[PATCH v3 09/10] ASoC: q6asm-dai: add gapless support
Srinivas Kandagatla
srinivas.kandagatla at linaro.org
Mon Jul 27 11:38:05 CEST 2020
Add support to gapless playback by implementing metadata,
next_track, drain and partial drain support.
Gapless on Q6ASM is implemented by opening 2 streams in a single
q6asm stream and toggling them on next track.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla at linaro.org>
---
sound/soc/qcom/qdsp6/q6asm-dai.c | 103 +++++++++++++++++++++++++++++--
sound/soc/qcom/qdsp6/q6asm.h | 1 +
2 files changed, 98 insertions(+), 6 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 420aaaa67788..4ecf9cb658ae 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -67,11 +67,14 @@ struct q6asm_dai_rtd {
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
+ uint32_t next_track_stream_id;
+ bool next_track;
uint32_t stream_id;
uint16_t session_id;
enum stream_state state;
uint32_t initial_samples_drop;
uint32_t trailing_samples_drop;
+ bool notify_on_drain;
};
struct q6asm_dai_data {
@@ -510,13 +513,19 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
struct q6asm_dai_rtd *prtd = priv;
struct snd_compr_stream *substream = prtd->cstream;
unsigned long flags;
+ u32 wflags = 0;
uint64_t avail;
- uint32_t bytes_written;
+ uint32_t bytes_written, bytes_to_write;
+ bool is_last_buffer = false;
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
spin_lock_irqsave(&prtd->lock, flags);
if (!prtd->bytes_sent) {
+ q6asm_stream_remove_initial_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->initial_samples_drop);
+
q6asm_write_async(prtd->audio_client, prtd->stream_id,
prtd->pcm_count, 0, 0, 0);
prtd->bytes_sent += prtd->pcm_count;
@@ -526,7 +535,30 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
break;
case ASM_CLIENT_EVENT_CMD_EOS_DONE:
- prtd->state = Q6ASM_STREAM_STOPPED;
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (prtd->notify_on_drain) {
+ if (substream->partial_drain) {
+ /*
+ * Close old stream and make it stale, switch
+ * the active stream now!
+ */
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id,
+ CMD_CLOSE);
+ /*
+ * vaild stream ids start from 1, So we are
+ * toggling this between 1 and 2.
+ */
+ prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
+ }
+
+ snd_compr_drain_notify(prtd->cstream);
+ prtd->notify_on_drain = false;
+
+ } else {
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
break;
case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
@@ -542,13 +574,32 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
}
avail = prtd->bytes_received - prtd->bytes_sent;
+ if (avail > prtd->pcm_count) {
+ bytes_to_write = prtd->pcm_count;
+ } else {
+ if (substream->partial_drain || prtd->notify_on_drain)
+ is_last_buffer = true;
+ bytes_to_write = avail;
+ }
+
+ if (bytes_to_write) {
+ if (substream->partial_drain && is_last_buffer) {
+ wflags |= ASM_LAST_BUFFER_FLAG;
+ q6asm_stream_remove_trailing_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->trailing_samples_drop);
+ }
- if (avail >= prtd->pcm_count) {
q6asm_write_async(prtd->audio_client, prtd->stream_id,
- prtd->pcm_count, 0, 0, 0);
- prtd->bytes_sent += prtd->pcm_count;
+ bytes_to_write, 0, 0, wflags);
+
+ prtd->bytes_sent += bytes_to_write;
}
+ if (prtd->notify_on_drain && is_last_buffer)
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id, CMD_EOS);
+
spin_unlock_irqrestore(&prtd->lock, flags);
break;
@@ -628,9 +679,15 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd = stream->private_data;
if (prtd->audio_client) {
- if (prtd->state)
+ if (prtd->state) {
q6asm_cmd(prtd->audio_client, prtd->stream_id,
CMD_CLOSE);
+ if (prtd->next_track_stream_id) {
+ q6asm_cmd(prtd->audio_client,
+ prtd->next_track_stream_id,
+ CMD_CLOSE);
+ }
+ }
snd_dma_free_pages(&prtd->dma_buffer);
q6asm_unmap_memory_regions(stream->direction,
@@ -905,6 +962,32 @@ static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
break;
case SNDRV_COMPRESS_ENCODER_DELAY:
prtd->initial_samples_drop = metadata->value[0];
+ if (prtd->next_track_stream_id) {
+ ret = q6asm_open_write(prtd->audio_client,
+ prtd->next_track_stream_id,
+ prtd->codec.id,
+ prtd->codec.profile,
+ prtd->bits_per_sample,
+ true);
+ if (ret < 0) {
+ dev_err(component->dev, "q6asm_open_write failed\n");
+ return ret;
+ }
+ ret = __q6asm_dai_compr_set_codec_params(component, stream,
+ &prtd->codec,
+ prtd->next_track_stream_id);
+ if (ret < 0) {
+ dev_err(component->dev, "q6asm_open_write failed\n");
+ return ret;
+ }
+
+ ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
+ prtd->next_track_stream_id,
+ prtd->initial_samples_drop);
+ prtd->next_track_stream_id = 0;
+
+ }
+
break;
default:
ret = -EINVAL;
@@ -938,6 +1021,14 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
CMD_PAUSE);
break;
+ case SND_COMPR_TRIGGER_NEXT_TRACK:
+ prtd->next_track = true;
+ prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
+ break;
+ case SND_COMPR_TRIGGER_DRAIN:
+ case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+ prtd->notify_on_drain = true;
+ break;
default:
ret = -EINVAL;
break;
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index f20e1441988f..82e584aa534f 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -33,6 +33,7 @@ enum {
#define MAX_SESSIONS 8
#define FORMAT_LINEAR_PCM 0x0000
+#define ASM_LAST_BUFFER_FLAG BIT(30)
struct q6asm_flac_cfg {
u32 sample_rate;
--
2.21.0
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