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