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 <jamr...@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.
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
To unsubscribe, visit link above, or email
ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
