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authorMark Brown <broonie@linaro.org>2014-01-16 12:42:57 +0000
committerMark Brown <broonie@linaro.org>2014-01-16 12:42:57 +0000
commit2f43a23ab9ea1865a663e100b0af20198decb4f1 (patch)
tree7501fb678115cbbdb544c43dc7b59f95408f1f52
parent7cfa7b547337faf5890c8c5f091e081fb79caf73 (diff)
parent55dcdb5051930dee75e9e2c0da90bc82ee3dcd77 (diff)
downloadop-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.h2
-rw-r--r--include/sound/soc.h4
-rw-r--r--sound/core/pcm_misc.c39
-rw-r--r--sound/soc/fsl/fsl_ssi.c3
-rw-r--r--sound/soc/fsl/mpc5200_psc_i2s.c3
-rw-r--r--sound/soc/s6000/s6000-i2s.c3
-rw-r--r--sound/soc/soc-pcm.c23
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)
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