xref: /third_party/ffmpeg/libavcodec/opus_silk.c (revision cabdff1a)
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        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
143static 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
201static 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
218static 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
305static 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
402static 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
415static 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
509static 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
738static 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
768static 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
786int 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
862void ff_silk_free(SilkContext **ps)
863{
864    av_freep(ps);
865}
866
867void 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
875int 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