mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 09:37:52 +01:00
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.
72 lines
2.0 KiB
C++
72 lines
2.0 KiB
C++
/*
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* Copyright (c) 2014 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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#define _USE_MATH_DEFINES
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#include "common_audio/window_generator.h"
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#include <cmath>
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#include <complex>
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#include "rtc_base/checks.h"
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using std::complex;
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namespace {
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// Modified Bessel function of order 0 for complex inputs.
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complex<float> I0(complex<float> x) {
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complex<float> y = x / 3.75f;
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y *= y;
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return 1.0f + y * (3.5156229f +
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y * (3.0899424f +
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y * (1.2067492f +
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y * (0.2659732f +
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y * (0.360768e-1f + y * 0.45813e-2f)))));
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}
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} // namespace
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namespace webrtc {
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void WindowGenerator::Hanning(int length, float* window) {
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RTC_CHECK_GT(length, 1);
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RTC_CHECK(window != nullptr);
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for (int i = 0; i < length; ++i) {
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window[i] =
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0.5f * (1 - cosf(2 * static_cast<float>(M_PI) * i / (length - 1)));
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}
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}
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void WindowGenerator::KaiserBesselDerived(float alpha,
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size_t length,
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float* window) {
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RTC_CHECK_GT(length, 1U);
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RTC_CHECK(window != nullptr);
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const size_t half = (length + 1) / 2;
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float sum = 0.0f;
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for (size_t i = 0; i <= half; ++i) {
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complex<float> r = (4.0f * i) / length - 1.0f;
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sum += I0(static_cast<float>(M_PI) * alpha * sqrt(1.0f - r * r)).real();
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window[i] = sum;
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}
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for (size_t i = length - 1; i >= half; --i) {
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window[length - i - 1] = sqrtf(window[length - i - 1] / sum);
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window[i] = window[length - i - 1];
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}
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if (length % 2 == 1) {
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window[half - 1] = sqrtf(window[half - 1] / sum);
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}
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}
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} // namespace webrtc
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