| commit | 10111bc49539589113c76318b7ba53591946e78a | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Thu Nov 17 14:48:48 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Nov 17 14:48:56 2016 |
| tree | 77f265acb012692c73cdea5dae0602818783e6b1 | |
| parent | dd31071e19bdfe0bedc335f070e1592cfd5d2eb6 [diff] |
Passed AudioMixer to AudioState::Config. This is a refactoring change in preparation for enabling AudioMixer with the goal to have a small CL as possible for passing audio through the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/ enables the mixer. An object of class AudioState is shared across different webrtc audio connections. It is created in tests and in WebRTCVoiceEngine. AudioState is constructed by passing a Config struct, where one argument is scoped_refptr<AudioMixer>. Populating this field has now been mandatory. Tests and WebRTCVoiceEngine create and pass either a AudioMixerImpl. WebRTCVoiceEngine passes a real AudioMixer, which is currently unused. An alternative would have tests pass a mocked audio mixer. We chose not to do that, because we believe that tests should use the real thing unless there are reasons against it. Construction time is not an issue, because the real mixer is relatively lightweight. We couldn't find a way to test any mixer-related changes in AudioState before the mixes is connected. The next dependent CL https://codereview.webrtc.org/2436033002/ contains unit tests for mixer usage. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2469743002 Cr-Commit-Position: refs/heads/master@{#15134}
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.