view libaf/af_tools.c @ 17149:9a0a376a54b1

Move audio packets reordering from codec interface to demuxers for real files (old and new format), pass only real extradata to the codec Enable cook codec from lavc, prefer lavc codecs for 14_4 and 28_8 formats. Disable internal 28_8, it's broken now and will be removed soon

#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);
}