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

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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 53 matching lines @@
 struct SenderInfo;
 
 namespace voe {
 
 class OutputMixer;
 class RtcEventLogProxy;
 class RtcpRttStatsProxy;
 class RtpPacketSenderProxy;
 class Statistics;
 class TransportFeedbackProxy;
-class TransmitMixer;
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
 
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
 // Example: a member can be set on thread T1 and read by an internal audio
 // thread T2. Accessing the member via this class ensures that we are
 // safe and also avoid TSan v2 warnings.
 class ChannelState {
  public:
@@ skipping 63 matching lines @@
       Channel*& channel,
       int32_t channelId,
       uint32_t instanceId,
       const VoEBase::ChannelConfig& config);
   Channel(int32_t channelId,
           uint32_t instanceId,
           const VoEBase::ChannelConfig& config);
   int32_t Init();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                OutputMixer& outputMixer,
-                               TransmitMixer& transmitMixer,
                                ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
                                VoiceEngineObserver* voiceEngineObserver,
                                rtc::CriticalSection* callbackCritSect);
   int32_t UpdateLocalTimeStamp();
 
   void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
   // TODO(ossu): Don't use! It's only here to confirm that the decoder factory
   // passed into AudioReceiveStream is the same as the one set when creating the
@@ skipping 301 matching lines @@
   std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
   // The rtp timestamp of the first played out audio frame.
   int64_t capture_start_rtp_time_stamp_;
   // The capture ntp time (in local timebase) of the first played out audio
   // frame.
   int64_t capture_start_ntp_time_ms_ GUARDED_BY(ts_stats_lock_);
 
   // uses
   Statistics* _engineStatisticsPtr;
   OutputMixer* _outputMixerPtr;
-  TransmitMixer* _transmitMixerPtr;
   ProcessThread* _moduleProcessThreadPtr;
   AudioDeviceModule* _audioDeviceModulePtr;
   VoiceEngineObserver* _voiceEngineObserverPtr;  // owned by base
   rtc::CriticalSection* _callbackCritSectPtr;    // owned by base
   Transport* _transportPtr;  // WebRtc socket or external transport
   RmsLevel rms_level_;
   int32_t _sendFrameType;  // Send data is voice, 1-voice, 0-otherwise
   // VoEBase
   bool _mixFileWithMicrophone;
   // VoEVolumeControl
@@ skipping 28 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_