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1
/*
2
 * Atrac 3 compatible decoder
3
 * Copyright (c) 2006-2008 Maxim Poliakovski
4
 * Copyright (c) 2006-2008 Benjamin Larsson
5
 *
6
 * This file is part of FFmpeg.
7
 *
8
 * FFmpeg is free software; you can redistribute it and/or
9
 * modify it under the terms of the GNU Lesser General Public
10
 * License as published by the Free Software Foundation; either
11
 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
14
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
16
 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
22

    
23
/**
24
 * @file libavcodec/atrac3.c
25
 * Atrac 3 compatible decoder.
26
 * This decoder handles Sony's ATRAC3 data.
27
 *
28
 * Container formats used to store atrac 3 data:
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 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
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 *
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 * To use this decoder, a calling application must supply the extradata
32
 * bytes provided in the containers above.
33
 */
34

    
35
#include <math.h>
36
#include <stddef.h>
37
#include <stdio.h>
38

    
39
#include "avcodec.h"
40
#include "get_bits.h"
41
#include "dsputil.h"
42
#include "bytestream.h"
43

    
44
#include "atrac.h"
45
#include "atrac3data.h"
46

    
47
#define JOINT_STEREO    0x12
48
#define STEREO          0x2
49

    
50

    
51
/* These structures are needed to store the parsed gain control data. */
52
typedef struct {
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    int   num_gain_data;
54
    int   levcode[8];
55
    int   loccode[8];
56
} gain_info;
57

    
58
typedef struct {
59
    gain_info   gBlock[4];
60
} gain_block;
61

    
62
typedef struct {
63
    int     pos;
64
    int     numCoefs;
65
    float   coef[8];
66
} tonal_component;
67

    
68
typedef struct {
69
    int               bandsCoded;
70
    int               numComponents;
71
    tonal_component   components[64];
72
    float             prevFrame[1024];
73
    int               gcBlkSwitch;
74
    gain_block        gainBlock[2];
75

    
76
    DECLARE_ALIGNED(16, float, spectrum)[1024];
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    DECLARE_ALIGNED(16, float, IMDCT_buf)[1024];
78

    
79
    float             delayBuf1[46]; ///<qmf delay buffers
80
    float             delayBuf2[46];
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    float             delayBuf3[46];
82
} channel_unit;
83

    
84
typedef struct {
85
    GetBitContext       gb;
86
    //@{
87
    /** stream data */
88
    int                 channels;
89
    int                 codingMode;
90
    int                 bit_rate;
91
    int                 sample_rate;
92
    int                 samples_per_channel;
93
    int                 samples_per_frame;
94

    
95
    int                 bits_per_frame;
96
    int                 bytes_per_frame;
97
    int                 pBs;
98
    channel_unit*       pUnits;
99
    //@}
100
    //@{
101
    /** joint-stereo related variables */
102
    int                 matrix_coeff_index_prev[4];
103
    int                 matrix_coeff_index_now[4];
104
    int                 matrix_coeff_index_next[4];
105
    int                 weighting_delay[6];
106
    //@}
107
    //@{
108
    /** data buffers */
109
    float               outSamples[2048];
110
    uint8_t*            decoded_bytes_buffer;
111
    float               tempBuf[1070];
112
    //@}
113
    //@{
114
    /** extradata */
115
    int                 atrac3version;
116
    int                 delay;
117
    int                 scrambled_stream;
118
    int                 frame_factor;
119
    //@}
120
} ATRAC3Context;
121

    
122
static DECLARE_ALIGNED(16, float,mdct_window)[512];
123
static VLC              spectral_coeff_tab[7];
124
static float            gain_tab1[16];
125
static float            gain_tab2[31];
126
static FFTContext       mdct_ctx;
127
static DSPContext       dsp;
128

    
129

    
130
/**
131
 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
132
 * caused by the reverse spectra of the QMF.
133
 *
134
 * @param pInput    float input
135
 * @param pOutput   float output
136
 * @param odd_band  1 if the band is an odd band
137
 */
138

    
139
static void IMLT(float *pInput, float *pOutput, int odd_band)
140
{
141
    int     i;
142

    
143
    if (odd_band) {
144
        /**
145
        * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
146
        * or it gives better compression to do it this way.
147
        * FIXME: It should be possible to handle this in ff_imdct_calc
148
        * for that to happen a modification of the prerotation step of
149
        * all SIMD code and C code is needed.
150
        * Or fix the functions before so they generate a pre reversed spectrum.
151
        */
152

    
153
        for (i=0; i<128; i++)
154
            FFSWAP(float, pInput[i], pInput[255-i]);
155
    }
156

    
157
    ff_imdct_calc(&mdct_ctx,pOutput,pInput);
158

    
159
    /* Perform windowing on the output. */
160
    dsp.vector_fmul(pOutput,mdct_window,512);
161

    
162
}
163

    
164

    
165
/**
166
 * Atrac 3 indata descrambling, only used for data coming from the rm container
167
 *
168
 * @param in        pointer to 8 bit array of indata
169
 * @param bits      amount of bits
170
 * @param out       pointer to 8 bit array of outdata
171
 */
172

    
173
static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
174
    int i, off;
175
    uint32_t c;
176
    const uint32_t* buf;
177
    uint32_t* obuf = (uint32_t*) out;
178

    
179
    off = (intptr_t)inbuffer & 3;
180
    buf = (const uint32_t*) (inbuffer - off);
181
    c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
182
    bytes += 3 + off;
183
    for (i = 0; i < bytes/4; i++)
184
        obuf[i] = c ^ buf[i];
185

    
186
    if (off)
187
        av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
188

    
189
    return off;
190
}
191

    
192

    
193
static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
194
    float enc_window[256];
195
    int i;
196

    
197
    /* Generate the mdct window, for details see
198
     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
199
    for (i=0 ; i<256; i++)
200
        enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
201

    
202
    if (!mdct_window[0])
203
        for (i=0 ; i<256; i++) {
204
            mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
205
            mdct_window[511-i] = mdct_window[i];
206
        }
207

    
208
    /* Initialize the MDCT transform. */
209
    ff_mdct_init(&mdct_ctx, 9, 1, 1.0);
210
}
211

    
212
/**
213
 * Atrac3 uninit, free all allocated memory
214
 */
215

    
216
static av_cold int atrac3_decode_close(AVCodecContext *avctx)
217
{
218
    ATRAC3Context *q = avctx->priv_data;
219

    
220
    av_free(q->pUnits);
221
    av_free(q->decoded_bytes_buffer);
222

    
223
    return 0;
224
}
225

    
226
/**
227
/ * Mantissa decoding
228
 *
229
 * @param gb            the GetBit context
230
 * @param selector      what table is the output values coded with
231
 * @param codingFlag    constant length coding or variable length coding
232
 * @param mantissas     mantissa output table
233
 * @param numCodes      amount of values to get
234
 */
235

    
236
static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
237
{
238
    int   numBits, cnt, code, huffSymb;
239

    
240
    if (selector == 1)
241
        numCodes /= 2;
242

    
243
    if (codingFlag != 0) {
244
        /* constant length coding (CLC) */
245
        numBits = CLCLengthTab[selector];
246

    
247
        if (selector > 1) {
248
            for (cnt = 0; cnt < numCodes; cnt++) {
249
                if (numBits)
250
                    code = get_sbits(gb, numBits);
251
                else
252
                    code = 0;
253
                mantissas[cnt] = code;
254
            }
255
        } else {
256
            for (cnt = 0; cnt < numCodes; cnt++) {
257
                if (numBits)
258
                    code = get_bits(gb, numBits); //numBits is always 4 in this case
259
                else
260
                    code = 0;
261
                mantissas[cnt*2] = seTab_0[code >> 2];
262
                mantissas[cnt*2+1] = seTab_0[code & 3];
263
            }
264
        }
265
    } else {
266
        /* variable length coding (VLC) */
267
        if (selector != 1) {
268
            for (cnt = 0; cnt < numCodes; cnt++) {
269
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
270
                huffSymb += 1;
271
                code = huffSymb >> 1;
272
                if (huffSymb & 1)
273
                    code = -code;
274
                mantissas[cnt] = code;
275
            }
276
        } else {
277
            for (cnt = 0; cnt < numCodes; cnt++) {
278
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
279
                mantissas[cnt*2] = decTable1[huffSymb*2];
280
                mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
281
            }
282
        }
283
    }
284
}
285

    
286
/**
287
 * Restore the quantized band spectrum coefficients
288
 *
289
 * @param gb            the GetBit context
290
 * @param pOut          decoded band spectrum
291
 * @return outSubbands   subband counter, fix for broken specification/files
292
 */
293

    
294
static int decodeSpectrum (GetBitContext *gb, float *pOut)
295
{
296
    int   numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
297
    int   subband_vlc_index[32], SF_idxs[32];
298
    int   mantissas[128];
299
    float SF;
300

    
301
    numSubbands = get_bits(gb, 5); // number of coded subbands
302
    codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
303

    
304
    /* Get the VLC selector table for the subbands, 0 means not coded. */
305
    for (cnt = 0; cnt <= numSubbands; cnt++)
306
        subband_vlc_index[cnt] = get_bits(gb, 3);
307

    
308
    /* Read the scale factor indexes from the stream. */
309
    for (cnt = 0; cnt <= numSubbands; cnt++) {
310
        if (subband_vlc_index[cnt] != 0)
311
            SF_idxs[cnt] = get_bits(gb, 6);
312
    }
313

    
314
    for (cnt = 0; cnt <= numSubbands; cnt++) {
315
        first = subbandTab[cnt];
316
        last = subbandTab[cnt+1];
317

    
318
        subbWidth = last - first;
319

    
320
        if (subband_vlc_index[cnt] != 0) {
321
            /* Decode spectral coefficients for this subband. */
322
            /* TODO: This can be done faster is several blocks share the
323
             * same VLC selector (subband_vlc_index) */
324
            readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
325

    
326
            /* Decode the scale factor for this subband. */
327
            SF = sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
328

    
329
            /* Inverse quantize the coefficients. */
330
            for (pIn=mantissas ; first<last; first++, pIn++)
331
                pOut[first] = *pIn * SF;
332
        } else {
333
            /* This subband was not coded, so zero the entire subband. */
334
            memset(pOut+first, 0, subbWidth*sizeof(float));
335
        }
336
    }
337

    
338
    /* Clear the subbands that were not coded. */
339
    first = subbandTab[cnt];
340
    memset(pOut+first, 0, (1024 - first) * sizeof(float));
341
    return numSubbands;
342
}
343

    
344
/**
345
 * Restore the quantized tonal components
346
 *
347
 * @param gb            the GetBit context
348
 * @param pComponent    tone component
349
 * @param numBands      amount of coded bands
350
 */
351

    
352
static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
353
{
354
    int i,j,k,cnt;
355
    int   components, coding_mode_selector, coding_mode, coded_values_per_component;
356
    int   sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
357
    int   band_flags[4], mantissa[8];
358
    float  *pCoef;
359
    float  scalefactor;
360
    int   component_count = 0;
361

    
362
    components = get_bits(gb,5);
363

    
364
    /* no tonal components */
365
    if (components == 0)
366
        return 0;
367

    
368
    coding_mode_selector = get_bits(gb,2);
369
    if (coding_mode_selector == 2)
370
        return -1;
371

    
372
    coding_mode = coding_mode_selector & 1;
373

    
374
    for (i = 0; i < components; i++) {
375
        for (cnt = 0; cnt <= numBands; cnt++)
376
            band_flags[cnt] = get_bits1(gb);
377

    
378
        coded_values_per_component = get_bits(gb,3);
379

    
380
        quant_step_index = get_bits(gb,3);
381
        if (quant_step_index <= 1)
382
            return -1;
383

    
384
        if (coding_mode_selector == 3)
385
            coding_mode = get_bits1(gb);
386

    
387
        for (j = 0; j < (numBands + 1) * 4; j++) {
388
            if (band_flags[j >> 2] == 0)
389
                continue;
390

    
391
            coded_components = get_bits(gb,3);
392

    
393
            for (k=0; k<coded_components; k++) {
394
                sfIndx = get_bits(gb,6);
395
                pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
396
                max_coded_values = 1024 - pComponent[component_count].pos;
397
                coded_values = coded_values_per_component + 1;
398
                coded_values = FFMIN(max_coded_values,coded_values);
399

    
400
                scalefactor = sf_table[sfIndx] * iMaxQuant[quant_step_index];
401

    
402
                readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
403

    
404
                pComponent[component_count].numCoefs = coded_values;
405

    
406
                /* inverse quant */
407
                pCoef = pComponent[component_count].coef;
408
                for (cnt = 0; cnt < coded_values; cnt++)
409
                    pCoef[cnt] = mantissa[cnt] * scalefactor;
410

    
411
                component_count++;
412
            }
413
        }
414
    }
415

    
416
    return component_count;
417
}
418

    
419
/**
420
 * Decode gain parameters for the coded bands
421
 *
422
 * @param gb            the GetBit context
423
 * @param pGb           the gainblock for the current band
424
 * @param numBands      amount of coded bands
425
 */
426

    
427
static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
428
{
429
    int   i, cf, numData;
430
    int   *pLevel, *pLoc;
431

    
432
    gain_info   *pGain = pGb->gBlock;
433

    
434
    for (i=0 ; i<=numBands; i++)
435
    {
436
        numData = get_bits(gb,3);
437
        pGain[i].num_gain_data = numData;
438
        pLevel = pGain[i].levcode;
439
        pLoc = pGain[i].loccode;
440

    
441
        for (cf = 0; cf < numData; cf++){
442
            pLevel[cf]= get_bits(gb,4);
443
            pLoc  [cf]= get_bits(gb,5);
444
            if(cf && pLoc[cf] <= pLoc[cf-1])
445
                return -1;
446
        }
447
    }
448

    
449
    /* Clear the unused blocks. */
450
    for (; i<4 ; i++)
451
        pGain[i].num_gain_data = 0;
452

    
453
    return 0;
454
}
455

    
456
/**
457
 * Apply gain parameters and perform the MDCT overlapping part
458
 *
459
 * @param pIn           input float buffer
460
 * @param pPrev         previous float buffer to perform overlap against
461
 * @param pOut          output float buffer
462
 * @param pGain1        current band gain info
463
 * @param pGain2        next band gain info
464
 */
465

    
466
static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
467
{
468
    /* gain compensation function */
469
    float  gain1, gain2, gain_inc;
470
    int   cnt, numdata, nsample, startLoc, endLoc;
471

    
472

    
473
    if (pGain2->num_gain_data == 0)
474
        gain1 = 1.0;
475
    else
476
        gain1 = gain_tab1[pGain2->levcode[0]];
477

    
478
    if (pGain1->num_gain_data == 0) {
479
        for (cnt = 0; cnt < 256; cnt++)
480
            pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
481
    } else {
482
        numdata = pGain1->num_gain_data;
483
        pGain1->loccode[numdata] = 32;
484
        pGain1->levcode[numdata] = 4;
485

    
486
        nsample = 0; // current sample = 0
487

    
488
        for (cnt = 0; cnt < numdata; cnt++) {
489
            startLoc = pGain1->loccode[cnt] * 8;
490
            endLoc = startLoc + 8;
491

    
492
            gain2 = gain_tab1[pGain1->levcode[cnt]];
493
            gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
494

    
495
            /* interpolate */
496
            for (; nsample < startLoc; nsample++)
497
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
498

    
499
            /* interpolation is done over eight samples */
500
            for (; nsample < endLoc; nsample++) {
501
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
502
                gain2 *= gain_inc;
503
            }
504
        }
505

    
506
        for (; nsample < 256; nsample++)
507
            pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
508
    }
509

    
510
    /* Delay for the overlapping part. */
511
    memcpy(pPrev, &pIn[256], 256*sizeof(float));
512
}
513

    
514
/**
515
 * Combine the tonal band spectrum and regular band spectrum
516
 * Return position of the last tonal coefficient
517
 *
518
 * @param pSpectrum     output spectrum buffer
519
 * @param numComponents amount of tonal components
520
 * @param pComponent    tonal components for this band
521
 */
522

    
523
static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
524
{
525
    int   cnt, i, lastPos = -1;
526
    float   *pIn, *pOut;
527

    
528
    for (cnt = 0; cnt < numComponents; cnt++){
529
        lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
530
        pIn = pComponent[cnt].coef;
531
        pOut = &(pSpectrum[pComponent[cnt].pos]);
532

    
533
        for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
534
            pOut[i] += pIn[i];
535
    }
536

    
537
    return lastPos;
538
}
539

    
540

    
541
#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
542

    
543
static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
544
{
545
    int    i, band, nsample, s1, s2;
546
    float    c1, c2;
547
    float    mc1_l, mc1_r, mc2_l, mc2_r;
548

    
549
    for (i=0,band = 0; band < 4*256; band+=256,i++) {
550
        s1 = pPrevCode[i];
551
        s2 = pCurrCode[i];
552
        nsample = 0;
553

    
554
        if (s1 != s2) {
555
            /* Selector value changed, interpolation needed. */
556
            mc1_l = matrixCoeffs[s1*2];
557
            mc1_r = matrixCoeffs[s1*2+1];
558
            mc2_l = matrixCoeffs[s2*2];
559
            mc2_r = matrixCoeffs[s2*2+1];
560

    
561
            /* Interpolation is done over the first eight samples. */
562
            for(; nsample < 8; nsample++) {
563
                c1 = su1[band+nsample];
564
                c2 = su2[band+nsample];
565
                c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
566
                su1[band+nsample] = c2;
567
                su2[band+nsample] = c1 * 2.0 - c2;
568
            }
569
        }
570

    
571
        /* Apply the matrix without interpolation. */
572
        switch (s2) {
573
            case 0:     /* M/S decoding */
574
                for (; nsample < 256; nsample++) {
575
                    c1 = su1[band+nsample];
576
                    c2 = su2[band+nsample];
577
                    su1[band+nsample] = c2 * 2.0;
578
                    su2[band+nsample] = (c1 - c2) * 2.0;
579
                }
580
                break;
581

    
582
            case 1:
583
                for (; nsample < 256; nsample++) {
584
                    c1 = su1[band+nsample];
585
                    c2 = su2[band+nsample];
586
                    su1[band+nsample] = (c1 + c2) * 2.0;
587
                    su2[band+nsample] = c2 * -2.0;
588
                }
589
                break;
590
            case 2:
591
            case 3:
592
                for (; nsample < 256; nsample++) {
593
                    c1 = su1[band+nsample];
594
                    c2 = su2[band+nsample];
595
                    su1[band+nsample] = c1 + c2;
596
                    su2[band+nsample] = c1 - c2;
597
                }
598
                break;
599
            default:
600
                assert(0);
601
        }
602
    }
603
}
604

    
605
static void getChannelWeights (int indx, int flag, float ch[2]){
606

    
607
    if (indx == 7) {
608
        ch[0] = 1.0;
609
        ch[1] = 1.0;
610
    } else {
611
        ch[0] = (float)(indx & 7) / 7.0;
612
        ch[1] = sqrt(2 - ch[0]*ch[0]);
613
        if(flag)
614
            FFSWAP(float, ch[0], ch[1]);
615
    }
616
}
617

    
618
static void channelWeighting (float *su1, float *su2, int *p3)
619
{
620
    int   band, nsample;
621
    /* w[x][y] y=0 is left y=1 is right */
622
    float w[2][2];
623

    
624
    if (p3[1] != 7 || p3[3] != 7){
625
        getChannelWeights(p3[1], p3[0], w[0]);
626
        getChannelWeights(p3[3], p3[2], w[1]);
627

    
628
        for(band = 1; band < 4; band++) {
629
            /* scale the channels by the weights */
630
            for(nsample = 0; nsample < 8; nsample++) {
631
                su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
632
                su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
633
            }
634

    
635
            for(; nsample < 256; nsample++) {
636
                su1[band*256+nsample] *= w[1][0];
637
                su2[band*256+nsample] *= w[1][1];
638
            }
639
        }
640
    }
641
}
642

    
643

    
644
/**
645
 * Decode a Sound Unit
646
 *
647
 * @param gb            the GetBit context
648
 * @param pSnd          the channel unit to be used
649
 * @param pOut          the decoded samples before IQMF in float representation
650
 * @param channelNum    channel number
651
 * @param codingMode    the coding mode (JOINT_STEREO or regular stereo/mono)
652
 */
653

    
654

    
655
static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
656
{
657
    int   band, result=0, numSubbands, lastTonal, numBands;
658

    
659
    if (codingMode == JOINT_STEREO && channelNum == 1) {
660
        if (get_bits(gb,2) != 3) {
661
            av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
662
            return -1;
663
        }
664
    } else {
665
        if (get_bits(gb,6) != 0x28) {
666
            av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
667
            return -1;
668
        }
669
    }
670

    
671
    /* number of coded QMF bands */
672
    pSnd->bandsCoded = get_bits(gb,2);
673

    
674
    result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
675
    if (result) return result;
676

    
677
    pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
678
    if (pSnd->numComponents == -1) return -1;
679

    
680
    numSubbands = decodeSpectrum (gb, pSnd->spectrum);
681

    
682
    /* Merge the decoded spectrum and tonal components. */
683
    lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
684

    
685

    
686
    /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
687
    numBands = (subbandTab[numSubbands] - 1) >> 8;
688
    if (lastTonal >= 0)
689
        numBands = FFMAX((lastTonal + 256) >> 8, numBands);
690

    
691

    
692
    /* Reconstruct time domain samples. */
693
    for (band=0; band<4; band++) {
694
        /* Perform the IMDCT step without overlapping. */
695
        if (band <= numBands) {
696
            IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
697
        } else
698
            memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
699

    
700
        /* gain compensation and overlapping */
701
        gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
702
                                    &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
703
                                    &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
704
    }
705

    
706
    /* Swap the gain control buffers for the next frame. */
707
    pSnd->gcBlkSwitch ^= 1;
708

    
709
    return 0;
710
}
711

    
712
/**
713
 * Frame handling
714
 *
715
 * @param q             Atrac3 private context
716
 * @param databuf       the input data
717
 */
718

    
719
static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
720
{
721
    int   result, i;
722
    float   *p1, *p2, *p3, *p4;
723
    uint8_t *ptr1;
724

    
725
    if (q->codingMode == JOINT_STEREO) {
726

    
727
        /* channel coupling mode */
728
        /* decode Sound Unit 1 */
729
        init_get_bits(&q->gb,databuf,q->bits_per_frame);
730

    
731
        result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
732
        if (result != 0)
733
            return (result);
734

    
735
        /* Framedata of the su2 in the joint-stereo mode is encoded in
736
         * reverse byte order so we need to swap it first. */
737
        if (databuf == q->decoded_bytes_buffer) {
738
            uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
739
            ptr1 = q->decoded_bytes_buffer;
740
            for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
741
                FFSWAP(uint8_t,*ptr1,*ptr2);
742
            }
743
        } else {
744
            const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
745
            for (i = 0; i < q->bytes_per_frame; i++)
746
                q->decoded_bytes_buffer[i] = *ptr2--;
747
        }
748

    
749
        /* Skip the sync codes (0xF8). */
750
        ptr1 = q->decoded_bytes_buffer;
751
        for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
752
            if (i >= q->bytes_per_frame)
753
                return -1;
754
        }
755

    
756

    
757
        /* set the bitstream reader at the start of the second Sound Unit*/
758
        init_get_bits(&q->gb,ptr1,q->bits_per_frame);
759

    
760
        /* Fill the Weighting coeffs delay buffer */
761
        memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
762
        q->weighting_delay[4] = get_bits1(&q->gb);
763
        q->weighting_delay[5] = get_bits(&q->gb,3);
764

    
765
        for (i = 0; i < 4; i++) {
766
            q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
767
            q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
768
            q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
769
        }
770

    
771
        /* Decode Sound Unit 2. */
772
        result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
773
        if (result != 0)
774
            return (result);
775

    
776
        /* Reconstruct the channel coefficients. */
777
        reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
778

    
779
        channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
780

    
781
    } else {
782
        /* normal stereo mode or mono */
783
        /* Decode the channel sound units. */
784
        for (i=0 ; i<q->channels ; i++) {
785

    
786
            /* Set the bitstream reader at the start of a channel sound unit. */
787
            init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
788

    
789
            result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
790
            if (result != 0)
791
                return (result);
792
        }
793
    }
794

    
795
    /* Apply the iQMF synthesis filter. */
796
    p1= q->outSamples;
797
    for (i=0 ; i<q->channels ; i++) {
798
        p2= p1+256;
799
        p3= p2+256;
800
        p4= p3+256;
801
        atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
802
        atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
803
        atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
804
        p1 +=1024;
805
    }
806

    
807
    return 0;
808
}
809

    
810

    
811
/**
812
 * Atrac frame decoding
813
 *
814
 * @param avctx     pointer to the AVCodecContext
815
 */
816

    
817
static int atrac3_decode_frame(AVCodecContext *avctx,
818
            void *data, int *data_size,
819
            AVPacket *avpkt) {
820
    const uint8_t *buf = avpkt->data;
821
    int buf_size = avpkt->size;
822
    ATRAC3Context *q = avctx->priv_data;
823
    int result = 0, i;
824
    const uint8_t* databuf;
825
    int16_t* samples = data;
826

    
827
    if (buf_size < avctx->block_align)
828
        return buf_size;
829

    
830
    /* Check if we need to descramble and what buffer to pass on. */
831
    if (q->scrambled_stream) {
832
        decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
833
        databuf = q->decoded_bytes_buffer;
834
    } else {
835
        databuf = buf;
836
    }
837

    
838
    result = decodeFrame(q, databuf);
839

    
840
    if (result != 0) {
841
        av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
842
        return -1;
843
    }
844

    
845
    if (q->channels == 1) {
846
        /* mono */
847
        for (i = 0; i<1024; i++)
848
            samples[i] = av_clip_int16(round(q->outSamples[i]));
849
        *data_size = 1024 * sizeof(int16_t);
850
    } else {
851
        /* stereo */
852
        for (i = 0; i < 1024; i++) {
853
            samples[i*2] = av_clip_int16(round(q->outSamples[i]));
854
            samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
855
        }
856
        *data_size = 2048 * sizeof(int16_t);
857
    }
858

    
859
    return avctx->block_align;
860
}
861

    
862

    
863
/**
864
 * Atrac3 initialization
865
 *
866
 * @param avctx     pointer to the AVCodecContext
867
 */
868

    
869
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
870
{
871
    int i;
872
    const uint8_t *edata_ptr = avctx->extradata;
873
    ATRAC3Context *q = avctx->priv_data;
874
    static VLC_TYPE atrac3_vlc_table[4096][2];
875
    static int vlcs_initialized = 0;
876

    
877
    /* Take data from the AVCodecContext (RM container). */
878
    q->sample_rate = avctx->sample_rate;
879
    q->channels = avctx->channels;
880
    q->bit_rate = avctx->bit_rate;
881
    q->bits_per_frame = avctx->block_align * 8;
882
    q->bytes_per_frame = avctx->block_align;
883

    
884
    /* Take care of the codec-specific extradata. */
885
    if (avctx->extradata_size == 14) {
886
        /* Parse the extradata, WAV format */
887
        av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown value always 1
888
        q->samples_per_channel = bytestream_get_le32(&edata_ptr);
889
        q->codingMode = bytestream_get_le16(&edata_ptr);
890
        av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
891
        q->frame_factor = bytestream_get_le16(&edata_ptr);  //Unknown always 1
892
        av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown always 0
893

    
894
        /* setup */
895
        q->samples_per_frame = 1024 * q->channels;
896
        q->atrac3version = 4;
897
        q->delay = 0x88E;
898
        if (q->codingMode)
899
            q->codingMode = JOINT_STEREO;
900
        else
901
            q->codingMode = STEREO;
902

    
903
        q->scrambled_stream = 0;
904

    
905
        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)) {
906
        } else {
907
            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);
908
            return -1;
909
        }
910

    
911
    } else if (avctx->extradata_size == 10) {
912
        /* Parse the extradata, RM format. */
913
        q->atrac3version = bytestream_get_be32(&edata_ptr);
914
        q->samples_per_frame = bytestream_get_be16(&edata_ptr);
915
        q->delay = bytestream_get_be16(&edata_ptr);
916
        q->codingMode = bytestream_get_be16(&edata_ptr);
917

    
918
        q->samples_per_channel = q->samples_per_frame / q->channels;
919
        q->scrambled_stream = 1;
920

    
921
    } else {
922
        av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
923
    }
924
    /* Check the extradata. */
925

    
926
    if (q->atrac3version != 4) {
927
        av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
928
        return -1;
929
    }
930

    
931
    if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
932
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
933
        return -1;
934
    }
935

    
936
    if (q->delay != 0x88E) {
937
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
938
        return -1;
939
    }
940

    
941
    if (q->codingMode == STEREO) {
942
        av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
943
    } else if (q->codingMode == JOINT_STEREO) {
944
        av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
945
    } else {
946
        av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
947
        return -1;
948
    }
949

    
950
    if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
951
        av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
952
        return -1;
953
    }
954

    
955

    
956
    if(avctx->block_align >= UINT_MAX/2)
957
        return -1;
958

    
959
    /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
960
     * this is for the bitstream reader. */
961
    if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)
962
        return AVERROR(ENOMEM);
963

    
964

    
965
    /* Initialize the VLC tables. */
966
    if (!vlcs_initialized) {
967
        for (i=0 ; i<7 ; i++) {
968
            spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
969
            spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i];
970
            init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
971
                huff_bits[i], 1, 1,
972
                huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
973
        }
974
        vlcs_initialized = 1;
975
    }
976

    
977
    init_atrac3_transforms(q);
978

    
979
    atrac_generate_tables();
980

    
981
    /* Generate gain tables. */
982
    for (i=0 ; i<16 ; i++)
983
        gain_tab1[i] = powf (2.0, (4 - i));
984

    
985
    for (i=-15 ; i<16 ; i++)
986
        gain_tab2[i+15] = powf (2.0, i * -0.125);
987

    
988
    /* init the joint-stereo decoding data */
989
    q->weighting_delay[0] = 0;
990
    q->weighting_delay[1] = 7;
991
    q->weighting_delay[2] = 0;
992
    q->weighting_delay[3] = 7;
993
    q->weighting_delay[4] = 0;
994
    q->weighting_delay[5] = 7;
995

    
996
    for (i=0; i<4; i++) {
997
        q->matrix_coeff_index_prev[i] = 3;
998
        q->matrix_coeff_index_now[i] = 3;
999
        q->matrix_coeff_index_next[i] = 3;
1000
    }
1001

    
1002
    dsputil_init(&dsp, avctx);
1003

    
1004
    q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
1005
    if (!q->pUnits) {
1006
        av_free(q->decoded_bytes_buffer);
1007
        return AVERROR(ENOMEM);
1008
    }
1009

    
1010
    avctx->sample_fmt = SAMPLE_FMT_S16;
1011
    return 0;
1012
}
1013

    
1014

    
1015
AVCodec atrac3_decoder =
1016
{
1017
    .name = "atrac3",
1018
    .type = CODEC_TYPE_AUDIO,
1019
    .id = CODEC_ID_ATRAC3,
1020
    .priv_data_size = sizeof(ATRAC3Context),
1021
    .init = atrac3_decode_init,
1022
    .close = atrac3_decode_close,
1023
    .decode = atrac3_decode_frame,
1024
    .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
1025
};