Support for remote audio into tracks

talk/app/webrtc/remoteaudiotrack.cc

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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  * 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:
+      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