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/*=============================================================================
//
// 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
//
//=============================================================================
*/
/* This audio filter changes the volume of the sound, and can be used
when the mixer doesn't support the PCM channel. It can handle
between 1 and 6 channels. The volume can be adjusted between -60dB
to +20dB and is set on a per channels basis. The is accessed through
AF_CONTROL_VOLUME_LEVEL.
The filter has support for soft-clipping, it is enabled by
AF_CONTROL_VOLUME_SOFTCLIPP. It has also a probing feature which
can be used to measure the power in the audio stream, both an
instantaneous value and the maximum value can be probed. The
probing is enable by AF_CONTROL_VOLUME_PROBE_ON_OFF and is done on a
per channel basis. The result from the probing is obtained using
AF_CONTROL_VOLUME_PROBE_GET and AF_CONTROL_VOLUME_PROBE_GET_MAX. The
probed values are calculated in dB.
*/
#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_volume_s
{
int enable[AF_NCH]; // Enable/disable / channel
float pow[AF_NCH]; // Estimated power level [dB]
float max[AF_NCH]; // Max Power level [dB]
float level[AF_NCH]; // Gain level for each channel
float time; // Forgetting factor for power estimate
int soft; // Enable/disable soft clipping
int fast; // Use fix-point volume control
}af_volume_t;
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_volume_t* s = (af_volume_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 = ((af_data_t*)arg)->nch;
if(s->fast && (((af_data_t*)arg)->format != (AF_FORMAT_FLOAT_NE))){
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
}
else{
// Cutoff set to 10Hz for forgetting factor
float x = 2.0*M_PI*15.0/(float)af->data->rate;
float t = 2.0-cos(x);
s->time = 1.0 - (t - sqrt(t*t - 1));
af_msg(AF_MSG_DEBUG0,"[volume] Forgetting factor = %0.5f\n",s->time);
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
}
return af_test_output(af,(af_data_t*)arg);
case AF_CONTROL_COMMAND_LINE:{
float v=0.0;
float vol[AF_NCH];
int i;
sscanf((char*)arg,"%f:%i", &v, &s->soft);
for(i=0;i<AF_NCH;i++) vol[i]=v;
return control(af,AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET, vol);
}
case AF_CONTROL_POST_CREATE:
s->fast = ((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) ==
AF_INIT_FLOAT) ? 0 : 1;
return AF_OK;
case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_SET:
memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int));
return AF_OK;
case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_GET:
memcpy((int*)arg,s->enable,AF_NCH*sizeof(int));
return AF_OK;
case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_SET:
s->soft = *(int*)arg;
return AF_OK;
case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_GET:
*(int*)arg = s->soft;
return AF_OK;
case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET:
return af_from_dB(AF_NCH,(float*)arg,s->level,20.0,-200.0,60.0);
case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_GET:
return af_to_dB(AF_NCH,s->level,(float*)arg,20.0);
case AF_CONTROL_VOLUME_PROBE | AF_CONTROL_GET:
return af_to_dB(AF_NCH,s->pow,(float*)arg,10.0);
case AF_CONTROL_VOLUME_PROBE_MAX | AF_CONTROL_GET:
return af_to_dB(AF_NCH,s->max,(float*)arg,10.0);
case AF_CONTROL_PRE_DESTROY:{
float m = 0.0;
int i;
if(!s->fast){
for(i=0;i<AF_NCH;i++)
m=max(m,s->max[i]);
af_to_dB(1, &m, &m, 10.0);
af_msg(AF_MSG_INFO,"[volume] The maximum volume was %0.2fdB \n", m);
}
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_volume_t* s = (af_volume_t*)af->setup; // Setup for this instance
int ch = 0; // Channel counter
register int nch = c->nch; // Number of channels
register int i = 0;
// Basic operation volume control only (used on slow machines)
if(af->data->format == (AF_FORMAT_S16_NE)){
int16_t* a = (int16_t*)c->audio; // Audio data
int len = c->len/2; // Number of samples
for(ch = 0; ch < nch ; ch++){
if(s->enable[ch]){
register int vol = (int)(255.0 * s->level[ch]);
for(i=ch;i<len;i+=nch){
register int x = (a[i] * vol) >> 8;
a[i]=clamp(x,SHRT_MIN,SHRT_MAX);
}
}
}
}
// Machine is fast and data is floating point
else if(af->data->format == (AF_FORMAT_FLOAT_NE)){
float* a = (float*)c->audio; // Audio data
int len = c->len/4; // Number of samples
for(ch = 0; ch < nch ; ch++){
// Volume control (fader)
if(s->enable[ch]){
float t = 1.0 - s->time;
for(i=ch;i<len;i+=nch){
register float x = a[i];
register float pow = x*x;
// Check maximum power value
if(pow > s->max[ch])
s->max[ch] = pow;
// Set volume
x *= s->level[ch];
// Peak meter
pow = x*x;
if(pow > s->pow[ch])
s->pow[ch] = pow;
else
s->pow[ch] = t*s->pow[ch] + pow*s->time; // LP filter
/* Soft clipping, the sound of a dream, thanks to Jon Wattes
post to Musicdsp.org */
if(s->soft)
x=af_softclip(x);
// Hard clipping
else
x=clamp(x,-1.0,1.0);
a[i] = x;
}
}
}
}
return c;
}
// Allocate memory and set function pointers
static int af_open(af_instance_t* af){
int i = 0;
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_volume_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
// Enable volume control and set initial volume to 0dB.
for(i=0;i<AF_NCH;i++){
((af_volume_t*)af->setup)->enable[i] = 1;
((af_volume_t*)af->setup)->level[i] = 1.0;
}
return AF_OK;
}
// Description of this filter
af_info_t af_info_volume = {
"Volume control audio filter",
"volume",
"Anders",
"",
AF_FLAGS_NOT_REENTRANT,
af_open
};
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