Change log:568b1c4cbe..e3b81fc1afFull diff:568b1c4cbe..e3b81fc1afChanged dependencies: * src/base:db9490a1bc..87130a8deb* src/build:8b27273109..2267fe9e91* src/ios:21c90fa060..1b4b90bfb0* src/testing:3ee68a77cf..889040ba67* src/third_party:60aff9b446..c917e53a50* src/third_party/catapult:db0acc015b..8e06404d69* src/tools:d6ad039ca3..a65b03f4a6DEPS diff:568b1c4cbe..e3b81fc1af/DEPS No update to Clang. TBR= BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Review-Url: https://codereview.webrtc.org/3010793002 Cr-Commit-Position: refs/heads/master@{#19610}
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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