33cc83595a
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate. That is the, initial probe will try to probe up to the max configured bitrate. ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to continue up to the max configured bitrate, regardless of of the max allocated bitrate. Bug: webrtc:14928 Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760 Reviewed-by: Diep Bui <diepbp@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42049}
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