videolan
/
ffmpeg
mirror of https://git.videolan.org/git/ffmpeg.git
1
Fork 0
Code Releases Activity

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>
(cherry picked from commit aa79d13f51)
This commit is contained in:
James Almer 2022-09-21 00:01:40 -03:00
parent c1b8ffbed8
commit 57e15b2e07
1 changed files with 14 additions and 5 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

19
libavformat/cafenc.c
View File

@ -52,7 +52,11 @@ 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->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:
@ -82,6 +86,8 @@ 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;
@ -109,7 +115,7 @@ 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");
@ -138,7 +144,10 @@ 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->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
@ -146,7 +155,7 @@ 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
@ -247,7 +256,7 @@ 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->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);