chromium/third_party/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.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/fixed_digital_level_estimator.h"

#include <algorithm>
#include <cmath>

#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"

namespace webrtc {
namespace {

constexpr float kInitialFilterStateLevel =;

// Instant attack.
constexpr float kAttackFilterConstant =;

// Limiter decay constant.
// Computed as `10 ** (-1/20 * subframe_duration / kDecayMs)` where:
// - `subframe_duration` is `kFrameDurationMs / kSubFramesInFrame`;
// - `kDecayMs` is defined in agc2_testing_common.h.
constexpr float kDecayFilterConstant =;

}  // namespace

FixedDigitalLevelEstimator::FixedDigitalLevelEstimator(
    size_t samples_per_channel,
    ApmDataDumper* apm_data_dumper)
    :{}

void FixedDigitalLevelEstimator::CheckParameterCombination() {}

std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
    DeinterleavedView<const float> float_frame) {}

void FixedDigitalLevelEstimator::SetSamplesPerChannel(
    size_t samples_per_channel) {}

void FixedDigitalLevelEstimator::Reset() {}

}  // namespace webrtc