+++ /dev/null
-ASoC Codec Class Driver
-=======================
-
-The codec class driver is generic and hardware independent code that configures
-the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
-It should contain no code that is specific to the target platform or machine.
-All platform and machine specific code should be added to the platform and
-machine drivers respectively.
-
-Each codec class driver *must* provide the following features:-
-
- 1) Codec DAI and PCM configuration
- 2) Codec control IO - using RegMap API
- 3) Mixers and audio controls
- 4) Codec audio operations
- 5) DAPM description.
- 6) DAPM event handler.
-
-Optionally, codec drivers can also provide:-
-
- 7) DAC Digital mute control.
-
-Its probably best to use this guide in conjunction with the existing codec
-driver code in sound/soc/codecs/
-
-ASoC Codec driver breakdown
-===========================
-
-1 - Codec DAI and PCM configuration
------------------------------------
-Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
-PCM capabilities and operations. This struct is exported so that it can be
-registered with the core by your machine driver.
-
-e.g.
-
-static struct snd_soc_dai_ops wm8731_dai_ops = {
- .prepare = wm8731_pcm_prepare,
- .hw_params = wm8731_hw_params,
- .shutdown = wm8731_shutdown,
- .digital_mute = wm8731_mute,
- .set_sysclk = wm8731_set_dai_sysclk,
- .set_fmt = wm8731_set_dai_fmt,
-};
-
-struct snd_soc_dai_driver wm8731_dai = {
- .name = "wm8731-hifi",
- .playback = {
- .stream_name = "Playback",
- .channels_min = 1,
- .channels_max = 2,
- .rates = WM8731_RATES,
- .formats = WM8731_FORMATS,},
- .capture = {
- .stream_name = "Capture",
- .channels_min = 1,
- .channels_max = 2,
- .rates = WM8731_RATES,
- .formats = WM8731_FORMATS,},
- .ops = &wm8731_dai_ops,
- .symmetric_rates = 1,
-};
-
-
-2 - Codec control IO
---------------------
-The codec can usually be controlled via an I2C or SPI style interface
-(AC97 combines control with data in the DAI). The codec driver should use the
-Regmap API for all codec IO. Please see include/linux/regmap.h and existing
-codec drivers for example regmap usage.
-
-
-3 - Mixers and audio controls
------------------------------
-All the codec mixers and audio controls can be defined using the convenience
-macros defined in soc.h.
-
- #define SOC_SINGLE(xname, reg, shift, mask, invert)
-
-Defines a single control as follows:-
-
- xname = Control name e.g. "Playback Volume"
- reg = codec register
- shift = control bit(s) offset in register
- mask = control bit size(s) e.g. mask of 7 = 3 bits
- invert = the control is inverted
-
-Other macros include:-
-
- #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
-
-A stereo control
-
- #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
-
-A stereo control spanning 2 registers
-
- #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
-
-Defines an single enumerated control as follows:-
-
- xreg = register
- xshift = control bit(s) offset in register
- xmask = control bit(s) size
- xtexts = pointer to array of strings that describe each setting
-
- #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
-
-Defines a stereo enumerated control
-
-
-4 - Codec Audio Operations
---------------------------
-The codec driver also supports the following ALSA PCM operations:-
-
-/* SoC audio ops */
-struct snd_soc_ops {
- int (*startup)(struct snd_pcm_substream *);
- void (*shutdown)(struct snd_pcm_substream *);
- int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
- int (*hw_free)(struct snd_pcm_substream *);
- int (*prepare)(struct snd_pcm_substream *);
-};
-
-Please refer to the ALSA driver PCM documentation for details.
-http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
-
-
-5 - DAPM description.
----------------------
-The Dynamic Audio Power Management description describes the codec power
-components and their relationships and registers to the ASoC core.
-Please read dapm.txt for details of building the description.
-
-Please also see the examples in other codec drivers.
-
-
-6 - DAPM event handler
-----------------------
-This function is a callback that handles codec domain PM calls and system
-domain PM calls (e.g. suspend and resume). It is used to put the codec
-to sleep when not in use.
-
-Power states:-
-
- SNDRV_CTL_POWER_D0: /* full On */
- /* vref/mid, clk and osc on, active */
-
- SNDRV_CTL_POWER_D1: /* partial On */
- SNDRV_CTL_POWER_D2: /* partial On */
-
- SNDRV_CTL_POWER_D3hot: /* Off, with power */
- /* everything off except vref/vmid, inactive */
-
- SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
-
-
-7 - Codec DAC digital mute control
-----------------------------------
-Most codecs have a digital mute before the DACs that can be used to
-minimise any system noise. The mute stops any digital data from
-entering the DAC.
-
-A callback can be created that is called by the core for each codec DAI
-when the mute is applied or freed.
-
-i.e.
-
-static int wm8974_mute(struct snd_soc_dai *dai, int mute)
-{
- struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
-
- if (mute)
- snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
- else
- snd_soc_write(codec, WM8974_DAC, mute_reg);
- return 0;
-}
--- /dev/null
+=======================
+ASoC Codec Class Driver
+=======================
+
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
+It should contain no code that is specific to the target platform or machine.
+All platform and machine specific code should be added to the platform and
+machine drivers respectively.
+
+Each codec class driver *must* provide the following features:-
+
+1. Codec DAI and PCM configuration
+2. Codec control IO - using RegMap API
+3. Mixers and audio controls
+4. Codec audio operations
+5. DAPM description.
+6. DAPM event handler.
+
+Optionally, codec drivers can also provide:-
+
+7. DAC Digital mute control.
+
+Its probably best to use this guide in conjunction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+Codec DAI and PCM configuration
+-------------------------------
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
+PCM capabilities and operations. This struct is exported so that it can be
+registered with the core by your machine driver.
+
+e.g.
+::
+
+ static struct snd_soc_dai_ops wm8731_dai_ops = {
+ .prepare = wm8731_pcm_prepare,
+ .hw_params = wm8731_hw_params,
+ .shutdown = wm8731_shutdown,
+ .digital_mute = wm8731_mute,
+ .set_sysclk = wm8731_set_dai_sysclk,
+ .set_fmt = wm8731_set_dai_fmt,
+ };
+
+ struct snd_soc_dai_driver wm8731_dai = {
+ .name = "wm8731-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8731_RATES,
+ .formats = WM8731_FORMATS,},
+ .ops = &wm8731_dai_ops,
+ .symmetric_rates = 1,
+ };
+
+
+Codec control IO
+----------------
+The codec can usually be controlled via an I2C or SPI style interface
+(AC97 combines control with data in the DAI). The codec driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
+
+
+Mixers and audio controls
+-------------------------
+All the codec mixers and audio controls can be defined using the convenience
+macros defined in soc.h.
+::
+
+ #define SOC_SINGLE(xname, reg, shift, mask, invert)
+
+Defines a single control as follows:-
+::
+
+ xname = Control name e.g. "Playback Volume"
+ reg = codec register
+ shift = control bit(s) offset in register
+ mask = control bit size(s) e.g. mask of 7 = 3 bits
+ invert = the control is inverted
+
+Other macros include:-
+::
+
+ #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
+
+A stereo control
+::
+
+ #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
+
+A stereo control spanning 2 registers
+::
+
+ #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
+
+Defines an single enumerated control as follows:-
+::
+
+ xreg = register
+ xshift = control bit(s) offset in register
+ xmask = control bit(s) size
+ xtexts = pointer to array of strings that describe each setting
+
+ #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
+
+Defines a stereo enumerated control
+
+
+Codec Audio Operations
+----------------------
+The codec driver also supports the following ALSA PCM operations:-
+::
+
+ /* SoC audio ops */
+ struct snd_soc_ops {
+ int (*startup)(struct snd_pcm_substream *);
+ void (*shutdown)(struct snd_pcm_substream *);
+ int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+ int (*hw_free)(struct snd_pcm_substream *);
+ int (*prepare)(struct snd_pcm_substream *);
+ };
+
+Please refer to the ALSA driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+
+DAPM description
+----------------
+The Dynamic Audio Power Management description describes the codec power
+components and their relationships and registers to the ASoC core.
+Please read dapm.txt for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+DAPM event handler
+------------------
+This function is a callback that handles codec domain PM calls and system
+domain PM calls (e.g. suspend and resume). It is used to put the codec
+to sleep when not in use.
+
+Power states:-
+::
+
+ SNDRV_CTL_POWER_D0: /* full On */
+ /* vref/mid, clk and osc on, active */
+
+ SNDRV_CTL_POWER_D1: /* partial On */
+ SNDRV_CTL_POWER_D2: /* partial On */
+
+ SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ /* everything off except vref/vmid, inactive */
+
+ SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
+
+
+Codec DAC digital mute control
+------------------------------
+Most codecs have a digital mute before the DACs that can be used to
+minimise any system noise. The mute stops any digital data from
+entering the DAC.
+
+A callback can be created that is called by the core for each codec DAI
+when the mute is applied or freed.
+
+i.e.
+::
+
+ static int wm8974_mute(struct snd_soc_dai *dai, int mute)
+ {
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+ if (mute)
+ snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+ else
+ snd_soc_write(codec, WM8974_DAC, mute_reg);
+ return 0;
+ }
projects / GitHub / moto-9609 / android_kernel_motorola_exynos9610.git / commitdiff