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

/*
 *  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.
 */

#include "modules/audio_processing/agc2/agc2_testing_common.h"

#include <math.h>

#include "rtc_base/checks.h"

namespace webrtc {
namespace test {

std::vector<double> LinSpace(double l, double r, int num_points) { … }

WhiteNoiseGenerator::WhiteNoiseGenerator(int min_amplitude, int max_amplitude)
    : … { … }

float WhiteNoiseGenerator::operator()() { … }

SineGenerator::SineGenerator(float amplitude,
                             float frequency_hz,
                             int sample_rate_hz)
    : … { … }

float SineGenerator::operator()() { … }

PulseGenerator::PulseGenerator(float pulse_amplitude,
                               float no_pulse_amplitude,
                               float frequency_hz,
                               int sample_rate_hz)
    : … { … }

float PulseGenerator::operator()() { … }

}  // namespace test
}  // namespace webrtc