FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavcodec/cook.c
Date: 2023-06-04 16:45:34
Exec Total Coverage
Lines: 442 539 82.0%
Functions: 29 29 100.0%
Branches: 194 271 71.6%
Line Branch Exec Source
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 #include "libavutil/tx.h"
50
51 #include "audiodsp.h"
52 #include "avcodec.h"
53 #include "get_bits.h"
54 #include "bytestream.h"
55 #include "codec_internal.h"
56 #include "decode.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 AVTXContext *mdct_ctx;
144 av_tx_fn mdct_fn;
145 float* mlt_window;
146
147 /* VLC data */
148 VLC envelope_quant_index[13];
149 VLC sqvh[7]; // scalar quantization
150
151 /* generate tables and related variables */
152 int gain_size_factor;
153 float gain_table[31];
154
155 /* data buffers */
156
157 uint8_t* decoded_bytes_buffer;
158 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
159 float decode_buffer_1[1024];
160 float decode_buffer_2[1024];
161 float decode_buffer_0[1060]; /* static allocation for joint decode */
162
163 const float *cplscales[5];
164 int num_subpackets;
165 COOKSubpacket subpacket[MAX_SUBPACKETS];
166 } COOKContext;
167
168 static float pow2tab[127];
169 static float rootpow2tab[127];
170
171 /*************** init functions ***************/
172
173 /* table generator */
174 5 static av_cold void init_pow2table(void)
175 {
176 /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
177 int i;
178 static const float exp2_tab[2] = {1, M_SQRT2};
179 5 float exp2_val = powf(2, -63);
180 5 float root_val = powf(2, -32);
181
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 640 for (i = -63; i < 64; i++) {
182
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 635 if (!(i & 1))
183 315 root_val *= 2;
184 635 pow2tab[63 + i] = exp2_val;
185 635 rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
186 635 exp2_val *= 2;
187 }
188 5 }
189
190 /* table generator */
191 6 static av_cold void init_gain_table(COOKContext *q)
192 {
193 int i;
194 6 q->gain_size_factor = q->samples_per_channel / 8;
195
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 192 for (i = 0; i < 31; i++)
196 186 q->gain_table[i] = pow(pow2tab[i + 48],
197 186 (1.0 / (double) q->gain_size_factor));
198 6 }
199
200 124 static av_cold int build_vlc(VLC *vlc, int nb_bits, const uint8_t counts[16],
201 const void *syms, int symbol_size, int offset,
202 void *logctx)
203 {
204 uint8_t lens[MAX_COOK_VLC_ENTRIES];
205 124 unsigned num = 0;
206
207
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 2108 for (int i = 0; i < 16; i++)
208
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 11636 for (unsigned count = num + counts[i]; num < count; num++)
209 9652 lens[num] = i + 1;
210
211 124 return ff_init_vlc_from_lengths(vlc, nb_bits, num, lens, 1,
212 syms, symbol_size, symbol_size,
213 offset, 0, logctx);
214 }
215
216 6 static av_cold int init_cook_vlc_tables(COOKContext *q)
217 {
218 int i, result;
219
220 6 result = 0;
221
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 84 for (i = 0; i < 13; i++) {
222 78 result |= build_vlc(&q->envelope_quant_index[i], QUANT_VLC_BITS,
223 78 envelope_quant_index_huffcounts[i],
224 78 envelope_quant_index_huffsyms[i], 1, -12, q->avctx);
225 }
226 6 av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
227
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 48 for (i = 0; i < 7; i++) {
228
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 42 int sym_size = 1 + (i == 3);
229 42 result |= build_vlc(&q->sqvh[i], vhvlcsize_tab[i],
230 42 cvh_huffcounts[i],
231 42 cvh_huffsyms[i], sym_size, 0, q->avctx);
232 }
233
234
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 12 for (i = 0; i < q->num_subpackets; i++) {
235
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 6 if (q->subpacket[i].joint_stereo == 1) {
236 8 result |= build_vlc(&q->subpacket[i].channel_coupling, COUPLING_VLC_BITS,
237 4 ccpl_huffcounts[q->subpacket[i].js_vlc_bits - 2],
238 4 ccpl_huffsyms[q->subpacket[i].js_vlc_bits - 2], 1,
239 4 0, q->avctx);
240 4 av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
241 }
242 }
243
244 6 av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
245 6 return result;
246 }
247
248 6 static av_cold int init_cook_mlt(COOKContext *q)
249 {
250 int j, ret;
251 6 int mlt_size = q->samples_per_channel;
252 6 const float scale = 1.0 / 32768.0;
253
254
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 6 if (!(q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))))
255 ✗ return AVERROR(ENOMEM);
256
257 /* Initialize the MLT window: simple sine window. */
258 6 ff_sine_window_init(q->mlt_window, mlt_size);
259
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 6150 for (j = 0; j < mlt_size; j++)
260 6144 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
261
262 /* Initialize the MDCT. */
263 6 ret = av_tx_init(&q->mdct_ctx, &q->mdct_fn, AV_TX_FLOAT_MDCT,
264 1, mlt_size, &scale, AV_TX_FULL_IMDCT);
265
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 6 if (ret < 0)
266 ✗ return ret;
267
268 6 return 0;
269 }
270
271 6 static av_cold void init_cplscales_table(COOKContext *q)
272 {
273 int i;
274
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 36 for (i = 0; i < 5; i++)
275 30 q->cplscales[i] = cplscales[i];
276 6 }
277
278 /*************** init functions end ***********/
279
280 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
281 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
282
283 /**
284 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
285 * Why? No idea, some checksum/error detection method maybe.
286 *
287 * Out buffer size: extra bytes are needed to cope with
288 * padding/misalignment.
289 * Subpackets passed to the decoder can contain two, consecutive
290 * half-subpackets, of identical but arbitrary size.
291 * 1234 1234 1234 1234 extraA extraB
292 * Case 1: AAAA BBBB 0 0
293 * Case 2: AAAA ABBB BB-- 3 3
294 * Case 3: AAAA AABB BBBB 2 2
295 * Case 4: AAAA AAAB BBBB BB-- 1 5
296 *
297 * Nice way to waste CPU cycles.
298 *
299 * @param inbuffer pointer to byte array of indata
300 * @param out pointer to byte array of outdata
301 * @param bytes number of bytes
302 */
303 240 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
304 {
305 static const uint32_t tab[4] = {
306 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
307 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
308 };
309 int i, off;
310 uint32_t c;
311 const uint32_t *buf;
312 240 uint32_t *obuf = (uint32_t *) out;
313 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
314 * I'm too lazy though, should be something like
315 * for (i = 0; i < bitamount / 64; i++)
316 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
317 * Buffer alignment needs to be checked. */
318
319 240 off = (intptr_t) inbuffer & 3;
320 240 buf = (const uint32_t *) (inbuffer - off);
321 240 c = tab[off];
322 240 bytes += 3 + off;
323
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 11520 for (i = 0; i < bytes / 4; i++)
324 11280 obuf[i] = c ^ buf[i];
325
326 240 return off;
327 }
328
329 6 static av_cold int cook_decode_close(AVCodecContext *avctx)
330 {
331 int i;
332 6 COOKContext *q = avctx->priv_data;
333 6 av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
334
335 /* Free allocated memory buffers. */
336 6 av_freep(&q->mlt_window);
337 6 av_freep(&q->decoded_bytes_buffer);
338
339 /* Free the transform. */
340 6 av_tx_uninit(&q->mdct_ctx);
341
342 /* Free the VLC tables. */
343
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 84 for (i = 0; i < 13; i++)
344 78 ff_free_vlc(&q->envelope_quant_index[i]);
345
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 48 for (i = 0; i < 7; i++)
346 42 ff_free_vlc(&q->sqvh[i]);
347
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 12 for (i = 0; i < q->num_subpackets; i++)
348 6 ff_free_vlc(&q->subpacket[i].channel_coupling);
349
350 6 av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
351
352 6 return 0;
353 }
354
355 /**
356 * Fill the gain array for the timedomain quantization.
357 *
358 * @param gb pointer to the GetBitContext
359 * @param gaininfo array[9] of gain indexes
360 */
361 240 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
362 {
363 int i, n;
364
365 240 n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update
366
367 240 i = 0;
368
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 241 while (n--) {
369 1 int index = get_bits(gb, 3);
370
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 1 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
371
372
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 8 while (i <= index)
373 7 gaininfo[i++] = gain;
374 }
375
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 2393 while (i <= 8)
376 2153 gaininfo[i++] = 0;
377 240 }
378
379 /**
380 * Create the quant index table needed for the envelope.
381 *
382 * @param q pointer to the COOKContext
383 * @param quant_index_table pointer to the array
384 */
385 240 static int decode_envelope(COOKContext *q, COOKSubpacket *p,
386 int *quant_index_table)
387 {
388 int i, j, vlc_index;
389
390 240 quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
391
392
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 10320 for (i = 1; i < p->total_subbands; i++) {
393 10080 vlc_index = i;
394
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 10080 if (i >= p->js_subband_start * 2) {
395 7440 vlc_index -= p->js_subband_start;
396 } else {
397 2640 vlc_index /= 2;
398
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 2640 if (vlc_index < 1)
399 240 vlc_index = 1;
400 }
401
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 10080 if (vlc_index > 13)
402 5520 vlc_index = 13; // the VLC tables >13 are identical to No. 13
403
404 10080 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
405 QUANT_VLC_BITS, 2);
406 10080 quant_index_table[i] = quant_index_table[i - 1] + j; // differential encoding
407
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 10080 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
408 ✗ av_log(q->avctx, AV_LOG_ERROR,
409 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
410 ✗ quant_index_table[i], i);
411 ✗ return AVERROR_INVALIDDATA;
412 }
413 }
414
415 240 return 0;
416 }
417
418 /**
419 * Calculate the category and category_index vector.
420 *
421 * @param q pointer to the COOKContext
422 * @param quant_index_table pointer to the array
423 * @param category pointer to the category array
424 * @param category_index pointer to the category_index array
425 */
426 240 static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
427 int *category, int *category_index)
428 {
429 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
430 240 int exp_index2[102] = { 0 };
431 240 int exp_index1[102] = { 0 };
432
433 240 int tmp_categorize_array[128 * 2] = { 0 };
434 240 int tmp_categorize_array1_idx = p->numvector_size;
435 240 int tmp_categorize_array2_idx = p->numvector_size;
436
437 240 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
438
439
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 240 if (bits_left > q->samples_per_channel)
440 240 bits_left = q->samples_per_channel +
441 240 ((bits_left - q->samples_per_channel) * 5) / 8;
442
443 240 bias = -32;
444
445 /* Estimate bias. */
446
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 1680 for (i = 32; i > 0; i = i / 2) {
447 1440 num_bits = 0;
448 1440 index = 0;
449
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 63360 for (j = p->total_subbands; j > 0; j--) {
450 61920 exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
451 61920 index++;
452 61920 num_bits += expbits_tab[exp_idx];
453 }
454
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 1440 if (num_bits >= bits_left - 32)
455 1333 bias += i;
456 }
457
458 /* Calculate total number of bits. */
459 240 num_bits = 0;
460
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 10560 for (i = 0; i < p->total_subbands; i++) {
461 10320 exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
462 10320 num_bits += expbits_tab[exp_idx];
463 10320 exp_index1[i] = exp_idx;
464 10320 exp_index2[i] = exp_idx;
465 }
466 240 tmpbias1 = tmpbias2 = num_bits;
467
468
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 30720 for (j = 1; j < p->numvector_size; j++) {
469
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 30480 if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
470 16408 int max = -999999;
471 16408 index = -1;
472
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 721952 for (i = 0; i < p->total_subbands; i++) {
473
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 705544 if (exp_index1[i] < 7) {
474 543313 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
475
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 543313 if (v >= max) {
476 186262 max = v;
477 186262 index = i;
478 }
479 }
480 }
481
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 16408 if (index == -1)
482 ✗ break;
483 16408 tmp_categorize_array[tmp_categorize_array1_idx++] = index;
484 16408 tmpbias1 -= expbits_tab[exp_index1[index]] -
485 16408 expbits_tab[exp_index1[index] + 1];
486 16408 ++exp_index1[index];
487 } else { /* <--- */
488 14072 int min = 999999;
489 14072 index = -1;
490
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 619168 for (i = 0; i < p->total_subbands; i++) {
491
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 605096 if (exp_index2[i] > 0) {
492 527917 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
493
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 527917 if (v < min) {
494 31097 min = v;
495 31097 index = i;
496 }
497 }
498 }
499
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 14072 if (index == -1)
500 ✗ break;
501 14072 tmp_categorize_array[--tmp_categorize_array2_idx] = index;
502 14072 tmpbias2 -= expbits_tab[exp_index2[index]] -
503 14072 expbits_tab[exp_index2[index] - 1];
504 14072 --exp_index2[index];
505 }
506 }
507
508
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 10560 for (i = 0; i < p->total_subbands; i++)
509 10320 category[i] = exp_index2[i];
510
511
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 30720 for (i = 0; i < p->numvector_size - 1; i++)
512 30480 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
513 240 }
514
515
516 /**
517 * Expand the category vector.
518 *
519 * @param q pointer to the COOKContext
520 * @param category pointer to the category array
521 * @param category_index pointer to the category_index array
522 */
523 240 static inline void expand_category(COOKContext *q, int *category,
524 int *category_index)
525 {
526 int i;
527
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 15141 for (i = 0; i < q->num_vectors; i++)
528 {
529 14901 int idx = category_index[i];
530
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 14901 if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
531 ✗ --category[idx];
532 }
533 240 }
534
535 /**
536 * The real requantization of the mltcoefs
537 *
538 * @param q pointer to the COOKContext
539 * @param index index
540 * @param quant_index quantisation index
541 * @param subband_coef_index array of indexes to quant_centroid_tab
542 * @param subband_coef_sign signs of coefficients
543 * @param mlt_p pointer into the mlt buffer
544 */
545 10320 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
546 int *subband_coef_index, int *subband_coef_sign,
547 float *mlt_p)
548 {
549 int i;
550 float f1;
551
552
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 216720 for (i = 0; i < SUBBAND_SIZE; i++) {
553
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 206400 if (subband_coef_index[i]) {
554 67248 f1 = quant_centroid_tab[index][subband_coef_index[i]];
555
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 67248 if (subband_coef_sign[i])
556 33713 f1 = -f1;
557 } else {
558 /* noise coding if subband_coef_index[i] == 0 */
559 139152 f1 = dither_tab[index];
560
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 139152 if (av_lfg_get(&q->random_state) < 0x80000000)
561 69511 f1 = -f1;
562 }
563 206400 mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
564 }
565 10320 }
566 /**
567 * Unpack the subband_coef_index and subband_coef_sign vectors.
568 *
569 * @param q pointer to the COOKContext
570 * @param category pointer to the category array
571 * @param subband_coef_index array of indexes to quant_centroid_tab
572 * @param subband_coef_sign signs of coefficients
573 */
574 8969 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
575 int *subband_coef_index, int *subband_coef_sign)
576 {
577 int i, j;
578 int vlc, vd, tmp, result;
579
580 8969 vd = vd_tab[category];
581 8969 result = 0;
582
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 65444 for (i = 0; i < vpr_tab[category]; i++) {
583 56475 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
584
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 56475 if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
585 ✗ vlc = 0;
586 ✗ result = 1;
587 }
588
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 235855 for (j = vd - 1; j >= 0; j--) {
589 179380 tmp = (vlc * invradix_tab[category]) / 0x100000;
590 179380 subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
591 179380 vlc = tmp;
592 }
593
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 235855 for (j = 0; j < vd; j++) {
594
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 179380 if (subband_coef_index[i * vd + j]) {
595
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 67248 if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
596 67248 subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
597 } else {
598 ✗ result = 1;
599 ✗ subband_coef_sign[i * vd + j] = 0;
600 }
601 } else {
602 112132 subband_coef_sign[i * vd + j] = 0;
603 }
604 }
605 }
606 8969 return result;
607 }
608
609
610 /**
611 * Fill the mlt_buffer with mlt coefficients.
612 *
613 * @param q pointer to the COOKContext
614 * @param category pointer to the category array
615 * @param quant_index_table pointer to the array
616 * @param mlt_buffer pointer to mlt coefficients
617 */
618 240 static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
619 int *quant_index_table, float *mlt_buffer)
620 {
621 /* A zero in this table means that the subband coefficient is
622 random noise coded. */
623 int subband_coef_index[SUBBAND_SIZE];
624 /* A zero in this table means that the subband coefficient is a
625 positive multiplicator. */
626 int subband_coef_sign[SUBBAND_SIZE];
627 int band, j;
628 240 int index = 0;
629
630
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 10560 for (band = 0; band < p->total_subbands; band++) {
631 10320 index = category[band];
632
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 10320 if (category[band] < 7) {
633
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 8969 if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
634 ✗ index = 7;
635 ✗ for (j = 0; j < p->total_subbands; j++)
636 ✗ category[band + j] = 7;
637 }
638 }
639
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 10320 if (index >= 7) {
640 1351 memset(subband_coef_index, 0, sizeof(subband_coef_index));
641 1351 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
642 }
643 10320 q->scalar_dequant(q, index, quant_index_table[band],
644 subband_coef_index, subband_coef_sign,
645 10320 &mlt_buffer[band * SUBBAND_SIZE]);
646 }
647
648 /* FIXME: should this be removed, or moved into loop above? */
649
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 240 if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
650 ✗ return;
651 }
652
653
654 240 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
655 {
656 240 int category_index[128] = { 0 };
657 240 int category[128] = { 0 };
658 int quant_index_table[102];
659 int res, i;
660
661
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 240 if ((res = decode_envelope(q, p, quant_index_table)) < 0)
662 ✗ return res;
663 240 q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
664 240 categorize(q, p, quant_index_table, category, category_index);
665 240 expand_category(q, category, category_index);
666
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 10560 for (i=0; i<p->total_subbands; i++) {
667
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 10320 if (category[i] > 7)
668 ✗ return AVERROR_INVALIDDATA;
669 }
670 240 decode_vectors(q, p, category, quant_index_table, mlt_buffer);
671
672 240 return 0;
673 }
674
675
676 /**
677 * the actual requantization of the timedomain samples
678 *
679 * @param q pointer to the COOKContext
680 * @param buffer pointer to the timedomain buffer
681 * @param gain_index index for the block multiplier
682 * @param gain_index_next index for the next block multiplier
683 */
684 14 static void interpolate_float(COOKContext *q, float *buffer,
685 int gain_index, int gain_index_next)
686 {
687 int i;
688 float fc1, fc2;
689 14 fc1 = pow2tab[gain_index + 63];
690
691
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 14 if (gain_index == gain_index_next) { // static gain
692
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 1548 for (i = 0; i < q->gain_size_factor; i++)
693 1536 buffer[i] *= fc1;
694 } else { // smooth gain
695 2 fc2 = q->gain_table[15 + (gain_index_next - gain_index)];
696
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 258 for (i = 0; i < q->gain_size_factor; i++) {
697 256 buffer[i] *= fc1;
698 256 fc1 *= fc2;
699 }
700 }
701 14 }
702
703 /**
704 * Apply transform window, overlap buffers.
705 *
706 * @param q pointer to the COOKContext
707 * @param inbuffer pointer to the mltcoefficients
708 * @param gains_ptr current and previous gains
709 * @param previous_buffer pointer to the previous buffer to be used for overlapping
710 */
711 480 static void imlt_window_float(COOKContext *q, float *inbuffer,
712 cook_gains *gains_ptr, float *previous_buffer)
713 {
714 480 const float fc = pow2tab[gains_ptr->previous[0] + 63];
715 int i;
716 /* The weird thing here, is that the two halves of the time domain
717 * buffer are swapped. Also, the newest data, that we save away for
718 * next frame, has the wrong sign. Hence the subtraction below.
719 * Almost sounds like a complex conjugate/reverse data/FFT effect.
720 */
721
722 /* Apply window and overlap */
723
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 492000 for (i = 0; i < q->samples_per_channel; i++)
724 491520 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
725 491520 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
726 480 }
727
728 /**
729 * The modulated lapped transform, this takes transform coefficients
730 * and transforms them into timedomain samples.
731 * Apply transform window, overlap buffers, apply gain profile
732 * and buffer management.
733 *
734 * @param q pointer to the COOKContext
735 * @param inbuffer pointer to the mltcoefficients
736 * @param gains_ptr current and previous gains
737 * @param previous_buffer pointer to the previous buffer to be used for overlapping
738 */
739 480 static void imlt_gain(COOKContext *q, float *inbuffer,
740 cook_gains *gains_ptr, float *previous_buffer)
741 {
742 480 float *buffer0 = q->mono_mdct_output;
743 480 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
744 int i;
745
746 /* Inverse modified discrete cosine transform */
747 480 q->mdct_fn(q->mdct_ctx, q->mono_mdct_output, inbuffer, sizeof(float));
748
749 480 q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
750
751 /* Apply gain profile */
752
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753
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 3840 if (gains_ptr->now[i] || gains_ptr->now[i + 1])
754 14 q->interpolate(q, &buffer1[q->gain_size_factor * i],
755 14 gains_ptr->now[i], gains_ptr->now[i + 1]);
756
757 /* Save away the current to be previous block. */
758 480 memcpy(previous_buffer, buffer0,
759 480 q->samples_per_channel * sizeof(*previous_buffer));
760 480 }
761
762
763 /**
764 * function for getting the jointstereo coupling information
765 *
766 * @param q pointer to the COOKContext
767 * @param decouple_tab decoupling array
768 */
769 240 static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
770 {
771 int i;
772 240 int vlc = get_bits1(&q->gb);
773 240 int start = cplband[p->js_subband_start];
774 240 int end = cplband[p->subbands - 1];
775 240 int length = end - start + 1;
776
777
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 240 if (start > end)
778 ✗ return 0;
779
780
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 240 if (vlc)
781
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 1806 for (i = 0; i < length; i++)
782 1677 decouple_tab[start + i] = get_vlc2(&q->gb,
783 1677 p->channel_coupling.table,
784 COUPLING_VLC_BITS, 3);
785 else
786
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 1554 for (i = 0; i < length; i++) {
787 1443 int v = get_bits(&q->gb, p->js_vlc_bits);
788
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 1443 if (v == (1<<p->js_vlc_bits)-1) {
789 ✗ av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
790 ✗ return AVERROR_INVALIDDATA;
791 }
792 1443 decouple_tab[start + i] = v;
793 }
794 240 return 0;
795 }
796
797 /**
798 * function decouples a pair of signals from a single signal via multiplication.
799 *
800 * @param q pointer to the COOKContext
801 * @param subband index of the current subband
802 * @param f1 multiplier for channel 1 extraction
803 * @param f2 multiplier for channel 2 extraction
804 * @param decode_buffer input buffer
805 * @param mlt_buffer1 pointer to left channel mlt coefficients
806 * @param mlt_buffer2 pointer to right channel mlt coefficients
807 */
808 7440 static void decouple_float(COOKContext *q,
809 COOKSubpacket *p,
810 int subband,
811 float f1, float f2,
812 float *decode_buffer,
813 float *mlt_buffer1, float *mlt_buffer2)
814 {
815 int j, tmp_idx;
816
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 156240 for (j = 0; j < SUBBAND_SIZE; j++) {
817 148800 tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
818 148800 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
819 148800 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
820 }
821 7440 }
822
823 /**
824 * function for decoding joint stereo data
825 *
826 * @param q pointer to the COOKContext
827 * @param mlt_buffer1 pointer to left channel mlt coefficients
828 * @param mlt_buffer2 pointer to right channel mlt coefficients
829 */
830 240 static int joint_decode(COOKContext *q, COOKSubpacket *p,
831 float *mlt_buffer_left, float *mlt_buffer_right)
832 {
833 int i, j, res;
834 240 int decouple_tab[SUBBAND_SIZE] = { 0 };
835 240 float *decode_buffer = q->decode_buffer_0;
836 int idx, cpl_tmp;
837 float f1, f2;
838 const float *cplscale;
839
840 240 memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
841
842 /* Make sure the buffers are zeroed out. */
843 240 memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
844 240 memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
845
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 240 if ((res = decouple_info(q, p, decouple_tab)) < 0)
846 ✗ return res;
847
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 240 if ((res = mono_decode(q, p, decode_buffer)) < 0)
848 ✗ return res;
849 /* The two channels are stored interleaved in decode_buffer. */
850
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 1680 for (i = 0; i < p->js_subband_start; i++) {
851
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 30240 for (j = 0; j < SUBBAND_SIZE; j++) {
852 28800 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
853 28800 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
854 }
855 }
856
857 /* When we reach js_subband_start (the higher frequencies)
858 the coefficients are stored in a coupling scheme. */
859 240 idx = (1 << p->js_vlc_bits) - 1;
860
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 7680 for (i = p->js_subband_start; i < p->subbands; i++) {
861 7440 cpl_tmp = cplband[i];
862 7440 idx -= decouple_tab[cpl_tmp];
863 7440 cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
864 7440 f1 = cplscale[decouple_tab[cpl_tmp] + 1];
865 7440 f2 = cplscale[idx];
866 7440 q->decouple(q, p, i, f1, f2, decode_buffer,
867 mlt_buffer_left, mlt_buffer_right);
868 7440 idx = (1 << p->js_vlc_bits) - 1;
869 }
870
871 240 return 0;
872 }
873
874 /**
875 * First part of subpacket decoding:
876 * decode raw stream bytes and read gain info.
877 *
878 * @param q pointer to the COOKContext
879 * @param inbuffer pointer to raw stream data
880 * @param gains_ptr array of current/prev gain pointers
881 */
882 240 static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
883 const uint8_t *inbuffer,
884 cook_gains *gains_ptr)
885 {
886 int offset;
887
888 240 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
889 240 p->bits_per_subpacket / 8);
890 240 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
891 p->bits_per_subpacket);
892 240 decode_gain_info(&q->gb, gains_ptr->now);
893
894 /* Swap current and previous gains */
895 240 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
896 240 }
897
898 /**
899 * Saturate the output signal and interleave.
900 *
901 * @param q pointer to the COOKContext
902 * @param out pointer to the output vector
903 */
904 476 static void saturate_output_float(COOKContext *q, float *out)
905 {
906 476 q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
907 476 FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
908 476 }
909
910
911 /**
912 * Final part of subpacket decoding:
913 * Apply modulated lapped transform, gain compensation,
914 * clip and convert to integer.
915 *
916 * @param q pointer to the COOKContext
917 * @param decode_buffer pointer to the mlt coefficients
918 * @param gains_ptr array of current/prev gain pointers
919 * @param previous_buffer pointer to the previous buffer to be used for overlapping
920 * @param out pointer to the output buffer
921 */
922 480 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
923 cook_gains *gains_ptr, float *previous_buffer,
924 float *out)
925 {
926 480 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
927
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 480 if (out)
928 476 q->saturate_output(q, out);
929 480 }
930
931
932 /**
933 * Cook subpacket decoding. This function returns one decoded subpacket,
934 * usually 1024 samples per channel.
935 *
936 * @param q pointer to the COOKContext
937 * @param inbuffer pointer to the inbuffer
938 * @param outbuffer pointer to the outbuffer
939 */
940 240 static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
941 const uint8_t *inbuffer, float **outbuffer)
942 {
943 240 int sub_packet_size = p->size;
944 int res;
945
946 240 memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
947 240 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
948
949
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 240 if (p->joint_stereo) {
950
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 240 if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
951 ✗ return res;
952 } else {
953 ✗ if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
954 ✗ return res;
955
956 ✗ if (p->num_channels == 2) {
957 ✗ decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
958 ✗ if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
959 ✗ return res;
960 }
961 }
962
963
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 240 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
964 240 p->mono_previous_buffer1,
965 238 outbuffer ? outbuffer[p->ch_idx] : NULL);
966
967
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 240 if (p->num_channels == 2) {
968
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 240 if (p->joint_stereo)
969
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 240 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
970 240 p->mono_previous_buffer2,
971 238 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
972 else
973 ✗ mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
974 ✗ p->mono_previous_buffer2,
975 ✗ outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
976 }
977
978 240 return 0;
979 }
980
981
982 240 static int cook_decode_frame(AVCodecContext *avctx, AVFrame *frame,
983 int *got_frame_ptr, AVPacket *avpkt)
984 {
985 240 const uint8_t *buf = avpkt->data;
986 240 int buf_size = avpkt->size;
987 240 COOKContext *q = avctx->priv_data;
988 240 float **samples = NULL;
989 int i, ret;
990 240 int offset = 0;
991 240 int chidx = 0;
992
993
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 240 if (buf_size < avctx->block_align)
994 ✗ return buf_size;
995
996 /* get output buffer */
997
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 240 if (q->discarded_packets >= 2) {
998 238 frame->nb_samples = q->samples_per_channel;
999
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 238 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1000 ✗ return ret;
1001 238 samples = (float **)frame->extended_data;
1002 }
1003
1004 /* estimate subpacket sizes */
1005 240 q->subpacket[0].size = avctx->block_align;
1006
1007
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 240 for (i = 1; i < q->num_subpackets; i++) {
1008 ✗ q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
1009 ✗ q->subpacket[0].size -= q->subpacket[i].size + 1;
1010 ✗ if (q->subpacket[0].size < 0) {
1011 ✗ av_log(avctx, AV_LOG_DEBUG,
1012 "frame subpacket size total > avctx->block_align!\n");
1013 ✗ return AVERROR_INVALIDDATA;
1014 }
1015 }
1016
1017 /* decode supbackets */
1018
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 480 for (i = 0; i < q->num_subpackets; i++) {
1019 240 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
1020 240 q->subpacket[i].bits_per_subpdiv;
1021 240 q->subpacket[i].ch_idx = chidx;
1022 240 av_log(avctx, AV_LOG_DEBUG,
1023 "subpacket[%i] size %i js %i %i block_align %i\n",
1024 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
1025 avctx->block_align);
1026
1027
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 240 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1028 ✗ return ret;
1029 240 offset += q->subpacket[i].size;
1030 240 chidx += q->subpacket[i].num_channels;
1031 480 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1032 240 i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1033 }
1034
1035 /* Discard the first two frames: no valid audio. */
1036
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 240 if (q->discarded_packets < 2) {
1037 2 q->discarded_packets++;
1038 2 *got_frame_ptr = 0;
1039 2 return avctx->block_align;
1040 }
1041
1042 238 *got_frame_ptr = 1;
1043
1044 238 return avctx->block_align;
1045 }
1046
1047 6 static void dump_cook_context(COOKContext *q)
1048 {
1049 //int i=0;
1050 #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1051 ff_dlog(q->avctx, "COOKextradata\n");
1052 ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1053 6 if (q->subpacket[0].cookversion > STEREO) {
1054 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1055 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1056 }
1057 ff_dlog(q->avctx, "COOKContext\n");
1058 PRINT("nb_channels", q->avctx->ch_layout.nb_channels);
1059 PRINT("bit_rate", (int)q->avctx->bit_rate);
1060 PRINT("sample_rate", q->avctx->sample_rate);
1061 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1062 PRINT("subbands", q->subpacket[0].subbands);
1063 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1064 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1065 PRINT("numvector_size", q->subpacket[0].numvector_size);
1066 PRINT("total_subbands", q->subpacket[0].total_subbands);
1067 6 }
1068
1069 /**
1070 * Cook initialization
1071 *
1072 * @param avctx pointer to the AVCodecContext
1073 */
1074 6 static av_cold int cook_decode_init(AVCodecContext *avctx)
1075 {
1076 static AVOnce init_static_once = AV_ONCE_INIT;
1077 6 COOKContext *q = avctx->priv_data;
1078 GetByteContext gb;
1079 6 int s = 0;
1080 6 unsigned int channel_mask = 0;
1081 6 int samples_per_frame = 0;
1082 int ret;
1083 6 int channels = avctx->ch_layout.nb_channels;
1084
1085 6 q->avctx = avctx;
1086
1087 /* Take care of the codec specific extradata. */
1088
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 6 if (avctx->extradata_size < 8) {
1089 ✗ av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1090 ✗ return AVERROR_INVALIDDATA;
1091 }
1092 6 av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1093
1094 6 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1095
1096 /* Take data from the AVCodecContext (RM container). */
1097
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 6 if (!channels) {
1098 ✗ av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1099 ✗ return AVERROR_INVALIDDATA;
1100 }
1101
1102
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 6 if (avctx->block_align >= INT_MAX / 8)
1103 ✗ return AVERROR(EINVAL);
1104
1105 /* Initialize RNG. */
1106 6 av_lfg_init(&q->random_state, 0);
1107
1108 6 ff_audiodsp_init(&q->adsp);
1109
1110
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 12 while (bytestream2_get_bytes_left(&gb)) {
1111
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 6 if (s >= FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
1112 ✗ avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1113 ✗ return AVERROR_PATCHWELCOME;
1114 }
1115 /* 8 for mono, 16 for stereo, ? for multichannel
1116 Swap to right endianness so we don't need to care later on. */
1117 6 q->subpacket[s].cookversion = bytestream2_get_be32(&gb);
1118 6 samples_per_frame = bytestream2_get_be16(&gb);
1119 6 q->subpacket[s].subbands = bytestream2_get_be16(&gb);
1120 6 bytestream2_get_be32(&gb); // Unknown unused
1121 6 q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
1122
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 6 if (q->subpacket[s].js_subband_start >= 51) {
1123 ✗ av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
1124 ✗ return AVERROR_INVALIDDATA;
1125 }
1126 6 q->subpacket[s].js_vlc_bits = bytestream2_get_be16(&gb);
1127
1128 /* Initialize extradata related variables. */
1129 6 q->subpacket[s].samples_per_channel = samples_per_frame / channels;
1130 6 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1131
1132 /* Initialize default data states. */
1133 6 q->subpacket[s].log2_numvector_size = 5;
1134 6 q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1135 6 q->subpacket[s].num_channels = 1;
1136
1137 /* Initialize version-dependent variables */
1138
1139 6 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1140 q->subpacket[s].cookversion);
1141 6 q->subpacket[s].joint_stereo = 0;
1142
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 6 switch (q->subpacket[s].cookversion) {
1143 ✗ case MONO:
1144 ✗ if (channels != 1) {
1145 ✗ avpriv_request_sample(avctx, "Container channels != 1");
1146 ✗ return AVERROR_PATCHWELCOME;
1147 }
1148 ✗ av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1149 ✗ break;
1150 2 case STEREO:
1151
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 2 if (channels != 1) {
1152 ✗ q->subpacket[s].bits_per_subpdiv = 1;
1153 ✗ q->subpacket[s].num_channels = 2;
1154 }
1155 2 av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1156 2 break;
1157 4 case JOINT_STEREO:
1158
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 4 if (channels != 2) {
1159 ✗ avpriv_request_sample(avctx, "Container channels != 2");
1160 ✗ return AVERROR_PATCHWELCOME;
1161 }
1162 4 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1163
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 4 if (avctx->extradata_size >= 16) {
1164 4 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1165 4 q->subpacket[s].js_subband_start;
1166 4 q->subpacket[s].joint_stereo = 1;
1167 4 q->subpacket[s].num_channels = 2;
1168 }
1169
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 4 if (q->subpacket[s].samples_per_channel > 256) {
1170 4 q->subpacket[s].log2_numvector_size = 6;
1171 }
1172
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 4 if (q->subpacket[s].samples_per_channel > 512) {
1173 4 q->subpacket[s].log2_numvector_size = 7;
1174 }
1175 4 break;
1176 ✗ case MC_COOK:
1177 ✗ av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1178 ✗ channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
1179
1180 ✗ if (av_popcount64(q->subpacket[s].channel_mask) > 1) {
1181 ✗ q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1182 ✗ q->subpacket[s].js_subband_start;
1183 ✗ q->subpacket[s].joint_stereo = 1;
1184 ✗ q->subpacket[s].num_channels = 2;
1185 ✗ q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1186
1187 ✗ if (q->subpacket[s].samples_per_channel > 256) {
1188 ✗ q->subpacket[s].log2_numvector_size = 6;
1189 }
1190 ✗ if (q->subpacket[s].samples_per_channel > 512) {
1191 ✗ q->subpacket[s].log2_numvector_size = 7;
1192 }
1193 } else
1194 ✗ q->subpacket[s].samples_per_channel = samples_per_frame;
1195
1196 ✗ break;
1197 ✗ default:
1198 ✗ avpriv_request_sample(avctx, "Cook version %d",
1199 q->subpacket[s].cookversion);
1200 ✗ return AVERROR_PATCHWELCOME;
1201 }
1202
1203
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 6 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1204 ✗ av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1205 ✗ return AVERROR_INVALIDDATA;
1206 } else
1207 6 q->samples_per_channel = q->subpacket[0].samples_per_channel;
1208
1209
1210 /* Initialize variable relations */
1211 6 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1212
1213 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1214
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 6 if (q->subpacket[s].total_subbands > 53) {
1215 ✗ avpriv_request_sample(avctx, "total_subbands > 53");
1216 ✗ return AVERROR_PATCHWELCOME;
1217 }
1218
1219
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 6 if ((q->subpacket[s].js_vlc_bits > 6) ||
1220
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 6 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1221 ✗ av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1222 ✗ q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1223 ✗ return AVERROR_INVALIDDATA;
1224 }
1225
1226
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 6 if (q->subpacket[s].subbands > 50) {
1227 ✗ avpriv_request_sample(avctx, "subbands > 50");
1228 ✗ return AVERROR_PATCHWELCOME;
1229 }
1230
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 6 if (q->subpacket[s].subbands == 0) {
1231 ✗ avpriv_request_sample(avctx, "subbands = 0");
1232 ✗ return AVERROR_PATCHWELCOME;
1233 }
1234 6 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1235 6 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1236 6 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1237 6 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1238
1239
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 6 if (q->num_subpackets + q->subpacket[s].num_channels > channels) {
1240 ✗ av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, channels);
1241 ✗ return AVERROR_INVALIDDATA;
1242 }
1243
1244 6 q->num_subpackets++;
1245 6 s++;
1246 }
1247
1248 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1249
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 6 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1250
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 6 q->samples_per_channel != 1024) {
1251 ✗ avpriv_request_sample(avctx, "samples_per_channel = %d",
1252 q->samples_per_channel);
1253 ✗ return AVERROR_PATCHWELCOME;
1254 }
1255
1256 /* Generate tables */
1257 6 ff_thread_once(&init_static_once, init_pow2table);
1258 6 init_gain_table(q);
1259 6 init_cplscales_table(q);
1260
1261
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 6 if ((ret = init_cook_vlc_tables(q)))
1262 ✗ return ret;
1263
1264 /* Pad the databuffer with:
1265 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1266 AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1267 6 q->decoded_bytes_buffer =
1268 6 av_mallocz(avctx->block_align
1269 6 + DECODE_BYTES_PAD1(avctx->block_align)
1270 6 + AV_INPUT_BUFFER_PADDING_SIZE);
1271
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 6 if (!q->decoded_bytes_buffer)
1272 ✗ return AVERROR(ENOMEM);
1273
1274 /* Initialize transform. */
1275
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 6 if ((ret = init_cook_mlt(q)))
1276 ✗ return ret;
1277
1278 /* Initialize COOK signal arithmetic handling */
1279 if (1) {
1280 6 q->scalar_dequant = scalar_dequant_float;
1281 6 q->decouple = decouple_float;
1282 6 q->imlt_window = imlt_window_float;
1283 6 q->interpolate = interpolate_float;
1284 6 q->saturate_output = saturate_output_float;
1285 }
1286
1287 6 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1288 6 av_channel_layout_uninit(&avctx->ch_layout);
1289
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 6 if (channel_mask)
1290 ✗ av_channel_layout_from_mask(&avctx->ch_layout, channel_mask);
1291 else
1292 6 av_channel_layout_default(&avctx->ch_layout, channels);
1293
1294
1295 6 dump_cook_context(q);
1296
1297 6 return 0;
1298 }
1299
1300 const FFCodec ff_cook_decoder = {
1301 .p.name = "cook",
1302 CODEC_LONG_NAME("Cook / Cooker / Gecko (RealAudio G2)"),
1303 .p.type = AVMEDIA_TYPE_AUDIO,
1304 .p.id = AV_CODEC_ID_COOK,
1305 .priv_data_size = sizeof(COOKContext),
1306 .init = cook_decode_init,
1307 .close = cook_decode_close,
1308 FF_CODEC_DECODE_CB(cook_decode_frame),
1309 .p.capabilities = AV_CODEC_CAP_DR1,
1310 .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1311 AV_SAMPLE_FMT_NONE },
1312 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1313 };
1314