external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-03-09 15:49:59 +00:00
Code Issues Releases Wiki Activity

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

Browse source 
* 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
No known key found for this signature in database
GPG key ID: 4AEE18F83AFDEB23
10 changed files with 109 additions and 38 deletions
Download patch file Download diff file Expand all files Collapse all files

16
trunk/src/kernel/srs_kernel_codec.cpp
View file

@ -488,6 +488,9 @@ 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");
@ -1407,20 +1410,13 @@ 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

@ -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,

10
trunk/src/kernel/srs_kernel_ts.hpp
View file

@ -97,7 +97,7 @@ enum SrsTsPidApply
SrsTsPidApplyAudio, // vor audio
};
// Table 2-29 - Stream type assignments
// Table 2-29 - Stream type assignments, hls-mpeg-ts-iso13818-1.pdf, page 66
enum SrsTsStream
{
// ITU-T | ISO/IEC Reserved
@ -106,8 +106,8 @@ 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
@ -1243,7 +1243,7 @@ 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;
@ -1259,6 +1259,10 @@ public:
public:
// get the video codec of ts muxer.
virtual SrsVideoCodecId video_codec();
public:
// Get and set the audio codec.
SrsAudioCodecId acodec();
void set_acodec(SrsAudioCodecId v);
};
// Used for HLS Encryption