b4e06d032e
API changes: - webrtc::AudioProcessing::Config::VoiceDetection removed - webrtc::AudioProcessingStats::voice_detected deprecated - cricket::AudioOptions::typing_detection deprecated - webrtc::StatsReport::StatsValueName:: kStatsValueNameTypingNoiseState deprecated PSA: https://groups.google.com/g/discuss-webrtc/c/7X6uwmJarE0 Bug: webrtc:11226,webrtc:11292 Change-Id: I8d008b56708cf62961b9857ec052b59fda3b41bf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250666 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35975}
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