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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
34 lines
1023 B
C++
34 lines
1023 B
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/agc2/agc2_testing_common.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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std::vector<double> LinSpace(const double l,
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const double r,
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size_t num_points) {
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RTC_CHECK(num_points >= 2);
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std::vector<double> points(num_points);
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const double step = (r - l) / (num_points - 1.0);
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points[0] = l;
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for (size_t i = 1; i < num_points - 1; i++) {
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points[i] = static_cast<double>(l) + i * step;
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}
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points[num_points - 1] = r;
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return points;
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}
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} // namespace test
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} // namespace webrtc
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