text: /*
   *  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.
   */
  
  #include "audio/audio_level.h"
  
  #include "api/audio/audio_frame.h"
  #include "common_audio/signal_processing/include/signal_processing_library.h"
  
  namespace webrtc {
  namespace voe {
  
  AudioLevel::AudioLevel()
      : abs_max_(0), count_(0), current_level_full_range_(0) {
  }
  
  AudioLevel::~AudioLevel() {}
  
  void AudioLevel::Reset() {
    rtc::CritScope cs(&crit_sect_);
    abs_max_ = 0;
    count_ = 0;
    current_level_full_range_ = 0;
    total_energy_ = 0.0;
    total_duration_ = 0.0;
  }
  
  int16_t AudioLevel::LevelFullRange() const {
    rtc::CritScope cs(&crit_sect_);
    return current_level_full_range_;
  }
  
  void AudioLevel::ResetLevelFullRange() {
    rtc::CritScope cs(&crit_sect_);
    abs_max_ = 0;
    count_ = 0;
    current_level_full_range_ = 0;
  }
  
  double AudioLevel::TotalEnergy() const {
    rtc::CritScope cs(&crit_sect_);
    return total_energy_;
  }
  
  double AudioLevel::TotalDuration() const {
    rtc::CritScope cs(&crit_sect_);
    return total_duration_;
  }
  
  void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
    // Check speech level (works for 2 channels as well)
    int16_t abs_value =
        audioFrame.muted()
            ? 0
            : WebRtcSpl_MaxAbsValueW16(
                  audioFrame.data(),
                  audioFrame.samples_per_channel_ * audioFrame.num_channels_);
  
    // Protect member access using a lock since this method is called on a
    // dedicated audio thread in the RecordedDataIsAvailable() callback.
    rtc::CritScope cs(&crit_sect_);
  
    if (abs_value > abs_max_)
      abs_max_ = abs_value;
  
    // Update level approximately 9 times per second, assuming audio frame
    // duration is approximately 10 ms. (The update frequency is every
    // 11th (= |kUpdateFrequency+1|) call: 1000/(11*10)=9.09..., we should
    // probably change this behavior, see https://crbug.com/webrtc/10784).
    if (count_++ == kUpdateFrequency) {
      current_level_full_range_ = abs_max_;
  
      count_ = 0;
  
      // Decay the absolute maximum (divide by 4)
      abs_max_ >>= 2;
    }
  
    // 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>(current_level_full_range_) / INT16_MAX;
    additional_energy *= additional_energy;
    total_energy_ += additional_energy * duration;
    total_duration_ += duration;
  }
  
  }  // namespace voe
  }  // namespace webrtc
  