Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/webrtc_audio_stream.cc

Index: remoting/protocol/webrtc_audio_stream.cc
diff --git a/remoting/protocol/webrtc_audio_stream.cc b/remoting/protocol/webrtc_audio_stream.cc
new file mode 100644
index 0000000000000000000000000000000000000000..5ede04ec81bcb85d81f86b6234df982b00ccfc9b
--- /dev/null
+++ b/remoting/protocol/webrtc_audio_stream.cc
@@ -0,0 +1,71 @@
+// Copyright 2016 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 "remoting/protocol/webrtc_audio_stream.h"
+
+#include "base/location.h"
+#include "base/logging.h"
+#include "base/single_thread_task_runner.h"
+#include "remoting/base/constants.h"
+#include "remoting/protocol/audio_source.h"
+#include "remoting/protocol/webrtc_audio_source_adapter.h"
+#include "remoting/protocol/webrtc_transport.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
+#include "third_party/webrtc/api/peerconnectioninterface.h"
+#include "third_party/webrtc/base/refcount.h"
+
+namespace remoting {
+namespace protocol {
+
+const char kAudioStreamLabel[] = "audio_stream";
+const char kAudioTrackLabel[] = "system_audio";
+
+WebrtcAudioStream::WebrtcAudioStream() {}
+
+WebrtcAudioStream::~WebrtcAudioStream() {
+  if (stream_) {
+    for (const auto& track : stream_->GetAudioTracks()) {
+      stream_->RemoveTrack(track.get());
+    }
+    peer_connection_->RemoveStream(stream_.get());
+  }
+}
+
+void WebrtcAudioStream::Start(
+    scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner,
+    std::unique_ptr<AudioSource> audio_source,
+    WebrtcTransport* webrtc_transport) {
+  DCHECK(webrtc_transport);
+
+  source_adapter_ =
+      new rtc::RefCountedObject<WebrtcAudioSourceAdapter>(audio_task_runner);
+  source_adapter_->Start(std::move(audio_source));
+
+  scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_connection_factory(
+      webrtc_transport->peer_connection_factory());
+  peer_connection_ = webrtc_transport->peer_connection();
+
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track =
+      peer_connection_factory->CreateAudioTrack(kAudioTrackLabel,
+                                                source_adapter_.get());
+
+  stream_ = peer_connection_factory->CreateLocalMediaStream(kAudioStreamLabel);
+
+  // AddTrack() may fail only if there is another track with the same name,
+  // which is impossible because it's a brand new stream.
+  bool result = stream_->AddTrack(audio_track.get());
+  DCHECK(result);
+
+  // AddStream() may fail if there is another stream with the same name or when
+  // the PeerConnection is closed, neither is expected.
+  result = peer_connection_->AddStream(stream_.get());
+  DCHECK(result);
+}
+
+void WebrtcAudioStream::Pause(bool pause) {
+  source_adapter_->Pause(pause);
+}
+
+}  // namespace protocol
+}  // namespace remoting