Adding stats that can be used to compute output audio levels.

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 58 matching lines @@
     uint32_t CaptureLevel() const;
 
     int32_t StopSend();
 
     // TODO(solenberg): Remove, once AudioMonitor is gone.
     int8_t AudioLevel() const;
 
     // 'virtual' to allow mocking.
     virtual int16_t AudioLevelFullRange() const;
 
+    // See description of "totalAudioEnergy" in the WebRTC stats spec.
+    double GetTotalInputEnergy() const;
+
+    double GetTotalInputDuration() const;
+
     bool IsRecordingCall();
 
     bool IsRecordingMic();
 
     int StartPlayingFileAsMicrophone(const char* fileName,
                                      bool loop,
                                      FileFormats format,
                                      int startPosition,
                                      float volumeScaling,
                                      int stopPosition,
@@ skipping 93 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;

[hlundin-webrtc, 2017/07/03 13:23:03]

0.0 here and below.

[Zach Stein, 2017/07/06 17:41:37]

Done.

+    double _totalInputDuration = 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