[PATCH v2 0/9] ASoC: qdsp6: add gapless compressed audio support
This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single q6asm session.
First few patches such as stream id per each command, gapless flags and silence meta data are for preparedness for adding gapless support. Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases.
This patchset is tested on RB3 aka DB845c platform.
This patchset as it is will support gapless however QDSP can also support switching decoders on a single stream. Patches to support such feature are send in different patchset which involves adding generic interfaces.
Thanks, srini
Changes since v1: - Fixed all the comments to use "q6asm" wording correctly. - dropped patches that are already applied
Srinivas Kandagatla (9): ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from q6asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in q6asm open ASoC: q6asm-dai: add next track metadata support ASoC: qdsp6-dai: add gapless support ASoC: q6asm-dai: add support to copy callback
sound/soc/qcom/qdsp6/q6asm-dai.c | 413 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 169 +++++++++---- sound/soc/qcom/qdsp6/q6asm.h | 48 ++-- 3 files changed, 467 insertions(+), 163 deletions(-)
Each q6asm session can have multiple streams, mixing usage of these names in variable are bit misleading to reader, so rename them accordingly.
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index e0983970cba9..51da3717a6a6 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -828,21 +828,21 @@ EXPORT_SYMBOL_GPL(q6asm_get_session_id); * @dev: Pointer to asm child device. * @cb: event callback. * @priv: private data associated with this client. - * @stream_id: stream id + * @session_id: session id * @perf_mode: performace mode for this client * * Return: Will be an error pointer on error or a valid audio client * on success. */ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, - void *priv, int stream_id, + void *priv, int session_id, int perf_mode) { struct q6asm *a = dev_get_drvdata(dev->parent); struct audio_client *ac; unsigned long flags;
- ac = q6asm_get_audio_client(a, stream_id + 1); + ac = q6asm_get_audio_client(a, session_id + 1); if (ac) { dev_err(dev, "Audio Client already active\n"); return ac; @@ -853,9 +853,9 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, return ERR_PTR(-ENOMEM);
spin_lock_irqsave(&a->slock, flags); - a->session[stream_id + 1] = ac; + a->session[session_id + 1] = ac; spin_unlock_irqrestore(&a->slock, flags); - ac->session = stream_id + 1; + ac->session = session_id + 1; ac->cb = cb; ac->dev = dev; ac->q6asm = a;
Each ASM session can have multiple streams attached to it, current design was to allow only one static stream id 1 per each session. However for use-case like gapless, we would need 2 streams to open per session.
This patch converts all the q6asm apis to take stream id as argument to allow multiple streams to open on a single session, This is useful for gapless playback cases.
Now the dai driver can specify which stream id for each command.
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm-dai.c | 86 ++++++++++++++++++----------- sound/soc/qcom/qdsp6/q6asm.c | 92 ++++++++++++++++++-------------- sound/soc/qcom/qdsp6/q6asm.h | 38 ++++++++----- 3 files changed, 133 insertions(+), 83 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 941f3216399c..fb0488e7beb9 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -67,6 +67,8 @@ struct q6asm_dai_rtd { uint16_t bits_per_sample; uint16_t source; /* Encoding source bit mask */ struct audio_client *audio_client; + /* Active */ + uint32_t stream_id; uint16_t session_id; enum stream_state state; }; @@ -184,8 +186,8 @@ 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->pcm_count, 0, 0, NO_TIMESTAMP); + 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; @@ -194,8 +196,8 @@ static void event_handler(uint32_t opcode, uint32_t token, 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->pcm_count, 0, 0, NO_TIMESTAMP); + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0);
break; } @@ -203,7 +205,7 @@ static void event_handler(uint32_t opcode, uint32_t token, prtd->pcm_irq_pos += prtd->pcm_count; snd_pcm_period_elapsed(substream); if (prtd->state == Q6ASM_STREAM_RUNNING) - q6asm_read(prtd->audio_client); + q6asm_read(prtd->audio_client, prtd->stream_id);
break; default: @@ -236,7 +238,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, /* rate and channels are sent to audio driver */ if (prtd->state) { /* clear the previous setup if any */ - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE); q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); q6routing_stream_close(soc_prtd->dai_link->id, @@ -255,11 +257,13 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, }
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM, + ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, + FORMAT_LINEAR_PCM, 0, prtd->bits_per_sample); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM, - prtd->bits_per_sample); + ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, + FORMAT_LINEAR_PCM, + prtd->bits_per_sample); }
if (ret < 0) { @@ -279,17 +283,19 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = q6asm_media_format_block_multi_ch_pcm( - prtd->audio_client, runtime->rate, - runtime->channels, NULL, + prtd->audio_client, prtd->stream_id, + runtime->rate, runtime->channels, NULL, prtd->bits_per_sample); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client, - runtime->rate, runtime->channels, - prtd->bits_per_sample); + prtd->stream_id, + runtime->rate, + runtime->channels, + prtd->bits_per_sample);
/* Queue the buffers */ for (i = 0; i < runtime->periods; i++) - q6asm_read(prtd->audio_client); + q6asm_read(prtd->audio_client, prtd->stream_id);
} if (ret < 0) @@ -311,15 +317,18 @@ static int q6asm_dai_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id, + 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: prtd->state = Q6ASM_STREAM_STOPPED; - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_EOS); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_PAUSE); break; default: ret = -EINVAL; @@ -364,6 +373,9 @@ static int q6asm_dai_open(struct snd_soc_component *component, return ret; }
+ /* DSP expects stream id from 1 */ + prtd->stream_id = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = q6asm_dai_hardware_playback; else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -430,7 +442,8 @@ static int q6asm_dai_close(struct snd_soc_component *component,
if (prtd->audio_client) { if (prtd->state) - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_cmd(prtd->audio_client, prtd->stream_id, + CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream, prtd->audio_client); @@ -502,8 +515,8 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, case ASM_CLIENT_EVENT_CMD_RUN_DONE: spin_lock_irqsave(&prtd->lock, flags); if (!prtd->bytes_sent) { - q6asm_write_async(prtd->audio_client, prtd->pcm_count, - 0, 0, NO_TIMESTAMP); + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); prtd->bytes_sent += prtd->pcm_count; }
@@ -528,8 +541,8 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, avail = prtd->bytes_received - prtd->bytes_sent;
if (avail >= prtd->pcm_count) { - q6asm_write_async(prtd->audio_client, - prtd->pcm_count, 0, 0, NO_TIMESTAMP); + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0); prtd->bytes_sent += prtd->pcm_count; }
@@ -563,6 +576,9 @@ static int q6asm_dai_compr_open(struct snd_soc_component *component, if (!prtd) return -ENOMEM;
+ /* DSP expects stream id from 1 */ + prtd->stream_id = 1; + prtd->cstream = stream; prtd->audio_client = q6asm_audio_client_alloc(dev, (q6asm_cb)compress_event_handler, @@ -610,7 +626,8 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
if (prtd->audio_client) { if (prtd->state) - q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_cmd(prtd->audio_client, prtd->stream_id, + CMD_CLOSE);
snd_dma_free_pages(&prtd->dma_buffer); q6asm_unmap_memory_regions(stream->direction, @@ -665,8 +682,9 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, 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, params->codec.id, - params->codec.profile, prtd->bits_per_sample); + ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, + params->codec.id, params->codec.profile, + prtd->bits_per_sample);
if (ret < 0) { dev_err(dev, "q6asm_open_write failed\n"); @@ -700,6 +718,7 @@ 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, &flac_cfg); if (ret < 0) { dev_err(dev, "FLAC CMD Format block failed:%d\n", ret); @@ -759,10 +778,12 @@ 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, &wma_cfg); + prtd->audio_client, prtd->stream_id, + &wma_cfg); else ret = q6asm_stream_media_format_block_wma_v10( - prtd->audio_client, &wma_cfg); + prtd->audio_client, prtd->stream_id, + &wma_cfg); if (ret < 0) { dev_err(dev, "WMA9 CMD failed:%d\n", ret); return -EIO; @@ -795,6 +816,7 @@ 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, &alac_cfg); if (ret < 0) { dev_err(dev, "ALAC CMD Format block failed:%d\n", ret); @@ -819,6 +841,7 @@ 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, &ape_cfg); if (ret < 0) { dev_err(dev, "APE CMD Format block failed:%d\n", ret); @@ -855,15 +878,18 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id, + 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: prtd->state = Q6ASM_STREAM_STOPPED; - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_EOS); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id, + CMD_PAUSE); break; default: ret = -EINVAL; diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 51da3717a6a6..f5d1f3c2c1ec 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -270,7 +270,6 @@ struct audio_client { wait_queue_head_t cmd_wait; struct aprv2_ibasic_rsp_result_t result; int perf_mode; - int stream_id; struct q6asm *q6asm; struct device *dev; }; @@ -862,8 +861,6 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, ac->priv = priv; ac->io_mode = ASM_SYNC_IO_MODE; ac->perf_mode = perf_mode; - /* DSP expects stream id from 1 */ - ac->stream_id = 1; ac->adev = a->adev; kref_init(&ac->refcount);
@@ -919,8 +916,9 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt) * * Return: Will be an negative value on error or zero on success */ -int q6asm_open_write(struct audio_client *ac, uint32_t format, - u32 codec_profile, uint16_t bits_per_sample) +int q6asm_open_write(struct audio_client *ac, uint32_t stream_id, + uint32_t format, u32 codec_profile, + uint16_t bits_per_sample) { struct asm_stream_cmd_open_write_v3 *open; struct apr_pkt *pkt; @@ -935,7 +933,7 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
pkt = p; open = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3; open->mode_flags = 0x00; @@ -998,8 +996,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format, } EXPORT_SYMBOL_GPL(q6asm_open_write);
-static int __q6asm_run(struct audio_client *ac, uint32_t flags, - uint32_t msw_ts, uint32_t lsw_ts, bool wait) +static int __q6asm_run(struct audio_client *ac, uint32_t stream_id, + uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts, + bool wait) { struct asm_session_cmd_run_v2 *run; struct apr_pkt *pkt; @@ -1014,7 +1013,7 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags, pkt = p; run = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2; run->flags = flags; @@ -1042,10 +1041,10 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags, * * Return: Will be an negative value on error or zero on success */ -int q6asm_run(struct audio_client *ac, uint32_t flags, +int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts) { - return __q6asm_run(ac, flags, msw_ts, lsw_ts, true); + return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, true); } EXPORT_SYMBOL_GPL(q6asm_run);
@@ -1053,16 +1052,17 @@ EXPORT_SYMBOL_GPL(q6asm_run); * q6asm_run_nowait() - start the audio client withou blocking * * @ac: audio client pointer + * @stream_id: stream id * @flags: flags associated with write * @msw_ts: timestamp msw * @lsw_ts: timestamp lsw * * Return: Will be an negative value on error or zero on success */ -int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, - uint32_t msw_ts, uint32_t lsw_ts) +int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id, + uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts) { - return __q6asm_run(ac, flags, msw_ts, lsw_ts, false); + return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, false); } EXPORT_SYMBOL_GPL(q6asm_run_nowait);
@@ -1070,6 +1070,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait); * q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration * * @ac: audio client pointer + * @stream_id: stream id * @rate: audio sample rate * @channels: number of audio channels. * @channel_map: channel map pointer @@ -1078,6 +1079,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait); * Return: Will be an negative value on error or zero on success */ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, + uint32_t stream_id, uint32_t rate, uint32_t channels, u8 channel_map[PCM_MAX_NUM_CHANNEL], uint16_t bits_per_sample) @@ -1096,7 +1098,7 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1125,8 +1127,8 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, } EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
- int q6asm_stream_media_format_block_flac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_flac_cfg *cfg) { struct asm_flac_fmt_blk_v2 *fmt; @@ -1142,7 +1144,7 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1163,6 +1165,7 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg) { struct asm_wmastdv9_fmt_blk_v2 *fmt; @@ -1178,7 +1181,7 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1200,6 +1203,7 @@ int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg) { struct asm_wmaprov10_fmt_blk_v2 *fmt; @@ -1215,7 +1219,7 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1238,6 +1242,7 @@ int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
int q6asm_stream_media_format_block_alac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_alac_cfg *cfg) { struct asm_alac_fmt_blk_v2 *fmt; @@ -1253,7 +1258,7 @@ int q6asm_stream_media_format_block_alac(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1279,6 +1284,7 @@ int q6asm_stream_media_format_block_alac(struct audio_client *ac, EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
int q6asm_stream_media_format_block_ape(struct audio_client *ac, + uint32_t stream_id, struct q6asm_ape_cfg *cfg) { struct asm_ape_fmt_blk_v2 *fmt; @@ -1294,7 +1300,7 @@ int q6asm_stream_media_format_block_ape(struct audio_client *ac, pkt = p; fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2; fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk); @@ -1321,6 +1327,7 @@ EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape); * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * * @ac: audio client pointer + * @stream_id: stream id * @rate: audio sample rate * @channels: number of audio channels. * @bits_per_sample: bits per sample @@ -1328,7 +1335,9 @@ EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape); * Return: Will be an negative value on error or zero on success */ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, - uint32_t rate, uint32_t channels, uint16_t bits_per_sample) + uint32_t stream_id, uint32_t rate, + uint32_t channels, + uint16_t bits_per_sample) { struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg; struct apr_pkt *pkt; @@ -1344,7 +1353,7 @@ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
pkt = p; enc_cfg = p + APR_HDR_SIZE; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM; enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2; @@ -1376,10 +1385,11 @@ EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support); * q6asm_read() - read data of period size from audio client * * @ac: audio client pointer + * @stream_id: stream id * * Return: Will be an negative value on error or zero on success */ -int q6asm_read(struct audio_client *ac) +int q6asm_read(struct audio_client *ac, uint32_t stream_id) { struct asm_data_cmd_read_v2 *read; struct audio_port_data *port; @@ -1400,7 +1410,7 @@ int q6asm_read(struct audio_client *ac)
spin_lock_irqsave(&ac->lock, flags); port = &ac->port[SNDRV_PCM_STREAM_CAPTURE]; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id); ab = &port->buf[port->dsp_buf]; pkt->hdr.opcode = ASM_DATA_CMD_READ_V2; read->buf_addr_lsw = lower_32_bits(ab->phys); @@ -1428,7 +1438,7 @@ int q6asm_read(struct audio_client *ac) } EXPORT_SYMBOL_GPL(q6asm_read);
-static int __q6asm_open_read(struct audio_client *ac, +static int __q6asm_open_read(struct audio_client *ac, uint32_t stream_id, uint32_t format, uint16_t bits_per_sample) { struct asm_stream_cmd_open_read_v3 *open; @@ -1444,7 +1454,7 @@ static int __q6asm_open_read(struct audio_client *ac, pkt = p; open = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3; /* Stream prio : High, provide meta info with encoded frames */ open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX; @@ -1475,15 +1485,16 @@ static int __q6asm_open_read(struct audio_client *ac, * q6asm_open_read() - Open audio client for reading * * @ac: audio client pointer + * @stream_id: stream id * @format: audio sample format * @bits_per_sample: bits per sample * * Return: Will be an negative value on error or zero on success */ -int q6asm_open_read(struct audio_client *ac, uint32_t format, - uint16_t bits_per_sample) +int q6asm_open_read(struct audio_client *ac, uint32_t stream_id, + uint32_t format, uint16_t bits_per_sample) { - return __q6asm_open_read(ac, format, bits_per_sample); + return __q6asm_open_read(ac, stream_id, format, bits_per_sample); } EXPORT_SYMBOL_GPL(q6asm_open_read);
@@ -1491,6 +1502,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read); * q6asm_write_async() - non blocking write * * @ac: audio client pointer + * @stream_id: stream id * @len: length in bytes * @msw_ts: timestamp msw * @lsw_ts: timestamp lsw @@ -1498,8 +1510,8 @@ EXPORT_SYMBOL_GPL(q6asm_open_read); * * Return: Will be an negative value on error or zero on success */ -int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, - uint32_t lsw_ts, uint32_t wflags) +int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len, + uint32_t msw_ts, uint32_t lsw_ts, uint32_t wflags) { struct asm_data_cmd_write_v2 *write; struct audio_port_data *port; @@ -1520,7 +1532,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
spin_lock_irqsave(&ac->lock, flags); port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK]; - q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id); + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
ab = &port->buf[port->dsp_buf]; pkt->hdr.token = port->dsp_buf; @@ -1567,9 +1579,9 @@ static void q6asm_reset_buf_state(struct audio_client *ac) spin_unlock_irqrestore(&ac->lock, flags); }
-static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait) +static int __q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd, + bool wait) { - int stream_id = ac->stream_id; struct apr_pkt pkt; int rc;
@@ -1616,13 +1628,14 @@ static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait) * q6asm_cmd() - run cmd on audio client * * @ac: audio client pointer + * @stream_id: stream id * @cmd: command to run on audio client. * * Return: Will be an negative value on error or zero on success */ -int q6asm_cmd(struct audio_client *ac, int cmd) +int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd) { - return __q6asm_cmd(ac, cmd, true); + return __q6asm_cmd(ac, stream_id, cmd, true); } EXPORT_SYMBOL_GPL(q6asm_cmd);
@@ -1630,13 +1643,14 @@ EXPORT_SYMBOL_GPL(q6asm_cmd); * q6asm_cmd_nowait() - non blocking, run cmd on audio client * * @ac: audio client pointer + * @stream_id: stream id * @cmd: command to run on audio client. * * Return: Will be an negative value on error or zero on success */ -int q6asm_cmd_nowait(struct audio_client *ac, int cmd) +int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd) { - return __q6asm_cmd(ac, cmd, false); + return __q6asm_cmd(ac, stream_id, cmd, false); } EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 38a207d6cd95..ceece124dd3d 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -93,37 +93,47 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, void *priv, int session_id, int perf_mode); void q6asm_audio_client_free(struct audio_client *ac); -int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, - uint32_t lsw_ts, uint32_t flags); -int q6asm_open_write(struct audio_client *ac, uint32_t format, - u32 codec_profile, uint16_t bits_per_sample); - -int q6asm_open_read(struct audio_client *ac, uint32_t format, +int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len, + uint32_t msw_ts, uint32_t lsw_ts, uint32_t flags); +int q6asm_open_write(struct audio_client *ac, uint32_t stream_id, + uint32_t format, u32 codec_profile, uint16_t bits_per_sample); + +int q6asm_open_read(struct audio_client *ac, uint32_t stream_id, + uint32_t format, uint16_t bits_per_sample); int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, - uint32_t rate, uint32_t channels, uint16_t bits_per_sample); -int q6asm_read(struct audio_client *ac); + uint32_t stream_id, uint32_t rate, + uint32_t channels, + uint16_t bits_per_sample); + +int q6asm_read(struct audio_client *ac, uint32_t stream_id);
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, + uint32_t stream_id, uint32_t rate, uint32_t channels, u8 channel_map[PCM_MAX_NUM_CHANNEL], uint16_t bits_per_sample); int q6asm_stream_media_format_block_flac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_flac_cfg *cfg); int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg); int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac, + uint32_t stream_id, struct q6asm_wma_cfg *cfg); int q6asm_stream_media_format_block_alac(struct audio_client *ac, + uint32_t stream_id, struct q6asm_alac_cfg *cfg); int q6asm_stream_media_format_block_ape(struct audio_client *ac, + uint32_t stream_id, struct q6asm_ape_cfg *cfg); -int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, - uint32_t lsw_ts); -int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts, - uint32_t lsw_ts); -int q6asm_cmd(struct audio_client *ac, int cmd); -int q6asm_cmd_nowait(struct audio_client *ac, int cmd); +int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags, + uint32_t msw_ts, uint32_t lsw_ts); +int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id, + uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); +int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd); +int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd); int q6asm_get_session_id(struct audio_client *ac); int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 941f3216399c..fb0488e7beb9 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -67,6 +67,8 @@ struct q6asm_dai_rtd { uint16_t bits_per_sample; uint16_t source; /* Encoding source bit mask */ struct audio_client *audio_client;
- /* Active */
nit-pick: what does this 'Active' comment try to say? the stream_id seems to be used for RUN/EOS/CLOSE operations.
- uint32_t stream_id; uint16_t session_id; enum stream_state state; };
@@ -184,8 +186,8 @@ 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->pcm_count, 0, 0, NO_TIMESTAMP);
q6asm_write_async(prtd->audio_client, prtd->stream_id,prtd->pcm_count, 0, 0, 0);
In the V1 review we discussed this
"
sound/soc/qcom/qdsp6/q6asm.h:#define NO_TIMESTAMP 0xFF00
is the change on the previous line intentional?
May be not!
Plan is that the users of these apis will send flags directly instead of boiler plating this!
This change should go as part of next patch("[PATCH 04/11] ASoC: q6asm: use flags directly from asm-dai") which would make it much clear! "
doesn't look like there was a change here?
Thanks Pierre for review,
On 21/07/2020 20:31, Pierre-Louis Bossart wrote:
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 941f3216399c..fb0488e7beb9 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -67,6 +67,8 @@ struct q6asm_dai_rtd { uint16_t bits_per_sample; uint16_t source; /* Encoding source bit mask */ struct audio_client *audio_client; + /* Active */
nit-pick: what does this 'Active' comment try to say? the stream_id seems to be used for RUN/EOS/CLOSE operations.
Active mean its the active stream id which is consuming the data at the point in time. As we toggle stream ids between 1 and 2. This active stream_id is used for every command sent to dsp.
+ uint32_t stream_id; uint16_t session_id; enum stream_state state; }; @@ -184,8 +186,8 @@ 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->pcm_count, 0, 0, NO_TIMESTAMP); + q6asm_write_async(prtd->audio_client, prtd->stream_id, + prtd->pcm_count, 0, 0, 0);
In the V1 review we discussed this
Sorry, I missed that! will address this in next version!
"
sound/soc/qcom/qdsp6/q6asm.h:#define NO_TIMESTAMP 0xFF00
is the change on the previous line intentional?
May be not!
Plan is that the users of these apis will send flags directly instead of boiler plating this!
This change should go as part of next patch("[PATCH 04/11] ASoC: q6asm: use flags directly from asm-dai") which would make it much clear! "
doesn't look like there was a change here?
use flags set by q6asm-dais directly!
This will be useful gapless case where write needs a special flag to indicate that last buffer.
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index f5d1f3c2c1ec..d6728304ce6a 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -1546,10 +1546,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len, write->mem_map_handle = ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
- if (wflags == NO_TIMESTAMP) - write->flags = (wflags & 0x800000FF); - else - write->flags = (0x80000000 | wflags); + write->flags = wflags;
port->dsp_buf++;
Add length to write command packet token so that we can track exactly how many bytes are consumed by DSP in the command reply.
This is useful in some use-cases where the end of the file/stream is not aligned with period size.
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +++- sound/soc/qcom/qdsp6/q6asm.c | 7 ++++--- sound/soc/qcom/qdsp6/q6asm.h | 3 +++ 3 files changed, 10 insertions(+), 4 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index fb0488e7beb9..6b9ceac2ceb2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -510,6 +510,7 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, struct snd_compr_stream *substream = prtd->cstream; unsigned long flags; uint64_t avail; + uint32_t bytes_written;
switch (opcode) { case ASM_CLIENT_EVENT_CMD_RUN_DONE: @@ -530,7 +531,8 @@ static void compress_event_handler(uint32_t opcode, uint32_t token, case ASM_CLIENT_EVENT_DATA_WRITE_DONE: spin_lock_irqsave(&prtd->lock, flags);
- prtd->copied_total += prtd->pcm_count; + bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT; + prtd->copied_total += bytes_written; snd_compr_fragment_elapsed(substream);
if (prtd->state != Q6ASM_STREAM_RUNNING) { diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index d6728304ce6a..205453d1c1fc 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -670,6 +670,7 @@ static int32_t q6asm_stream_callback(struct apr_device *adev, if (ac->io_mode & ASM_SYNC_IO_MODE) { phys_addr_t phys; unsigned long flags; + int token = hdr->token & ASM_WRITE_TOKEN_MASK;
spin_lock_irqsave(&ac->lock, flags);
@@ -681,12 +682,12 @@ static int32_t q6asm_stream_callback(struct apr_device *adev, goto done; }
- phys = port->buf[hdr->token].phys; + phys = port->buf[token].phys;
if (lower_32_bits(phys) != result->opcode || upper_32_bits(phys) != result->status) { dev_err(ac->dev, "Expected addr %pa\n", - &port->buf[hdr->token].phys); + &port->buf[token].phys); spin_unlock_irqrestore(&ac->lock, flags); ret = -EINVAL; goto done; @@ -1535,7 +1536,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len, q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
ab = &port->buf[port->dsp_buf]; - pkt->hdr.token = port->dsp_buf; + pkt->hdr.token = port->dsp_buf | (len << ASM_WRITE_TOKEN_LEN_SHIFT); pkt->hdr.opcode = ASM_DATA_CMD_WRITE_V2; write->buf_addr_lsw = lower_32_bits(ab->phys); write->buf_addr_msw = upper_32_bits(ab->phys); diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index ceece124dd3d..0379580f0742 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -20,6 +20,9 @@ #define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008 #define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009 #define ASM_CLIENT_EVENT_DATA_READ_DONE 0x100a +#define ASM_WRITE_TOKEN_MASK GENMASK(15, 0) +#define ASM_WRITE_TOKEN_LEN_MASK GENMASK(31, 16) +#define ASM_WRITE_TOKEN_LEN_SHIFT 16
enum { LEGACY_PCM_MODE = 0,
This patch adds support to ASM_DATA_CMD_REMOVE_INITIAL_SILENCE and ASM_DATA_CMD_REMOVE_TRAILING_SILENCE q6asm command to support compressed metadata for gapless playback.
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm.c | 53 ++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 6 ++++ 2 files changed, 59 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 205453d1c1fc..14ec7dad5b65 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -51,6 +51,8 @@ #define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D #define ASM_MEDIA_FMT_ALAC 0x00012f31 #define ASM_MEDIA_FMT_APE 0x00012f32 +#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67 +#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68
#define ASM_LEGACY_STREAM_SESSION 0 @@ -639,6 +641,8 @@ static int32_t q6asm_stream_callback(struct apr_device *adev, case ASM_STREAM_CMD_OPEN_READWRITE_V2: case ASM_STREAM_CMD_SET_ENCDEC_PARAM: case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2: + case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE: + case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE: if (result->status != 0) { dev_err(ac->dev, "cmd = 0x%x returned error = 0x%x\n", @@ -1324,6 +1328,55 @@ int q6asm_stream_media_format_block_ape(struct audio_client *ac, } EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+static int q6asm_stream_remove_silence(struct audio_client *ac, uint32_t stream_id, + uint32_t cmd, + uint32_t num_samples) +{ + uint32_t *samples; + struct apr_pkt *pkt; + void *p; + int rc, pkt_size; + + pkt_size = APR_HDR_SIZE + sizeof(uint32_t); + p = kzalloc(pkt_size, GFP_ATOMIC); + if (!p) + return -ENOMEM; + + pkt = p; + samples = p + APR_HDR_SIZE; + + q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id); + + pkt->hdr.opcode = cmd; + *samples = num_samples; + rc = apr_send_pkt(ac->adev, pkt); + if (rc == pkt_size) + rc = 0; + + kfree(pkt); + + return rc; +} + +int q6asm_stream_remove_initial_silence(struct audio_client *ac, + uint32_t stream_id, + uint32_t initial_samples) +{ + return q6asm_stream_remove_silence(ac, stream_id, + ASM_DATA_CMD_REMOVE_INITIAL_SILENCE, + initial_samples); +} +EXPORT_SYMBOL_GPL(q6asm_stream_remove_initial_silence); + +int q6asm_stream_remove_trailing_silence(struct audio_client *ac, uint32_t stream_id, + uint32_t trailing_samples) +{ + return q6asm_stream_remove_silence(ac, stream_id, + ASM_DATA_CMD_REMOVE_TRAILING_SILENCE, + trailing_samples); +} +EXPORT_SYMBOL_GPL(q6asm_stream_remove_trailing_silence); + /** * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 0379580f0742..e315f7ff5e54 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -135,6 +135,12 @@ int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id, uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts); +int q6asm_stream_remove_initial_silence(struct audio_client *ac, + uint32_t stream_id, + uint32_t initial_samples); +int q6asm_stream_remove_trailing_silence(struct audio_client *ac, + uint32_t stream_id, + uint32_t trailing_samples); int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd); int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd); int q6asm_get_session_id(struct audio_client *ac);
This patch adds support to gapless flag to q6asm_open_write().
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm.c | 4 +++- sound/soc/qcom/qdsp6/q6asm.h | 2 +- 3 files changed, 6 insertions(+), 4 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 6b9ceac2ceb2..a493350c8cc2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -259,7 +259,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, FORMAT_LINEAR_PCM, - 0, prtd->bits_per_sample); + 0, prtd->bits_per_sample, false); } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, FORMAT_LINEAR_PCM, @@ -686,7 +686,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, 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); + prtd->bits_per_sample, true);
if (ret < 0) { dev_err(dev, "q6asm_open_write failed\n"); diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 14ec7dad5b65..22ac99029e56 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -923,7 +923,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt) */ int q6asm_open_write(struct audio_client *ac, uint32_t stream_id, uint32_t format, u32 codec_profile, - uint16_t bits_per_sample) + uint16_t bits_per_sample, bool is_gapless) { struct asm_stream_cmd_open_write_v3 *open; struct apr_pkt *pkt; @@ -943,6 +943,8 @@ int q6asm_open_write(struct audio_client *ac, uint32_t stream_id, pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3; open->mode_flags = 0x00; open->mode_flags |= ASM_LEGACY_STREAM_SESSION; + if (is_gapless) + open->mode_flags |= BIT(ASM_SHIFT_GAPLESS_MODE_FLAG);
/* source endpoint : matrix */ open->sink_endpointype = ASM_END_POINT_DEVICE_MATRIX; diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index e315f7ff5e54..69513ac1fa55 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -100,7 +100,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len, uint32_t msw_ts, uint32_t lsw_ts, uint32_t flags); int q6asm_open_write(struct audio_client *ac, uint32_t stream_id, uint32_t format, u32 codec_profile, - uint16_t bits_per_sample); + uint16_t bits_per_sample, bool is_gapless);
int q6asm_open_read(struct audio_client *ac, uint32_t stream_id, uint32_t format, uint16_t bits_per_sample);
This patch adds support to metadata required to do a gapless playback.
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm-dai.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index a493350c8cc2..c4b4684b7824 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -71,6 +71,8 @@ struct q6asm_dai_rtd { uint32_t stream_id; uint16_t session_id; enum stream_state state; + uint32_t initial_samples_drop; + uint32_t trailing_samples_drop; };
struct q6asm_dai_data { @@ -869,6 +871,28 @@ 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, + struct snd_compr_metadata *metadata) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + int ret = 0; + + switch (metadata->key) { + case SNDRV_COMPRESS_ENCODER_PADDING: + prtd->trailing_samples_drop = metadata->value[0]; + break; + case SNDRV_COMPRESS_ENCODER_DELAY: + prtd->initial_samples_drop = metadata->value[0]; + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + static int q6asm_dai_compr_trigger(struct snd_soc_component *component, struct snd_compr_stream *stream, int cmd) { @@ -985,6 +1009,7 @@ static struct snd_compress_ops q6asm_dai_compress_ops = { .open = q6asm_dai_compr_open, .free = q6asm_dai_compr_free, .set_params = q6asm_dai_compr_set_params, + .set_metadata = q6asm_dai_compr_set_metadata, .pointer = q6asm_dai_compr_pointer, .trigger = q6asm_dai_compr_trigger, .get_caps = q6asm_dai_compr_get_caps,
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@linaro.org --- sound/soc/qcom/qdsp6/q6asm-dai.c | 253 ++++++++++++++++++++++--------- sound/soc/qcom/qdsp6/q6asm.h | 1 + 2 files changed, 182 insertions(+), 72 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c4b4684b7824..50055c113f10 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -53,7 +53,7 @@ enum stream_state { struct q6asm_dai_rtd { struct snd_pcm_substream *substream; struct snd_compr_stream *cstream; - struct snd_compr_params codec_param; + struct snd_codec codec; struct snd_dma_buffer dma_buffer; spinlock_t lock; phys_addr_t phys; @@ -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 { @@ -510,14 +513,20 @@ 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: 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; @@ -527,7 +536,26 @@ 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); + 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: @@ -543,13 +571,32 @@ 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 +676,15 @@ 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 +698,13 @@ 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_codec *codec, + 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; @@ -667,53 +718,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, struct snd_dec_alac *alac; struct snd_dec_ape *ape;
- 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; - } - } + codec_options = &(prtd->codec.options);
- 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; - } + memcpy(&prtd->codec, codec, sizeof(*codec));
- switch (params->codec.id) { + switch (codec->id) { case SND_AUDIOCODEC_FLAC:
memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg)); flac = &codec_options->flac_d;
- flac_cfg.ch_cfg = params->codec.ch_in; - flac_cfg.sample_rate = params->codec.sample_rate; + flac_cfg.ch_cfg = codec->ch_in; + flac_cfg.sample_rate = codec->sample_rate; flac_cfg.stream_info_present = 1; flac_cfg.sample_size = flac->sample_size; flac_cfg.min_blk_size = flac->min_blk_size; @@ -722,7 +738,7 @@ 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); @@ -735,10 +751,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
- wma_cfg.sample_rate = params->codec.sample_rate; - wma_cfg.num_channels = params->codec.ch_in; - wma_cfg.bytes_per_sec = params->codec.bit_rate / 8; - wma_cfg.block_align = params->codec.align; + wma_cfg.sample_rate = codec->sample_rate; + wma_cfg.num_channels = codec->ch_in; + wma_cfg.bytes_per_sec = codec->bit_rate / 8; + wma_cfg.block_align = codec->align; wma_cfg.bits_per_sample = prtd->bits_per_sample; wma_cfg.enc_options = wma->encoder_option; wma_cfg.adv_enc_options = wma->adv_encoder_option; @@ -752,7 +768,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, return -EINVAL;
/* check the codec profile */ - switch (params->codec.profile) { + switch (codec->profile) { case SND_AUDIOPROFILE_WMA9: wma_cfg.fmtag = 0x161; wma_v9 = 1; @@ -776,17 +792,17 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
default: dev_err(dev, "Unknown WMA profile:%x\n", - params->codec.profile); + codec->profile); return -EIO; }
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); @@ -798,10 +814,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&alac_cfg, 0x0, sizeof(alac_cfg)); alac = &codec_options->alac_d;
- alac_cfg.sample_rate = params->codec.sample_rate; - alac_cfg.avg_bit_rate = params->codec.bit_rate; + alac_cfg.sample_rate = codec->sample_rate; + alac_cfg.avg_bit_rate = codec->bit_rate; alac_cfg.bit_depth = prtd->bits_per_sample; - alac_cfg.num_channels = params->codec.ch_in; + alac_cfg.num_channels = codec->ch_in;
alac_cfg.frame_length = alac->frame_length; alac_cfg.pb = alac->pb; @@ -811,7 +827,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, alac_cfg.compatible_version = alac->compatible_version; alac_cfg.max_frame_bytes = alac->max_frame_bytes;
- switch (params->codec.ch_in) { + switch (codec->ch_in) { case 1: alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO; break; @@ -820,7 +836,7 @@ 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); @@ -832,8 +848,8 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&ape_cfg, 0x0, sizeof(ape_cfg)); ape = &codec_options->ape_d;
- ape_cfg.sample_rate = params->codec.sample_rate; - ape_cfg.num_channels = params->codec.ch_in; + ape_cfg.sample_rate = codec->sample_rate; + ape_cfg.num_channels = codec->ch_in; ape_cfg.bits_per_sample = prtd->bits_per_sample;
ape_cfg.compatible_version = ape->compatible_version; @@ -845,7 +861,7 @@ 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 +873,64 @@ 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, + ¶ms->codec, + 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); @@ -871,7 +945,8 @@ 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; @@ -884,6 +959,32 @@ 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) { + 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; @@ -917,6 +1018,14 @@ 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;
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) {/**
why the kernel-doc style comment?
[...]
-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_codec *codec,int stream_id)
not sure I get why you added the __ prefix, does it have any semantic meaning?
{ 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;
@@ -667,53 +718,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, struct snd_dec_alac *alac; struct snd_dec_ape *ape;
- 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;}- }
- codec_options = &(prtd->codec.options);
- 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;- }
- memcpy(&prtd->codec, codec, sizeof(*codec));
- switch (params->codec.id) {
switch (codec->id) { case SND_AUDIOCODEC_FLAC:
memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg)); flac = &codec_options->flac_d;
flac_cfg.ch_cfg = params->codec.ch_in;flac_cfg.sample_rate = params->codec.sample_rate;
all these indirection changes could have gone in a earlier patch, this adds a lot of changes that make this patch long to review. Same comment for all formats
flac_cfg.ch_cfg = codec->ch_in;flac_cfg.sample_rate = codec->sample_rate;
[...]
-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)@@ -884,6 +959,32 @@ 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) {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;
@@ -917,6 +1018,14 @@ 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);
it's rather odd, the initialization above uses next_track_stream_id = 0?
break;- case SND_COMPR_TRIGGER_DRAIN:
- case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
prtd->notify_on_drain = true;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;
Thanks Pierre for quick review!
On 21/07/2020 20:53, Pierre-Louis Bossart wrote:
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) { + /**
why the kernel-doc style comment?
Nothing intentional, will fix this!
[...]
-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_codec *codec, + int stream_id)
not sure I get why you added the __ prefix, does it have any semantic meaning?
Nope, just to mark them it as internal function, as the function name is very much similar to q6asm_dai_compr_set_params() callback! I will try to come up with better naming and also move the indirection changes to a separate patch!
{ 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; @@ -667,53 +718,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, struct snd_dec_alac *alac; struct snd_dec_ape *ape; - 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; - } - } + codec_options = &(prtd->codec.options); - 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; - } + memcpy(&prtd->codec, codec, sizeof(*codec)); - switch (params->codec.id) { + switch (codec->id) { case SND_AUDIOCODEC_FLAC: memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg)); flac = &codec_options->flac_d; - flac_cfg.ch_cfg = params->codec.ch_in; - flac_cfg.sample_rate = params->codec.sample_rate;
all these indirection changes could have gone in a earlier patch, this adds a lot of changes that make this patch long to review. Same comment for all formats
+ flac_cfg.ch_cfg = codec->ch_in; + flac_cfg.sample_rate = codec->sample_rate; flac_cfg.stream_info_present = 1; flac_cfg.sample_size = flac->sample_size; flac_cfg.min_blk_size = flac->min_blk_size;
[...]
...
@@ -917,6 +1018,14 @@ 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);
it's rather odd, the initialization above uses next_track_stream_id = 0?
Vaild stream ids start from 1, So we are toggling this between 1 and 2. So when we set next_track_stream_id to 0, that means we have opened the new next stream id and is set to prtd->stream_id. This logic is to ensure that we are not going to open next stream id twice!
+ 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;
@@ -917,6 +1018,14 @@ 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);
it's rather odd, the initialization above uses next_track_stream_id = 0?
Vaild stream ids start from 1, So we are toggling this between 1 and 2. So when we set next_track_stream_id to 0, that means we have opened the new next stream id and is set to prtd->stream_id. This logic is to ensure that we are not going to open next stream id twice!
ok, adding a comment would be good to show this was intentional and not a mistake.
During gapless playback, its possible for previous track to end at unaligned boundary, starting next track on the same boundary can lead to unaligned address exception in dsp.
So implement copy callback for finer control on the buffer offsets.
Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6asm-dai.c | 65 +++++++++++++++++++++++++++++--- 1 file changed, 60 insertions(+), 5 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 50055c113f10..b5c719682919 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -1052,16 +1052,71 @@ static int q6asm_dai_compr_pointer(struct snd_soc_component *component, return 0; }
-static int q6asm_dai_compr_ack(struct snd_soc_component *component, - struct snd_compr_stream *stream, - size_t count) +static int q6asm_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *stream, char __user *buf, + size_t count) { struct snd_compr_runtime *runtime = stream->runtime; struct q6asm_dai_rtd *prtd = runtime->private_data; unsigned long flags; + u32 wflags = 0; + int avail, bytes_in_flight = 0; + void *dstn; + size_t copy; + u32 app_pointer; + u32 bytes_received; + + bytes_received = prtd->bytes_received; + + /** + * Make sure that next track data pointer is aligned at 32 bit boundary + * This is a Mandatory requirement from DSP data buffers alignment + */ + if (prtd->next_track) + bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count); + + app_pointer = bytes_received/prtd->pcm_size; + app_pointer = bytes_received - (app_pointer * prtd->pcm_size); + dstn = prtd->dma_buffer.area + app_pointer; + + if (count < prtd->pcm_size - app_pointer) { + if (copy_from_user(dstn, buf, count)) + return -EFAULT; + } else { + copy = prtd->pcm_size - app_pointer; + if (copy_from_user(dstn, buf, copy)) + return -EFAULT; + if (copy_from_user(prtd->dma_buffer.area, buf + copy, + count - copy)) + return -EFAULT; + }
spin_lock_irqsave(&prtd->lock, flags); - prtd->bytes_received += count; + + bytes_in_flight = prtd->bytes_received - prtd->copied_total; + + if (prtd->next_track) { + prtd->next_track = false; + prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count); + prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count); + } + + prtd->bytes_received = bytes_received + count; + + /* Kick off the data to dsp if its starving!! */ + if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) { + uint32_t bytes_to_write = prtd->pcm_count; + + avail = prtd->bytes_received - prtd->bytes_sent; + + if (avail < prtd->pcm_count) + bytes_to_write = avail; + + q6asm_write_async(prtd->audio_client, prtd->stream_id, + bytes_to_write, 0, 0, wflags); + prtd->bytes_sent += bytes_to_write; + } + spin_unlock_irqrestore(&prtd->lock, flags);
return count; @@ -1124,7 +1179,7 @@ static struct snd_compress_ops q6asm_dai_compress_ops = { .get_caps = q6asm_dai_compr_get_caps, .get_codec_caps = q6asm_dai_compr_get_codec_caps, .mmap = q6asm_dai_compr_mmap, - .ack = q6asm_dai_compr_ack, + .copy = q6asm_compr_copy, };
static int q6asm_dai_pcm_new(struct snd_soc_component *component,
+static int q6asm_compr_copy(struct snd_soc_component *component,
struct snd_compr_stream *stream, char __user *buf,size_t count)- u32 wflags = 0;
- int avail, bytes_in_flight = 0;
- void *dstn;
- size_t copy;
- u32 app_pointer;
- u32 bytes_received;
- bytes_received = prtd->bytes_received;
- /**
/*
* Make sure that next track data pointer is aligned at 32 bit boundary* This is a Mandatory requirement from DSP data buffers alignment*/- if (prtd->next_track)
bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);- app_pointer = bytes_received/prtd->pcm_size;
- app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
- dstn = prtd->dma_buffer.area + app_pointer;
- if (count < prtd->pcm_size - app_pointer) {
if (copy_from_user(dstn, buf, count))return -EFAULT;- } else {
copy = prtd->pcm_size - app_pointer;if (copy_from_user(dstn, buf, copy))return -EFAULT;if (copy_from_user(prtd->dma_buffer.area, buf + copy,count - copy))return -EFAULT;- }
participants (2)
-
Pierre-Louis Bossart -
Srinivas Kandagatla