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

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 242 matching lines @@
   int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
   int StopRecordingPlayout();
 
   void SetMixWithMicStatus(bool mix);
 
   // Muting, Volume and Level.
   void SetInputMute(bool enable);
   void SetChannelOutputVolumeScaling(float scaling);
   int GetSpeechOutputLevel() const;
   int GetSpeechOutputLevelFullRange() const;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec.
+  double GetTotalOutputEnergy() const;
+  double GetTotalOutputDuration() const;
 
   // Stats.
   int GetNetworkStatistics(NetworkStatistics& stats);
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
   // Audio+Video Sync.
   uint32_t GetDelayEstimate() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
@@ skipping 190 matching lines @@
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
   acm2::CodecManager codec_manager_;
   acm2::RentACodec rent_a_codec_;
   std::unique_ptr<AudioSinkInterface> audio_sink_;
   AudioLevel _outputAudioLevel;
+  double _totalOutputEnergy = 0.0;
+  double _totalOutputDuration = 0.0;
   bool _externalTransport;
   // Downsamples to the codec rate if necessary.
   PushResampler<int16_t> input_resampler_;
   std::unique_ptr<FilePlayer> input_file_player_;
   std::unique_ptr<FilePlayer> output_file_player_;
   std::unique_ptr<FileRecorder> output_file_recorder_;
   int _inputFilePlayerId;
   int _outputFilePlayerId;
   int _outputFileRecorderId;
   bool _outputFileRecording;
@@ skipping 70 matching lines @@
 
   bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false;
 
   rtc::TaskQueue* encoder_queue_ = nullptr;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_