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mpv/libaf/af_comp.c
uau 7deec05ea0 libaf: change filter input/output ratio calculations
Change the audio filters to use a double instead of rationals for the
ratio of output to input size. The rationals could overflow when
calculating the overall ratio of a filter chain and gave no real
advantage compared to doubles.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24916 b3059339-0415-0410-9bf9-f77b7e298cf2
2007-11-01 06:52:01 +00:00

161 lines
4.6 KiB
C

/*=============================================================================
//
// This software has been released under the terms of the GNU General Public
// license. See http://www.gnu.org/copyleft/gpl.html for details.
//
// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
//
//=============================================================================
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <inttypes.h>
#include <math.h>
#include <limits.h>
#include "af.h"
// Data for specific instances of this filter
typedef struct af_comp_s
{
int enable[AF_NCH]; // Enable/disable / channel
float time[AF_NCH]; // Forgetting factor for power estimate
float pow[AF_NCH]; // Estimated power level [dB]
float tresh[AF_NCH]; // Threshold [dB]
int attack[AF_NCH]; // Attack time [ms]
int release[AF_NCH]; // Release time [ms]
float ratio[AF_NCH]; // Compression ratio
}af_comp_t;
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_comp_t* s = (af_comp_t*)af->setup;
int i;
switch(cmd){
case AF_CONTROL_REINIT:
// 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;
// Time constant set to 0.1s
// s->alpha = (1.0/0.2)/(2.0*M_PI*(float)((af_data_t*)arg)->rate);
return af_test_output(af,(af_data_t*)arg);
case AF_CONTROL_COMMAND_LINE:{
/* float v=-10.0; */
/* float vol[AF_NCH]; */
/* float s=0.0; */
/* float clipp[AF_NCH]; */
/* int i; */
/* sscanf((char*)arg,"%f:%f", &v, &s); */
/* for(i=0;i<AF_NCH;i++){ */
/* vol[i]=v; */
/* clipp[i]=s; */
/* } */
/* if(AF_OK != control(af,AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_SET, clipp)) */
/* return AF_ERROR; */
/* return control(af,AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET, vol); */
}
case AF_CONTROL_COMP_ON_OFF | AF_CONTROL_SET:
memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int));
return AF_OK;
case AF_CONTROL_COMP_ON_OFF | AF_CONTROL_GET:
memcpy((int*)arg,s->enable,AF_NCH*sizeof(int));
return AF_OK;
case AF_CONTROL_COMP_THRESH | AF_CONTROL_SET:
return af_from_dB(AF_NCH,(float*)arg,s->tresh,20.0,-60.0,-1.0);
case AF_CONTROL_COMP_THRESH | AF_CONTROL_GET:
return af_to_dB(AF_NCH,s->tresh,(float*)arg,10.0);
case AF_CONTROL_COMP_ATTACK | AF_CONTROL_SET:
return af_from_ms(AF_NCH,(float*)arg,s->attack,af->data->rate,500.0,0.1);
case AF_CONTROL_COMP_ATTACK | AF_CONTROL_GET:
return af_to_ms(AF_NCH,s->attack,(float*)arg,af->data->rate);
case AF_CONTROL_COMP_RELEASE | AF_CONTROL_SET:
return af_from_ms(AF_NCH,(float*)arg,s->release,af->data->rate,3000.0,10.0);
case AF_CONTROL_COMP_RELEASE | AF_CONTROL_GET:
return af_to_ms(AF_NCH,s->release,(float*)arg,af->data->rate);
case AF_CONTROL_COMP_RATIO | AF_CONTROL_SET:
for(i=0;i<AF_NCH;i++)
s->ratio[i] = clamp(((float*)arg)[i],1.0,10.0);
return AF_OK;
case AF_CONTROL_COMP_RATIO | AF_CONTROL_GET:
for(i=0;i<AF_NCH;i++)
((float*)arg)[i] = s->ratio[i];
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_comp_t* s = (af_comp_t*)af->setup; // Setup for this instance
float* a = (float*)c->audio; // Audio data
int len = c->len/4; // Number of samples
int ch = 0; // Channel counter
register int nch = c->nch; // Number of channels
register int i = 0;
// Compress/expand
for(ch = 0; ch < nch ; ch++){
if(s->enable[ch]){
float t = 1.0 - s->time[ch];
for(i=ch;i<len;i+=nch){
register float x = a[i];
register float pow = x*x;
s->pow[ch] = t*s->pow[ch] +
pow*s->time[ch]; // LP filter
if(pow < s->pow[ch]){
;
}
else{
;
}
a[i] = x;
}
}
}
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_comp_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
return AF_OK;
}
// Description of this filter
af_info_t af_info_comp = {
"Compressor/expander audio filter",
"comp",
"Anders",
"",
AF_FLAGS_NOT_REENTRANT,
af_open
};