Change log:946d80b37f..f52ff80530Full diff:946d80b37f..f52ff80530Changed dependencies: * src/base:058098bf6b..a3848e11ae* src/build:465f7fab27..0119b14e14* src/ios:50b48ba029..79eaa19c53* src/testing:57bea64dc9..e9068583fe* src/third_party:b1265e559b..3c251d88b3* src/third_party/catapult:596aa47eec..dd30f4f383* src/third_party/gtest-parallel:de7390a463..965cfdccf9* src/tools:87be0ad4d0..02f1273b3bDEPS diff:946d80b37f..f52ff80530/DEPS No update to Clang. TBR= BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Review-Url: https://codereview.webrtc.org/3013103002 . Cr-Commit-Position: refs/heads/master@{#19837}
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://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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