Instead of the longest frame since the last GetStats call, the longest frame received during last 10 seconds should be returned in GetStats(). Previous way is not a good idea because there are maybe several consumers of GetStats calls. If not all of them parse timing frame reports, some of them may be lost. Also, streamline reporting of TimingFrames via GetStats (remove separate methods and use VideoReceiveStream::Stats struct instead). BUG=webrtc:7594 Review-Url: https://codereview.webrtc.org/3008983002 Cr-Commit-Position: refs/heads/master@{#19650}
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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