola_codec_write(chip, 1, LOLA_VERB_SET_GRANULARITY_STEPS, + val, 0); + if (err < 0) + return err; + /* this can be a very slow function !!! */ + usleep_range(400 * val, 20000); + return lola_codec_flush(chip); +} + +/* + * Clock widget handling + */ + +int __devinit lola_init_clock_widget(struct lola *chip, int nid) +{ + unsigned int val; + int i, j, nitems, nb_verbs, idx, idx_list; + int err; + + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + + if ((val & 0xfff00000) != 0x01f00000) { /* test SubType and Type */ + snd_printdd("No valid clock widget\n"); + return 0; + } + + chip->clock.nid = nid; + chip->clock.items = val & 0xff; + snd_printdd("clock_list nid=%x, entries=%d\n", nid, + chip->clock.items); + if (chip->clock.items > MAX_SAMPLE_CLOCK_COUNT) { + printk(KERN_ERR SFX "CLOCK_LIST too big: %d\n", + chip->clock.items); + return -EINVAL; + } + + nitems = chip->clock.items; + nb_verbs = (nitems + 3) / 4; + idx = 0; + idx_list = 0; + for (i = 0; i < nb_verbs; i++) { + unsigned int res_ex; + unsigned short items[4]; + + err = lola_codec_read(chip, nid, LOLA_VERB_GET_CLOCK_LIST, + idx, 0, &val, &res_ex); + if (err < 0) { + printk(KERN_ERR SFX "Can't read CLOCK_LIST\n"); + return -EINVAL; + } + + items[0] = val & 0xfff; + items[1] = (val >> 16) & 0xfff; + items[2] = res_ex & 0xfff; + items[3] = (res_ex >> 16) & 0xfff; + + for (j = 0; j < 4; j++) { + unsigned char type = items[j] >> 8; + unsigned int freq = items[j] & 0xff; + int format = LOLA_CLOCK_FORMAT_NONE; + bool add_clock = true; + if (type == LOLA_CLOCK_TYPE_INTERNAL) { + freq = sample_rate_convert(freq); + if (freq < chip->sample_rate_min) + add_clock = false; + else if (freq == 48000) { + chip->clock.cur_index = idx_list; + chip->clock.cur_freq = 48000; + chip->clock.cur_valid = true; + } + } else if (type == LOLA_CLOCK_TYPE_VIDEO) { + freq = sample_rate_convert(freq); + if (freq < chip->sample_rate_min) + add_clock = false; + /* video clock has a format (0:NTSC, 1:PAL)*/ + if (items[j] & 0x80) + format = LOLA_CLOCK_FORMAT_NTSC; + else + format = LOLA_CLOCK_FORMAT_PAL; + } + if (add_clock) { + struct lola_sample_clock *sc; + sc = &chip->clock.sample_clock[idx_list]; + sc->type = type; + sc->format = format; + sc->freq = freq; + /* keep the index used with the board */ + chip->clock.idx_lookup[idx_list] = idx; + idx_list++; + } else { + chip->clock.items--; + } + if (++idx >= nitems) + break; + } + } + return 0; +} + +/* enable unsolicited events of the clock widget */ +int lola_enable_clock_events(struct lola *chip) +{ + unsigned int res; + int err; + + err = lola_codec_read(chip, chip->clock.nid, + LOLA_VERB_SET_UNSOLICITED_ENABLE, + LOLA_UNSOLICITED_ENABLE | LOLA_UNSOLICITED_TAG, + 0, &res, NULL); + if (err < 0) + return err; + if (res) { + printk(KERN_WARNING SFX "error in enable_clock_events %d\n", + res); + return -EINVAL; + } + return 0; +} + +int lola_set_clock_index(struct lola *chip, unsigned int idx) +{ + unsigned int res; + int err; + + err = lola_codec_read(chip, chip->clock.nid, + LOLA_VERB_SET_CLOCK_SELECT, + chip->clock.idx_lookup[idx], + 0, &res, NULL); + if (err < 0) + return err; + if (res) { + printk(KERN_WARNING SFX "error in set_clock %d\n", res); + return -EINVAL; + } + return 0; +} + +bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val) +{ + unsigned int tag; + + /* the current EXTERNAL clock information gets updated by interrupt + * with an unsolicited response + */ + if (!val) + return false; + tag = (val >> LOLA_UNSOL_RESP_TAG_OFFSET) & LOLA_UNSOLICITED_TAG_MASK; + if (tag != LOLA_UNSOLICITED_TAG) + return false; + + /* only for current = external clocks */ + if (chip->clock.sample_clock[chip->clock.cur_index].type != + LOLA_CLOCK_TYPE_INTERNAL) { + chip->clock.cur_freq = sample_rate_convert(val & 0x7f); + chip->clock.cur_valid = (val & 0x100) != 0; + } + return true; +} + +int lola_set_clock(struct lola *chip, int idx) +{ + int freq = 0; + bool valid = false; + + if (idx == chip->clock.cur_index) { + /* current clock is allowed */ + freq = chip->clock.cur_freq; + valid = chip->clock.cur_valid; + } else if (chip->clock.sample_clock[idx].type == + LOLA_CLOCK_TYPE_INTERNAL) { + /* internal clocks allowed */ + freq = chip->clock.sample_clock[idx].freq; + valid = true; + } + + if (!freq || !valid) + return -EINVAL; + + if (!check_gran_clock_compatibility(chip, chip->granularity, freq)) + return -EINVAL; + + if (idx != chip->clock.cur_index) { + int err = lola_set_clock_index(chip, idx); + if (err < 0) + return err; + /* update new settings */ + chip->clock.cur_index = idx; + chip->clock.cur_freq = freq; + chip->clock.cur_valid = true; + } + return 0; +} + +int lola_set_sample_rate(struct lola *chip, int rate) +{ + int i; + + if (chip->clock.cur_freq == rate && chip->clock.cur_valid) + return 0; + /* search for new dwClockIndex */ + for (i = 0; i < chip->clock.items; i++) { + if (chip->clock.sample_clock[i].type == LOLA_CLOCK_TYPE_INTERNAL && + chip->clock.sample_clock[i].freq == rate) + break; + } + if (i >= chip->clock.items) + return -EINVAL; + return lola_set_clock(chip, i); +} + diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c new file mode 100644 index 00000000000..0d09689d7b8 --- /dev/null +++ b/sound/pci/lola/lola_mixer.c @@ -0,0 +1,826 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License along with + * this program; if not, write to the Free Software Foundation, Inc., 59 + * Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "lola.h" + +static int __devinit lola_init_pin(struct lola *chip, struct lola_pin *pin, + int dir, int nid) +{ + unsigned int val; + int err; + + pin->nid = nid; + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + val &= 0x00f00fff; /* test TYPE and bits 0..11 */ + if (val == 0x00400200) /* Type = 4, Digital = 1 */ + pin->is_analog = false; + else if (val == 0x0040000a && dir == CAPT) /* Dig=0, InAmp/ovrd */ + pin->is_analog = true; + else if (val == 0x0040000c && dir == PLAY) /* Dig=0, OutAmp/ovrd */ + pin->is_analog = true; + else { + printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n", val, nid); + return -EINVAL; + } + + /* analog parameters only following, so continue in case of Digital pin + */ + if (!pin->is_analog) + return 0; + + if (dir == PLAY) + err = lola_read_param(chip, nid, LOLA_PAR_AMP_OUT_CAP, &val); + else + err = lola_read_param(chip, nid, LOLA_PAR_AMP_IN_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read AMP-caps for 0x%x\n", nid); + return err; + } + + pin->amp_mute = LOLA_AMP_MUTE_CAPABLE(val); + pin->amp_step_size = LOLA_AMP_STEP_SIZE(val); + pin->amp_num_steps = LOLA_AMP_NUM_STEPS(val); + if (pin->amp_num_steps) { + /* zero as mute state */ + pin->amp_num_steps++; + pin->amp_step_size++; + } + pin->amp_offset = LOLA_AMP_OFFSET(val); + + err = lola_codec_read(chip, nid, LOLA_VERB_GET_MAX_LEVEL, 0, 0, &val, + NULL); + if (err < 0) { + printk(KERN_ERR SFX "Can't get MAX_LEVEL 0x%x\n", nid); + return err; + } + pin->max_level = val & 0x3ff; /* 10 bits */ + + pin->config_default_reg = 0; + pin->fixed_gain_list_len = 0; + pin->cur_gain_step = 0; + + return 0; +} + +int __devinit lola_init_pins(struct lola *chip, int dir, int *nidp) +{ + int i, err, nid; + nid = *nidp; + for (i = 0; i < chip->pin[dir].num_pins; i++, nid++) { + err = lola_init_pin(chip, &chip->pin[dir].pins[i], dir, nid); + if (err < 0) + return err; + } + *nidp = nid; + return 0; +} + +void lola_free_mixer(struct lola *chip) +{ + if (chip->mixer.array_saved) + vfree(chip->mixer.array_saved); +} + +int __devinit lola_init_mixer_widget(struct lola *chip, int nid) +{ + unsigned int val; + int err; + + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + + if ((val & 0xfff00000) != 0x02f00000) { /* test SubType and Type */ + snd_printdd("No valid mixer widget\n"); + return 0; + } + + chip->mixer.nid = nid; + chip->mixer.caps = val; + chip->mixer.array = (struct lola_mixer_array __iomem *) + (chip->bar[BAR1].remap_addr + LOLA_BAR1_SOURCE_GAIN_ENABLE); + + /* reserve memory to copy mixer data for sleep mode transitions */ + chip->mixer.array_saved = vmalloc(sizeof(struct lola_mixer_array)); + + /* mixer matrix sources are physical input data and play streams */ + chip->mixer.src_stream_outs = chip->pcm[PLAY].num_streams; + chip->mixer.src_phys_ins = chip->pin[CAPT].num_pins; + + /* mixer matrix destinations are record streams and physical output */ + chip->mixer.dest_stream_ins = chip->pcm[CAPT].num_streams; + chip->mixer.dest_phys_outs = chip->pin[PLAY].num_pins; + + /* mixer matrix can have unused areas between PhysIn and + * Play or Record and PhysOut zones + */ + chip->mixer.src_stream_out_ofs = chip->mixer.src_phys_ins + + LOLA_MIXER_SRC_INPUT_PLAY_SEPARATION(val); + chip->mixer.dest_phys_out_ofs = chip->mixer.dest_stream_ins + + LOLA_MIXER_DEST_REC_OUTPUT_SEPATATION(val); + + /* example : MixerMatrix of LoLa881 + * 0-------8------16-------8------16 + * | | | | | + * | INPUT | | INPUT | | + * | -> |unused | -> |unused | + * | RECORD| | OUTPUT| | + * | | | | | + * 8-------------------------------- + * | | | | | + * | | | | | + * |unused |unused |unused |unused | + * | | | | | + * | | | | | + * 16------------------------------- + * | | | | | + * | PLAY | | PLAY | | + * | -> |unused | -> |unused | + * | RECORD| | OUTPUT| | + * | | | | | + * 8-------------------------------- + * | | | | | + * | | | | | + * |unused |unused |unused |unused | + * | | | | | + * | | | | | + * 16------------------------------- + */ + if (chip->mixer.src_stream_out_ofs > MAX_AUDIO_INOUT_COUNT || + chip->mixer.dest_phys_out_ofs > MAX_STREAM_IN_COUNT) { + printk(KERN_ERR SFX "Invalid mixer widget size\n"); + return -EINVAL; + } + + chip->mixer.src_mask = ((1U << chip->mixer.src_phys_ins) - 1) | + (((1U << chip->mixer.src_stream_outs) - 1) + << chip->mixer.src_stream_out_ofs); + chip->mixer.dest_mask = ((1U << chip->mixer.dest_stream_ins) - 1) | + (((1U << chip->mixer.dest_phys_outs) - 1) + << chip->mixer.dest_phys_out_ofs); + + return 0; +} + +static int lola_mixer_set_src_gain(struct lola *chip, unsigned int id, + unsigned short gain, bool on) +{ + unsigned int oldval, val; + + if (!(chip->mixer.src_mask & (1 << id))) + return -EINVAL; + writew(gain, &chip->mixer.array->src_gain[id]); + oldval = val = readl(&chip->mixer.array->src_gain_enable); + if (on) + val |= (1 << id); + else + val &= ~(1 << id); + writel(val, &chip->mixer.array->src_gain_enable); + lola_codec_flush(chip); + /* inform micro-controller about the new source gain */ + return lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, id, 0); +} + +#if 0 /* not used */ +static int lola_mixer_set_src_gains(struct lola *chip, unsigned int mask, + unsigned short *gains) +{ + int i; + + if ((chip->mixer.src_mask & mask) != mask) + return -EINVAL; + for (i = 0; i < LOLA_MIXER_DIM; i++) { + if (mask & (1 << i)) { + writew(*gains, &chip->mixer.array->src_gain[i]); + gains++; + } + } + writel(mask, &chip->mixer.array->src_gain_enable); + lola_codec_flush(chip); + if (chip->mixer.caps & LOLA_PEAK_METER_CAN_AGC_MASK) { + /* update for all srcs at once */ + return lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, 0x80, 0); + } + /* update manually */ + for (i = 0; i < LOLA_MIXER_DIM; i++) { + if (mask & (1 << i)) { + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, i, 0); + } + } + return 0; +} +#endif /* not used */ + +static int lola_mixer_set_mapping_gain(struct lola *chip, + unsigned int src, unsigned int dest, + unsigned short gain, bool on) +{ + unsigned int val; + + if (!(chip->mixer.src_mask & (1 << src)) || + !(chip->mixer.dest_mask & (1 << dest))) + return -EINVAL; + if (on) + writew(gain, &chip->mixer.array->dest_mix_gain[dest][src]); + val = readl(&chip->mixer.array->dest_mix_gain_enable[dest]); + if (on) + val |= (1 << src); + else + val &= ~(1 << src); + writel(val, &chip->mixer.array->dest_mix_gain_enable[dest]); + lola_codec_flush(chip); + return lola_codec_write(chip, chip->mixer.nid, LOLA_VERB_SET_MIX_GAIN, + src, dest); +} + +static int lola_mixer_set_dest_gains(struct lola *chip, unsigned int id, + unsigned int mask, unsigned short *gains) +{ + int i; + + if (!(chip->mixer.dest_mask & (1 << id)) || + (chip->mixer.src_mask & mask) != mask) + return -EINVAL; + for (i = 0; i < LOLA_MIXER_DIM; i++) { + if (mask & (1 << i)) { + writew(*gains, &chip->mixer.array->dest_mix_gain[id][i]); + gains++; + } + } + writel(mask, &chip->mixer.array->dest_mix_gain_enable[id]); + lola_codec_flush(chip); + /* update for all dests at once */ + return lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_DESTINATION_GAIN, id, 0); +} + +/* + */ + +static int set_analog_volume(struct lola *chip, int dir, + unsigned int idx, unsigned int val, + bool external_call); + +int lola_setup_all_analog_gains(struct lola *chip, int dir, bool mute) +{ + struct lola_pin *pin; + int idx, max_idx; + + pin = chip->pin[dir].pins; + max_idx = chip->pin[dir].num_pins; + for (idx = 0; idx < max_idx; idx++) { + if (pin[idx].is_analog) { + unsigned int val = mute ? 0 : pin[idx].cur_gain_step; + /* set volume and do not save the value */ + set_analog_volume(chip, dir, idx, val, false); + } + } + return lola_codec_flush(chip); +} + +void lola_save_mixer(struct lola *chip) +{ + /* mute analog output */ + if (chip->mixer.array_saved) { + /* store contents of mixer array */ + memcpy_fromio(chip->mixer.array_saved, chip->mixer.array, + sizeof(*chip->mixer.array)); + } + lola_setup_all_analog_gains(chip, PLAY, true); /* output mute */ +} + +void lola_restore_mixer(struct lola *chip) +{ + int i; + + /*lola_reset_setups(chip);*/ + if (chip->mixer.array_saved) { + /* restore contents of mixer array */ + memcpy_toio(chip->mixer.array, chip->mixer.array_saved, + sizeof(*chip->mixer.array)); + /* inform micro-controller about all restored values + * and ignore return values + */ + for (i = 0; i < chip->mixer.src_phys_ins; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, + i, 0); + for (i = 0; i < chip->mixer.src_stream_outs; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_SOURCE_GAIN, + chip->mixer.src_stream_out_ofs + i, 0); + for (i = 0; i < chip->mixer.dest_stream_ins; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_DESTINATION_GAIN, + i, 0); + for (i = 0; i < chip->mixer.dest_phys_outs; i++) + lola_codec_write(chip, chip->mixer.nid, + LOLA_VERB_SET_DESTINATION_GAIN, + chip->mixer.dest_phys_out_ofs + i, 0); + lola_codec_flush(chip); + } +} + +/* + */ + +static int set_analog_volume(struct lola *chip, int dir, + unsigned int idx, unsigned int val, + bool external_call) +{ + struct lola_pin *pin; + int err; + + if (idx >= chip->pin[dir].num_pins) + return -EINVAL; + pin = &chip->pin[dir].pins[idx]; + if (!pin->is_analog || pin->amp_num_steps <= val) + return -EINVAL; + if (external_call && pin->cur_gain_step == val) + return 0; + if (external_call) + lola_codec_flush(chip); + err = lola_codec_write(chip, pin->nid, + LOLA_VERB_SET_AMP_GAIN_MUTE, val, 0); + if (err < 0) + return err; + if (external_call) + pin->cur_gain_step = val; + return 0; +} + +int lola_set_src_config(struct lola *chip, unsigned int src_mask, bool update) +{ + int ret = 0; + int success = 0; + int n, err; + + /* SRC can be activated and the dwInputSRCMask is valid? */ + if ((chip->input_src_caps_mask & src_mask) != src_mask) + return -EINVAL; + /* handle all even Inputs - SRC is a stereo setting !!! */ + for (n = 0; n < chip->pin[CAPT].num_pins; n += 2) { + unsigned int mask = 3U << n; /* handle the stereo case */ + unsigned int new_src, src_state; + if (!(chip->input_src_caps_mask & mask)) + continue; + /* if one IO needs SRC, both stereo IO will get SRC */ + new_src = (src_mask & mask) != 0; + if (update) { + src_state = (chip->input_src_mask & mask) != 0; + if (src_state == new_src) + continue; /* nothing to change for this IO */ + } + err = lola_codec_write(chip, chip->pcm[CAPT].streams[n].nid, + LOLA_VERB_SET_SRC, new_src, 0); + if (!err) + success++; + else + ret = err; + } + if (success) + ret = lola_codec_flush(chip); + if (!ret) + chip->input_src_mask = src_mask; + return ret; +} + +/* + */ +static int init_mixer_values(struct lola *chip) +{ + int i; + + /* all src on */ + lola_set_src_config(chip, (1 << chip->pin[CAPT].num_pins) - 1, false); + + /* clear all matrix */ + memset_io(chip->mixer.array, 0, sizeof(*chip->mixer.array)); + /* set src gain to 0dB */ + for (i = 0; i < chip->mixer.src_phys_ins; i++) + lola_mixer_set_src_gain(chip, i, 336, true); /* 0dB */ + for (i = 0; i < chip->mixer.src_stream_outs; i++) + lola_mixer_set_src_gain(chip, + i + chip->mixer.src_stream_out_ofs, + 336, true); /* 0dB */ + /* set 1:1 dest gain */ + for (i = 0; i < chip->mixer.dest_stream_ins; i++) { + int src = i % chip->mixer.src_phys_ins; + lola_mixer_set_mapping_gain(chip, src, i, 336, true); + } + for (i = 0; i < chip->mixer.src_stream_outs; i++) { + int src = chip->mixer.src_stream_out_ofs + i; + int dst = chip->mixer.dest_phys_out_ofs + + i % chip->mixer.dest_phys_outs; + lola_mixer_set_mapping_gain(chip, src, dst, 336, true); + } + return 0; +} + +/* + * analog mixer control element + */ +static int lola_analog_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chip->pin[dir].num_pins; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = chip->pin[dir].pins[0].amp_num_steps; + return 0; +} + +static int lola_analog_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + int i; + + for (i = 0; i < chip->pin[dir].num_pins; i++) + ucontrol->value.integer.value[i] = + chip->pin[dir].pins[i].cur_gain_step; + return 0; +} + +static int lola_analog_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + int i, err; + + for (i = 0; i < chip->pin[dir].num_pins; i++) { + err = set_analog_volume(chip, dir, i, + ucontrol->value.integer.value[i], + true); + if (err < 0) + return err; + } + return 0; +} + +static int lola_analog_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int dir = kcontrol->private_value; + unsigned int val1, val2; + struct lola_pin *pin; + + if (size < 4 * sizeof(unsigned int)) + return -ENOMEM; + pin = &chip->pin[dir].pins[0]; + + val2 = pin->amp_step_size * 25; + val1 = -1 * (int)pin->amp_offset * (int)val2; +#ifdef TLV_DB_SCALE_MUTE + val2 |= TLV_DB_SCALE_MUTE; +#endif + if (put_user(SNDRV_CTL_TLVT_DB_SCALE, tlv)) + return -EFAULT; + if (put_user(2 * sizeof(unsigned int), tlv + 1)) + return -EFAULT; + if (put_user(val1, tlv + 2)) + return -EFAULT; + if (put_user(val2, tlv + 3)) + return -EFAULT; + return 0; +} + +static struct snd_kcontrol_new lola_analog_mixer __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK), + .info = lola_analog_vol_info, + .get = lola_analog_vol_get, + .put = lola_analog_vol_put, + .tlv.c = lola_analog_vol_tlv, +}; + +static int __devinit create_analog_mixer(struct lola *chip, int dir, char *name) +{ + if (!chip->pin[dir].num_pins) + return 0; + lola_analog_mixer.name = name; + lola_analog_mixer.private_value = dir; + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_analog_mixer, chip)); +} + +/* + */ +static int lola_input_src_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = chip->pin[CAPT].num_pins; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int lola_input_src_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int i; + + for (i = 0; i < chip->pin[CAPT].num_pins; i++) + ucontrol->value.integer.value[i] = + !!(chip->input_src_mask & (1 << i)); + return 0; +} + +static int lola_input_src_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + int i; + unsigned int mask; + + mask = 0; + for (i = 0; i < chip->pin[CAPT].num_pins; i++) + if (ucontrol->value.integer.value[i]) + mask |= 1 << i; + return lola_set_src_config(chip, mask, true); +} + +static struct snd_kcontrol_new lola_input_src_mixer __devinitdata = { + .name = "Analog Capture Switch", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = lola_input_src_info, + .get = lola_input_src_get, + .put = lola_input_src_put, +}; + +static int __devinit create_input_src_mixer(struct lola *chip) +{ + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_input_src_mixer, chip)); +} + +/* + * src gain mixer + */ +static int lola_src_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int count = (kcontrol->private_value >> 8) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = count; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 409; + return 0; +} + +static int lola_src_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int ofs = kcontrol->private_value & 0xff; + unsigned int count = (kcontrol->private_value >> 8) & 0xff; + unsigned int mask, i; + + mask = readl(&chip->mixer.array->src_gain_enable); + for (i = 0; i < count; i++) { + unsigned int idx = ofs + i; + unsigned short val; + if (!(chip->mixer.src_mask & (1 << idx))) + return -EINVAL; + if (mask & (1 << idx)) + val = readw(&chip->mixer.array->src_gain[idx]) + 1; + else + val = 0; + ucontrol->value.integer.value[i] = val; + } + return 0; +} + +static int lola_src_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int ofs = kcontrol->private_value & 0xff; + unsigned int count = (kcontrol->private_value >> 8) & 0xff; + int i, err; + + for (i = 0; i < count; i++) { + unsigned int idx = ofs + i; + unsigned short val = ucontrol->value.integer.value[i]; + if (val) + val--; + err = lola_mixer_set_src_gain(chip, idx, val, !!val); + if (err < 0) + return err; + } + return 0; +} + +/* raw value: 0 = -84dB, 336 = 0dB, 408=18dB, incremented 1 for mute */ +static const DECLARE_TLV_DB_SCALE(lola_src_gain_tlv, -8425, 25, 1); + +static struct snd_kcontrol_new lola_src_gain_mixer __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .info = lola_src_gain_info, + .get = lola_src_gain_get, + .put = lola_src_gain_put, + .tlv.p = lola_src_gain_tlv, +}; + +static int __devinit create_src_gain_mixer(struct lola *chip, + int num, int ofs, char *name) +{ + lola_src_gain_mixer.name = name; + lola_src_gain_mixer.private_value = ofs + (num << 8); + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_src_gain_mixer, chip)); +} + +/* + * destination gain (matrix-like) mixer + */ +static int lola_dest_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int src_num = (kcontrol->private_value >> 8) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = src_num; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 433; + return 0; +} + +static int lola_dest_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int src_ofs = kcontrol->private_value & 0xff; + unsigned int src_num = (kcontrol->private_value >> 8) & 0xff; + unsigned int dst_ofs = (kcontrol->private_value >> 16) & 0xff; + unsigned int dst, mask, i; + + dst = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + dst_ofs; + mask = readl(&chip->mixer.array->dest_mix_gain_enable[dst]); + for (i = 0; i < src_num; i++) { + unsigned int src = src_ofs + i; + unsigned short val; + if (!(chip->mixer.src_mask & (1 << src))) + return -EINVAL; + if (mask & (1 << dst)) + val = readw(&chip->mixer.array->dest_mix_gain[dst][src]) + 1; + else + val = 0; + ucontrol->value.integer.value[i] = val; + } + return 0; +} + +static int lola_dest_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct lola *chip = snd_kcontrol_chip(kcontrol); + unsigned int src_ofs = kcontrol->private_value & 0xff; + unsigned int src_num = (kcontrol->private_value >> 8) & 0xff; + unsigned int dst_ofs = (kcontrol->private_value >> 16) & 0xff; + unsigned int dst, mask; + unsigned short gains[MAX_STREAM_COUNT]; + int i, num; + + mask = 0; + num = 0; + for (i = 0; i < src_num; i++) { + unsigned short val = ucontrol->value.integer.value[i]; + if (val) { + gains[num++] = val - 1; + mask |= 1 << i; + } + } + mask <<= src_ofs; + dst = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + dst_ofs; + return lola_mixer_set_dest_gains(chip, dst, mask, gains); +} + +static const DECLARE_TLV_DB_SCALE(lola_dest_gain_tlv, -8425, 25, 1); + +static struct snd_kcontrol_new lola_dest_gain_mixer __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .info = lola_dest_gain_info, + .get = lola_dest_gain_get, + .put = lola_dest_gain_put, + .tlv.p = lola_dest_gain_tlv, +}; + +static int __devinit create_dest_gain_mixer(struct lola *chip, + int src_num, int src_ofs, + int num, int ofs, char *name) +{ + lola_dest_gain_mixer.count = num; + lola_dest_gain_mixer.name = name; + lola_dest_gain_mixer.private_value = + src_ofs + (src_num << 8) + (ofs << 16) + (num << 24); + return snd_ctl_add(chip->card, + snd_ctl_new1(&lola_dest_gain_mixer, chip)); +} + +/* + */ +int __devinit lola_create_mixer(struct lola *chip) +{ + int err; + + err = create_analog_mixer(chip, PLAY, "Analog Playback Volume"); + if (err < 0) + return err; + err = create_analog_mixer(chip, CAPT, "Analog Capture Volume"); + if (err < 0) + return err; + err = create_input_src_mixer(chip); + if (err < 0) + return err; + err = create_src_gain_mixer(chip, chip->mixer.src_phys_ins, 0, + "Line Source Gain Volume"); + if (err < 0) + return err; + err = create_src_gain_mixer(chip, chip->mixer.src_stream_outs, + chip->mixer.src_stream_out_ofs, + "Stream Source Gain Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_phys_ins, 0, + chip->mixer.dest_stream_ins, 0, + "Line Capture Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_stream_outs, + chip->mixer.src_stream_out_ofs, + chip->mixer.dest_stream_ins, 0, + "Stream-Loopback Capture Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_phys_ins, 0, + chip->mixer.dest_phys_outs, + chip->mixer.dest_phys_out_ofs, + "Line-Loopback Playback Volume"); + if (err < 0) + return err; + err = create_dest_gain_mixer(chip, + chip->mixer.src_stream_outs, + chip->mixer.src_stream_out_ofs, + chip->mixer.dest_phys_outs, + chip->mixer.dest_phys_out_ofs, + "Stream Playback Volume"); + if (err < 0) + return err; + + return init_mixer_values(chip); +} diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c new file mode 100644 index 00000000000..bf17f921d45 --- /dev/null +++ b/sound/pci/lola/lola_pcm.c @@ -0,0 +1,607 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License along with + * this program; if not, write to the Free Software Foundation, Inc., 59 + * Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#include +#include +#include +#include +#include +#include +#include "lola.h" + +#define BDL_SIZE 4096 +#define LOLA_MAX_BDL_ENTRIES (BDL_SIZE / 16) +#define LOLA_MAX_FRAG 32 +#define LOLA_MAX_BUF_SIZE (1024*1024*1024) + +static struct lola_pcm *lola_get_pcm(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + return &chip->pcm[substream->stream]; +} + +static struct lola_stream *lola_get_stream(struct snd_pcm_substream *substream) +{ + struct lola_pcm *pcm = lola_get_pcm(substream); + unsigned int idx = substream->number; + return &pcm->streams[idx]; +} + +static unsigned int lola_get_lrc(struct lola *chip) +{ + return lola_readl(chip, BAR1, LRC); +} + +static unsigned int lola_get_tstamp(struct lola *chip, bool quick_no_sync) +{ + unsigned int tstamp = lola_get_lrc(chip) >> 8; + if (chip->granularity) { + unsigned int wait_banks = quick_no_sync ? 0 : 8; + tstamp += (wait_banks + 1) * chip->granularity - 1; + tstamp -= tstamp % chip->granularity; + } + return tstamp << 8; +} + +/* clear any pending interrupt status */ +static void lola_stream_clear_pending_irq(struct lola *chip, + struct lola_stream *str) +{ + unsigned int val = lola_dsd_read(chip, str->dsd, STS); + val &= LOLA_DSD_STS_DESE | LOLA_DSD_STS_BCIS; + if (val) + lola_dsd_write(chip, str->dsd, STS, val); +} + +static void lola_stream_start(struct lola *chip, struct lola_stream *str, + unsigned int tstamp) +{ + lola_stream_clear_pending_irq(chip, str); + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_SRUN | + LOLA_DSD_CTL_IOCE | + LOLA_DSD_CTL_DEIE | + LOLA_DSD_CTL_VLRCV | + tstamp); +} + +static void lola_stream_stop(struct lola *chip, struct lola_stream *str, + unsigned int tstamp) +{ + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_IOCE | + LOLA_DSD_CTL_DEIE | + LOLA_DSD_CTL_VLRCV | + tstamp); + lola_stream_clear_pending_irq(chip, str); +} + +static void lola_stream_clear(struct lola *chip, struct lola_stream *str) +{ + lola_dsd_write(chip, str->dsd, CTL, 0); + lola_stream_clear_pending_irq(chip, str); +} + +static void wait_for_srst_clear(struct lola *chip, struct lola_stream *str) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(50); + while (time_before(jiffies, end_time)) { + unsigned int val; + val = lola_dsd_read(chip, str->dsd, CTL); + if (!(val & LOLA_DSD_CTL_SRST)) + return; + msleep(1); + } + printk(KERN_WARNING SFX "SRST not clear (stream %d)\n", str->dsd); +} + +static void lola_stream_reset(struct lola *chip, struct lola_stream *str) +{ + lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRST); + lola_dsd_write(chip, str->dsd, LVI, 0); + lola_dsd_write(chip, str->dsd, BDPU, 0); + lola_dsd_write(chip, str->dsd, BDPL, 0); + wait_for_srst_clear(chip, str); +} + +static int lola_stream_wait_for_fifo_ready(struct lola *chip, + struct lola_stream *str) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(50); + while (time_before(jiffies, end_time)) { + unsigned int val = lola_dsd_read(chip, str->dsd, STS); + if (val & LOLA_DSD_STS_FIFORDY) + return 0; + msleep(1); + } + printk(KERN_WARNING SFX "FIFO not ready (stream %d)\n", str->dsd); + return -EIO; +} + +static struct snd_pcm_hardware lola_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_FLOAT_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = LOLA_MAX_BUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = LOLA_MAX_BUF_SIZE / 2, + .periods_min = 2, + .periods_max = LOLA_MAX_FRAG, + .fifo_size = 0, +}; + +static int lola_pcm_open(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_pcm *pcm = lola_get_pcm(substream); + struct lola_stream *str = lola_get_stream(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + mutex_lock(&chip->open_mutex); + if (str->opened) { + mutex_unlock(&chip->open_mutex); + return -EBUSY; + } + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + PAGE_SIZE, &str->bdl); + if (err < 0) { + mutex_unlock(&chip->open_mutex); + printk(KERN_ERR SFX "Can't allocate BDL\n"); + return err; + } + str->substream = substream; + str->master = NULL; + str->opened = 1; + runtime->hw = lola_pcm_hw; + runtime->hw.channels_max = pcm->num_streams - str->index; + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + 128); + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + 128); + mutex_unlock(&chip->open_mutex); + return 0; +} + +static void lola_cleanup_slave_streams(struct lola_pcm *pcm, + struct lola_stream *str) +{ + int i; + for (i = str->index + 1; i < pcm->num_streams; i++) { + struct lola_stream *s = &pcm->streams[i]; + if (s->master != str) + break; + s->master = NULL; + s->opened = 0; + } +} + +static int lola_pcm_close(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_stream *str = lola_get_stream(substream); + + mutex_lock(&chip->open_mutex); + if (str->substream == substream) { + str->substream = NULL; + str->opened = 0; + } + snd_dma_free_pages(&str->bdl); + mutex_unlock(&chip->open_mutex); + return 0; +} + +static int lola_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct lola_stream *str = lola_get_stream(substream); + + str->bufsize = 0; + str->period_bytes = 0; + str->format_verb = 0; + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int lola_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_pcm *pcm = lola_get_pcm(substream); + struct lola_stream *str = lola_get_stream(substream); + + mutex_lock(&chip->open_mutex); + lola_stream_reset(chip, str); + lola_cleanup_slave_streams(pcm, str); + mutex_unlock(&chip->open_mutex); + return snd_pcm_lib_free_pages(substream); +} + +/* + * set up a BDL entry + */ +static int setup_bdle(struct snd_pcm_substream *substream, + struct lola_stream *str, u32 **bdlp, + int ofs, int size) +{ + u32 *bdl = *bdlp; + + while (size > 0) { + dma_addr_t addr; + int chunk; + + if (str->frags >= LOLA_MAX_BDL_ENTRIES) + return -EINVAL; + + addr = snd_pcm_sgbuf_get_addr(substream, ofs); + /* program the address field of the BDL entry */ + bdl[0] = cpu_to_le32((u32)addr); + bdl[1] = cpu_to_le32(upper_32_bits(addr)); + /* program the size field of the BDL entry */ + chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size); + bdl[2] = cpu_to_le32(chunk); + /* program the IOC to enable interrupt + * only when the whole fragment is processed + */ + size -= chunk; + bdl[3] = size ? 0 : cpu_to_le32(0x01); + bdl += 4; + str->frags++; + ofs += chunk; + } + *bdlp = bdl; + return ofs; +} + +/* + * set up BDL entries + */ +static int lola_setup_periods(struct lola *chip, + struct snd_pcm_substream *substream, + struct lola_stream *str) +{ + u32 *bdl; + int i, ofs, periods, period_bytes; + + period_bytes = str->period_bytes; + periods = str->bufsize / period_bytes; + + /* program the initial BDL entries */ + bdl = (u32 *)str->bdl.area; + ofs = 0; + str->frags = 0; + for (i = 0; i < periods; i++) { + ofs = setup_bdle(substream, str, &bdl, ofs, period_bytes); + if (ofs < 0) + goto error; + } + return 0; + + error: + snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n", + str->bufsize, period_bytes); + return -EINVAL; +} + +static unsigned int lola_get_format_verb(struct snd_pcm_substream *substream) +{ + unsigned int verb; + + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + verb = 0x00000000; + break; + case SNDRV_PCM_FORMAT_S24_LE: + verb = 0x00000200; + break; + case SNDRV_PCM_FORMAT_S32_LE: + verb = 0x00000300; + break; + case SNDRV_PCM_FORMAT_FLOAT_LE: + verb = 0x00001300; + break; + default: + return 0; + } + verb |= substream->runtime->channels; + return verb; +} + +static int lola_set_stream_config(struct lola *chip, + struct lola_stream *str, + int channels) +{ + int i, err; + unsigned int verb, val; + + /* set format info for all channels + * (with only one command for the first channel) + */ + err = lola_codec_read(chip, str->nid, LOLA_VERB_SET_STREAM_FORMAT, + str->format_verb, 0, &val, NULL); + if (err < 0) { + printk(KERN_ERR SFX "Cannot set stream format 0x%x\n", + str->format_verb); + return err; + } + + /* update stream - channel config */ + for (i = 0; i < channels; i++) { + verb = (str->index << 6) | i; + err = lola_codec_read(chip, str[i].nid, + LOLA_VERB_SET_CHANNEL_STREAMID, 0, verb, + &val, NULL); + if (err < 0) { + printk(KERN_ERR SFX "Cannot set stream channel %d\n", i); + return err; + } + } + return 0; +} + +/* + * set up the SD for streaming + */ +static int lola_setup_controller(struct lola *chip, struct lola_stream *str) +{ + /* make sure the run bit is zero for SD */ + lola_stream_clear(chip, str); + /* set up BDL */ + lola_dsd_write(chip, str->dsd, BDPL, (u32)str->bdl.addr); + lola_dsd_write(chip, str->dsd, BDPU, upper_32_bits(str->bdl.addr)); + /* program the stream LVI (last valid index) of the BDL */ + lola_dsd_write(chip, str->dsd, LVI, str->frags - 1); + lola_stream_stop(chip, str, lola_get_tstamp(chip, false)); + lola_stream_wait_for_fifo_ready(chip, str); + + return 0; +} + +static int lola_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_pcm *pcm = lola_get_pcm(substream); + struct lola_stream *str = lola_get_stream(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int bufsize, period_bytes, format_verb; + int i, err; + + mutex_lock(&chip->open_mutex); + lola_stream_reset(chip, str); + lola_cleanup_slave_streams(pcm, str); + if (str->index + runtime->channels >= pcm->num_streams) { + mutex_unlock(&chip->open_mutex); + return -EINVAL; + } + for (i = 1; i < runtime->channels; i++) { + str[i].master = str; + str[i].opened = 1; + } + mutex_unlock(&chip->open_mutex); + + bufsize = snd_pcm_lib_buffer_bytes(substream); + period_bytes = snd_pcm_lib_period_bytes(substream); + format_verb = lola_get_format_verb(substream); + + if (bufsize != str->bufsize || + period_bytes != str->period_bytes || + format_verb != str->format_verb) { + str->bufsize = bufsize; + str->period_bytes = period_bytes; + str->format_verb = format_verb; + err = lola_setup_periods(chip, substream, str); + if (err < 0) + return err; + } + + err = lola_set_stream_config(chip, str, runtime->channels); + if (err < 0) + return err; + + return lola_setup_controller(chip, str); +} + +static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_stream *str; + struct snd_pcm_substream *s; + unsigned int start; + unsigned int tstamp; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + start = 1; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + start = 0; + break; + default: + return -EINVAL; + } + + tstamp = lola_get_tstamp(chip, false); + spin_lock(&chip->reg_lock); + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + str = lola_get_stream(s); + if (start) + lola_stream_start(chip, str, tstamp); + else + lola_stream_stop(chip, str, tstamp); + str->running = start; + snd_pcm_trigger_done(s, substream); + } + spin_unlock(&chip->reg_lock); + return 0; +} + +static snd_pcm_uframes_t lola_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct lola *chip = snd_pcm_substream_chip(substream); + struct lola_stream *str = lola_get_stream(substream); + unsigned int pos = lola_dsd_read(chip, str->dsd, LPIB); + + if (pos >= str->bufsize) + pos = 0; + return bytes_to_frames(substream->runtime, pos); +} + +void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits) +{ + int i; + + for (i = 0; bits && i < pcm->num_streams; i++) { + if (bits & (1 << i)) { + struct lola_stream *str = &pcm->streams[i]; + if (str->substream && str->running) + snd_pcm_period_elapsed(str->substream); + bits &= ~(1 << i); + } + } +} + +static struct snd_pcm_ops lola_pcm_ops = { + .open = lola_pcm_open, + .close = lola_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = lola_pcm_hw_params, + .hw_free = lola_pcm_hw_free, + .prepare = lola_pcm_prepare, + .trigger = lola_pcm_trigger, + .pointer = lola_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + +int __devinit lola_create_pcm(struct lola *chip) +{ + struct snd_pcm *pcm; + int i, err; + + err = snd_pcm_new(chip->card, "Digigram Lola", 0, + chip->pcm[SNDRV_PCM_STREAM_PLAYBACK].num_streams, + chip->pcm[SNDRV_PCM_STREAM_CAPTURE].num_streams, + &pcm); + if (err < 0) + return err; + strlcpy(pcm->name, "Digigram Lola", sizeof(pcm->name)); + pcm->private_data = chip; + for (i = 0; i < 2; i++) { + if (chip->pcm[i].num_streams) + snd_pcm_set_ops(pcm, i, &lola_pcm_ops); + } + /* buffer pre-allocation */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, + snd_dma_pci_data(chip->pci), + 1024 * 64, 32 * 1024 * 1024); + return 0; +} + +void lola_free_pcm(struct lola *chip) +{ + /* nothing to do */ +} + +/* + */ + +static int lola_init_stream(struct lola *chip, struct lola_stream *str, + int idx, int nid, int dir) +{ + unsigned int val; + int err; + + str->nid = nid; + str->index = idx; + str->dsd = idx; + if (dir == PLAY) + str->dsd += MAX_STREAM_IN_COUNT; + err = lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read wcaps for 0x%x\n", nid); + return err; + } + if (dir == PLAY) { + /* test TYPE and bits 0..11 (no test bit9 : Digital = 0/1) */ + if ((val & 0x00f00dff) != 0x00000010) { + printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n", + val, nid); + return -EINVAL; + } + } else { + /* test TYPE and bits 0..11 (no test bit9 : Digital = 0/1) + * (bug : ignore bit8: Conn list = 0/1) + */ + if ((val & 0x00f00cff) != 0x00100010) { + printk(KERN_ERR SFX "Invalid wcaps 0x%x for 0x%x\n", + val, nid); + return -EINVAL; + } + /* test bit9:DIGITAL and bit12:SRC_PRESENT*/ + if ((val & 0x00001200) == 0x00001200) + chip->input_src_caps_mask |= (1 << idx); + } + + err = lola_read_param(chip, nid, LOLA_PAR_STREAM_FORMATS, &val); + if (err < 0) { + printk(KERN_ERR SFX "Can't read FORMATS 0x%x\n", nid); + return err; + } + val &= 3; + if (val == 3) + str->can_float = true; + if (!(val & 1)) { + printk(KERN_ERR SFX "Invalid formats 0x%x for 0x%x", val, nid); + return -EINVAL; + } + return 0; +} + +int __devinit lola_init_pcm(struct lola *chip, int dir, int *nidp) +{ + struct lola_pcm *pcm = &chip->pcm[dir]; + int i, nid, err; + + nid = *nidp; + for (i = 0; i < pcm->num_streams; i++, nid++) { + err = lola_init_stream(chip, &pcm->streams[i], i, nid, dir); + if (err < 0) + return err; + } + *nidp = nid; + return 0; +} diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c new file mode 100644 index 00000000000..247fa247717 --- /dev/null +++ b/sound/pci/lola/lola_proc.c @@ -0,0 +1,86 @@ +/* + * Support for Digigram Lola PCI-e boards + * + * Copyright (c) 2011 Takashi Iwai + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the Free + * Software Foundation; either version 2 of the License, or (at your option) + * any later version. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License along with + * this program; if not, write to the Free Software Foundation, Inc., 59 + * Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#include +#include +#include +#include +#include +#include +#include "lola.h" + +/* direct codec access for debugging */ +static void lola_proc_codec_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + char line[64]; + unsigned int id, verb, data, extdata; + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%i %i %i %i", &id, &verb, &data, &extdata) != 4) + continue; + lola_codec_read(chip, id, verb, data, extdata, + &chip->debug_res, + &chip->debug_res_ex); + } +} + +static void lola_proc_codec_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + snd_iprintf(buffer, "0x%x 0x%x\n", chip->debug_res, chip->debug_res_ex); +} + +/* + * dump some registers + */ +static void lola_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + int i; + + for (i = 0; i < 0x40; i += 4) { + snd_iprintf(buffer, "BAR0 %02x: %08x\n", i, + readl(chip->bar[BAR0].remap_addr + i)); + } + for (i = 0; i < 0x30; i += 4) { + snd_iprintf(buffer, "BAR1 %02x: %08x\n", i, + readl(chip->bar[BAR1].remap_addr + i)); + } + for (i = 0x80; i < 0xa0; i += 4) { + snd_iprintf(buffer, "BAR1 %02x: %08x\n", i, + readl(chip->bar[BAR1].remap_addr + i)); + } +} + +void __devinit lola_proc_debug_new(struct lola *chip) +{ + struct snd_info_entry *entry; + + if (!snd_card_proc_new(chip->card, "codec", &entry)) { + snd_info_set_text_ops(entry, chip, lola_proc_codec_read); + entry->mode |= S_IWUSR; + entry->c.text.write = lola_proc_codec_write; + } + if (!snd_card_proc_new(chip->card, "regs", &entry)) + snd_info_set_text_ops(entry, chip, lola_proc_regs_read); +} -- cgit v1.2.3 From 1c5d7b312f9d0e18e2051eac058257db3586f54d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:39:37 +0200 Subject: ALSA: lola - Make SRC helper global Make lola_sample_rate_convert() global so that it can be accessed from other files. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.h | 1 + sound/pci/lola/lola_clock.c | 8 ++++---- 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index 4734c7c6f83..23831a1a95a 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -499,6 +499,7 @@ int lola_set_clock_index(struct lola *chip, unsigned int idx); int lola_set_clock(struct lola *chip, int idx); int lola_set_sample_rate(struct lola *chip, int rate); bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val); +unsigned int lola_sample_rate_convert(unsigned int coded); /* mixer */ int lola_init_pins(struct lola *chip, int dir, int *nidp); diff --git a/sound/pci/lola/lola_clock.c b/sound/pci/lola/lola_clock.c index a364dc67e55..d2cce39c347 100644 --- a/sound/pci/lola/lola_clock.c +++ b/sound/pci/lola/lola_clock.c @@ -24,7 +24,7 @@ #include #include "lola.h" -static unsigned int sample_rate_convert(unsigned int coded) +unsigned int lola_sample_rate_convert(unsigned int coded) { unsigned int freq; @@ -172,7 +172,7 @@ int __devinit lola_init_clock_widget(struct lola *chip, int nid) int format = LOLA_CLOCK_FORMAT_NONE; bool add_clock = true; if (type == LOLA_CLOCK_TYPE_INTERNAL) { - freq = sample_rate_convert(freq); + freq = lola_sample_rate_convert(freq); if (freq < chip->sample_rate_min) add_clock = false; else if (freq == 48000) { @@ -181,7 +181,7 @@ int __devinit lola_init_clock_widget(struct lola *chip, int nid) chip->clock.cur_valid = true; } } else if (type == LOLA_CLOCK_TYPE_VIDEO) { - freq = sample_rate_convert(freq); + freq = lola_sample_rate_convert(freq); if (freq < chip->sample_rate_min) add_clock = false; /* video clock has a format (0:NTSC, 1:PAL)*/ @@ -263,7 +263,7 @@ bool lola_update_ext_clock_freq(struct lola *chip, unsigned int val) /* only for current = external clocks */ if (chip->clock.sample_clock[chip->clock.cur_index].type != LOLA_CLOCK_TYPE_INTERNAL) { - chip->clock.cur_freq = sample_rate_convert(val & 0x7f); + chip->clock.cur_freq = lola_sample_rate_convert(val & 0x7f); chip->clock.cur_valid = (val & 0x100) != 0; } return true; -- cgit v1.2.3 From c772bbe69a0171f12ad2adde5c150b4e211365ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:32:48 +0200 Subject: ALSA: lola - Changes in proc file The codec proc file becomes a read only that shows the codec widgets in a text form. A new proc file, codec_rw, is introduced instead for accessing the Lola verb directly by reading and writing to it. Also, regs proc file shows the contents of DSD, too. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola_proc.c | 148 +++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 142 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c index 247fa247717..9d7daf897c9 100644 --- a/sound/pci/lola/lola_proc.c +++ b/sound/pci/lola/lola_proc.c @@ -26,9 +26,126 @@ #include #include "lola.h" +static void print_audio_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid, const char *name) +{ + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x %s wcaps 0x%x\n", nid, name, val); + lola_read_param(chip, nid, LOLA_PAR_STREAM_FORMATS, &val); + snd_iprintf(buffer, " Formats: 0x%x\n", val); +} + +static void print_pin_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid, unsigned int ampcap, + const char *name) +{ + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x %s wcaps 0x%x\n", nid, name, val); + if (val == 0x00400200) + return; + lola_read_param(chip, nid, ampcap, &val); + snd_iprintf(buffer, " Amp-Caps: 0x%x\n", val); + snd_iprintf(buffer, " mute=%d, step-size=%d, steps=%d, ofs=%d\n", + LOLA_AMP_MUTE_CAPABLE(val), + LOLA_AMP_STEP_SIZE(val), + LOLA_AMP_NUM_STEPS(val), + LOLA_AMP_OFFSET(val)); + lola_codec_read(chip, nid, LOLA_VERB_GET_MAX_LEVEL, 0, 0, &val, NULL); + snd_iprintf(buffer, " Max-level: 0x%x\n", val); +} + +static void print_clock_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid) +{ + int i, j, num_clocks; + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x [Clock] wcaps 0x%x\n", nid, val); + num_clocks = val & 0xff; + for (i = 0; i < num_clocks; i += 4) { + unsigned int res_ex; + unsigned short items[4]; + const char *name; + + lola_codec_read(chip, nid, LOLA_VERB_GET_CLOCK_LIST, + i, 0, &val, &res_ex); + items[0] = val & 0xfff; + items[1] = (val >> 16) & 0xfff; + items[2] = res_ex & 0xfff; + items[3] = (res_ex >> 16) & 0xfff; + for (j = 0; j < 4; j++) { + unsigned char type = items[j] >> 8; + unsigned int freq = items[j] & 0xff; + if (i + j >= num_clocks) + break; + if (type == LOLA_CLOCK_TYPE_INTERNAL) { + name = "Internal"; + freq = lola_sample_rate_convert(freq); + } else if (type == LOLA_CLOCK_TYPE_VIDEO) { + name = "Video"; + freq = lola_sample_rate_convert(freq); + } else { + name = "Other"; + } + snd_iprintf(buffer, " Clock %d: Type %d:%s, freq=%d\n", + i + j, type, name, freq); + } + } +} + +static void print_mixer_widget(struct snd_info_buffer *buffer, + struct lola *chip, int nid) +{ + unsigned int val; + + lola_read_param(chip, nid, LOLA_PAR_AUDIO_WIDGET_CAP, &val); + snd_iprintf(buffer, "Node 0x%02x [Mixer] wcaps 0x%x\n", nid, val); +} + +static void lola_proc_codec_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct lola *chip = entry->private_data; + unsigned int val; + int i, nid; + + lola_read_param(chip, 0, LOLA_PAR_VENDOR_ID, &val); + snd_iprintf(buffer, "Vendor: 0x%08x\n", val); + lola_read_param(chip, 1, LOLA_PAR_FUNCTION_TYPE, &val); + snd_iprintf(buffer, "Function Type: %d\n", val); + lola_read_param(chip, 1, LOLA_PAR_SPECIFIC_CAPS, &val); + snd_iprintf(buffer, "Specific-Caps: 0x%08x\n", val); + snd_iprintf(buffer, " Pins-In %d, Pins-Out %d\n", + chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins); + nid = 2; + for (i = 0; i < chip->pcm[CAPT].num_streams; i++, nid++) + print_audio_widget(buffer, chip, nid, "[Audio-In]"); + for (i = 0; i < chip->pcm[PLAY].num_streams; i++, nid++) + print_audio_widget(buffer, chip, nid, "[Audio-Out]"); + for (i = 0; i < chip->pin[CAPT].num_pins; i++, nid++) + print_pin_widget(buffer, chip, nid, LOLA_PAR_AMP_IN_CAP, + "[Pin-In]"); + for (i = 0; i < chip->pin[PLAY].num_pins; i++, nid++) + print_pin_widget(buffer, chip, nid, LOLA_PAR_AMP_OUT_CAP, + "[Pin-Out]"); + if (LOLA_AFG_CLOCK_WIDGET_PRESENT(chip->lola_caps)) { + print_clock_widget(buffer, chip, nid); + nid++; + } + if (LOLA_AFG_MIXER_WIDGET_PRESENT(chip->lola_caps)) { + print_mixer_widget(buffer, chip, nid); + nid++; + } +} + /* direct codec access for debugging */ -static void lola_proc_codec_write(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) +static void lola_proc_codec_rw_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct lola *chip = entry->private_data; char line[64]; @@ -42,8 +159,8 @@ static void lola_proc_codec_write(struct snd_info_entry *entry, } } -static void lola_proc_codec_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) +static void lola_proc_codec_rw_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct lola *chip = entry->private_data; snd_iprintf(buffer, "0x%x 0x%x\n", chip->debug_res, chip->debug_res_ex); @@ -62,24 +179,43 @@ static void lola_proc_regs_read(struct snd_info_entry *entry, snd_iprintf(buffer, "BAR0 %02x: %08x\n", i, readl(chip->bar[BAR0].remap_addr + i)); } + snd_iprintf(buffer, "\n"); for (i = 0; i < 0x30; i += 4) { snd_iprintf(buffer, "BAR1 %02x: %08x\n", i, readl(chip->bar[BAR1].remap_addr + i)); } + snd_iprintf(buffer, "\n"); for (i = 0x80; i < 0xa0; i += 4) { snd_iprintf(buffer, "BAR1 %02x: %08x\n", i, readl(chip->bar[BAR1].remap_addr + i)); } + snd_iprintf(buffer, "\n"); + for (i = 0; i < 32; i++) { + snd_iprintf(buffer, "DSD %02x STS %08x\n", i, + lola_dsd_read(chip, i, STS)); + snd_iprintf(buffer, "DSD %02x LPIB %08x\n", i, + lola_dsd_read(chip, i, LPIB)); + snd_iprintf(buffer, "DSD %02x CTL %08x\n", i, + lola_dsd_read(chip, i, CTL)); + snd_iprintf(buffer, "DSD %02x LVIL %08x\n", i, + lola_dsd_read(chip, i, LVI)); + snd_iprintf(buffer, "DSD %02x BDPL %08x\n", i, + lola_dsd_read(chip, i, BDPL)); + snd_iprintf(buffer, "DSD %02x BDPU %08x\n", i, + lola_dsd_read(chip, i, BDPU)); + } } void __devinit lola_proc_debug_new(struct lola *chip) { struct snd_info_entry *entry; - if (!snd_card_proc_new(chip->card, "codec", &entry)) { + if (!snd_card_proc_new(chip->card, "codec", &entry)) snd_info_set_text_ops(entry, chip, lola_proc_codec_read); + if (!snd_card_proc_new(chip->card, "codec_rw", &entry)) { + snd_info_set_text_ops(entry, chip, lola_proc_codec_rw_read); entry->mode |= S_IWUSR; - entry->c.text.write = lola_proc_codec_write; + entry->c.text.write = lola_proc_codec_rw_write; } if (!snd_card_proc_new(chip->card, "regs", &entry)) snd_info_set_text_ops(entry, chip, lola_proc_regs_read); -- cgit v1.2.3 From a426c787233c87587b6cac797ac840162bdb09c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:36:09 +0200 Subject: ALSA: lola - Suppress the debug print Use snd_printdd() for less important debug messages. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index f59ce085a1a..16c6cad2e7c 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -447,9 +447,9 @@ static int lola_parse_tree(struct lola *chip) chip->lola_caps = val; chip->pin[CAPT].num_pins = LOLA_AFG_INPUT_PIN_COUNT(chip->lola_caps); chip->pin[PLAY].num_pins = LOLA_AFG_OUTPUT_PIN_COUNT(chip->lola_caps); - snd_printd(SFX "speccaps=0x%x, pins in=%d, out=%d\n", - chip->lola_caps, - chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins); + snd_printdd(SFX "speccaps=0x%x, pins in=%d, out=%d\n", + chip->lola_caps, + chip->pin[CAPT].num_pins, chip->pin[PLAY].num_pins); if (chip->pin[CAPT].num_pins > MAX_AUDIO_INOUT_COUNT || chip->pin[PLAY].num_pins > MAX_AUDIO_INOUT_COUNT) { @@ -600,7 +600,7 @@ static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, chip->pcm[CAPT].num_streams = (dever >> 0) & 0x3ff; chip->pcm[PLAY].num_streams = (dever >> 10) & 0x3ff; chip->version = (dever >> 24) & 0xff; - snd_printd(SFX "streams in=%d, out=%d, version=0x%x\n", + snd_printdd(SFX "streams in=%d, out=%d, version=0x%x\n", chip->pcm[CAPT].num_streams, chip->pcm[PLAY].num_streams, chip->version); -- cgit v1.2.3 From 333ff3971f374d9a057fd6d1492a2ef78989974d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:41:02 +0200 Subject: ALSA: lola - Use a single BDL Use a single BDL for both buffers instead of allocating for each. Also a few tune-up to avoid the stream stalls in the PCM code and the prelimianry work for SG-buffer support are added, too. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.h | 2 +- sound/pci/lola/lola_pcm.c | 78 +++++++++++++++++++++++++++++------------------ 2 files changed, 49 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index 23831a1a95a..1041b0f0e20 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -299,7 +299,6 @@ struct lola_stream { unsigned int bufsize; unsigned int period_bytes; unsigned int frags; - struct snd_dma_buffer bdl; /* BDL buffer */ /* format + channel setup */ unsigned int format_verb; @@ -314,6 +313,7 @@ struct lola_stream { struct lola_pcm { unsigned int num_streams; + struct snd_dma_buffer bdl; /* BDL buffer */ struct lola_stream streams[MAX_STREAM_COUNT]; }; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index bf17f921d45..a656439915c 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -26,10 +26,28 @@ #include #include "lola.h" -#define BDL_SIZE 4096 -#define LOLA_MAX_BDL_ENTRIES (BDL_SIZE / 16) -#define LOLA_MAX_FRAG 32 +/* #define USE_SG_BUFFER */ + +#define LOLA_MAX_BDL_ENTRIES 8 +#define LOLA_MAX_FRAG 8 #define LOLA_MAX_BUF_SIZE (1024*1024*1024) +#define LOLA_BDL_ENTRY_SIZE (16 * 16) + +#ifdef USE_SG_BUFFER +#define get_addr(substream, ofs) \ + snd_pcm_sgbuf_get_addr(substream, ofs) +#define get_size(substream, ofs, size) \ + snd_pcm_sgbuf_get_chunk_size(substream, ofs, size) +#define ops_page snd_pcm_sgbuf_ops_page +#define PREALLOC_TYPE SNDRV_DMA_TYPE_DEV_SG +#else +#define get_addr(substream, ofs) \ + ((substream)->runtime->dma_addr + ofs) +#define get_size(substream, ofs, size) \ + (size) +#define ops_page NULL +#define PREALLOC_TYPE SNDRV_DMA_TYPE_DEV +#endif static struct lola_pcm *lola_get_pcm(struct snd_pcm_substream *substream) { @@ -101,7 +119,7 @@ static void lola_stream_clear(struct lola *chip, struct lola_stream *str) static void wait_for_srst_clear(struct lola *chip, struct lola_stream *str) { - unsigned long end_time = jiffies + msecs_to_jiffies(50); + unsigned long end_time = jiffies + msecs_to_jiffies(200); while (time_before(jiffies, end_time)) { unsigned int val; val = lola_dsd_read(chip, str->dsd, CTL); @@ -115,16 +133,16 @@ static void wait_for_srst_clear(struct lola *chip, struct lola_stream *str) static void lola_stream_reset(struct lola *chip, struct lola_stream *str) { lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRST); + wait_for_srst_clear(chip, str); lola_dsd_write(chip, str->dsd, LVI, 0); lola_dsd_write(chip, str->dsd, BDPU, 0); lola_dsd_write(chip, str->dsd, BDPL, 0); - wait_for_srst_clear(chip, str); } static int lola_stream_wait_for_fifo_ready(struct lola *chip, struct lola_stream *str) { - unsigned long end_time = jiffies + msecs_to_jiffies(50); + unsigned long end_time = jiffies + msecs_to_jiffies(1000); while (time_before(jiffies, end_time)) { unsigned int val = lola_dsd_read(chip, str->dsd, STS); if (val & LOLA_DSD_STS_FIFORDY) @@ -164,21 +182,12 @@ static int lola_pcm_open(struct snd_pcm_substream *substream) struct lola_pcm *pcm = lola_get_pcm(substream); struct lola_stream *str = lola_get_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; - int err; mutex_lock(&chip->open_mutex); if (str->opened) { mutex_unlock(&chip->open_mutex); return -EBUSY; } - err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - PAGE_SIZE, &str->bdl); - if (err < 0) { - mutex_unlock(&chip->open_mutex); - printk(KERN_ERR SFX "Can't allocate BDL\n"); - return err; - } str->substream = substream; str->master = NULL; str->opened = 1; @@ -216,7 +225,6 @@ static int lola_pcm_close(struct snd_pcm_substream *substream) str->substream = NULL; str->opened = 0; } - snd_dma_free_pages(&str->bdl); mutex_unlock(&chip->open_mutex); return 0; } @@ -262,12 +270,12 @@ static int setup_bdle(struct snd_pcm_substream *substream, if (str->frags >= LOLA_MAX_BDL_ENTRIES) return -EINVAL; - addr = snd_pcm_sgbuf_get_addr(substream, ofs); + addr = get_addr(substream, ofs); /* program the address field of the BDL entry */ bdl[0] = cpu_to_le32((u32)addr); bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ - chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size); + chunk = get_size(substream, ofs, size); bdl[2] = cpu_to_le32(chunk); /* program the IOC to enable interrupt * only when the whole fragment is processed @@ -285,7 +293,7 @@ static int setup_bdle(struct snd_pcm_substream *substream, /* * set up BDL entries */ -static int lola_setup_periods(struct lola *chip, +static int lola_setup_periods(struct lola *chip, struct lola_pcm *pcm, struct snd_pcm_substream *substream, struct lola_stream *str) { @@ -296,7 +304,7 @@ static int lola_setup_periods(struct lola *chip, periods = str->bufsize / period_bytes; /* program the initial BDL entries */ - bdl = (u32 *)str->bdl.area; + bdl = (u32 *)(pcm->bdl.area + LOLA_BDL_ENTRY_SIZE * str->index); ofs = 0; str->frags = 0; for (i = 0; i < periods; i++) { @@ -371,13 +379,14 @@ static int lola_set_stream_config(struct lola *chip, /* * set up the SD for streaming */ -static int lola_setup_controller(struct lola *chip, struct lola_stream *str) +static int lola_setup_controller(struct lola *chip, struct lola_pcm *pcm, + struct lola_stream *str) { - /* make sure the run bit is zero for SD */ - lola_stream_clear(chip, str); + dma_addr_t bdl; /* set up BDL */ - lola_dsd_write(chip, str->dsd, BDPL, (u32)str->bdl.addr); - lola_dsd_write(chip, str->dsd, BDPU, upper_32_bits(str->bdl.addr)); + bdl = pcm->bdl.addr + LOLA_BDL_ENTRY_SIZE * str->index; + lola_dsd_write(chip, str->dsd, BDPL, (u32)bdl); + lola_dsd_write(chip, str->dsd, BDPU, upper_32_bits(bdl)); /* program the stream LVI (last valid index) of the BDL */ lola_dsd_write(chip, str->dsd, LVI, str->frags - 1); lola_stream_stop(chip, str, lola_get_tstamp(chip, false)); @@ -418,7 +427,7 @@ static int lola_pcm_prepare(struct snd_pcm_substream *substream) str->bufsize = bufsize; str->period_bytes = period_bytes; str->format_verb = format_verb; - err = lola_setup_periods(chip, substream, str); + err = lola_setup_periods(chip, pcm, substream, str); if (err < 0) return err; } @@ -427,7 +436,7 @@ static int lola_pcm_prepare(struct snd_pcm_substream *substream) if (err < 0) return err; - return lola_setup_controller(chip, str); + return lola_setup_controller(chip, pcm, str); } static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -504,7 +513,7 @@ static struct snd_pcm_ops lola_pcm_ops = { .prepare = lola_pcm_prepare, .trigger = lola_pcm_trigger, .pointer = lola_pcm_pointer, - .page = snd_pcm_sgbuf_ops_page, + .page = ops_page, }; int __devinit lola_create_pcm(struct lola *chip) @@ -512,6 +521,14 @@ int __devinit lola_create_pcm(struct lola *chip) struct snd_pcm *pcm; int i, err; + for (i = 0; i < 2; i++) { + err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + PAGE_SIZE, &chip->pcm[i].bdl); + if (err < 0) + return err; + } + err = snd_pcm_new(chip->card, "Digigram Lola", 0, chip->pcm[SNDRV_PCM_STREAM_PLAYBACK].num_streams, chip->pcm[SNDRV_PCM_STREAM_CAPTURE].num_streams, @@ -525,7 +542,7 @@ int __devinit lola_create_pcm(struct lola *chip) snd_pcm_set_ops(pcm, i, &lola_pcm_ops); } /* buffer pre-allocation */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, + snd_pcm_lib_preallocate_pages_for_all(pcm, PREALLOC_TYPE, snd_dma_pci_data(chip->pci), 1024 * 64, 32 * 1024 * 1024); return 0; @@ -533,7 +550,8 @@ int __devinit lola_create_pcm(struct lola *chip) void lola_free_pcm(struct lola *chip) { - /* nothing to do */ + snd_dma_free_pages(&chip->pcm[0].bdl); + snd_dma_free_pages(&chip->pcm[1].bdl); } /* -- cgit v1.2.3 From 0f8f56c959c9c4a65a5ad13f1b49d97db30b5c2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:47:03 +0200 Subject: ALSA: lola - Fix PCM stalls Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 17 ++++++-- sound/pci/lola/lola.h | 1 + sound/pci/lola/lola_pcm.c | 103 +++++++++++++++++++++++++++------------------- 3 files changed, 74 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 16c6cad2e7c..4d8221505b9 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -73,6 +73,7 @@ static int corb_send_verb(struct lola *chip, unsigned int nid, chip->last_data = data; chip->last_extdata = extdata; data |= (nid << 20) | (verb << 8); + spin_lock_irqsave(&chip->reg_lock, flags); if (chip->rirb.cmds < LOLA_CORB_ENTRIES - 1) { unsigned int wp = chip->corb.wp + 1; @@ -338,8 +339,6 @@ static int setup_corb_rirb(struct lola *chip) chip->corb.buf = (u32 *)chip->rb.area; chip->rirb.addr = chip->rb.addr + 2048; chip->rirb.buf = (u32 *)(chip->rb.area + 2048); - lola_writel(chip, BAR0, CORBLBASE, (u32)chip->corb.addr); - lola_writel(chip, BAR0, CORBUBASE, upper_32_bits(chip->corb.addr)); /* disable ringbuffer DMAs */ lola_writeb(chip, BAR0, RIRBCTL, 0); @@ -496,6 +495,10 @@ static int lola_parse_tree(struct lola *chip) if (!chip->cold_reset) { lola_reset_setups(chip); chip->cold_reset = 1; + } else { + /* set the granularity if it is not the default */ + if (chip->granularity != LOLA_GRANULARITY_MIN) + lola_set_granularity(chip, chip->granularity, true); } return 0; @@ -561,8 +564,14 @@ static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, chip->pci = pci; chip->irq = -1; - chip->sample_rate_min = 48000; - chip->granularity = LOLA_GRANULARITY_MIN; + /* below a sample_rate of 16kHz the analogue audio quality + * is NOT excellent + */ + chip->sample_rate_min = 16000; + /* for instance use always max granularity which is compatible + * with all sample rates + */ + chip->granularity = LOLA_GRANULARITY_MAX; err = pci_request_regions(pci, DRVNAME); if (err < 0) { diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index 1041b0f0e20..839779dc32f 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -305,6 +305,7 @@ struct lola_stream { /* flags */ unsigned int opened:1; + unsigned int prepared:1; unsigned int running:1; }; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index a656439915c..ef738e6c991 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -29,7 +29,6 @@ /* #define USE_SG_BUFFER */ #define LOLA_MAX_BDL_ENTRIES 8 -#define LOLA_MAX_FRAG 8 #define LOLA_MAX_BUF_SIZE (1024*1024*1024) #define LOLA_BDL_ENTRY_SIZE (16 * 16) @@ -111,12 +110,6 @@ static void lola_stream_stop(struct lola *chip, struct lola_stream *str, lola_stream_clear_pending_irq(chip, str); } -static void lola_stream_clear(struct lola *chip, struct lola_stream *str) -{ - lola_dsd_write(chip, str->dsd, CTL, 0); - lola_stream_clear_pending_irq(chip, str); -} - static void wait_for_srst_clear(struct lola *chip, struct lola_stream *str) { unsigned long end_time = jiffies + msecs_to_jiffies(200); @@ -130,22 +123,15 @@ static void wait_for_srst_clear(struct lola *chip, struct lola_stream *str) printk(KERN_WARNING SFX "SRST not clear (stream %d)\n", str->dsd); } -static void lola_stream_reset(struct lola *chip, struct lola_stream *str) +static int lola_stream_wait_for_fifo(struct lola *chip, + struct lola_stream *str, + bool ready) { - lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRST); - wait_for_srst_clear(chip, str); - lola_dsd_write(chip, str->dsd, LVI, 0); - lola_dsd_write(chip, str->dsd, BDPU, 0); - lola_dsd_write(chip, str->dsd, BDPL, 0); -} - -static int lola_stream_wait_for_fifo_ready(struct lola *chip, - struct lola_stream *str) -{ - unsigned long end_time = jiffies + msecs_to_jiffies(1000); + unsigned int val = ready ? LOLA_DSD_STS_FIFORDY : 0; + unsigned long end_time = jiffies + msecs_to_jiffies(200); while (time_before(jiffies, end_time)) { - unsigned int val = lola_dsd_read(chip, str->dsd, STS); - if (val & LOLA_DSD_STS_FIFORDY) + unsigned int reg = lola_dsd_read(chip, str->dsd, STS); + if ((reg & LOLA_DSD_STS_FIFORDY) == val) return 0; msleep(1); } @@ -153,6 +139,23 @@ static int lola_stream_wait_for_fifo_ready(struct lola *chip, return -EIO; } +static void lola_stream_reset(struct lola *chip, struct lola_stream *str) +{ + if (str->prepared) { + str->prepared = 0; + + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE); + lola_stream_wait_for_fifo(chip, str, false); + lola_stream_clear_pending_irq(chip, str); + lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRST); + lola_dsd_write(chip, str->dsd, LVI, 0); + lola_dsd_write(chip, str->dsd, BDPU, 0); + lola_dsd_write(chip, str->dsd, BDPL, 0); + wait_for_srst_clear(chip, str); + } +} + static struct snd_pcm_hardware lola_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | @@ -163,16 +166,16 @@ static struct snd_pcm_hardware lola_pcm_hw = { SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_FLOAT_LE), - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = LOLA_MAX_BUF_SIZE, .period_bytes_min = 128, .period_bytes_max = LOLA_MAX_BUF_SIZE / 2, .periods_min = 2, - .periods_max = LOLA_MAX_FRAG, + .periods_max = LOLA_MAX_BDL_ENTRIES, .fifo_size = 0, }; @@ -194,10 +197,13 @@ static int lola_pcm_open(struct snd_pcm_substream *substream) runtime->hw = lola_pcm_hw; runtime->hw.channels_max = pcm->num_streams - str->index; snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - 128); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - 128); + /* period size = multiple of chip->granularity (8, 16 or 32 frames) + * use LOLA_GRANULARITY_MAX = 32 for instance + */ + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + LOLA_GRANULARITY_MAX); + snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + LOLA_GRANULARITY_MAX); mutex_unlock(&chip->open_mutex); return 0; } @@ -383,16 +389,24 @@ static int lola_setup_controller(struct lola *chip, struct lola_pcm *pcm, struct lola_stream *str) { dma_addr_t bdl; + + if (str->prepared) + return -EINVAL; + /* set up BDL */ bdl = pcm->bdl.addr + LOLA_BDL_ENTRY_SIZE * str->index; lola_dsd_write(chip, str->dsd, BDPL, (u32)bdl); lola_dsd_write(chip, str->dsd, BDPU, upper_32_bits(bdl)); /* program the stream LVI (last valid index) of the BDL */ lola_dsd_write(chip, str->dsd, LVI, str->frags - 1); - lola_stream_stop(chip, str, lola_get_tstamp(chip, false)); - lola_stream_wait_for_fifo_ready(chip, str); + lola_stream_clear_pending_irq(chip, str); - return 0; + lola_dsd_write(chip, str->dsd, CTL, + LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE | LOLA_DSD_CTL_SRUN); + + str->prepared = 1; + + return lola_stream_wait_for_fifo(chip, str, true); } static int lola_pcm_prepare(struct snd_pcm_substream *substream) @@ -421,22 +435,25 @@ static int lola_pcm_prepare(struct snd_pcm_substream *substream) period_bytes = snd_pcm_lib_period_bytes(substream); format_verb = lola_get_format_verb(substream); - if (bufsize != str->bufsize || - period_bytes != str->period_bytes || - format_verb != str->format_verb) { - str->bufsize = bufsize; - str->period_bytes = period_bytes; - str->format_verb = format_verb; - err = lola_setup_periods(chip, pcm, substream, str); - if (err < 0) - return err; - } + str->bufsize = bufsize; + str->period_bytes = period_bytes; + str->format_verb = format_verb; + + err = lola_setup_periods(chip, pcm, substream, str); + if (err < 0) + return err; err = lola_set_stream_config(chip, str, runtime->channels); if (err < 0) return err; - return lola_setup_controller(chip, pcm, str); + err = lola_setup_controller(chip, pcm, str); + if (err < 0) { + lola_stream_reset(chip, str); + return err; + } + + return 0; } static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -- cgit v1.2.3 From fe3d393eda2a716c2ea7fcd63e105657911ed245 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:48:59 +0200 Subject: ALSA: lola - Add Lola-specific module options Added granularity and sample_rate_min module options. The former controls the h/w access granularity. As default, it's set to the max value 32. The latter controls the minimum sample rate in Hz, as default 16000. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 45 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 35 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 4d8221505b9..7c13ace3c33 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -32,6 +32,7 @@ #include #include "lola.h" +/* Standard options */ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; @@ -43,6 +44,28 @@ MODULE_PARM_DESC(id, "ID string for Digigram Lola driver."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Digigram Lola driver."); +/* Lola-specific options */ + +/* for instance use always max granularity which is compatible + * with all sample rates + */ +static int granularity[SNDRV_CARDS] = { + [0 ... (SNDRV_CARDS - 1)] = LOLA_GRANULARITY_MAX +}; + +/* below a sample_rate of 16kHz the analogue audio quality is NOT excellent */ +static int sample_rate_min[SNDRV_CARDS] = { + [0 ... (SNDRV_CARDS - 1) ] = 16000 +}; + +module_param_array(granularity, int, NULL, 0444); +MODULE_PARM_DESC(granularity, "Granularity value"); +module_param_array(sample_rate_min, int, NULL, 0444); +MODULE_PARM_DESC(sample_rate_min, "Minimal sample rate"); + +/* + */ + MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram, Lola}}"); MODULE_DESCRIPTION("Digigram Lola driver"); @@ -536,7 +559,7 @@ static int lola_dev_free(struct snd_device *device) } static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, - struct lola **rchip) + int dev, struct lola **rchip) { struct lola *chip; int err; @@ -564,14 +587,16 @@ static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, chip->pci = pci; chip->irq = -1; - /* below a sample_rate of 16kHz the analogue audio quality - * is NOT excellent - */ - chip->sample_rate_min = 16000; - /* for instance use always max granularity which is compatible - * with all sample rates - */ - chip->granularity = LOLA_GRANULARITY_MAX; + chip->sample_rate_min = sample_rate_min[dev]; + + chip->granularity = granularity[dev]; + /* FIXME */ + if (chip->granularity != LOLA_GRANULARITY_MAX) { + snd_printk(KERN_WARNING SFX + "Only %d granularity is supported for now\n", + LOLA_GRANULARITY_MAX); + chip->granularity = LOLA_GRANULARITY_MAX; + } err = pci_request_regions(pci, DRVNAME); if (err < 0) { @@ -674,7 +699,7 @@ static int __devinit lola_probe(struct pci_dev *pci, snd_card_set_dev(card, &pci->dev); - err = lola_create(card, pci, &chip); + err = lola_create(card, pci, dev, &chip); if (err < 0) goto out_free; card->private_data = chip; -- cgit v1.2.3 From 972505ccde9a3bdc8455f4cf3d3f37d17f8f10a3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:50:51 +0200 Subject: ALSA: lola - Use SG-buffer Completely switch to SG-buffer now, as it's working stably. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 1 - sound/pci/lola/lola_pcm.c | 26 ++++---------------------- 2 files changed, 4 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 7c13ace3c33..3d2516b11f2 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -588,7 +588,6 @@ static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->sample_rate_min = sample_rate_min[dev]; - chip->granularity = granularity[dev]; /* FIXME */ if (chip->granularity != LOLA_GRANULARITY_MAX) { diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index ef738e6c991..dc2e811dcb3 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -26,28 +26,10 @@ #include #include "lola.h" -/* #define USE_SG_BUFFER */ - #define LOLA_MAX_BDL_ENTRIES 8 #define LOLA_MAX_BUF_SIZE (1024*1024*1024) #define LOLA_BDL_ENTRY_SIZE (16 * 16) -#ifdef USE_SG_BUFFER -#define get_addr(substream, ofs) \ - snd_pcm_sgbuf_get_addr(substream, ofs) -#define get_size(substream, ofs, size) \ - snd_pcm_sgbuf_get_chunk_size(substream, ofs, size) -#define ops_page snd_pcm_sgbuf_ops_page -#define PREALLOC_TYPE SNDRV_DMA_TYPE_DEV_SG -#else -#define get_addr(substream, ofs) \ - ((substream)->runtime->dma_addr + ofs) -#define get_size(substream, ofs, size) \ - (size) -#define ops_page NULL -#define PREALLOC_TYPE SNDRV_DMA_TYPE_DEV -#endif - static struct lola_pcm *lola_get_pcm(struct snd_pcm_substream *substream) { struct lola *chip = snd_pcm_substream_chip(substream); @@ -276,12 +258,12 @@ static int setup_bdle(struct snd_pcm_substream *substream, if (str->frags >= LOLA_MAX_BDL_ENTRIES) return -EINVAL; - addr = get_addr(substream, ofs); + addr = snd_pcm_sgbuf_get_addr(substream, ofs); /* program the address field of the BDL entry */ bdl[0] = cpu_to_le32((u32)addr); bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ - chunk = get_size(substream, ofs, size); + chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size); bdl[2] = cpu_to_le32(chunk); /* program the IOC to enable interrupt * only when the whole fragment is processed @@ -530,7 +512,7 @@ static struct snd_pcm_ops lola_pcm_ops = { .prepare = lola_pcm_prepare, .trigger = lola_pcm_trigger, .pointer = lola_pcm_pointer, - .page = ops_page, + .page = snd_pcm_sgbuf_ops_page, }; int __devinit lola_create_pcm(struct lola *chip) @@ -559,7 +541,7 @@ int __devinit lola_create_pcm(struct lola *chip) snd_pcm_set_ops(pcm, i, &lola_pcm_ops); } /* buffer pre-allocation */ - snd_pcm_lib_preallocate_pages_for_all(pcm, PREALLOC_TYPE, + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), 1024 * 64, 32 * 1024 * 1024); return 0; -- cgit v1.2.3 From 8bd172dc96fba8ba5a7560afdc1ff7461c278e86 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:51:56 +0200 Subject: ALSA: lola - Allow granularity changes Add some sanity checks. Change PCM parameters appropriately per granularity. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 27 ++++++++++++++++++++++----- sound/pci/lola/lola.h | 1 + sound/pci/lola/lola_pcm.c | 6 ++++-- 3 files changed, 27 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 3d2516b11f2..8ee22bee10c 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -587,14 +587,31 @@ static int __devinit lola_create(struct snd_card *card, struct pci_dev *pci, chip->pci = pci; chip->irq = -1; - chip->sample_rate_min = sample_rate_min[dev]; chip->granularity = granularity[dev]; - /* FIXME */ - if (chip->granularity != LOLA_GRANULARITY_MAX) { + switch (chip->granularity) { + case 8: + chip->sample_rate_max = 48000; + break; + case 16: + chip->sample_rate_max = 96000; + break; + case 32: + chip->sample_rate_max = 192000; + break; + default: snd_printk(KERN_WARNING SFX - "Only %d granularity is supported for now\n", - LOLA_GRANULARITY_MAX); + "Invalid granularity %d, reset to %d\n", + chip->granularity, LOLA_GRANULARITY_MAX); chip->granularity = LOLA_GRANULARITY_MAX; + chip->sample_rate_max = 192000; + break; + } + chip->sample_rate_min = sample_rate_min[dev]; + if (chip->sample_rate_min > chip->sample_rate_max) { + snd_printk(KERN_WARNING SFX + "Invalid sample_rate_min %d, reset to 16000\n", + chip->sample_rate_min); + chip->sample_rate_min = 16000; } err = pci_request_regions(pci, DRVNAME); diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index 839779dc32f..bc8110ff6b4 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -367,6 +367,7 @@ struct lola { /* parameters */ unsigned int granularity; unsigned int sample_rate_min; + unsigned int sample_rate_max; /* flags */ unsigned int running :1; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index dc2e811dcb3..4bb5b5bd637 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -178,14 +178,16 @@ static int lola_pcm_open(struct snd_pcm_substream *substream) str->opened = 1; runtime->hw = lola_pcm_hw; runtime->hw.channels_max = pcm->num_streams - str->index; + runtime->hw.rate_min = chip->sample_rate_min; + runtime->hw.rate_max = chip->sample_rate_max; snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); /* period size = multiple of chip->granularity (8, 16 or 32 frames) * use LOLA_GRANULARITY_MAX = 32 for instance */ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - LOLA_GRANULARITY_MAX); + chip->granularity); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - LOLA_GRANULARITY_MAX); + chip->granularity); mutex_unlock(&chip->open_mutex); return 0; } -- cgit v1.2.3 From 7e79f2267605e59492fef92b966dddc3c6a68b41 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 16:59:27 +0200 Subject: ALSA: lola - Add SRC refcounting Added the refcounting for the exclusive SRC control. Also, fixed the possible stall after PCM pause operations. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.h | 4 +++- sound/pci/lola/lola_pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 46 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index bc8110ff6b4..180c2c12462 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -306,6 +306,7 @@ struct lola_stream { /* flags */ unsigned int opened:1; unsigned int prepared:1; + unsigned int paused:1; unsigned int running:1; }; @@ -356,6 +357,8 @@ struct lola { /* clock */ struct lola_clock_widget clock; + int ref_count_rate; + unsigned int sample_rate; /* mixer */ struct lola_mixer_widget mixer; @@ -370,7 +373,6 @@ struct lola { unsigned int sample_rate_max; /* flags */ - unsigned int running :1; unsigned int initialized :1; unsigned int cold_reset :1; diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 4bb5b5bd637..6be6b7e8f56 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -126,6 +126,22 @@ static void lola_stream_reset(struct lola *chip, struct lola_stream *str) if (str->prepared) { str->prepared = 0; + if (str->paused) { + /* finish pause - prepare for a new resume + * move this code later to trigger function, + * as this is also needed when resuming from pause + */ + str->paused = 0; + /* implement later loop for all streams */ + lola_stream_wait_for_fifo(chip, str, false); + lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRUN | + LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE); + /* end loop */ + /* implement later once more loop for all streams */ + lola_stream_wait_for_fifo(chip, str, true); + /* end loop */ + /* end finish pause */ + } lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE); lola_stream_wait_for_fifo(chip, str, false); @@ -178,12 +194,17 @@ static int lola_pcm_open(struct snd_pcm_substream *substream) str->opened = 1; runtime->hw = lola_pcm_hw; runtime->hw.channels_max = pcm->num_streams - str->index; - runtime->hw.rate_min = chip->sample_rate_min; - runtime->hw.rate_max = chip->sample_rate_max; + if (chip->sample_rate) { + /* sample rate is locked */ + runtime->hw.rate_min = chip->sample_rate; + runtime->hw.rate_max = chip->sample_rate; + } else { + runtime->hw.rate_min = chip->sample_rate_min; + runtime->hw.rate_max = chip->sample_rate_max; + } + chip->ref_count_rate++; snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - /* period size = multiple of chip->granularity (8, 16 or 32 frames) - * use LOLA_GRANULARITY_MAX = 32 for instance - */ + /* period size = multiple of chip->granularity (8, 16 or 32 frames)*/ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, chip->granularity); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, @@ -215,6 +236,10 @@ static int lola_pcm_close(struct snd_pcm_substream *substream) str->substream = NULL; str->opened = 0; } + if (--chip->ref_count_rate == 0) { + /* release sample rate */ + chip->sample_rate = 0; + } mutex_unlock(&chip->open_mutex); return 0; } @@ -427,6 +452,11 @@ static int lola_pcm_prepare(struct snd_pcm_substream *substream) if (err < 0) return err; + err = lola_set_sample_rate(chip, runtime->rate); + if (err < 0) + return err; + chip->sample_rate = runtime->rate; /* sample rate gets locked */ + err = lola_set_stream_config(chip, str, runtime->channels); if (err < 0) return err; @@ -447,6 +477,7 @@ static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_substream *s; unsigned int start; unsigned int tstamp; + bool sync_streams; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -463,7 +494,12 @@ static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return -EINVAL; } - tstamp = lola_get_tstamp(chip, false); + /* + * sample correct synchronization is only needed starting several + * streams on stop or if only one stream do as quick as possible + */ + sync_streams = (start && snd_pcm_stream_linked(substream)); + tstamp = lola_get_tstamp(chip, !sync_streams); spin_lock(&chip->reg_lock); snd_pcm_group_for_each_entry(s, substream) { if (s->pcm->card != substream->pcm->card) @@ -474,6 +510,7 @@ static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) else lola_stream_stop(chip, str, tstamp); str->running = start; + str->paused = !start; snd_pcm_trigger_done(s, substream); } spin_unlock(&chip->reg_lock); -- cgit v1.2.3 From c7aad3c317afc05418414c95e877173799145d0b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 17:02:35 +0200 Subject: ALSA: lola - Add sync in loop implementation For assuring the synchronized state with the pause operation, loop over the all linked streams and waits until all get ready in a loop. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola_pcm.c | 76 ++++++++++++++++++++++++++++++++++++----------- 1 file changed, 59 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 6be6b7e8f56..5c0014cbff0 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -121,27 +121,69 @@ static int lola_stream_wait_for_fifo(struct lola *chip, return -EIO; } -static void lola_stream_reset(struct lola *chip, struct lola_stream *str) +/* sync for FIFO ready/empty for all linked streams; + * clear paused flag when FIFO gets ready again + */ +static int lola_sync_wait_for_fifo(struct lola *chip, + struct snd_pcm_substream *substream, + bool ready) { - if (str->prepared) { - str->prepared = 0; + unsigned int val = ready ? LOLA_DSD_STS_FIFORDY : 0; + unsigned long end_time = jiffies + msecs_to_jiffies(200); + struct snd_pcm_substream *s; + int pending = 0; + + while (time_before(jiffies, end_time)) { + pending = 0; + snd_pcm_group_for_each_entry(s, substream) { + struct lola_stream *str; + if (s->pcm->card != substream->pcm->card) + continue; + str = lola_get_stream(s); + if (str->prepared && str->paused) { + unsigned int reg; + reg = lola_dsd_read(chip, str->dsd, STS); + if ((reg & LOLA_DSD_STS_FIFORDY) != val) { + pending = str->dsd + 1; + break; + } + if (ready) + str->paused = 0; + } + } + if (!pending) + return 0; + msleep(1); + } + printk(KERN_WARNING SFX "FIFO not ready (pending %d)\n", pending - 1); + return -EIO; +} - if (str->paused) { - /* finish pause - prepare for a new resume - * move this code later to trigger function, - * as this is also needed when resuming from pause - */ - str->paused = 0; - /* implement later loop for all streams */ - lola_stream_wait_for_fifo(chip, str, false); +/* finish pause - prepare for a new resume */ +static void lola_sync_pause(struct lola *chip, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_substream *s; + + lola_sync_wait_for_fifo(chip, substream, false); + snd_pcm_group_for_each_entry(s, substream) { + struct lola_stream *str; + if (s->pcm->card != substream->pcm->card) + continue; + str = lola_get_stream(s); + if (str->paused && str->prepared) lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_SRUN | LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE); - /* end loop */ - /* implement later once more loop for all streams */ - lola_stream_wait_for_fifo(chip, str, true); - /* end loop */ - /* end finish pause */ - } + } + lola_sync_wait_for_fifo(chip, substream, true); +} + +static void lola_stream_reset(struct lola *chip, struct lola_stream *str) +{ + if (str->prepared) { + if (str->paused) + lola_sync_pause(chip, str->substream); + str->prepared = 0; lola_dsd_write(chip, str->dsd, CTL, LOLA_DSD_CTL_IOCE | LOLA_DSD_CTL_DEIE); lola_stream_wait_for_fifo(chip, str, false); -- cgit v1.2.3 From 2db3002029d6c8fbd8ad4ab21f501475cd8c96f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 17:05:08 +0200 Subject: ALSA: lola - Rename to Digital SRC Capture Switch Renamed to Digial SRC Capture Switch for more correct representation. Also fixed analog volume control on Lola161611 and lola881. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.h | 1 + sound/pci/lola/lola_mixer.c | 15 ++++++++++++++- sound/pci/lola/lola_pcm.c | 4 ++-- 3 files changed, 17 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index 180c2c12462..400ab5ef4c6 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -242,6 +242,7 @@ struct lola_pin { struct lola_pin_array { unsigned int num_pins; + unsigned int num_analog_pins; struct lola_pin pins[MAX_PINS]; }; diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index 0d09689d7b8..5d518f1a712 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -99,6 +99,8 @@ int __devinit lola_init_pins(struct lola *chip, int dir, int *nidp) err = lola_init_pin(chip, &chip->pin[dir].pins[i], dir, nid); if (err < 0) return err; + if (chip->pin[dir].pins[i].is_analog) + chip->pin[dir].num_analog_pins++; } *nidp = nid; return 0; @@ -540,6 +542,9 @@ static int __devinit create_analog_mixer(struct lola *chip, int dir, char *name) { if (!chip->pin[dir].num_pins) return 0; + /* no analog volumes on digital only adapters */ + if (chip->pin[dir].num_pins != chip->pin[dir].num_analog_pins) + return 0; lola_analog_mixer.name = name; lola_analog_mixer.private_value = dir; return snd_ctl_add(chip->card, @@ -547,6 +552,7 @@ static int __devinit create_analog_mixer(struct lola *chip, int dir, char *name) } /* + * Hardware sample rate converter on digital input */ static int lola_input_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -587,15 +593,22 @@ static int lola_input_src_put(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new lola_input_src_mixer __devinitdata = { - .name = "Analog Capture Switch", + .name = "Digital SRC Capture Switch", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .info = lola_input_src_info, .get = lola_input_src_get, .put = lola_input_src_put, }; +/* + * Lola16161 or Lola881 can have Hardware sample rate converters + * on its digital input pins + */ static int __devinit create_input_src_mixer(struct lola *chip) { + if (!chip->input_src_caps_mask) + return 0; + return snd_ctl_add(chip->card, snd_ctl_new1(&lola_input_src_mixer, chip)); } diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 5c0014cbff0..aa21750202f 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -472,7 +472,7 @@ static int lola_pcm_prepare(struct snd_pcm_substream *substream) mutex_lock(&chip->open_mutex); lola_stream_reset(chip, str); lola_cleanup_slave_streams(pcm, str); - if (str->index + runtime->channels >= pcm->num_streams) { + if (str->index + runtime->channels > pcm->num_streams) { mutex_unlock(&chip->open_mutex); return -EINVAL; } @@ -538,7 +538,7 @@ static int lola_pcm_trigger(struct snd_pcm_substream *substream, int cmd) /* * sample correct synchronization is only needed starting several - * streams on stop or if only one stream do as quick as possible + * streams. On stop or if only one stream do as quick as possible */ sync_streams = (start && snd_pcm_stream_linked(substream)); tstamp = lola_get_tstamp(chip, !sync_streams); -- cgit v1.2.3 From fe4af1b55ed76e3051bbd8abce7d6b3c504272a5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 17:06:53 +0200 Subject: ALSA: lola - Implement polling_mode like hd-audio Also protect the call of lola_update_rirb() with spinlock. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 12 +++++++++++- sound/pci/lola/lola.h | 5 +++-- 2 files changed, 14 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 8ee22bee10c..34b24286d27 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -153,8 +153,14 @@ static int rirb_get_response(struct lola *chip, unsigned int *val, { unsigned long timeout; + again: timeout = jiffies + msecs_to_jiffies(1000); for (;;) { + if (chip->polling_mode) { + spin_lock_irq(&chip->reg_lock); + lola_update_rirb(chip); + spin_unlock_irq(&chip->reg_lock); + } if (!chip->rirb.cmds) { *val = chip->res; if (extval) @@ -175,9 +181,13 @@ static int rirb_get_response(struct lola *chip, unsigned int *val, break; udelay(20); cond_resched(); - lola_update_rirb(chip); } printk(KERN_WARNING SFX "RIRB response error\n"); + if (!chip->polling_mode) { + printk(KERN_WARNING SFX "switching to polling mode\n"); + chip->polling_mode = 1; + goto again; + } return -EIO; } diff --git a/sound/pci/lola/lola.h b/sound/pci/lola/lola.h index 400ab5ef4c6..d5708e29b16 100644 --- a/sound/pci/lola/lola.h +++ b/sound/pci/lola/lola.h @@ -374,8 +374,9 @@ struct lola { unsigned int sample_rate_max; /* flags */ - unsigned int initialized :1; - unsigned int cold_reset :1; + unsigned int initialized:1; + unsigned int cold_reset:1; + unsigned int polling_mode:1; /* for debugging */ unsigned int debug_res; -- cgit v1.2.3 From f044785d0afc6c173ca7298c1a1cb4559423112f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 18:21:01 +0200 Subject: ALSA: lola - Add missing inclusion of linux/delay.h Signed-off-by: Takashi Iwai --- sound/pci/lola/lola_pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index aa21750202f..c44db68eecb 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include "lola.h" -- cgit v1.2.3 From f2e0192519b31974009c0da55dc39c497240df06 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 May 2011 18:48:29 +0200 Subject: ALSA: lola - Yet another linux/delay.h inclusion Signed-off-by: Takashi Iwai --- sound/pci/lola/lola_clock.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/lola/lola_clock.c b/sound/pci/lola/lola_clock.c index d2cce39c347..72f8ef0ac86 100644 --- a/sound/pci/lola/lola_clock.c +++ b/sound/pci/lola/lola_clock.c @@ -20,6 +20,7 @@ #include #include +#include #include #include #include "lola.h" -- cgit v1.2.3 From 1307394afd2d6cf8f41f5e691b43ffc9e027630c Mon Sep 17 00:00:00 2001 From: Mike Rapoport Date: Tue, 26 Apr 2011 11:52:42 +0300 Subject: ASoC: tegra: TrimSlice machine support Signed-off-by: Mike Rapoport Acked-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 8 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/trimslice.c | 228 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 238 insertions(+) create mode 100644 sound/soc/tegra/trimslice.c (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 0f5bd8298a0..035d39a4beb 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -30,3 +30,11 @@ config SND_SOC_TEGRA_WM8903 boards using the WM8093 codec. Currently, the supported boards are Harmony, Ventana, Seaboard, Kaen, and Aebl. +config SND_SOC_TEGRA_TRIMSLICE + tristate "SoC Audio support for TrimSlice board" + depends on SND_SOC_TEGRA && MACH_TRIMSLICE && I2C + select SND_SOC_TEGRA_I2S + select SND_SOC_TLV320AIC23 + help + Say Y or M here if you want to add support for SoC audio on the + TrimSlice platform. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 13bef8d572c..fa6574d92a3 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -11,5 +11,7 @@ obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o +snd-soc-tegra-trimslice-objs := trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o +obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c new file mode 100644 index 00000000000..8fc07e9adf2 --- /dev/null +++ b/sound/soc/tegra/trimslice.c @@ -0,0 +1,228 @@ +/* + * trimslice.c - TrimSlice machine ASoC driver + * + * Copyright (C) 2011 - CompuLab, Ltd. + * Author: Mike Rapoport + * + * Based on code copyright/by: + * Author: Stephen Warren + * Copyright (C) 2010-2011 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include + +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "../codecs/tlv320aic23.h" + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-trimslice" + +struct tegra_trimslice { + struct tegra_asoc_utils_data util_data; +}; + +static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); + int srate, mclk; + int err; + + srate = params_rate(params); + mclk = 128 * srate; + + err = tegra_asoc_utils_set_rate(&trimslice->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "codec_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + dev_err(card->dev, "cpu_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static struct snd_soc_ops trimslice_asoc_ops = { + .hw_params = trimslice_asoc_hw_params, +}; + +static const struct snd_soc_dapm_widget trimslice_dapm_widgets[] = { + SND_SOC_DAPM_HP("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route trimslice_audio_map[] = { + {"Line Out", NULL, "LOUT"}, + {"Line Out", NULL, "ROUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, +}; + +static int trimslice_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_nc_pin(dapm, "LHPOUT"); + snd_soc_dapm_nc_pin(dapm, "RHPOUT"); + snd_soc_dapm_nc_pin(dapm, "MICIN"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .codec_name = "tlv320aic23-codec.2-001a", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "tlv320aic23-hifi", + .init = trimslice_asoc_init, + .ops = &trimslice_asoc_ops, +}; + +static struct snd_soc_card snd_soc_trimslice = { + .name = "tegra-trimslice", + .dai_link = &trimslice_tlv320aic23_dai, + .num_links = 1, + + .dapm_widgets = trimslice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(trimslice_dapm_widgets), + .dapm_routes = trimslice_audio_map, + .num_dapm_routes = ARRAY_SIZE(trimslice_audio_map), +}; + +static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_trimslice; + struct tegra_trimslice *trimslice; + int ret; + + trimslice = kzalloc(sizeof(struct tegra_trimslice), GFP_KERNEL); + if (!trimslice) { + dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n"); + return -ENOMEM; + } + + ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); + if (ret) + goto err_free_trimslice; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, trimslice); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&trimslice->util_data); +err_free_trimslice: + kfree(trimslice); + return ret; +} + +static int __devexit tegra_snd_trimslice_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&trimslice->util_data); + + kfree(trimslice); + + return 0; +} + +static struct platform_driver tegra_snd_trimslice_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_snd_trimslice_probe, + .remove = __devexit_p(tegra_snd_trimslice_remove), +}; + +static int __init snd_tegra_trimslice_init(void) +{ + return platform_driver_register(&tegra_snd_trimslice_driver); +} +module_init(snd_tegra_trimslice_init); + +static void __exit snd_tegra_trimslice_exit(void) +{ + platform_driver_unregister(&tegra_snd_trimslice_driver); +} +module_exit(snd_tegra_trimslice_exit); + +MODULE_AUTHOR("Mike Rapoport "); +MODULE_DESCRIPTION("Trimslice machine ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.3 From 0aaae527c71e6af571093d90474f37c8662008d5 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 30 Apr 2011 19:45:47 +0200 Subject: ASoC: Free the card's DAPM context Free the card's DAPM context when the card is removed. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a823654ef36..6a3cb53185c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2047,6 +2047,8 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) if (card->remove) card->remove(card); + snd_soc_dapm_free(&card->dapm); + kfree(card->rtd); snd_card_free(card->snd_card); return 0; -- cgit v1.2.3 From 8eecaf62445e175572ffabaab090b471001c5a2c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 30 Apr 2011 19:45:48 +0200 Subject: ASoC: Move DAPM debugfs directory creation to snd_soc_dapm_debugfs_init Move the creation of the DAPM debugfs directory to snd_soc_dapm_debugfs_init instead of having the same duplicated code in both codec and card DAPM setup. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 16 ++-------------- sound/soc/soc-dapm.c | 13 ++++++++++--- 2 files changed, 12 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6a3cb53185c..983ec640d4d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -302,13 +302,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) printk(KERN_WARNING "ASoC: Failed to create codec register debugfs file\n"); - codec->dapm.debugfs_dapm = debugfs_create_dir("dapm", - codec->debugfs_codec_root); - if (!codec->dapm.debugfs_dapm) - printk(KERN_WARNING - "Failed to create DAPM debugfs directory\n"); - - snd_soc_dapm_debugfs_init(&codec->dapm); + snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); } static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) @@ -1926,13 +1920,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); #ifdef CONFIG_DEBUG_FS - card->dapm.debugfs_dapm = debugfs_create_dir("dapm", - card->debugfs_card_root); - if (!card->dapm.debugfs_dapm) - printk(KERN_WARNING - "Failed to create card DAPM debugfs directory\n"); - - snd_soc_dapm_debugfs_init(&card->dapm); + snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); #endif snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 378f08adee5..ffed456b214 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1254,13 +1254,19 @@ static const struct file_operations dapm_bias_fops = { .llseek = default_llseek, }; -void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm) +void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, + struct dentry *parent) { struct snd_soc_dapm_widget *w; struct dentry *d; - if (!dapm->debugfs_dapm) + dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); + + if (!dapm->debugfs_dapm) { + printk(KERN_WARNING + "Failed to create DAPM debugfs directory\n"); return; + } d = debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm, @@ -1283,7 +1289,8 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm) } } #else -void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm) +void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, + struct dentry *parent) { } #endif -- cgit v1.2.3 From d5d1e0bef4385a0cd726613c0fc1909cd23efd39 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 30 Apr 2011 19:45:49 +0200 Subject: ASoC: Move DAPM widget debugfs entry creation to snd_soc_dapm_new_widgets Currently debugfs entries for a DAPM widgets are only added in snd_soc_dapm_debugfs_init. If a widget is added later (for example in the dai_link's probe callback) it will not show up in debugfs. This patch moves the creation of the widget debugfs entry to snd_soc_dapm_new_widgets where it will be added after the widget has been properly instantiated. As a side-effect this will also reduce the number of times the DAPM widget list is iterated during a card's instantiation. Since it is possible that snd_soc_dapm_new_widgets is invoked form the codecs or cards probe callbacks, the creation of the debugfs dapm directory has to be moved before these are called. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 13 +++++++------ sound/soc/soc-dapm.c | 35 +++++++++++++++++++++++------------ 2 files changed, 30 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 983ec640d4d..fbd011ebe19 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1493,6 +1493,8 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; + soc_init_codec_debugfs(codec); + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { @@ -1513,8 +1515,6 @@ static int soc_probe_codec(struct snd_soc_card *card, snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, driver->num_dapm_routes); - soc_init_codec_debugfs(codec); - /* mark codec as probed and add to card codec list */ codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); @@ -1523,6 +1523,7 @@ static int soc_probe_codec(struct snd_soc_card *card, return 0; err_probe: + soc_cleanup_codec_debugfs(codec); module_put(codec->dev->driver->owner); return ret; @@ -1873,6 +1874,10 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->dapm.card = card; list_add(&card->dapm.list, &card->dapm_list); +#ifdef CONFIG_DEBUG_FS + snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); +#endif + #ifdef CONFIG_PM_SLEEP /* deferred resume work */ INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); @@ -1919,10 +1924,6 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); -#ifdef CONFIG_DEBUG_FS - snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); -#endif - snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ffed456b214..8454cc2967d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1257,7 +1257,6 @@ static const struct file_operations dapm_bias_fops = { void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent) { - struct snd_soc_dapm_widget *w; struct dentry *d; dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); @@ -1274,25 +1273,35 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, if (!d) dev_warn(dapm->dev, "ASoC: Failed to create bias level debugfs file\n"); +} - list_for_each_entry(w, &dapm->card->widgets, list) { - if (!w->name || w->dapm != dapm) - continue; +static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct dentry *d; - d = debugfs_create_file(w->name, 0444, - dapm->debugfs_dapm, w, - &dapm_widget_power_fops); - if (!d) - dev_warn(w->dapm->dev, - "ASoC: Failed to create %s debugfs file\n", - w->name); - } + if (!dapm->debugfs_dapm || !w->name) + return; + + d = debugfs_create_file(w->name, 0444, + dapm->debugfs_dapm, w, + &dapm_widget_power_fops); + if (!d) + dev_warn(w->dapm->dev, + "ASoC: Failed to create %s debugfs file\n", + w->name); } + #else void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent) { } + +static inline void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) +{ +} + #endif /* test and update the power status of a mux widget */ @@ -1765,6 +1774,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) } w->new = 1; + + dapm_debugfs_add_widget(w); } dapm_power_widgets(dapm, SND_SOC_DAPM_STREAM_NOP); -- cgit v1.2.3 From 6c45e126567eb9f96519ca97917ce317fcbe5218 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 30 Apr 2011 19:45:50 +0200 Subject: ASoC: Remove DAPM debugfs entries before freeing widgets Remove the DAPM debugfs entries before freeing the context's widgets, otherwise a use after free situation might occur. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8454cc2967d..169e1767ebd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1292,6 +1292,11 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) w->name); } +static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) +{ + debugfs_remove_recursive(dapm->debugfs_dapm); +} + #else void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent) @@ -1302,6 +1307,10 @@ static inline void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) { } +static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) +{ +} + #endif /* test and update the power status of a mux widget */ @@ -2445,6 +2454,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { snd_soc_dapm_sys_remove(dapm->dev); + dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); } -- cgit v1.2.3 From 82cfecdc03499be63262d60daf859b4cc1ea3fba Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 28 Apr 2011 17:37:58 -0600 Subject: ASoC: s/w->kcontrols/w->kcontrol_news/g A future change will modify struct snd_soc_dapm_widget to store the actual kcontrol pointers for each kcontrol_new in a field named kcontrols. Rename the existing kcontrols field to enable this. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/soc-dapm.c | 28 +++++++++++++++------------- 2 files changed, 16 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 06b6981b8d6..19241576b6b 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -120,7 +120,7 @@ */ #define PM860X_DAPM_OUTPUT(wname, wevent) \ { .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ - .shift = 0, .invert = 0, .kcontrols = NULL, \ + .shift = 0, .invert = 0, .kcontrol_news = NULL, \ .num_kcontrols = 0, .event = wevent, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 169e1767ebd..35cc1ed00a4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -187,7 +187,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_mixer_named_ctl: { int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) - w->kcontrols[i].private_value; + w->kcontrol_news[i].private_value; unsigned int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; @@ -204,7 +204,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } break; case snd_soc_dapm_mux: { - struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; + struct soc_enum *e = (struct soc_enum *) + w->kcontrol_news[i].private_value; int val, item, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) @@ -220,7 +221,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } break; case snd_soc_dapm_virt_mux: { - struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; + struct soc_enum *e = (struct soc_enum *) + w->kcontrol_news[i].private_value; p->connect = 0; /* since a virtual mux has no backing registers to @@ -235,7 +237,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, break; case snd_soc_dapm_value_mux: { struct soc_enum *e = (struct soc_enum *) - w->kcontrols[i].private_value; + w->kcontrol_news[i].private_value; int val, item; val = snd_soc_read(w->codec, e->reg); @@ -310,11 +312,11 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, /* search for mixer kcontrol */ for (i = 0; i < dest->num_kcontrols; i++) { - if (!strcmp(control_name, dest->kcontrols[i].name)) { + if (!strcmp(control_name, dest->kcontrol_news[i].name)) { list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); - path->name = dest->kcontrols[i].name; + path->name = dest->kcontrol_news[i].name; dapm_set_path_status(dest, path, i); return 0; } @@ -349,7 +351,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, list_for_each_entry(path, &w->sources, list_sink) { /* mixer/mux paths name must match control name */ - if (path->name != (char*)w->kcontrols[i].name) + if (path->name != (char *)w->kcontrol_news[i].name) continue; /* add dapm control with long name. @@ -358,7 +360,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, * for dapm_mixer_named_ctl this is simply the * kcontrol name. */ - name_len = strlen(w->kcontrols[i].name) + 1; + name_len = strlen(w->kcontrol_news[i].name) + 1; if (w->id != snd_soc_dapm_mixer_named_ctl) name_len += 1 + strlen(w->name); @@ -377,17 +379,17 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, */ snprintf(path->long_name, name_len, "%s %s", w->name + prefix_len, - w->kcontrols[i].name); + w->kcontrol_news[i].name); break; case snd_soc_dapm_mixer_named_ctl: snprintf(path->long_name, name_len, "%s", - w->kcontrols[i].name); + w->kcontrol_news[i].name); break; } path->long_name[name_len - 1] = '\0'; - path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, + path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, path->long_name, prefix); ret = snd_ctl_add(card, path->kcontrol); if (ret < 0) { @@ -433,7 +435,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, * process but we're also using the same prefix for widgets so * cut the prefix off the front of the widget name. */ - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name + prefix_len, + kcontrol = snd_soc_cnew(&w->kcontrol_news[0], w, w->name + prefix_len, prefix); ret = snd_ctl_add(card, kcontrol); @@ -1648,7 +1650,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: ret = dapm_connect_mux(dapm, wsource, wsink, path, control, - &wsink->kcontrols[0]); + &wsink->kcontrol_news[0]); if (ret != 0) goto err; break; -- cgit v1.2.3 From fad598887dc0d89ffee3e51281a8143beb2ae58c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 28 Apr 2011 17:37:59 -0600 Subject: ASoC: Add w->kcontrols, and populate it Future changes will need reference to the kcontrol created for a given kcontrol_new. Store the created kcontrol values now. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 35cc1ed00a4..85b2c94535f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -400,6 +400,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, path->long_name = NULL; return ret; } + w->kcontrols[i] = path->kcontrol; } } return ret; @@ -442,6 +443,8 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, if (ret < 0) goto err; + w->kcontrols[0] = kcontrol; + list_for_each_entry(path, &w->sources, list_sink) path->kcontrol = kcontrol; @@ -1480,6 +1483,7 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) kfree(p->long_name); kfree(p); } + kfree(w->kcontrols); kfree(w->name); kfree(w); } @@ -1730,6 +1734,14 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) if (w->new) continue; + if (w->num_kcontrols) { + w->kcontrols = kzalloc(w->num_kcontrols * + sizeof(struct snd_kcontrol *), + GFP_KERNEL); + if (!w->kcontrols) + return -ENOMEM; + } + switch(w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: -- cgit v1.2.3 From fafd2176f72148e83c64a1f818ff33fceed83d08 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 28 Apr 2011 17:38:00 -0600 Subject: ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol A future change will allow multiple widgets to be affected by the same control. For example, a single register bit that controls separate muxes in both the L and R audio paths. This change updates the code that handles relevant controls to be able to iterate over a list of affected widgets. Note that only the put functions need significant modification to implement the iteration; the get functions do not need to iterate, nor unify the results, since all affected widgets reference the same kcontrol. When creating the list of widgets, always create a 1-sized list, since the control sharing is not implemented in this change. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 187 ++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 131 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 85b2c94535f..79b836c1045 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -333,6 +333,8 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; const char *prefix; + struct snd_soc_dapm_widget_list *wlist; + size_t wlistsize; if (dapm->codec) prefix = dapm->codec->name_prefix; @@ -354,6 +356,18 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, if (path->name != (char *)w->kcontrol_news[i].name) continue; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *), + wlist = kzalloc(wlistsize, GFP_KERNEL); + if (wlist == NULL) { + dev_err(dapm->dev, + "asoc: can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist->num_widgets = 1; + wlist->widgets[0] = w; + /* add dapm control with long name. * for dapm_mixer this is the concatenation of the * mixer and kcontrol name. @@ -366,8 +380,10 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, path->long_name = kmalloc(name_len, GFP_KERNEL); - if (path->long_name == NULL) + if (path->long_name == NULL) { + kfree(wlist); return -ENOMEM; + } switch (w->id) { default: @@ -389,13 +405,15 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, path->long_name[name_len - 1] = '\0'; - path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, - path->long_name, prefix); + path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i], + wlist, path->long_name, + prefix); ret = snd_ctl_add(card, path->kcontrol); if (ret < 0) { dev_err(dapm->dev, "asoc: failed to add dapm kcontrol %s: %d\n", path->long_name, ret); + kfree(wlist); kfree(path->long_name); path->long_name = NULL; return ret; @@ -416,12 +434,25 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, const char *prefix; size_t prefix_len; int ret = 0; + struct snd_soc_dapm_widget_list *wlist; + size_t wlistsize; if (!w->num_kcontrols) { dev_err(dapm->dev, "asoc: mux %s has no controls\n", w->name); return -EINVAL; } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *), + wlist = kzalloc(wlistsize, GFP_KERNEL); + if (wlist == NULL) { + dev_err(dapm->dev, + "asoc: can't allocate widget list for %s\n", w->name); + return -ENOMEM; + } + wlist->num_widgets = 1; + wlist->widgets[0] = w; + if (dapm->codec) prefix = dapm->codec->name_prefix; else @@ -436,8 +467,8 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, * process but we're also using the same prefix for widgets so * cut the prefix off the front of the widget name. */ - kcontrol = snd_soc_cnew(&w->kcontrol_news[0], w, w->name + prefix_len, - prefix); + kcontrol = snd_soc_cnew(&w->kcontrol_news[0], wlist, + w->name + prefix_len, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) @@ -452,6 +483,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, err: dev_err(dapm->dev, "asoc: failed to add kcontrol %s\n", w->name); + kfree(wlist); return ret; } @@ -1818,7 +1850,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -1857,7 +1890,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -1868,6 +1903,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, unsigned int val; int connect, change; struct snd_soc_dapm_update update; + int wi; val = (ucontrol->value.integer.value[0] & mask); @@ -1876,31 +1912,36 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mask = mask << shift; val = val << shift; - mutex_lock(&widget->codec->mutex); - widget->value = val; + if (val) + /* new connection */ + connect = invert ? 0 : 1; + else + /* old connection must be powered down */ + connect = invert ? 1 : 0; + + mutex_lock(&codec->mutex); change = snd_soc_test_bits(widget->codec, reg, mask, val); if (change) { - if (val) - /* new connection */ - connect = invert ? 0:1; - else - /* old connection must be powered down */ - connect = invert ? 1:0; + for (wi = 0; wi < wlist->num_widgets; wi++) { + widget = wlist->widgets[wi]; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + widget->value = val; - dapm_mixer_update_power(widget, kcontrol, connect); + update.kcontrol = kcontrol; + update.widget = widget; + update.reg = reg; + update.mask = mask; + update.val = val; + widget->dapm->update = &update; - widget->dapm->update = NULL; + dapm_mixer_update_power(widget, kcontrol, connect); + + widget->dapm->update = NULL; + } } - mutex_unlock(&widget->codec->mutex); + mutex_unlock(&codec->mutex); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -1917,7 +1958,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, bitmask; @@ -1945,11 +1987,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask, bitmask; struct snd_soc_dapm_update update; + int wi; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1965,22 +2010,29 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - mutex_lock(&widget->codec->mutex); - widget->value = val; + mutex_lock(&codec->mutex); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + if (change) { + for (wi = 0; wi < wlist->num_widgets; wi++) { + widget = wlist->widgets[wi]; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = e->reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + widget->value = val; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + update.kcontrol = kcontrol; + update.widget = widget; + update.reg = e->reg; + update.mask = mask; + update.val = val; + widget->dapm->update = &update; - widget->dapm->update = NULL; + dapm_mux_update_power(widget, kcontrol, change, mux, e); - mutex_unlock(&widget->codec->mutex); + widget->dapm->update = NULL; + } + } + + mutex_unlock(&codec->mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); @@ -1995,7 +2047,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; ucontrol->value.enumerated.item[0] = widget->value; @@ -2013,22 +2066,33 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; int ret = 0; + int wi; if (ucontrol->value.enumerated.item[0] >= e->max) return -EINVAL; - mutex_lock(&widget->codec->mutex); + mutex_lock(&codec->mutex); change = widget->value != ucontrol->value.enumerated.item[0]; - widget->value = ucontrol->value.enumerated.item[0]; - dapm_mux_update_power(widget, kcontrol, change, widget->value, e); + if (change) { + for (wi = 0; wi < wlist->num_widgets; wi++) { + widget = wlist->widgets[wi]; + + widget->value = ucontrol->value.enumerated.item[0]; + + dapm_mux_update_power(widget, kcontrol, change, + widget->value, e); + } + } - mutex_unlock(&widget->codec->mutex); + mutex_unlock(&codec->mutex); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); @@ -2049,7 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, mux; @@ -2089,11 +2154,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double); int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = widget->codec; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val, mux, change; unsigned int mask; struct snd_soc_dapm_update update; + int wi; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -2107,22 +2175,29 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - mutex_lock(&widget->codec->mutex); - widget->value = val; + mutex_lock(&codec->mutex); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + if (change) { + for (wi = 0; wi < wlist->num_widgets; wi++) { + widget = wlist->widgets[wi]; - update.kcontrol = kcontrol; - update.widget = widget; - update.reg = e->reg; - update.mask = mask; - update.val = val; - widget->dapm->update = &update; + widget->value = val; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + update.kcontrol = kcontrol; + update.widget = widget; + update.reg = e->reg; + update.mask = mask; + update.val = val; + widget->dapm->update = &update; - widget->dapm->update = NULL; + dapm_mux_update_power(widget, kcontrol, change, mux, e); - mutex_unlock(&widget->codec->mutex); + widget->dapm->update = NULL; + } + } + + mutex_unlock(&codec->mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); -- cgit v1.2.3 From af46800b9a3947724baeffb1a1649276971297c7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 28 Apr 2011 17:38:01 -0600 Subject: ASoC: Implement mux control sharing Control sharing is enabled when two widgets include pointers to the same kcontrol_new in their definition. Specifically: static const struct snd_kcontrol_new adcinput_mux = SOC_DAPM_ENUM("ADC Input", adcinput_enum); static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = { SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), }; This is useful when a single register bit or field affects multiple muxes at once. The common case is to have separate control bits or fields for each mux (channel). An alternative way of looking at this is that the mux is a stereo (or even n-channel) mux, rather than independant mono muxes. Without this change, a separate kcontrol will be created for each DAPM_MUX. This has the following disadvantages: * Confuses the user/programmer with redundant controls that don't map to separate hardware. * When one of the controls is changed, ASoC fails to update the DAPM logic for paths solely affected by the other controls impacted by the same register bits. This causes some paths not to be correctly powered up or down. Prior to this change, to work around this, the user or programmer had to manually toggle all duplicate controls away from the intended setting, and then back to it. Control sharing implies that the control is named based on the kcontrol_new itself, not any of the widgets that are affected by it. Control sharing is implemented by: When creating kcontrols, if a kcontrol does not yet exist for a particular kcontrol_new, then a new kcontrol is created with a list of widgets containing just a single entry. This is the normal case. However, if a kcontrol does already exists for the given kcontrol_new, the current widget is simply added to that kcontrol's list of affected widgets. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 106 ++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 75 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 79b836c1045..456617e6378 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -324,6 +324,28 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, return -ENODEV; } +static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, + const struct snd_kcontrol_new *kcontrol_new, + struct snd_kcontrol **kcontrol) +{ + struct snd_soc_dapm_widget *w; + int i; + + *kcontrol = NULL; + + list_for_each_entry(w, &dapm->card->widgets, list) { + for (i = 0; i < w->num_kcontrols; i++) { + if (&w->kcontrol_news[i] == kcontrol_new) { + if (w->kcontrols) + *kcontrol = w->kcontrols[i]; + return 1; + } + } + } + + return 0; +} + /* create new dapm mixer control */ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) @@ -433,58 +455,80 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_card *card = dapm->card->snd_card; const char *prefix; size_t prefix_len; - int ret = 0; + int ret; struct snd_soc_dapm_widget_list *wlist; + int shared, wlistentries; size_t wlistsize; + char *name; - if (!w->num_kcontrols) { - dev_err(dapm->dev, "asoc: mux %s has no controls\n", w->name); + if (w->num_kcontrols != 1) { + dev_err(dapm->dev, + "asoc: mux %s has incorrect number of controls\n", + w->name); return -EINVAL; } + shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0], + &kcontrol); + if (kcontrol) { + wlist = kcontrol->private_data; + wlistentries = wlist->num_widgets + 1; + } else { + wlist = NULL; + wlistentries = 1; + } wlistsize = sizeof(struct snd_soc_dapm_widget_list) + - sizeof(struct snd_soc_dapm_widget *), - wlist = kzalloc(wlistsize, GFP_KERNEL); + wlistentries * sizeof(struct snd_soc_dapm_widget *), + wlist = krealloc(wlist, wlistsize, GFP_KERNEL); if (wlist == NULL) { dev_err(dapm->dev, "asoc: can't allocate widget list for %s\n", w->name); return -ENOMEM; } - wlist->num_widgets = 1; - wlist->widgets[0] = w; + wlist->num_widgets = wlistentries; + wlist->widgets[wlistentries - 1] = w; - if (dapm->codec) - prefix = dapm->codec->name_prefix; - else - prefix = NULL; - - if (prefix) - prefix_len = strlen(prefix) + 1; - else - prefix_len = 0; + if (!kcontrol) { + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; + + if (shared) { + name = w->kcontrol_news[0].name; + prefix_len = 0; + } else { + name = w->name; + if (prefix) + prefix_len = strlen(prefix) + 1; + else + prefix_len = 0; + } - /* The control will get a prefix from the control creation - * process but we're also using the same prefix for widgets so - * cut the prefix off the front of the widget name. - */ - kcontrol = snd_soc_cnew(&w->kcontrol_news[0], wlist, - w->name + prefix_len, prefix); - ret = snd_ctl_add(card, kcontrol); + /* + * The control will get a prefix from the control creation + * process but we're also using the same prefix for widgets so + * cut the prefix off the front of the widget name. + */ + kcontrol = snd_soc_cnew(&w->kcontrol_news[0], wlist, + name + prefix_len, prefix); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "asoc: failed to add kcontrol %s\n", w->name); + kfree(wlist); + return ret; + } + } - if (ret < 0) - goto err; + kcontrol->private_data = wlist; w->kcontrols[0] = kcontrol; list_for_each_entry(path, &w->sources, list_sink) path->kcontrol = kcontrol; - return ret; - -err: - dev_err(dapm->dev, "asoc: failed to add kcontrol %s\n", w->name); - kfree(wlist); - return ret; + return 0; } /* create new dapm volume control */ -- cgit v1.2.3 From abc9d5aa085a4316e1abcd3c3f12ff0bb0133d57 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 3 May 2011 19:29:52 +0100 Subject: ASoC: Use shared controls for input signal path in WM8915 Gives finer grained power management. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 45 +++++++++++++++++++++++++-------------------- 1 file changed, 25 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 4a3c5cc77e5..0dc1dd70a13 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -991,10 +991,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8915_POWER_MANAGEMENT_1, 8, 0), SND_SOC_DAPM_PGA("IN1L PGA", WM8915_POWER_MANAGEMENT_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R PGA", WM8915_POWER_MANAGEMENT_2, 4, 0, NULL, 0), -SND_SOC_DAPM_PGA("ADC", SND_SOC_NOPM, 0, 0, NULL, 0), - -SND_SOC_DAPM_MUX("IN1 Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN2 Mux", SND_SOC_NOPM, 0, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux), SND_SOC_DAPM_PGA("IN1L", WM8915_POWER_MANAGEMENT_7, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R", WM8915_POWER_MANAGEMENT_7, 3, 0, NULL, 0), @@ -1172,28 +1172,33 @@ static const struct snd_soc_dapm_route wm8915_dapm_routes[] = { { "DMIC1L", NULL, "DMIC1" }, { "DMIC1R", NULL, "DMIC1" }, - { "ADC", NULL, "ADCL" }, - { "ADC", NULL, "ADCR" }, + { "IN1L Mux", "ADC", "ADCL" }, + { "IN1L Mux", "DMIC1", "DMIC1L" }, + { "IN1L Mux", "DMIC2", "DMIC2L" }, + + { "IN1R Mux", "ADC", "ADCR" }, + { "IN1R Mux", "DMIC1", "DMIC1R" }, + { "IN1R Mux", "DMIC2", "DMIC2R" }, - { "IN1 Mux", "ADC", "ADC" }, - { "IN1 Mux", "DMIC1", "DMIC1" }, - { "IN1 Mux", "DMIC2", "DMIC2" }, + { "IN2L Mux", "ADC", "ADCL" }, + { "IN2L Mux", "DMIC1", "DMIC1L" }, + { "IN2L Mux", "DMIC2", "DMIC2L" }, - { "IN2 Mux", "ADC", "ADC" }, - { "IN2 Mux", "DMIC1", "DMIC1" }, - { "IN2 Mux", "DMIC2", "DMIC2" }, + { "IN2R Mux", "ADC", "ADCR" }, + { "IN2R Mux", "DMIC1", "DMIC1R" }, + { "IN2R Mux", "DMIC2", "DMIC2R" }, - { "Left Sidetone", "IN1", "IN1 Mux" }, - { "Left Sidetone", "IN2", "IN2 Mux" }, + { "Left Sidetone", "IN1", "IN1L Mux" }, + { "Left Sidetone", "IN2", "IN2L Mux" }, - { "Right Sidetone", "IN1", "IN1 Mux" }, - { "Right Sidetone", "IN2", "IN2 Mux" }, + { "Right Sidetone", "IN1", "IN1R Mux" }, + { "Right Sidetone", "IN2", "IN2R Mux" }, - { "DSP1TXL", "IN1 Switch", "IN1 Mux" }, - { "DSP1TXR", "IN1 Switch", "IN1 Mux" }, + { "DSP1TXL", "IN1 Switch", "IN1L Mux" }, + { "DSP1TXR", "IN1 Switch", "IN1R Mux" }, - { "DSP2TXL", "IN1 Switch", "IN2 Mux" }, - { "DSP2TXR", "IN1 Switch", "IN2 Mux" }, + { "DSP2TXL", "IN1 Switch", "IN2L Mux" }, + { "DSP2TXR", "IN1 Switch", "IN2R Mux" }, { "AIF1TX0", NULL, "DSP1TXL" }, { "AIF1TX1", NULL, "DSP1TXR" }, -- cgit v1.2.3 From e1a02066080da55026c193603c7468c2d95d14a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 3 May 2011 19:31:20 +0100 Subject: ASoC: Remove outdated FIXME from WM8915 Actually the current code is perfectly sensible given the hardware. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 0dc1dd70a13..ccc9bd83279 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -1001,7 +1001,6 @@ SND_SOC_DAPM_PGA("IN1R", WM8915_POWER_MANAGEMENT_7, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("IN2L", WM8915_POWER_MANAGEMENT_7, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("IN2R", WM8915_POWER_MANAGEMENT_7, 7, 0, NULL, 0), -/* FIXME - these need to be concentrator widgets */ SND_SOC_DAPM_SUPPLY("DMIC2", WM8915_POWER_MANAGEMENT_7, 9, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC1", WM8915_POWER_MANAGEMENT_7, 8, 0, NULL, 0), -- cgit v1.2.3 From 447ee6a7cbbfb5ae7ab8f9b8b058b4a04fe398bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 May 2011 18:28:50 +0200 Subject: ALSA: hda - Use position_fix=3 as default for AMD chipsets AMD chipsets often behave pretty badly regarding the DMA position reporting. It results in the bad quality audio recording. Using position_fix=3 works well in general for them, so let's enable it as default for AMD. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f95ff6e029a..010fd318196 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2364,6 +2364,11 @@ static int __devinit check_position_fix(struct azx *chip, int fix) case AZX_DRIVER_ATI: /* Use link position directly, avoid any transfer problem. */ return POS_FIX_VIACOMBO; + case AZX_DRIVER_GENERIC: + /* AMD chipsets behave often badly, too */ + if (chip->pci->vendor == PCI_VENDOR_ID_AMD) + return POS_FIX_VIACOMBO; + break; } return POS_FIX_AUTO; -- cgit v1.2.3 From f428c94c842343b8580751cf637794050ca12d77 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 4 May 2011 09:53:10 -0700 Subject: ALSA: lola - fix lola build sound/pci/lola/Makefile was trying to build lola modules even when PCI and SND_LOLA were not enabled, causing build errors: ERROR: "snd_pcm_hw_constraint_step" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_period_elapsed" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_dma_alloc_pages" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_hw_constraint_integer" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_sgbuf_ops_page" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_set_ops" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_lib_free_pages" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_lib_ioctl" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_lib_malloc_pages" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_sgbuf_get_chunk_size" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_dma_free_pages" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_lib_preallocate_pages_for_all" [sound/pci/lola/snd-lola.ko] undefined! ERROR: "snd_pcm_new" [sound/pci/lola/snd-lola.ko] undefined! Fix the Makefile to build only when CONFIG_SND_LOLA is enabled. Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/pci/lola/Makefile | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/lola/Makefile b/sound/pci/lola/Makefile index 674715a8496..8178a2a59d0 100644 --- a/sound/pci/lola/Makefile +++ b/sound/pci/lola/Makefile @@ -1,3 +1,4 @@ snd-lola-y := lola.o lola_pcm.o lola_clock.o lola_mixer.o snd-lola-$(CONFIG_SND_DEBUG) += lola_proc.o -obj-m = snd-lola.o + +obj-$(CONFIG_SND_LOLA) += snd-lola.o -- cgit v1.2.3 From 77530150fba769d7b7e260b3f16ed2294c1737b6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 5 May 2011 16:59:09 +0200 Subject: ASoC: Create codec DAPM widgets before calling the codecs probe function This allows to create DAPM routes depending on those widgets in the codecs probe function. This is helpful when supporting similar codecs with minor differences in the DAPM routing with the same driver. Something similar has already been done for cards in commit a841ebb9 (ASoC: Create card DAPM widgets early so they can be used in callbacks). Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 133edeb9728..a477e218aa2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1495,6 +1495,10 @@ static int soc_probe_codec(struct snd_soc_card *card, soc_init_codec_debugfs(codec); + if (driver->dapm_widgets) + snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, + driver->num_dapm_widgets); + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { @@ -1508,9 +1512,6 @@ static int soc_probe_codec(struct snd_soc_card *card, if (driver->controls) snd_soc_add_controls(codec, driver->controls, driver->num_controls); - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, - driver->num_dapm_widgets); if (driver->dapm_routes) snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, driver->num_dapm_routes); -- cgit v1.2.3 From 64d27069755db41daa36e4770d88ebc57617559d Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 5 May 2011 14:18:11 +0100 Subject: ASoC: soc-cache: Allow codec->cache_bypass to be used with snd_soc_hw_bulk_write_raw() If we specifically want to write a block of data to the hw bypassing the cache, then allow this to happen inside snd_soc_hw_bulk_write_raw(). Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a217db25670..687beec5647 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -404,12 +404,13 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r { int ret; - /* Ensure that the base register is volatile. Subsequently - * any other register that is touched by this routine should be - * volatile as well to ensure that we don't get out of sync with - * the cache. + /* To ensure that we don't get out of sync with the cache, check + * whether the base register is volatile or if we've directly asked + * to bypass the cache. Out of bounds registers are considered + * volatile. */ - if (!snd_soc_codec_volatile_register(codec, reg) + if (!codec->cache_bypass + && !snd_soc_codec_volatile_register(codec, reg) && reg < codec->driver->reg_cache_size) return -EINVAL; -- cgit v1.2.3 From 0b4cd2e01c63cf303bba570c9896331e4932a9df Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 5 May 2011 16:59:10 +0200 Subject: ASoC: SSM2602: Cleanup coeff handling Drop unused field from the coeff struct, precalculate the srate register at compile-time and cleanup up the naming. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 75 ++++++++++++++++++++++------------------------ 1 file changed, 35 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 565bc72fb17..73b447eba29 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -167,83 +167,78 @@ static int ssm2602_add_widgets(struct snd_soc_codec *codec) return 0; } -struct _coeff_div { +struct ssm2602_coeff { u32 mclk; u32 rate; - u16 fs; - u8 sr:4; - u8 bosr:1; - u8 usb:1; + u8 srate; }; -/* codec mclk clock divider coefficients */ -static const struct _coeff_div coeff_div[] = { +#define SSM2602_COEFF_SRATE(sr, bosr, usb) (((sr) << 2) | ((bosr) << 1) | (usb)) + +/* codec mclk clock coefficients */ +static const struct ssm2602_coeff ssm2602_coeff_table[] = { /* 48k */ - {12288000, 48000, 256, 0x0, 0x0, 0x0}, - {18432000, 48000, 384, 0x0, 0x1, 0x0}, - {12000000, 48000, 250, 0x0, 0x0, 0x1}, + {12288000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x0)}, + {18432000, 48000, SSM2602_COEFF_SRATE(0x0, 0x1, 0x0)}, + {12000000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x1)}, /* 32k */ - {12288000, 32000, 384, 0x6, 0x0, 0x0}, - {18432000, 32000, 576, 0x6, 0x1, 0x0}, - {12000000, 32000, 375, 0x6, 0x0, 0x1}, + {12288000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x0)}, + {18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)}, + {12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)}, /* 8k */ - {12288000, 8000, 1536, 0x3, 0x0, 0x0}, - {18432000, 8000, 2304, 0x3, 0x1, 0x0}, - {11289600, 8000, 1408, 0xb, 0x0, 0x0}, - {16934400, 8000, 2112, 0xb, 0x1, 0x0}, - {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + {12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)}, + {18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)}, + {11289600, 8000, SSM2602_COEFF_SRATE(0xb, 0x0, 0x0)}, + {16934400, 8000, SSM2602_COEFF_SRATE(0xb, 0x1, 0x0)}, + {12000000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x1)}, /* 96k */ - {12288000, 96000, 128, 0x7, 0x0, 0x0}, - {18432000, 96000, 192, 0x7, 0x1, 0x0}, - {12000000, 96000, 125, 0x7, 0x0, 0x1}, + {12288000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x0)}, + {18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)}, + {12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)}, /* 44.1k */ - {11289600, 44100, 256, 0x8, 0x0, 0x0}, - {16934400, 44100, 384, 0x8, 0x1, 0x0}, - {12000000, 44100, 272, 0x8, 0x1, 0x1}, + {11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)}, + {16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)}, + {12000000, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x1)}, /* 88.2k */ - {11289600, 88200, 128, 0xf, 0x0, 0x0}, - {16934400, 88200, 192, 0xf, 0x1, 0x0}, - {12000000, 88200, 136, 0xf, 0x1, 0x1}, + {11289600, 88200, SSM2602_COEFF_SRATE(0xf, 0x0, 0x0)}, + {16934400, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x0)}, + {12000000, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x1)}, }; -static inline int get_coeff(int mclk, int rate) +static inline int ssm2602_get_coeff(int mclk, int rate) { int i; - for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { - if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) - return i; + for (i = 0; i < ARRAY_SIZE(ssm2602_coeff_table); i++) { + if (ssm2602_coeff_table[i].rate == rate && + ssm2602_coeff_table[i].mclk == mclk) + return ssm2602_coeff_table[i].srate; } - return i; + return -EINVAL; } static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - u16 srate; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, SSM2602_IFACE) & 0xfff3; - int i = get_coeff(ssm2602->sysclk, params_rate(params)); + int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params)); if (substream == ssm2602->slave_substream) { dev_dbg(codec->dev, "Ignoring hw_params for slave substream\n"); return 0; } - /*no match is found*/ - if (i == ARRAY_SIZE(coeff_div)) - return -EINVAL; - - srate = (coeff_div[i].sr << 2) | - (coeff_div[i].bosr << 1) | coeff_div[i].usb; + if (srate < 0) + return srate; snd_soc_write(codec, SSM2602_ACTIVE, 0); snd_soc_write(codec, SSM2602_SRATE, srate); -- cgit v1.2.3 From ffd13c39c74c140925a11bb4595f41badf78142d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 5 May 2011 16:59:11 +0200 Subject: ASoC: SSM2602: Remove duplicate control There are currently two controls which allow selecting the capture source, one as a normal control, the other as part of a DAPM_MUX widget. Remove the normal control. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 73b447eba29..30229cf1bb5 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -99,8 +99,6 @@ SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0), -SOC_ENUM("Capture Source", ssm2602_enum[0]), - SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), }; -- cgit v1.2.3 From 5663940e2a9a9c9031cdba9ca170060de14da83f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 5 May 2011 16:59:13 +0200 Subject: ASoC: SSM2602: Remove unused struct and define Those are leftovers from a pre-multicomponent era. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.h | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h index 42a47d0f8e2..b98c6916803 100644 --- a/sound/soc/codecs/ssm2602.h +++ b/sound/soc/codecs/ssm2602.h @@ -117,11 +117,5 @@ #define SSM2602_CACHEREGNUM 10 #define SSM2602_SYSCLK 0 -#define SSM2602_DAI 0 - -struct ssm2602_setup_data { - int i2c_bus; - unsigned short i2c_address; -}; #endif -- cgit v1.2.3 From 7164bdb643cd9c919d69a5ea55b6f8477b90657c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 8 May 2011 09:24:41 -0700 Subject: ASoC: SSM2602: Fix default register cache Some of the values in the default register cache did not represent the codecs state after reset. This patch fixes it. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 30229cf1bb5..a09d66c0e70 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -60,8 +60,8 @@ struct ssm2602_priv { * There is no point in caching the reset register */ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { - 0x0017, 0x0017, 0x0079, 0x0079, - 0x0000, 0x0000, 0x0000, 0x000a, + 0x0097, 0x0097, 0x0079, 0x0079, + 0x000a, 0x0008, 0x009f, 0x000a, 0x0000, 0x0000 }; -- cgit v1.2.3 From f6c1f2d5e5d50366910fd687e88d07ebaabe00ab Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 8 May 2011 09:24:42 -0700 Subject: ASoC: SSM2602: Do not power the codec up in probe It is not required to have the codec powered at this stage and DAPM will power the ADC and DAC down again after probe has run anyway. Thus we avoid some unnecessary writes by this change. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index a09d66c0e70..e1ebf3d0493 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -509,8 +509,6 @@ static int ssm2602_probe(struct snd_soc_codec *codec) return ret; } - /*power on device*/ - snd_soc_write(codec, SSM2602_ACTIVE, 0); /* set the update bits */ reg = snd_soc_read(codec, SSM2602_LINVOL); snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); @@ -523,7 +521,6 @@ static int ssm2602_probe(struct snd_soc_codec *codec) /*select Line in as default input*/ snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); - snd_soc_write(codec, SSM2602_PWR, 0); snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); -- cgit v1.2.3 From b1f7b2b56b98d9eedc6a1b127d5bc5e9c51d2f73 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 8 May 2011 09:24:43 -0700 Subject: ASoC: SSM2602: Add SSM2604 support The SSM2604 is basically a lightweight variant of the SSM2602 with a compatible register layout. Thus we can easily support both devices by the same driver, by providing a slightly set of controls, widgets and routes. Compared to the SSM2602 the SSM2604 has no microphone input and no headphone output. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 169 +++++++++++++++++++++++++++++++-------------- 1 file changed, 116 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index e1ebf3d0493..b0306cf09df 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -45,12 +45,19 @@ #define SSM2602_VERSION "0.1" +enum ssm2602_type { + SSM2602, + SSM2604, +}; + /* codec private data */ struct ssm2602_priv { unsigned int sysclk; enum snd_soc_control_type control_type; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; + + enum ssm2602_type type; }; /* @@ -80,90 +87,97 @@ static const struct soc_enum ssm2602_enum[] = { SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), }; -static const struct snd_kcontrol_new ssm2602_snd_controls[] = { +static const struct snd_kcontrol_new ssm260x_snd_controls[] = { +SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), +SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), +SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), +SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0), + +SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), +}; + +static const struct snd_kcontrol_new ssm2602_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V, 0, 127, 0), SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V, 7, 1, 0), -SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), -SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), +SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0), SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), - -SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), - -SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), -SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0), - -SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), }; /* Output Mixer */ -static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { +static const struct snd_kcontrol_new ssm260x_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), -SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0), +SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0), }; /* Input mux */ static const struct snd_kcontrol_new ssm2602_input_mux_controls = SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]); -static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = { -SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, - &ssm2602_output_mixer_controls[0], - ARRAY_SIZE(ssm2602_output_mixer_controls)), +static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), +SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("LOUT"), -SND_SOC_DAPM_OUTPUT("LHPOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), -SND_SOC_DAPM_OUTPUT("RHPOUT"), -SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), +SND_SOC_DAPM_INPUT("RLINEIN"), +SND_SOC_DAPM_INPUT("LLINEIN"), +}; + +static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1, + ssm260x_output_mixer_controls, + ARRAY_SIZE(ssm260x_output_mixer_controls)), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls), -SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1), + +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), SND_SOC_DAPM_INPUT("MICIN"), -SND_SOC_DAPM_INPUT("RLINEIN"), -SND_SOC_DAPM_INPUT("LLINEIN"), }; -static const struct snd_soc_dapm_route audio_conn[] = { - /* output mixer */ +static const struct snd_soc_dapm_widget ssm2604_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + ssm260x_output_mixer_controls, + ARRAY_SIZE(ssm260x_output_mixer_controls) - 1), /* Last element is the mic */ +}; + +static const struct snd_soc_dapm_route ssm260x_routes[] = { {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, + + {"ROUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + + {"Line Input", NULL, "LLINEIN"}, + {"Line Input", NULL, "RLINEIN"}, +}; + +static const struct snd_soc_dapm_route ssm2602_routes[] = { {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, - /* outputs */ {"RHPOUT", NULL, "Output Mixer"}, - {"ROUT", NULL, "Output Mixer"}, {"LHPOUT", NULL, "Output Mixer"}, - {"LOUT", NULL, "Output Mixer"}, - /* input mux */ {"Input Mux", "Line", "Line Input"}, {"Input Mux", "Mic", "Mic Bias"}, {"ADC", NULL, "Input Mux"}, - /* inputs */ - {"Line Input", NULL, "LLINEIN"}, - {"Line Input", NULL, "RLINEIN"}, {"Mic Bias", NULL, "MICIN"}, }; -static int ssm2602_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets, - ARRAY_SIZE(ssm2602_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_conn, ARRAY_SIZE(audio_conn)); - - return 0; -} +static const struct snd_soc_dapm_route ssm2604_routes[] = { + {"ADC", NULL, "Line Input"}, +}; struct ssm2602_coeff { u32 mclk; @@ -491,9 +505,47 @@ static int ssm2602_resume(struct snd_soc_codec *codec) } static int ssm2602_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret, reg; + + reg = snd_soc_read(codec, SSM2602_LOUT1V); + snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); + reg = snd_soc_read(codec, SSM2602_ROUT1V); + snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); + + ret = snd_soc_add_controls(codec, ssm2602_snd_controls, + ARRAY_SIZE(ssm2602_snd_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets, + ARRAY_SIZE(ssm2602_dapm_widgets)); + if (ret) + return ret; + + return snd_soc_dapm_add_routes(dapm, ssm2602_routes, + ARRAY_SIZE(ssm2602_routes)); +} + +static int ssm2604_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, ssm2604_dapm_widgets, + ARRAY_SIZE(ssm2604_dapm_widgets)); + if (ret) + return ret; + + return snd_soc_dapm_add_routes(dapm, ssm2604_routes, + ARRAY_SIZE(ssm2604_routes)); +} + +static int ssm260x_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - int ret = 0, reg; + int ret, reg; pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); @@ -514,19 +566,20 @@ static int ssm2602_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); reg = snd_soc_read(codec, SSM2602_RINVOL); snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); - reg = snd_soc_read(codec, SSM2602_LOUT1V); - snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); - reg = snd_soc_read(codec, SSM2602_ROUT1V); - snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); /*select Line in as default input*/ snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); - snd_soc_add_controls(codec, ssm2602_snd_controls, - ARRAY_SIZE(ssm2602_snd_controls)); - ssm2602_add_widgets(codec); + switch (ssm2602->type) { + case SSM2602: + ret = ssm2602_probe(codec); + break; + case SSM2604: + ret = ssm2604_probe(codec); + break; + } - return 0; + return ret; } /* remove everything here */ @@ -537,7 +590,7 @@ static int ssm2602_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { - .probe = ssm2602_probe, + .probe = ssm260x_probe, .remove = ssm2602_remove, .suspend = ssm2602_suspend, .resume = ssm2602_resume, @@ -545,6 +598,13 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .reg_cache_size = ARRAY_SIZE(ssm2602_reg), .reg_word_size = sizeof(u16), .reg_cache_default = ssm2602_reg, + + .controls = ssm260x_snd_controls, + .num_controls = ARRAY_SIZE(ssm260x_snd_controls), + .dapm_widgets = ssm260x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm260x_dapm_widgets), + .dapm_routes = ssm260x_routes, + .num_dapm_routes = ARRAY_SIZE(ssm260x_routes), }; #if defined(CONFIG_SPI_MASTER) @@ -559,6 +619,7 @@ static int __devinit ssm2602_spi_probe(struct spi_device *spi) spi_set_drvdata(spi, ssm2602); ssm2602->control_type = SND_SOC_SPI; + ssm2602->type = SSM2602; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); @@ -603,6 +664,7 @@ static int __devinit ssm2602_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, ssm2602); ssm2602->control_type = SND_SOC_I2C; + ssm2602->type = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ssm2602, &ssm2602_dai, 1); @@ -619,7 +681,8 @@ static int __devexit ssm2602_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id ssm2602_i2c_id[] = { - { "ssm2602", 0 }, + { "ssm2602", SSM2602 }, + { "ssm2604", SSM2604 }, { } }; MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); @@ -669,6 +732,6 @@ static void __exit ssm2602_exit(void) } module_exit(ssm2602_exit); -MODULE_DESCRIPTION("ASoC ssm2602 driver"); +MODULE_DESCRIPTION("ASoC SSM2602/SSM2604 driver"); MODULE_AUTHOR("Cliff Cai"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 7dcf2760bf9d127e6b582977d72316ca58612854 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 8 May 2011 09:24:44 -0700 Subject: ASoC: SSM2602: Add entry for the ssm2603 to the device id table The SSM2603 is mostly register compatible with the SSM2602 and can be supported by the current driver without any changes. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index b0306cf09df..d8287219eea 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -682,6 +682,7 @@ static int __devexit ssm2602_i2c_remove(struct i2c_client *client) static const struct i2c_device_id ssm2602_i2c_id[] = { { "ssm2602", SSM2602 }, + { "ssm2603", SSM2602 }, { "ssm2604", SSM2604 }, { } }; @@ -732,6 +733,6 @@ static void __exit ssm2602_exit(void) } module_exit(ssm2602_exit); -MODULE_DESCRIPTION("ASoC SSM2602/SSM2604 driver"); +MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver"); MODULE_AUTHOR("Cliff Cai"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 2a43801a76893286ead35e742e486591e75803a2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 8 May 2011 09:24:45 -0700 Subject: ASoC: SSM2602: Model power supply for the digital core as a DAPM widget Model the power supply for the digital core as a DAPM_SUPPLY widget. This allows to cleanup the code a bit. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 25 ++++++------------------- 1 file changed, 6 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index d8287219eea..763d392f12a 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -126,6 +126,8 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), +SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0), + SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), SND_SOC_DAPM_INPUT("RLINEIN"), @@ -152,6 +154,9 @@ SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, }; static const struct snd_soc_dapm_route ssm260x_routes[] = { + {"DAC", NULL, "Digital Core Power"}, + {"ADC", NULL, "Digital Core Power"}, + {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, @@ -252,7 +257,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, if (srate < 0) return srate; - snd_soc_write(codec, SSM2602_ACTIVE, 0); snd_soc_write(codec, SSM2602_SRATE, srate); /* bit size */ @@ -270,7 +274,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, break; } snd_soc_write(codec, SSM2602_IFACE, iface); - snd_soc_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); return 0; } @@ -312,17 +315,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - /* set active */ - snd_soc_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); - - return 0; -} - static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -330,16 +322,13 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - /* deactivate */ - if (!codec->active) - snd_soc_write(codec, SSM2602_ACTIVE, 0); - if (ssm2602->master_substream == substream) ssm2602->master_substream = ssm2602->slave_substream; ssm2602->slave_substream = NULL; } + static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -446,7 +435,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ - snd_soc_write(codec, SSM2602_ACTIVE, 0); snd_soc_write(codec, SSM2602_PWR, 0xffff); break; @@ -464,7 +452,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, static struct snd_soc_dai_ops ssm2602_dai_ops = { .startup = ssm2602_startup, - .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, .digital_mute = ssm2602_mute, -- cgit v1.2.3 From f3eee00da39cba3c8a4db7048458969c620ac6d8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 8 May 2011 09:24:46 -0700 Subject: ASoC: SSM2602: Provide dB ranges for the volume controls Also fix the maximum value for the capture volume control. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 763d392f12a..70099c9d63c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -40,6 +40,7 @@ #include #include #include +#include #include "ssm2602.h" @@ -87,8 +88,18 @@ static const struct soc_enum ssm2602_enum[] = { SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), }; +static const unsigned int ssm260x_outmix_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 47, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 48, 127, TLV_DB_SCALE_ITEM(-7400, 100, 0), +}; + +static const DECLARE_TLV_DB_SCALE(ssm260x_inpga_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(ssm260x_sidetone_tlv, -1500, 300, 0); + static const struct snd_kcontrol_new ssm260x_snd_controls[] = { -SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), +SOC_DOUBLE_R_TLV("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 45, 0, + ssm260x_inpga_tlv), SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1), @@ -98,12 +109,12 @@ SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), }; static const struct snd_kcontrol_new ssm2602_snd_controls[] = { -SOC_DOUBLE_R("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V, - 0, 127, 0), +SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V, + 0, 127, 0, ssm260x_outmix_tlv), SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V, 7, 1, 0), - -SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), +SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1, + ssm260x_sidetone_tlv), SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0), -- cgit v1.2.3 From 14219d06592025541559027d0fd8f96ef75f313c Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Mon, 9 May 2011 23:39:26 +0200 Subject: ALSA: tea575x: unify read/write functions Implement generic read/write functions to access TEA575x tuners. They're now implemented 4 times (once in es1968 and 3 times in fm801). This also allows mute to work on all cards. Also improve tuner detection/initialization. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/i2c/other/tea575x-tuner.c | 98 ++++++++++++++++++++++++++++++----------- 1 file changed, 72 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 9f35f378a17..31f9795daca 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -77,11 +77,65 @@ static struct v4l2_queryctrl radio_qctrl[] = { * lowlevel part */ +static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) +{ + u16 l; + u8 data; + + tea->ops->set_direction(tea, 1); + udelay(16); + + for (l = 25; l > 0; l--) { + data = (val >> 24) & TEA575X_DATA; + val <<= 1; /* shift data */ + tea->ops->set_pins(tea, data | TEA575X_WREN); + udelay(2); + tea->ops->set_pins(tea, data | TEA575X_WREN | TEA575X_CLK); + udelay(2); + tea->ops->set_pins(tea, data | TEA575X_WREN); + udelay(2); + } + + if (!tea->mute) + tea->ops->set_pins(tea, 0); +} + +static unsigned int snd_tea575x_read(struct snd_tea575x *tea) +{ + u16 l, rdata; + u32 data = 0; + + tea->ops->set_direction(tea, 0); + tea->ops->set_pins(tea, 0); + udelay(16); + + for (l = 24; l--;) { + tea->ops->set_pins(tea, TEA575X_CLK); + udelay(2); + if (!l) + tea->tuned = tea->ops->get_pins(tea) & TEA575X_MOST ? 0 : 1; + tea->ops->set_pins(tea, 0); + udelay(2); + data <<= 1; /* shift data */ + rdata = tea->ops->get_pins(tea); + if (!l) + tea->stereo = (rdata & TEA575X_MOST) ? 0 : 1; + if (rdata & TEA575X_DATA) + data++; + udelay(2); + } + + if (tea->mute) + tea->ops->set_pins(tea, TEA575X_WREN); + + return data; +} + static void snd_tea575x_get_freq(struct snd_tea575x *tea) { unsigned long freq; - freq = tea->ops->read(tea) & TEA575X_BIT_FREQ_MASK; + freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK; /* freq *= 12.5 */ freq *= 125; freq /= 10; @@ -111,7 +165,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= freq & TEA575X_BIT_FREQ_MASK; - tea->ops->write(tea, tea->val); + snd_tea575x_write(tea, tea->val); } /* @@ -139,7 +193,7 @@ static int vidioc_g_tuner(struct file *file, void *priv, if (v->index > 0) return -EINVAL; - tea->ops->read(tea); + snd_tea575x_read(tea); strcpy(v->name, "FM"); v->type = V4L2_TUNER_RADIO; @@ -233,10 +287,8 @@ static int vidioc_g_ctrl(struct file *file, void *priv, switch (ctrl->id) { case V4L2_CID_AUDIO_MUTE: - if (tea->ops->mute) { - ctrl->value = tea->mute; - return 0; - } + ctrl->value = tea->mute; + return 0; } return -EINVAL; } @@ -248,11 +300,11 @@ static int vidioc_s_ctrl(struct file *file, void *priv, switch (ctrl->id) { case V4L2_CID_AUDIO_MUTE: - if (tea->ops->mute) { - tea->ops->mute(tea, ctrl->value); + if (tea->mute != ctrl->value) { tea->mute = ctrl->value; - return 0; + snd_tea575x_set_freq(tea); } + return 0; } return -EINVAL; } @@ -317,18 +369,16 @@ static struct video_device tea575x_radio = { /* * initialize all the tea575x chips */ -void snd_tea575x_init(struct snd_tea575x *tea) +int snd_tea575x_init(struct snd_tea575x *tea) { int retval; - unsigned int val; struct video_device *tea575x_radio_inst; - val = tea->ops->read(tea); - if (val == 0x1ffffff || val == 0) { - snd_printk(KERN_ERR - "tea575x-tuner: Cannot find TEA575x chip\n"); - return; - } + tea->mute = 1; + + snd_tea575x_write(tea, 0x55AA); + if (snd_tea575x_read(tea) != 0x55AA) + return -ENODEV; tea->in_use = 0; tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; @@ -337,7 +387,7 @@ void snd_tea575x_init(struct snd_tea575x *tea) tea575x_radio_inst = video_device_alloc(); if (tea575x_radio_inst == NULL) { printk(KERN_ERR "tea575x-tuner: not enough memory\n"); - return; + return -ENOMEM; } memcpy(tea575x_radio_inst, &tea575x_radio, sizeof(tea575x_radio)); @@ -352,17 +402,13 @@ void snd_tea575x_init(struct snd_tea575x *tea) if (retval) { printk(KERN_ERR "tea575x-tuner: can't register video device!\n"); kfree(tea575x_radio_inst); - return; + return retval; } snd_tea575x_set_freq(tea); - - /* mute on init */ - if (tea->ops->mute) { - tea->ops->mute(tea, 1); - tea->mute = 1; - } tea->vd = tea575x_radio_inst; + + return 0; } void snd_tea575x_exit(struct snd_tea575x *tea) -- cgit v1.2.3 From 72587173cc31bb96db63e347b94ad0ef84878cea Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Mon, 9 May 2011 23:39:37 +0200 Subject: ALSA: es1968: convert TEA575x support to new interface Use common functions to access TEA575x tuner - remove original read/write functions and provide new pin manipulation functions instead. Tested with SF64-PCE2 card. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 95 ++++++++++++++---------------------------------------- 1 file changed, 24 insertions(+), 71 deletions(-) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index faf9138970a..75c5e0ed695 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2591,96 +2591,48 @@ static int __devinit snd_es1968_input_register(struct es1968 *chip) #define STR_WREN 0x0100 /* GPIO8 */ #define STR_MOST 0x0200 /* GPIO9 */ -static void snd_es1968_tea575x_write(struct snd_tea575x *tea, unsigned int val) +static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; - u16 l, bits; - u16 omask, odir; + u16 val = 0; - omask = inw(io + IO_MASK); - odir = (inw(io + IO_DIR) & ~STR_DATA) | (STR_CLK | STR_WREN); - outw(odir | STR_DATA, io + IO_DIR); - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - udelay(16); + val |= (pins & TEA575X_DATA) ? STR_DATA : 0; + val |= (pins & TEA575X_CLK) ? STR_CLK : 0; + val |= (pins & TEA575X_WREN) ? STR_WREN : 0; - for (l = 25; l; l--) { - bits = ((val >> 18) & STR_DATA) | STR_WREN; - val <<= 1; /* shift data */ - outw(bits, io); /* start strobe */ - udelay(2); - outw(bits | STR_CLK, io); /* HI level */ - udelay(2); - outw(bits, io); /* LO level */ - udelay(4); - } - - if (!tea->mute) - outw(0, io); - - udelay(4); - outw(omask, io + IO_MASK); - outw(odir, io + IO_DIR); - msleep(125); + outw(val, io); } -static unsigned int snd_es1968_tea575x_read(struct snd_tea575x *tea) +static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; - u16 l, rdata; - u32 data = 0; - u16 omask; - - omask = inw(io + IO_MASK); - outw(~(STR_CLK | STR_WREN), io + IO_MASK); - outw(0, io); - udelay(16); - - for (l = 24; l--;) { - outw(STR_CLK, io); /* HI state */ - udelay(2); - if (!l) - tea->tuned = inw(io) & STR_MOST ? 0 : 1; - outw(0, io); /* LO state */ - udelay(2); - data <<= 1; /* shift data */ - rdata = inw(io); - if (!l) - tea->stereo = (rdata & STR_MOST) ? 0 : 1; - else if (l && rdata & STR_DATA) - data++; - udelay(2); - } - - if (tea->mute) - outw(STR_WREN, io); - - udelay(4); - outw(omask, io + IO_MASK); + u16 val = inw(io); - return data & 0x3ffe; + return (val & STR_DATA) ? TEA575X_DATA : 0 | + (val & STR_MOST) ? TEA575X_MOST : 0; } -static void snd_es1968_tea575x_mute(struct snd_tea575x *tea, unsigned int mute) +static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool output) { struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; - u16 omask; + u16 odir = inw(io + IO_DIR); - omask = inw(io + IO_MASK); - outw(~STR_WREN, io + IO_MASK); - tea->mute = mute; - outw(tea->mute ? STR_WREN : 0, io); - udelay(4); - outw(omask, io + IO_MASK); - msleep(125); + if (output) { + outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); + outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + } else { + outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); + outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + } } static struct snd_tea575x_ops snd_es1968_tea_ops = { - .write = snd_es1968_tea575x_write, - .read = snd_es1968_tea575x_read, - .mute = snd_es1968_tea575x_mute, + .set_pins = snd_es1968_tea575x_set_pins, + .get_pins = snd_es1968_tea575x_get_pins, + .set_direction = snd_es1968_tea575x_set_direction, }; #endif @@ -2845,7 +2797,8 @@ static int __devinit snd_es1968_create(struct snd_card *card, chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; chip->tea.ops = &snd_es1968_tea_ops; - snd_tea575x_init(&chip->tea); + if (!snd_tea575x_init(&chip->tea)) + printk(KERN_INFO "es1968: detected TEA575x radio\n"); #endif *chip_ret = chip; -- cgit v1.2.3 From 938a1566b162aa485acd8dea2ebd92d92dfa0a45 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Mon, 9 May 2011 23:39:51 +0200 Subject: ALSA: fm801: convert TEA575x support to new interface Use common functions to access TEA575x tuner - remove original read/write functions and provide new pin manipulation functions instead. Also convert the original triple implementation to a simple GPIO pin map. Tested with SF256-PCP and SF64-PCR (added the GPIO pin for MO/ST signal for them). SF256-PCS untested (pin for MO/ST signal is a guess). Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 336 +++++++++--------------------------------------------- 1 file changed, 57 insertions(+), 279 deletions(-) (limited to 'sound') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index f4dc1c77202..1db5ead62a9 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -717,308 +717,86 @@ static int __devinit snd_fm801_pcm(struct fm801 *chip, int device, struct snd_pc #ifdef TEA575X_RADIO -/* 256PCS GPIO numbers */ -#define TEA_256PCS_DATA 1 -#define TEA_256PCS_WRITE_ENABLE 2 /* inverted */ -#define TEA_256PCS_BUS_CLOCK 3 - -static void snd_fm801_tea575x_256pcs_write(struct snd_tea575x *tea, unsigned int val) -{ - struct fm801 *chip = tea->private_data; - unsigned short reg; - int i = 25; - - spin_lock_irq(&chip->reg_lock); - reg = inw(FM801_REG(chip, GPIO_CTRL)); - /* use GPIO lines and set write enable bit */ - reg |= FM801_GPIO_GS(TEA_256PCS_DATA) | - FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) | - FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK); - /* all of lines are in the write direction */ - /* clear data and clock lines */ - reg &= ~(FM801_GPIO_GD(TEA_256PCS_DATA) | - FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) | - FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) | - FM801_GPIO_GP(TEA_256PCS_DATA) | - FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK) | - FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE)); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - - while (i--) { - if (val & (1 << i)) - reg |= FM801_GPIO_GP(TEA_256PCS_DATA); - else - reg &= ~FM801_GPIO_GP(TEA_256PCS_DATA); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - } +/* GPIO to TEA575x maps */ +struct snd_fm801_tea575x_gpio { + u8 data, clk, wren, most; +}; - /* and reset the write enable bit */ - reg |= FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE) | - FM801_GPIO_GP(TEA_256PCS_DATA); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - spin_unlock_irq(&chip->reg_lock); -} +static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = { + { .data = 1, .clk = 3, .wren = 2, .most = 0 }, /* SF256-PCS */ + { .data = 1, .clk = 0, .wren = 2, .most = 3 }, /* SF256-PCP */ + { .data = 2, .clk = 0, .wren = 1, .most = 3 }, /* SF64-PCR */ +}; -static unsigned int snd_fm801_tea575x_256pcs_read(struct snd_tea575x *tea) +static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct fm801 *chip = tea->private_data; - unsigned short reg; - unsigned int val = 0; - int i; - - spin_lock_irq(&chip->reg_lock); - reg = inw(FM801_REG(chip, GPIO_CTRL)); - /* use GPIO lines, set data direction to input */ - reg |= FM801_GPIO_GS(TEA_256PCS_DATA) | - FM801_GPIO_GS(TEA_256PCS_WRITE_ENABLE) | - FM801_GPIO_GS(TEA_256PCS_BUS_CLOCK) | - FM801_GPIO_GD(TEA_256PCS_DATA) | - FM801_GPIO_GP(TEA_256PCS_DATA) | - FM801_GPIO_GP(TEA_256PCS_WRITE_ENABLE); - /* all of lines are in the write direction, except data */ - /* clear data, write enable and clock lines */ - reg &= ~(FM801_GPIO_GD(TEA_256PCS_WRITE_ENABLE) | - FM801_GPIO_GD(TEA_256PCS_BUS_CLOCK) | - FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK)); - - for (i = 0; i < 24; i++) { - reg &= ~FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - reg |= FM801_GPIO_GP(TEA_256PCS_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - val <<= 1; - if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCS_DATA)) - val |= 1; - } - - spin_unlock_irq(&chip->reg_lock); - - return val; -} - -/* 256PCPR GPIO numbers */ -#define TEA_256PCPR_BUS_CLOCK 0 -#define TEA_256PCPR_DATA 1 -#define TEA_256PCPR_WRITE_ENABLE 2 /* inverted */ + unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); + struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; -static void snd_fm801_tea575x_256pcpr_write(struct snd_tea575x *tea, unsigned int val) -{ - struct fm801 *chip = tea->private_data; - unsigned short reg; - int i = 25; + reg &= ~(FM801_GPIO_GP(gpio.data) | + FM801_GPIO_GP(gpio.clk) | + FM801_GPIO_GP(gpio.wren)); - spin_lock_irq(&chip->reg_lock); - reg = inw(FM801_REG(chip, GPIO_CTRL)); - /* use GPIO lines and set write enable bit */ - reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) | - FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) | - FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK); - /* all of lines are in the write direction */ - /* clear data and clock lines */ - reg &= ~(FM801_GPIO_GD(TEA_256PCPR_DATA) | - FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) | - FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) | - FM801_GPIO_GP(TEA_256PCPR_DATA) | - FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK) | - FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE)); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - - while (i--) { - if (val & (1 << i)) - reg |= FM801_GPIO_GP(TEA_256PCPR_DATA); - else - reg &= ~FM801_GPIO_GP(TEA_256PCPR_DATA); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - } + reg |= (pins & TEA575X_DATA) ? FM801_GPIO_GP(gpio.data) : 0; + reg |= (pins & TEA575X_CLK) ? FM801_GPIO_GP(gpio.clk) : 0; + /* WRITE_ENABLE is inverted */ + reg |= (pins & TEA575X_WREN) ? 0 : FM801_GPIO_GP(gpio.wren); - /* and reset the write enable bit */ - reg |= FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE) | - FM801_GPIO_GP(TEA_256PCPR_DATA); outw(reg, FM801_REG(chip, GPIO_CTRL)); - spin_unlock_irq(&chip->reg_lock); } -static unsigned int snd_fm801_tea575x_256pcpr_read(struct snd_tea575x *tea) +static u8 snd_fm801_tea575x_get_pins(struct snd_tea575x *tea) { struct fm801 *chip = tea->private_data; - unsigned short reg; - unsigned int val = 0; - int i; - - spin_lock_irq(&chip->reg_lock); - reg = inw(FM801_REG(chip, GPIO_CTRL)); - /* use GPIO lines, set data direction to input */ - reg |= FM801_GPIO_GS(TEA_256PCPR_DATA) | - FM801_GPIO_GS(TEA_256PCPR_WRITE_ENABLE) | - FM801_GPIO_GS(TEA_256PCPR_BUS_CLOCK) | - FM801_GPIO_GD(TEA_256PCPR_DATA) | - FM801_GPIO_GP(TEA_256PCPR_DATA) | - FM801_GPIO_GP(TEA_256PCPR_WRITE_ENABLE); - /* all of lines are in the write direction, except data */ - /* clear data, write enable and clock lines */ - reg &= ~(FM801_GPIO_GD(TEA_256PCPR_WRITE_ENABLE) | - FM801_GPIO_GD(TEA_256PCPR_BUS_CLOCK) | - FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK)); - - for (i = 0; i < 24; i++) { - reg &= ~FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - reg |= FM801_GPIO_GP(TEA_256PCPR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - val <<= 1; - if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_256PCPR_DATA)) - val |= 1; - } - - spin_unlock_irq(&chip->reg_lock); + unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); + struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; - return val; + return (reg & FM801_GPIO_GP(gpio.data)) ? TEA575X_DATA : 0 | + (reg & FM801_GPIO_GP(gpio.most)) ? TEA575X_MOST : 0; } -/* 64PCR GPIO numbers */ -#define TEA_64PCR_BUS_CLOCK 0 -#define TEA_64PCR_WRITE_ENABLE 1 /* inverted */ -#define TEA_64PCR_DATA 2 - -static void snd_fm801_tea575x_64pcr_write(struct snd_tea575x *tea, unsigned int val) +static void snd_fm801_tea575x_set_direction(struct snd_tea575x *tea, bool output) { struct fm801 *chip = tea->private_data; - unsigned short reg; - int i = 25; + unsigned short reg = inw(FM801_REG(chip, GPIO_CTRL)); + struct snd_fm801_tea575x_gpio gpio = snd_fm801_tea575x_gpios[(chip->tea575x_tuner & TUNER_TYPE_MASK) - 1]; - spin_lock_irq(&chip->reg_lock); - reg = inw(FM801_REG(chip, GPIO_CTRL)); /* use GPIO lines and set write enable bit */ - reg |= FM801_GPIO_GS(TEA_64PCR_DATA) | - FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) | - FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK); - /* all of lines are in the write direction */ - /* clear data and clock lines */ - reg &= ~(FM801_GPIO_GD(TEA_64PCR_DATA) | - FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) | - FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) | - FM801_GPIO_GP(TEA_64PCR_DATA) | - FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK) | - FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE)); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - - while (i--) { - if (val & (1 << i)) - reg |= FM801_GPIO_GP(TEA_64PCR_DATA); - else - reg &= ~FM801_GPIO_GP(TEA_64PCR_DATA); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - } - - /* and reset the write enable bit */ - reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE) | - FM801_GPIO_GP(TEA_64PCR_DATA); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - spin_unlock_irq(&chip->reg_lock); -} - -static unsigned int snd_fm801_tea575x_64pcr_read(struct snd_tea575x *tea) -{ - struct fm801 *chip = tea->private_data; - unsigned short reg; - unsigned int val = 0; - int i; - - spin_lock_irq(&chip->reg_lock); - reg = inw(FM801_REG(chip, GPIO_CTRL)); - /* use GPIO lines, set data direction to input */ - reg |= FM801_GPIO_GS(TEA_64PCR_DATA) | - FM801_GPIO_GS(TEA_64PCR_WRITE_ENABLE) | - FM801_GPIO_GS(TEA_64PCR_BUS_CLOCK) | - FM801_GPIO_GD(TEA_64PCR_DATA) | - FM801_GPIO_GP(TEA_64PCR_DATA) | - FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); - /* all of lines are in the write direction, except data */ - /* clear data, write enable and clock lines */ - reg &= ~(FM801_GPIO_GD(TEA_64PCR_WRITE_ENABLE) | - FM801_GPIO_GD(TEA_64PCR_BUS_CLOCK) | - FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK)); - - for (i = 0; i < 24; i++) { - reg &= ~FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - reg |= FM801_GPIO_GP(TEA_64PCR_BUS_CLOCK); - outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - val <<= 1; - if (inw(FM801_REG(chip, GPIO_CTRL)) & FM801_GPIO_GP(TEA_64PCR_DATA)) - val |= 1; + reg |= FM801_GPIO_GS(gpio.data) | + FM801_GPIO_GS(gpio.wren) | + FM801_GPIO_GS(gpio.clk) | + FM801_GPIO_GS(gpio.most); + if (output) { + /* all of lines are in the write direction */ + /* clear data and clock lines */ + reg &= ~(FM801_GPIO_GD(gpio.data) | + FM801_GPIO_GD(gpio.wren) | + FM801_GPIO_GD(gpio.clk) | + FM801_GPIO_GP(gpio.data) | + FM801_GPIO_GP(gpio.clk) | + FM801_GPIO_GP(gpio.wren)); + } else { + /* use GPIO lines, set data direction to input */ + reg |= FM801_GPIO_GD(gpio.data) | + FM801_GPIO_GD(gpio.most) | + FM801_GPIO_GP(gpio.data) | + FM801_GPIO_GP(gpio.most) | + FM801_GPIO_GP(gpio.wren); + /* all of lines are in the write direction, except data */ + /* clear data, write enable and clock lines */ + reg &= ~(FM801_GPIO_GD(gpio.wren) | + FM801_GPIO_GD(gpio.clk) | + FM801_GPIO_GP(gpio.clk)); } - spin_unlock_irq(&chip->reg_lock); - - return val; -} - -static void snd_fm801_tea575x_64pcr_mute(struct snd_tea575x *tea, - unsigned int mute) -{ - struct fm801 *chip = tea->private_data; - unsigned short reg; - - spin_lock_irq(&chip->reg_lock); - - reg = inw(FM801_REG(chip, GPIO_CTRL)); - if (mute) - /* 0xf800 (mute) */ - reg &= ~FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); - else - /* 0xf802 (unmute) */ - reg |= FM801_GPIO_GP(TEA_64PCR_WRITE_ENABLE); outw(reg, FM801_REG(chip, GPIO_CTRL)); - udelay(1); - - spin_unlock_irq(&chip->reg_lock); } -static struct snd_tea575x_ops snd_fm801_tea_ops[3] = { - { - /* 1 = MediaForte 256-PCS */ - .write = snd_fm801_tea575x_256pcs_write, - .read = snd_fm801_tea575x_256pcs_read, - }, - { - /* 2 = MediaForte 256-PCPR */ - .write = snd_fm801_tea575x_256pcpr_write, - .read = snd_fm801_tea575x_256pcpr_read, - }, - { - /* 3 = MediaForte 64-PCR */ - .write = snd_fm801_tea575x_64pcr_write, - .read = snd_fm801_tea575x_64pcr_read, - .mute = snd_fm801_tea575x_64pcr_mute, - } +static struct snd_tea575x_ops snd_fm801_tea_ops = { + .set_pins = snd_fm801_tea575x_set_pins, + .get_pins = snd_fm801_tea575x_get_pins, + .set_direction = snd_fm801_tea575x_set_direction, }; #endif @@ -1456,7 +1234,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->tea.card = card; chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1]; + chip->tea.ops = &snd_fm801_tea_ops; snd_tea575x_init(&chip->tea); } #endif -- cgit v1.2.3 From d7ba858a7f7a95d1617756a83ff0717767f624fd Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Tue, 10 May 2011 23:24:15 +0200 Subject: ALSA: fm801: implement TEA575x tuner autodetection Autodetect TEA575x tuner connection type during init. This allows tuner to work out-of-the box. tea575x_tuner module parameter remains functional to force tuner type. Tested with SF256-PCP and SF64-PCR. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/fm801.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 1db5ead62a9..5aa3fd6f25f 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -53,7 +53,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * /* * Enable TEA575x tuner * 1 = MediaForte 256-PCS - * 2 = MediaForte 256-PCPR + * 2 = MediaForte 256-PCP * 3 = MediaForte 64-PCR * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 @@ -67,7 +67,7 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); #define TUNER_ONLY (1<<4) #define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) @@ -720,12 +720,13 @@ static int __devinit snd_fm801_pcm(struct fm801 *chip, int device, struct snd_pc /* GPIO to TEA575x maps */ struct snd_fm801_tea575x_gpio { u8 data, clk, wren, most; + char *name; }; static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = { - { .data = 1, .clk = 3, .wren = 2, .most = 0 }, /* SF256-PCS */ - { .data = 1, .clk = 0, .wren = 2, .most = 3 }, /* SF256-PCP */ - { .data = 2, .clk = 0, .wren = 1, .most = 3 }, /* SF64-PCR */ + { .data = 1, .clk = 3, .wren = 2, .most = 0, .name = "SF256-PCS" }, + { .data = 1, .clk = 0, .wren = 2, .most = 3, .name = "SF256-PCP" }, + { .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" }, }; static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) @@ -1229,14 +1230,24 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO + chip->tea.card = card; + chip->tea.freq_fixup = 10700; + chip->tea.private_data = chip; + chip->tea.ops = &snd_fm801_tea_ops; if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { - chip->tea.card = card; - chip->tea.freq_fixup = 10700; - chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops; - snd_tea575x_init(&chip->tea); - } + if (snd_tea575x_init(&chip->tea)) + snd_printk(KERN_ERR "TEA575x radio not found\n"); + } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) + /* autodetect tuner connection */ + for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { + chip->tea575x_tuner = tea575x_tuner; + if (!snd_tea575x_init(&chip->tea)) { + snd_printk(KERN_INFO "detected TEA575x radio type %s\n", + snd_fm801_tea575x_gpios[tea575x_tuner - 1].name); + break; + } + } #endif *rchip = chip; -- cgit v1.2.3 From 3a691b28a0ca3cf4d9010c6158318159e0275d2c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 11 May 2011 10:44:51 +0200 Subject: ALSA: add Apple iSight microphone driver This adds an experimental driver for the front and rear microphones of the Apple iSight web camera. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/Kconfig | 12 + sound/firewire/Makefile | 2 + sound/firewire/isight.c | 744 ++++++++++++++++++++++++++++++++++++++++ sound/firewire/iso-resources.c | 5 + sound/firewire/packets-buffer.c | 2 + 5 files changed, 765 insertions(+) create mode 100644 sound/firewire/isight.c (limited to 'sound') diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index e486f48660f..312f90d7ec6 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -22,4 +22,16 @@ config SND_FIREWIRE_SPEAKERS To compile this driver as a module, choose M here: the module will be called snd-firewire-speakers. +config SND_ISIGHT + tristate "Apple iSight microphone (EXPERIMENTAL)" + depends on EXPERIMENTAL + select SND_PCM + select SND_FIREWIRE_LIB + help + Say Y here to include support for the front and rear microphones + of the Apple iSight web camera. + + To compile this driver as a module, choose M here: the module + will be called snd-isight. + endif # SND_FIREWIRE diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index e5b1634d9ad..d71ed8935f7 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,6 +1,8 @@ snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp.o snd-firewire-speakers-objs := speakers.o +snd-isight-objs := isight.o obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o obj-$(CONFIG_SND_FIREWIRE_SPEAKERS) += snd-firewire-speakers.o +obj-$(CONFIG_SND_ISIGHT) += snd-isight.o diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c new file mode 100644 index 00000000000..a6f19f57a1c --- /dev/null +++ b/sound/firewire/isight.c @@ -0,0 +1,744 @@ +/* + * Apple iSight audio driver + * + * Copyright (c) Clemens Ladisch + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "lib.h" +#include "iso-resources.h" +#include "packets-buffer.h" + +#define OUI_APPLE 0x000a27 +#define MODEL_APPLE_ISIGHT 0x000008 +#define SW_ISIGHT_AUDIO 0x000010 + +#define REG_AUDIO_ENABLE 0x000 +#define AUDIO_ENABLE 0x80000000 +#define REG_DEF_AUDIO_GAIN 0x204 +#define REG_GAIN_RAW_START 0x210 +#define REG_GAIN_RAW_END 0x214 +#define REG_GAIN_DB_START 0x218 +#define REG_GAIN_DB_END 0x21c +#define REG_SAMPLE_RATE_INQUIRY 0x280 +#define REG_ISO_TX_CONFIG 0x300 +#define SPEED_SHIFT 16 +#define REG_SAMPLE_RATE 0x400 +#define RATE_48000 0x80000000 +#define REG_GAIN 0x500 +#define REG_MUTE 0x504 + +#define MAX_FRAMES_PER_PACKET 475 + +#define QUEUE_LENGTH 20 + +struct isight { + struct snd_card *card; + struct fw_unit *unit; + struct fw_device *device; + u64 audio_base; + struct fw_address_handler iris_handler; + struct snd_pcm_substream *pcm; + struct mutex mutex; + struct iso_packets_buffer buffer; + struct fw_iso_resources resources; + struct fw_iso_context *context; + bool pcm_running; + bool first_packet; + int packet_index; + u32 total_samples; + unsigned int buffer_pointer; + unsigned int period_counter; + s32 gain_min, gain_max; + unsigned int gain_tlv[4]; +}; + +struct audio_payload { + __be32 sample_count; + __be32 signature; + __be32 sample_total; + __be32 reserved; + __be16 samples[2 * MAX_FRAMES_PER_PACKET]; +}; + +MODULE_DESCRIPTION("iSight audio driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); + +static struct fw_iso_packet audio_packet = { + .payload_length = sizeof(struct audio_payload), + .interrupt = 1, +}; + +static void isight_update_pointers(struct isight *isight, unsigned int count) +{ + struct snd_pcm_runtime *runtime = isight->pcm->runtime; + unsigned int ptr; + + smp_wmb(); /* update buffer data before buffer pointer */ + + ptr = isight->buffer_pointer; + ptr += count; + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + ACCESS_ONCE(isight->buffer_pointer) = ptr; + + isight->period_counter += count; + if (isight->period_counter >= runtime->period_size) { + isight->period_counter -= runtime->period_size; + snd_pcm_period_elapsed(isight->pcm); + } +} + +static void isight_samples(struct isight *isight, + const __be16 *samples, unsigned int count) +{ + struct snd_pcm_runtime *runtime; + unsigned int count1; + + if (!ACCESS_ONCE(isight->pcm_running)) + return; + + runtime = isight->pcm->runtime; + if (isight->buffer_pointer + count <= runtime->buffer_size) { + memcpy(runtime->dma_area + isight->buffer_pointer * 4, + samples, count * 4); + } else { + count1 = runtime->buffer_size - isight->buffer_pointer; + memcpy(runtime->dma_area + isight->buffer_pointer * 4, + samples, count1 * 4); + samples += count1 * 2; + memcpy(runtime->dma_area, samples, (count - count1) * 4); + } + + isight_update_pointers(isight, count); +} + +static void isight_pcm_abort(struct isight *isight) +{ + unsigned long flags; + + snd_pcm_stream_lock_irqsave(isight->pcm, flags); + if (snd_pcm_running(isight->pcm)) + snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(isight->pcm, flags); +} + +static void isight_dropped_samples(struct isight *isight, unsigned int total) +{ + struct snd_pcm_runtime *runtime; + u32 dropped; + unsigned int count1; + + if (!ACCESS_ONCE(isight->pcm_running)) + return; + + runtime = isight->pcm->runtime; + dropped = total - isight->total_samples; + if (dropped < runtime->buffer_size) { + if (isight->buffer_pointer + dropped <= runtime->buffer_size) { + memset(runtime->dma_area + isight->buffer_pointer * 4, + 0, dropped * 4); + } else { + count1 = runtime->buffer_size - isight->buffer_pointer; + memset(runtime->dma_area + isight->buffer_pointer * 4, + 0, count1 * 4); + memset(runtime->dma_area, 0, (dropped - count1) * 4); + } + isight_update_pointers(isight, dropped); + } else { + isight_pcm_abort(isight); + } +} + +static void isight_packet(struct fw_iso_context *context, u32 cycle, + size_t header_length, void *header, void *data) +{ + struct isight *isight = data; + const struct audio_payload *payload; + unsigned int index, length, count, total; + int err; + + if (isight->packet_index < 0) + return; + index = isight->packet_index; + payload = isight->buffer.packets[index].buffer; + length = be32_to_cpup(header) >> 16; + + if (likely(length >= 16 && + payload->signature == cpu_to_be32(0x73676874/*"sght"*/))) { + count = be32_to_cpu(payload->sample_count); + if (likely(count <= (length - 16) / 4)) { + total = be32_to_cpu(payload->sample_total); + if (unlikely(total != isight->total_samples)) { + if (!isight->first_packet) + isight_dropped_samples(isight, total); + isight->first_packet = false; + isight->total_samples = total; + } + + isight_samples(isight, payload->samples, count); + isight->total_samples += count; + } + } + + if (++index >= QUEUE_LENGTH) + index = 0; + + err = fw_iso_context_queue(isight->context, &audio_packet, + &isight->buffer.iso_buffer, + isight->buffer.packets[index].offset); + if (err < 0) { + dev_err(&isight->unit->device, "queueing error: %d\n", err); + isight_pcm_abort(isight); + isight->packet_index = -1; + return; + } + + isight->packet_index = index; +} + +static int isight_connect(struct isight *isight) +{ + int ch, err, rcode, errors = 0; + __be32 value; + +retry_after_bus_reset: + ch = fw_iso_resources_allocate(&isight->resources, + sizeof(struct audio_payload), + isight->device->max_speed); + if (ch < 0) { + err = ch; + goto error; + } + + value = cpu_to_be32(ch | (isight->device->max_speed << SPEED_SHIFT)); + for (;;) { + rcode = fw_run_transaction( + isight->device->card, + TCODE_WRITE_QUADLET_REQUEST, + isight->device->node_id, + isight->resources.generation, + isight->device->max_speed, + isight->audio_base + REG_ISO_TX_CONFIG, + &value, 4); + if (rcode == RCODE_COMPLETE) { + return 0; + } else if (rcode == RCODE_GENERATION) { + fw_iso_resources_free(&isight->resources); + goto retry_after_bus_reset; + } else if (rcode_is_permanent_error(rcode) || ++errors >= 3) { + err = -EIO; + goto err_resources; + } + msleep(5); + } + +err_resources: + fw_iso_resources_free(&isight->resources); +error: + return err; +} + +static int isight_open(struct snd_pcm_substream *substream) +{ + static const struct snd_pcm_hardware hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 4 * 1024 * 1024, + .period_bytes_min = MAX_FRAMES_PER_PACKET * 4, + .period_bytes_max = 1024 * 1024, + .periods_min = 2, + .periods_max = UINT_MAX, + }; + struct isight *isight = substream->private_data; + + substream->runtime->hw = hardware; + + return iso_packets_buffer_init(&isight->buffer, isight->unit, + QUEUE_LENGTH, + sizeof(struct audio_payload), + DMA_FROM_DEVICE); +} + +static int isight_close(struct snd_pcm_substream *substream) +{ + struct isight *isight = substream->private_data; + + iso_packets_buffer_destroy(&isight->buffer, isight->unit); + + return 0; +} + +static int isight_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} + +static void isight_stop_streaming(struct isight *isight) +{ + __be32 value; + + if (!isight->context) + return; + + fw_iso_context_stop(isight->context); + fw_iso_context_destroy(isight->context); + isight->context = NULL; + + value = 0; + snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_AUDIO_ENABLE, + &value, 4); + + fw_iso_resources_free(&isight->resources); +} + +static int isight_hw_free(struct snd_pcm_substream *substream) +{ + struct isight *isight = substream->private_data; + + mutex_lock(&isight->mutex); + isight_stop_streaming(isight); + mutex_unlock(&isight->mutex); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int isight_start_streaming(struct isight *isight) +{ + __be32 sample_rate; + unsigned int i; + int err; + + if (isight->context) { + if (isight->packet_index < 0) + isight_stop_streaming(isight); + else + return 0; + } + + sample_rate = cpu_to_be32(RATE_48000); + err = snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_SAMPLE_RATE, + &sample_rate, 4); + if (err < 0) + return err; + + err = isight_connect(isight); + if (err < 0) + goto error; + + isight->context = fw_iso_context_create(isight->device->card, + FW_ISO_CONTEXT_RECEIVE, + isight->resources.channel, + isight->device->max_speed, + 4, isight_packet, isight); + if (IS_ERR(isight->context)) { + err = PTR_ERR(isight->context); + isight->context = NULL; + goto err_resources; + } + + for (i = 0; i < QUEUE_LENGTH; ++i) { + err = fw_iso_context_queue(isight->context, &audio_packet, + &isight->buffer.iso_buffer, + isight->buffer.packets[i].offset); + if (err < 0) + goto err_context; + } + + isight->first_packet = true; + isight->packet_index = 0; + + err = fw_iso_context_start(isight->context, -1, 0, + FW_ISO_CONTEXT_MATCH_ALL_TAGS/*?*/); + if (err < 0) + goto err_context; + + return 0; + +err_context: + fw_iso_context_destroy(isight->context); + isight->context = NULL; +err_resources: + fw_iso_resources_free(&isight->resources); +error: + return err; +} + +static int isight_prepare(struct snd_pcm_substream *substream) +{ + struct isight *isight = substream->private_data; + int err; + + isight->buffer_pointer = 0; + isight->period_counter = 0; + + mutex_lock(&isight->mutex); + err = isight_start_streaming(isight); + mutex_unlock(&isight->mutex); + + return err; +} + +static int isight_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct isight *isight = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ACCESS_ONCE(isight->pcm_running) = true; + break; + case SNDRV_PCM_TRIGGER_STOP: + ACCESS_ONCE(isight->pcm_running) = false; + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t isight_pointer(struct snd_pcm_substream *substream) +{ + struct isight *isight = substream->private_data; + + return ACCESS_ONCE(isight->buffer_pointer); +} + +static int isight_create_pcm(struct isight *isight) +{ + static struct snd_pcm_ops ops = { + .open = isight_open, + .close = isight_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = isight_hw_params, + .hw_free = isight_hw_free, + .prepare = isight_prepare, + .trigger = isight_trigger, + .pointer = isight_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, + }; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(isight->card, "iSight", 0, 0, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = isight; + strcpy(pcm->name, "iSight"); + isight->pcm = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + isight->pcm->ops = &ops; + + return 0; +} + +static int isight_gain_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + struct isight *isight = ctl->private_data; + + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = isight->gain_min; + info->value.integer.max = isight->gain_max; + + return 0; +} + +static int isight_gain_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct isight *isight = ctl->private_data; + __be32 gain; + int err; + + err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, + isight->audio_base + REG_GAIN, &gain, 4); + if (err < 0) + return err; + + value->value.integer.value[0] = (s32)be32_to_cpu(gain); + + return 0; +} + +static int isight_gain_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct isight *isight = ctl->private_data; + __be32 gain; + + if (value->value.integer.value[0] < isight->gain_min || + value->value.integer.value[0] > isight->gain_max) + return -EINVAL; + + gain = cpu_to_be32(value->value.integer.value[0]); + return snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_GAIN, &gain, 4); +} + +static int isight_mute_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct isight *isight = ctl->private_data; + __be32 mute; + int err; + + err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, + isight->audio_base + REG_MUTE, &mute, 4); + if (err < 0) + return err; + + value->value.integer.value[0] = !mute; + + return 0; +} + +static int isight_mute_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct isight *isight = ctl->private_data; + __be32 mute; + + mute = (__force __be32)!value->value.integer.value[0]; + return snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_MUTE, &mute, 4); +} + +static int isight_create_mixer(struct isight *isight) +{ + static const struct snd_kcontrol_new gain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = isight_gain_info, + .get = isight_gain_get, + .put = isight_gain_put, + }; + static const struct snd_kcontrol_new mute_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Switch", + .info = snd_ctl_boolean_mono_info, + .get = isight_mute_get, + .put = isight_mute_put, + }; + __be32 value; + struct snd_kcontrol *ctl; + int err; + + err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, + isight->audio_base + REG_GAIN_RAW_START, + &value, 4); + if (err < 0) + return err; + isight->gain_min = be32_to_cpu(value); + + err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, + isight->audio_base + REG_GAIN_RAW_END, + &value, 4); + if (err < 0) + return err; + isight->gain_max = be32_to_cpu(value); + + isight->gain_tlv[0] = SNDRV_CTL_TLVT_DB_MINMAX; + isight->gain_tlv[1] = 2 * sizeof(unsigned int); + err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, + isight->audio_base + REG_GAIN_DB_START, + &value, 4); + if (err < 0) + return err; + isight->gain_tlv[2] = (s32)be32_to_cpu(value) * 100; + err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, + isight->audio_base + REG_GAIN_DB_END, + &value, 4); + if (err < 0) + return err; + isight->gain_tlv[3] = (s32)be32_to_cpu(value) * 100; + + ctl = snd_ctl_new1(&gain_control, isight); + if (ctl) + ctl->tlv.p = isight->gain_tlv; + err = snd_ctl_add(isight->card, ctl); + if (err < 0) + return err; + + err = snd_ctl_add(isight->card, snd_ctl_new1(&mute_control, isight)); + if (err < 0) + return err; + + return 0; +} + +static void isight_card_free(struct snd_card *card) +{ + struct isight *isight = card->private_data; + + fw_iso_resources_destroy(&isight->resources); + fw_unit_put(isight->unit); + fw_device_put(isight->device); + mutex_destroy(&isight->mutex); +} + +static u64 get_unit_base(struct fw_unit *unit) +{ + struct fw_csr_iterator i; + int key, value; + + fw_csr_iterator_init(&i, unit->directory); + while (fw_csr_iterator_next(&i, &key, &value)) + if (key == CSR_OFFSET) + return CSR_REGISTER_BASE + value * 4; + return 0; +} + +static int isight_probe(struct device *unit_dev) +{ + struct fw_unit *unit = fw_unit(unit_dev); + struct fw_device *fw_dev = fw_parent_device(unit); + struct snd_card *card; + struct isight *isight; + int err; + + err = snd_card_create(-1, NULL, THIS_MODULE, sizeof(*isight), &card); + if (err < 0) + return err; + snd_card_set_dev(card, unit_dev); + + isight = card->private_data; + isight->card = card; + mutex_init(&isight->mutex); + isight->unit = fw_unit_get(unit); + isight->device = fw_device_get(fw_dev); + isight->audio_base = get_unit_base(unit); + if (!isight->audio_base) { + dev_err(&unit->device, "audio unit base not found\n"); + err = -ENXIO; + goto err_unit; + } + fw_iso_resources_init(&isight->resources, unit); + + card->private_free = isight_card_free; + + strcpy(card->driver, "iSight"); + strcpy(card->shortname, "Apple iSight"); + snprintf(card->longname, sizeof(card->longname), + "Apple iSight (GUID %08x%08x) at %s, S%d", + fw_dev->config_rom[3], fw_dev->config_rom[4], + dev_name(&unit->device), 100 << fw_dev->max_speed); + strcpy(card->mixername, "iSight"); + + err = isight_create_pcm(isight); + if (err < 0) + goto error; + + err = isight_create_mixer(isight); + if (err < 0) + goto error; + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(unit_dev, isight); + + return 0; + +err_unit: + fw_unit_put(isight->unit); + fw_device_put(isight->device); + mutex_destroy(&isight->mutex); +error: + snd_card_free(card); + return err; +} + +static int isight_remove(struct device *dev) +{ + struct isight *isight = dev_get_drvdata(dev); + + snd_card_disconnect(isight->card); + + mutex_lock(&isight->mutex); + isight_pcm_abort(isight); + isight_stop_streaming(isight); + mutex_unlock(&isight->mutex); + + snd_card_free_when_closed(isight->card); + + return 0; +} + +static void isight_bus_reset(struct fw_unit *unit) +{ + struct isight *isight = dev_get_drvdata(&unit->device); + + mutex_lock(&isight->mutex); + if (fw_iso_resources_update(&isight->resources) < 0) { + isight_pcm_abort(isight); + isight_stop_streaming(isight); + } + mutex_unlock(&isight->mutex); +} + +static const struct ieee1394_device_id isight_id_table[] = { + { + .match_flags = IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .specifier_id = OUI_APPLE, + .version = SW_ISIGHT_AUDIO, + }, + { } +}; +MODULE_DEVICE_TABLE(ieee1394, isight_id_table); + +static struct fw_driver isight_driver = { + .driver = { + .owner = THIS_MODULE, + .name = KBUILD_MODNAME, + .bus = &fw_bus_type, + .probe = isight_probe, + .remove = isight_remove, + }, + .update = isight_bus_reset, + .id_table = isight_id_table, +}; + +static int __init alsa_isight_init(void) +{ + return driver_register(&isight_driver.driver); +} + +static void __exit alsa_isight_exit(void) +{ + driver_unregister(&isight_driver.driver); +} + +module_init(alsa_isight_init); +module_exit(alsa_isight_exit); diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index 775dbd5f344..9d4a6714f9e 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -36,6 +36,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit) return 0; } +EXPORT_SYMBOL(fw_iso_resources_init); /** * fw_iso_resources_destroy - destroy a resource manager @@ -48,6 +49,7 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r) mutex_destroy(&r->mutex); fw_unit_put(r->unit); } +EXPORT_SYMBOL(fw_iso_resources_destroy); static unsigned int packet_bandwidth(unsigned int max_payload_bytes, int speed) { @@ -152,6 +154,7 @@ retry_after_bus_reset: return channel; } +EXPORT_SYMBOL(fw_iso_resources_allocate); /** * fw_iso_resources_update - update resource allocations after a bus reset @@ -203,6 +206,7 @@ int fw_iso_resources_update(struct fw_iso_resources *r) return channel; } +EXPORT_SYMBOL(fw_iso_resources_update); /** * fw_iso_resources_free - frees allocated resources @@ -230,3 +234,4 @@ void fw_iso_resources_free(struct fw_iso_resources *r) mutex_unlock(&r->mutex); } +EXPORT_SYMBOL(fw_iso_resources_free); diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c index 1e20e60ba6a..3c61ca2e615 100644 --- a/sound/firewire/packets-buffer.c +++ b/sound/firewire/packets-buffer.c @@ -60,6 +60,7 @@ err_packets: error: return err; } +EXPORT_SYMBOL(iso_packets_buffer_init); /** * iso_packets_buffer_destroy - frees packet buffer resources @@ -72,3 +73,4 @@ void iso_packets_buffer_destroy(struct iso_packets_buffer *b, fw_iso_buffer_destroy(&b->iso_buffer, fw_parent_device(unit)->card); kfree(b->packets); } +EXPORT_SYMBOL(iso_packets_buffer_destroy); -- cgit v1.2.3 From 03c29680d49662859d14d64f8673550fa3fb2ed1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 11 May 2011 10:47:30 +0200 Subject: ALSA: isight: fix isight_pcm_abort() crashes Fix crashes in isight_pcm_abort() that happen when the driver tries to access isight->pcm->runtime which does not exist when the device is not open. Introduce a new field pcm_active to track this state. Signed-off-by: Clemens Ladisch Reported-by: Stefan Richter Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index a6f19f57a1c..0230605c917 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -56,6 +56,7 @@ struct isight { struct iso_packets_buffer buffer; struct fw_iso_resources resources; struct fw_iso_context *context; + bool pcm_active; bool pcm_running; bool first_packet; int packet_index; @@ -131,10 +132,12 @@ static void isight_pcm_abort(struct isight *isight) { unsigned long flags; - snd_pcm_stream_lock_irqsave(isight->pcm, flags); - if (snd_pcm_running(isight->pcm)) - snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(isight->pcm, flags); + if (ACCESS_ONCE(isight->pcm_active)) { + snd_pcm_stream_lock_irqsave(isight->pcm, flags); + if (snd_pcm_running(isight->pcm)) + snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(isight->pcm, flags); + } } static void isight_dropped_samples(struct isight *isight, unsigned int total) @@ -295,8 +298,17 @@ static int isight_close(struct snd_pcm_substream *substream) static int isight_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + struct isight *isight = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; + + ACCESS_ONCE(isight->pcm_active) = true; + + return 0; } static void isight_stop_streaming(struct isight *isight) @@ -322,6 +334,8 @@ static int isight_hw_free(struct snd_pcm_substream *substream) { struct isight *isight = substream->private_data; + ACCESS_ONCE(isight->pcm_active) = false; + mutex_lock(&isight->mutex); isight_stop_streaming(isight); mutex_unlock(&isight->mutex); -- cgit v1.2.3 From 898732d1f1c7181fd3e94e7d7a784edb48d09d95 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 11 May 2011 10:48:24 +0200 Subject: ALSA: isight: fix packet requeueing After handling a received packet, we want to resubmit the same packet, so do not increase the packet index too early. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 0230605c917..4e334919a70 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -198,9 +198,6 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle, } } - if (++index >= QUEUE_LENGTH) - index = 0; - err = fw_iso_context_queue(isight->context, &audio_packet, &isight->buffer.iso_buffer, isight->buffer.packets[index].offset); @@ -211,6 +208,8 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle, return; } + if (++index >= QUEUE_LENGTH) + index = 0; isight->packet_index = index; } -- cgit v1.2.3 From f2934cd499ba2c7f605787508b4cfcfa3a45b0a4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 11 May 2011 10:49:02 +0200 Subject: ALSA: isight: fix divide error when queueing packets Set the .header_size field when queueing packets to avoid a division by zero. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 4e334919a70..10a9b9b0b2c 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -82,6 +82,7 @@ MODULE_LICENSE("GPL v2"); static struct fw_iso_packet audio_packet = { .payload_length = sizeof(struct audio_payload), .interrupt = 1, + .header_length = 4, }; static void isight_update_pointers(struct isight *isight, unsigned int count) -- cgit v1.2.3 From 8839eedafd2e91e5b124730825e9b39b1ff493dd Mon Sep 17 00:00:00 2001 From: Stefan Richter Date: Wed, 11 May 2011 10:49:58 +0200 Subject: ALSA: isight: add AudioEnable register write which is needed to get the iSight to talk. Signed-off-by: Stefan Richter Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 10a9b9b0b2c..1a8da2614db 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -345,7 +345,7 @@ static int isight_hw_free(struct snd_pcm_substream *substream) static int isight_start_streaming(struct isight *isight) { - __be32 sample_rate; + __be32 value; unsigned int i; int err; @@ -356,10 +356,10 @@ static int isight_start_streaming(struct isight *isight) return 0; } - sample_rate = cpu_to_be32(RATE_48000); + value = cpu_to_be32(RATE_48000); err = snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, isight->audio_base + REG_SAMPLE_RATE, - &sample_rate, 4); + &value, 4); if (err < 0) return err; @@ -367,6 +367,13 @@ static int isight_start_streaming(struct isight *isight) if (err < 0) goto error; + value = cpu_to_be32(AUDIO_ENABLE); + err = snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_AUDIO_ENABLE, + &value, 4); + if (err < 0) + goto err_resources; + isight->context = fw_iso_context_create(isight->device->card, FW_ISO_CONTEXT_RECEIVE, isight->resources.channel, @@ -400,6 +407,10 @@ err_context: fw_iso_context_destroy(isight->context); isight->context = NULL; err_resources: + value = 0; + snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + REG_AUDIO_ENABLE, + &value, 4); fw_iso_resources_free(&isight->resources); error: return err; -- cgit v1.2.3 From ac34dad26e6786257ef54d8df4f883825bea02eb Mon Sep 17 00:00:00 2001 From: Stefan Richter Date: Wed, 11 May 2011 10:52:21 +0200 Subject: ALSA: isight: wrap up register accesses Signed-off-by: Stefan Richter [cl: removed superfluous variable] Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 76 +++++++++++++++++++------------------------------ 1 file changed, 30 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 1a8da2614db..4d2edcfdbbc 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -5,6 +5,7 @@ * Licensed under the terms of the GNU General Public License, version 2. */ +#include #include #include #include @@ -311,23 +312,28 @@ static int isight_hw_params(struct snd_pcm_substream *substream, return 0; } -static void isight_stop_streaming(struct isight *isight) +static int reg_read(struct isight *isight, int offset, __be32 *value) { - __be32 value; + return snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, + isight->audio_base + offset, value, 4); +} + +static int reg_write(struct isight *isight, int offset, __be32 value) +{ + return snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, + isight->audio_base + offset, &value, 4); +} +static void isight_stop_streaming(struct isight *isight) +{ if (!isight->context) return; fw_iso_context_stop(isight->context); fw_iso_context_destroy(isight->context); isight->context = NULL; - - value = 0; - snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, - isight->audio_base + REG_AUDIO_ENABLE, - &value, 4); - fw_iso_resources_free(&isight->resources); + reg_write(isight, REG_AUDIO_ENABLE, 0); } static int isight_hw_free(struct snd_pcm_substream *substream) @@ -345,7 +351,6 @@ static int isight_hw_free(struct snd_pcm_substream *substream) static int isight_start_streaming(struct isight *isight) { - __be32 value; unsigned int i; int err; @@ -356,21 +361,15 @@ static int isight_start_streaming(struct isight *isight) return 0; } - value = cpu_to_be32(RATE_48000); - err = snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, - isight->audio_base + REG_SAMPLE_RATE, - &value, 4); + err = reg_write(isight, REG_SAMPLE_RATE, cpu_to_be32(RATE_48000)); if (err < 0) - return err; + goto error; err = isight_connect(isight); if (err < 0) goto error; - value = cpu_to_be32(AUDIO_ENABLE); - err = snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, - isight->audio_base + REG_AUDIO_ENABLE, - &value, 4); + err = reg_write(isight, REG_AUDIO_ENABLE, cpu_to_be32(AUDIO_ENABLE)); if (err < 0) goto err_resources; @@ -407,11 +406,8 @@ err_context: fw_iso_context_destroy(isight->context); isight->context = NULL; err_resources: - value = 0; - snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, - isight->audio_base + REG_AUDIO_ENABLE, - &value, 4); fw_iso_resources_free(&isight->resources); + reg_write(isight, REG_AUDIO_ENABLE, 0); error: return err; } @@ -503,8 +499,7 @@ static int isight_gain_get(struct snd_kcontrol *ctl, __be32 gain; int err; - err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, - isight->audio_base + REG_GAIN, &gain, 4); + err = reg_read(isight, REG_GAIN, &gain); if (err < 0) return err; @@ -517,15 +512,13 @@ static int isight_gain_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct isight *isight = ctl->private_data; - __be32 gain; if (value->value.integer.value[0] < isight->gain_min || value->value.integer.value[0] > isight->gain_max) return -EINVAL; - gain = cpu_to_be32(value->value.integer.value[0]); - return snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, - isight->audio_base + REG_GAIN, &gain, 4); + return reg_write(isight, REG_GAIN, + cpu_to_be32(value->value.integer.value[0])); } static int isight_mute_get(struct snd_kcontrol *ctl, @@ -535,8 +528,7 @@ static int isight_mute_get(struct snd_kcontrol *ctl, __be32 mute; int err; - err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, - isight->audio_base + REG_MUTE, &mute, 4); + err = reg_read(isight, REG_MUTE, &mute); if (err < 0) return err; @@ -549,11 +541,9 @@ static int isight_mute_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct isight *isight = ctl->private_data; - __be32 mute; - mute = (__force __be32)!value->value.integer.value[0]; - return snd_fw_transaction(isight->unit, TCODE_WRITE_QUADLET_REQUEST, - isight->audio_base + REG_MUTE, &mute, 4); + return reg_write(isight, REG_MUTE, + (__force __be32)!value->value.integer.value[0]); } static int isight_create_mixer(struct isight *isight) @@ -578,31 +568,25 @@ static int isight_create_mixer(struct isight *isight) struct snd_kcontrol *ctl; int err; - err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, - isight->audio_base + REG_GAIN_RAW_START, - &value, 4); + err = reg_read(isight, REG_GAIN_RAW_START, &value); if (err < 0) return err; isight->gain_min = be32_to_cpu(value); - err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, - isight->audio_base + REG_GAIN_RAW_END, - &value, 4); + err = reg_read(isight, REG_GAIN_RAW_END, &value); if (err < 0) return err; isight->gain_max = be32_to_cpu(value); isight->gain_tlv[0] = SNDRV_CTL_TLVT_DB_MINMAX; isight->gain_tlv[1] = 2 * sizeof(unsigned int); - err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, - isight->audio_base + REG_GAIN_DB_START, - &value, 4); + + err = reg_read(isight, REG_GAIN_DB_START, &value); if (err < 0) return err; isight->gain_tlv[2] = (s32)be32_to_cpu(value) * 100; - err = snd_fw_transaction(isight->unit, TCODE_READ_QUADLET_REQUEST, - isight->audio_base + REG_GAIN_DB_END, - &value, 4); + + err = reg_read(isight, REG_GAIN_DB_END, &value); if (err < 0) return err; isight->gain_tlv[3] = (s32)be32_to_cpu(value) * 100; -- cgit v1.2.3 From aee70400184b6a8d39243b02c244aed61259a46b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 11 May 2011 10:53:12 +0200 Subject: ALSA: isight: fix hang when unplugging a running device When aborting a PCM stream, the xrun is signaled only if the stream is running. When disconnecting a PCM stream, calling snd_card_disconnect() too early would change the stream into a non-running state and thus prevent the xrun from being noticed by user space. To prevent this, move the snd_card_disconnect() call after the xrun. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 4d2edcfdbbc..96267437d37 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -692,10 +692,9 @@ static int isight_remove(struct device *dev) { struct isight *isight = dev_get_drvdata(dev); - snd_card_disconnect(isight->card); - mutex_lock(&isight->mutex); isight_pcm_abort(isight); + snd_card_disconnect(isight->card); isight_stop_streaming(isight); mutex_unlock(&isight->mutex); -- cgit v1.2.3 From 3cabffd72c303c3b5bbbbe88c95b49043898d1f3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 11 May 2011 10:54:41 +0200 Subject: ALSA: isight: remove experimental status Experiments have shown this driver to work now. Signed-off-by: Clemens Ladisch Tested-by: Stefan Richter Signed-off-by: Takashi Iwai --- sound/firewire/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 312f90d7ec6..26071489970 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -23,8 +23,7 @@ config SND_FIREWIRE_SPEAKERS will be called snd-firewire-speakers. config SND_ISIGHT - tristate "Apple iSight microphone (EXPERIMENTAL)" - depends on EXPERIMENTAL + tristate "Apple iSight microphone" select SND_PCM select SND_FIREWIRE_LIB help -- cgit v1.2.3 From f3f7c1837f6bcae3601fc535b339426868bf1549 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 11 May 2011 11:07:09 +0200 Subject: ALSA: isight: fix locking Lockdep complains about conflicts between isight->mutex, ALSA's register_mutex, mm->mmap_sem, and pcm->open_mutex. This can be fixed by moving the calls to isight_pcm_abort(), snd_card_disconnect(), and fw_iso_resources_update() out of isight->mutex. These functions are designed to be called asynchronously; the mutex needs to protect only the device streaming state modified by isight_start/stop_streaming(). Signed-off-by: Clemens Ladisch Reported-by: Stefan Richter Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 96267437d37..86ee16ca365 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -692,9 +692,11 @@ static int isight_remove(struct device *dev) { struct isight *isight = dev_get_drvdata(dev); - mutex_lock(&isight->mutex); isight_pcm_abort(isight); + snd_card_disconnect(isight->card); + + mutex_lock(&isight->mutex); isight_stop_streaming(isight); mutex_unlock(&isight->mutex); @@ -707,12 +709,13 @@ static void isight_bus_reset(struct fw_unit *unit) { struct isight *isight = dev_get_drvdata(&unit->device); - mutex_lock(&isight->mutex); if (fw_iso_resources_update(&isight->resources) < 0) { isight_pcm_abort(isight); + + mutex_lock(&isight->mutex); isight_stop_streaming(isight); + mutex_unlock(&isight->mutex); } - mutex_unlock(&isight->mutex); } static const struct ieee1394_device_id isight_id_table[] = { -- cgit v1.2.3 From 3afb1b3e6fa358d8c7e652b80be6d0cde07b4b40 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 May 2011 00:01:58 +0200 Subject: ASoC: Fix NULL vs. 0 warning in SSM2602 sparse complains if 0 is used as a NULL pointer constant. Signed-off-by: Mark Brown Acked-by: Mike Frysinger Acked-by: Liam Girdwood --- sound/soc/codecs/ssm2602.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 70099c9d63c..84f4ad56855 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -137,7 +137,7 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), -SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0), +SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), -- cgit v1.2.3 From 051e994e9506e64259df34e91b028fe2470fbf2a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 May 2011 13:47:47 +0200 Subject: ASoC: Don't squash 16x8 registers down to 8 bits Currently we'll force all registers to fit in 8 bits before passing down to the I/O function. Looks like a cut'n'paste bug. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 687beec5647..6e5e30a17f8 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -319,7 +319,6 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; - reg &= 0xff; return do_hw_write(codec, reg, value, data, 3); } -- cgit v1.2.3 From 063b7cc43fb2413238095b81f9b4a2ee2c52056b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 May 2011 23:55:21 +0200 Subject: ASoC: Remove byte swap in 4x12 SPI write snd_soc_4_12_spi_write() contains a byte swap. Since this code was written for an Analog CODEC on a Blackfin reference board it appears that this is done because while Blackfin is little endian the CODEC is big endian (as are most CODECs). Push this up into the generic 4x12 write function and use cpu_to_be16() to do the byte swap so things are more regular and things work on both CPU endiannesses. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 6e5e30a17f8..73907e5f078 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -101,12 +101,11 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; + u16 data; - data[0] = (reg << 4) | ((value >> 8) & 0x000f); - data[1] = value & 0x00ff; + data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - return do_hw_write(codec, reg, value, data, 2); + return do_hw_write(codec, reg, value, &data, 2); } #if defined(CONFIG_SPI_MASTER) @@ -115,8 +114,8 @@ static int snd_soc_4_12_spi_write(void *control_data, const char *data, { u8 msg[2]; - msg[0] = data[1]; - msg[1] = data[0]; + msg[0] = data[0]; + msg[1] = data[1]; return do_spi_write(control_data, msg, len); } -- cgit v1.2.3 From 6e28f976ec73aec688bf89a63ac719ba765e9360 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 May 2011 23:33:48 +0200 Subject: ASoC: Use spi_write() for SPI writes do_spi_write() is just an open coded copy of do_spi_write() so we can delete it and just call spi_write() directly. Indeed, as a result of recent refactoring all the SPI write functions are just very long wrappers around spi_write() which don't add anything except for some pointless copies so we can just use spi_write() as the hw_write operation directly. It should be as type safe to do this as it is to do the same thing with I2C and it saves us a bunch of code. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 132 ++------------------------------------------------ 1 file changed, 4 insertions(+), 128 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 73907e5f078..01e5b984643 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,30 +20,6 @@ #include -#if defined(CONFIG_SPI_MASTER) -static int do_spi_write(void *control_data, const void *msg, - int len) -{ - struct spi_device *spi = control_data; - struct spi_transfer t; - struct spi_message m; - - if (len <= 0) - return 0; - - spi_message_init(&m); - memset(&t, 0, sizeof t); - - t.tx_buf = msg; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} -#endif - static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value, const void *data, int len) { @@ -108,21 +84,6 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, &data, 2); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_4_12_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[2]; - - msg[0] = data[0]; - msg[1] = data[1]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_4_12_spi_write NULL -#endif - static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -140,21 +101,6 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, data, 2); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_7_9_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[2]; - - msg[0] = data[0]; - msg[1] = data[1]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_7_9_spi_write NULL -#endif - static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -173,21 +119,6 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, return do_hw_read(codec, reg); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_8_8_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[2]; - - msg[0] = data[0]; - msg[1] = data[1]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_8_8_spi_write NULL -#endif - static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -206,22 +137,6 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, return do_hw_read(codec, reg); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_8_16_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[3]; - - msg[0] = data[0]; - msg[1] = data[1]; - msg[2] = data[2]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_8_16_spi_write NULL -#endif - #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int do_i2c_read(struct snd_soc_codec *codec, void *reg, int reglen, @@ -322,22 +237,6 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, data, 3); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_16_8_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[3]; - - msg[0] = data[0]; - msg[1] = data[1]; - msg[2] = data[2]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_16_8_spi_write NULL -#endif - #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, unsigned int r) @@ -374,23 +273,6 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, return do_hw_write(codec, reg, value, data, 4); } -#if defined(CONFIG_SPI_MASTER) -static int snd_soc_16_16_spi_write(void *control_data, const char *data, - int len) -{ - u8 msg[4]; - - msg[0] = data[0]; - msg[1] = data[1]; - msg[2] = data[2]; - msg[3] = data[3]; - - return do_spi_write(control_data, msg, len); -} -#else -#define snd_soc_16_16_spi_write NULL -#endif - /* Primitive bulk write support for soc-cache. The data pointed to by * `data' needs to already be in the form the hardware expects * including any leading register specific data. Any data written @@ -420,7 +302,7 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r #endif #if defined(CONFIG_SPI_MASTER) case SND_SOC_SPI: - ret = do_spi_write(codec->control_data, data, len); + ret = spi_write(codec->control_data, data, len); break; #endif default: @@ -439,43 +321,36 @@ static struct { int addr_bits; int data_bits; int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - int (*spi_write)(void *, const char *, int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, .write = snd_soc_4_12_write, .read = snd_soc_4_12_read, - .spi_write = snd_soc_4_12_spi_write, }, { .addr_bits = 7, .data_bits = 9, .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, - .spi_write = snd_soc_7_9_spi_write, }, { .addr_bits = 8, .data_bits = 8, .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, .i2c_read = snd_soc_8_8_read_i2c, - .spi_write = snd_soc_8_8_spi_write, }, { .addr_bits = 8, .data_bits = 16, .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, .i2c_read = snd_soc_8_16_read_i2c, - .spi_write = snd_soc_8_16_spi_write, }, { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .read = snd_soc_16_8_read, .i2c_read = snd_soc_16_8_read_i2c, - .spi_write = snd_soc_16_8_spi_write, }, { .addr_bits = 16, .data_bits = 16, .write = snd_soc_16_16_write, .read = snd_soc_16_16_read, .i2c_read = snd_soc_16_16_read_i2c, - .spi_write = snd_soc_16_16_spi_write, }, }; @@ -536,8 +411,9 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, break; case SND_SOC_SPI: - if (io_types[i].spi_write) - codec->hw_write = io_types[i].spi_write; +#ifdef CONFIG_SPI_MASTER + codec->hw_write = (hw_write_t)spi_write; +#endif codec->control_data = container_of(codec->dev, struct spi_device, -- cgit v1.2.3 From ca629928b9d5b28789c4b59729113e9d2b1bc7c0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 May 2011 14:34:53 +0200 Subject: ASoC: Disable WM8994/58 microphone detection over suspend It will be non-functional with the basises and clocks off anyway, if the system needs microphone detection enabled over suspend then it should be causing the CODEC to ignore suspend using the APIs for that to prevent the biases being disabled. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b6d47e77151..e6dfa103ad2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2416,8 +2416,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; + switch (control->type) { + case WM8994: + snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); + break; + case WM8958: + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, 0); + break; + } + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -2435,6 +2446,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8994_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; unsigned int val, mask; @@ -2473,6 +2485,19 @@ static int wm8994_resume(struct snd_soc_codec *codec) i + 1, ret); } + switch (control->type) { + case WM8994: + if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) + snd_soc_update_bits(codec, WM8994_MICBIAS, + WM8994_MICD_ENA, WM8994_MICD_ENA); + break; + case WM8958: + if (wm8994->jack_cb) + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, WM8958_MICD_ENA); + break; + } + return 0; } #else -- cgit v1.2.3 From 93864cf04283eb2899ef13ded472a9f24538303f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 May 2011 18:11:36 +0300 Subject: ASoC: tlv320dac33: Update e-mail address Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 4 ++-- sound/soc/codecs/tlv320dac33.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 90c361ef598..faa5e9fb147 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1,7 +1,7 @@ /* * ALSA SoC Texas Instruments TLV320DAC33 codec driver * - * Author: Peter Ujfalusi + * Author: Peter Ujfalusi * * Copyright: (C) 2009 Nokia Corporation * @@ -1658,5 +1658,5 @@ module_exit(dac33_module_exit); MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); -MODULE_AUTHOR("Peter Ujfalusi "); +MODULE_AUTHOR("Peter Ujfalusi "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h index 7c318b5da43..ed69670747b 100644 --- a/sound/soc/codecs/tlv320dac33.h +++ b/sound/soc/codecs/tlv320dac33.h @@ -1,7 +1,7 @@ /* * ALSA SoC Texas Instruments TLV320DAC33 codec driver * - * Author: Peter Ujfalusi + * Author: Peter Ujfalusi * * Copyright: (C) 2009 Nokia Corporation * -- cgit v1.2.3 From b4079ef40a5ae87dc0d29bded7a682b1cbe626ad Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 May 2011 18:12:41 +0300 Subject: ASoC: tpa6130a2: Update e-mail address Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tpa6130a2.c | 4 ++-- sound/soc/codecs/tpa6130a2.h | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 1f1ac8110be..239e0c46106 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -3,7 +3,7 @@ * * Copyright (C) Nokia Corporation * - * Author: Peter Ujfalusi + * Author: Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -495,7 +495,7 @@ static void __exit tpa6130a2_exit(void) i2c_del_driver(&tpa6130a2_i2c_driver); } -MODULE_AUTHOR("Peter Ujfalusi"); +MODULE_AUTHOR("Peter Ujfalusi "); MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h index 5df49c8756b..417444020ba 100644 --- a/sound/soc/codecs/tpa6130a2.h +++ b/sound/soc/codecs/tpa6130a2.h @@ -3,7 +3,7 @@ * * Copyright (C) Nokia Corporation * - * Author: Peter Ujfalusi + * Author: Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- cgit v1.2.3 From 56a874291606f996da26063f7e90ef2864884bc8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 May 2011 18:14:06 +0300 Subject: ASoC: omap-mcbsp: Update e-mail address Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcbsp.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2175f09e57b..afbc7be803d 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula - * Peter Ujfalusi + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 37dc7211ed3..9a7dedd6f5a 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula - * Peter Ujfalusi + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- cgit v1.2.3 From 1c7687b9958369e4e41018f4e2b4e0f901e13722 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 May 2011 18:14:43 +0300 Subject: ASoC: omap-pcm: Update e-mail address Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/omap/omap-pcm.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 8caeb8d305c..0c6eb30e92b 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula - * Peter Ujfalusi + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index fea0515331f..a0ed1dbb52d 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -4,7 +4,7 @@ * Copyright (C) 2008 Nokia Corporation * * Contact: Jarkko Nikula - * Peter Ujfalusi + * Peter Ujfalusi * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License -- cgit v1.2.3 From 0ac3a014b8cb0ea9a80d4f5c551ad1a7fa258a3c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 May 2011 18:15:10 +0300 Subject: ASoC: RX51: Update e-mail address Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/rx51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index d0986220eff..0aae998b654 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -3,7 +3,7 @@ * * Copyright (C) 2008 - 2009 Nokia Corporation * - * Contact: Peter Ujfalusi + * Contact: Peter Ujfalusi * Eduardo Valentin * Jarkko Nikula * -- cgit v1.2.3 From 9e53d856af0db8acc8a1bc6a9e3298f6bbb8944b Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Sun, 8 May 2011 09:48:24 -0700 Subject: ASoC: fix wm8958-dsp2 printk format warnings Fix printk format warnings in wm8958-dsp2.c: sound/soc/codecs/wm8958-dsp2.c:103: warning: format '%d' expects type 'int', but argument 4 has type 'size_t' sound/soc/codecs/wm8958-dsp2.c:111: warning: format '%d' expects type 'int', but argument 3 has type 'size_t' sound/soc/codecs/wm8958-dsp2.c:144: warning: format '%d' expects type 'int', but argument 5 has type 'size_t' Signed-off-by: Randy Dunlap Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8958-dsp2.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 74983ee2b87..5d4bc7a21df 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -99,7 +99,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, len = fw->size - len; while (len) { if (len < 12) { - dev_err(codec->dev, "%s short data block of %d\n", + dev_err(codec->dev, "%s short data block of %zd\n", name, len); goto err; } @@ -107,7 +107,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, memcpy(&data32, data + 4, sizeof(data32)); block_len = be32_to_cpu(data32); if (block_len + 8 > len) { - dev_err(codec->dev, "%d byte block longer than file\n", + dev_err(codec->dev, "%zd byte block longer than file\n", block_len); goto err; } @@ -141,7 +141,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, case WM_FW_BLOCK_I: case WM_FW_BLOCK_A: case WM_FW_BLOCK_C: - dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name, + dev_dbg(codec->dev, "%s: %zd bytes of %x@%x\n", name, block_len, (data32 >> 24) & 0xff, data32 & 0xffffff); -- cgit v1.2.3 From 22de71ba03311cdc1063757c50a1488cb90a1fca Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 12 May 2011 16:14:04 +0100 Subject: ASoC: core - allow ASoC more flexible machine name Allow ASoC machine drivers to register a driver name and a longname. This allows user space to determine the flavour of machine driver. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a477e218aa2..c1a4cf480c7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1926,9 +1926,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) card->num_dapm_routes); snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), - "%s", card->name); - snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->name); + snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), + "%s", card->long_name ? card->long_name : card->name); + snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), + "%s", card->driver_name); if (card->late_probe) { ret = card->late_probe(card); -- cgit v1.2.3 From d5e4b0adf6e694f9a8fb31f51fc381bd8be17eae Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Thu, 12 May 2011 16:26:20 +0100 Subject: ASoC: DMIC codec - Add input widget Digital microphones can have some additional elements in their audio path (like microphone bias). An input widget is required for digital microphone CODEC driver to allow external connections in machine drivers. Signed-off-by: Misael Lopez Cruz Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/dmic.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 57e9dac88d3..f9a87737ec1 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -39,7 +39,31 @@ static struct snd_soc_dai_driver dmic_dai = { }, }; -static struct snd_soc_codec_driver soc_dmic = {}; +static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_INPUT("DMic"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DMIC AIF", NULL, "DMic"}, +}; + +static int dmic_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, + ARRAY_SIZE(dmic_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + snd_soc_dapm_new_widgets(dapm); + + return 0; +} + +static struct snd_soc_codec_driver soc_dmic = { + .probe = dmic_probe, +}; static int __devinit dmic_dev_probe(struct platform_device *pdev) { -- cgit v1.2.3 From 1f71a3ba8f845d2b2538e79bafb99ac835c84ea6 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 28 Mar 2011 19:23:23 +0100 Subject: ASoC: twl6040 - fix LINEGAIN volume control Fix the TWL6040 LINEGAIN volume control to match the TRM. Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/twl6040.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 255901c4460..4c336636d4f 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -960,9 +960,9 @@ static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0); /* * AFMGAIN volume control: - * from 18 to 24 dB in 6 dB steps + * from -18 to 24 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 1800, 600, 0); +static DECLARE_TLV_DB_SCALE(afm_amp_tlv, -1800, 600, 0); /* * HSGAIN volume control: @@ -1049,7 +1049,7 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* AFM gains */ SOC_DOUBLE_TLV("Aux FM Volume", - TWL6040_REG_LINEGAIN, 0, 4, 0xF, 0, afm_amp_tlv), + TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv), /* Playback gains */ SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume", -- cgit v1.2.3 From b417382419c67fda6463e02eafb8efebd5f8759e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 12 May 2011 13:04:55 +0300 Subject: ASoC: omap-pcm: Period wakeup disabling on OMAP2+ Allow disabling ALSA period wakeup interrupts. This can only be done on OMAP2+ (2/3/4), since there we can chain the DMA. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 0c6eb30e92b..e6a6b991d05 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, @@ -195,7 +196,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); - else + else if (!substream->runtime->no_period_wakeup) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); if (!(cpu_class_is_omap1())) { -- cgit v1.2.3 From d491297752c3a36f6cfabd4cd578c0cfa2098044 Mon Sep 17 00:00:00 2001 From: Sanjeev Premi Date: Wed, 11 May 2011 19:25:35 +0530 Subject: ASoC: omap-mcbsp: Remove restrictive checks for cpu type Current checks for cpu type were too restrictive leading to failures for other silicons in same family. The problem was found while testing audio playback on AM37x and AM35x processors. But should exist on OMAP36xx as well. Signed-off-by: Sanjeev Premi cc: Mark Brown cc: Liam Girdwood cc: Jarkko Nikula Acked-by: Jarkko Nikula Acked-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-mcbsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index afbc7be803d..07b77235724 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x() || cpu_is_omap44xx()) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (cpu_is_omap343x()) { + if (cpu_is_omap34xx()) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (omap_mcbsp_get_dma_op_mode(bus_id) == -- cgit v1.2.3 From 2557f7427d4bd1fc00166556e3047c5f3ed91958 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 16:18:37 +0200 Subject: ALSA: hda - Fix auto-mic for CX2064x codecs The wrong id is assigned for external/internal mics in the auto-mic selection parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 35bebe59908..eecc154d0b6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3420,13 +3420,13 @@ static void cx_auto_check_auto_mic(struct hda_codec *codec) if (is_ext_mic(codec, cfg->inputs[0].pin) && is_int_mic(codec, cfg->inputs[1].pin)) { spec->auto_mic = 1; - spec->auto_mic_ext = 1; + spec->auto_mic_ext = 0; return; } - if (is_int_mic(codec, cfg->inputs[1].pin) && - is_ext_mic(codec, cfg->inputs[0].pin)) { + if (is_int_mic(codec, cfg->inputs[0].pin) && + is_ext_mic(codec, cfg->inputs[1].pin)) { spec->auto_mic = 1; - spec->auto_mic_ext = 0; + spec->auto_mic_ext = 1; return; } } -- cgit v1.2.3 From da33986651e137b1ea2ec21794e32bc5c57b03d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 16:24:15 +0200 Subject: ALSA: hda - Turn on EAPD dynamically per jack plug in Conexant auto mode Instead of keeping always EAPD on, turn on/off appropriately at jack plugging in Conexant auto-parser mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index eecc154d0b6..d63e15b8937 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3325,6 +3325,9 @@ static void cx_auto_parse_output(struct hda_codec *codec) spec->vmaster_nid = spec->private_dac_nids[0]; } +static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins, + hda_nid_t *pins, bool on); + /* auto-mute/unmute speaker and line outs according to headphone jack */ static void cx_auto_hp_automute(struct hda_codec *codec) { @@ -3341,11 +3344,13 @@ static void cx_auto_hp_automute(struct hda_codec *codec) break; } } + cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, present); for (i = 0; i < cfg->line_outs; i++) { snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, present ? 0 : PIN_OUT); } + cx_auto_turn_eapd(codec, cfg->line_outs, cfg->line_out_pins, !present); for (i = 0; !present && i < cfg->line_outs; i++) if (snd_hda_jack_detect(codec, cfg->line_out_pins[i])) present = 1; @@ -3354,6 +3359,7 @@ static void cx_auto_hp_automute(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, present ? 0 : PIN_OUT); } + cx_auto_turn_eapd(codec, cfg->speaker_outs, cfg->speaker_pins, !present); } /* automatic switch internal and external mic */ @@ -3517,14 +3523,15 @@ static int cx_auto_parse_auto_config(struct hda_codec *codec) return 0; } -static void cx_auto_turn_on_eapd(struct hda_codec *codec, int num_pins, - hda_nid_t *pins) +static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins, + hda_nid_t *pins, bool on) { int i; for (i = 0; i < num_pins; i++) { if (snd_hda_query_pin_caps(codec, pins[i]) & AC_PINCAP_EAPD) snd_hda_codec_write(codec, pins[i], 0, - AC_VERB_SET_EAPD_BTLENABLE, 0x02); + AC_VERB_SET_EAPD_BTLENABLE, + on ? 0x02 : 0); } } @@ -3565,6 +3572,13 @@ static void cx_auto_init_output(struct hda_codec *codec) for (i = 0; i < cfg->speaker_outs; i++) snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* turn on EAPD */ + cx_auto_turn_eapd(codec, cfg->line_outs, cfg->line_out_pins, + true); + cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, + true); + cx_auto_turn_eapd(codec, cfg->speaker_outs, cfg->speaker_pins, + true); } for (i = 0; i < spec->dac_info_filled; i++) { @@ -3573,11 +3587,6 @@ static void cx_auto_init_output(struct hda_codec *codec) nid = spec->multiout.dac_nids[0]; select_connection(codec, spec->dac_info[i].pin, nid); } - - /* turn on EAPD */ - cx_auto_turn_on_eapd(codec, cfg->line_outs, cfg->line_out_pins); - cx_auto_turn_on_eapd(codec, cfg->hp_outs, cfg->hp_pins); - cx_auto_turn_on_eapd(codec, cfg->speaker_outs, cfg->speaker_pins); } static void cx_auto_init_input(struct hda_codec *codec) -- cgit v1.2.3 From 0ad1b5b619e4f053dfdcef9dfc68cc5142d86961 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 16:43:12 +0200 Subject: ALSA: hda - Check AMP CAP at initialization of Conexant auto-parser Some codecs have no mute caps in audio I/O widgets. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d63e15b8937..19416e305be 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3549,11 +3549,17 @@ static void cx_auto_init_output(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; - int i; + int i, val; - for (i = 0; i < spec->multiout.num_dacs; i++) - snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + for (i = 0; i < spec->multiout.num_dacs; i++) { + nid = spec->multiout.dac_nids[i]; + if (query_amp_caps(codec, nid, HDA_OUTPUT) & AC_AMPCAP_MUTE) + val = AMP_OUT_MUTE; + else + val = AMP_OUT_ZERO; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, val); + } for (i = 0; i < cfg->hp_outs; i++) snd_hda_codec_write(codec, cfg->hp_pins[i], 0, @@ -3593,11 +3599,17 @@ static void cx_auto_init_input(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - int i; + int i, val; - for (i = 0; i < spec->num_adc_nids; i++) - snd_hda_codec_write(codec, spec->adc_nids[i], 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)); + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t nid = spec->adc_nids[i]; + if (query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE) + val = AMP_IN_MUTE(0); + else + val = AMP_IN_UNMUTE(0); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + val); + } for (i = 0; i < cfg->num_inputs; i++) { unsigned int type; -- cgit v1.2.3 From 6764bcef4cb964456615565ff974ed917de3c12d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 16:52:25 +0200 Subject: ALSA: hda - Add auto-parser support to cxt5051 / CX20561 Hermosa Extend the existing auto-parser for CX2064x for cxt5051 codec. Now the auto-parser supports ADC-switching for this codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 279 ++++++++++++++++++++++++++++++++--------- 1 file changed, 217 insertions(+), 62 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 19416e305be..cdb9f499d7f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -89,6 +89,8 @@ struct conexant_spec { unsigned int cur_adc_stream_tag; unsigned int cur_adc_format; + const struct hda_pcm_stream *capture_stream; + /* capture source */ const struct hda_input_mux *input_mux; const hda_nid_t *capsrc_nids; @@ -106,6 +108,7 @@ struct conexant_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; + hda_nid_t imux_adcs[HDA_MAX_NUM_INPUTS]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; struct pin_dac_pair dac_info[8]; int dac_info_filled; @@ -119,6 +122,8 @@ struct conexant_spec { unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int adc_switching:1; + unsigned int ext_mic_present; unsigned int recording; void (*capture_prepare)(struct hda_codec *codec); @@ -319,13 +324,19 @@ static int conexant_build_pcms(struct hda_codec *codec) spec->multiout.max_channels; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; - if (codec->vendor_id == 0x14f15051) - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - cx5051_pcm_analog_capture; - else - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - conexant_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids; + if (spec->capture_stream) + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *spec->capture_stream; + else { + if (codec->vendor_id == 0x14f15051) + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + cx5051_pcm_analog_capture; + else { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + conexant_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = + spec->num_adc_nids; + } + } info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; if (spec->multiout.dig_out_nid) { @@ -564,6 +575,7 @@ static const struct hda_codec_ops conexant_patch_ops = { #define set_beep_amp(spec, nid, idx, dir) /* NOP */ #endif +static int patch_conexant_auto(struct hda_codec *codec); /* * EAPD control * the private value = nid | (invert << 8) @@ -1906,6 +1918,7 @@ enum { CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_TOSHIBA, /* Toshiba M300 & co */ CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ + CXT5051_AUTO, /* auto-parser */ CXT5051_MODELS }; @@ -1917,6 +1930,7 @@ static const char *const cxt5051_models[CXT5051_MODELS] = { [CXT5051_F700] = "hp-700", [CXT5051_TOSHIBA] = "toshiba", [CXT5051_IDEAPAD] = "ideapad", + [CXT5051_AUTO] = "auto", }; static const struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1937,6 +1951,17 @@ static int patch_cxt5051(struct hda_codec *codec) struct conexant_spec *spec; int board_config; + board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, + cxt5051_models, + cxt5051_cfg_tbl); + if (board_config < 0) + board_config = CXT5051_AUTO; + if (board_config == CXT5051_AUTO) { + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + return patch_conexant_auto(codec); + } + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; @@ -1967,9 +1992,6 @@ static int patch_cxt5051(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; - board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, - cxt5051_models, - cxt5051_cfg_tbl); spec->auto_mic = AUTO_MIC_PORTB | AUTO_MIC_PORTC; switch (board_config) { case CXT5051_HP: @@ -3195,6 +3217,44 @@ static int patch_cxt5066(struct hda_codec *codec) * Automatic parser for CX20641 & co */ +static int cx_auto_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct conexant_spec *spec = codec->spec; + hda_nid_t adc = spec->imux_adcs[spec->cur_mux[0]]; + if (spec->adc_switching) { + spec->cur_adc = adc; + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + } + snd_hda_codec_setup_stream(codec, adc, stream_tag, 0, format); + return 0; +} + +static int cx_auto_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct conexant_spec *spec = codec->spec; + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + spec->cur_adc = 0; + return 0; +} + +static const struct hda_pcm_stream cx_auto_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0, /* fill later */ + .ops = { + .prepare = cx_auto_capture_pcm_prepare, + .cleanup = cx_auto_capture_pcm_cleanup + }, +}; + static const hda_nid_t cx_auto_adc_nids[] = { 0x14 }; /* get the connection index of @nid in the widget @mux */ @@ -3211,6 +3271,15 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, return -1; } +static int has_multi_connection(struct hda_codec *codec, hda_nid_t mux) +{ + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int nums; + + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); + return nums > 1; +} + /* get an unassigned DAC from the given list. * Return the nid if found and reduce the DAC list, or return zero if * not found @@ -3362,25 +3431,89 @@ static void cx_auto_hp_automute(struct hda_codec *codec) cx_auto_turn_eapd(codec, cfg->speaker_outs, cfg->speaker_pins, !present); } +static int cx_auto_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->private_imux, uinfo); +} + +static int cx_auto_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_mux[0]; + return 0; +} + +static int cx_auto_mux_enum_update(struct hda_codec *codec, + const struct hda_input_mux *imux, + unsigned int idx) +{ + struct conexant_spec *spec = codec->spec; + hda_nid_t adc; + + if (!imux->num_items) + return 0; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + if (spec->cur_mux[0] == idx) + return 0; + adc = spec->imux_adcs[idx]; + if (has_multi_connection(codec, adc)) + snd_hda_codec_write(codec, adc, 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[idx].index); + if (spec->cur_adc && spec->cur_adc != adc) { + /* stream is running, let's swap the current ADC */ + __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); + spec->cur_adc = adc; + snd_hda_codec_setup_stream(codec, adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + } + spec->cur_mux[0] = idx; + return 1; +} + +static int cx_auto_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + + return cx_auto_mux_enum_update(codec, &spec->private_imux, + ucontrol->value.enumerated.item[0]); +} + +static const struct snd_kcontrol_new cx_auto_capture_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = cx_auto_mux_enum_info, + .get = cx_auto_mux_enum_get, + .put = cx_auto_mux_enum_put + }, + {} +}; + /* automatic switch internal and external mic */ static void cx_auto_automic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - struct hda_input_mux *imux = &spec->private_imux; int ext_idx = spec->auto_mic_ext; if (!spec->auto_mic) return; - if (snd_hda_jack_detect(codec, cfg->inputs[ext_idx].pin)) { - snd_hda_codec_write(codec, spec->adc_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[ext_idx].index); - } else { - snd_hda_codec_write(codec, spec->adc_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[!ext_idx].index); - } + if (snd_hda_jack_detect(codec, cfg->inputs[ext_idx].pin)) + cx_auto_mux_enum_update(codec, &spec->private_imux, ext_idx); + else + cx_auto_mux_enum_update(codec, &spec->private_imux, !ext_idx); } static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) @@ -3442,22 +3575,33 @@ static void cx_auto_parse_input(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; struct hda_input_mux *imux; - int i; + int i, j; imux = &spec->private_imux; for (i = 0; i < cfg->num_inputs; i++) { - int idx = get_connection_index(codec, spec->adc_nids[0], + for (j = 0; j < spec->num_adc_nids; j++) { + hda_nid_t adc = spec->adc_nids[j]; + int idx = get_connection_index(codec, adc, cfg->inputs[i].pin); - if (idx >= 0) { - const char *label; - label = hda_get_autocfg_input_label(codec, cfg, i); - snd_hda_add_imux_item(imux, label, idx, NULL); + if (idx >= 0) { + const char *label; + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, idx, NULL); + spec->imux_adcs[i] = adc; + break; + } } } if (imux->num_items == 2 && cfg->num_inputs == 2) cx_auto_check_auto_mic(codec); - if (imux->num_items > 1 && !spec->auto_mic) - spec->input_mux = imux; + if (imux->num_items > 1 && !spec->auto_mic) { + for (i = 1; i < imux->num_items; i++) { + if (spec->imux_adcs[i] != spec->imux_adcs[0]) { + spec->adc_switching = 1; + break; + } + } + } } /* get digital-input audio widget corresponding to the given pin */ @@ -3736,39 +3880,37 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) return 0; } +static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, + const char *label, const char *pfx, + int cidx) +{ + struct conexant_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t adc_nid = spec->adc_nids[i]; + int idx = get_connection_index(codec, adc_nid, nid); + if (idx < 0) + continue; + return cx_auto_add_volume_idx(codec, label, pfx, + cidx, adc_nid, HDA_INPUT, idx); + } + return 0; +} + static int cx_auto_build_input_controls(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; static const char *prev_label; - int i, err, cidx, conn_len; - hda_nid_t conn[HDA_MAX_CONNECTIONS]; - - int multi_adc_volume = 0; /* If the ADC nid has several input volumes */ - int adc_nid = spec->adc_nids[0]; - - conn_len = snd_hda_get_connections(codec, adc_nid, conn, - HDA_MAX_CONNECTIONS); - if (conn_len < 0) - return conn_len; - - multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1; - if (!multi_adc_volume) { - err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid, - HDA_INPUT); - if (err < 0) - return err; - } + int i, err, cidx; prev_label = NULL; cidx = 0; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; const char *label; - int j; int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP; - if (!pin_amp && !multi_adc_volume) - continue; label = hda_get_autocfg_input_label(codec, cfg, i); if (label == prev_label) @@ -3784,18 +3926,23 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) return err; } - if (!multi_adc_volume) - continue; - for (j = 0; j < conn_len; j++) { - if (conn[j] == nid) { - err = cx_auto_add_volume_idx(codec, label, - " Capture", cidx, adc_nid, HDA_INPUT, j); - if (err < 0) - return err; - break; - } + if (cfg->num_inputs == 1) { + err = cx_auto_add_capture_volume(codec, nid, + "Capture", "", cidx); + } else { + err = cx_auto_add_capture_volume(codec, nid, + label, " Capture", cidx); } + if (err < 0) + return err; + } + + if (spec->private_imux.num_items > 1 && !spec->auto_mic) { + err = snd_hda_add_new_ctls(codec, cx_auto_capture_mixers); + if (err < 0) + return err; } + return 0; } @@ -3833,15 +3980,23 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; - spec->adc_nids = cx_auto_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(cx_auto_adc_nids); - spec->capsrc_nids = spec->adc_nids; + if (codec->vendor_id == 0x14f15051) { + codec->pin_amp_workaround = 1; + spec->adc_nids = cxt5051_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(cxt5051_adc_nids); + spec->capsrc_nids = spec->adc_nids; + } else { + spec->adc_nids = cx_auto_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(cx_auto_adc_nids); + spec->capsrc_nids = spec->adc_nids; + } err = cx_auto_parse_auto_config(codec); if (err < 0) { kfree(codec->spec); codec->spec = NULL; return err; } + spec->capture_stream = &cx_auto_pcm_analog_capture; codec->patch_ops = cx_auto_patch_ops; if (spec->beep_amp) snd_hda_attach_beep_device(codec, spec->beep_amp); -- cgit v1.2.3 From 1f8458a26293b692955f8dff671a3ed50dd9c603 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 17:22:05 +0200 Subject: ALSA: hda - Add auto-parser support to cxt5045 / CX20549 Venice Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 84 ++++++++++++++++++++++++++++++++---------- 1 file changed, 65 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index cdb9f499d7f..623cd9be547 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1059,6 +1059,7 @@ enum { #ifdef CONFIG_SND_DEBUG CXT5045_TEST, #endif + CXT5045_AUTO, CXT5045_MODELS }; @@ -1071,6 +1072,7 @@ static const char * const cxt5045_models[CXT5045_MODELS] = { #ifdef CONFIG_SND_DEBUG [CXT5045_TEST] = "test", #endif + [CXT5045_AUTO] = "auto", }; static const struct snd_pci_quirk cxt5045_cfg_tbl[] = { @@ -1097,6 +1099,14 @@ static int patch_cxt5045(struct hda_codec *codec) struct conexant_spec *spec; int board_config; + board_config = snd_hda_check_board_config(codec, CXT5045_MODELS, + cxt5045_models, + cxt5045_cfg_tbl); + if (board_config < 0) + board_config = CXT5045_AUTO; + if (board_config == CXT5045_AUTO) + return patch_conexant_auto(codec); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; @@ -1123,9 +1133,6 @@ static int patch_cxt5045(struct hda_codec *codec) codec->patch_ops = conexant_patch_ops; - board_config = snd_hda_check_board_config(codec, CXT5045_MODELS, - cxt5045_models, - cxt5045_cfg_tbl); switch (board_config) { case CXT5045_LAPTOP_HPSENSE: codec->patch_ops.unsol_event = cxt5045_hp_unsol_event; @@ -1956,11 +1963,8 @@ static int patch_cxt5051(struct hda_codec *codec) cxt5051_cfg_tbl); if (board_config < 0) board_config = CXT5051_AUTO; - if (board_config == CXT5051_AUTO) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); + if (board_config == CXT5051_AUTO) return patch_conexant_auto(codec); - } spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) @@ -3688,15 +3692,15 @@ static void select_connection(struct hda_codec *codec, hda_nid_t pin, AC_VERB_SET_CONNECT_SEL, idx); } -static void cx_auto_init_output(struct hda_codec *codec) +static void mute_outputs(struct hda_codec *codec, int num_nids, + const hda_nid_t *nids) { - struct conexant_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t nid; int i, val; - for (i = 0; i < spec->multiout.num_dacs; i++) { - nid = spec->multiout.dac_nids[i]; + for (i = 0; i < num_nids; i++) { + hda_nid_t nid = nids[i]; + if (!(get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) + continue; if (query_amp_caps(codec, nid, HDA_OUTPUT) & AC_AMPCAP_MUTE) val = AMP_OUT_MUTE; else @@ -3704,10 +3708,22 @@ static void cx_auto_init_output(struct hda_codec *codec) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val); } +} +static void cx_auto_init_output(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + int i; + + mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids); for (i = 0; i < cfg->hp_outs; i++) snd_hda_codec_write(codec, cfg->hp_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); + mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); + mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins); if (spec->auto_mute) { for (i = 0; i < cfg->hp_outs; i++) { snd_hda_codec_write(codec, cfg->hp_pins[i], 0, @@ -3747,6 +3763,8 @@ static void cx_auto_init_input(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) { hda_nid_t nid = spec->adc_nids[i]; + if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) + continue; if (query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE) val = AMP_IN_MUTE(0); else @@ -3839,6 +3857,19 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename, #define cx_auto_add_pb_volume(codec, nid, str, idx) \ cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) +static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, + hda_nid_t pin, const char *name, int idx) +{ + unsigned int caps; + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + caps = query_amp_caps(codec, pin, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, pin, name, idx); + return 0; +} + static int cx_auto_build_output_controls(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -3847,8 +3878,10 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) static const char * const texts[3] = { "Front", "Surround", "CLFE" }; if (spec->dac_info_filled == 1) - return cx_auto_add_pb_volume(codec, spec->dac_info[0].dac, - "Master", 0); + return try_add_pb_volume(codec, spec->dac_info[0].dac, + spec->dac_info[0].pin, + "Master", 0); + for (i = 0; i < spec->dac_info_filled; i++) { const char *label; int idx, type; @@ -3872,8 +3905,9 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) idx = num_spk++; break; } - err = cx_auto_add_pb_volume(codec, spec->dac_info[i].dac, - label, idx); + err = try_add_pb_volume(codec, spec->dac_info[i].dac, + spec->dac_info[i].pin, + label, idx); if (err < 0) return err; } @@ -3976,19 +4010,31 @@ static int patch_conexant_auto(struct hda_codec *codec) struct conexant_spec *spec; int err; + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; codec->spec = spec; - if (codec->vendor_id == 0x14f15051) { + switch (codec->vendor_id) { + case 0x14f15051: codec->pin_amp_workaround = 1; spec->adc_nids = cxt5051_adc_nids; spec->num_adc_nids = ARRAY_SIZE(cxt5051_adc_nids); spec->capsrc_nids = spec->adc_nids; - } else { + break; + case 0x14f15045: + codec->pin_amp_workaround = 1; + spec->adc_nids = cxt5045_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(cxt5045_adc_nids); + spec->capsrc_nids = spec->adc_nids; + break; + default: spec->adc_nids = cx_auto_adc_nids; spec->num_adc_nids = ARRAY_SIZE(cx_auto_adc_nids); spec->capsrc_nids = spec->adc_nids; + break; } err = cx_auto_parse_auto_config(codec); if (err < 0) { -- cgit v1.2.3 From f6100bb4b849c45caa8d6b3706756d3b8a560c92 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 18:26:39 +0200 Subject: ALSA: hda - Clean up input-mux handling in Conexant auto-parser Keep the registered input-pins in imux_pins[], and fix the inconsistent use of sepc->auto_mic_ext. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 623cd9be547..b575c9989f3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -75,7 +75,7 @@ struct conexant_spec { unsigned int cur_eapd; unsigned int hp_present; unsigned int auto_mic; - int auto_mic_ext; /* autocfg.inputs[] index for ext mic */ + int auto_mic_ext; /* imux_pins[] index for ext mic */ unsigned int need_dac_fix; hda_nid_t slave_dig_outs[2]; @@ -109,6 +109,7 @@ struct conexant_spec { struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; hda_nid_t imux_adcs[HDA_MAX_NUM_INPUTS]; + hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; struct pin_dac_pair dac_info[8]; int dac_info_filled; @@ -3509,12 +3510,11 @@ static const struct snd_kcontrol_new cx_auto_capture_mixers[] = { static void cx_auto_automic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; int ext_idx = spec->auto_mic_ext; if (!spec->auto_mic) return; - if (snd_hda_jack_detect(codec, cfg->inputs[ext_idx].pin)) + if (snd_hda_jack_detect(codec, spec->imux_pins[ext_idx])) cx_auto_mux_enum_update(codec, &spec->private_imux, ext_idx); else cx_auto_mux_enum_update(codec, &spec->private_imux, !ext_idx); @@ -3558,16 +3558,15 @@ static int is_ext_mic(struct hda_codec *codec, hda_nid_t pin) static void cx_auto_check_auto_mic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - if (is_ext_mic(codec, cfg->inputs[0].pin) && - is_int_mic(codec, cfg->inputs[1].pin)) { + if (is_ext_mic(codec, spec->imux_pins[0]) && + is_int_mic(codec, spec->imux_pins[1])) { spec->auto_mic = 1; spec->auto_mic_ext = 0; return; } - if (is_int_mic(codec, cfg->inputs[0].pin) && - is_ext_mic(codec, cfg->inputs[1].pin)) { + if (is_int_mic(codec, spec->imux_pins[0]) && + is_ext_mic(codec, spec->imux_pins[1])) { spec->auto_mic = 1; spec->auto_mic_ext = 1; return; @@ -3590,8 +3589,10 @@ static void cx_auto_parse_input(struct hda_codec *codec) if (idx >= 0) { const char *label; label = hda_get_autocfg_input_label(codec, cfg, i); + spec->imux_adcs[imux->num_items] = adc; + spec->imux_pins[imux->num_items] = + cfg->inputs[i].pin; snd_hda_add_imux_item(imux, label, idx, NULL); - spec->imux_adcs[i] = adc; break; } } -- cgit v1.2.3 From 5c9887e08761eecd4dafbdff354ef7c703efb201 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 18:36:37 +0200 Subject: ALSA: hda - Parse more deep input-source routes in Conexant auto-parser Handle not only a single-depth input-route but two-level depth routes (PIN->MUX->ADC), too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 73 ++++++++++++++++++++++++++++++------------ 1 file changed, 52 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b575c9989f3..655bd23e80d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3276,15 +3276,6 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, return -1; } -static int has_multi_connection(struct hda_codec *codec, hda_nid_t mux) -{ - hda_nid_t conn[HDA_MAX_NUM_INPUTS]; - int nums; - - nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); - return nums > 1; -} - /* get an unassigned DAC from the given list. * Return the nid if found and reduce the DAC list, or return zero if * not found @@ -3455,6 +3446,51 @@ static int cx_auto_mux_enum_get(struct snd_kcontrol *kcontrol, return 0; } +/* look for the route the given pin from mux and return the index; + * if do_select is set, actually select the route. + */ +static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux, + hda_nid_t pin, bool do_select, int depth) +{ + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int i, nums; + + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); + for (i = 0; i < nums; i++) + if (conn[i] == pin) { + if (do_select) + snd_hda_codec_write(codec, mux, 0, + AC_VERB_SET_CONNECT_SEL, i); + return i; + } + depth++; + if (depth == 2) + return -1; + for (i = 0; i < nums; i++) { + int ret = __select_input_connection(codec, conn[i], pin, + do_select, depth); + if (ret >= 0) { + if (do_select) + snd_hda_codec_write(codec, mux, 0, + AC_VERB_SET_CONNECT_SEL, i); + return ret; + } + } + return -1; +} + +static void select_input_connection(struct hda_codec *codec, hda_nid_t mux, + hda_nid_t pin) +{ + __select_input_connection(codec, mux, pin, true, 0); +} + +static int get_input_connection(struct hda_codec *codec, hda_nid_t mux, + hda_nid_t pin) +{ + return __select_input_connection(codec, mux, pin, false, 0); +} + static int cx_auto_mux_enum_update(struct hda_codec *codec, const struct hda_input_mux *imux, unsigned int idx) @@ -3469,10 +3505,8 @@ static int cx_auto_mux_enum_update(struct hda_codec *codec, if (spec->cur_mux[0] == idx) return 0; adc = spec->imux_adcs[idx]; - if (has_multi_connection(codec, adc)) - snd_hda_codec_write(codec, adc, 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[idx].index); + select_input_connection(codec, spec->imux_adcs[idx], + spec->imux_pins[idx]); if (spec->cur_adc && spec->cur_adc != adc) { /* stream is running, let's swap the current ADC */ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); @@ -3584,8 +3618,8 @@ static void cx_auto_parse_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { for (j = 0; j < spec->num_adc_nids; j++) { hda_nid_t adc = spec->adc_nids[j]; - int idx = get_connection_index(codec, adc, - cfg->inputs[i].pin); + int idx = get_input_connection(codec, adc, + cfg->inputs[i].pin); if (idx >= 0) { const char *label; label = hda_get_autocfg_input_label(codec, cfg, i); @@ -3791,11 +3825,8 @@ static void cx_auto_init_input(struct hda_codec *codec) AC_USRSP_EN | CONEXANT_MIC_EVENT); cx_auto_automic(codec); } else { - for (i = 0; i < spec->num_adc_nids; i++) { - snd_hda_codec_write(codec, spec->adc_nids[i], 0, - AC_VERB_SET_CONNECT_SEL, - spec->private_imux.items[0].index); - } + select_input_connection(codec, spec->imux_adcs[0], + spec->imux_pins[0]); } } @@ -3924,7 +3955,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < spec->num_adc_nids; i++) { hda_nid_t adc_nid = spec->adc_nids[i]; - int idx = get_connection_index(codec, adc_nid, nid); + int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; return cx_auto_add_volume_idx(codec, label, pfx, -- cgit v1.2.3 From fa5dadcbe00fd6c86a149df886d4ae2cc30c33ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 May 2011 19:33:18 +0200 Subject: ALSA: hda - Add support of auto-parser to cxt5047 / CX20551 Waikiki Similarly like other Conexant codecs, now model=auto is supported for cxt5047. But the auto-parser mode isn't activated as default yet, since BIOS pin-configs seem often broken on machines with this codec. User need to pass model=auto explicitly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 655bd23e80d..fc02751f22d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1512,6 +1512,7 @@ enum { #ifdef CONFIG_SND_DEBUG CXT5047_TEST, #endif + CXT5047_AUTO, CXT5047_MODELS }; @@ -1522,6 +1523,7 @@ static const char * const cxt5047_models[CXT5047_MODELS] = { #ifdef CONFIG_SND_DEBUG [CXT5047_TEST] = "test", #endif + [CXT5047_AUTO] = "auto", }; static const struct snd_pci_quirk cxt5047_cfg_tbl[] = { @@ -1537,6 +1539,16 @@ static int patch_cxt5047(struct hda_codec *codec) struct conexant_spec *spec; int board_config; + board_config = snd_hda_check_board_config(codec, CXT5047_MODELS, + cxt5047_models, + cxt5047_cfg_tbl); +#if 0 /* not enabled as default, as BIOS often broken for this codec */ + if (board_config < 0) + board_config = CXT5047_AUTO; +#endif + if (board_config == CXT5047_AUTO) + return patch_conexant_auto(codec); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; @@ -1560,9 +1572,6 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops = conexant_patch_ops; - board_config = snd_hda_check_board_config(codec, CXT5047_MODELS, - cxt5047_models, - cxt5047_cfg_tbl); switch (board_config) { case CXT5047_LAPTOP: spec->num_mixers = 2; @@ -4062,6 +4071,12 @@ static int patch_conexant_auto(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(cxt5045_adc_nids); spec->capsrc_nids = spec->adc_nids; break; + case 0x14f15047: + codec->pin_amp_workaround = 1; + spec->adc_nids = cxt5047_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(cxt5047_adc_nids); + spec->capsrc_nids = cxt5047_capsrc_nids; + break; default: spec->adc_nids = cx_auto_adc_nids; spec->num_adc_nids = ARRAY_SIZE(cx_auto_adc_nids); -- cgit v1.2.3 From ea27316e4cd13b25727715c0db8adb0b1661f5e7 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Thu, 12 May 2011 22:17:56 +0200 Subject: ALSA: tea575x: remove freq_fixup from struct freq_fixup is a constant, no need to hold it in struct snd_tea575x and set in each driver. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/i2c/other/tea575x-tuner.c | 8 ++++---- sound/pci/es1968.c | 1 - sound/pci/fm801.c | 1 - 3 files changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 31f9795daca..98ccec27e7b 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -141,9 +141,9 @@ static void snd_tea575x_get_freq(struct snd_tea575x *tea) freq /= 10; /* crystal fixup */ if (tea->tea5759) - freq += tea->freq_fixup; + freq += TEA575X_FMIF; else - freq -= tea->freq_fixup; + freq -= TEA575X_FMIF; tea->freq = freq * 16; /* from kHz */ } @@ -156,9 +156,9 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) freq /= 16; /* to kHz */ /* crystal fixup */ if (tea->tea5759) - freq -= tea->freq_fixup; + freq -= TEA575X_FMIF; else - freq += tea->freq_fixup; + freq += TEA575X_FMIF; /* freq /= 12.5 */ freq *= 10; freq /= 125; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 75c5e0ed695..3f3ff1bae81 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2794,7 +2794,6 @@ static int __devinit snd_es1968_create(struct snd_card *card, #ifdef CONFIG_SND_ES1968_RADIO chip->tea.card = card; - chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; chip->tea.ops = &snd_es1968_tea_ops; if (!snd_tea575x_init(&chip->tea)) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 5aa3fd6f25f..1d4e71cc18b 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1231,7 +1231,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, #ifdef TEA575X_RADIO chip->tea.card = card; - chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && -- cgit v1.2.3 From 3d11ba5593b801b1db85e9680d585713e6039112 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Thu, 12 May 2011 22:18:09 +0200 Subject: ALSA: tea575x: remove unused card from struct struct snd_card *card is present in struct snd_tea575x but never used. Remove it. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/es1968.c | 1 - sound/pci/fm801.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 3f3ff1bae81..ec7f8371a81 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2793,7 +2793,6 @@ static int __devinit snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO - chip->tea.card = card; chip->tea.private_data = chip; chip->tea.ops = &snd_es1968_tea_ops; if (!snd_tea575x_init(&chip->tea)) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 1d4e71cc18b..ae5c270bfa7 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1230,7 +1230,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - chip->tea.card = card; chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && -- cgit v1.2.3 From 10ca72014741554ad37c149ff0d9e93c1e3d5b7d Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Thu, 12 May 2011 22:18:22 +0200 Subject: ALSA: tea575x: use better card and bus names Provide real card and bus_info instead of hardcoded values. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/i2c/other/tea575x-tuner.c | 5 +++-- sound/pci/es1968.c | 2 ++ sound/pci/fm801.c | 2 ++ 3 files changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 98ccec27e7b..4831800239d 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -178,8 +178,9 @@ static int vidioc_querycap(struct file *file, void *priv, struct snd_tea575x *tea = video_drvdata(file); strlcpy(v->driver, "tea575x-tuner", sizeof(v->driver)); - strlcpy(v->card, tea->tea5759 ? "TEA5759" : "TEA5757", sizeof(v->card)); - sprintf(v->bus_info, "PCI"); + strlcpy(v->card, tea->card, sizeof(v->card)); + strlcat(v->card, tea->tea5759 ? " TEA5759" : " TEA5757", sizeof(v->card)); + strlcpy(v->bus_info, tea->bus_info, sizeof(v->bus_info)); v->version = RADIO_VERSION; v->capabilities = V4L2_CAP_TUNER | V4L2_CAP_RADIO; return 0; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index ec7f8371a81..ab0a6156a70 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2795,6 +2795,8 @@ static int __devinit snd_es1968_create(struct snd_card *card, #ifdef CONFIG_SND_ES1968_RADIO chip->tea.private_data = chip; chip->tea.ops = &snd_es1968_tea_ops; + strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); + sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); if (!snd_tea575x_init(&chip->tea)) printk(KERN_INFO "es1968: detected TEA575x radio\n"); #endif diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index ae5c270bfa7..05553da2f6c 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1232,6 +1232,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, #ifdef TEA575X_RADIO chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; + sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { if (snd_tea575x_init(&chip->tea)) @@ -1246,6 +1247,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, break; } } + strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card)); #endif *rchip = chip; -- cgit v1.2.3 From fdb62b500d27b2a2d6a111769e5c44c1c9f4d909 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sat, 14 May 2011 22:51:01 +0200 Subject: ALSA: fm801: clean-up radio-related Kconfig Remove TEA575X_RADIO define from fm801.c. Also update Kconfig help text to include all supported cards. Signed-off-by: Ondrej Zary Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 4 ++-- sound/pci/fm801.c | 9 ++++----- 2 files changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index f9f533fcccc..ef41825a7b1 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -560,8 +560,8 @@ config SND_FM801_TEA575X_BOOL depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_FM801 help Say Y here to include support for soundcards based on the ForteMedia - FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media - Forte SF256-PCS-02) into the snd-fm801 driver. + FM801 chip with a TEA5757 tuner (MediaForte SF256-PCS, SF256-PCP and + SF64-PCR) into the snd-fm801 driver. config SND_TEA575X tristate diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 05553da2f6c..eacd4901a30 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -38,7 +38,6 @@ #ifdef CONFIG_SND_FM801_TEA575X_BOOL #include -#define TEA575X_RADIO 1 #endif MODULE_AUTHOR("Jaroslav Kysela "); @@ -196,7 +195,7 @@ struct fm801 { spinlock_t reg_lock; struct snd_info_entry *proc_entry; -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_TEA575X_BOOL struct snd_tea575x tea; #endif @@ -715,7 +714,7 @@ static int __devinit snd_fm801_pcm(struct fm801 *chip, int device, struct snd_pc * TEA5757 radio */ -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_TEA575X_BOOL /* GPIO to TEA575x maps */ struct snd_fm801_tea575x_gpio { @@ -1150,7 +1149,7 @@ static int snd_fm801_free(struct fm801 *chip) outw(cmdw, FM801_REG(chip, IRQ_MASK)); __end_hw: -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_TEA575X_BOOL snd_tea575x_exit(&chip->tea); #endif if (chip->irq >= 0) @@ -1229,7 +1228,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); -#ifdef TEA575X_RADIO +#ifdef CONFIG_SND_FM801_TEA575X_BOOL chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); -- cgit v1.2.3 From 22ce5f74a954d02e56352ecfa45f8d7c817693e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 15 May 2011 12:19:29 +0200 Subject: ALSA: hda - Search ADC NIDs dynamically in Conexant auto-parser Instead of giving fixed arrays, look for ADC nids dynamically in the tree in Conexant auto-parser code. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 40 ++++++++++++++++++++++++++-------------- 1 file changed, 26 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fc02751f22d..ebce2fe3aec 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -110,6 +110,7 @@ struct conexant_spec { struct hda_input_mux private_imux; hda_nid_t imux_adcs[HDA_MAX_NUM_INPUTS]; hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; + hda_nid_t private_adc_nids[HDA_MAX_NUM_INPUTS]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; struct pin_dac_pair dac_info[8]; int dac_info_filled; @@ -4034,6 +4035,28 @@ static int cx_auto_build_controls(struct hda_codec *codec) return conexant_build_controls(codec); } +static int cx_auto_search_adcs(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + hda_nid_t nid, end_nid; + + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int caps = get_wcaps(codec, nid); + if (get_wcaps_type(caps) != AC_WID_AUD_IN) + continue; + if (caps & AC_WCAP_DIGITAL) + continue; + if (snd_BUG_ON(spec->num_adc_nids >= + ARRAY_SIZE(spec->private_adc_nids))) + break; + spec->private_adc_nids[spec->num_adc_nids++] = nid; + } + spec->adc_nids = spec->private_adc_nids; + return 0; +} + + static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = conexant_build_pcms, @@ -4058,29 +4081,18 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + err = cx_auto_search_adcs(codec); + if (err < 0) + return err; switch (codec->vendor_id) { case 0x14f15051: codec->pin_amp_workaround = 1; - spec->adc_nids = cxt5051_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(cxt5051_adc_nids); - spec->capsrc_nids = spec->adc_nids; break; case 0x14f15045: codec->pin_amp_workaround = 1; - spec->adc_nids = cxt5045_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(cxt5045_adc_nids); - spec->capsrc_nids = spec->adc_nids; break; case 0x14f15047: codec->pin_amp_workaround = 1; - spec->adc_nids = cxt5047_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(cxt5047_adc_nids); - spec->capsrc_nids = cxt5047_capsrc_nids; - break; - default: - spec->adc_nids = cx_auto_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(cx_auto_adc_nids); - spec->capsrc_nids = spec->adc_nids; break; } err = cx_auto_parse_auto_config(codec); -- cgit v1.2.3 From 1387cde51d0946eb3d8091b63f025c40cc1acdf4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 15 May 2011 12:21:20 +0200 Subject: ALSA: hda - Enable codec->pin_amp_workaround always for Conexant auto-parser It can (must for some) be used for all Conexnat codecs safely. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ebce2fe3aec..fb759bff91a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4081,20 +4081,10 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; err = cx_auto_search_adcs(codec); if (err < 0) return err; - switch (codec->vendor_id) { - case 0x14f15051: - codec->pin_amp_workaround = 1; - break; - case 0x14f15045: - codec->pin_amp_workaround = 1; - break; - case 0x14f15047: - codec->pin_amp_workaround = 1; - break; - } err = cx_auto_parse_auto_config(codec); if (err < 0) { kfree(codec->spec); -- cgit v1.2.3 From 9b842cd86843db0f6ba4badb30282044afaed6c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 15 May 2011 12:35:04 +0200 Subject: ALSA: hda - Don't use auto-parser for cxt5045 / 5051 as default Just for safety reason (for avoiding any possible regressions), don't enable auto-parser as default for cxt5045 and 5051, as well as 5047. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fb759bff91a..6c9abee9ce8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1104,8 +1104,10 @@ static int patch_cxt5045(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, CXT5045_MODELS, cxt5045_models, cxt5045_cfg_tbl); +#if 0 /* use the old method just for safety */ if (board_config < 0) board_config = CXT5045_AUTO; +#endif if (board_config == CXT5045_AUTO) return patch_conexant_auto(codec); @@ -1972,8 +1974,10 @@ static int patch_cxt5051(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, cxt5051_models, cxt5051_cfg_tbl); +#if 0 /* use the old method just for safety */ if (board_config < 0) board_config = CXT5051_AUTO; +#endif if (board_config == CXT5051_AUTO) return patch_conexant_auto(codec); -- cgit v1.2.3 From f7391fce6a1553ed3375ee4d73f2deaa3fd69732 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 May 2011 19:25:42 +0200 Subject: ASoC: Reintroduce do_spi_write() There is an unfortunate difference in return values between spi_write() and i2c_master_send() so we need an adaptor function to translate. Reported-by: Lars-Peter Clausen Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 01e5b984643..06b7b81a160 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,6 +20,20 @@ #include +#ifdef CONFIG_SPI_MASTER +static int do_spi_write(void *control, const char *data, int len) +{ + struct spi_device *spi = control; + int ret; + + ret = spi_write(spi, data, len); + if (ret < 0) + return ret; + + return len; +} +#endif + static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value, const void *data, int len) { @@ -412,7 +426,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_SPI: #ifdef CONFIG_SPI_MASTER - codec->hw_write = (hw_write_t)spi_write; + codec->hw_write = do_spi_write; #endif codec->control_data = container_of(codec->dev, -- cgit v1.2.3 From 770939c37feb4b9c88f4de8482ff922af64d20ee Mon Sep 17 00:00:00 2001 From: Jin Park Date: Thu, 12 May 2011 14:58:36 +0900 Subject: ASoC: codecs: max98088: Fixed invalid register definitions in mixer controls Fixed invalid register definitions in mixer controls such as left speaker mixer, left hp mixer and left rec mixer. Signed-off-by: Jin Park Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index bb58bdb4985..87cd5f68807 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -808,10 +808,10 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { /* Left speaker mixer switch */ static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1, 0), @@ -836,10 +836,10 @@ static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = { /* Left headphone mixer switch */ static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1, 0), @@ -864,10 +864,10 @@ static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = { /* Left earpiece/receiver mixer switch */ static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = { - SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), - SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), - SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0), SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1, 0), SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1, 0), SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1, 0), -- cgit v1.2.3 From 938b4fbc916667cbe0410c325e7163c99a08c05f Mon Sep 17 00:00:00 2001 From: Jin Park Date: Thu, 12 May 2011 14:58:37 +0900 Subject: ASoC: codecs: max98088: Moved the EX Limiter Mode from dapm widget to control Moved the EX Limiter Mode from dapm widget to control, because it was not required DAPM route. Signed-off-by: Jin Park Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 87cd5f68807..16eb90666c4 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum = ARRAY_SIZE(max98088_exmode_texts), max98088_exmode_texts, max98088_exmode_values); -static const struct snd_kcontrol_new max98088_exmode_controls = - SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum); static const char *max98088_ex_thresh[] = { /* volts PP */ "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; @@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0), SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0), + SOC_ENUM("EX Limiter Mode", max98088_exmode_enum), SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum), SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum), @@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = { SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0), - SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0, - &max98088_exmode_controls), - SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), SND_SOC_DAPM_OUTPUT("SPKL"), -- cgit v1.2.3 From 25709f6d83cc23d6f2912194c77ebf850310223e Mon Sep 17 00:00:00 2001 From: Jin Park Date: Thu, 12 May 2011 14:58:38 +0900 Subject: ASoC: codecs: max98088: Added digital mute function in DAI1 and DAI2 Added digital mute function in DAI1 and DAI2. Signed-off-by: Jin Park Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/max98088.h | 13 +++++++++++++ 2 files changed, 45 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 16eb90666c4..4173b67c94d 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1566,6 +1566,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int reg; + + if (mute) + reg = M98088_DAI_MUTE; + else + reg = 0; + + snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + +static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int reg; + + if (mute) + reg = M98088_DAI_MUTE; + else + reg = 0; + + snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + static void max98088_sync_cache(struct snd_soc_codec *codec) { u16 *reg_cache = codec->reg_cache; @@ -1627,12 +1657,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops = { .set_sysclk = max98088_dai_set_sysclk, .set_fmt = max98088_dai1_set_fmt, .hw_params = max98088_dai1_hw_params, + .digital_mute = max98088_dai1_digital_mute, }; static struct snd_soc_dai_ops max98088_dai2_ops = { .set_sysclk = max98088_dai_set_sysclk, .set_fmt = max98088_dai2_set_fmt, .hw_params = max98088_dai2_hw_params, + .digital_mute = max98088_dai2_digital_mute, }; static struct snd_soc_dai_driver max98088_dai[] = { diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index 56554c797fe..be89a4f4aab 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -133,6 +133,19 @@ #define M98088_REC_LINEMODE (1<<7) #define M98088_REC_LINEMODE_MASK (1<<7) +/* M98088_REG_2D_MIX_SPK_CNTL */ + #define M98088_MIX_SPKR_GAIN_MASK (3<<2) + #define M98088_MIX_SPKR_GAIN_SHIFT 2 + #define M98088_MIX_SPKL_GAIN_MASK (3<<0) + #define M98088_MIX_SPKL_GAIN_SHIFT 0 + +/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */ + #define M98088_DAI_MUTE (1<<7) + #define M98088_DAI_MUTE_MASK (1<<7) + #define M98088_DAI_VOICE_GAIN_MASK (3<<4) + #define M98088_DAI_ATTENUATION_MASK (0xF<<0) + #define M98088_DAI_ATTENUATION_SHIFT 0 + /* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */ #define M98088_MICPRE_MASK (3<<5) #define M98088_MICPRE_SHIFT 5 -- cgit v1.2.3 From cf27f29ae2a4a0feed75262361f9f7dff20230b1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 May 2011 11:33:02 +0200 Subject: ALSA: hda - Build boost controls from selector widget in Cxt auto-parser When the intermediate selector widget in the capture path provides the boost volume, create the corresponding volume control. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 79 ++++++++++++++++++++++++++++++++---------- 1 file changed, 61 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6c9abee9ce8..28664e1c1ae 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -108,6 +108,7 @@ struct conexant_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; + int imux_cfg_idx[HDA_MAX_NUM_INPUTS]; /* corresponding autocfg.input */ hda_nid_t imux_adcs[HDA_MAX_NUM_INPUTS]; hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; hda_nid_t private_adc_nids[HDA_MAX_NUM_INPUTS]; @@ -3464,30 +3465,42 @@ static int cx_auto_mux_enum_get(struct snd_kcontrol *kcontrol, * if do_select is set, actually select the route. */ static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux, - hda_nid_t pin, bool do_select, int depth) + hda_nid_t pin, hda_nid_t *srcp, + bool do_select, int depth) { hda_nid_t conn[HDA_MAX_NUM_INPUTS]; int i, nums; + switch (get_wcaps_type(get_wcaps(codec, mux))) { + case AC_WID_AUD_IN: + case AC_WID_AUD_SEL: + case AC_WID_AUD_MIX: + break; + default: + return -1; + } + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); for (i = 0; i < nums; i++) if (conn[i] == pin) { if (do_select) snd_hda_codec_write(codec, mux, 0, AC_VERB_SET_CONNECT_SEL, i); + if (srcp) + *srcp = mux; return i; } depth++; if (depth == 2) return -1; for (i = 0; i < nums; i++) { - int ret = __select_input_connection(codec, conn[i], pin, + int ret = __select_input_connection(codec, conn[i], pin, srcp, do_select, depth); if (ret >= 0) { if (do_select) snd_hda_codec_write(codec, mux, 0, AC_VERB_SET_CONNECT_SEL, i); - return ret; + return i; } } return -1; @@ -3496,13 +3509,13 @@ static int __select_input_connection(struct hda_codec *codec, hda_nid_t mux, static void select_input_connection(struct hda_codec *codec, hda_nid_t mux, hda_nid_t pin) { - __select_input_connection(codec, mux, pin, true, 0); + __select_input_connection(codec, mux, pin, NULL, true, 0); } static int get_input_connection(struct hda_codec *codec, hda_nid_t mux, hda_nid_t pin) { - return __select_input_connection(codec, mux, pin, false, 0); + return __select_input_connection(codec, mux, pin, NULL, false, 0); } static int cx_auto_mux_enum_update(struct hda_codec *codec, @@ -3637,6 +3650,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) if (idx >= 0) { const char *label; label = hda_get_autocfg_input_label(codec, cfg, i); + spec->imux_cfg_idx[imux->num_items] = i; spec->imux_adcs[imux->num_items] = adc; spec->imux_pins[imux->num_items] = cfg->inputs[i].pin; @@ -3978,35 +3992,64 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, return 0; } +static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, + const char *label, int cidx) +{ + struct conexant_spec *spec = codec->spec; + hda_nid_t mux, nid; + int con; + + nid = spec->imux_pins[idx]; + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + return cx_auto_add_volume(codec, label, " Boost", cidx, + nid, HDA_INPUT); + con = __select_input_connection(codec, spec->imux_adcs[idx], nid, &mux, + false, 0); + if (con < 0) + return 0; + if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) + return cx_auto_add_volume(codec, label, " Boost", 0, + mux, HDA_OUTPUT); + return 0; +} + static int cx_auto_build_input_controls(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - static const char *prev_label; + struct hda_input_mux *imux = &spec->private_imux; + const char *prev_label; + int input_conn[HDA_MAX_NUM_INPUTS]; int i, err, cidx; + int multi_connection; + + multi_connection = 0; + for (i = 0; i < imux->num_items; i++) { + cidx = get_input_connection(codec, spec->imux_adcs[i], + spec->imux_pins[i]); + input_conn[i] = (spec->imux_adcs[i] << 8) | cidx; + if (i > 0 && input_conn[i] != input_conn[0]) + multi_connection = 1; + } prev_label = NULL; cidx = 0; - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; + for (i = 0; i < imux->num_items; i++) { + hda_nid_t nid = spec->imux_pins[i]; const char *label; - int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP; - label = hda_get_autocfg_input_label(codec, cfg, i); + label = hda_get_autocfg_input_label(codec, &spec->autocfg, + spec->imux_cfg_idx[i]); if (label == prev_label) cidx++; else cidx = 0; prev_label = label; - if (pin_amp) { - err = cx_auto_add_volume(codec, label, " Boost", cidx, - nid, HDA_INPUT); - if (err < 0) - return err; - } + err = cx_auto_add_boost_volume(codec, i, label, cidx); + if (err < 0) + return err; - if (cfg->num_inputs == 1) { + if (!multi_connection) { err = cx_auto_add_capture_volume(codec, nid, "Capture", "", cidx); } else { -- cgit v1.2.3 From f9759301c68a274302e434daa135926c25ca51ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 May 2011 11:45:15 +0200 Subject: ALSA: hda - Don't create multiple same volume/boost controls in Cxt auto-parser Check the routing more exactly for avoiding the duplicated controls for the very same effect for multiple capture routes in Conexant auto-parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 28664e1c1ae..ed983a0b0dc 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -109,6 +109,7 @@ struct conexant_spec { struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; int imux_cfg_idx[HDA_MAX_NUM_INPUTS]; /* corresponding autocfg.input */ + hda_nid_t imux_boost_nid[HDA_MAX_NUM_INPUTS]; /* boost widget */ hda_nid_t imux_adcs[HDA_MAX_NUM_INPUTS]; hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; hda_nid_t private_adc_nids[HDA_MAX_NUM_INPUTS]; @@ -3651,6 +3652,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) const char *label; label = hda_get_autocfg_input_label(codec, cfg, i); spec->imux_cfg_idx[imux->num_items] = i; + spec->imux_boost_nid[imux->num_items] = 0; spec->imux_adcs[imux->num_items] = adc; spec->imux_pins[imux->num_items] = cfg->inputs[i].pin; @@ -3997,7 +3999,7 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, { struct conexant_spec *spec = codec->spec; hda_nid_t mux, nid; - int con; + int i, con; nid = spec->imux_pins[idx]; if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) @@ -4007,9 +4009,16 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, false, 0); if (con < 0) return 0; - if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) + for (i = 0; i < idx; i++) { + if (spec->imux_boost_nid[i] == mux) + return 0; /* already present */ + } + + if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) { + spec->imux_boost_nid[idx] = mux; return cx_auto_add_volume(codec, label, " Boost", 0, mux, HDA_OUTPUT); + } return 0; } @@ -4050,6 +4059,8 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) return err; if (!multi_connection) { + if (i > 0) + continue; err = cx_auto_add_capture_volume(codec, nid, "Capture", "", cidx); } else { -- cgit v1.2.3 From fea4a4f9737883ed660bd99b9b5cff9120455094 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 May 2011 11:49:12 +0200 Subject: ALSA: hda - Add support of auto-parser to cxt5066 codecs Still experimental. Not enabled as default unless model=auto is passed. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ed983a0b0dc..2b12d72bf91 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3040,6 +3040,7 @@ enum { CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ + CXT5066_AUTO, /* BIOS auto-parser */ CXT5066_MODELS }; @@ -3052,6 +3053,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_THINKPAD] = "thinkpad", [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", + [CXT5066_AUTO] = "auto", }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -3089,6 +3091,15 @@ static int patch_cxt5066(struct hda_codec *codec) struct conexant_spec *spec; int board_config; + board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, + cxt5066_models, cxt5066_cfg_tbl); +#if 0 /* use the old method just for safety */ + if (board_config < 0) + board_config = CXT5066_AUTO; +#endif + if (board_config == CXT5066_AUTO) + return patch_conexant_auto(codec); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; @@ -3119,8 +3130,6 @@ static int patch_cxt5066(struct hda_codec *codec) set_beep_amp(spec, 0x13, 0, HDA_OUTPUT); - board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, - cxt5066_models, cxt5066_cfg_tbl); switch (board_config) { default: case CXT5066_LAPTOP: -- cgit v1.2.3 From e033ebfb399227e01686260ac271029011bc6b47 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 16 May 2011 12:09:29 +0200 Subject: ALSA: HDA: Use one dmic only for Dell Studio 1558 There are no signs of a dmic at node 0x0b, so the user is left with an additional internal mic which does not exist. This commit removes that non-existing mic. Cc: stable@kernel.org (2.6.32+) BugLink: http://bugs.launchpad.net/bugs/731706 Reported-by: James Page Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a2884bb8803..38f43175168 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1604,7 +1604,7 @@ static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe, "Dell Studio XPS 1645", STAC_DELL_M6_BOTH), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413, - "Dell Studio 1558", STAC_DELL_M6_BOTH), + "Dell Studio 1558", STAC_DELL_M6_DMIC), {} /* terminator */ }; -- cgit v1.2.3 From d0b48af6c2b887354d0893e598d92911ce52620e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 14 May 2011 17:21:28 -0700 Subject: ASoC: Ensure output PGA is enabled for line outputs in wm_hubs Also fix a left/right typo while we're at it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Date: Sun, 15 May 2011 12:18:38 -0700 Subject: ASoC: Add some missing volume update bit sets for wm_hubs devices Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm_hubs.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c7c2fec482a..e55b298c14a 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -836,17 +836,21 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, WM8993_IN2_VU, WM8993_IN2_VU); + snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT, + WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT, WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME, - WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC); + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC, + WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC); snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC, WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC); snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME, - WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC); + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU, + WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU); snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU); -- cgit v1.2.3 From d7fdae7c6533b9a409158c736781cdd352b76793 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 15 May 2011 18:02:53 -0700 Subject: ASoC: Skip noop reconfiguration of WM8958 DSP2 algorithms If we're setting the currently applied value for one of the DSP algorithm configurations we can just skip all the handling as the control set is a noop. This ensures we do not disrupt a running DSP. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8958-dsp2.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 5d4bc7a21df..ca26779bb6b 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -508,6 +508,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->mbc_ena[mbc] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -628,6 +631,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->vss_ena[vss] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -689,6 +695,16 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (hpf < 3) { + if (wm8994->hpf1_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } else { + if (wm8994->hpf2_ena[hpf % 3] == + ucontrol->value.integer.value[0]) + return 0; + } + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; @@ -782,6 +798,9 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (wm8994->enh_eq_ena[eq] == ucontrol->value.integer.value[0]) + return 0; + if (ucontrol->value.integer.value[0] > 1) return -EINVAL; -- cgit v1.2.3 From fa63e477ddb09075f68cd5fe4db8f8045627a787 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 15 May 2011 18:18:56 -0700 Subject: ASoC: Don't restart an already running WM8958 DSP2 Don't want to upset the DSP. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8958-dsp2.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index ca26779bb6b..0293763debe 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -362,6 +362,10 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start) path, wm8994->dsp_active, start, pwr_reg, reg); if (start && ena) { + /* If the DSP is already running then noop */ + if (reg & WM8958_DSP2_ENA) + return; + /* If either AIFnCLK is not yet enabled postpone */ if (!(snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA_MASK) && -- cgit v1.2.3 From 9d03545d886bedd2c81b8f28ae0cc68c356d02f7 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 13 May 2011 19:16:52 +0300 Subject: ASoC: Fix wrong data type access in a few codec drivers Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol") changed the control private data type that is passed to snd_soc_cnew when creating dapm mixer and mux controls. Commit did not update a few codec drivers that are using their own put callbacks and thus are accessing a wrong data type. Signed-off-by: Jarkko Nikula Tested-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 3 ++- sound/soc/codecs/wm8903.c | 3 ++- sound/soc/codecs/wm8993.c | 3 ++- sound/soc/codecs/wm8994.c | 6 ++++-- sound/soc/codecs/wm8995.c | 4 ++-- 5 files changed, 12 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6c43c13f043..c3d96fc8c26 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 957cd66177d..43e3d760766 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 056aef90434..9e5ff789b80 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, static int class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; struct snd_soc_codec *codec = widget->codec; struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e6dfa103ad2..970a95c5360 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -877,7 +877,8 @@ static const char *hp_mux_text[] = { static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; @@ -1004,7 +1005,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec = w->codec; int ret; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 67eaaecbb42..5ad873fda81 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -305,11 +305,11 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source, static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget *w; + struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_widget *w = wlist->widgets[0]; struct snd_soc_codec *codec; int ret; - w = snd_kcontrol_chip(kcontrol); codec = w->codec; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); wm8995_update_class_w(codec); -- cgit v1.2.3 From 34e268d87d60da7968d7300aae7d575be9d16ae7 Mon Sep 17 00:00:00 2001 From: Stephen Boyd Date: Thu, 12 May 2011 16:50:10 -0700 Subject: ASoC: Silence DEBUG_STRICT_USER_COPY_CHECKS=y warning Enabling DEBUG_STRICT_USER_COPY_CHECKS causes the following warning: In file included from arch/x86/include/asm/uaccess.h:573, from include/linux/poll.h:14, from include/sound/pcm.h:29, from include/sound/ac97_codec.h:31, from sound/soc/soc-core.c:34: In function 'copy_from_user', inlined from 'codec_reg_write_file' at sound/soc/soc-core.c:252: arch/x86/include/asm/uaccess_64.h:65: warning: call to 'copy_from_user_overflow' declared with attribute warning: copy_from_user() buffer size is not provably correct presumably due to buf_size being signed causing GCC to fail to see that buf_size can't become negative. Acked-by: Liam Girdwood Signed-off-by: Stephen Boyd Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c1a4cf480c7..5968745213f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -242,7 +242,7 @@ static ssize_t codec_reg_write_file(struct file *file, const char __user *user_buf, size_t count, loff_t *ppos) { char buf[32]; - int buf_size; + size_t buf_size; char *start = buf; unsigned long reg, value; int step = 1; -- cgit v1.2.3 From 47ad1f4e40c4374243a56c8df168f4fb551c7915 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 09:15:55 +0200 Subject: ALSA: hda - Code refactoring in patch_conexant.c Use a struct instead of each array for managing input-source info for auto-parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 64 ++++++++++++++++++++++-------------------- 1 file changed, 34 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2b12d72bf91..390bc02dcf5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -55,6 +55,13 @@ struct pin_dac_pair { int type; }; +struct imux_info { + hda_nid_t pin; /* input pin NID */ + hda_nid_t adc; /* connected ADC NID */ + hda_nid_t boost; /* optional boost volume NID */ + int index; /* corresponding to autocfg.input */ +}; + struct conexant_spec { const struct snd_kcontrol_new *mixers[5]; @@ -108,10 +115,7 @@ struct conexant_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; - int imux_cfg_idx[HDA_MAX_NUM_INPUTS]; /* corresponding autocfg.input */ - hda_nid_t imux_boost_nid[HDA_MAX_NUM_INPUTS]; /* boost widget */ - hda_nid_t imux_adcs[HDA_MAX_NUM_INPUTS]; - hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS]; + struct imux_info imux_info[HDA_MAX_NUM_INPUTS]; hda_nid_t private_adc_nids[HDA_MAX_NUM_INPUTS]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; struct pin_dac_pair dac_info[8]; @@ -3254,7 +3258,7 @@ static int cx_auto_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; - hda_nid_t adc = spec->imux_adcs[spec->cur_mux[0]]; + hda_nid_t adc = spec->imux_info[spec->cur_mux[0]].adc; if (spec->adc_switching) { spec->cur_adc = adc; spec->cur_adc_stream_tag = stream_tag; @@ -3541,9 +3545,9 @@ static int cx_auto_mux_enum_update(struct hda_codec *codec, idx = imux->num_items - 1; if (spec->cur_mux[0] == idx) return 0; - adc = spec->imux_adcs[idx]; - select_input_connection(codec, spec->imux_adcs[idx], - spec->imux_pins[idx]); + adc = spec->imux_info[idx].adc; + select_input_connection(codec, spec->imux_info[idx].adc, + spec->imux_info[idx].pin); if (spec->cur_adc && spec->cur_adc != adc) { /* stream is running, let's swap the current ADC */ __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); @@ -3585,7 +3589,7 @@ static void cx_auto_automic(struct hda_codec *codec) if (!spec->auto_mic) return; - if (snd_hda_jack_detect(codec, spec->imux_pins[ext_idx])) + if (snd_hda_jack_detect(codec, spec->imux_info[ext_idx].pin)) cx_auto_mux_enum_update(codec, &spec->private_imux, ext_idx); else cx_auto_mux_enum_update(codec, &spec->private_imux, !ext_idx); @@ -3630,14 +3634,14 @@ static void cx_auto_check_auto_mic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - if (is_ext_mic(codec, spec->imux_pins[0]) && - is_int_mic(codec, spec->imux_pins[1])) { + if (is_ext_mic(codec, spec->imux_info[0].pin) && + is_int_mic(codec, spec->imux_info[1].pin)) { spec->auto_mic = 1; spec->auto_mic_ext = 0; return; } - if (is_int_mic(codec, spec->imux_pins[0]) && - is_ext_mic(codec, spec->imux_pins[1])) { + if (is_int_mic(codec, spec->imux_info[0].pin) && + is_ext_mic(codec, spec->imux_info[1].pin)) { spec->auto_mic = 1; spec->auto_mic_ext = 1; return; @@ -3660,10 +3664,10 @@ static void cx_auto_parse_input(struct hda_codec *codec) if (idx >= 0) { const char *label; label = hda_get_autocfg_input_label(codec, cfg, i); - spec->imux_cfg_idx[imux->num_items] = i; - spec->imux_boost_nid[imux->num_items] = 0; - spec->imux_adcs[imux->num_items] = adc; - spec->imux_pins[imux->num_items] = + spec->imux_info[imux->num_items].index = i; + spec->imux_info[imux->num_items].boost = 0; + spec->imux_info[imux->num_items].adc = adc; + spec->imux_info[imux->num_items].pin = cfg->inputs[i].pin; snd_hda_add_imux_item(imux, label, idx, NULL); break; @@ -3674,7 +3678,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) cx_auto_check_auto_mic(codec); if (imux->num_items > 1 && !spec->auto_mic) { for (i = 1; i < imux->num_items; i++) { - if (spec->imux_adcs[i] != spec->imux_adcs[0]) { + if (spec->imux_info[i].adc != spec->imux_info[0].adc) { spec->adc_switching = 1; break; } @@ -3864,8 +3868,8 @@ static void cx_auto_init_input(struct hda_codec *codec) AC_USRSP_EN | CONEXANT_MIC_EVENT); cx_auto_automic(codec); } else { - select_input_connection(codec, spec->imux_adcs[0], - spec->imux_pins[0]); + select_input_connection(codec, spec->imux_info[0].adc, + spec->imux_info[0].pin); } } @@ -4010,21 +4014,21 @@ static int cx_auto_add_boost_volume(struct hda_codec *codec, int idx, hda_nid_t mux, nid; int i, con; - nid = spec->imux_pins[idx]; + nid = spec->imux_info[idx].pin; if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) return cx_auto_add_volume(codec, label, " Boost", cidx, nid, HDA_INPUT); - con = __select_input_connection(codec, spec->imux_adcs[idx], nid, &mux, - false, 0); + con = __select_input_connection(codec, spec->imux_info[idx].adc, nid, + &mux, false, 0); if (con < 0) return 0; for (i = 0; i < idx; i++) { - if (spec->imux_boost_nid[i] == mux) + if (spec->imux_info[i].boost == mux) return 0; /* already present */ } if (get_wcaps(codec, mux) & AC_WCAP_OUT_AMP) { - spec->imux_boost_nid[idx] = mux; + spec->imux_info[idx].boost = mux; return cx_auto_add_volume(codec, label, " Boost", 0, mux, HDA_OUTPUT); } @@ -4042,9 +4046,9 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) multi_connection = 0; for (i = 0; i < imux->num_items; i++) { - cidx = get_input_connection(codec, spec->imux_adcs[i], - spec->imux_pins[i]); - input_conn[i] = (spec->imux_adcs[i] << 8) | cidx; + cidx = get_input_connection(codec, spec->imux_info[i].adc, + spec->imux_info[i].pin); + input_conn[i] = (spec->imux_info[i].adc << 8) | cidx; if (i > 0 && input_conn[i] != input_conn[0]) multi_connection = 1; } @@ -4052,11 +4056,11 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) prev_label = NULL; cidx = 0; for (i = 0; i < imux->num_items; i++) { - hda_nid_t nid = spec->imux_pins[i]; + hda_nid_t nid = spec->imux_info[i].pin; const char *label; label = hda_get_autocfg_input_label(codec, &spec->autocfg, - spec->imux_cfg_idx[i]); + spec->imux_info[i].index); if (label == prev_label) cidx++; else -- cgit v1.2.3 From a3a85d3983f7e18c46fba9f92c21d8a713de9791 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 09:17:52 +0200 Subject: ALSA: hda - Add missing Front/Surround/CLFE as slaves for Cxt auto-parser Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 390bc02dcf5..703dda69ad7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -475,12 +475,18 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = { static const char * const slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", + "Front Playback Volume", + "Surround Playback Volume", + "CLFE Playback Volume", NULL }; static const char * const slave_sws[] = { "Headphone Playback Switch", "Speaker Playback Switch", + "Front Playback Switch", + "Surround Playback Switch", + "CLFE Playback Switch", NULL }; -- cgit v1.2.3 From 03697e2acce9b8818cdb5fc0ebd5e5199dea1c32 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 09:53:31 +0200 Subject: ALSA: hda - Add automute-mode enum to Conexant auto-parser Implement the same functionality as Realtek's auto-mute mode control. Now Conexant auto-parser can also mutes line-out and provide the enum control for different automute behavior. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 5 + sound/pci/hda/patch_conexant.c | 259 +++++++++++++++++++++++++++++++++-------- 2 files changed, 215 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1ed6ee5a1e7..01a7cf6b5fb 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -493,6 +493,11 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); +static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) +{ + return !!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT); +} + /* flags for hda_nid_item */ #define HDA_NID_ITEM_AMP (1<<0) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 703dda69ad7..ac595363e0e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -39,6 +39,7 @@ #define CONEXANT_HP_EVENT 0x37 #define CONEXANT_MIC_EVENT 0x38 +#define CONEXANT_LINE_EVENT 0x39 /* Conexant 5051 specific */ @@ -81,6 +82,7 @@ struct conexant_spec { */ unsigned int cur_eapd; unsigned int hp_present; + unsigned int line_present; unsigned int auto_mic; int auto_mic_ext; /* imux_pins[] index for ext mic */ unsigned int need_dac_fix; @@ -123,6 +125,9 @@ struct conexant_spec { unsigned int port_d_mode; unsigned int auto_mute:1; /* used in auto-parser */ + unsigned int detect_line:1; /* Line-out detection enabled */ + unsigned int automute_lines:1; /* automute line-out as well */ + unsigned int automute_hp_lo:1; /* both HP and LO available */ unsigned int dell_automute:1; unsigned int dell_vostro:1; unsigned int ideapad:1; @@ -3420,48 +3425,193 @@ static void cx_auto_parse_output(struct hda_codec *codec) spec->multiout.dac_nids = spec->private_dac_nids; spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (cfg->hp_outs > 0) - spec->auto_mute = 1; + for (i = 0; i < cfg->hp_outs; i++) { + if (is_jack_detectable(codec, cfg->hp_pins[i])) { + spec->auto_mute = 1; + break; + } + } + if (spec->auto_mute && cfg->line_out_pins[0] && + cfg->line_out_pins[0] != cfg->hp_pins[0] && + cfg->line_out_pins[0] != cfg->speaker_pins[0]) { + for (i = 0; i < cfg->line_outs; i++) { + if (is_jack_detectable(codec, cfg->line_out_pins[i])) { + spec->detect_line = 1; + break; + } + } + spec->automute_lines = spec->detect_line; + } + spec->vmaster_nid = spec->private_dac_nids[0]; } static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins, hda_nid_t *pins, bool on); +static void do_automute(struct hda_codec *codec, int num_pins, + hda_nid_t *pins, bool on) +{ + int i; + for (i = 0; i < num_pins; i++) + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + on ? PIN_OUT : 0); + cx_auto_turn_eapd(codec, num_pins, pins, on); +} + +static int detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) +{ + int i, present = 0; + + for (i = 0; i < num_pins; i++) { + hda_nid_t nid = pins[i]; + if (!nid || !is_jack_detectable(codec, nid)) + break; + snd_hda_input_jack_report(codec, nid); + present |= snd_hda_jack_detect(codec, nid); + } + return present; +} + /* auto-mute/unmute speaker and line outs according to headphone jack */ +static void cx_auto_update_speakers(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int on; + + if (!spec->auto_mute) + on = 0; + else + on = spec->hp_present | spec->line_present; + cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, on); + do_automute(codec, cfg->speaker_outs, cfg->speaker_pins, !on); + + /* toggle line-out mutes if needed, too */ + /* if LO is a copy of either HP or Speaker, don't need to handle it */ + if (cfg->line_out_pins[0] == cfg->hp_pins[0] || + cfg->line_out_pins[0] == cfg->speaker_pins[0]) + return; + if (!spec->automute_lines || !spec->auto_mute) + on = 0; + else + on = spec->hp_present; + do_automute(codec, cfg->line_outs, cfg->line_out_pins, !on); +} + static void cx_auto_hp_automute(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - int i, present; if (!spec->auto_mute) return; - present = 0; - for (i = 0; i < cfg->hp_outs; i++) { - if (snd_hda_jack_detect(codec, cfg->hp_pins[i])) { - present = 1; - break; - } - } - cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, present); - for (i = 0; i < cfg->line_outs; i++) { - snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - present ? 0 : PIN_OUT); + spec->hp_present = detect_jacks(codec, cfg->hp_outs, cfg->hp_pins); + cx_auto_update_speakers(codec); +} + +static void cx_auto_line_automute(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (!spec->auto_mute || !spec->detect_line) + return; + spec->line_present = detect_jacks(codec, cfg->line_outs, + cfg->line_out_pins); + cx_auto_update_speakers(codec); +} + +static int cx_automute_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + static const char * const texts2[] = { + "Disabled", "Enabled" + }; + static const char * const texts3[] = { + "Disabled", "Speaker Only", "Line-Out+Speaker" + }; + const char * const *texts; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + if (spec->automute_hp_lo) { + uinfo->value.enumerated.items = 3; + texts = texts3; + } else { + uinfo->value.enumerated.items = 2; + texts = texts2; } - cx_auto_turn_eapd(codec, cfg->line_outs, cfg->line_out_pins, !present); - for (i = 0; !present && i < cfg->line_outs; i++) - if (snd_hda_jack_detect(codec, cfg->line_out_pins[i])) - present = 1; - for (i = 0; i < cfg->speaker_outs; i++) { - snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - present ? 0 : PIN_OUT); + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int cx_automute_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + unsigned int val; + if (!spec->auto_mute) + val = 0; + else if (!spec->automute_lines) + val = 1; + else + val = 2; + ucontrol->value.enumerated.item[0] = val; + return 0; +} + +static int cx_automute_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + + switch (ucontrol->value.enumerated.item[0]) { + case 0: + if (!spec->auto_mute) + return 0; + spec->auto_mute = 0; + break; + case 1: + if (spec->auto_mute && !spec->automute_lines) + return 0; + spec->auto_mute = 1; + spec->automute_lines = 0; + break; + case 2: + if (!spec->automute_hp_lo) + return -EINVAL; + if (spec->auto_mute && spec->automute_lines) + return 0; + spec->auto_mute = 1; + spec->automute_lines = 1; + break; + default: + return -EINVAL; } - cx_auto_turn_eapd(codec, cfg->speaker_outs, cfg->speaker_pins, !present); + cx_auto_update_speakers(codec); + return 1; } +static const struct snd_kcontrol_new cx_automute_mode_enum[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Auto-Mute Mode", + .info = cx_automute_mode_info, + .get = cx_automute_mode_get, + .put = cx_automute_mode_put, + }, + { } +}; + static int cx_auto_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -3607,7 +3757,9 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) switch (res >> 26) { case CONEXANT_HP_EVENT: cx_auto_hp_automute(codec); - snd_hda_input_jack_report(codec, nid); + break; + case CONEXANT_LINE_EVENT: + cx_auto_line_automute(codec); break; case CONEXANT_MIC_EVENT: cx_auto_automic(codec); @@ -3630,7 +3782,7 @@ static int is_ext_mic(struct hda_codec *codec, hda_nid_t pin) unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && snd_hda_get_input_pin_attr(def_conf) >= INPUT_PIN_ATTR_NORMAL && - (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_PRES_DETECT); + is_jack_detectable(codec, pin); } /* check whether the pin config is suitable for auto-mic switching; @@ -3794,6 +3946,16 @@ static void mute_outputs(struct hda_codec *codec, int num_nids, } } +static void enable_unsol_pins(struct hda_codec *codec, int num_pins, + hda_nid_t *pins, unsigned int tag) +{ + int i; + for (i = 0; i < num_pins; i++) + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | tag); +} + static void cx_auto_init_output(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -3808,35 +3970,27 @@ static void cx_auto_init_output(struct hda_codec *codec) mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins); - if (spec->auto_mute) { - for (i = 0; i < cfg->hp_outs; i++) { - snd_hda_codec_write(codec, cfg->hp_pins[i], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | CONEXANT_HP_EVENT); - } - cx_auto_hp_automute(codec); - } else { - for (i = 0; i < cfg->line_outs; i++) - snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - for (i = 0; i < cfg->speaker_outs; i++) - snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - /* turn on EAPD */ - cx_auto_turn_eapd(codec, cfg->line_outs, cfg->line_out_pins, - true); - cx_auto_turn_eapd(codec, cfg->hp_outs, cfg->hp_pins, - true); - cx_auto_turn_eapd(codec, cfg->speaker_outs, cfg->speaker_pins, - true); - } - for (i = 0; i < spec->dac_info_filled; i++) { nid = spec->dac_info[i].dac; if (!nid) nid = spec->multiout.dac_nids[0]; select_connection(codec, spec->dac_info[i].pin, nid); } + if (spec->auto_mute) { + enable_unsol_pins(codec, cfg->hp_outs, cfg->hp_pins, + CONEXANT_HP_EVENT); + spec->hp_present = detect_jacks(codec, cfg->hp_outs, + cfg->hp_pins); + if (spec->detect_line) { + enable_unsol_pins(codec, cfg->line_outs, + cfg->line_out_pins, + CONEXANT_LINE_EVENT); + spec->line_present = + detect_jacks(codec, cfg->line_outs, + cfg->line_out_pins); + } + } + cx_auto_update_speakers(codec); } static void cx_auto_init_input(struct hda_codec *codec) @@ -3992,6 +4146,13 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) if (err < 0) return err; } + + if (spec->auto_mute) { + err = snd_hda_add_new_ctls(codec, cx_automute_mode_enum); + if (err < 0) + return err; + } + return 0; } -- cgit v1.2.3 From 06dec2282b1366136442950958b517db691533a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 10:00:16 +0200 Subject: ALSA: hda - Use is_jack_detectable() helper Replaced the open-code with the new helper function. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 9 ++++----- sound/pci/hda/patch_realtek.c | 9 +++------ 2 files changed, 7 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 8b735052bc1..43fd403e9bd 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -331,8 +331,8 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t pin = cfg->inputs[idx].pin; - unsigned int val = snd_hda_query_pin_caps(codec, pin); - if (!(val & AC_PINCAP_PRES_DETECT)) + unsigned int val; + if (!is_jack_detectable(codec, pin)) return 0; val = snd_hda_codec_get_pincfg(codec, pin); return (snd_hda_get_input_pin_attr(val) != INPUT_PIN_ATTR_INT); @@ -847,15 +847,14 @@ static void cs_automute(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int caps, hp_present; + unsigned int hp_present; hda_nid_t nid; int i; hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; - caps = snd_hda_query_pin_caps(codec, nid); - if (!(caps & AC_PINCAP_PRES_DETECT)) + if (!is_jack_detectable(codec, nid)) continue; hp_present = snd_hda_jack_detect(codec, nid); if (hp_present) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4dd0cccc5a9..473ddc72b5c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1560,8 +1560,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; - if (!(snd_hda_query_pin_caps(codec, nid) & - AC_PINCAP_PRES_DETECT)) + if (!is_jack_detectable(codec, nid)) continue; snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n", nid); @@ -1576,8 +1575,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) cfg->line_out_pins[0] != cfg->speaker_pins[0]) { for (i = 0; i < cfg->line_outs; i++) { hda_nid_t nid = cfg->line_out_pins[i]; - if (!(snd_hda_query_pin_caps(codec, nid) & - AC_PINCAP_PRES_DETECT)) + if (!is_jack_detectable(codec, nid)) continue; snd_printdd("realtek: Enable Line-Out auto-muting " "on NID 0x%x\n", nid); @@ -19596,8 +19594,7 @@ static void alc680_rec_autoswitch(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { nid = cfg->inputs[i].pin; - if (!(snd_hda_query_pin_caps(codec, nid) & - AC_PINCAP_PRES_DETECT)) + if (!is_jack_detectable(codec, nid)) continue; if (snd_hda_jack_detect(codec, nid)) { if (cfg->inputs[i].type < type_found) { -- cgit v1.2.3 From 1682c8174692180fcb19317704a0d32c8731eb33 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 10:02:27 +0200 Subject: ALSA: hda - Use get_wcaps_type() Replace the open-code with get_wcaps_type() macro. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 43fd403e9bd..4ff70c130da 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -349,8 +349,7 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, hda_nid_t pins[2]; unsigned int type; int j, nums; - type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) - >> AC_WCAP_TYPE_SHIFT; + type = get_wcaps_type(get_wcaps(codec, nid)); if (type != AC_WID_AUD_IN) continue; nums = snd_hda_get_connections(codec, nid, pins, -- cgit v1.2.3 From 52d3cb88d75701f800db16561ff12c7692b56e55 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 10:04:08 +0200 Subject: ALSA: hda - Fix initialization of spec->automute_lines in patch_realtek.c spec->automute_lines shouldn't be set unless the line-detection is available. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 473ddc72b5c..1f00cdbf234 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1584,7 +1584,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) AC_USRSP_EN | ALC880_FRONT_EVENT); spec->detect_line = 1; } - spec->automute_lines = 1; + spec->automute_lines = spec->detect_line; } if (spec->automute) { -- cgit v1.2.3 From 43c1b2e9209cc824177a5a13e34fb21dfab3455a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 10:35:15 +0200 Subject: ALSA: hda - Add support of dock-mic detection to Conexant auto-parser In addition to the normal external mic jack, check also the mic jack on a docking-station as well, and select the input source appropriately. The similar functionality was already implemented in patch_sigmatel.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 98 +++++++++++++++++++++++++----------------- 1 file changed, 59 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ac595363e0e..2a04fea0169 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -85,6 +85,8 @@ struct conexant_spec { unsigned int line_present; unsigned int auto_mic; int auto_mic_ext; /* imux_pins[] index for ext mic */ + int auto_mic_dock; /* imux_pins[] index for dock mic */ + int auto_mic_int; /* imux_pins[] index for int mic */ unsigned int need_dac_fix; hda_nid_t slave_dig_outs[2]; @@ -3737,18 +3739,27 @@ static const struct snd_kcontrol_new cx_auto_capture_mixers[] = { {} }; +static bool select_automic(struct hda_codec *codec, int idx, bool detect) +{ + struct conexant_spec *spec = codec->spec; + if (idx < 0) + return false; + if (detect && !snd_hda_jack_detect(codec, spec->imux_info[idx].pin)) + return false; + cx_auto_mux_enum_update(codec, &spec->private_imux, idx); + return true; +} + /* automatic switch internal and external mic */ static void cx_auto_automic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - int ext_idx = spec->auto_mic_ext; if (!spec->auto_mic) return; - if (snd_hda_jack_detect(codec, spec->imux_info[ext_idx].pin)) - cx_auto_mux_enum_update(codec, &spec->private_imux, ext_idx); - else - cx_auto_mux_enum_update(codec, &spec->private_imux, !ext_idx); + if (!select_automic(codec, spec->auto_mic_ext, true)) + if (!select_automic(codec, spec->auto_mic_dock, true)) + select_automic(codec, spec->auto_mic_int, false); } static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) @@ -3768,42 +3779,43 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) } } -/* return true if it's an internal-mic pin */ -static int is_int_mic(struct hda_codec *codec, hda_nid_t pin) -{ - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); - return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && - snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT; -} - -/* return true if it's an external-mic pin */ -static int is_ext_mic(struct hda_codec *codec, hda_nid_t pin) -{ - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); - return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && - snd_hda_get_input_pin_attr(def_conf) >= INPUT_PIN_ATTR_NORMAL && - is_jack_detectable(codec, pin); -} - /* check whether the pin config is suitable for auto-mic switching; - * auto-mic is enabled only when one int-mic and one-ext mic exist + * auto-mic is enabled only when one int-mic and one ext- and/or + * one dock-mic exist */ static void cx_auto_check_auto_mic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; + int pset[INPUT_PIN_ATTR_NORMAL + 1]; + int i; - if (is_ext_mic(codec, spec->imux_info[0].pin) && - is_int_mic(codec, spec->imux_info[1].pin)) { - spec->auto_mic = 1; - spec->auto_mic_ext = 0; - return; - } - if (is_int_mic(codec, spec->imux_info[0].pin) && - is_ext_mic(codec, spec->imux_info[1].pin)) { - spec->auto_mic = 1; - spec->auto_mic_ext = 1; - return; + for (i = 0; i < INPUT_PIN_ATTR_NORMAL; i++) + pset[i] = -1; + for (i = 0; i < spec->private_imux.num_items; i++) { + hda_nid_t pin = spec->imux_info[i].pin; + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); + int attr; + if (get_defcfg_device(def_conf) != AC_JACK_MIC_IN) + return; /* no-mic input */ + attr = snd_hda_get_input_pin_attr(def_conf); + if (attr == INPUT_PIN_ATTR_UNUSED) + continue; + if (attr > INPUT_PIN_ATTR_NORMAL) + attr = INPUT_PIN_ATTR_NORMAL; + if (attr != INPUT_PIN_ATTR_INT && + !is_jack_detectable(codec, pin)) + continue; + if (pset[attr] >= 0) + return; /* already occupied */ + pset[attr] = i; } + if (pset[INPUT_PIN_ATTR_INT] < 0 || + (pset[INPUT_PIN_ATTR_NORMAL] < 0 && pset[INPUT_PIN_ATTR_DOCK])) + return; /* no input to switch*/ + spec->auto_mic = 1; + spec->auto_mic_ext = pset[INPUT_PIN_ATTR_NORMAL]; + spec->auto_mic_dock = pset[INPUT_PIN_ATTR_DOCK]; + spec->auto_mic_int = pset[INPUT_PIN_ATTR_INT]; } static void cx_auto_parse_input(struct hda_codec *codec) @@ -3832,7 +3844,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) } } } - if (imux->num_items == 2 && cfg->num_inputs == 2) + if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items) cx_auto_check_auto_mic(codec); if (imux->num_items > 1 && !spec->auto_mic) { for (i = 1; i < imux->num_items; i++) { @@ -4022,10 +4034,18 @@ static void cx_auto_init_input(struct hda_codec *codec) } if (spec->auto_mic) { - int ext_idx = spec->auto_mic_ext; - snd_hda_codec_write(codec, cfg->inputs[ext_idx].pin, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | CONEXANT_MIC_EVENT); + if (spec->auto_mic_ext >= 0) { + snd_hda_codec_write(codec, + cfg->inputs[spec->auto_mic_ext].pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | CONEXANT_MIC_EVENT); + } + if (spec->auto_mic_dock >= 0) { + snd_hda_codec_write(codec, + cfg->inputs[spec->auto_mic_dock].pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | CONEXANT_MIC_EVENT); + } cx_auto_automic(codec); } else { select_input_connection(codec, spec->imux_info[0].adc, -- cgit v1.2.3 From e35d9d6a153493055fc888add70786154f00edd4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 11:28:16 +0200 Subject: ALSA: hda - Check unsol-cap in is_jack_detectalbe() Also replace more open-codes with this function. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 3 ++- sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 11 ++++------- 4 files changed, 8 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 01a7cf6b5fb..08ec073444e 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -495,7 +495,8 @@ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) { - return !!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT); + return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) && + (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP); } /* flags for hda_nid_item */ diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4ff70c130da..26a1521045b 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -922,7 +922,7 @@ static void init_output(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + if (is_jack_detectable(codec, nid)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1f00cdbf234..24bc8a67a39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1628,7 +1628,7 @@ static void alc_init_auto_mic(struct hda_codec *codec) } if (!ext || !fixed) return; - if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) + if (!is_jack_detectable(codec, ext)) return; /* no unsol support */ snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n", ext, fixed); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 38f43175168..8d26a5102b0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3114,8 +3114,7 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { - wid_caps = get_wcaps(codec, pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) + if (is_jack_detectable(codec, pins[i])) spec->hp_detect = 1; } nid = dac_nids[i]; @@ -3611,7 +3610,7 @@ static int stac_check_auto_mic(struct hda_codec *codec) return 0; if (!fixed || (!ext && !dock)) return 0; /* no input to switch */ - if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) + if (!is_jack_detectable(codec, ext)) return 0; /* no unsol support */ if (set_mic_route(codec, &spec->ext_mic, ext) || set_mic_route(codec, &spec->int_mic, fixed) || @@ -3926,13 +3925,11 @@ static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; hda_nid_t pin = cfg->hp_pins[0]; - unsigned int wid_caps; if (! pin) return 0; - wid_caps = get_wcaps(codec, pin); - if (wid_caps & AC_WCAP_UNSOL_CAP) + if (is_jack_detectable(codec, pin)) spec->hp_detect = 1; return 0; @@ -4143,7 +4140,7 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, struct sigmatel_event *event; int tag; - if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) + if (!is_jack_detectable(codec, nid)) return 0; event = stac_get_event(codec, nid); if (event) { -- cgit v1.2.3 From 8ed99d976812d1e14a254b9ac1fe6255af8270ff Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 12:05:02 +0200 Subject: ALSA: hda - Add dock-mic detection support to Realtek auto-parser In addition to the normal mic jack, the mic (or line-in) jack on the docking-station is checked also as a candidate for auto-selection. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 82 ++++++++++++++++++++++++++++++++++--------- 1 file changed, 65 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 24bc8a67a39..cee89b1ef39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -360,6 +360,7 @@ struct alc_spec { const struct hda_input_mux *input_mux; unsigned int cur_mux[3]; struct alc_mic_route ext_mic; + struct alc_mic_route dock_mic; struct alc_mic_route int_mic; /* channel model */ @@ -1057,6 +1058,7 @@ static int alc_init_jacks(struct hda_codec *codec) int err; unsigned int hp_nid = spec->autocfg.hp_pins[0]; unsigned int mic_nid = spec->ext_mic.pin; + unsigned int dock_nid = spec->dock_mic.pin; if (hp_nid) { err = snd_hda_input_jack_add(codec, hp_nid, @@ -1073,6 +1075,13 @@ static int alc_init_jacks(struct hda_codec *codec) return err; snd_hda_input_jack_report(codec, mic_nid); } + if (dock_nid) { + err = snd_hda_input_jack_add(codec, dock_nid, + SND_JACK_MICROPHONE, NULL); + if (err < 0) + return err; + snd_hda_input_jack_report(codec, dock_nid); + } #endif /* CONFIG_SND_HDA_INPUT_JACK */ return 0; } @@ -1217,7 +1226,7 @@ static void alc_dual_mic_adc_auto_switch(struct hda_codec *codec) static void alc_mic_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - struct alc_mic_route *dead, *alive; + struct alc_mic_route *dead1, *dead2, *alive; unsigned int present, type; hda_nid_t cap_nid; @@ -1235,13 +1244,24 @@ static void alc_mic_automute(struct hda_codec *codec) cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0]; + alive = &spec->int_mic; + dead1 = &spec->ext_mic; + dead2 = &spec->dock_mic; + present = snd_hda_jack_detect(codec, spec->ext_mic.pin); if (present) { alive = &spec->ext_mic; - dead = &spec->int_mic; - } else { - alive = &spec->int_mic; - dead = &spec->ext_mic; + dead1 = &spec->int_mic; + dead2 = &spec->dock_mic; + } + if (!present && spec->dock_mic.pin > 0) { + present = snd_hda_jack_detect(codec, spec->dock_mic.pin); + if (present) { + alive = &spec->dock_mic; + dead1 = &spec->int_mic; + dead2 = &spec->ext_mic; + } + snd_hda_input_jack_report(codec, spec->dock_mic.pin); } type = get_wcaps_type(get_wcaps(codec, cap_nid)); @@ -1250,9 +1270,14 @@ static void alc_mic_automute(struct hda_codec *codec) snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT, alive->mux_idx, HDA_AMP_MUTE, 0); - snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT, - dead->mux_idx, - HDA_AMP_MUTE, HDA_AMP_MUTE); + if (dead1->pin > 0) + snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT, + dead1->mux_idx, + HDA_AMP_MUTE, HDA_AMP_MUTE); + if (dead2->pin > 0) + snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT, + dead2->mux_idx, + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { /* MUX style (e.g. ALC880) */ snd_hda_codec_write_cache(codec, cap_nid, 0, @@ -1598,15 +1623,10 @@ static void alc_init_auto_mic(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t fixed, ext; + hda_nid_t fixed, ext, dock; int i; - /* there must be only two mic inputs exclusively */ - for (i = 0; i < cfg->num_inputs; i++) - if (cfg->inputs[i].type >= AUTO_PIN_LINE_IN) - return; - - fixed = ext = 0; + fixed = ext = dock = 0; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; unsigned int defcfg; @@ -1615,26 +1635,45 @@ static void alc_init_auto_mic(struct hda_codec *codec) case INPUT_PIN_ATTR_INT: if (fixed) return; /* already occupied */ + if (cfg->inputs[i].type != AUTO_PIN_MIC) + return; /* invalid type */ fixed = nid; break; case INPUT_PIN_ATTR_UNUSED: return; /* invalid entry */ + case INPUT_PIN_ATTR_DOCK: + if (dock) + return; /* already occupied */ + if (cfg->inputs[i].type > AUTO_PIN_LINE_IN) + return; /* invalid type */ + dock = nid; + break; default: if (ext) return; /* already occupied */ + if (cfg->inputs[i].type != AUTO_PIN_MIC) + return; /* invalid type */ ext = nid; break; } } + if (!ext && dock) { + ext = dock; + dock = 0; + } if (!ext || !fixed) return; if (!is_jack_detectable(codec, ext)) return; /* no unsol support */ - snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n", - ext, fixed); + if (dock && !is_jack_detectable(codec, dock)) + return; /* no unsol support */ + snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n", + ext, fixed, dock); spec->ext_mic.pin = ext; + spec->dock_mic.pin = dock; spec->int_mic.pin = fixed; spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */ + spec->dock_mic.mux_idx = MUX_IDX_UNDEF; /* set later */ spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */ spec->auto_mic = 1; snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0, @@ -5716,6 +5755,12 @@ static void fixup_automic_adc(struct hda_codec *codec) spec->capsrc_nids += i; spec->adc_nids += i; spec->num_adc_nids = 1; + /* optional dock-mic */ + eidx = get_connection_index(codec, cap, spec->dock_mic.pin); + if (eidx < 0) + spec->dock_mic.pin = 0; + else + spec->dock_mic.mux_idx = eidx; return; } snd_printd(KERN_INFO "hda_codec: %s: " @@ -5743,6 +5788,8 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin) struct alc_spec *spec = codec->spec; int i; + if (!pin) + return 0; for (i = 0; i < spec->num_adc_nids; i++) { hda_nid_t cap = spec->capsrc_nids ? spec->capsrc_nids[i] : spec->adc_nids[i]; @@ -5783,6 +5830,7 @@ static void fixup_dual_adc_switch(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; init_capsrc_for_pin(codec, spec->ext_mic.pin); + init_capsrc_for_pin(codec, spec->dock_mic.pin); init_capsrc_for_pin(codec, spec->int_mic.pin); } -- cgit v1.2.3 From 1f83ac5ac9fb3e8b2e053de380f55ba412228577 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 12:29:09 +0200 Subject: ALSA: hda - Handle dock line-in as auto-detectable for IDT codecs When a docking-station has a line-in jack, we can handle it also as a detectable jack just like mic-in. This will improve the usability of HP laptops with a docking-station. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8d26a5102b0..7f81cc2274f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3520,14 +3520,18 @@ static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *fixed, hda_nid_t *ext, hda_nid_t *dock) { unsigned int cfg; + unsigned int type; if (!nid) return 0; cfg = snd_hda_codec_get_pincfg(codec, nid); + type = get_defcfg_device(cfg); switch (snd_hda_get_input_pin_attr(cfg)) { case INPUT_PIN_ATTR_INT: if (*fixed) return 1; /* already occupied */ + if (type != AC_JACK_MIC_IN) + return 1; /* invalid type */ *fixed = nid; break; case INPUT_PIN_ATTR_UNUSED: @@ -3535,11 +3539,15 @@ static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, case INPUT_PIN_ATTR_DOCK: if (*dock) return 1; /* already occupied */ + if (type != AC_JACK_MIC_IN && type != AC_JACK_LINE_IN) + return 1; /* invalid type */ *dock = nid; break; default: if (*ext) return 1; /* already occupied */ + if (type != AC_JACK_MIC_IN) + return 1; /* invalid type */ *ext = nid; break; } @@ -3595,10 +3603,6 @@ static int stac_check_auto_mic(struct hda_codec *codec) hda_nid_t fixed, ext, dock; int i; - for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].type >= AUTO_PIN_LINE_IN) - return 0; /* must be exclusively mics */ - } fixed = ext = dock = 0; for (i = 0; i < cfg->num_inputs; i++) if (check_mic_pin(codec, cfg->inputs[i].pin, -- cgit v1.2.3 From b55fcb508dfc9f00056beb33d4c466bc9032dd05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 12:57:46 +0200 Subject: ALSA: hda - Handle dock line-in as auto-detecable for Cxt auto-parser Similar process like in patch_realtek.c and patch_sigmatel.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2a04fea0169..4f37477d3c7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3794,17 +3794,19 @@ static void cx_auto_check_auto_mic(struct hda_codec *codec) for (i = 0; i < spec->private_imux.num_items; i++) { hda_nid_t pin = spec->imux_info[i].pin; unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); - int attr; - if (get_defcfg_device(def_conf) != AC_JACK_MIC_IN) - return; /* no-mic input */ + int type, attr; attr = snd_hda_get_input_pin_attr(def_conf); if (attr == INPUT_PIN_ATTR_UNUSED) - continue; + return; /* invalid entry */ if (attr > INPUT_PIN_ATTR_NORMAL) attr = INPUT_PIN_ATTR_NORMAL; if (attr != INPUT_PIN_ATTR_INT && !is_jack_detectable(codec, pin)) - continue; + return; /* non-detectable pin */ + type = get_defcfg_device(def_conf); + if (type != AC_JACK_MIC_IN && + (attr != INPUT_PIN_ATTR_DOCK || type != AC_JACK_LINE_IN)) + return; /* no valid input type */ if (pset[attr] >= 0) return; /* already occupied */ pset[attr] = i; -- cgit v1.2.3 From 20c304ed84e05a91b2acae36d428d621d3c1d1c6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 May 2011 18:41:25 +0200 Subject: ALSA: hda - Enable snoop bit for AMD controllers AMD Hudson controllers give noisy outputs when the buffer data is rewritten on the fly as PulseAudio does. This seems fixed by the snoop bit enabled just like ATI chipset. Also, disable 64bit DMA as now, to be sure. We can revisit this later. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 010fd318196..ecadb5b7d64 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1092,7 +1092,13 @@ static void azx_init_pci(struct azx *chip) ? "Failed" : "OK"); } break; - + default: + /* AMD Hudson needs the similar snoop, as it seems... */ + if (chip->pci->vendor == PCI_VENDOR_ID_AMD) + update_pci_byte(chip->pci, + ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, + 0x07, ATI_SB450_HDAUDIO_ENABLE_SNOOP); + break; } } @@ -2566,6 +2572,13 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap &= ~ICH6_GCAP_64OK; pci_dev_put(p_smbus); } + } else { + /* FIXME: not sure whether this is really needed, but + * Hudson isn't stable enough for allowing everything... + * let's check later again. + */ + if (chip->pci->vendor == PCI_VENDOR_ID_AMD) + gcap &= ~ICH6_GCAP_64OK; } /* disable 64bit DMA address for Teradici */ -- cgit v1.2.3 From 60ed286a61b43e3fedf18e63f2d71b758f3505aa Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 May 2011 11:28:39 +0200 Subject: ALSA: usb-audio: make hwc_debug a noop in case HW_CONST_DEBUG is not set Just defining it to nothing is dangerous as it can alter the code execution flow, for example when used in as only function in a conditional code block. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/debug.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/debug.h b/sound/usb/debug.h index 343ec2d9ee6..58030176f00 100644 --- a/sound/usb/debug.h +++ b/sound/usb/debug.h @@ -8,7 +8,7 @@ #ifdef HW_CONST_DEBUG #define hwc_debug(fmt, args...) printk(KERN_DEBUG fmt, ##args) #else -#define hwc_debug(fmt, args...) /**/ +#define hwc_debug(fmt, args...) do { } while(0) #endif #endif /* __USBAUDIO_DEBUG_H */ -- cgit v1.2.3 From ee95cb6121dae17bc199cd566503dff1b2dd243b Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 May 2011 11:28:40 +0200 Subject: ALSA: usb-audio: include format.h in format.c Just in case a prototype changes, we'll be warned. This also fixes a sparse warning. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/format.c b/sound/usb/format.c index 5b792d2c806..95d5d7c4f95 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -30,6 +30,7 @@ #include "helper.h" #include "debug.h" #include "clock.h" +#include "format.h" /* * parse the audio format type I descriptor -- cgit v1.2.3 From 759e890f5c25ef087d866b330261e793b03ef7a4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 May 2011 11:28:41 +0200 Subject: ALSA: usb-audio: remove invalid extra mixers for Komplete Audio 6 This was a flaw in the reading of the spec tables - Native Instrument's "Komplete Audio 6" device has no such extra controls. This patch also fixes the device name in two comments. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 17 ----------------- sound/usb/quirks-table.h | 2 +- sound/usb/quirks.c | 2 +- 3 files changed, 2 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 4a7ad7ed62f..73dcc8256bc 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -398,17 +398,6 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new snd_nativeinstruments_ak6_mixers[] = { - { - .name = "Direct Monitor Channel 1+2", - .private_value = _MAKE_NI_CONTROL(0x03, 0x03), - }, - { - .name = "Direct Monitor Channel 3+4", - .private_value = _MAKE_NI_CONTROL(0x03, 0x05), - }, -}; - static struct snd_kcontrol_new snd_nativeinstruments_ta6_mixers[] = { { .name = "Direct Thru Channel A", @@ -537,12 +526,6 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_xonar_u1_controls_create(mixer); break; - case USB_ID(0x17cc, 0x1001): /* Audio Kontrol 6 */ - err = snd_nativeinstruments_create_mixer(mixer, - snd_nativeinstruments_ak6_mixers, - ARRAY_SIZE(snd_nativeinstruments_ak6_mixers)); - break; - case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 54e18c181a1..5c1a176c426 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2332,7 +2332,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Native Instruments MK2 series */ { - /* Audio Kontrol 6 */ + /* Komplete Audio 6 */ .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = 0x17cc, .idProduct = 0x1000, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2452edd2f14..c1a5d7d4580 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -539,7 +539,7 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, /* Access Music VirusTI Desktop */ return snd_usb_accessmusic_boot_quirk(dev); - case USB_ID(0x17cc, 0x1000): /* Audio Kontrol 6 */ + case USB_ID(0x17cc, 0x1000): /* Komplete Audio 6 */ case USB_ID(0x17cc, 0x1010): /* Traktor Audio 6 */ case USB_ID(0x17cc, 0x1020): /* Traktor Audio 10 */ return snd_usb_nativeinstruments_boot_quirk(dev); -- cgit v1.2.3 From 56a9eb1f30c3c7b543a5684e91b47d6ae952feba Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 May 2011 11:28:42 +0200 Subject: ALSA: usb-audio: Add quirk for KORG PANDORA PX5D MIDI interface MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Daniel Mack Reported-and-tested-by: Frédéric Jaume Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5c1a176c426..78c3e8dbca6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2179,6 +2179,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* KORG devices */ +{ + USB_DEVICE_VENDOR_SPEC(0x0944, 0x0200), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "KORG, Inc.", + /* .product_name = "PANDORA PX5D", */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + } +}, + /* AKAI devices */ { USB_DEVICE(0x09e8, 0x0062), -- cgit v1.2.3 From 0ef283247a0cf0fd2e8370ee467030292eb3129e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 May 2011 11:28:43 +0200 Subject: ALSA: usb-audio: add quirks for Roland GR-55 Signed-off-by: Daniel Mack Reported-by: Jeffrey Scott Flesher Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 78c3e8dbca6..690767c0aad 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1651,6 +1651,32 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0127), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "GR-55", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- cgit v1.2.3 From 3bc6fbc7439a88969de97d979795ce7847950668 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 May 2011 11:28:44 +0200 Subject: ALSA: usb-audio: assume valid clock If the interface can't report a clock's validity, assume that it's valid. Signed-off-by: Daniel Mack Reported-by: Vicente Joel Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7754a103454..075195e8661 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -104,6 +104,15 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) int err; unsigned char data; struct usb_device *dev = chip->dev; + struct uac_clock_source_descriptor *cs_desc = + snd_usb_find_clock_source(chip->ctrl_intf, source_id); + + if (!cs_desc) + return 0; + + /* If a clock source can't tell us whether it's valid, we assume it is */ + if (!uac2_control_is_readable(cs_desc->bmControls, UAC2_CS_CONTROL_CLOCK_VALID)) + return 1; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, @@ -114,7 +123,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) if (err < 0) { snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n", __func__, source_id); - return err; + return 0; } return !!data; -- cgit v1.2.3 From c91d9cda558fc348205fa972c8b864f8579ef258 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 May 2011 11:28:45 +0200 Subject: ALSA: usb-audio: handle "Fast Track Ultra" with USB_DEVICE_VENDOR_SPEC() That way, the class compliant MIDI interface is also handled. Signed-off-by: Daniel Mack Reported-and-tested-by: Grant Diffey Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 690767c0aad..78792a8900c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1979,7 +1979,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2080), + USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra", */ @@ -2046,7 +2046,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0763, 0x2081), + USB_DEVICE_VENDOR_SPEC(0x0763, 0x2081), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ -- cgit v1.2.3 From b478b998447d3eb95e03beab1ad132e3e6bc74d2 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 18 May 2011 11:51:15 +0200 Subject: ALSA: hda - Add support of ALC898/899 codec These are compatible with ALC882 codec. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cee89b1ef39..fbf2d437e8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19610,6 +19610,15 @@ static int patch_alc888(struct hda_codec *codec) return patch_alc882(codec); } +static int patch_alc899(struct hda_codec *codec) +{ + if ((alc_read_coef_idx(codec, 0) & 0x2000) != 0x2000) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("ALC898", GFP_KERNEL); + } + return patch_alc882(codec); +} + /* * ALC680 support */ @@ -20097,6 +20106,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, + { .id = 0x10ec0899, .name = "ALC899", .patch = patch_alc899 }, {} /* terminator */ }; -- cgit v1.2.3 From 296f03380e986f910b20dfb5ad7743902e7d840e Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 18 May 2011 11:52:36 +0200 Subject: ALSA: hda - Add support of ALC221 / ALC276 codecs Compatible with ALC269. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fbf2d437e8c..fc0c04699d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -20071,6 +20071,7 @@ static int patch_alc680(struct hda_codec *codec) * patch entries */ static const struct hda_codec_preset snd_hda_preset_realtek[] = { + { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, @@ -20079,6 +20080,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 }, { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 }, + { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, -- cgit v1.2.3 From b896b4ebf0c136b51b184ea9f39247701e332005 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 18 May 2011 11:53:16 +0200 Subject: ALSA: hda - Fix no sound after Windows boot with ALC269 Change power control register to default. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc0c04699d5..473342da306 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -15193,14 +15193,21 @@ static int alc269_fill_coef(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0xd); if ((val & 0x0c00) >> 10 != 0x1) { /* Capless ramp up clock control */ - alc_write_coef_idx(codec, 0xd, val | 1<<10); + alc_write_coef_idx(codec, 0xd, val | (1<<10)); } val = alc_read_coef_idx(codec, 0x17); if ((val & 0x01c0) >> 6 != 0x4) { /* Class D power on reset */ - alc_write_coef_idx(codec, 0x17, val | 1<<7); + alc_write_coef_idx(codec, 0x17, val | (1<<7)); } } + + val = alc_read_coef_idx(codec, 0xd); /* Class D */ + alc_write_coef_idx(codec, 0xd, val | (1<<14)); + + val = alc_read_coef_idx(codec, 0x4); /* HP */ + alc_write_coef_idx(codec, 0x4, val | (1<<11)); + return 0; } -- cgit v1.2.3 From 7cdd8d73139ec935a8e91806131a5b91e26c653e Mon Sep 17 00:00:00 2001 From: Mathieu Bouffard Date: Wed, 18 May 2011 17:09:17 +0200 Subject: ALSA: usb-audio - Add support for USB X-Fi S51 Pro USB X-Fi S51 Pro volume and mute from the volume knob on the unit. Compiled and tested with 2.6.39-rc7-git12 Signed-off-by: Mathieu Bouffard Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 73dcc8256bc..9146cffa6ed 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -61,6 +61,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ + { USB_ID(0x041e, 0x30df), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; @@ -188,6 +189,12 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, !value, 0, NULL, 0, 100); + /* USB X-Fi S51 Pro */ + if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + !value, 0, NULL, 0, 100); else err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, @@ -234,9 +241,13 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) /* USB X-Fi S51 doesn't have a CMSS LED */ if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0) continue; + /* USB X-Fi S51 Pro doesn't have one either */ + if ((mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) && i == 0) + continue; if (i > 1 && /* Live24ext has 2 LEDs only */ (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || + mixer->chip->usb_id == USB_ID(0x041e, 0x30df) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, @@ -512,6 +523,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x041e, 0x3020): case USB_ID(0x041e, 0x3040): case USB_ID(0x041e, 0x3042): + case USB_ID(0x041e, 0x30df): case USB_ID(0x041e, 0x3048): err = snd_audigy2nx_controls_create(mixer); if (err < 0) -- cgit v1.2.3 From ec08b14483de0702ca43e3a8506e149486975f9b Mon Sep 17 00:00:00 2001 From: Ben Gardiner Date: Wed, 18 May 2011 10:03:34 -0400 Subject: ALSA: sound, core, pcm_lib: xrun_log: log also in_interrupt When debugging pcm drivers I found the "period" or "hw" prefix printed by either XRUN_DEBUG_PERIODUPDATE or XRUN_DEBUG_PERIODUPDATE events, respectively to be very useful is observing the interplay between interrupt-context updates and syscall-context updates. Similarly, when debugging overruns with XRUN_DEBUG_LOG it is useful to see the context of the last 10 positions. Add an in_interrupt member to hwptr_log_entry which stores the value of the in_interrupt parameter of snd_pcm_update_hw_ptr0 when the log entry is created. Print a "[Q]" prefix when dumping the log entries if in_interrupt was true. Signed-off-by: Ben Gardiner Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 64449cb8f87..c8c8091f9a7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -189,6 +189,7 @@ static void xrun(struct snd_pcm_substream *substream) #define XRUN_LOG_CNT 10 struct hwptr_log_entry { + unsigned int in_interrupt; unsigned long jiffies; snd_pcm_uframes_t pos; snd_pcm_uframes_t period_size; @@ -204,7 +205,7 @@ struct snd_pcm_hwptr_log { }; static void xrun_log(struct snd_pcm_substream *substream, - snd_pcm_uframes_t pos) + snd_pcm_uframes_t pos, int in_interrupt) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hwptr_log *log = runtime->hwptr_log; @@ -220,6 +221,7 @@ static void xrun_log(struct snd_pcm_substream *substream, return; } entry = &log->entries[log->idx]; + entry->in_interrupt = in_interrupt; entry->jiffies = jiffies; entry->pos = pos; entry->period_size = runtime->period_size; @@ -246,9 +248,11 @@ static void xrun_log_show(struct snd_pcm_substream *substream) entry = &log->entries[idx]; if (entry->period_size == 0) break; - snd_printd("hwptr log: %s: j=%lu, pos=%ld/%ld/%ld, " + snd_printd("hwptr log: %s: %sj=%lu, pos=%ld/%ld/%ld, " "hwptr=%ld/%ld\n", - name, entry->jiffies, (unsigned long)entry->pos, + name, entry->in_interrupt ? "[Q] " : "", + entry->jiffies, + (unsigned long)entry->pos, (unsigned long)entry->period_size, (unsigned long)entry->buffer_size, (unsigned long)entry->old_hw_ptr, @@ -326,7 +330,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } pos -= pos % runtime->min_align; if (xrun_debug(substream, XRUN_DEBUG_LOG)) - xrun_log(substream, pos); + xrun_log(substream, pos, in_interrupt); hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; if (in_interrupt) { -- cgit v1.2.3 From 539494253547d078089cc15490e83f6e5f2e7213 Mon Sep 17 00:00:00 2001 From: Taylor Hutt Date: Tue, 17 May 2011 18:03:54 -0700 Subject: ASoC: Max98095: Move existing NULL check before pointer dereference. Visual inspection shows that max98095_put_eq_enum() and max98095_put_bq_enum() each have a possible NULL deref of 'pdata'. This change moves the NULL check above the use. Signed-off-by: Taylor Hutt Acked-by: Peter Hsiang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index a6cc94e1750..e1d282d477d 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1872,16 +1872,14 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, BUG_ON(channel > 1); - cdata = &max98095->dai[channel]; + if (!pdata || !max98095->eq_textcnt) + return 0; if (sel >= pdata->eq_cfgcnt) return -EINVAL; + cdata = &max98095->dai[channel]; cdata->eq_sel = sel; - - if (!pdata || !max98095->eq_textcnt) - return 0; - fs = cdata->rate; /* Find the selected configuration with nearest sample rate */ @@ -2020,16 +2018,14 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, BUG_ON(channel > 1); - cdata = &max98095->dai[channel]; + if (!pdata || !max98095->bq_textcnt) + return 0; if (sel >= pdata->bq_cfgcnt) return -EINVAL; + cdata = &max98095->dai[channel]; cdata->bq_sel = sel; - - if (!pdata || !max98095->bq_textcnt) - return 0; - fs = cdata->rate; /* Find the selected configuration with nearest sample rate */ -- cgit v1.2.3 From 217658f46c2dbfe260f8f5976f2a201911a2f4c6 Mon Sep 17 00:00:00 2001 From: Ben Gardiner Date: Wed, 18 May 2011 23:52:38 -0400 Subject: ALSA: sound, core, pcm_lib: fix xrun_log The xrun_log function was augmented with the in_interrupt parameter whereas the empty macro definition used when xrun logging is disabled was not. Add a third parameter to the empty macro definition so as to not cause compiler errors when xrun logging (CONFIG_SND_PCM_XRUN_DEBUG) is disabled. Signed-off-by: Ben Gardiner Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index c8c8091f9a7..abfeff1611c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -266,7 +266,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ #define hw_ptr_error(substream, fmt, args...) do { } while (0) -#define xrun_log(substream, pos) do { } while (0) +#define xrun_log(substream, pos, in_interrupt) do { } while (0) #define xrun_log_show(substream) do { } while (0) #endif -- cgit v1.2.3 From 07acecc11139efbc8d0401576e81dc8c188cb092 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 19 May 2011 11:46:03 +0200 Subject: ALSA: HDA: Add jack detection for HDMI Just as for headphones and microphone jacks, this patch adds reporting of HDMI jack status through the input layer. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/patch_hdmi.c | 11 +++++++++++ 2 files changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c63f376ba9a..8edd998509f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5055,6 +5055,8 @@ static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, return "Line-out"; case SND_JACK_HEADSET: return "Headset"; + case SND_JACK_VIDEOOUT: + return "HDMI/DP"; default: return "Misc"; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 6eb209d4b5b..7348296985f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -33,6 +33,7 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" @@ -720,6 +721,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) &spec->sink_eld[index]); /* TODO: do real things about ELD */ } + + snd_hda_input_jack_report(codec, tag); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -912,6 +915,7 @@ static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) { struct hdmi_spec *spec = codec->spec; + int err; if (spec->num_pins >= MAX_HDMI_PINS) { snd_printk(KERN_WARNING @@ -919,6 +923,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) return -E2BIG; } + err = snd_hda_input_jack_add(codec, pin_nid, + SND_JACK_VIDEOOUT, NULL); + if (err < 0) + return err; + snd_hda_input_jack_report(codec, pin_nid); + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); spec->pin[spec->num_pins] = pin_nid; @@ -1120,6 +1130,7 @@ static void generic_hdmi_free(struct hda_codec *codec) for (i = 0; i < spec->num_pins; i++) snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); + snd_hda_input_jack_free(codec); kfree(spec); } -- cgit v1.2.3 From df1fe13289eca445b77471c204408bec67313be3 Mon Sep 17 00:00:00 2001 From: Dmitry Eremin-Solenikov Date: Thu, 19 May 2011 18:48:27 +0400 Subject: ALSA: intel8x0m: enable AMD8111 modem AMD 8111 southbridges contain a controller for MC'97 modem. Enable support for this controller in intel8x0m driver. Signed-off-by: Dmitry Eremin-Solenikov Signed-off-by: Takashi Iwai --- sound/pci/intel8x0m.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 2ae8d29500a..bae3a279f75 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -235,8 +235,8 @@ static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { { PCI_VDEVICE(NVIDIA, 0x0069), DEVICE_NFORCE }, /* NFORCE2 */ { PCI_VDEVICE(NVIDIA, 0x0089), DEVICE_NFORCE }, /* NFORCE2s */ { PCI_VDEVICE(NVIDIA, 0x00d9), DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(AMD, 0x746e), DEVICE_INTEL }, /* AMD8111 */ #if 0 - { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ #endif { 0, } @@ -1261,9 +1261,9 @@ static struct shortname_table { { PCI_DEVICE_ID_NVIDIA_MCP2_MODEM, "NVidia nForce2" }, { PCI_DEVICE_ID_NVIDIA_MCP2S_MODEM, "NVidia nForce2s" }, { PCI_DEVICE_ID_NVIDIA_MCP3_MODEM, "NVidia nForce3" }, + { 0x746e, "AMD AMD8111" }, #if 0 { 0x5455, "ALi M5455" }, - { 0x746d, "AMD AMD8111" }, #endif { 0 }, }; -- cgit v1.2.3 From 39dfe1387060dbad30255fc36c8b3b67d3b359d7 Mon Sep 17 00:00:00 2001 From: Madis Janson Date: Thu, 19 May 2011 18:32:41 +0200 Subject: ALSA: hda - Enable Realtek ALC269 codec input layer beep This fixes the input layer beep not working on some EeePC 1000 models by adding the subsystem id into whitelist. Otherwise the corresponding ALSA mixer is not enabled and stays muted, resulting in no console beep. Signed-off-by: Madis Janson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 473342da306..acb9c898d30 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5943,6 +5943,7 @@ static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), + SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; -- cgit v1.2.3 From 00d2701070c91728988bbfa414a346a23acd8275 Mon Sep 17 00:00:00 2001 From: Dmitry Artamonow Date: Wed, 18 May 2011 19:25:09 +0400 Subject: ASoC: Asahi Kasei AK4641 codec driver A driver for the AK4641 codec used in iPAQ hx4700 and Glofiish M800 among others. Signed-off-by: Harald Welte Signed-off-by: Philipp Zabel Signed-off-by: Dmitry Artamonow Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4641.c | 664 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4641.h | 47 ++++ 4 files changed, 717 insertions(+) create mode 100644 sound/soc/codecs/ak4641.c create mode 100644 sound/soc/codecs/ak4641.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2a6971891d3..98175a096df 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C @@ -139,6 +140,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4641 + tristate + config SND_SOC_AK4642 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4cb2f42dbff..fd8558406ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o @@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c new file mode 100644 index 00000000000..ed96f247c2d --- /dev/null +++ b/sound/soc/codecs/ak4641.c @@ -0,0 +1,664 @@ +/* + * ak4641.c -- AK4641 ALSA Soc Audio driver + * + * Copyright (C) 2008 Harald Welte + * Copyright (C) 2011 Dmitry Artamonow + * + * Based on ak4535.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ak4641.h" + +/* codec private data */ +struct ak4641_priv { + struct snd_soc_codec *codec; + unsigned int sysclk; + int deemph; + int playback_fs; +}; + +/* + * ak4641 register cache + */ +static const u8 ak4641_reg[AK4641_CACHEREGNUM] = { + 0x00, 0x80, 0x00, 0x80, + 0x02, 0x00, 0x11, 0x05, + 0x00, 0x00, 0x36, 0x10, + 0x00, 0x00, 0x57, 0x00, + 0x88, 0x88, 0x08, 0x08 +}; + +static const int deemph_settings[] = {44100, 0, 48000, 32000}; + +static int ak4641_set_deemph(struct snd_soc_codec *codec) +{ + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + int i, best = 0; + + for (i = 0 ; i < ARRAY_SIZE(deemph_settings); i++) { + /* if deemphasis is on, select the nearest available rate */ + if (ak4641->deemph && deemph_settings[i] != 0 && + abs(deemph_settings[i] - ak4641->playback_fs) < + abs(deemph_settings[best] - ak4641->playback_fs)) + best = i; + + if (!ak4641->deemph && deemph_settings[i] == 0) + best = i; + } + + dev_dbg(codec->dev, "Set deemphasis %d\n", best); + + return snd_soc_update_bits(codec, AK4641_DAC, 0x3, best); +} + +static int ak4641_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + ak4641->deemph = deemph; + + return ak4641_set_deemph(codec); +} + +static int ak4641_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = ak4641->deemph; + return 0; +}; + +static const char *ak4641_mono_out[] = {"(L + R)/2", "Hi-Z"}; +static const char *ak4641_hp_out[] = {"Stereo", "Mono"}; +static const char *ak4641_mic_select[] = {"Internal", "External"}; +static const char *ak4641_mic_or_dac[] = {"Microphone", "Voice DAC"}; + + +static const DECLARE_TLV_DB_SCALE(mono_gain_tlv, -1700, 2300, 0); +static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(master_tlv, -12750, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_stereo_sidetone_tlv, -2700, 300, 0); +static const DECLARE_TLV_DB_SCALE(mic_mono_sidetone_tlv, -400, 400, 0); +static const DECLARE_TLV_DB_SCALE(capture_tlv, -800, 50, 0); +static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0); +static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0); + + +static const struct soc_enum ak4641_mono_out_enum = + SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out); +static const struct soc_enum ak4641_hp_out_enum = + SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out); +static const struct soc_enum ak4641_mic_select_enum = + SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select); +static const struct soc_enum ak4641_mic_or_dac_enum = + SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac); + +static const struct snd_kcontrol_new ak4641_snd_controls[] = { + SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum), + SOC_SINGLE_TLV("Mono 1 Gain Volume", AK4641_SIG1, 7, 1, 1, + mono_gain_tlv), + SOC_ENUM("Headphone Output", ak4641_hp_out_enum), + SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0, + ak4641_get_deemph, ak4641_put_deemph), + + SOC_SINGLE_TLV("Mic Boost Volume", AK4641_MIC, 0, 1, 0, mic_boost_tlv), + + SOC_SINGLE("ALC Operation Time", AK4641_TIMER, 0, 3, 0), + SOC_SINGLE("ALC Recovery Time", AK4641_TIMER, 2, 3, 0), + SOC_SINGLE("ALC ZC Time", AK4641_TIMER, 4, 3, 0), + + SOC_SINGLE("ALC 1 Switch", AK4641_ALC1, 5, 1, 0), + + SOC_SINGLE_TLV("ALC Volume", AK4641_ALC2, 0, 71, 0, alc_tlv), + SOC_SINGLE("Left Out Enable Switch", AK4641_SIG2, 1, 1, 0), + SOC_SINGLE("Right Out Enable Switch", AK4641_SIG2, 0, 1, 0), + + SOC_SINGLE_TLV("Capture Volume", AK4641_PGA, 0, 71, 0, capture_tlv), + + SOC_DOUBLE_R_TLV("Master Playback Volume", AK4641_LATT, + AK4641_RATT, 0, 255, 1, master_tlv), + + SOC_SINGLE_TLV("AUX In Volume", AK4641_VOL, 0, 15, 0, aux_in_tlv), + + SOC_SINGLE("Equalizer Switch", AK4641_DAC, 2, 1, 0), + SOC_SINGLE_TLV("EQ1 100 Hz Volume", AK4641_EQLO, 0, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ2 250 Hz Volume", AK4641_EQLO, 4, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ3 1 kHz Volume", AK4641_EQMID, 0, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ4 3.5 kHz Volume", AK4641_EQMID, 4, 15, 1, eq_tlv), + SOC_SINGLE_TLV("EQ5 10 kHz Volume", AK4641_EQHI, 0, 15, 1, eq_tlv), +}; + +/* Mono 1 Mixer */ +static const struct snd_kcontrol_new ak4641_mono1_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("Mic Mono Sidetone Volume", AK4641_VOL, 7, 1, 0, + mic_mono_sidetone_tlv), + SOC_DAPM_SINGLE("Mic Mono Sidetone Switch", AK4641_SIG1, 4, 1, 0), + SOC_DAPM_SINGLE("Mono Playback Switch", AK4641_SIG1, 5, 1, 0), +}; + +/* Stereo Mixer */ +static const struct snd_kcontrol_new ak4641_stereo_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("Mic Sidetone Volume", AK4641_VOL, 4, 7, 0, + mic_stereo_sidetone_tlv), + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4641_SIG2, 4, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", AK4641_SIG2, 7, 1, 0), + SOC_DAPM_SINGLE("Aux Bypass Switch", AK4641_SIG2, 5, 1, 0), +}; + +/* Input Mixer */ +static const struct snd_kcontrol_new ak4641_input_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Capture Switch", AK4641_MIC, 2, 1, 0), + SOC_DAPM_SINGLE("Aux Capture Switch", AK4641_MIC, 5, 1, 0), +}; + +/* Mic mux */ +static const struct snd_kcontrol_new ak4641_mic_mux_control = + SOC_DAPM_ENUM("Mic Select", ak4641_mic_select_enum); + +/* Input mux */ +static const struct snd_kcontrol_new ak4641_input_mux_control = + SOC_DAPM_ENUM("Input Select", ak4641_mic_or_dac_enum); + +/* mono 2 switch */ +static const struct snd_kcontrol_new ak4641_mono2_control = + SOC_DAPM_SINGLE("Switch", AK4641_SIG1, 0, 1, 0); + +/* ak4641 dapm widgets */ +static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, + &ak4641_stereo_mixer_controls[0], + ARRAY_SIZE(ak4641_stereo_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, + &ak4641_mono1_mixer_controls[0], + ARRAY_SIZE(ak4641_mono1_mixer_controls)), + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, + &ak4641_input_mixer_controls[0], + ARRAY_SIZE(ak4641_input_mixer_controls)), + SND_SOC_DAPM_MUX("Mic Mux", SND_SOC_NOPM, 0, 0, + &ak4641_mic_mux_control), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4641_input_mux_control), + SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, + &ak4641_mono2_control), + + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("MOUT1"), + SND_SOC_DAPM_OUTPUT("MOUT2"), + SND_SOC_DAPM_OUTPUT("MICOUT"), + + SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AK4641_PM1, 0, 0), + SND_SOC_DAPM_PGA("Mic", AK4641_PM1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX In", AK4641_PM1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", AK4641_PM1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line Out", AK4641_PM1, 4, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AK4641_PM2, 0, 0), + SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0), + + SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0), + SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), + + SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0), + SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0), + + SND_SOC_DAPM_INPUT("MICIN"), + SND_SOC_DAPM_INPUT("MICEXT"), + SND_SOC_DAPM_INPUT("AUX"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const struct snd_soc_dapm_route ak4641_audio_map[] = { + /* Stereo Mixer */ + {"Stereo Mixer", "Playback Switch", "DAC"}, + {"Stereo Mixer", "Mic Sidetone Switch", "Input Mux"}, + {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, + + /* Mono 1 Mixer */ + {"Mono1 Mixer", "Mic Mono Sidetone Switch", "Input Mux"}, + {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, + + /* Mic */ + {"Mic", NULL, "AIN"}, + {"Mic Mux", "Internal", "Mic Int Bias"}, + {"Mic Mux", "External", "Mic Ext Bias"}, + {"Mic Int Bias", NULL, "MICIN"}, + {"Mic Ext Bias", NULL, "MICEXT"}, + {"MICOUT", NULL, "Mic Mux"}, + + /* Input Mux */ + {"Input Mux", "Microphone", "Mic"}, + {"Input Mux", "Voice DAC", "Voice DAC"}, + + /* Line Out */ + {"LOUT", NULL, "Line Out"}, + {"ROUT", NULL, "Line Out"}, + {"Line Out", NULL, "Stereo Mixer"}, + + /* Mono 1 Out */ + {"MOUT1", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono1 Mixer"}, + + /* Mono 2 Out */ + {"MOUT2", NULL, "Mono 2 Enable"}, + {"Mono 2 Enable", "Switch", "Mono Out 2"}, + {"Mono Out 2", NULL, "Stereo Mixer"}, + + {"Voice ADC", NULL, "Mono 2 Enable"}, + + /* Aux In */ + {"AUX In", NULL, "AUX"}, + + /* ADC */ + {"ADC", NULL, "Input Mixer"}, + {"Input Mixer", "Mic Capture Switch", "Mic"}, + {"Input Mixer", "Aux Capture Switch", "AUX In"}, +}; + +static int ak4641_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + + ak4641->sysclk = freq; + return 0; +} + +static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); + int rate = params_rate(params), fs = 256; + u8 mode2; + + if (rate) + fs = ak4641->sysclk / rate; + else + return -EINVAL; + + /* set fs */ + switch (fs) { + case 1024: + mode2 = (0x2 << 5); + break; + case 512: + mode2 = (0x1 << 5); + break; + case 256: + mode2 = (0x0 << 5); + break; + default: + dev_err(codec->dev, "Error: unsupported fs=%d\n", fs); + return -EINVAL; + } + + snd_soc_update_bits(codec, AK4641_MODE2, (0x3 << 5), mode2); + + /* Update de-emphasis filter for the new rate */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ak4641->playback_fs = rate; + ak4641_set_deemph(codec); + }; + + return 0; +} + +static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 btif; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + btif = (0x3 << 5); + break; + case SND_SOC_DAIFMT_LEFT_J: + btif = (0x2 << 5); + break; + case SND_SOC_DAIFMT_DSP_A: /* MSB after FRM */ + btif = (0x0 << 5); + break; + case SND_SOC_DAIFMT_DSP_B: /* MSB during FRM */ + btif = (0x1 << 5); + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); +} + +static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode1 = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode1 = 0x02; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode1 = 0x01; + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, AK4641_MODE1, mode1); +} + +static int ak4641_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_update_bits(codec, AK4641_DAC, 0x20, mute ? 0x20 : 0); +} + +static int ak4641_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ak4641_platform_data *pdata = codec->dev->platform_data; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + /* unmute */ + snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0); + break; + case SND_SOC_BIAS_PREPARE: + /* mute */ + snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 1); + mdelay(1); + if (pdata && gpio_is_valid(pdata->gpio_npdn)) + gpio_set_value(pdata->gpio_npdn, 1); + mdelay(1); + + ret = snd_soc_cache_sync(codec); + if (ret) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0x80); + snd_soc_update_bits(codec, AK4641_PM2, 0x80, 0); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0); + if (pdata && gpio_is_valid(pdata->gpio_npdn)) + gpio_set_value(pdata->gpio_npdn, 0); + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + codec->cache_sync = 1; + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define AK4641_RATES (SNDRV_PCM_RATE_8000_48000) +#define AK4641_RATES_BT (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000) +#define AK4641_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops ak4641_i2s_dai_ops = { + .hw_params = ak4641_i2s_hw_params, + .set_fmt = ak4641_i2s_set_dai_fmt, + .digital_mute = ak4641_mute, + .set_sysclk = ak4641_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops ak4641_pcm_dai_ops = { + .hw_params = NULL, /* rates are controlled by BT chip */ + .set_fmt = ak4641_pcm_set_dai_fmt, + .digital_mute = ak4641_mute, + .set_sysclk = ak4641_set_dai_sysclk, +}; + +struct snd_soc_dai_driver ak4641_dai[] = { +{ + .name = "ak4641-hifi", + .id = 1, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4641_RATES, + .formats = AK4641_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4641_RATES, + .formats = AK4641_FORMATS, + }, + .ops = &ak4641_i2s_dai_ops, + .symmetric_rates = 1, +}, +{ + .name = "ak4641-voice", + .id = 1, + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = AK4641_RATES_BT, + .formats = AK4641_FORMATS, + }, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 1, + .rates = AK4641_RATES_BT, + .formats = AK4641_FORMATS, + }, + .ops = &ak4641_pcm_dai_ops, + .symmetric_rates = 1, +}, +}; + +static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak4641_resume(struct snd_soc_codec *codec) +{ + ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int ak4641_probe(struct snd_soc_codec *codec) +{ + struct ak4641_platform_data *pdata = codec->dev->platform_data; + int ret; + + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + ret = gpio_request_one(pdata->gpio_power, + GPIOF_OUT_INIT_LOW, "ak4641 power"); + if (ret) + goto err_out; + } + if (gpio_is_valid(pdata->gpio_npdn)) { + ret = gpio_request_one(pdata->gpio_npdn, + GPIOF_OUT_INIT_LOW, "ak4641 npdn"); + if (ret) + goto err_gpio; + + udelay(1); /* > 150 ns */ + gpio_set_value(pdata->gpio_npdn, 1); + } + } + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err_register; + } + + /* power on device */ + ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; + +err_register: + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) + gpio_set_value(pdata->gpio_power, 0); + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } +err_gpio: + if (pdata && gpio_is_valid(pdata->gpio_power)) + gpio_free(pdata->gpio_power); +err_out: + return ret; +} + +static int ak4641_remove(struct snd_soc_codec *codec) +{ + struct ak4641_platform_data *pdata = codec->dev->platform_data; + + ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF); + + if (pdata) { + if (gpio_is_valid(pdata->gpio_power)) { + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_power); + } + if (gpio_is_valid(pdata->gpio_npdn)) + gpio_free(pdata->gpio_npdn); + } + return 0; +} + + +static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { + .probe = ak4641_probe, + .remove = ak4641_remove, + .suspend = ak4641_suspend, + .resume = ak4641_resume, + .controls = ak4641_snd_controls, + .num_controls = ARRAY_SIZE(ak4641_snd_controls), + .dapm_widgets = ak4641_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4641_dapm_widgets), + .dapm_routes = ak4641_audio_map, + .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map), + .set_bias_level = ak4641_set_bias_level, + .reg_cache_size = ARRAY_SIZE(ak4641_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = ak4641_reg, + .reg_cache_step = 1, +}; + + +static int __devinit ak4641_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4641_priv *ak4641; + int ret; + + ak4641 = kzalloc(sizeof(struct ak4641_priv), GFP_KERNEL); + if (!ak4641) + return -ENOMEM; + + i2c_set_clientdata(i2c, ak4641); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641, + ak4641_dai, ARRAY_SIZE(ak4641_dai)); + if (ret < 0) + kfree(ak4641); + + return ret; +} + +static int __devexit ak4641_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + kfree(i2c_get_clientdata(i2c)); + return 0; +} + +static const struct i2c_device_id ak4641_i2c_id[] = { + { "ak4641", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id); + +static struct i2c_driver ak4641_i2c_driver = { + .driver = { + .name = "ak4641", + .owner = THIS_MODULE, + }, + .probe = ak4641_i2c_probe, + .remove = __devexit_p(ak4641_i2c_remove), + .id_table = ak4641_i2c_id, +}; + +static int __init ak4641_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&ak4641_i2c_driver); + if (ret != 0) + pr_err("Failed to register AK4641 I2C driver: %d\n", ret); + + return ret; +} +module_init(ak4641_modinit); + +static void __exit ak4641_exit(void) +{ + i2c_del_driver(&ak4641_i2c_driver); +} +module_exit(ak4641_exit); + +MODULE_DESCRIPTION("SoC AK4641 driver"); +MODULE_AUTHOR("Harald Welte "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4641.h b/sound/soc/codecs/ak4641.h new file mode 100644 index 00000000000..4a263248efe --- /dev/null +++ b/sound/soc/codecs/ak4641.h @@ -0,0 +1,47 @@ +/* + * ak4641.h -- AK4641 SoC Audio driver + * + * Copyright 2008 Harald Welte + * + * Based on ak4535.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4641_H +#define _AK4641_H + +/* AK4641 register space */ + +#define AK4641_PM1 0x00 +#define AK4641_PM2 0x01 +#define AK4641_SIG1 0x02 +#define AK4641_SIG2 0x03 +#define AK4641_MODE1 0x04 +#define AK4641_MODE2 0x05 +#define AK4641_DAC 0x06 +#define AK4641_MIC 0x07 +#define AK4641_TIMER 0x08 +#define AK4641_ALC1 0x09 +#define AK4641_ALC2 0x0a +#define AK4641_PGA 0x0b +#define AK4641_LATT 0x0c +#define AK4641_RATT 0x0d +#define AK4641_VOL 0x0e +#define AK4641_STATUS 0x0f +#define AK4641_EQLO 0x10 +#define AK4641_EQMID 0x11 +#define AK4641_EQHI 0x12 +#define AK4641_BTIF 0x13 + +#define AK4641_CACHEREGNUM 0x14 + + + +#define AK4641_DAI_HIFI 0 +#define AK4641_DAI_VOICE 1 + + +#endif -- cgit v1.2.3 From c26f642e2683fb1a367686bc0bac9f5947885cb6 Mon Sep 17 00:00:00 2001 From: Dmitry Artamonow Date: Wed, 18 May 2011 19:25:10 +0400 Subject: ASoC: add iPAQ hx4700 machine driver AK4641 connected via I2S and I2C, jack detection via GPIO. Signed-off-by: Philipp Zabel Signed-off-by: Dmitry Artamonow Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/hx4700.c | 255 +++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 266 insertions(+) create mode 100644 sound/soc/pxa/hx4700.c (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 580f4857130..33ebc46b45b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -155,6 +155,15 @@ config SND_SOC_RAUMFELD help Say Y if you want to add support for SoC audio on Raumfeld devices +config SND_PXA2XX_SOC_HX4700 + tristate "SoC Audio support for HP iPAQ hx4700" + depends on SND_PXA2XX_SOC && MACH_H4700 + select SND_PXA2XX_SOC_I2S + select SND_SOC_AK4641 + help + Say Y if you want to add support for SoC audio on the + HP iPAQ hx4700. + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 07660165ec8..af357623be9 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o snd-soc-saarb-objs := saarb.o snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o +snd-soc-hx4700-objs := hx4700.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-z2-objs := z2.o @@ -37,6 +38,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c new file mode 100644 index 00000000000..65c124831a0 --- /dev/null +++ b/sound/soc/pxa/hx4700.c @@ -0,0 +1,255 @@ +/* + * SoC audio for HP iPAQ hx4700 + * + * Copyright (c) 2009 Philipp Zabel + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include "pxa2xx-i2s.h" + +#include "../codecs/ak4641.h" + +static struct snd_soc_jack hs_jack; + +/* Headphones jack detection DAPM pin */ +static struct snd_soc_jack_pin hs_jack_pin[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + /* disable speaker when hp jack is inserted */ + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + +/* Headphones jack detection GPIO */ +static struct snd_soc_jack_gpio hs_jack_gpio = { + .gpio = GPIO75_HX4700_EARPHONE_nDET, + .invert = true, + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, +}; + +/* + * iPAQ hx4700 uses I2S for capture and playback. + */ +static int hx4700_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the I2S system clock as output */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + /* inform codec driver about clock freq * + * (PXA I2S always uses divider 256) */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params), + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops hx4700_ops = { + .hw_params = hx4700_hw_params, +}; + +static int hx4700_spk_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(GPIO107_HX4700_SPK_nSD, !!SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +static int hx4700_hp_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(GPIO92_HX4700_HP_DRIVER, !!SND_SOC_DAPM_EVENT_ON(event)); + return 0; +} + +/* hx4700 machine dapm widgets */ +static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power), + SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power), + SND_SOC_DAPM_MIC("Built-in Microphone", NULL), +}; + +/* hx4700 machine audio_map */ +static const struct snd_soc_dapm_route hx4700_audio_map[] = { + + /* Headphone connected to LOUT, ROUT */ + {"Headphone Jack", NULL, "LOUT"}, + {"Headphone Jack", NULL, "ROUT"}, + + /* Speaker connected to MOUT2 */ + {"Speaker", NULL, "MOUT2"}, + + /* Microphone connected to MICIN */ + {"MICIN", NULL, "Built-in Microphone"}, + {"AIN", NULL, "MICOUT"}, +}; + +/* + * Logic for a ak4641 as connected on a HP iPAQ hx4700 + */ +static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int err; + + /* NC codec pins */ + /* FIXME: is anything connected here? */ + snd_soc_dapm_nc_pin(dapm, "MOUT1"); + snd_soc_dapm_nc_pin(dapm, "MICEXT"); + snd_soc_dapm_nc_pin(dapm, "AUX"); + + /* Jack detection API stuff */ + err = snd_soc_jack_new(codec, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack); + if (err) + return err; + + err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin), + hs_jack_pin); + if (err) + return err; + + err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio); + + return err; +} + +/* hx4700 digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link hx4700_dai = { + .name = "ak4641", + .stream_name = "AK4641", + .cpu_dai_name = "pxa2xx-i2s", + .codec_dai_name = "ak4641-hifi", + .platform_name = "pxa-pcm-audio", + .codec_name = "ak4641.0-0012", + .init = hx4700_ak4641_init, + .ops = &hx4700_ops, +}; + +/* hx4700 audio machine driver */ +static struct snd_soc_card snd_soc_card_hx4700 = { + .name = "iPAQ hx4700", + .dai_link = &hx4700_dai, + .num_links = 1, + .dapm_widgets = hx4700_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(hx4700_dapm_widgets), + .dapm_routes = hx4700_audio_map, + .num_dapm_routes = ARRAY_SIZE(hx4700_audio_map), +}; + +static struct gpio hx4700_audio_gpios[] = { + { GPIO107_HX4700_SPK_nSD, GPIOF_OUT_INIT_HIGH, "SPK_POWER" }, + { GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" }, +}; + +static int __devinit hx4700_audio_probe(struct platform_device *pdev) +{ + int ret; + + if (!machine_is_h4700()) + return -ENODEV; + + ret = gpio_request_array(hx4700_audio_gpios, + ARRAY_SIZE(hx4700_audio_gpios)); + if (ret) + return ret; + + snd_soc_card_hx4700.dev = &pdev->dev; + ret = snd_soc_register_card(&snd_soc_card_hx4700); + if (ret) + return ret; + + return 0; +} + +static int __devexit hx4700_audio_remove(struct platform_device *pdev) +{ + snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio); + snd_soc_unregister_card(&snd_soc_card_hx4700); + + gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); + gpio_set_value(GPIO107_HX4700_SPK_nSD, 0); + + gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios)); + return 0; +} + +static struct platform_driver hx4700_audio_driver = { + .driver = { + .name = "hx4700-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = hx4700_audio_probe, + .remove = __devexit_p(hx4700_audio_remove), +}; + +static int __init hx4700_modinit(void) +{ + return platform_driver_register(&hx4700_audio_driver); +} +module_init(hx4700_modinit); + +static void __exit hx4700_modexit(void) +{ + platform_driver_unregister(&hx4700_audio_driver); +} + +module_exit(hx4700_modexit); + +MODULE_AUTHOR("Philipp Zabel"); +MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:hx4700-audio"); -- cgit v1.2.3 From a0c8326397262f1817ee6c5212ad6adf43e3df36 Mon Sep 17 00:00:00 2001 From: Ben Gardiner Date: Wed, 18 May 2011 09:27:45 -0400 Subject: ASoC: davinci-mcasp: enable ping-pong SRAM buffers The davinci-i2s driver copies the platform data for playback and capture sram sizes which is in turn used by davinci-pcm to allocate ping-pong buffers. Copy also the platform data in davinci-mcasp probe. Signed-off-by: Ben Gardiner Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4ddc6d3b667..8566238db2a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -909,6 +909,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_size = pdata->sram_size_playback; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + mem->start); @@ -925,6 +926,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_size = pdata->sram_size_capture; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + mem->start); -- cgit v1.2.3 From 5a2d227fdc7a02ed1b4cebba391d8fb9ad57caaf Mon Sep 17 00:00:00 2001 From: Adrian Wilkins Date: Thu, 19 May 2011 21:52:38 +0100 Subject: ALSA: hda - Fix input-src parse in patch_analog.c Compare pin type enum to the pin type and not the array index. Fixes bug#0005368. Signed-off-by: Adrian Wilkins Cc: (2.6.37 and later) Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 981c631624d..f1b3875c57d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3070,6 +3070,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; + int type = cfg->inputs[i].type; switch (nid) { case 0x15: /* port-C */ snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0); @@ -3079,7 +3080,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) break; } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - i == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN); + type == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN); if (nid != AD1988_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- cgit v1.2.3 From acb373da7cc8aac7f7f4f35056db16606da01d42 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Fri, 20 May 2011 13:04:04 +0800 Subject: ALSA: hda - Remove PCM mixer elements from Virtual Master of realtek Afer commit aa202455eec51699e44f658530728162cefa1307 , none of realtek codec has hardware volume control "PCM Playback Volume" and "PCM Playback Switch". As Virtual Master require all slave controls must have same number of step and dB range. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index acb9c898d30..7a4e10002f5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3155,7 +3155,6 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", - "PCM Playback Volume", NULL, }; @@ -3170,7 +3169,6 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", - "PCM Playback Switch", NULL, }; -- cgit v1.2.3 From bfe9fc8aebc997ce8bcf8ac0586c84a247812064 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Fri, 20 May 2011 14:32:04 +0800 Subject: ALSA: emu10k1 - Add dB range to Bass and Treble for SB Live! As the "Wave", "Wave Surround" or "Front" Playback Volume must be changed to 70% (i.e. -12 dB) so that distortion won't occur when increase Bass and Treble from 50% to 100%, so the maximum gain in Bass and Treble are +12 dB. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emufx.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 7a9401462c1..dae4050ede5 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -303,6 +303,9 @@ static const u32 db_table[101] = { static const DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1); static const DECLARE_TLV_DB_LINEAR(snd_emu10k1_db_linear, TLV_DB_GAIN_MUTE, 0); +/* EMU10K1 bass/treble db gain */ +static const DECLARE_TLV_DB_SCALE(snd_emu10k1_bass_treble_db_scale, -1200, 60, 0); + static const u32 onoff_table[2] = { 0x00000000, 0x00000001 }; @@ -2163,6 +2166,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) ctl->min = 0; ctl->max = 40; ctl->value[0] = ctl->value[1] = 20; + ctl->tlv = snd_emu10k1_bass_treble_db_scale; ctl->translation = EMU10K1_GPR_TRANSLATION_BASS; ctl = &controls[i + 1]; ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -2172,6 +2176,7 @@ static int __devinit _snd_emu10k1_init_efx(struct snd_emu10k1 *emu) ctl->min = 0; ctl->max = 40; ctl->value[0] = ctl->value[1] = 20; + ctl->tlv = snd_emu10k1_bass_treble_db_scale; ctl->translation = EMU10K1_GPR_TRANSLATION_TREBLE; #define BASS_GPR 0x8c -- cgit v1.2.3 From 591e610d651079117bd71c625605a3ebd868b1a1 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 20 May 2011 15:35:43 +0800 Subject: ALSA: hda - add Intel Panther Point HDMI codec id Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7348296985f..32290187322 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1688,6 +1688,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862803, .name = "Eaglelake HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -1733,6 +1734,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862802"); MODULE_ALIAS("snd-hda-codec-id:80862803"); MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:80862805"); +MODULE_ALIAS("snd-hda-codec-id:80862806"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From e28fb9c603f321e0bb4c6c43d58b31de052d6de6 Mon Sep 17 00:00:00 2001 From: Ralf Baechle Date: Fri, 20 May 2011 09:34:24 +0100 Subject: SOUND: OSS: Remove Au1550 driver. This driver does no longer build since at least 2.6.30 and there is a modern ALSA replacement for it. RIP, Rot In Pieces. Signed-off-by: Ralf Baechle Signed-off-by: Takashi Iwai --- sound/oss/Kconfig | 4 - sound/oss/Makefile | 1 - sound/oss/ac97_codec.c | 1203 -------------------------- sound/oss/au1550_ac97.c | 2147 ----------------------------------------------- 4 files changed, 3355 deletions(-) delete mode 100644 sound/oss/ac97_codec.c delete mode 100644 sound/oss/au1550_ac97.c (limited to 'sound') diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 76c09021807..6c93e051f9a 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -22,10 +22,6 @@ config SOUND_VWSND for more info on this driver's capabilities. -config SOUND_AU1550_AC97 - tristate "Au1550/Au1200 AC97 Sound" - depends on SOC_AU1550 || SOC_AU1200 - config SOUND_MSNDCLAS tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" depends on (m || !STANDALONE) && ISA diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 90ffb99c6b1..77f21b68bf0 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -25,7 +25,6 @@ obj-$(CONFIG_SOUND_WAVEARTIST) += waveartist.o obj-$(CONFIG_SOUND_MSNDCLAS) += msnd.o msnd_classic.o obj-$(CONFIG_SOUND_MSNDPIN) += msnd.o msnd_pinnacle.o obj-$(CONFIG_SOUND_VWSND) += vwsnd.o -obj-$(CONFIG_SOUND_AU1550_AC97) += au1550_ac97.o ac97_codec.o obj-$(CONFIG_SOUND_BCM_CS4297A) += swarm_cs4297a.o obj-$(CONFIG_DMASOUND) += dmasound/ diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c deleted file mode 100644 index 0cd23d94888..00000000000 --- a/sound/oss/ac97_codec.c +++ /dev/null @@ -1,1203 +0,0 @@ -/* - * ac97_codec.c: Generic AC97 mixer/modem module - * - * Derived from ac97 mixer in maestro and trident driver. - * - * Copyright 2000 Silicon Integrated System Corporation - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - * - ************************************************************************** - * - * The Intel Audio Codec '97 specification is available at: - * http://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf - * - ************************************************************************** - * - * History - * May 02, 2003 Liam Girdwood - * Removed non existent WM9700 - * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 - * WM9712 and WM9717 - * Mar 28, 2002 Randolph Bentson - * corrections to support WM9707 in ViewPad 1000 - * v0.4 Mar 15 2000 Ollie Lho - * dual codecs support verified with 4 channels output - * v0.3 Feb 22 2000 Ollie Lho - * bug fix for record mask setting - * v0.2 Feb 10 2000 Ollie Lho - * add ac97_read_proc for /proc/driver/{vendor}/ac97 - * v0.1 Jan 14 2000 Ollie Lho - * Isolated from trident.c to support multiple ac97 codec - */ -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#define CODEC_ID_BUFSZ 14 - -static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel); -static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel, - unsigned int left, unsigned int right); -static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val ); -static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask); -static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg); - -static int ac97_init_mixer(struct ac97_codec *codec); - -static int wolfson_init03(struct ac97_codec * codec); -static int wolfson_init04(struct ac97_codec * codec); -static int wolfson_init05(struct ac97_codec * codec); -static int wolfson_init11(struct ac97_codec * codec); -static int wolfson_init13(struct ac97_codec * codec); -static int tritech_init(struct ac97_codec * codec); -static int tritech_maestro_init(struct ac97_codec * codec); -static int sigmatel_9708_init(struct ac97_codec *codec); -static int sigmatel_9721_init(struct ac97_codec *codec); -static int sigmatel_9744_init(struct ac97_codec *codec); -static int ad1886_init(struct ac97_codec *codec); -static int eapd_control(struct ac97_codec *codec, int); -static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode); -static int cmedia_init(struct ac97_codec * codec); -static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode); -static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode); - - -/* - * AC97 operations. - * - * If you are adding a codec then you should be able to use - * eapd_ops - any codec that supports EAPD amp control (most) - * null_ops - any ancient codec that supports nothing - * - * The three functions are - * init - used for non AC97 standard initialisation - * amplifier - used to do amplifier control (1=on 0=off) - * digital - switch to digital modes (0 = analog) - * - * Not all codecs support all features, not all drivers use all the - * operations yet - */ - -static struct ac97_ops null_ops = { NULL, NULL, NULL }; -static struct ac97_ops default_ops = { NULL, eapd_control, NULL }; -static struct ac97_ops default_digital_ops = { NULL, eapd_control, generic_digital_control}; -static struct ac97_ops wolfson_ops03 = { wolfson_init03, NULL, NULL }; -static struct ac97_ops wolfson_ops04 = { wolfson_init04, NULL, NULL }; -static struct ac97_ops wolfson_ops05 = { wolfson_init05, NULL, NULL }; -static struct ac97_ops wolfson_ops11 = { wolfson_init11, NULL, NULL }; -static struct ac97_ops wolfson_ops13 = { wolfson_init13, NULL, NULL }; -static struct ac97_ops tritech_ops = { tritech_init, NULL, NULL }; -static struct ac97_ops tritech_m_ops = { tritech_maestro_init, NULL, NULL }; -static struct ac97_ops sigmatel_9708_ops = { sigmatel_9708_init, NULL, NULL }; -static struct ac97_ops sigmatel_9721_ops = { sigmatel_9721_init, NULL, NULL }; -static struct ac97_ops sigmatel_9744_ops = { sigmatel_9744_init, NULL, NULL }; -static struct ac97_ops crystal_digital_ops = { NULL, eapd_control, crystal_digital_control }; -static struct ac97_ops ad1886_ops = { ad1886_init, eapd_control, NULL }; -static struct ac97_ops cmedia_ops = { NULL, eapd_control, NULL}; -static struct ac97_ops cmedia_digital_ops = { cmedia_init, eapd_control, cmedia_digital_control}; - -/* sorted by vendor/device id */ -static const struct { - u32 id; - char *name; - struct ac97_ops *ops; - int flags; -} ac97_codec_ids[] = { - {0x41445303, "Analog Devices AD1819", &null_ops}, - {0x41445340, "Analog Devices AD1881", &null_ops}, - {0x41445348, "Analog Devices AD1881A", &null_ops}, - {0x41445360, "Analog Devices AD1885", &default_ops}, - {0x41445361, "Analog Devices AD1886", &ad1886_ops}, - {0x41445370, "Analog Devices AD1981", &null_ops}, - {0x41445372, "Analog Devices AD1981A", &null_ops}, - {0x41445374, "Analog Devices AD1981B", &null_ops}, - {0x41445460, "Analog Devices AD1885", &default_ops}, - {0x41445461, "Analog Devices AD1886", &ad1886_ops}, - {0x414B4D00, "Asahi Kasei AK4540", &null_ops}, - {0x414B4D01, "Asahi Kasei AK4542", &null_ops}, - {0x414B4D02, "Asahi Kasei AK4543", &null_ops}, - {0x414C4326, "ALC100P", &null_ops}, - {0x414C4710, "ALC200/200P", &null_ops}, - {0x414C4720, "ALC650", &default_digital_ops}, - {0x434D4941, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME }, - {0x434D4942, "CMedia", &cmedia_ops, AC97_NO_PCM_VOLUME }, - {0x434D4961, "CMedia", &cmedia_digital_ops, AC97_NO_PCM_VOLUME }, - {0x43525900, "Cirrus Logic CS4297", &default_ops}, - {0x43525903, "Cirrus Logic CS4297", &default_ops}, - {0x43525913, "Cirrus Logic CS4297A rev A", &default_ops}, - {0x43525914, "Cirrus Logic CS4297A rev B", &default_ops}, - {0x43525923, "Cirrus Logic CS4298", &null_ops}, - {0x4352592B, "Cirrus Logic CS4294", &null_ops}, - {0x4352592D, "Cirrus Logic CS4294", &null_ops}, - {0x43525931, "Cirrus Logic CS4299 rev A", &crystal_digital_ops}, - {0x43525933, "Cirrus Logic CS4299 rev C", &crystal_digital_ops}, - {0x43525934, "Cirrus Logic CS4299 rev D", &crystal_digital_ops}, - {0x43585430, "CXT48", &default_ops, AC97_DELUDED_MODEM }, - {0x43585442, "CXT66", &default_ops, AC97_DELUDED_MODEM }, - {0x44543031, "Diamond Technology DT0893", &default_ops}, - {0x45838308, "ESS Allegro ES1988", &null_ops}, - {0x49434511, "ICE1232", &null_ops}, /* I hope --jk */ - {0x4e534331, "National Semiconductor LM4549", &null_ops}, - {0x53494c22, "Silicon Laboratory Si3036", &null_ops}, - {0x53494c23, "Silicon Laboratory Si3038", &null_ops}, - {0x545200FF, "TriTech TR?????", &tritech_m_ops}, - {0x54524102, "TriTech TR28022", &null_ops}, - {0x54524103, "TriTech TR28023", &null_ops}, - {0x54524106, "TriTech TR28026", &null_ops}, - {0x54524108, "TriTech TR28028", &tritech_ops}, - {0x54524123, "TriTech TR A5", &null_ops}, - {0x574D4C03, "Wolfson WM9703/07/08/17", &wolfson_ops03}, - {0x574D4C04, "Wolfson WM9704M/WM9704Q", &wolfson_ops04}, - {0x574D4C05, "Wolfson WM9705/WM9710", &wolfson_ops05}, - {0x574D4C09, "Wolfson WM9709", &null_ops}, - {0x574D4C12, "Wolfson WM9711/9712", &wolfson_ops11}, - {0x574D4C13, "Wolfson WM9713", &wolfson_ops13, AC97_DEFAULT_POWER_OFF}, - {0x83847600, "SigmaTel STAC????", &null_ops}, - {0x83847604, "SigmaTel STAC9701/3/4/5", &null_ops}, - {0x83847605, "SigmaTel STAC9704", &null_ops}, - {0x83847608, "SigmaTel STAC9708", &sigmatel_9708_ops}, - {0x83847609, "SigmaTel STAC9721/23", &sigmatel_9721_ops}, - {0x83847644, "SigmaTel STAC9744/45", &sigmatel_9744_ops}, - {0x83847652, "SigmaTel STAC9752/53", &default_ops}, - {0x83847656, "SigmaTel STAC9756/57", &sigmatel_9744_ops}, - {0x83847666, "SigmaTel STAC9750T", &sigmatel_9744_ops}, - {0x83847684, "SigmaTel STAC9783/84?", &null_ops}, - {0x57454301, "Winbond 83971D", &null_ops}, -}; - -/* this table has default mixer values for all OSS mixers. */ -static struct mixer_defaults { - int mixer; - unsigned int value; -} mixer_defaults[SOUND_MIXER_NRDEVICES] = { - /* all values 0 -> 100 in bytes */ - {SOUND_MIXER_VOLUME, 0x4343}, - {SOUND_MIXER_BASS, 0x4343}, - {SOUND_MIXER_TREBLE, 0x4343}, - {SOUND_MIXER_PCM, 0x4343}, - {SOUND_MIXER_SPEAKER, 0x4343}, - {SOUND_MIXER_LINE, 0x4343}, - {SOUND_MIXER_MIC, 0x0000}, - {SOUND_MIXER_CD, 0x4343}, - {SOUND_MIXER_ALTPCM, 0x4343}, - {SOUND_MIXER_IGAIN, 0x4343}, - {SOUND_MIXER_LINE1, 0x4343}, - {SOUND_MIXER_PHONEIN, 0x4343}, - {SOUND_MIXER_PHONEOUT, 0x4343}, - {SOUND_MIXER_VIDEO, 0x4343}, - {-1,0} -}; - -/* table to scale scale from OSS mixer value to AC97 mixer register value */ -static struct ac97_mixer_hw { - unsigned char offset; - int scale; -} ac97_hw[SOUND_MIXER_NRDEVICES]= { - [SOUND_MIXER_VOLUME] = {AC97_MASTER_VOL_STEREO,64}, - [SOUND_MIXER_BASS] = {AC97_MASTER_TONE, 16}, - [SOUND_MIXER_TREBLE] = {AC97_MASTER_TONE, 16}, - [SOUND_MIXER_PCM] = {AC97_PCMOUT_VOL, 32}, - [SOUND_MIXER_SPEAKER] = {AC97_PCBEEP_VOL, 16}, - [SOUND_MIXER_LINE] = {AC97_LINEIN_VOL, 32}, - [SOUND_MIXER_MIC] = {AC97_MIC_VOL, 32}, - [SOUND_MIXER_CD] = {AC97_CD_VOL, 32}, - [SOUND_MIXER_ALTPCM] = {AC97_HEADPHONE_VOL, 64}, - [SOUND_MIXER_IGAIN] = {AC97_RECORD_GAIN, 16}, - [SOUND_MIXER_LINE1] = {AC97_AUX_VOL, 32}, - [SOUND_MIXER_PHONEIN] = {AC97_PHONE_VOL, 32}, - [SOUND_MIXER_PHONEOUT] = {AC97_MASTER_VOL_MONO, 64}, - [SOUND_MIXER_VIDEO] = {AC97_VIDEO_VOL, 32}, -}; - -/* the following tables allow us to go from OSS <-> ac97 quickly. */ -enum ac97_recsettings { - AC97_REC_MIC=0, - AC97_REC_CD, - AC97_REC_VIDEO, - AC97_REC_AUX, - AC97_REC_LINE, - AC97_REC_STEREO, /* combination of all enabled outputs.. */ - AC97_REC_MONO, /*.. or the mono equivalent */ - AC97_REC_PHONE -}; - -static const unsigned int ac97_rm2oss[] = { - [AC97_REC_MIC] = SOUND_MIXER_MIC, - [AC97_REC_CD] = SOUND_MIXER_CD, - [AC97_REC_VIDEO] = SOUND_MIXER_VIDEO, - [AC97_REC_AUX] = SOUND_MIXER_LINE1, - [AC97_REC_LINE] = SOUND_MIXER_LINE, - [AC97_REC_STEREO]= SOUND_MIXER_IGAIN, - [AC97_REC_PHONE] = SOUND_MIXER_PHONEIN -}; - -/* indexed by bit position */ -static const unsigned int ac97_oss_rm[] = { - [SOUND_MIXER_MIC] = AC97_REC_MIC, - [SOUND_MIXER_CD] = AC97_REC_CD, - [SOUND_MIXER_VIDEO] = AC97_REC_VIDEO, - [SOUND_MIXER_LINE1] = AC97_REC_AUX, - [SOUND_MIXER_LINE] = AC97_REC_LINE, - [SOUND_MIXER_IGAIN] = AC97_REC_STEREO, - [SOUND_MIXER_PHONEIN] = AC97_REC_PHONE -}; - -static LIST_HEAD(codecs); -static LIST_HEAD(codec_drivers); -static DEFINE_MUTEX(codec_mutex); - -/* reads the given OSS mixer from the ac97 the caller must have insured that the ac97 knows - about that given mixer, and should be holding a spinlock for the card */ -static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel) -{ - u16 val; - int ret = 0; - int scale; - struct ac97_mixer_hw *mh = &ac97_hw[oss_channel]; - - val = codec->codec_read(codec , mh->offset); - - if (val & AC97_MUTE) { - ret = 0; - } else if (AC97_STEREO_MASK & (1 << oss_channel)) { - /* nice stereo mixers .. */ - int left,right; - - left = (val >> 8) & 0x7f; - right = val & 0x7f; - - if (oss_channel == SOUND_MIXER_IGAIN) { - right = (right * 100) / mh->scale; - left = (left * 100) / mh->scale; - } else { - /* these may have 5 or 6 bit resolution */ - if(oss_channel == SOUND_MIXER_VOLUME || oss_channel == SOUND_MIXER_ALTPCM) - scale = (1 << codec->bit_resolution); - else - scale = mh->scale; - - right = 100 - ((right * 100) / scale); - left = 100 - ((left * 100) / scale); - } - ret = left | (right << 8); - } else if (oss_channel == SOUND_MIXER_SPEAKER) { - ret = 100 - ((((val & 0x1e)>>1) * 100) / mh->scale); - } else if (oss_channel == SOUND_MIXER_PHONEIN) { - ret = 100 - (((val & 0x1f) * 100) / mh->scale); - } else if (oss_channel == SOUND_MIXER_PHONEOUT) { - scale = (1 << codec->bit_resolution); - ret = 100 - (((val & 0x1f) * 100) / scale); - } else if (oss_channel == SOUND_MIXER_MIC) { - ret = 100 - (((val & 0x1f) * 100) / mh->scale); - /* the low bit is optional in the tone sliders and masking - it lets us avoid the 0xf 'bypass'.. */ - } else if (oss_channel == SOUND_MIXER_BASS) { - ret = 100 - ((((val >> 8) & 0xe) * 100) / mh->scale); - } else if (oss_channel == SOUND_MIXER_TREBLE) { - ret = 100 - (((val & 0xe) * 100) / mh->scale); - } - -#ifdef DEBUG - printk("ac97_codec: read OSS mixer %2d (%s ac97 register 0x%02x), " - "0x%04x -> 0x%04x\n", - oss_channel, codec->id ? "Secondary" : "Primary", - mh->offset, val, ret); -#endif - - return ret; -} - -/* write the OSS encoded volume to the given OSS encoded mixer, again caller's job to - make sure all is well in arg land, call with spinlock held */ -static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel, - unsigned int left, unsigned int right) -{ - u16 val = 0; - int scale; - struct ac97_mixer_hw *mh = &ac97_hw[oss_channel]; - -#ifdef DEBUG - printk("ac97_codec: wrote OSS mixer %2d (%s ac97 register 0x%02x), " - "left vol:%2d, right vol:%2d:", - oss_channel, codec->id ? "Secondary" : "Primary", - mh->offset, left, right); -#endif - - if (AC97_STEREO_MASK & (1 << oss_channel)) { - /* stereo mixers */ - if (left == 0 && right == 0) { - val = AC97_MUTE; - } else { - if (oss_channel == SOUND_MIXER_IGAIN) { - right = (right * mh->scale) / 100; - left = (left * mh->scale) / 100; - if (right >= mh->scale) - right = mh->scale-1; - if (left >= mh->scale) - left = mh->scale-1; - } else { - /* these may have 5 or 6 bit resolution */ - if (oss_channel == SOUND_MIXER_VOLUME || - oss_channel == SOUND_MIXER_ALTPCM) - scale = (1 << codec->bit_resolution); - else - scale = mh->scale; - - right = ((100 - right) * scale) / 100; - left = ((100 - left) * scale) / 100; - if (right >= scale) - right = scale-1; - if (left >= scale) - left = scale-1; - } - val = (left << 8) | right; - } - } else if (oss_channel == SOUND_MIXER_BASS) { - val = codec->codec_read(codec , mh->offset) & ~0x0f00; - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val |= (left << 8) & 0x0e00; - } else if (oss_channel == SOUND_MIXER_TREBLE) { - val = codec->codec_read(codec , mh->offset) & ~0x000f; - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val |= left & 0x000e; - } else if(left == 0) { - val = AC97_MUTE; - } else if (oss_channel == SOUND_MIXER_SPEAKER) { - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val = left << 1; - } else if (oss_channel == SOUND_MIXER_PHONEIN) { - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val = left; - } else if (oss_channel == SOUND_MIXER_PHONEOUT) { - scale = (1 << codec->bit_resolution); - left = ((100 - left) * scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val = left; - } else if (oss_channel == SOUND_MIXER_MIC) { - val = codec->codec_read(codec , mh->offset) & ~0x801f; - left = ((100 - left) * mh->scale) / 100; - if (left >= mh->scale) - left = mh->scale-1; - val |= left; - /* the low bit is optional in the tone sliders and masking - it lets us avoid the 0xf 'bypass'.. */ - } -#ifdef DEBUG - printk(" 0x%04x", val); -#endif - - codec->codec_write(codec, mh->offset, val); - -#ifdef DEBUG - val = codec->codec_read(codec, mh->offset); - printk(" -> 0x%04x\n", val); -#endif -} - -/* a thin wrapper for write_mixer */ -static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val ) -{ - unsigned int left,right; - - /* cleanse input a little */ - right = ((val >> 8) & 0xff) ; - left = (val & 0xff) ; - - if (right > 100) right = 100; - if (left > 100) left = 100; - - codec->mixer_state[oss_mixer] = (right << 8) | left; - codec->write_mixer(codec, oss_mixer, left, right); -} - -/* read or write the recmask, the ac97 can really have left and right recording - inputs independently set, but OSS doesn't seem to want us to express that to - the user. the caller guarantees that we have a supported bit set, and they - must be holding the card's spinlock */ -static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask) -{ - unsigned int val; - - if (rw) { - /* read it from the card */ - val = codec->codec_read(codec, AC97_RECORD_SELECT); -#ifdef DEBUG - printk("ac97_codec: ac97 recmask to set to 0x%04x\n", val); -#endif - return (1 << ac97_rm2oss[val & 0x07]); - } - - /* else, write the first set in the mask as the - output */ - /* clear out current set value first (AC97 supports only 1 input!) */ - val = (1 << ac97_rm2oss[codec->codec_read(codec, AC97_RECORD_SELECT) & 0x07]); - if (mask != val) - mask &= ~val; - - val = ffs(mask); - val = ac97_oss_rm[val-1]; - val |= val << 8; /* set both channels */ - -#ifdef DEBUG - printk("ac97_codec: setting ac97 recmask to 0x%04x\n", val); -#endif - - codec->codec_write(codec, AC97_RECORD_SELECT, val); - - return 0; -}; - -static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg) -{ - int i, val = 0; - - if (cmd == SOUND_MIXER_INFO) { - mixer_info info; - memset(&info, 0, sizeof(info)); - strlcpy(info.id, codec->name, sizeof(info.id)); - strlcpy(info.name, codec->name, sizeof(info.name)); - info.modify_counter = codec->modcnt; - if (copy_to_user((void __user *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - if (cmd == SOUND_OLD_MIXER_INFO) { - _old_mixer_info info; - memset(&info, 0, sizeof(info)); - strlcpy(info.id, codec->name, sizeof(info.id)); - strlcpy(info.name, codec->name, sizeof(info.name)); - if (copy_to_user((void __user *)arg, &info, sizeof(info))) - return -EFAULT; - return 0; - } - - if (_IOC_TYPE(cmd) != 'M' || _SIOC_SIZE(cmd) != sizeof(int)) - return -EINVAL; - - if (cmd == OSS_GETVERSION) - return put_user(SOUND_VERSION, (int __user *)arg); - - if (_SIOC_DIR(cmd) == _SIOC_READ) { - switch (_IOC_NR(cmd)) { - case SOUND_MIXER_RECSRC: /* give them the current record source */ - if (!codec->recmask_io) { - val = 0; - } else { - val = codec->recmask_io(codec, 1, 0); - } - break; - - case SOUND_MIXER_DEVMASK: /* give them the supported mixers */ - val = codec->supported_mixers; - break; - - case SOUND_MIXER_RECMASK: /* Arg contains a bit for each supported recording source */ - val = codec->record_sources; - break; - - case SOUND_MIXER_STEREODEVS: /* Mixer channels supporting stereo */ - val = codec->stereo_mixers; - break; - - case SOUND_MIXER_CAPS: - val = SOUND_CAP_EXCL_INPUT; - break; - - default: /* read a specific mixer */ - i = _IOC_NR(cmd); - - if (!supported_mixer(codec, i)) - return -EINVAL; - - /* do we ever want to touch the hardware? */ - /* val = codec->read_mixer(codec, i); */ - val = codec->mixer_state[i]; - break; - } - return put_user(val, (int __user *)arg); - } - - if (_SIOC_DIR(cmd) == (_SIOC_WRITE|_SIOC_READ)) { - codec->modcnt++; - if (get_user(val, (int __user *)arg)) - return -EFAULT; - - switch (_IOC_NR(cmd)) { - case SOUND_MIXER_RECSRC: /* Arg contains a bit for each recording source */ - if (!codec->recmask_io) return -EINVAL; - if (!val) return 0; - if (!(val &= codec->record_sources)) return -EINVAL; - - codec->recmask_io(codec, 0, val); - - return 0; - default: /* write a specific mixer */ - i = _IOC_NR(cmd); - - if (!supported_mixer(codec, i)) - return -EINVAL; - - ac97_set_mixer(codec, i, val); - - return 0; - } - } - return -EINVAL; -} - -/** - * codec_id - Turn id1/id2 into a PnP string - * @id1: Vendor ID1 - * @id2: Vendor ID2 - * @buf: CODEC_ID_BUFSZ byte buffer - * - * Fills buf with a zero terminated PnP ident string for the id1/id2 - * pair. For convenience the return is the passed in buffer pointer. - */ - -static char *codec_id(u16 id1, u16 id2, char *buf) -{ - if(id1&0x8080) { - snprintf(buf, CODEC_ID_BUFSZ, "0x%04x:0x%04x", id1, id2); - } else { - buf[0] = (id1 >> 8); - buf[1] = (id1 & 0xFF); - buf[2] = (id2 >> 8); - snprintf(buf+3, CODEC_ID_BUFSZ - 3, "%d", id2&0xFF); - } - return buf; -} - -/** - * ac97_check_modem - Check if the Codec is a modem - * @codec: codec to check - * - * Return true if the device is an AC97 1.0 or AC97 2.0 modem - */ - -static int ac97_check_modem(struct ac97_codec *codec) -{ - /* Check for an AC97 1.0 soft modem (ID1) */ - if(codec->codec_read(codec, AC97_RESET) & 2) - return 1; - /* Check for an AC97 2.x soft modem */ - codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0L); - if(codec->codec_read(codec, AC97_EXTENDED_MODEM_ID) & 1) - return 1; - return 0; -} - - -/** - * ac97_alloc_codec - Allocate an AC97 codec - * - * Returns a new AC97 codec structure. AC97 codecs may become - * refcounted soon so this interface is needed. Returns with - * one reference taken. - */ - -struct ac97_codec *ac97_alloc_codec(void) -{ - struct ac97_codec *codec = kzalloc(sizeof(struct ac97_codec), GFP_KERNEL); - if(!codec) - return NULL; - - spin_lock_init(&codec->lock); - INIT_LIST_HEAD(&codec->list); - return codec; -} - -EXPORT_SYMBOL(ac97_alloc_codec); - -/** - * ac97_release_codec - Release an AC97 codec - * @codec: codec to release - * - * Release an allocated AC97 codec. This will be refcounted in - * time but for the moment is trivial. Calls the unregister - * handler if the codec is now defunct. - */ - -void ac97_release_codec(struct ac97_codec *codec) -{ - /* Remove from the list first, we don't want to be - "rediscovered" */ - mutex_lock(&codec_mutex); - list_del(&codec->list); - mutex_unlock(&codec_mutex); - /* - * The driver needs to deal with internal - * locking to avoid accidents here. - */ - if(codec->driver) - codec->driver->remove(codec, codec->driver); - kfree(codec); -} - -EXPORT_SYMBOL(ac97_release_codec); - -/** - * ac97_probe_codec - Initialize and setup AC97-compatible codec - * @codec: (in/out) Kernel info for a single AC97 codec - * - * Reset the AC97 codec, then initialize the mixer and - * the rest of the @codec structure. - * - * The codec_read and codec_write fields of @codec are - * required to be setup and working when this function - * is called. All other fields are set by this function. - * - * codec_wait field of @codec can optionally be provided - * when calling this function. If codec_wait is not %NULL, - * this function will call codec_wait any time it is - * necessary to wait for the audio chip to reach the - * codec-ready state. If codec_wait is %NULL, then - * the default behavior is to call schedule_timeout. - * Currently codec_wait is used to wait for AC97 codec - * reset to complete. - * - * Some codecs will power down when a register reset is - * performed. We now check for such codecs. - * - * Returns 1 (true) on success, or 0 (false) on failure. - */ - -int ac97_probe_codec(struct ac97_codec *codec) -{ - u16 id1, id2; - u16 audio; - int i; - char cidbuf[CODEC_ID_BUFSZ]; - u16 f; - struct list_head *l; - struct ac97_driver *d; - - /* wait for codec-ready state */ - if (codec->codec_wait) - codec->codec_wait(codec); - else - udelay(10); - - /* will the codec power down if register reset ? */ - id1 = codec->codec_read(codec, AC97_VENDOR_ID1); - id2 = codec->codec_read(codec, AC97_VENDOR_ID2); - codec->name = NULL; - codec->codec_ops = &null_ops; - for (i = 0; i < ARRAY_SIZE(ac97_codec_ids); i++) { - if (ac97_codec_ids[i].id == ((id1 << 16) | id2)) { - codec->type = ac97_codec_ids[i].id; - codec->name = ac97_codec_ids[i].name; - codec->codec_ops = ac97_codec_ids[i].ops; - codec->flags = ac97_codec_ids[i].flags; - break; - } - } - - codec->model = (id1 << 16) | id2; - if ((codec->flags & AC97_DEFAULT_POWER_OFF) == 0) { - /* reset codec and wait for the ready bit before we continue */ - codec->codec_write(codec, AC97_RESET, 0L); - if (codec->codec_wait) - codec->codec_wait(codec); - else - udelay(10); - } - - /* probing AC97 codec, AC97 2.0 says that bit 15 of register 0x00 (reset) should - * be read zero. - * - * FIXME: is the following comment outdated? -jgarzik - * Probing of AC97 in this way is not reliable, it is not even SAFE !! - */ - if ((audio = codec->codec_read(codec, AC97_RESET)) & 0x8000) { - printk(KERN_ERR "ac97_codec: %s ac97 codec not present\n", - (codec->id & 0x2) ? (codec->id&1 ? "4th" : "Tertiary") - : (codec->id&1 ? "Secondary": "Primary")); - return 0; - } - - /* probe for Modem Codec */ - codec->modem = ac97_check_modem(codec); - - /* enable SPDIF */ - f = codec->codec_read(codec, AC97_EXTENDED_STATUS); - if((codec->codec_ops == &null_ops) && (f & 4)) - codec->codec_ops = &default_digital_ops; - - /* A device which thinks its a modem but isn't */ - if(codec->flags & AC97_DELUDED_MODEM) - codec->modem = 0; - - if (codec->name == NULL) - codec->name = "Unknown"; - printk(KERN_INFO "ac97_codec: AC97 %s codec, id: %s (%s)\n", - codec->modem ? "Modem" : (audio ? "Audio" : ""), - codec_id(id1, id2, cidbuf), codec->name); - - if(!ac97_init_mixer(codec)) - return 0; - - /* - * Attach last so the caller can override the mixer - * callbacks. - */ - - mutex_lock(&codec_mutex); - list_add(&codec->list, &codecs); - - list_for_each(l, &codec_drivers) { - d = list_entry(l, struct ac97_driver, list); - if ((codec->model ^ d->codec_id) & d->codec_mask) - continue; - if(d->probe(codec, d) == 0) - { - codec->driver = d; - break; - } - } - - mutex_unlock(&codec_mutex); - return 1; -} - -static int ac97_init_mixer(struct ac97_codec *codec) -{ - u16 cap; - int i; - - cap = codec->codec_read(codec, AC97_RESET); - - /* mixer masks */ - codec->supported_mixers = AC97_SUPPORTED_MASK; - codec->stereo_mixers = AC97_STEREO_MASK; - codec->record_sources = AC97_RECORD_MASK; - if (!(cap & 0x04)) - codec->supported_mixers &= ~(SOUND_MASK_BASS|SOUND_MASK_TREBLE); - if (!(cap & 0x10)) - codec->supported_mixers &= ~SOUND_MASK_ALTPCM; - - - /* detect bit resolution */ - codec->codec_write(codec, AC97_MASTER_VOL_STEREO, 0x2020); - if(codec->codec_read(codec, AC97_MASTER_VOL_STEREO) == 0x2020) - codec->bit_resolution = 6; - else - codec->bit_resolution = 5; - - /* generic OSS to AC97 wrapper */ - codec->read_mixer = ac97_read_mixer; - codec->write_mixer = ac97_write_mixer; - codec->recmask_io = ac97_recmask_io; - codec->mixer_ioctl = ac97_mixer_ioctl; - - /* initialize mixer channel volumes */ - for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) { - struct mixer_defaults *md = &mixer_defaults[i]; - if (md->mixer == -1) - break; - if (!supported_mixer(codec, md->mixer)) - continue; - ac97_set_mixer(codec, md->mixer, md->value); - } - - /* codec specific initialization for 4-6 channel output or secondary codec stuff */ - if (codec->codec_ops->init != NULL) { - codec->codec_ops->init(codec); - } - - /* - * Volume is MUTE only on this device. We have to initialise - * it but its useless beyond that. - */ - if(codec->flags & AC97_NO_PCM_VOLUME) - { - codec->supported_mixers &= ~SOUND_MASK_PCM; - printk(KERN_WARNING "AC97 codec does not have proper volume support.\n"); - } - return 1; -} - -#define AC97_SIGMATEL_ANALOG 0x6c /* Analog Special */ -#define AC97_SIGMATEL_DAC2INVERT 0x6e -#define AC97_SIGMATEL_BIAS1 0x70 -#define AC97_SIGMATEL_BIAS2 0x72 -#define AC97_SIGMATEL_MULTICHN 0x74 /* Multi-Channel programming */ -#define AC97_SIGMATEL_CIC1 0x76 -#define AC97_SIGMATEL_CIC2 0x78 - - -static int sigmatel_9708_init(struct ac97_codec * codec) -{ - u16 codec72, codec6c; - - codec72 = codec->codec_read(codec, AC97_SIGMATEL_BIAS2) & 0x8000; - codec6c = codec->codec_read(codec, AC97_SIGMATEL_ANALOG); - - if ((codec72==0) && (codec6c==0)) { - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1000); - codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0007); - } else if ((codec72==0x8000) && (codec6c==0)) { - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x1001); - codec->codec_write(codec, AC97_SIGMATEL_DAC2INVERT, 0x0008); - } else if ((codec72==0x8000) && (codec6c==0x0080)) { - /* nothing */ - } - codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000); - return 0; -} - - -static int sigmatel_9721_init(struct ac97_codec * codec) -{ - /* Only set up secondary codec */ - if (codec->id == 0) - return 0; - - codec->codec_write(codec, AC97_SURROUND_MASTER, 0L); - - /* initialize SigmaTel STAC9721/23 as secondary codec, decoding AC link - sloc 3,4 = 0x01, slot 7,8 = 0x00, */ - codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x00); - - /* we don't have the crystal when we are on an AMR card, so use - BIT_CLK as our clock source. Write the magic word ABBA and read - back to enable register 0x78 */ - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_read(codec, AC97_SIGMATEL_CIC1); - - /* sync all the clocks*/ - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x3802); - - return 0; -} - - -static int sigmatel_9744_init(struct ac97_codec * codec) -{ - // patch for SigmaTel - codec->codec_write(codec, AC97_SIGMATEL_CIC1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_CIC2, 0x0000); // is this correct? --jk - codec->codec_write(codec, AC97_SIGMATEL_BIAS1, 0xabba); - codec->codec_write(codec, AC97_SIGMATEL_BIAS2, 0x0002); - codec->codec_write(codec, AC97_SIGMATEL_MULTICHN, 0x0000); - return 0; -} - -static int cmedia_init(struct ac97_codec *codec) -{ - /* Initialise the CMedia 9739 */ - /* - We could set various options here - Register 0x20 bit 0x100 sets mic as center bass - Also do multi_channel_ctrl &=~0x3000 |=0x1000 - - For now we set up the GPIO and PC beep - */ - - u16 v; - - /* MIC */ - codec->codec_write(codec, 0x64, 0x3000); - v = codec->codec_read(codec, 0x64); - v &= ~0x8000; - codec->codec_write(codec, 0x64, v); - codec->codec_write(codec, 0x70, 0x0100); - codec->codec_write(codec, 0x72, 0x0020); - return 0; -} - -#define AC97_WM97XX_FMIXER_VOL 0x72 -#define AC97_WM97XX_RMIXER_VOL 0x74 -#define AC97_WM97XX_TEST 0x5a -#define AC97_WM9704_RPCM_VOL 0x70 -#define AC97_WM9711_OUT3VOL 0x16 - -static int wolfson_init03(struct ac97_codec * codec) -{ - /* this is known to work for the ViewSonic ViewPad 1000 */ - codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808); - codec->codec_write(codec, AC97_GENERAL_PURPOSE, 0x8000); - return 0; -} - -static int wolfson_init04(struct ac97_codec * codec) -{ - codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808); - codec->codec_write(codec, AC97_WM97XX_RMIXER_VOL, 0x0808); - - // patch for DVD noise - codec->codec_write(codec, AC97_WM97XX_TEST, 0x0200); - - // init vol as PCM vol - codec->codec_write(codec, AC97_WM9704_RPCM_VOL, - codec->codec_read(codec, AC97_PCMOUT_VOL)); - - /* set rear surround volume */ - codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000); - return 0; -} - -/* WM9705, WM9710 */ -static int wolfson_init05(struct ac97_codec * codec) -{ - /* set front mixer volume */ - codec->codec_write(codec, AC97_WM97XX_FMIXER_VOL, 0x0808); - return 0; -} - -/* WM9711, WM9712 */ -static int wolfson_init11(struct ac97_codec * codec) -{ - /* stop pop's during suspend/resume */ - codec->codec_write(codec, AC97_WM97XX_TEST, - codec->codec_read(codec, AC97_WM97XX_TEST) & 0xffbf); - - /* set out3 volume */ - codec->codec_write(codec, AC97_WM9711_OUT3VOL, 0x0808); - return 0; -} - -/* WM9713 */ -static int wolfson_init13(struct ac97_codec * codec) -{ - codec->codec_write(codec, AC97_RECORD_GAIN, 0x00a0); - codec->codec_write(codec, AC97_POWER_CONTROL, 0x0000); - codec->codec_write(codec, AC97_EXTENDED_MODEM_ID, 0xDA00); - codec->codec_write(codec, AC97_EXTEND_MODEM_STAT, 0x3810); - codec->codec_write(codec, AC97_PHONE_VOL, 0x0808); - codec->codec_write(codec, AC97_PCBEEP_VOL, 0x0808); - - return 0; -} - -static int tritech_init(struct ac97_codec * codec) -{ - codec->codec_write(codec, 0x26, 0x0300); - codec->codec_write(codec, 0x26, 0x0000); - codec->codec_write(codec, AC97_SURROUND_MASTER, 0x0000); - codec->codec_write(codec, AC97_RESERVED_3A, 0x0000); - return 0; -} - - -/* copied from drivers/sound/maestro.c */ -static int tritech_maestro_init(struct ac97_codec * codec) -{ - /* no idea what this does */ - codec->codec_write(codec, 0x2A, 0x0001); - codec->codec_write(codec, 0x2C, 0x0000); - codec->codec_write(codec, 0x2C, 0XFFFF); - return 0; -} - - - -/* - * Presario700 workaround - * for Jack Sense/SPDIF Register mis-setting causing - * no audible output - * by Santiago Nullo 04/05/2002 - */ - -#define AC97_AD1886_JACK_SENSE 0x72 - -static int ad1886_init(struct ac97_codec * codec) -{ - /* from AD1886 Specs */ - codec->codec_write(codec, AC97_AD1886_JACK_SENSE, 0x0010); - return 0; -} - - - - -/* - * This is basically standard AC97. It should work as a default for - * almost all modern codecs. Note that some cards wire EAPD *backwards* - * That side of it is up to the card driver not us to cope with. - * - */ - -static int eapd_control(struct ac97_codec * codec, int on) -{ - if(on) - codec->codec_write(codec, AC97_POWER_CONTROL, - codec->codec_read(codec, AC97_POWER_CONTROL)|0x8000); - else - codec->codec_write(codec, AC97_POWER_CONTROL, - codec->codec_read(codec, AC97_POWER_CONTROL)&~0x8000); - return 0; -} - -static int generic_digital_control(struct ac97_codec *codec, int slots, int rate, int mode) -{ - u16 reg; - - reg = codec->codec_read(codec, AC97_SPDIF_CONTROL); - - switch(rate) - { - /* Off by default */ - default: - case 0: - reg = codec->codec_read(codec, AC97_EXTENDED_STATUS); - codec->codec_write(codec, AC97_EXTENDED_STATUS, (reg & ~AC97_EA_SPDIF)); - if(rate == 0) - return 0; - return -EINVAL; - case 1: - reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_48K; - break; - case 2: - reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_44K; - break; - case 3: - reg = (reg & AC97_SC_SPSR_MASK) | AC97_SC_SPSR_32K; - break; - } - - reg &= ~AC97_SC_CC_MASK; - reg |= (mode & AUDIO_CCMASK) << 6; - - if(mode & AUDIO_DIGITAL) - reg |= 2; - if(mode & AUDIO_PRO) - reg |= 1; - if(mode & AUDIO_DRS) - reg |= 0x4000; - - codec->codec_write(codec, AC97_SPDIF_CONTROL, reg); - - reg = codec->codec_read(codec, AC97_EXTENDED_STATUS); - reg &= (AC97_EA_SLOT_MASK); - reg |= AC97_EA_VRA | AC97_EA_SPDIF | slots; - codec->codec_write(codec, AC97_EXTENDED_STATUS, reg); - - reg = codec->codec_read(codec, AC97_EXTENDED_STATUS); - if(!(reg & 0x0400)) - { - codec->codec_write(codec, AC97_EXTENDED_STATUS, reg & ~ AC97_EA_SPDIF); - return -EINVAL; - } - return 0; -} - -/* - * Crystal digital audio control (CS4299) - */ - -static int crystal_digital_control(struct ac97_codec *codec, int slots, int rate, int mode) -{ - u16 cv; - - if(mode & AUDIO_DIGITAL) - return -EINVAL; - - switch(rate) - { - case 0: cv = 0x0; break; /* SPEN off */ - case 48000: cv = 0x8004; break; /* 48KHz digital */ - case 44100: cv = 0x8104; break; /* 44.1KHz digital */ - case 32768: /* 32Khz */ - default: - return -EINVAL; - } - codec->codec_write(codec, 0x68, cv); - return 0; -} - -/* - * CMedia digital audio control - * Needs more work. - */ - -static int cmedia_digital_control(struct ac97_codec *codec, int slots, int rate, int mode) -{ - u16 cv; - - if(mode & AUDIO_DIGITAL) - return -EINVAL; - - switch(rate) - { - case 0: cv = 0x0001; break; /* SPEN off */ - case 48000: cv = 0x0009; break; /* 48KHz digital */ - default: - return -EINVAL; - } - codec->codec_write(codec, 0x2A, 0x05c4); - codec->codec_write(codec, 0x6C, cv); - - /* Switch on mix to surround */ - cv = codec->codec_read(codec, 0x64); - cv &= ~0x0200; - if(mode) - cv |= 0x0200; - codec->codec_write(codec, 0x64, cv); - return 0; -} - - -/* copied from drivers/sound/maestro.c */ -#if 0 /* there has been 1 person on the planet with a pt101 that we - know of. If they care, they can put this back in :) */ -static int pt101_init(struct ac97_codec * codec) -{ - printk(KERN_INFO "ac97_codec: PT101 Codec detected, initializing but _not_ installing mixer device.\n"); - /* who knows.. */ - codec->codec_write(codec, 0x2A, 0x0001); - codec->codec_write(codec, 0x2C, 0x0000); - codec->codec_write(codec, 0x2C, 0xFFFF); - codec->codec_write(codec, 0x10, 0x9F1F); - codec->codec_write(codec, 0x12, 0x0808); - codec->codec_write(codec, 0x14, 0x9F1F); - codec->codec_write(codec, 0x16, 0x9F1F); - codec->codec_write(codec, 0x18, 0x0404); - codec->codec_write(codec, 0x1A, 0x0000); - codec->codec_write(codec, 0x1C, 0x0000); - codec->codec_write(codec, 0x02, 0x0404); - codec->codec_write(codec, 0x04, 0x0808); - codec->codec_write(codec, 0x0C, 0x801F); - codec->codec_write(codec, 0x0E, 0x801F); - return 0; -} -#endif - - -EXPORT_SYMBOL(ac97_probe_codec); - -MODULE_LICENSE("GPL"); - diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c deleted file mode 100644 index a8f626d99c5..00000000000 --- a/sound/oss/au1550_ac97.c +++ /dev/null @@ -1,2147 +0,0 @@ -/* - * au1550_ac97.c -- Sound driver for Alchemy Au1550 MIPS Internet Edge - * Processor. - * - * Copyright 2004 Embedded Edge, LLC - * dan@embeddededge.com - * - * Mostly copied from the au1000.c driver and some from the - * PowerMac dbdma driver. - * We assume the processor can do memory coherent DMA. - * - * Ported to 2.6 by Matt Porter - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED - * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF - * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN - * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, - * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT - * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF - * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON - * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT - * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF - * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. - * - * You should have received a copy of the GNU General Public License along - * with this program; if not, write to the Free Software Foundation, Inc., - * 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#undef DEBUG - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include -#include -#include - -#undef OSS_DOCUMENTED_MIXER_SEMANTICS - -/* misc stuff */ -#define POLL_COUNT 0x50000 -#define AC97_EXT_DACS (AC97_EXTID_SDAC | AC97_EXTID_CDAC | AC97_EXTID_LDAC) - -/* The number of DBDMA ring descriptors to allocate. No sense making - * this too large....if you can't keep up with a few you aren't likely - * to be able to with lots of them, either. - */ -#define NUM_DBDMA_DESCRIPTORS 4 - -#define err(format, arg...) printk(KERN_ERR format "\n" , ## arg) - -/* Boot options - * 0 = no VRA, 1 = use VRA if codec supports it - */ -static DEFINE_MUTEX(au1550_ac97_mutex); -static int vra = 1; -module_param(vra, bool, 0); -MODULE_PARM_DESC(vra, "if 1 use VRA if codec supports it"); - -static struct au1550_state { - /* soundcore stuff */ - int dev_audio; - - struct ac97_codec *codec; - unsigned codec_base_caps; /* AC'97 reg 00h, "Reset Register" */ - unsigned codec_ext_caps; /* AC'97 reg 28h, "Extended Audio ID" */ - int no_vra; /* do not use VRA */ - - spinlock_t lock; - struct mutex open_mutex; - struct mutex sem; - fmode_t open_mode; - wait_queue_head_t open_wait; - - struct dmabuf { - u32 dmanr; - unsigned sample_rate; - unsigned src_factor; - unsigned sample_size; - int num_channels; - int dma_bytes_per_sample; - int user_bytes_per_sample; - int cnt_factor; - - void *rawbuf; - unsigned buforder; - unsigned numfrag; - unsigned fragshift; - void *nextIn; - void *nextOut; - int count; - unsigned total_bytes; - unsigned error; - wait_queue_head_t wait; - - /* redundant, but makes calculations easier */ - unsigned fragsize; - unsigned dma_fragsize; - unsigned dmasize; - unsigned dma_qcount; - - /* OSS stuff */ - unsigned mapped:1; - unsigned ready:1; - unsigned stopped:1; - unsigned ossfragshift; - int ossmaxfrags; - unsigned subdivision; - } dma_dac, dma_adc; -} au1550_state; - -static unsigned -ld2(unsigned int x) -{ - unsigned r = 0; - - if (x >= 0x10000) { - x >>= 16; - r += 16; - } - if (x >= 0x100) { - x >>= 8; - r += 8; - } - if (x >= 0x10) { - x >>= 4; - r += 4; - } - if (x >= 4) { - x >>= 2; - r += 2; - } - if (x >= 2) - r++; - return r; -} - -static void -au1550_delay(int msec) -{ - if (in_interrupt()) - return; - - schedule_timeout_uninterruptible(msecs_to_jiffies(msec)); -} - -static u16 -rdcodec(struct ac97_codec *codec, u8 addr) -{ - struct au1550_state *s = codec->private_data; - unsigned long flags; - u32 cmd, val; - u16 data; - int i; - - spin_lock_irqsave(&s->lock, flags); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) - err("rdcodec: codec cmd pending expired!"); - - cmd = (u32)PSC_AC97CDC_INDX(addr); - cmd |= PSC_AC97CDC_RD; /* read command */ - au_writel(cmd, PSC_AC97CDC); - au_sync(); - - /* now wait for the data - */ - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) { - err("rdcodec: read poll expired!"); - data = 0; - goto out; - } - - /* wait for command done? - */ - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97EVNT); - au_sync(); - if (val & PSC_AC97EVNT_CD) - break; - } - if (i == POLL_COUNT) { - err("rdcodec: read cmdwait expired!"); - data = 0; - goto out; - } - - data = au_readl(PSC_AC97CDC) & 0xffff; - au_sync(); - - /* Clear command done event. - */ - au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT); - au_sync(); - - out: - spin_unlock_irqrestore(&s->lock, flags); - - return data; -} - - -static void -wrcodec(struct ac97_codec *codec, u8 addr, u16 data) -{ - struct au1550_state *s = codec->private_data; - unsigned long flags; - u32 cmd, val; - int i; - - spin_lock_irqsave(&s->lock, flags); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) - err("wrcodec: codec cmd pending expired!"); - - cmd = (u32)PSC_AC97CDC_INDX(addr); - cmd |= (u32)data; - au_writel(cmd, PSC_AC97CDC); - au_sync(); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (!(val & PSC_AC97STAT_CP)) - break; - } - if (i == POLL_COUNT) - err("wrcodec: codec cmd pending expired!"); - - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97EVNT); - au_sync(); - if (val & PSC_AC97EVNT_CD) - break; - } - if (i == POLL_COUNT) - err("wrcodec: read cmdwait expired!"); - - /* Clear command done event. - */ - au_writel(PSC_AC97EVNT_CD, PSC_AC97EVNT); - au_sync(); - - spin_unlock_irqrestore(&s->lock, flags); -} - -static void -waitcodec(struct ac97_codec *codec) -{ - u16 temp; - u32 val; - int i; - - /* codec_wait is used to wait for a ready state after - * an AC97C_RESET. - */ - au1550_delay(10); - - /* first poll the CODEC_READY tag bit - */ - for (i = 0; i < POLL_COUNT; i++) { - val = au_readl(PSC_AC97STAT); - au_sync(); - if (val & PSC_AC97STAT_CR) - break; - } - if (i == POLL_COUNT) { - err("waitcodec: CODEC_READY poll expired!"); - return; - } - - /* get AC'97 powerdown control/status register - */ - temp = rdcodec(codec, AC97_POWER_CONTROL); - - /* If anything is powered down, power'em up - */ - if (temp & 0x7f00) { - /* Power on - */ - wrcodec(codec, AC97_POWER_CONTROL, 0); - au1550_delay(100); - - /* Reread - */ - temp = rdcodec(codec, AC97_POWER_CONTROL); - } - - /* Check if Codec REF,ANL,DAC,ADC ready - */ - if ((temp & 0x7f0f) != 0x000f) - err("codec reg 26 status (0x%x) not ready!!", temp); -} - -/* stop the ADC before calling */ -static void -set_adc_rate(struct au1550_state *s, unsigned rate) -{ - struct dmabuf *adc = &s->dma_adc; - struct dmabuf *dac = &s->dma_dac; - unsigned adc_rate, dac_rate; - u16 ac97_extstat; - - if (s->no_vra) { - /* calc SRC factor - */ - adc->src_factor = ((96000 / rate) + 1) >> 1; - adc->sample_rate = 48000 / adc->src_factor; - return; - } - - adc->src_factor = 1; - - ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS); - - rate = rate > 48000 ? 48000 : rate; - - /* enable VRA - */ - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ac97_extstat | AC97_EXTSTAT_VRA); - - /* now write the sample rate - */ - wrcodec(s->codec, AC97_PCM_LR_ADC_RATE, (u16) rate); - - /* read it back for actual supported rate - */ - adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE); - - pr_debug("set_adc_rate: set to %d Hz\n", adc_rate); - - /* some codec's don't allow unequal DAC and ADC rates, in which case - * writing one rate reg actually changes both. - */ - dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE); - if (dac->num_channels > 2) - wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, dac_rate); - if (dac->num_channels > 4) - wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, dac_rate); - - adc->sample_rate = adc_rate; - dac->sample_rate = dac_rate; -} - -/* stop the DAC before calling */ -static void -set_dac_rate(struct au1550_state *s, unsigned rate) -{ - struct dmabuf *dac = &s->dma_dac; - struct dmabuf *adc = &s->dma_adc; - unsigned adc_rate, dac_rate; - u16 ac97_extstat; - - if (s->no_vra) { - /* calc SRC factor - */ - dac->src_factor = ((96000 / rate) + 1) >> 1; - dac->sample_rate = 48000 / dac->src_factor; - return; - } - - dac->src_factor = 1; - - ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS); - - rate = rate > 48000 ? 48000 : rate; - - /* enable VRA - */ - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ac97_extstat | AC97_EXTSTAT_VRA); - - /* now write the sample rate - */ - wrcodec(s->codec, AC97_PCM_FRONT_DAC_RATE, (u16) rate); - - /* I don't support different sample rates for multichannel, - * so make these channels the same. - */ - if (dac->num_channels > 2) - wrcodec(s->codec, AC97_PCM_SURR_DAC_RATE, (u16) rate); - if (dac->num_channels > 4) - wrcodec(s->codec, AC97_PCM_LFE_DAC_RATE, (u16) rate); - /* read it back for actual supported rate - */ - dac_rate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE); - - pr_debug("set_dac_rate: set to %d Hz\n", dac_rate); - - /* some codec's don't allow unequal DAC and ADC rates, in which case - * writing one rate reg actually changes both. - */ - adc_rate = rdcodec(s->codec, AC97_PCM_LR_ADC_RATE); - - dac->sample_rate = dac_rate; - adc->sample_rate = adc_rate; -} - -static void -stop_dac(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_dac; - u32 stat; - unsigned long flags; - - if (db->stopped) - return; - - spin_lock_irqsave(&s->lock, flags); - - au_writel(PSC_AC97PCR_TP, PSC_AC97PCR); - au_sync(); - - /* Wait for Transmit Busy to show disabled. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_TB) != 0); - - au1xxx_dbdma_reset(db->dmanr); - - db->stopped = 1; - - spin_unlock_irqrestore(&s->lock, flags); -} - -static void -stop_adc(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_adc; - unsigned long flags; - u32 stat; - - if (db->stopped) - return; - - spin_lock_irqsave(&s->lock, flags); - - au_writel(PSC_AC97PCR_RP, PSC_AC97PCR); - au_sync(); - - /* Wait for Receive Busy to show disabled. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_RB) != 0); - - au1xxx_dbdma_reset(db->dmanr); - - db->stopped = 1; - - spin_unlock_irqrestore(&s->lock, flags); -} - - -static void -set_xmit_slots(int num_channels) -{ - u32 ac97_config, stat; - - ac97_config = au_readl(PSC_AC97CFG); - au_sync(); - ac97_config &= ~(PSC_AC97CFG_TXSLOT_MASK | PSC_AC97CFG_DE_ENABLE); - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - switch (num_channels) { - case 6: /* stereo with surround and center/LFE, - * slots 3,4,6,7,8,9 - */ - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(6); - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(9); - - case 4: /* stereo with surround, slots 3,4,7,8 */ - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(7); - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(8); - - case 2: /* stereo, slots 3,4 */ - case 1: /* mono */ - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(3); - ac97_config |= PSC_AC97CFG_TXSLOT_ENA(4); - } - - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - ac97_config |= PSC_AC97CFG_DE_ENABLE; - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - /* Wait for Device ready. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_DR) == 0); -} - -static void -set_recv_slots(int num_channels) -{ - u32 ac97_config, stat; - - ac97_config = au_readl(PSC_AC97CFG); - au_sync(); - ac97_config &= ~(PSC_AC97CFG_RXSLOT_MASK | PSC_AC97CFG_DE_ENABLE); - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - /* Always enable slots 3 and 4 (stereo). Slot 6 is - * optional Mic ADC, which we don't support yet. - */ - ac97_config |= PSC_AC97CFG_RXSLOT_ENA(3); - ac97_config |= PSC_AC97CFG_RXSLOT_ENA(4); - - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - ac97_config |= PSC_AC97CFG_DE_ENABLE; - au_writel(ac97_config, PSC_AC97CFG); - au_sync(); - - /* Wait for Device ready. - */ - do { - stat = au_readl(PSC_AC97STAT); - au_sync(); - } while ((stat & PSC_AC97STAT_DR) == 0); -} - -/* Hold spinlock for both start_dac() and start_adc() calls */ -static void -start_dac(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_dac; - - if (!db->stopped) - return; - - set_xmit_slots(db->num_channels); - au_writel(PSC_AC97PCR_TC, PSC_AC97PCR); - au_sync(); - au_writel(PSC_AC97PCR_TS, PSC_AC97PCR); - au_sync(); - - au1xxx_dbdma_start(db->dmanr); - - db->stopped = 0; -} - -static void -start_adc(struct au1550_state *s) -{ - struct dmabuf *db = &s->dma_adc; - int i; - - if (!db->stopped) - return; - - /* Put two buffers on the ring to get things started. - */ - for (i=0; i<2; i++) { - au1xxx_dbdma_put_dest(db->dmanr, virt_to_phys(db->nextIn), - db->dma_fragsize, DDMA_FLAGS_IE); - - db->nextIn += db->dma_fragsize; - if (db->nextIn >= db->rawbuf + db->dmasize) - db->nextIn -= db->dmasize; - } - - set_recv_slots(db->num_channels); - au1xxx_dbdma_start(db->dmanr); - au_writel(PSC_AC97PCR_RC, PSC_AC97PCR); - au_sync(); - au_writel(PSC_AC97PCR_RS, PSC_AC97PCR); - au_sync(); - - db->stopped = 0; -} - -static int -prog_dmabuf(struct au1550_state *s, struct dmabuf *db) -{ - unsigned user_bytes_per_sec; - unsigned bufs; - unsigned rate = db->sample_rate; - - if (!db->rawbuf) { - db->ready = db->mapped = 0; - db->buforder = 5; /* 32 * PAGE_SIZE */ - db->rawbuf = kmalloc((PAGE_SIZE << db->buforder), GFP_KERNEL); - if (!db->rawbuf) - return -ENOMEM; - } - - db->cnt_factor = 1; - if (db->sample_size == 8) - db->cnt_factor *= 2; - if (db->num_channels == 1) - db->cnt_factor *= 2; - db->cnt_factor *= db->src_factor; - - db->count = 0; - db->dma_qcount = 0; - db->nextIn = db->nextOut = db->rawbuf; - - db->user_bytes_per_sample = (db->sample_size>>3) * db->num_channels; - db->dma_bytes_per_sample = 2 * ((db->num_channels == 1) ? - 2 : db->num_channels); - - user_bytes_per_sec = rate * db->user_bytes_per_sample; - bufs = PAGE_SIZE << db->buforder; - if (db->ossfragshift) { - if ((1000 << db->ossfragshift) < user_bytes_per_sec) - db->fragshift = ld2(user_bytes_per_sec/1000); - else - db->fragshift = db->ossfragshift; - } else { - db->fragshift = ld2(user_bytes_per_sec / 100 / - (db->subdivision ? db->subdivision : 1)); - if (db->fragshift < 3) - db->fragshift = 3; - } - - db->fragsize = 1 << db->fragshift; - db->dma_fragsize = db->fragsize * db->cnt_factor; - db->numfrag = bufs / db->dma_fragsize; - - while (db->numfrag < 4 && db->fragshift > 3) { - db->fragshift--; - db->fragsize = 1 << db->fragshift; - db->dma_fragsize = db->fragsize * db->cnt_factor; - db->numfrag = bufs / db->dma_fragsize; - } - - if (db->ossmaxfrags >= 4 && db->ossmaxfrags < db->numfrag) - db->numfrag = db->ossmaxfrags; - - db->dmasize = db->dma_fragsize * db->numfrag; - memset(db->rawbuf, 0, bufs); - - pr_debug("prog_dmabuf: rate=%d, samplesize=%d, channels=%d\n", - rate, db->sample_size, db->num_channels); - pr_debug("prog_dmabuf: fragsize=%d, cnt_factor=%d, dma_fragsize=%d\n", - db->fragsize, db->cnt_factor, db->dma_fragsize); - pr_debug("prog_dmabuf: numfrag=%d, dmasize=%d\n", db->numfrag, db->dmasize); - - db->ready = 1; - return 0; -} - -static int -prog_dmabuf_adc(struct au1550_state *s) -{ - stop_adc(s); - return prog_dmabuf(s, &s->dma_adc); - -} - -static int -prog_dmabuf_dac(struct au1550_state *s) -{ - stop_dac(s); - return prog_dmabuf(s, &s->dma_dac); -} - - -static void dac_dma_interrupt(int irq, void *dev_id) -{ - struct au1550_state *s = (struct au1550_state *) dev_id; - struct dmabuf *db = &s->dma_dac; - u32 ac97c_stat; - - spin_lock(&s->lock); - - ac97c_stat = au_readl(PSC_AC97STAT); - if (ac97c_stat & (AC97C_XU | AC97C_XO | AC97C_TE)) - pr_debug("AC97C status = 0x%08x\n", ac97c_stat); - db->dma_qcount--; - - if (db->count >= db->fragsize) { - if (au1xxx_dbdma_put_source(db->dmanr, - virt_to_phys(db->nextOut), db->fragsize, - DDMA_FLAGS_IE) == 0) { - err("qcount < 2 and no ring room!"); - } - db->nextOut += db->fragsize; - if (db->nextOut >= db->rawbuf + db->dmasize) - db->nextOut -= db->dmasize; - db->count -= db->fragsize; - db->total_bytes += db->dma_fragsize; - db->dma_qcount++; - } - - /* wake up anybody listening */ - if (waitqueue_active(&db->wait)) - wake_up(&db->wait); - - spin_unlock(&s->lock); -} - - -static void adc_dma_interrupt(int irq, void *dev_id) -{ - struct au1550_state *s = (struct au1550_state *)dev_id; - struct dmabuf *dp = &s->dma_adc; - u32 obytes; - char *obuf; - - spin_lock(&s->lock); - - /* Pull the buffer from the dma queue. - */ - au1xxx_dbdma_get_dest(dp->dmanr, (void *)(&obuf), &obytes); - - if ((dp->count + obytes) > dp->dmasize) { - /* Overrun. Stop ADC and log the error - */ - spin_unlock(&s->lock); - stop_adc(s); - dp->error++; - err("adc overrun"); - return; - } - - /* Put a new empty buffer on the destination DMA. - */ - au1xxx_dbdma_put_dest(dp->dmanr, virt_to_phys(dp->nextIn), - dp->dma_fragsize, DDMA_FLAGS_IE); - - dp->nextIn += dp->dma_fragsize; - if (dp->nextIn >= dp->rawbuf + dp->dmasize) - dp->nextIn -= dp->dmasize; - - dp->count += obytes; - dp->total_bytes += obytes; - - /* wake up anybody listening - */ - if (waitqueue_active(&dp->wait)) - wake_up(&dp->wait); - - spin_unlock(&s->lock); -} - -static loff_t -au1550_llseek(struct file *file, loff_t offset, int origin) -{ - return -ESPIPE; -} - - -static int -au1550_open_mixdev(struct inode *inode, struct file *file) -{ - mutex_lock(&au1550_ac97_mutex); - file->private_data = &au1550_state; - mutex_unlock(&au1550_ac97_mutex); - return 0; -} - -static int -au1550_release_mixdev(struct inode *inode, struct file *file) -{ - return 0; -} - -static int -mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, - unsigned long arg) -{ - return codec->mixer_ioctl(codec, cmd, arg); -} - -static long -au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) -{ - struct au1550_state *s = file->private_data; - struct ac97_codec *codec = s->codec; - int ret; - - mutex_lock(&au1550_ac97_mutex); - ret = mixdev_ioctl(codec, cmd, arg); - mutex_unlock(&au1550_ac97_mutex); - - return ret; -} - -static /*const */ struct file_operations au1550_mixer_fops = { - .owner = THIS_MODULE, - .llseek = au1550_llseek, - .unlocked_ioctl = au1550_ioctl_mixdev, - .open = au1550_open_mixdev, - .release = au1550_release_mixdev, -}; - -static int -drain_dac(struct au1550_state *s, int nonblock) -{ - unsigned long flags; - int count, tmo; - - if (s->dma_dac.mapped || !s->dma_dac.ready || s->dma_dac.stopped) - return 0; - - for (;;) { - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - spin_unlock_irqrestore(&s->lock, flags); - if (count <= s->dma_dac.fragsize) - break; - if (signal_pending(current)) - break; - if (nonblock) - return -EBUSY; - tmo = 1000 * count / (s->no_vra ? - 48000 : s->dma_dac.sample_rate); - tmo /= s->dma_dac.dma_bytes_per_sample; - au1550_delay(tmo); - } - if (signal_pending(current)) - return -ERESTARTSYS; - return 0; -} - -static inline u8 S16_TO_U8(s16 ch) -{ - return (u8) (ch >> 8) + 0x80; -} -static inline s16 U8_TO_S16(u8 ch) -{ - return (s16) (ch - 0x80) << 8; -} - -/* - * Translates user samples to dma buffer suitable for AC'97 DAC data: - * If mono, copy left channel to right channel in dma buffer. - * If 8 bit samples, cvt to 16-bit before writing to dma buffer. - * If interpolating (no VRA), duplicate every audio frame src_factor times. - */ -static int -translate_from_user(struct dmabuf *db, char* dmabuf, char* userbuf, - int dmacount) -{ - int sample, i; - int interp_bytes_per_sample; - int num_samples; - int mono = (db->num_channels == 1); - char usersample[12]; - s16 ch, dmasample[6]; - - if (db->sample_size == 16 && !mono && db->src_factor == 1) { - /* no translation necessary, just copy - */ - if (copy_from_user(dmabuf, userbuf, dmacount)) - return -EFAULT; - return dmacount; - } - - interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor; - num_samples = dmacount / interp_bytes_per_sample; - - for (sample = 0; sample < num_samples; sample++) { - if (copy_from_user(usersample, userbuf, - db->user_bytes_per_sample)) { - return -EFAULT; - } - - for (i = 0; i < db->num_channels; i++) { - if (db->sample_size == 8) - ch = U8_TO_S16(usersample[i]); - else - ch = *((s16 *) (&usersample[i * 2])); - dmasample[i] = ch; - if (mono) - dmasample[i + 1] = ch; /* right channel */ - } - - /* duplicate every audio frame src_factor times - */ - for (i = 0; i < db->src_factor; i++) - memcpy(dmabuf, dmasample, db->dma_bytes_per_sample); - - userbuf += db->user_bytes_per_sample; - dmabuf += interp_bytes_per_sample; - } - - return num_samples * interp_bytes_per_sample; -} - -/* - * Translates AC'97 ADC samples to user buffer: - * If mono, send only left channel to user buffer. - * If 8 bit samples, cvt from 16 to 8 bit before writing to user buffer. - * If decimating (no VRA), skip over src_factor audio frames. - */ -static int -translate_to_user(struct dmabuf *db, char* userbuf, char* dmabuf, - int dmacount) -{ - int sample, i; - int interp_bytes_per_sample; - int num_samples; - int mono = (db->num_channels == 1); - char usersample[12]; - - if (db->sample_size == 16 && !mono && db->src_factor == 1) { - /* no translation necessary, just copy - */ - if (copy_to_user(userbuf, dmabuf, dmacount)) - return -EFAULT; - return dmacount; - } - - interp_bytes_per_sample = db->dma_bytes_per_sample * db->src_factor; - num_samples = dmacount / interp_bytes_per_sample; - - for (sample = 0; sample < num_samples; sample++) { - for (i = 0; i < db->num_channels; i++) { - if (db->sample_size == 8) - usersample[i] = - S16_TO_U8(*((s16 *) (&dmabuf[i * 2]))); - else - *((s16 *) (&usersample[i * 2])) = - *((s16 *) (&dmabuf[i * 2])); - } - - if (copy_to_user(userbuf, usersample, - db->user_bytes_per_sample)) { - return -EFAULT; - } - - userbuf += db->user_bytes_per_sample; - dmabuf += interp_bytes_per_sample; - } - - return num_samples * interp_bytes_per_sample; -} - -/* - * Copy audio data to/from user buffer from/to dma buffer, taking care - * that we wrap when reading/writing the dma buffer. Returns actual byte - * count written to or read from the dma buffer. - */ -static int -copy_dmabuf_user(struct dmabuf *db, char* userbuf, int count, int to_user) -{ - char *bufptr = to_user ? db->nextOut : db->nextIn; - char *bufend = db->rawbuf + db->dmasize; - int cnt, ret; - - if (bufptr + count > bufend) { - int partial = (int) (bufend - bufptr); - if (to_user) { - if ((cnt = translate_to_user(db, userbuf, - bufptr, partial)) < 0) - return cnt; - ret = cnt; - if ((cnt = translate_to_user(db, userbuf + partial, - db->rawbuf, - count - partial)) < 0) - return cnt; - ret += cnt; - } else { - if ((cnt = translate_from_user(db, bufptr, userbuf, - partial)) < 0) - return cnt; - ret = cnt; - if ((cnt = translate_from_user(db, db->rawbuf, - userbuf + partial, - count - partial)) < 0) - return cnt; - ret += cnt; - } - } else { - if (to_user) - ret = translate_to_user(db, userbuf, bufptr, count); - else - ret = translate_from_user(db, bufptr, userbuf, count); - } - - return ret; -} - - -static ssize_t -au1550_read(struct file *file, char *buffer, size_t count, loff_t *ppos) -{ - struct au1550_state *s = file->private_data; - struct dmabuf *db = &s->dma_adc; - DECLARE_WAITQUEUE(wait, current); - ssize_t ret; - unsigned long flags; - int cnt, usercnt, avail; - - if (db->mapped) - return -ENXIO; - if (!access_ok(VERIFY_WRITE, buffer, count)) - return -EFAULT; - ret = 0; - - count *= db->cnt_factor; - - mutex_lock(&s->sem); - add_wait_queue(&db->wait, &wait); - - while (count > 0) { - /* wait for samples in ADC dma buffer - */ - do { - spin_lock_irqsave(&s->lock, flags); - if (db->stopped) - start_adc(s); - avail = db->count; - if (avail <= 0) - __set_current_state(TASK_INTERRUPTIBLE); - spin_unlock_irqrestore(&s->lock, flags); - if (avail <= 0) { - if (file->f_flags & O_NONBLOCK) { - if (!ret) - ret = -EAGAIN; - goto out; - } - mutex_unlock(&s->sem); - schedule(); - if (signal_pending(current)) { - if (!ret) - ret = -ERESTARTSYS; - goto out2; - } - mutex_lock(&s->sem); - } - } while (avail <= 0); - - /* copy from nextOut to user - */ - if ((cnt = copy_dmabuf_user(db, buffer, - count > avail ? - avail : count, 1)) < 0) { - if (!ret) - ret = -EFAULT; - goto out; - } - - spin_lock_irqsave(&s->lock, flags); - db->count -= cnt; - db->nextOut += cnt; - if (db->nextOut >= db->rawbuf + db->dmasize) - db->nextOut -= db->dmasize; - spin_unlock_irqrestore(&s->lock, flags); - - count -= cnt; - usercnt = cnt / db->cnt_factor; - buffer += usercnt; - ret += usercnt; - } /* while (count > 0) */ - -out: - mutex_unlock(&s->sem); -out2: - remove_wait_queue(&db->wait, &wait); - set_current_state(TASK_RUNNING); - return ret; -} - -static ssize_t -au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos) -{ - struct au1550_state *s = file->private_data; - struct dmabuf *db = &s->dma_dac; - DECLARE_WAITQUEUE(wait, current); - ssize_t ret = 0; - unsigned long flags; - int cnt, usercnt, avail; - - pr_debug("write: count=%d\n", count); - - if (db->mapped) - return -ENXIO; - if (!access_ok(VERIFY_READ, buffer, count)) - return -EFAULT; - - count *= db->cnt_factor; - - mutex_lock(&s->sem); - add_wait_queue(&db->wait, &wait); - - while (count > 0) { - /* wait for space in playback buffer - */ - do { - spin_lock_irqsave(&s->lock, flags); - avail = (int) db->dmasize - db->count; - if (avail <= 0) - __set_current_state(TASK_INTERRUPTIBLE); - spin_unlock_irqrestore(&s->lock, flags); - if (avail <= 0) { - if (file->f_flags & O_NONBLOCK) { - if (!ret) - ret = -EAGAIN; - goto out; - } - mutex_unlock(&s->sem); - schedule(); - if (signal_pending(current)) { - if (!ret) - ret = -ERESTARTSYS; - goto out2; - } - mutex_lock(&s->sem); - } - } while (avail <= 0); - - /* copy from user to nextIn - */ - if ((cnt = copy_dmabuf_user(db, (char *) buffer, - count > avail ? - avail : count, 0)) < 0) { - if (!ret) - ret = -EFAULT; - goto out; - } - - spin_lock_irqsave(&s->lock, flags); - db->count += cnt; - db->nextIn += cnt; - if (db->nextIn >= db->rawbuf + db->dmasize) - db->nextIn -= db->dmasize; - - /* If the data is available, we want to keep two buffers - * on the dma queue. If the queue count reaches zero, - * we know the dma has stopped. - */ - while ((db->dma_qcount < 2) && (db->count >= db->fragsize)) { - if (au1xxx_dbdma_put_source(db->dmanr, - virt_to_phys(db->nextOut), db->fragsize, - DDMA_FLAGS_IE) == 0) { - err("qcount < 2 and no ring room!"); - } - db->nextOut += db->fragsize; - if (db->nextOut >= db->rawbuf + db->dmasize) - db->nextOut -= db->dmasize; - db->total_bytes += db->dma_fragsize; - if (db->dma_qcount == 0) - start_dac(s); - db->dma_qcount++; - } - spin_unlock_irqrestore(&s->lock, flags); - - count -= cnt; - usercnt = cnt / db->cnt_factor; - buffer += usercnt; - ret += usercnt; - } /* while (count > 0) */ - -out: - mutex_unlock(&s->sem); -out2: - remove_wait_queue(&db->wait, &wait); - set_current_state(TASK_RUNNING); - return ret; -} - - -/* No kernel lock - we have our own spinlock */ -static unsigned int -au1550_poll(struct file *file, struct poll_table_struct *wait) -{ - struct au1550_state *s = file->private_data; - unsigned long flags; - unsigned int mask = 0; - - if (file->f_mode & FMODE_WRITE) { - if (!s->dma_dac.ready) - return 0; - poll_wait(file, &s->dma_dac.wait, wait); - } - if (file->f_mode & FMODE_READ) { - if (!s->dma_adc.ready) - return 0; - poll_wait(file, &s->dma_adc.wait, wait); - } - - spin_lock_irqsave(&s->lock, flags); - - if (file->f_mode & FMODE_READ) { - if (s->dma_adc.count >= (signed)s->dma_adc.dma_fragsize) - mask |= POLLIN | POLLRDNORM; - } - if (file->f_mode & FMODE_WRITE) { - if (s->dma_dac.mapped) { - if (s->dma_dac.count >= - (signed)s->dma_dac.dma_fragsize) - mask |= POLLOUT | POLLWRNORM; - } else { - if ((signed) s->dma_dac.dmasize >= - s->dma_dac.count + (signed)s->dma_dac.dma_fragsize) - mask |= POLLOUT | POLLWRNORM; - } - } - spin_unlock_irqrestore(&s->lock, flags); - return mask; -} - -static int -au1550_mmap(struct file *file, struct vm_area_struct *vma) -{ - struct au1550_state *s = file->private_data; - struct dmabuf *db; - unsigned long size; - int ret = 0; - - mutex_lock(&au1550_ac97_mutex); - mutex_lock(&s->sem); - if (vma->vm_flags & VM_WRITE) - db = &s->dma_dac; - else if (vma->vm_flags & VM_READ) - db = &s->dma_adc; - else { - ret = -EINVAL; - goto out; - } - if (vma->vm_pgoff != 0) { - ret = -EINVAL; - goto out; - } - size = vma->vm_end - vma->vm_start; - if (size > (PAGE_SIZE << db->buforder)) { - ret = -EINVAL; - goto out; - } - if (remap_pfn_range(vma, vma->vm_start, page_to_pfn(virt_to_page(db->rawbuf)), - size, vma->vm_page_prot)) { - ret = -EAGAIN; - goto out; - } - vma->vm_flags &= ~VM_IO; - db->mapped = 1; -out: - mutex_unlock(&s->sem); - mutex_unlock(&au1550_ac97_mutex); - return ret; -} - -#ifdef DEBUG -static struct ioctl_str_t { - unsigned int cmd; - const char *str; -} ioctl_str[] = { - {SNDCTL_DSP_RESET, "SNDCTL_DSP_RESET"}, - {SNDCTL_DSP_SYNC, "SNDCTL_DSP_SYNC"}, - {SNDCTL_DSP_SPEED, "SNDCTL_DSP_SPEED"}, - {SNDCTL_DSP_STEREO, "SNDCTL_DSP_STEREO"}, - {SNDCTL_DSP_GETBLKSIZE, "SNDCTL_DSP_GETBLKSIZE"}, - {SNDCTL_DSP_SAMPLESIZE, "SNDCTL_DSP_SAMPLESIZE"}, - {SNDCTL_DSP_CHANNELS, "SNDCTL_DSP_CHANNELS"}, - {SOUND_PCM_WRITE_CHANNELS, "SOUND_PCM_WRITE_CHANNELS"}, - {SOUND_PCM_WRITE_FILTER, "SOUND_PCM_WRITE_FILTER"}, - {SNDCTL_DSP_POST, "SNDCTL_DSP_POST"}, - {SNDCTL_DSP_SUBDIVIDE, "SNDCTL_DSP_SUBDIVIDE"}, - {SNDCTL_DSP_SETFRAGMENT, "SNDCTL_DSP_SETFRAGMENT"}, - {SNDCTL_DSP_GETFMTS, "SNDCTL_DSP_GETFMTS"}, - {SNDCTL_DSP_SETFMT, "SNDCTL_DSP_SETFMT"}, - {SNDCTL_DSP_GETOSPACE, "SNDCTL_DSP_GETOSPACE"}, - {SNDCTL_DSP_GETISPACE, "SNDCTL_DSP_GETISPACE"}, - {SNDCTL_DSP_NONBLOCK, "SNDCTL_DSP_NONBLOCK"}, - {SNDCTL_DSP_GETCAPS, "SNDCTL_DSP_GETCAPS"}, - {SNDCTL_DSP_GETTRIGGER, "SNDCTL_DSP_GETTRIGGER"}, - {SNDCTL_DSP_SETTRIGGER, "SNDCTL_DSP_SETTRIGGER"}, - {SNDCTL_DSP_GETIPTR, "SNDCTL_DSP_GETIPTR"}, - {SNDCTL_DSP_GETOPTR, "SNDCTL_DSP_GETOPTR"}, - {SNDCTL_DSP_MAPINBUF, "SNDCTL_DSP_MAPINBUF"}, - {SNDCTL_DSP_MAPOUTBUF, "SNDCTL_DSP_MAPOUTBUF"}, - {SNDCTL_DSP_SETSYNCRO, "SNDCTL_DSP_SETSYNCRO"}, - {SNDCTL_DSP_SETDUPLEX, "SNDCTL_DSP_SETDUPLEX"}, - {SNDCTL_DSP_GETODELAY, "SNDCTL_DSP_GETODELAY"}, - {SNDCTL_DSP_GETCHANNELMASK, "SNDCTL_DSP_GETCHANNELMASK"}, - {SNDCTL_DSP_BIND_CHANNEL, "SNDCTL_DSP_BIND_CHANNEL"}, - {OSS_GETVERSION, "OSS_GETVERSION"}, - {SOUND_PCM_READ_RATE, "SOUND_PCM_READ_RATE"}, - {SOUND_PCM_READ_CHANNELS, "SOUND_PCM_READ_CHANNELS"}, - {SOUND_PCM_READ_BITS, "SOUND_PCM_READ_BITS"}, - {SOUND_PCM_READ_FILTER, "SOUND_PCM_READ_FILTER"} -}; -#endif - -static int -dma_count_done(struct dmabuf *db) -{ - if (db->stopped) - return 0; - - return db->dma_fragsize - au1xxx_get_dma_residue(db->dmanr); -} - - -static int -au1550_ioctl(struct file *file, unsigned int cmd, unsigned long arg) -{ - struct au1550_state *s = file->private_data; - unsigned long flags; - audio_buf_info abinfo; - count_info cinfo; - int count; - int val, mapped, ret, diff; - - mapped = ((file->f_mode & FMODE_WRITE) && s->dma_dac.mapped) || - ((file->f_mode & FMODE_READ) && s->dma_adc.mapped); - -#ifdef DEBUG - for (count = 0; count < ARRAY_SIZE(ioctl_str); count++) { - if (ioctl_str[count].cmd == cmd) - break; - } - if (count < ARRAY_SIZE(ioctl_str)) - pr_debug("ioctl %s, arg=0x%lxn", ioctl_str[count].str, arg); - else - pr_debug("ioctl 0x%x unknown, arg=0x%lx\n", cmd, arg); -#endif - - switch (cmd) { - case OSS_GETVERSION: - return put_user(SOUND_VERSION, (int *) arg); - - case SNDCTL_DSP_SYNC: - if (file->f_mode & FMODE_WRITE) - return drain_dac(s, file->f_flags & O_NONBLOCK); - return 0; - - case SNDCTL_DSP_SETDUPLEX: - return 0; - - case SNDCTL_DSP_GETCAPS: - return put_user(DSP_CAP_DUPLEX | DSP_CAP_REALTIME | - DSP_CAP_TRIGGER | DSP_CAP_MMAP, (int *)arg); - - case SNDCTL_DSP_RESET: - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - synchronize_irq(); - s->dma_dac.count = s->dma_dac.total_bytes = 0; - s->dma_dac.nextIn = s->dma_dac.nextOut = - s->dma_dac.rawbuf; - } - if (file->f_mode & FMODE_READ) { - stop_adc(s); - synchronize_irq(); - s->dma_adc.count = s->dma_adc.total_bytes = 0; - s->dma_adc.nextIn = s->dma_adc.nextOut = - s->dma_adc.rawbuf; - } - return 0; - - case SNDCTL_DSP_SPEED: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val >= 0) { - if (file->f_mode & FMODE_READ) { - stop_adc(s); - set_adc_rate(s, val); - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - set_dac_rate(s, val); - } - if (s->open_mode & FMODE_READ) - if ((ret = prog_dmabuf_adc(s))) - return ret; - if (s->open_mode & FMODE_WRITE) - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return put_user((file->f_mode & FMODE_READ) ? - s->dma_adc.sample_rate : - s->dma_dac.sample_rate, - (int *)arg); - - case SNDCTL_DSP_STEREO: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - s->dma_adc.num_channels = val ? 2 : 1; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - s->dma_dac.num_channels = val ? 2 : 1; - if (s->codec_ext_caps & AC97_EXT_DACS) { - /* disable surround and center/lfe in AC'97 - */ - u16 ext_stat = rdcodec(s->codec, - AC97_EXTENDED_STATUS); - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ext_stat | (AC97_EXTSTAT_PRI | - AC97_EXTSTAT_PRJ | - AC97_EXTSTAT_PRK)); - } - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return 0; - - case SNDCTL_DSP_CHANNELS: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val != 0) { - if (file->f_mode & FMODE_READ) { - if (val < 0 || val > 2) - return -EINVAL; - stop_adc(s); - s->dma_adc.num_channels = val; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - switch (val) { - case 1: - case 2: - break; - case 3: - case 5: - return -EINVAL; - case 4: - if (!(s->codec_ext_caps & - AC97_EXTID_SDAC)) - return -EINVAL; - break; - case 6: - if ((s->codec_ext_caps & - AC97_EXT_DACS) != AC97_EXT_DACS) - return -EINVAL; - break; - default: - return -EINVAL; - } - - stop_dac(s); - if (val <= 2 && - (s->codec_ext_caps & AC97_EXT_DACS)) { - /* disable surround and center/lfe - * channels in AC'97 - */ - u16 ext_stat = - rdcodec(s->codec, - AC97_EXTENDED_STATUS); - wrcodec(s->codec, - AC97_EXTENDED_STATUS, - ext_stat | (AC97_EXTSTAT_PRI | - AC97_EXTSTAT_PRJ | - AC97_EXTSTAT_PRK)); - } else if (val >= 4) { - /* enable surround, center/lfe - * channels in AC'97 - */ - u16 ext_stat = - rdcodec(s->codec, - AC97_EXTENDED_STATUS); - ext_stat &= ~AC97_EXTSTAT_PRJ; - if (val == 6) - ext_stat &= - ~(AC97_EXTSTAT_PRI | - AC97_EXTSTAT_PRK); - wrcodec(s->codec, - AC97_EXTENDED_STATUS, - ext_stat); - } - - s->dma_dac.num_channels = val; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - } - return put_user(val, (int *) arg); - - case SNDCTL_DSP_GETFMTS: /* Returns a mask */ - return put_user(AFMT_S16_LE | AFMT_U8, (int *) arg); - - case SNDCTL_DSP_SETFMT: /* Selects ONE fmt */ - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val != AFMT_QUERY) { - if (file->f_mode & FMODE_READ) { - stop_adc(s); - if (val == AFMT_S16_LE) - s->dma_adc.sample_size = 16; - else { - val = AFMT_U8; - s->dma_adc.sample_size = 8; - } - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - if (val == AFMT_S16_LE) - s->dma_dac.sample_size = 16; - else { - val = AFMT_U8; - s->dma_dac.sample_size = 8; - } - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - } else { - if (file->f_mode & FMODE_READ) - val = (s->dma_adc.sample_size == 16) ? - AFMT_S16_LE : AFMT_U8; - else - val = (s->dma_dac.sample_size == 16) ? - AFMT_S16_LE : AFMT_U8; - } - return put_user(val, (int *) arg); - - case SNDCTL_DSP_POST: - return 0; - - case SNDCTL_DSP_GETTRIGGER: - val = 0; - spin_lock_irqsave(&s->lock, flags); - if (file->f_mode & FMODE_READ && !s->dma_adc.stopped) - val |= PCM_ENABLE_INPUT; - if (file->f_mode & FMODE_WRITE && !s->dma_dac.stopped) - val |= PCM_ENABLE_OUTPUT; - spin_unlock_irqrestore(&s->lock, flags); - return put_user(val, (int *) arg); - - case SNDCTL_DSP_SETTRIGGER: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - if (val & PCM_ENABLE_INPUT) { - spin_lock_irqsave(&s->lock, flags); - start_adc(s); - spin_unlock_irqrestore(&s->lock, flags); - } else - stop_adc(s); - } - if (file->f_mode & FMODE_WRITE) { - if (val & PCM_ENABLE_OUTPUT) { - spin_lock_irqsave(&s->lock, flags); - start_dac(s); - spin_unlock_irqrestore(&s->lock, flags); - } else - stop_dac(s); - } - return 0; - - case SNDCTL_DSP_GETOSPACE: - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - abinfo.fragsize = s->dma_dac.fragsize; - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - count -= dma_count_done(&s->dma_dac); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - abinfo.bytes = (s->dma_dac.dmasize - count) / - s->dma_dac.cnt_factor; - abinfo.fragstotal = s->dma_dac.numfrag; - abinfo.fragments = abinfo.bytes >> s->dma_dac.fragshift; - pr_debug("ioctl SNDCTL_DSP_GETOSPACE: bytes=%d, fragments=%d\n", abinfo.bytes, abinfo.fragments); - return copy_to_user((void *) arg, &abinfo, - sizeof(abinfo)) ? -EFAULT : 0; - - case SNDCTL_DSP_GETISPACE: - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - abinfo.fragsize = s->dma_adc.fragsize; - spin_lock_irqsave(&s->lock, flags); - count = s->dma_adc.count; - count += dma_count_done(&s->dma_adc); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - abinfo.bytes = count / s->dma_adc.cnt_factor; - abinfo.fragstotal = s->dma_adc.numfrag; - abinfo.fragments = abinfo.bytes >> s->dma_adc.fragshift; - return copy_to_user((void *) arg, &abinfo, - sizeof(abinfo)) ? -EFAULT : 0; - - case SNDCTL_DSP_NONBLOCK: - spin_lock(&file->f_lock); - file->f_flags |= O_NONBLOCK; - spin_unlock(&file->f_lock); - return 0; - - case SNDCTL_DSP_GETODELAY: - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - count -= dma_count_done(&s->dma_dac); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - count /= s->dma_dac.cnt_factor; - return put_user(count, (int *) arg); - - case SNDCTL_DSP_GETIPTR: - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - spin_lock_irqsave(&s->lock, flags); - cinfo.bytes = s->dma_adc.total_bytes; - count = s->dma_adc.count; - if (!s->dma_adc.stopped) { - diff = dma_count_done(&s->dma_adc); - count += diff; - cinfo.bytes += diff; - cinfo.ptr = virt_to_phys(s->dma_adc.nextIn) + diff - - virt_to_phys(s->dma_adc.rawbuf); - } else - cinfo.ptr = virt_to_phys(s->dma_adc.nextIn) - - virt_to_phys(s->dma_adc.rawbuf); - if (s->dma_adc.mapped) - s->dma_adc.count &= (s->dma_adc.dma_fragsize-1); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - cinfo.blocks = count >> s->dma_adc.fragshift; - return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)); - - case SNDCTL_DSP_GETOPTR: - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - spin_lock_irqsave(&s->lock, flags); - cinfo.bytes = s->dma_dac.total_bytes; - count = s->dma_dac.count; - if (!s->dma_dac.stopped) { - diff = dma_count_done(&s->dma_dac); - count -= diff; - cinfo.bytes += diff; - cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) + diff - - virt_to_phys(s->dma_dac.rawbuf); - } else - cinfo.ptr = virt_to_phys(s->dma_dac.nextOut) - - virt_to_phys(s->dma_dac.rawbuf); - if (s->dma_dac.mapped) - s->dma_dac.count &= (s->dma_dac.dma_fragsize-1); - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - cinfo.blocks = count >> s->dma_dac.fragshift; - return copy_to_user((void *) arg, &cinfo, sizeof(cinfo)); - - case SNDCTL_DSP_GETBLKSIZE: - if (file->f_mode & FMODE_WRITE) - return put_user(s->dma_dac.fragsize, (int *) arg); - else - return put_user(s->dma_adc.fragsize, (int *) arg); - - case SNDCTL_DSP_SETFRAGMENT: - if (get_user(val, (int *) arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - s->dma_adc.ossfragshift = val & 0xffff; - s->dma_adc.ossmaxfrags = (val >> 16) & 0xffff; - if (s->dma_adc.ossfragshift < 4) - s->dma_adc.ossfragshift = 4; - if (s->dma_adc.ossfragshift > 15) - s->dma_adc.ossfragshift = 15; - if (s->dma_adc.ossmaxfrags < 4) - s->dma_adc.ossmaxfrags = 4; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - s->dma_dac.ossfragshift = val & 0xffff; - s->dma_dac.ossmaxfrags = (val >> 16) & 0xffff; - if (s->dma_dac.ossfragshift < 4) - s->dma_dac.ossfragshift = 4; - if (s->dma_dac.ossfragshift > 15) - s->dma_dac.ossfragshift = 15; - if (s->dma_dac.ossmaxfrags < 4) - s->dma_dac.ossmaxfrags = 4; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return 0; - - case SNDCTL_DSP_SUBDIVIDE: - if ((file->f_mode & FMODE_READ && s->dma_adc.subdivision) || - (file->f_mode & FMODE_WRITE && s->dma_dac.subdivision)) - return -EINVAL; - if (get_user(val, (int *) arg)) - return -EFAULT; - if (val != 1 && val != 2 && val != 4) - return -EINVAL; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - s->dma_adc.subdivision = val; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - s->dma_dac.subdivision = val; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return 0; - - case SOUND_PCM_READ_RATE: - return put_user((file->f_mode & FMODE_READ) ? - s->dma_adc.sample_rate : - s->dma_dac.sample_rate, - (int *)arg); - - case SOUND_PCM_READ_CHANNELS: - if (file->f_mode & FMODE_READ) - return put_user(s->dma_adc.num_channels, (int *)arg); - else - return put_user(s->dma_dac.num_channels, (int *)arg); - - case SOUND_PCM_READ_BITS: - if (file->f_mode & FMODE_READ) - return put_user(s->dma_adc.sample_size, (int *)arg); - else - return put_user(s->dma_dac.sample_size, (int *)arg); - - case SOUND_PCM_WRITE_FILTER: - case SNDCTL_DSP_SETSYNCRO: - case SOUND_PCM_READ_FILTER: - return -EINVAL; - } - - return mixdev_ioctl(s->codec, cmd, arg); -} - -static long -au1550_unlocked_ioctl(struct file *file, unsigned int cmd, unsigned long arg) -{ - int ret; - - mutex_lock(&au1550_ac97_mutex); - ret = au1550_ioctl(file, cmd, arg); - mutex_unlock(&au1550_ac97_mutex); - - return ret; -} - -static int -au1550_open(struct inode *inode, struct file *file) -{ - int minor = MINOR(inode->i_rdev); - DECLARE_WAITQUEUE(wait, current); - struct au1550_state *s = &au1550_state; - int ret; - -#ifdef DEBUG - if (file->f_flags & O_NONBLOCK) - pr_debug("open: non-blocking\n"); - else - pr_debug("open: blocking\n"); -#endif - - file->private_data = s; - mutex_lock(&au1550_ac97_mutex); - /* wait for device to become free */ - mutex_lock(&s->open_mutex); - while (s->open_mode & file->f_mode) { - ret = -EBUSY; - if (file->f_flags & O_NONBLOCK) - goto out; - add_wait_queue(&s->open_wait, &wait); - __set_current_state(TASK_INTERRUPTIBLE); - mutex_unlock(&s->open_mutex); - schedule(); - remove_wait_queue(&s->open_wait, &wait); - set_current_state(TASK_RUNNING); - ret = -ERESTARTSYS; - if (signal_pending(current)) - goto out2; - mutex_lock(&s->open_mutex); - } - - stop_dac(s); - stop_adc(s); - - if (file->f_mode & FMODE_READ) { - s->dma_adc.ossfragshift = s->dma_adc.ossmaxfrags = - s->dma_adc.subdivision = s->dma_adc.total_bytes = 0; - s->dma_adc.num_channels = 1; - s->dma_adc.sample_size = 8; - set_adc_rate(s, 8000); - if ((minor & 0xf) == SND_DEV_DSP16) - s->dma_adc.sample_size = 16; - } - - if (file->f_mode & FMODE_WRITE) { - s->dma_dac.ossfragshift = s->dma_dac.ossmaxfrags = - s->dma_dac.subdivision = s->dma_dac.total_bytes = 0; - s->dma_dac.num_channels = 1; - s->dma_dac.sample_size = 8; - set_dac_rate(s, 8000); - if ((minor & 0xf) == SND_DEV_DSP16) - s->dma_dac.sample_size = 16; - } - - if (file->f_mode & FMODE_READ) { - if ((ret = prog_dmabuf_adc(s))) - goto out; - } - if (file->f_mode & FMODE_WRITE) { - if ((ret = prog_dmabuf_dac(s))) - goto out; - } - - s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE); - mutex_init(&s->sem); - ret = 0; -out: - mutex_unlock(&s->open_mutex); -out2: - mutex_unlock(&au1550_ac97_mutex); - return ret; -} - -static int -au1550_release(struct inode *inode, struct file *file) -{ - struct au1550_state *s = file->private_data; - - mutex_lock(&au1550_ac97_mutex); - - if (file->f_mode & FMODE_WRITE) { - mutex_unlock(&au1550_ac97_mutex); - drain_dac(s, file->f_flags & O_NONBLOCK); - mutex_lock(&au1550_ac97_mutex); - } - - mutex_lock(&s->open_mutex); - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - kfree(s->dma_dac.rawbuf); - s->dma_dac.rawbuf = NULL; - } - if (file->f_mode & FMODE_READ) { - stop_adc(s); - kfree(s->dma_adc.rawbuf); - s->dma_adc.rawbuf = NULL; - } - s->open_mode &= ((~file->f_mode) & (FMODE_READ|FMODE_WRITE)); - mutex_unlock(&s->open_mutex); - wake_up(&s->open_wait); - mutex_unlock(&au1550_ac97_mutex); - return 0; -} - -static /*const */ struct file_operations au1550_audio_fops = { - .owner = THIS_MODULE, - .llseek = au1550_llseek, - .read = au1550_read, - .write = au1550_write, - .poll = au1550_poll, - .unlocked_ioctl = au1550_unlocked_ioctl, - .mmap = au1550_mmap, - .open = au1550_open, - .release = au1550_release, -}; - -MODULE_AUTHOR("Advanced Micro Devices (AMD), dan@embeddededge.com"); -MODULE_DESCRIPTION("Au1550 AC97 Audio Driver"); -MODULE_LICENSE("GPL"); - - -static int __devinit -au1550_probe(void) -{ - struct au1550_state *s = &au1550_state; - int val; - - memset(s, 0, sizeof(struct au1550_state)); - - init_waitqueue_head(&s->dma_adc.wait); - init_waitqueue_head(&s->dma_dac.wait); - init_waitqueue_head(&s->open_wait); - mutex_init(&s->open_mutex); - spin_lock_init(&s->lock); - - s->codec = ac97_alloc_codec(); - if(s->codec == NULL) { - err("Out of memory"); - return -1; - } - s->codec->private_data = s; - s->codec->id = 0; - s->codec->codec_read = rdcodec; - s->codec->codec_write = wrcodec; - s->codec->codec_wait = waitcodec; - - if (!request_mem_region(CPHYSADDR(AC97_PSC_SEL), - 0x30, "Au1550 AC97")) { - err("AC'97 ports in use"); - } - - /* Allocate the DMA Channels - */ - if ((s->dma_dac.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_MEM_CHAN, - DBDMA_AC97_TX_CHAN, dac_dma_interrupt, (void *)s)) == 0) { - err("Can't get DAC DMA"); - goto err_dma1; - } - au1xxx_dbdma_set_devwidth(s->dma_dac.dmanr, 16); - if (au1xxx_dbdma_ring_alloc(s->dma_dac.dmanr, - NUM_DBDMA_DESCRIPTORS) == 0) { - err("Can't get DAC DMA descriptors"); - goto err_dma1; - } - - if ((s->dma_adc.dmanr = au1xxx_dbdma_chan_alloc(DBDMA_AC97_RX_CHAN, - DBDMA_MEM_CHAN, adc_dma_interrupt, (void *)s)) == 0) { - err("Can't get ADC DMA"); - goto err_dma2; - } - au1xxx_dbdma_set_devwidth(s->dma_adc.dmanr, 16); - if (au1xxx_dbdma_ring_alloc(s->dma_adc.dmanr, - NUM_DBDMA_DESCRIPTORS) == 0) { - err("Can't get ADC DMA descriptors"); - goto err_dma2; - } - - pr_info("DAC: DMA%d, ADC: DMA%d", DBDMA_AC97_TX_CHAN, DBDMA_AC97_RX_CHAN); - - /* register devices */ - - if ((s->dev_audio = register_sound_dsp(&au1550_audio_fops, -1)) < 0) - goto err_dev1; - if ((s->codec->dev_mixer = - register_sound_mixer(&au1550_mixer_fops, -1)) < 0) - goto err_dev2; - - /* The GPIO for the appropriate PSC was configured by the - * board specific start up. - * - * configure PSC for AC'97 - */ - au_writel(0, AC97_PSC_CTRL); /* Disable PSC */ - au_sync(); - au_writel((PSC_SEL_CLK_SERCLK | PSC_SEL_PS_AC97MODE), AC97_PSC_SEL); - au_sync(); - - /* cold reset the AC'97 - */ - au_writel(PSC_AC97RST_RST, PSC_AC97RST); - au_sync(); - au1550_delay(10); - au_writel(0, PSC_AC97RST); - au_sync(); - - /* need to delay around 500msec(bleech) to give - some CODECs enough time to wakeup */ - au1550_delay(500); - - /* warm reset the AC'97 to start the bitclk - */ - au_writel(PSC_AC97RST_SNC, PSC_AC97RST); - au_sync(); - udelay(100); - au_writel(0, PSC_AC97RST); - au_sync(); - - /* Enable PSC - */ - au_writel(PSC_CTRL_ENABLE, AC97_PSC_CTRL); - au_sync(); - - /* Wait for PSC ready. - */ - do { - val = au_readl(PSC_AC97STAT); - au_sync(); - } while ((val & PSC_AC97STAT_SR) == 0); - - /* Configure AC97 controller. - * Deep FIFO, 16-bit sample, DMA, make sure DMA matches fifo size. - */ - val = PSC_AC97CFG_SET_LEN(16); - val |= PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8; - - /* Enable device so we can at least - * talk over the AC-link. - */ - au_writel(val, PSC_AC97CFG); - au_writel(PSC_AC97MSK_ALLMASK, PSC_AC97MSK); - au_sync(); - val |= PSC_AC97CFG_DE_ENABLE; - au_writel(val, PSC_AC97CFG); - au_sync(); - - /* Wait for Device ready. - */ - do { - val = au_readl(PSC_AC97STAT); - au_sync(); - } while ((val & PSC_AC97STAT_DR) == 0); - - /* codec init */ - if (!ac97_probe_codec(s->codec)) - goto err_dev3; - - s->codec_base_caps = rdcodec(s->codec, AC97_RESET); - s->codec_ext_caps = rdcodec(s->codec, AC97_EXTENDED_ID); - pr_info("AC'97 Base/Extended ID = %04x/%04x", - s->codec_base_caps, s->codec_ext_caps); - - if (!(s->codec_ext_caps & AC97_EXTID_VRA)) { - /* codec does not support VRA - */ - s->no_vra = 1; - } else if (!vra) { - /* Boot option says disable VRA - */ - u16 ac97_extstat = rdcodec(s->codec, AC97_EXTENDED_STATUS); - wrcodec(s->codec, AC97_EXTENDED_STATUS, - ac97_extstat & ~AC97_EXTSTAT_VRA); - s->no_vra = 1; - } - if (s->no_vra) - pr_info("no VRA, interpolating and decimating"); - - /* set mic to be the recording source */ - val = SOUND_MASK_MIC; - mixdev_ioctl(s->codec, SOUND_MIXER_WRITE_RECSRC, - (unsigned long) &val); - - return 0; - - err_dev3: - unregister_sound_mixer(s->codec->dev_mixer); - err_dev2: - unregister_sound_dsp(s->dev_audio); - err_dev1: - au1xxx_dbdma_chan_free(s->dma_adc.dmanr); - err_dma2: - au1xxx_dbdma_chan_free(s->dma_dac.dmanr); - err_dma1: - release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30); - - ac97_release_codec(s->codec); - return -1; -} - -static void __devinit -au1550_remove(void) -{ - struct au1550_state *s = &au1550_state; - - if (!s) - return; - synchronize_irq(); - au1xxx_dbdma_chan_free(s->dma_adc.dmanr); - au1xxx_dbdma_chan_free(s->dma_dac.dmanr); - release_mem_region(CPHYSADDR(AC97_PSC_SEL), 0x30); - unregister_sound_dsp(s->dev_audio); - unregister_sound_mixer(s->codec->dev_mixer); - ac97_release_codec(s->codec); -} - -static int __init -init_au1550(void) -{ - return au1550_probe(); -} - -static void __exit -cleanup_au1550(void) -{ - au1550_remove(); -} - -module_init(init_au1550); -module_exit(cleanup_au1550); - -#ifndef MODULE - -static int __init -au1550_setup(char *options) -{ - char *this_opt; - - if (!options || !*options) - return 0; - - while ((this_opt = strsep(&options, ","))) { - if (!*this_opt) - continue; - if (!strncmp(this_opt, "vra", 3)) { - vra = 1; - } - } - - return 1; -} - -__setup("au1550_audio=", au1550_setup); - -#endif /* MODULE */ -- cgit v1.2.3 From 6ae759e889d9fd1733f4392f5083ac8c899253c5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 19 May 2011 17:44:46 -0600 Subject: ASoC: spdif-dit: Add missing MODULE_* MODULE_ALIAS is required so that the module will auto-load based on a platform_device registration in the board file. While we're at it, add some other MODULE_*. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transciever.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 4c32b54913a..6a1a7e705cd 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -21,7 +21,7 @@ #include #include -MODULE_LICENSE("GPL"); +#define DRV_NAME "spdif-dit" #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE @@ -56,7 +56,7 @@ static struct platform_driver spdif_dit_driver = { .probe = spdif_dit_probe, .remove = spdif_dit_remove, .driver = { - .name = "spdif-dit", + .name = DRV_NAME, .owner = THIS_MODULE, }, }; @@ -74,3 +74,7 @@ static void __exit dit_exit(void) module_init(dit_modinit); module_exit(dit_exit); +MODULE_AUTHOR("Steve Chen "); +MODULE_DESCRIPTION("SPDIF dummy codec driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.3 From 0dfe8da4921fdae6b9c77e320731e490d4b15a99 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 16 May 2011 14:19:27 -0600 Subject: ASoC: Tegra: Fix compile when debugfs not enabled The prototype of the inline dummy version of tegra_i2s_debug_add was not consistent with the real version. Reported-by: Rhyland-Klein Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 4f5e2c90b02..6b817e20548 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -114,7 +114,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) { } -- cgit v1.2.3 From 314c3ff47643c7c1f11317f81a80902af80795e3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 May 2011 16:27:02 +0200 Subject: Revert "ALSA: hda - Use position_fix=3 as default for AMD chipsets" This reverts commit 447ee6a7cbbfb5ae7ab8f9b8b058b4a04fe398bf. The workaround introduced by this commit seems bogus. The AMD chipsets don't provide proper position-buffer nor FIFO value required by VIACOMBO fix. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ecadb5b7d64..0c1996dd137 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2370,11 +2370,6 @@ static int __devinit check_position_fix(struct azx *chip, int fix) case AZX_DRIVER_ATI: /* Use link position directly, avoid any transfer problem. */ return POS_FIX_VIACOMBO; - case AZX_DRIVER_GENERIC: - /* AMD chipsets behave often badly, too */ - if (chip->pci->vendor == PCI_VENDOR_ID_AMD) - return POS_FIX_VIACOMBO; - break; } return POS_FIX_AUTO; -- cgit v1.2.3 From 50e3bbf9898840eead86f90a43b3625a2b2f4112 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 May 2011 16:29:09 +0200 Subject: ALSA: hda - Use LPIB for ATI/AMD chipsets as default ATI and AMD chipsets seem not providing the proper position-buffer information, and it also doesn't provide FIFO register required by VIACOMBO fix. It's better to use LPIB for these. Reported-by: David Henningsson Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0c1996dd137..43a036716d2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2367,9 +2367,16 @@ static int __devinit check_position_fix(struct azx *chip, int fix) /* Check VIA/ATI HD Audio Controller exist */ switch (chip->driver_type) { case AZX_DRIVER_VIA: - case AZX_DRIVER_ATI: /* Use link position directly, avoid any transfer problem. */ return POS_FIX_VIACOMBO; + case AZX_DRIVER_ATI: + /* ATI chipsets don't work well with position-buffer */ + return POS_FIX_LPIB; + case AZX_DRIVER_GENERIC: + /* AMD chipsets also don't work with position-buffer */ + if (chip->pci->vendor == PCI_VENDOR_ID_AMD) + return POS_FIX_LPIB; + break; } return POS_FIX_AUTO; -- cgit v1.2.3 From 2b39535b9e54888649923beaab443af212b6c0fd Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 20 May 2011 15:47:40 +0300 Subject: ASoC: core: Don't set "(null)" as a driver name Commit 22de71b ("ASoC: core - allow ASoC more flexible machine name") writes "(null)" to driver name string in struct snd_card if card->driver_name is NULL. This causes segmentation faults with some user space ALSA utilities like aplay and arecord. Fix this by using the card->name if no driver name is specified. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5968745213f..bb7cd581294 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1930,7 +1930,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), - "%s", card->driver_name); + "%s", card->driver_name ? card->driver_name : card->name); if (card->late_probe) { ret = card->late_probe(card); -- cgit v1.2.3