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/*
* 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() {
MutexLock lock(&mutex_);
abs_max_ = 0;
count_ = 0;
current_level_full_range_ = 0;
total_energy_ = 0.0;
total_duration_ = 0.0;
}
int16_t AudioLevel::LevelFullRange() const {
MutexLock lock(&mutex_);
return current_level_full_range_;
}
void AudioLevel::ResetLevelFullRange() {
MutexLock lock(&mutex_);
abs_max_ = 0;
count_ = 0;
current_level_full_range_ = 0;
}
double AudioLevel::TotalEnergy() const {
MutexLock lock(&mutex_);
return total_energy_;
}
double AudioLevel::TotalDuration() const {
MutexLock lock(&mutex_);
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.
MutexLock lock(&mutex_);
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
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