view libao2/pl_volnorm.c @ 13249:a6642a4330fa

ensure that avi files have a valid header as soon as possible. without this, the header says 0x0 video size, which works with mplayer when the video size is stored in the codec data, but it does NOT work with other players or with codecs that don't store size (e.g. snow). actually i don't like having seeks in the muxer module, but i don't know any other way to implement this fix without major changes to mencoder. if you have a better fix, please reverse this and commit yours.

/* Normalizer plugin
 *
 * Limitations:
 *  - only AFMT_S16_LE supported
 *  - no parameters yet => tweak the values by editing the #defines
 *
 * License: GPLv2
 * Author: pl <[e-mail]> (c) 2002 and beyond...
 *
 * Sources: some ideas from volnorm plugin for xmms
 *
 * */

#define PLUGIN

/* Values for AVG:
 * 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)
 *
 * */
#define AVG 1

#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 <[e-mail]>",
        ""
};

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 5.0
static float mul;


#if AVG==1
// "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


#elif AVG==2
// Size of the memory array
// FIXME: should depend on the frequency of the data (should be a few seconds)
#define NSAMPLES 128

// Indicates where to write (in 0..NSAMPLES-1)
static int idx;
// The array
static struct {
    float avg;		// average level of the sample
    int32_t len;	// sample size (weight)
} mem[NSAMPLES];

// 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

#else
// Kab00m !
#error "Unknown AVG"
#endif


// Some limits
#define MIN_S16 -32768
#define MAX_S16  32767

// "Ideal" level
#define MID_S16 (MAX_S16 * 0.25)

// Silence level
// FIXME: should be relative to the level of the samples
#define SIL_S16 (MAX_S16 * 0.01)


// 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,void *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_NE):
      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(){
  int i;
  mul = MUL_INIT;
  switch(ao_plugin_data.format) {
    case(AFMT_S16_NE):
#if AVG==1
      lastavg = MID_S16;
#elif AVG==2
      for(i=0; i < NSAMPLES; ++i) {
	      mem[i].len = 0;
	      mem[i].avg = 0;
      }
      idx = 0;
#endif

      break;
    default:
      fprintf(stderr,"[pl_volnorm] internal inconsistency - 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_NE): {
#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, tmp;
    float curavg, newavg;

#if AVG==1
    float neededmul;
#elif AVG==2
    float avg;
    int32_t totallen;
#endif

    // Evaluate current samples average level
    curavg = 0.0;
    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
#if AVG==1
    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);
    }
#elif AVG==2
    avg = 0.0;
    totallen = 0;

    for (i = 0; i < NSAMPLES; ++i) {
        avg += mem[i].avg * (float) mem[i].len;
        totallen += mem[i].len;
    }

    if (totallen > MIN_SAMPLE_SIZE) {
    	avg /= (float) totallen;
    	if (avg >= SIL_S16) {
    	    mul = (float) MID_S16 / avg;
    	    CLAMP(mul, MUL_MIN, MUL_MAX);
    	}
    }
#endif

    // Scale & clamp the samples
    for (i = 0; i < len ; ++i) {
      tmp = mul * data[i];
      CLAMP(tmp, MIN_S16, MAX_S16);
      data[i] = tmp;
    }

    // Evaluation of newavg (not 100% accurate because of values clamping)
    newavg = mul * curavg;

    // Stores computed values for future smoothing
#if AVG==1
    lastavg = (1.0 - SMOOTH_LASTAVG) * lastavg + SMOOTH_LASTAVG * newavg;
    //printf("\rmul=%02.1f ", mul);
#elif AVG==2
    mem[idx].len = len;
    mem[idx].avg = newavg;
    idx = (idx + 1) % NSAMPLES;
    //printf("\rmul=%02.1f (%04dKiB) ", mul, totallen/1024);
#endif
    //fflush(stdout);

    break;
  }
  default:
    return 0;
  }
  return 1;

}