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/* Normalizer plugin
*
* Limitations:
* - only AFMT_S16_LE supported
* - no parameters yet => tweak the values by editing the #defines
*
* License: GPLv2
* Author: pl <p_l@gmx.fr> (c) 2002 and beyond...
*
* Sources: some ideas from volnorm for xmms
*
* */
#define PLUGIN
#include <stdio.h>
#include <stdlib.h>
#include <inttypes.h>
#include <math.h> // for sqrt()
#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
static ao_info_t info = {
"Volume normalizer",
"volnorm",
"pl <p_l@gmx.fr>",
""
};
LIBAO_PLUGIN_EXTERN(volnorm)
// mul is the value by which the samples are scaled
// and has to be in [MUL_MIN, MUL_MAX]
#define MUL_INIT 1.0
#define MUL_MIN 0.1
#define MUL_MAX 15.0
static float mul;
// "history" value of the filter
static float lastavg;
// SMOOTH_* must be in ]0.0, 1.0[
// The new value accounts for SMOOTH_MUL in the value and history
#define SMOOTH_MUL 0.06
#define SMOOTH_LASTAVG 0.06
// ideal average level
#define MID_S16 (INT16_MAX * 0.25)
// silence level
#define SIL_S16 (INT16_MAX * 0.02)
// local data
static struct {
int inuse; // This plugin is in use TRUE, FALSE
int format; // sample fomat
} pl_volnorm = {0, 0};
// minimal interface
static int control(int cmd,int arg){
switch(cmd){
case AOCONTROL_PLUGIN_SET_LEN:
return CONTROL_OK;
}
return CONTROL_UNKNOWN;
}
// minimal interface
// open & setup audio device
// return: 1=success 0=fail
static int init(){
switch(ao_plugin_data.format){
case(AFMT_S16_LE):
break;
default:
fprintf(stderr,"[pl_volnorm] Audio format not yet supported.\n");
return 0;
}
pl_volnorm.format = ao_plugin_data.format;
pl_volnorm.inuse = 1;
reset();
printf("[pl_volnorm] Normalizer plugin in use.\n");
return 1;
}
// close plugin
static void uninit(){
pl_volnorm.inuse=0;
}
// empty buffers
static void reset(){
mul = MUL_INIT;
switch(ao_plugin_data.format) {
case(AFMT_S16_LE):
lastavg = MID_S16;
break;
default:
fprintf(stderr,"[pl_volnorm] internal inconsistency - please bugreport.\n");
*(char *) 0 = 0;
}
}
// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
switch(pl_volnorm.format){
case(AFMT_S16_LE): {
#define CLAMP(x,m,M) do { if ((x)<(m)) (x) = (m); else if ((x)>(M)) (x) = (M); } while(0)
int16_t* data=(int16_t*)ao_plugin_data.data;
int len=ao_plugin_data.len / 2; // 16 bits samples
int32_t i;
register int32_t tmp;
register float curavg;
float newavg;
float neededmul;
// average of the current samples
curavg = 0.0;
for (i = 0; i < len ; ++i) {
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
if (curavg > SIL_S16) {
neededmul = MID_S16 / ( curavg * mul);
mul = (1.0 - SMOOTH_MUL) * mul + SMOOTH_MUL * neededmul;
// Clamp the mul coefficient
CLAMP(mul, MUL_MIN, MUL_MAX);
}
// Scale & clamp the samples
for (i=0; i < len ; ++i) {
tmp = data[i] * mul;
CLAMP(tmp, INT16_MIN, INT16_MAX);
data[i] = tmp;
}
// Evaluation of newavg (not 100% accurate because of values clamping)
newavg = mul * curavg;
#if 0
printf("time = %d len = %d curavg = %6.0f lastavg = %6.0f newavg = %6.0f\n"
" needed_m = %2.2f m = %2.2f\n\n",
time(NULL), len, curavg, lastavg, newavg, neededmul, mul);
#endif
lastavg = (1.0 - SMOOTH_LASTAVG) * lastavg + SMOOTH_LASTAVG * newavg;
break;
}
default:
return 0;
}
return 1;
}
|
