aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/atrac3.c
blob: b4809111f8a4f7fb1f1919ef3284b305f4a500ef (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
/*
 * ATRAC3 compatible decoder
 * Copyright (c) 2006-2008 Maxim Poliakovski
 * Copyright (c) 2006-2008 Benjamin Larsson
 *
 * 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
 * ATRAC3 compatible decoder.
 * This decoder handles Sony's ATRAC3 data.
 *
 * Container formats used to store ATRAC3 data:
 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
 *
 * To use this decoder, a calling application must supply the extradata
 * bytes provided in the containers above.
 */

#include <math.h>
#include <stddef.h>
#include <stdio.h>

#include "libavutil/attributes.h"
#include "libavutil/float_dsp.h"
#include "libavutil/libm.h"
#include "libavutil/mem_internal.h"
#include "libavutil/thread.h"

#include "avcodec.h"
#include "bytestream.h"
#include "fft.h"
#include "get_bits.h"
#include "internal.h"

#include "atrac.h"
#include "atrac3data.h"

#define MIN_CHANNELS    1
#define MAX_CHANNELS    8
#define MAX_JS_PAIRS    8 / 2

#define JOINT_STEREO    0x12
#define SINGLE          0x2

#define SAMPLES_PER_FRAME 1024
#define MDCT_SIZE          512

#define ATRAC3_VLC_BITS 8

typedef struct GainBlock {
    AtracGainInfo g_block[4];
} GainBlock;

typedef struct TonalComponent {
    int pos;
    int num_coefs;
    float coef[8];
} TonalComponent;

typedef struct ChannelUnit {
    int            bands_coded;
    int            num_components;
    float          prev_frame[SAMPLES_PER_FRAME];
    int            gc_blk_switch;
    TonalComponent components[64];
    GainBlock      gain_block[2];

    DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
    DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];

    float          delay_buf1[46]; ///<qmf delay buffers
    float          delay_buf2[46];
    float          delay_buf3[46];
} ChannelUnit;

typedef struct ATRAC3Context {
    GetBitContext gb;
    //@{
    /** stream data */
    int coding_mode;

    ChannelUnit *units;
    //@}
    //@{
    /** joint-stereo related variables */
    int matrix_coeff_index_prev[MAX_JS_PAIRS][4];
    int matrix_coeff_index_now[MAX_JS_PAIRS][4];
    int matrix_coeff_index_next[MAX_JS_PAIRS][4];
    int weighting_delay[MAX_JS_PAIRS][6];
    //@}
    //@{
    /** data buffers */
    uint8_t *decoded_bytes_buffer;
    float temp_buf[1070];
    //@}
    //@{
    /** extradata */
    int scrambled_stream;
    //@}

    AtracGCContext    gainc_ctx;
    FFTContext        mdct_ctx;
    void (*vector_fmul)(float *dst, const float *src0, const float *src1,
                        int len);
} ATRAC3Context;

static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
static VLC_TYPE atrac3_vlc_table[7 * 1 << ATRAC3_VLC_BITS][2];
static VLC   spectral_coeff_tab[7];

/**
 * Regular 512 points IMDCT without overlapping, with the exception of the
 * swapping of odd bands caused by the reverse spectra of the QMF.
 *
 * @param odd_band  1 if the band is an odd band
 */
static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
{
    int i;

    if (odd_band) {
        /**
         * Reverse the odd bands before IMDCT, this is an effect of the QMF
         * transform or it gives better compression to do it this way.
         * FIXME: It should be possible to handle this in imdct_calc
         * for that to happen a modification of the prerotation step of
         * all SIMD code and C code is needed.
         * Or fix the functions before so they generate a pre reversed spectrum.
         */
        for (i = 0; i < 128; i++)
            FFSWAP(float, input[i], input[255 - i]);
    }

    q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);

    /* Perform windowing on the output. */
    q->vector_fmul(output, output, mdct_window, MDCT_SIZE);
}

/*
 * indata descrambling, only used for data coming from the rm container
 */
static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
{
    int i, off;
    uint32_t c;
    const uint32_t *buf;
    uint32_t *output = (uint32_t *)out;

    off = (intptr_t)input & 3;
    buf = (const uint32_t *)(input - off);
    if (off)
        c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
    else
        c = av_be2ne32(0x537F6103U);
    bytes += 3 + off;
    for (i = 0; i < bytes / 4; i++)
        output[i] = c ^ buf[i];

    if (off)
        avpriv_request_sample(NULL, "Offset of %d", off);

    return off;
}

static av_cold void init_imdct_window(void)
{
    int i, j;

    /* generate the mdct window, for details see
     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
    for (i = 0, j = 255; i < 128; i++, j--) {
        float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
        float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
        float w  = 0.5 * (wi * wi + wj * wj);
        mdct_window[i] = mdct_window[511 - i] = wi / w;
        mdct_window[j] = mdct_window[511 - j] = wj / w;
    }
}

static av_cold int atrac3_decode_close(AVCodecContext *avctx)
{
    ATRAC3Context *q = avctx->priv_data;

    av_freep(&q->units);
    av_freep(&q->decoded_bytes_buffer);

    ff_mdct_end(&q->mdct_ctx);

    return 0;
}

/**
 * Mantissa decoding
 *
 * @param selector     which table the output values are coded with
 * @param coding_flag  constant length coding or variable length coding
 * @param mantissas    mantissa output table
 * @param num_codes    number of values to get
 */
static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
                                       int coding_flag, int *mantissas,
                                       int num_codes)
{
    int i, code, huff_symb;

    if (selector == 1)
        num_codes /= 2;

    if (coding_flag != 0) {
        /* constant length coding (CLC) */
        int num_bits = clc_length_tab[selector];

        if (selector > 1) {
            for (i = 0; i < num_codes; i++) {
                if (num_bits)
                    code = get_sbits(gb, num_bits);
                else
                    code = 0;
                mantissas[i] = code;
            }
        } else {
            for (i = 0; i < num_codes; i++) {
                if (num_bits)
                    code = get_bits(gb, num_bits); // num_bits is always 4 in this case
                else
                    code = 0;
                mantissas[i * 2    ] = mantissa_clc_tab[code >> 2];
                mantissas[i * 2 + 1] = mantissa_clc_tab[code &  3];
            }
        }
    } else {
        /* variable length coding (VLC) */
        if (selector != 1) {
            for (i = 0; i < num_codes; i++) {
                mantissas[i] = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
                                        ATRAC3_VLC_BITS, 1);
            }
        } else {
            for (i = 0; i < num_codes; i++) {
                huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
                                     ATRAC3_VLC_BITS, 1);
                mantissas[i * 2    ] = mantissa_vlc_tab[huff_symb * 2    ];
                mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
            }
        }
    }
}

/**
 * Restore the quantized band spectrum coefficients
 *
 * @return subband count, fix for broken specification/files
 */
static int decode_spectrum(GetBitContext *gb, float *output)
{
    int num_subbands, coding_mode, i, j, first, last, subband_size;
    int subband_vlc_index[32], sf_index[32];
    int mantissas[128];
    float scale_factor;

    num_subbands = get_bits(gb, 5);  // number of coded subbands
    coding_mode  = get_bits1(gb);    // coding Mode: 0 - VLC/ 1-CLC

    /* get the VLC selector table for the subbands, 0 means not coded */
    for (i = 0; i <= num_subbands; i++)
        subband_vlc_index[i] = get_bits(gb, 3);

    /* read the scale factor indexes from the stream */
    for (i = 0; i <= num_subbands; i++) {
        if (subband_vlc_index[i] != 0)
            sf_index[i] = get_bits(gb, 6);
    }

    for (i = 0; i <= num_subbands; i++) {
        first = subband_tab[i    ];
        last  = subband_tab[i + 1];

        subband_size = last - first;

        if (subband_vlc_index[i] != 0) {
            /* decode spectral coefficients for this subband */
            /* TODO: This can be done faster is several blocks share the
             * same VLC selector (subband_vlc_index) */
            read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
                                       mantissas, subband_size);

            /* decode the scale factor for this subband */
            scale_factor = ff_atrac_sf_table[sf_index[i]] *
                           inv_max_quant[subband_vlc_index[i]];

            /* inverse quantize the coefficients */
            for (j = 0; first < last; first++, j++)
                output[first] = mantissas[j] * scale_factor;
        } else {
            /* this subband was not coded, so zero the entire subband */
            memset(output + first, 0, subband_size * sizeof(*output));
        }
    }

    /* clear the subbands that were not coded */
    first = subband_tab[i];
    memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
    return num_subbands;
}

/**
 * Restore the quantized tonal components
 *
 * @param components tonal components
 * @param num_bands  number of coded bands
 */
static int decode_tonal_components(GetBitContext *gb,
                                   TonalComponent *components, int num_bands)
{
    int i, b, c, m;
    int nb_components, coding_mode_selector, coding_mode;
    int band_flags[4], mantissa[8];
    int component_count = 0;

    nb_components = get_bits(gb, 5);

    /* no tonal components */
    if (nb_components == 0)
        return 0;

    coding_mode_selector = get_bits(gb, 2);
    if (coding_mode_selector == 2)
        return AVERROR_INVALIDDATA;

    coding_mode = coding_mode_selector & 1;

    for (i = 0; i < nb_components; i++) {
        int coded_values_per_component, quant_step_index;

        for (b = 0; b <= num_bands; b++)
            band_flags[b] = get_bits1(gb);

        coded_values_per_component = get_bits(gb, 3);

        quant_step_index = get_bits(gb, 3);
        if (quant_step_index <= 1)
            return AVERROR_INVALIDDATA;

        if (coding_mode_selector == 3)
            coding_mode = get_bits1(gb);

        for (b = 0; b < (num_bands + 1) * 4; b++) {
            int coded_components;

            if (band_flags[b >> 2] == 0)
                continue;

            coded_components = get_bits(gb, 3);

            for (c = 0; c < coded_components; c++) {
                TonalComponent *cmp = &components[component_count];
                int sf_index, coded_values, max_coded_values;
                float scale_factor;

                sf_index = get_bits(gb, 6);
                if (component_count >= 64)
                    return AVERROR_INVALIDDATA;

                cmp->pos = b * 64 + get_bits(gb, 6);

                max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
                coded_values     = coded_values_per_component + 1;
                coded_values     = FFMIN(max_coded_values, coded_values);

                scale_factor = ff_atrac_sf_table[sf_index] *
                               inv_max_quant[quant_step_index];

                read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
                                           mantissa, coded_values);

                cmp->num_coefs = coded_values;

                /* inverse quant */
                for (m = 0; m < coded_values; m++)
                    cmp->coef[m] = mantissa[m] * scale_factor;

                component_count++;
            }
        }
    }

    return component_count;
}

/**
 * Decode gain parameters for the coded bands
 *
 * @param block      the gainblock for the current band
 * @param num_bands  amount of coded bands
 */
static int decode_gain_control(GetBitContext *gb, GainBlock *block,
                               int num_bands)
{
    int b, j;
    int *level, *loc;

    AtracGainInfo *gain = block->g_block;

    for (b = 0; b <= num_bands; b++) {
        gain[b].num_points = get_bits(gb, 3);
        level              = gain[b].lev_code;
        loc                = gain[b].loc_code;

        for (j = 0; j < gain[b].num_points; j++) {
            level[j] = get_bits(gb, 4);
            loc[j]   = get_bits(gb, 5);
            if (j && loc[j] <= loc[j - 1])
                return AVERROR_INVALIDDATA;
        }
    }

    /* Clear the unused blocks. */
    for (; b < 4 ; b++)
        gain[b].num_points = 0;

    return 0;
}

/**
 * Combine the tonal band spectrum and regular band spectrum
 *
 * @param spectrum        output spectrum buffer
 * @param num_components  number of tonal components
 * @param components      tonal components for this band
 * @return                position of the last tonal coefficient
 */
static int add_tonal_components(float *spectrum, int num_components,
                                TonalComponent *components)
{
    int i, j, last_pos = -1;
    float *input, *output;

    for (i = 0; i < num_components; i++) {
        last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
        input    = components[i].coef;
        output   = &spectrum[components[i].pos];

        for (j = 0; j < components[i].num_coefs; j++)
            output[j] += input[j];
    }

    return last_pos;
}

#define INTERPOLATE(old, new, nsample) \
    ((old) + (nsample) * 0.125 * ((new) - (old)))

static void reverse_matrixing(float *su1, float *su2, int *prev_code,
                              int *curr_code)
{
    int i, nsample, band;
    float mc1_l, mc1_r, mc2_l, mc2_r;

    for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
        int s1 = prev_code[i];
        int s2 = curr_code[i];
        nsample = band;

        if (s1 != s2) {
            /* Selector value changed, interpolation needed. */
            mc1_l = matrix_coeffs[s1 * 2    ];
            mc1_r = matrix_coeffs[s1 * 2 + 1];
            mc2_l = matrix_coeffs[s2 * 2    ];
            mc2_r = matrix_coeffs[s2 * 2 + 1];

            /* Interpolation is done over the first eight samples. */
            for (; nsample < band + 8; nsample++) {
                float c1 = su1[nsample];
                float c2 = su2[nsample];
                c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
                     c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
                su1[nsample] = c2;
                su2[nsample] = c1 * 2.0 - c2;
            }
        }

        /* Apply the matrix without interpolation. */
        switch (s2) {
        case 0:     /* M/S decoding */
            for (; nsample < band + 256; nsample++) {
                float c1 = su1[nsample];
                float c2 = su2[nsample];
                su1[nsample] =  c2       * 2.0;
                su2[nsample] = (c1 - c2) * 2.0;
            }
            break;
        case 1:
            for (; nsample < band + 256; nsample++) {
                float c1 = su1[nsample];
                float c2 = su2[nsample];
                su1[nsample] = (c1 + c2) *  2.0;
                su2[nsample] =  c2       * -2.0;
            }
            break;
        case 2:
        case 3:
            for (; nsample < band + 256; nsample++) {
                float c1 = su1[nsample];
                float c2 = su2[nsample];
                su1[nsample] = c1 + c2;
                su2[nsample] = c1 - c2;
            }
            break;
        default:
            av_assert1(0);
        }
    }
}

static void get_channel_weights(int index, int flag, float ch[2])
{
    if (index == 7) {
        ch[0] = 1.0;
        ch[1] = 1.0;
    } else {
        ch[0] = (index & 7) / 7.0;
        ch[1] = sqrt(2 - ch[0] * ch[0]);
        if (flag)
            FFSWAP(float, ch[0], ch[1]);
    }
}

static void channel_weighting(float *su1, float *su2, int *p3)
{
    int band, nsample;
    /* w[x][y] y=0 is left y=1 is right */
    float w[2][2];

    if (p3[1] != 7 || p3[3] != 7) {
        get_channel_weights(p3[1], p3[0], w[0]);
        get_channel_weights(p3[3], p3[2], w[1]);

        for (band = 256; band < 4 * 256; band += 256) {
            for (nsample = band; nsample < band + 8; nsample++) {
                su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
                su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
            }
            for(; nsample < band + 256; nsample++) {
                su1[nsample] *= w[1][0];
                su2[nsample] *= w[1][1];
            }
        }
    }
}

/**
 * Decode a Sound Unit
 *
 * @param snd           the channel unit to be used
 * @param output        the decoded samples before IQMF in float representation
 * @param channel_num   channel number
 * @param coding_mode   the coding mode (JOINT_STEREO or single channels)
 */
static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
                                     ChannelUnit *snd, float *output,
                                     int channel_num, int coding_mode)
{
    int band, ret, num_subbands, last_tonal, num_bands;
    GainBlock *gain1 = &snd->gain_block[    snd->gc_blk_switch];
    GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];

    if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) {
        if (get_bits(gb, 2) != 3) {
            av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
            return AVERROR_INVALIDDATA;
        }
    } else {
        if (get_bits(gb, 6) != 0x28) {
            av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
            return AVERROR_INVALIDDATA;
        }
    }

    /* number of coded QMF bands */
    snd->bands_coded = get_bits(gb, 2);

    ret = decode_gain_control(gb, gain2, snd->bands_coded);
    if (ret)
        return ret;

    snd->num_components = decode_tonal_components(gb, snd->components,
                                                  snd->bands_coded);
    if (snd->num_components < 0)
        return snd->num_components;

    num_subbands = decode_spectrum(gb, snd->spectrum);

    /* Merge the decoded spectrum and tonal components. */
    last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
                                      snd->components);


    /* calculate number of used MLT/QMF bands according to the amount of coded
       spectral lines */
    num_bands = (subband_tab[num_subbands] - 1) >> 8;
    if (last_tonal >= 0)
        num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);


    /* Reconstruct time domain samples. */
    for (band = 0; band < 4; band++) {
        /* Perform the IMDCT step without overlapping. */
        if (band <= num_bands)
            imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
        else
            memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));

        /* gain compensation and overlapping */
        ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
                                   &snd->prev_frame[band * 256],
                                   &gain1->g_block[band], &gain2->g_block[band],
                                   256, &output[band * 256]);
    }

    /* Swap the gain control buffers for the next frame. */
    snd->gc_blk_switch ^= 1;

    return 0;
}

static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
                        float **out_samples)
{
    ATRAC3Context *q = avctx->priv_data;
    int ret, i, ch;
    uint8_t *ptr1;

    if (q->coding_mode == JOINT_STEREO) {
        /* channel coupling mode */

        /* Decode sound unit pairs (channels are expected to be even).
         * Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */
        const uint8_t *js_databuf;
        int js_pair, js_block_align;

        js_block_align = (avctx->block_align / avctx->channels) * 2; /* block pair */

        for (ch = 0; ch < avctx->channels; ch = ch + 2) {
            js_pair = ch/2;
            js_databuf = databuf + js_pair * js_block_align; /* align to current pair */

            /* Set the bitstream reader at the start of first channel sound unit. */
            init_get_bits(&q->gb,
                          js_databuf, js_block_align * 8);

            /* decode Sound Unit 1 */
            ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch],
                                            out_samples[ch], ch, JOINT_STEREO);
            if (ret != 0)
                return ret;

            /* Framedata of the su2 in the joint-stereo mode is encoded in
             * reverse byte order so we need to swap it first. */
            if (js_databuf == q->decoded_bytes_buffer) {
                uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1;
                ptr1          = q->decoded_bytes_buffer;
                for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--)
                    FFSWAP(uint8_t, *ptr1, *ptr2);
            } else {
                const uint8_t *ptr2 = js_databuf + js_block_align - 1;
                for (i = 0; i < js_block_align; i++)
                    q->decoded_bytes_buffer[i] = *ptr2--;
            }

            /* Skip the sync codes (0xF8). */
            ptr1 = q->decoded_bytes_buffer;
            for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
                if (i >= js_block_align)
                    return AVERROR_INVALIDDATA;
            }


            /* set the bitstream reader at the start of the second Sound Unit */
            ret = init_get_bits8(&q->gb,
                           ptr1, q->decoded_bytes_buffer + js_block_align - ptr1);
            if (ret < 0)
                return ret;

            /* Fill the Weighting coeffs delay buffer */
            memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2],
                    4 * sizeof(*q->weighting_delay[js_pair]));
            q->weighting_delay[js_pair][4] = get_bits1(&q->gb);
            q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3);

            for (i = 0; i < 4; i++) {
                q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i];
                q->matrix_coeff_index_now[js_pair][i]  = q->matrix_coeff_index_next[js_pair][i];
                q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2);
            }

            /* Decode Sound Unit 2. */
            ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1],
                                            out_samples[ch+1], ch+1, JOINT_STEREO);
            if (ret != 0)
                return ret;

            /* Reconstruct the channel coefficients. */
            reverse_matrixing(out_samples[ch], out_samples[ch+1],
                              q->matrix_coeff_index_prev[js_pair],
                              q->matrix_coeff_index_now[js_pair]);

            channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]);
        }
    } else {
        /* single channels */
        /* Decode the channel sound units. */
        for (i = 0; i < avctx->channels; i++) {
            /* Set the bitstream reader at the start of a channel sound unit. */
            init_get_bits(&q->gb,
                          databuf + i * avctx->block_align / avctx->channels,
                          avctx->block_align * 8 / avctx->channels);

            ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
                                            out_samples[i], i, q->coding_mode);
            if (ret != 0)
                return ret;
        }
    }

    /* Apply the iQMF synthesis filter. */
    for (i = 0; i < avctx->channels; i++) {
        float *p1 = out_samples[i];
        float *p2 = p1 + 256;
        float *p3 = p2 + 256;
        float *p4 = p3 + 256;
        ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
        ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
        ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
    }

    return 0;
}

static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
                           int size, float **out_samples)
{
    ATRAC3Context *q = avctx->priv_data;
    int ret, i;

    /* Set the bitstream reader at the start of a channel sound unit. */
    init_get_bits(&q->gb, databuf, size * 8);
    /* single channels */
    /* Decode the channel sound units. */
    for (i = 0; i < avctx->channels; i++) {
        ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
                                        out_samples[i], i, q->coding_mode);
        if (ret != 0)
            return ret;
        while (i < avctx->channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) {
            skip_bits(&q->gb, 1);
        }
    }

    /* Apply the iQMF synthesis filter. */
    for (i = 0; i < avctx->channels; i++) {
        float *p1 = out_samples[i];
        float *p2 = p1 + 256;
        float *p3 = p2 + 256;
        float *p4 = p3 + 256;
        ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
        ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
        ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
    }

    return 0;
}

static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)
{
    AVFrame *frame     = data;
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    ATRAC3Context *q = avctx->priv_data;
    int ret;
    const uint8_t *databuf;

    if (buf_size < avctx->block_align) {
        av_log(avctx, AV_LOG_ERROR,
               "Frame too small (%d bytes). Truncated file?\n", buf_size);
        return AVERROR_INVALIDDATA;
    }

    /* get output buffer */
    frame->nb_samples = SAMPLES_PER_FRAME;
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;

    /* Check if we need to descramble and what buffer to pass on. */
    if (q->scrambled_stream) {
        decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
        databuf = q->decoded_bytes_buffer;
    } else {
        databuf = buf;
    }

    ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
    if (ret) {
        av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
        return ret;
    }

    *got_frame_ptr = 1;

    return avctx->block_align;
}

static int atrac3al_decode_frame(AVCodecContext *avctx, void *data,
                                 int *got_frame_ptr, AVPacket *avpkt)
{
    AVFrame *frame = data;
    int ret;

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

    ret = al_decode_frame(avctx, avpkt->data, avpkt->size,
                          (float **)frame->extended_data);
    if (ret) {
        av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
        return ret;
    }

    *got_frame_ptr = 1;

    return avpkt->size;
}

static av_cold void atrac3_init_static_data(void)
{
    VLC_TYPE (*table)[2] = atrac3_vlc_table;
    const uint8_t (*hufftabs)[2] = atrac3_hufftabs;
    int i;

    init_imdct_window();
    ff_atrac_generate_tables();

    /* Initialize the VLC tables. */
    for (i = 0; i < 7; i++) {
        spectral_coeff_tab[i].table           = table;
        spectral_coeff_tab[i].table_allocated = 256;
        ff_init_vlc_from_lengths(&spectral_coeff_tab[i], ATRAC3_VLC_BITS, huff_tab_sizes[i],
                                 &hufftabs[0][1], 2,
                                 &hufftabs[0][0], 2, 1,
                                 -31, INIT_VLC_USE_NEW_STATIC, NULL);
        hufftabs += huff_tab_sizes[i];
        table += 256;
    }
}

static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
    static AVOnce init_static_once = AV_ONCE_INIT;
    int i, js_pair, ret;
    int version, delay, samples_per_frame, frame_factor;
    const uint8_t *edata_ptr = avctx->extradata;
    ATRAC3Context *q = avctx->priv_data;
    AVFloatDSPContext *fdsp;

    if (avctx->channels < MIN_CHANNELS || avctx->channels > MAX_CHANNELS) {
        av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
        return AVERROR(EINVAL);
    }

    /* Take care of the codec-specific extradata. */
    if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) {
        version           = 4;
        samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
        delay             = 0x88E;
        q->coding_mode    = SINGLE;
    } else if (avctx->extradata_size == 14) {
        /* Parse the extradata, WAV format */
        av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
               bytestream_get_le16(&edata_ptr));  // Unknown value always 1
        edata_ptr += 4;                             // samples per channel
        q->coding_mode = bytestream_get_le16(&edata_ptr);
        av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
               bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
        frame_factor = bytestream_get_le16(&edata_ptr);  // Unknown always 1
        av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
               bytestream_get_le16(&edata_ptr));  // Unknown always 0

        /* setup */
        samples_per_frame    = SAMPLES_PER_FRAME * avctx->channels;
        version              = 4;
        delay                = 0x88E;
        q->coding_mode       = q->coding_mode ? JOINT_STEREO : SINGLE;
        q->scrambled_stream  = 0;

        if (avctx->block_align !=  96 * avctx->channels * frame_factor &&
            avctx->block_align != 152 * avctx->channels * frame_factor &&
            avctx->block_align != 192 * avctx->channels * frame_factor) {
            av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
                   "configuration %d/%d/%d\n", avctx->block_align,
                   avctx->channels, frame_factor);
            return AVERROR_INVALIDDATA;
        }
    } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
        /* Parse the extradata, RM format. */
        version                = bytestream_get_be32(&edata_ptr);
        samples_per_frame      = bytestream_get_be16(&edata_ptr);
        delay                  = bytestream_get_be16(&edata_ptr);
        q->coding_mode         = bytestream_get_be16(&edata_ptr);
        q->scrambled_stream    = 1;

    } else {
        av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
               avctx->extradata_size);
        return AVERROR(EINVAL);
    }

    /* Check the extradata */

    if (version != 4) {
        av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
        return AVERROR_INVALIDDATA;
    }

    if (samples_per_frame != SAMPLES_PER_FRAME * avctx->channels) {
        av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
               samples_per_frame);
        return AVERROR_INVALIDDATA;
    }

    if (delay != 0x88E) {
        av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
               delay);
        return AVERROR_INVALIDDATA;
    }

    if (q->coding_mode == SINGLE)
        av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
    else if (q->coding_mode == JOINT_STEREO) {
        if (avctx->channels % 2 == 1) { /* Joint stereo channels must be even */
            av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
            return AVERROR_INVALIDDATA;
        }
        av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
    } else {
        av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
               q->coding_mode);
        return AVERROR_INVALIDDATA;
    }

    if (avctx->block_align > 4096 || avctx->block_align <= 0)
        return AVERROR(EINVAL);

    q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
                                         AV_INPUT_BUFFER_PADDING_SIZE);
    if (!q->decoded_bytes_buffer)
        return AVERROR(ENOMEM);

    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

    /* initialize the MDCT transform */
    if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
        return ret;
    }

    /* init the joint-stereo decoding data */
    for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) {
        q->weighting_delay[js_pair][0] = 0;
        q->weighting_delay[js_pair][1] = 7;
        q->weighting_delay[js_pair][2] = 0;
        q->weighting_delay[js_pair][3] = 7;
        q->weighting_delay[js_pair][4] = 0;
        q->weighting_delay[js_pair][5] = 7;

        for (i = 0; i < 4; i++) {
            q->matrix_coeff_index_prev[js_pair][i] = 3;
            q->matrix_coeff_index_now[js_pair][i]  = 3;
            q->matrix_coeff_index_next[js_pair][i] = 3;
        }
    }

    ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
    fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
    if (!fdsp)
        return AVERROR(ENOMEM);
    q->vector_fmul = fdsp->vector_fmul;
    av_free(fdsp);

    q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
    if (!q->units)
        return AVERROR(ENOMEM);

    ff_thread_once(&init_static_once, atrac3_init_static_data);

    return 0;
}

const AVCodec ff_atrac3_decoder = {
    .name             = "atrac3",
    .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
    .type             = AVMEDIA_TYPE_AUDIO,
    .id               = AV_CODEC_ID_ATRAC3,
    .priv_data_size   = sizeof(ATRAC3Context),
    .init             = atrac3_decode_init,
    .close            = atrac3_decode_close,
    .decode           = atrac3_decode_frame,
    .capabilities     = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
    .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                        AV_SAMPLE_FMT_NONE },
    .caps_internal    = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};

const AVCodec ff_atrac3al_decoder = {
    .name             = "atrac3al",
    .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
    .type             = AVMEDIA_TYPE_AUDIO,
    .id               = AV_CODEC_ID_ATRAC3AL,
    .priv_data_size   = sizeof(ATRAC3Context),
    .init             = atrac3_decode_init,
    .close            = atrac3_decode_close,
    .decode           = atrac3al_decode_frame,
    .capabilities     = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
    .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                        AV_SAMPLE_FMT_NONE },
    .caps_internal    = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};