| commit | 051f678808d03fb2e68fe2a084b1deeb326c71bf | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Mon Oct 31 10:26:40 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Oct 31 10:26:48 2016 |
| tree | bfcfeaaa2d7ba1676302f3550334be8a3b3eafc3 | |
| parent | 9aa78832f99eb0b76bb4655194fec5c291918187 [diff] |
Add a NeededFrequency() method to the AudioMixer::Source interface. This change will allow for a audio source to report its sampling rate to the audio mixer. It is needed in order to mix at a lower sampling rate. Mixing at a lower sampling rate can in many cases lead to big efficiency improvements, as reported by experiments. The code affected is all implementations of the Source interface: AudioReceiveStream and a mock class. The AudioReceiveStream now queries its underlying voe::Channel object for the needed frequency. Note that the changes to the mixing algorithm are done in a later CL. BUG=webrtc:6346 NOTRY=True TBR=solenberg@webrtc.org Review-Url: https://codereview.webrtc.org/2448113009 Cr-Commit-Position: refs/heads/master@{#14839}
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.