chromium/third_party/webrtc/modules/audio_processing/agc2/limiter.cc

/*
 *  Copyright (c) 2018 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 "modules/audio_processing/agc2/limiter.h"

#include <algorithm>
#include <array>
#include <cmath>

#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"

namespace webrtc {
namespace {

// This constant affects the way scaling factors are interpolated for the first
// sub-frame of a frame. Only in the case in which the first sub-frame has an
// estimated level which is greater than the that of the previous analyzed
// sub-frame, linear interpolation is replaced with a power function which
// reduces the chances of over-shooting (and hence saturation), however reducing
// the fixed gain effectiveness.
constexpr float kAttackFirstSubframeInterpolationPower = …;

void InterpolateFirstSubframe(float last_factor,
                              float current_factor,
                              rtc::ArrayView<float> subframe) { … }

void ComputePerSampleSubframeFactors(
    const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
    MonoView<float> per_sample_scaling_factors) { … }

void ScaleSamples(MonoView<const float> per_sample_scaling_factors,
                  DeinterleavedView<float> signal) { … }
}  // namespace

Limiter::Limiter(ApmDataDumper* apm_data_dumper,
                 size_t samples_per_channel,
                 absl::string_view histogram_name)
    : … { … }

Limiter::~Limiter() = default;

void Limiter::Process(DeinterleavedView<float> signal) { … }

InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const { … }

void Limiter::SetSamplesPerChannel(size_t samples_per_channel) { … }

void Limiter::Reset() { … }

float Limiter::LastAudioLevel() const { … }

}  // namespace webrtc