view libaf/af_equalizer.c @ 30903:565f6d0f0d2a

Make all instances of codec_patch unconditional, otherwise compilation will break on systems that do not have win32 dlls enabled. Fixes compilation bug introduced by r30942 10l to the anonymous guy who explains the importance of commit messages and would like to have romance novels in these very messages.

/*
 * Equalizer filter, implementation of a 10 band time domain graphic
 * equalizer using IIR filters. The IIR filters are implemented using a
 * Direct Form II approach, but has been modified (b1 == 0 always) to
 * save computation.
 *
 * 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 <stdio.h>
#include <stdlib.h>

#include <inttypes.h>
#include <math.h>

#include "af.h"

#define L   	2      // Storage for filter taps
#define KM  	10     // Max number of bands

#define Q   1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
			 gives 4dB suppression @ Fc*2 and Fc/2 */

/* Center frequencies for band-pass filters
   The different frequency bands are:
   nr.    	center frequency
   0  	31.25 Hz
   1 	62.50 Hz
   2	125.0 Hz
   3	250.0 Hz
   4	500.0 Hz
   5	1.000 kHz
   6	2.000 kHz
   7	4.000 kHz
   8	8.000 kHz
   9 	16.00 kHz
*/
#define CF  	{31.25,62.5,125,250,500,1000,2000,4000,8000,16000}

// Maximum and minimum gain for the bands
#define G_MAX	+12.0
#define G_MIN	-12.0

// Data for specific instances of this filter
typedef struct af_equalizer_s
{
  float   a[KM][L];        	// A weights
  float   b[KM][L];	     	// B weights
  float   wq[AF_NCH][KM][L];  	// Circular buffer for W data
  float   g[AF_NCH][KM];      	// Gain factor for each channel and band
  int     K; 		   	// Number of used eq bands
  int     channels;        	// Number of channels
  float   gain_factor;     // applied at output to avoid clipping
} af_equalizer_t;

// 2nd order Band-pass Filter design
static void bp2(float* a, float* b, float fc, float q){
  double th= 2.0 * M_PI * fc;
  double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));

  a[0] = (1.0 + C) * cos(th);
  a[1] = -1 * C;

  b[0] = (1.0 - C)/2.0;
  b[1] = -1.0050;
}

// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
  af_equalizer_t* s   = (af_equalizer_t*)af->setup;

  switch(cmd){
  case AF_CONTROL_REINIT:{
    int k =0, i =0;
    float F[KM] = CF;

    s->gain_factor=0.0;

    // Sanity check
    if(!arg) return AF_ERROR;

    af->data->rate   = ((af_data_t*)arg)->rate;
    af->data->nch    = ((af_data_t*)arg)->nch;
    af->data->format = AF_FORMAT_FLOAT_NE;
    af->data->bps    = 4;

    // Calculate number of active filters
    s->K=KM;
    while(F[s->K-1] > (float)af->data->rate/2.2)
      s->K--;

    if(s->K != KM)
      mp_msg(MSGT_AFILTER, MSGL_INFO, "[equalizer] Limiting the number of filters to"
	     " %i due to low sample rate.\n",s->K);

    // Generate filter taps
    for(k=0;k<s->K;k++)
      bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);

    // Calculate how much this plugin adds to the overall time delay
    af->delay = 2 * af->data->nch * af->data->bps;

    // Calculate gain factor to prevent clipping at output
    for(k=0;k<AF_NCH;k++)
    {
        for(i=0;i<KM;i++)
        {
            if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
        }
    }

    s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;

    if(s->gain_factor > 0.0)
    {
        s->gain_factor=0.1+(s->gain_factor/12.0);
    }else{
        s->gain_factor=1;
    }

    return af_test_output(af,arg);
  }
  case AF_CONTROL_COMMAND_LINE:{
    float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0};
    int i,j;
    sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1],
	   &g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]);
    for(i=0;i<AF_NCH;i++){
      for(j=0;j<KM;j++){
	((af_equalizer_t*)af->setup)->g[i][j] =
	  pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
      }
    }
    return AF_OK;
  }
  case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
    float* gain = ((af_control_ext_t*)arg)->arg;
    int    ch   = ((af_control_ext_t*)arg)->ch;
    int    k;
    if(ch >= AF_NCH || ch < 0)
      return AF_ERROR;

    for(k = 0 ; k<KM ; k++)
      s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;

    return AF_OK;
  }
  case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
    float* gain = ((af_control_ext_t*)arg)->arg;
    int    ch   = ((af_control_ext_t*)arg)->ch;
    int    k;
    if(ch >= AF_NCH || ch < 0)
      return AF_ERROR;

    for(k = 0 ; k<KM ; k++)
      gain[k] = log10(s->g[ch][k]+1.0) * 20.0;

    return AF_OK;
  }
  }
  return AF_UNKNOWN;
}

// Deallocate memory
static void uninit(struct af_instance_s* af)
{
  if(af->data)
    free(af->data);
  if(af->setup)
    free(af->setup);
}

// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
  af_data_t*       c 	= data;			    	// Current working data
  af_equalizer_t*  s 	= (af_equalizer_t*)af->setup; 	// Setup
  uint32_t  	   ci  	= af->data->nch; 	    	// Index for channels
  uint32_t	   nch 	= af->data->nch;   	    	// Number of channels

  while(ci--){
    float*	g   = s->g[ci];      // Gain factor
    float*	in  = ((float*)c->audio)+ci;
    float*	out = ((float*)c->audio)+ci;
    float* 	end = in + c->len/4; // Block loop end

    while(in < end){
      register int	k  = 0;		// Frequency band index
      register float 	yt = *in; 	// Current input sample
      in+=nch;

      // Run the filters
      for(;k<s->K;k++){
 	// Pointer to circular buffer wq
 	register float* wq = s->wq[ci][k];
 	// Calculate output from AR part of current filter
 	register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
 	// Calculate output form MA part of current filter
 	yt+=(w + wq[1]*s->b[k][1])*g[k];
 	// Update circular buffer
 	wq[1] = wq[0];
	wq[0] = w;
      }
      // Calculate output
      *out=yt*s->gain_factor;
      out+=nch;
    }
  }
  return c;
}

// Allocate memory and set function pointers
static int af_open(af_instance_t* af){
  af->control=control;
  af->uninit=uninit;
  af->play=play;
  af->mul=1;
  af->data=calloc(1,sizeof(af_data_t));
  af->setup=calloc(1,sizeof(af_equalizer_t));
  if(af->data == NULL || af->setup == NULL)
    return AF_ERROR;
  return AF_OK;
}

// Description of this filter
af_info_t af_info_equalizer = {
  "Equalizer audio filter",
  "equalizer",
  "Anders",
  "",
  AF_FLAGS_NOT_REENTRANT,
  af_open
};