Move total audio energy and duration tracking to AudioLevel and protect with existing critial secti…

webrtc/voice_engine/transmit_mixer.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 179 matching lines @@
     std::unique_ptr<FilePlayer> file_player_;
     std::unique_ptr<FileRecorder> file_recorder_;
     std::unique_ptr<FileRecorder> file_call_recorder_;
     int _filePlayerId = 0;
     int _fileRecorderId = 0;
     int _fileCallRecorderId = 0;
     bool _filePlaying = false;
     bool _fileRecording = false;
     bool _fileCallRecording = false;
     voe::AudioLevel _audioLevel;
-    double totalInputEnergy_ = 0.0;
-    double totalInputDuration_ = 0.0;
     // protect file instances and their variables in MixedParticipants()
     rtc::CriticalSection _critSect;
     rtc::CriticalSection _callbackCritSect;
 
 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     MonitorModule<TransmitMixer> _monitorModule;
     webrtc::TypingDetection _typingDetection;
     bool _typingNoiseWarningPending = false;
     bool _typingNoiseDetected = false;
 #endif
 
     int _instanceId = 0;
     bool _mixFileWithMicrophone = false;
     uint32_t _captureLevel = 0;
     bool stereo_codec_ = false;
     bool swap_stereo_channels_ = false;
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H