[alsa-devel] [PATCH] ALSA: pcm - introduce soc_delay

[Vinod Koul]

In many modern SoCs the audio DSP can buffer the PCM ring buffer data. Today we
have no means to represent this buffering and ALSA wrongly detects an overrun
when hw_ptr reaches app_ptr value, though DSP may still have some buffered data.

This patch tries to add a new field "soc_delay" to represent buffering done in
DSPs. This value is also used for the xrun calculations in ALSA.

Signed-off-by: Vinod Koul <vinod.koul at linux.intel.com>

--
Once we are okay with this approach, I will send a follow up patch which adds
this notion in ASoC and uses this to compute cpu_dai delay. The codec_dai delay
along with FIFO delay from cpu_dai should be added and reresented by today's
notion of delay.
---
 include/sound/pcm.h     |    1 +
 sound/core/pcm_lib.c    |   14 +++++++++++---
 sound/core/pcm_native.c |    6 +++---
 3 files changed, 15 insertions(+), 6 deletions(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index a55d5db..405deb7 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -281,6 +281,7 @@ struct snd_pcm_runtime {
 	unsigned long hw_ptr_jiffies;	/* Time when hw_ptr is updated */
 	unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */
 	snd_pcm_sframes_t delay;	/* extra delay; typically FIFO size */
+	snd_pcm_sframes_t soc_delay;	/* extra delay; typically delay incurred in soc */
 
 	/* -- HW params -- */
 	snd_pcm_access_t access;	/* access mode */
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 8f312fa..4977012 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -292,7 +292,15 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream,
 			return -EPIPE;
 		}
 	} else {
-		if (avail >= runtime->stop_threshold) {
+		snd_pcm_uframes_t actual_avail;
+		if (avail < runtime->soc_delay)
+			actual_avail = avail;
+		else
+			actual_avail = avail - runtime->soc_delay;
+		if (actual_avail  >= runtime->stop_threshold) {
+			snd_printd(KERN_ERR  "avail > stop_threshold!!\n");
+			snd_printd(KERN_ERR  "actual_avail %ld, avail %ld, soc_delay %ld!!\n",
+					actual_avail, avail,  runtime->soc_delay);
 			xrun(substream);
 			return -EPIPE;
 		}
@@ -440,9 +448,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 	if (runtime->hw.info & SNDRV_PCM_INFO_BATCH)
 		goto no_jiffies_check;
 	hdelta = delta;
-	if (hdelta < runtime->delay)
+	if (hdelta < (runtime->delay + runtime->soc_delay))
 		goto no_jiffies_check;
-	hdelta -= runtime->delay;
+	hdelta -= (runtime->delay + runtime->soc_delay);
 	jdelta = curr_jiffies - runtime->hw_ptr_jiffies;
 	if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
 		delta = jdelta /
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 53b5ada..fc2d664 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -606,13 +606,13 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
 		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
 		    runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
 			status->delay = runtime->buffer_size - status->avail;
-			status->delay += runtime->delay;
+			status->delay += runtime->delay + runtime->soc_delay;
 		} else
 			status->delay = 0;
 	} else {
 		status->avail = snd_pcm_capture_avail(runtime);
 		if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
-			status->delay = status->avail + runtime->delay;
+			status->delay = status->avail + runtime->delay + runtime->soc_delay;
 		else
 			status->delay = 0;
 	}
@@ -2442,7 +2442,7 @@ static int snd_pcm_delay(struct snd_pcm_substream *substream,
 			n = snd_pcm_playback_hw_avail(runtime);
 		else
 			n = snd_pcm_capture_avail(runtime);
-		n += runtime->delay;
+		n += runtime->delay + runtime->soc_delay;
 		break;
 	case SNDRV_PCM_STATE_XRUN:
 		err = -EPIPE;
-- 
1.7.0.4