This reverts commit 91fc422d0884da7b2b64549791b043258d7c5555. Reason for revert: owners.SyntaxErrorInOwnersFile: /b/s/w/ir/cache/builder/presubmit/src/media/OWNERS:11 syntax error: per-file globs cannot span directories or use escapes: "per-file engine/simulcast*=sprang@webrtc.org" Original change's description: > Add sprang@ as owner of simulcast.cc/h > > Bug: None > Change-Id: I41817d76726f526afcde5c934abd1f401b180a3c > Reviewed-on: https://webrtc-review.googlesource.com/86682 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23812} TBR=sprang@webrtc.org,stefan@webrtc.org Change-Id: Ia95e48769c4c2c81b3e3758038c8bfcb8c352589 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: None Reviewed-on: https://webrtc-review.googlesource.com/86900 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23818}
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
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.
Authoritative list of directories that contain the native API header files.
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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