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author | Mark Brown <broonie@linaro.org> | 2014-01-16 12:42:57 +0000 |
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committer | Mark Brown <broonie@linaro.org> | 2014-01-16 12:42:57 +0000 |
commit | 2f43a23ab9ea1865a663e100b0af20198decb4f1 (patch) | |
tree | 7501fb678115cbbdb544c43dc7b59f95408f1f52 | |
parent | 7cfa7b547337faf5890c8c5f091e081fb79caf73 (diff) | |
parent | 55dcdb5051930dee75e9e2c0da90bc82ee3dcd77 (diff) | |
download | op-kernel-dev-2f43a23ab9ea1865a663e100b0af20198decb4f1.zip op-kernel-dev-2f43a23ab9ea1865a663e100b0af20198decb4f1.tar.gz |
Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai
-rw-r--r-- | include/sound/pcm.h | 2 | ||||
-rw-r--r-- | include/sound/soc.h | 4 | ||||
-rw-r--r-- | sound/core/pcm_misc.c | 39 | ||||
-rw-r--r-- | sound/soc/fsl/fsl_ssi.c | 3 | ||||
-rw-r--r-- | sound/soc/fsl/mpc5200_psc_i2s.c | 3 | ||||
-rw-r--r-- | sound/soc/s6000/s6000-i2s.c | 3 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 23 |
7 files changed, 59 insertions, 18 deletions
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index fe6ca40..4883499 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -900,6 +900,8 @@ extern const struct snd_pcm_hw_constraint_list snd_pcm_known_rates; int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime); unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate); unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit); +unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a, + unsigned int rates_b); static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream, struct snd_dma_buffer *bufp) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..03ce45b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -895,6 +895,10 @@ struct snd_soc_dai_link { /* This DAI link can route to other DAI links at runtime (Frontend)*/ unsigned int dynamic:1; + /* DPCM capture and Playback support */ + unsigned int dpcm_capture:1; + unsigned int dpcm_playback:1; + /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 43f24cc..4560ca0 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -514,3 +514,42 @@ unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit) return 0; } EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate); + +static unsigned int snd_pcm_rate_mask_sanitize(unsigned int rates) +{ + if (rates & SNDRV_PCM_RATE_CONTINUOUS) + return SNDRV_PCM_RATE_CONTINUOUS; + else if (rates & SNDRV_PCM_RATE_KNOT) + return SNDRV_PCM_RATE_KNOT; + return rates; +} + +/** + * snd_pcm_rate_mask_intersect - computes the intersection between two rate masks + * @rates_a: The first rate mask + * @rates_b: The second rate mask + * + * This function computes the rates that are supported by both rate masks passed + * to the function. It will take care of the special handling of + * SNDRV_PCM_RATE_CONTINUOUS and SNDRV_PCM_RATE_KNOT. + * + * Return: A rate mask containing the rates that are supported by both rates_a + * and rates_b. + */ +unsigned int snd_pcm_rate_mask_intersect(unsigned int rates_a, + unsigned int rates_b) +{ + rates_a = snd_pcm_rate_mask_sanitize(rates_a); + rates_b = snd_pcm_rate_mask_sanitize(rates_b); + + if (rates_a & SNDRV_PCM_RATE_CONTINUOUS) + return rates_b; + else if (rates_b & SNDRV_PCM_RATE_CONTINUOUS) + return rates_a; + else if (rates_a & SNDRV_PCM_RATE_KNOT) + return rates_b; + else if (rates_b & SNDRV_PCM_RATE_KNOT) + return rates_a; + return rates_a & rates_b; +} +EXPORT_SYMBOL_GPL(snd_pcm_rate_mask_intersect); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b2ebaf8..76e56b3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -80,8 +80,7 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) * ALSA that we support all rates and let the codec driver decide what rates * are really supported. */ -#define FSLSSI_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) +#define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index f4efaad..5d07e8a 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -26,8 +26,7 @@ * ALSA that we support all rates and let the codec driver decide what rates * are really supported. */ -#define PSC_I2S_RATES (SNDRV_PCM_RATE_5512 | SNDRV_PCM_RATE_8000_192000 | \ - SNDRV_PCM_RATE_CONTINUOUS) +#define PSC_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS /** * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index 73bb99f..7eba797 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -405,8 +405,7 @@ static int s6000_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } -#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_192000) +#define S6000_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS #define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops s6000_i2s_dai_ops = { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e95ef95..5932971 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -240,14 +240,15 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, cpu_stream->channels_max); - hw->formats = codec_stream->formats & cpu_stream->formats; - hw->rates = codec_stream->rates & cpu_stream->rates; - if (codec_stream->rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - hw->rates |= cpu_stream->rates; - if (cpu_stream->rates - & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - hw->rates |= codec_stream->rates; + if (hw->formats) + hw->formats &= codec_stream->formats & cpu_stream->formats; + else + hw->formats = codec_stream->formats & cpu_stream->formats; + hw->rates = snd_pcm_rate_mask_intersect(codec_stream->rates, + cpu_stream->rates); + + hw->rate_min = 0; + hw->rate_max = UINT_MAX; snd_pcm_limit_hw_rates(runtime); @@ -2140,10 +2141,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int ret = 0, playback = 0, capture = 0; if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - if (cpu_dai->driver->playback.channels_min) - playback = 1; - if (cpu_dai->driver->capture.channels_min) - capture = 1; + playback = rtd->dai_link->dpcm_playback; + capture = rtd->dai_link->dpcm_capture; } else { if (codec_dai->driver->playback.channels_min && cpu_dai->driver->playback.channels_min) |