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diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
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+++ b/webrtc/voice_engine/channel_proxy.h
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+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
+#define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
+
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/voice_engine/channel_manager.h"
+#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+
+#include <string>
+#include <vector>
+
+namespace webrtc {
+
+class AudioSinkInterface;
+class PacketRouter;
+class RtpPacketSender;
+class TransportFeedbackObserver;
+
+namespace voe {
+
+class Channel;
+
+// This class provides the "view" of a voe::Channel that we need to implement
+// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
+// purposes:
+//  1. Allow mocking just the interfaces used, instead of the entire
+//     voe::Channel class.
+//  2. Provide a refined interface for the stream classes, including assumptions
+//     on return values and input adaptation.
+class ChannelProxy {
+ public:
+  ChannelProxy();
+  explicit ChannelProxy(const ChannelOwner& channel_owner);
+  virtual ~ChannelProxy();
+
+  virtual void SetRTCPStatus(bool enable);
+  virtual void SetLocalSSRC(uint32_t ssrc);
+  virtual void SetRTCP_CNAME(const std::string& c_name);
+  virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
+  virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
+  virtual void EnableSendTransportSequenceNumber(int id);
+  virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
+  virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
+  virtual void SetCongestionControlObjects(
+      RtpPacketSender* rtp_packet_sender,
+      TransportFeedbackObserver* transport_feedback_observer,
+      PacketRouter* packet_router);
+
+  virtual CallStatistics GetRTCPStatistics() const;
+  virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
+  virtual NetworkStatistics GetNetworkStatistics() const;
+  virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
+  virtual int32_t GetSpeechOutputLevelFullRange() const;
+  virtual uint32_t GetDelayEstimate() const;
+
+  virtual bool SetSendTelephoneEventPayloadType(int payload_type);
+  virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
+
+  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+
+ private:
+  Channel* channel() const;
+
+  rtc::ThreadChecker thread_checker_;
+  ChannelOwner channel_owner_;
+};
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_