FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavcodec/atrac3.c
Date: 2026-01-16 07:34:38
Exec Total Coverage
Lines: 290 451 64.3%
Functions: 16 19 84.2%
Branches: 128 230 55.7%
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
38 #include "libavutil/attributes.h"
39 #include "libavutil/float_dsp.h"
40 #include "libavutil/libm.h"
41 #include "libavutil/mem.h"
42 #include "libavutil/mem_internal.h"
43 #include "libavutil/thread.h"
44 #include "libavutil/tx.h"
45
46 #include "avcodec.h"
47 #include "bytestream.h"
48 #include "codec_internal.h"
49 #include "decode.h"
50 #include "get_bits.h"
51
52 #include "atrac.h"
53 #include "atrac3data.h"
54
55 #define MIN_CHANNELS 1
56 #define MAX_CHANNELS 8
57 #define MAX_JS_PAIRS 8 / 2
58
59 #define JOINT_STEREO 0x12
60 #define SINGLE 0x2
61
62 #define SAMPLES_PER_FRAME 1024
63 #define MDCT_SIZE 512
64
65 #define ATRAC3_VLC_BITS 8
66
67 typedef struct GainBlock {
68 AtracGainInfo g_block[4];
69 } GainBlock;
70
71 typedef struct TonalComponent {
72 int pos;
73 int num_coefs;
74 float coef[8];
75 } TonalComponent;
76
77 typedef struct ChannelUnit {
78 int bands_coded;
79 int num_components;
80 float prev_frame[SAMPLES_PER_FRAME];
81 int gc_blk_switch;
82 TonalComponent components[64];
83 GainBlock gain_block[2];
84
85 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
86 DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
87
88 float delay_buf1[46]; ///<qmf delay buffers
89 float delay_buf2[46];
90 float delay_buf3[46];
91 } ChannelUnit;
92
93 typedef struct ATRAC3Context {
94 GetBitContext gb;
95 //@{
96 /** stream data */
97 int coding_mode;
98
99 ChannelUnit *units;
100 //@}
101 //@{
102 /** joint-stereo related variables */
103 int matrix_coeff_index_prev[MAX_JS_PAIRS][4];
104 int matrix_coeff_index_now[MAX_JS_PAIRS][4];
105 int matrix_coeff_index_next[MAX_JS_PAIRS][4];
106 int weighting_delay[MAX_JS_PAIRS][6];
107 //@}
108 //@{
109 /** data buffers */
110 uint8_t *decoded_bytes_buffer;
111 float temp_buf[1070];
112 //@}
113 //@{
114 /** extradata */
115 int scrambled_stream;
116 //@}
117
118 AtracGCContext gainc_ctx;
119 AVTXContext *mdct_ctx;
120 av_tx_fn mdct_fn;
121 void (*vector_fmul)(float *dst, const float *src0, const float *src1,
122 int len);
123 } ATRAC3Context;
124
125 static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
126 static VLCElem atrac3_vlc_table[7 * 1 << ATRAC3_VLC_BITS];
127 static VLC spectral_coeff_tab[7];
128
129 /**
130 * Regular 512 points IMDCT without overlapping, with the exception of the
131 * swapping of odd bands caused by the reverse spectra of the QMF.
132 *
133 * @param odd_band 1 if the band is an odd band
134 */
135 2804 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
136 {
137 int i;
138
139
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 2804 if (odd_band) {
140 /**
141 * Reverse the odd bands before IMDCT, this is an effect of the QMF
142 * transform or it gives better compression to do it this way.
143 * FIXME: It should be possible to handle this in imdct_calc
144 * for that to happen a modification of the prerotation step of
145 * all SIMD code and C code is needed.
146 * Or fix the functions before so they generate a pre reversed spectrum.
147 */
148
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 109392 for (i = 0; i < 128; i++)
149 108544 FFSWAP(float, input[i], input[255 - i]);
150 }
151
152 2804 q->mdct_fn(q->mdct_ctx, output, input, sizeof(float));
153
154 /* Perform windowing on the output. */
155 2804 q->vector_fmul(output, output, mdct_window, MDCT_SIZE);
156 2804 }
157
158 /*
159 * indata descrambling, only used for data coming from the rm container
160 */
161 ✗ static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
162 {
163 int i, off;
164 uint32_t c;
165 const uint32_t *buf;
166 ✗ uint32_t *output = (uint32_t *)out;
167
168 ✗ off = (intptr_t)input & 3;
169 ✗ buf = (const uint32_t *)(input - off);
170 ✗ if (off)
171 ✗ c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
172 else
173 ✗ c = av_be2ne32(0x537F6103U);
174 ✗ bytes += 3 + off;
175 ✗ for (i = 0; i < bytes / 4; i++)
176 ✗ output[i] = c ^ buf[i];
177
178 ✗ if (off)
179 ✗ avpriv_request_sample(NULL, "Offset of %d", off);
180
181 ✗ return off;
182 }
183
184 6 static av_cold void init_imdct_window(void)
185 {
186 int i, j;
187
188 /* generate the mdct window, for details see
189 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
190
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 774 for (i = 0, j = 255; i < 128; i++, j--) {
191 768 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
192 768 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
193 768 float w = 0.5 * (wi * wi + wj * wj);
194 768 mdct_window[i] = mdct_window[511 - i] = wi / w;
195 768 mdct_window[j] = mdct_window[511 - j] = wj / w;
196 }
197 6 }
198
199 9 static av_cold int atrac3_decode_close(AVCodecContext *avctx)
200 {
201 9 ATRAC3Context *q = avctx->priv_data;
202
203 9 av_freep(&q->units);
204 9 av_freep(&q->decoded_bytes_buffer);
205
206 9 av_tx_uninit(&q->mdct_ctx);
207
208 9 return 0;
209 }
210
211 /**
212 * Mantissa decoding
213 *
214 * @param selector which table the output values are coded with
215 * @param coding_flag constant length coding or variable length coding
216 * @param mantissas mantissa output table
217 * @param num_codes number of values to get
218 */
219 28209 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
220 int coding_flag, int *mantissas,
221 int num_codes)
222 {
223 int i, code, huff_symb;
224
225
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 28209 if (selector == 1)
226 15517 num_codes /= 2;
227
228
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 28209 if (coding_flag != 0) {
229 /* constant length coding (CLC) */
230 ✗ int num_bits = clc_length_tab[selector];
231
232 ✗ if (selector > 1) {
233 ✗ for (i = 0; i < num_codes; i++) {
234 ✗ if (num_bits)
235 ✗ code = get_sbits(gb, num_bits);
236 else
237 ✗ code = 0;
238 ✗ mantissas[i] = code;
239 }
240 } else {
241 ✗ for (i = 0; i < num_codes; i++) {
242 ✗ if (num_bits)
243 ✗ code = get_bits(gb, num_bits); // num_bits is always 4 in this case
244 else
245 ✗ code = 0;
246 ✗ mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
247 ✗ mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
248 }
249 }
250 } else {
251 /* variable length coding (VLC) */
252
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 28209 if (selector != 1) {
253
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 192516 for (i = 0; i < num_codes; i++) {
254 179824 mantissas[i] = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
255 ATRAC3_VLC_BITS, 1);
256 }
257 } else {
258
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 250405 for (i = 0; i < num_codes; i++) {
259 234888 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
260 ATRAC3_VLC_BITS, 1);
261 234888 mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
262 234888 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
263 }
264 }
265 }
266 28209 }
267
268 /**
269 * Restore the quantized band spectrum coefficients
270 *
271 * @return subband count, fix for broken specification/files
272 */
273 1108 static int decode_spectrum(GetBitContext *gb, float *output)
274 {
275 int num_subbands, coding_mode, i, j, first, last, subband_size;
276 int subband_vlc_index[32], sf_index[32];
277 int mantissas[128];
278 float scale_factor;
279
280 1108 num_subbands = get_bits(gb, 5); // number of coded subbands
281 1108 coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
282
283 /* get the VLC selector table for the subbands, 0 means not coded */
284
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 29317 for (i = 0; i <= num_subbands; i++)
285 28209 subband_vlc_index[i] = get_bits(gb, 3);
286
287 /* read the scale factor indexes from the stream */
288
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 29317 for (i = 0; i <= num_subbands; i++) {
289
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 28209 if (subband_vlc_index[i] != 0)
290 28209 sf_index[i] = get_bits(gb, 6);
291 }
292
293
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 29317 for (i = 0; i <= num_subbands; i++) {
294 28209 first = subband_tab[i ];
295 28209 last = subband_tab[i + 1];
296
297 28209 subband_size = last - first;
298
299
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 28209 if (subband_vlc_index[i] != 0) {
300 /* decode spectral coefficients for this subband */
301 /* TODO: This can be done faster is several blocks share the
302 * same VLC selector (subband_vlc_index) */
303 28209 read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
304 mantissas, subband_size);
305
306 /* decode the scale factor for this subband */
307 28209 scale_factor = ff_atrac_sf_table[sf_index[i]] *
308 28209 inv_max_quant[subband_vlc_index[i]];
309
310 /* inverse quantize the coefficients */
311
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 677809 for (j = 0; first < last; first++, j++)
312 649600 output[first] = mantissas[j] * scale_factor;
313 } else {
314 /* this subband was not coded, so zero the entire subband */
315 ✗ memset(output + first, 0, subband_size * sizeof(*output));
316 }
317 }
318
319 /* clear the subbands that were not coded */
320 1108 first = subband_tab[i];
321 1108 memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
322 1108 return num_subbands;
323 }
324
325 /**
326 * Restore the quantized tonal components
327 *
328 * @param components tonal components
329 * @param num_bands number of coded bands
330 */
331 1108 static int decode_tonal_components(GetBitContext *gb,
332 TonalComponent *components, int num_bands)
333 {
334 int i, b, c, m;
335 int nb_components, coding_mode_selector, coding_mode;
336 int band_flags[4], mantissa[8];
337 1108 int component_count = 0;
338
339 1108 nb_components = get_bits(gb, 5);
340
341 /* no tonal components */
342
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 1108 if (nb_components == 0)
343 1108 return 0;
344
345 ✗ coding_mode_selector = get_bits(gb, 2);
346 ✗ if (coding_mode_selector == 2)
347 ✗ return AVERROR_INVALIDDATA;
348
349 ✗ coding_mode = coding_mode_selector & 1;
350
351 ✗ for (i = 0; i < nb_components; i++) {
352 int coded_values_per_component, quant_step_index;
353
354 ✗ for (b = 0; b <= num_bands; b++)
355 ✗ band_flags[b] = get_bits1(gb);
356
357 ✗ coded_values_per_component = get_bits(gb, 3);
358
359 ✗ quant_step_index = get_bits(gb, 3);
360 ✗ if (quant_step_index <= 1)
361 ✗ return AVERROR_INVALIDDATA;
362
363 ✗ if (coding_mode_selector == 3)
364 ✗ coding_mode = get_bits1(gb);
365
366 ✗ for (b = 0; b < (num_bands + 1) * 4; b++) {
367 int coded_components;
368
369 ✗ if (band_flags[b >> 2] == 0)
370 ✗ continue;
371
372 ✗ coded_components = get_bits(gb, 3);
373
374 ✗ for (c = 0; c < coded_components; c++) {
375 ✗ TonalComponent *cmp = &components[component_count];
376 int sf_index, coded_values, max_coded_values;
377 float scale_factor;
378
379 ✗ sf_index = get_bits(gb, 6);
380 ✗ if (component_count >= 64)
381 ✗ return AVERROR_INVALIDDATA;
382
383 ✗ cmp->pos = b * 64 + get_bits(gb, 6);
384
385 ✗ max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
386 ✗ coded_values = coded_values_per_component + 1;
387 ✗ coded_values = FFMIN(max_coded_values, coded_values);
388
389 ✗ scale_factor = ff_atrac_sf_table[sf_index] *
390 ✗ inv_max_quant[quant_step_index];
391
392 ✗ read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
393 mantissa, coded_values);
394
395 ✗ cmp->num_coefs = coded_values;
396
397 /* inverse quant */
398 ✗ for (m = 0; m < coded_values; m++)
399 ✗ cmp->coef[m] = mantissa[m] * scale_factor;
400
401 ✗ component_count++;
402 }
403 }
404 }
405
406 ✗ return component_count;
407 }
408
409 /**
410 * Decode gain parameters for the coded bands
411 *
412 * @param block the gainblock for the current band
413 * @param num_bands amount of coded bands
414 */
415 1108 static int decode_gain_control(GetBitContext *gb, GainBlock *block,
416 int num_bands)
417 {
418 int b, j;
419 int *level, *loc;
420
421 1108 AtracGainInfo *gain = block->g_block;
422
423
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 3912 for (b = 0; b <= num_bands; b++) {
424 2804 gain[b].num_points = get_bits(gb, 3);
425 2804 level = gain[b].lev_code;
426 2804 loc = gain[b].loc_code;
427
428
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 3678 for (j = 0; j < gain[b].num_points; j++) {
429 874 level[j] = get_bits(gb, 4);
430 874 loc[j] = get_bits(gb, 5);
431
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 874 if (j && loc[j] <= loc[j - 1])
432 ✗ return AVERROR_INVALIDDATA;
433 }
434 }
435
436 /* Clear the unused blocks. */
437
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 2736 for (; b < 4 ; b++)
438 1628 gain[b].num_points = 0;
439
440 1108 return 0;
441 }
442
443 /**
444 * Combine the tonal band spectrum and regular band spectrum
445 *
446 * @param spectrum output spectrum buffer
447 * @param num_components number of tonal components
448 * @param components tonal components for this band
449 * @return position of the last tonal coefficient
450 */
451 1108 static int add_tonal_components(float *spectrum, int num_components,
452 TonalComponent *components)
453 {
454 1108 int i, j, last_pos = -1;
455 float *input, *output;
456
457
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 1108 for (i = 0; i < num_components; i++) {
458 ✗ last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
459 ✗ input = components[i].coef;
460 ✗ output = &spectrum[components[i].pos];
461
462 ✗ for (j = 0; j < components[i].num_coefs; j++)
463 ✗ output[j] += input[j];
464 }
465
466 1108 return last_pos;
467 }
468
469 #define INTERPOLATE(old, new, nsample) \
470 ((old) + (nsample) * 0.125 * ((new) - (old)))
471
472 260 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
473 int *curr_code)
474 {
475 int i, nsample, band;
476 float mc1_l, mc1_r, mc2_l, mc2_r;
477
478
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 1300 for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
479 1040 int s1 = prev_code[i];
480 1040 int s2 = curr_code[i];
481 1040 nsample = band;
482
483
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 1040 if (s1 != s2) {
484 /* Selector value changed, interpolation needed. */
485 38 mc1_l = matrix_coeffs[s1 * 2 ];
486 38 mc1_r = matrix_coeffs[s1 * 2 + 1];
487 38 mc2_l = matrix_coeffs[s2 * 2 ];
488 38 mc2_r = matrix_coeffs[s2 * 2 + 1];
489
490 /* Interpolation is done over the first eight samples. */
491
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 342 for (; nsample < band + 8; nsample++) {
492 304 float c1 = su1[nsample];
493 304 float c2 = su2[nsample];
494 304 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
495 304 c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
496 304 su1[nsample] = c2;
497 304 su2[nsample] = c1 * 2.0 - c2;
498 }
499 }
500
501 /* Apply the matrix without interpolation. */
502
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 1040 switch (s2) {
503 45 case 0: /* M/S decoding */
504
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 11413 for (; nsample < band + 256; nsample++) {
505 11368 float c1 = su1[nsample];
506 11368 float c2 = su2[nsample];
507 11368 su1[nsample] = c2 * 2.0;
508 11368 su2[nsample] = (c1 - c2) * 2.0;
509 }
510 45 break;
511 ✗ case 1:
512 ✗ for (; nsample < band + 256; nsample++) {
513 ✗ float c1 = su1[nsample];
514 ✗ float c2 = su2[nsample];
515 ✗ su1[nsample] = (c1 + c2) * 2.0;
516 ✗ su2[nsample] = c2 * -2.0;
517 }
518 ✗ break;
519 995 case 2:
520 case 3:
521
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 255563 for (; nsample < band + 256; nsample++) {
522 254568 float c1 = su1[nsample];
523 254568 float c2 = su2[nsample];
524 254568 su1[nsample] = c1 + c2;
525 254568 su2[nsample] = c1 - c2;
526 }
527 995 break;
528 ✗ default:
529 ✗ av_unreachable("curr_code/matrix_coeff_index_* values are stored in two bits");
530 }
531 }
532 260 }
533
534 492 static void get_channel_weights(int index, int flag, float ch[2])
535 {
536
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 492 if (index == 7) {
537 53 ch[0] = 1.0;
538 53 ch[1] = 1.0;
539 } else {
540 439 ch[0] = (index & 7) / 7.0;
541 439 ch[1] = sqrt(2 - ch[0] * ch[0]);
542
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 439 if (flag)
543 115 FFSWAP(float, ch[0], ch[1]);
544 }
545 492 }
546
547 260 static void channel_weighting(float *su1, float *su2, int *p3)
548 {
549 int band, nsample;
550 /* w[x][y] y=0 is left y=1 is right */
551 float w[2][2];
552
553
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 260 if (p3[1] != 7 || p3[3] != 7) {
554 246 get_channel_weights(p3[1], p3[0], w[0]);
555 246 get_channel_weights(p3[3], p3[2], w[1]);
556
557
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 984 for (band = 256; band < 4 * 256; band += 256) {
558
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 6642 for (nsample = band; nsample < band + 8; nsample++) {
559 5904 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
560 5904 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
561 }
562
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 183762 for(; nsample < band + 256; nsample++) {
563 183024 su1[nsample] *= w[1][0];
564 183024 su2[nsample] *= w[1][1];
565 }
566 }
567 }
568 260 }
569
570 /**
571 * Decode a Sound Unit
572 *
573 * @param snd the channel unit to be used
574 * @param output the decoded samples before IQMF in float representation
575 * @param channel_num channel number
576 * @param coding_mode the coding mode (JOINT_STEREO or single channels)
577 */
578 1108 static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
579 ChannelUnit *snd, float *output,
580 int channel_num, int coding_mode)
581 {
582 int band, ret, num_subbands, last_tonal, num_bands;
583 1108 GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
584 1108 GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
585
586
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 1108 if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) {
587
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 260 if (get_bits(gb, 2) != 3) {
588 ✗ av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
589 ✗ return AVERROR_INVALIDDATA;
590 }
591 } else {
592
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 848 if (get_bits(gb, 6) != 0x28) {
593 ✗ av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
594 ✗ return AVERROR_INVALIDDATA;
595 }
596 }
597
598 /* number of coded QMF bands */
599 1108 snd->bands_coded = get_bits(gb, 2);
600
601 1108 ret = decode_gain_control(gb, gain2, snd->bands_coded);
602
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 1108 if (ret)
603 ✗ return ret;
604
605 1108 snd->num_components = decode_tonal_components(gb, snd->components,
606 snd->bands_coded);
607
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 1108 if (snd->num_components < 0)
608 ✗ return snd->num_components;
609
610 1108 num_subbands = decode_spectrum(gb, snd->spectrum);
611
612 /* Merge the decoded spectrum and tonal components. */
613 1108 last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
614 1108 snd->components);
615
616
617 /* calculate number of used MLT/QMF bands according to the amount of coded
618 spectral lines */
619 1108 num_bands = (subband_tab[num_subbands + 1] - 1) >> 8;
620
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 1108 if (last_tonal >= 0)
621 ✗ num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
622
623
624 /* Reconstruct time domain samples. */
625
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 5540 for (band = 0; band < 4; band++) {
626 /* Perform the IMDCT step without overlapping. */
627
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 4432 if (band <= num_bands)
628 2804 imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
629 else
630 1628 memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
631
632 /* gain compensation and overlapping */
633 4432 ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
634 4432 &snd->prev_frame[band * 256],
635 &gain1->g_block[band], &gain2->g_block[band],
636 4432 256, &output[band * 256]);
637 }
638
639 /* Swap the gain control buffers for the next frame. */
640 1108 snd->gc_blk_switch ^= 1;
641
642 1108 return 0;
643 }
644
645 554 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
646 float **out_samples)
647 {
648 554 ATRAC3Context *q = avctx->priv_data;
649 int ret, i, ch;
650 uint8_t *ptr1;
651 554 int channels = avctx->ch_layout.nb_channels;
652
653
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 554 if (q->coding_mode == JOINT_STEREO) {
654 /* channel coupling mode */
655
656 /* Decode sound unit pairs (channels are expected to be even).
657 * Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */
658 const uint8_t *js_databuf;
659 int js_pair, js_block_align;
660
661 260 js_block_align = (avctx->block_align / channels) * 2; /* block pair */
662
663
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 520 for (ch = 0; ch < channels; ch = ch + 2) {
664 260 js_pair = ch/2;
665 260 js_databuf = databuf + js_pair * js_block_align; /* align to current pair */
666
667 /* Set the bitstream reader at the start of first channel sound unit. */
668 260 init_get_bits(&q->gb,
669 js_databuf, js_block_align * 8);
670
671 /* decode Sound Unit 1 */
672 260 ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch],
673 260 out_samples[ch], ch, JOINT_STEREO);
674
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 260 if (ret != 0)
675 ✗ return ret;
676
677 /* Framedata of the su2 in the joint-stereo mode is encoded in
678 * reverse byte order so we need to swap it first. */
679
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 260 if (js_databuf == q->decoded_bytes_buffer) {
680 ✗ uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1;
681 ✗ ptr1 = q->decoded_bytes_buffer;
682 ✗ for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--)
683 ✗ FFSWAP(uint8_t, *ptr1, *ptr2);
684 } else {
685 260 const uint8_t *ptr2 = js_databuf + js_block_align - 1;
686
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 50180 for (i = 0; i < js_block_align; i++)
687 49920 q->decoded_bytes_buffer[i] = *ptr2--;
688 }
689
690 /* Skip the sync codes (0xF8). */
691 260 ptr1 = q->decoded_bytes_buffer;
692
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 260 for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
693 ✗ if (i >= js_block_align)
694 ✗ return AVERROR_INVALIDDATA;
695 }
696
697
698 /* set the bitstream reader at the start of the second Sound Unit */
699 260 ret = init_get_bits8(&q->gb,
700 260 ptr1, q->decoded_bytes_buffer + js_block_align - ptr1);
701
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 260 if (ret < 0)
702 ✗ return ret;
703
704 /* Fill the Weighting coeffs delay buffer */
705 260 memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2],
706 4 * sizeof(*q->weighting_delay[js_pair]));
707 260 q->weighting_delay[js_pair][4] = get_bits1(&q->gb);
708 260 q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3);
709
710
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 1300 for (i = 0; i < 4; i++) {
711 1040 q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i];
712 1040 q->matrix_coeff_index_now[js_pair][i] = q->matrix_coeff_index_next[js_pair][i];
713 1040 q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2);
714 }
715
716 /* Decode Sound Unit 2. */
717 260 ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1],
718 260 out_samples[ch+1], ch+1, JOINT_STEREO);
719
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 260 if (ret != 0)
720 ✗ return ret;
721
722 /* Reconstruct the channel coefficients. */
723 260 reverse_matrixing(out_samples[ch], out_samples[ch+1],
724 260 q->matrix_coeff_index_prev[js_pair],
725 260 q->matrix_coeff_index_now[js_pair]);
726
727 260 channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]);
728 }
729 } else {
730 /* single channels */
731 /* Decode the channel sound units. */
732
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 882 for (i = 0; i < channels; i++) {
733 /* Set the bitstream reader at the start of a channel sound unit. */
734 588 init_get_bits(&q->gb,
735 588 databuf + i * avctx->block_align / channels,
736 588 avctx->block_align * 8 / channels);
737
738 588 ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
739 588 out_samples[i], i, q->coding_mode);
740
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 588 if (ret != 0)
741 ✗ return ret;
742 }
743 }
744
745 /* Apply the iQMF synthesis filter. */
746
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 1662 for (i = 0; i < channels; i++) {
747 1108 float *p1 = out_samples[i];
748 1108 float *p2 = p1 + 256;
749 1108 float *p3 = p2 + 256;
750 1108 float *p4 = p3 + 256;
751 1108 ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
752 1108 ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
753 1108 ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
754 }
755
756 554 return 0;
757 }
758
759 ✗ static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
760 int size, float **out_samples)
761 {
762 ✗ ATRAC3Context *q = avctx->priv_data;
763 ✗ int channels = avctx->ch_layout.nb_channels;
764 int ret, i;
765
766 /* Set the bitstream reader at the start of a channel sound unit. */
767 ✗ init_get_bits(&q->gb, databuf, size * 8);
768 /* single channels */
769 /* Decode the channel sound units. */
770 ✗ for (i = 0; i < channels; i++) {
771 ✗ ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
772 ✗ out_samples[i], i, q->coding_mode);
773 ✗ if (ret != 0)
774 ✗ return ret;
775 ✗ while (i < channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) {
776 ✗ skip_bits(&q->gb, 1);
777 }
778 }
779
780 /* Apply the iQMF synthesis filter. */
781 ✗ for (i = 0; i < channels; i++) {
782 ✗ float *p1 = out_samples[i];
783 ✗ float *p2 = p1 + 256;
784 ✗ float *p3 = p2 + 256;
785 ✗ float *p4 = p3 + 256;
786 ✗ ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
787 ✗ ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
788 ✗ ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
789 }
790
791 ✗ return 0;
792 }
793
794 557 static int atrac3_decode_frame(AVCodecContext *avctx, AVFrame *frame,
795 int *got_frame_ptr, AVPacket *avpkt)
796 {
797 557 const uint8_t *buf = avpkt->data;
798 557 int buf_size = avpkt->size;
799 557 ATRAC3Context *q = avctx->priv_data;
800 int ret;
801 const uint8_t *databuf;
802
803
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 557 if (buf_size < avctx->block_align) {
804 3 av_log(avctx, AV_LOG_ERROR,
805 "Frame too small (%d bytes). Truncated file?\n", buf_size);
806 3 return AVERROR_INVALIDDATA;
807 }
808
809 /* get output buffer */
810 554 frame->nb_samples = SAMPLES_PER_FRAME;
811
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 554 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
812 ✗ return ret;
813
814 /* Check if we need to descramble and what buffer to pass on. */
815
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 554 if (q->scrambled_stream) {
816 ✗ decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
817 ✗ databuf = q->decoded_bytes_buffer;
818 } else {
819 554 databuf = buf;
820 }
821
822 554 ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
823
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 554 if (ret) {
824 ✗ av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
825 ✗ return ret;
826 }
827
828 554 *got_frame_ptr = 1;
829
830 554 return avctx->block_align;
831 }
832
833 ✗ static int atrac3al_decode_frame(AVCodecContext *avctx, AVFrame *frame,
834 int *got_frame_ptr, AVPacket *avpkt)
835 {
836 int ret;
837
838 ✗ frame->nb_samples = SAMPLES_PER_FRAME;
839 ✗ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
840 ✗ return ret;
841
842 ✗ ret = al_decode_frame(avctx, avpkt->data, avpkt->size,
843 ✗ (float **)frame->extended_data);
844 ✗ if (ret) {
845 ✗ av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
846 ✗ return ret;
847 }
848
849 ✗ *got_frame_ptr = 1;
850
851 ✗ return avpkt->size;
852 }
853
854 6 static av_cold void atrac3_init_static_data(void)
855 {
856 6 VLCElem *table = atrac3_vlc_table;
857 6 const uint8_t (*hufftabs)[2] = atrac3_hufftabs;
858 int i;
859
860 6 init_imdct_window();
861 6 ff_atrac_generate_tables();
862
863 /* Initialize the VLC tables. */
864
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 48 for (i = 0; i < 7; i++) {
865 42 spectral_coeff_tab[i].table = table;
866 42 spectral_coeff_tab[i].table_allocated = 256;
867 42 ff_vlc_init_from_lengths(&spectral_coeff_tab[i], ATRAC3_VLC_BITS, huff_tab_sizes[i],
868 42 &hufftabs[0][1], 2,
869 42 &hufftabs[0][0], 2, 1,
870 -31, VLC_INIT_USE_STATIC, NULL);
871 42 hufftabs += huff_tab_sizes[i];
872 42 table += 256;
873 }
874 6 }
875
876 9 static av_cold int atrac3_decode_init(AVCodecContext *avctx)
877 {
878 static AVOnce init_static_once = AV_ONCE_INIT;
879 int i, js_pair, ret;
880 int version, delay, samples_per_frame, frame_factor;
881 9 const uint8_t *edata_ptr = avctx->extradata;
882 9 ATRAC3Context *q = avctx->priv_data;
883 AVFloatDSPContext *fdsp;
884 9 float scale = 1.0 / 32768;
885 9 int channels = avctx->ch_layout.nb_channels;
886
887
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 9 if (channels < MIN_CHANNELS || channels > MAX_CHANNELS) {
888 ✗ av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
889 ✗ return AVERROR(EINVAL);
890 }
891
892 /* Take care of the codec-specific extradata. */
893
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 9 if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) {
894 ✗ version = 4;
895 ✗ samples_per_frame = SAMPLES_PER_FRAME * channels;
896 ✗ delay = 0x88E;
897 ✗ q->coding_mode = SINGLE;
898
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 9 } else if (avctx->extradata_size == 14) {
899 /* Parse the extradata, WAV format */
900 9 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
901 bytestream_get_le16(&edata_ptr)); // Unknown value always 1
902 9 edata_ptr += 4; // samples per channel
903 9 q->coding_mode = bytestream_get_le16(&edata_ptr);
904 9 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
905 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
906 9 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
907 9 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
908 bytestream_get_le16(&edata_ptr)); // Unknown always 0
909
910 /* setup */
911 9 samples_per_frame = SAMPLES_PER_FRAME * channels;
912 9 version = 4;
913 9 delay = 0x88E;
914
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 9 q->coding_mode = q->coding_mode ? JOINT_STEREO : SINGLE;
915 9 q->scrambled_stream = 0;
916
917
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 9 if (avctx->block_align != 96 * channels * frame_factor &&
918
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 7 avctx->block_align != 152 * channels * frame_factor &&
919
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 5 avctx->block_align != 192 * channels * frame_factor) {
920 ✗ av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
921 "configuration %d/%d/%d\n", avctx->block_align,
922 channels, frame_factor);
923 ✗ return AVERROR_INVALIDDATA;
924 }
925 ✗ } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
926 /* Parse the extradata, RM format. */
927 ✗ version = bytestream_get_be32(&edata_ptr);
928 ✗ samples_per_frame = bytestream_get_be16(&edata_ptr);
929 ✗ delay = bytestream_get_be16(&edata_ptr);
930 ✗ q->coding_mode = bytestream_get_be16(&edata_ptr);
931 ✗ q->scrambled_stream = 1;
932
933 } else {
934 ✗ av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
935 avctx->extradata_size);
936 ✗ return AVERROR(EINVAL);
937 }
938
939 /* Check the extradata */
940
941
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 9 if (version != 4) {
942 ✗ av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
943 ✗ return AVERROR_INVALIDDATA;
944 }
945
946
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 9 if (samples_per_frame != SAMPLES_PER_FRAME * channels) {
947 ✗ av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
948 samples_per_frame);
949 ✗ return AVERROR_INVALIDDATA;
950 }
951
952
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 9 if (delay != 0x88E) {
953 ✗ av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
954 delay);
955 ✗ return AVERROR_INVALIDDATA;
956 }
957
958
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 9 if (q->coding_mode == SINGLE)
959 7 av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
960
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 2 else if (q->coding_mode == JOINT_STEREO) {
961
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 2 if (channels % 2 == 1) { /* Joint stereo channels must be even */
962 ✗ av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
963 ✗ return AVERROR_INVALIDDATA;
964 }
965 2 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
966 } else {
967 ✗ av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
968 q->coding_mode);
969 ✗ return AVERROR_INVALIDDATA;
970 }
971
972
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 9 if (avctx->block_align > 4096 || avctx->block_align <= 0)
973 ✗ return AVERROR(EINVAL);
974
975 9 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
976 AV_INPUT_BUFFER_PADDING_SIZE);
977
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 9 if (!q->decoded_bytes_buffer)
978 ✗ return AVERROR(ENOMEM);
979
980 9 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
981
982 /* initialize the MDCT transform */
983
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 9 if ((ret = av_tx_init(&q->mdct_ctx, &q->mdct_fn, AV_TX_FLOAT_MDCT, 1, 256,
984 &scale, AV_TX_FULL_IMDCT)) < 0) {
985 ✗ av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
986 ✗ return ret;
987 }
988
989 /* init the joint-stereo decoding data */
990
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 45 for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) {
991 36 q->weighting_delay[js_pair][0] = 0;
992 36 q->weighting_delay[js_pair][1] = 7;
993 36 q->weighting_delay[js_pair][2] = 0;
994 36 q->weighting_delay[js_pair][3] = 7;
995 36 q->weighting_delay[js_pair][4] = 0;
996 36 q->weighting_delay[js_pair][5] = 7;
997
998
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 180 for (i = 0; i < 4; i++) {
999 144 q->matrix_coeff_index_prev[js_pair][i] = 3;
1000 144 q->matrix_coeff_index_now[js_pair][i] = 3;
1001 144 q->matrix_coeff_index_next[js_pair][i] = 3;
1002 }
1003 }
1004
1005 9 ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
1006 9 fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1007
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 9 if (!fdsp)
1008 ✗ return AVERROR(ENOMEM);
1009 9 q->vector_fmul = fdsp->vector_fmul;
1010 9 av_free(fdsp);
1011
1012 9 q->units = av_calloc(channels, sizeof(*q->units));
1013
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 9 if (!q->units)
1014 ✗ return AVERROR(ENOMEM);
1015
1016 9 ff_thread_once(&init_static_once, atrac3_init_static_data);
1017
1018 9 return 0;
1019 }
1020
1021 const FFCodec ff_atrac3_decoder = {
1022 .p.name = "atrac3",
1023 CODEC_LONG_NAME("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
1024 .p.type = AVMEDIA_TYPE_AUDIO,
1025 .p.id = AV_CODEC_ID_ATRAC3,
1026 .priv_data_size = sizeof(ATRAC3Context),
1027 .init = atrac3_decode_init,
1028 .close = atrac3_decode_close,
1029 FF_CODEC_DECODE_CB(atrac3_decode_frame),
1030 .p.capabilities = AV_CODEC_CAP_DR1,
1031 CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP),
1032 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1033 };
1034
1035 const FFCodec ff_atrac3al_decoder = {
1036 .p.name = "atrac3al",
1037 CODEC_LONG_NAME("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
1038 .p.type = AVMEDIA_TYPE_AUDIO,
1039 .p.id = AV_CODEC_ID_ATRAC3AL,
1040 .priv_data_size = sizeof(ATRAC3Context),
1041 .init = atrac3_decode_init,
1042 .close = atrac3_decode_close,
1043 FF_CODEC_DECODE_CB(atrac3al_decode_frame),
1044 .p.capabilities = AV_CODEC_CAP_DR1,
1045 CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP),
1046 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1047 };
1048