Add Audio support in Chromoting host when using WebRTC.

remoting/protocol/webrtc_audio_stream.h

Index: remoting/protocol/webrtc_audio_stream.h
diff --git a/remoting/protocol/webrtc_audio_stream.h b/remoting/protocol/webrtc_audio_stream.h
new file mode 100644
index 0000000000000000000000000000000000000000..6c5981a3aa44c045661037d50d24a4c45cadd1e2
--- /dev/null
+++ b/remoting/protocol/webrtc_audio_stream.h
@@ -0,0 +1,54 @@
+// 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.
+
+#ifndef REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_
+#define REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_
+
+#include <memory>
+
+#include "base/macros.h"
+#include "base/memory/ref_counted.h"
+#include "remoting/protocol/audio_stream.h"
+
+namespace base {
+class SingleThreadTaskRunner;
+}  // namespace webrtc
+
+namespace webrtc {
+class MediaStreamInterface;
+class PeerConnectionInterface;
+}  // namespace webrtc
+
+namespace remoting {
+namespace protocol {
+
+class AudioSource;
+class WebrtcAudioSourceAdapter;
+class WebrtcTransport;
+
+class WebrtcAudioStream : public AudioStream {
+ public:
+  WebrtcAudioStream();
+  ~WebrtcAudioStream() override;
+
+  void Start(scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner,
+             std::unique_ptr<AudioSource> audio_source,
+             WebrtcTransport* webrtc_transport);
+
+  // AudioStream interface.
+  void Pause(bool pause) override;
+
+ private:
+  scoped_refptr<WebrtcAudioSourceAdapter> source_adapter_;
+
+  scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  scoped_refptr<webrtc::MediaStreamInterface> stream_;
+
+  DISALLOW_COPY_AND_ASSIGN(WebrtcAudioStream);
+};
+
+}  // namespace protocol
+}  // namespace remoting
+
+#endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_STREAM_H_