postfilt.h

g729 coding ipaddressing

/**************************************************************************
*
* ROUTINE
*				postfilt & postfilt2
*
* FUNCTION
*				 
*				reduce coder noise (sample wise AGC version)
*
* SYNOPSIS
*
*				subroutine postfilt(s, l, alpha, beta, powerin
*								powerout, dp1, dp2, dp3)
*
*	formal 
*						data	I/O
*		name			type	type	function
*		-------------------------------------------------------------------
*		s				float	i/o 	speech input/postfiltered output
*		l				int 	i		subframe size
*		alpha			float	i		filter parameter
*		beta			float	i		filter parameter
*		powerin 		float	i/o 	input power estimate
*		powerout		float	i/o 	output power estimate
*		dp1 			float	i/o 	filter memory
*		dp2 			float	i/o 	filter memory
*		dp3 			float	i/o 	filter memory
*
*	external 
*						data	I/O
*		name			type	type	function
*		-------------------------------------------------------------------
*		no				int 	i
*		fci[]			float	i
*
***************************************************************************
*
* DESCRIPTION
*
*		Adaptive postfilter routine to reduce perceptual coder noise.
*		The postfilter emphasis the spectral regions predicted by the
*		short-term LPC analysis.  This tends to mask coder noise by
*		concentrating it under the formant peaks.  Unfortunately, acoustic
*		background noise may also be enhanced because LPC analysis often
*		models acoustic noise instead of speech formants.  In addtion,
*		postfiltering can be detrimental to tandem coding if not taken
*		into consideration.  (To overcome these problems, we hope to
*       eventually incorporate the postfilter's enhancement properties
*		into the analysis process.)
*
*		Adaptive spectral tilt compensation is applied to flatten the 
*		overall tilt of the postfilter.  [[Slight high frequency boost is 
*		applied for output shaping.  A pitch postfilter is used to reduce 
*		pitch buzz.]]  Finally, AGC compensates for the filter gains using 
*		a time constant set by parameter tc that should be dependent on
*		frame length.
*
*
***************************************************************************
*
* CALLED BY
*
*		celp
*
* CALLS
*
*		bwexp	zerofilt   polefilt   pctorc
*
***************************************************************************
*
* REFERENCES
*
*		Chen & Gersho, Real-Time Vector APC Speech Coding at 4800 bps
*       with Adaptive Postfiltering, ICASSP '87
*
*       Juin-Hwey (Raymond) Chen, "Low-Bit-Rate Predictive Coding of
*       Speech Waveforms Based on Vector Quantization," PhD Dissertation,
*		UCSB ECE Dept., March 1987.
*
*       Ramamoorthy, Jayant, Cox, & Sondhi, "Enhancement of ADPCM Speech
*		Coding with Backward-Adaptive Algorithms for Postfiltering and
*       Noise Feedback," IEEE JOSAIC, Feb. 1988, pp. 364-382.
*
**************************************************************************/
#define TC	0.01

static void postfilt(float s[], int l, float alpha, float beta,
	float *powerin, float *powerout, float a_dp1[],
	float a_dp2[], float a_dp3[])
{
  int n;
  float ast[2];
  float pcexp1[MAXNO + 1], pcexp2[MAXNO + 1], rcexp2[MAXNO];

#ifdef POSTFIL2
  float scale;
#else
  float newpowerin[MAXL + 1], newpowerout[MAXL + 1];
#endif

  /*					*estimate input power							*/

#ifdef POSTFIL2
  for (n = 0; n < l; n++)
	*powerin = *powerin * (1.0 - TC) + TC * s[n] * s[n];
#else
  newpowerin[0] = *powerin;
  for (n = 0; n < l; n++)
	newpowerin[n + 1] = (float) ((1.0 - TC) * newpowerin[n] + TC * s[n] * s[n]);
  *powerin = newpowerin[l];
#endif

  /* *BW expansion														*/
  bwexp(beta, fci, pcexp1, no);
  bwexp(alpha, fci, pcexp2, no);

  /* *pole-zero postfilter												*/
  zerofilt(pcexp1, no, a_dp1, s, l);
  polefilt(pcexp2, no, a_dp2, s, l);

  /* *find spectral tilt (1st order fit) of postfilter
	 *(denominator dominates the tilt)									*/
  pctorc(pcexp2, rcexp2, no);

  /* *tilt compensation by a scaled zero
     *(don't allow hF roll-off)                                         */
  ast[0] = 1.0;
  ast[1] = (float) ((rcexp2[0] > 0.) ? -0.5 * rcexp2[0] : 0);
  zerofilt(ast, 1, a_dp3, s, l);

  /* *estimate output power 											*/

#ifdef POSTFIL2
  for (n = 0; n < l; n++)
	*powerout = *powerout * (1.0 - TC) + TC * s[n] * s[n];

  /* *block wise automatic gain control 								*/

  if (*powerout > 0.0)
	for (scale = sqrt(*powerin / *powerout), n = 0; n < l; n++)
	  s[n] *= scale;
#else
  newpowerout[0] = *powerout;
  for (n = 0; n < l; n++)
	newpowerout[n + 1] = (float) ((1.0 - TC) * newpowerout[n] + TC * s[n] * s[n]);
  *powerout = newpowerout[l];

  /* *sample wise automatic gain control								*/

  for (n = 0; n < l; n++)
  {
	if (newpowerout[n + 1] > 0.0)
	  s[n] *= (float) sqrt(newpowerin[n + 1] / newpowerout[n + 1]);
  }
#endif
}
#undef TC