| Message ID | 20200707163641.17113-8-srinivas.kandagatla@linaro.org (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | ASoC: qdsp6: add gapless compressed audio support | expand |
> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c > index c3558288242a..8c214436a2c2 100644 > --- a/sound/soc/qcom/qdsp6/q6asm-dai.c > +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c > @@ -258,7 +258,7 @@ 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); nit-pick: it's a bit ironic that is_gapless is false for PCM, when there is no gap in the first place.. > } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { > ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, > FORMAT_LINEAR_PCM, > @@ -685,7 +685,7 @@ 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);
Thanks Pierre for review, On 07/07/2020 17:57, Pierre-Louis Bossart wrote: > >> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c >> b/sound/soc/qcom/qdsp6/q6asm-dai.c >> index c3558288242a..8c214436a2c2 100644 >> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c >> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c >> @@ -258,7 +258,7 @@ 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); > > nit-pick: it's a bit ironic that is_gapless is false for PCM, when there > is no gap in the first place.. I think this is to do with same apis reused for both compressed and pcm. Probably we can live with it for now! --srini > >> } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { >> ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, >> FORMAT_LINEAR_PCM, >> @@ -685,7 +685,7 @@ 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);
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c3558288242a..8c214436a2c2 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -258,7 +258,7 @@ 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, @@ -685,7 +685,7 @@ 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 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(-)