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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
 *
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 * 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.
12
 *
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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
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 * 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 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 "bitstream.h"
41
#include "dsputil.h"
42
#include "bytestream.h"
43

    
44
#include "atrac3data.h"
45

    
46
#define JOINT_STEREO    0x12
47
#define STEREO          0x2
48

    
49

    
50
/* These structures are needed to store the parsed gain control data. */
51
typedef struct {
52
    int   num_gain_data;
53
    int   levcode[8];
54
    int   loccode[8];
55
} gain_info;
56

    
57
typedef struct {
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    gain_info   gBlock[4];
59
} gain_block;
60

    
61
typedef struct {
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    int     pos;
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    int     numCoefs;
64
    float   coef[8];
65
} tonal_component;
66

    
67
typedef struct {
68
    int               bandsCoded;
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    int               numComponents;
70
    tonal_component   components[64];
71
    float             prevFrame[1024];
72
    int               gcBlkSwitch;
73
    gain_block        gainBlock[2];
74

    
75
    DECLARE_ALIGNED_16(float, spectrum[1024]);
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    DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);
77

    
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    float             delayBuf1[46]; ///<qmf delay buffers
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    float             delayBuf2[46];
80
    float             delayBuf3[46];
81
} channel_unit;
82

    
83
typedef struct {
84
    GetBitContext       gb;
85
    //@{
86
    /** stream data */
87
    int                 channels;
88
    int                 codingMode;
89
    int                 bit_rate;
90
    int                 sample_rate;
91
    int                 samples_per_channel;
92
    int                 samples_per_frame;
93

    
94
    int                 bits_per_frame;
95
    int                 bytes_per_frame;
96
    int                 pBs;
97
    channel_unit*       pUnits;
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    //@}
99
    //@{
100
    /** joint-stereo related variables */
101
    int                 matrix_coeff_index_prev[4];
102
    int                 matrix_coeff_index_now[4];
103
    int                 matrix_coeff_index_next[4];
104
    int                 weighting_delay[6];
105
    //@}
106
    //@{
107
    /** data buffers */
108
    float               outSamples[2048];
109
    uint8_t*            decoded_bytes_buffer;
110
    float               tempBuf[1070];
111
    //@}
112
    //@{
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    /** extradata */
114
    int                 atrac3version;
115
    int                 delay;
116
    int                 scrambled_stream;
117
    int                 frame_factor;
118
    //@}
119
} ATRAC3Context;
120

    
121
static DECLARE_ALIGNED_16(float,mdct_window[512]);
122
static float            qmf_window[48];
123
static VLC              spectral_coeff_tab[7];
124
static float            SFTable[64];
125
static float            gain_tab1[16];
126
static float            gain_tab2[31];
127
static MDCTContext      mdct_ctx;
128
static DSPContext       dsp;
129

    
130

    
131
/* quadrature mirror synthesis filter */
132

    
133
/**
134
 * Quadrature mirror synthesis filter.
135
 *
136
 * @param inlo      lower part of spectrum
137
 * @param inhi      higher part of spectrum
138
 * @param nIn       size of spectrum buffer
139
 * @param pOut      out buffer
140
 * @param delayBuf  delayBuf buffer
141
 * @param temp      temp buffer
142
 */
143

    
144

    
145
static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
146
{
147
    int   i, j;
148
    float   *p1, *p3;
149

    
150
    memcpy(temp, delayBuf, 46*sizeof(float));
151

    
152
    p3 = temp + 46;
153

    
154
    /* loop1 */
155
    for(i=0; i<nIn; i+=2){
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        p3[2*i+0] = inlo[i  ] + inhi[i  ];
157
        p3[2*i+1] = inlo[i  ] - inhi[i  ];
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        p3[2*i+2] = inlo[i+1] + inhi[i+1];
159
        p3[2*i+3] = inlo[i+1] - inhi[i+1];
160
    }
161

    
162
    /* loop2 */
163
    p1 = temp;
164
    for (j = nIn; j != 0; j--) {
165
        float s1 = 0.0;
166
        float s2 = 0.0;
167

    
168
        for (i = 0; i < 48; i += 2) {
169
            s1 += p1[i] * qmf_window[i];
170
            s2 += p1[i+1] * qmf_window[i+1];
171
        }
172

    
173
        pOut[0] = s2;
174
        pOut[1] = s1;
175

    
176
        p1 += 2;
177
        pOut += 2;
178
    }
179

    
180
    /* Update the delay buffer. */
181
    memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
182
}
183

    
184
/**
185
 * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
186
 * caused by the reverse spectra of the QMF.
187
 *
188
 * @param pInput    float input
189
 * @param pOutput   float output
190
 * @param odd_band  1 if the band is an odd band
191
 */
192

    
193
static void IMLT(float *pInput, float *pOutput, int odd_band)
194
{
195
    int     i;
196

    
197
    if (odd_band) {
198
        /**
199
        * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
200
        * or it gives better compression to do it this way.
201
        * FIXME: It should be possible to handle this in ff_imdct_calc
202
        * for that to happen a modification of the prerotation step of
203
        * all SIMD code and C code is needed.
204
        * Or fix the functions before so they generate a pre reversed spectrum.
205
        */
206

    
207
        for (i=0; i<128; i++)
208
            FFSWAP(float, pInput[i], pInput[255-i]);
209
    }
210

    
211
    ff_imdct_calc(&mdct_ctx,pOutput,pInput);
212

    
213
    /* Perform windowing on the output. */
214
    dsp.vector_fmul(pOutput,mdct_window,512);
215

    
216
}
217

    
218

    
219
/**
220
 * Atrac 3 indata descrambling, only used for data coming from the rm container
221
 *
222
 * @param in        pointer to 8 bit array of indata
223
 * @param bits      amount of bits
224
 * @param out       pointer to 8 bit array of outdata
225
 */
226

    
227
static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
228
    int i, off;
229
    uint32_t c;
230
    const uint32_t* buf;
231
    uint32_t* obuf = (uint32_t*) out;
232

    
233
    off = (int)((long)inbuffer & 3);
234
    buf = (const uint32_t*) (inbuffer - off);
235
    c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
236
    bytes += 3 + off;
237
    for (i = 0; i < bytes/4; i++)
238
        obuf[i] = c ^ buf[i];
239

    
240
    if (off)
241
        av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
242

    
243
    return off;
244
}
245

    
246

    
247
static void init_atrac3_transforms(ATRAC3Context *q) {
248
    float enc_window[256];
249
    float s;
250
    int i;
251

    
252
    /* Generate the mdct window, for details see
253
     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
254
    for (i=0 ; i<256; i++)
255
        enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
256

    
257
    if (!mdct_window[0])
258
        for (i=0 ; i<256; i++) {
259
            mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
260
            mdct_window[511-i] = mdct_window[i];
261
        }
262

    
263
    /* Generate the QMF window. */
264
    for (i=0 ; i<24; i++) {
265
        s = qmf_48tap_half[i] * 2.0;
266
        qmf_window[i] = s;
267
        qmf_window[47 - i] = s;
268
    }
269

    
270
    /* Initialize the MDCT transform. */
271
    ff_mdct_init(&mdct_ctx, 9, 1);
272
}
273

    
274
/**
275
 * Atrac3 uninit, free all allocated memory
276
 */
277

    
278
static int atrac3_decode_close(AVCodecContext *avctx)
279
{
280
    ATRAC3Context *q = avctx->priv_data;
281

    
282
    av_free(q->pUnits);
283
    av_free(q->decoded_bytes_buffer);
284

    
285
    return 0;
286
}
287

    
288
/**
289
/ * Mantissa decoding
290
 *
291
 * @param gb            the GetBit context
292
 * @param selector      what table is the output values coded with
293
 * @param codingFlag    constant length coding or variable length coding
294
 * @param mantissas     mantissa output table
295
 * @param numCodes      amount of values to get
296
 */
297

    
298
static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
299
{
300
    int   numBits, cnt, code, huffSymb;
301

    
302
    if (selector == 1)
303
        numCodes /= 2;
304

    
305
    if (codingFlag != 0) {
306
        /* constant length coding (CLC) */
307
        numBits = CLCLengthTab[selector];
308

    
309
        if (selector > 1) {
310
            for (cnt = 0; cnt < numCodes; cnt++) {
311
                if (numBits)
312
                    code = get_sbits(gb, numBits);
313
                else
314
                    code = 0;
315
                mantissas[cnt] = code;
316
            }
317
        } else {
318
            for (cnt = 0; cnt < numCodes; cnt++) {
319
                if (numBits)
320
                    code = get_bits(gb, numBits); //numBits is always 4 in this case
321
                else
322
                    code = 0;
323
                mantissas[cnt*2] = seTab_0[code >> 2];
324
                mantissas[cnt*2+1] = seTab_0[code & 3];
325
            }
326
        }
327
    } else {
328
        /* variable length coding (VLC) */
329
        if (selector != 1) {
330
            for (cnt = 0; cnt < numCodes; cnt++) {
331
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
332
                huffSymb += 1;
333
                code = huffSymb >> 1;
334
                if (huffSymb & 1)
335
                    code = -code;
336
                mantissas[cnt] = code;
337
            }
338
        } else {
339
            for (cnt = 0; cnt < numCodes; cnt++) {
340
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
341
                mantissas[cnt*2] = decTable1[huffSymb*2];
342
                mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
343
            }
344
        }
345
    }
346
}
347

    
348
/**
349
 * Restore the quantized band spectrum coefficients
350
 *
351
 * @param gb            the GetBit context
352
 * @param pOut          decoded band spectrum
353
 * @return outSubbands   subband counter, fix for broken specification/files
354
 */
355

    
356
static int decodeSpectrum (GetBitContext *gb, float *pOut)
357
{
358
    int   numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
359
    int   subband_vlc_index[32], SF_idxs[32];
360
    int   mantissas[128];
361
    float SF;
362

    
363
    numSubbands = get_bits(gb, 5); // number of coded subbands
364
    codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
365

    
366
    /* Get the VLC selector table for the subbands, 0 means not coded. */
367
    for (cnt = 0; cnt <= numSubbands; cnt++)
368
        subband_vlc_index[cnt] = get_bits(gb, 3);
369

    
370
    /* Read the scale factor indexes from the stream. */
371
    for (cnt = 0; cnt <= numSubbands; cnt++) {
372
        if (subband_vlc_index[cnt] != 0)
373
            SF_idxs[cnt] = get_bits(gb, 6);
374
    }
375

    
376
    for (cnt = 0; cnt <= numSubbands; cnt++) {
377
        first = subbandTab[cnt];
378
        last = subbandTab[cnt+1];
379

    
380
        subbWidth = last - first;
381

    
382
        if (subband_vlc_index[cnt] != 0) {
383
            /* Decode spectral coefficients for this subband. */
384
            /* TODO: This can be done faster is several blocks share the
385
             * same VLC selector (subband_vlc_index) */
386
            readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
387

    
388
            /* Decode the scale factor for this subband. */
389
            SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
390

    
391
            /* Inverse quantize the coefficients. */
392
            for (pIn=mantissas ; first<last; first++, pIn++)
393
                pOut[first] = *pIn * SF;
394
        } else {
395
            /* This subband was not coded, so zero the entire subband. */
396
            memset(pOut+first, 0, subbWidth*sizeof(float));
397
        }
398
    }
399

    
400
    /* Clear the subbands that were not coded. */
401
    first = subbandTab[cnt];
402
    memset(pOut+first, 0, (1024 - first) * sizeof(float));
403
    return numSubbands;
404
}
405

    
406
/**
407
 * Restore the quantized tonal components
408
 *
409
 * @param gb            the GetBit context
410
 * @param pComponent    tone component
411
 * @param numBands      amount of coded bands
412
 */
413

    
414
static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
415
{
416
    int i,j,k,cnt;
417
    int   components, coding_mode_selector, coding_mode, coded_values_per_component;
418
    int   sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
419
    int   band_flags[4], mantissa[8];
420
    float  *pCoef;
421
    float  scalefactor;
422
    int   component_count = 0;
423

    
424
    components = get_bits(gb,5);
425

    
426
    /* no tonal components */
427
    if (components == 0)
428
        return 0;
429

    
430
    coding_mode_selector = get_bits(gb,2);
431
    if (coding_mode_selector == 2)
432
        return -1;
433

    
434
    coding_mode = coding_mode_selector & 1;
435

    
436
    for (i = 0; i < components; i++) {
437
        for (cnt = 0; cnt <= numBands; cnt++)
438
            band_flags[cnt] = get_bits1(gb);
439

    
440
        coded_values_per_component = get_bits(gb,3);
441

    
442
        quant_step_index = get_bits(gb,3);
443
        if (quant_step_index <= 1)
444
            return -1;
445

    
446
        if (coding_mode_selector == 3)
447
            coding_mode = get_bits1(gb);
448

    
449
        for (j = 0; j < (numBands + 1) * 4; j++) {
450
            if (band_flags[j >> 2] == 0)
451
                continue;
452

    
453
            coded_components = get_bits(gb,3);
454

    
455
            for (k=0; k<coded_components; k++) {
456
                sfIndx = get_bits(gb,6);
457
                pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
458
                max_coded_values = 1024 - pComponent[component_count].pos;
459
                coded_values = coded_values_per_component + 1;
460
                coded_values = FFMIN(max_coded_values,coded_values);
461

    
462
                scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];
463

    
464
                readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
465

    
466
                pComponent[component_count].numCoefs = coded_values;
467

    
468
                /* inverse quant */
469
                pCoef = pComponent[component_count].coef;
470
                for (cnt = 0; cnt < coded_values; cnt++)
471
                    pCoef[cnt] = mantissa[cnt] * scalefactor;
472

    
473
                component_count++;
474
            }
475
        }
476
    }
477

    
478
    return component_count;
479
}
480

    
481
/**
482
 * Decode gain parameters for the coded bands
483
 *
484
 * @param gb            the GetBit context
485
 * @param pGb           the gainblock for the current band
486
 * @param numBands      amount of coded bands
487
 */
488

    
489
static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
490
{
491
    int   i, cf, numData;
492
    int   *pLevel, *pLoc;
493

    
494
    gain_info   *pGain = pGb->gBlock;
495

    
496
    for (i=0 ; i<=numBands; i++)
497
    {
498
        numData = get_bits(gb,3);
499
        pGain[i].num_gain_data = numData;
500
        pLevel = pGain[i].levcode;
501
        pLoc = pGain[i].loccode;
502

    
503
        for (cf = 0; cf < numData; cf++){
504
            pLevel[cf]= get_bits(gb,4);
505
            pLoc  [cf]= get_bits(gb,5);
506
            if(cf && pLoc[cf] <= pLoc[cf-1])
507
                return -1;
508
        }
509
    }
510

    
511
    /* Clear the unused blocks. */
512
    for (; i<4 ; i++)
513
        pGain[i].num_gain_data = 0;
514

    
515
    return 0;
516
}
517

    
518
/**
519
 * Apply gain parameters and perform the MDCT overlapping part
520
 *
521
 * @param pIn           input float buffer
522
 * @param pPrev         previous float buffer to perform overlap against
523
 * @param pOut          output float buffer
524
 * @param pGain1        current band gain info
525
 * @param pGain2        next band gain info
526
 */
527

    
528
static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
529
{
530
    /* gain compensation function */
531
    float  gain1, gain2, gain_inc;
532
    int   cnt, numdata, nsample, startLoc, endLoc;
533

    
534

    
535
    if (pGain2->num_gain_data == 0)
536
        gain1 = 1.0;
537
    else
538
        gain1 = gain_tab1[pGain2->levcode[0]];
539

    
540
    if (pGain1->num_gain_data == 0) {
541
        for (cnt = 0; cnt < 256; cnt++)
542
            pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
543
    } else {
544
        numdata = pGain1->num_gain_data;
545
        pGain1->loccode[numdata] = 32;
546
        pGain1->levcode[numdata] = 4;
547

    
548
        nsample = 0; // current sample = 0
549

    
550
        for (cnt = 0; cnt < numdata; cnt++) {
551
            startLoc = pGain1->loccode[cnt] * 8;
552
            endLoc = startLoc + 8;
553

    
554
            gain2 = gain_tab1[pGain1->levcode[cnt]];
555
            gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
556

    
557
            /* interpolate */
558
            for (; nsample < startLoc; nsample++)
559
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
560

    
561
            /* interpolation is done over eight samples */
562
            for (; nsample < endLoc; nsample++) {
563
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
564
                gain2 *= gain_inc;
565
            }
566
        }
567

    
568
        for (; nsample < 256; nsample++)
569
            pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
570
    }
571

    
572
    /* Delay for the overlapping part. */
573
    memcpy(pPrev, &pIn[256], 256*sizeof(float));
574
}
575

    
576
/**
577
 * Combine the tonal band spectrum and regular band spectrum
578
 * Return position of the last tonal coefficient
579
 *
580
 * @param pSpectrum     output spectrum buffer
581
 * @param numComponents amount of tonal components
582
 * @param pComponent    tonal components for this band
583
 */
584

    
585
static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
586
{
587
    int   cnt, i, lastPos = -1;
588
    float   *pIn, *pOut;
589

    
590
    for (cnt = 0; cnt < numComponents; cnt++){
591
        lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
592
        pIn = pComponent[cnt].coef;
593
        pOut = &(pSpectrum[pComponent[cnt].pos]);
594

    
595
        for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
596
            pOut[i] += pIn[i];
597
    }
598

    
599
    return lastPos;
600
}
601

    
602

    
603
#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
604

    
605
static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
606
{
607
    int    i, band, nsample, s1, s2;
608
    float    c1, c2;
609
    float    mc1_l, mc1_r, mc2_l, mc2_r;
610

    
611
    for (i=0,band = 0; band < 4*256; band+=256,i++) {
612
        s1 = pPrevCode[i];
613
        s2 = pCurrCode[i];
614
        nsample = 0;
615

    
616
        if (s1 != s2) {
617
            /* Selector value changed, interpolation needed. */
618
            mc1_l = matrixCoeffs[s1*2];
619
            mc1_r = matrixCoeffs[s1*2+1];
620
            mc2_l = matrixCoeffs[s2*2];
621
            mc2_r = matrixCoeffs[s2*2+1];
622

    
623
            /* Interpolation is done over the first eight samples. */
624
            for(; nsample < 8; nsample++) {
625
                c1 = su1[band+nsample];
626
                c2 = su2[band+nsample];
627
                c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
628
                su1[band+nsample] = c2;
629
                su2[band+nsample] = c1 * 2.0 - c2;
630
            }
631
        }
632

    
633
        /* Apply the matrix without interpolation. */
634
        switch (s2) {
635
            case 0:     /* M/S decoding */
636
                for (; nsample < 256; nsample++) {
637
                    c1 = su1[band+nsample];
638
                    c2 = su2[band+nsample];
639
                    su1[band+nsample] = c2 * 2.0;
640
                    su2[band+nsample] = (c1 - c2) * 2.0;
641
                }
642
                break;
643

    
644
            case 1:
645
                for (; nsample < 256; nsample++) {
646
                    c1 = su1[band+nsample];
647
                    c2 = su2[band+nsample];
648
                    su1[band+nsample] = (c1 + c2) * 2.0;
649
                    su2[band+nsample] = c2 * -2.0;
650
                }
651
                break;
652
            case 2:
653
            case 3:
654
                for (; nsample < 256; nsample++) {
655
                    c1 = su1[band+nsample];
656
                    c2 = su2[band+nsample];
657
                    su1[band+nsample] = c1 + c2;
658
                    su2[band+nsample] = c1 - c2;
659
                }
660
                break;
661
            default:
662
                assert(0);
663
        }
664
    }
665
}
666

    
667
static void getChannelWeights (int indx, int flag, float ch[2]){
668

    
669
    if (indx == 7) {
670
        ch[0] = 1.0;
671
        ch[1] = 1.0;
672
    } else {
673
        ch[0] = (float)(indx & 7) / 7.0;
674
        ch[1] = sqrt(2 - ch[0]*ch[0]);
675
        if(flag)
676
            FFSWAP(float, ch[0], ch[1]);
677
    }
678
}
679

    
680
static void channelWeighting (float *su1, float *su2, int *p3)
681
{
682
    int   band, nsample;
683
    /* w[x][y] y=0 is left y=1 is right */
684
    float w[2][2];
685

    
686
    if (p3[1] != 7 || p3[3] != 7){
687
        getChannelWeights(p3[1], p3[0], w[0]);
688
        getChannelWeights(p3[3], p3[2], w[1]);
689

    
690
        for(band = 1; band < 4; band++) {
691
            /* scale the channels by the weights */
692
            for(nsample = 0; nsample < 8; nsample++) {
693
                su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
694
                su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
695
            }
696

    
697
            for(; nsample < 256; nsample++) {
698
                su1[band*256+nsample] *= w[1][0];
699
                su2[band*256+nsample] *= w[1][1];
700
            }
701
        }
702
    }
703
}
704

    
705

    
706
/**
707
 * Decode a Sound Unit
708
 *
709
 * @param gb            the GetBit context
710
 * @param pSnd          the channel unit to be used
711
 * @param pOut          the decoded samples before IQMF in float representation
712
 * @param channelNum    channel number
713
 * @param codingMode    the coding mode (JOINT_STEREO or regular stereo/mono)
714
 */
715

    
716

    
717
static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
718
{
719
    int   band, result=0, numSubbands, lastTonal, numBands;
720

    
721
    if (codingMode == JOINT_STEREO && channelNum == 1) {
722
        if (get_bits(gb,2) != 3) {
723
            av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
724
            return -1;
725
        }
726
    } else {
727
        if (get_bits(gb,6) != 0x28) {
728
            av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
729
            return -1;
730
        }
731
    }
732

    
733
    /* number of coded QMF bands */
734
    pSnd->bandsCoded = get_bits(gb,2);
735

    
736
    result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
737
    if (result) return result;
738

    
739
    pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
740
    if (pSnd->numComponents == -1) return -1;
741

    
742
    numSubbands = decodeSpectrum (gb, pSnd->spectrum);
743

    
744
    /* Merge the decoded spectrum and tonal components. */
745
    lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
746

    
747

    
748
    /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
749
    numBands = (subbandTab[numSubbands] - 1) >> 8;
750
    if (lastTonal >= 0)
751
        numBands = FFMAX((lastTonal + 256) >> 8, numBands);
752

    
753

    
754
    /* Reconstruct time domain samples. */
755
    for (band=0; band<4; band++) {
756
        /* Perform the IMDCT step without overlapping. */
757
        if (band <= numBands) {
758
            IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
759
        } else
760
            memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
761

    
762
        /* gain compensation and overlapping */
763
        gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
764
                                    &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
765
                                    &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
766
    }
767

    
768
    /* Swap the gain control buffers for the next frame. */
769
    pSnd->gcBlkSwitch ^= 1;
770

    
771
    return 0;
772
}
773

    
774
/**
775
 * Frame handling
776
 *
777
 * @param q             Atrac3 private context
778
 * @param databuf       the input data
779
 */
780

    
781
static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
782
{
783
    int   result, i;
784
    float   *p1, *p2, *p3, *p4;
785
    uint8_t *ptr1;
786

    
787
    if (q->codingMode == JOINT_STEREO) {
788

    
789
        /* channel coupling mode */
790
        /* decode Sound Unit 1 */
791
        init_get_bits(&q->gb,databuf,q->bits_per_frame);
792

    
793
        result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
794
        if (result != 0)
795
            return (result);
796

    
797
        /* Framedata of the su2 in the joint-stereo mode is encoded in
798
         * reverse byte order so we need to swap it first. */
799
        if (databuf == q->decoded_bytes_buffer) {
800
            uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
801
            ptr1 = q->decoded_bytes_buffer;
802
        for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
803
            FFSWAP(uint8_t,*ptr1,*ptr2);
804
        }
805
        } else {
806
            const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
807
            for (i = 0; i < q->bytes_per_frame; i++)
808
                q->decoded_bytes_buffer[i] = *ptr2--;
809
        }
810

    
811
        /* Skip the sync codes (0xF8). */
812
        ptr1 = q->decoded_bytes_buffer;
813
        for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
814
            if (i >= q->bytes_per_frame)
815
                return -1;
816
        }
817

    
818

    
819
        /* set the bitstream reader at the start of the second Sound Unit*/
820
        init_get_bits(&q->gb,ptr1,q->bits_per_frame);
821

    
822
        /* Fill the Weighting coeffs delay buffer */
823
        memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
824
        q->weighting_delay[4] = get_bits1(&q->gb);
825
        q->weighting_delay[5] = get_bits(&q->gb,3);
826

    
827
        for (i = 0; i < 4; i++) {
828
            q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
829
            q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
830
            q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
831
        }
832

    
833
        /* Decode Sound Unit 2. */
834
        result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
835
        if (result != 0)
836
            return (result);
837

    
838
        /* Reconstruct the channel coefficients. */
839
        reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
840

    
841
        channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
842

    
843
    } else {
844
        /* normal stereo mode or mono */
845
        /* Decode the channel sound units. */
846
        for (i=0 ; i<q->channels ; i++) {
847

    
848
            /* Set the bitstream reader at the start of a channel sound unit. */
849
            init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
850

    
851
            result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
852
            if (result != 0)
853
                return (result);
854
        }
855
    }
856

    
857
    /* Apply the iQMF synthesis filter. */
858
    p1= q->outSamples;
859
    for (i=0 ; i<q->channels ; i++) {
860
        p2= p1+256;
861
        p3= p2+256;
862
        p4= p3+256;
863
        iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
864
        iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
865
        iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
866
        p1 +=1024;
867
    }
868

    
869
    return 0;
870
}
871

    
872

    
873
/**
874
 * Atrac frame decoding
875
 *
876
 * @param avctx     pointer to the AVCodecContext
877
 */
878

    
879
static int atrac3_decode_frame(AVCodecContext *avctx,
880
            void *data, int *data_size,
881
            const uint8_t *buf, int buf_size) {
882
    ATRAC3Context *q = avctx->priv_data;
883
    int result = 0, i;
884
    const uint8_t* databuf;
885
    int16_t* samples = data;
886

    
887
    if (buf_size < avctx->block_align)
888
        return buf_size;
889

    
890
    /* Check if we need to descramble and what buffer to pass on. */
891
    if (q->scrambled_stream) {
892
        decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
893
        databuf = q->decoded_bytes_buffer;
894
    } else {
895
        databuf = buf;
896
    }
897

    
898
    result = decodeFrame(q, databuf);
899

    
900
    if (result != 0) {
901
        av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
902
        return -1;
903
    }
904

    
905
    if (q->channels == 1) {
906
        /* mono */
907
        for (i = 0; i<1024; i++)
908
            samples[i] = av_clip_int16(round(q->outSamples[i]));
909
        *data_size = 1024 * sizeof(int16_t);
910
    } else {
911
        /* stereo */
912
        for (i = 0; i < 1024; i++) {
913
            samples[i*2] = av_clip_int16(round(q->outSamples[i]));
914
            samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
915
        }
916
        *data_size = 2048 * sizeof(int16_t);
917
    }
918

    
919
    return avctx->block_align;
920
}
921

    
922

    
923
/**
924
 * Atrac3 initialization
925
 *
926
 * @param avctx     pointer to the AVCodecContext
927
 */
928

    
929
static int atrac3_decode_init(AVCodecContext *avctx)
930
{
931
    int i;
932
    const uint8_t *edata_ptr = avctx->extradata;
933
    ATRAC3Context *q = avctx->priv_data;
934

    
935
    /* Take data from the AVCodecContext (RM container). */
936
    q->sample_rate = avctx->sample_rate;
937
    q->channels = avctx->channels;
938
    q->bit_rate = avctx->bit_rate;
939
    q->bits_per_frame = avctx->block_align * 8;
940
    q->bytes_per_frame = avctx->block_align;
941

    
942
    /* Take care of the codec-specific extradata. */
943
    if (avctx->extradata_size == 14) {
944
        /* Parse the extradata, WAV format */
945
        av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown value always 1
946
        q->samples_per_channel = bytestream_get_le32(&edata_ptr);
947
        q->codingMode = bytestream_get_le16(&edata_ptr);
948
        av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
949
        q->frame_factor = bytestream_get_le16(&edata_ptr);  //Unknown always 1
950
        av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown always 0
951

    
952
        /* setup */
953
        q->samples_per_frame = 1024 * q->channels;
954
        q->atrac3version = 4;
955
        q->delay = 0x88E;
956
        if (q->codingMode)
957
            q->codingMode = JOINT_STEREO;
958
        else
959
            q->codingMode = STEREO;
960

    
961
        q->scrambled_stream = 0;
962

    
963
        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)) {
964
        } else {
965
            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);
966
            return -1;
967
        }
968

    
969
    } else if (avctx->extradata_size == 10) {
970
        /* Parse the extradata, RM format. */
971
        q->atrac3version = bytestream_get_be32(&edata_ptr);
972
        q->samples_per_frame = bytestream_get_be16(&edata_ptr);
973
        q->delay = bytestream_get_be16(&edata_ptr);
974
        q->codingMode = bytestream_get_be16(&edata_ptr);
975

    
976
        q->samples_per_channel = q->samples_per_frame / q->channels;
977
        q->scrambled_stream = 1;
978

    
979
    } else {
980
        av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
981
    }
982
    /* Check the extradata. */
983

    
984
    if (q->atrac3version != 4) {
985
        av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
986
        return -1;
987
    }
988

    
989
    if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
990
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
991
        return -1;
992
    }
993

    
994
    if (q->delay != 0x88E) {
995
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
996
        return -1;
997
    }
998

    
999
    if (q->codingMode == STEREO) {
1000
        av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
1001
    } else if (q->codingMode == JOINT_STEREO) {
1002
        av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
1003
    } else {
1004
        av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
1005
        return -1;
1006
    }
1007

    
1008
    if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
1009
        av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
1010
        return -1;
1011
    }
1012

    
1013

    
1014
    if(avctx->block_align >= UINT_MAX/2)
1015
        return -1;
1016

    
1017
    /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
1018
     * this is for the bitstream reader. */
1019
    if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)
1020
        return AVERROR(ENOMEM);
1021

    
1022

    
1023
    /* Initialize the VLC tables. */
1024
    for (i=0 ; i<7 ; i++) {
1025
        init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
1026
            huff_bits[i], 1, 1,
1027
            huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);
1028
    }
1029

    
1030
    init_atrac3_transforms(q);
1031

    
1032
    /* Generate the scale factors. */
1033
    for (i=0 ; i<64 ; i++)
1034
        SFTable[i] = pow(2.0, (i - 15) / 3.0);
1035

    
1036
    /* Generate gain tables. */
1037
    for (i=0 ; i<16 ; i++)
1038
        gain_tab1[i] = powf (2.0, (4 - i));
1039

    
1040
    for (i=-15 ; i<16 ; i++)
1041
        gain_tab2[i+15] = powf (2.0, i * -0.125);
1042

    
1043
    /* init the joint-stereo decoding data */
1044
    q->weighting_delay[0] = 0;
1045
    q->weighting_delay[1] = 7;
1046
    q->weighting_delay[2] = 0;
1047
    q->weighting_delay[3] = 7;
1048
    q->weighting_delay[4] = 0;
1049
    q->weighting_delay[5] = 7;
1050

    
1051
    for (i=0; i<4; i++) {
1052
        q->matrix_coeff_index_prev[i] = 3;
1053
        q->matrix_coeff_index_now[i] = 3;
1054
        q->matrix_coeff_index_next[i] = 3;
1055
    }
1056

    
1057
    dsputil_init(&dsp, avctx);
1058

    
1059
    q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
1060
    if (!q->pUnits) {
1061
        av_free(q->decoded_bytes_buffer);
1062
        return AVERROR(ENOMEM);
1063
    }
1064

    
1065
    avctx->sample_fmt = SAMPLE_FMT_S16;
1066
    return 0;
1067
}
1068

    
1069

    
1070
AVCodec atrac3_decoder =
1071
{
1072
    .name = "atrac3",
1073
    .type = CODEC_TYPE_AUDIO,
1074
    .id = CODEC_ID_ATRAC3,
1075
    .priv_data_size = sizeof(ATRAC3Context),
1076
    .init = atrac3_decode_init,
1077
    .close = atrac3_decode_close,
1078
    .decode = atrac3_decode_frame,
1079
    .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
1080
};