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authorLinus Torvalds <torvalds@linux-foundation.org>2012-03-22 13:00:13 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2012-03-22 13:00:13 -0700
commitb2094ef840697bc8ca5d17a83b7e30fad5f1e9fa (patch)
tree64e5f7253b6a85b6d5d36f95c0d3c67c1798918d /include
parent424a6f6ef990b7e9f56f6627bfc6c46b493faeb4 (diff)
parent6681bc0deba495fad0d6fb349e40524abd1b1732 (diff)
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull updates of sound stuff from Takashi Iwai: "Here is the first big update chunk of sound stuff for 3.4-rc1. In the common sound infrastructure, there are a few changes for dynamic PCM support (used in ASoC) and a few clean-ups. Majority of changes are found, as usual, in HD-audio and ASoC. Some highlights of HD-audio changes: - All the long-standing static quirk codes for Realtek codec were finally removed by fixing and extending the Realtek auto-parser. - The mute-LED control is standardized over all HD-audio codec drivers using the extended vmaster hook. - The vmaster slave mixer elements are initialized to 0dB as default so that the user won't be annoyed by the silent output after updates, e.g. due to the additions of new elements. - Other many fix-ups for the misc HD-audio devices. In the ASoC side, this is a very active release, including a quite a few framework enhancements. Some highlights: - Support for widgets not associated with a CODEC, an important part of the dynamic PCM framework. - A library factoring out the common code shared by dmaengine based DMA drivers contributed by Lars-Peter Clausen. This will save a lot of code and make it much easier to deploy enhancements to dmaengine. - Support for binary controls, used for providing runtime configuration of algorithm coefficients. - A new DAPM widget type for regulator supplies allowing drivers for devices that can power down unused supplies while active to do without any per-driver code. - DAPM widgets for DAIs, initially giving a speed boost for playback startup and shutdown and also the basis for CODEC<->CODEC DAI link support. - Support for specifying the number of significant bits on audio interfaces, useful for allowing applications to know how much effort to put into generating data for a larger sample format. - Conversion of the FSI driver used on some SH processors to DMAEngine. - Conversion of EP93xx drivers to DMAEngine. - New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics WM2200. - Move audmux driver from arc/arm to sound/soc - McBSP move from arch/ to sound/ and updates Also, a few small updates and fixes for other drivers like au88x0, ymfpci, USB 6fire, USB usx2yaudio are included." * tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits) ASoC: wm8994: Provide VMID mode control and fix default sequence ASoC: wm8994: Add missing break in resume ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF ASoC: pxa-ssp: atomically set stream active masks ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master ASoC: Samsung: Added to support mono recording ALSA: hda - Fix build with CONFIG_PM=n ALSA: au88x0 - Avoid possible Oops at unbinding ALSA: usb-audio - Fix build error by consitification of rate list ASoC: core: Fix obscure leak of runtime array ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link() ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list ASoC: wm8996: Add 44.1kHz support ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE ASoC: mx27vis-aic32x4: Convert it to platform driver ALSA: hda - fix printing of high HDMI sample rates ALSA: ymfpci - Fix legacy registers on S3/S4 resume ALSA: control - Fixe a trailing white space error ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook() ALSA: hda - Add "Mute-LED Mode" enum control ...
Diffstat (limited to 'include')
-rw-r--r--include/linux/mfd/wm8994/pdata.h3
-rw-r--r--include/linux/platform_data/omap-abe-twl6040.h49
-rw-r--r--include/linux/regmap.h4
-rw-r--r--include/sound/control.h7
-rw-r--r--include/sound/dmaengine_pcm.h49
-rw-r--r--include/sound/jack.h3
-rw-r--r--include/sound/max9768.h24
-rw-r--r--include/sound/pcm.h9
-rw-r--r--include/sound/sh_fsi.h12
-rw-r--r--include/sound/soc-dai.h11
-rw-r--r--include/sound/soc-dapm.h33
-rw-r--r--include/sound/soc.h45
-rw-r--r--include/sound/version.h2
-rw-r--r--include/sound/wm2200.h41
-rw-r--r--include/sound/wm8962.h6
-rw-r--r--include/sound/ymfpci.h2
16 files changed, 279 insertions, 21 deletions
diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h
index 3fb1f407d5e..dc3e0501168 100644
--- a/include/linux/mfd/wm8994/pdata.h
+++ b/include/linux/mfd/wm8994/pdata.h
@@ -185,6 +185,9 @@ struct wm8994_pdata {
unsigned int jd_scthr:2;
unsigned int jd_thr:2;
+ /* Configure WM1811 jack detection for use with external capacitor */
+ unsigned int jd_ext_cap:1;
+
/* WM8958 microphone bias configuration */
int micbias[2];
diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h
new file mode 100644
index 00000000000..5d298ac10fc
--- /dev/null
+++ b/include/linux/platform_data/omap-abe-twl6040.h
@@ -0,0 +1,49 @@
+/**
+ * omap-abe-twl6040.h - ASoC machine driver OMAP4+ devices, header.
+ *
+ * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com
+ * All rights reserved.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.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
+ */
+
+#ifndef _OMAP_ABE_TWL6040_H_
+#define _OMAP_ABE_TWL6040_H_
+
+/* To select if only one channel is connected in a stereo port */
+#define ABE_TWL6040_LEFT (1 << 0)
+#define ABE_TWL6040_RIGHT (1 << 1)
+
+struct omap_abe_twl6040_data {
+ char *card_name;
+ /* Feature flags for connected audio pins */
+ u8 has_hs;
+ u8 has_hf;
+ bool has_ep;
+ u8 has_aux;
+ u8 has_vibra;
+ bool has_dmic;
+ bool has_hsmic;
+ bool has_mainmic;
+ bool has_submic;
+ u8 has_afm;
+ /* Other features */
+ bool jack_detection; /* board can detect jack events */
+ int mclk_freq; /* MCLK frequency speed for twl6040 */
+};
+
+#endif /* _OMAP_ABE_TWL6040_H_ */
diff --git a/include/linux/regmap.h b/include/linux/regmap.h
index eb93921cdd3..c47898d47c8 100644
--- a/include/linux/regmap.h
+++ b/include/linux/regmap.h
@@ -143,12 +143,16 @@ int regmap_update_bits(struct regmap *map, unsigned int reg,
int regmap_update_bits_check(struct regmap *map, unsigned int reg,
unsigned int mask, unsigned int val,
bool *change);
+int regmap_get_val_bytes(struct regmap *map);
int regcache_sync(struct regmap *map);
void regcache_cache_only(struct regmap *map, bool enable);
void regcache_cache_bypass(struct regmap *map, bool enable);
void regcache_mark_dirty(struct regmap *map);
+int regmap_register_patch(struct regmap *map, const struct reg_default *regs,
+ int num_regs);
+
/**
* Description of an IRQ for the generic regmap irq_chip.
*
diff --git a/include/sound/control.h b/include/sound/control.h
index b2796e83c7a..8332e865c75 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -40,7 +40,7 @@ struct snd_kcontrol_new {
snd_ctl_elem_iface_t iface; /* interface identifier */
unsigned int device; /* device/client number */
unsigned int subdevice; /* subdevice (substream) number */
- unsigned char *name; /* ASCII name of item */
+ const unsigned char *name; /* ASCII name of item */
unsigned int index; /* index of item */
unsigned int access; /* access rights */
unsigned int count; /* count of same elements */
@@ -227,6 +227,11 @@ snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
}
+int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
+ void (*hook)(void *private_data, int),
+ void *private_data);
+void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kctl);
+
/*
* Helper functions for jack-detection controls
*/
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
new file mode 100644
index 00000000000..a8fcaa6d531
--- /dev/null
+++ b/include/sound/dmaengine_pcm.h
@@ -0,0 +1,49 @@
+/*
+ * Copyright (C) 2012, Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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.
+ *
+ * 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.
+ *
+ */
+#ifndef __SOUND_DMAENGINE_PCM_H__
+#define __SOUND_DMAENGINE_PCM_H__
+
+#include <sound/pcm.h>
+#include <linux/dmaengine.h>
+
+/**
+ * snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM
+ * substream
+ * @substream: PCM substream
+ */
+static inline enum dma_transfer_direction
+snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return DMA_MEM_TO_DEV;
+ else
+ return DMA_DEV_TO_MEM;
+}
+
+void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data);
+void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream);
+
+int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
+ const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config);
+int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream);
+
+int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
+ dma_filter_fn filter_fn, void *filter_data);
+int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream);
+
+struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream);
+
+#endif
diff --git a/include/sound/jack.h b/include/sound/jack.h
index 63c790742db..58916573db5 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -53,6 +53,9 @@ enum snd_jack_types {
SND_JACK_BTN_5 = 0x0200,
};
+/* Keep in sync with definitions above */
+#define SND_JACK_SWITCH_TYPES 6
+
struct snd_jack {
struct input_dev *input_dev;
int registered;
diff --git a/include/sound/max9768.h b/include/sound/max9768.h
new file mode 100644
index 00000000000..0f78b41d030
--- /dev/null
+++ b/include/sound/max9768.h
@@ -0,0 +1,24 @@
+/*
+ * Platform data for MAX9768
+ * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K.
+ * same licence as the driver
+ */
+
+#ifndef __SOUND_MAX9768_PDATA_H__
+#define __SOUND_MAX9768_PDATA_H__
+
+/**
+ * struct max9768_pdata - optional platform specific MAX9768 configuration
+ * @shdn_gpio: GPIO to SHDN pin. If not valid, pin must be hardwired HIGH
+ * @mute_gpio: GPIO to MUTE pin. If not valid, control for mute won't be added
+ * @flags: configuration flags, e.g. set classic PWM mode (check datasheet
+ * regarding "filterless modulation" which is default).
+ */
+struct max9768_pdata {
+ int shdn_gpio;
+ int mute_gpio;
+ unsigned flags;
+#define MAX9768_FLAG_CLASSIC_PWM (1 << 0)
+};
+
+#endif /* __SOUND_MAX9768_PDATA_H__*/
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 0cf91b2f08c..0d1112815be 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -264,7 +264,7 @@ struct snd_pcm_hw_constraint_ratdens {
struct snd_pcm_hw_constraint_list {
unsigned int count;
- unsigned int *list;
+ const unsigned int *list;
unsigned int mask;
};
@@ -454,6 +454,7 @@ struct snd_pcm {
void *private_data;
void (*private_free) (struct snd_pcm *pcm);
struct device *dev; /* actual hw device this belongs to */
+ bool internal; /* pcm is for internal use only */
#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
struct snd_pcm_oss oss;
#endif
@@ -475,6 +476,9 @@ extern const struct file_operations snd_pcm_f_ops[2];
int snd_pcm_new(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
struct snd_pcm **rpcm);
+int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
+ int playback_count, int capture_count,
+ struct snd_pcm **rpcm);
int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count);
int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree);
@@ -781,7 +785,8 @@ void snd_interval_muldivk(const struct snd_interval *a, const struct snd_interva
unsigned int k, struct snd_interval *c);
void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
const struct snd_interval *b, struct snd_interval *c);
-int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask);
+int snd_interval_list(struct snd_interval *i, unsigned int count,
+ const unsigned int *list, unsigned int mask);
int snd_interval_ratnum(struct snd_interval *i,
unsigned int rats_count, struct snd_ratnum *rats,
unsigned int *nump, unsigned int *denp);
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index 9b1aacaa82f..b457e87fbd0 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -72,10 +72,16 @@
#define SH_FSI_BPFMD_32 (5 << 4)
#define SH_FSI_BPFMD_16 (6 << 4)
+struct sh_fsi_port_info {
+ unsigned long flags;
+ int tx_id;
+ int rx_id;
+ int (*set_rate)(struct device *dev, int rate, int enable);
+};
+
struct sh_fsi_platform_info {
- unsigned long porta_flags;
- unsigned long portb_flags;
- int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
+ struct sh_fsi_port_info port_a;
+ struct sh_fsi_port_info port_b;
};
/*
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 2413acc5488..c429f248cf4 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -17,6 +17,7 @@
#include <linux/list.h>
struct snd_pcm_substream;
+struct snd_soc_dapm_widget;
/*
* DAI hardware audio formats.
@@ -238,6 +239,9 @@ struct snd_soc_dai {
unsigned char pop_wait:1;
unsigned char probed:1;
+ struct snd_soc_dapm_widget *playback_widget;
+ struct snd_soc_dapm_widget *capture_widget;
+
/* DAI DMA data */
void *playback_dma_data;
void *capture_dma_data;
@@ -246,10 +250,9 @@ struct snd_soc_dai {
unsigned int rate;
/* parent platform/codec */
- union {
- struct snd_soc_platform *platform;
- struct snd_soc_codec *codec;
- };
+ struct snd_soc_platform *platform;
+ struct snd_soc_codec *codec;
+
struct snd_soc_card *card;
struct list_head list;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index d26a9b78477..e46107fffeb 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -243,6 +243,10 @@
{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .event = wevent, \
.event_flags = wflags}
+#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \
+{ .id = snd_soc_dapm_regulator_supply, .name = wname, \
+ .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
@@ -322,6 +326,8 @@ struct snd_soc_dapm_context;
int dapm_reg_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
+int dapm_regulator_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event);
/* dapm controls */
int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
@@ -346,11 +352,12 @@ int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
-int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
- const struct snd_soc_dapm_widget *widget);
int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget,
int num);
+int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
+ struct snd_soc_dai *dai);
+int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
/* dapm path setup */
int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
@@ -361,10 +368,16 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_route *route, int num);
/* dapm events */
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
- const char *stream, int event);
+int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
+ struct snd_soc_dai *dai, int event);
void snd_soc_dapm_shutdown(struct snd_soc_card *card);
+/* external DAPM widget events */
+int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kcontrol, int connect);
+int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
+ struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e);
+
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm,
@@ -411,9 +424,11 @@ enum snd_soc_dapm_type {
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
+ snd_soc_dapm_regulator_supply, /* external regulator */
snd_soc_dapm_aif_in, /* audio interface input */
snd_soc_dapm_aif_out, /* audio interface output */
snd_soc_dapm_siggen, /* signal generator */
+ snd_soc_dapm_dai, /* link to DAI structure */
};
/*
@@ -434,8 +449,8 @@ struct snd_soc_dapm_route {
/* dapm audio path between two widgets */
struct snd_soc_dapm_path {
- char *name;
- char *long_name;
+ const char *name;
+ const char *long_name;
/* source (input) and sink (output) widgets */
struct snd_soc_dapm_widget *source;
@@ -458,13 +473,15 @@ struct snd_soc_dapm_path {
/* dapm widget */
struct snd_soc_dapm_widget {
enum snd_soc_dapm_type id;
- char *name; /* widget name */
- char *sname; /* stream name */
+ const char *name; /* widget name */
+ const char *sname; /* stream name */
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct list_head list;
struct snd_soc_dapm_context *dapm;
+ void *priv; /* widget specific data */
+
/* dapm control */
short reg; /* negative reg = no direct dapm */
unsigned char shift; /* bits to shift */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 0992dff5595..2ebf7877c14 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -185,6 +185,20 @@
.rreg = xreg_right, .shift = xshift, \
.min = xmin, .max = xmax} }
+#define SND_SOC_BYTES(xname, xbase, xregs) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
+ .put = snd_soc_bytes_put, .private_value = \
+ ((unsigned long)&(struct soc_bytes) \
+ {.base = xbase, .num_regs = xregs }) }
+
+#define SND_SOC_BYTES_MASK(xname, xbase, xregs, xmask) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
+ .put = snd_soc_bytes_put, .private_value = \
+ ((unsigned long)&(struct soc_bytes) \
+ {.base = xbase, .num_regs = xregs, \
+ .mask = xmask }) }
/*
* Simplified versions of above macros, declaring a struct and calculating
@@ -366,12 +380,16 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
*Controls
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
- void *data, char *long_name,
+ void *data, const char *long_name,
const char *prefix);
-int snd_soc_add_controls(struct snd_soc_codec *codec,
+int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_card_controls(struct snd_soc_card *soc_card,
+ const struct snd_kcontrol_new *controls, int num_controls);
+int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
+ const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
@@ -409,6 +427,13 @@ int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
+int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
+int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol);
+
/**
* struct snd_soc_reg_access - Describes whether a given register is
@@ -505,6 +530,7 @@ struct snd_soc_pcm_stream {
unsigned int rate_max; /* max rate */
unsigned int channels_min; /* min channels */
unsigned int channels_max; /* max channels */
+ unsigned int sig_bits; /* number of bits of content */
};
/* SoC audio ops */
@@ -559,6 +585,7 @@ struct snd_soc_codec {
unsigned int ac97_created:1; /* Codec has been created by SoC */
unsigned int sysfs_registered:1; /* codec has been sysfs registered */
unsigned int cache_init:1; /* codec cache has been initialized */
+ unsigned int using_regmap:1; /* using regmap access */
u32 cache_only; /* Suppress writes to hardware */
u32 cache_sync; /* Cache needs to be synced to hardware */
@@ -637,6 +664,8 @@ struct snd_soc_codec_driver {
/* codec stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+ bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */
+
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@@ -689,6 +718,7 @@ struct snd_soc_platform {
int id;
struct device *dev;
struct snd_soc_platform_driver *driver;
+ struct mutex mutex;
unsigned int suspended:1; /* platform is suspended */
unsigned int probed:1;
@@ -698,6 +728,11 @@ struct snd_soc_platform {
struct list_head card_list;
struct snd_soc_dapm_context dapm;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_platform_root;
+ struct dentry *debugfs_dapm;
+#endif
};
struct snd_soc_dai_link {
@@ -875,6 +910,12 @@ struct soc_mixer_control {
unsigned int reg, rreg, shift, rshift, invert;
};
+struct soc_bytes {
+ int base;
+ int num_regs;
+ u32 mask;
+};
+
/* enumerated kcontrol */
struct soc_enum {
unsigned short reg;
diff --git a/include/sound/version.h b/include/sound/version.h
index 8fc5321e1ec..cc75024c108 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
/* include/version.h */
-#define CONFIG_SND_VERSION "1.0.24"
+#define CONFIG_SND_VERSION "1.0.25"
#define CONFIG_SND_DATE ""
diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h
new file mode 100644
index 00000000000..79bf55be7ff
--- /dev/null
+++ b/include/sound/wm2200.h
@@ -0,0 +1,41 @@
+/*
+ * linux/sound/wm2200.h -- Platform data for WM2200
+ *
+ * Copyright 2012 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM2200_H
+#define __LINUX_SND_WM2200_H
+
+#define WM2200_GPIO_SET 0x10000
+
+enum wm2200_in_mode {
+ WM2200_IN_SE = 0,
+ WM2200_IN_DIFF = 1,
+ WM2200_IN_DMIC = 2,
+};
+
+enum wm2200_dmic_sup {
+ WM2200_DMIC_SUP_MICVDD = 0,
+ WM2200_DMIC_SUP_MICBIAS1 = 1,
+ WM2200_DMIC_SUP_MICBIAS2 = 2,
+};
+
+struct wm2200_pdata {
+ int reset; /** GPIO controlling /RESET, if any */
+ int ldo_ena; /** GPIO controlling LODENA, if any */
+ int irq_flags;
+
+ int gpio_defaults[4];
+
+ enum wm2200_in_mode in_mode[3];
+ enum wm2200_dmic_sup dmic_sup[3];
+
+ int micbias_cfg[2]; /** Register value to configure MICBIAS */
+};
+
+#endif
diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h
index 1750bed7c2f..79e6d427b85 100644
--- a/include/sound/wm8962.h
+++ b/include/sound/wm8962.h
@@ -49,6 +49,12 @@ struct wm8962_pdata {
bool irq_active_low;
bool spk_mono; /* Speaker outputs tied together as mono */
+
+ /**
+ * This flag should be set if one or both IN4 inputs is wired
+ * in a DC measurement configuration.
+ */
+ bool in4_dc_measure;
};
#endif
diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h
index 444cd6ba0ba..41199664666 100644
--- a/include/sound/ymfpci.h
+++ b/include/sound/ymfpci.h
@@ -366,6 +366,8 @@ struct snd_ymfpci {
#ifdef CONFIG_PM
u32 *saved_regs;
u32 saved_ydsxgr_mode;
+ u16 saved_dsxg_legacy;
+ u16 saved_dsxg_elegacy;
#endif
};