summaryrefslogtreecommitdiff
path: root/libavcodec/atrac3.c
blob: b29778974701a4e4129aae1b0df3f82dfa65628b (plain)
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
/*
 * Atrac 3 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
 * Atrac 3 compatible decoder.
 * This decoder handles Sony's ATRAC3 data.
 *
 * Container formats used to store atrac 3 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 "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "bytestream.h"
#include "fft.h"
#include "fmtconvert.h"

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

#define JOINT_STEREO    0x12
#define STEREO          0x2

#define SAMPLES_PER_FRAME 1024
#define MDCT_SIZE          512

/* These structures are needed to store the parsed gain control data. */
typedef struct {
    int   num_gain_data;
    int   levcode[8];
    int   loccode[8];
} gain_info;

typedef struct {
    gain_info   gBlock[4];
} gain_block;

typedef struct {
    int     pos;
    int     numCoefs;
    float   coef[8];
} tonal_component;

typedef struct {
    int               bandsCoded;
    int               numComponents;
    tonal_component   components[64];
    float             prevFrame[SAMPLES_PER_FRAME];
    int               gcBlkSwitch;
    gain_block        gainBlock[2];

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

    float             delayBuf1[46]; ///<qmf delay buffers
    float             delayBuf2[46];
    float             delayBuf3[46];
} channel_unit;

typedef struct {
    AVFrame             frame;
    GetBitContext       gb;
    //@{
    /** stream data */
    int                 channels;
    int                 codingMode;
    int                 bit_rate;
    int                 sample_rate;
    int                 samples_per_channel;
    int                 samples_per_frame;

    int                 bits_per_frame;
    int                 bytes_per_frame;
    int                 pBs;
    channel_unit*       pUnits;
    //@}
    //@{
    /** joint-stereo related variables */
    int                 matrix_coeff_index_prev[4];
    int                 matrix_coeff_index_now[4];
    int                 matrix_coeff_index_next[4];
    int                 weighting_delay[6];
    //@}
    //@{
    /** data buffers */
    float              *outSamples[2];
    uint8_t*            decoded_bytes_buffer;
    float               tempBuf[1070];
    //@}
    //@{
    /** extradata */
    int                 atrac3version;
    int                 delay;
    int                 scrambled_stream;
    int                 frame_factor;
    //@}

    FFTContext          mdct_ctx;
    FmtConvertContext   fmt_conv;
} ATRAC3Context;

static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
static VLC              spectral_coeff_tab[7];
static float            gain_tab1[16];
static float            gain_tab2[31];
static DSPContext       dsp;


/**
 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
 * caused by the reverse spectra of the QMF.
 *
 * @param pInput    float input
 * @param pOutput   float output
 * @param odd_band  1 if the band is an odd band
 */

static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, 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, pInput[i], pInput[255-i]);
    }

    q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);

    /* Perform windowing on the output. */
    dsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);

}


/**
 * Atrac 3 indata descrambling, only used for data coming from the rm container
 *
 * @param inbuffer  pointer to 8 bit array of indata
 * @param out       pointer to 8 bit array of outdata
 * @param bytes     amount of bytes
 */

static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
    int i, off;
    uint32_t c;
    const uint32_t* buf;
    uint32_t* obuf = (uint32_t*) out;

    off = (intptr_t)inbuffer & 3;
    buf = (const uint32_t*) (inbuffer - off);
    c = av_be2ne32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
    bytes += 3 + off;
    for (i = 0; i < bytes/4; i++)
        obuf[i] = c ^ buf[i];

    if (off)
        av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);

    return off;
}


static av_cold int init_atrac3_transforms(ATRAC3Context *q, int is_float) {
    float enc_window[256];
    int i;

    /* Generate the mdct window, for details see
     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
    for (i=0 ; i<256; i++)
        enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;

    if (!mdct_window[0])
        for (i=0 ; i<256; i++) {
            mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
            mdct_window[511-i] = mdct_window[i];
        }

    /* Initialize the MDCT transform. */
    return ff_mdct_init(&q->mdct_ctx, 9, 1, is_float ? 1.0 / 32768 : 1.0);
}

/**
 * Atrac3 uninit, free all allocated memory
 */

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

    av_free(q->pUnits);
    av_free(q->decoded_bytes_buffer);
    av_freep(&q->outSamples[0]);

    ff_mdct_end(&q->mdct_ctx);

    return 0;
}

/**
/ * Mantissa decoding
 *
 * @param gb            the GetBit context
 * @param selector      what table is the output values coded with
 * @param codingFlag    constant length coding or variable length coding
 * @param mantissas     mantissa output table
 * @param numCodes      amount of values to get
 */

static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
{
    int   numBits, cnt, code, huffSymb;

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

    if (codingFlag != 0) {
        /* constant length coding (CLC) */
        numBits = CLCLengthTab[selector];

        if (selector > 1) {
            for (cnt = 0; cnt < numCodes; cnt++) {
                if (numBits)
                    code = get_sbits(gb, numBits);
                else
                    code = 0;
                mantissas[cnt] = code;
            }
        } else {
            for (cnt = 0; cnt < numCodes; cnt++) {
                if (numBits)
                    code = get_bits(gb, numBits); //numBits is always 4 in this case
                else
                    code = 0;
                mantissas[cnt*2] = seTab_0[code >> 2];
                mantissas[cnt*2+1] = seTab_0[code & 3];
            }
        }
    } else {
        /* variable length coding (VLC) */
        if (selector != 1) {
            for (cnt = 0; cnt < numCodes; cnt++) {
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
                huffSymb += 1;
                code = huffSymb >> 1;
                if (huffSymb & 1)
                    code = -code;
                mantissas[cnt] = code;
            }
        } else {
            for (cnt = 0; cnt < numCodes; cnt++) {
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
                mantissas[cnt*2] = decTable1[huffSymb*2];
                mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
            }
        }
    }
}

/**
 * Restore the quantized band spectrum coefficients
 *
 * @param gb            the GetBit context
 * @param pOut          decoded band spectrum
 * @return outSubbands   subband counter, fix for broken specification/files
 */

static int decodeSpectrum (GetBitContext *gb, float *pOut)
{
    int   numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
    int   subband_vlc_index[32], SF_idxs[32];
    int   mantissas[128];
    float SF;

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

    /* Get the VLC selector table for the subbands, 0 means not coded. */
    for (cnt = 0; cnt <= numSubbands; cnt++)
        subband_vlc_index[cnt] = get_bits(gb, 3);

    /* Read the scale factor indexes from the stream. */
    for (cnt = 0; cnt <= numSubbands; cnt++) {
        if (subband_vlc_index[cnt] != 0)
            SF_idxs[cnt] = get_bits(gb, 6);
    }

    for (cnt = 0; cnt <= numSubbands; cnt++) {
        first = subbandTab[cnt];
        last = subbandTab[cnt+1];

        subbWidth = last - first;

        if (subband_vlc_index[cnt] != 0) {
            /* Decode spectral coefficients for this subband. */
            /* TODO: This can be done faster is several blocks share the
             * same VLC selector (subband_vlc_index) */
            readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);

            /* Decode the scale factor for this subband. */
            SF = ff_atrac_sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];

            /* Inverse quantize the coefficients. */
            for (pIn=mantissas ; first<last; first++, pIn++)
                pOut[first] = *pIn * SF;
        } else {
            /* This subband was not coded, so zero the entire subband. */
            memset(pOut+first, 0, subbWidth*sizeof(float));
        }
    }

    /* Clear the subbands that were not coded. */
    first = subbandTab[cnt];
    memset(pOut+first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float));
    return numSubbands;
}

/**
 * Restore the quantized tonal components
 *
 * @param gb            the GetBit context
 * @param pComponent    tone component
 * @param numBands      amount of coded bands
 */

static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
{
    int i,j,k,cnt;
    int   components, coding_mode_selector, coding_mode, coded_values_per_component;
    int   sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
    int   band_flags[4], mantissa[8];
    float  *pCoef;
    float  scalefactor;
    int   component_count = 0;

    components = get_bits(gb,5);

    /* no tonal components */
    if (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 < components; i++) {
        for (cnt = 0; cnt <= numBands; cnt++)
            band_flags[cnt] = 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 (j = 0; j < (numBands + 1) * 4; j++) {
            if (band_flags[j >> 2] == 0)
                continue;

            coded_components = get_bits(gb,3);

            for (k=0; k<coded_components; k++) {
                sfIndx = get_bits(gb,6);
                if (component_count >= 64)
                    return AVERROR_INVALIDDATA;
                pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
                max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
                coded_values = coded_values_per_component + 1;
                coded_values = FFMIN(max_coded_values,coded_values);

                scalefactor = ff_atrac_sf_table[sfIndx] * iMaxQuant[quant_step_index];

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

                pComponent[component_count].numCoefs = coded_values;

                /* inverse quant */
                pCoef = pComponent[component_count].coef;
                for (cnt = 0; cnt < coded_values; cnt++)
                    pCoef[cnt] = mantissa[cnt] * scalefactor;

                component_count++;
            }
        }
    }

    return component_count;
}

/**
 * Decode gain parameters for the coded bands
 *
 * @param gb            the GetBit context
 * @param pGb           the gainblock for the current band
 * @param numBands      amount of coded bands
 */

static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
{
    int   i, cf, numData;
    int   *pLevel, *pLoc;

    gain_info   *pGain = pGb->gBlock;

    for (i=0 ; i<=numBands; i++)
    {
        numData = get_bits(gb,3);
        pGain[i].num_gain_data = numData;
        pLevel = pGain[i].levcode;
        pLoc = pGain[i].loccode;

        for (cf = 0; cf < numData; cf++){
            pLevel[cf]= get_bits(gb,4);
            pLoc  [cf]= get_bits(gb,5);
            if(cf && pLoc[cf] <= pLoc[cf-1])
                return AVERROR_INVALIDDATA;
        }
    }

    /* Clear the unused blocks. */
    for (; i<4 ; i++)
        pGain[i].num_gain_data = 0;

    return 0;
}

/**
 * Apply gain parameters and perform the MDCT overlapping part
 *
 * @param pIn           input float buffer
 * @param pPrev         previous float buffer to perform overlap against
 * @param pOut          output float buffer
 * @param pGain1        current band gain info
 * @param pGain2        next band gain info
 */

static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
{
    /* gain compensation function */
    float  gain1, gain2, gain_inc;
    int   cnt, numdata, nsample, startLoc, endLoc;


    if (pGain2->num_gain_data == 0)
        gain1 = 1.0;
    else
        gain1 = gain_tab1[pGain2->levcode[0]];

    if (pGain1->num_gain_data == 0) {
        for (cnt = 0; cnt < 256; cnt++)
            pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
    } else {
        numdata = pGain1->num_gain_data;
        pGain1->loccode[numdata] = 32;
        pGain1->levcode[numdata] = 4;

        nsample = 0; // current sample = 0

        for (cnt = 0; cnt < numdata; cnt++) {
            startLoc = pGain1->loccode[cnt] * 8;
            endLoc = startLoc + 8;

            gain2 = gain_tab1[pGain1->levcode[cnt]];
            gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];

            /* interpolate */
            for (; nsample < startLoc; nsample++)
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;

            /* interpolation is done over eight samples */
            for (; nsample < endLoc; nsample++) {
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
                gain2 *= gain_inc;
            }
        }

        for (; nsample < 256; nsample++)
            pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
    }

    /* Delay for the overlapping part. */
    memcpy(pPrev, &pIn[256], 256*sizeof(float));
}

/**
 * Combine the tonal band spectrum and regular band spectrum
 * Return position of the last tonal coefficient
 *
 * @param pSpectrum     output spectrum buffer
 * @param numComponents amount of tonal components
 * @param pComponent    tonal components for this band
 */

static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
{
    int   cnt, i, lastPos = -1;
    float   *pIn, *pOut;

    for (cnt = 0; cnt < numComponents; cnt++){
        lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
        pIn = pComponent[cnt].coef;
        pOut = &(pSpectrum[pComponent[cnt].pos]);

        for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
            pOut[i] += pIn[i];
    }

    return lastPos;
}


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

static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
{
    int    i, band, nsample, s1, s2;
    float    c1, c2;
    float    mc1_l, mc1_r, mc2_l, mc2_r;

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

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

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

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

            case 1:
                for (; nsample < 256; nsample++) {
                    c1 = su1[band+nsample];
                    c2 = su2[band+nsample];
                    su1[band+nsample] = (c1 + c2) * 2.0;
                    su2[band+nsample] = c2 * -2.0;
                }
                break;
            case 2:
            case 3:
                for (; nsample < 256; nsample++) {
                    c1 = su1[band+nsample];
                    c2 = su2[band+nsample];
                    su1[band+nsample] = c1 + c2;
                    su2[band+nsample] = c1 - c2;
                }
                break;
            default:
                assert(0);
        }
    }
}

static void getChannelWeights (int indx, int flag, float ch[2]){

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

static void channelWeighting (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){
        getChannelWeights(p3[1], p3[0], w[0]);
        getChannelWeights(p3[3], p3[2], w[1]);

        for(band = 1; band < 4; band++) {
            /* scale the channels by the weights */
            for(nsample = 0; nsample < 8; nsample++) {
                su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
                su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
            }

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


/**
 * Decode a Sound Unit
 *
 * @param gb            the GetBit context
 * @param pSnd          the channel unit to be used
 * @param pOut          the decoded samples before IQMF in float representation
 * @param channelNum    channel number
 * @param codingMode    the coding mode (JOINT_STEREO or regular stereo/mono)
 */


static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
{
    int   band, result=0, numSubbands, lastTonal, numBands;

    if (codingMode == JOINT_STEREO && channelNum == 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 */
    pSnd->bandsCoded = get_bits(gb,2);

    result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
    if (result) return result;

    pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
    if (pSnd->numComponents == -1) return -1;

    numSubbands = decodeSpectrum (gb, pSnd->spectrum);

    /* Merge the decoded spectrum and tonal components. */
    lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);


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


    /* Reconstruct time domain samples. */
    for (band=0; band<4; band++) {
        /* Perform the IMDCT step without overlapping. */
        if (band <= numBands) {
            IMLT(q, &(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
        } else
            memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));

        /* gain compensation and overlapping */
        gainCompensateAndOverlap(pSnd->IMDCT_buf, &pSnd->prevFrame[band * 256],
                                 &pOut[band * 256],
                                 &pSnd->gainBlock[1 - pSnd->gcBlkSwitch].gBlock[band],
                                 &pSnd->gainBlock[    pSnd->gcBlkSwitch].gBlock[band]);
    }

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

    return 0;
}

/**
 * Frame handling
 *
 * @param q             Atrac3 private context
 * @param databuf       the input data
 */

static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
                       float **out_samples)
{
    int   result, i;
    float   *p1, *p2, *p3, *p4;
    uint8_t *ptr1;

    if (q->codingMode == JOINT_STEREO) {

        /* channel coupling mode */
        /* decode Sound Unit 1 */
        init_get_bits(&q->gb,databuf,q->bits_per_frame);

        result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
        if (result != 0)
            return result;

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


        /* set the bitstream reader at the start of the second Sound Unit*/
        init_get_bits(&q->gb,ptr1,q->bits_per_frame);

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

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

        /* Decode Sound Unit 2. */
        result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
        if (result != 0)
            return result;

        /* Reconstruct the channel coefficients. */
        reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);

        channelWeighting(out_samples[0], out_samples[1], q->weighting_delay);

    } else {
        /* normal stereo mode or mono */
        /* Decode the channel sound units. */
        for (i=0 ; i<q->channels ; i++) {

            /* Set the bitstream reader at the start of a channel sound unit. */
            init_get_bits(&q->gb,
                          databuf + i * q->bytes_per_frame / q->channels,
                          q->bits_per_frame / q->channels);

            result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
            if (result != 0)
                return result;
        }
    }

    /* Apply the iQMF synthesis filter. */
    for (i=0 ; i<q->channels ; i++) {
        p1 = out_samples[i];
        p2= p1+256;
        p3= p2+256;
        p4= p3+256;
        ff_atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
        ff_atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
        ff_atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
    }

    return 0;
}


/**
 * Atrac frame decoding
 *
 * @param avctx     pointer to the AVCodecContext
 */

static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    ATRAC3Context *q = avctx->priv_data;
    int result;
    const uint8_t* databuf;
    float   *samples_flt;
    int16_t *samples_s16;

    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 */
    q->frame.nb_samples = SAMPLES_PER_FRAME;
    if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) {
        av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
        return result;
    }
    samples_flt = (float   *)q->frame.data[0];
    samples_s16 = (int16_t *)q->frame.data[0];

    /* 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;
    }

    if (q->channels == 1 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
        result = decodeFrame(q, databuf, &samples_flt);
    else
        result = decodeFrame(q, databuf, q->outSamples);

    if (result != 0) {
        av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
        return result;
    }

    /* interleave */
    if (q->channels == 2 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
        q->fmt_conv.float_interleave(samples_flt,
                                     (const float **)q->outSamples,
                                     SAMPLES_PER_FRAME, 2);
    } else if (avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
        q->fmt_conv.float_to_int16_interleave(samples_s16,
                                              (const float **)q->outSamples,
                                              SAMPLES_PER_FRAME, q->channels);
    }

    *got_frame_ptr   = 1;
    *(AVFrame *)data = q->frame;

    return avctx->block_align;
}


/**
 * Atrac3 initialization
 *
 * @param avctx     pointer to the AVCodecContext
 */

static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
    int i, ret;
    const uint8_t *edata_ptr = avctx->extradata;
    ATRAC3Context *q = avctx->priv_data;
    static VLC_TYPE atrac3_vlc_table[4096][2];
    static int vlcs_initialized = 0;

    /* Take data from the AVCodecContext (RM container). */
    q->sample_rate = avctx->sample_rate;
    q->channels = avctx->channels;
    q->bit_rate = avctx->bit_rate;
    q->bits_per_frame = avctx->block_align * 8;
    q->bytes_per_frame = avctx->block_align;

    /* Take care of the codec-specific extradata. */
    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
        q->samples_per_channel = bytestream_get_le32(&edata_ptr);
        q->codingMode = bytestream_get_le16(&edata_ptr);
        av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
        q->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 */
        q->samples_per_frame = SAMPLES_PER_FRAME * q->channels;
        q->atrac3version = 4;
        q->delay = 0x88E;
        if (q->codingMode)
            q->codingMode = JOINT_STEREO;
        else
            q->codingMode = STEREO;

        q->scrambled_stream = 0;

        if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) {
        } else {
            av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor);
            return AVERROR_INVALIDDATA;
        }

    } else if (avctx->extradata_size == 10) {
        /* Parse the extradata, RM format. */
        q->atrac3version = bytestream_get_be32(&edata_ptr);
        q->samples_per_frame = bytestream_get_be16(&edata_ptr);
        q->delay = bytestream_get_be16(&edata_ptr);
        q->codingMode = bytestream_get_be16(&edata_ptr);

        q->samples_per_channel = q->samples_per_frame / q->channels;
        q->scrambled_stream = 1;

    } else {
        av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
    }
    /* Check the extradata. */

    if (q->atrac3version != 4) {
        av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
        return AVERROR_INVALIDDATA;
    }

    if (q->samples_per_frame != SAMPLES_PER_FRAME && q->samples_per_frame != SAMPLES_PER_FRAME*2) {
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
        return AVERROR_INVALIDDATA;
    }

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

    if (q->codingMode == STEREO) {
        av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
    } else if (q->codingMode == JOINT_STEREO) {
        av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
    } else {
        av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
        return AVERROR_INVALIDDATA;
    }

    if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
        av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
        return AVERROR(EINVAL);
    }


    if(avctx->block_align >= UINT_MAX/2)
        return AVERROR(EINVAL);

    /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
     * this is for the bitstream reader. */
    if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)
        return AVERROR(ENOMEM);


    /* Initialize the VLC tables. */
    if (!vlcs_initialized) {
        for (i=0 ; i<7 ; i++) {
            spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
            spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i];
            init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
                huff_bits[i], 1, 1,
                huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
        }
        vlcs_initialized = 1;
    }

    if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT)
        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
    else
        avctx->sample_fmt = AV_SAMPLE_FMT_S16;

    if ((ret = init_atrac3_transforms(q, avctx->sample_fmt == AV_SAMPLE_FMT_FLT))) {
        av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
        av_freep(&q->decoded_bytes_buffer);
        return ret;
    }

    ff_atrac_generate_tables();

    /* Generate gain tables. */
    for (i=0 ; i<16 ; i++)
        gain_tab1[i] = powf (2.0, (4 - i));

    for (i=-15 ; i<16 ; i++)
        gain_tab2[i+15] = powf (2.0, i * -0.125);

    /* init the joint-stereo decoding data */
    q->weighting_delay[0] = 0;
    q->weighting_delay[1] = 7;
    q->weighting_delay[2] = 0;
    q->weighting_delay[3] = 7;
    q->weighting_delay[4] = 0;
    q->weighting_delay[5] = 7;

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

    ff_dsputil_init(&dsp, avctx);
    ff_fmt_convert_init(&q->fmt_conv, avctx);

    q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
    if (!q->pUnits) {
        atrac3_decode_close(avctx);
        return AVERROR(ENOMEM);
    }

    if (avctx->channels > 1 || avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
        q->outSamples[0] = av_mallocz(SAMPLES_PER_FRAME * avctx->channels * sizeof(*q->outSamples[0]));
        q->outSamples[1] = q->outSamples[0] + SAMPLES_PER_FRAME;
        if (!q->outSamples[0]) {
            atrac3_decode_close(avctx);
            return AVERROR(ENOMEM);
        }
    }

    avcodec_get_frame_defaults(&q->frame);
    avctx->coded_frame = &q->frame;

    return 0;
}


AVCodec ff_atrac3_decoder =
{
    .name = "atrac3",
    .type = AVMEDIA_TYPE_AUDIO,
    .id = CODEC_ID_ATRAC3,
    .priv_data_size = sizeof(ATRAC3Context),
    .init = atrac3_decode_init,
    .close = atrac3_decode_close,
    .decode = atrac3_decode_frame,
    .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
    .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
};