| Message ID | 20220922231445.6525-1-jamrial@gmail.com |
|---|---|
| State | Accepted |
| Commit | aa79d13f51aa820c7e5f07784a2512434e68bc46 |
| Headers | show |
| Series | [FFmpeg-devel,v2] avformat/cafenc: derive Opus frame size from the relevant stream parameters | expand |
| Context | Check | Description |
|---|---|---|
| andriy/make_x86 | success | Make finished |
| andriy/make_fate_x86 | success | Make fate finished |
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.
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);
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(-)