| 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.
|
|
|