81da488ab6
The audio_device_module field was currently unused. The audio_mixer field is going to be used to pass an AudioMixer to AudioState. In the hopefully-not-very-far future, the toplevel WebRTC API will allow passing a custom AudioMixer, e.g. for spatialized audio (audio in space). If no mixer is passed, a default mixer is created (the one in modules/audio_mixer). The only object which will have a permanent reference to the mixer is AudioState. AudioState is created in WebRTCVoiceEngine with a configuration object, which already contains a VoiceEngine pointer. In this CL, we extend this config object with a mixer pointer. In summary: in an upcoming CL, a mixer will be either created in or passed to WebRTCVoiceEngine. This mixer will be passed to the ctor of AudioState in a config struct. BUG=webrtc:6346 NOTRY=True Review-Url: https://codereview.webrtc.org/2456363002 Cr-Commit-Position: refs/heads/master@{#14973}
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