1 /*
2 * COOK compatible decoder
3 * Copyright (c) 2003 Sascha Sommer
4 * Copyright (c) 2005 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 * 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/channel_layout.h"
46 #include "libavutil/lfg.h"
47 #include "libavutil/mem_internal.h"
48 #include "libavutil/thread.h"
49
50 #include "audiodsp.h"
51 #include "avcodec.h"
52 #include "get_bits.h"
53 #include "bytestream.h"
54 #include "codec_internal.h"
55 #include "fft.h"
56 #include "internal.h"
57 #include "sinewin.h"
58 #include "unary.h"
59
60 #include "cookdata.h"
61
62 /* the different Cook versions */
63 #define MONO 0x1000001
64 #define STEREO 0x1000002
65 #define JOINT_STEREO 0x1000003
66 #define MC_COOK 0x2000000
67
68 #define SUBBAND_SIZE 20
69 #define MAX_SUBPACKETS 5
70
71 #define QUANT_VLC_BITS 9
72 #define COUPLING_VLC_BITS 6
73
74 typedef struct cook_gains {
75 int *now;
76 int *previous;
77 } cook_gains;
78
79 typedef struct COOKSubpacket {
80 int ch_idx;
81 int size;
82 int num_channels;
83 int cookversion;
84 int subbands;
85 int js_subband_start;
86 int js_vlc_bits;
87 int samples_per_channel;
88 int log2_numvector_size;
89 unsigned int channel_mask;
90 VLC channel_coupling;
91 int joint_stereo;
92 int bits_per_subpacket;
93 int bits_per_subpdiv;
94 int total_subbands;
95 int numvector_size; // 1 << log2_numvector_size;
96
97 float mono_previous_buffer1[1024];
98 float mono_previous_buffer2[1024];
99
100 cook_gains gains1;
101 cook_gains gains2;
102 int gain_1[9];
103 int gain_2[9];
104 int gain_3[9];
105 int gain_4[9];
106 } COOKSubpacket;
107
108 typedef struct cook {
109 /*
110 * The following 5 functions provide the lowlevel arithmetic on
111 * the internal audio buffers.
112 */
113 void (*scalar_dequant)(struct cook *q, int index, int quant_index,
114 int *subband_coef_index, int *subband_coef_sign,
115 float *mlt_p);
116
117 void (*decouple)(struct cook *q,
118 COOKSubpacket *p,
119 int subband,
120 float f1, float f2,
121 float *decode_buffer,
122 float *mlt_buffer1, float *mlt_buffer2);
123
124 void (*imlt_window)(struct cook *q, float *buffer1,
125 cook_gains *gains_ptr, float *previous_buffer);
126
127 void (*interpolate)(struct cook *q, float *buffer,
128 int gain_index, int gain_index_next);
129
130 void (*saturate_output)(struct cook *q, float *out);
131
132 AVCodecContext* avctx;
133 AudioDSPContext adsp;
134 GetBitContext gb;
135 /* stream data */
136 int num_vectors;
137 int samples_per_channel;
138 /* states */
139 AVLFG random_state;
140 int discarded_packets;
141
142 /* transform data */
143 FFTContext mdct_ctx;
144 float* mlt_window;
145
146 /* VLC data */
147 VLC envelope_quant_index[13];
148 VLC sqvh[7]; // scalar quantization
149
150 /* generate tables and related variables */
151 int gain_size_factor;
152 float gain_table[31];
153
154 /* data buffers */
155
156 uint8_t* decoded_bytes_buffer;
157 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
158 float decode_buffer_1[1024];
159 float decode_buffer_2[1024];
160 float decode_buffer_0[1060]; /* static allocation for joint decode */
161
162 const float *cplscales[5];
163 int num_subpackets;
164 COOKSubpacket subpacket[MAX_SUBPACKETS];
165 } COOKContext;
166
167 static float pow2tab[127];
168 static float rootpow2tab[127];
169
170 /*************** init functions ***************/
171
172 /* table generator */
init_pow2table(void)173 static av_cold void init_pow2table(void)
174 {
175 /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
176 int i;
177 static const float exp2_tab[2] = {1, M_SQRT2};
178 float exp2_val = powf(2, -63);
179 float root_val = powf(2, -32);
180 for (i = -63; i < 64; i++) {
181 if (!(i & 1))
182 root_val *= 2;
183 pow2tab[63 + i] = exp2_val;
184 rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
185 exp2_val *= 2;
186 }
187 }
188
189 /* table generator */
init_gain_table(COOKContext * q)190 static av_cold void init_gain_table(COOKContext *q)
191 {
192 int i;
193 q->gain_size_factor = q->samples_per_channel / 8;
194 for (i = 0; i < 31; i++)
195 q->gain_table[i] = pow(pow2tab[i + 48],
196 (1.0 / (double) q->gain_size_factor));
197 }
198
build_vlc(VLC * vlc,int nb_bits,const uint8_t counts[16],const void * syms,int symbol_size,int offset,void * logctx)199 static av_cold int build_vlc(VLC *vlc, int nb_bits, const uint8_t counts[16],
200 const void *syms, int symbol_size, int offset,
201 void *logctx)
202 {
203 uint8_t lens[MAX_COOK_VLC_ENTRIES];
204 unsigned num = 0;
205
206 for (int i = 0; i < 16; i++)
207 for (unsigned count = num + counts[i]; num < count; num++)
208 lens[num] = i + 1;
209
210 return ff_init_vlc_from_lengths(vlc, nb_bits, num, lens, 1,
211 syms, symbol_size, symbol_size,
212 offset, 0, logctx);
213 }
214
init_cook_vlc_tables(COOKContext * q)215 static av_cold int init_cook_vlc_tables(COOKContext *q)
216 {
217 int i, result;
218
219 result = 0;
220 for (i = 0; i < 13; i++) {
221 result |= build_vlc(&q->envelope_quant_index[i], QUANT_VLC_BITS,
222 envelope_quant_index_huffcounts[i],
223 envelope_quant_index_huffsyms[i], 1, -12, q->avctx);
224 }
225 av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
226 for (i = 0; i < 7; i++) {
227 int sym_size = 1 + (i == 3);
228 result |= build_vlc(&q->sqvh[i], vhvlcsize_tab[i],
229 cvh_huffcounts[i],
230 cvh_huffsyms[i], sym_size, 0, q->avctx);
231 }
232
233 for (i = 0; i < q->num_subpackets; i++) {
234 if (q->subpacket[i].joint_stereo == 1) {
235 result |= build_vlc(&q->subpacket[i].channel_coupling, COUPLING_VLC_BITS,
236 ccpl_huffcounts[q->subpacket[i].js_vlc_bits - 2],
237 ccpl_huffsyms[q->subpacket[i].js_vlc_bits - 2], 1,
238 0, q->avctx);
239 av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
240 }
241 }
242
243 av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
244 return result;
245 }
246
init_cook_mlt(COOKContext * q)247 static av_cold int init_cook_mlt(COOKContext *q)
248 {
249 int j, ret;
250 int mlt_size = q->samples_per_channel;
251
252 if (!(q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))))
253 return AVERROR(ENOMEM);
254
255 /* Initialize the MLT window: simple sine window. */
256 ff_sine_window_init(q->mlt_window, mlt_size);
257 for (j = 0; j < mlt_size; j++)
258 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
259
260 /* Initialize the MDCT. */
261 ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0);
262 if (ret < 0)
263 return ret;
264 av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
265 av_log2(mlt_size) + 1);
266
267 return 0;
268 }
269
init_cplscales_table(COOKContext * q)270 static av_cold void init_cplscales_table(COOKContext *q)
271 {
272 int i;
273 for (i = 0; i < 5; i++)
274 q->cplscales[i] = cplscales[i];
275 }
276
277 /*************** init functions end ***********/
278
279 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
280 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
281
282 /**
283 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
284 * Why? No idea, some checksum/error detection method maybe.
285 *
286 * Out buffer size: extra bytes are needed to cope with
287 * padding/misalignment.
288 * Subpackets passed to the decoder can contain two, consecutive
289 * half-subpackets, of identical but arbitrary size.
290 * 1234 1234 1234 1234 extraA extraB
291 * Case 1: AAAA BBBB 0 0
292 * Case 2: AAAA ABBB BB-- 3 3
293 * Case 3: AAAA AABB BBBB 2 2
294 * Case 4: AAAA AAAB BBBB BB-- 1 5
295 *
296 * Nice way to waste CPU cycles.
297 *
298 * @param inbuffer pointer to byte array of indata
299 * @param out pointer to byte array of outdata
300 * @param bytes number of bytes
301 */
decode_bytes(const uint8_t * inbuffer,uint8_t * out,int bytes)302 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
303 {
304 static const uint32_t tab[4] = {
305 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
306 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
307 };
308 int i, off;
309 uint32_t c;
310 const uint32_t *buf;
311 uint32_t *obuf = (uint32_t *) out;
312 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
313 * I'm too lazy though, should be something like
314 * for (i = 0; i < bitamount / 64; i++)
315 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
316 * Buffer alignment needs to be checked. */
317
318 off = (intptr_t) inbuffer & 3;
319 buf = (const uint32_t *) (inbuffer - off);
320 c = tab[off];
321 bytes += 3 + off;
322 for (i = 0; i < bytes / 4; i++)
323 obuf[i] = c ^ buf[i];
324
325 return off;
326 }
327
cook_decode_close(AVCodecContext * avctx)328 static av_cold int cook_decode_close(AVCodecContext *avctx)
329 {
330 int i;
331 COOKContext *q = avctx->priv_data;
332 av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
333
334 /* Free allocated memory buffers. */
335 av_freep(&q->mlt_window);
336 av_freep(&q->decoded_bytes_buffer);
337
338 /* Free the transform. */
339 ff_mdct_end(&q->mdct_ctx);
340
341 /* Free the VLC tables. */
342 for (i = 0; i < 13; i++)
343 ff_free_vlc(&q->envelope_quant_index[i]);
344 for (i = 0; i < 7; i++)
345 ff_free_vlc(&q->sqvh[i]);
346 for (i = 0; i < q->num_subpackets; i++)
347 ff_free_vlc(&q->subpacket[i].channel_coupling);
348
349 av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
350
351 return 0;
352 }
353
354 /**
355 * Fill the gain array for the timedomain quantization.
356 *
357 * @param gb pointer to the GetBitContext
358 * @param gaininfo array[9] of gain indexes
359 */
decode_gain_info(GetBitContext * gb,int * gaininfo)360 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
361 {
362 int i, n;
363
364 n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update
365
366 i = 0;
367 while (n--) {
368 int index = get_bits(gb, 3);
369 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
370
371 while (i <= index)
372 gaininfo[i++] = gain;
373 }
374 while (i <= 8)
375 gaininfo[i++] = 0;
376 }
377
378 /**
379 * Create the quant index table needed for the envelope.
380 *
381 * @param q pointer to the COOKContext
382 * @param quant_index_table pointer to the array
383 */
decode_envelope(COOKContext * q,COOKSubpacket * p,int * quant_index_table)384 static int decode_envelope(COOKContext *q, COOKSubpacket *p,
385 int *quant_index_table)
386 {
387 int i, j, vlc_index;
388
389 quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
390
391 for (i = 1; i < p->total_subbands; i++) {
392 vlc_index = i;
393 if (i >= p->js_subband_start * 2) {
394 vlc_index -= p->js_subband_start;
395 } else {
396 vlc_index /= 2;
397 if (vlc_index < 1)
398 vlc_index = 1;
399 }
400 if (vlc_index > 13)
401 vlc_index = 13; // the VLC tables >13 are identical to No. 13
402
403 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
404 QUANT_VLC_BITS, 2);
405 quant_index_table[i] = quant_index_table[i - 1] + j; // differential encoding
406 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
407 av_log(q->avctx, AV_LOG_ERROR,
408 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
409 quant_index_table[i], i);
410 return AVERROR_INVALIDDATA;
411 }
412 }
413
414 return 0;
415 }
416
417 /**
418 * Calculate the category and category_index vector.
419 *
420 * @param q pointer to the COOKContext
421 * @param quant_index_table pointer to the array
422 * @param category pointer to the category array
423 * @param category_index pointer to the category_index array
424 */
categorize(COOKContext * q,COOKSubpacket * p,const int * quant_index_table,int * category,int * category_index)425 static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
426 int *category, int *category_index)
427 {
428 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
429 int exp_index2[102] = { 0 };
430 int exp_index1[102] = { 0 };
431
432 int tmp_categorize_array[128 * 2] = { 0 };
433 int tmp_categorize_array1_idx = p->numvector_size;
434 int tmp_categorize_array2_idx = p->numvector_size;
435
436 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
437
438 if (bits_left > q->samples_per_channel)
439 bits_left = q->samples_per_channel +
440 ((bits_left - q->samples_per_channel) * 5) / 8;
441
442 bias = -32;
443
444 /* Estimate bias. */
445 for (i = 32; i > 0; i = i / 2) {
446 num_bits = 0;
447 index = 0;
448 for (j = p->total_subbands; j > 0; j--) {
449 exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
450 index++;
451 num_bits += expbits_tab[exp_idx];
452 }
453 if (num_bits >= bits_left - 32)
454 bias += i;
455 }
456
457 /* Calculate total number of bits. */
458 num_bits = 0;
459 for (i = 0; i < p->total_subbands; i++) {
460 exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
461 num_bits += expbits_tab[exp_idx];
462 exp_index1[i] = exp_idx;
463 exp_index2[i] = exp_idx;
464 }
465 tmpbias1 = tmpbias2 = num_bits;
466
467 for (j = 1; j < p->numvector_size; j++) {
468 if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
469 int max = -999999;
470 index = -1;
471 for (i = 0; i < p->total_subbands; i++) {
472 if (exp_index1[i] < 7) {
473 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
474 if (v >= max) {
475 max = v;
476 index = i;
477 }
478 }
479 }
480 if (index == -1)
481 break;
482 tmp_categorize_array[tmp_categorize_array1_idx++] = index;
483 tmpbias1 -= expbits_tab[exp_index1[index]] -
484 expbits_tab[exp_index1[index] + 1];
485 ++exp_index1[index];
486 } else { /* <--- */
487 int min = 999999;
488 index = -1;
489 for (i = 0; i < p->total_subbands; i++) {
490 if (exp_index2[i] > 0) {
491 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
492 if (v < min) {
493 min = v;
494 index = i;
495 }
496 }
497 }
498 if (index == -1)
499 break;
500 tmp_categorize_array[--tmp_categorize_array2_idx] = index;
501 tmpbias2 -= expbits_tab[exp_index2[index]] -
502 expbits_tab[exp_index2[index] - 1];
503 --exp_index2[index];
504 }
505 }
506
507 for (i = 0; i < p->total_subbands; i++)
508 category[i] = exp_index2[i];
509
510 for (i = 0; i < p->numvector_size - 1; i++)
511 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
512 }
513
514
515 /**
516 * Expand the category vector.
517 *
518 * @param q pointer to the COOKContext
519 * @param category pointer to the category array
520 * @param category_index pointer to the category_index array
521 */
expand_category(COOKContext * q,int * category,int * category_index)522 static inline void expand_category(COOKContext *q, int *category,
523 int *category_index)
524 {
525 int i;
526 for (i = 0; i < q->num_vectors; i++)
527 {
528 int idx = category_index[i];
529 if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
530 --category[idx];
531 }
532 }
533
534 /**
535 * The real requantization of the mltcoefs
536 *
537 * @param q pointer to the COOKContext
538 * @param index index
539 * @param quant_index quantisation index
540 * @param subband_coef_index array of indexes to quant_centroid_tab
541 * @param subband_coef_sign signs of coefficients
542 * @param mlt_p pointer into the mlt buffer
543 */
scalar_dequant_float(COOKContext * q,int index,int quant_index,int * subband_coef_index,int * subband_coef_sign,float * mlt_p)544 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
545 int *subband_coef_index, int *subband_coef_sign,
546 float *mlt_p)
547 {
548 int i;
549 float f1;
550
551 for (i = 0; i < SUBBAND_SIZE; i++) {
552 if (subband_coef_index[i]) {
553 f1 = quant_centroid_tab[index][subband_coef_index[i]];
554 if (subband_coef_sign[i])
555 f1 = -f1;
556 } else {
557 /* noise coding if subband_coef_index[i] == 0 */
558 f1 = dither_tab[index];
559 if (av_lfg_get(&q->random_state) < 0x80000000)
560 f1 = -f1;
561 }
562 mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
563 }
564 }
565 /**
566 * Unpack the subband_coef_index and subband_coef_sign vectors.
567 *
568 * @param q pointer to the COOKContext
569 * @param category pointer to the category array
570 * @param subband_coef_index array of indexes to quant_centroid_tab
571 * @param subband_coef_sign signs of coefficients
572 */
unpack_SQVH(COOKContext * q,COOKSubpacket * p,int category,int * subband_coef_index,int * subband_coef_sign)573 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
574 int *subband_coef_index, int *subband_coef_sign)
575 {
576 int i, j;
577 int vlc, vd, tmp, result;
578
579 vd = vd_tab[category];
580 result = 0;
581 for (i = 0; i < vpr_tab[category]; i++) {
582 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
583 if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
584 vlc = 0;
585 result = 1;
586 }
587 for (j = vd - 1; j >= 0; j--) {
588 tmp = (vlc * invradix_tab[category]) / 0x100000;
589 subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
590 vlc = tmp;
591 }
592 for (j = 0; j < vd; j++) {
593 if (subband_coef_index[i * vd + j]) {
594 if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
595 subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
596 } else {
597 result = 1;
598 subband_coef_sign[i * vd + j] = 0;
599 }
600 } else {
601 subband_coef_sign[i * vd + j] = 0;
602 }
603 }
604 }
605 return result;
606 }
607
608
609 /**
610 * Fill the mlt_buffer with mlt coefficients.
611 *
612 * @param q pointer to the COOKContext
613 * @param category pointer to the category array
614 * @param quant_index_table pointer to the array
615 * @param mlt_buffer pointer to mlt coefficients
616 */
decode_vectors(COOKContext * q,COOKSubpacket * p,int * category,int * quant_index_table,float * mlt_buffer)617 static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
618 int *quant_index_table, float *mlt_buffer)
619 {
620 /* A zero in this table means that the subband coefficient is
621 random noise coded. */
622 int subband_coef_index[SUBBAND_SIZE];
623 /* A zero in this table means that the subband coefficient is a
624 positive multiplicator. */
625 int subband_coef_sign[SUBBAND_SIZE];
626 int band, j;
627 int index = 0;
628
629 for (band = 0; band < p->total_subbands; band++) {
630 index = category[band];
631 if (category[band] < 7) {
632 if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
633 index = 7;
634 for (j = 0; j < p->total_subbands; j++)
635 category[band + j] = 7;
636 }
637 }
638 if (index >= 7) {
639 memset(subband_coef_index, 0, sizeof(subband_coef_index));
640 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
641 }
642 q->scalar_dequant(q, index, quant_index_table[band],
643 subband_coef_index, subband_coef_sign,
644 &mlt_buffer[band * SUBBAND_SIZE]);
645 }
646
647 /* FIXME: should this be removed, or moved into loop above? */
648 if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
649 return;
650 }
651
652
mono_decode(COOKContext * q,COOKSubpacket * p,float * mlt_buffer)653 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
654 {
655 int category_index[128] = { 0 };
656 int category[128] = { 0 };
657 int quant_index_table[102];
658 int res, i;
659
660 if ((res = decode_envelope(q, p, quant_index_table)) < 0)
661 return res;
662 q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
663 categorize(q, p, quant_index_table, category, category_index);
664 expand_category(q, category, category_index);
665 for (i=0; i<p->total_subbands; i++) {
666 if (category[i] > 7)
667 return AVERROR_INVALIDDATA;
668 }
669 decode_vectors(q, p, category, quant_index_table, mlt_buffer);
670
671 return 0;
672 }
673
674
675 /**
676 * the actual requantization of the timedomain samples
677 *
678 * @param q pointer to the COOKContext
679 * @param buffer pointer to the timedomain buffer
680 * @param gain_index index for the block multiplier
681 * @param gain_index_next index for the next block multiplier
682 */
interpolate_float(COOKContext * q,float * buffer,int gain_index,int gain_index_next)683 static void interpolate_float(COOKContext *q, float *buffer,
684 int gain_index, int gain_index_next)
685 {
686 int i;
687 float fc1, fc2;
688 fc1 = pow2tab[gain_index + 63];
689
690 if (gain_index == gain_index_next) { // static gain
691 for (i = 0; i < q->gain_size_factor; i++)
692 buffer[i] *= fc1;
693 } else { // smooth gain
694 fc2 = q->gain_table[15 + (gain_index_next - gain_index)];
695 for (i = 0; i < q->gain_size_factor; i++) {
696 buffer[i] *= fc1;
697 fc1 *= fc2;
698 }
699 }
700 }
701
702 /**
703 * Apply transform window, overlap buffers.
704 *
705 * @param q pointer to the COOKContext
706 * @param inbuffer pointer to the mltcoefficients
707 * @param gains_ptr current and previous gains
708 * @param previous_buffer pointer to the previous buffer to be used for overlapping
709 */
imlt_window_float(COOKContext * q,float * inbuffer,cook_gains * gains_ptr,float * previous_buffer)710 static void imlt_window_float(COOKContext *q, float *inbuffer,
711 cook_gains *gains_ptr, float *previous_buffer)
712 {
713 const float fc = pow2tab[gains_ptr->previous[0] + 63];
714 int i;
715 /* The weird thing here, is that the two halves of the time domain
716 * buffer are swapped. Also, the newest data, that we save away for
717 * next frame, has the wrong sign. Hence the subtraction below.
718 * Almost sounds like a complex conjugate/reverse data/FFT effect.
719 */
720
721 /* Apply window and overlap */
722 for (i = 0; i < q->samples_per_channel; i++)
723 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
724 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
725 }
726
727 /**
728 * The modulated lapped transform, this takes transform coefficients
729 * and transforms them into timedomain samples.
730 * Apply transform window, overlap buffers, apply gain profile
731 * and buffer management.
732 *
733 * @param q pointer to the COOKContext
734 * @param inbuffer pointer to the mltcoefficients
735 * @param gains_ptr current and previous gains
736 * @param previous_buffer pointer to the previous buffer to be used for overlapping
737 */
imlt_gain(COOKContext * q,float * inbuffer,cook_gains * gains_ptr,float * previous_buffer)738 static void imlt_gain(COOKContext *q, float *inbuffer,
739 cook_gains *gains_ptr, float *previous_buffer)
740 {
741 float *buffer0 = q->mono_mdct_output;
742 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
743 int i;
744
745 /* Inverse modified discrete cosine transform */
746 q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
747
748 q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
749
750 /* Apply gain profile */
751 for (i = 0; i < 8; i++)
752 if (gains_ptr->now[i] || gains_ptr->now[i + 1])
753 q->interpolate(q, &buffer1[q->gain_size_factor * i],
754 gains_ptr->now[i], gains_ptr->now[i + 1]);
755
756 /* Save away the current to be previous block. */
757 memcpy(previous_buffer, buffer0,
758 q->samples_per_channel * sizeof(*previous_buffer));
759 }
760
761
762 /**
763 * function for getting the jointstereo coupling information
764 *
765 * @param q pointer to the COOKContext
766 * @param decouple_tab decoupling array
767 */
decouple_info(COOKContext * q,COOKSubpacket * p,int * decouple_tab)768 static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
769 {
770 int i;
771 int vlc = get_bits1(&q->gb);
772 int start = cplband[p->js_subband_start];
773 int end = cplband[p->subbands - 1];
774 int length = end - start + 1;
775
776 if (start > end)
777 return 0;
778
779 if (vlc)
780 for (i = 0; i < length; i++)
781 decouple_tab[start + i] = get_vlc2(&q->gb,
782 p->channel_coupling.table,
783 COUPLING_VLC_BITS, 3);
784 else
785 for (i = 0; i < length; i++) {
786 int v = get_bits(&q->gb, p->js_vlc_bits);
787 if (v == (1<<p->js_vlc_bits)-1) {
788 av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
789 return AVERROR_INVALIDDATA;
790 }
791 decouple_tab[start + i] = v;
792 }
793 return 0;
794 }
795
796 /**
797 * function decouples a pair of signals from a single signal via multiplication.
798 *
799 * @param q pointer to the COOKContext
800 * @param subband index of the current subband
801 * @param f1 multiplier for channel 1 extraction
802 * @param f2 multiplier for channel 2 extraction
803 * @param decode_buffer input buffer
804 * @param mlt_buffer1 pointer to left channel mlt coefficients
805 * @param mlt_buffer2 pointer to right channel mlt coefficients
806 */
decouple_float(COOKContext * q,COOKSubpacket * p,int subband,float f1,float f2,float * decode_buffer,float * mlt_buffer1,float * mlt_buffer2)807 static void decouple_float(COOKContext *q,
808 COOKSubpacket *p,
809 int subband,
810 float f1, float f2,
811 float *decode_buffer,
812 float *mlt_buffer1, float *mlt_buffer2)
813 {
814 int j, tmp_idx;
815 for (j = 0; j < SUBBAND_SIZE; j++) {
816 tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
817 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
818 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
819 }
820 }
821
822 /**
823 * function for decoding joint stereo data
824 *
825 * @param q pointer to the COOKContext
826 * @param mlt_buffer1 pointer to left channel mlt coefficients
827 * @param mlt_buffer2 pointer to right channel mlt coefficients
828 */
joint_decode(COOKContext * q,COOKSubpacket * p,float * mlt_buffer_left,float * mlt_buffer_right)829 static int joint_decode(COOKContext *q, COOKSubpacket *p,
830 float *mlt_buffer_left, float *mlt_buffer_right)
831 {
832 int i, j, res;
833 int decouple_tab[SUBBAND_SIZE] = { 0 };
834 float *decode_buffer = q->decode_buffer_0;
835 int idx, cpl_tmp;
836 float f1, f2;
837 const float *cplscale;
838
839 memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
840
841 /* Make sure the buffers are zeroed out. */
842 memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
843 memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
844 if ((res = decouple_info(q, p, decouple_tab)) < 0)
845 return res;
846 if ((res = mono_decode(q, p, decode_buffer)) < 0)
847 return res;
848 /* The two channels are stored interleaved in decode_buffer. */
849 for (i = 0; i < p->js_subband_start; i++) {
850 for (j = 0; j < SUBBAND_SIZE; j++) {
851 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
852 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
853 }
854 }
855
856 /* When we reach js_subband_start (the higher frequencies)
857 the coefficients are stored in a coupling scheme. */
858 idx = (1 << p->js_vlc_bits) - 1;
859 for (i = p->js_subband_start; i < p->subbands; i++) {
860 cpl_tmp = cplband[i];
861 idx -= decouple_tab[cpl_tmp];
862 cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
863 f1 = cplscale[decouple_tab[cpl_tmp] + 1];
864 f2 = cplscale[idx];
865 q->decouple(q, p, i, f1, f2, decode_buffer,
866 mlt_buffer_left, mlt_buffer_right);
867 idx = (1 << p->js_vlc_bits) - 1;
868 }
869
870 return 0;
871 }
872
873 /**
874 * First part of subpacket decoding:
875 * decode raw stream bytes and read gain info.
876 *
877 * @param q pointer to the COOKContext
878 * @param inbuffer pointer to raw stream data
879 * @param gains_ptr array of current/prev gain pointers
880 */
decode_bytes_and_gain(COOKContext * q,COOKSubpacket * p,const uint8_t * inbuffer,cook_gains * gains_ptr)881 static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
882 const uint8_t *inbuffer,
883 cook_gains *gains_ptr)
884 {
885 int offset;
886
887 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
888 p->bits_per_subpacket / 8);
889 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
890 p->bits_per_subpacket);
891 decode_gain_info(&q->gb, gains_ptr->now);
892
893 /* Swap current and previous gains */
894 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
895 }
896
897 /**
898 * Saturate the output signal and interleave.
899 *
900 * @param q pointer to the COOKContext
901 * @param out pointer to the output vector
902 */
saturate_output_float(COOKContext * q,float * out)903 static void saturate_output_float(COOKContext *q, float *out)
904 {
905 q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
906 FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
907 }
908
909
910 /**
911 * Final part of subpacket decoding:
912 * Apply modulated lapped transform, gain compensation,
913 * clip and convert to integer.
914 *
915 * @param q pointer to the COOKContext
916 * @param decode_buffer pointer to the mlt coefficients
917 * @param gains_ptr array of current/prev gain pointers
918 * @param previous_buffer pointer to the previous buffer to be used for overlapping
919 * @param out pointer to the output buffer
920 */
mlt_compensate_output(COOKContext * q,float * decode_buffer,cook_gains * gains_ptr,float * previous_buffer,float * out)921 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
922 cook_gains *gains_ptr, float *previous_buffer,
923 float *out)
924 {
925 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
926 if (out)
927 q->saturate_output(q, out);
928 }
929
930
931 /**
932 * Cook subpacket decoding. This function returns one decoded subpacket,
933 * usually 1024 samples per channel.
934 *
935 * @param q pointer to the COOKContext
936 * @param inbuffer pointer to the inbuffer
937 * @param outbuffer pointer to the outbuffer
938 */
decode_subpacket(COOKContext * q,COOKSubpacket * p,const uint8_t * inbuffer,float ** outbuffer)939 static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
940 const uint8_t *inbuffer, float **outbuffer)
941 {
942 int sub_packet_size = p->size;
943 int res;
944
945 memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
946 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
947
948 if (p->joint_stereo) {
949 if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
950 return res;
951 } else {
952 if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
953 return res;
954
955 if (p->num_channels == 2) {
956 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
957 if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
958 return res;
959 }
960 }
961
962 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
963 p->mono_previous_buffer1,
964 outbuffer ? outbuffer[p->ch_idx] : NULL);
965
966 if (p->num_channels == 2) {
967 if (p->joint_stereo)
968 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
969 p->mono_previous_buffer2,
970 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
971 else
972 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
973 p->mono_previous_buffer2,
974 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
975 }
976
977 return 0;
978 }
979
980
cook_decode_frame(AVCodecContext * avctx,AVFrame * frame,int * got_frame_ptr,AVPacket * avpkt)981 static int cook_decode_frame(AVCodecContext *avctx, AVFrame *frame,
982 int *got_frame_ptr, AVPacket *avpkt)
983 {
984 const uint8_t *buf = avpkt->data;
985 int buf_size = avpkt->size;
986 COOKContext *q = avctx->priv_data;
987 float **samples = NULL;
988 int i, ret;
989 int offset = 0;
990 int chidx = 0;
991
992 if (buf_size < avctx->block_align)
993 return buf_size;
994
995 /* get output buffer */
996 if (q->discarded_packets >= 2) {
997 frame->nb_samples = q->samples_per_channel;
998 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
999 return ret;
1000 samples = (float **)frame->extended_data;
1001 }
1002
1003 /* estimate subpacket sizes */
1004 q->subpacket[0].size = avctx->block_align;
1005
1006 for (i = 1; i < q->num_subpackets; i++) {
1007 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
1008 q->subpacket[0].size -= q->subpacket[i].size + 1;
1009 if (q->subpacket[0].size < 0) {
1010 av_log(avctx, AV_LOG_DEBUG,
1011 "frame subpacket size total > avctx->block_align!\n");
1012 return AVERROR_INVALIDDATA;
1013 }
1014 }
1015
1016 /* decode supbackets */
1017 for (i = 0; i < q->num_subpackets; i++) {
1018 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
1019 q->subpacket[i].bits_per_subpdiv;
1020 q->subpacket[i].ch_idx = chidx;
1021 av_log(avctx, AV_LOG_DEBUG,
1022 "subpacket[%i] size %i js %i %i block_align %i\n",
1023 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
1024 avctx->block_align);
1025
1026 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1027 return ret;
1028 offset += q->subpacket[i].size;
1029 chidx += q->subpacket[i].num_channels;
1030 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1031 i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1032 }
1033
1034 /* Discard the first two frames: no valid audio. */
1035 if (q->discarded_packets < 2) {
1036 q->discarded_packets++;
1037 *got_frame_ptr = 0;
1038 return avctx->block_align;
1039 }
1040
1041 *got_frame_ptr = 1;
1042
1043 return avctx->block_align;
1044 }
1045
dump_cook_context(COOKContext * q)1046 static void dump_cook_context(COOKContext *q)
1047 {
1048 //int i=0;
1049 #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1050 ff_dlog(q->avctx, "COOKextradata\n");
1051 ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1052 if (q->subpacket[0].cookversion > STEREO) {
1053 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1054 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1055 }
1056 ff_dlog(q->avctx, "COOKContext\n");
1057 PRINT("nb_channels", q->avctx->ch_layout.nb_channels);
1058 PRINT("bit_rate", (int)q->avctx->bit_rate);
1059 PRINT("sample_rate", q->avctx->sample_rate);
1060 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1061 PRINT("subbands", q->subpacket[0].subbands);
1062 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1063 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1064 PRINT("numvector_size", q->subpacket[0].numvector_size);
1065 PRINT("total_subbands", q->subpacket[0].total_subbands);
1066 }
1067
1068 /**
1069 * Cook initialization
1070 *
1071 * @param avctx pointer to the AVCodecContext
1072 */
cook_decode_init(AVCodecContext * avctx)1073 static av_cold int cook_decode_init(AVCodecContext *avctx)
1074 {
1075 static AVOnce init_static_once = AV_ONCE_INIT;
1076 COOKContext *q = avctx->priv_data;
1077 GetByteContext gb;
1078 int s = 0;
1079 unsigned int channel_mask = 0;
1080 int samples_per_frame = 0;
1081 int ret;
1082 int channels = avctx->ch_layout.nb_channels;
1083
1084 q->avctx = avctx;
1085
1086 /* Take care of the codec specific extradata. */
1087 if (avctx->extradata_size < 8) {
1088 av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1089 return AVERROR_INVALIDDATA;
1090 }
1091 av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1092
1093 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1094
1095 /* Take data from the AVCodecContext (RM container). */
1096 if (!channels) {
1097 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1098 return AVERROR_INVALIDDATA;
1099 }
1100
1101 if (avctx->block_align >= INT_MAX / 8)
1102 return AVERROR(EINVAL);
1103
1104 /* Initialize RNG. */
1105 av_lfg_init(&q->random_state, 0);
1106
1107 ff_audiodsp_init(&q->adsp);
1108
1109 while (bytestream2_get_bytes_left(&gb)) {
1110 if (s >= FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
1111 avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1112 return AVERROR_PATCHWELCOME;
1113 }
1114 /* 8 for mono, 16 for stereo, ? for multichannel
1115 Swap to right endianness so we don't need to care later on. */
1116 q->subpacket[s].cookversion = bytestream2_get_be32(&gb);
1117 samples_per_frame = bytestream2_get_be16(&gb);
1118 q->subpacket[s].subbands = bytestream2_get_be16(&gb);
1119 bytestream2_get_be32(&gb); // Unknown unused
1120 q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
1121 if (q->subpacket[s].js_subband_start >= 51) {
1122 av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
1123 return AVERROR_INVALIDDATA;
1124 }
1125 q->subpacket[s].js_vlc_bits = bytestream2_get_be16(&gb);
1126
1127 /* Initialize extradata related variables. */
1128 q->subpacket[s].samples_per_channel = samples_per_frame / channels;
1129 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1130
1131 /* Initialize default data states. */
1132 q->subpacket[s].log2_numvector_size = 5;
1133 q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1134 q->subpacket[s].num_channels = 1;
1135
1136 /* Initialize version-dependent variables */
1137
1138 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1139 q->subpacket[s].cookversion);
1140 q->subpacket[s].joint_stereo = 0;
1141 switch (q->subpacket[s].cookversion) {
1142 case MONO:
1143 if (channels != 1) {
1144 avpriv_request_sample(avctx, "Container channels != 1");
1145 return AVERROR_PATCHWELCOME;
1146 }
1147 av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1148 break;
1149 case STEREO:
1150 if (channels != 1) {
1151 q->subpacket[s].bits_per_subpdiv = 1;
1152 q->subpacket[s].num_channels = 2;
1153 }
1154 av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1155 break;
1156 case JOINT_STEREO:
1157 if (channels != 2) {
1158 avpriv_request_sample(avctx, "Container channels != 2");
1159 return AVERROR_PATCHWELCOME;
1160 }
1161 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1162 if (avctx->extradata_size >= 16) {
1163 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1164 q->subpacket[s].js_subband_start;
1165 q->subpacket[s].joint_stereo = 1;
1166 q->subpacket[s].num_channels = 2;
1167 }
1168 if (q->subpacket[s].samples_per_channel > 256) {
1169 q->subpacket[s].log2_numvector_size = 6;
1170 }
1171 if (q->subpacket[s].samples_per_channel > 512) {
1172 q->subpacket[s].log2_numvector_size = 7;
1173 }
1174 break;
1175 case MC_COOK:
1176 av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1177 channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
1178
1179 if (av_popcount64(q->subpacket[s].channel_mask) > 1) {
1180 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1181 q->subpacket[s].js_subband_start;
1182 q->subpacket[s].joint_stereo = 1;
1183 q->subpacket[s].num_channels = 2;
1184 q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1185
1186 if (q->subpacket[s].samples_per_channel > 256) {
1187 q->subpacket[s].log2_numvector_size = 6;
1188 }
1189 if (q->subpacket[s].samples_per_channel > 512) {
1190 q->subpacket[s].log2_numvector_size = 7;
1191 }
1192 } else
1193 q->subpacket[s].samples_per_channel = samples_per_frame;
1194
1195 break;
1196 default:
1197 avpriv_request_sample(avctx, "Cook version %d",
1198 q->subpacket[s].cookversion);
1199 return AVERROR_PATCHWELCOME;
1200 }
1201
1202 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1203 av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1204 return AVERROR_INVALIDDATA;
1205 } else
1206 q->samples_per_channel = q->subpacket[0].samples_per_channel;
1207
1208
1209 /* Initialize variable relations */
1210 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1211
1212 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1213 if (q->subpacket[s].total_subbands > 53) {
1214 avpriv_request_sample(avctx, "total_subbands > 53");
1215 return AVERROR_PATCHWELCOME;
1216 }
1217
1218 if ((q->subpacket[s].js_vlc_bits > 6) ||
1219 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1220 av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1221 q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1222 return AVERROR_INVALIDDATA;
1223 }
1224
1225 if (q->subpacket[s].subbands > 50) {
1226 avpriv_request_sample(avctx, "subbands > 50");
1227 return AVERROR_PATCHWELCOME;
1228 }
1229 if (q->subpacket[s].subbands == 0) {
1230 avpriv_request_sample(avctx, "subbands = 0");
1231 return AVERROR_PATCHWELCOME;
1232 }
1233 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1234 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1235 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1236 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1237
1238 if (q->num_subpackets + q->subpacket[s].num_channels > channels) {
1239 av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, channels);
1240 return AVERROR_INVALIDDATA;
1241 }
1242
1243 q->num_subpackets++;
1244 s++;
1245 }
1246
1247 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1248 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1249 q->samples_per_channel != 1024) {
1250 avpriv_request_sample(avctx, "samples_per_channel = %d",
1251 q->samples_per_channel);
1252 return AVERROR_PATCHWELCOME;
1253 }
1254
1255 /* Generate tables */
1256 ff_thread_once(&init_static_once, init_pow2table);
1257 init_gain_table(q);
1258 init_cplscales_table(q);
1259
1260 if ((ret = init_cook_vlc_tables(q)))
1261 return ret;
1262
1263 /* Pad the databuffer with:
1264 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1265 AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1266 q->decoded_bytes_buffer =
1267 av_mallocz(avctx->block_align
1268 + DECODE_BYTES_PAD1(avctx->block_align)
1269 + AV_INPUT_BUFFER_PADDING_SIZE);
1270 if (!q->decoded_bytes_buffer)
1271 return AVERROR(ENOMEM);
1272
1273 /* Initialize transform. */
1274 if ((ret = init_cook_mlt(q)))
1275 return ret;
1276
1277 /* Initialize COOK signal arithmetic handling */
1278 if (1) {
1279 q->scalar_dequant = scalar_dequant_float;
1280 q->decouple = decouple_float;
1281 q->imlt_window = imlt_window_float;
1282 q->interpolate = interpolate_float;
1283 q->saturate_output = saturate_output_float;
1284 }
1285
1286 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1287 av_channel_layout_uninit(&avctx->ch_layout);
1288 if (channel_mask)
1289 av_channel_layout_from_mask(&avctx->ch_layout, channel_mask);
1290 else
1291 av_channel_layout_default(&avctx->ch_layout, channels);
1292
1293
1294 dump_cook_context(q);
1295
1296 return 0;
1297 }
1298
1299 const FFCodec ff_cook_decoder = {
1300 .p.name = "cook",
1301 .p.long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1302 .p.type = AVMEDIA_TYPE_AUDIO,
1303 .p.id = AV_CODEC_ID_COOK,
1304 .priv_data_size = sizeof(COOKContext),
1305 .init = cook_decode_init,
1306 .close = cook_decode_close,
1307 FF_CODEC_DECODE_CB(cook_decode_frame),
1308 .p.capabilities = AV_CODEC_CAP_DR1,
1309 .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1310 AV_SAMPLE_FMT_NONE },
1311 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1312 };
1313