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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.
58 lines
2.0 KiB
C++
58 lines
2.0 KiB
C++
/*
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* Copyright (c) 2013 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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// MSVC++ requires this to be set before any other includes to get M_PI.
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#define _USE_MATH_DEFINES
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#include "common_audio/resampler/sinusoidal_linear_chirp_source.h"
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#include <math.h>
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namespace webrtc {
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SinusoidalLinearChirpSource::SinusoidalLinearChirpSource(int sample_rate,
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size_t samples,
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double max_frequency,
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double delay_samples)
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: sample_rate_(sample_rate),
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total_samples_(samples),
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max_frequency_(max_frequency),
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current_index_(0),
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delay_samples_(delay_samples) {
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// Chirp rate.
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double duration = static_cast<double>(total_samples_) / sample_rate_;
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k_ = (max_frequency_ - kMinFrequency) / duration;
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}
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void SinusoidalLinearChirpSource::Run(size_t frames, float* destination) {
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for (size_t i = 0; i < frames; ++i, ++current_index_) {
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// Filter out frequencies higher than Nyquist.
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if (Frequency(current_index_) > 0.5 * sample_rate_) {
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destination[i] = 0;
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} else {
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// Calculate time in seconds.
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if (current_index_ < delay_samples_) {
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destination[i] = 0;
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} else {
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// Sinusoidal linear chirp.
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double t = (current_index_ - delay_samples_) / sample_rate_;
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destination[i] = sin(2 * M_PI * (kMinFrequency * t + (k_ / 2) * t * t));
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}
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}
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}
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}
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double SinusoidalLinearChirpSource::Frequency(size_t position) {
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return kMinFrequency + (position - delay_samples_) *
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(max_frequency_ - kMinFrequency) / total_samples_;
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}
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} // namespace webrtc
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