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Merge branch v5.0.116 into develop

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1. MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269 (#296) (#3333)
2. MP3: Add config examples for MP3. #296
3. Script: Refine GitHub actions.
This commit is contained in:
winlin 2022-12-25 16:23:23 +08:00
commit b5aaf67c93
16 changed files with 212 additions and 55 deletions
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16
trunk/src/kernel/srs_kernel_codec.cpp
View file

@ -595,6 +595,9 @@ srs_error_t SrsFrame::initialize(SrsCodecConfig* c)
srs_error_t SrsFrame::add_sample(char* bytes, int size)
{
srs_error_t err = srs_success;
// Ignore empty sample.
if (!bytes || size <= 0) return err;
if (nb_samples >= SrsMaxNbSamples) {
return srs_error_new(ERROR_HLS_DECODE_ERROR, "Frame samples overflow");
@ -2063,20 +2066,13 @@ srs_error_t SrsFormat::audio_mp3_demux(SrsBuffer* stream, int64_t timestamp)
// we always decode aac then mp3.
srs_assert(acodec->id == SrsAudioCodecIdMP3);
// Update the RAW MP3 data.
// Update the RAW MP3 data. Note the start is 12 bits syncword 0xFFF, so we should not skip any bytes, for detail
// please see ISO_IEC_11172-3-MP3-1993.pdf page 20 and 26.
raw = stream->data() + stream->pos();
nb_raw = stream->size() - stream->pos();
stream->skip(1);
if (stream->empty()) {
return err;
}
char* data = stream->data() + stream->pos();
int size = stream->size() - stream->pos();
// mp3 payload.
if ((err = audio->add_sample(data, size)) != srs_success) {
if ((err = audio->add_sample(raw, nb_raw)) != srs_success) {
return srs_error_wrap(err, "add audio frame");
}

25
trunk/src/kernel/srs_kernel_ts.cpp
View file

@ -2676,8 +2676,8 @@ SrsTsContextWriter::SrsTsContextWriter(ISrsStreamWriter* w, SrsTsContext* c, Srs
{
writer = w;
context = c;
acodec = ac;
acodec_ = ac;
vcodec = vc;
}
@ -2692,7 +2692,7 @@ 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");
@ -2707,7 +2707,7 @@ 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");
@ -2725,6 +2725,16 @@ void SrsTsContextWriter::update_video_codec(SrsVideoCodecId v)
vcodec = v;
}
SrsAudioCodecId SrsTsContextWriter::acodec()
{
return acodec_;
}
void SrsTsContextWriter::set_acodec(SrsAudioCodecId v)
{
acodec_ = v;
}
SrsEncFileWriter::SrsEncFileWriter()
{
memset(iv,0,16);
@ -3217,6 +3227,13 @@ 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,

8
trunk/src/kernel/srs_kernel_ts.hpp
View file

@ -107,8 +107,8 @@ enum SrsTsStream
// ISO/IEC 11172 Video
// ITU-T Rec. H.262 | ISO/IEC 13818-2 Video or ISO/IEC 11172-2 constrained parameter video stream
// ISO/IEC 11172 Audio
SrsTsStreamAudioMp3 = 0x03,
// ISO/IEC 13818-3 Audio
SrsTsStreamAudioMp3 = 0x04,
// ITU-T Rec. H.222.0 | ISO/IEC 13818-1 private_sections
// ITU-T Rec. H.222.0 | ISO/IEC 13818-1 PES packets containing private data
// ISO/IEC 13522 MHEG
@ -1260,7 +1260,7 @@ private:
// User must config the codec in right way.
// @see https://github.com/ossrs/srs/issues/301
SrsVideoCodecId vcodec;
SrsAudioCodecId acodec;
SrsAudioCodecId acodec_;
private:
SrsTsContext* context;
ISrsStreamWriter* writer;
@ -1277,6 +1277,10 @@ public:
// Get or update the video codec of ts muxer.
virtual SrsVideoCodecId video_codec();
virtual void update_video_codec(SrsVideoCodecId v);
public:
// Get and set the audio codec.
SrsAudioCodecId acodec();
void set_acodec(SrsAudioCodecId v);
};
// Used for HLS Encryption