| /*************************************************************************/ |
| /* */ |
| /* 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 ([email protected]) */ |
| /* 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; |
| |
| } |
