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libtgvoip/webrtc_dsp/modules/audio_processing/agc2/agc2_testing_common.cc

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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.
*/
#include "modules/audio_processing/agc2/agc2_testing_common.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
std::vector<double> LinSpace(const double l,
const double r,
size_t num_points) {
RTC_CHECK(num_points >= 2);
std::vector<double> points(num_points);
const double step = (r - l) / (num_points - 1.0);
points[0] = l;
for (size_t i = 1; i < num_points - 1; i++) {
points[i] = static_cast<double>(l) + i * step;
}
points[num_points - 1] = r;
return points;
}
} // namespace test
} // namespace webrtc