tor-browser

The Tor Browser
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audio_level.cc (3146B)

      1 /*
      2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
      3 *
      4 *  Use of this source code is governed by a BSD-style license
      5 *  that can be found in the LICENSE file in the root of the source
      6 *  tree. An additional intellectual property rights grant can be found
      7 *  in the file PATENTS.  All contributing project authors may
      8 *  be found in the AUTHORS file in the root of the source tree.
      9 */
     10 
     11 #include "audio/audio_level.h"
     12 
     13 #include <cstdint>
     14 
     15 #include "api/audio/audio_frame.h"
     16 #include "common_audio/signal_processing/include/signal_processing_library.h"
     17 #include "rtc_base/synchronization/mutex.h"
     18 
     19 namespace webrtc {
     20 namespace voe {
     21 
     22 AudioLevel::AudioLevel()
     23    : abs_max_(0), count_(0), current_level_full_range_(0) {}
     24 
     25 AudioLevel::~AudioLevel() {}
     26 
     27 void AudioLevel::Reset() {
     28  MutexLock lock(&mutex_);
     29  abs_max_ = 0;
     30  count_ = 0;
     31  current_level_full_range_ = 0;
     32  total_energy_ = 0.0;
     33  total_duration_ = 0.0;
     34 }
     35 
     36 int16_t AudioLevel::LevelFullRange() const {
     37  MutexLock lock(&mutex_);
     38  return current_level_full_range_;
     39 }
     40 
     41 void AudioLevel::ResetLevelFullRange() {
     42  MutexLock lock(&mutex_);
     43  abs_max_ = 0;
     44  count_ = 0;
     45  current_level_full_range_ = 0;
     46 }
     47 
     48 double AudioLevel::TotalEnergy() const {
     49  MutexLock lock(&mutex_);
     50  return total_energy_;
     51 }
     52 
     53 double AudioLevel::TotalDuration() const {
     54  MutexLock lock(&mutex_);
     55  return total_duration_;
     56 }
     57 
     58 void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
     59  // Check speech level (works for 2 channels as well)
     60  int16_t abs_value =
     61      audioFrame.muted()
     62          ? 0
     63          : WebRtcSpl_MaxAbsValueW16(
     64                audioFrame.data(),
     65                audioFrame.samples_per_channel_ * audioFrame.num_channels_);
     66 
     67  // Protect member access using a lock since this method is called on a
     68  // dedicated audio thread in the RecordedDataIsAvailable() callback.
     69  MutexLock lock(&mutex_);
     70 
     71  if (abs_value > abs_max_)
     72    abs_max_ = abs_value;
     73 
     74  // Update level approximately 9 times per second, assuming audio frame
     75  // duration is approximately 10 ms. (The update frequency is every
     76  // 11th (= |kUpdateFrequency+1|) call: 1000/(11*10)=9.09..., we should
     77  // probably change this behavior, see https://crbug.com/webrtc/10784).
     78  if (count_++ == kUpdateFrequency) {
     79    current_level_full_range_ = abs_max_;
     80 
     81    count_ = 0;
     82 
     83    // Decay the absolute maximum (divide by 4)
     84    abs_max_ >>= 2;
     85  }
     86 
     87  // See the description for "totalAudioEnergy" in the WebRTC stats spec
     88  // (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
     89  // for an explanation of these formulas. In short, we need a value that can
     90  // be used to compute RMS audio levels over different time intervals, by
     91  // taking the difference between the results from two getStats calls. To do
     92  // this, the value needs to be of units "squared sample value * time".
     93  double additional_energy =
     94      static_cast<double>(current_level_full_range_) / INT16_MAX;
     95  additional_energy *= additional_energy;
     96  total_energy_ += additional_energy * duration;
     97  total_duration_ += duration;
     98 }
     99 
    100 }  // namespace voe
    101 }  // namespace webrtc