| Index: talk/app/webrtc/remoteaudiotrack.cc
|
| diff --git a/talk/app/webrtc/remoteaudiotrack.cc b/talk/app/webrtc/remoteaudiotrack.cc
|
| index df27f05e4859b2f423a950485aef50915501dd41..2c4481c80d71b8c9f6b5eeef0bbbef03bf3f848a 100644
|
| --- a/talk/app/webrtc/remoteaudiotrack.cc
|
| +++ b/talk/app/webrtc/remoteaudiotrack.cc
|
| @@ -26,3 +26,70 @@
|
| */
|
|
|
| #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
|
|
|