Support for remote audio into tracks

webrtc/voice_engine/channel_proxy.h

 /*
  *  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 ~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_