Added WebRTC audio quality test using fake device; improved fake device

media/base/media_switches.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/base/media_switches.h"
 
 namespace switches {
 
 // Allow users to specify a custom buffer size for debugging purpose.
 const char kAudioBufferSize[] = "audio-buffer-size";
@@ skipping 79 matching lines @@
 
 // Use fake device for Media Stream to replace actual camera and microphone.
 const char kUseFakeDeviceForMediaStream[] = "use-fake-device-for-media-stream";
 
 // Use an .y4m file to play as the webcam. See the comments in
 // media/video/capture/file_video_capture_device.h for more details.
 const char kUseFileForFakeVideoCapture[] = "use-file-for-fake-video-capture";
 
 // Play a .wav file as the microphone. Note that for WebRTC calls we'll treat
 // the bits as if they came from the microphone, which means you should disable
-// audio processing (lest your audio file will play back distorted). Also the
-// wav file will have to have the same number of channels, bits per sample and
-// sampling frequency as the system's input device.
+// audio processing (lest your audio file will play back distorted). The input
+// file is converted to suit Chrome's audio buses if necessary, so most sane
+// .wav files should work.
 const char kUseFileForFakeAudioCapture[] = "use-file-for-fake-audio-capture";
 
 // Enables support for inband text tracks in media content.
 const char kEnableInbandTextTracks[] = "enable-inband-text-tracks";
 
 }  // namespace switches