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

webrtc/voice_engine/transmit_mixer.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 295 matching lines @@
     {
         rtc::CritScope cs(&_critSect);
         file_recording =  _fileRecording;
     }
     if (file_recording)
     {
         RecordAudioToFile(_audioFrame.sample_rate_hz_);
     }
 
     // --- Measure audio level of speech after all processing.
-    _audioLevel.ComputeLevel(_audioFrame);
-
-    // See the description for "totalAudioEnergy" in the WebRTC stats spec
-    // (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
-    // 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>(_audioLevel.LevelFullRange()) / INT16_MAX;
-    additional_energy *= additional_energy;
     double sample_duration = static_cast<double>(nSamples) / samplesPerSec;
-    totalInputEnergy_ += additional_energy * sample_duration;
-    totalInputDuration_ += sample_duration;
+    _audioLevel.ComputeLevel(_audioFrame, sample_duration);
 
     return 0;
 }
 
 void TransmitMixer::ProcessAndEncodeAudio() {
   RTC_DCHECK_GT(_audioFrame.samples_per_channel_, 0);
   for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid();
        it.Increment()) {
     Channel* const channel = it.GetChannel();
     if (channel->Sending()) {
@@ skipping 525 matching lines @@
     return _audioLevel.Level();
 }
 
 int16_t TransmitMixer::AudioLevelFullRange() const
 {
     // Speech + file level [0,32767]
     return _audioLevel.LevelFullRange();
 }
 
 double TransmitMixer::GetTotalInputEnergy() const {
-  return totalInputEnergy_;
+  return _audioLevel.TotalEnergy();
 }
 
 double TransmitMixer::GetTotalInputDuration() const {
-  return totalInputDuration_;
+  return _audioLevel.TotalDuration();
 }
 
 bool TransmitMixer::IsRecordingCall()
 {
     return _fileCallRecording;
 }
 
 bool TransmitMixer::IsRecordingMic()
 {
     rtc::CritScope cs(&_critSect);
@@ skipping 154 matching lines @@
 void TransmitMixer::EnableStereoChannelSwapping(bool enable) {
   swap_stereo_channels_ = enable;
 }
 
 bool TransmitMixer::IsStereoChannelSwappingEnabled() {
   return swap_stereo_channels_;
 }
 
 }  // namespace voe
 }  // namespace webrtc