Care should be taken when landing this, because it will affect users of WebRTC. I'm thinking primarily of Chromium. Chromium will start to support High profile and Baseline profile using SW codecs with this CL. Clients who do SDP munging without looking at the H264 profile might switch from Constrained Baseline to High profile with this change. Bug: webrtc:8317 Change-Id: Idca3a6b761a66d9e521b913b850c6ae14381f1f4 Reviewed-on: https://webrtc-review.googlesource.com/6341 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20190}
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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