Fix threading issues in the audio pipeline for WebRTC remoting protocol.

remoting/protocol/webrtc_audio_sink_adapter.h

 // 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 "base/threading/thread_task_runner_handle.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
 
+namespace base {
+class SingleThreadTaskRunner;
+}  // namespace base
+
 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<base::SingleThreadTaskRunner> task_runner_;
+  base::WeakPtr<AudioStub> audio_stub_;
   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_