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Printf warn log when SRT audio duration too large (#3186)

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* SRT: print warning log when audio duration too large

* Fix #3164: SRT: Choppy when audio ts gap is too large. v5.0.65

Co-authored-by: winlin <winlin@vip.126.com>
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john 2022-09-22 07:37:22 -05:00 committed by GitHub
parent 4acb246c57
commit b328142140
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4 changed files with 34 additions and 6 deletions
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1
trunk/doc/CHANGELOG.md
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@ -7,6 +7,7 @@ The changelog for SRS.
## SRS 5.0 Changelog
* v5.0, 2022-09-22, Fix [#3164](https://github.com/ossrs/srs/issues/3164): SRT: Choppy when audio ts gap is too large. v5.0.65
* v5.0, 2022-09-16, APM: Support distributed tracing by Tencent Cloud APM. v5.0.64
* v5.0, 2022-09-16, For [#3179](https://github.com/ossrs/srs/issues/3179): WebRTC: Make sure the same m-lines order for offer and answer. v5.0.63
* v5.0, 2022-09-10, For [#3174](https://github.com/ossrs/srs/issues/3174): WebRTC: Support Unity to publish or play stream. v5.0.62

29
trunk/src/app/srs_app_srt_source.cpp
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@ -18,6 +18,7 @@ using namespace std;
#include <srs_protocol_rtmp_stack.hpp>
#include <srs_app_source.hpp>
#include <srs_app_statistic.hpp>
#include <srs_app_pithy_print.hpp>
SrsSrtPacket::SrsSrtPacket()
{
@ -250,14 +251,21 @@ SrsRtmpFromSrtBridge::SrsRtmpFromSrtBridge(SrsLiveSource* source) : ISrsSrtSourc
sps_ = "";
pps_ = "";
live_source_ = source;
req_ = NULL;
live_source_ = source;
video_streamid_ = 1;
audio_streamid_ = 2;
pp_audio_duration_ = new SrsAlonePithyPrint();
}
SrsRtmpFromSrtBridge::~SrsRtmpFromSrtBridge()
{
srs_freep(ts_ctx_);
srs_freep(req_);
srs_freep(pp_audio_duration_);
}
srs_error_t SrsRtmpFromSrtBridge::on_publish()
@ -450,7 +458,7 @@ srs_error_t SrsRtmpFromSrtBridge::check_sps_pps_change(SrsTsMessage* msg)
}
SrsMessageHeader header;
header.initialize_video(nb_flv, dts, 1);
header.initialize_video(nb_flv, dts, video_streamid_);
SrsCommonMessage rtmp;
if ((err = rtmp.create(&header, flv, nb_flv)) != srs_success) {
return srs_error_wrap(err, "create rtmp");
@ -488,7 +496,7 @@ srs_error_t SrsRtmpFromSrtBridge::on_h264_frame(SrsTsMessage* msg, vector<pair<c
}
SrsCommonMessage rtmp;
rtmp.header.initialize_video(frame_size, dts, 1/*streamid*/);
rtmp.header.initialize_video(frame_size, dts, video_streamid_);
rtmp.create_payload(frame_size);
rtmp.size = frame_size;
SrsBuffer payload(rtmp.payload, rtmp.size);
@ -530,6 +538,7 @@ srs_error_t SrsRtmpFromSrtBridge::on_ts_audio(SrsTsMessage* msg, SrsBuffer* avs)
uint32_t pts = (uint32_t)(msg->pts / 90);
int frame_idx = 0;
int duration_ms = 0;
// send each frame.
while (!avs->empty()) {
@ -567,6 +576,7 @@ srs_error_t SrsRtmpFromSrtBridge::on_ts_audio(SrsTsMessage* msg, SrsBuffer* avs)
default: sample_rate = 44100; break;
}
uint32_t frame_pts = (double)pts + (frame_idx * (1024.0 * 1000.0 / sample_rate));
duration_ms += 1024.0 * 1000.0 / sample_rate;
++frame_idx;
if ((err = check_audio_sh_change(msg, frame_pts)) != srs_success) {
@ -577,6 +587,15 @@ srs_error_t SrsRtmpFromSrtBridge::on_ts_audio(SrsTsMessage* msg, SrsBuffer* avs)
return srs_error_wrap(err, "audio frame");
}
}
pp_audio_duration_->elapse();
if ((duration_ms >= 200) && pp_audio_duration_->can_print()) {
// MPEG-TS always merge multi audio frame into one pes packet, may cause high latency and AV synchronization errors
// @see https://github.com/ossrs/srs/issues/3164
srs_warn("srt to rtmp, audio duration=%dms too large, audio frames=%d, may cause high latency and AV synchronization errors, "
"read https://ossrs.io/lts/en-us/docs/v5/doc/srt-codec#ffmpeg-push-srt-stream", duration_ms, frame_idx);
}
return err;
}
@ -595,7 +614,7 @@ srs_error_t SrsRtmpFromSrtBridge::check_audio_sh_change(SrsTsMessage* msg, uint3
int rtmp_len = audio_sh_.size() + 2;
SrsCommonMessage rtmp;
rtmp.header.initialize_audio(rtmp_len, pts, 1);
rtmp.header.initialize_audio(rtmp_len, pts, audio_streamid_);
rtmp.create_payload(rtmp_len);
rtmp.size = rtmp_len;
@ -619,7 +638,7 @@ srs_error_t SrsRtmpFromSrtBridge::on_aac_frame(SrsTsMessage* msg, uint32_t pts,
int rtmp_len = frame_size + 2/* 2 bytes of flv audio tag header*/;
SrsCommonMessage rtmp;
rtmp.header.initialize_audio(rtmp_len, pts, 2/*streamid*/);
rtmp.header.initialize_audio(rtmp_len, pts, audio_streamid_);
rtmp.create_payload(rtmp_len);
rtmp.size = rtmp_len;

8
trunk/src/app/srs_app_srt_source.hpp
View file

@ -20,6 +20,7 @@ class SrsSharedPtrMessage;
class SrsRequest;
class SrsLiveSource;
class SrsSrtSource;
class SrsAlonePithyPrint;
// The SRT packet with shared message.
class SrsSrtPacket
@ -136,6 +137,13 @@ private:
SrsRequest* req_;
SrsLiveSource* live_source_;
// SRT to rtmp, video stream id.
int video_streamid_;
// SRT to rtmp, audio stream id.
int audio_streamid_;
// Cycle print when audio duration too large because mpegts may merge multi audio frame in one pes packet.
SrsAlonePithyPrint* pp_audio_duration_;
};
class SrsSrtSource

2
trunk/src/core/srs_core_version5.hpp
View file

@ -9,6 +9,6 @@
#define VERSION_MAJOR 5
#define VERSION_MINOR 0
#define VERSION_REVISION 64
#define VERSION_REVISION 65
#endif