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| author | James Almer <jamrial@gmail.com> | 2022-09-21 00:01:40 -0300 |
|---|---|---|
| committer | James Almer <jamrial@gmail.com> | 2022-09-24 12:16:15 -0300 |
| commit | aa79d13f51aa820c7e5f07784a2512434e68bc46 (patch) | |
| tree | 6687dac2df3a70c251312d609884bcc3dd78c3c4 /libavformat/cafenc.c | |
| parent | 257eea3db44bb4862b0fa56e12f5627cb3f8fed4 (diff) | |
| download | ffmpeg-aa79d13f51aa820c7e5f07784a2512434e68bc46.tar.gz | |
avformat/cafenc: derive Opus frame size from the relevant stream parameters
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 after being encoded.
Fixes ticket #9930.
Signed-off-by: James Almer <jamrial@gmail.com>
Diffstat (limited to 'libavformat/cafenc.c')
| -rw-r--r-- | libavformat/cafenc.c | 19 |
1 files 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); |