diff libaf/af_equalizer.c @ 8073:c0e556f9986b

Adding equalizer filter + some cosmetics

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/libaf/af_equalizer.c	Sun Nov 03 09:51:02 2002 +0000
@@ -0,0 +1,217 @@
+/*=============================================================================
+//	
+//  This software has been released under the terms of the GNU Public
+//  license. See http://www.gnu.org/copyleft/gpl.html for details.
+//
+//  Copyright 2001 Anders Johansson [e-mail]
+//
+//=============================================================================
+*/
+
+/* 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.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <unistd.h>
+#include <inttypes.h>
+#include <math.h>
+
+#include "../config.h"
+#include "../mp_msg.h"
+#include "../libao2/afmt.h"
+
+#include "af.h"
+#include "equalizer.h"
+
+#define NCH 	6     // Max number of channels
+#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
+#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[NCH][KM][L];  // Circular buffer for W data
+  float   g[NCH][KM];      // Gain factor for each channel and band
+  int     K; 		   // Number of used eq bands
+  int     channels;        // Number of channels
+} 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;
+    float F[KM] = CF;
+    
+    // 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 = AFMT_S16_LE;
+    af->data->bps    = 2;
+    
+    // Calculate number of active filters
+    s->K=KM;
+    while(F[s->K-1] > (float)af->data->rate/2.0)
+      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 += 2000.0/((float)af->data->rate);
+
+    // Only AFMT_S16_LE is supported 
+    if(af->data->format != ((af_data_t*)arg)->format || 
+       af->data->bps != ((af_data_t*)arg)->bps)
+      return AF_FALSE;
+    return AF_OK;
+  }
+  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<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_SET_GAIN:{
+    float gain = ((equalizer_t*)arg)->gain;
+    int   ch   = ((equalizer_t*)arg)->channel;
+    int   band = ((equalizer_t*)arg)->band;
+    if(ch > NCH || ch < 0 || band > KM || band < 0)
+      return AF_ERROR;
+    
+    s->g[ch][band] = pow(10.0,clamp(gain,G_MIN,G_MAX)/20.0)-1.0;
+    return AF_OK;
+  }
+  case AF_CONTROL_EQUALIZER_GET_GAIN:{
+    int ch     =((equalizer_t*)arg)->channel;
+    int band   =((equalizer_t*)arg)->band;
+    if(ch > NCH || ch < 0 || band > KM || band < 0)
+      return AF_ERROR;
+    
+    ((equalizer_t*)arg)->gain = log10(s->g[ch][band]+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 
+    int16_t*	in  = ((int16_t*)c->audio)+ci;
+    int16_t*	out = ((int16_t*)c->audio)+ci;
+    int16_t* 	end = in + c->len/2; // Block loop end
+
+    while(in < end){
+      register uint32_t	k  = 0;   	   // Frequency band index
+      register float 	yt = (float)(*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=(int16_t)(yt/(4.0*10.0));
+      out+=nch;
+    }
+  }
+  return c;
+}
+
+// Allocate memory and set function pointers
+static int open(af_instance_t* af){
+  af->control=control;
+  af->uninit=uninit;
+  af->play=play;
+  af->mul.n=1;
+  af->mul.d=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,
+  open
+};
+
+
+
+
+
+
+