Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index 16122709c31fe976ddde3c63a477d78957f72070..bdb3a92b44645470f41cf484715f23ec1066c573 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -696,7 +696,18 @@ MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted( |
// Measure audio level (0-9) |
// TODO(henrik.lundin) Use the |muted| information here too. |
+ // TODO(deadbeef): Use RmsLevel for |_outputAudioLevel| as well. |
_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". |
hbos
2017/07/10 09:58:24
Ditto link.
Zach Stein
2017/07/10 18:35:20
Done.
|
+ double additional_energy = |
+ static_cast<double>(_outputAudioLevel.LevelFullRange()) / INT16_MAX; |
+ additional_energy *= additional_energy; |
+ _totalOutputEnergy += additional_energy * 0.01; |
+ _totalOutputDuration += 0.01; |
hbos
2017/07/10 09:58:24
nit: Can you have a constant double kAudioSampleDu
Zach Stein
2017/07/10 18:35:20
Done.
|
if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) { |
// The first frame with a valid rtp timestamp. |
@@ -2370,6 +2381,14 @@ 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; |