Change log:e3b81fc1af..d96af20ab5Full diff:e3b81fc1af..d96af20ab5Changed dependencies: * src/build:2267fe9e91..79e6eb4ea4* src/ios:1b4b90bfb0..5b1f1341d4* src/testing:889040ba67..bc3a93604e* src/third_party:c917e53a50..c86e0bc867* src/third_party/catapult:8e06404d69..1831170b35* src/tools:a65b03f4a6..9215ac4334DEPS diff:e3b81fc1af..d96af20ab5/DEPS No update to Clang. TBR= BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Review-Url: https://codereview.webrtc.org/3012613002 Cr-Commit-Position: refs/heads/master@{#19616}
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://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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