Rename af_volnorm to af_drc

The previous name of this filter was misleading, because it doesn’t actually
normalize volume levels. What it does is closer to performing low-quality
dynamic range compression, hence it is now called af_drc.

1 files changed, 0 insertions, 353 deletions

diff --git a/audio/filter/af_volnorm.c b/audio/filter/af_volnorm.c
deleted file mode 100644
index f49bbc185a..0000000000
--- a/audio/filter/af_volnorm.c
+++ /dev/null
@@ -1,353 +0,0 @@
-/*
- * 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 0.01
-
-// 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* 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   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-
-    if(((struct mp_audio*)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,(struct mp_audio*)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* af)
-{
-    free(af->data);
-    free(af->setup);
-}
-
-static void method1_int16(af_volnorm_t *s, struct mp_audio *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, struct mp_audio *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, struct mp_audio *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, struct mp_audio *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 struct mp_audio* play(struct af_instance* af, struct mp_audio* 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(struct af_instance* af){
-  int i = 0;
-  af->control=control;
-  af->uninit=uninit;
-  af->play=play;
-  af->mul=1;
-  af->data=calloc(1,sizeof(struct mp_audio));
-  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
-struct af_info af_info_volnorm = {
-    "Volume normalizer filter",
-    "volnorm",
-    "Alex Beregszaszi & Pierre Lombard",
-    "",
-    AF_FLAGS_NOT_REENTRANT,
-    af_open
-};