Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai

7 files changed, 59 insertions, 18 deletions

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index fe6ca400b9ad..4883499ab38b 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 5a049d969c59..03ce45bf8ade 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 43f24cce3dec..4560ca0e5651 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 b2ebaf811599..76e56b39db01 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 f4efaadb80a2..5d07e8a74a21 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 73bb99f0109a..7eba7979b9af 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 e95ef9586723..5932971cf54d 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)