cd02ebaea0
Before this change, if both app and encoder provided bitrate limits, WebRTC ignored the limits provided by encoder. Now intersection of these sets is used. Also changed DCHECKs in GetEncoderBitrateLimits to allow zero values of min_bitrate_bps and min_start_bitrate_bps. Bug: none Change-Id: Ib8be965ea43f51013b0a0f82fd4256a372432dda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166600 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30338}
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.
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct