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Merge branch v4.0.269 into 5.0release

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1. MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269 (#296) (#3333)
This commit is contained in:
winlin 2022-12-25 12:10:03 +08:00
commit 05d7400cd5
9 changed files with 108 additions and 21 deletions
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1
trunk/doc/CHANGELOG.md
View file

@ -118,6 +118,7 @@ The changelog for SRS.
## SRS 4.0 Changelog
* v4.0, 2022-12-24, For [#296](https://github.com/ossrs/srs/issues/296): MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269
* v4.0, 2022-11-22, Pick [#3079](https://github.com/ossrs/srs/issues/3079): WebRTC: Fix no audio and video issue for Firefox. v4.0.268
* v4.0, 2022-10-10, For [#2901](https://github.com/ossrs/srs/issues/2901): Edge: Fast disconnect and reconnect. v4.0.267
* v4.0, 2022-09-27, For [#3167](https://github.com/ossrs/srs/issues/3167): WebRTC: Refine sequence jitter algorithm. v4.0.266

28
trunk/src/app/srs_app_hls.cpp
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@ -202,6 +202,7 @@ SrsHlsMuxer::SrsHlsMuxer()
async = new SrsAsyncCallWorker();
context = new SrsTsContext();
segments = new SrsFragmentWindow();
latest_acodec_ = SrsAudioCodecIdForbidden;
memset(key, 0, 16);
memset(iv, 0, 16);
@ -263,6 +264,24 @@ int SrsHlsMuxer::deviation()
return deviation_ts;
}
SrsAudioCodecId SrsHlsMuxer::latest_acodec()
{
// If current context writer exists, we query from it.
if (current && current->tscw) return current->tscw->acodec();
// Get the configured or updated config.
return latest_acodec_;
}
void SrsHlsMuxer::set_latest_acodec(SrsAudioCodecId v)
{
// Refresh the codec in context writer for current segment.
if (current && current->tscw) current->tscw->set_acodec(v);
// Refresh the codec for future segments.
latest_acodec_ = v;
}
srs_error_t SrsHlsMuxer::initialize()
{
return srs_success;
@ -371,6 +390,8 @@ srs_error_t SrsHlsMuxer::segment_open()
srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str());
}
}
// Now that we know the latest audio codec in stream, use it.
if (latest_acodec_ != SrsAudioCodecIdForbidden) default_acodec = latest_acodec_;
// load the default vcodec from config.
SrsVideoCodecId default_vcodec = SrsVideoCodecIdAVC;
@ -963,6 +984,13 @@ srs_error_t SrsHlsController::on_sequence_header()
srs_error_t SrsHlsController::write_audio(SrsAudioFrame* frame, int64_t pts)
{
srs_error_t err = srs_success;
// Refresh the codec ASAP.
if (muxer->latest_acodec() != frame->acodec()->id) {
srs_trace("HLS: Switch audio codec %d(%s) to %d(%s)", muxer->latest_acodec(), srs_audio_codec_id2str(muxer->latest_acodec()).c_str(),
frame->acodec()->id, srs_audio_codec_id2str(frame->acodec()->id).c_str());
muxer->set_latest_acodec(frame->acodec()->id);
}
// write audio to cache.
if ((err = tsmc->cache_audio(frame, pts)) != srs_success) {

6
trunk/src/app/srs_app_hls.hpp
View file

@ -156,6 +156,9 @@ private:
SrsHlsSegment* current;
// The ts context, to keep cc continous between ts.
SrsTsContext* context;
private:
// Latest audio codec, parsed from stream.
SrsAudioCodecId latest_acodec_;
public:
SrsHlsMuxer();
virtual ~SrsHlsMuxer();
@ -166,6 +169,9 @@ public:
virtual std::string ts_url();
virtual srs_utime_t duration();
virtual int deviation();
public:
SrsAudioCodecId latest_acodec();
void set_latest_acodec(SrsAudioCodecId v);
public:
// Initialize the hls muxer.
virtual srs_error_t initialize();

4
trunk/src/app/srs_app_http_stream.cpp
View file

@ -829,7 +829,9 @@ void SrsLiveStream::http_hooks_on_stop(ISrsHttpMessage* r)
srs_error_t SrsLiveStream::streaming_send_messages(ISrsBufferEncoder* enc, SrsSharedPtrMessage** msgs, int nb_msgs)
{
srs_error_t err = srs_success;
// TODO: In gop cache, we know both the audio and video codec, so we should notice the encoder, which might depends
// on setting the correct codec information, for example, HTTP-TS or HLS will write PMT.
for (int i = 0; i < nb_msgs; i++) {
SrsSharedPtrMessage* msg = msgs[i];

2
trunk/src/core/srs_core_version4.hpp
View file

@ -9,6 +9,6 @@
#define VERSION_MAJOR 4
#define VERSION_MINOR 0
#define VERSION_REVISION 268
#define VERSION_REVISION 269
#endif

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

@ -543,6 +543,9 @@ 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");
@ -1472,20 +1475,13 @@ 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

@ -2641,8 +2641,8 @@ SrsTsContextWriter::SrsTsContextWriter(ISrsStreamWriter* w, SrsTsContext* c, Srs
{
writer = w;
context = c;
acodec = ac;
acodec_ = ac;
vcodec = vc;
}
@ -2657,7 +2657,7 @@ 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");
@ -2672,7 +2672,7 @@ 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");
@ -2685,6 +2685,16 @@ 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);
@ -3122,6 +3132,13 @@ 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
@ -1259,7 +1259,7 @@ 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;
@ -1275,6 +1275,10 @@ 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

39
trunk/src/utest/srs_utest_kernel.cpp
View file

@ -3469,11 +3469,23 @@ VOID TEST(KernelCodecTest, AVFrame)
EXPECT_TRUE(20 == f.samples[1].size);
EXPECT_TRUE(2 == f.nb_samples);
}
if (true) {
SrsAudioFrame f;
EXPECT_TRUE(0 == f.nb_samples);
HELPER_EXPECT_SUCCESS(f.add_sample((char*)1, 0));
EXPECT_TRUE(0 == f.nb_samples);
HELPER_EXPECT_SUCCESS(f.add_sample(NULL, 1));
EXPECT_TRUE(0 == f.nb_samples);
}
if (true) {
SrsAudioFrame f;
for (int i = 0; i < SrsMaxNbSamples; i++) {
HELPER_EXPECT_SUCCESS(f.add_sample((char*)(int64_t)i, i*10));
HELPER_EXPECT_SUCCESS(f.add_sample((char*)(int64_t)(i + 1), i*10 + 1));
}
srs_error_t err = f.add_sample((char*)1, 1);
@ -3580,18 +3592,39 @@ VOID TEST(KernelCodecTest, AudioFormat)
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\x00", 0));
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\x00", 1));
}
// For MP3
if (true) {
SrsFormat f;
HELPER_EXPECT_SUCCESS(f.initialize());
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\x20", 1));
EXPECT_TRUE(0 == f.nb_raw);
EXPECT_TRUE(0 == f.audio->nb_samples);
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\x20\x00", 2));
EXPECT_TRUE(1 == f.nb_raw);
EXPECT_TRUE(0 == f.audio->nb_samples);
EXPECT_TRUE(1 == f.audio->nb_samples);
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\x20\x00\x00", 3));
EXPECT_TRUE(2 == f.nb_raw);
EXPECT_TRUE(1 == f.audio->nb_samples);
}
// For AAC
if (true) {
SrsFormat f;
HELPER_EXPECT_SUCCESS(f.initialize());
HELPER_EXPECT_FAILED(f.on_audio(0, (char*)"\xa0", 1));
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\xaf\x00\x12\x10", 4));
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\xa0\x01", 2));
EXPECT_TRUE(0 == f.nb_raw);
EXPECT_TRUE(0 == f.audio->nb_samples);
HELPER_EXPECT_SUCCESS(f.on_audio(0, (char*)"\xa0\x01\x00", 3));
EXPECT_TRUE(1 == f.nb_raw);
EXPECT_TRUE(1 == f.audio->nb_samples);
}
if (true) {
SrsFormat f;