view libaf/filter.c @ 29794:df1826dcdb2d

Disable audio when initializing the filter chain fails (can happen e.g. when the hwmpa decoder is used but the hardware does not support hardware MPEG audio). Otherwise this will lead to a crash later on when the decode code tries to access the audio filter chain.

/*
 * design and implementation of different types of digital filters
 *
 * Copyright (C) 2001 Anders Johansson [e-mail]
 *
 * This file is part of MPlayer.
 *
 * MPlayer 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.
 *
 * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#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;
}

/* C implementation of parallel FIR filter y(k)=w(k) * x(k) (where * denotes convolution)

   n  number of filter taps, where mod(n,4)==0
   d  number of filters
   xi current index in xq
   w  filter taps k by n big
   x  input signal must be a circular buffers which are indexed backwards
   y  output buffer
   s  output buffer stride
*/
FLOAT_TYPE* af_filter_pfir(unsigned int n, unsigned int d, unsigned int xi,
                           const FLOAT_TYPE** w, const FLOAT_TYPE** x, FLOAT_TYPE* y,
                           unsigned int s)
{
  register const FLOAT_TYPE* xt = *x + xi;
  register const FLOAT_TYPE* wt = *w;
  register int    nt = 2*n;
  while(d-- > 0){
    *y = af_filter_fir(n,wt,xt);
    wt+=n;
    xt+=nt;
    y+=s;
  }
  return y;
}

/* Add new data to circular queue designed to be used with a parallel
   FIR filter, with d filters. xq is the circular queue, in pointing
   at the new samples, xi current index in xq and n the length of the
   filter. xq must be n*2 by k big, s is the index for in.
*/
int af_filter_updatepq(unsigned int n, unsigned int d, unsigned int xi,
                       FLOAT_TYPE** xq, const FLOAT_TYPE* in, unsigned int s)
{
  register FLOAT_TYPE* txq = *xq + xi;
  register int nt = n*2;

  while(d-- >0){
    *txq= *(txq+n) = *in;
    txq+=nt;
    in+=s;
  }
  return (++xi)&(n-1);
}

/******************************************************************************
*  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;
}

/* Design polyphase FIR filter from prototype filter

   n     length of prototype filter
   k     number of polyphase components
   w     prototype filter taps
   pw    Parallel FIR filter
   g     Filter gain
   flags FWD forward indexing
         REW reverse indexing
	 ODD multiply every 2nd filter tap by -1 => HP filter

   returns 0 if OK, -1 if fail
*/
int af_filter_design_pfir(unsigned int n, unsigned int k, const FLOAT_TYPE* w,
                          FLOAT_TYPE** pw, FLOAT_TYPE g, unsigned int flags)
{
  int l = (int)n/k;	// Length of individual FIR filters
  int i;     	// Counters
  int j;
  FLOAT_TYPE t;	// g * w[i]

  // Sanity check
  if(l<1 || k<1 || !w || !pw)
    return -1;

  // Do the stuff
  if(flags&REW){
    for(j=l-1;j>-1;j--){//Columns
      for(i=0;i<(int)k;i++){//Rows
	t=g *  *w++;
	pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? -1 : 1) : 1);
      }
    }
  }
  else{
    for(j=0;j<l;j++){//Columns
      for(i=0;i<(int)k;i++){//Rows
	t=g *  *w++;
	pw[i][j]=t * ((flags & ODD) ? ((j & 1) ? 1 : -1) : 1);
      }
    }
  }
  return -1;
}

/******************************************************************************
*  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;
}