98e71f57ea
Traditionally, we'd back off to 85% of the measured throughput in response to an overuse. However, this backoff doesn't appear to be sufficient to drain the queues in some low-bitrate scenarios, and the problem has gotten a bit worse with the RobustThroughputEstimator. (The new estimate looks more stable. The old estimator had more variation, the lowest points were lower, causing backoffs to lower rates.) With this change, we back off to 0.85*thoughput-5kbps. The difference is negligible except at low bitrates. Bug: webrtc:13402,b/298636540 Change-Id: I53328953c056b8ad77f6c7561d6017f171b2dfbc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321701 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40827}
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