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[alsa-devel] soc-of.h file
by dinesh 08 Jul '08

08 Jul '08

I need to use function of_snd_soc_register_platform of soc-of.o i got soc-of.c file from patch [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers with but i am not able to find soc-of.h file please provide this to me. Regards, Dinesh Dua

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[alsa-devel] [PATCH] asoc: at32 - DAI struct merge and enable_pin() change.
by Liam Girdwood 08 Jul '08

08 Jul '08

This adds support for the recent DAI struct merge and new DAPM pin API. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at32/playpaq_wm8510.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index fd62f25..fee5f8e 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -341,15 +341,15 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) } - /* always connected endpoints */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); - snd_soc_dapm_sync_endpoints(codec); + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); /* Make CSB show PLL rate */ - codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, WM8510_OPCLKDIV_1 | 4); return 0; -- 1.5.4.2

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[alsa-devel] [PATCH] asoc: n810 - fix build error.
by Liam Girdwood 08 Jul '08

08 Jul '08

This patch adds a missing " that was recently introduced (removed) in the DAI struct merge patch series. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/omap/n810.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index e53c055..02cec96 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -63,7 +63,7 @@ static void n810_ext_control(struct snd_soc_codec *codec) if (n810_dmic_func) snd_soc_dapm_enable_pin(codec, "DMic"); else - snd_soc_dapm_disable_pin(codec, "DMic); + snd_soc_dapm_disable_pin(codec, "DMic"); snd_soc_dapm_sync(codec); } -- 1.5.4.2

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[alsa-devel] [PATCH 02/02] asoc: machines - add Digital Audio Interface (DAI) control functions.
by Liam Girdwood 08 Jul '08

08 Jul '08

This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct within the machine drivers. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at32/playpaq_wm8510.c | 16 ++++----- sound/soc/at91/eti_b1_wm8731.c | 18 +++++----- sound/soc/davinci/davinci-evm.c | 6 ++-- sound/soc/fsl/mpc8610_hpcd.c | 58 +++++++++++++++--------------------- sound/soc/omap/n810.c | 6 ++-- sound/soc/pxa/corgi.c | 8 ++-- sound/soc/pxa/poodle.c | 8 ++-- sound/soc/pxa/spitz.c | 8 ++-- sound/soc/s3c24xx/neo1973_wm8753.c | 26 ++++++++-------- 9 files changed, 71 insertions(+), 83 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 18ac7d7..fd62f25 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -210,14 +210,14 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, /* * set CPU and CODEC DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CODEC DAI format (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CPU DAI format (%d)\n", @@ -233,14 +233,13 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", cd.cmr_div, cd.period); - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, - AT32_SSC_CMR_DIV, cd.cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, cd.period); if (ret < 0) { pr_warning("playpaq_wm8510: " @@ -260,7 +259,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); if (ret < 0) { pr_warning ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", @@ -270,7 +269,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_get_rate(CODEC_CLK), pll_out); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", @@ -279,8 +278,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, } - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, - WM8510_MCLKDIV, mclk_div); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", ret); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 9d42139..d532de9 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -58,13 +58,13 @@ static int eti_b1_startup(struct snd_pcm_substream *substream) int ret; /* cpu clock is the AT91 master clock sent to the SSC */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK, + ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, 60000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -96,13 +96,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, int cmr_div, period; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; @@ -141,17 +141,17 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, } /* set the MCK divider for BCLK */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); if (ret < 0) return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* set the BCLK divider for DACLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_TCMR_PERIOD, period); } else { /* set the BCLK divider for ADCLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_RCMR_PERIOD, period); } if (ret < 0) @@ -163,13 +163,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4249e6a..5e2c306 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -38,19 +38,19 @@ static int evm_hw_params(struct snd_pcm_substream *substream, int ret = 0; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; /* set the codec system clock */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 59d7e49..4bdc9d8 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -103,55 +103,45 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the CPU driver what the serial protocol is. */ - if (cpu_dai->dai_ops.set_fmt) { - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; } /* Tell the codec driver what the serial protocol is. */ - if (codec_dai->dai_ops.set_fmt) { - ret = codec_dai->dai_ops.set_fmt(codec_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; } /* * Tell the CPU driver what the clock frequency is, and whether it's a * slave or master. */ - if (cpu_dai->dai_ops.set_sysclk) { - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, - machine_data->clk_frequency, - machine_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; } /* * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - if (codec_dai->dai_ops.set_sysclk) { - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; } return 0; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d1233c0..e53c055 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -91,7 +91,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, int err; /* Set codec DAI configuration */ - err = codec_dai->dai_ops.set_fmt(codec_dai, + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -99,7 +99,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set cpu DAI configuration */ - err = cpu_dai->dai_ops.set_fmt(cpu_dai, + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -107,7 +107,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set the codec system clock for DAC and ADC */ - err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000, + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); return err; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 782afbf..c029446 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -143,25 +143,25 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index ce25b6b..65a4e9a 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -122,25 +122,25 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index fd1abc7..6438579 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -141,25 +141,25 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 51a4ce3..4d7a9aa 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -108,44 +108,44 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set MCLK division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS); if (ret < 0) return ret; /* set codec BCLK division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); if (ret < 0) return ret; /* set prescaler division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; /* codec PLL input is PCLK/4 */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -161,7 +161,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); } /* @@ -194,24 +194,24 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, /* todo: gg check mode (DSP_B) against CSR datasheet */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set codec PCM division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); if (ret < 0) return ret; /* configue and enable PLL for 12.288MHz output */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -227,7 +227,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { -- 1.5.4.2

1 0

[alsa-devel] [PATCH 01/02] asoc: core - add Digital Audio Interface (DAI) control functions.
by Liam Girdwood 08 Jul '08

08 Jul '08

This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 21 ++++++++ sound/soc/soc-core.c | 140 +++++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 153 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 778e57e..1890d87 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,27 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + /* *Controls */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d626b4..83f1190 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -434,8 +434,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) else { codec_dai->pop_wait = 0; cancel_delayed_work(&socdev->delayed_work); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } } else { /* no delayed work - do we need to power up codec */ @@ -454,8 +453,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_START); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } else { /* codec already powered - power on widgets */ @@ -467,8 +465,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + + snd_soc_dai_digital_mute(codec_dai, 0); } } @@ -566,8 +564,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* apply codec digital mute */ - if (!codec->active && codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 1); + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ if (machine->ops && machine->ops->hw_free) @@ -1703,6 +1701,132 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); +/** + * snd_soc_dai_set_sysclk - configure DAI system or master clock. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the DAI master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); + +/** + * snd_soc_dai_set_clkdiv - configure DAI clock dividers. + * @dai: DAI + * @clk_id: DAI specific clock divider ID + * @div: new clock divisor. + * + * Configures the clock dividers. This is used to derive the best DAI bit and + * frame clocks from the system or master clock. It's best to set the DAI bit + * and frame clocks as low as possible to save system power. + */ +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div) +{ + if (dai->dai_ops.set_clkdiv) + return dai->dai_ops.set_clkdiv(dai, div_id, div); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); + +/** + * snd_soc_dai_set_pll - configure DAI PLL. + * @dai: DAI + * @pll_id: DAI specific PLL ID + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + if (dai->dai_ops.set_pll) + return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); + +/** + * snd_soc_dai_set_fmt - configure DAI hardware audio format. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @fmt: SND_SOC_DAIFMT_ format value. + * + * Configures the DAI hardware format and clocking. + */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->dai_ops.set_fmt) + return dai->dai_ops.set_fmt(dai, fmt); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @mask: DAI specific mask representing used slots. + * @slots: Number of slots in use. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tdm_slot(dai, mask, slots); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); + +/** + * snd_soc_dai_set_tristate - configure DAI system or master clock. + * @dai: DAI + * @tristate: tristate enable + * + * Tristates the DAI so that others can use it. + */ +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tristate(dai, tristate); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); + +/** + * snd_soc_dai_digital_mute - configure DAI system or master clock. + * @dai: DAI + * @mute: mute enable + * + * Mutes the DAI DAC. + */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +{ + if (dai->dai_ops.digital_mute) + return dai->dai_ops.digital_mute(dai, mute); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -- 1.5.4.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: Switch DAPM to use of standard DEBUG macro
by Mark Brown 08 Jul '08

08 Jul '08

DAPM contains debug output controlled by a DAPM_DEBUG macro. Change this to be controlled by the standard DEBUG, dropping the custom dbg() macro as we go. Also fix the error printed when configuring an unknown pin to be an unconditionally displayed error rather than debug output. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 22 ++++++++++------------ 1 files changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 94296b5..d18ebc6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -45,13 +45,10 @@ #include <sound/initval.h> /* debug */ -#define DAPM_DEBUG 0 -#if DAPM_DEBUG +#ifdef DEBUG #define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -#define dbg(format, arg...) printk(format, ## arg) #else #define dump_dapm(codec, action) -#define dbg(format, arg...) #endif /* dapm power sequences - make this per codec in the future */ @@ -233,7 +230,8 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(); } - dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); + pr_debug("reg %x old %x new %x change %d\n", widget->reg, + old, new, change); return change; } @@ -591,8 +589,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* call any power change event handlers */ if (power_change) { if (w->event) { - dbg("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", w->name, w->event_flags); if (power) { /* power up event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { @@ -634,7 +632,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return ret; } -#if DAPM_DEBUG +#ifdef DEBUG static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) { struct snd_soc_dapm_widget *w; @@ -887,13 +885,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, list_for_each_entry(w, &codec->dapm_widgets, list) { if (!strcmp(w->name, pin)) { - dbg("dapm: %s: pin %s\n", codec->name, pin); + pr_debug("dapm: %s: pin %s\n", codec->name, pin); w->connected = status; return 0; } } - dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); return -EINVAL; } @@ -1397,8 +1395,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, { if (!w->sname) continue; - dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname, - stream, event); + pr_debug("widget %s\n %s stream %s event %d\n", + w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { case SND_SOC_DAPM_STREAM_START: -- 1.5.6

2 2

[alsa-devel] [PATCH] hda: removed redundant gpio_mask
by Matthew Ranostay 08 Jul '08

08 Jul '08

An gpio_mask value was defined twice needlessly. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index c4f3489..a6d1388 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -3669,7 +3669,6 @@ again: /* GPIO0 High = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; - spec->gpio_mask = 0x01; spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids;

1 0

[alsa-devel] [PATCH 01/10] asoc: core - 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 in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 71 ++++++++++++++++--------------------------------- sound/soc/soc-core.c | 50 +++++++++++++++++----------------- 2 files changed, 48 insertions(+), 73 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 340223a..778e57e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -221,8 +221,7 @@ struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; -struct snd_soc_codec_dai; -struct snd_soc_cpu_dai; +struct snd_soc_dai; struct snd_soc_codec; struct snd_soc_machine_config; struct soc_enum; @@ -317,50 +316,24 @@ struct snd_soc_ops { /* ASoC DAI ops */ struct snd_soc_dai_ops { /* DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, + int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec_dai *codec_dai, + int (*set_pll)(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, - int div_id, int div); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); /* DAI format configuration */ - int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_codec_dai *, int tristate); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* digital mute */ - int (*digital_mute)(struct snd_soc_codec_dai *, int mute); + int (*digital_mute)(struct snd_soc_dai *dai, int mute); }; -/* SoC Codec DAI */ -struct snd_soc_codec_dai { - char *name; - int id; - unsigned char type; - - /* DAI capabilities */ - struct snd_soc_pcm_stream playback; - struct snd_soc_pcm_stream capture; - - /* DAI runtime info */ - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; - - /* DAI private data */ - void *private_data; -}; - -/* SoC CPU DAI */ -struct snd_soc_cpu_dai { - +/* SoC DAI (Digital Audio Interface) */ +struct snd_soc_dai { /* DAI description */ char *name; unsigned int id; @@ -368,13 +341,13 @@ struct snd_soc_cpu_dai { /* DAI callbacks */ int (*probe)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); void (*remove)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* ops */ struct snd_soc_ops ops; @@ -386,7 +359,9 @@ struct snd_soc_cpu_dai { /* DAI runtime info */ struct snd_pcm_runtime *runtime; - unsigned char active:1; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; void *dma_data; /* DAI private data */ @@ -428,7 +403,7 @@ struct snd_soc_codec { struct delayed_work delayed_work; /* codec DAI's */ - struct snd_soc_codec_dai *dai; + struct snd_soc_dai *dai; unsigned int num_dai; }; @@ -447,12 +422,12 @@ struct snd_soc_platform { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, + int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); @@ -466,8 +441,8 @@ struct snd_soc_dai_link { char *stream_name; /* Stream name */ /* DAI */ - struct snd_soc_codec_dai *codec_dai; - struct snd_soc_cpu_dai *cpu_dai; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bdbbc6a..4d626b4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -134,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -272,7 +272,7 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, delayed_work.work); struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; mutex_lock(&pcm_mutex); @@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; int ret = 0; @@ -489,8 +489,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -559,8 +559,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -594,8 +594,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->ops.trigger) { @@ -651,7 +651,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); } @@ -664,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) machine->suspend_pre(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) @@ -690,7 +690,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) codec_dev->suspend(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); } @@ -726,7 +726,7 @@ static void soc_resume_deferred(struct work_struct *work) machine->resume_pre(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); } @@ -747,13 +747,13 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) @@ -803,7 +803,7 @@ static int soc_probe(struct platform_device *pdev) } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) @@ -838,7 +838,7 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -867,7 +867,7 @@ static int soc_remove(struct platform_device *pdev) codec_dev->remove(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -895,8 +895,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = dai_link->codec_dai; + struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; struct snd_soc_pcm_runtime *rtd; struct snd_pcm *pcm; char new_name[64]; @@ -1211,7 +1211,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; #ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; #endif -- 1.5.4.2

2 1

[alsa-devel] [PATCH] asoc: core - refactored DAPM pin control API.
by Liam Girdwood 07 Jul '08

07 Jul '08

Refactored snd_soc_dapm_set_endpoint() to snd_soc_dapm_enable_pin() and snd_soc_dapm_disable_pin(). Renamed snd_soc_dapm_sync_endpoints() to snd_soc_dapm_sync(). Renamed snd_soc_dapm_get_endpoint_status() to snd_soc_dapm_get_pin_status(). Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 11 ++-- sound/soc/at91/eti_b1_wm8731.c | 10 ++-- sound/soc/codecs/tlv320aic3x.c | 4 +- sound/soc/davinci/davinci-evm.c | 16 +++--- sound/soc/omap/n810.c | 21 ++++-- sound/soc/pxa/corgi.c | 42 +++++++------ sound/soc/pxa/poodle.c | 24 ++++---- sound/soc/pxa/spitz.c | 62 ++++++++++---------- sound/soc/pxa/tosa.c | 30 +++++---- sound/soc/s3c24xx/neo1973_wm8753.c | 116 ++++++++++++++++++------------------ sound/soc/sh/exmmb-ac97.c | 2 +- sound/soc/sh/exmmb-i2s.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/soc-dapm.c | 81 ++++++++++++++++--------- 14 files changed, 227 insertions(+), 196 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f8223fa..64adbe5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -227,12 +227,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); -/* dapm audio endpoint control */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *pin, int status); -int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, - char *pin); -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec); +/* dapm audio pin control and status */ +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_sync(struct snd_soc_codec *codec); /* dapm widget types */ enum snd_soc_dapm_type { diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 83614dd..c8b6bc2 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -216,14 +216,14 @@ static int eti_b1_wm8731_init(struct snd_soc_codec *codec) snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 528c26a..b735ddb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -29,7 +29,7 @@ * --------------------------------------- * * Hence the machine layer should disable unsupported inputs/outputs by - * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc. + * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc. */ #include <linux/module.h> @@ -206,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, } if (found) - snd_soc_dapm_sync_endpoints(widget->codec); + snd_soc_dapm_sync(widget->codec); } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4c70a0e..091eae3 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -103,17 +103,17 @@ static int evm_aic3x_init(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line In", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c168a64..182d690 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -49,10 +49,17 @@ static int n810_jack_func; static void n810_ext_control(struct snd_soc_codec *codec) { - snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); + if (n810_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + + if (n810_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int n810_startup(struct snd_pcm_substream *substream) @@ -204,9 +211,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { @@ -223,7 +230,7 @@ static int n810_aic33_init(struct snd_soc_codec *codec) /* Set up N810 specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index edeea63..db18ef6 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -50,47 +50,51 @@ static int corgi_spk_func; static void corgi_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic = 0, line = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: - hp = 1; /* set = unmute headphone */ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_MIC: - mic = 1; /* reset = mute headphone */ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - line = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - hs = 1; - mic = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (corgi_spk_func == CORGI_SPK_ON) - spk = 1; - - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic); - snd_soc_dapm_set_endpoint(codec, "Line Jack", line); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int corgi_startup(struct snd_pcm_substream *substream) @@ -285,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { @@ -303,7 +307,7 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) /* Set up corgi specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 810f1fe..36cbf69 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -48,8 +48,6 @@ static int poodle_spk_func; static void poodle_ext_control(struct snd_soc_codec *codec) { - int spk = 0; - /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec) POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); } else { locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 0); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); } - if (poodle_spk_func == POODLE_SPK_ON) - spk = 1; - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); + if (poodle_spk_func == POODLE_SPK_ON) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int poodle_startup(struct snd_pcm_substream *substream) @@ -248,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "MICIN", 1); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { @@ -267,7 +265,7 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 092b5c7..ec18163 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -51,60 +51,60 @@ static int spitz_spk_func; static void spitz_ext_control(struct snd_soc_codec *codec) { if (spitz_spk_func == SPITZ_SPK_ON) - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); else - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0); + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* set up jack connection */ switch (spitz_jack_func) { case SPITZ_HP: /* enable and unmute hp jack, disable mic bias */ - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HP_OFF: /* jack removed, everything off */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int spitz_startup(struct snd_pcm_substream *substream) @@ -291,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONO", 0); + snd_soc_dapm_disable_pin(codec, "RINPUT1"); + snd_soc_dapm_disable_pin(codec, "LINPUT2"); + snd_soc_dapm_disable_pin(codec, "RINPUT2"); + snd_soc_dapm_disable_pin(codec, "LINPUT3"); + snd_soc_dapm_disable_pin(codec, "RINPUT3"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONO"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { @@ -314,7 +314,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 465ff0f..dba7689 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -52,29 +52,31 @@ static int tosa_spk_func; static void tosa_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic_int = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (tosa_jack_func) { case TOSA_HP: - hp = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_MIC_INT: - mic_int = 1; + snd_soc_dapm_enable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_HEADSET: - hs = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (tosa_spk_func == TOSA_SPK_ON) - spk = 1; + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); - snd_soc_dapm_set_endpoint(codec, "Speaker", spk); - snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int tosa_startup(struct snd_pcm_substream *substream) @@ -191,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { @@ -209,7 +211,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) /* set up tosa specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 3485123..f053e85 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -250,77 +250,77 @@ static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) switch (neo1973_scenario) { case NEO_AUDIO_OFF: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; default: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } @@ -511,12 +511,12 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); - snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT4", 0); - snd_soc_dapm_set_endpoint(codec, "LINE1", 0); - snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + snd_soc_dapm_disable_pin(codec, "LOUT2"); + snd_soc_dapm_disable_pin(codec, "ROUT2"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "OUT4"); + snd_soc_dapm_disable_pin(codec, "LINE1"); + snd_soc_dapm_disable_pin(codec, "LINE2"); /* set endpoints to default mode */ @@ -539,7 +539,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) err = snd_soc_dapm_add_routes(codec, dapm_routes, ARRAY_SIZE(dapm_routes)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/sh/exmmb-ac97.c b/sound/soc/sh/exmmb-ac97.c index e8d710f..b0b4a20 100644 --- a/sound/soc/sh/exmmb-ac97.c +++ b/sound/soc/sh/exmmb-ac97.c @@ -46,7 +46,7 @@ static int exm7760_ac97_machine_init(struct snd_soc_codec *codec) { MSG("machine_init() enter\n"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); MSG("machine_init() leave\n"); return 0; diff --git a/sound/soc/sh/exmmb-i2s.c b/sound/soc/sh/exmmb-i2s.c index db1341e..f253f94 100644 --- a/sound/soc/sh/exmmb-i2s.c +++ b/sound/soc/sh/exmmb-i2s.c @@ -116,7 +116,7 @@ static int exm7760_i2s_machine_init(struct snd_soc_codec *codec) cs4251x_gpio_mode(codec, CS4251X_RXP_6, CS4251X_GPIO_MODE_GPOLOW, 0, 0); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 2f91de8..846d1b3 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -25,7 +25,7 @@ extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) { - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9e83357..0813c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -857,8 +857,25 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) } } +static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, + char *pin, int status) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, pin)) { + dbg("dapm: %s: pin %s\n", codec->name, pin); + w->connected = status; + return 0; + } + } + + dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + return -EINVAL; +} + /** - * snd_soc_dapm_sync_endpoints - scan and power dapm paths + * snd_soc_dapm_sync - scan and power dapm paths * @codec: audio codec * * Walks all dapm audio paths and powers widgets according to their @@ -866,11 +883,11 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) * * Returns 0 for success. */ -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) +int snd_soc_dapm_sync(struct snd_soc_codec *codec) { return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); +EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, const char *sink, const char *control, const char *source) @@ -1418,53 +1435,57 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, } /** - * snd_soc_dapm_set_endpoint - set audio endpoint status - * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) - * @status: point status - * - * Set audio endpoint status - connected or disconnected. + * snd_soc_dapm_enable_pin - enable pin. + * @snd_soc_codec: SoC codec + * @pin: pin name * - * Returns 0 for success else error. + * Enables input/output pin and it's parents or children widgets iff there is + * a valid audio route and active audio stream. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *endpoint, int status) +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) { - struct snd_soc_dapm_widget *w; - - list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) { - w->connected = status; - return 0; - } - } + return snd_soc_dapm_set_pin(codec, pin, 1); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); - return -ENODEV; +/** + * snd_soc_dapm_disable_pin - disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Disables input/output pin and it's parents or children widgets. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** - * snd_soc_dapm_get_endpoint_status - get audio endpoint status + * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) + * @pin: audio signal pin endpoint (or start point) * - * Get audio endpoint status - connected or disconnected. + * Get audio pin status - connected or disconnected. * - * Returns status + * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, - char *endpoint) +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) { struct snd_soc_dapm_widget *w; list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) + if (!strcmp(w->name, pin)) return w->connected; } return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); /** * snd_soc_dapm_free - free dapm resources -- 1.5.4.2

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[alsa-devel] [PATCH 10/10] asoc: sh - 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 SuperH platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/sh/ssi.c | 8 ++++---- 4 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 7a3ce80..9faa126 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -326,7 +326,7 @@ static void camelot_pcm_free(struct snd_pcm *pcm) } static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index b7b676b..df7bc34 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE -struct snd_soc_cpu_dai sh4_hac_dai[] = { +struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 846d1b3..92bfaf4 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,7 +20,7 @@ #define IPSEL 0xFE400034 /* platform specific structs can be declared here */ -extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_dai sh4_hac_dai[2]; extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 3388bc3..55c3464 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -208,7 +208,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, +static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; @@ -222,7 +222,7 @@ static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, * This divider is used to generate the SSI_SCK (I2S bitclock) from the * clock at the HAC_BIT_CLK ("oversampling clock") pin. */ -static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr; @@ -245,7 +245,7 @@ static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) return 0; } -static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr = SSIREG(SSICR); @@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) -struct snd_soc_cpu_dai sh4_ssi_dai[] = { +struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, -- 1.5.4.2

1 0