Make fake audio file playback loop by default.

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 "build/build_config.h"
 #include "media/base/media_switches.h"
 
 namespace switches {
 
 // Allow users to specify a custom buffer size for debugging purpose.
@@ skipping 75 matching lines @@
 const char kUseFakeDeviceForMediaStream[] = "use-fake-device-for-media-stream";
 
 // Use an .y4m file to play as the webcam. See the comments in
 // media/capture/video/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). The input
 // file is converted to suit Chrome's audio buses if necessary, so most sane
-// .wav files should work.
+// .wav files should work. You can pass either <path> to play the file looping
+// or <path>%noloop to stop after playing the file to completion.
 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";
 
 // When running tests on a system without the required hardware or libraries,
 // this flag will cause the tests to fail. Otherwise, they silently succeed.
 const char kRequireAudioHardwareForTesting[] =
     "require-audio-hardware-for-testing";
 
 // Allows clients to override the threshold for when the media renderer will
 // declare the underflow state for the video stream when audio is present.
 // TODO(dalecurtis): Remove once experiments for http://crbug.com/470940 finish.
 const char kVideoUnderflowThresholdMs[] = "video-underflow-threshold-ms";
 
 // Disables the new rendering algorithm for webrtc, which is designed to improve
 // the rendering smoothness.
 const char kDisableRTCSmoothnessAlgorithm[] =
     "disable-rtc-smoothness-algorithm";
 
 }  // namespace switches
 
 namespace media {
 
 // Use shared block-based buffering for media.
 const base::Feature kUseNewMediaCache{"use-new-media-cache",
                                       base::FEATURE_ENABLED_BY_DEFAULT};
 
 }  // namespace media