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* 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 <vector>
#include "absl/strings/string_view.h"
#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class ApmDataDumper;
class Limiter {
Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
absl::string_view histogram_name_prefix);
Limiter(const Limiter& limiter) = delete;
Limiter& operator=(const Limiter& limiter) = delete;
// Applies limiter and hard-clipping to `signal`.
void Process(AudioFrameView<float> signal);
InterpolatedGainCurve::Stats GetGainCurveStats() const;
// Supported rates must be
// * supported by FixedDigitalLevelEstimator
// * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
// so that samples_per_channel fit in the
// per_sample_scaling_factors_ array.
void SetSampleRate(int sample_rate_hz);
// Resets the internal state.
void Reset();
float LastAudioLevel() const;
const InterpolatedGainCurve interp_gain_curve_;
FixedDigitalLevelEstimator level_estimator_;
ApmDataDumper* const apm_data_dumper_ = nullptr;
// Work array containing the sub-frame scaling factors to be interpolated.
std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
std::array<float, kMaximalNumberOfSamplesPerChannel>
per_sample_scaling_factors_ = {};
float last_scaling_factor_ = 1.f;
} // namespace webrtc