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If experimental min bitrate value is not configured, set the min bitrate for the first simulcast stream equal to kDefaultMinVideoBitrateBps (=30kbps). Min bitrate depends on resolution. At absence of the experimental min bitrate override, we got high min bitrate values for high resolutions (600kbps for VP8 720p, for example) before. That led to encode pauses [1] which is an undesired behavior. [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/simulcast_rate_allocator.cc;l=173;drc=f317f7106a7a15a04da7cd30c2e2ddb1b3025bc6 Bug: webrtc:351644568, b/352504711 Change-Id: Ifc93cc230fb194d2c9a739368d415f24385939fd No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357420 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42649}
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