limiter.h - mozsearch

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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 <vector>

#include "absl/strings/string_view.h"

#include "api/audio/audio_frame.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 {

 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_