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MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269 (#296) (#3333)

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* MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269 (#296)

1. Refresh HLS audio codec if changed in stream.
2. Refresh TS audio codec if changed in stream.
3. Fix mp3 demux bug in SrsFormat::audio_mp3_demux.
4. Use 3(MPEG1) not 4(MPEG2) as PMT stream type, follow FFmpeg.
5. MP3: Update utest for mp3 sample parsing.
6. MP3: Ignore empty frame sample.
7. UTest: Fix utest failed, do not copy files.
This commit is contained in:
Winlin 2022-12-25 11:43:26 +08:00 • committed by GitHub
parent 2573a25101
commit 577cd299e1
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GPG key ID: 4AEE18F83AFDEB23
10 changed files with 109 additions and 38 deletions
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25
trunk/src/kernel/srs_kernel_ts.cpp
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@ -2598,8 +2598,8 @@ SrsTsContextWriter::SrsTsContextWriter(ISrsStreamWriter* w, SrsTsContext* c, Srs
{
writer = w;
context = c;
acodec = ac;
acodec_ = ac;
vcodec = vc;
}
@ -2614,7 +2614,7 @@ srs_error_t SrsTsContextWriter::write_audio(SrsTsMessage* audio)
srs_info("hls: write audio pts=%" PRId64 ", dts=%" PRId64 ", size=%d",
audio->pts, audio->dts, audio->PES_packet_length);
if ((err = context->encode(writer, audio, vcodec, acodec)) != srs_success) {
if ((err = context->encode(writer, audio, vcodec, acodec_)) != srs_success) {
return srs_error_wrap(err, "ts: write audio");
}
srs_info("hls encode audio ok");
@ -2629,7 +2629,7 @@ srs_error_t SrsTsContextWriter::write_video(SrsTsMessage* video)
srs_info("hls: write video pts=%" PRId64 ", dts=%" PRId64 ", size=%d",
video->pts, video->dts, video->PES_packet_length);
if ((err = context->encode(writer, video, vcodec, acodec)) != srs_success) {
if ((err = context->encode(writer, video, vcodec, acodec_)) != srs_success) {
return srs_error_wrap(err, "ts: write video");
}
srs_info("hls encode video ok");
@ -2642,6 +2642,16 @@ SrsVideoCodecId SrsTsContextWriter::video_codec()
return vcodec;
}
SrsAudioCodecId SrsTsContextWriter::acodec()
{
return acodec_;
}
void SrsTsContextWriter::set_acodec(SrsAudioCodecId v)
{
acodec_ = v;
}
SrsEncFileWriter::SrsEncFileWriter()
{
memset(iv,0,16);
@ -3079,6 +3089,13 @@ srs_error_t SrsTsTransmuxer::write_audio(int64_t timestamp, char* data, int size
if (format->acodec->id == SrsAudioCodecIdAAC && format->audio->aac_packet_type == SrsAudioAacFrameTraitSequenceHeader) {
return err;
}
// Switch audio codec if not AAC.
if (tscw->acodec() != format->acodec->id) {
srs_trace("TS: Switch audio codec %d(%s) to %d(%s)", tscw->acodec(), srs_audio_codec_id2str(tscw->acodec()).c_str(),
format->acodec->id, srs_audio_codec_id2str(format->acodec->id).c_str());
tscw->set_acodec(format->acodec->id);
}
// the dts calc from rtmp/flv header.
// @remark for http ts stream, the timestamp is always monotonically increase,