GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/atrac3.c Lines: 291 451 64.5 %
Date: 2019-11-22 03:34:36 Branches: 132 234 56.4 %

Line Branch Exec Source
1
/*
2
 * ATRAC3 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.
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
25
 * ATRAC3 compatible decoder.
26
 * This decoder handles Sony's ATRAC3 data.
27
 *
28
 * Container formats used to store ATRAC3 data:
29
 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
30
 *
31
 * 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 "libavutil/attributes.h"
40
#include "libavutil/float_dsp.h"
41
#include "libavutil/libm.h"
42
#include "avcodec.h"
43
#include "bytestream.h"
44
#include "fft.h"
45
#include "get_bits.h"
46
#include "internal.h"
47
48
#include "atrac.h"
49
#include "atrac3data.h"
50
51
#define MIN_CHANNELS    1
52
#define MAX_CHANNELS    8
53
#define MAX_JS_PAIRS    8 / 2
54
55
#define JOINT_STEREO    0x12
56
#define SINGLE          0x2
57
58
#define SAMPLES_PER_FRAME 1024
59
#define MDCT_SIZE          512
60
61
typedef struct GainBlock {
62
    AtracGainInfo g_block[4];
63
} GainBlock;
64
65
typedef struct TonalComponent {
66
    int pos;
67
    int num_coefs;
68
    float coef[8];
69
} TonalComponent;
70
71
typedef struct ChannelUnit {
72
    int            bands_coded;
73
    int            num_components;
74
    float          prev_frame[SAMPLES_PER_FRAME];
75
    int            gc_blk_switch;
76
    TonalComponent components[64];
77
    GainBlock      gain_block[2];
78
79
    DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
80
    DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
81
82
    float          delay_buf1[46]; ///<qmf delay buffers
83
    float          delay_buf2[46];
84
    float          delay_buf3[46];
85
} ChannelUnit;
86
87
typedef struct ATRAC3Context {
88
    GetBitContext gb;
89
    //@{
90
    /** stream data */
91
    int coding_mode;
92
93
    ChannelUnit *units;
94
    //@}
95
    //@{
96
    /** joint-stereo related variables */
97
    int matrix_coeff_index_prev[MAX_JS_PAIRS][4];
98
    int matrix_coeff_index_now[MAX_JS_PAIRS][4];
99
    int matrix_coeff_index_next[MAX_JS_PAIRS][4];
100
    int weighting_delay[MAX_JS_PAIRS][6];
101
    //@}
102
    //@{
103
    /** data buffers */
104
    uint8_t *decoded_bytes_buffer;
105
    float temp_buf[1070];
106
    //@}
107
    //@{
108
    /** extradata */
109
    int scrambled_stream;
110
    //@}
111
112
    AtracGCContext    gainc_ctx;
113
    FFTContext        mdct_ctx;
114
    AVFloatDSPContext *fdsp;
115
} ATRAC3Context;
116
117
static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
118
static VLC_TYPE atrac3_vlc_table[4096][2];
119
static VLC   spectral_coeff_tab[7];
120
121
/**
122
 * Regular 512 points IMDCT without overlapping, with the exception of the
123
 * swapping of odd bands caused by the reverse spectra of the QMF.
124
 *
125
 * @param odd_band  1 if the band is an odd band
126
 */
127
2800
static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
128
{
129
    int i;
130
131
2800
    if (odd_band) {
132
        /**
133
         * Reverse the odd bands before IMDCT, this is an effect of the QMF
134
         * transform or it gives better compression to do it this way.
135
         * FIXME: It should be possible to handle this in imdct_calc
136
         * for that to happen a modification of the prerotation step of
137
         * all SIMD code and C code is needed.
138
         * Or fix the functions before so they generate a pre reversed spectrum.
139
         */
140
109392
        for (i = 0; i < 128; i++)
141
108544
            FFSWAP(float, input[i], input[255 - i]);
142
    }
143
144
2800
    q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
145
146
    /* Perform windowing on the output. */
147
2800
    q->fdsp->vector_fmul(output, output, mdct_window, MDCT_SIZE);
148
2800
}
149
150
/*
151
 * indata descrambling, only used for data coming from the rm container
152
 */
153
static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
154
{
155
    int i, off;
156
    uint32_t c;
157
    const uint32_t *buf;
158
    uint32_t *output = (uint32_t *)out;
159
160
    off = (intptr_t)input & 3;
161
    buf = (const uint32_t *)(input - off);
162
    if (off)
163
        c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
164
    else
165
        c = av_be2ne32(0x537F6103U);
166
    bytes += 3 + off;
167
    for (i = 0; i < bytes / 4; i++)
168
        output[i] = c ^ buf[i];
169
170
    if (off)
171
        avpriv_request_sample(NULL, "Offset of %d", off);
172
173
    return off;
174
}
175
176
4
static av_cold void init_imdct_window(void)
177
{
178
    int i, j;
179
180
    /* generate the mdct window, for details see
181
     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
182
516
    for (i = 0, j = 255; i < 128; i++, j--) {
183
512
        float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
184
512
        float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
185
512
        float w  = 0.5 * (wi * wi + wj * wj);
186
512
        mdct_window[i] = mdct_window[511 - i] = wi / w;
187
512
        mdct_window[j] = mdct_window[511 - j] = wj / w;
188
    }
189
4
}
190
191
7
static av_cold int atrac3_decode_close(AVCodecContext *avctx)
192
{
193
7
    ATRAC3Context *q = avctx->priv_data;
194
195
7
    av_freep(&q->units);
196
7
    av_freep(&q->decoded_bytes_buffer);
197
7
    av_freep(&q->fdsp);
198
199
7
    ff_mdct_end(&q->mdct_ctx);
200
201
7
    return 0;
202
}
203
204
/**
205
 * Mantissa decoding
206
 *
207
 * @param selector     which table the output values are coded with
208
 * @param coding_flag  constant length coding or variable length coding
209
 * @param mantissas    mantissa output table
210
 * @param num_codes    number of values to get
211
 */
212
28209
static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
213
                                       int coding_flag, int *mantissas,
214
                                       int num_codes)
215
{
216
    int i, code, huff_symb;
217
218
28209
    if (selector == 1)
219
15517
        num_codes /= 2;
220
221
28209
    if (coding_flag != 0) {
222
        /* constant length coding (CLC) */
223
        int num_bits = clc_length_tab[selector];
224
225
        if (selector > 1) {
226
            for (i = 0; i < num_codes; i++) {
227
                if (num_bits)
228
                    code = get_sbits(gb, num_bits);
229
                else
230
                    code = 0;
231
                mantissas[i] = code;
232
            }
233
        } else {
234
            for (i = 0; i < num_codes; i++) {
235
                if (num_bits)
236
                    code = get_bits(gb, num_bits); // num_bits is always 4 in this case
237
                else
238
                    code = 0;
239
                mantissas[i * 2    ] = mantissa_clc_tab[code >> 2];
240
                mantissas[i * 2 + 1] = mantissa_clc_tab[code &  3];
241
            }
242
        }
243
    } else {
244
        /* variable length coding (VLC) */
245
28209
        if (selector != 1) {
246
192516
            for (i = 0; i < num_codes; i++) {
247
179824
                huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
248
179824
                                     spectral_coeff_tab[selector-1].bits, 3);
249
179824
                huff_symb += 1;
250
179824
                code = huff_symb >> 1;
251
179824
                if (huff_symb & 1)
252
139733
                    code = -code;
253
179824
                mantissas[i] = code;
254
            }
255
        } else {
256
250405
            for (i = 0; i < num_codes; i++) {
257
234888
                huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
258
234888
                                     spectral_coeff_tab[selector - 1].bits, 3);
259
234888
                mantissas[i * 2    ] = mantissa_vlc_tab[huff_symb * 2    ];
260
234888
                mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
261
            }
262
        }
263
    }
264
28209
}
265
266
/**
267
 * Restore the quantized band spectrum coefficients
268
 *
269
 * @return subband count, fix for broken specification/files
270
 */
271
1108
static int decode_spectrum(GetBitContext *gb, float *output)
272
{
273
    int num_subbands, coding_mode, i, j, first, last, subband_size;
274
    int subband_vlc_index[32], sf_index[32];
275
    int mantissas[128];
276
    float scale_factor;
277
278
1108
    num_subbands = get_bits(gb, 5);  // number of coded subbands
279
1108
    coding_mode  = get_bits1(gb);    // coding Mode: 0 - VLC/ 1-CLC
280
281
    /* get the VLC selector table for the subbands, 0 means not coded */
282
29317
    for (i = 0; i <= num_subbands; i++)
283
28209
        subband_vlc_index[i] = get_bits(gb, 3);
284
285
    /* read the scale factor indexes from the stream */
286
29317
    for (i = 0; i <= num_subbands; i++) {
287
28209
        if (subband_vlc_index[i] != 0)
288
28209
            sf_index[i] = get_bits(gb, 6);
289
    }
290
291
29317
    for (i = 0; i <= num_subbands; i++) {
292
28209
        first = subband_tab[i    ];
293
28209
        last  = subband_tab[i + 1];
294
295
28209
        subband_size = last - first;
296
297
28209
        if (subband_vlc_index[i] != 0) {
298
            /* decode spectral coefficients for this subband */
299
            /* TODO: This can be done faster is several blocks share the
300
             * same VLC selector (subband_vlc_index) */
301
28209
            read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
302
                                       mantissas, subband_size);
303
304
            /* decode the scale factor for this subband */
305
28209
            scale_factor = ff_atrac_sf_table[sf_index[i]] *
306
28209
                           inv_max_quant[subband_vlc_index[i]];
307
308
            /* inverse quantize the coefficients */
309
677809
            for (j = 0; first < last; first++, j++)
310
649600
                output[first] = mantissas[j] * scale_factor;
311
        } else {
312
            /* this subband was not coded, so zero the entire subband */
313
            memset(output + first, 0, subband_size * sizeof(*output));
314
        }
315
    }
316
317
    /* clear the subbands that were not coded */
318
1108
    first = subband_tab[i];
319
1108
    memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
320
1108
    return num_subbands;
321
}
322
323
/**
324
 * Restore the quantized tonal components
325
 *
326
 * @param components tonal components
327
 * @param num_bands  number of coded bands
328
 */
329
1108
static int decode_tonal_components(GetBitContext *gb,
330
                                   TonalComponent *components, int num_bands)
331
{
332
    int i, b, c, m;
333
    int nb_components, coding_mode_selector, coding_mode;
334
    int band_flags[4], mantissa[8];
335
1108
    int component_count = 0;
336
337
1108
    nb_components = get_bits(gb, 5);
338
339
    /* no tonal components */
340
1108
    if (nb_components == 0)
341
1108
        return 0;
342
343
    coding_mode_selector = get_bits(gb, 2);
344
    if (coding_mode_selector == 2)
345
        return AVERROR_INVALIDDATA;
346
347
    coding_mode = coding_mode_selector & 1;
348
349
    for (i = 0; i < nb_components; i++) {
350
        int coded_values_per_component, quant_step_index;
351
352
        for (b = 0; b <= num_bands; b++)
353
            band_flags[b] = get_bits1(gb);
354
355
        coded_values_per_component = get_bits(gb, 3);
356
357
        quant_step_index = get_bits(gb, 3);
358
        if (quant_step_index <= 1)
359
            return AVERROR_INVALIDDATA;
360
361
        if (coding_mode_selector == 3)
362
            coding_mode = get_bits1(gb);
363
364
        for (b = 0; b < (num_bands + 1) * 4; b++) {
365
            int coded_components;
366
367
            if (band_flags[b >> 2] == 0)
368
                continue;
369
370
            coded_components = get_bits(gb, 3);
371
372
            for (c = 0; c < coded_components; c++) {
373
                TonalComponent *cmp = &components[component_count];
374
                int sf_index, coded_values, max_coded_values;
375
                float scale_factor;
376
377
                sf_index = get_bits(gb, 6);
378
                if (component_count >= 64)
379
                    return AVERROR_INVALIDDATA;
380
381
                cmp->pos = b * 64 + get_bits(gb, 6);
382
383
                max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
384
                coded_values     = coded_values_per_component + 1;
385
                coded_values     = FFMIN(max_coded_values, coded_values);
386
387
                scale_factor = ff_atrac_sf_table[sf_index] *
388
                               inv_max_quant[quant_step_index];
389
390
                read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
391
                                           mantissa, coded_values);
392
393
                cmp->num_coefs = coded_values;
394
395
                /* inverse quant */
396
                for (m = 0; m < coded_values; m++)
397
                    cmp->coef[m] = mantissa[m] * scale_factor;
398
399
                component_count++;
400
            }
401
        }
402
    }
403
404
    return component_count;
405
}
406
407
/**
408
 * Decode gain parameters for the coded bands
409
 *
410
 * @param block      the gainblock for the current band
411
 * @param num_bands  amount of coded bands
412
 */
413
1108
static int decode_gain_control(GetBitContext *gb, GainBlock *block,
414
                               int num_bands)
415
{
416
    int b, j;
417
    int *level, *loc;
418
419
1108
    AtracGainInfo *gain = block->g_block;
420
421
3912
    for (b = 0; b <= num_bands; b++) {
422
2804
        gain[b].num_points = get_bits(gb, 3);
423
2804
        level              = gain[b].lev_code;
424
2804
        loc                = gain[b].loc_code;
425
426
3678
        for (j = 0; j < gain[b].num_points; j++) {
427
874
            level[j] = get_bits(gb, 4);
428
874
            loc[j]   = get_bits(gb, 5);
429

874
            if (j && loc[j] <= loc[j - 1])
430
                return AVERROR_INVALIDDATA;
431
        }
432
    }
433
434
    /* Clear the unused blocks. */
435
2736
    for (; b < 4 ; b++)
436
1628
        gain[b].num_points = 0;
437
438
1108
    return 0;
439
}
440
441
/**
442
 * Combine the tonal band spectrum and regular band spectrum
443
 *
444
 * @param spectrum        output spectrum buffer
445
 * @param num_components  number of tonal components
446
 * @param components      tonal components for this band
447
 * @return                position of the last tonal coefficient
448
 */
449
1108
static int add_tonal_components(float *spectrum, int num_components,
450
                                TonalComponent *components)
451
{
452
1108
    int i, j, last_pos = -1;
453
    float *input, *output;
454
455
1108
    for (i = 0; i < num_components; i++) {
456
        last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
457
        input    = components[i].coef;
458
        output   = &spectrum[components[i].pos];
459
460
        for (j = 0; j < components[i].num_coefs; j++)
461
            output[j] += input[j];
462
    }
463
464
1108
    return last_pos;
465
}
466
467
#define INTERPOLATE(old, new, nsample) \
468
    ((old) + (nsample) * 0.125 * ((new) - (old)))
469
470
260
static void reverse_matrixing(float *su1, float *su2, int *prev_code,
471
                              int *curr_code)
472
{
473
    int i, nsample, band;
474
    float mc1_l, mc1_r, mc2_l, mc2_r;
475
476
1300
    for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
477
1040
        int s1 = prev_code[i];
478
1040
        int s2 = curr_code[i];
479
1040
        nsample = band;
480
481
1040
        if (s1 != s2) {
482
            /* Selector value changed, interpolation needed. */
483
38
            mc1_l = matrix_coeffs[s1 * 2    ];
484
38
            mc1_r = matrix_coeffs[s1 * 2 + 1];
485
38
            mc2_l = matrix_coeffs[s2 * 2    ];
486
38
            mc2_r = matrix_coeffs[s2 * 2 + 1];
487
488
            /* Interpolation is done over the first eight samples. */
489
342
            for (; nsample < band + 8; nsample++) {
490
304
                float c1 = su1[nsample];
491
304
                float c2 = su2[nsample];
492
304
                c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
493
304
                     c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
494
304
                su1[nsample] = c2;
495
304
                su2[nsample] = c1 * 2.0 - c2;
496
            }
497
        }
498
499
        /* Apply the matrix without interpolation. */
500

1040
        switch (s2) {
501
45
        case 0:     /* M/S decoding */
502
11413
            for (; nsample < band + 256; nsample++) {
503
11368
                float c1 = su1[nsample];
504
11368
                float c2 = su2[nsample];
505
11368
                su1[nsample] =  c2       * 2.0;
506
11368
                su2[nsample] = (c1 - c2) * 2.0;
507
            }
508
45
            break;
509
        case 1:
510
            for (; nsample < band + 256; nsample++) {
511
                float c1 = su1[nsample];
512
                float c2 = su2[nsample];
513
                su1[nsample] = (c1 + c2) *  2.0;
514
                su2[nsample] =  c2       * -2.0;
515
            }
516
            break;
517
995
        case 2:
518
        case 3:
519
255563
            for (; nsample < band + 256; nsample++) {
520
254568
                float c1 = su1[nsample];
521
254568
                float c2 = su2[nsample];
522
254568
                su1[nsample] = c1 + c2;
523
254568
                su2[nsample] = c1 - c2;
524
            }
525
995
            break;
526
1040
        default:
527
            av_assert1(0);
528
        }
529
    }
530
260
}
531
532
492
static void get_channel_weights(int index, int flag, float ch[2])
533
{
534
492
    if (index == 7) {
535
53
        ch[0] = 1.0;
536
53
        ch[1] = 1.0;
537
    } else {
538
439
        ch[0] = (index & 7) / 7.0;
539
439
        ch[1] = sqrt(2 - ch[0] * ch[0]);
540
439
        if (flag)
541
115
            FFSWAP(float, ch[0], ch[1]);
542
    }
543
492
}
544
545
260
static void channel_weighting(float *su1, float *su2, int *p3)
546
{
547
    int band, nsample;
548
    /* w[x][y] y=0 is left y=1 is right */
549
    float w[2][2];
550
551

260
    if (p3[1] != 7 || p3[3] != 7) {
552
246
        get_channel_weights(p3[1], p3[0], w[0]);
553
246
        get_channel_weights(p3[3], p3[2], w[1]);
554
555
984
        for (band = 256; band < 4 * 256; band += 256) {
556
6642
            for (nsample = band; nsample < band + 8; nsample++) {
557
5904
                su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
558
5904
                su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
559
            }
560
183762
            for(; nsample < band + 256; nsample++) {
561
183024
                su1[nsample] *= w[1][0];
562
183024
                su2[nsample] *= w[1][1];
563
            }
564
        }
565
    }
566
260
}
567
568
/**
569
 * Decode a Sound Unit
570
 *
571
 * @param snd           the channel unit to be used
572
 * @param output        the decoded samples before IQMF in float representation
573
 * @param channel_num   channel number
574
 * @param coding_mode   the coding mode (JOINT_STEREO or single channels)
575
 */
576
1108
static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
577
                                     ChannelUnit *snd, float *output,
578
                                     int channel_num, int coding_mode)
579
{
580
    int band, ret, num_subbands, last_tonal, num_bands;
581
1108
    GainBlock *gain1 = &snd->gain_block[    snd->gc_blk_switch];
582
1108
    GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
583
584

1108
    if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) {
585
260
        if (get_bits(gb, 2) != 3) {
586
            av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
587
            return AVERROR_INVALIDDATA;
588
        }
589
    } else {
590
848
        if (get_bits(gb, 6) != 0x28) {
591
            av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
592
            return AVERROR_INVALIDDATA;
593
        }
594
    }
595
596
    /* number of coded QMF bands */
597
1108
    snd->bands_coded = get_bits(gb, 2);
598
599
1108
    ret = decode_gain_control(gb, gain2, snd->bands_coded);
600
1108
    if (ret)
601
        return ret;
602
603
1108
    snd->num_components = decode_tonal_components(gb, snd->components,
604
                                                  snd->bands_coded);
605
1108
    if (snd->num_components < 0)
606
        return snd->num_components;
607
608
1108
    num_subbands = decode_spectrum(gb, snd->spectrum);
609
610
    /* Merge the decoded spectrum and tonal components. */
611
1108
    last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
612
1108
                                      snd->components);
613
614
615
    /* calculate number of used MLT/QMF bands according to the amount of coded
616
       spectral lines */
617
1108
    num_bands = (subband_tab[num_subbands] - 1) >> 8;
618
1108
    if (last_tonal >= 0)
619
        num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
620
621
622
    /* Reconstruct time domain samples. */
623
5540
    for (band = 0; band < 4; band++) {
624
        /* Perform the IMDCT step without overlapping. */
625
4432
        if (band <= num_bands)
626
2800
            imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
627
        else
628
1632
            memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
629
630
        /* gain compensation and overlapping */
631
4432
        ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
632
4432
                                   &snd->prev_frame[band * 256],
633
                                   &gain1->g_block[band], &gain2->g_block[band],
634
4432
                                   256, &output[band * 256]);
635
    }
636
637
    /* Swap the gain control buffers for the next frame. */
638
1108
    snd->gc_blk_switch ^= 1;
639
640
1108
    return 0;
641
}
642
643
554
static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
644
                        float **out_samples)
645
{
646
554
    ATRAC3Context *q = avctx->priv_data;
647
    int ret, i, ch;
648
    uint8_t *ptr1;
649
650
554
    if (q->coding_mode == JOINT_STEREO) {
651
        /* channel coupling mode */
652
653
        /* Decode sound unit pairs (channels are expected to be even).
654
         * Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */
655
        const uint8_t *js_databuf;
656
        int js_pair, js_block_align;
657
658
260
        js_block_align = (avctx->block_align / avctx->channels) * 2; /* block pair */
659
660
520
        for (ch = 0; ch < avctx->channels; ch = ch + 2) {
661
260
            js_pair = ch/2;
662
260
            js_databuf = databuf + js_pair * js_block_align; /* align to current pair */
663
664
            /* Set the bitstream reader at the start of first channel sound unit. */
665
260
            init_get_bits(&q->gb,
666
                          js_databuf, js_block_align * 8);
667
668
            /* decode Sound Unit 1 */
669
260
            ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch],
670
260
                                            out_samples[ch], ch, JOINT_STEREO);
671
260
            if (ret != 0)
672
                return ret;
673
674
            /* Framedata of the su2 in the joint-stereo mode is encoded in
675
             * reverse byte order so we need to swap it first. */
676
260
            if (js_databuf == q->decoded_bytes_buffer) {
677
                uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1;
678
                ptr1          = q->decoded_bytes_buffer;
679
                for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--)
680
                    FFSWAP(uint8_t, *ptr1, *ptr2);
681
            } else {
682
260
                const uint8_t *ptr2 = js_databuf + js_block_align - 1;
683
50180
                for (i = 0; i < js_block_align; i++)
684
49920
                    q->decoded_bytes_buffer[i] = *ptr2--;
685
            }
686
687
            /* Skip the sync codes (0xF8). */
688
260
            ptr1 = q->decoded_bytes_buffer;
689
260
            for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
690
                if (i >= js_block_align)
691
                    return AVERROR_INVALIDDATA;
692
            }
693
694
695
            /* set the bitstream reader at the start of the second Sound Unit */
696
260
            ret = init_get_bits8(&q->gb,
697
260
                           ptr1, q->decoded_bytes_buffer + js_block_align - ptr1);
698
260
            if (ret < 0)
699
                return ret;
700
701
            /* Fill the Weighting coeffs delay buffer */
702
260
            memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2],
703
                    4 * sizeof(*q->weighting_delay[js_pair]));
704
260
            q->weighting_delay[js_pair][4] = get_bits1(&q->gb);
705
260
            q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3);
706
707
1300
            for (i = 0; i < 4; i++) {
708
1040
                q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i];
709
1040
                q->matrix_coeff_index_now[js_pair][i]  = q->matrix_coeff_index_next[js_pair][i];
710
1040
                q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2);
711
            }
712
713
            /* Decode Sound Unit 2. */
714
260
            ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1],
715
260
                                            out_samples[ch+1], ch+1, JOINT_STEREO);
716
260
            if (ret != 0)
717
                return ret;
718
719
            /* Reconstruct the channel coefficients. */
720
260
            reverse_matrixing(out_samples[ch], out_samples[ch+1],
721
260
                              q->matrix_coeff_index_prev[js_pair],
722
260
                              q->matrix_coeff_index_now[js_pair]);
723
724
260
            channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]);
725
        }
726
    } else {
727
        /* single channels */
728
        /* Decode the channel sound units. */
729
882
        for (i = 0; i < avctx->channels; i++) {
730
            /* Set the bitstream reader at the start of a channel sound unit. */
731
588
            init_get_bits(&q->gb,
732
588
                          databuf + i * avctx->block_align / avctx->channels,
733
588
                          avctx->block_align * 8 / avctx->channels);
734
735
588
            ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
736
588
                                            out_samples[i], i, q->coding_mode);
737
588
            if (ret != 0)
738
                return ret;
739
        }
740
    }
741
742
    /* Apply the iQMF synthesis filter. */
743
1662
    for (i = 0; i < avctx->channels; i++) {
744
1108
        float *p1 = out_samples[i];
745
1108
        float *p2 = p1 + 256;
746
1108
        float *p3 = p2 + 256;
747
1108
        float *p4 = p3 + 256;
748
1108
        ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
749
1108
        ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
750
1108
        ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
751
    }
752
753
554
    return 0;
754
}
755
756
static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
757
                           int size, float **out_samples)
758
{
759
    ATRAC3Context *q = avctx->priv_data;
760
    int ret, i;
761
762
    /* Set the bitstream reader at the start of a channel sound unit. */
763
    init_get_bits(&q->gb, databuf, size * 8);
764
    /* single channels */
765
    /* Decode the channel sound units. */
766
    for (i = 0; i < avctx->channels; i++) {
767
        ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
768
                                        out_samples[i], i, q->coding_mode);
769
        if (ret != 0)
770
            return ret;
771
        while (i < avctx->channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) {
772
            skip_bits(&q->gb, 1);
773
        }
774
    }
775
776
    /* Apply the iQMF synthesis filter. */
777
    for (i = 0; i < avctx->channels; i++) {
778
        float *p1 = out_samples[i];
779
        float *p2 = p1 + 256;
780
        float *p3 = p2 + 256;
781
        float *p4 = p3 + 256;
782
        ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
783
        ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
784
        ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
785
    }
786
787
    return 0;
788
}
789
790
557
static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
791
                               int *got_frame_ptr, AVPacket *avpkt)
792
{
793
557
    AVFrame *frame     = data;
794
557
    const uint8_t *buf = avpkt->data;
795
557
    int buf_size = avpkt->size;
796
557
    ATRAC3Context *q = avctx->priv_data;
797
    int ret;
798
    const uint8_t *databuf;
799
800
557
    if (buf_size < avctx->block_align) {
801
3
        av_log(avctx, AV_LOG_ERROR,
802
               "Frame too small (%d bytes). Truncated file?\n", buf_size);
803
3
        return AVERROR_INVALIDDATA;
804
    }
805
806
    /* get output buffer */
807
554
    frame->nb_samples = SAMPLES_PER_FRAME;
808
554
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
809
        return ret;
810
811
    /* Check if we need to descramble and what buffer to pass on. */
812
554
    if (q->scrambled_stream) {
813
        decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
814
        databuf = q->decoded_bytes_buffer;
815
    } else {
816
554
        databuf = buf;
817
    }
818
819
554
    ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
820
554
    if (ret) {
821
        av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
822
        return ret;
823
    }
824
825
554
    *got_frame_ptr = 1;
826
827
554
    return avctx->block_align;
828
}
829
830
static int atrac3al_decode_frame(AVCodecContext *avctx, void *data,
831
                                 int *got_frame_ptr, AVPacket *avpkt)
832
{
833
    AVFrame *frame = data;
834
    int ret;
835
836
    frame->nb_samples = SAMPLES_PER_FRAME;
837
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
838
        return ret;
839
840
    ret = al_decode_frame(avctx, avpkt->data, avpkt->size,
841
                          (float **)frame->extended_data);
842
    if (ret) {
843
        av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
844
        return ret;
845
    }
846
847
    *got_frame_ptr = 1;
848
849
    return avpkt->size;
850
}
851
852
4
static av_cold void atrac3_init_static_data(void)
853
{
854
    int i;
855
856
4
    init_imdct_window();
857
4
    ff_atrac_generate_tables();
858
859
    /* Initialize the VLC tables. */
860
32
    for (i = 0; i < 7; i++) {
861
28
        spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
862
28
        spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
863
28
                                                atrac3_vlc_offs[i    ];
864
28
        init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
865
                 huff_bits[i],  1, 1,
866
                 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
867
    }
868
4
}
869
870
7
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
871
{
872
    static int static_init_done;
873
    int i, js_pair, ret;
874
    int version, delay, samples_per_frame, frame_factor;
875
7
    const uint8_t *edata_ptr = avctx->extradata;
876
7
    ATRAC3Context *q = avctx->priv_data;
877
878

7
    if (avctx->channels < MIN_CHANNELS || avctx->channels > MAX_CHANNELS) {
879
        av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
880
        return AVERROR(EINVAL);
881
    }
882
883
7
    if (!static_init_done)
884
4
        atrac3_init_static_data();
885
7
    static_init_done = 1;
886
887
    /* Take care of the codec-specific extradata. */
888
7
    if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) {
889
        version           = 4;
890
        samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
891
        delay             = 0x88E;
892
        q->coding_mode    = SINGLE;
893
7
    } else if (avctx->extradata_size == 14) {
894
        /* Parse the extradata, WAV format */
895
7
        av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
896
               bytestream_get_le16(&edata_ptr));  // Unknown value always 1
897
7
        edata_ptr += 4;                             // samples per channel
898
7
        q->coding_mode = bytestream_get_le16(&edata_ptr);
899
7
        av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
900
               bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
901
7
        frame_factor = bytestream_get_le16(&edata_ptr);  // Unknown always 1
902
7
        av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
903
               bytestream_get_le16(&edata_ptr));  // Unknown always 0
904
905
        /* setup */
906
7
        samples_per_frame    = SAMPLES_PER_FRAME * avctx->channels;
907
7
        version              = 4;
908
7
        delay                = 0x88E;
909
7
        q->coding_mode       = q->coding_mode ? JOINT_STEREO : SINGLE;
910
7
        q->scrambled_stream  = 0;
911
912
7
        if (avctx->block_align !=  96 * avctx->channels * frame_factor &&
913
5
            avctx->block_align != 152 * avctx->channels * frame_factor &&
914
3
            avctx->block_align != 192 * avctx->channels * frame_factor) {
915
            av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
916
                   "configuration %d/%d/%d\n", avctx->block_align,
917
                   avctx->channels, frame_factor);
918
            return AVERROR_INVALIDDATA;
919
        }
920
    } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
921
        /* Parse the extradata, RM format. */
922
        version                = bytestream_get_be32(&edata_ptr);
923
        samples_per_frame      = bytestream_get_be16(&edata_ptr);
924
        delay                  = bytestream_get_be16(&edata_ptr);
925
        q->coding_mode         = bytestream_get_be16(&edata_ptr);
926
        q->scrambled_stream    = 1;
927
928
    } else {
929
        av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
930
               avctx->extradata_size);
931
        return AVERROR(EINVAL);
932
    }
933
934
    /* Check the extradata */
935
936
7
    if (version != 4) {
937
        av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
938
        return AVERROR_INVALIDDATA;
939
    }
940
941
7
    if (samples_per_frame != SAMPLES_PER_FRAME * avctx->channels) {
942
        av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
943
               samples_per_frame);
944
        return AVERROR_INVALIDDATA;
945
    }
946
947
7
    if (delay != 0x88E) {
948
        av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
949
               delay);
950
        return AVERROR_INVALIDDATA;
951
    }
952
953
7
    if (q->coding_mode == SINGLE)
954
5
        av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
955
2
    else if (q->coding_mode == JOINT_STEREO) {
956
2
        if (avctx->channels % 2 == 1) { /* Joint stereo channels must be even */
957
            av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
958
            return AVERROR_INVALIDDATA;
959
        }
960
2
        av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
961
    } else {
962
        av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
963
               q->coding_mode);
964
        return AVERROR_INVALIDDATA;
965
    }
966
967

7
    if (avctx->block_align > 1024 || avctx->block_align <= 0)
968
        return AVERROR(EINVAL);
969
970
7
    q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
971
                                         AV_INPUT_BUFFER_PADDING_SIZE);
972
7
    if (!q->decoded_bytes_buffer)
973
        return AVERROR(ENOMEM);
974
975
7
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
976
977
    /* initialize the MDCT transform */
978
7
    if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
979
        av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
980
        av_freep(&q->decoded_bytes_buffer);
981
        return ret;
982
    }
983
984
    /* init the joint-stereo decoding data */
985
35
    for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) {
986
28
        q->weighting_delay[js_pair][0] = 0;
987
28
        q->weighting_delay[js_pair][1] = 7;
988
28
        q->weighting_delay[js_pair][2] = 0;
989
28
        q->weighting_delay[js_pair][3] = 7;
990
28
        q->weighting_delay[js_pair][4] = 0;
991
28
        q->weighting_delay[js_pair][5] = 7;
992
993
140
        for (i = 0; i < 4; i++) {
994
112
            q->matrix_coeff_index_prev[js_pair][i] = 3;
995
112
            q->matrix_coeff_index_now[js_pair][i]  = 3;
996
112
            q->matrix_coeff_index_next[js_pair][i] = 3;
997
        }
998
    }
999
1000
7
    ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
1001
7
    q->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1002
1003
7
    q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
1004

7
    if (!q->units || !q->fdsp) {
1005
        atrac3_decode_close(avctx);
1006
        return AVERROR(ENOMEM);
1007
    }
1008
1009
7
    return 0;
1010
}
1011
1012
AVCodec ff_atrac3_decoder = {
1013
    .name             = "atrac3",
1014
    .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
1015
    .type             = AVMEDIA_TYPE_AUDIO,
1016
    .id               = AV_CODEC_ID_ATRAC3,
1017
    .priv_data_size   = sizeof(ATRAC3Context),
1018
    .init             = atrac3_decode_init,
1019
    .close            = atrac3_decode_close,
1020
    .decode           = atrac3_decode_frame,
1021
    .capabilities     = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1022
    .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1023
                                                        AV_SAMPLE_FMT_NONE },
1024
};
1025
1026
AVCodec ff_atrac3al_decoder = {
1027
    .name             = "atrac3al",
1028
    .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
1029
    .type             = AVMEDIA_TYPE_AUDIO,
1030
    .id               = AV_CODEC_ID_ATRAC3AL,
1031
    .priv_data_size   = sizeof(ATRAC3Context),
1032
    .init             = atrac3_decode_init,
1033
    .close            = atrac3_decode_close,
1034
    .decode           = atrac3al_decode_frame,
1035
    .capabilities     = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1036
    .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1037
                                                        AV_SAMPLE_FMT_NONE },
1038
};