Follows https://webrtc-review.googlesource.com/c/src/+/129768 closely. Adds an ENCODER and sets it up to parse SDP config for multistream opus. E.g. this is the new SDP syntax for 6.1 surround sound: "multiopus/48000/6 channel_mapping=0,4,1,2,3,5 num_streams=4 coupled_streams=2" Bug: webrtc:8649 Change-Id: I3fc341e76f5c41dab0243cf65f6461e4c3d9d67d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132001 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27775}
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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