mirror of
https://github.com/mpv-player/mpv
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77456b04d9
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14276 b3059339-0415-0410-9bf9-f77b7e298cf2
435 lines
13 KiB
C
435 lines
13 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU General Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
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//
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//=============================================================================
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*/
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/* Design and implementation of different types of digital filters
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*/
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#include <string.h>
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#include <math.h>
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#include "dsp.h"
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/******************************************************************************
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* FIR filter implementations
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******************************************************************************/
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/* C implementation of FIR filter y=w*x
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n number of filter taps, where mod(n,4)==0
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w filter taps
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x input signal must be a circular buffer which is indexed backwards
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*/
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inline _ftype_t af_filter_fir(register unsigned int n, _ftype_t* w, _ftype_t* x)
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{
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register _ftype_t y; // Output
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y = 0.0;
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do{
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n--;
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y+=w[n]*x[n];
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}while(n != 0);
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return y;
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}
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/* C implementation of parallel FIR filter y(k)=w(k) * x(k) (where * denotes convolution)
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n number of filter taps, where mod(n,4)==0
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d number of filters
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xi current index in xq
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w filter taps k by n big
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x input signal must be a circular buffers which are indexed backwards
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y output buffer
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s output buffer stride
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*/
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inline _ftype_t* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** w, _ftype_t** x, _ftype_t* y, unsigned int s)
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{
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register _ftype_t* xt = *x + xi;
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register _ftype_t* wt = *w;
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register int nt = 2*n;
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while(d-- > 0){
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*y = af_filter_fir(n,wt,xt);
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wt+=n;
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xt+=nt;
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y+=s;
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}
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return y;
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}
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/* Add new data to circular queue designed to be used with a parallel
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FIR filter, with d filters. xq is the circular queue, in pointing
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at the new samples, xi current index in xq and n the length of the
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filter. xq must be n*2 by k big, s is the index for in.
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*/
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inline int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s)
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{
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register _ftype_t* txq = *xq + xi;
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register int nt = n*2;
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while(d-- >0){
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*txq= *(txq+n) = *in;
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txq+=nt;
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in+=s;
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}
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return (++xi)&(n-1);
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}
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/******************************************************************************
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* FIR filter design
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******************************************************************************/
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/* Design FIR filter using the Window method
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n filter length must be odd for HP and BS filters
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w buffer for the filter taps (must be n long)
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fc cutoff frequencies (1 for LP and HP, 2 for BP and BS)
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0 < fc < 1 where 1 <=> Fs/2
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flags window and filter type as defined in filter.h
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variables are ored together: i.e. LP|HAMMING will give a
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low pass filter designed using a hamming window
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opt beta constant used only when designing using kaiser windows
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returns 0 if OK, -1 if fail
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*/
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int af_filter_design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _ftype_t opt)
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{
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unsigned int o = n & 1; // Indicator for odd filter length
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unsigned int end = ((n + 1) >> 1) - o; // Loop end
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unsigned int i; // Loop index
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_ftype_t k1 = 2 * M_PI; // 2*pi*fc1
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_ftype_t k2 = 0.5 * (_ftype_t)(1 - o);// Constant used if the filter has even length
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_ftype_t k3; // 2*pi*fc2 Constant used in BP and BS design
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_ftype_t g = 0.0; // Gain
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_ftype_t t1,t2,t3; // Temporary variables
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_ftype_t fc1,fc2; // Cutoff frequencies
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// Sanity check
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if(!w || (n == 0)) return -1;
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// Get window coefficients
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switch(flags & WINDOW_MASK){
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case(BOXCAR):
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af_window_boxcar(n,w); break;
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case(TRIANG):
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af_window_triang(n,w); break;
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case(HAMMING):
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af_window_hamming(n,w); break;
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case(HANNING):
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af_window_hanning(n,w); break;
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case(BLACKMAN):
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af_window_blackman(n,w); break;
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case(FLATTOP):
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af_window_flattop(n,w); break;
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case(KAISER):
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af_window_kaiser(n,w,opt); break;
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default:
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return -1;
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}
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if(flags & (LP | HP)){
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fc1=*fc;
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// Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
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fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
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k1 *= fc1;
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if(flags & LP){ // Low pass filter
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// If the filter length is odd, there is one point which is exactly
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// in the middle. The value at this point is 2*fCutoff*sin(x)/x,
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// where x is zero. To make sure nothing strange happens, we set this
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// value separately.
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if (o){
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w[end] = fc1 * w[end] * 2.0;
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g=w[end];
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}
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// Create filter
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for (i=0 ; i<end ; i++){
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t1 = (_ftype_t)(i+1) - k2;
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w[end-i-1] = w[n-end+i] = w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
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g += 2*w[end-i-1]; // Total gain in filter
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}
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}
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else{ // High pass filter
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if (!o) // High pass filters must have odd length
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return -1;
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w[end] = 1.0 - (fc1 * w[end] * 2.0);
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g= w[end];
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// Create filter
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for (i=0 ; i<end ; i++){
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t1 = (_ftype_t)(i+1);
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w[end-i-1] = w[n-end+i] = -1 * w[end-i-1] * sin(k1 * t1)/(M_PI * t1); // Sinc
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g += ((i&1) ? (2*w[end-i-1]) : (-2*w[end-i-1])); // Total gain in filter
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}
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}
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}
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if(flags & (BP | BS)){
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fc1=fc[0];
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fc2=fc[1];
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// Cutoff frequencies must be < 1.0 where 1.0 <=> Fs/2
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fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25;
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fc2 = ((fc2 <= 1.0) && (fc2 > 0.0)) ? fc2/2 : 0.25;
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k3 = k1 * fc2; // 2*pi*fc2
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k1 *= fc1; // 2*pi*fc1
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if(flags & BP){ // Band pass
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// Calculate center tap
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if (o){
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g=w[end]*(fc1+fc2);
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w[end] = (fc2 - fc1) * w[end] * 2.0;
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}
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// Create filter
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for (i=0 ; i<end ; i++){
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t1 = (_ftype_t)(i+1) - k2;
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t2 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
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t3 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
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g += w[end-i-1] * (t3 + t2); // Total gain in filter
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w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
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}
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}
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else{ // Band stop
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if (!o) // Band stop filters must have odd length
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return -1;
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w[end] = 1.0 - (fc2 - fc1) * w[end] * 2.0;
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g= w[end];
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// Create filter
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for (i=0 ; i<end ; i++){
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t1 = (_ftype_t)(i+1);
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t2 = sin(k1 * t1)/(M_PI * t1); // Sinc fc1
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t3 = sin(k3 * t1)/(M_PI * t1); // Sinc fc2
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w[end-i-1] = w[n-end+i] = w[end-i-1] * (t2 - t3);
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g += 2*w[end-i-1]; // Total gain in filter
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}
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}
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}
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// Normalize gain
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g=1/g;
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for (i=0; i<n; i++)
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w[i] *= g;
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return 0;
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}
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/* Design polyphase FIR filter from prototype filter
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n length of prototype filter
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k number of polyphase components
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w prototype filter taps
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pw Parallel FIR filter
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g Filter gain
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flags FWD forward indexing
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REW reverse indexing
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ODD multiply every 2nd filter tap by -1 => HP filter
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returns 0 if OK, -1 if fail
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*/
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int af_filter_design_pfir(unsigned int n, unsigned int k, _ftype_t* w, _ftype_t** pw, _ftype_t g, unsigned int flags)
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{
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int l = (int)n/k; // Length of individual FIR filters
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int i; // Counters
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int j;
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_ftype_t t; // g * w[i]
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// Sanity check
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if(l<1 || k<1 || !w || !pw)
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return -1;
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// Do the stuff
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if(flags&REW){
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for(j=l-1;j>-1;j--){//Columns
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for(i=0;i<(int)k;i++){//Rows
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t=g * *w++;
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pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? -1 : 1) : 1);
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}
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}
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}
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else{
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for(j=0;j<l;j++){//Columns
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for(i=0;i<(int)k;i++){//Rows
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t=g * *w++;
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pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? 1 : -1) : 1);
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}
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}
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}
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return -1;
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}
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/******************************************************************************
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* IIR filter design
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******************************************************************************/
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/* Helper functions for the bilinear transform */
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/* Pre-warp the coefficients of a numerator or denominator.
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Note that a0 is assumed to be 1, so there is no wrapping
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of it.
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*/
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void af_filter_prewarp(_ftype_t* a, _ftype_t fc, _ftype_t fs)
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{
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_ftype_t wp;
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wp = 2.0 * fs * tan(M_PI * fc / fs);
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a[2] = a[2]/(wp * wp);
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a[1] = a[1]/wp;
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}
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/* Transform the numerator and denominator coefficients of s-domain
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biquad section into corresponding z-domain coefficients.
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The transfer function for z-domain is:
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1 + alpha1 * z^(-1) + alpha2 * z^(-2)
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H(z) = -------------------------------------
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1 + beta1 * z^(-1) + beta2 * z^(-2)
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Store the 4 IIR coefficients in array pointed by coef in following
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order:
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beta1, beta2 (denominator)
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alpha1, alpha2 (numerator)
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Arguments:
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a - s-domain numerator coefficients
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b - s-domain denominator coefficients
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k - filter gain factor. Initially set to 1 and modified by each
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biquad section in such a way, as to make it the
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coefficient by which to multiply the overall filter gain
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in order to achieve a desired overall filter gain,
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specified in initial value of k.
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fs - sampling rate (Hz)
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coef - array of z-domain coefficients to be filled in.
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Return: On return, set coef z-domain coefficients and k to the gain
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required to maintain overall gain = 1.0;
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*/
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void af_filter_bilinear(_ftype_t* a, _ftype_t* b, _ftype_t* k, _ftype_t fs, _ftype_t *coef)
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{
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_ftype_t ad, bd;
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/* alpha (Numerator in s-domain) */
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ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0];
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/* beta (Denominator in s-domain) */
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bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0];
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/* Update gain constant for this section */
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*k *= ad/bd;
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/* Denominator */
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*coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */
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*coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */
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/* Numerator */
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*coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */
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*coef = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad; /* alpha2 */
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}
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/* IIR filter design using bilinear transform and prewarp. Transforms
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2nd order s domain analog filter into a digital IIR biquad link. To
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create a filter fill in a, b, Q and fs and make space for coef and k.
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Example Butterworth design:
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Below are Butterworth polynomials, arranged as a series of 2nd
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order sections:
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Note: n is filter order.
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n Polynomials
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-------------------------------------------------------------------
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2 s^2 + 1.4142s + 1
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4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
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6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
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For n=4 we have following equation for the filter transfer function:
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1 1
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T(s) = --------------------------- * ----------------------------
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s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1
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The filter consists of two 2nd order sections since highest s power
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is 2. Now we can take the coefficients, or the numbers by which s
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is multiplied and plug them into a standard formula to be used by
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bilinear transform.
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Our standard form for each 2nd order section is:
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a2 * s^2 + a1 * s + a0
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H(s) = ----------------------
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b2 * s^2 + b1 * s + b0
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Note that Butterworth numerator is 1 for all filter sections, which
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means s^2 = 0 and s^1 = 0
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Let's convert standard Butterworth polynomials into this form:
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0 + 0 + 1 0 + 0 + 1
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--------------------------- * --------------------------
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1 + ((1/Q) * 0.765367) + 1 1 + ((1/Q) * 1.847759) + 1
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Section 1:
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a2 = 0; a1 = 0; a0 = 1;
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b2 = 1; b1 = 0.765367; b0 = 1;
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Section 2:
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a2 = 0; a1 = 0; a0 = 1;
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b2 = 1; b1 = 1.847759; b0 = 1;
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Q is filter quality factor or resonance, in the range of 1 to
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1000. The overall filter Q is a product of all 2nd order stages.
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For example, the 6th order filter (3 stages, or biquads) with
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individual Q of 2 will have filter Q = 2 * 2 * 2 = 8.
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Arguments:
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a - s-domain numerator coefficients, a[1] is always assumed to be 1.0
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b - s-domain denominator coefficients
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Q - Q value for the filter
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k - filter gain factor. Initially set to 1 and modified by each
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biquad section in such a way, as to make it the
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coefficient by which to multiply the overall filter gain
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in order to achieve a desired overall filter gain,
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specified in initial value of k.
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fs - sampling rate (Hz)
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coef - array of z-domain coefficients to be filled in.
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Note: Upon return from each call, the k argument will be set to a
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value, by which to multiply our actual signal in order for the gain
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to be one. On second call to szxform() we provide k that was
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changed by the previous section. During actual audio filtering
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k can be used for gain compensation.
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return -1 if fail 0 if success.
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*/
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int af_filter_szxform(_ftype_t* a, _ftype_t* b, _ftype_t Q, _ftype_t fc, _ftype_t fs, _ftype_t *k, _ftype_t *coef)
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{
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_ftype_t at[3];
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_ftype_t bt[3];
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if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
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return -1;
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memcpy(at,a,3*sizeof(_ftype_t));
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memcpy(bt,b,3*sizeof(_ftype_t));
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bt[1]/=Q;
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/* Calculate a and b and overwrite the original values */
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af_filter_prewarp(at, fc, fs);
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af_filter_prewarp(bt, fc, fs);
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/* Execute bilinear transform */
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af_filter_bilinear(at, bt, k, fs, coef);
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return 0;
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}
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