| commit | 81da488ab6c9c79e4a74e29d18c574285312241f | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Tue Nov 08 12:26:30 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Nov 08 12:26:37 2016 |
| tree | 8119611370dfe8472c8091a74522038950d0c6b9 | |
| parent | 40532a164663f03b812ec7ccc893da7a4bdc26d3 [diff] |
Added audio mixer and removed audio device module in AudioState::Config. 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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.