9e3cd624a046821e3530c1c8db5d65bc65946945
[libav.git] / libavcodec / cook.c
1 /*
2 * COOK compatible decoder
3 * Copyright (c) 2003 Sascha Sommer
4 * Copyright (c) 2005 Benjamin Larsson
5 *
6 * This file is part of Libav.
7 *
8 * Libav 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 * Libav 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 Libav; 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 * Cook compatible decoder. Bastardization of the G.722.1 standard.
26 * This decoder handles RealNetworks, RealAudio G2 data.
27 * Cook is identified by the codec name cook in RM files.
28 *
29 * To use this decoder, a calling application must supply the extradata
30 * bytes provided from the RM container; 8+ bytes for mono streams and
31 * 16+ for stereo streams (maybe more).
32 *
33 * Codec technicalities (all this assume a buffer length of 1024):
34 * Cook works with several different techniques to achieve its compression.
35 * In the timedomain the buffer is divided into 8 pieces and quantized. If
36 * two neighboring pieces have different quantization index a smooth
37 * quantization curve is used to get a smooth overlap between the different
38 * pieces.
39 * To get to the transformdomain Cook uses a modulated lapped transform.
40 * The transform domain has 50 subbands with 20 elements each. This
41 * means only a maximum of 50*20=1000 coefficients are used out of the 1024
42 * available.
43 */
44
45 #include "libavutil/lfg.h"
46 #include "avcodec.h"
47 #include "get_bits.h"
48 #include "dsputil.h"
49 #include "bytestream.h"
50 #include "fft.h"
51 #include "libavutil/audioconvert.h"
52 #include "sinewin.h"
53
54 #include "cookdata.h"
55
56 /* the different Cook versions */
57 #define MONO 0x1000001
58 #define STEREO 0x1000002
59 #define JOINT_STEREO 0x1000003
60 #define MC_COOK 0x2000000 //multichannel Cook, not supported
61
62 #define SUBBAND_SIZE 20
63 #define MAX_SUBPACKETS 5
64
65 typedef struct {
66 int *now;
67 int *previous;
68 } cook_gains;
69
70 typedef struct {
71 int ch_idx;
72 int size;
73 int num_channels;
74 int cookversion;
75 int samples_per_frame;
76 int subbands;
77 int js_subband_start;
78 int js_vlc_bits;
79 int samples_per_channel;
80 int log2_numvector_size;
81 unsigned int channel_mask;
82 VLC ccpl; ///< channel coupling
83 int joint_stereo;
84 int bits_per_subpacket;
85 int bits_per_subpdiv;
86 int total_subbands;
87 int numvector_size; ///< 1 << log2_numvector_size;
88
89 float mono_previous_buffer1[1024];
90 float mono_previous_buffer2[1024];
91 /** gain buffers */
92 cook_gains gains1;
93 cook_gains gains2;
94 int gain_1[9];
95 int gain_2[9];
96 int gain_3[9];
97 int gain_4[9];
98 } COOKSubpacket;
99
100 typedef struct cook {
101 /*
102 * The following 5 functions provide the lowlevel arithmetic on
103 * the internal audio buffers.
104 */
105 void (* scalar_dequant)(struct cook *q, int index, int quant_index,
106 int* subband_coef_index, int* subband_coef_sign,
107 float* mlt_p);
108
109 void (* decouple) (struct cook *q,
110 COOKSubpacket *p,
111 int subband,
112 float f1, float f2,
113 float *decode_buffer,
114 float *mlt_buffer1, float *mlt_buffer2);
115
116 void (* imlt_window) (struct cook *q, float *buffer1,
117 cook_gains *gains_ptr, float *previous_buffer);
118
119 void (* interpolate) (struct cook *q, float* buffer,
120 int gain_index, int gain_index_next);
121
122 void (* saturate_output) (struct cook *q, int chan, float *out);
123
124 AVCodecContext* avctx;
125 GetBitContext gb;
126 /* stream data */
127 int nb_channels;
128 int bit_rate;
129 int sample_rate;
130 int num_vectors;
131 int samples_per_channel;
132 /* states */
133 AVLFG random_state;
134
135 /* transform data */
136 FFTContext mdct_ctx;
137 float* mlt_window;
138
139 /* VLC data */
140 VLC envelope_quant_index[13];
141 VLC sqvh[7]; //scalar quantization
142
143 /* generatable tables and related variables */
144 int gain_size_factor;
145 float gain_table[23];
146
147 /* data buffers */
148
149 uint8_t* decoded_bytes_buffer;
150 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
151 float decode_buffer_1[1024];
152 float decode_buffer_2[1024];
153 float decode_buffer_0[1060]; /* static allocation for joint decode */
154
155 const float *cplscales[5];
156 int num_subpackets;
157 COOKSubpacket subpacket[MAX_SUBPACKETS];
158 } COOKContext;
159
160 static float pow2tab[127];
161 static float rootpow2tab[127];
162
163 /*************** init functions ***************/
164
165 /* table generator */
166 static av_cold void init_pow2table(void){
167 int i;
168 for (i=-63 ; i<64 ; i++){
169 pow2tab[63+i]= pow(2, i);
170 rootpow2tab[63+i]=sqrt(pow(2, i));
171 }
172 }
173
174 /* table generator */
175 static av_cold void init_gain_table(COOKContext *q) {
176 int i;
177 q->gain_size_factor = q->samples_per_channel/8;
178 for (i=0 ; i<23 ; i++) {
179 q->gain_table[i] = pow(pow2tab[i+52] ,
180 (1.0/(double)q->gain_size_factor));
181 }
182 }
183
184
185 static av_cold int init_cook_vlc_tables(COOKContext *q) {
186 int i, result;
187
188 result = 0;
189 for (i=0 ; i<13 ; i++) {
190 result |= init_vlc (&q->envelope_quant_index[i], 9, 24,
191 envelope_quant_index_huffbits[i], 1, 1,
192 envelope_quant_index_huffcodes[i], 2, 2, 0);
193 }
194 av_log(q->avctx,AV_LOG_DEBUG,"sqvh VLC init\n");
195 for (i=0 ; i<7 ; i++) {
196 result |= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
197 cvh_huffbits[i], 1, 1,
198 cvh_huffcodes[i], 2, 2, 0);
199 }
200
201 for(i=0;i<q->num_subpackets;i++){
202 if (q->subpacket[i].joint_stereo==1){
203 result |= init_vlc (&q->subpacket[i].ccpl, 6, (1<<q->subpacket[i].js_vlc_bits)-1,
204 ccpl_huffbits[q->subpacket[i].js_vlc_bits-2], 1, 1,
205 ccpl_huffcodes[q->subpacket[i].js_vlc_bits-2], 2, 2, 0);
206 av_log(q->avctx,AV_LOG_DEBUG,"subpacket %i Joint-stereo VLC used.\n",i);
207 }
208 }
209
210 av_log(q->avctx,AV_LOG_DEBUG,"VLC tables initialized.\n");
211 return result;
212 }
213
214 static av_cold int init_cook_mlt(COOKContext *q) {
215 int j, ret;
216 int mlt_size = q->samples_per_channel;
217
218 if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
219 return AVERROR(ENOMEM);
220
221 /* Initialize the MLT window: simple sine window. */
222 ff_sine_window_init(q->mlt_window, mlt_size);
223 for(j=0 ; j<mlt_size ; j++)
224 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
225
226 /* Initialize the MDCT. */
227 if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0/32768.0))) {
228 av_free(q->mlt_window);
229 return ret;
230 }
231 av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
232 av_log2(mlt_size)+1);
233
234 return 0;
235 }
236
237 static const float *maybe_reformat_buffer32 (COOKContext *q, const float *ptr, int n)
238 {
239 if (1)
240 return ptr;
241 }
242
243 static av_cold void init_cplscales_table (COOKContext *q) {
244 int i;
245 for (i=0;i<5;i++)
246 q->cplscales[i] = maybe_reformat_buffer32 (q, cplscales[i], (1<<(i+2))-1);
247 }
248
249 /*************** init functions end ***********/
250
251 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4)
252 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
253
254 /**
255 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
256 * Why? No idea, some checksum/error detection method maybe.
257 *
258 * Out buffer size: extra bytes are needed to cope with
259 * padding/misalignment.
260 * Subpackets passed to the decoder can contain two, consecutive
261 * half-subpackets, of identical but arbitrary size.
262 * 1234 1234 1234 1234 extraA extraB
263 * Case 1: AAAA BBBB 0 0
264 * Case 2: AAAA ABBB BB-- 3 3
265 * Case 3: AAAA AABB BBBB 2 2
266 * Case 4: AAAA AAAB BBBB BB-- 1 5
267 *
268 * Nice way to waste CPU cycles.
269 *
270 * @param inbuffer pointer to byte array of indata
271 * @param out pointer to byte array of outdata
272 * @param bytes number of bytes
273 */
274
275 static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
276 int i, off;
277 uint32_t c;
278 const uint32_t* buf;
279 uint32_t* obuf = (uint32_t*) out;
280 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
281 * I'm too lazy though, should be something like
282 * for(i=0 ; i<bitamount/64 ; i++)
283 * (int64_t)out[i] = 0x37c511f237c511f2^av_be2ne64(int64_t)in[i]);
284 * Buffer alignment needs to be checked. */
285
286 off = (intptr_t)inbuffer & 3;
287 buf = (const uint32_t*) (inbuffer - off);
288 c = av_be2ne32((0x37c511f2 >> (off*8)) | (0x37c511f2 << (32-(off*8))));
289 bytes += 3 + off;
290 for (i = 0; i < bytes/4; i++)
291 obuf[i] = c ^ buf[i];
292
293 return off;
294 }
295
296 /**
297 * Cook uninit
298 */
299
300 static av_cold int cook_decode_close(AVCodecContext *avctx)
301 {
302 int i;
303 COOKContext *q = avctx->priv_data;
304 av_log(avctx,AV_LOG_DEBUG, "Deallocating memory.\n");
305
306 /* Free allocated memory buffers. */
307 av_free(q->mlt_window);
308 av_free(q->decoded_bytes_buffer);
309
310 /* Free the transform. */
311 ff_mdct_end(&q->mdct_ctx);
312
313 /* Free the VLC tables. */
314 for (i=0 ; i<13 ; i++) {
315 free_vlc(&q->envelope_quant_index[i]);
316 }
317 for (i=0 ; i<7 ; i++) {
318 free_vlc(&q->sqvh[i]);
319 }
320 for (i=0 ; i<q->num_subpackets ; i++) {
321 free_vlc(&q->subpacket[i].ccpl);
322 }
323
324 av_log(avctx,AV_LOG_DEBUG,"Memory deallocated.\n");
325
326 return 0;
327 }
328
329 /**
330 * Fill the gain array for the timedomain quantization.
331 *
332 * @param gb pointer to the GetBitContext
333 * @param gaininfo array[9] of gain indexes
334 */
335
336 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
337 {
338 int i, n;
339
340 while (get_bits1(gb)) {}
341 n = get_bits_count(gb) - 1; //amount of elements*2 to update
342
343 i = 0;
344 while (n--) {
345 int index = get_bits(gb, 3);
346 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
347
348 while (i <= index) gaininfo[i++] = gain;
349 }
350 while (i <= 8) gaininfo[i++] = 0;
351 }
352
353 /**
354 * Create the quant index table needed for the envelope.
355 *
356 * @param q pointer to the COOKContext
357 * @param quant_index_table pointer to the array
358 */
359
360 static void decode_envelope(COOKContext *q, COOKSubpacket *p, int* quant_index_table) {
361 int i,j, vlc_index;
362
363 quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize
364
365 for (i=1 ; i < p->total_subbands ; i++){
366 vlc_index=i;
367 if (i >= p->js_subband_start * 2) {
368 vlc_index-=p->js_subband_start;
369 } else {
370 vlc_index/=2;
371 if(vlc_index < 1) vlc_index = 1;
372 }
373 if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13
374
375 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table,
376 q->envelope_quant_index[vlc_index-1].bits,2);
377 quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding
378 }
379 }
380
381 /**
382 * Calculate the category and category_index vector.
383 *
384 * @param q pointer to the COOKContext
385 * @param quant_index_table pointer to the array
386 * @param category pointer to the category array
387 * @param category_index pointer to the category_index array
388 */
389
390 static void categorize(COOKContext *q, COOKSubpacket *p, int* quant_index_table,
391 int* category, int* category_index){
392 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
393 int exp_index2[102];
394 int exp_index1[102];
395
396 int tmp_categorize_array[128*2];
397 int tmp_categorize_array1_idx=p->numvector_size;
398 int tmp_categorize_array2_idx=p->numvector_size;
399
400 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
401
402 if(bits_left > q->samples_per_channel) {
403 bits_left = q->samples_per_channel +
404 ((bits_left - q->samples_per_channel)*5)/8;
405 //av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
406 }
407
408 memset(&exp_index1, 0, sizeof(exp_index1));
409 memset(&exp_index2, 0, sizeof(exp_index2));
410 memset(&tmp_categorize_array, 0, sizeof(tmp_categorize_array));
411
412 bias=-32;
413
414 /* Estimate bias. */
415 for (i=32 ; i>0 ; i=i/2){
416 num_bits = 0;
417 index = 0;
418 for (j=p->total_subbands ; j>0 ; j--){
419 exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
420 index++;
421 num_bits+=expbits_tab[exp_idx];
422 }
423 if(num_bits >= bits_left - 32){
424 bias+=i;
425 }
426 }
427
428 /* Calculate total number of bits. */
429 num_bits=0;
430 for (i=0 ; i<p->total_subbands ; i++) {
431 exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
432 num_bits += expbits_tab[exp_idx];
433 exp_index1[i] = exp_idx;
434 exp_index2[i] = exp_idx;
435 }
436 tmpbias1 = tmpbias2 = num_bits;
437
438 for (j = 1 ; j < p->numvector_size ; j++) {
439 if (tmpbias1 + tmpbias2 > 2*bits_left) { /* ---> */
440 int max = -999999;
441 index=-1;
442 for (i=0 ; i<p->total_subbands ; i++){
443 if (exp_index1[i] < 7) {
444 v = (-2*exp_index1[i]) - quant_index_table[i] + bias;
445 if ( v >= max) {
446 max = v;
447 index = i;
448 }
449 }
450 }
451 if(index==-1)break;
452 tmp_categorize_array[tmp_categorize_array1_idx++] = index;
453 tmpbias1 -= expbits_tab[exp_index1[index]] -
454 expbits_tab[exp_index1[index]+1];
455 ++exp_index1[index];
456 } else { /* <--- */
457 int min = 999999;
458 index=-1;
459 for (i=0 ; i<p->total_subbands ; i++){
460 if(exp_index2[i] > 0){
461 v = (-2*exp_index2[i])-quant_index_table[i]+bias;
462 if ( v < min) {
463 min = v;
464 index = i;
465 }
466 }
467 }
468 if(index == -1)break;
469 tmp_categorize_array[--tmp_categorize_array2_idx] = index;
470 tmpbias2 -= expbits_tab[exp_index2[index]] -
471 expbits_tab[exp_index2[index]-1];
472 --exp_index2[index];
473 }
474 }
475
476 for(i=0 ; i<p->total_subbands ; i++)
477 category[i] = exp_index2[i];
478
479 for(i=0 ; i<p->numvector_size-1 ; i++)
480 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
481
482 }
483
484
485 /**
486 * Expand the category vector.
487 *
488 * @param q pointer to the COOKContext
489 * @param category pointer to the category array
490 * @param category_index pointer to the category_index array
491 */
492
493 static inline void expand_category(COOKContext *q, int* category,
494 int* category_index){
495 int i;
496 for(i=0 ; i<q->num_vectors ; i++){
497 ++category[category_index[i]];
498 }
499 }
500
501 /**
502 * The real requantization of the mltcoefs
503 *
504 * @param q pointer to the COOKContext
505 * @param index index
506 * @param quant_index quantisation index
507 * @param subband_coef_index array of indexes to quant_centroid_tab
508 * @param subband_coef_sign signs of coefficients
509 * @param mlt_p pointer into the mlt buffer
510 */
511
512 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
513 int* subband_coef_index, int* subband_coef_sign,
514 float* mlt_p){
515 int i;
516 float f1;
517
518 for(i=0 ; i<SUBBAND_SIZE ; i++) {
519 if (subband_coef_index[i]) {
520 f1 = quant_centroid_tab[index][subband_coef_index[i]];
521 if (subband_coef_sign[i]) f1 = -f1;
522 } else {
523 /* noise coding if subband_coef_index[i] == 0 */
524 f1 = dither_tab[index];
525 if (av_lfg_get(&q->random_state) < 0x80000000) f1 = -f1;
526 }
527 mlt_p[i] = f1 * rootpow2tab[quant_index+63];
528 }
529 }
530 /**
531 * Unpack the subband_coef_index and subband_coef_sign vectors.
532 *
533 * @param q pointer to the COOKContext
534 * @param category pointer to the category array
535 * @param subband_coef_index array of indexes to quant_centroid_tab
536 * @param subband_coef_sign signs of coefficients
537 */
538
539 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, int* subband_coef_index,
540 int* subband_coef_sign) {
541 int i,j;
542 int vlc, vd ,tmp, result;
543
544 vd = vd_tab[category];
545 result = 0;
546 for(i=0 ; i<vpr_tab[category] ; i++){
547 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
548 if (p->bits_per_subpacket < get_bits_count(&q->gb)){
549 vlc = 0;
550 result = 1;
551 }
552 for(j=vd-1 ; j>=0 ; j--){
553 tmp = (vlc * invradix_tab[category])/0x100000;
554 subband_coef_index[vd*i+j] = vlc - tmp * (kmax_tab[category]+1);
555 vlc = tmp;
556 }
557 for(j=0 ; j<vd ; j++){
558 if (subband_coef_index[i*vd + j]) {
559 if(get_bits_count(&q->gb) < p->bits_per_subpacket){
560 subband_coef_sign[i*vd+j] = get_bits1(&q->gb);
561 } else {
562 result=1;
563 subband_coef_sign[i*vd+j]=0;
564 }
565 } else {
566 subband_coef_sign[i*vd+j]=0;
567 }
568 }
569 }
570 return result;
571 }
572
573
574 /**
575 * Fill the mlt_buffer with mlt coefficients.
576 *
577 * @param q pointer to the COOKContext
578 * @param category pointer to the category array
579 * @param quant_index_table pointer to the array
580 * @param mlt_buffer pointer to mlt coefficients
581 */
582
583
584 static void decode_vectors(COOKContext* q, COOKSubpacket* p, int* category,
585 int *quant_index_table, float* mlt_buffer){
586 /* A zero in this table means that the subband coefficient is
587 random noise coded. */
588 int subband_coef_index[SUBBAND_SIZE];
589 /* A zero in this table means that the subband coefficient is a
590 positive multiplicator. */
591 int subband_coef_sign[SUBBAND_SIZE];
592 int band, j;
593 int index=0;
594
595 for(band=0 ; band<p->total_subbands ; band++){
596 index = category[band];
597 if(category[band] < 7){
598 if(unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)){
599 index=7;
600 for(j=0 ; j<p->total_subbands ; j++) category[band+j]=7;
601 }
602 }
603 if(index>=7) {
604 memset(subband_coef_index, 0, sizeof(subband_coef_index));
605 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
606 }
607 q->scalar_dequant(q, index, quant_index_table[band],
608 subband_coef_index, subband_coef_sign,
609 &mlt_buffer[band * SUBBAND_SIZE]);
610 }
611
612 if(p->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){
613 return;
614 } /* FIXME: should this be removed, or moved into loop above? */
615 }
616
617
618 /**
619 * function for decoding mono data
620 *
621 * @param q pointer to the COOKContext
622 * @param mlt_buffer pointer to mlt coefficients
623 */
624
625 static void mono_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer) {
626
627 int category_index[128];
628 int quant_index_table[102];
629 int category[128];
630
631 memset(&category, 0, sizeof(category));
632 memset(&category_index, 0, sizeof(category_index));
633
634 decode_envelope(q, p, quant_index_table);
635 q->num_vectors = get_bits(&q->gb,p->log2_numvector_size);
636 categorize(q, p, quant_index_table, category, category_index);
637 expand_category(q, category, category_index);
638 decode_vectors(q, p, category, quant_index_table, mlt_buffer);
639 }
640
641
642 /**
643 * the actual requantization of the timedomain samples
644 *
645 * @param q pointer to the COOKContext
646 * @param buffer pointer to the timedomain buffer
647 * @param gain_index index for the block multiplier
648 * @param gain_index_next index for the next block multiplier
649 */
650
651 static void interpolate_float(COOKContext *q, float* buffer,
652 int gain_index, int gain_index_next){
653 int i;
654 float fc1, fc2;
655 fc1 = pow2tab[gain_index+63];
656
657 if(gain_index == gain_index_next){ //static gain
658 for(i=0 ; i<q->gain_size_factor ; i++){
659 buffer[i]*=fc1;
660 }
661 } else { //smooth gain
662 fc2 = q->gain_table[11 + (gain_index_next-gain_index)];
663 for(i=0 ; i<q->gain_size_factor ; i++){
664 buffer[i]*=fc1;
665 fc1*=fc2;
666 }
667 }
668 }
669
670 /**
671 * Apply transform window, overlap buffers.
672 *
673 * @param q pointer to the COOKContext
674 * @param inbuffer pointer to the mltcoefficients
675 * @param gains_ptr current and previous gains
676 * @param previous_buffer pointer to the previous buffer to be used for overlapping
677 */
678
679 static void imlt_window_float (COOKContext *q, float *inbuffer,
680 cook_gains *gains_ptr, float *previous_buffer)
681 {
682 const float fc = pow2tab[gains_ptr->previous[0] + 63];
683 int i;
684 /* The weird thing here, is that the two halves of the time domain
685 * buffer are swapped. Also, the newest data, that we save away for
686 * next frame, has the wrong sign. Hence the subtraction below.
687 * Almost sounds like a complex conjugate/reverse data/FFT effect.
688 */
689
690 /* Apply window and overlap */
691 for(i = 0; i < q->samples_per_channel; i++){
692 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
693 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
694 }
695 }
696
697 /**
698 * The modulated lapped transform, this takes transform coefficients
699 * and transforms them into timedomain samples.
700 * Apply transform window, overlap buffers, apply gain profile
701 * and buffer management.
702 *
703 * @param q pointer to the COOKContext
704 * @param inbuffer pointer to the mltcoefficients
705 * @param gains_ptr current and previous gains
706 * @param previous_buffer pointer to the previous buffer to be used for overlapping
707 */
708
709 static void imlt_gain(COOKContext *q, float *inbuffer,
710 cook_gains *gains_ptr, float* previous_buffer)
711 {
712 float *buffer0 = q->mono_mdct_output;
713 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
714 int i;
715
716 /* Inverse modified discrete cosine transform */
717 q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
718
719 q->imlt_window (q, buffer1, gains_ptr, previous_buffer);
720
721 /* Apply gain profile */
722 for (i = 0; i < 8; i++) {
723 if (gains_ptr->now[i] || gains_ptr->now[i + 1])
724 q->interpolate(q, &buffer1[q->gain_size_factor * i],
725 gains_ptr->now[i], gains_ptr->now[i + 1]);
726 }
727
728 /* Save away the current to be previous block. */
729 memcpy(previous_buffer, buffer0,
730 q->samples_per_channel * sizeof(*previous_buffer));
731 }
732
733
734 /**
735 * function for getting the jointstereo coupling information
736 *
737 * @param q pointer to the COOKContext
738 * @param decouple_tab decoupling array
739 *
740 */
741 static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
742 {
743 int i;
744 int vlc = get_bits1(&q->gb);
745 int start = cplband[p->js_subband_start];
746 int end = cplband[p->subbands-1];
747 int length = end - start + 1;
748
749 if (start > end)
750 return;
751
752 if (vlc) {
753 for (i = 0; i < length; i++)
754 decouple_tab[start + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
755 } else {
756 for (i = 0; i < length; i++)
757 decouple_tab[start + i] = get_bits(&q->gb, p->js_vlc_bits);
758 }
759 }
760
761 /*
762 * function decouples a pair of signals from a single signal via multiplication.
763 *
764 * @param q pointer to the COOKContext
765 * @param subband index of the current subband
766 * @param f1 multiplier for channel 1 extraction
767 * @param f2 multiplier for channel 2 extraction
768 * @param decode_buffer input buffer
769 * @param mlt_buffer1 pointer to left channel mlt coefficients
770 * @param mlt_buffer2 pointer to right channel mlt coefficients
771 */
772 static void decouple_float (COOKContext *q,
773 COOKSubpacket *p,
774 int subband,
775 float f1, float f2,
776 float *decode_buffer,
777 float *mlt_buffer1, float *mlt_buffer2)
778 {
779 int j, tmp_idx;
780 for (j=0 ; j<SUBBAND_SIZE ; j++) {
781 tmp_idx = ((p->js_subband_start + subband)*SUBBAND_SIZE)+j;
782 mlt_buffer1[SUBBAND_SIZE*subband + j] = f1 * decode_buffer[tmp_idx];
783 mlt_buffer2[SUBBAND_SIZE*subband + j] = f2 * decode_buffer[tmp_idx];
784 }
785 }
786
787 /**
788 * function for decoding joint stereo data
789 *
790 * @param q pointer to the COOKContext
791 * @param mlt_buffer1 pointer to left channel mlt coefficients
792 * @param mlt_buffer2 pointer to right channel mlt coefficients
793 */
794
795 static void joint_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer1,
796 float* mlt_buffer2) {
797 int i,j;
798 int decouple_tab[SUBBAND_SIZE];
799 float *decode_buffer = q->decode_buffer_0;
800 int idx, cpl_tmp;
801 float f1,f2;
802 const float* cplscale;
803
804 memset(decouple_tab, 0, sizeof(decouple_tab));
805 memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
806
807 /* Make sure the buffers are zeroed out. */
808 memset(mlt_buffer1, 0, 1024 * sizeof(*mlt_buffer1));
809 memset(mlt_buffer2, 0, 1024 * sizeof(*mlt_buffer2));
810 decouple_info(q, p, decouple_tab);
811 mono_decode(q, p, decode_buffer);
812
813 /* The two channels are stored interleaved in decode_buffer. */
814 for (i=0 ; i<p->js_subband_start ; i++) {
815 for (j=0 ; j<SUBBAND_SIZE ; j++) {
816 mlt_buffer1[i*20+j] = decode_buffer[i*40+j];
817 mlt_buffer2[i*20+j] = decode_buffer[i*40+20+j];
818 }
819 }
820
821 /* When we reach js_subband_start (the higher frequencies)
822 the coefficients are stored in a coupling scheme. */
823 idx = (1 << p->js_vlc_bits) - 1;
824 for (i=p->js_subband_start ; i<p->subbands ; i++) {
825 cpl_tmp = cplband[i];
826 idx -=decouple_tab[cpl_tmp];
827 cplscale = q->cplscales[p->js_vlc_bits-2]; //choose decoupler table
828 f1 = cplscale[decouple_tab[cpl_tmp]];
829 f2 = cplscale[idx-1];
830 q->decouple (q, p, i, f1, f2, decode_buffer, mlt_buffer1, mlt_buffer2);
831 idx = (1 << p->js_vlc_bits) - 1;
832 }
833 }
834
835 /**
836 * First part of subpacket decoding:
837 * decode raw stream bytes and read gain info.
838 *
839 * @param q pointer to the COOKContext
840 * @param inbuffer pointer to raw stream data
841 * @param gains_ptr array of current/prev gain pointers
842 */
843
844 static inline void
845 decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer,
846 cook_gains *gains_ptr)
847 {
848 int offset;
849
850 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
851 p->bits_per_subpacket/8);
852 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
853 p->bits_per_subpacket);
854 decode_gain_info(&q->gb, gains_ptr->now);
855
856 /* Swap current and previous gains */
857 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
858 }
859
860 /**
861 * Saturate the output signal and interleave.
862 *
863 * @param q pointer to the COOKContext
864 * @param chan channel to saturate
865 * @param out pointer to the output vector
866 */
867 static void saturate_output_float(COOKContext *q, int chan, float *out)
868 {
869 int j;
870 float *output = q->mono_mdct_output + q->samples_per_channel;
871 for (j = 0; j < q->samples_per_channel; j++) {
872 out[chan + q->nb_channels * j] = av_clipf(output[j], -1.0, 1.0);
873 }
874 }
875
876 /**
877 * Final part of subpacket decoding:
878 * Apply modulated lapped transform, gain compensation,
879 * clip and convert to integer.
880 *
881 * @param q pointer to the COOKContext
882 * @param decode_buffer pointer to the mlt coefficients
883 * @param gains_ptr array of current/prev gain pointers
884 * @param previous_buffer pointer to the previous buffer to be used for overlapping
885 * @param out pointer to the output buffer
886 * @param chan 0: left or single channel, 1: right channel
887 */
888
889 static inline void
890 mlt_compensate_output(COOKContext *q, float *decode_buffer,
891 cook_gains *gains_ptr, float *previous_buffer,
892 float *out, int chan)
893 {
894 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
895 q->saturate_output (q, chan, out);
896 }
897
898
899 /**
900 * Cook subpacket decoding. This function returns one decoded subpacket,
901 * usually 1024 samples per channel.
902 *
903 * @param q pointer to the COOKContext
904 * @param inbuffer pointer to the inbuffer
905 * @param outbuffer pointer to the outbuffer
906 */
907 static void decode_subpacket(COOKContext *q, COOKSubpacket *p,
908 const uint8_t *inbuffer, float *outbuffer)
909 {
910 int sub_packet_size = p->size;
911 /* packet dump */
912 // for (i=0 ; i<sub_packet_size ; i++) {
913 // av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]);
914 // }
915 // av_log(q->avctx, AV_LOG_ERROR, "\n");
916 memset(q->decode_buffer_1,0,sizeof(q->decode_buffer_1));
917 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
918
919 if (p->joint_stereo) {
920 joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2);
921 } else {
922 mono_decode(q, p, q->decode_buffer_1);
923
924 if (p->num_channels == 2) {
925 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size/2, &p->gains2);
926 mono_decode(q, p, q->decode_buffer_2);
927 }
928 }
929
930 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
931 p->mono_previous_buffer1, outbuffer, p->ch_idx);
932
933 if (p->num_channels == 2) {
934 if (p->joint_stereo) {
935 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
936 p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
937 } else {
938 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
939 p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
940 }
941 }
942
943 }
944
945
946 /**
947 * Cook frame decoding
948 *
949 * @param avctx pointer to the AVCodecContext
950 */
951
952 static int cook_decode_frame(AVCodecContext *avctx,
953 void *data, int *data_size,
954 AVPacket *avpkt) {
955 const uint8_t *buf = avpkt->data;
956 int buf_size = avpkt->size;
957 COOKContext *q = avctx->priv_data;
958 int i, out_size;
959 int offset = 0;
960 int chidx = 0;
961
962 if (buf_size < avctx->block_align)
963 return buf_size;
964
965 out_size = q->nb_channels * q->samples_per_channel *
966 av_get_bytes_per_sample(avctx->sample_fmt);
967 if (*data_size < out_size) {
968 av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
969 return AVERROR(EINVAL);
970 }
971
972 /* estimate subpacket sizes */
973 q->subpacket[0].size = avctx->block_align;
974
975 for(i=1;i<q->num_subpackets;i++){
976 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
977 q->subpacket[0].size -= q->subpacket[i].size + 1;
978 if (q->subpacket[0].size < 0) {
979 av_log(avctx,AV_LOG_DEBUG,"frame subpacket size total > avctx->block_align!\n");
980 return AVERROR_INVALIDDATA;
981 }
982 }
983
984 /* decode supbackets */
985 for(i=0;i<q->num_subpackets;i++){
986 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
987 q->subpacket[i].ch_idx = chidx;
988 av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
989 decode_subpacket(q, &q->subpacket[i], buf + offset, data);
990 offset += q->subpacket[i].size;
991 chidx += q->subpacket[i].num_channels;
992 av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
993 }
994 *data_size = out_size;
995
996 /* Discard the first two frames: no valid audio. */
997 if (avctx->frame_number < 2) *data_size = 0;
998
999 return avctx->block_align;
1000 }
1001
1002 #ifdef DEBUG
1003 static void dump_cook_context(COOKContext *q)
1004 {
1005 //int i=0;
1006 #define PRINT(a,b) av_log(q->avctx,AV_LOG_ERROR," %s = %d\n", a, b);
1007 av_log(q->avctx,AV_LOG_ERROR,"COOKextradata\n");
1008 av_log(q->avctx,AV_LOG_ERROR,"cookversion=%x\n",q->subpacket[0].cookversion);
1009 if (q->subpacket[0].cookversion > STEREO) {
1010 PRINT("js_subband_start",q->subpacket[0].js_subband_start);
1011 PRINT("js_vlc_bits",q->subpacket[0].js_vlc_bits);
1012 }
1013 av_log(q->avctx,AV_LOG_ERROR,"COOKContext\n");
1014 PRINT("nb_channels",q->nb_channels);
1015 PRINT("bit_rate",q->bit_rate);
1016 PRINT("sample_rate",q->sample_rate);
1017 PRINT("samples_per_channel",q->subpacket[0].samples_per_channel);
1018 PRINT("samples_per_frame",q->subpacket[0].samples_per_frame);
1019 PRINT("subbands",q->subpacket[0].subbands);
1020 PRINT("js_subband_start",q->subpacket[0].js_subband_start);
1021 PRINT("log2_numvector_size",q->subpacket[0].log2_numvector_size);
1022 PRINT("numvector_size",q->subpacket[0].numvector_size);
1023 PRINT("total_subbands",q->subpacket[0].total_subbands);
1024 }
1025 #endif
1026
1027 static av_cold int cook_count_channels(unsigned int mask){
1028 int i;
1029 int channels = 0;
1030 for(i = 0;i<32;i++){
1031 if(mask & (1<<i))
1032 ++channels;
1033 }
1034 return channels;
1035 }
1036
1037 /**
1038 * Cook initialization
1039 *
1040 * @param avctx pointer to the AVCodecContext
1041 */
1042
1043 static av_cold int cook_decode_init(AVCodecContext *avctx)
1044 {
1045 COOKContext *q = avctx->priv_data;
1046 const uint8_t *edata_ptr = avctx->extradata;
1047 const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
1048 int extradata_size = avctx->extradata_size;
1049 int s = 0;
1050 unsigned int channel_mask = 0;
1051 int ret;
1052 q->avctx = avctx;
1053
1054 /* Take care of the codec specific extradata. */
1055 if (extradata_size <= 0) {
1056 av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");
1057 return AVERROR_INVALIDDATA;
1058 }
1059 av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
1060
1061 /* Take data from the AVCodecContext (RM container). */
1062 q->sample_rate = avctx->sample_rate;
1063 q->nb_channels = avctx->channels;
1064 q->bit_rate = avctx->bit_rate;
1065
1066 /* Initialize RNG. */
1067 av_lfg_init(&q->random_state, 0);
1068
1069 while(edata_ptr < edata_ptr_end){
1070 /* 8 for mono, 16 for stereo, ? for multichannel
1071 Swap to right endianness so we don't need to care later on. */
1072 if (extradata_size >= 8){
1073 q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
1074 q->subpacket[s].samples_per_frame = bytestream_get_be16(&edata_ptr);
1075 q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
1076 extradata_size -= 8;
1077 }
1078 if (avctx->extradata_size >= 8){
1079 bytestream_get_be32(&edata_ptr); //Unknown unused
1080 q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
1081 q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
1082 extradata_size -= 8;
1083 }
1084
1085 /* Initialize extradata related variables. */
1086 q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame / q->nb_channels;
1087 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1088
1089 /* Initialize default data states. */
1090 q->subpacket[s].log2_numvector_size = 5;
1091 q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1092 q->subpacket[s].num_channels = 1;
1093
1094 /* Initialize version-dependent variables */
1095
1096 av_log(avctx,AV_LOG_DEBUG,"subpacket[%i].cookversion=%x\n",s,q->subpacket[s].cookversion);
1097 q->subpacket[s].joint_stereo = 0;
1098 switch (q->subpacket[s].cookversion) {
1099 case MONO:
1100 if (q->nb_channels != 1) {
1101 av_log_ask_for_sample(avctx, "Container channels != 1.\n");
1102 return AVERROR(ENOTSUP);
1103 }
1104 av_log(avctx,AV_LOG_DEBUG,"MONO\n");
1105 break;
1106 case STEREO:
1107 if (q->nb_channels != 1) {
1108 q->subpacket[s].bits_per_subpdiv = 1;
1109 q->subpacket[s].num_channels = 2;
1110 }
1111 av_log(avctx,AV_LOG_DEBUG,"STEREO\n");
1112 break;
1113 case JOINT_STEREO:
1114 if (q->nb_channels != 2) {
1115 av_log_ask_for_sample(avctx, "Container channels != 2.\n");
1116 return AVERROR(ENOTSUP);
1117 }
1118 av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
1119 if (avctx->extradata_size >= 16){
1120 q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
1121 q->subpacket[s].joint_stereo = 1;
1122 q->subpacket[s].num_channels = 2;
1123 }
1124 if (q->subpacket[s].samples_per_channel > 256) {
1125 q->subpacket[s].log2_numvector_size = 6;
1126 }
1127 if (q->subpacket[s].samples_per_channel > 512) {
1128 q->subpacket[s].log2_numvector_size = 7;
1129 }
1130 break;
1131 case MC_COOK:
1132 av_log(avctx,AV_LOG_DEBUG,"MULTI_CHANNEL\n");
1133 if(extradata_size >= 4)
1134 channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
1135
1136 if(cook_count_channels(q->subpacket[s].channel_mask) > 1){
1137 q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
1138 q->subpacket[s].joint_stereo = 1;
1139 q->subpacket[s].num_channels = 2;
1140 q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame >> 1;
1141
1142 if (q->subpacket[s].samples_per_channel > 256) {
1143 q->subpacket[s].log2_numvector_size = 6;
1144 }
1145 if (q->subpacket[s].samples_per_channel > 512) {
1146 q->subpacket[s].log2_numvector_size = 7;
1147 }
1148 }else
1149 q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame;
1150
1151 break;
1152 default:
1153 av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
1154 return AVERROR(ENOTSUP);
1155 }
1156
1157 if(s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1158 av_log(avctx,AV_LOG_ERROR,"different number of samples per channel!\n");
1159 return AVERROR_INVALIDDATA;
1160 } else
1161 q->samples_per_channel = q->subpacket[0].samples_per_channel;
1162
1163
1164 /* Initialize variable relations */
1165 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1166
1167 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1168 if (q->subpacket[s].total_subbands > 53) {
1169 av_log_ask_for_sample(avctx, "total_subbands > 53\n");
1170 return AVERROR(ENOTSUP);
1171 }
1172
1173 if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 2*q->subpacket[s].joint_stereo)) {
1174 av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1175 q->subpacket[s].js_vlc_bits, 2*q->subpacket[s].joint_stereo);
1176 return AVERROR_INVALIDDATA;
1177 }
1178
1179 if (q->subpacket[s].subbands > 50) {
1180 av_log_ask_for_sample(avctx, "subbands > 50\n");
1181 return AVERROR(ENOTSUP);
1182 }
1183 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1184 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1185 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1186 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1187
1188 q->num_subpackets++;
1189 s++;
1190 if (s > MAX_SUBPACKETS) {
1191 av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
1192 return AVERROR(ENOTSUP);
1193 }
1194 }
1195 /* Generate tables */
1196 init_pow2table();
1197 init_gain_table(q);
1198 init_cplscales_table(q);
1199
1200 if ((ret = init_cook_vlc_tables(q)))
1201 return ret;
1202
1203
1204 if(avctx->block_align >= UINT_MAX/2)
1205 return AVERROR(EINVAL);
1206
1207 /* Pad the databuffer with:
1208 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1209 FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1210 q->decoded_bytes_buffer =
1211 av_mallocz(avctx->block_align
1212 + DECODE_BYTES_PAD1(avctx->block_align)
1213 + FF_INPUT_BUFFER_PADDING_SIZE);
1214 if (q->decoded_bytes_buffer == NULL)
1215 return AVERROR(ENOMEM);
1216
1217 /* Initialize transform. */
1218 if ((ret = init_cook_mlt(q)))
1219 return ret;
1220
1221 /* Initialize COOK signal arithmetic handling */
1222 if (1) {
1223 q->scalar_dequant = scalar_dequant_float;
1224 q->decouple = decouple_float;
1225 q->imlt_window = imlt_window_float;
1226 q->interpolate = interpolate_float;
1227 q->saturate_output = saturate_output_float;
1228 }
1229
1230 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1231 if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) {
1232 } else {
1233 av_log_ask_for_sample(avctx,
1234 "unknown amount of samples_per_channel = %d\n",
1235 q->samples_per_channel);
1236 return AVERROR(ENOTSUP);
1237 }
1238
1239 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
1240 if (channel_mask)
1241 avctx->channel_layout = channel_mask;
1242 else
1243 avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1244
1245 #ifdef DEBUG
1246 dump_cook_context(q);
1247 #endif
1248 return 0;
1249 }
1250
1251
1252 AVCodec ff_cook_decoder =
1253 {
1254 .name = "cook",
1255 .type = AVMEDIA_TYPE_AUDIO,
1256 .id = CODEC_ID_COOK,
1257 .priv_data_size = sizeof(COOKContext),
1258 .init = cook_decode_init,
1259 .close = cook_decode_close,
1260 .decode = cook_decode_frame,
1261 .long_name = NULL_IF_CONFIG_SMALL("COOK"),
1262 };