Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index 16122709c31fe976ddde3c63a477d78957f72070..f69246fc85138f3969bc0e98a384f28e9f1ad937 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -50,6 +50,7 @@ namespace voe { |
namespace { |
+constexpr double kAudioSampleDurationSeconds = 0.01; |
constexpr int64_t kMaxRetransmissionWindowMs = 1000; |
constexpr int64_t kMinRetransmissionWindowMs = 30; |
@@ -696,7 +697,20 @@ 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 (see |
+ // https://crbug.com/webrtc/7517). |
_outputAudioLevel.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>(_outputAudioLevel.LevelFullRange()) / INT16_MAX; |
+ additional_energy *= additional_energy; |
+ totalOutputEnergy_ += additional_energy * kAudioSampleDurationSeconds; |
+ totalOutputDuration_ += kAudioSampleDurationSeconds; |
if (capture_start_rtp_time_stamp_ < 0 && audioFrame->timestamp_ != 0) { |
// The first frame with a valid rtp timestamp. |
@@ -2370,6 +2384,14 @@ 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; |