Change log:d35e92ac3a..5198fef411Full diff:d35e92ac3a..5198fef411Changed dependencies: * src/base:14e5130655..21d4e642d8* src/build:277f67400d..8e7ce53a80* src/ios:a4ab0e8543..170f7c8508* src/testing:8370632ba2..4869e7efc6* src/third_party:6dad1d2bc8..bdcd9569fa* src/tools:5ee3add9c7..8a68b2d221DEPS diff:d35e92ac3a..5198fef411/DEPS No update to Clang. TBR= BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Review-Url: https://codereview.webrtc.org/3007613002 Cr-Commit-Position: refs/heads/master@{#19534}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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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://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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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