8f5787a919
* VoEBase contains only stub methods (until downstream code is updated). * voe::Channel and ChannelProxy classes remain, but are now created internally to the streams. As a result, internal::Audio[Receive|Send]Stream can have a ChannelProxy injected for testing. * Stream classes share Call::module_process_thread_ for their RtpRtcp modules, rather than using a separate thread shared only among audio streams. * voe::Channel instances use Call::worker_queue_ for encoding packets, rather than having a separate queue for audio (send) streams. Bug: webrtc:4690 Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d Reviewed-on: https://webrtc-review.googlesource.com/34640 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21578}
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