Add audio support in WebrtcConnectionToHost, audio unittest

remoting/protocol/webrtc_audio_sink_adapter.h

Index: remoting/protocol/webrtc_audio_sink_adapter.h
diff --git a/remoting/protocol/webrtc_audio_sink_adapter.h b/remoting/protocol/webrtc_audio_sink_adapter.h
new file mode 100644
index 0000000000000000000000000000000000000000..4f4969a1809f61c77142efcd51fdeb962216fb3c
--- /dev/null
+++ b/remoting/protocol/webrtc_audio_sink_adapter.h
@@ -0,0 +1,39 @@
+// 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_SINK_ADAPTER_H_
+#define REMOTING_PROTOCOL_WEBRTC_AUDIO_SINK_ADAPTER_H_
+
+#include "base/memory/ref_counted.h"
+#include "base/memory/weak_ptr.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
+
+namespace remoting {
+namespace protocol {
+
+class AudioStub;
+
+class WebrtcAudioSinkAdapter : public webrtc::AudioTrackSinkInterface {
+ public:
+  WebrtcAudioSinkAdapter(scoped_refptr<webrtc::MediaStreamInterface> stream,
+                         base::WeakPtr<AudioStub> audio_stub);
+  ~WebrtcAudioSinkAdapter() override;
+
+  void OnData(const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              size_t number_of_channels,
+              size_t number_of_frames) override;
+
+ private:
+  scoped_refptr<webrtc::MediaStreamInterface> media_stream_;
+  scoped_refptr<webrtc::AudioTrackInterface> audio_track_;
+
+  base::WeakPtr<AudioStub> audio_stub_;
+};
+
+}  // namespace protocol
+}  // namespace remoting
+
+#endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_SINK_ADAPTER_H_