b0d8a3728c
This ensure BWE works as intended with transport sequence numbers on audio. Tested with webrtc_perf_tests --gtest_filter=CallPerfTest.Min_Bitrate_VideoAndAudio and --gtest_filter=Rampup* Bug: webrtc:14854, webrtc:7135, b/266786240 Change-Id: I3b7a743149c22035e582a2157b5f0a93747857cf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291523 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39208}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation