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libtgvoip/webrtc_dsp/modules/audio_processing/vad/gmm.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

62 lines
1.8 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/gmm.h"
#include <math.h>
namespace webrtc {
static const int kMaxDimension = 10;
static void RemoveMean(const double* in,
const double* mean_vec,
int dimension,
double* out) {
for (int n = 0; n < dimension; ++n)
out[n] = in[n] - mean_vec[n];
}
static double ComputeExponent(const double* in,
const double* covar_inv,
int dimension) {
double q = 0;
for (int i = 0; i < dimension; ++i) {
double v = 0;
for (int j = 0; j < dimension; j++)
v += (*covar_inv++) * in[j];
q += v * in[i];
}
q *= -0.5;
return q;
}
double EvaluateGmm(const double* x, const GmmParameters& gmm_parameters) {
if (gmm_parameters.dimension > kMaxDimension) {
return -1; // This is invalid pdf so the caller can check this.
}
double f = 0;
double v[kMaxDimension];
const double* mean_vec = gmm_parameters.mean;
const double* covar_inv = gmm_parameters.covar_inverse;
for (int n = 0; n < gmm_parameters.num_mixtures; n++) {
RemoveMean(x, mean_vec, gmm_parameters.dimension, v);
double q = ComputeExponent(v, covar_inv, gmm_parameters.dimension) +
gmm_parameters.weight[n];
f += exp(q);
mean_vec += gmm_parameters.dimension;
covar_inv += gmm_parameters.dimension * gmm_parameters.dimension;
}
return f;
}
} // namespace webrtc