postfilt.c
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上传日期:2021-02-19
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文件大小:5k
源码类别:
语音压缩
开发平台:
C/C++
- /**************************************************************************
- *
- * 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
- #include <math.h>
- #include "ccsub.h"
- extern int no;
- extern float fci[MAXNO+1];
- postfilt(s, l, alpha, beta, powerin, powerout, dp1, dp2, dp3)
- int l;
- float s[], alpha, beta, *powerin, *powerout;
- float dp1[], dp2[], 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] = (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, dp1, s, l);
- polefilt(pcexp2, no, 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] = (rcexp2[0] > 0.) ? -0.5 * rcexp2[0] : 0;
- zerofilt(ast, 1, 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] = (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] *= sqrt(newpowerin[n + 1] / newpowerout[n + 1]);
- }
- #endif
- }