ASoC: qcom: q6asm-dai: schedule all available frames to avoid dsp under-runs

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.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Tested-by: Alexey Klimov <alexey.klimov@linaro.org> # RB5, RB3
Link: https://patch.msgid.link/20251023102444.88158-7-srinivas.kandagatla@oss.qualcomm.com
Signed-off-by: Mark Brown <broonie@kernel.org>

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 0eae8c6e42b8aa8ee9ef4ace4a07b5352ac65131..db2ea8973ac98626a7763ec45ff95f8aea364c02 100644 (file)
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -64,6 +64,7 @@ struct q6asm_dai_rtd {
        uint64_t bytes_received;
        uint64_t copied_total;
        uint16_t bits_per_sample;
+       snd_pcm_uframes_t queue_ptr;
        uint16_t source; /* Encoding source bit mask */
        struct audio_client *audio_client;
        uint32_t next_track_stream_id;
@@ -85,6 +86,7 @@ struct q6asm_dai_data {
 static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
        .info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
                                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                               SNDRV_PCM_INFO_NO_REWINDS | SNDRV_PCM_INFO_SYNC_APPLPTR |
                                SNDRV_PCM_INFO_MMAP_VALID |
                                SNDRV_PCM_INFO_INTERLEAVED |
                                SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
@@ -108,6 +110,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_playback = {
        .info =                 (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
                                SNDRV_PCM_INFO_BLOCK_TRANSFER |
                                SNDRV_PCM_INFO_MMAP_VALID |
+                               SNDRV_PCM_INFO_NO_REWINDS | SNDRV_PCM_INFO_SYNC_APPLPTR |
                                SNDRV_PCM_INFO_INTERLEAVED |
                                SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
        .formats =              (SNDRV_PCM_FMTBIT_S16_LE |
@@ -182,9 +185,6 @@ static void event_handler(uint32_t opcode, uint32_t token,
 
        switch (opcode) {
        case ASM_CLIENT_EVENT_CMD_RUN_DONE:
-               if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-                       q6asm_write_async(prtd->audio_client, prtd->stream_id,
-                                  prtd->pcm_count, 0, 0, 0);
                break;
        case ASM_CLIENT_EVENT_CMD_EOS_DONE:
                prtd->state = Q6ASM_STREAM_STOPPED;
@@ -192,10 +192,6 @@ static void event_handler(uint32_t opcode, uint32_t token,
        case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
                prtd->pcm_irq_pos += prtd->pcm_count;
                snd_pcm_period_elapsed(substream);
-               if (prtd->state == Q6ASM_STREAM_RUNNING)
-                       q6asm_write_async(prtd->audio_client, prtd->stream_id,
-                                          prtd->pcm_count, 0, 0, 0);
-
                break;
                }
        case ASM_CLIENT_EVENT_DATA_READ_DONE:
@@ -311,6 +307,29 @@ open_err:
        return ret;
 }
 
+static int q6asm_dai_ack(struct snd_soc_component *component, struct snd_pcm_substream *substream)
+{
+       struct snd_pcm_runtime *runtime = substream->runtime;
+       struct q6asm_dai_rtd *prtd = runtime->private_data;
+       int i, ret = 0, avail_periods;
+
+       if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && prtd->state == Q6ASM_STREAM_RUNNING) {
+               avail_periods = (runtime->control->appl_ptr - prtd->queue_ptr)/runtime->period_size;
+               for (i = 0; i < avail_periods; i++) {
+                       ret = q6asm_write_async(prtd->audio_client, prtd->stream_id,
+                                          prtd->pcm_count, 0, 0, 0);
+
+                       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 q6asm_dai_trigger(struct snd_soc_component *component,
                             struct snd_pcm_substream *substream, int cmd)
 {
@@ -1215,6 +1234,7 @@ static const struct snd_soc_component_driver q6asm_fe_dai_component = {
        .close                  = q6asm_dai_close,
        .prepare                = q6asm_dai_prepare,
        .trigger                = q6asm_dai_trigger,
+       .ack                    = q6asm_dai_ack,
        .pointer                = q6asm_dai_pointer,
        .pcm_construct          = q6asm_dai_pcm_new,
        .compress_ops           = &q6asm_dai_compress_ops,