bf2c049a12
we currently check for bandwidth overuse once for every RTP packet. This CL creates an experiment to test processing all packets in the RTCP feedback before checking for overuse. This can be thought of as checking for overuse per RTCP packet instead of per RTP packet. The change is not expected to have a large impact, but enabling the experiment will make the delay-based BWE slightly less sensitive. This means that we'll be less likely to back down incorrectly after a brief network transient, at the cost of sometimes missing real overuse (especially when the network queues are short). In the latter case, the loss-based estimator is expected to detect the overuse. The experiment is off by default. BUG=webrtc:7508 Review-Url: https://codereview.webrtc.org/2835573003 Cr-Commit-Position: refs/heads/master@{#17968}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc