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avformat/cafenc: derive Opus frame size from the relevant stream parameters

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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>
This commit is contained in:
James Almer
2022-09-21 00:01:40 -03:00
parent 257eea3db4
commit aa79d13f51
1 changed files with 14 additions and 5 deletions
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19
libavformat/cafenc.c
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@@ -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) { switch (codec_id) {
case AV_CODEC_ID_PCM_S8: case AV_CODEC_ID_PCM_S8:
case AV_CODEC_ID_PCM_S16LE: 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; return 320;
case AV_CODEC_ID_MP1: case AV_CODEC_ID_MP1:
return 384; return 384;
case AV_CODEC_ID_OPUS:
return frame_size * 48000 / sample_rate;
case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3: case AV_CODEC_ID_MP3:
return 1152; return 1152;
@@ -110,7 +116,7 @@ static int caf_write_header(AVFormatContext *s)
AVDictionaryEntry *t = NULL; AVDictionaryEntry *t = NULL;
unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, par->codec_id); unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, par->codec_id);
int64_t chunk_size = 0; 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) { if (s->nb_streams != 1) {
av_log(s, AV_LOG_ERROR, "CAF files have exactly one stream\n"); 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) 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 ffio_wfourcc(pb, "caff"); //< mFileType
avio_wb16(pb, 1); //< mFileVersion avio_wb16(pb, 1); //< mFileVersion
@@ -147,7 +156,7 @@ static int caf_write_header(AVFormatContext *s)
ffio_wfourcc(pb, "desc"); //< Audio Description chunk ffio_wfourcc(pb, "desc"); //< Audio Description chunk
avio_wb64(pb, 32); //< mChunkSize 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_wl32(pb, codec_tag); //< mFormatID
avio_wb32(pb, codec_flags(par->codec_id)); //< mFormatFlags avio_wb32(pb, codec_flags(par->codec_id)); //< mFormatFlags
avio_wb32(pb, par->block_align); //< mBytesPerPacket 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_seek(pb, caf->data, SEEK_SET);
avio_wb64(pb, file_size - caf->data - 8); avio_wb64(pb, file_size - caf->data - 8);
if (!par->block_align) { 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) { if (!packet_size) {
packet_size = st->duration / (caf->packets - 1); packet_size = st->duration / (caf->packets - 1);
avio_seek(pb, FRAME_SIZE_OFFSET, SEEK_SET); avio_seek(pb, FRAME_SIZE_OFFSET, SEEK_SET);