Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().

webrtc/audio/audio_receive_stream.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 23 matching lines @@
   AudioReceiveStream(CongestionController* congestion_controller,
                      const webrtc::AudioReceiveStream::Config& config,
                      const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
   ~AudioReceiveStream() override;
 
   // webrtc::AudioReceiveStream implementation.
   void Start() override;
   void Stop() override;
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
+  void SetGain(float gain) override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   const PacketTime& packet_time);
   const webrtc::AudioReceiveStream::Config& config() const;
 
  private:
   VoiceEngine* voice_engine() const;
 
   rtc::ThreadChecker thread_checker_;
   RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
   const webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_