b3808dcc36
This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31 Now using rtc_source_set to be able to generate runtime deps Original change's description: > Split perf-test-specific resources in low_bandwidth_audio_test > > Bug: chromium:755660 > Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6 > Reviewed-on: https://webrtc-review.googlesource.com/61961 > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22439} No-Try: True Bug: chromium:755660 Change-Id: I66eda6f016c98e2a8a99f230d9e0323cc09e4976 Reviewed-on: https://webrtc-review.googlesource.com/62020 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22450}
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