view libaf/af_extrastereo.c @ 24589:9118be6575da

demux_audio.c: Fix timestamp handling The code calculated the pts values of audio packets by adding the length of the current packet to the pts of the previous one. The length of the previous packet should be added instead. This broke WAV timestamps near the end of the stream where a short packet occurs. Change the code to store the pts of the next packet instead of the last one. This fixes the WAV timestamps and allows some simplifications. MP3 timestamps are not affected as packets are always treated as constant decoded length, and FLAC timestamps still have worse problems (FLAC is treated as as if it was constant bitrate even though it isn't). Also store the timestamps as double instead of float.

/*=============================================================================
//	
//  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 <inttypes.h>
#include <math.h>
#include <limits.h>

#include "af.h"

// Data for specific instances of this filter
typedef struct af_extrastereo_s
{
    float mul;
}af_extrastereo_t;

static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data);
static af_data_t* play_float(struct af_instance_s* af, af_data_t* data);

// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
  af_extrastereo_t* s   = (af_extrastereo_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    = 2;
    if (((af_data_t*)arg)->format == AF_FORMAT_FLOAT_NE)
    {
	af->data->format = AF_FORMAT_FLOAT_NE;
	af->data->bps = 4;
	af->play = play_float;
    }// else
    {
	af->data->format = AF_FORMAT_S16_NE;
	af->data->bps = 2;
	af->play = play_s16;
    }

    return af_test_output(af,(af_data_t*)arg);
  }
  case AF_CONTROL_COMMAND_LINE:{
    float f;
    sscanf((char*)arg,"%f", &f);
    s->mul = f;
    return AF_OK;
  }
  case AF_CONTROL_ES_MUL | AF_CONTROL_SET:
    s->mul = *(float*)arg;
    return AF_OK;
  case AF_CONTROL_ES_MUL | AF_CONTROL_GET:
    *(float*)arg = s->mul;
    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);
}

// Filter data through filter
static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data)
{
  af_extrastereo_t *s = af->setup;
  register int i = 0;
  int16_t *a = (int16_t*)data->audio;	// Audio data
  int len = data->len/2;		// Number of samples
  int avg, l, r;
  
  for (i = 0; i < len; i+=2)
  {
    avg = (a[i] + a[i + 1]) / 2;
    
    l = avg + (int)(s->mul * (a[i] - avg));
    r = avg + (int)(s->mul * (a[i + 1] - avg));

    a[i] = clamp(l, SHRT_MIN, SHRT_MAX);
    a[i + 1] = clamp(r, SHRT_MIN, SHRT_MAX);
  }

  return data;
}

static af_data_t* play_float(struct af_instance_s* af, af_data_t* data)
{
  af_extrastereo_t *s = af->setup;
  register int i = 0;
  float *a = (float*)data->audio;	// Audio data
  int len = data->len/4;		// Number of samples
  float avg, l, r;
  
  for (i = 0; i < len; i+=2)
  {
    avg = (a[i] + a[i + 1]) / 2;
    
    l = avg + (s->mul * (a[i] - avg));
    r = avg + (s->mul * (a[i + 1] - avg));
    
    a[i] = af_softclip(l);
    a[i + 1] = af_softclip(r);
  }

  return data;
}

// Allocate memory and set function pointers
static int af_open(af_instance_t* af){
  af->control=control;
  af->uninit=uninit;
  af->play=play_s16;
  af->mul.n=1;
  af->mul.d=1;
  af->data=calloc(1,sizeof(af_data_t));
  af->setup=calloc(1,sizeof(af_extrastereo_t));
  if(af->data == NULL || af->setup == NULL)
    return AF_ERROR;

  ((af_extrastereo_t*)af->setup)->mul = 2.5;
  return AF_OK;
}

// Description of this filter
af_info_t af_info_extrastereo = {
    "Extra stereo",
    "extrastereo",
    "Alex Beregszaszi & Pierre Lombard",
    "",
    AF_FLAGS_NOT_REENTRANT,
    af_open
};