aboutsummaryrefslogblamecommitdiffstats
path: root/libavcodec/amrwbdec.c
blob: 941d065127a407250c8c9d06a28fef2e2d8bfa35 (plain) (tree)
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
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026
1027
1028
1029
1030
1031
1032
1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
1047
1048
1049
1050
1051
1052
1053
1054
1055
1056
1057
1058
1059
1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
1075
1076
1077
1078
1079
1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
1094
1095
1096
1097
1098
1099
1100
1101
1102
1103
1104
1105
1106
1107
1108
1109
1110
1111
1112
1113
1114
1115
1116
1117
1118
1119
1120
1121
1122
1123
1124
1125
1126
1127
1128
1129
1130
1131
1132
1133
1134
1135
1136
1137
1138
1139
1140
1141
1142
1143
1144
1145
1146
1147
1148
1149
1150
1151
1152
1153
1154
1155
1156
1157
1158
1159
1160
1161
1162
1163
1164
1165
1166
1167
1168
1169
1170
1171
1172
1173
1174
1175
1176
1177
1178
1179
1180
1181
1182
1183
1184
1185
1186
1187
1188
1189
1190
1191
1192
1193
1194
1195
1196
1197
1198
1199
1200
1201
1202
1203
1204
1205
1206
1207
1208
1209
1210
1211
1212
1213
1214
1215
1216
1217
1218
1219
1220
1221
1222
1223
1224
1225
1226
1227
1228
1229
1230
1231
1232
1233
1234
1235

















































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































































                                                                                                                                         
                                                                              
  
/*
 * AMR wideband decoder
 * Copyright (c) 2010 Marcelo Galvao Povoa
 *
 * 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
 */

/**
 * @file
 * AMR wideband decoder
 */

#include "libavutil/lfg.h"

#include "avcodec.h"
#include "get_bits.h"
#include "lsp.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"

#define AMR_USE_16BIT_TABLES
#include "amr.h"

#include "amrwbdata.h"

typedef struct {
    AMRWBFrame                             frame; ///< AMRWB parameters decoded from bitstream
    enum Mode                        fr_cur_mode; ///< mode index of current frame
    uint8_t                           fr_quality; ///< frame quality index (FQI)
    float                      isf_cur[LP_ORDER]; ///< working ISF vector from current frame
    float                   isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
    float               isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
    double                      isp[4][LP_ORDER]; ///< ISP vectors from current frame
    double               isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame

    float                   lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector

    uint8_t                       base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
    uint8_t                        pitch_lag_int; ///< integer part of pitch lag of the previous subframe

    float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
    float                            *excitation; ///< points to current excitation in excitation_buf[]

    float           pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
    float           fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe

    float                    prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
    float                          pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
    float                          fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes

    float                              tilt_coef; ///< {beta_1} related to the voicing of the previous subframe

    float                 prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
    uint8_t                    prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
    float                           prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold

    float samples_az[LP_ORDER + AMRWB_SFR_SIZE];         ///< low-band samples and memory from synthesis at 12.8kHz
    float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE];     ///< low-band samples and memory processed for upsampling
    float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz

    float          hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
    float                           demph_mem[1]; ///< previous value in the de-emphasis filter
    float               bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
    float                 lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter

    AVLFG                                   prng; ///< random number generator for white noise excitation
    uint8_t                          first_frame; ///< flag active during decoding of the first frame
} AMRWBContext;

static av_cold int amrwb_decode_init(AVCodecContext *avctx)
{
    AMRWBContext *ctx = avctx->priv_data;
    int i;

    avctx->sample_fmt = SAMPLE_FMT_FLT;

    av_lfg_init(&ctx->prng, 1);

    ctx->excitation  = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
    ctx->first_frame = 1;

    for (i = 0; i < LP_ORDER; i++)
        ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));

    for (i = 0; i < 4; i++)
        ctx->prediction_error[i] = MIN_ENERGY;

    return 0;
}

/**
 * Decode the frame header in the "MIME/storage" format. This format
 * is simpler and does not carry the auxiliary information of the frame
 *
 * @param[in] ctx                  The Context
 * @param[in] buf                  Pointer to the input buffer
 *
 * @return The decoded header length in bytes
 */
static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
{
    GetBitContext gb;
    init_get_bits(&gb, buf, 8);

    /* Decode frame header (1st octet) */
    skip_bits(&gb, 1);  // padding bit
    ctx->fr_cur_mode  = get_bits(&gb, 4);
    ctx->fr_quality   = get_bits1(&gb);
    skip_bits(&gb, 2);  // padding bits

    return 1;
}

/**
 * Decodes quantized ISF vectors using 36-bit indexes (6K60 mode only)
 *
 * @param[in]  ind                 Array of 5 indexes
 * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
 *
 */
static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
{
    int i;

    for (i = 0; i < 9; i++)
        isf_q[i]      = dico1_isf[ind[0]][i]      * (1.0f / (1 << 15));

    for (i = 0; i < 7; i++)
        isf_q[i + 9]  = dico2_isf[ind[1]][i]      * (1.0f / (1 << 15));

    for (i = 0; i < 5; i++)
        isf_q[i]     += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));

    for (i = 0; i < 4; i++)
        isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));

    for (i = 0; i < 7; i++)
        isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
}

/**
 * Decodes quantized ISF vectors using 46-bit indexes (except 6K60 mode)
 *
 * @param[in]  ind                 Array of 7 indexes
 * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
 *
 */
static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
{
    int i;

    for (i = 0; i < 9; i++)
        isf_q[i]       = dico1_isf[ind[0]][i]  * (1.0f / (1 << 15));

    for (i = 0; i < 7; i++)
        isf_q[i + 9]   = dico2_isf[ind[1]][i]  * (1.0f / (1 << 15));

    for (i = 0; i < 3; i++)
        isf_q[i]      += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));

    for (i = 0; i < 3; i++)
        isf_q[i + 3]  += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));

    for (i = 0; i < 3; i++)
        isf_q[i + 6]  += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));

    for (i = 0; i < 3; i++)
        isf_q[i + 9]  += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));

    for (i = 0; i < 4; i++)
        isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
}

/**
 * Apply mean and past ISF values using the prediction factor
 * Updates past ISF vector
 *
 * @param[in,out] isf_q            Current quantized ISF
 * @param[in,out] isf_past         Past quantized ISF
 *
 */
static void isf_add_mean_and_past(float *isf_q, float *isf_past)
{
    int i;
    float tmp;

    for (i = 0; i < LP_ORDER; i++) {
        tmp = isf_q[i];
        isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
        isf_q[i] += PRED_FACTOR * isf_past[i];
        isf_past[i] = tmp;
    }
}

/**
 * Interpolate the fourth ISP vector from current and past frames
 * to obtain a ISP vector for each subframe
 *
 * @param[in,out] isp_q            ISPs for each subframe
 * @param[in]     isp4_past        Past ISP for subframe 4
 */
static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
{
    int i, k;

    for (k = 0; k < 3; k++) {
        float c = isfp_inter[k];
        for (i = 0; i < LP_ORDER; i++)
            isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
    }
}

/**
 * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes)
 * Calculate integer lag and fractional lag always using 1/4 resolution
 * In 1st and 3rd subframes the index is relative to last subframe integer lag
 *
 * @param[out]    lag_int          Decoded integer pitch lag
 * @param[out]    lag_frac         Decoded fractional pitch lag
 * @param[in]     pitch_index      Adaptive codebook pitch index
 * @param[in,out] base_lag_int     Base integer lag used in relative subframes
 * @param[in]     subframe         Current subframe index (0 to 3)
 */
static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
                                  uint8_t *base_lag_int, int subframe)
{
    if (subframe == 0 || subframe == 2) {
        if (pitch_index < 376) {
            *lag_int  = (pitch_index + 137) >> 2;
            *lag_frac = pitch_index - (*lag_int << 2) + 136;
        } else if (pitch_index < 440) {
            *lag_int  = (pitch_index + 257 - 376) >> 1;
            *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
            /* the actual resolution is 1/2 but expressed as 1/4 */
        } else {
            *lag_int  = pitch_index - 280;
            *lag_frac = 0;
        }
        /* minimum lag for next subframe */
        *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
                                AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
        // XXX: the spec states clearly that *base_lag_int should be
        // the nearest integer to *lag_int (minus 8), but the ref code
        // actually always uses its floor, I'm following the latter
    } else {
        *lag_int  = (pitch_index + 1) >> 2;
        *lag_frac = pitch_index - (*lag_int << 2);
        *lag_int += *base_lag_int;
    }
}

/**
 * Decode a adaptive codebook index into pitch lag for 8k85 and 6k60 modes
 * Description is analogous to decode_pitch_lag_high, but in 6k60 relative
 * index is used for all subframes except the first
 */
static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
                                 uint8_t *base_lag_int, int subframe, enum Mode mode)
{
    if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
        if (pitch_index < 116) {
            *lag_int  = (pitch_index + 69) >> 1;
            *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
        } else {
            *lag_int  = pitch_index - 24;
            *lag_frac = 0;
        }
        // XXX: same problem as before
        *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
                                AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
    } else {
        *lag_int  = (pitch_index + 1) >> 1;
        *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
        *lag_int += *base_lag_int;
    }
}

/**
 * Find the pitch vector by interpolating the past excitation at the
 * pitch delay, which is obtained in this function
 *
 * @param[in,out] ctx              The context
 * @param[in]     amr_subframe     Current subframe data
 * @param[in]     subframe         Current subframe index (0 to 3)
 */
static void decode_pitch_vector(AMRWBContext *ctx,
                                const AMRWBSubFrame *amr_subframe,
                                const int subframe)
{
    int pitch_lag_int, pitch_lag_frac;
    int i;
    float *exc     = ctx->excitation;
    enum Mode mode = ctx->fr_cur_mode;

    if (mode <= MODE_8k85) {
        decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
                              &ctx->base_pitch_lag, subframe, mode);
    } else
        decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
                              &ctx->base_pitch_lag, subframe);

    ctx->pitch_lag_int = pitch_lag_int;
    pitch_lag_int += pitch_lag_frac > 0;

    /* Calculate the pitch vector by interpolating the past excitation at the
       pitch lag using a hamming windowed sinc function */
    ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int,
                          ac_inter, 4,
                          pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
                          LP_ORDER, AMRWB_SFR_SIZE + 1);

    /* Check which pitch signal path should be used
     * 6k60 and 8k85 modes have the ltp flag set to 0 */
    if (amr_subframe->ltp) {
        memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
    } else {
        for (i = 0; i < AMRWB_SFR_SIZE; i++)
            ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
                                   0.18 * exc[i + 1];
        memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
    }
}

/** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
#define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))

/** Get the bit at specified position */
#define BIT_POS(x, p) (((x) >> (p)) & 1)

/**
 * The next six functions decode_[i]p_track decode exactly i pulses
 * positions and amplitudes (-1 or 1) in a subframe track using
 * an encoded pulse indexing (TS 26.190 section 5.8.2)
 *
 * The results are given in out[], in which a negative number means
 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) )
 *
 * @param[out] out                 Output buffer (writes i elements)
 * @param[in]  code                Pulse index (no. of bits varies, see below)
 * @param[in]  m                   (log2) Number of potential positions
 * @param[in]  off                 Offset for decoded positions
 */
static inline void decode_1p_track(int *out, int code, int m, int off)
{
    int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits

    out[0] = BIT_POS(code, m) ? -pos : pos;
}

static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
{
    int pos0 = BIT_STR(code, m, m) + off;
    int pos1 = BIT_STR(code, 0, m) + off;

    out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
    out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
    out[1] = pos0 > pos1 ? -out[1] : out[1];
}

static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
{
    int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);

    decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
                    m - 1, off + half_2p);
    decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
}

static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
{
    int half_4p, subhalf_2p;
    int b_offset = 1 << (m - 1);

    switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
    case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
        half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
        subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);

        decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
                        m - 2, off + half_4p + subhalf_2p);
        decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
                        m - 1, off + half_4p);
        break;
    case 1: /* 1 pulse in A, 3 pulses in B */
        decode_1p_track(out, BIT_STR(code,  3*m - 2, m),
                        m - 1, off);
        decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
                        m - 1, off + b_offset);
        break;
    case 2: /* 2 pulses in each half */
        decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
                        m - 1, off);
        decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
                        m - 1, off + b_offset);
        break;
    case 3: /* 3 pulses in A, 1 pulse in B */
        decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
                        m - 1, off);
        decode_1p_track(out + 3, BIT_STR(code, 0, m),
                        m - 1, off + b_offset);
        break;
    }
}

static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
{
    int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);

    decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
                    m - 1, off + half_3p);

    decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
}

static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
{
    int b_offset = 1 << (m - 1);
    /* which half has more pulses in cases 0 to 2 */
    int half_more  = BIT_POS(code, 6*m - 5) << (m - 1);
    int half_other = b_offset - half_more;

    switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
    case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
        decode_1p_track(out, BIT_STR(code, 0, m),
                        m - 1, off + half_more);
        decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
                        m - 1, off + half_more);
        break;
    case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
        decode_1p_track(out, BIT_STR(code, 0, m),
                        m - 1, off + half_other);
        decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
                        m - 1, off + half_more);
        break;
    case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
        decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
                        m - 1, off + half_other);
        decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
                        m - 1, off + half_more);
        break;
    case 3: /* 3 pulses in A, 3 pulses in B */
        decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
                        m - 1, off);
        decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
                        m - 1, off + b_offset);
        break;
    }
}

/**
 * Decode the algebraic codebook index to pulse positions and signs,
 * then construct the algebraic codebook vector
 *
 * @param[out] fixed_vector        Buffer for the fixed codebook excitation
 * @param[in]  pulse_hi            MSBs part of the pulse index array (higher modes only)
 * @param[in]  pulse_lo            LSBs part of the pulse index array
 * @param[in]  mode                Mode of the current frame
 */
static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
                                const uint16_t *pulse_lo, const enum Mode mode)
{
    /* sig_pos stores for each track the decoded pulse position indexes
     * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
    int sig_pos[4][6];
    int spacing = (mode == MODE_6k60) ? 2 : 4;
    int i, j;

    switch (mode) {
    case MODE_6k60:
        for (i = 0; i < 2; i++)
            decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
        break;
    case MODE_8k85:
        for (i = 0; i < 4; i++)
            decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
        break;
    case MODE_12k65:
        for (i = 0; i < 4; i++)
            decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
        break;
    case MODE_14k25:
        for (i = 0; i < 2; i++)
            decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
        for (i = 2; i < 4; i++)
            decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
        break;
    case MODE_15k85:
        for (i = 0; i < 4; i++)
            decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
        break;
    case MODE_18k25:
        for (i = 0; i < 4; i++)
            decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
                           ((int) pulse_hi[i] << 14), 4, 1);
        break;
    case MODE_19k85:
        for (i = 0; i < 2; i++)
            decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
                           ((int) pulse_hi[i] << 10), 4, 1);
        for (i = 2; i < 4; i++)
            decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
                           ((int) pulse_hi[i] << 14), 4, 1);
        break;
    case MODE_23k05:
    case MODE_23k85:
        for (i = 0; i < 4; i++)
            decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
                           ((int) pulse_hi[i] << 11), 4, 1);
        break;
    }

    memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);

    for (i = 0; i < 4; i++)
        for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
            int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;

            fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
        }
}

/**
 * Decode pitch gain and fixed gain correction factor
 *
 * @param[in]  vq_gain             Vector-quantized index for gains
 * @param[in]  mode                Mode of the current frame
 * @param[out] fixed_gain_factor   Decoded fixed gain correction factor
 * @param[out] pitch_gain          Decoded pitch gain
 */
static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
                         float *fixed_gain_factor, float *pitch_gain)
{
    const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
                                                qua_gain_7b[vq_gain]);

    *pitch_gain        = gains[0] * (1.0f / (1 << 14));
    *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
}

/**
 * Apply pitch sharpening filters to the fixed codebook vector
 *
 * @param[in]     ctx              The context
 * @param[in,out] fixed_vector     Fixed codebook excitation
 */
// XXX: Spec states this procedure should be applied when the pitch
// lag is less than 64, but this checking seems absent in reference and AMR-NB
static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
{
    int i;

    /* Tilt part */
    for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
        fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;

    /* Periodicity enhancement part */
    for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
        fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
}

/**
 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced)
 *
 * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
 * @param[in] p_gain, f_gain       Pitch and fixed gains
 */
// XXX: There is something wrong with the precision here! The magnitudes
// of the energies are not correct. Please check the reference code carefully
static float voice_factor(float *p_vector, float p_gain,
                          float *f_vector, float f_gain)
{
    double p_ener = (double) ff_dot_productf(p_vector, p_vector,
                                             AMRWB_SFR_SIZE) * p_gain * p_gain;
    double f_ener = (double) ff_dot_productf(f_vector, f_vector,
                                             AMRWB_SFR_SIZE) * f_gain * f_gain;

    return (p_ener - f_ener) / (p_ener + f_ener);
}

/**
 * Reduce fixed vector sparseness by smoothing with one of three IR filters
 * Also known as "adaptive phase dispersion"
 *
 * @param[in]     ctx              The context
 * @param[in,out] fixed_vector     Unfiltered fixed vector
 * @param[out]    buf              Space for modified vector if necessary
 *
 * @return The potentially overwritten filtered fixed vector address
 */
static float *anti_sparseness(AMRWBContext *ctx,
                              float *fixed_vector, float *buf)
{
    int ir_filter_nr;

    if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
        return fixed_vector;

    if (ctx->pitch_gain[0] < 0.6) {
        ir_filter_nr = 0;      // strong filtering
    } else if (ctx->pitch_gain[0] < 0.9) {
        ir_filter_nr = 1;      // medium filtering
    } else
        ir_filter_nr = 2;      // no filtering

    /* detect 'onset' */
    if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
        if (ir_filter_nr < 2)
            ir_filter_nr++;
    } else {
        int i, count = 0;

        for (i = 0; i < 6; i++)
            if (ctx->pitch_gain[i] < 0.6)
                count++;

        if (count > 2)
            ir_filter_nr = 0;

        if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
            ir_filter_nr--;
    }

    /* update ir filter strength history */
    ctx->prev_ir_filter_nr = ir_filter_nr;

    ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);

    if (ir_filter_nr < 2) {
        int i;
        const float *coef = ir_filters_lookup[ir_filter_nr];

        /* Circular convolution code in the reference
         * decoder was modified to avoid using one
         * extra array. The filtered vector is given by:
         *
         * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
         */

        memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
        for (i = 0; i < AMRWB_SFR_SIZE; i++)
            if (fixed_vector[i])
                ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
                                  AMRWB_SFR_SIZE);
        fixed_vector = buf;
    }

    return fixed_vector;
}

/**
 * Calculate a stability factor {teta} based on distance between
 * current and past isf. A value of 1 shows maximum signal stability
 */
static float stability_factor(const float *isf, const float *isf_past)
{
    int i;
    float acc = 0.0;

    for (i = 0; i < LP_ORDER - 1; i++)
        acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);

    // XXX: This part is not so clear from the reference code
    // the result is more accurate changing the "/ 256" to "* 512"
    return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
}

/**
 * Apply a non-linear fixed gain smoothing in order to reduce
 * fluctuation in the energy of excitation
 *
 * @param[in]     fixed_gain       Unsmoothed fixed gain
 * @param[in,out] prev_tr_gain     Previous threshold gain (updated)
 * @param[in]     voice_fac        Frame voicing factor
 * @param[in]     stab_fac         Frame stability factor
 *
 * @return The smoothed gain
 */
static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
                            float voice_fac,  float stab_fac)
{
    float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
    float g0;

    // XXX: the following fixed-point constants used to in(de)crement
    // gain by 1.5dB were taken from the reference code, maybe it could
    // be simpler
    if (fixed_gain < *prev_tr_gain) {
        g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
                     (6226 * (1.0f / (1 << 15)))); // +1.5 dB
    } else
        g0 = FFMAX(*prev_tr_gain, fixed_gain *
                    (27536 * (1.0f / (1 << 15)))); // -1.5 dB

    *prev_tr_gain = g0; // update next frame threshold

    return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
}

/**
 * Filter the fixed_vector to emphasize the higher frequencies
 *
 * @param[in,out] fixed_vector     Fixed codebook vector
 * @param[in]     voice_fac        Frame voicing factor
 */
static void pitch_enhancer(float *fixed_vector, float voice_fac)
{
    int i;
    float cpe  = 0.125 * (1 + voice_fac);
    float last = fixed_vector[0]; // holds c(i - 1)

    fixed_vector[0] -= cpe * fixed_vector[1];

    for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
        float cur = fixed_vector[i];

        fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
        last = cur;
    }

    fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
}

/**
 * Conduct 16th order linear predictive coding synthesis from excitation
 *
 * @param[in]     ctx              Pointer to the AMRWBContext
 * @param[in]     lpc              Pointer to the LPC coefficients
 * @param[out]    excitation       Buffer for synthesis final excitation
 * @param[in]     fixed_gain       Fixed codebook gain for synthesis
 * @param[in]     fixed_vector     Algebraic codebook vector
 * @param[in,out] samples          Pointer to the output samples and memory
 */
static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
                      float fixed_gain, const float *fixed_vector,
                      float *samples)
{
    ff_weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
                            ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);

    /* emphasize pitch vector contribution in low bitrate modes */
    if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
        int i;
        float energy = ff_dot_productf(excitation, excitation,
                                       AMRWB_SFR_SIZE);

        // XXX: Weird part in both ref code and spec. A unknown parameter
        // {beta} seems to be identical to the current pitch gain
        float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];

        for (i = 0; i < AMRWB_SFR_SIZE; i++)
            excitation[i] += pitch_factor * ctx->pitch_vector[i];

        ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
                                                energy, AMRWB_SFR_SIZE);
    }

    ff_celp_lp_synthesis_filterf(samples, lpc, excitation,
                                 AMRWB_SFR_SIZE, LP_ORDER);
}

/**
 * Apply to synthesis a de-emphasis filter of the form:
 * H(z) = 1 / (1 - m * z^-1)
 *
 * @param[out]    out              Output buffer
 * @param[in]     in               Input samples array with in[-1]
 * @param[in]     m                Filter coefficient
 * @param[in,out] mem              State from last filtering
 */
static void de_emphasis(float *out, float *in, float m, float mem[1])
{
    int i;

    out[0] = in[0] + m * mem[0];

    for (i = 1; i < AMRWB_SFR_SIZE; i++)
         out[i] = in[i] + out[i - 1] * m;

    mem[0] = out[AMRWB_SFR_SIZE - 1];
}

/**
 * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
 * a FIR interpolation filter. Uses past data from before *in address
 *
 * @param[out] out                 Buffer for interpolated signal
 * @param[in]  in                  Current signal data (length 0.8*o_size)
 * @param[in]  o_size              Output signal length
 */
static void upsample_5_4(float *out, const float *in, int o_size)
{
    const float *in0 = in - UPS_FIR_SIZE + 1;
    int i, j, k;
    int int_part = 0, frac_part;

    i = 0;
    for (j = 0; j < o_size / 5; j++) {
        out[i] = in[int_part];
        frac_part = 4;
        i++;

        for (k = 1; k < 5; k++) {
            out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4 - frac_part],
                                     UPS_MEM_SIZE);
            int_part++;
            frac_part--;
            i++;
        }
    }
}

/**
 * Calculate the high-band gain based on encoded index (23k85 mode) or
 * on the low-band speech signal and the Voice Activity Detection flag
 *
 * @param[in] ctx                  The context
 * @param[in] synth                LB speech synthesis at 12.8k
 * @param[in] hb_idx               Gain index for mode 23k85 only
 * @param[in] vad                  VAD flag for the frame
 */
static float find_hb_gain(AMRWBContext *ctx, const float *synth,
                          uint16_t hb_idx, uint8_t vad)
{
    int wsp = (vad > 0);
    float tilt;

    if (ctx->fr_cur_mode == MODE_23k85)
        return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));

    tilt = ff_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
           ff_dot_productf(synth, synth, AMRWB_SFR_SIZE);

    /* return gain bounded by [0.1, 1.0] */
    return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
}

/**
 * Generate the high-band excitation with the same energy from the lower
 * one and scaled by the given gain
 *
 * @param[in]  ctx                 The context
 * @param[out] hb_exc              Buffer for the excitation
 * @param[in]  synth_exc           Low-band excitation used for synthesis
 * @param[in]  hb_gain             Wanted excitation gain
 */
static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
                                 const float *synth_exc, float hb_gain)
{
    int i;
    float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);

    /* Generate a white-noise excitation */
    for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
        hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);

    ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
                                            energy * hb_gain * hb_gain,
                                            AMRWB_SFR_SIZE_16k);
}

/**
 * Calculate the auto-correlation for the ISF difference vector
 */
static float auto_correlation(float *diff_isf, float mean, int lag)
{
    int i;
    float sum = 0.0;

    for (i = 7; i < LP_ORDER - 2; i++) {
        float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
        sum += prod * prod;
    }
    return sum;
}

/**
 * Extrapolate a ISF vector to the 16kHz range (20th order LP)
 * used at mode 6k60 LP filter for the high frequency band
 *
 * @param[out] out                 Buffer for extrapolated isf
 * @param[in]  isf                 Input isf vector
 */
static void extrapolate_isf(float out[LP_ORDER_16k], float isf[LP_ORDER])
{
    float diff_isf[LP_ORDER - 2], diff_mean;
    float *diff_hi = diff_isf - LP_ORDER + 1; // diff array for extrapolated indexes
    float corr_lag[3];
    float est, scale;
    int i, i_max_corr;

    memcpy(out, isf, (LP_ORDER - 1) * sizeof(float));
    out[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];

    /* Calculate the difference vector */
    for (i = 0; i < LP_ORDER - 2; i++)
        diff_isf[i] = isf[i + 1] - isf[i];

    diff_mean = 0.0;
    for (i = 2; i < LP_ORDER - 2; i++)
        diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));

    /* Find which is the maximum autocorrelation */
    i_max_corr = 0;
    for (i = 0; i < 3; i++) {
        corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);

        if (corr_lag[i] > corr_lag[i_max_corr])
            i_max_corr = i;
    }
    i_max_corr++;

    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
        out[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
                            - isf[i - 2 - i_max_corr];

    /* Calculate an estimate for ISF(18) and scale ISF based on the error */
    est   = 7965 + (out[2] - out[3] - out[4]) / 6.0;
    scale = 0.5 * (FFMIN(est, 7600) - out[LP_ORDER - 2]) /
            (out[LP_ORDER_16k - 2] - out[LP_ORDER - 2]);

    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
        diff_hi[i] = scale * (out[i] - out[i - 1]);

    /* Stability insurance */
    for (i = LP_ORDER; i < LP_ORDER_16k - 1; i++)
        if (diff_hi[i] + diff_hi[i - 1] < 5.0) {
            if (diff_hi[i] > diff_hi[i - 1]) {
                diff_hi[i - 1] = 5.0 - diff_hi[i];
            } else
                diff_hi[i] = 5.0 - diff_hi[i - 1];
        }

    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
        out[i] = out[i - 1] + diff_hi[i] * (1.0f / (1 << 15));

    /* Scale the ISF vector for 16000 Hz */
    for (i = 0; i < LP_ORDER_16k - 1; i++)
        out[i] *= 0.8;
}

/**
 * Spectral expand the LP coefficients using the equation:
 *   y[i] = x[i] * (gamma ** i)
 *
 * @param[out] out                 Output buffer (may use input array)
 * @param[in]  lpc                 LP coefficients array
 * @param[in]  gamma               Weighting factor
 * @param[in]  size                LP array size
 */
static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
{
    int i;
    float fac = gamma;

    for (i = 0; i < size; i++) {
        out[i] = lpc[i] * fac;
        fac   *= gamma;
    }
}

/**
 * Conduct 20th order linear predictive coding synthesis for the high
 * frequency band excitation at 16kHz
 *
 * @param[in]     ctx              The context
 * @param[in]     subframe         Current subframe index (0 to 3)
 * @param[in,out] samples          Pointer to the output speech samples
 * @param[in]     exc              Generated white-noise scaled excitation
 * @param[in]     isf              Current frame isf vector
 * @param[in]     isf_past         Past frame final isf vector
 */
static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
                         const float *exc, const float *isf, const float *isf_past)
{
    float hb_lpc[LP_ORDER_16k];
    enum Mode mode = ctx->fr_cur_mode;

    if (mode == MODE_6k60) {
        float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
        double e_isp[LP_ORDER_16k];

        ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
                                1.0 - isfp_inter[subframe], LP_ORDER);

        extrapolate_isf(e_isf, e_isf);

        e_isf[LP_ORDER_16k - 1] *= 2.0;
        ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
        ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);

        lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
    } else {
        lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
    }

    ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
                                 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
}

/**
 * Apply to high-band samples a 15th order filter
 * The filter characteristic depends on the given coefficients
 *
 * @param[out]    out              Buffer for filtered output
 * @param[in]     fir_coef         Filter coefficients
 * @param[in,out] mem              State from last filtering (updated)
 * @param[in]     in               Input speech data (high-band)
 *
 * @remark It is safe to pass the same array in in and out parameters
 */
static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
                          float mem[HB_FIR_SIZE], const float *in)
{
    int i, j;
    float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples

    memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
    memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));

    for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
        out[i] = 0.0;
        for (j = 0; j <= HB_FIR_SIZE; j++)
            out[i] += data[i + j] * fir_coef[j];
    }

    memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
}

/**
 * Update context state before the next subframe
 */
static void update_sub_state(AMRWBContext *ctx)
{
    memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
            (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));

    memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
    memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));

    memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
            LP_ORDER * sizeof(float));
    memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
            UPS_MEM_SIZE * sizeof(float));
    memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
            LP_ORDER_16k * sizeof(float));
}

static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
                              AVPacket *avpkt)
{
    AMRWBContext *ctx  = avctx->priv_data;
    AMRWBFrame   *cf   = &ctx->frame;
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    int expected_fr_size, header_size;
    float *buf_out = data;
    float spare_vector[AMRWB_SFR_SIZE];      // extra stack space to hold result from anti-sparseness processing
    float fixed_gain_factor;                 // fixed gain correction factor (gamma)
    float *synth_fixed_vector;               // pointer to the fixed vector that synthesis should use
    float synth_fixed_gain;                  // the fixed gain that synthesis should use
    float voice_fac, stab_fac;               // parameters used for gain smoothing
    float synth_exc[AMRWB_SFR_SIZE];         // post-processed excitation for synthesis
    float hb_exc[AMRWB_SFR_SIZE_16k];        // excitation for the high frequency band
    float hb_samples[AMRWB_SFR_SIZE_16k];    // filtered high-band samples from synthesis
    float hb_gain;
    int sub, i;

    header_size      = decode_mime_header(ctx, buf);
    expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;

    if (buf_size < expected_fr_size) {
        av_log(avctx, AV_LOG_ERROR,
            "Frame too small (%d bytes). Truncated file?\n", buf_size);
        *data_size = 0;
        return buf_size;
    }

    if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
        av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");

    if (ctx->fr_cur_mode == MODE_SID) /* Comfort noise frame */
        av_log_missing_feature(avctx, "SID mode", 1);

    if (ctx->fr_cur_mode >= MODE_SID)
        return -1;

    ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
        buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);

    /* Decode the quantized ISF vector */
    if (ctx->fr_cur_mode == MODE_6k60) {
        decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
    } else {
        decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
    }

    isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
    ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);

    stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);

    ctx->isf_cur[LP_ORDER - 1] *= 2.0;
    ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);

    /* Generate a ISP vector for each subframe */
    if (ctx->first_frame) {
        ctx->first_frame = 0;
        memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
    }
    interpolate_isp(ctx->isp, ctx->isp_sub4_past);

    for (sub = 0; sub < 4; sub++)
        ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);

    for (sub = 0; sub < 4; sub++) {
        const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
        float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;

        /* Decode adaptive codebook (pitch vector) */
        decode_pitch_vector(ctx, cur_subframe, sub);
        /* Decode innovative codebook (fixed vector) */
        decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
                            cur_subframe->pul_il, ctx->fr_cur_mode);

        pitch_sharpening(ctx, ctx->fixed_vector);

        decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
                     &fixed_gain_factor, &ctx->pitch_gain[0]);

        ctx->fixed_gain[0] =
            ff_amr_set_fixed_gain(fixed_gain_factor,
                       ff_dot_productf(ctx->fixed_vector, ctx->fixed_vector,
                                       AMRWB_SFR_SIZE) / AMRWB_SFR_SIZE,
                       ctx->prediction_error,
                       ENERGY_MEAN, energy_pred_fac);

        /* Calculate voice factor and store tilt for next subframe */
        voice_fac      = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
                                      ctx->fixed_vector, ctx->fixed_gain[0]);
        ctx->tilt_coef = voice_fac * 0.25 + 0.25;

        /* Construct current excitation */
        for (i = 0; i < AMRWB_SFR_SIZE; i++) {
            ctx->excitation[i] *= ctx->pitch_gain[0];
            ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
            ctx->excitation[i] = truncf(ctx->excitation[i]);
        }

        /* Post-processing of excitation elements */
        synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
                                          voice_fac, stab_fac);

        synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
                                             spare_vector);

        pitch_enhancer(synth_fixed_vector, voice_fac);

        synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
                  synth_fixed_vector, &ctx->samples_az[LP_ORDER]);

        /* Synthesis speech post-processing */
        de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
                    &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);

        ff_acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
            &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
            hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);

        upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
                     AMRWB_SFR_SIZE_16k);

        /* High frequency band (6.4 - 7.0 kHz) generation part */
        ff_acelp_apply_order_2_transfer_function(hb_samples,
            &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
            hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);

        hb_gain = find_hb_gain(ctx, hb_samples,
                               cur_subframe->hb_gain, cf->vad);

        scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);

        hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
                     hb_exc, ctx->isf_cur, ctx->isf_past_final);

        /* High-band post-processing filters */
        hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
                      &ctx->samples_hb[LP_ORDER_16k]);

        if (ctx->fr_cur_mode == MODE_23k85)
            hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
                          hb_samples);

        /* Add the low and high frequency bands */
        for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
            sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));

        /* Update buffers and history */
        update_sub_state(ctx);
    }

    /* update state for next frame */
    memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
    memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));

    /* report how many samples we got */
    *data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float);

    return expected_fr_size;
}

AVCodec amrwb_decoder = {
    .name           = "amrwb",
    .type           = CODEC_TYPE_AUDIO,
    .id             = CODEC_ID_AMR_WB,
    .priv_data_size = sizeof(AMRWBContext),
    .init           = amrwb_decode_init,
    .decode         = amrwb_decode_frame,
    .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"),
    .sample_fmts    = (enum AVSampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
};