aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/amrnbdec.c
blob: 5c359a8f3d0f98d0690d342c2b2ad34c5b44edf8 (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
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
/*
 * AMR narrowband decoder
 * Copyright (c) 2006-2007 Robert Swain
 * Copyright (c) 2009 Colin McQuillan
 *
 * This file is part of Libav.
 *
 * Libav 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.
 *
 * Libav 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 Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */


/**
 * @file
 * AMR narrowband decoder
 *
 * This decoder uses floats for simplicity and so is not bit-exact. One
 * difference is that differences in phase can accumulate. The test sequences
 * in 3GPP TS 26.074 can still be useful.
 *
 * - Comparing this file's output to the output of the ref decoder gives a
 *   PSNR of 30 to 80. Plotting the output samples shows a difference in
 *   phase in some areas.
 *
 * - Comparing both decoders against their input, this decoder gives a similar
 *   PSNR. If the test sequence homing frames are removed (this decoder does
 *   not detect them), the PSNR is at least as good as the reference on 140
 *   out of 169 tests.
 */


#include <string.h>
#include <math.h>

#include "libavutil/channel_layout.h"
#include "avcodec.h"
#include "dsputil.h"
#include "libavutil/common.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "lsp.h"
#include "amr.h"
#include "internal.h"

#include "amrnbdata.h"

#define AMR_BLOCK_SIZE              160   ///< samples per frame
#define AMR_SAMPLE_BOUND        32768.0   ///< threshold for synthesis overflow

/**
 * Scale from constructed speech to [-1,1]
 *
 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
 * upscales by two (section 6.2.2).
 *
 * Fundamentally, this scale is determined by energy_mean through
 * the fixed vector contribution to the excitation vector.
 */
#define AMR_SAMPLE_SCALE  (2.0 / 32768.0)

/** Prediction factor for 12.2kbit/s mode */
#define PRED_FAC_MODE_12k2             0.65

#define LSF_R_FAC          (8000.0 / 32768.0) ///< LSF residual tables to Hertz
#define MIN_LSF_SPACING    (50.0488 / 8000.0) ///< Ensures stability of LPC filter
#define PITCH_LAG_MIN_MODE_12k2          18   ///< Lower bound on decoded lag search in 12.2kbit/s mode

/** Initial energy in dB. Also used for bad frames (unimplemented). */
#define MIN_ENERGY -14.0

/** Maximum sharpening factor
 *
 * The specification says 0.8, which should be 13107, but the reference C code
 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in bitexact G.729.)
 */
#define SHARP_MAX 0.79449462890625

/** Number of impulse response coefficients used for tilt factor */
#define AMR_TILT_RESPONSE   22
/** Tilt factor = 1st reflection coefficient * gamma_t */
#define AMR_TILT_GAMMA_T   0.8
/** Adaptive gain control factor used in post-filter */
#define AMR_AGC_ALPHA      0.9

typedef struct AMRContext {
    AVFrame                         avframe; ///< AVFrame for decoded samples
    AMRNBFrame                        frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
    uint8_t             bad_frame_indicator; ///< bad frame ? 1 : 0
    enum Mode                cur_frame_mode;

    int16_t     prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
    double          lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
    double   prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame

    float         lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
    float          lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector

    float           lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes

    uint8_t                   pitch_lag_int; ///< integer part of pitch lag from current subframe

    float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
    float                       *excitation; ///< pointer to the current excitation vector in excitation_buf

    float   pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
    float   fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)

    float               prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
    float                     pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
    float                     fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes

    float                              beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
    uint8_t                      diff_count; ///< the number of subframes for which diff has been above 0.65
    uint8_t                      hang_count; ///< the number of subframes since a hangover period started

    float            prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
    uint8_t               prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
    uint8_t                 ir_filter_onset; ///< flag for impulse response filter strength

    float                postfilter_mem[10]; ///< previous intermediate values in the formant filter
    float                          tilt_mem; ///< previous input to tilt compensation filter
    float                    postfilter_agc; ///< previous factor used for adaptive gain control
    float                  high_pass_mem[2]; ///< previous intermediate values in the high-pass filter

    float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples

} AMRContext;

/** Double version of ff_weighted_vector_sumf() */
static void weighted_vector_sumd(double *out, const double *in_a,
                                 const double *in_b, double weight_coeff_a,
                                 double weight_coeff_b, int length)
{
    int i;

    for (i = 0; i < length; i++)
        out[i] = weight_coeff_a * in_a[i]
               + weight_coeff_b * in_b[i];
}

static av_cold int amrnb_decode_init(AVCodecContext *avctx)
{
    AMRContext *p = avctx->priv_data;
    int i;

    if (avctx->channels > 1) {
        av_log_missing_feature(avctx, "multi-channel AMR", 0);
        return AVERROR_PATCHWELCOME;
    }

    avctx->channels       = 1;
    avctx->channel_layout = AV_CH_LAYOUT_MONO;
    avctx->sample_rate    = 8000;
    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

    // p->excitation always points to the same position in p->excitation_buf
    p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];

    for (i = 0; i < LP_FILTER_ORDER; i++) {
        p->prev_lsp_sub4[i] =    lsp_sub4_init[i] * 1000 / (float)(1 << 15);
        p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
    }

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

    avcodec_get_frame_defaults(&p->avframe);
    avctx->coded_frame = &p->avframe;

    return 0;
}


/**
 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
 *
 * The order of speech bits is specified by 3GPP TS 26.101.
 *
 * @param p the context
 * @param buf               pointer to the input buffer
 * @param buf_size          size of the input buffer
 *
 * @return the frame mode
 */
static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
                                  int buf_size)
{
    enum Mode mode;

    // Decode the first octet.
    mode = buf[0] >> 3 & 0x0F;                      // frame type
    p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit

    if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
        return NO_DATA;
    }

    if (mode < MODE_DTX)
        ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
                           amr_unpacking_bitmaps_per_mode[mode]);

    return mode;
}


/// @name AMR pitch LPC coefficient decoding functions
/// @{

/**
 * Interpolate the LSF vector (used for fixed gain smoothing).
 * The interpolation is done over all four subframes even in MODE_12k2.
 *
 * @param[in,out] lsf_q     LSFs in [0,1] for each subframe
 * @param[in]     lsf_new   New LSFs in [0,1] for subframe 4
 */
static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
{
    int i;

    for (i = 0; i < 4; i++)
        ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
                                0.25 * (3 - i), 0.25 * (i + 1),
                                LP_FILTER_ORDER);
}

/**
 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
 *
 * @param p the context
 * @param lsp output LSP vector
 * @param lsf_no_r LSF vector without the residual vector added
 * @param lsf_quantizer pointers to LSF dictionary tables
 * @param quantizer_offset offset in tables
 * @param sign for the 3 dictionary table
 * @param update store data for computing the next frame's LSFs
 */
static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
                                 const float lsf_no_r[LP_FILTER_ORDER],
                                 const int16_t *lsf_quantizer[5],
                                 const int quantizer_offset,
                                 const int sign, const int update)
{
    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
    int i;

    for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
        memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
               2 * sizeof(*lsf_r));

    if (sign) {
        lsf_r[4] *= -1;
        lsf_r[5] *= -1;
    }

    if (update)
        memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));

    for (i = 0; i < LP_FILTER_ORDER; i++)
        lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);

    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);

    if (update)
        interpolate_lsf(p->lsf_q, lsf_q);

    ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
}

/**
 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
 *
 * @param p                 pointer to the AMRContext
 */
static void lsf2lsp_5(AMRContext *p)
{
    const uint16_t *lsf_param = p->frame.lsf;
    float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
    const int16_t *lsf_quantizer[5];
    int i;

    lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
    lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
    lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
    lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
    lsf_quantizer[4] = lsf_5_5[lsf_param[4]];

    for (i = 0; i < LP_FILTER_ORDER; i++)
        lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];

    lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
    lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);

    // interpolate LSP vectors at subframes 1 and 3
    weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
    weighted_vector_sumd(p->lsp[2], p->lsp[1]       , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
}

/**
 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
 *
 * @param p                 pointer to the AMRContext
 */
static void lsf2lsp_3(AMRContext *p)
{
    const uint16_t *lsf_param = p->frame.lsf;
    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
    const int16_t *lsf_quantizer;
    int i, j;

    lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
    memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));

    lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
    memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));

    lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
    memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));

    // calculate mean-removed LSF vector and add mean
    for (i = 0; i < LP_FILTER_ORDER; i++)
        lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);

    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);

    // store data for computing the next frame's LSFs
    interpolate_lsf(p->lsf_q, lsf_q);
    memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));

    ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);

    // interpolate LSP vectors at subframes 1, 2 and 3
    for (i = 1; i <= 3; i++)
        for(j = 0; j < LP_FILTER_ORDER; j++)
            p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
                (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
}

/// @}


/// @name AMR pitch vector decoding functions
/// @{

/**
 * Like ff_decode_pitch_lag(), but with 1/6 resolution
 */
static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
                                 const int prev_lag_int, const int subframe)
{
    if (subframe == 0 || subframe == 2) {
        if (pitch_index < 463) {
            *lag_int  = (pitch_index + 107) * 10923 >> 16;
            *lag_frac = pitch_index - *lag_int * 6 + 105;
        } else {
            *lag_int  = pitch_index - 368;
            *lag_frac = 0;
        }
    } else {
        *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
        *lag_frac = pitch_index - *lag_int * 6 - 3;
        *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
                            PITCH_DELAY_MAX - 9);
    }
}

static void decode_pitch_vector(AMRContext *p,
                                const AMRNBSubframe *amr_subframe,
                                const int subframe)
{
    int pitch_lag_int, pitch_lag_frac;
    enum Mode mode = p->cur_frame_mode;

    if (p->cur_frame_mode == MODE_12k2) {
        decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
                             amr_subframe->p_lag, p->pitch_lag_int,
                             subframe);
    } else
        ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
                            amr_subframe->p_lag,
                            p->pitch_lag_int, subframe,
                            mode != MODE_4k75 && mode != MODE_5k15,
                            mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));

    p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t

    pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);

    pitch_lag_int += pitch_lag_frac > 0;

    /* Calculate the pitch vector by interpolating the past excitation at the
       pitch lag using a b60 hamming windowed sinc function.   */
    ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
                          ff_b60_sinc, 6,
                          pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
                          10, AMR_SUBFRAME_SIZE);

    memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
}

/// @}


/// @name AMR algebraic code book (fixed) vector decoding functions
/// @{

/**
 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
 */
static void decode_10bit_pulse(int code, int pulse_position[8],
                               int i1, int i2, int i3)
{
    // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
    // the 3 pulses and the upper 7 bits being coded in base 5
    const uint8_t *positions = base_five_table[code >> 3];
    pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
    pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
    pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
}

/**
 * Decode the algebraic codebook index to pulse positions and signs and
 * construct the algebraic codebook vector for MODE_10k2.
 *
 * @param fixed_index          positions of the eight pulses
 * @param fixed_sparse         pointer to the algebraic codebook vector
 */
static void decode_8_pulses_31bits(const int16_t *fixed_index,
                                   AMRFixed *fixed_sparse)
{
    int pulse_position[8];
    int i, temp;

    decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
    decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);

    // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
    // the 2 pulses and the upper 5 bits being coded in base 5
    temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
    pulse_position[3] = temp % 5;
    pulse_position[7] = temp / 5;
    if (pulse_position[7] & 1)
        pulse_position[3] = 4 - pulse_position[3];
    pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
    pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);

    fixed_sparse->n = 8;
    for (i = 0; i < 4; i++) {
        const int pos1   = (pulse_position[i]     << 2) + i;
        const int pos2   = (pulse_position[i + 4] << 2) + i;
        const float sign = fixed_index[i] ? -1.0 : 1.0;
        fixed_sparse->x[i    ] = pos1;
        fixed_sparse->x[i + 4] = pos2;
        fixed_sparse->y[i    ] = sign;
        fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
    }
}

/**
 * Decode the algebraic codebook index to pulse positions and signs,
 * then construct the algebraic codebook vector.
 *
 *                              nb of pulses | bits encoding pulses
 * For MODE_4k75 or MODE_5k15,             2 | 1-3, 4-6, 7
 *                  MODE_5k9,              2 | 1,   2-4, 5-6, 7-9
 *                  MODE_6k7,              3 | 1-3, 4,   5-7, 8,  9-11
 *      MODE_7k4 or MODE_7k95,             4 | 1-3, 4-6, 7-9, 10, 11-13
 *
 * @param fixed_sparse pointer to the algebraic codebook vector
 * @param pulses       algebraic codebook indexes
 * @param mode         mode of the current frame
 * @param subframe     current subframe number
 */
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
                                const enum Mode mode, const int subframe)
{
    assert(MODE_4k75 <= mode && mode <= MODE_12k2);

    if (mode == MODE_12k2) {
        ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
    } else if (mode == MODE_10k2) {
        decode_8_pulses_31bits(pulses, fixed_sparse);
    } else {
        int *pulse_position = fixed_sparse->x;
        int i, pulse_subset;
        const int fixed_index = pulses[0];

        if (mode <= MODE_5k15) {
            pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
            pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
            pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
            fixed_sparse->n = 2;
        } else if (mode == MODE_5k9) {
            pulse_subset      = ((fixed_index & 1) << 1) + 1;
            pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
            pulse_subset      = (fixed_index  >> 4) & 3;
            pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
            fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
        } else if (mode == MODE_6k7) {
            pulse_position[0] = (fixed_index        & 7) * 5;
            pulse_subset      = (fixed_index  >> 2) & 2;
            pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
            pulse_subset      = (fixed_index  >> 6) & 2;
            pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
            fixed_sparse->n = 3;
        } else { // mode <= MODE_7k95
            pulse_position[0] = gray_decode[ fixed_index        & 7];
            pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
            pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
            pulse_subset      = (fixed_index >> 9) & 1;
            pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
            fixed_sparse->n = 4;
        }
        for (i = 0; i < fixed_sparse->n; i++)
            fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
    }
}

/**
 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
 *
 * @param p the context
 * @param subframe unpacked amr subframe
 * @param mode mode of the current frame
 * @param fixed_sparse sparse respresentation of the fixed vector
 */
static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
                             AMRFixed *fixed_sparse)
{
    // The spec suggests the current pitch gain is always used, but in other
    // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
    // so the codebook gain cannot depend on the quantized pitch gain.
    if (mode == MODE_12k2)
        p->beta = FFMIN(p->pitch_gain[4], 1.0);

    fixed_sparse->pitch_lag  = p->pitch_lag_int;
    fixed_sparse->pitch_fac  = p->beta;

    // Save pitch sharpening factor for the next subframe
    // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
    // the fact that the gains for two subframes are jointly quantized.
    if (mode != MODE_4k75 || subframe & 1)
        p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
}
/// @}


/// @name AMR gain decoding functions
/// @{

/**
 * fixed gain smoothing
 * Note that where the spec specifies the "spectrum in the q domain"
 * in section 6.1.4, in fact frequencies should be used.
 *
 * @param p the context
 * @param lsf LSFs for the current subframe, in the range [0,1]
 * @param lsf_avg averaged LSFs
 * @param mode mode of the current frame
 *
 * @return fixed gain smoothed
 */
static float fixed_gain_smooth(AMRContext *p , const float *lsf,
                               const float *lsf_avg, const enum Mode mode)
{
    float diff = 0.0;
    int i;

    for (i = 0; i < LP_FILTER_ORDER; i++)
        diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];

    // If diff is large for ten subframes, disable smoothing for a 40-subframe
    // hangover period.
    p->diff_count++;
    if (diff <= 0.65)
        p->diff_count = 0;

    if (p->diff_count > 10) {
        p->hang_count = 0;
        p->diff_count--; // don't let diff_count overflow
    }

    if (p->hang_count < 40) {
        p->hang_count++;
    } else if (mode < MODE_7k4 || mode == MODE_10k2) {
        const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
        const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
                                       p->fixed_gain[2] + p->fixed_gain[3] +
                                       p->fixed_gain[4]) * 0.2;
        return smoothing_factor * p->fixed_gain[4] +
               (1.0 - smoothing_factor) * fixed_gain_mean;
    }
    return p->fixed_gain[4];
}

/**
 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
 *
 * @param p the context
 * @param amr_subframe unpacked amr subframe
 * @param mode mode of the current frame
 * @param subframe current subframe number
 * @param fixed_gain_factor decoded gain correction factor
 */
static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
                         const enum Mode mode, const int subframe,
                         float *fixed_gain_factor)
{
    if (mode == MODE_12k2 || mode == MODE_7k95) {
        p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
            * (1.0 / 16384.0);
        *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
            * (1.0 /  2048.0);
    } else {
        const uint16_t *gains;

        if (mode >= MODE_6k7) {
            gains = gains_high[amr_subframe->p_gain];
        } else if (mode >= MODE_5k15) {
            gains = gains_low [amr_subframe->p_gain];
        } else {
            // gain index is only coded in subframes 0,2 for MODE_4k75
            gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
        }

        p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
        *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
    }
}

/// @}


/// @name AMR preprocessing functions
/// @{

/**
 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
 *
 * @param out vector with filter applied
 * @param in source vector
 * @param filter phase filter coefficients
 *
 *  out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
 */
static void apply_ir_filter(float *out, const AMRFixed *in,
                            const float *filter)
{
    float filter1[AMR_SUBFRAME_SIZE],     ///< filters at pitch lag*1 and *2
          filter2[AMR_SUBFRAME_SIZE];
    int   lag = in->pitch_lag;
    float fac = in->pitch_fac;
    int i;

    if (lag < AMR_SUBFRAME_SIZE) {
        ff_celp_circ_addf(filter1, filter, filter, lag, fac,
                          AMR_SUBFRAME_SIZE);

        if (lag < AMR_SUBFRAME_SIZE >> 1)
            ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
                              AMR_SUBFRAME_SIZE);
    }

    memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
    for (i = 0; i < in->n; i++) {
        int   x = in->x[i];
        float y = in->y[i];
        const float *filterp;

        if (x >= AMR_SUBFRAME_SIZE - lag) {
            filterp = filter;
        } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
            filterp = filter1;
        } else
            filterp = filter2;

        ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
    }
}

/**
 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
 * Also know as "adaptive phase dispersion".
 *
 * This implements 3GPP TS 26.090 section 6.1(5).
 *
 * @param p the context
 * @param fixed_sparse algebraic codebook vector
 * @param fixed_vector unfiltered fixed vector
 * @param fixed_gain smoothed gain
 * @param out space for modified vector if necessary
 */
static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
                                    const float *fixed_vector,
                                    float fixed_gain, float *out)
{
    int ir_filter_nr;

    if (p->pitch_gain[4] < 0.6) {
        ir_filter_nr = 0;      // strong filtering
    } else if (p->pitch_gain[4] < 0.9) {
        ir_filter_nr = 1;      // medium filtering
    } else
        ir_filter_nr = 2;      // no filtering

    // detect 'onset'
    if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
        p->ir_filter_onset = 2;
    } else if (p->ir_filter_onset)
        p->ir_filter_onset--;

    if (!p->ir_filter_onset) {
        int i, count = 0;

        for (i = 0; i < 5; i++)
            if (p->pitch_gain[i] < 0.6)
                count++;
        if (count > 2)
            ir_filter_nr = 0;

        if (ir_filter_nr > p->prev_ir_filter_nr + 1)
            ir_filter_nr--;
    } else if (ir_filter_nr < 2)
        ir_filter_nr++;

    // Disable filtering for very low level of fixed_gain.
    // Note this step is not specified in the technical description but is in
    // the reference source in the function Ph_disp.
    if (fixed_gain < 5.0)
        ir_filter_nr = 2;

    if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
         && ir_filter_nr < 2) {
        apply_ir_filter(out, fixed_sparse,
                        (p->cur_frame_mode == MODE_7k95 ?
                             ir_filters_lookup_MODE_7k95 :
                             ir_filters_lookup)[ir_filter_nr]);
        fixed_vector = out;
    }

    // update ir filter strength history
    p->prev_ir_filter_nr       = ir_filter_nr;
    p->prev_sparse_fixed_gain  = fixed_gain;

    return fixed_vector;
}

/// @}


/// @name AMR synthesis functions
/// @{

/**
 * Conduct 10th order linear predictive coding synthesis.
 *
 * @param p             pointer to the AMRContext
 * @param lpc           pointer to the LPC coefficients
 * @param fixed_gain    fixed codebook gain for synthesis
 * @param fixed_vector  algebraic codebook vector
 * @param samples       pointer to the output speech samples
 * @param overflow      16-bit overflow flag
 */
static int synthesis(AMRContext *p, float *lpc,
                     float fixed_gain, const float *fixed_vector,
                     float *samples, uint8_t overflow)
{
    int i;
    float excitation[AMR_SUBFRAME_SIZE];

    // if an overflow has been detected, the pitch vector is scaled down by a
    // factor of 4
    if (overflow)
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
            p->pitch_vector[i] *= 0.25;

    ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
                            p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);

    // emphasize pitch vector contribution
    if (p->pitch_gain[4] > 0.5 && !overflow) {
        float energy = ff_scalarproduct_float_c(excitation, excitation,
                                                AMR_SUBFRAME_SIZE);
        float pitch_factor =
            p->pitch_gain[4] *
            (p->cur_frame_mode == MODE_12k2 ?
                0.25 * FFMIN(p->pitch_gain[4], 1.0) :
                0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));

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

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

    ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
                                 LP_FILTER_ORDER);

    // detect overflow
    for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
        if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
            return 1;
        }

    return 0;
}

/// @}


/// @name AMR update functions
/// @{

/**
 * Update buffers and history at the end of decoding a subframe.
 *
 * @param p             pointer to the AMRContext
 */
static void update_state(AMRContext *p)
{
    memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));

    memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
            (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));

    memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
    memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));

    memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
            LP_FILTER_ORDER * sizeof(float));
}

/// @}


/// @name AMR Postprocessing functions
/// @{

/**
 * Get the tilt factor of a formant filter from its transfer function
 *
 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
 */
static float tilt_factor(float *lpc_n, float *lpc_d)
{
    float rh0, rh1; // autocorrelation at lag 0 and 1

    // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
    float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
    float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response

    hf[0] = 1.0;
    memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
    ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
                                 LP_FILTER_ORDER);

    rh0 = ff_scalarproduct_float_c(hf, hf,     AMR_TILT_RESPONSE);
    rh1 = ff_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1);

    // The spec only specifies this check for 12.2 and 10.2 kbit/s
    // modes. But in the ref source the tilt is always non-negative.
    return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
}

/**
 * Perform adaptive post-filtering to enhance the quality of the speech.
 * See section 6.2.1.
 *
 * @param p             pointer to the AMRContext
 * @param lpc           interpolated LP coefficients for this subframe
 * @param buf_out       output of the filter
 */
static void postfilter(AMRContext *p, float *lpc, float *buf_out)
{
    int i;
    float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input

    float speech_gain       = ff_scalarproduct_float_c(samples, samples,
                                                       AMR_SUBFRAME_SIZE);

    float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
    const float *gamma_n, *gamma_d;                       // Formant filter factor table
    float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients

    if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
        gamma_n = ff_pow_0_7;
        gamma_d = ff_pow_0_75;
    } else {
        gamma_n = ff_pow_0_55;
        gamma_d = ff_pow_0_7;
    }

    for (i = 0; i < LP_FILTER_ORDER; i++) {
         lpc_n[i] = lpc[i] * gamma_n[i];
         lpc_d[i] = lpc[i] * gamma_d[i];
    }

    memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
    ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
                                 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
    memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
           sizeof(float) * LP_FILTER_ORDER);

    ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
                                      pole_out + LP_FILTER_ORDER,
                                      AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);

    ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
                         AMR_SUBFRAME_SIZE);

    ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
                             AMR_AGC_ALPHA, &p->postfilter_agc);
}

/// @}

static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
                              int *got_frame_ptr, AVPacket *avpkt)
{

    AMRContext *p = avctx->priv_data;        // pointer to private data
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    float *buf_out;                          // pointer to the output data buffer
    int i, subframe, ret;
    float fixed_gain_factor;
    AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
    float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
    float synth_fixed_gain;                  // the fixed gain that synthesis should use
    const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use

    /* get output buffer */
    p->avframe.nb_samples = AMR_BLOCK_SIZE;
    if ((ret = ff_get_buffer(avctx, &p->avframe)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return ret;
    }
    buf_out = (float *)p->avframe.data[0];

    p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
    if (p->cur_frame_mode == NO_DATA) {
        av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
        return AVERROR_INVALIDDATA;
    }
    if (p->cur_frame_mode == MODE_DTX) {
        av_log_missing_feature(avctx, "dtx mode", 1);
        return AVERROR_PATCHWELCOME;
    }

    if (p->cur_frame_mode == MODE_12k2) {
        lsf2lsp_5(p);
    } else
        lsf2lsp_3(p);

    for (i = 0; i < 4; i++)
        ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);

    for (subframe = 0; subframe < 4; subframe++) {
        const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];

        decode_pitch_vector(p, amr_subframe, subframe);

        decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
                            p->cur_frame_mode, subframe);

        // The fixed gain (section 6.1.3) depends on the fixed vector
        // (section 6.1.2), but the fixed vector calculation uses
        // pitch sharpening based on the on the pitch gain (section 6.1.3).
        // So the correct order is: pitch gain, pitch sharpening, fixed gain.
        decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
                     &fixed_gain_factor);

        pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);

        if (fixed_sparse.pitch_lag == 0) {
            av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
            return AVERROR_INVALIDDATA;
        }
        ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
                            AMR_SUBFRAME_SIZE);

        p->fixed_gain[4] =
            ff_amr_set_fixed_gain(fixed_gain_factor,
                                  ff_scalarproduct_float_c(p->fixed_vector,
                                                           p->fixed_vector,
                                                           AMR_SUBFRAME_SIZE) /
                                  AMR_SUBFRAME_SIZE,
                       p->prediction_error,
                       energy_mean[p->cur_frame_mode], energy_pred_fac);

        // The excitation feedback is calculated without any processing such
        // as fixed gain smoothing. This isn't mentioned in the specification.
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
            p->excitation[i] *= p->pitch_gain[4];
        ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
                            AMR_SUBFRAME_SIZE);

        // In the ref decoder, excitation is stored with no fractional bits.
        // This step prevents buzz in silent periods. The ref encoder can
        // emit long sequences with pitch factor greater than one. This
        // creates unwanted feedback if the excitation vector is nonzero.
        // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
            p->excitation[i] = truncf(p->excitation[i]);

        // Smooth fixed gain.
        // The specification is ambiguous, but in the reference source, the
        // smoothed value is NOT fed back into later fixed gain smoothing.
        synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
                                             p->lsf_avg, p->cur_frame_mode);

        synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
                                             synth_fixed_gain, spare_vector);

        if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
            // overflow detected -> rerun synthesis scaling pitch vector down
            // by a factor of 4, skipping pitch vector contribution emphasis
            // and adaptive gain control
            synthesis(p, p->lpc[subframe], synth_fixed_gain,
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);

        postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);

        // update buffers and history
        ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
        update_state(p);
    }

    ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
                                             highpass_poles,
                                             highpass_gain * AMR_SAMPLE_SCALE,
                                             p->high_pass_mem, AMR_BLOCK_SIZE);

    /* Update averaged lsf vector (used for fixed gain smoothing).
     *
     * Note that lsf_avg should not incorporate the current frame's LSFs
     * for fixed_gain_smooth.
     * The specification has an incorrect formula: the reference decoder uses
     * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
    ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
                            0.84, 0.16, LP_FILTER_ORDER);

    *got_frame_ptr   = 1;
    *(AVFrame *)data = p->avframe;

    /* return the amount of bytes consumed if everything was OK */
    return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
}


AVCodec ff_amrnb_decoder = {
    .name           = "amrnb",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_AMR_NB,
    .priv_data_size = sizeof(AMRContext),
    .init           = amrnb_decode_init,
    .decode         = amrnb_decode_frame,
    .capabilities   = CODEC_CAP_DR1,
    .long_name      = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
                                                     AV_SAMPLE_FMT_NONE },
};