This was first reviewed in https://codereview.webrtc.org/2790493004/. It got reverted in https://codereview.webrtc.org/2791453004/ due to upstreaming error. Bug: None TBR: niklas.enbom@webrtc.org Change-Id: Ia5e9bf86e004258b2aa7822bd489d357fcb8f906 Reviewed-on: https://chromium-review.googlesource.com/645634 Reviewed-by: Minyue Li (BackIn2018March) <minyue@webrtc.org> Commit-Queue: Minyue Li (BackIn2018March) <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19642}
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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