diff options
Diffstat (limited to 'Documentation/sound/alsa')
| -rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 20 | ||||
| -rw-r--r-- | Documentation/sound/alsa/HD-Audio-Models.txt | 2 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/codec.txt | 45 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/machine.txt | 38 | ||||
| -rw-r--r-- | Documentation/sound/alsa/soc/platform.txt | 12 |
5 files changed, 50 insertions, 67 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index d0eb696..3c1eddd 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -974,13 +974,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. See hdspm.txt for details. - Module snd-hifier - ----------------- - - Module for the MediaTek/TempoTec HiFier Fantasia sound card. - - This module supports autoprobe and multiple cards. - Module snd-ice1712 ------------------ @@ -1531,15 +1524,20 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-oxygen ----------------- - Module for sound cards based on the C-Media CMI8788 chip: + Module for sound cards based on the C-Media CMI8786/8787/8788 chip: * Asound A-8788 + * Asus Xonar DG * AuzenTech X-Meridian + * AuzenTech X-Meridian 2G * Bgears b-Enspirer * Club3D Theatron DTS * HT-Omega Claro (plus) * HT-Omega Claro halo (XT) + * Kuroutoshikou CMI8787-HG2PCI * Razer Barracuda AC-1 * Sondigo Inferno + * TempoTec HiFier Fantasia + * TempoTec HiFier Serenade This module supports autoprobe and multiple cards. @@ -2006,9 +2004,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-virtuoso ------------------- - Module for sound cards based on the Asus AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, DS, HDAV1.3 (Deluxe), Essence ST - (Deluxe) and Essence STX. + Module for sound cards based on the Asus AV66/AV100/AV200 chips, + i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), Essence STX, + HDAV1.3 (Deluxe), and HDAV1.3 Slim. This module supports autoprobe and multiple cards. diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 37c6aad..0caf77e 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -149,7 +149,6 @@ ALC882/883/885/888/889 acer-aspire-7730g Acer Aspire 7730G acer-aspire-8930g Acer Aspire 8930G medion Medion Laptops - medion-md2 Medion MD2 targa-dig Targa/MSI targa-2ch-dig Targa/MSI with 2-channel targa-8ch-dig Targa/MSI with 8-channel (MSI GX620) @@ -297,6 +296,7 @@ Conexant 5066 ============= laptop Basic Laptop config (default) hp-laptop HP laptops, e g G60 + asus Asus K52JU, Lenovo G560 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index 37ba3a7..bce23a4 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -27,42 +27,38 @@ ASoC Codec driver breakdown 1 - Codec DAI and PCM configuration ----------------------------------- -Each codec driver must have a struct snd_soc_codec_dai to define its DAI and +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. -struct snd_soc_codec_dai wm8731_dai = { - .name = "WM8731", - /* playback capabilities */ +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 capabilities */ .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - /* pcm operations - see section 4 below */ - .ops = { - .prepare = wm8731_pcm_prepare, - .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, - }, - /* DAI operations - see DAI.txt */ - .dai_ops = { - .digital_mute = wm8731_mute, - .set_sysclk = wm8731_set_dai_sysclk, - .set_fmt = wm8731_set_dai_fmt, - } + .ops = &wm8731_dai_ops, + .symmetric_rates = 1, }; -EXPORT_SYMBOL_GPL(wm8731_dai); 2 - Codec control IO @@ -186,13 +182,14 @@ when the mute is applied or freed. i.e. -static int wm8974_mute(struct snd_soc_codec *codec, - struct snd_soc_codec_dai *dai, int mute) +static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; - if(mute) - wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); + 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 - wm8974_write(codec, WM8974_DAC, mute_reg); + snd_soc_write(codec, WM8974_DAC, mute_reg); return 0; } diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 2524c75..3e2ec9c 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -12,6 +12,8 @@ the following struct:- struct snd_soc_card { char *name; + ... + int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -22,12 +24,13 @@ struct snd_soc_card { int (*resume_pre)(struct platform_device *pdev); int (*resume_post)(struct platform_device *pdev); - /* machine stream operations */ - struct snd_soc_ops *ops; + ... /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + + ... }; probe()/remove() @@ -42,11 +45,6 @@ of any machine audio tasks that have to be done before or after the codec, DAIs and DMA is suspended and resumed. Optional. -Machine operations ------------------- -The machine specific audio operations can be set here. Again this is optional. - - Machine DAI Configuration ------------------------- The machine DAI configuration glues all the codec and CPU DAIs together. It can @@ -61,8 +59,10 @@ struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. static struct snd_soc_dai_link corgi_dai = { .name = "WM8731", .stream_name = "WM8731", - .cpu_dai = &pxa_i2s_dai, - .codec_dai = &wm8731_dai, + .cpu_dai_name = "pxa-is2-dai", + .codec_dai_name = "wm8731-hifi", + .platform_name = "pxa-pcm-audio", + .codec_name = "wm8713-codec.0-001a", .init = corgi_wm8731_init, .ops = &corgi_ops, }; @@ -77,26 +77,6 @@ static struct snd_soc_card snd_soc_corgi = { }; -Machine Audio Subsystem ------------------------ - -The machine soc device glues the platform, machine and codec driver together. -Private data can also be set here. e.g. - -/* corgi audio private data */ -static struct wm8731_setup_data corgi_wm8731_setup = { - .i2c_address = 0x1b, -}; - -/* corgi audio subsystem */ -static struct snd_soc_device corgi_snd_devdata = { - .machine = &snd_soc_corgi, - .platform = &pxa2xx_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &corgi_wm8731_setup, -}; - - Machine Power Map ----------------- diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index 06d8359..d57efad 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -20,9 +20,10 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -The platform driver exports its DMA functionality via struct snd_soc_platform:- +The platform driver exports its DMA functionality via struct +snd_soc_platform_driver:- -struct snd_soc_platform { +struct snd_soc_platform_driver { char *name; int (*probe)(struct platform_device *pdev); @@ -34,6 +35,13 @@ struct snd_soc_platform { int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); + /* + * For platform caused delay reporting. + * Optional. + */ + snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, + struct snd_soc_dai *); + /* platform stream ops */ struct snd_pcm_ops *pcm_ops; }; |
