Statistics
| Branch: | Revision:

ffmpeg / libavcodec / atrac3.c @ 10e26bc7

History | View | Annotate | Download (32.1 KB)

1
/*
2
 * Atrac 3 compatible decoder
3
 * Copyright (c) 2006-2007 Maxim Poliakovski
4
 * Copyright (c) 2006-2007 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.
12
 *
13
 * 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.
17
 *
18
 * You should have received a copy of the GNU Lesser General Public
19
 * License along with FFmpeg; if not, write to the Free Software
20
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21
 */
22

    
23
/**
24
 * @file atrac3.c
25
 * Atrac 3 compatible decoder.
26
 * This decoder handles RealNetworks, RealAudio atrc data.
27
 * Atrac 3 is identified by the codec name atrc in RealMedia files.
28
 *
29
 * To use this decoder, a calling application must supply the extradata
30
 * bytes provided from the RealMedia container: 10 bytes or 14 bytes
31
 * from the WAV container.
32
 */
33

    
34
#include <math.h>
35
#include <stddef.h>
36
#include <stdio.h>
37

    
38
#include "avcodec.h"
39
#include "bitstream.h"
40
#include "dsputil.h"
41
#include "bytestream.h"
42

    
43
#include "atrac3data.h"
44

    
45
#define JOINT_STEREO    0x12
46
#define STEREO          0x2
47

    
48

    
49
/* These structures are needed to store the parsed gain control data. */
50
typedef struct {
51
    int   num_gain_data;
52
    int   levcode[8];
53
    int   loccode[8];
54
} gain_info;
55

    
56
typedef struct {
57
    gain_info   gBlock[4];
58
} gain_block;
59

    
60
typedef struct {
61
    int     pos;
62
    int     numCoefs;
63
    float   coef[8];
64
} tonal_component;
65

    
66
typedef struct {
67
    int               bandsCoded;
68
    int               numComponents;
69
    tonal_component   components[64];
70
    float             prevFrame[1024];
71
    int               gcBlkSwitch;
72
    gain_block        gainBlock[2];
73

    
74
    DECLARE_ALIGNED_16(float, spectrum[1024]);
75
    DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);
76

    
77
    float             delayBuf1[46]; ///<qmf delay buffers
78
    float             delayBuf2[46];
79
    float             delayBuf3[46];
80
} channel_unit;
81

    
82
typedef struct {
83
    GetBitContext       gb;
84
    //@{
85
    /** stream data */
86
    int                 channels;
87
    int                 codingMode;
88
    int                 bit_rate;
89
    int                 sample_rate;
90
    int                 samples_per_channel;
91
    int                 samples_per_frame;
92

    
93
    int                 bits_per_frame;
94
    int                 bytes_per_frame;
95
    int                 pBs;
96
    channel_unit*       pUnits;
97
    //@}
98
    //@{
99
    /** joint-stereo related variables */
100
    int                 matrix_coeff_index_prev[4];
101
    int                 matrix_coeff_index_now[4];
102
    int                 matrix_coeff_index_next[4];
103
    int                 weighting_delay[6];
104
    //@}
105
    //@{
106
    /** data buffers */
107
    float               outSamples[2048];
108
    uint8_t*            decoded_bytes_buffer;
109
    float               tempBuf[1070];
110
    DECLARE_ALIGNED_16(float,mdct_tmp[512]);
111
    //@}
112
    //@{
113
    /** 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){
156
        p3[2*i+0] = inlo[i  ] + inhi[i  ];
157
        p3[2*i+1] = inlo[i  ] - inhi[i  ];
158
        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
 * @param mdct_tmp  aligned temporary buffer for the mdct
192
 */
193

    
194
static void IMLT(float *pInput, float *pOutput, int odd_band, float* mdct_tmp)
195
{
196
    int     i;
197

    
198
    if (odd_band) {
199
        /**
200
        * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
201
        * or it gives better compression to do it this way.
202
        * FIXME: It should be possible to handle this in ff_imdct_calc
203
        * for that to happen a modification of the prerotation step of
204
        * all SIMD code and C code is needed.
205
        * Or fix the functions before so they generate a pre reversed spectrum.
206
        */
207

    
208
        for (i=0; i<128; i++)
209
            FFSWAP(float, pInput[i], pInput[255-i]);
210
    }
211

    
212
    mdct_ctx.fft.imdct_calc(&mdct_ctx,pOutput,pInput,mdct_tmp);
213

    
214
    /* Perform windowing on the output. */
215
    dsp.vector_fmul(pOutput,mdct_window,512);
216

    
217
}
218

    
219

    
220
/**
221
 * Atrac 3 indata descrambling, only used for data coming from the rm container
222
 *
223
 * @param in        pointer to 8 bit array of indata
224
 * @param bits      amount of bits
225
 * @param out       pointer to 8 bit array of outdata
226
 */
227

    
228
static int decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){
229
    int i, off;
230
    uint32_t c;
231
    uint32_t* buf;
232
    uint32_t* obuf = (uint32_t*) out;
233

    
234
    off = (int)((long)inbuffer & 3);
235
    buf = (uint32_t*) (inbuffer - off);
236
    c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
237
    bytes += 3 + off;
238
    for (i = 0; i < bytes/4; i++)
239
        obuf[i] = c ^ buf[i];
240

    
241
    if (off)
242
        av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
243

    
244
    return off;
245
}
246

    
247

    
248
static void init_atrac3_transforms(ATRAC3Context *q) {
249
    float enc_window[256];
250
    float s;
251
    int i;
252

    
253
    /* Generate the mdct window, for details see
254
     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
255
    for (i=0 ; i<256; i++)
256
        enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
257

    
258
    if (!mdct_window[0])
259
        for (i=0 ; i<256; i++) {
260
            mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
261
            mdct_window[511-i] = mdct_window[i];
262
        }
263

    
264
    /* Generate the QMF window. */
265
    for (i=0 ; i<24; i++) {
266
        s = qmf_48tap_half[i] * 2.0;
267
        qmf_window[i] = s;
268
        qmf_window[47 - i] = s;
269
    }
270

    
271
    /* Initialize the MDCT transform. */
272
    ff_mdct_init(&mdct_ctx, 9, 1);
273
}
274

    
275
/**
276
 * Atrac3 uninit, free all allocated memory
277
 */
278

    
279
static int atrac3_decode_close(AVCodecContext *avctx)
280
{
281
    ATRAC3Context *q = avctx->priv_data;
282

    
283
    av_free(q->pUnits);
284
    av_free(q->decoded_bytes_buffer);
285

    
286
    return 0;
287
}
288

    
289
/**
290
/ * Mantissa decoding
291
 *
292
 * @param gb            the GetBit context
293
 * @param selector      what table is the output values coded with
294
 * @param codingFlag    constant length coding or variable length coding
295
 * @param mantissas     mantissa output table
296
 * @param numCodes      amount of values to get
297
 */
298

    
299
static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
300
{
301
    int   numBits, cnt, code, huffSymb;
302

    
303
    if (selector == 1)
304
        numCodes /= 2;
305

    
306
    if (codingFlag != 0) {
307
        /* constant length coding (CLC) */
308
        //FIXME we don't have any samples coded in CLC mode
309
        numBits = CLCLengthTab[selector];
310

    
311
        if (selector > 1) {
312
            for (cnt = 0; cnt < numCodes; cnt++) {
313
                if (numBits)
314
                    code = get_sbits(gb, numBits);
315
                else
316
                    code = 0;
317
                mantissas[cnt] = code;
318
            }
319
        } else {
320
            for (cnt = 0; cnt < numCodes; cnt++) {
321
                if (numBits)
322
                    code = get_bits(gb, numBits); //numBits is always 4 in this case
323
                else
324
                    code = 0;
325
                mantissas[cnt*2] = seTab_0[code >> 2];
326
                mantissas[cnt*2+1] = seTab_0[code & 3];
327
            }
328
        }
329
    } else {
330
        /* variable length coding (VLC) */
331
        if (selector != 1) {
332
            for (cnt = 0; cnt < numCodes; cnt++) {
333
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
334
                huffSymb += 1;
335
                code = huffSymb >> 1;
336
                if (huffSymb & 1)
337
                    code = -code;
338
                mantissas[cnt] = code;
339
            }
340
        } else {
341
            for (cnt = 0; cnt < numCodes; cnt++) {
342
                huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
343
                mantissas[cnt*2] = decTable1[huffSymb*2];
344
                mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
345
            }
346
        }
347
    }
348
}
349

    
350
/**
351
 * Restore the quantized band spectrum coefficients
352
 *
353
 * @param gb            the GetBit context
354
 * @param pOut          decoded band spectrum
355
 * @return outSubbands   subband counter, fix for broken specification/files
356
 */
357

    
358
static int decodeSpectrum (GetBitContext *gb, float *pOut)
359
{
360
    int   numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
361
    int   subband_vlc_index[32], SF_idxs[32];
362
    int   mantissas[128];
363
    float SF;
364

    
365
    numSubbands = get_bits(gb, 5); // number of coded subbands
366
    codingMode = get_bits(gb, 1); // coding Mode: 0 - VLC/ 1-CLC
367

    
368
    /* Get the VLC selector table for the subbands, 0 means not coded. */
369
    for (cnt = 0; cnt <= numSubbands; cnt++)
370
        subband_vlc_index[cnt] = get_bits(gb, 3);
371

    
372
    /* Read the scale factor indexes from the stream. */
373
    for (cnt = 0; cnt <= numSubbands; cnt++) {
374
        if (subband_vlc_index[cnt] != 0)
375
            SF_idxs[cnt] = get_bits(gb, 6);
376
    }
377

    
378
    for (cnt = 0; cnt <= numSubbands; cnt++) {
379
        first = subbandTab[cnt];
380
        last = subbandTab[cnt+1];
381

    
382
        subbWidth = last - first;
383

    
384
        if (subband_vlc_index[cnt] != 0) {
385
            /* Decode spectral coefficients for this subband. */
386
            /* TODO: This can be done faster is several blocks share the
387
             * same VLC selector (subband_vlc_index) */
388
            readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
389

    
390
            /* Decode the scale factor for this subband. */
391
            SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
392

    
393
            /* Inverse quantize the coefficients. */
394
            for (pIn=mantissas ; first<last; first++, pIn++)
395
                pOut[first] = *pIn * SF;
396
        } else {
397
            /* This subband was not coded, so zero the entire subband. */
398
            memset(pOut+first, 0, subbWidth*sizeof(float));
399
        }
400
    }
401

    
402
    /* Clear the subbands that were not coded. */
403
    first = subbandTab[cnt];
404
    memset(pOut+first, 0, (1024 - first) * sizeof(float));
405
    return numSubbands;
406
}
407

    
408
/**
409
 * Restore the quantized tonal components
410
 *
411
 * @param gb            the GetBit context
412
 * @param numComponents tonal components to report back
413
 * @param pComponent    tone component
414
 * @param numBands      amount of coded bands
415
 */
416

    
417
static int decodeTonalComponents (GetBitContext *gb, int *numComponents, tonal_component *pComponent, int numBands)
418
{
419
    int i,j,k,cnt;
420
    int   component_count, components, coding_mode_selector, coding_mode, coded_values_per_component;
421
    int   sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
422
    int   band_flags[4], mantissa[8];
423
    float  *pCoef;
424
    float  scalefactor;
425

    
426
    component_count = 0;
427
    *numComponents = 0;
428

    
429
    components = get_bits(gb,5);
430

    
431
    /* no tonal components */
432
    if (components == 0)
433
        return 0;
434

    
435
    coding_mode_selector = get_bits(gb,2);
436
    if (coding_mode_selector == 2)
437
        return -1;
438

    
439
    coding_mode = coding_mode_selector & 1;
440

    
441
    for (i = 0; i < components; i++) {
442
        for (cnt = 0; cnt <= numBands; cnt++)
443
            band_flags[cnt] = get_bits1(gb);
444

    
445
        coded_values_per_component = get_bits(gb,3);
446

    
447
        quant_step_index = get_bits(gb,3);
448
        if (quant_step_index <= 1)
449
            return -1;
450

    
451
        if (coding_mode_selector == 3)
452
            coding_mode = get_bits1(gb);
453

    
454
        for (j = 0; j < (numBands + 1) * 4; j++) {
455
            if (band_flags[j >> 2] == 0)
456
                continue;
457

    
458
            coded_components = get_bits(gb,3);
459

    
460
            for (k=0; k<coded_components; k++) {
461
                sfIndx = get_bits(gb,6);
462
                pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
463
                max_coded_values = 1024 - pComponent[component_count].pos;
464
                coded_values = coded_values_per_component + 1;
465
                coded_values = FFMIN(max_coded_values,coded_values);
466

    
467
                scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];
468

    
469
                readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
470

    
471
                pComponent[component_count].numCoefs = coded_values;
472

    
473
                /* inverse quant */
474
                pCoef = pComponent[k].coef;
475
                for (cnt = 0; cnt < coded_values; cnt++)
476
                    pCoef[cnt] = mantissa[cnt] * scalefactor;
477

    
478
                component_count++;
479
            }
480
        }
481
    }
482

    
483
    *numComponents = component_count;
484

    
485
    return 0;
486
}
487

    
488
/**
489
 * Decode gain parameters for the coded bands
490
 *
491
 * @param gb            the GetBit context
492
 * @param pGb           the gainblock for the current band
493
 * @param numBands      amount of coded bands
494
 */
495

    
496
static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
497
{
498
    int   i, cf, numData;
499
    int   *pLevel, *pLoc;
500

    
501
    gain_info   *pGain = pGb->gBlock;
502

    
503
    for (i=0 ; i<=numBands; i++)
504
    {
505
        numData = get_bits(gb,3);
506
        pGain[i].num_gain_data = numData;
507
        pLevel = pGain[i].levcode;
508
        pLoc = pGain[i].loccode;
509

    
510
        for (cf = 0; cf < numData; cf++){
511
            pLevel[cf]= get_bits(gb,4);
512
            pLoc  [cf]= get_bits(gb,5);
513
            if(cf && pLoc[cf] <= pLoc[cf-1])
514
                return -1;
515
        }
516
    }
517

    
518
    /* Clear the unused blocks. */
519
    for (; i<4 ; i++)
520
        pGain[i].num_gain_data = 0;
521

    
522
    return 0;
523
}
524

    
525
/**
526
 * Apply gain parameters and perform the MDCT overlapping part
527
 *
528
 * @param pIn           input float buffer
529
 * @param pPrev         previous float buffer to perform overlap against
530
 * @param pOut          output float buffer
531
 * @param pGain1        current band gain info
532
 * @param pGain2        next band gain info
533
 */
534

    
535
static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
536
{
537
    /* gain compensation function */
538
    float  gain1, gain2, gain_inc;
539
    int   cnt, numdata, nsample, startLoc, endLoc;
540

    
541

    
542
    if (pGain2->num_gain_data == 0)
543
        gain1 = 1.0;
544
    else
545
        gain1 = gain_tab1[pGain2->levcode[0]];
546

    
547
    if (pGain1->num_gain_data == 0) {
548
        for (cnt = 0; cnt < 256; cnt++)
549
            pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
550
    } else {
551
        numdata = pGain1->num_gain_data;
552
        pGain1->loccode[numdata] = 32;
553
        pGain1->levcode[numdata] = 4;
554

    
555
        nsample = 0; // current sample = 0
556

    
557
        for (cnt = 0; cnt < numdata; cnt++) {
558
            startLoc = pGain1->loccode[cnt] * 8;
559
            endLoc = startLoc + 8;
560

    
561
            gain2 = gain_tab1[pGain1->levcode[cnt]];
562
            gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
563

    
564
            /* interpolate */
565
            for (; nsample < startLoc; nsample++)
566
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
567

    
568
            /* interpolation is done over eight samples */
569
            for (; nsample < endLoc; nsample++) {
570
                pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
571
                gain2 *= gain_inc;
572
            }
573
        }
574

    
575
        for (; nsample < 256; nsample++)
576
            pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
577
    }
578

    
579
    /* Delay for the overlapping part. */
580
    memcpy(pPrev, &pIn[256], 256*sizeof(float));
581
}
582

    
583
/**
584
 * Combine the tonal band spectrum and regular band spectrum
585
 *
586
 * @param pSpectrum     output spectrum buffer
587
 * @param numComponents amount of tonal components
588
 * @param pComponent    tonal components for this band
589
 */
590

    
591
static void addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
592
{
593
    int   cnt, i;
594
    float   *pIn, *pOut;
595

    
596
    for (cnt = 0; cnt < numComponents; cnt++){
597
        pIn = pComponent[cnt].coef;
598
        pOut = &(pSpectrum[pComponent[cnt].pos]);
599

    
600
        for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
601
            pOut[i] += pIn[i];
602
    }
603
}
604

    
605

    
606
#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
607

    
608
static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
609
{
610
    int    i, band, nsample, s1, s2;
611
    float    c1, c2;
612
    float    mc1_l, mc1_r, mc2_l, mc2_r;
613

    
614
    for (i=0,band = 0; band < 4*256; band+=256,i++) {
615
        s1 = pPrevCode[i];
616
        s2 = pCurrCode[i];
617
        nsample = 0;
618

    
619
        if (s1 != s2) {
620
            /* Selector value changed, interpolation needed. */
621
            mc1_l = matrixCoeffs[s1*2];
622
            mc1_r = matrixCoeffs[s1*2+1];
623
            mc2_l = matrixCoeffs[s2*2];
624
            mc2_r = matrixCoeffs[s2*2+1];
625

    
626
            /* Interpolation is done over the first eight samples. */
627
            for(; nsample < 8; nsample++) {
628
                c1 = su1[band+nsample];
629
                c2 = su2[band+nsample];
630
                c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
631
                su1[band+nsample] = c2;
632
                su2[band+nsample] = c1 * 2.0 - c2;
633
            }
634
        }
635

    
636
        /* Apply the matrix without interpolation. */
637
        switch (s2) {
638
            case 0:     /* M/S decoding */
639
                for (; nsample < 256; nsample++) {
640
                    c1 = su1[band+nsample];
641
                    c2 = su2[band+nsample];
642
                    su1[band+nsample] = c2 * 2.0;
643
                    su2[band+nsample] = (c1 - c2) * 2.0;
644
                }
645
                break;
646

    
647
            case 1:
648
                for (; nsample < 256; nsample++) {
649
                    c1 = su1[band+nsample];
650
                    c2 = su2[band+nsample];
651
                    su1[band+nsample] = (c1 + c2) * 2.0;
652
                    su2[band+nsample] = c2 * -2.0;
653
                }
654
                break;
655
            case 2:
656
            case 3:
657
                for (; nsample < 256; nsample++) {
658
                    c1 = su1[band+nsample];
659
                    c2 = su2[band+nsample];
660
                    su1[band+nsample] = c1 + c2;
661
                    su2[band+nsample] = c1 - c2;
662
                }
663
                break;
664
            default:
665
                assert(0);
666
        }
667
    }
668
}
669

    
670
static void getChannelWeights (int indx, int flag, float ch[2]){
671

    
672
    if (indx == 7) {
673
        ch[0] = 1.0;
674
        ch[1] = 1.0;
675
    } else {
676
        ch[0] = (float)(indx & 7) / 7.0;
677
        ch[1] = sqrt(2 - ch[0]*ch[0]);
678
        if(flag)
679
            FFSWAP(float, ch[0], ch[1]);
680
    }
681
}
682

    
683
static void channelWeighting (float *su1, float *su2, int *p3)
684
{
685
    int   band, nsample;
686
    /* w[x][y] y=0 is left y=1 is right */
687
    float w[2][2];
688

    
689
    if (p3[1] != 7 || p3[3] != 7){
690
        getChannelWeights(p3[1], p3[0], w[0]);
691
        getChannelWeights(p3[3], p3[2], w[1]);
692

    
693
        for(band = 1; band < 4; band++) {
694
            /* scale the channels by the weights */
695
            for(nsample = 0; nsample < 8; nsample++) {
696
                su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
697
                su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
698
            }
699

    
700
            for(; nsample < 256; nsample++) {
701
                su1[band*256+nsample] *= w[1][0];
702
                su2[band*256+nsample] *= w[1][1];
703
            }
704
        }
705
    }
706
}
707

    
708

    
709
/**
710
 * Decode a Sound Unit
711
 *
712
 * @param gb            the GetBit context
713
 * @param pSnd          the channel unit to be used
714
 * @param pOut          the decoded samples before IQMF in float representation
715
 * @param channelNum    channel number
716
 * @param codingMode    the coding mode (JOINT_STEREO or regular stereo/mono)
717
 */
718

    
719

    
720
static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
721
{
722
    int   band, result=0, numSubbands, numBands;
723

    
724
    if (codingMode == JOINT_STEREO && channelNum == 1) {
725
        if (get_bits(gb,2) != 3) {
726
            av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
727
            return -1;
728
        }
729
    } else {
730
        if (get_bits(gb,6) != 0x28) {
731
            av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
732
            return -1;
733
        }
734
    }
735

    
736
    /* number of coded QMF bands */
737
    pSnd->bandsCoded = get_bits(gb,2);
738

    
739
    result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
740
    if (result) return result;
741

    
742
    result = decodeTonalComponents (gb, &pSnd->numComponents, pSnd->components, pSnd->bandsCoded);
743
    if (result) return result;
744

    
745
    numSubbands = decodeSpectrum (gb, pSnd->spectrum);
746

    
747
    /* Merge the decoded spectrum and tonal components. */
748
    addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
749

    
750

    
751
    /* Convert number of subbands into number of MLT/QMF bands */
752
    numBands = (subbandTab[numSubbands] - 1) >> 8;
753

    
754

    
755
    /* Reconstruct time domain samples. */
756
    for (band=0; band<4; band++) {
757
        /* Perform the IMDCT step without overlapping. */
758
        if (band <= numBands) {
759
            IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1,q->mdct_tmp);
760
        } else
761
            memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
762

    
763
        /* gain compensation and overlapping */
764
        gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
765
                                    &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
766
                                    &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
767
    }
768

    
769
    /* Swap the gain control buffers for the next frame. */
770
    pSnd->gcBlkSwitch ^= 1;
771

    
772
    return 0;
773
}
774

    
775
/**
776
 * Frame handling
777
 *
778
 * @param q             Atrac3 private context
779
 * @param databuf       the input data
780
 */
781

    
782
static int decodeFrame(ATRAC3Context *q, uint8_t* databuf)
783
{
784
    int   result, i;
785
    float   *p1, *p2, *p3, *p4;
786
    uint8_t    *ptr1, *ptr2;
787

    
788
    if (q->codingMode == JOINT_STEREO) {
789

    
790
        /* channel coupling mode */
791
        /* decode Sound Unit 1 */
792
        init_get_bits(&q->gb,databuf,q->bits_per_frame);
793

    
794
        result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
795
        if (result != 0)
796
            return (result);
797

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

    
806
        /* Skip the sync codes (0xF8). */
807
        ptr1 = databuf;
808
        for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
809
            if (i >= q->bytes_per_frame)
810
                return -1;
811
        }
812

    
813

    
814
        /* set the bitstream reader at the start of the second Sound Unit*/
815
        init_get_bits(&q->gb,ptr1,q->bits_per_frame);
816

    
817
        /* Fill the Weighting coeffs delay buffer */
818
        memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
819
        q->weighting_delay[4] = get_bits(&q->gb,1);
820
        q->weighting_delay[5] = get_bits(&q->gb,3);
821

    
822
        for (i = 0; i < 4; i++) {
823
            q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
824
            q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
825
            q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
826
        }
827

    
828
        /* Decode Sound Unit 2. */
829
        result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
830
        if (result != 0)
831
            return (result);
832

    
833
        /* Reconstruct the channel coefficients. */
834
        reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
835

    
836
        channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
837

    
838
    } else {
839
        /* normal stereo mode or mono */
840
        /* Decode the channel sound units. */
841
        for (i=0 ; i<q->channels ; i++) {
842

    
843
            /* Set the bitstream reader at the start of a channel sound unit. */
844
            init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
845

    
846
            result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
847
            if (result != 0)
848
                return (result);
849
        }
850
    }
851

    
852
    /* Apply the iQMF synthesis filter. */
853
    p1= q->outSamples;
854
    for (i=0 ; i<q->channels ; i++) {
855
        p2= p1+256;
856
        p3= p2+256;
857
        p4= p3+256;
858
        iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
859
        iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
860
        iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
861
        p1 +=1024;
862
    }
863

    
864
    return 0;
865
}
866

    
867

    
868
/**
869
 * Atrac frame decoding
870
 *
871
 * @param avctx     pointer to the AVCodecContext
872
 */
873

    
874
static int atrac3_decode_frame(AVCodecContext *avctx,
875
            void *data, int *data_size,
876
            uint8_t *buf, int buf_size) {
877
    ATRAC3Context *q = avctx->priv_data;
878
    int result = 0, i;
879
    uint8_t* databuf;
880
    int16_t* samples = data;
881

    
882
    if (buf_size < avctx->block_align)
883
        return buf_size;
884

    
885
    /* Check if we need to descramble and what buffer to pass on. */
886
    if (q->scrambled_stream) {
887
        decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
888
        databuf = q->decoded_bytes_buffer;
889
    } else {
890
        databuf = buf;
891
    }
892

    
893
    result = decodeFrame(q, databuf);
894

    
895
    if (result != 0) {
896
        av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
897
        return -1;
898
    }
899

    
900
    if (q->channels == 1) {
901
        /* mono */
902
        for (i = 0; i<1024; i++)
903
            samples[i] = av_clip(round(q->outSamples[i]), -32768, 32767);
904
        *data_size = 1024 * sizeof(int16_t);
905
    } else {
906
        /* stereo */
907
        for (i = 0; i < 1024; i++) {
908
            samples[i*2] = av_clip(round(q->outSamples[i]), -32768, 32767);
909
            samples[i*2+1] = av_clip(round(q->outSamples[1024+i]), -32768, 32767);
910
        }
911
        *data_size = 2048 * sizeof(int16_t);
912
    }
913

    
914
    return avctx->block_align;
915
}
916

    
917

    
918
/**
919
 * Atrac3 initialization
920
 *
921
 * @param avctx     pointer to the AVCodecContext
922
 */
923

    
924
static int atrac3_decode_init(AVCodecContext *avctx)
925
{
926
    int i;
927
    uint8_t *edata_ptr = avctx->extradata;
928
    ATRAC3Context *q = avctx->priv_data;
929

    
930
    /* Take data from the AVCodecContext (RM container). */
931
    q->sample_rate = avctx->sample_rate;
932
    q->channels = avctx->channels;
933
    q->bit_rate = avctx->bit_rate;
934
    q->bits_per_frame = avctx->block_align * 8;
935
    q->bytes_per_frame = avctx->block_align;
936

    
937
    /* Take care of the codec-specific extradata. */
938
    if (avctx->extradata_size == 14) {
939
        /* Parse the extradata, WAV format */
940
        av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown value always 1
941
        q->samples_per_channel = bytestream_get_le32(&edata_ptr);
942
        q->codingMode = bytestream_get_le16(&edata_ptr);
943
        av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
944
        q->frame_factor = bytestream_get_le16(&edata_ptr);  //Unknown always 1
945
        av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown always 0
946

    
947
        /* setup */
948
        q->samples_per_frame = 1024 * q->channels;
949
        q->atrac3version = 4;
950
        q->delay = 0x88E;
951
        if (q->codingMode)
952
            q->codingMode = JOINT_STEREO;
953
        else
954
            q->codingMode = STEREO;
955

    
956
        q->scrambled_stream = 0;
957

    
958
        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)) {
959
        } else {
960
            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);
961
            return -1;
962
        }
963

    
964
    } else if (avctx->extradata_size == 10) {
965
        /* Parse the extradata, RM format. */
966
        q->atrac3version = bytestream_get_be32(&edata_ptr);
967
        q->samples_per_frame = bytestream_get_be16(&edata_ptr);
968
        q->delay = bytestream_get_be16(&edata_ptr);
969
        q->codingMode = bytestream_get_be16(&edata_ptr);
970

    
971
        q->samples_per_channel = q->samples_per_frame / q->channels;
972
        q->scrambled_stream = 1;
973

    
974
    } else {
975
        av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
976
    }
977
    /* Check the extradata. */
978

    
979
    if (q->atrac3version != 4) {
980
        av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
981
        return -1;
982
    }
983

    
984
    if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
985
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
986
        return -1;
987
    }
988

    
989
    if (q->delay != 0x88E) {
990
        av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
991
        return -1;
992
    }
993

    
994
    if (q->codingMode == STEREO) {
995
        av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
996
    } else if (q->codingMode == JOINT_STEREO) {
997
        av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
998
    } else {
999
        av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
1000
        return -1;
1001
    }
1002

    
1003
    if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
1004
        av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
1005
        return -1;
1006
    }
1007

    
1008

    
1009
    if(avctx->block_align >= UINT_MAX/2)
1010
        return -1;
1011

    
1012
    /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
1013
     * this is for the bitstream reader. */
1014
    if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)
1015
        return -1;
1016

    
1017

    
1018
    /* Initialize the VLC tables. */
1019
    for (i=0 ; i<7 ; i++) {
1020
        init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
1021
            huff_bits[i], 1, 1,
1022
            huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);
1023
    }
1024

    
1025
    init_atrac3_transforms(q);
1026

    
1027
    /* Generate the scale factors. */
1028
    for (i=0 ; i<64 ; i++)
1029
        SFTable[i] = pow(2.0, (i - 15) / 3.0);
1030

    
1031
    /* Generate gain tables. */
1032
    for (i=0 ; i<16 ; i++)
1033
        gain_tab1[i] = powf (2.0, (4 - i));
1034

    
1035
    for (i=-15 ; i<16 ; i++)
1036
        gain_tab2[i+15] = powf (2.0, i * -0.125);
1037

    
1038
    /* init the joint-stereo decoding data */
1039
    q->weighting_delay[0] = 0;
1040
    q->weighting_delay[1] = 7;
1041
    q->weighting_delay[2] = 0;
1042
    q->weighting_delay[3] = 7;
1043
    q->weighting_delay[4] = 0;
1044
    q->weighting_delay[5] = 7;
1045

    
1046
    for (i=0; i<4; i++) {
1047
        q->matrix_coeff_index_prev[i] = 3;
1048
        q->matrix_coeff_index_now[i] = 3;
1049
        q->matrix_coeff_index_next[i] = 3;
1050
    }
1051

    
1052
    dsputil_init(&dsp, avctx);
1053

    
1054
    q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
1055

    
1056
    return 0;
1057
}
1058

    
1059

    
1060
AVCodec atrac3_decoder =
1061
{
1062
    .name = "atrac 3",
1063
    .type = CODEC_TYPE_AUDIO,
1064
    .id = CODEC_ID_ATRAC3,
1065
    .priv_data_size = sizeof(ATRAC3Context),
1066
    .init = atrac3_decode_init,
1067
    .close = atrac3_decode_close,
1068
    .decode = atrac3_decode_frame,
1069
};