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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_AGC2_TESTING_COMMON_H_
#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
#include <vector>
#include "rtc_base/basictypes.h"
namespace webrtc {
namespace test {
// Level Estimator test parameters.
constexpr float kDecayMs = 500.f;
// Limiter parameters.
constexpr float kLimiterMaxInputLevelDbFs = 1.f;
constexpr float kLimiterKneeSmoothnessDb = 1.f;
constexpr float kLimiterCompressionRatio = 5.f;
std::vector<double> LinSpace(const double l, const double r, size_t num_points);
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_