Attempt to deflake CcFbSendRateAdaptation tests Increase packet_queue_length from 8 to 20 and simulation duration from 10s to 15s in RtcpRateIncreaseWhenAudioStopSending to prevent test flakiness on slow/constrained link emulations. Bug: None Change-Id: I7e8288fd2f040e117aec501a902a171d80490354 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/486900 Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Diep Bui <diepbp@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Commit-Queue: Diep Bui <diepbp@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48123}
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.
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.