mplayer.hg: libaf/af_equalizer.c comparison

comparison libaf/af_equalizer.c @ 8607:d6f40a06867b

Changes includes: - Improved runtime control system - 3 New filter panning, compressor/limiter and a noise gate - The compressor/limiter and the noise gate are not yet finished - The panning filter does combined mixing and channel routing and can be used to down-mix from stereo to mono (for example) - Improvements to volume and channel - volume now has a very good soft clipping using sin() - channel can handle generic routing of audio data - Conversion of all filters to handle floating point data - Cleanup of message printing - Fix for the sig 11 bug reported by Denes

author	anders
date	Sat, 28 Dec 2002 13:59:53 +0000
parents	e8832e66babd
children	14090f7300a8

comparison

equal deleted inserted replaced

-:d80edba39db9
+:d6f40a06867b
 #include <unistd.h>
 #include <inttypes.h>
 #include <math.h>
 #include "af.h"
-#include "equalizer.h"
-#define NCH	AF_NCH // 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
+/* 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   a[KM][L];        	// A weights
-float   b[KM][L];        // B weights
+float   b[KM][L];	     	// B weights
-float   wq[NCH][KM][L];  // Circular buffer for W data
+float   wq[AF_NCH][KM][L];  	// Circular buffer for W data
-float   g[NCH][KM];      // Gain factor for each channel and band
+float   g[AF_NCH][KM];      	// Gain factor for each channel and band
-int     K; 		   // Number of used eq bands
+int     K; 		   	// Number of used eq bands
-int     channels;        // Number of channels
+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;
 // 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_NE | AF_FORMAT_SI;
+af->data->format = AF_FORMAT_NE | AF_FORMAT_F;
-af->data->bps    = 2;
+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)
-af_msg(AF_MSG_INFO,"Limiting the number of filters to %i due to low sample rate.\n",s->K);
+af_msg(AF_MSG_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 += 2000.0/((float)af->data->rate);
-// Only signed 16 bit little endian is supported
+return af_test_output(af,arg);
-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(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_SET_GAIN:{
+case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
-float gain = ((equalizer_t*)arg)->gain;
+float* gain = ((af_control_ext_t*)arg)->arg;
-int   ch   = ((equalizer_t*)arg)->channel;
+int    ch   = ((af_control_ext_t*)arg)->ch;
-int   band = ((equalizer_t*)arg)->band;
+int    k;
-if(ch > NCH || ch < 0 || band > KM || band < 0)
+if(ch > AF_NCH || ch < 0)
 return AF_ERROR;
-s->g[ch][band] = pow(10.0,clamp(gain,G_MIN,G_MAX)/20.0)-1.0;
+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_GET_GAIN:{
+case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
-int ch     =((equalizer_t*)arg)->channel;
+float* gain = ((af_control_ext_t*)arg)->arg;
-int band   =((equalizer_t*)arg)->band;
+int    ch   = ((af_control_ext_t*)arg)->ch;
-if(ch > NCH || ch < 0 || band > KM || band < 0)
+int    k;
+if(ch > AF_NCH || ch < 0)
 return AF_ERROR;
-((equalizer_t*)arg)->gain = log10(s->g[ch][band]+1.0) * 20.0;
+for(k = 0 ; k<KM ; k++)
+gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
 return AF_OK;
 }
 }
 return AF_UNKNOWN;
 }
 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;
+float*	in  = ((float*)c->audio)+ci;
-int16_t*	out = ((int16_t*)c->audio)+ci;
+float*	out = ((float*)c->audio)+ci;
-int16_t* 	end = in + c->len/2; // Block loop end
+float* 	end = in + c->len/4; // Block loop end
 while(in < end){
-register uint32_t	k  = 0;   	   // Frequency band index
+register uint32_t	k  = 0;		// Frequency band index
-register float 	yt = (float)(*in); // Current input sample
+register float 	yt = *in; 	// Current input sample
 in+=nch;
 // Run the filters
 for(;k<s->K;k++){
 	// Pointer to circular buffer wq
 	// Update circular buffer
 	wq[1] = wq[0];
 	wq[0] = w;
 }
 // Calculate output
-*out=(int16_t)(yt/(4.0*10.0));
+*out=yt/(4.0*10.0);
 out+=nch;
 }
 }
 return c;
 }

Mercurial > mplayer.hg

comparison libaf/af_equalizer.c @ 8607:d6f40a06867b