This reverts commit 3dc4d4a21f80cdf44c508412d784b254957696eb. Reason for revert: breaks internal project Original change's description: > Move clients of WebRtcSession to use PeerConnection > > This change is part of the work to merge WebRtcSession into > PeerConnection. To make that work easier, this moves all clients > of WebRtcSession to use shims added to PeerConnection. That way > when the classes are merged they won't need to be modified. > > Bug: webrtc:8183 > Change-Id: I43de7acf7e38c9fcf2dbf55d50eb05e73767c251 > Reviewed-on: https://webrtc-review.googlesource.com/4320 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20030} TBR=steveanton@webrtc.org,deadbeef@webrtc.org Change-Id: I13f335b24c26753429cd08a4ca3e295eed5660ff No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8183 Reviewed-on: https://webrtc-review.googlesource.com/4700 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20035}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
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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