051f678808
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}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.