From a37f1b8fdc912600c24f9d0d45d7046e50a031e4 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Tue, 30 Dec 2014 11:12:35 -0800 Subject: ASoC: tegra: Add platform driver for rt5677 audio codec The driver supports NVIDIA Tegra Ryu board Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 10 ++ sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 347 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 359 insertions(+) create mode 100644 sound/soc/tegra/tegra_rt5677.c (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 31198cf7f88d..a6768f832c6f 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -128,3 +128,13 @@ config SND_SOC_TEGRA_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the MAX98090 codec, such as Venice2. + +config SND_SOC_TEGRA_RT5677 + tristate "SoC Audio support for Tegra boards using a RT5677 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5677 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5677 codec, such as Ryu. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 5ae588cd96c4..9171655ad843 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-rt5640-objs := tegra_rt5640.o +snd-soc-tegra-rt5677-objs := tegra_rt5677.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o @@ -27,6 +28,7 @@ snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c new file mode 100644 index 000000000000..e4cf978a6e3a --- /dev/null +++ b/sound/soc/tegra/tegra_rt5677.c @@ -0,0 +1,347 @@ +/* +* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * + * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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, see . + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include "../codecs/rt5677.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5677" + +struct tegra_rt5677 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; + int gpio_hp_en; + int gpio_mic_present; + int gpio_dmic_clk_en; +}; + +static int tegra_rt5677_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_card *card = rtd->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, err; + + srate = params_rate(params); + mclk = 256 * srate; + + 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; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_hp_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_hp_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static struct snd_soc_ops tegra_rt5677_ops = { + .hw_params = tegra_rt5677_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5677_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5677_hp_jack_pins = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; +static struct snd_soc_jack_gpio tegra_rt5677_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack tegra_rt5677_mic_jack; + +static struct snd_soc_jack_pin tegra_rt5677_mic_jack_pins = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio tegra_rt5677_mic_jack_gpio = { + .name = "Headset Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1 +}; + +static const struct snd_soc_dapm_widget tegra_rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", tegra_rt5677_event_hp), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic 1", NULL), + SND_SOC_DAPM_MIC("Internal Mic 2", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5677_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic 1"), + SOC_DAPM_PIN_SWITCH("Internal Mic 2"), +}; + +static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack); + snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_pins); + + if (gpio_is_valid(machine->gpio_mic_present)) { + tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; + snd_soc_jack_add_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static int tegra_rt5677_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + if (gpio_is_valid(machine->gpio_mic_present)) { + snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5677_dai = { + .name = "RT5677", + .stream_name = "RT5677 PCM", + .codec_dai_name = "rt5677-aif1", + .init = tegra_rt5677_asoc_init, + .ops = &tegra_rt5677_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5677 = { + .name = "tegra-rt5677", + .owner = THIS_MODULE, + .remove = tegra_rt5677_card_remove, + .dai_link = &tegra_rt5677_dai, + .num_links = 1, + .controls = tegra_rt5677_controls, + .num_controls = ARRAY_SIZE(tegra_rt5677_controls), + .dapm_widgets = tegra_rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5677_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5677_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5677; + struct tegra_rt5677 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5677), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_mic_present = of_get_named_gpio(np, + "nvidia,mic-present-gpios", 0); + if (machine->gpio_mic_present == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_hp_en = of_get_named_gpio(np, "nvidia,hp-en-gpios", 0); + if (machine->gpio_hp_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_en, + GPIOF_OUT_INIT_LOW, "hp_en"); + if (ret) { + dev_err(card->dev, "cannot get hp_en gpio\n"); + return ret; + } + } + + machine->gpio_dmic_clk_en = of_get_named_gpio(np, + "nvidia,dmic-clk-en-gpios", 0); + if (machine->gpio_dmic_clk_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_dmic_clk_en)) { + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_dmic_clk_en, + GPIOF_OUT_INIT_HIGH, "dmic_clk_en"); + if (ret) { + dev_err(card->dev, "cannot get dmic_clk_en gpio\n"); + return ret; + } + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_rt5677_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5677_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5677_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5677_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + 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(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5677_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5677_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5677", }, + {}, +}; + +static struct platform_driver tegra_rt5677_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5677_of_match, + }, + .probe = tegra_rt5677_probe, + .remove = tegra_rt5677_remove, +}; +module_platform_driver(tegra_rt5677_driver); + +MODULE_AUTHOR("Anatol Pomozov "); +MODULE_DESCRIPTION("Tegra+RT5677 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5677_of_match); -- cgit v1.2.3 From bec78c5f4ae228c4cbd432e97cadb8827fd8f1f9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 12 Jan 2015 10:27:17 +0100 Subject: ASoC: mc13783: Update set_tdm_slot() semantics The mc13783 driver uses inverted semantics for the tx_mask and rx_mask parameter of the set_tdm_slot() callback compared to rest of ASoC. This patch updates the driver's semantics to be consistent with the rest of ASoC, i.e. a set bit means a active slot and a cleared bit means a inactive slot. This will allow us to use the set_tdm_slot() API in a more generic way. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 10 +++++----- sound/soc/fsl/imx-mc13783.c | 3 +-- 2 files changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index c1e441c2c8af..2ffb9a0570dc 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -328,16 +328,16 @@ static int mc13783_set_tdm_slot_dac(struct snd_soc_dai *dai, } switch (rx_mask) { - case 0xfffffffc: + case 0x03: val |= SSI_NETWORK_DAC_RXSLOT_0_1; break; - case 0xfffffff3: + case 0x0c: val |= SSI_NETWORK_DAC_RXSLOT_2_3; break; - case 0xffffffcf: + case 0x30: val |= SSI_NETWORK_DAC_RXSLOT_4_5; break; - case 0xffffff3f: + case 0xc0: val |= SSI_NETWORK_DAC_RXSLOT_6_7; break; default: @@ -360,7 +360,7 @@ static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, if (slots != 4) return -EINVAL; - if (tx_mask != 0xfffffffc) + if (tx_mask != 0x3) return -EINVAL; val |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 6bf5bce01a92..9589452e995e 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -37,8 +37,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xfffffffc, 0xfffffffc, - 4, 16); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 4, 16); if (ret) return ret; -- cgit v1.2.3 From d0077aaf2206f3c3524d71a9f38b408dca63852f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 12 Jan 2015 10:27:18 +0100 Subject: ASoC: fsl: Update set_tdm_slot() semantics The fsl-ssi and imx-ssi drivers use inverted semantics for the tx_mask and rx_mask parameter of the set_tdm_slot() callback compared to rest of ASoC. This patch updates the driver's semantics to be consistent with the rest of ASoC, i.e. a set bit means a active slot and a cleared bit means a inactive slot. This will allow us to use the set_tdm_slot() API in a more generic way. Signed-off-by: Lars-Peter Clausen Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/fsl_utils.c | 6 +++--- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-ssi.c | 4 ++-- sound/soc/fsl/wm1133-ev1.c | 4 ++-- 6 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9ce70fc67b09..0d0203b34d8b 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -69,7 +69,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a65f17d57ffb..8841e59a9869 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -992,8 +992,8 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN); - regmap_write(regs, CCSR_SSI_STMSK, tx_mask); - regmap_write(regs, CCSR_SSI_SRMSK, rx_mask); + regmap_write(regs, CCSR_SSI_STMSK, ~tx_mask); + regmap_write(regs, CCSR_SSI_SRMSK, ~rx_mask); regmap_update_bits(regs, CCSR_SSI_SCR, CCSR_SSI_SCR_SSIEN, val); diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 2ac7755da876..5fd4463dbf05 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -94,7 +94,7 @@ EXPORT_SYMBOL(fsl_asoc_get_dma_channel); * @rx_mask: bitmask representing active RX slots. * * This function used to generate the TDM slot TX/RX mask. And the TX/RX - * mask will use a 0 bit for an active slot as default, and the default + * mask will use a 1 bit for an active slot as default, and the default * active bits are at the LSB of the mask value. */ int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots, @@ -105,9 +105,9 @@ int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots, return -EINVAL; if (tx_mask) - *tx_mask = ~((1 << slots) - 1); + *tx_mask = ((1 << slots) - 1); if (rx_mask) - *rx_mask = ~((1 << slots) - 1); + *rx_mask = ((1 << slots) - 1); return 0; } diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 9589452e995e..9e6493d4e7ff 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -45,7 +45,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 2, 16); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16); if (ret) return ret; diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index fa801e17c51e..6aeaac33871a 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -74,8 +74,8 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, sccr |= SSI_STCCR_DC(slots - 1); writel(sccr, ssi->base + SSI_SRCCR); - writel(tx_mask, ssi->base + SSI_STMSK); - writel(rx_mask, ssi->base + SSI_SRMSK); + writel(~tx_mask, ssi->base + SSI_STMSK); + writel(~rx_mask, ssi->base + SSI_SRMSK); return 0; } diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 804749a6c61e..ca7b774b13ee 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -116,10 +116,10 @@ static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream, /* TODO: The SSI driver should figure this out for us */ switch (channels) { case 2: - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0); break; case 1: - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0); + snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 1, 0); break; default: return -EINVAL; -- cgit v1.2.3 From bbcdb69dfcbd8842ee2a54265abd3e53cb3089e2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 12 Jan 2015 10:27:19 +0100 Subject: ASoC: fsl: Remove fsl_asoc_xlate_tdm_slot_mask() Now that the fsl DAI drivers uses the same semantics as the rest of a ASoC the custom fsl_asoc_xlate_tdm_slot_mask() callback can be removed as it is identical to the generic one. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_utils.c | 27 --------------------------- sound/soc/fsl/fsl_utils.h | 3 --- sound/soc/fsl/imx-ssi.c | 1 - 3 files changed, 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 5fd4463dbf05..b9e42b503a37 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -86,33 +86,6 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, } EXPORT_SYMBOL(fsl_asoc_get_dma_channel); -/** - * fsl_asoc_xlate_tdm_slot_mask - generate TDM slot TX/RX mask. - * - * @slots: Number of slots in use. - * @tx_mask: bitmask representing active TX slots. - * @rx_mask: bitmask representing active RX slots. - * - * This function used to generate the TDM slot TX/RX mask. And the TX/RX - * mask will use a 1 bit for an active slot as default, and the default - * active bits are at the LSB of the mask value. - */ -int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots, - unsigned int *tx_mask, - unsigned int *rx_mask) -{ - if (!slots) - return -EINVAL; - - if (tx_mask) - *tx_mask = ((1 << slots) - 1); - if (rx_mask) - *rx_mask = ((1 << slots) - 1); - - return 0; -} -EXPORT_SYMBOL_GPL(fsl_asoc_xlate_tdm_slot_mask); - MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Freescale ASoC utility code"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/fsl/fsl_utils.h b/sound/soc/fsl/fsl_utils.h index df535db40313..1687b66ef18e 100644 --- a/sound/soc/fsl/fsl_utils.h +++ b/sound/soc/fsl/fsl_utils.h @@ -22,7 +22,4 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, const char *name, struct snd_soc_dai_link *dai, unsigned int *dma_channel_id, unsigned int *dma_id); -int fsl_asoc_xlate_tdm_slot_mask(unsigned int slots, - unsigned int *tx_mask, - unsigned int *rx_mask); #endif /* _FSL_UTILS_H */ diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 6aeaac33871a..461ce27b884f 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -340,7 +340,6 @@ static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { .set_fmt = imx_ssi_set_dai_fmt, .set_clkdiv = imx_ssi_set_dai_clkdiv, .set_sysclk = imx_ssi_set_dai_sysclk, - .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask, .set_tdm_slot = imx_ssi_set_dai_tdm_slot, .trigger = imx_ssi_trigger, }; -- cgit v1.2.3 From e46c93669349072f5caca853f5618cfa01b86008 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 12 Jan 2015 10:27:20 +0100 Subject: ASoC: Update snd_soc_dai_set_tdm_slot() documentation There have been some conflicting interpretations of how snd_soc_dai_set_tdm_slot() is supposed to work. This patch updates the documentation to be more specific on the exact semantics to avoid such problems in the future. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 985052b3fbed..64e047dc7cdf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2119,15 +2119,27 @@ static int snd_soc_xlate_tdm_slot_mask(unsigned int slots, } /** - * snd_soc_dai_set_tdm_slot - configure DAI TDM. - * @dai: DAI + * snd_soc_dai_set_tdm_slot() - Configures a DAI for TDM operation + * @dai: The DAI to configure * @tx_mask: bitmask representing active TX slots. * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. * @slot_width: Width in bits for each slot. * - * Configures a DAI for TDM operation. Both mask and slots are codec and DAI - * specific. + * This function configures the specified DAI for TDM operation. @slot contains + * the total number of slots of the TDM stream and @slot_with the width of each + * slot in bit clock cycles. @tx_mask and @rx_mask are bitmasks specifying the + * active slots of the TDM stream for the specified DAI, i.e. which slots the + * DAI should write to or read from. If a bit is set the corresponding slot is + * active, if a bit is cleared the corresponding slot is inactive. Bit 0 maps to + * the first slot, bit 1 to the second slot and so on. The first active slot + * maps to the first channel of the DAI, the second active slot to the second + * channel and so on. + * + * TDM mode can be disabled by passing 0 for @slots. In this case @tx_mask, + * @rx_mask and @slot_width will be ignored. + * + * Returns 0 on success, a negative error code otherwise. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) -- cgit v1.2.3 From ddf9ea21f5fa0832c9711ae13dd467d1f5c4cd87 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Thu, 22 Jan 2015 15:47:16 -0800 Subject: ASoC: ts3a227e: Remap keys to match Android headset specification ts3a227e datasheet says typical key resistance is key1 50 Ohm key2 135 Ohm key3 240 Ohm key4 470 Ohm The android headset specification expect buttons impedance: A (MEDIA) 0-70 Ohm D (VOICECOMMAND) 110-180 Ohm B (VOLUMEUP) 210-290 Ohm C (VOLUMEDOWN) 360-680 Ohm Thus key mapping should be key1 - MEDIA key2 - VOICECOMMAND key3 - VOLUMEUP key3 - VOLUMEDOWN Signed-off-by: Anatol Pomozov Acked-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/ts3a227e.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 1d1205702d23..b55034f63de5 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -221,9 +221,9 @@ int ts3a227e_enable_jack_detect(struct snd_soc_component *component, struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ts3a227e->jack = jack; ts3a227e_jack_report(ts3a227e); -- cgit v1.2.3 From 39552d7ad1409d07ef278a97adbfbee02a272d25 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Thu, 22 Jan 2015 15:47:24 -0800 Subject: ASoC: ts3a227e: Add dts property that allows to specify micbias voltage The voltage controls key press threshold. Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- sound/soc/codecs/ts3a227e.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index b55034f63de5..65f8ec2094fb 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -79,6 +79,10 @@ static const int ts3a227e_buttons[] = { /* TS3A227E_REG_SETTING_2 0x05 */ #define KP_ENABLE 0x04 +/* TS3A227E_REG_SETTING_3 0x06 */ +#define MICBIAS_SETTING_SFT (3) +#define MICBIAS_SETTING_MASK (0x7 << MICBIAS_SETTING_SFT) + /* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ #define TYPE_3_POLE 0x01 #define TYPE_4_POLE_OMTP 0x02 @@ -248,6 +252,21 @@ static const struct regmap_config ts3a227e_regmap_config = { .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), }; +static int ts3a227e_parse_dt(struct ts3a227e *ts3a227e, struct device_node *np) +{ + u32 micbias; + int err; + + err = of_property_read_u32(np, "ti,micbias", &micbias); + if (!err) { + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, + MICBIAS_SETTING_MASK, + (micbias & 0x07) << MICBIAS_SETTING_SFT); + } + + return 0; +} + static int ts3a227e_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -265,6 +284,14 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, if (IS_ERR(ts3a227e->regmap)) return PTR_ERR(ts3a227e->regmap); + if (dev->of_node) { + ret = ts3a227e_parse_dt(ts3a227e, dev->of_node); + if (ret) { + dev_err(dev, "Failed to parse device tree: %d\n", ret); + return ret; + } + } + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "TS3A227E", ts3a227e); -- cgit v1.2.3 From 9503112d909cbbc2865a28c2586c436254169da8 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Mon, 2 Feb 2015 16:48:05 +0200 Subject: ASoC: tlv320aic3x: Add support for tlv320aic3104 Disables GPIO support and LINE2 input and renames Mic3 input to Mic2, if tlv320aic3104 mode is seleced. Devicetree binding document is updated accordingly. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 345 +++++++++++++++++++++++++++++------------ 1 file changed, 244 insertions(+), 101 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b7ebce054b4e..cb92cdba0324 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -87,6 +87,7 @@ struct aic3x_priv { #define AIC3X_MODEL_3X 0 #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 +#define AIC3X_MODEL_3104 3 u16 model; /* Selects the micbias voltage */ @@ -316,52 +317,37 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * only for swapped L-to-R and R-to-L routes. See below stereo controls * for direct L-to-L and R-to-R routes. */ - SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", - LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer PGAR Bypass Volume", PGAR_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer DACR1 Playback Volume", DACR1_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", - LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer PGAL Bypass Volume", PGAL_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer DACL1 Playback Volume", DACL1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", - LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer PGAR Bypass Volume", PGAR_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer DACR1 Playback Volume", DACR1_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", - LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer PGAL Bypass Volume", PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer DACL1 Playback Volume", DACL1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", - LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer PGAR Bypass Volume", PGAR_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer DACR1 Playback Volume", DACR1_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", - LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer PGAL Bypass Volume", PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer DACL1 Playback Volume", DACL1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), /* Stereo output controls for direct L-to-L and R-to-R routes */ - SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", - LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("Line PGA Bypass Volume", PGAL_2_LLOPM_VOL, PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), @@ -369,9 +355,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", - LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HP PGA Bypass Volume", PGAL_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -379,9 +362,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", - LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HPCOM PGA Bypass Volume", PGAL_2_HPLCOM_VOL, PGAR_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), @@ -424,6 +404,45 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), }; +/* For other than tlv320aic3104 */ +static const struct snd_kcontrol_new aic3x_extra_snd_controls[] = { + /* + * Output controls that map to output mixer switches. Note these are + * only for swapped L-to-R and R-to-L routes. See below stereo controls + * for direct L-to-L and R-to-R routes. + */ + SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", + LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", + LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", + LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", + LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", + LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", + LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + + /* Stereo output controls for direct L-to-L and R-to-R routes */ + SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", + LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", + LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", + LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), +}; + static const struct snd_kcontrol_new aic3x_mono_controls[] = { SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, @@ -464,22 +483,24 @@ SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum); /* Left Line Mixer */ static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), }; /* Right Line Mixer */ static const struct snd_kcontrol_new aic3x_right_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), }; /* Mono Mixer */ @@ -494,42 +515,46 @@ static const struct snd_kcontrol_new aic3x_mono_mixer_controls[] = { /* Left HP Mixer */ static const struct snd_kcontrol_new aic3x_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLOUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), }; /* Right HP Mixer */ static const struct snd_kcontrol_new aic3x_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), }; /* Left HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_left_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), }; /* Right HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_right_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPRCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; /* Left PGA Mixer */ @@ -550,6 +575,22 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), }; +/* Left PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), +}; + +/* Right PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), +}; + /* Left Line1 Mux */ static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum); @@ -593,26 +634,56 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_left_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1l_mux_controls), SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1r_mux_controls), - SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line2_mux_controls), /* Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", LINE1R_2_RADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_right_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1l_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1r_mux_controls), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("LLOUT"), + SND_SOC_DAPM_OUTPUT("RLOUT"), + SND_SOC_DAPM_OUTPUT("HPLOUT"), + SND_SOC_DAPM_OUTPUT("HPROUT"), + SND_SOC_DAPM_OUTPUT("HPLCOM"), + SND_SOC_DAPM_OUTPUT("HPRCOM"), + + SND_SOC_DAPM_INPUT("LINE1L"), + SND_SOC_DAPM_INPUT("LINE1R"), + + /* + * Virtual output pin to detection block inside codec. This can be + * used to keep codec bias on if gpio or detection features are needed. + * Force pin on or construct a path with an input jack and mic bias + * widgets. + */ + SND_SOC_DAPM_OUTPUT("Detection"), +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3x_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_left_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line2_mux_controls), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), @@ -637,11 +708,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), - /* Mic Bias */ - SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_line_mixer_controls[0], @@ -662,27 +728,46 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_hpcom_mixer_controls[0], ARRAY_SIZE(aic3x_right_hpcom_mixer_controls)), - SND_SOC_DAPM_OUTPUT("LLOUT"), - SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("HPLOUT"), - SND_SOC_DAPM_OUTPUT("HPROUT"), - SND_SOC_DAPM_OUTPUT("HPLCOM"), - SND_SOC_DAPM_OUTPUT("HPRCOM"), - SND_SOC_DAPM_INPUT("MIC3L"), SND_SOC_DAPM_INPUT("MIC3R"), - SND_SOC_DAPM_INPUT("LINE1L"), - SND_SOC_DAPM_INPUT("LINE1R"), SND_SOC_DAPM_INPUT("LINE2L"), SND_SOC_DAPM_INPUT("LINE2R"), +}; - /* - * Virtual output pin to detection block inside codec. This can be - * used to keep codec bias on if gpio or detection features are needed. - * Force pin on or construct a path with an input jack and mic bias - * widgets. - */ - SND_SOC_DAPM_OUTPUT("Detection"), +/* For tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3104_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_left_pga_mixer_controls)), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_right_pga_mixer_controls)), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_line_mixer_controls[0], + ARRAY_SIZE(aic3x_left_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_line_mixer_controls[0], + ARRAY_SIZE(aic3x_right_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hpcom_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hpcom_mixer_controls) - 2), + + SND_SOC_DAPM_INPUT("MIC2L"), + SND_SOC_DAPM_INPUT("MIC2R"), }; static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { @@ -712,17 +797,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line1R Mux", "single-ended", "LINE1R"}, {"Left Line1R Mux", "differential", "LINE1R"}, - {"Left Line2L Mux", "single-ended", "LINE2L"}, - {"Left Line2L Mux", "differential", "LINE2L"}, - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, - {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, - {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Left ADC", NULL, "Left PGA Mixer"}, - {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -730,25 +808,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line1L Mux", "single-ended", "LINE1L"}, {"Right Line1L Mux", "differential", "LINE1L"}, - {"Right Line2R Mux", "single-ended", "LINE2R"}, - {"Right Line2R Mux", "differential", "LINE2R"}, - {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, - {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, - {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, - {"Right ADC", NULL, "GPIO1 dmic modclk"}, - - /* - * Logical path between digital mic enable and GPIO1 modulator clock - * output function - */ - {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, /* Left DAC Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, @@ -761,10 +824,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right DAC Mux", "DAC_R3", "Right DAC"}, /* Left Line Output */ - {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -773,10 +834,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"LLOUT", NULL, "Left Line Out"}, /* Right Line Output */ - {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -785,10 +844,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"RLOUT", NULL, "Right Line Out"}, /* Left HP Output */ - {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -797,10 +854,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLOUT", NULL, "Left HP Out"}, /* Right HP Output */ - {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -809,10 +864,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPROUT", NULL, "Right HP Out"}, /* Left HPCOM Output */ - {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -823,10 +876,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLCOM", NULL, "Left HP Com"}, /* Right HPCOM Output */ - {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -839,6 +890,72 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra[] = { + /* Left Input */ + {"Left Line2L Mux", "single-ended", "LINE2L"}, + {"Left Line2L Mux", "differential", "LINE2L"}, + + {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, + {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Left ADC", NULL, "GPIO1 dmic modclk"}, + + /* Right Input */ + {"Right Line2R Mux", "single-ended", "LINE2R"}, + {"Right Line2R Mux", "differential", "LINE2R"}, + + {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Right ADC", NULL, "GPIO1 dmic modclk"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, + + /* Left Line Output */ + {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right Line Output */ + {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HP Output */ + {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HP Output */ + {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HPCOM Output */ + {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HPCOM Output */ + {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra_3104[] = { + /* Left Input */ + {"Left PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Left PGA Mixer", "Mic2R Switch", "MIC2R"}, + + /* Right Input */ + {"Right PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Right PGA Mixer", "Mic2R Switch", "MIC2R"}, +}; + static const struct snd_soc_dapm_route intercon_mono[] = { /* Mono Output */ {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, @@ -867,17 +984,31 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, ARRAY_SIZE(aic3x_dapm_mono_widgets)); snd_soc_dapm_add_routes(dapm, intercon_mono, ARRAY_SIZE(intercon_mono)); break; case AIC3X_MODEL_3007: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); break; + case AIC3X_MODEL_3104: + snd_soc_dapm_new_controls(dapm, aic3104_extra_dapm_widgets, + ARRAY_SIZE(aic3104_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra_3104, + ARRAY_SIZE(intercon_extra_3104)); + break; } return 0; @@ -1438,23 +1569,33 @@ static int aic3x_probe(struct snd_soc_codec *codec) aic3x_init(codec); if (aic3x->setup) { - /* setup GPIO functions */ - snd_soc_write(codec, AIC3X_GPIO1_REG, - (aic3x->setup->gpio_func[0] & 0xf) << 4); - snd_soc_write(codec, AIC3X_GPIO2_REG, - (aic3x->setup->gpio_func[1] & 0xf) << 4); + if (aic3x->model != AIC3X_MODEL_3104) { + /* setup GPIO functions */ + snd_soc_write(codec, AIC3X_GPIO1_REG, + (aic3x->setup->gpio_func[0] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO2_REG, + (aic3x->setup->gpio_func[1] & 0xf) << 4); + } else { + dev_warn(codec->dev, "GPIO functionality is not supported on tlv320aic3104\n"); + } } switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, aic3x_mono_controls, ARRAY_SIZE(aic3x_mono_controls)); break; case AIC3X_MODEL_3007: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); break; + case AIC3X_MODEL_3104: + break; } /* set mic bias voltage */ @@ -1522,6 +1663,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3106", AIC3X_MODEL_3X }, + { "tlv320aic3104", AIC3X_MODEL_3104 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1673,6 +1815,7 @@ static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic33" }, { .compatible = "ti,tlv320aic3007" }, { .compatible = "ti,tlv320aic3106" }, + { .compatible = "ti,tlv320aic3104" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- cgit v1.2.3 From b8255930e0fbda841890ff6bb7154aa5fd62e143 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 4 Feb 2015 12:15:46 +0200 Subject: ASoC: tlv320aic3x: Fix bad comment before intercon_extra_3104 definition MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The intercon_extra_3104 is obviously for tlv320aic3104. Reported-by: Benoît Thébaudeau Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cb92cdba0324..ed35e8f1f04c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -945,7 +945,7 @@ static const struct snd_soc_dapm_route intercon_extra[] = { {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, }; -/* For other than tlv320aic3104 */ +/* For tlv320aic3104 */ static const struct snd_soc_dapm_route intercon_extra_3104[] = { /* Left Input */ {"Left PGA Mixer", "Mic2L Switch", "MIC2L"}, -- cgit v1.2.3 From e7a0332f716d92b9a80c46fbd7d624c502984ca7 Mon Sep 17 00:00:00 2001 From: "Lad, Prabhakar" Date: Wed, 4 Feb 2015 17:29:30 +0000 Subject: ASoC: ts3a227e: fix sparse warning this patch fixes following sparse warning: ts3a227e.c:222:5: warning: symbol 'ts3a227e_enable_jack_detect' was not declared. Should it be static? Signed-off-by: Lad, Prabhakar Signed-off-by: Mark Brown --- sound/soc/codecs/ts3a227e.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 1d1205702d23..6528bfe5c2ff 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -20,6 +20,8 @@ #include #include +#include "ts3a227e.h" + struct ts3a227e { struct regmap *regmap; struct snd_soc_jack *jack; -- cgit v1.2.3