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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 "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:
+ 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;
+ ~Limiter();
+
+ // 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;
+
+ 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_