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

webrtc/voice_engine/transmit_mixer.cc

Index: webrtc/voice_engine/transmit_mixer.cc
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 6796f8457c5e5b3060920c6be9995b8f739612b2..95e9225183dcd7efa5ce7cbd72e351a10572fc5f 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -308,6 +308,20 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
 
     // --- 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;
+
     return 0;
 }
 
@@ -851,6 +865,14 @@ int16_t TransmitMixer::AudioLevelFullRange() const
     return _audioLevel.LevelFullRange();
 }
 
+double TransmitMixer::GetTotalInputEnergy() const {
+  return _totalInputEnergy;
+}
+
+double TransmitMixer::GetTotalInputDuration() const {
+  return _totalInputDuration;
+}
+
 bool TransmitMixer::IsRecordingCall()
 {
     return _fileCallRecording;