This reverts commit 28addd03c1040fc5a634ed92a7edde2aea811e79. Reason for revert: Speculative revert: looks like all win bots turned purple Original change's description: > Make it possible to isolate bwe_simulations_tests > > Add missing data dependencies and add it to gn_isolate_map.pyl > > Bug: chromium:749648 > Change-Id: I6b6c1bb2e4d647471a2747042788a691ce2e1e5d > Reviewed-on: https://webrtc-review.googlesource.com/8721 > Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org> > Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20258} TBR=kjellander@webrtc.org,ehmaldonado@webrtc.org,stefan@webrtc.org Change-Id: I8d07560ba3a60b97cf6525723a0f9888a72a1b1d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: chromium:749648 Reviewed-on: https://webrtc-review.googlesource.com/8860 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20259}
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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