| 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..84390d3f5fc84172e1eab7f634555e2b8cc1df4d 100644
|
| --- a/webrtc/voice_engine/transmit_mixer.cc
|
| +++ b/webrtc/voice_engine/transmit_mixer.cc
|
| @@ -308,6 +308,19 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
|
|
|
| // --- Measure audio level of speech after all processing.
|
| _audioLevel.ComputeLevel(_audioFrame);
|
| +
|
| + // TODO(zstein): Extract helper to share with voice_engine/channel.cc
|
| + // 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>(_audioLevel.LevelFullRange()) / INT16_MAX;
|
| + additional_energy *= additional_energy;
|
| + _totalInputEnergy += additional_energy * 0.01;
|
| + _totalInputDuration += 0.01;
|
| +
|
| return 0;
|
| }
|
|
|
| @@ -851,6 +864,14 @@ 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;
|
|
|