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> Making WebRTC able to play and record audio to files for tests. > > By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to > WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to > play out audio to a file and feed audio in from a file. We want to do > so we can better test WebRTC-using applications by recording what the > audio stack outputs and feeding known audio in for quality tests. > > R=henrika@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/20609004 TBR=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6396 4adac7df-926f-26a2-2b94-8c16560cd09d
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.