mpv/libaf/af_volnorm.c

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/*
* Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
*
* 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 <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <inttypes.h>
#include <math.h>
#include <limits.h>
#include "af.h"
// Methods:
// 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1)
// 2: uses several samples to smooth the variations (standard weighted mean
// on past samples)
// Size of the memory array
// FIXME: should depend on the frequency of the data (should be a few seconds)
#define NSAMPLES 128
// If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we
// choose to ignore the computed value as it's not significant enough
// FIXME: should depend on the frequency of the data (0.5s maybe)
#define MIN_SAMPLE_SIZE 32000
// 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 5.0
// Silence level
// FIXME: should be relative to the level of the samples
#define SIL_S16 (SHRT_MAX * 0.01)
#define SIL_FLOAT (INT_MAX * 0.01) // FIXME
// smooth must be in ]0.0, 1.0[
#define SMOOTH_MUL 0.06
#define SMOOTH_LASTAVG 0.06
#define DEFAULT_TARGET 0.25
// Data for specific instances of this filter
typedef struct af_volume_s
{
int method; // method used
float mul;
// method 1
float lastavg; // history value of the filter
// method 2
int idx;
struct {
float avg; // average level of the sample
int len; // sample size (weight)
} mem[NSAMPLES];
// "Ideal" level
float mid_s16;
float mid_float;
}af_volnorm_t;
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_volnorm_t* s = (af_volnorm_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(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
}else{
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:{
int i = 0;
float target = DEFAULT_TARGET;
sscanf((char*)arg,"%d:%f", &i, &target);
if (i != 1 && i != 2)
return AF_ERROR;
s->method = i-1;
s->mid_s16 = ((float)SHRT_MAX) * target;
s->mid_float = ((float)INT_MAX) * target;
return AF_OK;
}
}
return AF_UNKNOWN;
}
// Deallocate memory
static void uninit(struct af_instance_s* af)
{
free(af->data);
free(af->setup);
}
static void method1_int16(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
int16_t *data = (int16_t*)c->audio; // Audio data
int len = c->len/2; // Number of samples
float curavg = 0.0, newavg, neededmul;
int tmp;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
if (curavg > SIL_S16)
{
neededmul = s->mid_s16 / (curavg * s->mul);
s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
// clamp the mul coefficient
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
{
tmp = s->mul * data[i];
tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
data[i] = tmp;
}
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
}
static void method1_float(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
float *data = (float*)c->audio; // Audio data
int len = c->len/4; // Number of samples
float curavg = 0.0, newavg, neededmul, tmp;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
if (curavg > SIL_FLOAT) // FIXME
{
neededmul = s->mid_float / (curavg * s->mul);
s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
// clamp the mul coefficient
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
data[i] *= s->mul;
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
}
static void method2_int16(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
int16_t *data = (int16_t*)c->audio; // Audio data
int len = c->len/2; // Number of samples
float curavg = 0.0, newavg, avg = 0.0;
int tmp, totallen = 0;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
for (i = 0; i < NSAMPLES; i++)
{
avg += s->mem[i].avg * (float)s->mem[i].len;
totallen += s->mem[i].len;
}
if (totallen > MIN_SAMPLE_SIZE)
{
avg /= (float)totallen;
if (avg >= SIL_S16)
{
s->mul = s->mid_s16 / avg;
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
{
tmp = s->mul * data[i];
tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
data[i] = tmp;
}
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->mem[s->idx].len = len;
s->mem[s->idx].avg = newavg;
s->idx = (s->idx + 1) % NSAMPLES;
}
static void method2_float(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
float *data = (float*)c->audio; // Audio data
int len = c->len/4; // Number of samples
float curavg = 0.0, newavg, avg = 0.0, tmp;
int totallen = 0;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
for (i = 0; i < NSAMPLES; i++)
{
avg += s->mem[i].avg * (float)s->mem[i].len;
totallen += s->mem[i].len;
}
if (totallen > MIN_SAMPLE_SIZE)
{
avg /= (float)totallen;
if (avg >= SIL_FLOAT)
{
s->mul = s->mid_float / avg;
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
data[i] *= s->mul;
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->mem[s->idx].len = len;
s->mem[s->idx].avg = newavg;
s->idx = (s->idx + 1) % NSAMPLES;
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
af_volnorm_t *s = af->setup;
if(af->data->format == (AF_FORMAT_S16_NE))
{
if (s->method)
method2_int16(s, data);
else
method1_int16(s, data);
}
else if(af->data->format == (AF_FORMAT_FLOAT_NE))
{
if (s->method)
method2_float(s, data);
else
method1_float(s, data);
}
return data;
}
// 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=1;
af->data=calloc(1,sizeof(af_data_t));
af->setup=calloc(1,sizeof(af_volnorm_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
((af_volnorm_t*)af->setup)->mul = MUL_INIT;
((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET;
((af_volnorm_t*)af->setup)->idx = 0;
((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET;
((af_volnorm_t*)af->setup)->mid_float = ((float)INT_MAX) * DEFAULT_TARGET;
for (i = 0; i < NSAMPLES; i++)
{
((af_volnorm_t*)af->setup)->mem[i].len = 0;
((af_volnorm_t*)af->setup)->mem[i].avg = 0;
}
return AF_OK;
}
// Description of this filter
af_info_t af_info_volnorm = {
"Volume normalizer filter",
"volnorm",
"Alex Beregszaszi & Pierre Lombard",
"",
AF_FLAGS_NOT_REENTRANT,
af_open
};