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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.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
#include <array>
#include <cstddef>
#include "absl/strings/string_view.h"
#include "api/audio/audio_view.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
namespace webrtc {
class ApmDataDumper;
class Limiter {
public:
// See `SetSamplesPerChannel()` for valid values for `samples_per_channel`.
Limiter(ApmDataDumper* apm_data_dumper,
size_t samples_per_channel,
absl::string_view histogram_name_prefix);
Limiter(const Limiter& limiter) = delete;
Limiter& operator=(const Limiter& limiter) = delete;
~Limiter();
// Applies limiter and hard-clipping to `signal`.
void Process(DeinterleavedView<float> signal);
InterpolatedGainCurve::Stats GetGainCurveStats() const;
// Supported values must be
// * Supported by FixedDigitalLevelEstimator
// * Below or equal to kMaximalNumberOfSamplesPerChannel so that samples
// fit in the per_sample_scaling_factors_ array.
void SetSamplesPerChannel(size_t samples_per_channel);
// Resets the internal state.
void Reset();
float LastAudioLevel() const;
private:
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
#endif // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
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