FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/atrac3.c
Date: 2021-09-24 20:55:06
Exec Total Coverage
Lines: 289 448 64.5%
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 #include <stdio.h>
38
39 #include "libavutil/attributes.h"
40 #include "libavutil/float_dsp.h"
41 #include "libavutil/libm.h"
42 #include "libavutil/mem_internal.h"
43 #include "libavutil/thread.h"
44
45 #include "avcodec.h"
46 #include "bytestream.h"
47 #include "fft.h"
48 #include "get_bits.h"
49 #include "internal.h"
50
51 #include "atrac.h"
52 #include "atrac3data.h"
53
54 #define MIN_CHANNELS 1
55 #define MAX_CHANNELS 8
56 #define MAX_JS_PAIRS 8 / 2
57
58 #define JOINT_STEREO 0x12
59 #define SINGLE 0x2
60
61 #define SAMPLES_PER_FRAME 1024
62 #define MDCT_SIZE 512
63
64 #define ATRAC3_VLC_BITS 8
65
66 typedef struct GainBlock {
67 AtracGainInfo g_block[4];
68 } GainBlock;
69
70 typedef struct TonalComponent {
71 int pos;
72 int num_coefs;
73 float coef[8];
74 } TonalComponent;
75
76 typedef struct ChannelUnit {
77 int bands_coded;
78 int num_components;
79 float prev_frame[SAMPLES_PER_FRAME];
80 int gc_blk_switch;
81 TonalComponent components[64];
82 GainBlock gain_block[2];
83
84 DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
85 DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
86
87 float delay_buf1[46]; ///<qmf delay buffers
88 float delay_buf2[46];
89 float delay_buf3[46];
90 } ChannelUnit;
91
92 typedef struct ATRAC3Context {
93 GetBitContext gb;
94 //@{
95 /** stream data */
96 int coding_mode;
97
98 ChannelUnit *units;
99 //@}
100 //@{
101 /** joint-stereo related variables */
102 int matrix_coeff_index_prev[MAX_JS_PAIRS][4];
103 int matrix_coeff_index_now[MAX_JS_PAIRS][4];
104 int matrix_coeff_index_next[MAX_JS_PAIRS][4];
105 int weighting_delay[MAX_JS_PAIRS][6];
106 //@}
107 //@{
108 /** data buffers */
109 uint8_t *decoded_bytes_buffer;
110 float temp_buf[1070];
111 //@}
112 //@{
113 /** extradata */
114 int scrambled_stream;
115 //@}
116
117 AtracGCContext gainc_ctx;
118 FFTContext mdct_ctx;
119 void (*vector_fmul)(float *dst, const float *src0, const float *src1,
120 int len);
121 } ATRAC3Context;
122
123 static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
124 static VLC_TYPE atrac3_vlc_table[7 * 1 << ATRAC3_VLC_BITS][2];
125 static VLC spectral_coeff_tab[7];
126
127 /**
128 * Regular 512 points IMDCT without overlapping, with the exception of the
129 * swapping of odd bands caused by the reverse spectra of the QMF.
130 *
131 * @param odd_band 1 if the band is an odd band
132 */
133 2800 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
134 {
135 int i;
136
137
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2800 if (odd_band) {
138 /**
139 * Reverse the odd bands before IMDCT, this is an effect of the QMF
140 * transform or it gives better compression to do it this way.
141 * FIXME: It should be possible to handle this in imdct_calc
142 * for that to happen a modification of the prerotation step of
143 * all SIMD code and C code is needed.
144 * Or fix the functions before so they generate a pre reversed spectrum.
145 */
146
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109392 for (i = 0; i < 128; i++)
147 108544 FFSWAP(float, input[i], input[255 - i]);
148 }
149
150 2800 q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
151
152 /* Perform windowing on the output. */
153 2800 q->vector_fmul(output, output, mdct_window, MDCT_SIZE);
154 2800 }
155
156 /*
157 * indata descrambling, only used for data coming from the rm container
158 */
159 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
160 {
161 int i, off;
162 uint32_t c;
163 const uint32_t *buf;
164 uint32_t *output = (uint32_t *)out;
165
166 off = (intptr_t)input & 3;
167 buf = (const uint32_t *)(input - off);
168 if (off)
169 c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
170 else
171 c = av_be2ne32(0x537F6103U);
172 bytes += 3 + off;
173 for (i = 0; i < bytes / 4; i++)
174 output[i] = c ^ buf[i];
175
176 if (off)
177 avpriv_request_sample(NULL, "Offset of %d", off);
178
179 return off;
180 }
181
182 4 static av_cold void init_imdct_window(void)
183 {
184 int i, j;
185
186 /* generate the mdct window, for details see
187 * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
188
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516 for (i = 0, j = 255; i < 128; i++, j--) {
189 512 float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
190 512 float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
191 512 float w = 0.5 * (wi * wi + wj * wj);
192 512 mdct_window[i] = mdct_window[511 - i] = wi / w;
193 512 mdct_window[j] = mdct_window[511 - j] = wj / w;
194 }
195 4 }
196
197 7 static av_cold int atrac3_decode_close(AVCodecContext *avctx)
198 {
199 7 ATRAC3Context *q = avctx->priv_data;
200
201 7 av_freep(&q->units);
202 7 av_freep(&q->decoded_bytes_buffer);
203
204 7 ff_mdct_end(&q->mdct_ctx);
205
206 7 return 0;
207 }
208
209 /**
210 * Mantissa decoding
211 *
212 * @param selector which table the output values are coded with
213 * @param coding_flag constant length coding or variable length coding
214 * @param mantissas mantissa output table
215 * @param num_codes number of values to get
216 */
217 28209 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
218 int coding_flag, int *mantissas,
219 int num_codes)
220 {
221 int i, code, huff_symb;
222
223
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28209 if (selector == 1)
224 15517 num_codes /= 2;
225
226
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28209 if (coding_flag != 0) {
227 /* constant length coding (CLC) */
228 int num_bits = clc_length_tab[selector];
229
230 if (selector > 1) {
231 for (i = 0; i < num_codes; i++) {
232 if (num_bits)
233 code = get_sbits(gb, num_bits);
234 else
235 code = 0;
236 mantissas[i] = code;
237 }
238 } else {
239 for (i = 0; i < num_codes; i++) {
240 if (num_bits)
241 code = get_bits(gb, num_bits); // num_bits is always 4 in this case
242 else
243 code = 0;
244 mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
245 mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
246 }
247 }
248 } else {
249 /* variable length coding (VLC) */
250
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28209 if (selector != 1) {
251
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192516 for (i = 0; i < num_codes; i++) {
252 179824 mantissas[i] = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
253 ATRAC3_VLC_BITS, 1);
254 }
255 } else {
256
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250405 for (i = 0; i < num_codes; i++) {
257 234888 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
258 ATRAC3_VLC_BITS, 1);
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
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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
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29317 for (i = 0; i <= num_subbands; i++) {
287
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28209 if (subband_vlc_index[i] != 0)
288 28209 sf_index[i] = get_bits(gb, 6);
289 }
290
291
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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
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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
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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
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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
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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
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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
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874 if (j && loc[j] <= loc[j - 1])
430 return AVERROR_INVALIDDATA;
431 }
432 }
433
434 /* Clear the unused blocks. */
435
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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
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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
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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
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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
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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
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1040 switch (s2) {
501 45 case 0: /* M/S decoding */
502
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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
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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
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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
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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
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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
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984 for (band = 256; band < 4 * 256; band += 256) {
556
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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
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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
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1108 if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) {
585
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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
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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
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1108 if (ret)
601 return ret;
602
603 1108 snd->num_components = decode_tonal_components(gb, snd->components,
604 snd->bands_coded);
605
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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
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1108 if (last_tonal >= 0)
619 num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
620
621
622 /* Reconstruct time domain samples. */
623
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5540 for (band = 0; band < 4; band++) {
624 /* Perform the IMDCT step without overlapping. */
625
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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588 if (ret != 0)
738 return ret;
739 }
740 }
741
742 /* Apply the iQMF synthesis filter. */
743
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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
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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
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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
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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
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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 4 VLC_TYPE (*table)[2] = atrac3_vlc_table;
855 4 const uint8_t (*hufftabs)[2] = atrac3_hufftabs;
856 int i;
857
858 4 init_imdct_window();
859 4 ff_atrac_generate_tables();
860
861 /* Initialize the VLC tables. */
862
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32 for (i = 0; i < 7; i++) {
863 28 spectral_coeff_tab[i].table = table;
864 28 spectral_coeff_tab[i].table_allocated = 256;
865 28 ff_init_vlc_from_lengths(&spectral_coeff_tab[i], ATRAC3_VLC_BITS, huff_tab_sizes[i],
866 28 &hufftabs[0][1], 2,
867 28 &hufftabs[0][0], 2, 1,
868 -31, INIT_VLC_USE_NEW_STATIC, NULL);
869 28 hufftabs += huff_tab_sizes[i];
870 28 table += 256;
871 }
872 4 }
873
874 7 static av_cold int atrac3_decode_init(AVCodecContext *avctx)
875 {
876 static AVOnce init_static_once = AV_ONCE_INIT;
877 int i, js_pair, ret;
878 int version, delay, samples_per_frame, frame_factor;
879 7 const uint8_t *edata_ptr = avctx->extradata;
880 7 ATRAC3Context *q = avctx->priv_data;
881 AVFloatDSPContext *fdsp;
882
883
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7 if (avctx->channels < MIN_CHANNELS || avctx->channels > MAX_CHANNELS) {
884 av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
885 return AVERROR(EINVAL);
886 }
887
888 /* Take care of the codec-specific extradata. */
889
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7 if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) {
890 version = 4;
891 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
892 delay = 0x88E;
893 q->coding_mode = SINGLE;
894
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7 } else if (avctx->extradata_size == 14) {
895 /* Parse the extradata, WAV format */
896 7 av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
897 bytestream_get_le16(&edata_ptr)); // Unknown value always 1
898 7 edata_ptr += 4; // samples per channel
899 7 q->coding_mode = bytestream_get_le16(&edata_ptr);
900 7 av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
901 bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
902 7 frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
903 7 av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
904 bytestream_get_le16(&edata_ptr)); // Unknown always 0
905
906 /* setup */
907 7 samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
908 7 version = 4;
909 7 delay = 0x88E;
910
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7 q->coding_mode = q->coding_mode ? JOINT_STEREO : SINGLE;
911 7 q->scrambled_stream = 0;
912
913
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7 if (avctx->block_align != 96 * avctx->channels * frame_factor &&
914
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5 avctx->block_align != 152 * avctx->channels * frame_factor &&
915
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3 avctx->block_align != 192 * avctx->channels * frame_factor) {
916 av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
917 "configuration %d/%d/%d\n", avctx->block_align,
918 avctx->channels, frame_factor);
919 return AVERROR_INVALIDDATA;
920 }
921 } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
922 /* Parse the extradata, RM format. */
923 version = bytestream_get_be32(&edata_ptr);
924 samples_per_frame = bytestream_get_be16(&edata_ptr);
925 delay = bytestream_get_be16(&edata_ptr);
926 q->coding_mode = bytestream_get_be16(&edata_ptr);
927 q->scrambled_stream = 1;
928
929 } else {
930 av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
931 avctx->extradata_size);
932 return AVERROR(EINVAL);
933 }
934
935 /* Check the extradata */
936
937
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7 if (version != 4) {
938 av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
939 return AVERROR_INVALIDDATA;
940 }
941
942
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7 if (samples_per_frame != SAMPLES_PER_FRAME * avctx->channels) {
943 av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
944 samples_per_frame);
945 return AVERROR_INVALIDDATA;
946 }
947
948
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7 if (delay != 0x88E) {
949 av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
950 delay);
951 return AVERROR_INVALIDDATA;
952 }
953
954
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7 if (q->coding_mode == SINGLE)
955 5 av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
956
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2 else if (q->coding_mode == JOINT_STEREO) {
957
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2 if (avctx->channels % 2 == 1) { /* Joint stereo channels must be even */
958 av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
959 return AVERROR_INVALIDDATA;
960 }
961 2 av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
962 } else {
963 av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
964 q->coding_mode);
965 return AVERROR_INVALIDDATA;
966 }
967
968
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7 if (avctx->block_align > 4096 || avctx->block_align <= 0)
969 return AVERROR(EINVAL);
970
971 7 q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
972 AV_INPUT_BUFFER_PADDING_SIZE);
973
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7 if (!q->decoded_bytes_buffer)
974 return AVERROR(ENOMEM);
975
976 7 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
977
978 /* initialize the MDCT transform */
979
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7 if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
980 av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
981 return ret;
982 }
983
984 /* init the joint-stereo decoding data */
985
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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
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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 fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1002
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7 if (!fdsp)
1003 return AVERROR(ENOMEM);
1004 7 q->vector_fmul = fdsp->vector_fmul;
1005 7 av_free(fdsp);
1006
1007 7 q->units = av_calloc(avctx->channels, sizeof(*q->units));
1008
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7 if (!q->units)
1009 return AVERROR(ENOMEM);
1010
1011 7 ff_thread_once(&init_static_once, atrac3_init_static_data);
1012
1013 7 return 0;
1014 }
1015
1016 const AVCodec ff_atrac3_decoder = {
1017 .name = "atrac3",
1018 .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
1019 .type = AVMEDIA_TYPE_AUDIO,
1020 .id = AV_CODEC_ID_ATRAC3,
1021 .priv_data_size = sizeof(ATRAC3Context),
1022 .init = atrac3_decode_init,
1023 .close = atrac3_decode_close,
1024 .decode = atrac3_decode_frame,
1025 .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1026 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1027 AV_SAMPLE_FMT_NONE },
1028 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1029 };
1030
1031 const AVCodec ff_atrac3al_decoder = {
1032 .name = "atrac3al",
1033 .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
1034 .type = AVMEDIA_TYPE_AUDIO,
1035 .id = AV_CODEC_ID_ATRAC3AL,
1036 .priv_data_size = sizeof(ATRAC3Context),
1037 .init = atrac3_decode_init,
1038 .close = atrac3_decode_close,
1039 .decode = atrac3al_decode_frame,
1040 .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1041 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1042 AV_SAMPLE_FMT_NONE },
1043 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1044 };
1045