aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/g729postfilter.c
blob: 26e937f0baf454370fa33d0b115d4a91297d0858 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
/*
 * G.729, G729 Annex D postfilter
 * Copyright (c) 2008 Vladimir Voroshilov
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <stdint.h>
#include <string.h>

#include "libavutil/common.h"
#include "libavutil/intmath.h"

#include "audiodsp.h"
#include "g729.h"
#include "g729postfilter.h"
#include "celp_math.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "celp_filters.h"

#define FRAC_BITS 15
#include "mathops.h"

/**
 * short interpolation filter (of length 33, according to spec)
 * for computing signal with non-integer delay
 */
static const int16_t ff_g729_interp_filt_short[(ANALYZED_FRAC_DELAYS+1)*SHORT_INT_FILT_LEN] = {
      0, 31650, 28469, 23705, 18050, 12266,  7041,  2873,
      0, -1597, -2147, -1992, -1492,  -933,  -484,  -188,
};

/**
 * long interpolation filter (of length 129, according to spec)
 * for computing signal with non-integer delay
 */
static const int16_t ff_g729_interp_filt_long[(ANALYZED_FRAC_DELAYS+1)*LONG_INT_FILT_LEN] = {
   0, 31915, 29436, 25569, 20676, 15206,  9639,  4439,
   0, -3390, -5579, -6549, -6414, -5392, -3773, -1874,
   0,  1595,  2727,  3303,  3319,  2850,  2030,  1023,
   0,  -887, -1527, -1860, -1876, -1614, -1150,  -579,
   0,   501,   859,  1041,  1044,   892,   631,   315,
   0,  -266,  -453,  -543,  -538,  -455,  -317,  -156,
   0,   130,   218,   258,   253,   212,   147,    72,
   0,   -59,  -101,  -122,  -123,  -106,   -77,   -40,
};

/**
 * formant_pp_factor_num_pow[i] = FORMANT_PP_FACTOR_NUM^(i+1)
 */
static const int16_t formant_pp_factor_num_pow[10]= {
  /* (0.15) */
  18022, 9912, 5451, 2998, 1649, 907, 499, 274, 151, 83
};

/**
 * formant_pp_factor_den_pow[i] = FORMANT_PP_FACTOR_DEN^(i+1)
 */
static const int16_t formant_pp_factor_den_pow[10] = {
  /* (0.15) */
  22938, 16057, 11240, 7868, 5508, 3856, 2699, 1889, 1322, 925
};

/**
 * \brief Residual signal calculation (4.2.1 if G.729)
 * \param out [out] output data filtered through A(z/FORMANT_PP_FACTOR_NUM)
 * \param filter_coeffs (3.12) A(z/FORMANT_PP_FACTOR_NUM) filter coefficients
 * \param in input speech data to process
 * \param subframe_size size of one subframe
 *
 * \note in buffer must contain 10 items of previous speech data before top of the buffer
 * \remark It is safe to pass the same buffer for input and output.
 */
static void residual_filter(int16_t* out, const int16_t* filter_coeffs, const int16_t* in,
                            int subframe_size)
{
    int i, n;

    for (n = subframe_size - 1; n >= 0; n--) {
        int sum = 0x800;
        for (i = 0; i < 10; i++)
            sum += filter_coeffs[i] * in[n - i - 1];

        out[n] = in[n] + (sum >> 12);
    }
}

/**
 * \brief long-term postfilter (4.2.1)
 * \param dsp initialized DSP context
 * \param pitch_delay_int integer part of the pitch delay in the first subframe
 * \param residual filtering input data
 * \param residual_filt [out] speech signal with applied A(z/FORMANT_PP_FACTOR_NUM) filter
 * \param subframe_size size of subframe
 *
 * \return 0 if long-term prediction gain is less than 3dB, 1 -  otherwise
 */
static int16_t long_term_filter(AudioDSPContext *adsp, int pitch_delay_int,
                                const int16_t* residual, int16_t *residual_filt,
                                int subframe_size)
{
    int i, k, tmp, tmp2;
    int sum;
    int L_temp0;
    int L_temp1;
    int64_t L64_temp0;
    int64_t L64_temp1;
    int16_t shift;
    int corr_int_num, corr_int_den;

    int ener;
    int16_t sh_ener;

    int16_t gain_num,gain_den; //selected signal's gain numerator and denominator
    int16_t sh_gain_num, sh_gain_den;
    int gain_num_square;

    int16_t gain_long_num,gain_long_den; //filtered through long interpolation filter signal's gain numerator and denominator
    int16_t sh_gain_long_num, sh_gain_long_den;

    int16_t best_delay_int, best_delay_frac;

    int16_t delayed_signal_offset;
    int lt_filt_factor_a, lt_filt_factor_b;

    int16_t * selected_signal;
    const int16_t * selected_signal_const; //Necessary to avoid compiler warning

    int16_t sig_scaled[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
    int16_t delayed_signal[ANALYZED_FRAC_DELAYS][SUBFRAME_SIZE+1];
    int corr_den[ANALYZED_FRAC_DELAYS][2];

    tmp = 0;
    for(i=0; i<subframe_size + RES_PREV_DATA_SIZE; i++)
        tmp |= FFABS(residual[i]);

    if(!tmp)
        shift = 3;
    else
        shift = av_log2(tmp) - 11;

    if (shift > 0)
        for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++)
            sig_scaled[i] = residual[i] >> shift;
    else
        for (i = 0; i < subframe_size + RES_PREV_DATA_SIZE; i++)
            sig_scaled[i] = (unsigned)residual[i] << -shift;

    /* Start of best delay searching code */
    gain_num = 0;

    ener = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE,
                                    sig_scaled + RES_PREV_DATA_SIZE,
                                    subframe_size);
    if (ener) {
        sh_ener = av_log2(ener) - 14;
        sh_ener = FFMAX(sh_ener, 0);
        ener >>= sh_ener;
        /* Search for best pitch delay.

                       sum{ r(n) * r(k,n) ] }^2
           R'(k)^2 := -------------------------
                       sum{ r(k,n) * r(k,n) }


           R(T)    :=  sum{ r(n) * r(n-T) ] }


           where
           r(n-T) is integer delayed signal with delay T
           r(k,n) is non-integer delayed signal with integer delay best_delay
           and fractional delay k */

        /* Find integer delay best_delay which maximizes correlation R(T).

           This is also equals to numerator of R'(0),
           since the fine search (second step) is done with 1/8
           precision around best_delay. */
        corr_int_num = 0;
        best_delay_int = pitch_delay_int - 1;
        for (i = pitch_delay_int - 1; i <= pitch_delay_int + 1; i++) {
            sum = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE,
                                           sig_scaled + RES_PREV_DATA_SIZE - i,
                                           subframe_size);
            if (sum > corr_int_num) {
                corr_int_num = sum;
                best_delay_int = i;
            }
        }
        if (corr_int_num) {
            /* Compute denominator of pseudo-normalized correlation R'(0). */
            corr_int_den = adsp->scalarproduct_int16(sig_scaled + RES_PREV_DATA_SIZE - best_delay_int,
                                                     sig_scaled + RES_PREV_DATA_SIZE - best_delay_int,
                                                    subframe_size);

            /* Compute signals with non-integer delay k (with 1/8 precision),
               where k is in [0;6] range.
               Entire delay is qual to best_delay+(k+1)/8
               This is archieved by applying an interpolation filter of
               legth 33 to source signal. */
            for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
                ff_acelp_interpolate(&delayed_signal[k][0],
                                     &sig_scaled[RES_PREV_DATA_SIZE - best_delay_int],
                                     ff_g729_interp_filt_short,
                                     ANALYZED_FRAC_DELAYS+1,
                                     8 - k - 1,
                                     SHORT_INT_FILT_LEN,
                                     subframe_size + 1);
            }

            /* Compute denominator of pseudo-normalized correlation R'(k).

                 corr_den[k][0] is square root of R'(k) denominator, for int(T) == int(T0)
                 corr_den[k][1] is square root of R'(k) denominator, for int(T) == int(T0)+1

              Also compute maximum value of above denominators over all k. */
            tmp = corr_int_den;
            for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
                sum = adsp->scalarproduct_int16(&delayed_signal[k][1],
                                               &delayed_signal[k][1],
                                               subframe_size - 1);
                corr_den[k][0] = sum + delayed_signal[k][0            ] * delayed_signal[k][0            ];
                corr_den[k][1] = sum + delayed_signal[k][subframe_size] * delayed_signal[k][subframe_size];

                tmp = FFMAX3(tmp, corr_den[k][0], corr_den[k][1]);
            }

            sh_gain_den = av_log2(tmp) - 14;
            if (sh_gain_den >= 0) {

                sh_gain_num =  FFMAX(sh_gain_den, sh_ener);
                /* Loop through all k and find delay that maximizes
                   R'(k) correlation.
                   Search is done in [int(T0)-1; intT(0)+1] range
                   with 1/8 precision. */
                delayed_signal_offset = 1;
                best_delay_frac = 0;
                gain_den = corr_int_den >> sh_gain_den;
                gain_num = corr_int_num >> sh_gain_num;
                gain_num_square = gain_num * gain_num;
                for (k = 0; k < ANALYZED_FRAC_DELAYS; k++) {
                    for (i = 0; i < 2; i++) {
                        int16_t gain_num_short, gain_den_short;
                        int gain_num_short_square;
                        /* Compute numerator of pseudo-normalized
                           correlation R'(k). */
                        sum = adsp->scalarproduct_int16(&delayed_signal[k][i],
                                                       sig_scaled + RES_PREV_DATA_SIZE,
                                                       subframe_size);
                        gain_num_short = FFMAX(sum >> sh_gain_num, 0);

                        /*
                                      gain_num_short_square                gain_num_square
                           R'(T)^2 = -----------------------, max R'(T)^2= --------------
                                           den                                 gain_den
                        */
                        gain_num_short_square = gain_num_short * gain_num_short;
                        gain_den_short = corr_den[k][i] >> sh_gain_den;

                        tmp = MULL(gain_num_short_square, gain_den, FRAC_BITS);
                        tmp2 = MULL(gain_num_square, gain_den_short, FRAC_BITS);

                        // R'(T)^2 > max R'(T)^2
                        if (tmp > tmp2) {
                            gain_num = gain_num_short;
                            gain_den = gain_den_short;
                            gain_num_square = gain_num_short_square;
                            delayed_signal_offset = i;
                            best_delay_frac = k + 1;
                        }
                    }
                }

                /*
                       R'(T)^2
                  2 * --------- < 1
                        R(0)
                */
                L64_temp0 =  (int64_t)gain_num_square  << ((sh_gain_num << 1) + 1);
                L64_temp1 = ((int64_t)gain_den * ener) << (sh_gain_den + sh_ener);
                if (L64_temp0 < L64_temp1)
                    gain_num = 0;
            } // if(sh_gain_den >= 0)
        } // if(corr_int_num)
    } // if(ener)
    /* End of best delay searching code  */

    if (!gain_num) {
        memcpy(residual_filt, residual + RES_PREV_DATA_SIZE, subframe_size * sizeof(int16_t));

        /* Long-term prediction gain is less than 3dB. Long-term postfilter is disabled. */
        return 0;
    }
    if (best_delay_frac) {
        /* Recompute delayed signal with an interpolation filter of length 129. */
        ff_acelp_interpolate(residual_filt,
                             &sig_scaled[RES_PREV_DATA_SIZE - best_delay_int + delayed_signal_offset],
                             ff_g729_interp_filt_long,
                             ANALYZED_FRAC_DELAYS + 1,
                             8 - best_delay_frac,
                             LONG_INT_FILT_LEN,
                             subframe_size + 1);
        /* Compute R'(k) correlation's numerator. */
        sum = adsp->scalarproduct_int16(residual_filt,
                                       sig_scaled + RES_PREV_DATA_SIZE,
                                       subframe_size);

        if (sum < 0) {
            gain_long_num = 0;
            sh_gain_long_num = 0;
        } else {
            tmp = av_log2(sum) - 14;
            tmp = FFMAX(tmp, 0);
            sum >>= tmp;
            gain_long_num = sum;
            sh_gain_long_num = tmp;
        }

        /* Compute R'(k) correlation's denominator. */
        sum = adsp->scalarproduct_int16(residual_filt, residual_filt, subframe_size);

        tmp = av_log2(sum) - 14;
        tmp = FFMAX(tmp, 0);
        sum >>= tmp;
        gain_long_den = sum;
        sh_gain_long_den = tmp;

        /* Select between original and delayed signal.
           Delayed signal will be selected if it increases R'(k)
           correlation. */
        L_temp0 = gain_num * gain_num;
        L_temp0 = MULL(L_temp0, gain_long_den, FRAC_BITS);

        L_temp1 = gain_long_num * gain_long_num;
        L_temp1 = MULL(L_temp1, gain_den, FRAC_BITS);

        tmp = ((sh_gain_long_num - sh_gain_num) * 2) - (sh_gain_long_den - sh_gain_den);
        if (tmp > 0)
            L_temp0 >>= tmp;
        else
            L_temp1 >>= FFMIN(-tmp, 31);

        /* Check if longer filter increases the values of R'(k). */
        if (L_temp1 > L_temp0) {
            /* Select long filter. */
            selected_signal = residual_filt;
            gain_num = gain_long_num;
            gain_den = gain_long_den;
            sh_gain_num = sh_gain_long_num;
            sh_gain_den = sh_gain_long_den;
        } else
            /* Select short filter. */
            selected_signal = &delayed_signal[best_delay_frac-1][delayed_signal_offset];

        /* Rescale selected signal to original value. */
        if (shift > 0)
            for (i = 0; i < subframe_size; i++)
                selected_signal[i] *= 1 << shift;
        else
            for (i = 0; i < subframe_size; i++)
                selected_signal[i] >>= -shift;

        /* necessary to avoid compiler warning */
        selected_signal_const = selected_signal;
    } // if(best_delay_frac)
    else
        selected_signal_const = residual + RES_PREV_DATA_SIZE - (best_delay_int + 1 - delayed_signal_offset);
#ifdef G729_BITEXACT
    tmp = sh_gain_num - sh_gain_den;
    if (tmp > 0)
        gain_den >>= tmp;
    else
        gain_num >>= -tmp;

    if (gain_num > gain_den)
        lt_filt_factor_a = MIN_LT_FILT_FACTOR_A;
    else {
        gain_num >>= 2;
        gain_den >>= 1;
        lt_filt_factor_a = (gain_den << 15) / (gain_den + gain_num);
    }
#else
    L64_temp0 = (((int64_t)gain_num) << sh_gain_num) >> 1;
    L64_temp1 = ((int64_t)gain_den) << sh_gain_den;
    lt_filt_factor_a = FFMAX((L64_temp1 << 15) / (L64_temp1 + L64_temp0), MIN_LT_FILT_FACTOR_A);
#endif

    /* Filter through selected filter. */
    lt_filt_factor_b = 32767 - lt_filt_factor_a + 1;

    ff_acelp_weighted_vector_sum(residual_filt, residual + RES_PREV_DATA_SIZE,
                                 selected_signal_const,
                                 lt_filt_factor_a, lt_filt_factor_b,
                                 1<<14, 15, subframe_size);

    // Long-term prediction gain is larger than 3dB.
    return 1;
}

/**
 * \brief Calculate reflection coefficient for tilt compensation filter (4.2.3).
 * \param dsp initialized DSP context
 * \param lp_gn (3.12) coefficients of A(z/FORMANT_PP_FACTOR_NUM) filter
 * \param lp_gd (3.12) coefficients of A(z/FORMANT_PP_FACTOR_DEN) filter
 * \param speech speech to update
 * \param subframe_size size of subframe
 *
 * \return (3.12) reflection coefficient
 *
 * \remark The routine also calculates the gain term for the short-term
 *         filter (gf) and multiplies the speech data by 1/gf.
 *
 * \note All members of lp_gn, except 10-19 must be equal to zero.
 */
static int16_t get_tilt_comp(AudioDSPContext *adsp, int16_t *lp_gn,
                             const int16_t *lp_gd, int16_t* speech,
                             int subframe_size)
{
    int rh1,rh0; // (3.12)
    int temp;
    int i;
    int gain_term;

    lp_gn[10] = 4096; //1.0 in (3.12)

    /* Apply 1/A(z/FORMANT_PP_FACTOR_DEN) filter to hf. */
    ff_celp_lp_synthesis_filter(lp_gn + 11, lp_gd + 1, lp_gn + 11, 22, 10, 0, 0, 0x800);
    /* Now lp_gn (starting with 10) contains impulse response
       of A(z/FORMANT_PP_FACTOR_NUM)/A(z/FORMANT_PP_FACTOR_DEN) filter. */

    rh0 = adsp->scalarproduct_int16(lp_gn + 10, lp_gn + 10, 20);
    rh1 = adsp->scalarproduct_int16(lp_gn + 10, lp_gn + 11, 20);

    /* downscale to avoid overflow */
    temp = av_log2(rh0) - 14;
    if (temp > 0) {
        rh0 >>= temp;
        rh1 >>= temp;
    }

    if (FFABS(rh1) > rh0 || !rh0)
        return 0;

    gain_term = 0;
    for (i = 0; i < 20; i++)
        gain_term += FFABS(lp_gn[i + 10]);
    gain_term >>= 2; // (3.12) -> (5.10)

    if (gain_term > 0x400) { // 1.0 in (5.10)
        temp = 0x2000000 / gain_term; // 1.0/gain_term in (0.15)
        for (i = 0; i < subframe_size; i++)
            speech[i] = (speech[i] * temp + 0x4000) >> 15;
    }

    return -(rh1 * (1 << 15)) / rh0;
}

/**
 * \brief Apply tilt compensation filter (4.2.3).
 * \param res_pst [in/out] residual signal (partially filtered)
 * \param k1 (3.12) reflection coefficient
 * \param subframe_size size of subframe
 * \param ht_prev_data previous data for 4.2.3, equation 86
 *
 * \return new value for ht_prev_data
*/
static int16_t apply_tilt_comp(int16_t* out, int16_t* res_pst, int refl_coeff,
                               int subframe_size, int16_t ht_prev_data)
{
    int tmp, tmp2;
    int i;
    int gt, ga;
    int fact, sh_fact;

    if (refl_coeff > 0) {
        gt = (refl_coeff * G729_TILT_FACTOR_PLUS + 0x4000) >> 15;
        fact = 0x2000; // 0.5 in (0.15)
        sh_fact = 14;
    } else {
        gt = (refl_coeff * G729_TILT_FACTOR_MINUS + 0x4000) >> 15;
        fact = 0x400; // 0.5 in (3.12)
        sh_fact = 11;
    }
    ga = (fact << 16) / av_clip_int16(32768 - FFABS(gt));
    gt >>= 1;

    /* Apply tilt compensation filter to signal. */
    tmp = res_pst[subframe_size - 1];

    for (i = subframe_size - 1; i >= 1; i--) {
        tmp2 = (gt * res_pst[i-1]) * 2 + 0x4000;
        tmp2 = res_pst[i] + (tmp2 >> 15);

        tmp2 = (tmp2 * ga + fact) >> sh_fact;
        out[i] = tmp2;
    }
    tmp2 = (gt * ht_prev_data) * 2 + 0x4000;
    tmp2 = res_pst[0] + (tmp2 >> 15);
    tmp2 = (tmp2 * ga + fact) >> sh_fact;
    out[0] = tmp2;

    return tmp;
}

void ff_g729_postfilter(AudioDSPContext *adsp, int16_t* ht_prev_data, int* voicing,
                     const int16_t *lp_filter_coeffs, int pitch_delay_int,
                     int16_t* residual, int16_t* res_filter_data,
                     int16_t* pos_filter_data, int16_t *speech, int subframe_size)
{
    int16_t residual_filt_buf[SUBFRAME_SIZE+11];
    int16_t lp_gn[33]; // (3.12)
    int16_t lp_gd[11]; // (3.12)
    int tilt_comp_coeff;
    int i;

    /* Zero-filling is necessary for tilt-compensation filter. */
    memset(lp_gn, 0, 33 * sizeof(int16_t));

    /* Calculate A(z/FORMANT_PP_FACTOR_NUM) filter coefficients. */
    for (i = 0; i < 10; i++)
        lp_gn[i + 11] = (lp_filter_coeffs[i + 1] * formant_pp_factor_num_pow[i] + 0x4000) >> 15;

    /* Calculate A(z/FORMANT_PP_FACTOR_DEN) filter coefficients. */
    for (i = 0; i < 10; i++)
        lp_gd[i + 1] = (lp_filter_coeffs[i + 1] * formant_pp_factor_den_pow[i] + 0x4000) >> 15;

    /* residual signal calculation (one-half of short-term postfilter) */
    memcpy(speech - 10, res_filter_data, 10 * sizeof(int16_t));
    residual_filter(residual + RES_PREV_DATA_SIZE, lp_gn + 11, speech, subframe_size);
    /* Save data to use it in the next subframe. */
    memcpy(res_filter_data, speech + subframe_size - 10, 10 * sizeof(int16_t));

    /* long-term filter. If long-term prediction gain is larger than 3dB (returned value is
       nonzero) then declare current subframe as periodic. */
    i = long_term_filter(adsp, pitch_delay_int,
                                                residual, residual_filt_buf + 10,
                                                subframe_size);
    *voicing = FFMAX(*voicing, i);

    /* shift residual for using in next subframe */
    memmove(residual, residual + subframe_size, RES_PREV_DATA_SIZE * sizeof(int16_t));

    /* short-term filter tilt compensation */
    tilt_comp_coeff = get_tilt_comp(adsp, lp_gn, lp_gd, residual_filt_buf + 10, subframe_size);

    /* Apply second half of short-term postfilter: 1/A(z/FORMANT_PP_FACTOR_DEN) */
    ff_celp_lp_synthesis_filter(pos_filter_data + 10, lp_gd + 1,
                                residual_filt_buf + 10,
                                subframe_size, 10, 0, 0, 0x800);
    memcpy(pos_filter_data, pos_filter_data + subframe_size, 10 * sizeof(int16_t));

    *ht_prev_data = apply_tilt_comp(speech, pos_filter_data + 10, tilt_comp_coeff,
                                    subframe_size, *ht_prev_data);
}

/**
 * \brief Adaptive gain control (4.2.4)
 * \param gain_before gain of speech before applying postfilters
 * \param gain_after  gain of speech after applying postfilters
 * \param speech [in/out] signal buffer
 * \param subframe_size length of subframe
 * \param gain_prev (3.12) previous value of gain coefficient
 *
 * \return (3.12) last value of gain coefficient
 */
int16_t ff_g729_adaptive_gain_control(int gain_before, int gain_after, int16_t *speech,
                                   int subframe_size, int16_t gain_prev)
{
    int gain; // (3.12)
    int n;
    int exp_before, exp_after;

    if(!gain_after && gain_before)
        return 0;

    if (gain_before) {

        exp_before  = 14 - av_log2(gain_before);
        gain_before = bidir_sal(gain_before, exp_before);

        exp_after  = 14 - av_log2(gain_after);
        gain_after = bidir_sal(gain_after, exp_after);

        if (gain_before < gain_after) {
            gain = (gain_before << 15) / gain_after;
            gain = bidir_sal(gain, exp_after - exp_before - 1);
        } else {
            gain = ((gain_before - gain_after) << 14) / gain_after + 0x4000;
            gain = bidir_sal(gain, exp_after - exp_before);
        }
        gain = av_clip_int16(gain);
        gain = (gain * G729_AGC_FAC1 + 0x4000) >> 15; // gain * (1-0.9875)
    } else
        gain = 0;

    for (n = 0; n < subframe_size; n++) {
        // gain_prev = gain + 0.9875 * gain_prev
        gain_prev = (G729_AGC_FACTOR * gain_prev + 0x4000) >> 15;
        gain_prev = av_clip_int16(gain + gain_prev);
        speech[n] = av_clip_int16((speech[n] * gain_prev + 0x2000) >> 14);
    }
    return gain_prev;
}