[FFmpeg-devel] [PATCH 1/7 v2] avcodec: add an Immersive Audio Model and Formats frame split bsf

[James Almer]

On 1/30/2024 7:11 PM, Andreas Rheinhardt wrote:
> James Almer:
>> On 1/30/2024 6:47 PM, Andreas Rheinhardt wrote:
>>>> +    *obu_size = get_leb(&gb);
>>> This stuff here should not a GetBitContext at all, as basically
>>> everything is byte-aligned (and the flags above are in known bits).
>>
>> I'm not going to write yet another leb() reading function to work on raw
>> bytes. We have enough scattered around and in fact we should try to
>> remove most.
>>
>>>> +static const enum AVCodecID iamf_stream_split_codec_ids[] = {
>>>> +    AV_CODEC_ID_PCM_S16LE, AV_CODEC_ID_PCM_S16BE,
>>>> +    AV_CODEC_ID_PCM_S24LE, AV_CODEC_ID_PCM_S24BE,
>>>> +    AV_CODEC_ID_PCM_S32LE, AV_CODEC_ID_PCM_S32BE,
>>>> +    AV_CODEC_ID_OPUS,      AV_CODEC_ID_AAC,
>>>> +    AV_CODEC_ID_FLAC,      AV_CODEC_ID_NONE,
>>>> +};
>>>> +
>>>> +const FFBitStreamFilter ff_iamf_stream_split_bsf = {
>>>> +    .p.name         = "iamf_stream_split",
>>>> +    .p.codec_ids    = iamf_stream_split_codec_ids,
>>>> +    .p.priv_class   = &iamf_stream_split_class,
>>>> +    .priv_data_size = sizeof(IAMFSplitContext),
>>>> +    .init           = iamf_stream_split_init,
>>>> +    .flush          = iamf_stream_split_flush,
>>>> +    .close          = iamf_stream_split_close,
>>>> +    .filter         = iamf_stream_split_filter,
>>>> +};
>>>
>>> This needs to add documentation for what this BSF is actually supposed
>>> to do. Right now it seems crazy: It parses the packet's data and expects
>>> to find OBU headers, although the input data is supposed to be PCM,
>>> Opus, AAC or Flac.
>>
>> It's not too different than aac_adtstoasc in that it takes audio from
>> those codecs listed above encapsulated in one form and returns it in
>> another form.
>> In this case, it takes OBUs containing one or more audio frames, removes
>> the OBU encapsulation, and propagates each raw audio frame in separate
>> packets.
>>
> 
> Then these packets do not really contain PCM, Opus or Flac at all.

It does, encapsulated in OBU. Not really different than AAC in ADTS, 
like i said.
I can remove the iamf_stream_split_codec_ids[] array in any case.