Change log:6995ce043e..e242527ec0Full diff:6995ce043e..e242527ec0Changed dependencies: * src/base:533dae7360..8322a45b6e* src/buildtools:26b7e66950..f6d165d9d8* src/ios:ec55a185d9..76b87738dd* src/testing:12a0d395a0..8fcd8f2775* src/third_party:55d2f862b6..3ef96a8e1f* src/third_party/catapult:6be590cea2..e1aa3179fa* src/tools:5f0140b10b..c8b855bc2fDEPS diff:6995ce043e..e242527ec0/DEPS No update to Clang. TBR=buildbot@webrtc.org, BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I176844ea9fb5b37938541877ac3f0e5b2b9fa596 Reviewed-on: https://webrtc-review.googlesource.com/2501 Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org> Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19913}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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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