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

[Andreas Rheinhardt <andreas.rheinhardt@outlook.com>]

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.

I disagree with this: While there are some codecs where several
different encapsulations are widely used and supported (in which case
one can actually find out information about the packetization by looking
at the extradata (H.26x) or the packets (AAC)), the aforementioned ones
are not among them. Your PCM-in-OBU is just not PCM.

- Andreas

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