aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/g729dec.c
blob: 33e1fb9c29d9ccb88389bcd478de23be974a888b (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
/*
 * G.729, G729 Annex D decoders
 * Copyright (c) 2008 Vladimir Voroshilov
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <inttypes.h>
#include <string.h>

#include "avcodec.h"
#include "libavutil/avutil.h"
#include "get_bits.h"
#include "audiodsp.h"
#include "codec_internal.h"
#include "decode.h"


#include "g729.h"
#include "lsp.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_pitch_delay.h"
#include "acelp_vectors.h"
#include "g729data.h"
#include "g729postfilter.h"

/**
 * minimum quantized LSF value (3.2.4)
 * 0.005 in Q13
 */
#define LSFQ_MIN                   40

/**
 * maximum quantized LSF value (3.2.4)
 * 3.135 in Q13
 */
#define LSFQ_MAX                   25681

/**
 * minimum LSF distance (3.2.4)
 * 0.0391 in Q13
 */
#define LSFQ_DIFF_MIN              321

/// interpolation filter length
#define INTERPOL_LEN              11

/**
 * minimum gain pitch value (3.8, Equation 47)
 * 0.2 in (1.14)
 */
#define SHARP_MIN                  3277

/**
 * maximum gain pitch value (3.8, Equation 47)
 * (EE) This does not comply with the specification.
 * Specification says about 0.8, which should be
 * 13107 in (1.14), but reference C code uses
 * 13017 (equals to 0.7945) instead of it.
 */
#define SHARP_MAX                  13017

/**
 * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26  * subframe_size) in (7.13)
 */
#define MR_ENERGY 1018156

#define DECISION_NOISE        0
#define DECISION_INTERMEDIATE 1
#define DECISION_VOICE        2

typedef enum {
    FORMAT_G729_8K = 0,
    FORMAT_G729D_6K4,
    FORMAT_COUNT,
} G729Formats;

typedef struct {
    uint8_t ac_index_bits[2];   ///< adaptive codebook index for second subframe (size in bits)
    uint8_t parity_bit;         ///< parity bit for pitch delay
    uint8_t gc_1st_index_bits;  ///< gain codebook (first stage) index (size in bits)
    uint8_t gc_2nd_index_bits;  ///< gain codebook (second stage) index (size in bits)
    uint8_t fc_signs_bits;      ///< number of pulses in fixed-codebook vector
    uint8_t fc_indexes_bits;    ///< size (in bits) of fixed-codebook index entry
    uint8_t block_size;
} G729FormatDescription;

typedef struct {
    /// past excitation signal buffer
    int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];

    int16_t* exc;               ///< start of past excitation data in buffer
    int pitch_delay_int_prev;   ///< integer part of previous subframe's pitch delay (4.1.3)

    /// (2.13) LSP quantizer outputs
    int16_t  past_quantizer_output_buf[MA_NP + 1][10];
    int16_t* past_quantizer_outputs[MA_NP + 1];

    int16_t lsfq[10];           ///< (2.13) quantized LSF coefficients from previous frame
    int16_t lsp_buf[2][10];     ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
    int16_t *lsp[2];            ///< pointers to lsp_buf

    int16_t quant_energy[4];    ///< (5.10) past quantized energy

    /// previous speech data for LP synthesis filter
    int16_t syn_filter_data[10];


    /// residual signal buffer (used in long-term postfilter)
    int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];

    /// previous speech data for residual calculation filter
    int16_t res_filter_data[SUBFRAME_SIZE+10];

    /// previous speech data for short-term postfilter
    int16_t pos_filter_data[SUBFRAME_SIZE+10];

    /// (1.14) pitch gain of current and five previous subframes
    int16_t past_gain_pitch[6];

    /// (14.1) gain code from current and previous subframe
    int16_t past_gain_code[2];

    /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
    int16_t voice_decision;

    int16_t onset;              ///< detected onset level (0-2)
    int16_t was_periodic;       ///< whether previous frame was declared as periodic or not (4.4)
    int16_t ht_prev_data;       ///< previous data for 4.2.3, equation 86
    int gain_coeff;             ///< (1.14) gain coefficient (4.2.4)
    uint16_t rand_value;        ///< random number generator value (4.4.4)
    int ma_predictor_prev;      ///< switched MA predictor of LSP quantizer from last good frame

    /// (14.14) high-pass filter data (past input)
    int hpf_f[2];

    /// high-pass filter data (past output)
    int16_t hpf_z[2];
}  G729ChannelContext;

typedef struct {
    AudioDSPContext adsp;

    G729ChannelContext *channel_context;
} G729Context;

static const G729FormatDescription format_g729_8k = {
    .ac_index_bits     = {8,5},
    .parity_bit        = 1,
    .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
    .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
    .fc_signs_bits     = 4,
    .fc_indexes_bits   = 13,
    .block_size        = G729_8K_BLOCK_SIZE,
};

static const G729FormatDescription format_g729d_6k4 = {
    .ac_index_bits     = {8,4},
    .parity_bit        = 0,
    .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
    .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
    .fc_signs_bits     = 2,
    .fc_indexes_bits   = 9,
    .block_size        = G729D_6K4_BLOCK_SIZE,
};

/**
 * @brief pseudo random number generator
 */
static inline uint16_t g729_prng(uint16_t value)
{
    return 31821 * value + 13849;
}

/**
 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
 * @param[out] lsfq (2.13) quantized LSF coefficients
 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
 * @param ma_predictor switched MA predictor of LSP quantizer
 * @param vq_1st first stage vector of quantizer
 * @param vq_2nd_low second stage lower vector of LSP quantizer
 * @param vq_2nd_high second stage higher vector of LSP quantizer
 */
static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
                       int16_t ma_predictor,
                       int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
{
    int i,j;
    static const uint8_t min_distance[2]={10, 5}; //(2.13)
    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];

    for (i = 0; i < 5; i++) {
        quantizer_output[i]     = cb_lsp_1st[vq_1st][i    ] + cb_lsp_2nd[vq_2nd_low ][i    ];
        quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
    }

    for (j = 0; j < 2; j++) {
        for (i = 1; i < 10; i++) {
            int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
            if (diff > 0) {
                quantizer_output[i - 1] -= diff;
                quantizer_output[i    ] += diff;
            }
        }
    }

    for (i = 0; i < 10; i++) {
        int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
        for (j = 0; j < MA_NP; j++)
            sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];

        lsfq[i] = sum >> 15;
    }

    ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
}

/**
 * Restores past LSP quantizer output using LSF from previous frame
 * @param[in,out] lsfq (2.13) quantized LSF coefficients
 * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
 * @param ma_predictor_prev MA predictor from previous frame
 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
 */
static void lsf_restore_from_previous(int16_t* lsfq,
                                      int16_t* past_quantizer_outputs[MA_NP + 1],
                                      int ma_predictor_prev)
{
    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
    int i,k;

    for (i = 0; i < 10; i++) {
        int tmp = lsfq[i] << 15;

        for (k = 0; k < MA_NP; k++)
            tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];

        quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
    }
}

/**
 * Constructs new excitation signal and applies phase filter to it
 * @param[out] out constructed speech signal
 * @param in original excitation signal
 * @param fc_cur (2.13) original fixed-codebook vector
 * @param gain_code (14.1) gain code
 * @param subframe_size length of the subframe
 */
static void g729d_get_new_exc(
        int16_t* out,
        const int16_t* in,
        const int16_t* fc_cur,
        int dstate,
        int gain_code,
        int subframe_size)
{
    int i;
    int16_t fc_new[SUBFRAME_SIZE];

    ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);

    for (i = 0; i < subframe_size; i++) {
        out[i]  = in[i];
        out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
        out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
    }
}

/**
 * Makes decision about onset in current subframe
 * @param past_onset decision result of previous subframe
 * @param past_gain_code gain code of current and previous subframe
 *
 * @return onset decision result for current subframe
 */
static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
{
    if ((past_gain_code[0] >> 1) > past_gain_code[1])
        return 2;

    return FFMAX(past_onset-1, 0);
}

/**
 * Makes decision about voice presence in current subframe
 * @param onset onset level
 * @param prev_voice_decision voice decision result from previous subframe
 * @param past_gain_pitch pitch gain of current and previous subframes
 *
 * @return voice decision result for current subframe
 */
static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
{
    int i, low_gain_pitch_cnt, voice_decision;

    if (past_gain_pitch[0] >= 14745) {       // 0.9
        voice_decision = DECISION_VOICE;
    } else if (past_gain_pitch[0] <= 9830) { // 0.6
        voice_decision = DECISION_NOISE;
    } else {
        voice_decision = DECISION_INTERMEDIATE;
    }

    for (i = 0, low_gain_pitch_cnt = 0; i < 6; i++)
        if (past_gain_pitch[i] < 9830)
            low_gain_pitch_cnt++;

    if (low_gain_pitch_cnt > 2 && !onset)
        voice_decision = DECISION_NOISE;

    if (!onset && voice_decision > prev_voice_decision + 1)
        voice_decision--;

    if (onset && voice_decision < DECISION_VOICE)
        voice_decision++;

    return voice_decision;
}

static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
{
    int64_t res = 0;

    while (order--)
        res += *v1++ * *v2++;

    if      (res > INT32_MAX) return INT32_MAX;
    else if (res < INT32_MIN) return INT32_MIN;

    return res;
}

static av_cold int decoder_init(AVCodecContext * avctx)
{
    G729Context *s = avctx->priv_data;
    G729ChannelContext *ctx;
    int channels = avctx->ch_layout.nb_channels;
    int c,i,k;

    if (channels < 1 || channels > 2) {
        av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", channels);
        return AVERROR(EINVAL);
    }
    avctx->sample_fmt = AV_SAMPLE_FMT_S16P;

    /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
    avctx->frame_size = SUBFRAME_SIZE << 1;

    ctx =
    s->channel_context = av_mallocz(sizeof(G729ChannelContext) * channels);
    if (!ctx)
        return AVERROR(ENOMEM);

    for (c = 0; c < channels; c++) {
        ctx->gain_coeff = 16384; // 1.0 in (1.14)

        for (k = 0; k < MA_NP + 1; k++) {
            ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
            for (i = 1; i < 11; i++)
                ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
        }

        ctx->lsp[0] = ctx->lsp_buf[0];
        ctx->lsp[1] = ctx->lsp_buf[1];
        memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));

        ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];

        ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;

        /* random seed initialization */
        ctx->rand_value = 21845;

        /* quantized prediction error */
        for (i = 0; i < 4; i++)
            ctx->quant_energy[i] = -14336; // -14 in (5.10)

        ctx++;
    }

    ff_audiodsp_init(&s->adsp);
    s->adsp.scalarproduct_int16 = scalarproduct_int16_c;

    return 0;
}

static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
                        int *got_frame_ptr, AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    int16_t *out_frame;
    GetBitContext gb;
    const G729FormatDescription *format;
    int c, i;
    int16_t *tmp;
    G729Formats packet_type;
    G729Context *s = avctx->priv_data;
    G729ChannelContext *ctx = s->channel_context;
    int channels = avctx->ch_layout.nb_channels;
    int16_t lp[2][11];           // (3.12)
    uint8_t ma_predictor;     ///< switched MA predictor of LSP quantizer
    uint8_t quantizer_1st;    ///< first stage vector of quantizer
    uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
    uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)

    int pitch_delay_int[2];      // pitch delay, integer part
    int pitch_delay_3x;          // pitch delay, multiplied by 3
    int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
    int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
    int j, ret;
    int gain_before, gain_after;

    frame->nb_samples = SUBFRAME_SIZE<<1;
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;

    if (buf_size && buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels) == 0) {
        packet_type = FORMAT_G729_8K;
        format = &format_g729_8k;
        //Reset voice decision
        ctx->onset = 0;
        ctx->voice_decision = DECISION_VOICE;
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
    } else if (buf_size == G729D_6K4_BLOCK_SIZE * channels && avctx->codec_id != AV_CODEC_ID_ACELP_KELVIN) {
        packet_type = FORMAT_G729D_6K4;
        format = &format_g729d_6k4;
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
    } else {
        av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
        return AVERROR_INVALIDDATA;
    }

    for (c = 0; c < channels; c++) {
        int frame_erasure = 0; ///< frame erasure detected during decoding
        int bad_pitch = 0;     ///< parity check failed
        int is_periodic = 0;   ///< whether one of the subframes is declared as periodic or not
        out_frame = (int16_t*)frame->data[c];
        if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
            if (*buf != ((avctx->ch_layout.nb_channels - 1 - c) * 0x80 | 2))
                avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
            buf++;
        }

        for (i = 0; i < format->block_size; i++)
            frame_erasure |= buf[i];
        frame_erasure = !frame_erasure;

        init_get_bits8(&gb, buf, format->block_size);

        ma_predictor     = get_bits(&gb, 1);
        quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
        quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
        quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);

        if (frame_erasure) {
            lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
                                      ctx->ma_predictor_prev);
        } else {
            lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
                       ma_predictor,
                       quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
            ctx->ma_predictor_prev = ma_predictor;
        }

        tmp = ctx->past_quantizer_outputs[MA_NP];
        memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
                MA_NP * sizeof(int16_t*));
        ctx->past_quantizer_outputs[0] = tmp;

        ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);

        ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);

        FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);

        for (i = 0; i < 2; i++) {
            int gain_corr_factor;

            uint8_t ac_index;      ///< adaptive codebook index
            uint8_t pulses_signs;  ///< fixed-codebook vector pulse signs
            int fc_indexes;        ///< fixed-codebook indexes
            uint8_t gc_1st_index;  ///< gain codebook (first stage) index
            uint8_t gc_2nd_index;  ///< gain codebook (second stage) index

            ac_index      = get_bits(&gb, format->ac_index_bits[i]);
            if (!i && format->parity_bit)
                bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
            fc_indexes    = get_bits(&gb, format->fc_indexes_bits);
            pulses_signs  = get_bits(&gb, format->fc_signs_bits);
            gc_1st_index  = get_bits(&gb, format->gc_1st_index_bits);
            gc_2nd_index  = get_bits(&gb, format->gc_2nd_index_bits);

            if (frame_erasure) {
                pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
            } else if (!i) {
                if (bad_pitch) {
                    pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
                } else {
                    pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
                }
            } else {
                int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
                                              PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);

                if (packet_type == FORMAT_G729D_6K4) {
                    pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
                } else {
                    pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
                }
            }

            /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
            pitch_delay_int[i]  = (pitch_delay_3x + 1) / 3;
            if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
                av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
                pitch_delay_int[i] = PITCH_DELAY_MAX;
            }

            if (frame_erasure) {
                ctx->rand_value = g729_prng(ctx->rand_value);
                fc_indexes   = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);

                ctx->rand_value = g729_prng(ctx->rand_value);
                pulses_signs = ctx->rand_value;
            }


            memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
            switch (packet_type) {
                case FORMAT_G729_8K:
                    ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
                                                ff_fc_4pulses_8bits_track_4,
                                                fc_indexes, pulses_signs, 3, 3);
                    break;
                case FORMAT_G729D_6K4:
                    ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
                                                ff_fc_2pulses_9bits_track2_gray,
                                                fc_indexes, pulses_signs, 1, 4);
                    break;
            }

            /*
              This filter enhances harmonic components of the fixed-codebook vector to
              improve the quality of the reconstructed speech.

                         / fc_v[i],                                    i < pitch_delay
              fc_v[i] = <
                         \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
            */
            if (SUBFRAME_SIZE > pitch_delay_int[i])
                ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
                                             fc + pitch_delay_int[i],
                                             fc, 1 << 14,
                                             av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
                                             0, 14,
                                             SUBFRAME_SIZE - pitch_delay_int[i]);

            memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
            ctx->past_gain_code[1] = ctx->past_gain_code[0];

            if (frame_erasure) {
                ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
                ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)

                gain_corr_factor = 0;
            } else {
                if (packet_type == FORMAT_G729D_6K4) {
                    ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
                                               cb_gain_2nd_6k4[gc_2nd_index][0];
                    gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
                                       cb_gain_2nd_6k4[gc_2nd_index][1];

                    /* Without check below overflow can occur in ff_acelp_update_past_gain.
                       It is not issue for G.729, because gain_corr_factor in it's case is always
                       greater than 1024, while in G.729D it can be even zero. */
                    gain_corr_factor = FFMAX(gain_corr_factor, 1024);
    #ifndef G729_BITEXACT
                    gain_corr_factor >>= 1;
    #endif
                } else {
                    ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
                                               cb_gain_2nd_8k[gc_2nd_index][0];
                    gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
                                       cb_gain_2nd_8k[gc_2nd_index][1];
                }

                /* Decode the fixed-codebook gain. */
                ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
                                                                   fc, MR_ENERGY,
                                                                   ctx->quant_energy,
                                                                   ma_prediction_coeff,
                                                                   SUBFRAME_SIZE, 4);
    #ifdef G729_BITEXACT
                /*
                  This correction required to get bit-exact result with
                  reference code, because gain_corr_factor in G.729D is
                  two times larger than in original G.729.

                  If bit-exact result is not issue then gain_corr_factor
                  can be simpler divided by 2 before call to g729_get_gain_code
                  instead of using correction below.
                */
                if (packet_type == FORMAT_G729D_6K4) {
                    gain_corr_factor >>= 1;
                    ctx->past_gain_code[0] >>= 1;
                }
    #endif
            }
            ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);

            /* Routine requires rounding to lowest. */
            ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
                                 ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
                                 ff_acelp_interp_filter, 6,
                                 (pitch_delay_3x % 3) << 1,
                                 10, SUBFRAME_SIZE);

            ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
                                         ctx->exc + i * SUBFRAME_SIZE, fc,
                                         (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
                                         ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
                                         1 << 13, 14, SUBFRAME_SIZE);

            memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));

            if (ff_celp_lp_synthesis_filter(
                synth+10,
                &lp[i][1],
                ctx->exc  + i * SUBFRAME_SIZE,
                SUBFRAME_SIZE,
                10,
                1,
                0,
                0x800))
                /* Overflow occurred, downscale excitation signal... */
                for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
                    ctx->exc_base[j] >>= 2;

            /* ... and make synthesis again. */
            if (packet_type == FORMAT_G729D_6K4) {
                int16_t exc_new[SUBFRAME_SIZE];

                ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
                ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);

                g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);

                ff_celp_lp_synthesis_filter(
                        synth+10,
                        &lp[i][1],
                        exc_new,
                        SUBFRAME_SIZE,
                        10,
                        0,
                        0,
                        0x800);
            } else {
                ff_celp_lp_synthesis_filter(
                        synth+10,
                        &lp[i][1],
                        ctx->exc  + i * SUBFRAME_SIZE,
                        SUBFRAME_SIZE,
                        10,
                        0,
                        0,
                        0x800);
            }
            /* Save data (without postfilter) for use in next subframe. */
            memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));

            /* Calculate gain of unfiltered signal for use in AGC. */
            gain_before = 0;
            for (j = 0; j < SUBFRAME_SIZE; j++)
                gain_before += FFABS(synth[j+10]);

            /* Call postfilter and also update voicing decision for use in next frame. */
            ff_g729_postfilter(
                    &s->adsp,
                    &ctx->ht_prev_data,
                    &is_periodic,
                    &lp[i][0],
                    pitch_delay_int[0],
                    ctx->residual,
                    ctx->res_filter_data,
                    ctx->pos_filter_data,
                    synth+10,
                    SUBFRAME_SIZE);

            /* Calculate gain of filtered signal for use in AGC. */
            gain_after = 0;
            for (j = 0; j < SUBFRAME_SIZE; j++)
                gain_after += FFABS(synth[j+10]);

            ctx->gain_coeff = ff_g729_adaptive_gain_control(
                    gain_before,
                    gain_after,
                    synth+10,
                    SUBFRAME_SIZE,
                    ctx->gain_coeff);

            if (frame_erasure) {
                ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
            } else {
                ctx->pitch_delay_int_prev = pitch_delay_int[i];
            }

            memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
            ff_acelp_high_pass_filter(
                    out_frame + i*SUBFRAME_SIZE,
                    ctx->hpf_f,
                    synth+10,
                    SUBFRAME_SIZE);
            memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
        }

        ctx->was_periodic = is_periodic;

        /* Save signal for use in next frame. */
        memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));

        buf += format->block_size;
        ctx++;
    }

    *got_frame_ptr = 1;
    return (format->block_size + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * channels;
}

static av_cold int decode_close(AVCodecContext *avctx)
{
    G729Context *s = avctx->priv_data;
    av_freep(&s->channel_context);

    return 0;
}

const FFCodec ff_g729_decoder = {
    .p.name         = "g729",
    CODEC_LONG_NAME("G.729"),
    .p.type         = AVMEDIA_TYPE_AUDIO,
    .p.id           = AV_CODEC_ID_G729,
    .priv_data_size = sizeof(G729Context),
    .init           = decoder_init,
    FF_CODEC_DECODE_CB(decode_frame),
    .close          = decode_close,
    .p.capabilities =
#if FF_API_SUBFRAMES
                      AV_CODEC_CAP_SUBFRAMES |
#endif
                      AV_CODEC_CAP_DR1,
};

const FFCodec ff_acelp_kelvin_decoder = {
    .p.name         = "acelp.kelvin",
    CODEC_LONG_NAME("Sipro ACELP.KELVIN"),
    .p.type         = AVMEDIA_TYPE_AUDIO,
    .p.id           = AV_CODEC_ID_ACELP_KELVIN,
    .priv_data_size = sizeof(G729Context),
    .init           = decoder_init,
    FF_CODEC_DECODE_CB(decode_frame),
    .close          = decode_close,
    .p.capabilities =
#if FF_API_SUBFRAMES
                      AV_CODEC_CAP_SUBFRAMES |
#endif
                      AV_CODEC_CAP_DR1,
};