-
-
Notifications
You must be signed in to change notification settings - Fork 5.4k
New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
RTC: Fix rtc to rtmp sync timestamp using sender report. #2470
Conversation
Codecov Report
@@ Coverage Diff @@
## 4.0release #2470 +/- ##
==============================================
+ Coverage 59.03% 59.33% +0.29%
==============================================
Files 122 122
Lines 51367 51556 +189
==============================================
+ Hits 30326 30592 +266
+ Misses 21041 20964 -77 | Impacted Files | Coverage Δ | |' Translated to English while maintaining the markdown structure: '| Impacted Files | Coverage Δ | | Translated to English while maintaining the markdown structure: '| trunk/src/app/srs_app_rtc_codec.cpp | Continue to review full report at Codecov.
Translated to English while maintaining the markdown structure: '>
|
bf2c3a1
to
5e87627
Compare
Referenced this article: https://mp.weixin.qq.com/s/Tmr2Qk-h2BRa276_YFaBig
WebRTC uses SenderReport to synchronize the timestamps of audio and video during streaming. Without receiving SenderReport, it is impossible to determine the synchronization time. Therefore, it is not forwarded to RtmpFromRtcBridge to avoid further issues with rtmp/hls/flv/dvr caused by timestamp problems in the source stream.
TRANS_BY_GPT3