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

webrtc/voice_engine/channel.cc

 /*
  *  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 678 matching lines @@
   {
     rtc::CritScope cs(&_fileCritSect);
 
     if (_outputFileRecording && output_file_recorder_) {
       output_file_recorder_->RecordAudioToFile(*audioFrame);
     }
   }
 
   // Measure audio level (0-9)
   // TODO(henrik.lundin) Use the |muted| information here too.
+  // TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| as well.
+  // TODO(zstein): Use sample_rate_hz_?

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

What would you do with sample_rate_hz_?

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

I wasn't sure if this method was intended to process 10ms of audio or if the
duration should be dynamic. Looking again, it looks like 10ms is guaranteed here
though, so I will remove this.

[hlundin-webrtc, 2017/07/07 06:48:44]

Acknowledged.

   _outputAudioLevel.ComputeLevel(*audioFrame);
+  // See the description for "totalAudioEnergy" in the WebRTC stats spec for an
+  // explanation of these formulas. In short, we need a value that can be used
+  // to compute RMS audio levels over different time intervals, by taking the
+  // difference between the results from two getStats calls. To do this, the
+  // value needs to be of units "squared sample value * time".
+  double additional_energy =
+      static_cast<double>(_outputAudioLevel.LevelFullRange()) / INT16_MAX;
+  additional_energy *= additional_energy;
+  _totalOutputEnergy += additional_energy * 0.01;
+  _totalOutputDuration += 0.01;
 
   if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) {
     // The first frame with a valid rtp timestamp.
     capture_start_rtp_time_stamp_ = audioFrame->timestamp_;
   }
 
   if (capture_start_rtp_time_stamp_ >= 0) {
     // audioFrame.timestamp_ should be valid from now on.
 
     // Compute elapsed time.
@@ skipping 1653 matching lines @@
 }
 
 int Channel::GetSpeechOutputLevel() const {
   return _outputAudioLevel.Level();
 }
 
 int Channel::GetSpeechOutputLevelFullRange() const {
   return _outputAudioLevel.LevelFullRange();
 }
 
+double Channel::GetTotalOutputEnergy() const {
+  return _totalOutputEnergy;
+}
+
+double Channel::GetTotalOutputDuration() const {
+  return _totalOutputDuration;
+}
+
 void Channel::SetInputMute(bool enable) {
   rtc::CritScope cs(&volume_settings_critsect_);
   input_mute_ = enable;
 }
 
 bool Channel::InputMute() const {
   rtc::CritScope cs(&volume_settings_critsect_);
   return input_mute_;
 }
 
@@ skipping 757 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc