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

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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 111 matching lines @@
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
     Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
     bool playout = false;
-    float volume_scale = 1.0f;
     bool vad = false;
     bool codec_fec = false;
     int max_encoding_bandwidth = 0;
     bool opus_dtx = false;
     int cn8_type = 13;
     int cn16_type = 105;
     int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     int neteq_capacity = -1;
@@ skipping 291 matching lines @@
   WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
   WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
   WEBRTC_STUB(SetMicVolume, (unsigned int));
   WEBRTC_STUB(GetMicVolume, (unsigned int&));
   WEBRTC_STUB(SetInputMute, (int, bool));
   WEBRTC_STUB(GetInputMute, (int, bool&));
   WEBRTC_STUB(GetSpeechInputLevel, (unsigned int&));
   WEBRTC_STUB(GetSpeechOutputLevel, (int, unsigned int&));
   WEBRTC_STUB(GetSpeechInputLevelFullRange, (unsigned int&));
   WEBRTC_STUB(GetSpeechOutputLevelFullRange, (int, unsigned int&));
-  WEBRTC_FUNC(SetChannelOutputVolumeScaling, (int channel, float scale)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->volume_scale= scale;
-    return 0;
-  }
-  WEBRTC_FUNC(GetChannelOutputVolumeScaling, (int channel, float& scale)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    scale = channels_[channel]->volume_scale;
-    return 0;
-  }
+  WEBRTC_STUB(SetChannelOutputVolumeScaling, (int channel, float scale));
+  WEBRTC_STUB(GetChannelOutputVolumeScaling, (int channel, float& scale));
   WEBRTC_STUB(SetOutputVolumePan, (int channel, float left, float right));
   WEBRTC_STUB(GetOutputVolumePan, (int channel, float& left, float& right));
 
   // webrtc::VoEAudioProcessing
   WEBRTC_FUNC(SetNsStatus, (bool enable, webrtc::NsModes mode)) {
     ns_enabled_ = enable;
     ns_mode_ = mode;
     return 0;
   }
   WEBRTC_FUNC(GetNsStatus, (bool& enabled, webrtc::NsModes& mode)) {
@@ skipping 129 matching lines @@
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   int playout_fail_channel_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_