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

webrtc/voice_engine/channel.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 32 matching lines @@
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
 namespace voe {
 
 namespace {
 
+constexpr double kAudioSampleDurationSeconds = 0.01;
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
 
 }  // namespace
 
 const int kTelephoneEventAttenuationdB = 10;
 
 class RtcEventLogProxy final : public webrtc::RtcEventLog {
  public:
   RtcEventLogProxy() : event_log_(nullptr) {}
@@ skipping 626 matching lines @@
   {
     rtc::CritScope cs(&_fileCritSect);
 
     if (_outputFileRecording && output_file_recorder_) {
       output_file_recorder_->RecordAudioToFile(*audioFrame);
     }
   }
 
   // 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.
     capture_start_rtp_time_stamp_ = audioFrame->timestamp_;
   }
 
   if (capture_start_rtp_time_stamp_ >= 0) {
     // audioFrame.timestamp_ should be valid from now on.
 
     // Compute elapsed time.
@@ skipping 1653 matching lines @@
 }
 
 int Channel::GetSpeechOutputLevel() const {
   return _outputAudioLevel.Level();
 }
 
 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;
 }
 
 bool Channel::InputMute() const {
   rtc::CritScope cs(&volume_settings_critsect_);
   return input_mute_;
 }
 
@@ skipping 757 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc