view libaf/af_volnorm.c @ 16429:84174804804b

Updates to NUT spec: 1. remove average_bitrate 2. add other_stream_header, for subtitles and metadata 3. add max_pts to index 4. index_ptr - a 64 bit integer to say the total length of all index packets 5. specify how to write "multiple" indexes 6. change forward_ptr behavior, starts right after forward_ptr, ends after checksum 7. remove stream_id <-> stream_class limitation. 8. time_base_nom must also be non zero. 9. rename time_base_nom and time_base_denom, now timebase means the length of a tick, not amounts of ticks 10. remove (old?) sample_rate_mul stuff. 11. specify what exactly the checksum covers. 12. specify that stream classes which have multiple streams must have an info packet.. (in new Semantic requirements section) 13. Rename 'timestamp' to pts. 14. Change date of draft... 15. Add myself to authors...

/*=============================================================================
//	
//  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 2004 Alex Beregszaszi & Pierre Lombard
//
//=============================================================================
*/

#include <stdio.h>
#include <stdlib.h>
#include <string.h> 

#include <unistd.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
// "Ideal" level
#define MID_S16 (SHRT_MAX * 0.25)
#define MID_FLOAT (INT_MAX * 0.25)

// 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

// 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];
}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;
    sscanf((char*)arg,"%d", &i);
    if (i != 1 && i != 2)
	return AF_ERROR;
    s->method = i-1;
    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);
}

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 = 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 = 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 = 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 = 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 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_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 = MID_S16;
  ((af_volnorm_t*)af->setup)->idx = 0;
  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,
    open
};