[PATCH 10/11] ASoC: qdsp6-dai: add gapless support

From: Srinivas Kandagatla
Date: Tue Jul 07 2020 - 12:37:12 EST

Next message: Suraj Upadhyay: "[PATCH 3/4] drm: edid: Convert logging to drm_* functions."
Previous message: Srinivas Kandagatla: "[PATCH 11/11] ASoC: q6asm-dai: add support to copy callback"
In reply to: Srinivas Kandagatla: "[PATCH 11/11] ASoC: q6asm-dai: add support to copy callback"
Next in thread: Pierre-Louis Bossart: "Re: [PATCH 10/11] ASoC: qdsp6-dai: add gapless support"
Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]

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 asm stream
and toggling them on next track.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx>
---
sound/soc/qcom/qdsp6/q6asm-dai.c | 212 +++++++++++++++++++++++--------
sound/soc/qcom/qdsp6/q6asm.h | 1 +
2 files changed, 158 insertions(+), 55 deletions(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index c4b4684b7824..f48eca227fb5 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -67,12 +67,15 @@ 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;
/* Active */
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 {
@@ -510,9 +513,11 @@ 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;
+ unsigned long flags = 0;
+ 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:
@@ -527,7 +532,25 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
break;

case ASM_CLIENT_EVENT_CMD_EOS_DONE:
- prtd->state = Q6ASM_STREAM_STOPPED;
+ if (prtd->notify_on_drain) {
+ if (substream->partial_drain && prtd->next_track_stream_id) {
+ /**
+ * Close old stream and make it stale, switch
+ * the active stream now!
+ */
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id,
+ CMD_CLOSE);
+ prtd->stream_id = prtd->next_track_stream_id;
+ prtd->next_track_stream_id = 0;
+ }
+
+ snd_compr_drain_notify(prtd->cstream);
+ prtd->notify_on_drain = false;
+
+ } else {
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ }
break;

case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
@@ -543,13 +566,32 @@ 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;

@@ -629,9 +671,15 @@ 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,
@@ -645,15 +693,13 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
return 0;
}

-static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
- struct snd_compr_stream *stream,
- struct snd_compr_params *params)
+static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_params *params,
+ int stream_id)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
- struct snd_soc_pcm_runtime *rtd = stream->private_data;
- int dir = stream->direction;
- struct q6asm_dai_data *pdata;
struct q6asm_flac_cfg flac_cfg;
struct q6asm_wma_cfg wma_cfg;
struct q6asm_alac_cfg alac_cfg;
@@ -669,43 +715,8 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,

codec_options = &(prtd->codec_param.codec.options);

-
memcpy(&prtd->codec_param, params, sizeof(*params));

- pdata = snd_soc_component_get_drvdata(component);
- if (!pdata)
- return -EINVAL;
-
- if (!prtd || !prtd->audio_client) {
- dev_err(dev, "private data null or audio client freed\n");
- return -EINVAL;
- }
-
- prtd->periods = runtime->fragments;
- prtd->pcm_count = runtime->fragment_size;
- prtd->pcm_size = runtime->fragments * runtime->fragment_size;
- prtd->bits_per_sample = 16;
- if (dir == SND_COMPRESS_PLAYBACK) {
- ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
- params->codec.id, params->codec.profile,
- prtd->bits_per_sample, true);
-
- if (ret < 0) {
- dev_err(dev, "q6asm_open_write failed\n");
- q6asm_audio_client_free(prtd->audio_client);
- prtd->audio_client = NULL;
- return ret;
- }
- }
-
- prtd->session_id = q6asm_get_session_id(prtd->audio_client);
- ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
- prtd->session_id, dir);
- if (ret) {
- dev_err(dev, "Stream reg failed ret:%d\n", ret);
- return ret;
- }
-
switch (params->codec.id) {
case SND_AUDIOCODEC_FLAC:

@@ -722,7 +733,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
flac_cfg.min_frame_size = flac->min_frame_size;

ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
- prtd->stream_id,
+ stream_id,
&flac_cfg);
if (ret < 0) {
dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
@@ -782,11 +793,11 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,

if (wma_v9)
ret = q6asm_stream_media_format_block_wma_v9(
- prtd->audio_client, prtd->stream_id,
+ prtd->audio_client, stream_id,
&wma_cfg);
else
ret = q6asm_stream_media_format_block_wma_v10(
- prtd->audio_client, prtd->stream_id,
+ prtd->audio_client, stream_id,
&wma_cfg);
if (ret < 0) {
dev_err(dev, "WMA9 CMD failed:%d\n", ret);
@@ -820,7 +831,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
break;
}
ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
- prtd->stream_id,
+ stream_id,
&alac_cfg);
if (ret < 0) {
dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
@@ -845,7 +856,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
ape_cfg.seek_table_present = ape->seek_table_present;

ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
- prtd->stream_id,
+ stream_id,
&ape_cfg);
if (ret < 0) {
dev_err(dev, "APE CMD Format block failed:%d\n", ret);
@@ -857,6 +868,63 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
break;
}

+ return 0;
+}
+
+static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_params *params)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+ int dir = stream->direction;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
+ int ret;
+
+ pdata = snd_soc_component_get_drvdata(component);
+ if (!pdata)
+ return -EINVAL;
+
+ if (!prtd || !prtd->audio_client) {
+ dev_err(dev, "private data null or audio client freed\n");
+ return -EINVAL;
+ }
+
+ prtd->periods = runtime->fragments;
+ prtd->pcm_count = runtime->fragment_size;
+ prtd->pcm_size = runtime->fragments * runtime->fragment_size;
+ prtd->bits_per_sample = 16;
+
+ if (dir == SND_COMPRESS_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
+ params->codec.profile, prtd->bits_per_sample,
+ true);
+
+ if (ret < 0) {
+ dev_err(dev, "q6asm_open_write failed\n");
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return ret;
+ }
+ }
+
+ prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+ ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
+ prtd->session_id, dir);
+ if (ret) {
+ dev_err(dev, "Stream reg failed ret:%d\n", ret);
+ return ret;
+ }
+
+ ret = __q6asm_dai_compr_set_codec_params(component, stream, params,
+ prtd->stream_id);
+ if (ret) {
+ dev_err(dev, "codec param setup failed ret:%d\n", ret);
+ return ret;
+ }
+
ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
(prtd->pcm_size / prtd->periods),
prtd->periods);
@@ -871,12 +939,13 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
return 0;
}

-static int q6asm_dai_compr_set_metadata(struct snd_compr_stream *stream,
+static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
struct snd_compr_metadata *metadata)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
- int ret = 0;
+ int stream_id = prtd->stream_id, ret = 0;

switch (metadata->key) {
case SNDRV_COMPRESS_ENCODER_PADDING:
@@ -884,6 +953,31 @@ static int q6asm_dai_compr_set_metadata(struct snd_compr_stream *stream,
break;
case SNDRV_COMPRESS_ENCODER_DELAY:
prtd->initial_samples_drop = metadata->value[0];
+ if (prtd->next_track_stream_id) {
+ stream_id = prtd->next_track_stream_id;
+ ret = q6asm_open_write(prtd->audio_client,
+ prtd->next_track_stream_id,
+ prtd->codec_param.codec.id,
+ prtd->codec_param.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_param,
+ prtd->next_track_stream_id);
+ if (ret < 0) {
+ dev_err(component->dev, "q6asm_open_write failed\n");
+ return ret;
+ }
+
+ }
+
+ q6asm_stream_remove_initial_silence(prtd->audio_client,
+ stream_id,
+ prtd->initial_samples_drop);
break;
default:
ret = -EINVAL;
@@ -917,6 +1011,14 @@ 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 69513ac1fa55..a8dddfeb28da 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -34,6 +34,7 @@ enum {
#define MAX_SESSIONS 8
#define NO_TIMESTAMP 0xFF00
#define FORMAT_LINEAR_PCM 0x0000
+#define ASM_LAST_BUFFER_FLAG BIT(30)

struct q6asm_flac_cfg {
u32 sample_rate;
--
2.21.0

Next message: Suraj Upadhyay: "[PATCH 3/4] drm: edid: Convert logging to drm_* functions."
Previous message: Srinivas Kandagatla: "[PATCH 11/11] ASoC: q6asm-dai: add support to copy callback"
In reply to: Srinivas Kandagatla: "[PATCH 11/11] ASoC: q6asm-dai: add support to copy callback"
Next in thread: Pierre-Louis Bossart: "Re: [PATCH 10/11] ASoC: qdsp6-dai: add gapless support"
Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]