blob: f1c5b68dc17c491c6589b1ade0a6d954f5a5bf38 [file] [log] [blame]
/*
* 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) {
WebRtcSpl_Init();
}
AudioLevel::~AudioLevel() {}
int16_t AudioLevel::LevelFullRange() const {
rtc::CritScope cs(&crit_sect_);
return current_level_full_range_;
}
void AudioLevel::Clear() {
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 10 times per second
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