b3cb8ab4ed
am: 9a337512d9 * commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits) Add tests for verifying transport feedback for audio and video. Eliminate defines in talk/ Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) Remove assert which was incorrectly added to TcpPort::OnSentPacket. Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. Update with new default boringssl no-aes cipher suites. Re-enable tests. Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ ) Re-land: "Use an explicit identifier in Config" Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ ) Remove libfuzzer trybot from default trybot set. Add ramp-up tests for transport sequence number with and w/o audio. Delete remnants of non-square pixel support from cricket::VideoFrame. Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop(). Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) Re-enable tests that failed under Linux_Msan. Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) Roll chromium_revision 346fea9..099be58 (369082:369139) Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan Add build_protobuf variable. ...
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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