ref: ddca83a2f0f45046c7008bb3e19da5294a1bd160
dir: /src/wavesynth/cst_sigpr.c/
/*************************************************************************/
/* */
/* Language Technologies Institute */
/* Carnegie Mellon University */
/* Copyright (c) 2001 */
/* All Rights Reserved. */
/* */
/* Permission is hereby granted, free of charge, to use and distribute */
/* this software and its documentation without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of this work, and to */
/* permit persons to whom this work is furnished to do so, subject to */
/* the following conditions: */
/* 1. The code must retain the above copyright notice, this list of */
/* conditions and the following disclaimer. */
/* 2. Any modifications must be clearly marked as such. */
/* 3. Original authors' names are not deleted. */
/* 4. The authors' names are not used to endorse or promote products */
/* derived from this software without specific prior written */
/* permission. */
/* */
/* CARNEGIE MELLON UNIVERSITY AND THE CONTRIBUTORS TO THIS WORK */
/* DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING */
/* ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT */
/* SHALL CARNEGIE MELLON UNIVERSITY NOR THE CONTRIBUTORS BE LIABLE */
/* FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES */
/* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN */
/* AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, */
/* ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF */
/* THIS SOFTWARE. */
/* */
/*************************************************************************/
/* Author: Alan W Black (awb@cs.cmu.edu) */
/* Date: January 2001 */
/*************************************************************************/
/* */
/* Signal processing functions */
/* */
/*************************************************************************/
#include "cst_math.h"
#include "cst_hrg.h"
#include "cst_wave.h"
#include "cst_sigpr.h"
#include "cst_sts.h"
cst_wave *lpc_resynth(cst_lpcres *lpcres)
{
cst_wave *w;
int i,j,r,o,k;
int ci,cr;
float *outbuf, *lpccoefs;
int pm_size_samps;
/* Get a new wave to build the signal into */
w = new_wave();
cst_wave_resize(w,lpcres->num_samples,1);
w->sample_rate = lpcres->sample_rate;
/* outbuf is a circular buffer with past relevant samples in it */
outbuf = cst_alloc(float,1+lpcres->num_channels);
/* unpacked lpc coefficients */
lpccoefs = cst_alloc(float,lpcres->num_channels);
for (r=0,o=lpcres->num_channels,i=0; i < lpcres->num_frames; i++)
{
pm_size_samps = lpcres->sizes[i];
/* Unpack the LPC coefficients */
for (k=0; k<lpcres->num_channels; k++)
{
lpccoefs[k] = (float)((((double)lpcres->frames[i][k])/65535.0)*
lpcres->lpc_range) + lpcres->lpc_min;
}
/* Note we don't zero the lead in from the previous part */
/* seems like you should but it makes it worse if you do */
/* memset(outbuf,0,sizeof(float)*(1+lpcres->num_channels)); */
/* resynthesis the signal */
for (j=0; j < pm_size_samps; j++,r++)
{
outbuf[o] = (float)cst_ulaw_to_short(lpcres->residual[r]);
cr = (o == 0 ? lpcres->num_channels : o-1);
for (ci=0; ci < lpcres->num_channels; ci++)
{
outbuf[o] += lpccoefs[ci] * outbuf[cr];
cr = (cr == 0 ? lpcres->num_channels : cr-1);
}
w->samples[r] = (short)(outbuf[o]);
o = (o == lpcres->num_channels ? 0 : o+1);
}
}
cst_free(outbuf);
cst_free(lpccoefs);
return w;
}
cst_wave *lpc_resynth_windows(cst_lpcres *lpcres)
{
cst_wave *w;
int i,j,r,o,k;
int ci,cr;
float *outbuf, *lpccoefs;
int pm_size_samps;
/* Get a new wave to build the signal into */
w = new_wave();
cst_wave_resize(w,lpcres->num_samples,1);
w->sample_rate = lpcres->sample_rate;
/* outbuf is a circular buffer with past relevant samples in it */
outbuf = cst_alloc(float,1+lpcres->num_channels);
/* unpacked lpc coefficients */
lpccoefs = cst_alloc(float,lpcres->num_channels);
for (r=0,o=lpcres->num_channels,i=0; i < lpcres->num_frames; i++)
{
pm_size_samps = lpcres->sizes[i];
/* Unpack the LPC coefficients */
for (k=0; k<lpcres->num_channels; k++)
{
lpccoefs[k] = ((float)(((double)lpcres->frames[i][k])/65535.0)*
lpcres->lpc_range) + lpcres->lpc_min;
}
memset(outbuf,0,sizeof(float)*(1+lpcres->num_channels));
/* resynthesis the signal */
for (j=0; j < pm_size_samps; j++,r++)
{
outbuf[o] = (float)cst_ulaw_to_short(lpcres->residual[r]);
cr = (o == 0 ? lpcres->num_channels : o-1);
for (ci=0; ci < lpcres->num_channels; ci++)
{
outbuf[o] += lpccoefs[ci] * outbuf[cr];
cr = (cr == 0 ? lpcres->num_channels : cr-1);
}
w->samples[r] = (short)(outbuf[o]);
o = (o == lpcres->num_channels ? 0 : o+1);
}
}
cst_free(outbuf);
cst_free(lpccoefs);
return w;
}
const static short ulaw_to_short_table[] =
{
-32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956,
-23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764,
-15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412,
-11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316,
-7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140,
-5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092,
-3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004,
-2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980,
-1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436,
-1372, -1308, -1244, -1180, -1116, -1052, -988, -924,
-876, -844, -812, -780, -748, -716, -684, -652,
-620, -588, -556, -524, -492, -460, -428, -396,
-372, -356, -340, -324, -308, -292, -276, -260,
-244, -228, -212, -196, -180, -164, -148, -132,
-120, -112, -104, -96, -88, -80, -72, -64,
-56, -48, -40, -32, -24, -16, -8, 0,
32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956,
23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764,
15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412,
11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316,
7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140,
5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092,
3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004,
2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980,
1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436,
1372, 1308, 1244, 1180, 1116, 1052, 988, 924,
876, 844, 812, 780, 748, 716, 684, 652,
620, 588, 556, 524, 492, 460, 428, 396,
372, 356, 340, 324, 308, 292, 276, 260,
244, 228, 212, 196, 180, 164, 148, 132,
120, 112, 104, 96, 88, 80, 72, 64,
56, 48, 40, 32, 24, 16, 8, 0 };
cst_wave *lpc_resynth_fixedpoint(cst_lpcres *lpcres)
{
/* The fixed point version, without floats */
cst_wave *w;
int i,j,r,o,k;
int stream_mark;
int ci,cr;
int *outbuf, *lpccoefs;
int pm_size_samps, ilpc_min, ilpc_range;
int rc = CST_AUDIO_STREAM_CONT;
/* Get a new wave to build the signal into */
w = new_wave();
cst_wave_resize(w,lpcres->num_samples,1);
w->sample_rate = lpcres->sample_rate;
/* outbuf is a circular buffer with past relevant samples in it */
outbuf = cst_alloc(int,1+lpcres->num_channels);
/* unpacked lpc coefficients */
lpccoefs = cst_alloc(int,lpcres->num_channels);
ilpc_min = (int)(lpcres->lpc_min*32768.0);
/* assume range is never > abs(16) */
ilpc_range = (int)(lpcres->lpc_range*2048.0);
stream_mark = 0;
for (r=0,o=lpcres->num_channels,i=0;
(rc == CST_AUDIO_STREAM_CONT) && (i < lpcres->num_frames);
i++)
{
pm_size_samps = lpcres->sizes[i];
if (lpcres->delayed_decoding)
{
/* do decoding for this frame */
add_residual_g721vuv(lpcres->sizes[i],
&lpcres->residual[r],
lpcres->sizes[i],
lpcres->packed_residuals[i]);
}
/* Unpack the LPC coefficients */
for (k=0; k<lpcres->num_channels; k++)
lpccoefs[k]=((lpcres->frames[i][k]/2*ilpc_range)/2048+ilpc_min)/2;
/* resynthesis the signal */
for (j=0; j < pm_size_samps; j++,r++)
{
outbuf[o] = (int)ulaw_to_short_table[lpcres->residual[r]];
outbuf[o] *= 16384;
cr = (o == 0 ? lpcres->num_channels : o-1);
for (ci=0; ci < lpcres->num_channels; ci++)
{
outbuf[o] += lpccoefs[ci]*outbuf[cr];
cr = (cr == 0 ? lpcres->num_channels : cr-1);
}
outbuf[o] /= 16384;
w->samples[r] = (short)outbuf[o];
o = (o == lpcres->num_channels ? 0 : o+1);
}
if (lpcres->asi && (r-stream_mark > lpcres->asi->min_buffsize))
{
rc = (*lpcres->asi->asc)(w,stream_mark,r-stream_mark,0,
lpcres->asi);
stream_mark = r;
}
}
if ((lpcres->asi) && (rc == CST_AUDIO_STREAM_CONT))
(*lpcres->asi->asc)(w,stream_mark,r-stream_mark,1,lpcres->asi);
cst_free(outbuf);
cst_free(lpccoefs);
w->num_samples = r; /* just to be safe */
if (rc == CST_AUDIO_STREAM_STOP)
{
delete_wave(w);
return NULL;
}
else
return w;
}
cst_wave *lpc_resynth_sfp(cst_lpcres *lpcres)
{
/* The fixed point spike excited, without floats */
cst_wave *w;
int i,j,r,o,k;
int ci,cr;
int *outbuf, *lpccoefs;
int pm_size_samps, ilpc_min, ilpc_range;
/* Get a new wave to build the signal into */
w = new_wave();
cst_wave_resize(w,lpcres->num_samples,1);
w->sample_rate = lpcres->sample_rate;
/* outbuf is a circular buffer with past relevant samples in it */
outbuf = cst_alloc(int,1+lpcres->num_channels);
/* unpacked lpc coefficients */
lpccoefs = cst_alloc(int,lpcres->num_channels);
ilpc_min = (int)(lpcres->lpc_min*32768.0);
/* assume range is never > abs(16) */
ilpc_range = (int)(lpcres->lpc_range*2048.0);
for (r=0,o=lpcres->num_channels,i=0; i < lpcres->num_frames; i++)
{
pm_size_samps = lpcres->sizes[i];
/* Unpack the LPC coefficients */
for (k=0; k<lpcres->num_channels; k++)
lpccoefs[k]=((lpcres->frames[i][k]/2*ilpc_range)/2048+ilpc_min)/2;
/* resynthesis the signal */
for (j=0; j < pm_size_samps; j++,r++)
{
outbuf[o] = (int)cst_ulaw_to_short(lpcres->residual[r]);
cr = (o == 0 ? lpcres->num_channels : o-1);
for (ci=0; ci < lpcres->num_channels; ci++)
{
outbuf[o] += (lpccoefs[ci]*outbuf[cr])/16384;
cr = (cr == 0 ? lpcres->num_channels : cr-1);
}
w->samples[r] = (short)outbuf[o];
o = (o == lpcres->num_channels ? 0 : o+1);
}
}
cst_free(outbuf);
cst_free(lpccoefs);
return w;
}