Change log:0f448170cb..568b1c4cbeFull diff:0f448170cb..568b1c4cbeChanged dependencies: * src/base:b55b56c514..db9490a1bc* src/build:815636f502..8b27273109* src/ios:7879f4ba18..21c90fa060* src/testing:61972623dc..3ee68a77cf* src/third_party:536d63f06f..60aff9b446* src/third_party/catapult:021853793c..db0acc015b* src/tools:bf7858c2d4..d6ad039ca3DEPS diff:0f448170cb..568b1c4cbe/DEPS No update to Clang. TBR= BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Review-Url: https://codereview.webrtc.org/3008953002 Cr-Commit-Position: refs/heads/master@{#19609}
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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