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1/*
2 * Copyright (c) 2012 Andrew D'Addesio
3 * Copyright (c) 2013-2014 Mozilla Corporation
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
23 * @file
24 * Opus SILK decoder
25 */
26
27#include <stdint.h>
28
29#include "opus.h"
30#include "opustab.h"
31
32typedef struct SilkFrame {
33 int coded;
34 int log_gain;
35 int16_t nlsf[16];
36 float lpc[16];
37
38 float output [2 * SILK_HISTORY];
39 float lpc_history[2 * SILK_HISTORY];
40 int primarylag;
41
42 int prev_voiced;
43} SilkFrame;
44
45struct SilkContext {
46 AVCodecContext *avctx;
47 int output_channels;
48
49 int midonly;
50 int subframes;
51 int sflength;
52 int flength;
53 int nlsf_interp_factor;
54
55 enum OpusBandwidth bandwidth;
56 int wb;
57
58 SilkFrame frame[2];
59 float prev_stereo_weights[2];
60 float stereo_weights[2];
61
62 int prev_coded_channels;
63};
64
65static inline void silk_stabilize_lsf(int16_t nlsf[16], int order, const uint16_t min_delta[17])
66{
67 int pass, i;
68 for (pass = 0; pass < 20; pass++) {
69 int k, min_diff = 0;
70 for (i = 0; i < order+1; i++) {
71 int low = i != 0 ? nlsf[i-1] : 0;
72 int high = i != order ? nlsf[i] : 32768;
73 int diff = (high - low) - (min_delta[i]);
74
75 if (diff < min_diff) {
76 min_diff = diff;
77 k = i;
78
79 if (pass == 20)
80 break;
81 }
82 }
83 if (min_diff == 0) /* no issues; stabilized */
84 return;
85
86 /* wiggle one or two LSFs */
87 if (k == 0) {
88 /* repel away from lower bound */
89 nlsf[0] = min_delta[0];
90 } else if (k == order) {
91 /* repel away from higher bound */
92 nlsf[order-1] = 32768 - min_delta[order];
93 } else {
94 /* repel away from current position */
95 int min_center = 0, max_center = 32768, center_val;
96
97 /* lower extent */
98 for (i = 0; i < k; i++)
99 min_center += min_delta[i];
100 min_center += min_delta[k] >> 1;
101
102 /* upper extent */
103 for (i = order; i > k; i--)
104 max_center -= min_delta[i];
105 max_center -= min_delta[k] >> 1;
106
107 /* move apart */
108 center_val = nlsf[k - 1] + nlsf[k];
109 center_val = (center_val >> 1) + (center_val & 1); // rounded divide by 2
110 center_val = FFMIN(max_center, FFMAX(min_center, center_val));
111
112 nlsf[k - 1] = center_val - (min_delta[k] >> 1);
113 nlsf[k] = nlsf[k - 1] + min_delta[k];
114 }
115 }
116
117 /* resort to the fall-back method, the standard method for LSF stabilization */
118
119 /* sort; as the LSFs should be nearly sorted, use insertion sort */
120 for (i = 1; i < order; i++) {
121 int j, value = nlsf[i];
122 for (j = i - 1; j >= 0 && nlsf[j] > value; j--)
123 nlsf[j + 1] = nlsf[j];
124 nlsf[j + 1] = value;
125 }
126
127 /* push forwards to increase distance */
128 if (nlsf[0] < min_delta[0])
129 nlsf[0] = min_delta[0];
130 for (i = 1; i < order; i++)
131 if (nlsf[i] < nlsf[i - 1] + min_delta[i])
132 nlsf[i] = nlsf[i - 1] + min_delta[i];
133
134 /* push backwards to increase distance */
135 if (nlsf[order-1] > 32768 - min_delta[order])
136 nlsf[order-1] = 32768 - min_delta[order];
137 for (i = order-2; i >= 0; i--)
138 if (nlsf[i] > nlsf[i + 1] - min_delta[i+1])
139 nlsf[i] = nlsf[i + 1] - min_delta[i+1];
140
141 return;
142}
143
144static inline int silk_is_lpc_stable(const int16_t lpc[16], int order)
145{
146 int k, j, DC_resp = 0;
147 int32_t lpc32[2][16]; // Q24
148 int totalinvgain = 1 << 30; // 1.0 in Q30
149 int32_t *row = lpc32[0], *prevrow;
150
151 /* initialize the first row for the Levinson recursion */
152 for (k = 0; k < order; k++) {
153 DC_resp += lpc[k];
154 row[k] = lpc[k] * 4096;
155 }
156
157 if (DC_resp >= 4096)
158 return 0;
159
160 /* check if prediction gain pushes any coefficients too far */
161 for (k = order - 1; 1; k--) {
162 int rc; // Q31; reflection coefficient
163 int gaindiv; // Q30; inverse of the gain (the divisor)
164 int gain; // gain for this reflection coefficient
165 int fbits; // fractional bits used for the gain
166 int error; // Q29; estimate of the error of our partial estimate of 1/gaindiv
167
168 if (FFABS(row[k]) > 16773022)
169 return 0;
170
171 rc = -(row[k] * 128);
172 gaindiv = (1 << 30) - MULH(rc, rc);
173
174 totalinvgain = MULH(totalinvgain, gaindiv) << 2;
175 if (k == 0)
176 return (totalinvgain >= 107374);
177
178 /* approximate 1.0/gaindiv */
179 fbits = opus_ilog(gaindiv);
180 gain = ((1 << 29) - 1) / (gaindiv >> (fbits + 1 - 16)); // Q<fbits-16>
181 error = (1 << 29) - MULL(gaindiv << (15 + 16 - fbits), gain, 16);
182 gain = ((gain << 16) + (error * gain >> 13));
183
184 /* switch to the next row of the LPC coefficients */
185 prevrow = row;
186 row = lpc32[k & 1];
187
188 for (j = 0; j < k; j++) {
189 int x = prevrow[j] - ROUND_MULL(prevrow[k - j - 1], rc, 31);
190 row[j] = ROUND_MULL(x, gain, fbits);
191 }
192 }
193}
194
195static void silk_lsp2poly(const int32_t lsp[16], int32_t pol[16], int half_order)
196{
197 int i, j;
198
199 pol[0] = 65536; // 1.0 in Q16
200 pol[1] = -lsp[0];
201
202 for (i = 1; i < half_order; i++) {
203 pol[i + 1] = pol[i - 1] * 2 - ROUND_MULL(lsp[2 * i], pol[i], 16);
204 for (j = i; j > 1; j--)
205 pol[j] += pol[j - 2] - ROUND_MULL(lsp[2 * i], pol[j - 1], 16);
206
207 pol[1] -= lsp[2 * i];
208 }
209}
210
211static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order)
212{
213 int i, k;
214 int32_t lsp[16]; // Q17; 2*cos(LSF)
215 int32_t p[9], q[9]; // Q16
216 int32_t lpc32[16]; // Q17
217 int16_t lpc[16]; // Q12
218
219 /* convert the LSFs to LSPs, i.e. 2*cos(LSF) */
220 for (k = 0; k < order; k++) {
221 int index = nlsf[k] >> 8;
222 int offset = nlsf[k] & 255;
223 int k2 = (order == 10) ? ff_silk_lsf_ordering_nbmb[k] : ff_silk_lsf_ordering_wb[k];
224
225 /* interpolate and round */
226 lsp[k2] = ff_silk_cosine[index] * 256;
227 lsp[k2] += (ff_silk_cosine[index + 1] - ff_silk_cosine[index]) * offset;
228 lsp[k2] = (lsp[k2] + 4) >> 3;
229 }
230
231 silk_lsp2poly(lsp , p, order >> 1);
232 silk_lsp2poly(lsp + 1, q, order >> 1);
233
234 /* reconstruct A(z) */
235 for (k = 0; k < order>>1; k++) {
236 lpc32[k] = -p[k + 1] - p[k] - q[k + 1] + q[k];
237 lpc32[order-k-1] = -p[k + 1] - p[k] + q[k + 1] - q[k];
238 }
239
240 /* limit the range of the LPC coefficients to each fit within an int16_t */
241 for (i = 0; i < 10; i++) {
242 int j;
243 unsigned int maxabs = 0;
244 for (j = 0, k = 0; j < order; j++) {
245 unsigned int x = FFABS(lpc32[k]);
246 if (x > maxabs) {
247 maxabs = x; // Q17
248 k = j;
249 }
250 }
251
252 maxabs = (maxabs + 16) >> 5; // convert to Q12
253
254 if (maxabs > 32767) {
255 /* perform bandwidth expansion */
256 unsigned int chirp, chirp_base; // Q16
257 maxabs = FFMIN(maxabs, 163838); // anything above this overflows chirp's numerator
258 chirp_base = chirp = 65470 - ((maxabs - 32767) << 14) / ((maxabs * (k+1)) >> 2);
259
260 for (k = 0; k < order; k++) {
261 lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
262 chirp = (chirp_base * chirp + 32768) >> 16;
263 }
264 } else break;
265 }
266
267 if (i == 10) {
268 /* time's up: just clamp */
269 for (k = 0; k < order; k++) {
270 int x = (lpc32[k] + 16) >> 5;
271 lpc[k] = av_clip_int16(x);
272 lpc32[k] = lpc[k] << 5; // shortcut mandated by the spec; drops lower 5 bits
273 }
274 } else {
275 for (k = 0; k < order; k++)
276 lpc[k] = (lpc32[k] + 16) >> 5;
277 }
278
279 /* if the prediction gain causes the LPC filter to become unstable,
280 apply further bandwidth expansion on the Q17 coefficients */
281 for (i = 1; i <= 16 && !silk_is_lpc_stable(lpc, order); i++) {
282 unsigned int chirp, chirp_base;
283 chirp_base = chirp = 65536 - (1 << i);
284
285 for (k = 0; k < order; k++) {
286 lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
287 lpc[k] = (lpc32[k] + 16) >> 5;
288 chirp = (chirp_base * chirp + 32768) >> 16;
289 }
290 }
291
292 for (i = 0; i < order; i++)
293 lpcf[i] = lpc[i] / 4096.0f;
294}
295
296static inline void silk_decode_lpc(SilkContext *s, SilkFrame *frame,
297 OpusRangeCoder *rc,
298 float lpc_leadin[16], float lpc[16],
299 int *lpc_order, int *has_lpc_leadin, int voiced)
300{
301 int i;
302 int order; // order of the LP polynomial; 10 for NB/MB and 16 for WB
303 int8_t lsf_i1, lsf_i2[16]; // stage-1 and stage-2 codebook indices
304 int16_t lsf_res[16]; // residual as a Q10 value
305 int16_t nlsf[16]; // Q15
306
307 *lpc_order = order = s->wb ? 16 : 10;
308
309 /* obtain LSF stage-1 and stage-2 indices */
310 lsf_i1 = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s1[s->wb][voiced]);
311 for (i = 0; i < order; i++) {
312 int index = s->wb ? ff_silk_lsf_s2_model_sel_wb [lsf_i1][i] :
313 ff_silk_lsf_s2_model_sel_nbmb[lsf_i1][i];
314 lsf_i2[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2[index]) - 4;
315 if (lsf_i2[i] == -4)
316 lsf_i2[i] -= ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
317 else if (lsf_i2[i] == 4)
318 lsf_i2[i] += ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
319 }
320
321 /* reverse the backwards-prediction step */
322 for (i = order - 1; i >= 0; i--) {
323 int qstep = s->wb ? 9830 : 11796;
324
325 lsf_res[i] = lsf_i2[i] * 1024;
326 if (lsf_i2[i] < 0) lsf_res[i] += 102;
327 else if (lsf_i2[i] > 0) lsf_res[i] -= 102;
328 lsf_res[i] = (lsf_res[i] * qstep) >> 16;
329
330 if (i + 1 < order) {
331 int weight = s->wb ? ff_silk_lsf_pred_weights_wb [ff_silk_lsf_weight_sel_wb [lsf_i1][i]][i] :
332 ff_silk_lsf_pred_weights_nbmb[ff_silk_lsf_weight_sel_nbmb[lsf_i1][i]][i];
333 lsf_res[i] += (lsf_res[i+1] * weight) >> 8;
334 }
335 }
336
337 /* reconstruct the NLSF coefficients from the supplied indices */
338 for (i = 0; i < order; i++) {
339 const uint8_t * codebook = s->wb ? ff_silk_lsf_codebook_wb [lsf_i1] :
340 ff_silk_lsf_codebook_nbmb[lsf_i1];
341 int cur, prev, next, weight_sq, weight, ipart, fpart, y, value;
342
343 /* find the weight of the residual */
344 /* TODO: precompute */
345 cur = codebook[i];
346 prev = i ? codebook[i - 1] : 0;
347 next = i + 1 < order ? codebook[i + 1] : 256;
348 weight_sq = (1024 / (cur - prev) + 1024 / (next - cur)) << 16;
349
350 /* approximate square-root with mandated fixed-point arithmetic */
351 ipart = opus_ilog(weight_sq);
352 fpart = (weight_sq >> (ipart-8)) & 127;
353 y = ((ipart & 1) ? 32768 : 46214) >> ((32 - ipart)>>1);
354 weight = y + ((213 * fpart * y) >> 16);
355
356 value = cur * 128 + (lsf_res[i] * 16384) / weight;
357 nlsf[i] = av_clip_uintp2(value, 15);
358 }
359
360 /* stabilize the NLSF coefficients */
361 silk_stabilize_lsf(nlsf, order, s->wb ? ff_silk_lsf_min_spacing_wb :
362 ff_silk_lsf_min_spacing_nbmb);
363
364 /* produce an interpolation for the first 2 subframes, */
365 /* and then convert both sets of NLSFs to LPC coefficients */
366 *has_lpc_leadin = 0;
367 if (s->subframes == 4) {
368 int offset = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_interpolation_offset);
369 if (offset != 4 && frame->coded) {
370 *has_lpc_leadin = 1;
371 if (offset != 0) {
372 int16_t nlsf_leadin[16];
373 for (i = 0; i < order; i++)
374 nlsf_leadin[i] = frame->nlsf[i] +
375 ((nlsf[i] - frame->nlsf[i]) * offset >> 2);
376 silk_lsf2lpc(nlsf_leadin, lpc_leadin, order);
377 } else /* avoid re-computation for a (roughly) 1-in-4 occurrence */
378 memcpy(lpc_leadin, frame->lpc, 16 * sizeof(float));
379 } else
380 offset = 4;
381 s->nlsf_interp_factor = offset;
382
383 silk_lsf2lpc(nlsf, lpc, order);
384 } else {
385 s->nlsf_interp_factor = 4;
386 silk_lsf2lpc(nlsf, lpc, order);
387 }
388
389 memcpy(frame->nlsf, nlsf, order * sizeof(nlsf[0]));
390 memcpy(frame->lpc, lpc, order * sizeof(lpc[0]));
391}
392
393static inline void silk_count_children(OpusRangeCoder *rc, int model, int32_t total,
394 int32_t child[2])
395{
396 if (total != 0) {
397 child[0] = ff_opus_rc_dec_cdf(rc,
398 ff_silk_model_pulse_location[model] + (((total - 1 + 5) * (total - 1)) >> 1));
399 child[1] = total - child[0];
400 } else {
401 child[0] = 0;
402 child[1] = 0;
403 }
404}
405
406static inline void silk_decode_excitation(SilkContext *s, OpusRangeCoder *rc,
407 float* excitationf,
408 int qoffset_high, int active, int voiced)
409{
410 int i;
411 uint32_t seed;
412 int shellblocks;
413 int ratelevel;
414 uint8_t pulsecount[20]; // total pulses in each shell block
415 uint8_t lsbcount[20] = {0}; // raw lsbits defined for each pulse in each shell block
416 int32_t excitation[320]; // Q23
417
418 /* excitation parameters */
419 seed = ff_opus_rc_dec_cdf(rc, ff_silk_model_lcg_seed);
420 shellblocks = ff_silk_shell_blocks[s->bandwidth][s->subframes >> 2];
421 ratelevel = ff_opus_rc_dec_cdf(rc, ff_silk_model_exc_rate[voiced]);
422
423 for (i = 0; i < shellblocks; i++) {
424 pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[ratelevel]);
425 if (pulsecount[i] == 17) {
426 while (pulsecount[i] == 17 && ++lsbcount[i] != 10)
427 pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[9]);
428 if (lsbcount[i] == 10)
429 pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[10]);
430 }
431 }
432
433 /* decode pulse locations using PVQ */
434 for (i = 0; i < shellblocks; i++) {
435 if (pulsecount[i] != 0) {
436 int a, b, c, d;
437 int32_t * location = excitation + 16*i;
438 int32_t branch[4][2];
439 branch[0][0] = pulsecount[i];
440
441 /* unrolled tail recursion */
442 for (a = 0; a < 1; a++) {
443 silk_count_children(rc, 0, branch[0][a], branch[1]);
444 for (b = 0; b < 2; b++) {
445 silk_count_children(rc, 1, branch[1][b], branch[2]);
446 for (c = 0; c < 2; c++) {
447 silk_count_children(rc, 2, branch[2][c], branch[3]);
448 for (d = 0; d < 2; d++) {
449 silk_count_children(rc, 3, branch[3][d], location);
450 location += 2;
451 }
452 }
453 }
454 }
455 } else
456 memset(excitation + 16*i, 0, 16*sizeof(int32_t));
457 }
458
459 /* decode least significant bits */
460 for (i = 0; i < shellblocks << 4; i++) {
461 int bit;
462 for (bit = 0; bit < lsbcount[i >> 4]; bit++)
463 excitation[i] = (excitation[i] << 1) |
464 ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_lsb);
465 }
466
467 /* decode signs */
468 for (i = 0; i < shellblocks << 4; i++) {
469 if (excitation[i] != 0) {
470 int sign = ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_sign[active +
471 voiced][qoffset_high][FFMIN(pulsecount[i >> 4], 6)]);
472 if (sign == 0)
473 excitation[i] *= -1;
474 }
475 }
476
477 /* assemble the excitation */
478 for (i = 0; i < shellblocks << 4; i++) {
479 int value = excitation[i];
480 excitation[i] = value * 256 | ff_silk_quant_offset[voiced][qoffset_high];
481 if (value < 0) excitation[i] += 20;
482 else if (value > 0) excitation[i] -= 20;
483
484 /* invert samples pseudorandomly */
485 seed = 196314165 * seed + 907633515;
486 if (seed & 0x80000000)
487 excitation[i] *= -1;
488 seed += value;
489
490 excitationf[i] = excitation[i] / 8388608.0f;
491 }
492}
493
494/** Maximum residual history according to 4.2.7.6.1 */
495#define SILK_MAX_LAG (288 + LTP_ORDER / 2)
496
497/** Order of the LTP filter */
498#define LTP_ORDER 5
499
500static void silk_decode_frame(SilkContext *s, OpusRangeCoder *rc,
501 int frame_num, int channel, int coded_channels, int active, int active1)
502{
503 /* per frame */
504 int voiced; // combines with active to indicate inactive, active, or active+voiced
505 int qoffset_high;
506 int order; // order of the LPC coefficients
507 float lpc_leadin[16], lpc_body[16], residual[SILK_MAX_LAG + SILK_HISTORY];
508 int has_lpc_leadin;
509 float ltpscale;
510
511 /* per subframe */
512 struct {
513 float gain;
514 int pitchlag;
515 float ltptaps[5];
516 } sf[4];
517
518 SilkFrame * const frame = s->frame + channel;
519
520 int i;
521
522 /* obtain stereo weights */
523 if (coded_channels == 2 && channel == 0) {
524 int n, wi[2], ws[2], w[2];
525 n = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s1);
526 wi[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n / 5);
527 ws[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
528 wi[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n % 5);
529 ws[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
530
531 for (i = 0; i < 2; i++)
532 w[i] = ff_silk_stereo_weights[wi[i]] +
533 (((ff_silk_stereo_weights[wi[i] + 1] - ff_silk_stereo_weights[wi[i]]) * 6554) >> 16)
534 * (ws[i]*2 + 1);
535
536 s->stereo_weights[0] = (w[0] - w[1]) / 8192.0;
537 s->stereo_weights[1] = w[1] / 8192.0;
538
539 /* and read the mid-only flag */
540 s->midonly = active1 ? 0 : ff_opus_rc_dec_cdf(rc, ff_silk_model_mid_only);
541 }
542
543 /* obtain frame type */
544 if (!active) {
545 qoffset_high = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_inactive);
546 voiced = 0;
547 } else {
548 int type = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_active);
549 qoffset_high = type & 1;
550 voiced = type >> 1;
551 }
552
553 /* obtain subframe quantization gains */
554 for (i = 0; i < s->subframes; i++) {
555 int log_gain; //Q7
556 int ipart, fpart, lingain;
557
558 if (i == 0 && (frame_num == 0 || !frame->coded)) {
559 /* gain is coded absolute */
560 int x = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_highbits[active + voiced]);
561 log_gain = (x<<3) | ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_lowbits);
562
563 if (frame->coded)
564 log_gain = FFMAX(log_gain, frame->log_gain - 16);
565 } else {
566 /* gain is coded relative */
567 int delta_gain = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_delta);
568 log_gain = av_clip_uintp2(FFMAX((delta_gain<<1) - 16,
569 frame->log_gain + delta_gain - 4), 6);
570 }
571
572 frame->log_gain = log_gain;
573
574 /* approximate 2**(x/128) with a Q7 (i.e. non-integer) input */
575 log_gain = (log_gain * 0x1D1C71 >> 16) + 2090;
576 ipart = log_gain >> 7;
577 fpart = log_gain & 127;
578 lingain = (1 << ipart) + ((-174 * fpart * (128-fpart) >>16) + fpart) * ((1<<ipart) >> 7);
579 sf[i].gain = lingain / 65536.0f;
580 }
581
582 /* obtain LPC filter coefficients */
583 silk_decode_lpc(s, frame, rc, lpc_leadin, lpc_body, &order, &has_lpc_leadin, voiced);
584
585 /* obtain pitch lags, if this is a voiced frame */
586 if (voiced) {
587 int lag_absolute = (!frame_num || !frame->prev_voiced);
588 int primarylag; // primary pitch lag for the entire SILK frame
589 int ltpfilter;
590 const int8_t * offsets;
591
592 if (!lag_absolute) {
593 int delta = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_delta);
594 if (delta)
595 primarylag = frame->primarylag + delta - 9;
596 else
597 lag_absolute = 1;
598 }
599
600 if (lag_absolute) {
601 /* primary lag is coded absolute */
602 int highbits, lowbits;
603 static const uint16_t *model[] = {
604 ff_silk_model_pitch_lowbits_nb, ff_silk_model_pitch_lowbits_mb,
605 ff_silk_model_pitch_lowbits_wb
606 };
607 highbits = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_highbits);
608 lowbits = ff_opus_rc_dec_cdf(rc, model[s->bandwidth]);
609
610 primarylag = ff_silk_pitch_min_lag[s->bandwidth] +
611 highbits*ff_silk_pitch_scale[s->bandwidth] + lowbits;
612 }
613 frame->primarylag = primarylag;
614
615 if (s->subframes == 2)
616 offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
617 ? ff_silk_pitch_offset_nb10ms[ff_opus_rc_dec_cdf(rc,
618 ff_silk_model_pitch_contour_nb10ms)]
619 : ff_silk_pitch_offset_mbwb10ms[ff_opus_rc_dec_cdf(rc,
620 ff_silk_model_pitch_contour_mbwb10ms)];
621 else
622 offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
623 ? ff_silk_pitch_offset_nb20ms[ff_opus_rc_dec_cdf(rc,
624 ff_silk_model_pitch_contour_nb20ms)]
625 : ff_silk_pitch_offset_mbwb20ms[ff_opus_rc_dec_cdf(rc,
626 ff_silk_model_pitch_contour_mbwb20ms)];
627
628 for (i = 0; i < s->subframes; i++)
629 sf[i].pitchlag = av_clip(primarylag + offsets[i],
630 ff_silk_pitch_min_lag[s->bandwidth],
631 ff_silk_pitch_max_lag[s->bandwidth]);
632
633 /* obtain LTP filter coefficients */
634 ltpfilter = ff_opus_rc_dec_cdf(rc, ff_silk_model_ltp_filter);
635 for (i = 0; i < s->subframes; i++) {
636 int index, j;
637 static const uint16_t *filter_sel[] = {
638 ff_silk_model_ltp_filter0_sel, ff_silk_model_ltp_filter1_sel,
639 ff_silk_model_ltp_filter2_sel
640 };
641 static const int8_t (*filter_taps[])[5] = {
642 ff_silk_ltp_filter0_taps, ff_silk_ltp_filter1_taps, ff_silk_ltp_filter2_taps
643 };
644 index = ff_opus_rc_dec_cdf(rc, filter_sel[ltpfilter]);
645 for (j = 0; j < 5; j++)
646 sf[i].ltptaps[j] = filter_taps[ltpfilter][index][j] / 128.0f;
647 }
648 }
649
650 /* obtain LTP scale factor */
651 if (voiced && frame_num == 0)
652 ltpscale = ff_silk_ltp_scale_factor[ff_opus_rc_dec_cdf(rc,
653 ff_silk_model_ltp_scale_index)] / 16384.0f;
654 else ltpscale = 15565.0f/16384.0f;
655
656 /* generate the excitation signal for the entire frame */
657 silk_decode_excitation(s, rc, residual + SILK_MAX_LAG, qoffset_high,
658 active, voiced);
659
660 /* skip synthesising the side channel if we want mono-only */
661 if (s->output_channels == channel)
662 return;
663
664 /* generate the output signal */
665 for (i = 0; i < s->subframes; i++) {
666 const float * lpc_coeff = (i < 2 && has_lpc_leadin) ? lpc_leadin : lpc_body;
667 float *dst = frame->output + SILK_HISTORY + i * s->sflength;
668 float *resptr = residual + SILK_MAX_LAG + i * s->sflength;
669 float *lpc = frame->lpc_history + SILK_HISTORY + i * s->sflength;
670 float sum;
671 int j, k;
672
673 if (voiced) {
674 int out_end;
675 float scale;
676
677 if (i < 2 || s->nlsf_interp_factor == 4) {
678 out_end = -i * s->sflength;
679 scale = ltpscale;
680 } else {
681 out_end = -(i - 2) * s->sflength;
682 scale = 1.0f;
683 }
684
685 /* when the LPC coefficients change, a re-whitening filter is used */
686 /* to produce a residual that accounts for the change */
687 for (j = - sf[i].pitchlag - LTP_ORDER/2; j < out_end; j++) {
688 sum = dst[j];
689 for (k = 0; k < order; k++)
690 sum -= lpc_coeff[k] * dst[j - k - 1];
691 resptr[j] = av_clipf(sum, -1.0f, 1.0f) * scale / sf[i].gain;
692 }
693
694 if (out_end) {
695 float rescale = sf[i-1].gain / sf[i].gain;
696 for (j = out_end; j < 0; j++)
697 resptr[j] *= rescale;
698 }
699
700 /* LTP synthesis */
701 for (j = 0; j < s->sflength; j++) {
702 sum = resptr[j];
703 for (k = 0; k < LTP_ORDER; k++)
704 sum += sf[i].ltptaps[k] * resptr[j - sf[i].pitchlag + LTP_ORDER/2 - k];
705 resptr[j] = sum;
706 }
707 }
708
709 /* LPC synthesis */
710 for (j = 0; j < s->sflength; j++) {
711 sum = resptr[j] * sf[i].gain;
712 for (k = 1; k <= order; k++)
713 sum += lpc_coeff[k - 1] * lpc[j - k];
714
715 lpc[j] = sum;
716 dst[j] = av_clipf(sum, -1.0f, 1.0f);
717 }
718 }
719
720 frame->prev_voiced = voiced;
721 memmove(frame->lpc_history, frame->lpc_history + s->flength, SILK_HISTORY * sizeof(float));
722 memmove(frame->output, frame->output + s->flength, SILK_HISTORY * sizeof(float));
723
724 frame->coded = 1;
725}
726
727static void silk_unmix_ms(SilkContext *s, float *l, float *r)
728{
729 float *mid = s->frame[0].output + SILK_HISTORY - s->flength;
730 float *side = s->frame[1].output + SILK_HISTORY - s->flength;
731 float w0_prev = s->prev_stereo_weights[0];
732 float w1_prev = s->prev_stereo_weights[1];
733 float w0 = s->stereo_weights[0];
734 float w1 = s->stereo_weights[1];
735 int n1 = ff_silk_stereo_interp_len[s->bandwidth];
736 int i;
737
738 for (i = 0; i < n1; i++) {
739 float interp0 = w0_prev + i * (w0 - w0_prev) / n1;
740 float interp1 = w1_prev + i * (w1 - w1_prev) / n1;
741 float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
742
743 l[i] = av_clipf((1 + interp1) * mid[i - 1] + side[i - 1] + interp0 * p0, -1.0, 1.0);
744 r[i] = av_clipf((1 - interp1) * mid[i - 1] - side[i - 1] - interp0 * p0, -1.0, 1.0);
745 }
746
747 for (; i < s->flength; i++) {
748 float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
749
750 l[i] = av_clipf((1 + w1) * mid[i - 1] + side[i - 1] + w0 * p0, -1.0, 1.0);
751 r[i] = av_clipf((1 - w1) * mid[i - 1] - side[i - 1] - w0 * p0, -1.0, 1.0);
752 }
753
754 memcpy(s->prev_stereo_weights, s->stereo_weights, sizeof(s->stereo_weights));
755}
756
757static void silk_flush_frame(SilkFrame *frame)
758{
759 if (!frame->coded)
760 return;
761
762 memset(frame->output, 0, sizeof(frame->output));
763 memset(frame->lpc_history, 0, sizeof(frame->lpc_history));
764
765 memset(frame->lpc, 0, sizeof(frame->lpc));
766 memset(frame->nlsf, 0, sizeof(frame->nlsf));
767
768 frame->log_gain = 0;
769
770 frame->primarylag = 0;
771 frame->prev_voiced = 0;
772 frame->coded = 0;
773}
774
775int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
776 float *output[2],
777 enum OpusBandwidth bandwidth,
778 int coded_channels,
779 int duration_ms)
780{
781 int active[2][6], redundancy[2];
782 int nb_frames, i, j;
783
784 if (bandwidth > OPUS_BANDWIDTH_WIDEBAND ||
785 coded_channels > 2 || duration_ms > 60) {
786 av_log(s->avctx, AV_LOG_ERROR, "Invalid parameters passed "
787 "to the SILK decoder.\n");
788 return AVERROR(EINVAL);
789 }
790
791 nb_frames = 1 + (duration_ms > 20) + (duration_ms > 40);
792 s->subframes = duration_ms / nb_frames / 5; // 5ms subframes
793 s->sflength = 20 * (bandwidth + 2);
794 s->flength = s->sflength * s->subframes;
795 s->bandwidth = bandwidth;
796 s->wb = bandwidth == OPUS_BANDWIDTH_WIDEBAND;
797
798 /* make sure to flush the side channel when switching from mono to stereo */
799 if (coded_channels > s->prev_coded_channels)
800 silk_flush_frame(&s->frame[1]);
801 s->prev_coded_channels = coded_channels;
802
803 /* read the LP-layer header bits */
804 for (i = 0; i < coded_channels; i++) {
805 for (j = 0; j < nb_frames; j++)
806 active[i][j] = ff_opus_rc_dec_log(rc, 1);
807
808 redundancy[i] = ff_opus_rc_dec_log(rc, 1);
809 if (redundancy[i]) {
810 avpriv_report_missing_feature(s->avctx, "LBRR frames");
811 return AVERROR_PATCHWELCOME;
812 }
813 }
814
815 for (i = 0; i < nb_frames; i++) {
816 for (j = 0; j < coded_channels && !s->midonly; j++)
817 silk_decode_frame(s, rc, i, j, coded_channels, active[j][i], active[1][i]);
818
819 /* reset the side channel if it is not coded */
820 if (s->midonly && s->frame[1].coded)
821 silk_flush_frame(&s->frame[1]);
822
823 if (coded_channels == 1 || s->output_channels == 1) {
824 for (j = 0; j < s->output_channels; j++) {
825 memcpy(output[j] + i * s->flength,
826 s->frame[0].output + SILK_HISTORY - s->flength - 2,
827 s->flength * sizeof(float));
828 }
829 } else {
830 silk_unmix_ms(s, output[0] + i * s->flength, output[1] + i * s->flength);
831 }
832
833 s->midonly = 0;
834 }
835
836 return nb_frames * s->flength;
837}
838
839void ff_silk_free(SilkContext **ps)
840{
841 av_freep(ps);
842}
843
844void ff_silk_flush(SilkContext *s)
845{
846 silk_flush_frame(&s->frame[0]);
847 silk_flush_frame(&s->frame[1]);
848
849 memset(s->prev_stereo_weights, 0, sizeof(s->prev_stereo_weights));
850}
851
852int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)
853{
854 SilkContext *s;
855
856 if (output_channels != 1 && output_channels != 2) {
857 av_log(avctx, AV_LOG_ERROR, "Invalid number of output channels: %d\n",
858 output_channels);
859 return AVERROR(EINVAL);
860 }
861
862 s = av_mallocz(sizeof(*s));
863 if (!s)
864 return AVERROR(ENOMEM);
865
866 s->avctx = avctx;
867 s->output_channels = output_channels;
868
869 ff_silk_flush(s);
870
871 *ps = s;
872
873 return 0;
874}
875
generated by cgit at 2024-05-17 20:31:09 (GMT)