mirror of
https://github.com/mpv-player/mpv
synced 2024-11-11 00:15:33 +01:00
7deec05ea0
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
176 lines
4.1 KiB
C
176 lines
4.1 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 2006 Michael Niedermayer
|
|
// Copyright 2004 Alex Beregszaszi & Pierre Lombard (original af_extrastereo.c upon which this is based)
|
|
//
|
|
//=============================================================================
|
|
*/
|
|
|
|
#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_sinesuppress_s
|
|
{
|
|
double freq;
|
|
double decay;
|
|
double real;
|
|
double imag;
|
|
double ref;
|
|
double pos;
|
|
}af_sinesuppress_t;
|
|
|
|
static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data);
|
|
//static af_data_t* play_float(struct af_instance_s* af, af_data_t* data);
|
|
|
|
// Initialization and runtime control
|
|
static int control(struct af_instance_s* af, int cmd, void* arg)
|
|
{
|
|
af_sinesuppress_t* s = (af_sinesuppress_t*)af->setup;
|
|
|
|
switch(cmd){
|
|
case AF_CONTROL_REINIT:{
|
|
// Sanity check
|
|
if(!arg) return AF_ERROR;
|
|
|
|
af->data->rate = ((af_data_t*)arg)->rate;
|
|
af->data->nch = 1;
|
|
#if 0
|
|
if (((af_data_t*)arg)->format == AF_FORMAT_FLOAT_NE)
|
|
{
|
|
af->data->format = AF_FORMAT_FLOAT_NE;
|
|
af->data->bps = 4;
|
|
af->play = play_float;
|
|
}// else
|
|
#endif
|
|
{
|
|
af->data->format = AF_FORMAT_S16_NE;
|
|
af->data->bps = 2;
|
|
af->play = play_s16;
|
|
}
|
|
|
|
return af_test_output(af,(af_data_t*)arg);
|
|
}
|
|
case AF_CONTROL_COMMAND_LINE:{
|
|
float f1,f2;
|
|
sscanf((char*)arg,"%f:%f", &f1,&f2);
|
|
s->freq = f1;
|
|
s->decay = f2;
|
|
return AF_OK;
|
|
}
|
|
case AF_CONTROL_SS_FREQ | AF_CONTROL_SET:
|
|
s->freq = *(float*)arg;
|
|
return AF_OK;
|
|
case AF_CONTROL_SS_FREQ | AF_CONTROL_GET:
|
|
*(float*)arg = s->freq;
|
|
return AF_OK;
|
|
case AF_CONTROL_SS_DECAY | AF_CONTROL_SET:
|
|
s->decay = *(float*)arg;
|
|
return AF_OK;
|
|
case AF_CONTROL_SS_DECAY | AF_CONTROL_GET:
|
|
*(float*)arg = s->decay;
|
|
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_s16(struct af_instance_s* af, af_data_t* data)
|
|
{
|
|
af_sinesuppress_t *s = af->setup;
|
|
register int i = 0;
|
|
int16_t *a = (int16_t*)data->audio; // Audio data
|
|
int len = data->len/2; // Number of samples
|
|
|
|
for (i = 0; i < len; i++)
|
|
{
|
|
double co= cos(s->pos);
|
|
double si= sin(s->pos);
|
|
|
|
s->real += co * a[i];
|
|
s->imag += si * a[i];
|
|
s->ref += co * co;
|
|
|
|
a[i] -= (s->real * co + s->imag * si) / s->ref;
|
|
|
|
s->real -= s->real * s->decay;
|
|
s->imag -= s->imag * s->decay;
|
|
s->ref -= s->ref * s->decay;
|
|
|
|
s->pos += 2 * M_PI * s->freq / data->rate;
|
|
}
|
|
|
|
af_msg(AF_MSG_VERBOSE,"[sinesuppress] f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref);
|
|
|
|
return data;
|
|
}
|
|
|
|
#if 0
|
|
static af_data_t* play_float(struct af_instance_s* af, af_data_t* data)
|
|
{
|
|
af_sinesuppress_t *s = af->setup;
|
|
register int i = 0;
|
|
float *a = (float*)data->audio; // Audio data
|
|
int len = data->len/4; // Number of samples
|
|
float avg, l, r;
|
|
|
|
for (i = 0; i < len; i+=2)
|
|
{
|
|
avg = (a[i] + a[i + 1]) / 2;
|
|
|
|
/* l = avg + (s->mul * (a[i] - avg));
|
|
r = avg + (s->mul * (a[i + 1] - avg));*/
|
|
|
|
a[i] = af_softclip(l);
|
|
a[i + 1] = af_softclip(r);
|
|
}
|
|
|
|
return data;
|
|
}
|
|
#endif
|
|
|
|
// Allocate memory and set function pointers
|
|
static int af_open(af_instance_t* af){
|
|
af->control=control;
|
|
af->uninit=uninit;
|
|
af->play=play_s16;
|
|
af->mul=1;
|
|
af->data=calloc(1,sizeof(af_data_t));
|
|
af->setup=calloc(1,sizeof(af_sinesuppress_t));
|
|
if(af->data == NULL || af->setup == NULL)
|
|
return AF_ERROR;
|
|
|
|
((af_sinesuppress_t*)af->setup)->freq = 50.0;
|
|
((af_sinesuppress_t*)af->setup)->decay = 0.0001;
|
|
return AF_OK;
|
|
}
|
|
|
|
// Description of this filter
|
|
af_info_t af_info_sinesuppress = {
|
|
"Sine Suppress",
|
|
"sinesuppress",
|
|
"Michael Niedermayer",
|
|
"",
|
|
0,
|
|
af_open
|
|
};
|