[PATCH v4 0/5] ASoC: q6apm: fix under runs and fragment sizes

[PATCH v4 0/5] ASoC: q6apm: fix under runs and fragment sizes

[srinivas.kandagatla]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

On Qualcomm Audioreach setup, some of the audio artifacts are seen in
both recording and playback. These patches fix issues by
1. Adjusting the fragment size that dsp can service.
2. schedule available playback buffers in time for dsp to not hit under runs 
3. remove some of the manual calculations done to get hardware pointer.

With these patches, am able to see significant Audio quality improvements.
I have few more patches to optimize the dsp drivers, but for now am
keeping this series simple to address the underruns and overruns issues
noticed in pipewire setup.

Any testing would be appreciated.

Please note that on pipewire min-latency has to be set to 512 which
reflects the DSP latency requirements of 10ms. You might see audio
artifacts like glitches if you try to play audio below 256 latency.

thanks,
Srini

Changes since v3:
	- updated period size aligment patch with 10ms period size
	  contstriants.

Changes since v2:
	- dropped patch which is causing regression with pluseaudio.
	- setup period sizes only for capture path
	- fix underruns/overruns in dsp pipelines.
	- add fixes tag
	- add patch to fix buffer alignment

Changes since v1:
	- added new patches to fix the fragment size, pointer
	  calculations
	- updated to schedule only available buffers.


Srinivas Kandagatla (5):
  ASoC: q6apm-dai: schedule all available frames to avoid dsp under-runs
  ASoC: q6apm: add q6apm_get_hw_pointer helper
  ASoC: q6apm-dai: make use of q6apm_get_hw_pointer
  ASoC: qdsp6: q6apm-dai: set 10 ms period and buffer alignment.
  ASoC: qdsp6: q6apm-dai: fix capture pipeline overruns.

 sound/soc/qcom/qdsp6/q6apm-dai.c | 61 +++++++++++++++++---------------
 sound/soc/qcom/qdsp6/q6apm.c     | 18 +++++++++-
 sound/soc/qcom/qdsp6/q6apm.h     |  3 ++
 3 files changed, 52 insertions(+), 30 deletions(-)

-- 
2.39.2

[PATCH v4 1/5] ASoC: q6apm-dai: schedule all available frames to avoid dsp under-runs

[srinivas.kandagatla]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

With the existing code, we are only setting up one period at a time, in a
ping-pong buffer style. This triggers lot of underruns in the dsp
leading to jitter noise during audio playback.

Fix this by scheduling all available periods, this will ensure that the dsp
has enough buffer feed and ultimatley fixing the underruns and audio
distortion.

Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Reported-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 28 +++++++++++++++++++++++-----
 1 file changed, 23 insertions(+), 5 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index c9404b5934c7..9d8e8e37c6de 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -70,6 +70,7 @@ struct q6apm_dai_rtd {
 	unsigned int bytes_received;
 	unsigned int copied_total;
 	uint16_t bits_per_sample;
+	snd_pcm_uframes_t queue_ptr;
 	bool next_track;
 	enum stream_state state;
 	struct q6apm_graph *graph;
@@ -134,8 +135,6 @@ static void event_handler(uint32_t opcode, uint32_t token, void *payload, void *
 		prtd->pos += prtd->pcm_count;
 		spin_unlock_irqrestore(&prtd->lock, flags);
 		snd_pcm_period_elapsed(substream);
-		if (prtd->state == Q6APM_STREAM_RUNNING)
-			q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, 0);
 
 		break;
 	case APM_CLIENT_EVENT_DATA_READ_DONE:
@@ -294,6 +293,27 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
 	return 0;
 }
 
+static int q6apm_dai_ack(struct snd_soc_component *component, struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct q6apm_dai_rtd *prtd = runtime->private_data;
+	int i, ret = 0, avail_periods;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		avail_periods = (runtime->control->appl_ptr - prtd->queue_ptr)/runtime->period_size;
+		for (i = 0; i < avail_periods; i++) {
+			ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+			if (ret < 0) {
+				dev_err(component->dev, "Error queuing playback buffer %d\n", ret);
+				return ret;
+			}
+			prtd->queue_ptr += runtime->period_size;
+		}
+	}
+
+	return ret;
+}
+
 static int q6apm_dai_trigger(struct snd_soc_component *component,
 			     struct snd_pcm_substream *substream, int cmd)
 {
@@ -305,9 +325,6 @@ static int q6apm_dai_trigger(struct snd_soc_component *component,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		 /* start writing buffers for playback only as we already queued capture buffers */
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, 0);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		/* TODO support be handled via SoftPause Module */
@@ -836,6 +853,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = {
 	.hw_params	= q6apm_dai_hw_params,
 	.pointer	= q6apm_dai_pointer,
 	.trigger	= q6apm_dai_trigger,
+	.ack		= q6apm_dai_ack,
 	.compress_ops	= &q6apm_dai_compress_ops,
 	.use_dai_pcm_id = true,
 };
-- 
2.39.2

[PATCH v4 2/5] ASoC: q6apm: add q6apm_get_hw_pointer helper

[srinivas.kandagatla]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

Implement an helper function in q6apm to be able to read the current
hardware pointer for both read and write buffers.

This should help q6apm-dai to get the hardware pointer consistently
without it doing manual calculation, which could go wrong in some race
conditions.

Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm.c | 18 +++++++++++++++++-
 sound/soc/qcom/qdsp6/q6apm.h |  3 +++
 2 files changed, 20 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 2a2a5bd98110..ca57413cb784 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -494,6 +494,19 @@ int q6apm_read(struct q6apm_graph *graph)
 }
 EXPORT_SYMBOL_GPL(q6apm_read);
 
+int q6apm_get_hw_pointer(struct q6apm_graph *graph, int dir)
+{
+	struct audioreach_graph_data *data;
+
+	if (dir == SNDRV_PCM_STREAM_PLAYBACK)
+		data = &graph->rx_data;
+	else
+		data = &graph->tx_data;
+
+	return (int)atomic_read(&data->hw_ptr);
+}
+EXPORT_SYMBOL_GPL(q6apm_get_hw_pointer);
+
 static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 {
 	struct data_cmd_rsp_rd_sh_mem_ep_data_buffer_done_v2 *rd_done;
@@ -520,7 +533,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 		done = data->payload;
 		phys = graph->rx_data.buf[token].phys;
 		mutex_unlock(&graph->lock);
-
+		/* token numbering starts at 0 */
+		atomic_set(&graph->rx_data.hw_ptr, token + 1);
 		if (lower_32_bits(phys) == done->buf_addr_lsw &&
 		    upper_32_bits(phys) == done->buf_addr_msw) {
 			graph->result.opcode = hdr->opcode;
@@ -553,6 +567,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
 		rd_done = data->payload;
 		phys = graph->tx_data.buf[hdr->token].phys;
 		mutex_unlock(&graph->lock);
+		/* token numbering starts at 0 */
+		atomic_set(&graph->tx_data.hw_ptr, hdr->token + 1);
 
 		if (upper_32_bits(phys) == rd_done->buf_addr_msw &&
 		    lower_32_bits(phys) == rd_done->buf_addr_lsw) {
diff --git a/sound/soc/qcom/qdsp6/q6apm.h b/sound/soc/qcom/qdsp6/q6apm.h
index c248c8d2b1ab..7ce08b401e31 100644
--- a/sound/soc/qcom/qdsp6/q6apm.h
+++ b/sound/soc/qcom/qdsp6/q6apm.h
@@ -2,6 +2,7 @@
 #ifndef __Q6APM_H__
 #define __Q6APM_H__
 #include <linux/types.h>
+#include <linux/atomic.h>
 #include <linux/slab.h>
 #include <linux/wait.h>
 #include <linux/kernel.h>
@@ -77,6 +78,7 @@ struct audioreach_graph_data {
 	uint32_t num_periods;
 	uint32_t dsp_buf;
 	uint32_t mem_map_handle;
+	atomic_t hw_ptr;
 };
 
 struct audioreach_graph {
@@ -150,4 +152,5 @@ int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph,
 int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
 int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
 int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph, uint32_t codec_id);
+int q6apm_get_hw_pointer(struct q6apm_graph *graph, int dir);
 #endif /* __APM_GRAPH_ */
-- 
2.39.2

[PATCH v4 3/5] ASoC: q6apm-dai: make use of q6apm_get_hw_pointer

[srinivas.kandagatla]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

With the existing code, the buffer position is only reset in pointer
callback, which leaves the possiblity of it going over the size of
buffer size and reporting incorrect position to userspace.

Without this patch, its possible to see errors like:
snd-x1e80100 sound: invalid position: pcmC0D0p:0, pos = 12288, buffer size = 12288, period size = 1536
snd-x1e80100 sound: invalid position: pcmC0D0p:0, pos = 12288, buffer size = 12288, period size = 1536

Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 23 ++++-------------------
 1 file changed, 4 insertions(+), 19 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 9d8e8e37c6de..90cb24947f31 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -64,7 +64,6 @@ struct q6apm_dai_rtd {
 	phys_addr_t phys;
 	unsigned int pcm_size;
 	unsigned int pcm_count;
-	unsigned int pos;       /* Buffer position */
 	unsigned int periods;
 	unsigned int bytes_sent;
 	unsigned int bytes_received;
@@ -124,23 +123,16 @@ static void event_handler(uint32_t opcode, uint32_t token, void *payload, void *
 {
 	struct q6apm_dai_rtd *prtd = priv;
 	struct snd_pcm_substream *substream = prtd->substream;
-	unsigned long flags;
 
 	switch (opcode) {
 	case APM_CLIENT_EVENT_CMD_EOS_DONE:
 		prtd->state = Q6APM_STREAM_STOPPED;
 		break;
 	case APM_CLIENT_EVENT_DATA_WRITE_DONE:
-		spin_lock_irqsave(&prtd->lock, flags);
-		prtd->pos += prtd->pcm_count;
-		spin_unlock_irqrestore(&prtd->lock, flags);
 		snd_pcm_period_elapsed(substream);
 
 		break;
 	case APM_CLIENT_EVENT_DATA_READ_DONE:
-		spin_lock_irqsave(&prtd->lock, flags);
-		prtd->pos += prtd->pcm_count;
-		spin_unlock_irqrestore(&prtd->lock, flags);
 		snd_pcm_period_elapsed(substream);
 		if (prtd->state == Q6APM_STREAM_RUNNING)
 			q6apm_read(prtd->graph);
@@ -247,7 +239,6 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
 	}
 
 	prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
-	prtd->pos = 0;
 	/* rate and channels are sent to audio driver */
 	ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg);
 	if (ret < 0) {
@@ -445,16 +436,12 @@ static snd_pcm_uframes_t q6apm_dai_pointer(struct snd_soc_component *component,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct q6apm_dai_rtd *prtd = runtime->private_data;
 	snd_pcm_uframes_t ptr;
-	unsigned long flags;
 
-	spin_lock_irqsave(&prtd->lock, flags);
-	if (prtd->pos == prtd->pcm_size)
-		prtd->pos = 0;
-
-	ptr =  bytes_to_frames(runtime, prtd->pos);
-	spin_unlock_irqrestore(&prtd->lock, flags);
+	ptr = q6apm_get_hw_pointer(prtd->graph, substream->stream) * runtime->period_size;
+	if (ptr)
+		return ptr - 1;
 
-	return ptr;
+	return 0;
 }
 
 static int q6apm_dai_hw_params(struct snd_soc_component *component,
@@ -669,8 +656,6 @@ static int q6apm_dai_compr_set_params(struct snd_soc_component *component,
 	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
 	prtd->bits_per_sample = 16;
 
-	prtd->pos = 0;
-
 	if (prtd->next_track != true) {
 		memcpy(&prtd->codec, codec, sizeof(*codec));
 
-- 
2.39.2

[PATCH v4 4/5] ASoC: qdsp6: q6apm-dai: set 10 ms period and buffer alignment.

[srinivas.kandagatla]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

DSP expects the periods to be aligned to fragment sizes, currently
setting up to hw constriants on periods bytes is not going to work
correctly as we can endup with periods sizes aligned to 32 bytes however
not aligned to fragment size.

Update the constriants to use fragment size, and also set at step of
10ms for period size to accommodate DSP requirements of 10ms latency.

Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 90cb24947f31..a636f9280645 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -231,7 +231,6 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
 	cfg.bit_width = prtd->bits_per_sample;
 	cfg.fmt = SND_AUDIOCODEC_PCM;
 	audioreach_set_default_channel_mapping(cfg.channel_map, runtime->channels);
-
 	if (prtd->state) {
 		/* clear the previous setup if any  */
 		q6apm_graph_stop(prtd->graph);
@@ -385,13 +384,14 @@ static int q6apm_dai_open(struct snd_soc_component *component,
 		}
 	}
 
-	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+	/* setup 10ms latency to accommodate DSP restrictions */
+	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 480);
 	if (ret < 0) {
 		dev_err(dev, "constraint for period bytes step ret = %d\n", ret);
 		goto err;
 	}
 
-	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+	ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 480);
 	if (ret < 0) {
 		dev_err(dev, "constraint for buffer bytes step ret = %d\n", ret);
 		goto err;
-- 
2.39.2

Re: [PATCH v4 4/5] ASoC: qdsp6: q6apm-dai: set 10 ms period and buffer alignment.

[Johan Hovold]

On Fri, Mar 14, 2025 at 02:32:19PM +0000, Srinivas Kandagatla wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
> 
> DSP expects the periods to be aligned to fragment sizes, currently
> setting up to hw constriants on periods bytes is not going to work
> correctly as we can endup with periods sizes aligned to 32 bytes however
> not aligned to fragment size.
> 
> Update the constriants to use fragment size, and also set at step of
> 10ms for period size to accommodate DSP requirements of 10ms latency.
> 
> Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")

For all of the patches, if the intention is that they should be
backported to stable they should have a CC stable tag here.

> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
> ---
>  sound/soc/qcom/qdsp6/q6apm-dai.c | 6 +++---
>  1 file changed, 3 insertions(+), 3 deletions(-)
> 
> diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
> index 90cb24947f31..a636f9280645 100644
> --- a/sound/soc/qcom/qdsp6/q6apm-dai.c
> +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
> @@ -231,7 +231,6 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
>  	cfg.bit_width = prtd->bits_per_sample;
>  	cfg.fmt = SND_AUDIOCODEC_PCM;
>  	audioreach_set_default_channel_mapping(cfg.channel_map, runtime->channels);
> -

nit: unrelated change

>  	if (prtd->state) {
>  		/* clear the previous setup if any  */
>  		q6apm_graph_stop(prtd->graph);

Johan

Re: [PATCH v4 4/5] ASoC: qdsp6: q6apm-dai: set 10 ms period and buffer alignment.

[Srinivas Kandagatla]

On 14/03/2025 16:46, Johan Hovold wrote:
> On Fri, Mar 14, 2025 at 02:32:19PM +0000, Srinivas Kandagatla wrote:
>> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
>>
>> DSP expects the periods to be aligned to fragment sizes, currently
>> setting up to hw constriants on periods bytes is not going to work
>> correctly as we can endup with periods sizes aligned to 32 bytes however
>> not aligned to fragment size.
>>
>> Update the constriants to use fragment size, and also set at step of
>> 10ms for period size to accommodate DSP requirements of 10ms latency.
>>
>> Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")
> 
> For all of the patches, if the intention is that they should be
> backported to stable they should have a CC stable tag here.

Ideally yes. It would nice to get these back ported to stable.
> 
>> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
>> ---
>>   sound/soc/qcom/qdsp6/q6apm-dai.c | 6 +++---
>>   1 file changed, 3 insertions(+), 3 deletions(-)
>>
>> diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
>> index 90cb24947f31..a636f9280645 100644
>> --- a/sound/soc/qcom/qdsp6/q6apm-dai.c
>> +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
>> @@ -231,7 +231,6 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
>>   	cfg.bit_width = prtd->bits_per_sample;
>>   	cfg.fmt = SND_AUDIOCODEC_PCM;
>>   	audioreach_set_default_channel_mapping(cfg.channel_map, runtime->channels);
>> -
> 
> nit: unrelated change
Fixed in v5.

--srini
> 
>>   	if (prtd->state) {
>>   		/* clear the previous setup if any  */
>>   		q6apm_graph_stop(prtd->graph);
> 
> Johan

[PATCH v4 5/5] ASoC: qdsp6: q6apm-dai: fix capture pipeline overruns.

[srinivas.kandagatla]

From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>

Period sizes less than 6k for capture path triggers overruns in the
dsp capture pipeline.

Change the period size and number of periods to value which DSP is happy with.

Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
---
 sound/soc/qcom/qdsp6/q6apm-dai.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index a636f9280645..292cab61831c 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -24,8 +24,8 @@
 #define PLAYBACK_MIN_PERIOD_SIZE	128
 #define CAPTURE_MIN_NUM_PERIODS		2
 #define CAPTURE_MAX_NUM_PERIODS		8
-#define CAPTURE_MAX_PERIOD_SIZE		4096
-#define CAPTURE_MIN_PERIOD_SIZE		320
+#define CAPTURE_MAX_PERIOD_SIZE		65536
+#define CAPTURE_MIN_PERIOD_SIZE		6144
 #define BUFFER_BYTES_MAX (PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE)
 #define BUFFER_BYTES_MIN (PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE)
 #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
-- 
2.39.2

Re: [PATCH v4 0/5] ASoC: q6apm: fix under runs and fragment sizes

[Johan Hovold]

On Fri, Mar 14, 2025 at 02:32:15PM +0000, Srinivas Kandagatla wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
> 
> On Qualcomm Audioreach setup, some of the audio artifacts are seen in
> both recording and playback. These patches fix issues by
> 1. Adjusting the fragment size that dsp can service.
> 2. schedule available playback buffers in time for dsp to not hit under runs 
> 3. remove some of the manual calculations done to get hardware pointer.
> 
> With these patches, am able to see significant Audio quality improvements.
> I have few more patches to optimize the dsp drivers, but for now am
> keeping this series simple to address the underruns and overruns issues
> noticed in pipewire setup.
> 
> Any testing would be appreciated.
> 
> Please note that on pipewire min-latency has to be set to 512 which
> reflects the DSP latency requirements of 10ms. You might see audio
> artifacts like glitches if you try to play audio below 256 latency.

> Changes since v3:
> 	- updated period size aligment patch with 10ms period size
> 	  contstriants.
> 
> Changes since v2:
> 	- dropped patch which is causing regression with pluseaudio.
> 	- setup period sizes only for capture path
> 	- fix underruns/overruns in dsp pipelines.
> 	- add fixes tag
> 	- add patch to fix buffer alignment
> 
> Changes since v1:
> 	- added new patches to fix the fragment size, pointer
> 	  calculations
> 	- updated to schedule only available buffers.
> 

I tested v4 on the X13s with PulseAudio and T14s with PipeWire.

I'm no longer noticing any regressions with PulseAudio, while this
series makes PipeWire actually usable.

I may be hearing some crackling with PipeWire when recording with the 
pavucontrol volume meters enabled which results in a quantum size of
256. But I guess that's to be expected. Playback seems to work also at
256 with v4.

Tested-by: Johan Hovold <johan+linaro@kernel.org>

Johan