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libtgvoip/webrtc_dsp/modules/audio_processing/vad/pitch_internal.cc
Grishka 5caaaafa42 Updated WebRTC APM
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.
2018-11-23 04:02:53 +03:00

52 lines
2.1 KiB
C++

/*
* Copyright (c) 2012 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/vad/pitch_internal.h"
#include <cmath>
// A 4-to-3 linear interpolation.
// The interpolation constants are derived as following:
// Input pitch parameters are updated every 7.5 ms. Within a 30-ms interval
// we are interested in pitch parameters of 0-5 ms, 10-15ms and 20-25ms. This is
// like interpolating 4-to-6 and keep the odd samples.
// The reason behind this is that LPC coefficients are computed for the first
// half of each 10ms interval.
static void PitchInterpolation(double old_val, const double* in, double* out) {
out[0] = 1. / 6. * old_val + 5. / 6. * in[0];
out[1] = 5. / 6. * in[1] + 1. / 6. * in[2];
out[2] = 0.5 * in[2] + 0.5 * in[3];
}
void GetSubframesPitchParameters(int sampling_rate_hz,
double* gains,
double* lags,
int num_in_frames,
int num_out_frames,
double* log_old_gain,
double* old_lag,
double* log_pitch_gain,
double* pitch_lag_hz) {
// Gain interpolation is in log-domain, also returned in log-domain.
for (int n = 0; n < num_in_frames; n++)
gains[n] = log(gains[n] + 1e-12);
// Interpolate lags and gains.
PitchInterpolation(*log_old_gain, gains, log_pitch_gain);
*log_old_gain = gains[num_in_frames - 1];
PitchInterpolation(*old_lag, lags, pitch_lag_hz);
*old_lag = lags[num_in_frames - 1];
// Convert pitch-lags to Hertz.
for (int n = 0; n < num_out_frames; n++) {
pitch_lag_hz[n] = (sampling_rate_hz) / (pitch_lag_hz[n]);
}
}