| commit | 059fb4480b4bc402c2dd52c04c5028db50802f58 | [log] [tgz] |
|---|---|---|
| author | solenberg <solenberg@webrtc.org> | Wed Oct 26 12:12:24 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Oct 26 12:12:29 2016 |
| tree | 8442fbbb332724acccef532806bb339bad2b7626 | |
| parent | 16b6d6dc5b367746a9f910d1cebf9f65e8dd2c7f [diff] |
- Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing. - Update MockAudioProcessing to current APM interface. - Replace calls to VoEAudioProcessing::Start/StopAecDump with direct calls on APM. - Add AudioProcessing* in WVoE, get it from VoE, so we can call directly on APM. BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/2446143002 Cr-Commit-Position: refs/heads/master@{#14786}
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.