From b7482f52789266e2548be5d0f6420c9fc12428d8 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Wed, 28 May 2008 17:58:06 +0100 Subject: [ALSA] ASoC: Add UDA1380 driver The UDA1380 codec is used by the HTC Magician and a number of Samsung reference boards. This driver has had a long out of tree history, having originally been written by Giorgio Padrin and converted to ASoC by Richard Purdie. Since conversion to ASoC most of the work on the driver has been done by Philipp Zabel with some review and updates for new APIs by Liam Girdwood and Mark Brown. Signed-off-by: Richard Purdie Signed-off-by: Philipp Zabel Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/uda1380.h | 89 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 89 insertions(+) create mode 100644 sound/soc/codecs/uda1380.h (limited to 'sound/soc/codecs/uda1380.h') diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h new file mode 100644 index 00000000000..f9d885c8bf0 --- /dev/null +++ b/sound/soc/codecs/uda1380.h @@ -0,0 +1,89 @@ +/* + * Audio support for Philips UDA1380 + * + * 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. + * + * Copyright (c) 2005 Giorgio Padrin + */ + +#ifndef _UDA1380_H +#define _UDA1380_H + +#define UDA1380_CLK 0x00 +#define UDA1380_IFACE 0x01 +#define UDA1380_PM 0x02 +#define UDA1380_AMIX 0x03 +#define UDA1380_HP 0x04 +#define UDA1380_MVOL 0x10 +#define UDA1380_MIXVOL 0x11 +#define UDA1380_MODE 0x12 +#define UDA1380_DEEMP 0x13 +#define UDA1380_MIXER 0x14 +#define UDA1380_INTSTAT 0x18 +#define UDA1380_DEC 0x20 +#define UDA1380_PGA 0x21 +#define UDA1380_ADC 0x22 +#define UDA1380_AGC 0x23 +#define UDA1380_DECSTAT 0x28 +#define UDA1380_RESET 0x7f + +#define UDA1380_CACHEREGNUM 0x24 + +/* Register flags */ +#define R00_EN_ADC 0x0800 +#define R00_EN_DEC 0x0400 +#define R00_EN_DAC 0x0200 +#define R00_EN_INT 0x0100 +#define R00_DAC_CLK 0x0010 +#define R01_SFORI_I2S 0x0000 +#define R01_SFORI_LSB16 0x0100 +#define R01_SFORI_LSB18 0x0200 +#define R01_SFORI_LSB20 0x0300 +#define R01_SFORI_MSB 0x0500 +#define R01_SFORI_MASK 0x0700 +#define R01_SFORO_I2S 0x0000 +#define R01_SFORO_LSB16 0x0001 +#define R01_SFORO_LSB18 0x0002 +#define R01_SFORO_LSB20 0x0003 +#define R01_SFORO_LSB24 0x0004 +#define R01_SFORO_MSB 0x0005 +#define R01_SFORO_MASK 0x0007 +#define R01_SEL_SOURCE 0x0040 +#define R01_SIM 0x0010 +#define R02_PON_PLL 0x8000 +#define R02_PON_HP 0x2000 +#define R02_PON_DAC 0x0400 +#define R02_PON_BIAS 0x0100 +#define R02_EN_AVC 0x0080 +#define R02_PON_AVC 0x0040 +#define R02_PON_LNA 0x0010 +#define R02_PON_PGAL 0x0008 +#define R02_PON_ADCL 0x0004 +#define R02_PON_PGAR 0x0002 +#define R02_PON_ADCR 0x0001 +#define R13_MTM 0x4000 +#define R14_SILENCE 0x0080 +#define R14_SDET_ON 0x0040 +#define R21_MT_ADC 0x8000 +#define R22_SEL_LNA 0x0008 +#define R22_SEL_MIC 0x0004 +#define R22_SKIP_DCFIL 0x0002 +#define R23_AGC_EN 0x0001 + +struct uda1380_setup_data { + unsigned short i2c_address; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ +#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ +#define UDA1380_DAI_CAPTURE 2 /* capture DAI */ + +extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_codec_device soc_codec_dev_uda1380; + +#endif /* _UDA1380_H */ -- cgit v1.2.3-70-g09d2 From e550e17ffeb8cf8db27724eaf2ad05f77388afb9 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:07:52 +0100 Subject: ALSA: asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the codec drivers. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ac97.h | 2 +- sound/soc/codecs/ak4535.c | 8 ++++---- sound/soc/codecs/ak4535.h | 2 +- sound/soc/codecs/cs4270.c | 8 ++++---- sound/soc/codecs/cs4270.h | 2 +- sound/soc/codecs/tlv320aic3x.c | 8 ++++---- sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/codecs/uda1380.c | 6 +++--- sound/soc/codecs/uda1380.h | 2 +- sound/soc/codecs/wm8510.c | 10 +++++----- sound/soc/codecs/wm8510.h | 2 +- sound/soc/codecs/wm8731.c | 8 ++++---- sound/soc/codecs/wm8731.h | 2 +- sound/soc/codecs/wm8750.c | 8 ++++---- sound/soc/codecs/wm8750.h | 2 +- sound/soc/codecs/wm8753.c | 28 ++++++++++++++-------------- sound/soc/codecs/wm8753.h | 2 +- sound/soc/codecs/wm8990.c | 12 ++++++------ sound/soc/codecs/wm8990.h | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9712.h | 2 +- sound/soc/codecs/wm9713.c | 10 +++++----- sound/soc/codecs/wm9713.h | 2 +- 24 files changed, 67 insertions(+), 67 deletions(-) (limited to 'sound/soc/codecs/uda1380.h') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e4516f3ce64..61fd96ca7bc 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -41,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ac97_dai = { +struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97, .playback = { diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h index 2bf6d69fd06..281aa42e2bb 100644 --- a/sound/soc/codecs/ac97.h +++ b/sound/soc/codecs/ac97.h @@ -14,6 +14,6 @@ #define __LINUX_SND_SOC_AC97_H extern struct snd_soc_codec_device soc_codec_dev_ac97; -extern struct snd_soc_codec_dai ac97_dai; +extern struct snd_soc_dai ac97_dai; #endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 469266e881d..b26003c4f3e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -329,7 +329,7 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec) return 0; } -static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -369,7 +369,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -394,7 +394,7 @@ static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; @@ -436,7 +436,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ak4535_dai = { +struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h index fc686ddf753..e9fe30e2c05 100644 --- a/sound/soc/codecs/ak4535.h +++ b/sound/soc/codecs/ak4535.h @@ -40,7 +40,7 @@ struct ak4535_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_dai ak4535_dai; extern struct snd_soc_codec_device soc_codec_dev_ak4535; #endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e73fcfd9f5c..9deb8c74fdf 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -201,7 +201,7 @@ static struct { * driver what the input settings can be. This would need to be implemented * for stand-alone mode to work. */ -static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, * data for playback only, but ASoC currently does not support different * formats for playback vs. record. */ -static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; @@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * board does not have the MUTEA or MUTEB pins connected to such circuitry, * then this function will do nothing. */ -static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +static int cs4270_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; int reg6; @@ -667,7 +667,7 @@ error: #endif /* USE_I2C*/ -struct snd_soc_codec_dai cs4270_dai = { +struct snd_soc_dai cs4270_dai = { .name = "CS4270", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h index 0ced49b7804..adc6cd9667d 100644 --- a/sound/soc/codecs/cs4270.h +++ b/sound/soc/codecs/cs4270.h @@ -16,7 +16,7 @@ * The ASoC codec DAI structure for the CS4270. Assign this structure to * the .codec_dai field of your machine driver's snd_soc_dai_link structure. */ -extern struct snd_soc_codec_dai cs4270_dai; +extern struct snd_soc_dai cs4270_dai; /* * The ASoC codec device structure for the CS4270. Assign this structure diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 954d39b7c04..b1dce5f459d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -814,7 +814,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, return 0; } -static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) +static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; @@ -831,7 +831,7 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -841,7 +841,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -990,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai aic3x_dai = { +struct snd_soc_dai aic3x_dai = { .name = "aic3x", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e6009461063..d76c079b86e 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -228,7 +228,7 @@ struct aic3x_setup_data { unsigned int gpio_func[2]; }; -extern struct snd_soc_codec_dai aic3x_dai; +extern struct snd_soc_dai aic3x_dai; extern struct snd_soc_codec_device soc_codec_dev_aic3x; #endif /* _AIC3X_H */ diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6d5335b14d5..a52d6d9e007 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -372,7 +372,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) return 0; } -static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -499,7 +499,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) uda1380_write(codec, UDA1380_CLK, clk); } -static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) +static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; @@ -542,7 +542,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai uda1380_dai[] = { +struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", .playback = { diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h index f9d885c8bf0..50c603e2c9f 100644 --- a/sound/soc/codecs/uda1380.h +++ b/sound/soc/codecs/uda1380.h @@ -83,7 +83,7 @@ struct uda1380_setup_data { #define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ #define UDA1380_DAI_CAPTURE 2 /* capture DAI */ -extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_dai uda1380_dai[3]; extern struct snd_soc_codec_device soc_codec_dev_uda1380; #endif /* _UDA1380_H */ diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index b549f6753ab..67325fd9544 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -332,7 +332,7 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -368,7 +368,7 @@ static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Configure WM8510 clock dividers. */ -static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -402,7 +402,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -510,7 +510,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8510_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; @@ -554,7 +554,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai wm8510_dai = { +struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c862e7b7d53..f5d2e42eb3f 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -97,7 +97,7 @@ struct wm8510_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_dai wm8510_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8510; #endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 3ff42ad65ed..369d39c3f74 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -318,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream) } } -static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; @@ -330,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -349,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, } -static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -443,7 +443,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8731_dai = { +struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 5bcab6a7afb..99f2e3c60e3 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -38,7 +38,7 @@ struct wm8731_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8731_dai; +extern struct snd_soc_dai wm8731_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8731; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index eb460c9aa63..e23cb09f0d1 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -536,7 +536,7 @@ static inline int get_coeff(int mclk, int rate) return -EINVAL; } -static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -554,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return -EINVAL; } -static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -647,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; @@ -692,7 +692,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8750_dai = { +struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h index a97a54a6348..8ef30e628b2 100644 --- a/sound/soc/codecs/wm8750.h +++ b/sound/soc/codecs/wm8750.h @@ -61,7 +61,7 @@ struct wm8750_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8750_dai; +extern struct snd_soc_dai wm8750_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8750; #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be01a738f18..8604809f0c3 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -740,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; @@ -863,7 +863,7 @@ static int get_coeff(int mclk, int rate) /* * Clock after PLL and dividers */ -static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -890,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -960,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1026,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1054,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1087,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1195,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1210,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_pcm_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) @@ -1218,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1233,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1250,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7; @@ -1316,7 +1316,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static const struct snd_soc_codec_dai wm8753_all_dai[] = { +static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", .id = 1, @@ -1456,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = { }, }; -struct snd_soc_codec_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[2]; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 95e2a1f5316..44f5f1ff0cc 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -120,7 +120,7 @@ struct wm8753_setup_data { #define WM8753_DAI_HIFI 0 #define WM8753_DAI_VOICE 1 -extern struct snd_soc_codec_dai wm8753_dai[2]; +extern struct snd_soc_dai wm8753_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm8753; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index a1371b73ba7..3ecce5168e9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1029,7 +1029,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg; @@ -1065,7 +1065,7 @@ static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Clock after PLL and dividers */ -static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1078,7 +1078,7 @@ static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1131,7 +1131,7 @@ static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1196,7 +1196,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8990_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 val; @@ -1329,7 +1329,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ -struct snd_soc_codec_dai wm8990_dai = { +struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", .id = 1, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index bf9f8823dfc..6bea5748528 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -825,7 +825,7 @@ struct wm8990_setup_data { #define WM8990_ADCCLK_DIV 2 #define WM8990_BCLK_DIV 3 -extern struct snd_soc_codec_dai wm8990_dai; +extern struct snd_soc_dai wm8990_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8990; #endif /* __WM8990REGISTERDEFS_H__ */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 47390113bd0..9fc8edd8222 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -532,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai wm9712_dai[] = { +struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h index 719105d61e6..d29e8a18ca6 100644 --- a/sound/soc/codecs/wm9712.h +++ b/sound/soc/codecs/wm9712.h @@ -8,7 +8,7 @@ #define WM9712_DAI_AC97_HIFI 0 #define WM9712_DAI_AC97_AUX 1 -extern struct snd_soc_codec_dai wm9712_dai[2]; +extern struct snd_soc_dai wm9712_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm9712; #endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a4806189044..38d1fe0971f 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -789,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -800,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, * Tristate the PCM DAI lines, tristate can be disabled by calling * wm9713_set_dai_fmt() */ -static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; @@ -816,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, * Configure WM9713 clock dividers. * Voice DAC needs 256 FS */ -static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -858,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1018,7 +1018,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) -struct snd_soc_codec_dai wm9713_dai[] = { +struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h index d357b6c8134..63b8d81756e 100644 --- a/sound/soc/codecs/wm9713.h +++ b/sound/soc/codecs/wm9713.h @@ -46,7 +46,7 @@ #define WM9713_DAI_PCM_VOICE 2 extern struct snd_soc_codec_device soc_codec_dev_wm9713; -extern struct snd_soc_codec_dai wm9713_dai[3]; +extern struct snd_soc_dai wm9713_dai[3]; int wm9713_reset(struct snd_soc_codec *codec, int try_warm); -- cgit v1.2.3-70-g09d2