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[alsa-devel] [PATCH 06/10] asoc: fsl - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the Freescale PPC platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_dma.h | 2 +- sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++------------ sound/soc/fsl/fsl_ssi.h | 4 ++-- sound/soc/fsl/mpc8610_hpcd.c | 4 ++-- 5 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 78de716..da2bc59 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -282,7 +282,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * once for each .dai_link in the machine driver's snd_soc_machine * structure. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { static u64 fsl_dma_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 430a6ce..385d4a4 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,7 +126,7 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_cpu_dai object +/* DMA information needed to create a snd_soc_dai object * * ssi_stx_phys: bus address of SSI STX register to use * ssi_srx_phys: bus address of SSI SRX register to use diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f588545..71bff33 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -82,7 +82,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; - struct snd_soc_cpu_dai cpu_dai; + struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; struct { @@ -479,7 +479,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) * @freq: the frequency of the given clock ID, currently ignored * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) */ -static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -497,7 +497,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, * * @format: one of SND_SOC_DAIFMT_xxx */ -static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -505,7 +505,7 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ -static struct snd_soc_cpu_dai fsl_ssi_dai_template = { +static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -569,15 +569,15 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, } /** - * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure + * fsl_ssi_create_dai: create a snd_soc_dai structure * - * This function is called by the machine driver to create a snd_soc_cpu_dai + * This function is called by the machine driver to create a snd_soc_dai * structure. The function creates an ssi_private object, which contains - * the snd_soc_cpu_dai. It also creates the sysfs statistics device. + * the snd_soc_dai. It also creates the sysfs statistics device. */ -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) { - struct snd_soc_cpu_dai *fsl_ssi_dai; + struct snd_soc_dai *fsl_ssi_dai; struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr; @@ -588,7 +588,7 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) return NULL; } memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, - sizeof(struct snd_soc_cpu_dai)); + sizeof(struct snd_soc_dai)); fsl_ssi_dai = &ssi_private->cpu_dai; dev_attr = &ssi_private->dev_attr; @@ -623,11 +623,11 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); /** - * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object + * fsl_ssi_destroy_dai: destroy the snd_soc_dai object * * This function undoes the operations of fsl_ssi_create_dai() */ -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) { struct fsl_ssi_private *ssi_private = container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index c5ce88e..83b44d7 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -217,8 +217,8 @@ struct fsl_ssi_info { struct device *dev; }; -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai); #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 8820c3f..59d7e49 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -96,8 +96,8 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct mpc8610_hpcd_data *machine_data = rtd->socdev->dev->platform_data; int ret = 0; -- 1.5.4.2

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[alsa-devel] [PATCH 05/10] asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

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 <lg(a)opensource.wolfsonmicro.com> --- 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(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e4516f3..61fd96c 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 2bf6d69..281aa42 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 469266e..b26003c 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 fc686dd..e9fe30e 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 e73fcfd..9deb8c7 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 0ced49b..adc6cd9 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 954d39b..b1dce5f 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 e600946..d76c079 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 6d5335b..a52d6d9 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 f9d885c..50c603e 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 b549f67..67325fd 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 c862e7b..f5d2e42 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 3ff42ad..369d39c 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 5bcab6a..99f2e3c 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 eb460c9..e23cb09 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 a97a54a..8ef30e6 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 be01a73..8604809 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 95e2a1f..44f5f1f 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 a1371b7..3ecce51 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 bf9f882..6bea574 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 4739011..9fc8edd 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 719105d..d29e8a1 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 a480618..38d1fe0 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 d357b6c..63b8d81 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); -- 1.5.4.2

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[alsa-devel] [PATCH 04/10] asoc: davinci - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the DaVinci platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/davinci/davinci-evm.c | 4 ++-- sound/soc/davinci/davinci-i2s.c | 14 +++++++------- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 091eae3..4249e6a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,8 +33,8 @@ static int evm_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_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; /* set codec DAI configuration */ diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c3b545c..5ebf1ff 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -147,7 +147,7 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) static int davinci_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; cpu_dai->dma_data = dev->dma_params[substream->stream]; @@ -155,7 +155,7 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream) return 0; } -static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcbsp_dev *dev = cpu_dai->private_data; @@ -296,11 +296,11 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) } static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -358,11 +358,11 @@ err_release_region: } static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -378,7 +378,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -struct snd_soc_cpu_dai davinci_i2s_dai = { +struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 9592d17..c5b0918 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,6 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H -extern struct snd_soc_cpu_dai davinci_i2s_dai; +extern struct snd_soc_dai davinci_i2s_dai; #endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 6a76927..6a5e56a 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -350,7 +350,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret; -- 1.5.4.2

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[alsa-devel] [PATCH 03/10] asoc: at91 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the AT91 platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at91/at91-pcm.c | 6 +++--- sound/soc/at91/at91-ssc.c | 12 ++++++------ sound/soc/at91/at91-ssc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 8 ++++---- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index ccac6bd..d47492b 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -318,7 +318,7 @@ static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, static u64 at91_pcm_dmamask = 0xffffffff; static int at91_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -367,7 +367,7 @@ static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; @@ -392,7 +392,7 @@ static int at91_pcm_suspend(struct platform_device *pdev, } static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index bc35d00..c3625b6 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -281,7 +281,7 @@ static void at91_ssc_shutdown(struct snd_pcm_substream *substream) /* * Record the SSC system clock rate. */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* @@ -303,7 +303,7 @@ static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record the DAI format for use in hw_params(). */ -static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -315,7 +315,7 @@ static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params(). */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -634,7 +634,7 @@ static int at91_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -662,7 +662,7 @@ static int at91_ssc_suspend(struct platform_device *pdev, } static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -700,7 +700,7 @@ static int at91_ssc_resume(struct platform_device *pdev, #define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_cpu_dai at91_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at91-ssc0", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index b188f97..6b7bf38 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -21,7 +21,7 @@ #define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at91_ssc_dai[]; +extern struct snd_soc_dai at91_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index ad971e7..9d42139 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -53,8 +53,8 @@ static struct clk *pllb_clk; static int eti_b1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* cpu clock is the AT91 master clock sent to the SSC */ @@ -87,8 +87,8 @@ static int eti_b1_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_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; #ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE -- 1.5.4.2

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[alsa-devel] [PATCH 02/10] asoc: at32 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for AT32 platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at32/at32-pcm.c | 6 +++--- sound/soc/at32/at32-ssc.c | 12 ++++++------ sound/soc/at32/at32-ssc.h | 2 +- sound/soc/at32/playpaq_wm8510.c | 6 +++--- 4 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c index cf76e89..435f1da 100644 --- a/sound/soc/at32/at32-pcm.c +++ b/sound/soc/at32/at32-pcm.c @@ -366,7 +366,7 @@ static struct snd_pcm_ops at32_pcm_ops = { static u64 at32_pcm_dmamask = 0xffffffff; static int at32_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -422,7 +422,7 @@ static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; @@ -447,7 +447,7 @@ static int at32_pcm_suspend(struct platform_device *pdev, static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c index 0ca4410..4ef6492 100644 --- a/sound/soc/at32/at32-ssc.c +++ b/sound/soc/at32/at32-ssc.c @@ -336,7 +336,7 @@ static void at32_ssc_shutdown(struct snd_pcm_substream *substream) /* * Set the SSC system clock rate */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* TODO: What the heck do I do here? */ @@ -348,7 +348,7 @@ static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record DAI format for use by hw_params() */ -static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -362,7 +362,7 @@ static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params() */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -670,7 +670,7 @@ static int at32_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; @@ -699,7 +699,7 @@ static int at32_ssc_suspend(struct platform_device *pdev, static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; u32 cr; @@ -746,7 +746,7 @@ static int at32_ssc_resume(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) -struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at32-ssc0", .id = 0, diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h index 3c6901a..3c052db 100644 --- a/sound/soc/at32/at32-ssc.h +++ b/sound/soc/at32/at32-ssc.h @@ -52,7 +52,7 @@ struct at32_ssc_info { #define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at32_ssc_dai[]; +extern struct snd_soc_dai at32_ssc_dai[]; diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index d6b9fd5..18ac7d7 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -82,7 +82,7 @@ static struct clk *_pll0; #if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( struct snd_pcm_hw_params *params, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; @@ -132,8 +132,8 @@ static int playpaq_wm8510_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_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; unsigned int pll_out = 0, bclk = 0, mclk_div = 0; -- 1.5.4.2

1 0

[alsa-devel] Repeated calls to trigger and prepare in ALSA driver
by DSP Tech 07 Jul '08

07 Jul '08

Hi All, I am trying to develop ALSA PCM driver for my hardware board which has I2S-DMA interface. I ran a OSS test application. I find that trigger() and prepare() calls are being called repeatedly. Consequently, the DMA tries to reset and start playback for every buffer. Have any of you faced this problem? Any clue in this matter would be helpful. Thanks _________________________________________________________________ Tried the new MSN Messenger? It’s cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

2 1

[alsa-devel] ASOC: codec_reg_show and variable length registers
by Jon Smirl 07 Jul '08

07 Jul '08

codec_reg_show() can't handle my TAS5504 codec. The codec has 104 (sparse out of 256) registers with variable lengths from 1 to 32 bytes. Should the register cache really be part of the core? The core could still make the sysfs entries and use a callback to print them. reg_cache_size/reg_cache_step would then become private. It's the sprintf that is the problem. I need to push that into my driver so that I can print out variable lengths. Or we need to make the core use an indirect table that encodes the lengths. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] HDA Intel ALC885 - Need new or edited model option
by Constrabus Gegami 06 Jul '08

06 Jul '08

I need a new or edited model option for this card: HDA Intel card in Alienware m15x laptop That uses codec: ALC885 Detailed specifications on the card: http://www.alienware.com/Configurator_Pages/Details_Pages/Details_Page.aspx… alsa-info.sh genterated output: http://pastebin.ca/1061774 (i switched back to no model options before i ran the script, i dont know if it leaves behind anything or if it even matters) Jack Observations: -1x auto-sensing headphone jack -1x auto-sensing microphone jack All current available model options either present me with: -working switchable internal speakers with non working headphone jack -working switchable headphone jack with non working internal speakers -non working headphone jack and internal speakers -working non switchable internal speakers with non working heaphone -working non switchable headphones with non working speakers -no working sound What is needed: - working switchable internal speakers with working headphone and mic jack (preferably switchable) Distro: Gentoo amd64 using alsa driver package (not in kernel modules) I hope that this will benefit others as well as I. If any more information is needed, please email me. -- --Constrabus

1 0

[alsa-devel] [PATCH] ASoC: Fix warning from strict_strtoul()
by Mark Brown 06 Jul '08

06 Jul '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9e83357..456d78a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -808,7 +808,11 @@ static ssize_t dapm_pop_time_store(struct device *dev, const char *buf, size_t count) { - if (strict_strtoul(buf, 10, &pop_time) < 0) + unsigned long val; + + if (strict_strtoul(buf, 10, &val) >= 0) + pop_time = val; + else printk(KERN_ERR "Unable to parse pop_time setting\n"); return count; -- 1.5.6

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[alsa-devel] EMU10K1 Problem. Im stumped.
by Travis Place 05 Jul '08

05 Jul '08

Just had a user on #alsa with an EMU10K1 problem. To summarize, the card shows up in linux (but doesnt work), yet the card works perfectly in windows. The correct modules are loading, but ALSA doesnt see the card. It had the user unload/reload snd-emu10k1 and check dmesg output, and the relevant result is as follows. EMU10K1_Audigy: probe of 0000:00:0a.0 failed with error -16 There complete alsa-info.sh output can be found at http://pastebin.ca/1062377 I have had the user run this both with and without the onboard AC97 card enabled (just to be sure). What seems odd to me, is that the card is at 00:0a.0 and the SB Live revision is showing as (rev 0a). Just doesnt look right to me.. The card is an Creative SBLive! 'SB0220'. I have one of these exact cards here, if its needed for testing, but its never caused me any issues. Anyway, any help would be appreciated. Thanks, Travis Place (wishie)

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