view libaf/af_tools.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.

#include <math.h>
#include <string.h> 
#include <af.h>

/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
   fail */
inline int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
{
  int i = 0; 
  // Sanity check
  if(!in || !out) 
    return AF_ERROR;

  for(i=0;i<n;i++){
    if(in[i]<=-200)
      out[i]=0.0;
    else
      out[i]=pow(10.0,clamp(in[i],mi,ma)/k);
  }
  return AF_OK;
}

/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
   fail */
inline int af_to_dB(int n, float* in, float* out, float k)
{
  int i = 0; 
  // Sanity check
  if(!in || !out) 
    return AF_ERROR;

  for(i=0;i<n;i++){
    if(in[i] == 0.0)
      out[i]=-200.0;
    else
      out[i]=k*log10(in[i]);
  }
  return AF_OK;
}

/* Convert from ms to sample time */
inline int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
{
  int i = 0; 
  // Sanity check
  if(!in || !out) 
    return AF_ERROR;

  for(i=0;i<n;i++)
    out[i]=(int)((float)rate * clamp(in[i],mi,ma)/1000.0);

  return AF_OK;
}

/* Convert from sample time to ms */
inline int af_to_ms(int n, int* in, float* out, int rate)
{
  int i = 0; 
  // Sanity check
  if(!in || !out || !rate) 
    return AF_ERROR;

  for(i=0;i<n;i++)
    out[i]=1000.0 * (float)in[i]/((float)rate);
  
  return AF_OK;
}

/* Helper function for testing the output format */
inline int af_test_output(struct af_instance_s* af, af_data_t* out)
{
  if((af->data->format != out->format) || 
     (af->data->bps    != out->bps)    ||
     (af->data->rate   != out->rate)   ||
     (af->data->nch    != out->nch)){
    memcpy(out,af->data,sizeof(af_data_t));
    return AF_FALSE;
  }
  return AF_OK;
}

/* Soft clipping, the sound of a dream, thanks to Jon Wattes
   post to Musicdsp.org */
inline float af_softclip(float a)
{
    if (a >= M_PI/2)
	return 1.0;
    else if (a <= -M_PI/2)
	return -1.0;
    else
	return sin(a);
}