view libaf/af_volnorm.c @ 36838:7df9dd22f234

Don't set win32 as audio driver if none has been given. Select from the list of audio drivers instead. Having win32 as selected item in the combo box although this isn't used by MPlayer by default is confusing as well. Besides that, there seem to be issues with this driver when changing from or to it during playback.

/*
 * 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 "libavutil/common.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 (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 = 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 = av_clipf(s->mul, MUL_MIN, MUL_MAX);
  }

  // Scale & clamp the samples
  for (i = 0; i < len; i++)
  {
    tmp = s->mul * data[i];
    tmp = av_clip_int16(tmp);
    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 = av_clipf(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 = av_clipf(s->mul, MUL_MIN, MUL_MAX);
    }
  }

  // Scale & clamp the samples
  for (i = 0; i < len; i++)
  {
    tmp = s->mul * data[i];
    tmp = av_clip_int16(tmp);
    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 = av_clipf(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 = 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
};