Change log:fc9df5d5ac..7b8dc77cd5Full diff:fc9df5d5ac..7b8dc77cd5Changed dependencies: * src/ios:314a0f2012..a9b9da8a76* src/testing:da54118c24..596142d858* src/third_party:6e6ee356d4..835e148124* src/tools:7776c991de..a705bd3076DEPS diff:fc9df5d5ac..7b8dc77cd5/DEPS No update to Clang. TBR=buildbot@webrtc.org, BUG=None CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal Change-Id: I83a95b2cbb0bab0bc13d4951b82c79a12afac8cb Reviewed-on: https://webrtc-review.googlesource.com/17240 Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org> Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20488}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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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