Change log:97fba65068..f181c11ae2Full diff:97fba65068..f181c11ae2Changed dependencies: * src/base:ce0467d52d..bdaf4d9893* src/build:866fc229b0..92ccaf1432* src/buildtools:84fdc99243..cbc33b9c0a* src/ios:65cfd9a7f5..c47b4b2b8c* src/testing:595dd3f22b..dcfc1809e9* src/third_party:19f9ecc451..478c4078e3* src/third_party/gtest-parallel:c82123114d..7676778438* src/tools:63d88c95a5..c0173fdd43DEPS diff:97fba65068..f181c11ae2/DEPS No update to Clang. TBR= BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Review-Url: https://codereview.webrtc.org/3006203002 Cr-Commit-Position: refs/heads/master@{#19722}
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
- 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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