Re: [RFC PATCH 1/6] ALSA: compress: add Sample Rate Converter codec support

[Jeff Brower <jbrower@xxxxxxxxxxxxxx>]

All-

"The sample rate converter is not an encoder ..."

Indeed, an encoder creates a compressed bitstream from audio data  
(typically linear PCM samples), normally for transmission of some  
type. A sample rate converter generates audio data from audio data,  
and is normally applied prior to an encoder because it can only accept  
a limited range of sample rates.

-Jeff

Quoting Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx>:

> On 8/6/24 12:26, Shengjiu Wang wrote:
> > Add Sample Rate Converter(SRC) codec support, define the output
> > format and rate for SRC.
> >
> > Signed-off-by: Shengjiu Wang <shengjiu.wang@xxxxxxx>
> > ---
> >  include/uapi/sound/compress_offload.h | 2 ++
> >  include/uapi/sound/compress_params.h  | 9 ++++++++-
> >  2 files changed, 10 insertions(+), 1 deletion(-)
> >
> > diff --git a/include/uapi/sound/compress_offload.h  
> > b/include/uapi/sound/compress_offload.h
> > index 98772b0cbcb7..8b2b72f94e26 100644
> > --- a/include/uapi/sound/compress_offload.h
> > +++ b/include/uapi/sound/compress_offload.h
> > @@ -112,10 +112,12 @@ struct snd_compr_codec_caps {
> >   * end of the track
> >   * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the  
> > encoder at the
> >   * beginning of the track
> > + * @SNDRV_COMPRESS_SRC_RATIO_MOD: Resampling Ratio Modifier for  
> > sample rate converter
> >   */
> >  enum sndrv_compress_encoder {
> >  	SNDRV_COMPRESS_ENCODER_PADDING = 1,
> >  	SNDRV_COMPRESS_ENCODER_DELAY = 2,
> > +	SNDRV_COMPRESS_SRC_RATIO_MOD = 3,
> >  };
>
> this sounds wrong to me. The sample rate converter is not an "encoder",
> and the properties for padding/delay are totally specific to an encoder
> function.
>
> The other point is that I am not sure how standard this ratio_mod
> parameter is. This could be totally specific to a specific
> implementation, and another ASRC might have different parameters.
>
> >  /**
> > diff --git a/include/uapi/sound/compress_params.h  
> > b/include/uapi/sound/compress_params.h
> > index ddc77322d571..0843773ea6b4 100644
> > --- a/include/uapi/sound/compress_params.h
> > +++ b/include/uapi/sound/compress_params.h
> > @@ -43,7 +43,8 @@
> >  #define SND_AUDIOCODEC_BESPOKE               ((__u32) 0x0000000E)
> >  #define SND_AUDIOCODEC_ALAC                  ((__u32) 0x0000000F)
> >  #define SND_AUDIOCODEC_APE                   ((__u32) 0x00000010)
> > -#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_APE
> > +#define SND_AUDIOCODEC_SRC                   ((__u32) 0x00000011)
> > +#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_SRC
>
> I am not sure this is wise to change such definitions?
> >  /*
> >   * Profile and modes are listed with bit masks. This allows for a
> > @@ -324,6 +325,11 @@ struct snd_dec_ape {
> >  	__u32 seek_table_present;
> >  } __attribute__((packed, aligned(4)));
> >
> > +struct snd_dec_src {
> > +	__u32 format_out;
> > +	__u32 rate_out;
> > +} __attribute__((packed, aligned(4)));
>
> Again I am not sure how standard those parameters are, and even if they
> were if their representation is reusable.
>
> And the compressed API has a good split between encoders and decoders, I
> am not sure how an SRC can be classified as either.
>
> >  union snd_codec_options {
> >  	struct snd_enc_wma wma;
> >  	struct snd_enc_vorbis vorbis;
> > @@ -334,6 +340,7 @@ union snd_codec_options {
> >  	struct snd_dec_wma wma_d;
> >  	struct snd_dec_alac alac_d;
> >  	struct snd_dec_ape ape_d;
> > +	struct snd_dec_src src;
> >  } __attribute__((packed, aligned(4)));
> >
> >  /** struct snd_codec_desc - description of codec capabilities