This reverts commit a5fbc23379823d74b8cf4bc18887ff40237989e8. Reason for revert: <INSERT REASONING HERE> Original change's description: > BWE allocation strategy > > This is reland of https://webrtc-review.googlesource.com/c/src/+/4860 with the fixed RampUpTest test > > Bug: webrtc:8243 > Change-Id: I4b90a449b00dd05feee974001e08fb40710b59ac > Reviewed-on: https://webrtc-review.googlesource.com/13124 > Commit-Queue: Alex Narest <alexnarest@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20345} TBR=stefan@webrtc.org,alexnarest@webrtc.org Change-Id: I8ed12cd2115ef63204e384cc93c9f4473daa54d1 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8243 Reviewed-on: https://webrtc-review.googlesource.com/14020 Commit-Queue: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Alex Narest <alexnarest@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20361}
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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