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rirgen.cpp
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rirgen.cpp
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/*
Program : Room Impulse Response Generator
Description : Computes the response of an acoustic source to one or more
microphones in a reverberant room using the image method [1,2].
[1] J.B. Allen and D.A. Berkley,
Image method for efficiently simulating small-room acoustics,
Journal Acoustic Society of America, 65(4), April 1979, p 943.
[2] P.M. Peterson,
Simulating the response of multiple microphones to a single
acoustic source in a reverberant room, Journal Acoustic
Society of America, 80(5), November 1986.
Author : dr.ir. E.A.P. Habets (ehabets@dereverberation.org)
Version : 2.1.20141124
History : 1.0.20030606 Initial version
1.1.20040803 + Microphone directivity
+ Improved phase accuracy [2]
1.2.20040312 + Reflection order
1.3.20050930 + Reverberation Time
1.4.20051114 + Supports multi-channels
1.5.20051116 + High-pass filter [1]
+ Microphone directivity control
1.6.20060327 + Minor improvements
1.7.20060531 + Minor improvements
1.8.20080713 + Minor improvements
1.9.20090822 + 3D microphone directivity control
2.0.20100920 + Calculation of the source-image position
changed in the code and tutorial.
This ensures a proper response to reflections
in case a directional microphone is used.
2.1.20120318 + Avoid the use of unallocated memory
2.1.20140721 + Fixed computation of alpha
2.1.20141124 + The window and sinc are now both centered
around t=0
Copyright (C) 2003-2014 E.A.P. Habets, The Netherlands.
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "rirgen.h"
#define _USE_MATH_DEFINES
#include <cmath>
#include <limits>
#include <cstdlib>
#include <iostream>
#define ROUND(x) ((x)>=0?(long)((x)+0.5):(long)((x)-0.5))
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
template<typename T> inline const T sinc(T const& x) {
return (x==0) ? 1 : std::sin(x)/x;
}
double sim_microphone(double x, double y, double z, double* angle, char mtype) {
if (mtype == 'b' || mtype == 'c' || mtype == 's' || mtype == 'h') {
double gain, vartheta, varphi, rho;
// Polar Pattern rho
// ---------------------------
// Bidirectional 0
// Hypercardioid 0.25
// Cardioid 0.5
// Subcardioid 0.75
// Omnidirectional 1
switch(mtype) {
case 'b':
rho = 0;
break;
case 'h':
rho = 0.25;
break;
case 'c':
rho = 0.5;
break;
case 's':
rho = 0.75;
break;
};
vartheta = acos(z/sqrt(pow(x,2)+pow(y,2)+pow(z,2)));
varphi = atan2(y,x);
gain = sin(M_PI/2-angle[1]) * sin(vartheta) * cos(angle[0]-varphi) + cos(M_PI/2-angle[1]) * cos(vartheta);
gain = rho + (1-rho) * gain;
return gain;
} else {
return 1;
}
}
std::vector< std::vector<double> > computeRIR(double c, double fs, const std::vector< std::vector<double> >& rr, const std::vector<double>& ss, const std::vector<double>& LL, const std::vector<double>& beta_input, const std::vector<double>& orientation, int isHighPassFilter, int nDimension, int nOrder, int nSamples, char microphone_type) {
// | Room Impulse Response Generator |\n"
// | |\n"
// | Computes the response of an acoustic source to one or more |\n"
// | microphones in a reverberant room using the image method [1,2]. |\n"
// | |\n"
// | Author : dr.ir. Emanuel Habets (ehabets@dereverberation.org) |\n"
// | |\n"
// | Version : 2.1.20141124 |\n"
// | |\n"
// | Copyright (C) 2003-2014 E.A.P. Habets, The Netherlands. |\n"
// | |\n"
// | [1] J.B. Allen and D.A. Berkley, |\n"
// | Image method for efficiently simulating small-room acoustics,|\n"
// | Journal Acoustic Society of America, |\n"
// | 65(4), April 1979, p 943. |\n"
// | |\n"
// | [2] P.M. Peterson, |\n"
// | Simulating the response of multiple microphones to a single |\n"
// | acoustic source in a reverberant room, Journal Acoustic |\n"
// | Society of America, 80(5), November 1986. |\n"
// --------------------------------------------------------------------\n\n"
// function [h, beta_hat] = rir_generator(c, fs, r, s, L, beta, nsample,\n"
// mtype, order, dim, orientation, hp_filter);\n\n"
// Input parameters:\n"
// c : sound velocity in m/s.\n"
// fs : sampling frequency in Hz.\n"
// r : M x 3 array specifying the (x,y,z) coordinates of the\n"
// receiver(s) in m.\n"
// s : 1 x 3 vector specifying the (x,y,z) coordinates of the\n"
// source in m.\n"
// L : 1 x 3 vector specifying the room dimensions (x,y,z) in m.\n"
// beta : 1 x 6 vector specifying the reflection coefficients\n"
// [beta_x1 beta_x2 beta_y1 beta_y2 beta_z1 beta_z2] or\n"
// beta = reverberation time (T_60) in seconds.\n"
// nsample : number of samples to calculate, default is T_60*fs.\n"
// mtype : [omnidirectional, subcardioid, cardioid, hypercardioid,\n"
// bidirectional], default is omnidirectional.\n"
// order : reflection order, default is -1, i.e. maximum order.\n"
// dim : room dimension (2 or 3), default is 3.\n"
// orientation : direction in which the microphones are pointed, specified using\n"
// azimuth and elevation angles (in radians), default is [0 0].\n"
// hp_filter : use 'false' to disable high-pass filter, the high-pass filter\n"
// is enabled by default.\n\n"
// Output parameters:\n"
// h : M x nsample matrix containing the calculated room impulse\n"
// response(s).\n"
// beta_hat : In case a reverberation time is specified as an input parameter\n"
// the corresponding reflection coefficient is returned.\n\n");
// Load parameters
int nMicrophones = rr.size();
double beta[6];
double angle[2];
double reverberation_time;
if (beta_input.size() == 1) {
double V = LL[0]*LL[1]*LL[2];
double S = 2*(LL[0]*LL[2]+LL[1]*LL[2]+LL[0]*LL[1]);
reverberation_time = beta_input[0];
if (reverberation_time != 0) {
double alfa = 24*V*log(10.0)/(c*S*reverberation_time);
// if (alfa > 1)
// mexErrMsgTxt("Error: The reflection coefficients cannot be calculated using the current "
// "room parameters, i.e. room size and reverberation time.\n Please "
// "specify the reflection coefficients or change the room parameters.");
for (int i=0;i<6;i++)
beta[i] = sqrt(1-alfa);
} else {
for (int i=0;i<6;i++)
beta[i] = 0;
}
} else {
for (int i=0;i<6;i++)
beta[i] = beta_input[i];
}
// 3D Microphone orientation (optional)
if (orientation.size()) {
angle[0] = orientation[0];
angle[1] = orientation[1];
} else {
angle[0] = 0;
angle[1] = 0;
}
// Room Dimension (optional)
// if (nDimension != 2 && nDimension != 3)
// mexErrMsgTxt("Invalid input arguments! (9)");
if (nDimension == 2) {
beta[4] = 0;
beta[5] = 0;
}
// Reflection order (optional)
if (nOrder < -1)
{
// mexErrMsgTxt("Invalid input arguments! (8)");
}
// Number of samples (optional)
if (nSamples == -1) {
if (beta_input.size() > 1) {
double V = LL[0]*LL[1]*LL[2];
double alpha = ((1-pow(beta[0],2))+(1-pow(beta[1],2)))*LL[1]*LL[2] +
((1-pow(beta[2],2))+(1-pow(beta[3],2)))*LL[0]*LL[2] +
((1-pow(beta[4],2))+(1-pow(beta[5],2)))*LL[0]*LL[1];
reverberation_time = 24*log(10.0)*V/(c*alpha);
if (reverberation_time < 0.128)
reverberation_time = 0.128;
}
nSamples = (int) (reverberation_time * fs);
}
// Create output vector
std::vector< std::vector<double> > imp(nMicrophones);
for (int idxMicrophone = 0; idxMicrophone < nMicrophones ; idxMicrophone++)
imp[idxMicrophone].resize(nSamples);
// Temporary variables and constants (high-pass filter)
const double W = 2*M_PI*100/fs; // The cut-off frequency equals 100 Hz
const double R1 = exp(-W);
const double B1 = 2*R1*cos(W);
const double B2 = -R1 * R1;
const double A1 = -(1+R1);
double X0;
double* Y = new double[3];
// Temporary variables and constants (image-method)
const double Fc = 1; // The cut-off frequency equals fs/2 - Fc is the normalized cut-off frequency.
const int Tw = 2 * ROUND(0.004*fs); // The width of the low-pass FIR equals 8 ms
const double cTs = c/fs;
double* LPI = new double[Tw];
double* r = new double[3];
double* s = new double[3];
double* L = new double[3];
double Rm[3];
double Rp_plus_Rm[3];
double refl[3];
double fdist,dist;
double gain;
int startPosition;
int n1, n2, n3;
int q, j, k;
int mx, my, mz;
int n;
s[0] = ss[0]/cTs; s[1] = ss[1]/cTs; s[2] = ss[2]/cTs;
L[0] = LL[0]/cTs; L[1] = LL[1]/cTs; L[2] = LL[2]/cTs;
for (int idxMicrophone = 0; idxMicrophone < nMicrophones ; idxMicrophone++)
{
// [x_1 x_2 ... x_N y_1 y_2 ... y_N z_1 z_2 ... z_N]
r[0] = rr[idxMicrophone][0] / cTs;
r[1] = rr[idxMicrophone][1] / cTs;
r[2] = rr[idxMicrophone][2] / cTs;
n1 = (int) ceil(nSamples/(2*L[0]));
n2 = (int) ceil(nSamples/(2*L[1]));
n3 = (int) ceil(nSamples/(2*L[2]));
// Generate room impulse response
for (mx = -n1 ; mx <= n1 ; mx++)
{
Rm[0] = 2*mx*L[0];
for (my = -n2 ; my <= n2 ; my++)
{
Rm[1] = 2*my*L[1];
for (mz = -n3 ; mz <= n3 ; mz++)
{
Rm[2] = 2*mz*L[2];
for (q = 0 ; q <= 1 ; q++)
{
Rp_plus_Rm[0] = (1-2*q)*s[0] - r[0] + Rm[0];
refl[0] = pow(beta[0], std::abs(mx-q)) * pow(beta[1], std::abs(mx));
for (j = 0 ; j <= 1 ; j++)
{
Rp_plus_Rm[1] = (1-2*j)*s[1] - r[1] + Rm[1];
refl[1] = pow(beta[2], std::abs(my-j)) * pow(beta[3], std::abs(my));
for (k = 0 ; k <= 1 ; k++)
{
Rp_plus_Rm[2] = (1-2*k)*s[2] - r[2] + Rm[2];
refl[2] = pow(beta[4],std::abs(mz-k)) * pow(beta[5], std::abs(mz));
dist = sqrt(pow(Rp_plus_Rm[0], 2) + pow(Rp_plus_Rm[1], 2) + pow(Rp_plus_Rm[2], 2));
if (std::abs(2*mx-q)+std::abs(2*my-j)+std::abs(2*mz-k) <= nOrder || nOrder == -1)
{
fdist = floor(dist);
if (fdist < nSamples)
{
gain = sim_microphone(Rp_plus_Rm[0], Rp_plus_Rm[1], Rp_plus_Rm[2], angle, microphone_type)
* refl[0]*refl[1]*refl[2]/(4*M_PI*dist*cTs);
for (n = 0 ; n < Tw ; n++)
LPI[n] = 0.5 * (1 - cos(2*M_PI*((n+1-(dist-fdist))/Tw))) * Fc * sinc(M_PI*Fc*(n+1-(dist-fdist)-(Tw/2)));
startPosition = (int) fdist-(Tw/2)+1;
for (n = 0 ; n < Tw; n++)
if (startPosition+n >= 0 && startPosition+n < nSamples)
imp[idxMicrophone][startPosition + n] += gain * LPI[n];
}
}
}
}
}
}
}
}
// 'Original' high-pass filter as proposed by Allen and Berkley.
if (isHighPassFilter == 1)
{
for (int idx = 0 ; idx < 3 ; idx++) {Y[idx] = 0;}
for (int idx = 0 ; idx < nSamples ; idx++)
{
X0 = imp[idxMicrophone][idx];
Y[2] = Y[1];
Y[1] = Y[0];
Y[0] = B1*Y[1] + B2*Y[2] + X0;
imp[idxMicrophone][idx] = Y[0] + A1*Y[1] + R1*Y[2];
}
}
}
delete[] Y;
delete[] LPI;
delete[] r;
delete[] s;
delete[] L;
return imp;
}