b4977de306
Made path from NetEq to AudioTransport ready for many-channel audio. If there is one stream, we can handle anything that fits in an AudioFrame. For many streams, the current limit is 6. Some multi-channel combinations are not supported: e.g. if we get stereo audio and attempt to play out 6 channels. Changes: * AudioFrameOperations - replaced the MonoTo* and *ToMono methods by UpmixChannels & DownmixChannels. * AudioMixer: removed DCHECKs for <= 2 channels and tweaked the mixing algorithm to handle many channels. Bug: webrtc:8649 Change-Id: Ib83e16d463694e35658caa09c27849e853d508fb Reviewed-on: https://webrtc-review.googlesource.com/c/106040 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26446}
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