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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 
32 typedef 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 
45 struct 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 
silk_stabilize_lsf(int16_t nlsf[16],int order,const uint16_t min_delta[17])65 static 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         nlsf[i] = FFMAX(nlsf[i], FFMIN(nlsf[i - 1] + min_delta[i], 32767));
132 
133     /* push backwards to increase distance */
134     if (nlsf[order-1] > 32768 - min_delta[order])
135         nlsf[order-1] = 32768 - min_delta[order];
136     for (i = order-2; i >= 0; i--)
137         if (nlsf[i] > nlsf[i + 1] - min_delta[i+1])
138             nlsf[i] = nlsf[i + 1] - min_delta[i+1];
139 
140     return;
141 }
142 
silk_is_lpc_stable(const int16_t lpc[16],int order)143 static inline int silk_is_lpc_stable(const int16_t lpc[16], int order)
144 {
145     int k, j, DC_resp = 0;
146     int32_t lpc32[2][16];       // Q24
147     int totalinvgain = 1 << 30; // 1.0 in Q30
148     int32_t *row = lpc32[0], *prevrow;
149 
150     /* initialize the first row for the Levinson recursion */
151     for (k = 0; k < order; k++) {
152         DC_resp += lpc[k];
153         row[k] = lpc[k] * 4096;
154     }
155 
156     if (DC_resp >= 4096)
157         return 0;
158 
159     /* check if prediction gain pushes any coefficients too far */
160     for (k = order - 1; 1; k--) {
161         int rc;      // Q31; reflection coefficient
162         int gaindiv; // Q30; inverse of the gain (the divisor)
163         int gain;    // gain for this reflection coefficient
164         int fbits;   // fractional bits used for the gain
165         int error;   // Q29; estimate of the error of our partial estimate of 1/gaindiv
166 
167         if (FFABS(row[k]) > 16773022)
168             return 0;
169 
170         rc      = -(row[k] * 128);
171         gaindiv = (1 << 30) - MULH(rc, rc);
172 
173         totalinvgain = MULH(totalinvgain, gaindiv) << 2;
174         if (k == 0)
175             return (totalinvgain >= 107374);
176 
177         /* approximate 1.0/gaindiv */
178         fbits = opus_ilog(gaindiv);
179         gain  = ((1 << 29) - 1) / (gaindiv >> (fbits + 1 - 16)); // Q<fbits-16>
180         error = (1 << 29) - MULL(gaindiv << (15 + 16 - fbits), gain, 16);
181         gain  = ((gain << 16) + (error * gain >> 13));
182 
183         /* switch to the next row of the LPC coefficients */
184         prevrow = row;
185         row = lpc32[k & 1];
186 
187         for (j = 0; j < k; j++) {
188             int x = av_sat_sub32(prevrow[j], ROUND_MULL(prevrow[k - j - 1], rc, 31));
189             int64_t tmp = ROUND_MULL(x, gain, fbits);
190 
191             /* per RFC 8251 section 6, if this calculation overflows, the filter
192                is considered unstable. */
193             if (tmp < INT32_MIN || tmp > INT32_MAX)
194                 return 0;
195 
196             row[j] = (int32_t)tmp;
197         }
198     }
199 }
200 
silk_lsp2poly(const int32_t lsp[],int32_t pol[],int half_order)201 static void silk_lsp2poly(const int32_t lsp[/* 2 * half_order - 1 */],
202                           int32_t pol[/* half_order + 1 */], int half_order)
203 {
204     int i, j;
205 
206     pol[0] = 65536; // 1.0 in Q16
207     pol[1] = -lsp[0];
208 
209     for (i = 1; i < half_order; i++) {
210         pol[i + 1] = pol[i - 1] * 2 - ROUND_MULL(lsp[2 * i], pol[i], 16);
211         for (j = i; j > 1; j--)
212             pol[j] += pol[j - 2] - ROUND_MULL(lsp[2 * i], pol[j - 1], 16);
213 
214         pol[1] -= lsp[2 * i];
215     }
216 }
217 
silk_lsf2lpc(const int16_t nlsf[16],float lpcf[16],int order)218 static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order)
219 {
220     int i, k;
221     int32_t lsp[16];     // Q17; 2*cos(LSF)
222     int32_t p[9], q[9];  // Q16
223     int32_t lpc32[16];   // Q17
224     int16_t lpc[16];     // Q12
225 
226     /* convert the LSFs to LSPs, i.e. 2*cos(LSF) */
227     for (k = 0; k < order; k++) {
228         int index = nlsf[k] >> 8;
229         int offset = nlsf[k] & 255;
230         int k2 = (order == 10) ? ff_silk_lsf_ordering_nbmb[k] : ff_silk_lsf_ordering_wb[k];
231 
232         /* interpolate and round */
233         lsp[k2]  = ff_silk_cosine[index] * 256;
234         lsp[k2] += (ff_silk_cosine[index + 1] - ff_silk_cosine[index]) * offset;
235         lsp[k2]  = (lsp[k2] + 4) >> 3;
236     }
237 
238     silk_lsp2poly(lsp    , p, order >> 1);
239     silk_lsp2poly(lsp + 1, q, order >> 1);
240 
241     /* reconstruct A(z) */
242     for (k = 0; k < order>>1; k++) {
243         int32_t p_tmp = p[k + 1] + p[k];
244         int32_t q_tmp = q[k + 1] - q[k];
245         lpc32[k]         = -q_tmp - p_tmp;
246         lpc32[order-k-1] =  q_tmp - p_tmp;
247     }
248 
249     /* limit the range of the LPC coefficients to each fit within an int16_t */
250     for (i = 0; i < 10; i++) {
251         int j;
252         unsigned int maxabs = 0;
253         for (j = 0, k = 0; j < order; j++) {
254             unsigned int x = FFABS(lpc32[k]);
255             if (x > maxabs) {
256                 maxabs = x; // Q17
257                 k      = j;
258             }
259         }
260 
261         maxabs = (maxabs + 16) >> 5; // convert to Q12
262 
263         if (maxabs > 32767) {
264             /* perform bandwidth expansion */
265             unsigned int chirp, chirp_base; // Q16
266             maxabs = FFMIN(maxabs, 163838); // anything above this overflows chirp's numerator
267             chirp_base = chirp = 65470 - ((maxabs - 32767) << 14) / ((maxabs * (k+1)) >> 2);
268 
269             for (k = 0; k < order; k++) {
270                 lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
271                 chirp    = (chirp_base * chirp + 32768) >> 16;
272             }
273         } else break;
274     }
275 
276     if (i == 10) {
277         /* time's up: just clamp */
278         for (k = 0; k < order; k++) {
279             int x = (lpc32[k] + 16) >> 5;
280             lpc[k] = av_clip_int16(x);
281             lpc32[k] = lpc[k] << 5; // shortcut mandated by the spec; drops lower 5 bits
282         }
283     } else {
284         for (k = 0; k < order; k++)
285             lpc[k] = (lpc32[k] + 16) >> 5;
286     }
287 
288     /* if the prediction gain causes the LPC filter to become unstable,
289        apply further bandwidth expansion on the Q17 coefficients */
290     for (i = 1; i <= 16 && !silk_is_lpc_stable(lpc, order); i++) {
291         unsigned int chirp, chirp_base;
292         chirp_base = chirp = 65536 - (1 << i);
293 
294         for (k = 0; k < order; k++) {
295             lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
296             lpc[k]   = (lpc32[k] + 16) >> 5;
297             chirp    = (chirp_base * chirp + 32768) >> 16;
298         }
299     }
300 
301     for (i = 0; i < order; i++)
302         lpcf[i] = lpc[i] / 4096.0f;
303 }
304 
silk_decode_lpc(SilkContext * s,SilkFrame * frame,OpusRangeCoder * rc,float lpc_leadin[16],float lpc[16],int * lpc_order,int * has_lpc_leadin,int voiced)305 static inline void silk_decode_lpc(SilkContext *s, SilkFrame *frame,
306                                    OpusRangeCoder *rc,
307                                    float lpc_leadin[16], float lpc[16],
308                                    int *lpc_order, int *has_lpc_leadin, int voiced)
309 {
310     int i;
311     int order;                   // order of the LP polynomial; 10 for NB/MB and 16 for WB
312     int8_t  lsf_i1, lsf_i2[16];  // stage-1 and stage-2 codebook indices
313     int16_t lsf_res[16];         // residual as a Q10 value
314     int16_t nlsf[16];            // Q15
315 
316     *lpc_order = order = s->wb ? 16 : 10;
317 
318     /* obtain LSF stage-1 and stage-2 indices */
319     lsf_i1 = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s1[s->wb][voiced]);
320     for (i = 0; i < order; i++) {
321         int index = s->wb ? ff_silk_lsf_s2_model_sel_wb  [lsf_i1][i] :
322                             ff_silk_lsf_s2_model_sel_nbmb[lsf_i1][i];
323         lsf_i2[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2[index]) - 4;
324         if (lsf_i2[i] == -4)
325             lsf_i2[i] -= ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
326         else if (lsf_i2[i] == 4)
327             lsf_i2[i] += ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
328     }
329 
330     /* reverse the backwards-prediction step */
331     for (i = order - 1; i >= 0; i--) {
332         int qstep = s->wb ? 9830 : 11796;
333 
334         lsf_res[i] = lsf_i2[i] * 1024;
335         if (lsf_i2[i] < 0)      lsf_res[i] += 102;
336         else if (lsf_i2[i] > 0) lsf_res[i] -= 102;
337         lsf_res[i] = (lsf_res[i] * qstep) >> 16;
338 
339         if (i + 1 < order) {
340             int weight = s->wb ? ff_silk_lsf_pred_weights_wb  [ff_silk_lsf_weight_sel_wb  [lsf_i1][i]][i] :
341                                  ff_silk_lsf_pred_weights_nbmb[ff_silk_lsf_weight_sel_nbmb[lsf_i1][i]][i];
342             lsf_res[i] += (lsf_res[i+1] * weight) >> 8;
343         }
344     }
345 
346     /* reconstruct the NLSF coefficients from the supplied indices */
347     for (i = 0; i < order; i++) {
348         const uint8_t * codebook = s->wb ? ff_silk_lsf_codebook_wb  [lsf_i1] :
349                                            ff_silk_lsf_codebook_nbmb[lsf_i1];
350         int cur, prev, next, weight_sq, weight, ipart, fpart, y, value;
351 
352         /* find the weight of the residual */
353         /* TODO: precompute */
354         cur = codebook[i];
355         prev = i ? codebook[i - 1] : 0;
356         next = i + 1 < order ? codebook[i + 1] : 256;
357         weight_sq = (1024 / (cur - prev) + 1024 / (next - cur)) << 16;
358 
359         /* approximate square-root with mandated fixed-point arithmetic */
360         ipart = opus_ilog(weight_sq);
361         fpart = (weight_sq >> (ipart-8)) & 127;
362         y = ((ipart & 1) ? 32768 : 46214) >> ((32 - ipart)>>1);
363         weight = y + ((213 * fpart * y) >> 16);
364 
365         value = cur * 128 + (lsf_res[i] * 16384) / weight;
366         nlsf[i] = av_clip_uintp2(value, 15);
367     }
368 
369     /* stabilize the NLSF coefficients */
370     silk_stabilize_lsf(nlsf, order, s->wb ? ff_silk_lsf_min_spacing_wb :
371                                             ff_silk_lsf_min_spacing_nbmb);
372 
373     /* produce an interpolation for the first 2 subframes, */
374     /* and then convert both sets of NLSFs to LPC coefficients */
375     *has_lpc_leadin = 0;
376     if (s->subframes == 4) {
377         int offset = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_interpolation_offset);
378         if (offset != 4 && frame->coded) {
379             *has_lpc_leadin = 1;
380             if (offset != 0) {
381                 int16_t nlsf_leadin[16];
382                 for (i = 0; i < order; i++)
383                     nlsf_leadin[i] = frame->nlsf[i] +
384                         ((nlsf[i] - frame->nlsf[i]) * offset >> 2);
385                 silk_lsf2lpc(nlsf_leadin, lpc_leadin, order);
386             } else  /* avoid re-computation for a (roughly) 1-in-4 occurrence */
387                 memcpy(lpc_leadin, frame->lpc, 16 * sizeof(float));
388         } else
389             offset = 4;
390         s->nlsf_interp_factor = offset;
391 
392         silk_lsf2lpc(nlsf, lpc, order);
393     } else {
394         s->nlsf_interp_factor = 4;
395         silk_lsf2lpc(nlsf, lpc, order);
396     }
397 
398     memcpy(frame->nlsf, nlsf, order * sizeof(nlsf[0]));
399     memcpy(frame->lpc,  lpc,  order * sizeof(lpc[0]));
400 }
401 
silk_count_children(OpusRangeCoder * rc,int model,int32_t total,int32_t child[2])402 static inline void silk_count_children(OpusRangeCoder *rc, int model, int32_t total,
403                                        int32_t child[2])
404 {
405     if (total != 0) {
406         child[0] = ff_opus_rc_dec_cdf(rc,
407                        ff_silk_model_pulse_location[model] + (((total - 1 + 5) * (total - 1)) >> 1));
408         child[1] = total - child[0];
409     } else {
410         child[0] = 0;
411         child[1] = 0;
412     }
413 }
414 
silk_decode_excitation(SilkContext * s,OpusRangeCoder * rc,float * excitationf,int qoffset_high,int active,int voiced)415 static inline void silk_decode_excitation(SilkContext *s, OpusRangeCoder *rc,
416                                           float* excitationf,
417                                           int qoffset_high, int active, int voiced)
418 {
419     int i;
420     uint32_t seed;
421     int shellblocks;
422     int ratelevel;
423     uint8_t pulsecount[20];     // total pulses in each shell block
424     uint8_t lsbcount[20] = {0}; // raw lsbits defined for each pulse in each shell block
425     int32_t excitation[320];    // Q23
426 
427     /* excitation parameters */
428     seed = ff_opus_rc_dec_cdf(rc, ff_silk_model_lcg_seed);
429     shellblocks = ff_silk_shell_blocks[s->bandwidth][s->subframes >> 2];
430     ratelevel = ff_opus_rc_dec_cdf(rc, ff_silk_model_exc_rate[voiced]);
431 
432     for (i = 0; i < shellblocks; i++) {
433         pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[ratelevel]);
434         if (pulsecount[i] == 17) {
435             while (pulsecount[i] == 17 && ++lsbcount[i] != 10)
436                 pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[9]);
437             if (lsbcount[i] == 10)
438                 pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[10]);
439         }
440     }
441 
442     /* decode pulse locations using PVQ */
443     for (i = 0; i < shellblocks; i++) {
444         if (pulsecount[i] != 0) {
445             int a, b, c, d;
446             int32_t * location = excitation + 16*i;
447             int32_t branch[4][2];
448             branch[0][0] = pulsecount[i];
449 
450             /* unrolled tail recursion */
451             for (a = 0; a < 1; a++) {
452                 silk_count_children(rc, 0, branch[0][a], branch[1]);
453                 for (b = 0; b < 2; b++) {
454                     silk_count_children(rc, 1, branch[1][b], branch[2]);
455                     for (c = 0; c < 2; c++) {
456                         silk_count_children(rc, 2, branch[2][c], branch[3]);
457                         for (d = 0; d < 2; d++) {
458                             silk_count_children(rc, 3, branch[3][d], location);
459                             location += 2;
460                         }
461                     }
462                 }
463             }
464         } else
465             memset(excitation + 16*i, 0, 16*sizeof(int32_t));
466     }
467 
468     /* decode least significant bits */
469     for (i = 0; i < shellblocks << 4; i++) {
470         int bit;
471         for (bit = 0; bit < lsbcount[i >> 4]; bit++)
472             excitation[i] = (excitation[i] << 1) |
473                             ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_lsb);
474     }
475 
476     /* decode signs */
477     for (i = 0; i < shellblocks << 4; i++) {
478         if (excitation[i] != 0) {
479             int sign = ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_sign[active +
480                                          voiced][qoffset_high][FFMIN(pulsecount[i >> 4], 6)]);
481             if (sign == 0)
482                 excitation[i] *= -1;
483         }
484     }
485 
486     /* assemble the excitation */
487     for (i = 0; i < shellblocks << 4; i++) {
488         int value = excitation[i];
489         excitation[i] = value * 256 | ff_silk_quant_offset[voiced][qoffset_high];
490         if (value < 0)      excitation[i] += 20;
491         else if (value > 0) excitation[i] -= 20;
492 
493         /* invert samples pseudorandomly */
494         seed = 196314165 * seed + 907633515;
495         if (seed & 0x80000000)
496             excitation[i] *= -1;
497         seed += value;
498 
499         excitationf[i] = excitation[i] / 8388608.0f;
500     }
501 }
502 
503 /** Maximum residual history according to 4.2.7.6.1 */
504 #define SILK_MAX_LAG  (288 + LTP_ORDER / 2)
505 
506 /** Order of the LTP filter */
507 #define LTP_ORDER 5
508 
silk_decode_frame(SilkContext * s,OpusRangeCoder * rc,int frame_num,int channel,int coded_channels,int active,int active1,int redundant)509 static void silk_decode_frame(SilkContext *s, OpusRangeCoder *rc,
510                               int frame_num, int channel, int coded_channels,
511                               int active, int active1, int redundant)
512 {
513     /* per frame */
514     int voiced;       // combines with active to indicate inactive, active, or active+voiced
515     int qoffset_high;
516     int order;                             // order of the LPC coefficients
517     float lpc_leadin[16], lpc_body[16], residual[SILK_MAX_LAG + SILK_HISTORY];
518     int has_lpc_leadin;
519     float ltpscale;
520 
521     /* per subframe */
522     struct {
523         float gain;
524         int pitchlag;
525         float ltptaps[5];
526     } sf[4];
527 
528     SilkFrame * const frame = s->frame + channel;
529 
530     int i;
531 
532     /* obtain stereo weights */
533     if (coded_channels == 2 && channel == 0) {
534         int n, wi[2], ws[2], w[2];
535         n     = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s1);
536         wi[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n / 5);
537         ws[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
538         wi[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n % 5);
539         ws[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
540 
541         for (i = 0; i < 2; i++)
542             w[i] = ff_silk_stereo_weights[wi[i]] +
543                    (((ff_silk_stereo_weights[wi[i] + 1] - ff_silk_stereo_weights[wi[i]]) * 6554) >> 16)
544                     * (ws[i]*2 + 1);
545 
546         s->stereo_weights[0] = (w[0] - w[1]) / 8192.0;
547         s->stereo_weights[1] = w[1]          / 8192.0;
548 
549         /* and read the mid-only flag */
550         s->midonly = active1 ? 0 : ff_opus_rc_dec_cdf(rc, ff_silk_model_mid_only);
551     }
552 
553     /* obtain frame type */
554     if (!active) {
555         qoffset_high = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_inactive);
556         voiced = 0;
557     } else {
558         int type = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_active);
559         qoffset_high = type & 1;
560         voiced = type >> 1;
561     }
562 
563     /* obtain subframe quantization gains */
564     for (i = 0; i < s->subframes; i++) {
565         int log_gain;     //Q7
566         int ipart, fpart, lingain;
567 
568         if (i == 0 && (frame_num == 0 || !frame->coded)) {
569             /* gain is coded absolute */
570             int x = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_highbits[active + voiced]);
571             log_gain = (x<<3) | ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_lowbits);
572 
573             if (frame->coded)
574                 log_gain = FFMAX(log_gain, frame->log_gain - 16);
575         } else {
576             /* gain is coded relative */
577             int delta_gain = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_delta);
578             log_gain = av_clip_uintp2(FFMAX((delta_gain<<1) - 16,
579                                      frame->log_gain + delta_gain - 4), 6);
580         }
581 
582         frame->log_gain = log_gain;
583 
584         /* approximate 2**(x/128) with a Q7 (i.e. non-integer) input */
585         log_gain = (log_gain * 0x1D1C71 >> 16) + 2090;
586         ipart = log_gain >> 7;
587         fpart = log_gain & 127;
588         lingain = (1 << ipart) + ((-174 * fpart * (128-fpart) >>16) + fpart) * ((1<<ipart) >> 7);
589         sf[i].gain = lingain / 65536.0f;
590     }
591 
592     /* obtain LPC filter coefficients */
593     silk_decode_lpc(s, frame, rc, lpc_leadin, lpc_body, &order, &has_lpc_leadin, voiced);
594 
595     /* obtain pitch lags, if this is a voiced frame */
596     if (voiced) {
597         int lag_absolute = (!frame_num || !frame->prev_voiced);
598         int primarylag;         // primary pitch lag for the entire SILK frame
599         int ltpfilter;
600         const int8_t * offsets;
601 
602         if (!lag_absolute) {
603             int delta = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_delta);
604             if (delta)
605                 primarylag = frame->primarylag + delta - 9;
606             else
607                 lag_absolute = 1;
608         }
609 
610         if (lag_absolute) {
611             /* primary lag is coded absolute */
612             int highbits, lowbits;
613             static const uint16_t * const model[] = {
614                 ff_silk_model_pitch_lowbits_nb, ff_silk_model_pitch_lowbits_mb,
615                 ff_silk_model_pitch_lowbits_wb
616             };
617             highbits = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_highbits);
618             lowbits  = ff_opus_rc_dec_cdf(rc, model[s->bandwidth]);
619 
620             primarylag = ff_silk_pitch_min_lag[s->bandwidth] +
621                          highbits*ff_silk_pitch_scale[s->bandwidth] + lowbits;
622         }
623         frame->primarylag = primarylag;
624 
625         if (s->subframes == 2)
626             offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
627                      ? ff_silk_pitch_offset_nb10ms[ff_opus_rc_dec_cdf(rc,
628                                                 ff_silk_model_pitch_contour_nb10ms)]
629                      : ff_silk_pitch_offset_mbwb10ms[ff_opus_rc_dec_cdf(rc,
630                                                 ff_silk_model_pitch_contour_mbwb10ms)];
631         else
632             offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
633                      ? ff_silk_pitch_offset_nb20ms[ff_opus_rc_dec_cdf(rc,
634                                                 ff_silk_model_pitch_contour_nb20ms)]
635                      : ff_silk_pitch_offset_mbwb20ms[ff_opus_rc_dec_cdf(rc,
636                                                 ff_silk_model_pitch_contour_mbwb20ms)];
637 
638         for (i = 0; i < s->subframes; i++)
639             sf[i].pitchlag = av_clip(primarylag + offsets[i],
640                                      ff_silk_pitch_min_lag[s->bandwidth],
641                                      ff_silk_pitch_max_lag[s->bandwidth]);
642 
643         /* obtain LTP filter coefficients */
644         ltpfilter = ff_opus_rc_dec_cdf(rc, ff_silk_model_ltp_filter);
645         for (i = 0; i < s->subframes; i++) {
646             int index, j;
647             static const uint16_t * const filter_sel[] = {
648                 ff_silk_model_ltp_filter0_sel, ff_silk_model_ltp_filter1_sel,
649                 ff_silk_model_ltp_filter2_sel
650             };
651             static const int8_t (* const filter_taps[])[5] = {
652                 ff_silk_ltp_filter0_taps, ff_silk_ltp_filter1_taps, ff_silk_ltp_filter2_taps
653             };
654             index = ff_opus_rc_dec_cdf(rc, filter_sel[ltpfilter]);
655             for (j = 0; j < 5; j++)
656                 sf[i].ltptaps[j] = filter_taps[ltpfilter][index][j] / 128.0f;
657         }
658     }
659 
660     /* obtain LTP scale factor */
661     if (voiced && frame_num == 0)
662         ltpscale = ff_silk_ltp_scale_factor[ff_opus_rc_dec_cdf(rc,
663                                          ff_silk_model_ltp_scale_index)] / 16384.0f;
664     else ltpscale = 15565.0f/16384.0f;
665 
666     /* generate the excitation signal for the entire frame */
667     silk_decode_excitation(s, rc, residual + SILK_MAX_LAG, qoffset_high,
668                            active, voiced);
669 
670     /* skip synthesising the output if we do not need it */
671     // TODO: implement error recovery
672     if (s->output_channels == channel || redundant)
673         return;
674 
675     /* generate the output signal */
676     for (i = 0; i < s->subframes; i++) {
677         const float * lpc_coeff = (i < 2 && has_lpc_leadin) ? lpc_leadin : lpc_body;
678         float *dst    = frame->output      + SILK_HISTORY + i * s->sflength;
679         float *resptr = residual           + SILK_MAX_LAG + i * s->sflength;
680         float *lpc    = frame->lpc_history + SILK_HISTORY + i * s->sflength;
681         float sum;
682         int j, k;
683 
684         if (voiced) {
685             int out_end;
686             float scale;
687 
688             if (i < 2 || s->nlsf_interp_factor == 4) {
689                 out_end = -i * s->sflength;
690                 scale   = ltpscale;
691             } else {
692                 out_end = -(i - 2) * s->sflength;
693                 scale   = 1.0f;
694             }
695 
696             /* when the LPC coefficients change, a re-whitening filter is used */
697             /* to produce a residual that accounts for the change */
698             for (j = - sf[i].pitchlag - LTP_ORDER/2; j < out_end; j++) {
699                 sum = dst[j];
700                 for (k = 0; k < order; k++)
701                     sum -= lpc_coeff[k] * dst[j - k - 1];
702                 resptr[j] = av_clipf(sum, -1.0f, 1.0f) * scale / sf[i].gain;
703             }
704 
705             if (out_end) {
706                 float rescale = sf[i-1].gain / sf[i].gain;
707                 for (j = out_end; j < 0; j++)
708                     resptr[j] *= rescale;
709             }
710 
711             /* LTP synthesis */
712             for (j = 0; j < s->sflength; j++) {
713                 sum = resptr[j];
714                 for (k = 0; k < LTP_ORDER; k++)
715                     sum += sf[i].ltptaps[k] * resptr[j - sf[i].pitchlag + LTP_ORDER/2 - k];
716                 resptr[j] = sum;
717             }
718         }
719 
720         /* LPC synthesis */
721         for (j = 0; j < s->sflength; j++) {
722             sum = resptr[j] * sf[i].gain;
723             for (k = 1; k <= order; k++)
724                 sum += lpc_coeff[k - 1] * lpc[j - k];
725 
726             lpc[j] = sum;
727             dst[j] = av_clipf(sum, -1.0f, 1.0f);
728         }
729     }
730 
731     frame->prev_voiced = voiced;
732     memmove(frame->lpc_history, frame->lpc_history + s->flength, SILK_HISTORY * sizeof(float));
733     memmove(frame->output,      frame->output      + s->flength, SILK_HISTORY * sizeof(float));
734 
735     frame->coded = 1;
736 }
737 
silk_unmix_ms(SilkContext * s,float * l,float * r)738 static void silk_unmix_ms(SilkContext *s, float *l, float *r)
739 {
740     float *mid    = s->frame[0].output + SILK_HISTORY - s->flength;
741     float *side   = s->frame[1].output + SILK_HISTORY - s->flength;
742     float w0_prev = s->prev_stereo_weights[0];
743     float w1_prev = s->prev_stereo_weights[1];
744     float w0      = s->stereo_weights[0];
745     float w1      = s->stereo_weights[1];
746     int n1        = ff_silk_stereo_interp_len[s->bandwidth];
747     int i;
748 
749     for (i = 0; i < n1; i++) {
750         float interp0 = w0_prev + i * (w0 - w0_prev) / n1;
751         float interp1 = w1_prev + i * (w1 - w1_prev) / n1;
752         float p0      = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
753 
754         l[i] = av_clipf((1 + interp1) * mid[i - 1] + side[i - 1] + interp0 * p0, -1.0, 1.0);
755         r[i] = av_clipf((1 - interp1) * mid[i - 1] - side[i - 1] - interp0 * p0, -1.0, 1.0);
756     }
757 
758     for (; i < s->flength; i++) {
759         float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
760 
761         l[i] = av_clipf((1 + w1) * mid[i - 1] + side[i - 1] + w0 * p0, -1.0, 1.0);
762         r[i] = av_clipf((1 - w1) * mid[i - 1] - side[i - 1] - w0 * p0, -1.0, 1.0);
763     }
764 
765     memcpy(s->prev_stereo_weights, s->stereo_weights, sizeof(s->stereo_weights));
766 }
767 
silk_flush_frame(SilkFrame * frame)768 static void silk_flush_frame(SilkFrame *frame)
769 {
770     if (!frame->coded)
771         return;
772 
773     memset(frame->output,      0, sizeof(frame->output));
774     memset(frame->lpc_history, 0, sizeof(frame->lpc_history));
775 
776     memset(frame->lpc,  0, sizeof(frame->lpc));
777     memset(frame->nlsf, 0, sizeof(frame->nlsf));
778 
779     frame->log_gain = 0;
780 
781     frame->primarylag  = 0;
782     frame->prev_voiced = 0;
783     frame->coded       = 0;
784 }
785 
ff_silk_decode_superframe(SilkContext * s,OpusRangeCoder * rc,float * output[2],enum OpusBandwidth bandwidth,int coded_channels,int duration_ms)786 int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
787                               float *output[2],
788                               enum OpusBandwidth bandwidth,
789                               int coded_channels,
790                               int duration_ms)
791 {
792     int active[2][6], redundancy[2];
793     int nb_frames, i, j;
794 
795     if (bandwidth > OPUS_BANDWIDTH_WIDEBAND ||
796         coded_channels > 2 || duration_ms > 60) {
797         av_log(s->avctx, AV_LOG_ERROR, "Invalid parameters passed "
798                "to the SILK decoder.\n");
799         return AVERROR(EINVAL);
800     }
801 
802     nb_frames = 1 + (duration_ms > 20) + (duration_ms > 40);
803     s->subframes = duration_ms / nb_frames / 5;         // 5ms subframes
804     s->sflength  = 20 * (bandwidth + 2);
805     s->flength   = s->sflength * s->subframes;
806     s->bandwidth = bandwidth;
807     s->wb        = bandwidth == OPUS_BANDWIDTH_WIDEBAND;
808 
809     /* make sure to flush the side channel when switching from mono to stereo */
810     if (coded_channels > s->prev_coded_channels)
811         silk_flush_frame(&s->frame[1]);
812     s->prev_coded_channels = coded_channels;
813 
814     /* read the LP-layer header bits */
815     for (i = 0; i < coded_channels; i++) {
816         for (j = 0; j < nb_frames; j++)
817             active[i][j] = ff_opus_rc_dec_log(rc, 1);
818 
819         redundancy[i] = ff_opus_rc_dec_log(rc, 1);
820     }
821 
822     /* read the per-frame LBRR flags */
823     for (i = 0; i < coded_channels; i++)
824         if (redundancy[i] && duration_ms > 20) {
825             redundancy[i] = ff_opus_rc_dec_cdf(rc, duration_ms == 40 ?
826                                                    ff_silk_model_lbrr_flags_40 : ff_silk_model_lbrr_flags_60);
827         }
828 
829     /* decode the LBRR frames */
830     for (i = 0; i < nb_frames; i++) {
831         for (j = 0; j < coded_channels; j++)
832             if (redundancy[j] & (1 << i)) {
833                 int active1 = (j == 0 && !(redundancy[1] & (1 << i))) ? 0 : 1;
834                 silk_decode_frame(s, rc, i, j, coded_channels, 1, active1, 1);
835             }
836     }
837 
838     for (i = 0; i < nb_frames; i++) {
839         for (j = 0; j < coded_channels && !s->midonly; j++)
840             silk_decode_frame(s, rc, i, j, coded_channels, active[j][i], active[1][i], 0);
841 
842         /* reset the side channel if it is not coded */
843         if (s->midonly && s->frame[1].coded)
844             silk_flush_frame(&s->frame[1]);
845 
846         if (coded_channels == 1 || s->output_channels == 1) {
847             for (j = 0; j < s->output_channels; j++) {
848                 memcpy(output[j] + i * s->flength,
849                        s->frame[0].output + SILK_HISTORY - s->flength - 2,
850                        s->flength * sizeof(float));
851             }
852         } else {
853             silk_unmix_ms(s, output[0] + i * s->flength, output[1] + i * s->flength);
854         }
855 
856         s->midonly        = 0;
857     }
858 
859     return nb_frames * s->flength;
860 }
861 
ff_silk_free(SilkContext ** ps)862 void ff_silk_free(SilkContext **ps)
863 {
864     av_freep(ps);
865 }
866 
ff_silk_flush(SilkContext * s)867 void ff_silk_flush(SilkContext *s)
868 {
869     silk_flush_frame(&s->frame[0]);
870     silk_flush_frame(&s->frame[1]);
871 
872     memset(s->prev_stereo_weights, 0, sizeof(s->prev_stereo_weights));
873 }
874 
ff_silk_init(AVCodecContext * avctx,SilkContext ** ps,int output_channels)875 int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)
876 {
877     SilkContext *s;
878 
879     if (output_channels != 1 && output_channels != 2) {
880         av_log(avctx, AV_LOG_ERROR, "Invalid number of output channels: %d\n",
881                output_channels);
882         return AVERROR(EINVAL);
883     }
884 
885     s = av_mallocz(sizeof(*s));
886     if (!s)
887         return AVERROR(ENOMEM);
888 
889     s->avctx           = avctx;
890     s->output_channels = output_channels;
891 
892     ff_silk_flush(s);
893 
894     *ps = s;
895 
896     return 0;
897 }
898