This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. Original change's description: > Added PeerConnectionObserver::OnRemoveTrack. > > This corresponds to processing the removal of a remote track step of > the spec, with processing the addition of a remote track already > covered by OnAddTrack. > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > Bug: webrtc:8260, webrtc:8315 > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > Reviewed-on: https://webrtc-review.googlesource.com/4722 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20214} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/7940 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20218}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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.
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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