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mpv/audio/filter/tools.c
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2014-04-13 18:03:01 +02:00

124 lines
3.2 KiB
C

/*
* 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 <math.h>
#include <string.h>
#include "common/common.h"
#include "af.h"
/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
* fail. input <= -200dB will become 0 gain. */
int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
{
int i = 0;
// Sanity check
if(!in || !out)
return AF_ERROR;
for(i=0;i<n;i++){
if(in[i]<=-200)
out[i]=0.0;
else
out[i]=pow(10.0,MPCLAMP(in[i],mi,ma)/k);
}
return AF_OK;
}
/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
* fail. gain=0 will become -200 dB. k is just a multiplier. */
int af_to_dB(int n, float* in, float* out, float k)
{
int i = 0;
// Sanity check
if(!in || !out)
return AF_ERROR;
for(i=0;i<n;i++){
if(in[i] == 0.0)
out[i]=-200.0;
else
out[i]=k*log10(in[i]);
}
return AF_OK;
}
/* Convert from ms to sample time */
int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
{
int i = 0;
// Sanity check
if(!in || !out)
return AF_ERROR;
for(i=0;i<n;i++)
out[i]=(int)((float)rate * MPCLAMP(in[i],mi,ma)/1000.0);
return AF_OK;
}
/* Convert from sample time to ms */
int af_to_ms(int n, int* in, float* out, int rate)
{
int i = 0;
// Sanity check
if(!in || !out || !rate)
return AF_ERROR;
for(i=0;i<n;i++)
out[i]=1000.0 * (float)in[i]/((float)rate);
return AF_OK;
}
/*
* test if output format matches
* af: audio filter
* out: needed format, will be overwritten by available
* format if they do not match
* returns: AF_FALSE if formats do not match, AF_OK if they match
*
* compares the format, rate and nch values of af->data with out
* Note: logically, *out=*af->data always happens, because out contains the
* format only, no actual audio data or memory allocations. *out always
* contains the parameters from af->data after the function returns.
*/
int af_test_output(struct af_instance* af, struct mp_audio* out)
{
if((af->data->format != out->format) ||
(af->data->bps != out->bps) ||
(af->data->rate != out->rate) ||
!mp_chmap_equals(&af->data->channels, &out->channels)){
*out = *af->data;
return AF_FALSE;
}
return AF_OK;
}
/* Soft clipping, the sound of a dream, thanks to Jon Wattes
post to Musicdsp.org */
float af_softclip(float a)
{
if (a >= M_PI/2)
return 1.0;
else if (a <= -M_PI/2)
return -1.0;
else
return sin(a);
}