| Message ID | 20220118212732.281657-3-daniel.baluta@oss.nxp.com (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | SOF: Add compress support implementation | expand |
On 1/18/22 3:27 PM, Daniel Baluta wrote: > From: Paul Olaru <paul.olaru@nxp.com> > > These functions are used by the userspace to determine what the firmware > supports and tools like cplay should use in terms of sample rate, bit > rate, buffer size and channel count. > > The current implementation uses i.MX8 tested scenarios! > > Signed-off-by: Paul Olaru <paul.olaru@nxp.com> > Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com> > --- > sound/soc/sof/compress.c | 74 ++++++++++++++++++++++++++++++++++++++++ > 1 file changed, 74 insertions(+) > > diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c > index 91a9c95929cd..e3f3f309f312 100644 > --- a/sound/soc/sof/compress.c > +++ b/sound/soc/sof/compress.c > @@ -308,6 +308,78 @@ static int sof_compr_pointer(struct snd_soc_component *component, > return 0; > } > > +static int sof_compr_get_caps(struct snd_soc_component *component, > + struct snd_compr_stream *cstream, > + struct snd_compr_caps *caps) > +{ > + caps->num_codecs = 3; > + caps->min_fragment_size = 3840; > + caps->max_fragment_size = 3840; > + caps->min_fragments = 2; > + caps->max_fragments = 2; > + caps->codecs[0] = SND_AUDIOCODEC_MP3; > + caps->codecs[1] = SND_AUDIOCODEC_AAC; > + caps->codecs[2] = SND_AUDIOCODEC_PCM; I don't think you can add this unconditionally for all devices/platforms, clearly this wouldn't be true for Intel for now. If the information is not part of a firmware manifest or topology, then it's likely we have to use an abstraction layer to add this for specific platforms. it's really a bit odd to hard-code all of this at the kernel level, this was not really what I had in mind when we come up with the concept of querying capabilities. I understand though that for testing this is convenient, so maybe this can become a set of fall-back properties in case the firmware doesn't advertise anything. > + > + return 0; > +} > + > +static struct snd_compr_codec_caps caps_pcm = { > + .num_descriptors = 1, > + .descriptor[0].max_ch = 2, > + .descriptor[0].sample_rates[0] = 48000, > + .descriptor[0].num_sample_rates = 1, > + .descriptor[0].bit_rate = {1536, 3072}, > + .descriptor[0].num_bitrates = 2, > + .descriptor[0].profiles = SND_AUDIOPROFILE_PCM, > + .descriptor[0].modes = 0, > + .descriptor[0].formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, > +}; > + > +static struct snd_compr_codec_caps caps_mp3 = { > + .num_descriptors = 1, > + .descriptor[0].max_ch = 2, > + .descriptor[0].sample_rates[0] = 48000, > + .descriptor[0].num_sample_rates = 1, > + .descriptor[0].bit_rate = {32, 40, 48, 56, 64, 80, 96, 112, 224, 256, 320}, > + .descriptor[0].num_bitrates = 11, > + .descriptor[0].profiles = 0, > + .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO, > + .descriptor[0].formats = 0, > +}; > + > +static struct snd_compr_codec_caps caps_aac = { > + .num_descriptors = 1, > + .descriptor[0].max_ch = 2, > + .descriptor[0].sample_rates[0] = 48000, > + .descriptor[0].num_sample_rates = 1, > + .descriptor[0].bit_rate = {128, 192}, > + .descriptor[0].num_bitrates = 2, > + .descriptor[0].profiles = 0, > + .descriptor[0].modes = 0, > + .descriptor[0].formats = SND_AUDIOSTREAMFORMAT_MP4ADTS | SND_AUDIOSTREAMFORMAT_MP2ADTS, > +}; > + > +static int sof_compr_get_codec_caps(struct snd_soc_component *component, > + struct snd_compr_stream *cstream, > + struct snd_compr_codec_caps *codec) > +{ > + switch (codec->codec) { > + case SND_AUDIOCODEC_MP3: > + *codec = caps_mp3; > + break; > + case SND_AUDIOCODEC_AAC: > + *codec = caps_aac; > + break; > + case SND_AUDIOCODEC_PCM: > + *codec = caps_pcm; > + break; > + default: > + return -EINVAL; > + } > + return 0; > +} > + > struct snd_compress_ops sof_compressed_ops = { > .open = sof_compr_open, > .free = sof_compr_free, > @@ -316,5 +388,7 @@ struct snd_compress_ops sof_compressed_ops = { > .trigger = sof_compr_trigger, > .pointer = sof_compr_pointer, > .copy = sof_compr_copy, > + .get_caps = sof_compr_get_caps, > + .get_codec_caps = sof_compr_get_codec_caps, > }; > EXPORT_SYMBOL(sof_compressed_ops); >
On 1/19/22 3:00 AM, Pierre-Louis Bossart wrote: > > > On 1/18/22 3:27 PM, Daniel Baluta wrote: >> From: Paul Olaru <paul.olaru@nxp.com> >> >> These functions are used by the userspace to determine what the firmware >> supports and tools like cplay should use in terms of sample rate, bit >> rate, buffer size and channel count. >> >> The current implementation uses i.MX8 tested scenarios! >> >> Signed-off-by: Paul Olaru <paul.olaru@nxp.com> >> Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com> >> --- >> sound/soc/sof/compress.c | 74 ++++++++++++++++++++++++++++++++++++++++ >> 1 file changed, 74 insertions(+) >> >> diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c >> index 91a9c95929cd..e3f3f309f312 100644 >> --- a/sound/soc/sof/compress.c >> +++ b/sound/soc/sof/compress.c >> @@ -308,6 +308,78 @@ static int sof_compr_pointer(struct snd_soc_component *component, >> return 0; >> } >> >> +static int sof_compr_get_caps(struct snd_soc_component *component, >> + struct snd_compr_stream *cstream, >> + struct snd_compr_caps *caps) >> +{ >> + caps->num_codecs = 3; >> + caps->min_fragment_size = 3840; >> + caps->max_fragment_size = 3840; >> + caps->min_fragments = 2; >> + caps->max_fragments = 2; >> + caps->codecs[0] = SND_AUDIOCODEC_MP3; >> + caps->codecs[1] = SND_AUDIOCODEC_AAC; >> + caps->codecs[2] = SND_AUDIOCODEC_PCM; > > I don't think you can add this unconditionally for all > devices/platforms, clearly this wouldn't be true for Intel for now. > > If the information is not part of a firmware manifest or topology, then > it's likely we have to use an abstraction layer to add this for specific > platforms. > > it's really a bit odd to hard-code all of this at the kernel level, this > was not really what I had in mind when we come up with the concept of > querying capabilities. I understand though that for testing this is > convenient, so maybe this can become a set of fall-back properties in > case the firmware doesn't advertise anything. I see your point. I think for the moment I will remove this patch until I will come with a better solution. One important thing is: where do we advertise the supported parameters: 1) topology. 2) codec component instance (codec adapter) inside FW. 3) Linux kernel side based on some info about the current running platform. Unfortunately, most of the existing users of this interface really do hardcode supported params: e.g intel/atom/sst/sst_drv_interface.c qcom/qdsp6/q6asm-dai.c uniphier/aio-compress.c But that's because I think they only support one single platform family which has same capabilities. > >> + >> + return 0; >> +} >> + >> +static struct snd_compr_codec_caps caps_pcm = { >> + .num_descriptors = 1, >> + .descriptor[0].max_ch = 2, >> + .descriptor[0].sample_rates[0] = 48000, >> + .descriptor[0].num_sample_rates = 1, >> + .descriptor[0].bit_rate = {1536, 3072}, >> + .descriptor[0].num_bitrates = 2, >> + .descriptor[0].profiles = SND_AUDIOPROFILE_PCM, >> + .descriptor[0].modes = 0, >> + .descriptor[0].formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, >> +}; >> + >> +static struct snd_compr_codec_caps caps_mp3 = { >> + .num_descriptors = 1, >> + .descriptor[0].max_ch = 2, >> + .descriptor[0].sample_rates[0] = 48000, >> + .descriptor[0].num_sample_rates = 1, >> + .descriptor[0].bit_rate = {32, 40, 48, 56, 64, 80, 96, 112, 224, 256, 320}, >> + .descriptor[0].num_bitrates = 11, >> + .descriptor[0].profiles = 0, >> + .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO, >> + .descriptor[0].formats = 0, >> +}; >> + >> +static struct snd_compr_codec_caps caps_aac = { >> + .num_descriptors = 1, >> + .descriptor[0].max_ch = 2, >> + .descriptor[0].sample_rates[0] = 48000, >> + .descriptor[0].num_sample_rates = 1, >> + .descriptor[0].bit_rate = {128, 192}, >> + .descriptor[0].num_bitrates = 2, >> + .descriptor[0].profiles = 0, >> + .descriptor[0].modes = 0, >> + .descriptor[0].formats = SND_AUDIOSTREAMFORMAT_MP4ADTS | SND_AUDIOSTREAMFORMAT_MP2ADTS, >> +}; >> + >> +static int sof_compr_get_codec_caps(struct snd_soc_component *component, >> + struct snd_compr_stream *cstream, >> + struct snd_compr_codec_caps *codec) >> +{ >> + switch (codec->codec) { >> + case SND_AUDIOCODEC_MP3: >> + *codec = caps_mp3; >> + break; >> + case SND_AUDIOCODEC_AAC: >> + *codec = caps_aac; >> + break; >> + case SND_AUDIOCODEC_PCM: >> + *codec = caps_pcm; >> + break; >> + default: >> + return -EINVAL; >> + } >> + return 0; >> +} >> + >> struct snd_compress_ops sof_compressed_ops = { >> .open = sof_compr_open, >> .free = sof_compr_free, >> @@ -316,5 +388,7 @@ struct snd_compress_ops sof_compressed_ops = { >> .trigger = sof_compr_trigger, >> .pointer = sof_compr_pointer, >> .copy = sof_compr_copy, >> + .get_caps = sof_compr_get_caps, >> + .get_codec_caps = sof_compr_get_codec_caps, >> }; >> EXPORT_SYMBOL(sof_compressed_ops); >>
>>> +static int sof_compr_get_caps(struct snd_soc_component *component, >>> + struct snd_compr_stream *cstream, >>> + struct snd_compr_caps *caps) >>> +{ >>> + caps->num_codecs = 3; >>> + caps->min_fragment_size = 3840; >>> + caps->max_fragment_size = 3840; >>> + caps->min_fragments = 2; >>> + caps->max_fragments = 2; >>> + caps->codecs[0] = SND_AUDIOCODEC_MP3; >>> + caps->codecs[1] = SND_AUDIOCODEC_AAC; >>> + caps->codecs[2] = SND_AUDIOCODEC_PCM; >> >> I don't think you can add this unconditionally for all >> devices/platforms, clearly this wouldn't be true for Intel for now. >> >> If the information is not part of a firmware manifest or topology, then >> it's likely we have to use an abstraction layer to add this for specific >> platforms. >> >> it's really a bit odd to hard-code all of this at the kernel level, this >> was not really what I had in mind when we come up with the concept of >> querying capabilities. I understand though that for testing this is >> convenient, so maybe this can become a set of fall-back properties in >> case the firmware doesn't advertise anything. > > I see your point. I think for the moment I will remove this patch > until I will come with a better solution. > > One important thing is: where do we advertise the supported parameters: > > 1) topology. > 2) codec component instance (codec adapter) inside FW. > 3) Linux kernel side based on some info about the current running platform. I like the topology approach because it doesn't require an IPC to retrieve the information and it doesn't require additional tables in the firmware - which would increase the size for no good reason. The topology also allows to remove support for the compressed path if there are any concerns with memory/mcps usages in a given platform. DSPs have finite resources and designers will need to trade off fancy noise suppression libraries, large SRAM buffers to reduce power and compressed audio codecs. I think we need to have something in the firmware manifest that lists the presence of the codec component in the build or installed libraries, so that we can verify that topology and firmware are aligned. Otherwise this will be really error-prone. We've had this problem for a while now that the topology can refer to components that are not part of the build, and it's problematic IMHO to see an IPC error as a result of a firmware/topology mistmatch. > Unfortunately, most of the existing users of this interface really do > hardcode supported params: > > e.g > > intel/atom/sst/sst_drv_interface.c > qcom/qdsp6/q6asm-dai.c > uniphier/aio-compress.c > > But that's because I think they only support one single platform family > which has same capabilities. I think Qualcomm probably has the same problems as us, the firmware can be modified, so hard-coding in the kernel is far from ideal. The unifier case is a bit different, IIRC they only support IEC formats which at the kernel level can be treated as PCM. I am not sure why the compressed interface was required here, but it's not wrong to use the compressed interface as a domain-specific transport mechanism. Others do it was well for audio over SPI and on the Intel side the probes are based on a custom transport format to multiplex all the PCM data on a single 'compressed' stream.
diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 91a9c95929cd..e3f3f309f312 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -308,6 +308,78 @@ static int sof_compr_pointer(struct snd_soc_component *component, return 0; } +static int sof_compr_get_caps(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + struct snd_compr_caps *caps) +{ + caps->num_codecs = 3; + caps->min_fragment_size = 3840; + caps->max_fragment_size = 3840; + caps->min_fragments = 2; + caps->max_fragments = 2; + caps->codecs[0] = SND_AUDIOCODEC_MP3; + caps->codecs[1] = SND_AUDIOCODEC_AAC; + caps->codecs[2] = SND_AUDIOCODEC_PCM; + + return 0; +} + +static struct snd_compr_codec_caps caps_pcm = { + .num_descriptors = 1, + .descriptor[0].max_ch = 2, + .descriptor[0].sample_rates[0] = 48000, + .descriptor[0].num_sample_rates = 1, + .descriptor[0].bit_rate = {1536, 3072}, + .descriptor[0].num_bitrates = 2, + .descriptor[0].profiles = SND_AUDIOPROFILE_PCM, + .descriptor[0].modes = 0, + .descriptor[0].formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, +}; + +static struct snd_compr_codec_caps caps_mp3 = { + .num_descriptors = 1, + .descriptor[0].max_ch = 2, + .descriptor[0].sample_rates[0] = 48000, + .descriptor[0].num_sample_rates = 1, + .descriptor[0].bit_rate = {32, 40, 48, 56, 64, 80, 96, 112, 224, 256, 320}, + .descriptor[0].num_bitrates = 11, + .descriptor[0].profiles = 0, + .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO, + .descriptor[0].formats = 0, +}; + +static struct snd_compr_codec_caps caps_aac = { + .num_descriptors = 1, + .descriptor[0].max_ch = 2, + .descriptor[0].sample_rates[0] = 48000, + .descriptor[0].num_sample_rates = 1, + .descriptor[0].bit_rate = {128, 192}, + .descriptor[0].num_bitrates = 2, + .descriptor[0].profiles = 0, + .descriptor[0].modes = 0, + .descriptor[0].formats = SND_AUDIOSTREAMFORMAT_MP4ADTS | SND_AUDIOSTREAMFORMAT_MP2ADTS, +}; + +static int sof_compr_get_codec_caps(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + struct snd_compr_codec_caps *codec) +{ + switch (codec->codec) { + case SND_AUDIOCODEC_MP3: + *codec = caps_mp3; + break; + case SND_AUDIOCODEC_AAC: + *codec = caps_aac; + break; + case SND_AUDIOCODEC_PCM: + *codec = caps_pcm; + break; + default: + return -EINVAL; + } + return 0; +} + struct snd_compress_ops sof_compressed_ops = { .open = sof_compr_open, .free = sof_compr_free, @@ -316,5 +388,7 @@ struct snd_compress_ops sof_compressed_ops = { .trigger = sof_compr_trigger, .pointer = sof_compr_pointer, .copy = sof_compr_copy, + .get_caps = sof_compr_get_caps, + .get_codec_caps = sof_compr_get_codec_caps, }; EXPORT_SYMBOL(sof_compressed_ops);