| Index: talk/app/webrtc/remoteaudiotrack.cc | 
| diff --git a/talk/app/webrtc/remoteaudiotrack.cc b/talk/app/webrtc/remoteaudiotrack.cc | 
| index 2c4481c80d71b8c9f6b5eeef0bbbef03bf3f848a..5f0b23e59e3553483f1c41180fdd6c6135c93594 100644 | 
| --- a/talk/app/webrtc/remoteaudiotrack.cc | 
| +++ b/talk/app/webrtc/remoteaudiotrack.cc | 
| @@ -25,71 +25,4 @@ | 
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | 
| */ | 
|  | 
| -#include "talk/app/webrtc/remoteaudiotrack.h" | 
| - | 
| -#include "talk/app/webrtc/remoteaudiosource.h" | 
| - | 
| -using rtc::scoped_refptr; | 
| - | 
| -namespace webrtc { | 
| - | 
| -// static | 
| -scoped_refptr<RemoteAudioTrack> RemoteAudioTrack::Create( | 
| -    const std::string& id, | 
| -    const scoped_refptr<RemoteAudioSource>& source) { | 
| -  return new rtc::RefCountedObject<RemoteAudioTrack>(id, source); | 
| -} | 
| - | 
| -RemoteAudioTrack::RemoteAudioTrack( | 
| -    const std::string& label, | 
| -    const scoped_refptr<RemoteAudioSource>& source) | 
| -    : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) { | 
| -  audio_source_->RegisterObserver(this); | 
| -  TrackState new_state = kInitializing; | 
| -  switch (audio_source_->state()) { | 
| -    case MediaSourceInterface::kLive: | 
| -    case MediaSourceInterface::kMuted: | 
| -      new_state = kLive; | 
| -      break; | 
| -    case MediaSourceInterface::kEnded: | 
| -      new_state = kEnded; | 
| -      break; | 
| -    case MediaSourceInterface::kInitializing: | 
| -    default: | 
| -      // kInitializing; | 
| -      break; | 
| -  } | 
| -  set_state(new_state); | 
| -} | 
| - | 
| -RemoteAudioTrack::~RemoteAudioTrack() { | 
| -  set_state(MediaStreamTrackInterface::kEnded); | 
| -  audio_source_->UnregisterObserver(this); | 
| -} | 
| - | 
| -std::string RemoteAudioTrack::kind() const { | 
| -  return MediaStreamTrackInterface::kAudioKind; | 
| -} | 
| - | 
| -AudioSourceInterface* RemoteAudioTrack::GetSource() const { | 
| -  return audio_source_.get(); | 
| -} | 
| - | 
| -void RemoteAudioTrack::AddSink(AudioTrackSinkInterface* sink) { | 
| -  audio_source_->AddSink(sink); | 
| -} | 
| - | 
| -void RemoteAudioTrack::RemoveSink(AudioTrackSinkInterface* sink) { | 
| -  audio_source_->RemoveSink(sink); | 
| -} | 
| - | 
| -bool RemoteAudioTrack::GetSignalLevel(int* level) { | 
| -  return false; | 
| -} | 
| - | 
| -void RemoteAudioTrack::OnChanged() { | 
| -  if (audio_source_->state() == MediaSourceInterface::kEnded) | 
| -    set_state(MediaStreamTrackInterface::kEnded); | 
| -} | 
| - | 
| -}  // namespace webrtc | 
| +// TODO(tommi): Delete this file when removed from build files in Chromium. | 
|  |