chromium/third_party/webrtc/modules/audio_processing/agc2/agc2_testing_common.h

/*
 *  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_AGC2_TESTING_COMMON_H_
#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_

#include <limits>
#include <vector>

#include "rtc_base/random.h"

namespace webrtc {
namespace test {

constexpr float kMinS16 = …;
constexpr float kMaxS16 = …;

// Level Estimator test parameters.
constexpr float kDecayMs = …;

// Limiter parameters.
constexpr float kLimiterMaxInputLevelDbFs = …;
constexpr float kLimiterKneeSmoothnessDb = …;
constexpr float kLimiterCompressionRatio = …;

// Returns evenly spaced `num_points` numbers over a specified interval [l, r].
std::vector<double> LinSpace(double l, double r, int num_points);

// Generates white noise.
class WhiteNoiseGenerator { … };

// Generates a sine function.
class SineGenerator { … };

// Generates periodic pulses.
class PulseGenerator { … };

}  // namespace test
}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_