| Message ID | 20200721170007.4554-7-srinivas.kandagatla@linaro.org (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | ALSA: compress: add support to change codec profile in gapless playback | expand |
On 7/21/20 12:00 PM, Srinivas Kandagatla wrote: > Make use of new set_codec_params callback to allow decoder switching > during gapless playback. > > Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> > --- > sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ++++++++++++++++++++++++++++++++ > 1 file changed, 33 insertions(+) > > diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c > index b5c719682919..a8cfb1996614 100644 > --- a/sound/soc/qcom/qdsp6/q6asm-dai.c > +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c > @@ -876,6 +876,37 @@ static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *componen > return 0; > } > > +static int q6asm_dai_compr_set_codec_params(struct snd_soc_component *component, > + struct snd_compr_stream *stream, > + struct snd_codec *codec) > +{ > + struct snd_compr_runtime *runtime = stream->runtime; > + struct q6asm_dai_rtd *prtd = runtime->private_data; > + int ret; > + > + ret = q6asm_open_write(prtd->audio_client, prtd->next_track_stream_id, > + codec->id, codec->profile, prtd->bits_per_sample, > + true); > + if (ret < 0) { > + pr_err("q6asm_open_write failed\n"); > + return ret; > + } > + > + ret = __q6asm_dai_compr_set_codec_params(component, stream, codec, > + prtd->next_track_stream_id); > + if (ret < 0) { > + pr_err("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; same comment as in the other patchset, the stream_id toggles between 1 and 2, it's not clear to me what 0 means. off-by-one bug or feature?
Thanks Pierre for quick review. On 21/07/2020 21:09, Pierre-Louis Bossart wrote: > > > On 7/21/20 12:00 PM, Srinivas Kandagatla wrote: >> Make use of new set_codec_params callback to allow decoder switching >> during gapless playback. >> >> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> >> --- >> sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ++++++++++++++++++++++++++++++++ >> 1 file changed, 33 insertions(+) >> >> diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c >> b/sound/soc/qcom/qdsp6/q6asm-dai.c >> index b5c719682919..a8cfb1996614 100644 >> --- a/sound/soc/qcom/qdsp6/q6asm-dai.c >> +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c >> @@ -876,6 +876,37 @@ static int >> __q6asm_dai_compr_set_codec_params(struct snd_soc_component *componen >> return 0; >> } >> +static int q6asm_dai_compr_set_codec_params(struct snd_soc_component >> *component, >> + struct snd_compr_stream *stream, >> + struct snd_codec *codec) >> +{ >> + struct snd_compr_runtime *runtime = stream->runtime; >> + struct q6asm_dai_rtd *prtd = runtime->private_data; >> + int ret; >> + >> + ret = q6asm_open_write(prtd->audio_client, >> prtd->next_track_stream_id, >> + codec->id, codec->profile, prtd->bits_per_sample, >> + true); >> + if (ret < 0) { >> + pr_err("q6asm_open_write failed\n"); >> + return ret; >> + } >> + >> + ret = __q6asm_dai_compr_set_codec_params(component, stream, codec, >> + prtd->next_track_stream_id); >> + if (ret < 0) { >> + pr_err("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; > > same comment as in the other patchset, the stream_id toggles between 1 > and 2, it's not clear to me what 0 means. Valid stream ids start from 1. to achieve gapless we toggle between 1 and 2. --srini > > off-by-one bug or feature?
On Tue, Jul 21, 2020 at 8:03 PM Srinivas Kandagatla <srinivas.kandagatla@linaro.org> wrote: > > Make use of new set_codec_params callback to allow decoder switching > during gapless playback. > > Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> > --- > sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ++++++++++++++++++++++++++++++++ > 1 file changed, 33 insertions(+) > > diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c > index b5c719682919..a8cfb1996614 100644 > --- a/sound/soc/qcom/qdsp6/q6asm-dai.c > +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c > @@ -876,6 +876,37 @@ static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *componen > return 0; > } > > +static int q6asm_dai_compr_set_codec_params(struct snd_soc_component *component, > + struct snd_compr_stream *stream, > + struct snd_codec *codec) > +{ > + struct snd_compr_runtime *runtime = stream->runtime; > + struct q6asm_dai_rtd *prtd = runtime->private_data; > + int ret; > + > + ret = q6asm_open_write(prtd->audio_client, prtd->next_track_stream_id, > + codec->id, codec->profile, prtd->bits_per_sample, > + true); > + if (ret < 0) { > + pr_err("q6asm_open_write failed\n"); Since you have component->dev here I think it is worth it to use dev_err instead of pr_err. Same for the rest of the code.
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index b5c719682919..a8cfb1996614 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -876,6 +876,37 @@ static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *componen return 0; } +static int q6asm_dai_compr_set_codec_params(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_codec *codec) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + int ret; + + ret = q6asm_open_write(prtd->audio_client, prtd->next_track_stream_id, + codec->id, codec->profile, prtd->bits_per_sample, + true); + if (ret < 0) { + pr_err("q6asm_open_write failed\n"); + return ret; + } + + ret = __q6asm_dai_compr_set_codec_params(component, stream, codec, + prtd->next_track_stream_id); + if (ret < 0) { + pr_err("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; + + return ret; +} + static int q6asm_dai_compr_set_params(struct snd_soc_component *component, struct snd_compr_stream *stream, struct snd_compr_params *params) @@ -1144,6 +1175,7 @@ static int q6asm_dai_compr_get_caps(struct snd_soc_component *component, caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; + caps->flags = SND_COMPR_CAP_FLAGS_DSP_CAN_SWITCH_DECODER; caps->num_codecs = 5; caps->codecs[0] = SND_AUDIOCODEC_MP3; caps->codecs[1] = SND_AUDIOCODEC_FLAC; @@ -1173,6 +1205,7 @@ 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_codec_params = q6asm_dai_compr_set_codec_params, .set_metadata = q6asm_dai_compr_set_metadata, .pointer = q6asm_dai_compr_pointer, .trigger = q6asm_dai_compr_trigger,
Make use of new set_codec_params callback to allow decoder switching during gapless playback. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 33 ++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+)