Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().

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.
  */
 
@@ skipping 55 matching lines @@
   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(int event, int duration_ms);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
+  virtual void SetInputMute(bool muted);
 
   virtual void RegisterExternalTransport(Transport* transport);
   virtual void DeRegisterExternalTransport();
   virtual bool ReceivedRTPPacket(const uint8_t* packet,
                                  size_t length,
                                  const PacketTime& packet_time);
   virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
 
   virtual const rtc::scoped_refptr<AudioDecoderFactory>&
   GetAudioDecoderFactory() const;
 
  private:
   Channel* channel() const;
 
   rtc::ThreadChecker thread_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_