dd31071e19
All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
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. This page is maintained by the Google Chrome team.
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://chromium.googlesource.com/external/webrtc
- 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