Re: [FFmpeg-devel] [PATCH v2] avformat/cafenc: derive Opus frame size from the relevant stream parameters

[James Almer <jamrial@gmail.com>]

On 9/22/2022 8:14 PM, James Almer wrote:
> Use the stream duration as last resort, as an off-by-one result of the
> "st->duration / (caf->packets - 1)" calculation can break playback on some
> devices.
> Also, don't write the sample_rate value propagated by encoders like libopus.
> The sample rate of the audio fed to it is irrelevant for the container after
> being encoded.
> 
> Fixes ticket #9930.
> 
> Signed-off-by: James Almer <jamrial@gmail.com>
> ---
>   libavformat/cafenc.c | 19 ++++++++++++++-----
>   1 file changed, 14 insertions(+), 5 deletions(-)
> 
> diff --git a/libavformat/cafenc.c b/libavformat/cafenc.c
> index fedb430b17..b90811d46f 100644
> --- a/libavformat/cafenc.c
> +++ b/libavformat/cafenc.c
> @@ -53,7 +53,11 @@ static uint32_t codec_flags(enum AVCodecID codec_id) {
>       }
>   }
>   
> -static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int block_align) {
> +static uint32_t samples_per_packet(const AVCodecParameters *par) {
> +    enum AVCodecID codec_id = par->codec_id;
> +    int channels = par->ch_layout.nb_channels, block_align = par->block_align;
> +    int frame_size = par->frame_size, sample_rate = par->sample_rate;
> +
>       switch (codec_id) {
>       case AV_CODEC_ID_PCM_S8:
>       case AV_CODEC_ID_PCM_S16LE:
> @@ -83,6 +87,8 @@ static uint32_t samples_per_packet(enum AVCodecID codec_id, int channels, int bl
>           return 320;
>       case AV_CODEC_ID_MP1:
>           return 384;
> +    case AV_CODEC_ID_OPUS:
> +        return frame_size * 48000 / sample_rate;
>       case AV_CODEC_ID_MP2:
>       case AV_CODEC_ID_MP3:
>           return 1152;
> @@ -110,7 +116,7 @@ static int caf_write_header(AVFormatContext *s)
>       AVDictionaryEntry *t = NULL;
>       unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, par->codec_id);
>       int64_t chunk_size = 0;
> -    int frame_size = par->frame_size;
> +    int frame_size = par->frame_size, sample_rate = par->sample_rate;
>   
>       if (s->nb_streams != 1) {
>           av_log(s, AV_LOG_ERROR, "CAF files have exactly one stream\n");
> @@ -139,7 +145,10 @@ static int caf_write_header(AVFormatContext *s)
>       }
>   
>       if (par->codec_id != AV_CODEC_ID_MP3 || frame_size != 576)
> -        frame_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
> +        frame_size = samples_per_packet(par);
> +
> +    if (par->codec_id == AV_CODEC_ID_OPUS)
> +        sample_rate = 48000;
>   
>       ffio_wfourcc(pb, "caff"); //< mFileType
>       avio_wb16(pb, 1);         //< mFileVersion
> @@ -147,7 +156,7 @@ static int caf_write_header(AVFormatContext *s)
>   
>       ffio_wfourcc(pb, "desc");                         //< Audio Description chunk
>       avio_wb64(pb, 32);                                //< mChunkSize
> -    avio_wb64(pb, av_double2int(par->sample_rate));   //< mSampleRate
> +    avio_wb64(pb, av_double2int(sample_rate));        //< mSampleRate
>       avio_wl32(pb, codec_tag);                         //< mFormatID
>       avio_wb32(pb, codec_flags(par->codec_id));        //< mFormatFlags
>       avio_wb32(pb, par->block_align);                  //< mBytesPerPacket
> @@ -248,7 +257,7 @@ static int caf_write_trailer(AVFormatContext *s)
>           avio_seek(pb, caf->data, SEEK_SET);
>           avio_wb64(pb, file_size - caf->data - 8);
>           if (!par->block_align) {
> -            int packet_size = samples_per_packet(par->codec_id, par->ch_layout.nb_channels, par->block_align);
> +            int packet_size = samples_per_packet(par);
>               if (!packet_size) {
>                   packet_size = st->duration / (caf->packets - 1);
>                   avio_seek(pb, FRAME_SIZE_OFFSET, SEEK_SET);

Will apply.
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