Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.

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 42 matching lines @@
 #define WEBRTC_BOOL_STUB(method, args) \
   bool method args override { return true; }
 
 #define WEBRTC_VOID_STUB(method, args) \
   void method args override {}
 
 #define WEBRTC_FUNC(method, args) int method args override
 
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEBase, public webrtc::VoECodec,
-      public webrtc::VoEHardware,
-      public webrtc::VoEVolumeControl {
+      public webrtc::VoEHardware {
  public:
   struct Channel {
     std::vector<webrtc::CodecInst> recv_codecs;
     size_t neteq_capacity = 0;
     bool neteq_fast_accelerate = false;
   };
 
   explicit FakeWebRtcVoiceEngine(webrtc::AudioProcessing* apm,
                                  webrtc::voe::TransmitMixer* transmit_mixer)
       : apm_(apm), transmit_mixer_(transmit_mixer) {
@@ skipping 142 matching lines @@
   WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
   WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
   WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
   bool BuiltInAECIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
   bool BuiltInAGCIsAvailable() const override { return false; }
   WEBRTC_STUB(EnableBuiltInNS, (bool enable));
   bool BuiltInNSIsAvailable() const override { return false; }
 
-  // webrtc::VoEVolumeControl
-  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_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));
-
   size_t GetNetEqCapacity() const {
     auto ch = channels_.find(last_channel_);
     RTC_DCHECK(ch != channels_.end());
     return ch->second->neteq_capacity;
   }
   bool GetNetEqFastAccelerate() const {
     auto ch = channels_.find(last_channel_);
     RTC_CHECK(ch != channels_.end());
     return ch->second->neteq_fast_accelerate;
   }
 
  private:
   bool inited_ = false;
   int last_channel_ = -1;
   std::map<int, Channel*> channels_;
   bool fail_create_channel_ = false;
   webrtc::AudioProcessing* apm_ = nullptr;
   webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_