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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
48#include "audiodsp.h"
49#include "avcodec.h"
50#include "get_bits.h"
51#include "bytestream.h"
52#include "fft.h"
53#include "internal.h"
54#include "sinewin.h"
55#include "unary.h"
56
57#include "cookdata.h"
58
59/* the different Cook versions */
60#define MONO 0x1000001
61#define STEREO 0x1000002
62#define JOINT_STEREO 0x1000003
63#define MC_COOK 0x2000000 // multichannel Cook, not supported
64
65#define SUBBAND_SIZE 20
66#define MAX_SUBPACKETS 5
67
68typedef struct cook_gains {
69 int *now;
70 int *previous;
71} cook_gains;
72
73typedef struct COOKSubpacket {
74 int ch_idx;
75 int size;
76 int num_channels;
77 int cookversion;
78 int subbands;
79 int js_subband_start;
80 int js_vlc_bits;
81 int samples_per_channel;
82 int log2_numvector_size;
83 unsigned int channel_mask;
84 VLC channel_coupling;
85 int joint_stereo;
86 int bits_per_subpacket;
87 int bits_per_subpdiv;
88 int total_subbands;
89 int numvector_size; // 1 << log2_numvector_size;
90
91 float mono_previous_buffer1[1024];
92 float mono_previous_buffer2[1024];
93
94 cook_gains gains1;
95 cook_gains gains2;
96 int gain_1[9];
97 int gain_2[9];
98 int gain_3[9];
99 int gain_4[9];
100} COOKSubpacket;
101
102typedef struct cook {
103 /*
104 * The following 5 functions provide the lowlevel arithmetic on
105 * the internal audio buffers.
106 */
107 void (*scalar_dequant)(struct cook *q, int index, int quant_index,
108 int *subband_coef_index, int *subband_coef_sign,
109 float *mlt_p);
110
111 void (*decouple)(struct cook *q,
112 COOKSubpacket *p,
113 int subband,
114 float f1, float f2,
115 float *decode_buffer,
116 float *mlt_buffer1, float *mlt_buffer2);
117
118 void (*imlt_window)(struct cook *q, float *buffer1,
119 cook_gains *gains_ptr, float *previous_buffer);
120
121 void (*interpolate)(struct cook *q, float *buffer,
122 int gain_index, int gain_index_next);
123
124 void (*saturate_output)(struct cook *q, float *out);
125
126 AVCodecContext* avctx;
127 AudioDSPContext adsp;
128 GetBitContext gb;
129 /* stream data */
130 int num_vectors;
131 int samples_per_channel;
132 /* states */
133 AVLFG random_state;
134 int discarded_packets;
135
136 /* transform data */
137 FFTContext mdct_ctx;
138 float* mlt_window;
139
140 /* VLC data */
141 VLC envelope_quant_index[13];
142 VLC sqvh[7]; // scalar quantization
143
144 /* generate tables and related variables */
145 int gain_size_factor;
146 float gain_table[23];
147
148 /* data buffers */
149
150 uint8_t* decoded_bytes_buffer;
151 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
152 float decode_buffer_1[1024];
153 float decode_buffer_2[1024];
154 float decode_buffer_0[1060]; /* static allocation for joint decode */
155
156 const float *cplscales[5];
157 int num_subpackets;
158 COOKSubpacket subpacket[MAX_SUBPACKETS];
159} COOKContext;
160
161static float pow2tab[127];
162static float rootpow2tab[127];
163
164/*************** init functions ***************/
165
166/* table generator */
167static av_cold void init_pow2table(void)
168{
169 /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
170 int i;
171 static const float exp2_tab[2] = {1, M_SQRT2};
172 float exp2_val = powf(2, -63);
173 float root_val = powf(2, -32);
174 for (i = -63; i < 64; i++) {
175 if (!(i & 1))
176 root_val *= 2;
177 pow2tab[63 + i] = exp2_val;
178 rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
179 exp2_val *= 2;
180 }
181}
182
183/* table generator */
184static av_cold void init_gain_table(COOKContext *q)
185{
186 int i;
187 q->gain_size_factor = q->samples_per_channel / 8;
188 for (i = 0; i < 23; i++)
189 q->gain_table[i] = pow(pow2tab[i + 52],
190 (1.0 / (double) q->gain_size_factor));
191}
192
193
194static av_cold int init_cook_vlc_tables(COOKContext *q)
195{
196 int i, result;
197
198 result = 0;
199 for (i = 0; i < 13; i++) {
200 result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
201 envelope_quant_index_huffbits[i], 1, 1,
202 envelope_quant_index_huffcodes[i], 2, 2, 0);
203 }
204 av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
205 for (i = 0; i < 7; i++) {
206 result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
207 cvh_huffbits[i], 1, 1,
208 cvh_huffcodes[i], 2, 2, 0);
209 }
210
211 for (i = 0; i < q->num_subpackets; i++) {
212 if (q->subpacket[i].joint_stereo == 1) {
213 result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
214 (1 << q->subpacket[i].js_vlc_bits) - 1,
215 ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
216 ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
217 av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
218 }
219 }
220
221 av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
222 return result;
223}
224
225static av_cold int init_cook_mlt(COOKContext *q)
226{
227 int j, ret;
228 int mlt_size = q->samples_per_channel;
229
230 if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
231 return AVERROR(ENOMEM);
232
233 /* Initialize the MLT window: simple sine window. */
234 ff_sine_window_init(q->mlt_window, mlt_size);
235 for (j = 0; j < mlt_size; j++)
236 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
237
238 /* Initialize the MDCT. */
239 if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
240 av_freep(&q->mlt_window);
241 return ret;
242 }
243 av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
244 av_log2(mlt_size) + 1);
245
246 return 0;
247}
248
249static av_cold void init_cplscales_table(COOKContext *q)
250{
251 int i;
252 for (i = 0; i < 5; i++)
253 q->cplscales[i] = cplscales[i];
254}
255
256/*************** init functions end ***********/
257
258#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
259#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
260
261/**
262 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
263 * Why? No idea, some checksum/error detection method maybe.
264 *
265 * Out buffer size: extra bytes are needed to cope with
266 * padding/misalignment.
267 * Subpackets passed to the decoder can contain two, consecutive
268 * half-subpackets, of identical but arbitrary size.
269 * 1234 1234 1234 1234 extraA extraB
270 * Case 1: AAAA BBBB 0 0
271 * Case 2: AAAA ABBB BB-- 3 3
272 * Case 3: AAAA AABB BBBB 2 2
273 * Case 4: AAAA AAAB BBBB BB-- 1 5
274 *
275 * Nice way to waste CPU cycles.
276 *
277 * @param inbuffer pointer to byte array of indata
278 * @param out pointer to byte array of outdata
279 * @param bytes number of bytes
280 */
281static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
282{
283 static const uint32_t tab[4] = {
284 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
285 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
286 };
287 int i, off;
288 uint32_t c;
289 const uint32_t *buf;
290 uint32_t *obuf = (uint32_t *) out;
291 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
292 * I'm too lazy though, should be something like
293 * for (i = 0; i < bitamount / 64; i++)
294 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
295 * Buffer alignment needs to be checked. */
296
297 off = (intptr_t) inbuffer & 3;
298 buf = (const uint32_t *) (inbuffer - off);
299 c = tab[off];
300 bytes += 3 + off;
301 for (i = 0; i < bytes / 4; i++)
302 obuf[i] = c ^ buf[i];
303
304 return off;
305}
306
307static av_cold int cook_decode_close(AVCodecContext *avctx)
308{
309 int i;
310 COOKContext *q = avctx->priv_data;
311 av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
312
313 /* Free allocated memory buffers. */
314 av_freep(&q->mlt_window);
315 av_freep(&q->decoded_bytes_buffer);
316
317 /* Free the transform. */
318 ff_mdct_end(&q->mdct_ctx);
319
320 /* Free the VLC tables. */
321 for (i = 0; i < 13; i++)
322 ff_free_vlc(&q->envelope_quant_index[i]);
323 for (i = 0; i < 7; i++)
324 ff_free_vlc(&q->sqvh[i]);
325 for (i = 0; i < q->num_subpackets; i++)
326 ff_free_vlc(&q->subpacket[i].channel_coupling);
327
328 av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
329
330 return 0;
331}
332
333/**
334 * Fill the gain array for the timedomain quantization.
335 *
336 * @param gb pointer to the GetBitContext
337 * @param gaininfo array[9] of gain indexes
338 */
339static void decode_gain_info(GetBitContext *gb, int *gaininfo)
340{
341 int i, n;
342
343 n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update
344
345 i = 0;
346 while (n--) {
347 int index = get_bits(gb, 3);
348 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
349
350 while (i <= index)
351 gaininfo[i++] = gain;
352 }
353 while (i <= 8)
354 gaininfo[i++] = 0;
355}
356
357/**
358 * Create the quant index table needed for the envelope.
359 *
360 * @param q pointer to the COOKContext
361 * @param quant_index_table pointer to the array
362 */
363static int decode_envelope(COOKContext *q, COOKSubpacket *p,
364 int *quant_index_table)
365{
366 int i, j, vlc_index;
367
368 quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
369
370 for (i = 1; i < p->total_subbands; i++) {
371 vlc_index = i;
372 if (i >= p->js_subband_start * 2) {
373 vlc_index -= p->js_subband_start;
374 } else {
375 vlc_index /= 2;
376 if (vlc_index < 1)
377 vlc_index = 1;
378 }
379 if (vlc_index > 13)
380 vlc_index = 13; // the VLC tables >13 are identical to No. 13
381
382 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
383 q->envelope_quant_index[vlc_index - 1].bits, 2);
384 quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
385 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
386 av_log(q->avctx, AV_LOG_ERROR,
387 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
388 quant_index_table[i], i);
389 return AVERROR_INVALIDDATA;
390 }
391 }
392
393 return 0;
394}
395
396/**
397 * Calculate the category and category_index vector.
398 *
399 * @param q pointer to the COOKContext
400 * @param quant_index_table pointer to the array
401 * @param category pointer to the category array
402 * @param category_index pointer to the category_index array
403 */
404static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
405 int *category, int *category_index)
406{
407 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
408 int exp_index2[102] = { 0 };
409 int exp_index1[102] = { 0 };
410
411 int tmp_categorize_array[128 * 2] = { 0 };
412 int tmp_categorize_array1_idx = p->numvector_size;
413 int tmp_categorize_array2_idx = p->numvector_size;
414
415 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
416
417 if (bits_left > q->samples_per_channel)
418 bits_left = q->samples_per_channel +
419 ((bits_left - q->samples_per_channel) * 5) / 8;
420
421 bias = -32;
422
423 /* Estimate bias. */
424 for (i = 32; i > 0; i = i / 2) {
425 num_bits = 0;
426 index = 0;
427 for (j = p->total_subbands; j > 0; j--) {
428 exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
429 index++;
430 num_bits += expbits_tab[exp_idx];
431 }
432 if (num_bits >= bits_left - 32)
433 bias += i;
434 }
435
436 /* Calculate total number of bits. */
437 num_bits = 0;
438 for (i = 0; i < p->total_subbands; i++) {
439 exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
440 num_bits += expbits_tab[exp_idx];
441 exp_index1[i] = exp_idx;
442 exp_index2[i] = exp_idx;
443 }
444 tmpbias1 = tmpbias2 = num_bits;
445
446 for (j = 1; j < p->numvector_size; j++) {
447 if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
448 int max = -999999;
449 index = -1;
450 for (i = 0; i < p->total_subbands; i++) {
451 if (exp_index1[i] < 7) {
452 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
453 if (v >= max) {
454 max = v;
455 index = i;
456 }
457 }
458 }
459 if (index == -1)
460 break;
461 tmp_categorize_array[tmp_categorize_array1_idx++] = index;
462 tmpbias1 -= expbits_tab[exp_index1[index]] -
463 expbits_tab[exp_index1[index] + 1];
464 ++exp_index1[index];
465 } else { /* <--- */
466 int min = 999999;
467 index = -1;
468 for (i = 0; i < p->total_subbands; i++) {
469 if (exp_index2[i] > 0) {
470 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
471 if (v < min) {
472 min = v;
473 index = i;
474 }
475 }
476 }
477 if (index == -1)
478 break;
479 tmp_categorize_array[--tmp_categorize_array2_idx] = index;
480 tmpbias2 -= expbits_tab[exp_index2[index]] -
481 expbits_tab[exp_index2[index] - 1];
482 --exp_index2[index];
483 }
484 }
485
486 for (i = 0; i < p->total_subbands; i++)
487 category[i] = exp_index2[i];
488
489 for (i = 0; i < p->numvector_size - 1; i++)
490 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
491}
492
493
494/**
495 * Expand the category vector.
496 *
497 * @param q pointer to the COOKContext
498 * @param category pointer to the category array
499 * @param category_index pointer to the category_index array
500 */
501static inline void expand_category(COOKContext *q, int *category,
502 int *category_index)
503{
504 int i;
505 for (i = 0; i < q->num_vectors; i++)
506 {
507 int idx = category_index[i];
508 if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
509 --category[idx];
510 }
511}
512
513/**
514 * The real requantization of the mltcoefs
515 *
516 * @param q pointer to the COOKContext
517 * @param index index
518 * @param quant_index quantisation index
519 * @param subband_coef_index array of indexes to quant_centroid_tab
520 * @param subband_coef_sign signs of coefficients
521 * @param mlt_p pointer into the mlt buffer
522 */
523static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
524 int *subband_coef_index, int *subband_coef_sign,
525 float *mlt_p)
526{
527 int i;
528 float f1;
529
530 for (i = 0; i < SUBBAND_SIZE; i++) {
531 if (subband_coef_index[i]) {
532 f1 = quant_centroid_tab[index][subband_coef_index[i]];
533 if (subband_coef_sign[i])
534 f1 = -f1;
535 } else {
536 /* noise coding if subband_coef_index[i] == 0 */
537 f1 = dither_tab[index];
538 if (av_lfg_get(&q->random_state) < 0x80000000)
539 f1 = -f1;
540 }
541 mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
542 }
543}
544/**
545 * Unpack the subband_coef_index and subband_coef_sign vectors.
546 *
547 * @param q pointer to the COOKContext
548 * @param category pointer to the category array
549 * @param subband_coef_index array of indexes to quant_centroid_tab
550 * @param subband_coef_sign signs of coefficients
551 */
552static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
553 int *subband_coef_index, int *subband_coef_sign)
554{
555 int i, j;
556 int vlc, vd, tmp, result;
557
558 vd = vd_tab[category];
559 result = 0;
560 for (i = 0; i < vpr_tab[category]; i++) {
561 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
562 if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
563 vlc = 0;
564 result = 1;
565 }
566 for (j = vd - 1; j >= 0; j--) {
567 tmp = (vlc * invradix_tab[category]) / 0x100000;
568 subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
569 vlc = tmp;
570 }
571 for (j = 0; j < vd; j++) {
572 if (subband_coef_index[i * vd + j]) {
573 if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
574 subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
575 } else {
576 result = 1;
577 subband_coef_sign[i * vd + j] = 0;
578 }
579 } else {
580 subband_coef_sign[i * vd + j] = 0;
581 }
582 }
583 }
584 return result;
585}
586
587
588/**
589 * Fill the mlt_buffer with mlt coefficients.
590 *
591 * @param q pointer to the COOKContext
592 * @param category pointer to the category array
593 * @param quant_index_table pointer to the array
594 * @param mlt_buffer pointer to mlt coefficients
595 */
596static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
597 int *quant_index_table, float *mlt_buffer)
598{
599 /* A zero in this table means that the subband coefficient is
600 random noise coded. */
601 int subband_coef_index[SUBBAND_SIZE];
602 /* A zero in this table means that the subband coefficient is a
603 positive multiplicator. */
604 int subband_coef_sign[SUBBAND_SIZE];
605 int band, j;
606 int index = 0;
607
608 for (band = 0; band < p->total_subbands; band++) {
609 index = category[band];
610 if (category[band] < 7) {
611 if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
612 index = 7;
613 for (j = 0; j < p->total_subbands; j++)
614 category[band + j] = 7;
615 }
616 }
617 if (index >= 7) {
618 memset(subband_coef_index, 0, sizeof(subband_coef_index));
619 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
620 }
621 q->scalar_dequant(q, index, quant_index_table[band],
622 subband_coef_index, subband_coef_sign,
623 &mlt_buffer[band * SUBBAND_SIZE]);
624 }
625
626 /* FIXME: should this be removed, or moved into loop above? */
627 if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
628 return;
629}
630
631
632static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
633{
634 int category_index[128] = { 0 };
635 int category[128] = { 0 };
636 int quant_index_table[102];
637 int res, i;
638
639 if ((res = decode_envelope(q, p, quant_index_table)) < 0)
640 return res;
641 q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
642 categorize(q, p, quant_index_table, category, category_index);
643 expand_category(q, category, category_index);
644 for (i=0; i<p->total_subbands; i++) {
645 if (category[i] > 7)
646 return AVERROR_INVALIDDATA;
647 }
648 decode_vectors(q, p, category, quant_index_table, mlt_buffer);
649
650 return 0;
651}
652
653
654/**
655 * the actual requantization of the timedomain samples
656 *
657 * @param q pointer to the COOKContext
658 * @param buffer pointer to the timedomain buffer
659 * @param gain_index index for the block multiplier
660 * @param gain_index_next index for the next block multiplier
661 */
662static void interpolate_float(COOKContext *q, float *buffer,
663 int gain_index, int gain_index_next)
664{
665 int i;
666 float fc1, fc2;
667 fc1 = pow2tab[gain_index + 63];
668
669 if (gain_index == gain_index_next) { // static gain
670 for (i = 0; i < q->gain_size_factor; i++)
671 buffer[i] *= fc1;
672 } else { // smooth gain
673 fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
674 for (i = 0; i < q->gain_size_factor; i++) {
675 buffer[i] *= fc1;
676 fc1 *= fc2;
677 }
678 }
679}
680
681/**
682 * Apply transform window, overlap buffers.
683 *
684 * @param q pointer to the COOKContext
685 * @param inbuffer pointer to the mltcoefficients
686 * @param gains_ptr current and previous gains
687 * @param previous_buffer pointer to the previous buffer to be used for overlapping
688 */
689static void imlt_window_float(COOKContext *q, float *inbuffer,
690 cook_gains *gains_ptr, float *previous_buffer)
691{
692 const float fc = pow2tab[gains_ptr->previous[0] + 63];
693 int i;
694 /* The weird thing here, is that the two halves of the time domain
695 * buffer are swapped. Also, the newest data, that we save away for
696 * next frame, has the wrong sign. Hence the subtraction below.
697 * Almost sounds like a complex conjugate/reverse data/FFT effect.
698 */
699
700 /* Apply window and overlap */
701 for (i = 0; i < q->samples_per_channel; i++)
702 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
703 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
704}
705
706/**
707 * The modulated lapped transform, this takes transform coefficients
708 * and transforms them into timedomain samples.
709 * Apply transform window, overlap buffers, apply gain profile
710 * and buffer management.
711 *
712 * @param q pointer to the COOKContext
713 * @param inbuffer pointer to the mltcoefficients
714 * @param gains_ptr current and previous gains
715 * @param previous_buffer pointer to the previous buffer to be used for overlapping
716 */
717static void imlt_gain(COOKContext *q, float *inbuffer,
718 cook_gains *gains_ptr, float *previous_buffer)
719{
720 float *buffer0 = q->mono_mdct_output;
721 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
722 int i;
723
724 /* Inverse modified discrete cosine transform */
725 q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
726
727 q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
728
729 /* Apply gain profile */
730 for (i = 0; i < 8; i++)
731 if (gains_ptr->now[i] || gains_ptr->now[i + 1])
732 q->interpolate(q, &buffer1[q->gain_size_factor * i],
733 gains_ptr->now[i], gains_ptr->now[i + 1]);
734
735 /* Save away the current to be previous block. */
736 memcpy(previous_buffer, buffer0,
737 q->samples_per_channel * sizeof(*previous_buffer));
738}
739
740
741/**
742 * function for getting the jointstereo coupling information
743 *
744 * @param q pointer to the COOKContext
745 * @param decouple_tab decoupling array
746 */
747static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
748{
749 int i;
750 int vlc = get_bits1(&q->gb);
751 int start = cplband[p->js_subband_start];
752 int end = cplband[p->subbands - 1];
753 int length = end - start + 1;
754
755 if (start > end)
756 return 0;
757
758 if (vlc)
759 for (i = 0; i < length; i++)
760 decouple_tab[start + i] = get_vlc2(&q->gb,
761 p->channel_coupling.table,
762 p->channel_coupling.bits, 2);
763 else
764 for (i = 0; i < length; i++) {
765 int v = get_bits(&q->gb, p->js_vlc_bits);
766 if (v == (1<<p->js_vlc_bits)-1) {
767 av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
768 return AVERROR_INVALIDDATA;
769 }
770 decouple_tab[start + i] = v;
771 }
772 return 0;
773}
774
775/**
776 * function decouples a pair of signals from a single signal via multiplication.
777 *
778 * @param q pointer to the COOKContext
779 * @param subband index of the current subband
780 * @param f1 multiplier for channel 1 extraction
781 * @param f2 multiplier for channel 2 extraction
782 * @param decode_buffer input buffer
783 * @param mlt_buffer1 pointer to left channel mlt coefficients
784 * @param mlt_buffer2 pointer to right channel mlt coefficients
785 */
786static void decouple_float(COOKContext *q,
787 COOKSubpacket *p,
788 int subband,
789 float f1, float f2,
790 float *decode_buffer,
791 float *mlt_buffer1, float *mlt_buffer2)
792{
793 int j, tmp_idx;
794 for (j = 0; j < SUBBAND_SIZE; j++) {
795 tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
796 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
797 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
798 }
799}
800
801/**
802 * function for decoding joint stereo data
803 *
804 * @param q pointer to the COOKContext
805 * @param mlt_buffer1 pointer to left channel mlt coefficients
806 * @param mlt_buffer2 pointer to right channel mlt coefficients
807 */
808static int joint_decode(COOKContext *q, COOKSubpacket *p,
809 float *mlt_buffer_left, float *mlt_buffer_right)
810{
811 int i, j, res;
812 int decouple_tab[SUBBAND_SIZE] = { 0 };
813 float *decode_buffer = q->decode_buffer_0;
814 int idx, cpl_tmp;
815 float f1, f2;
816 const float *cplscale;
817
818 memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
819
820 /* Make sure the buffers are zeroed out. */
821 memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
822 memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
823 if ((res = decouple_info(q, p, decouple_tab)) < 0)
824 return res;
825 if ((res = mono_decode(q, p, decode_buffer)) < 0)
826 return res;
827 /* The two channels are stored interleaved in decode_buffer. */
828 for (i = 0; i < p->js_subband_start; i++) {
829 for (j = 0; j < SUBBAND_SIZE; j++) {
830 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
831 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
832 }
833 }
834
835 /* When we reach js_subband_start (the higher frequencies)
836 the coefficients are stored in a coupling scheme. */
837 idx = (1 << p->js_vlc_bits) - 1;
838 for (i = p->js_subband_start; i < p->subbands; i++) {
839 cpl_tmp = cplband[i];
840 idx -= decouple_tab[cpl_tmp];
841 cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
842 f1 = cplscale[decouple_tab[cpl_tmp] + 1];
843 f2 = cplscale[idx];
844 q->decouple(q, p, i, f1, f2, decode_buffer,
845 mlt_buffer_left, mlt_buffer_right);
846 idx = (1 << p->js_vlc_bits) - 1;
847 }
848
849 return 0;
850}
851
852/**
853 * First part of subpacket decoding:
854 * decode raw stream bytes and read gain info.
855 *
856 * @param q pointer to the COOKContext
857 * @param inbuffer pointer to raw stream data
858 * @param gains_ptr array of current/prev gain pointers
859 */
860static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
861 const uint8_t *inbuffer,
862 cook_gains *gains_ptr)
863{
864 int offset;
865
866 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
867 p->bits_per_subpacket / 8);
868 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
869 p->bits_per_subpacket);
870 decode_gain_info(&q->gb, gains_ptr->now);
871
872 /* Swap current and previous gains */
873 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
874}
875
876/**
877 * Saturate the output signal and interleave.
878 *
879 * @param q pointer to the COOKContext
880 * @param out pointer to the output vector
881 */
882static void saturate_output_float(COOKContext *q, float *out)
883{
884 q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
885 FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
886}
887
888
889/**
890 * Final part of subpacket decoding:
891 * Apply modulated lapped transform, gain compensation,
892 * clip and convert to integer.
893 *
894 * @param q pointer to the COOKContext
895 * @param decode_buffer pointer to the mlt coefficients
896 * @param gains_ptr array of current/prev gain pointers
897 * @param previous_buffer pointer to the previous buffer to be used for overlapping
898 * @param out pointer to the output buffer
899 */
900static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
901 cook_gains *gains_ptr, float *previous_buffer,
902 float *out)
903{
904 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
905 if (out)
906 q->saturate_output(q, out);
907}
908
909
910/**
911 * Cook subpacket decoding. This function returns one decoded subpacket,
912 * usually 1024 samples per channel.
913 *
914 * @param q pointer to the COOKContext
915 * @param inbuffer pointer to the inbuffer
916 * @param outbuffer pointer to the outbuffer
917 */
918static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
919 const uint8_t *inbuffer, float **outbuffer)
920{
921 int sub_packet_size = p->size;
922 int res;
923
924 memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
925 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
926
927 if (p->joint_stereo) {
928 if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
929 return res;
930 } else {
931 if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
932 return res;
933
934 if (p->num_channels == 2) {
935 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
936 if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
937 return res;
938 }
939 }
940
941 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
942 p->mono_previous_buffer1,
943 outbuffer ? outbuffer[p->ch_idx] : NULL);
944
945 if (p->num_channels == 2) {
946 if (p->joint_stereo)
947 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
948 p->mono_previous_buffer2,
949 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
950 else
951 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
952 p->mono_previous_buffer2,
953 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
954 }
955
956 return 0;
957}
958
959
960static int cook_decode_frame(AVCodecContext *avctx, void *data,
961 int *got_frame_ptr, AVPacket *avpkt)
962{
963 AVFrame *frame = data;
964 const uint8_t *buf = avpkt->data;
965 int buf_size = avpkt->size;
966 COOKContext *q = avctx->priv_data;
967 float **samples = NULL;
968 int i, ret;
969 int offset = 0;
970 int chidx = 0;
971
972 if (buf_size < avctx->block_align)
973 return buf_size;
974
975 /* get output buffer */
976 if (q->discarded_packets >= 2) {
977 frame->nb_samples = q->samples_per_channel;
978 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
979 return ret;
980 samples = (float **)frame->extended_data;
981 }
982
983 /* estimate subpacket sizes */
984 q->subpacket[0].size = avctx->block_align;
985
986 for (i = 1; i < q->num_subpackets; i++) {
987 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
988 q->subpacket[0].size -= q->subpacket[i].size + 1;
989 if (q->subpacket[0].size < 0) {
990 av_log(avctx, AV_LOG_DEBUG,
991 "frame subpacket size total > avctx->block_align!\n");
992 return AVERROR_INVALIDDATA;
993 }
994 }
995
996 /* decode supbackets */
997 for (i = 0; i < q->num_subpackets; i++) {
998 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
999 q->subpacket[i].bits_per_subpdiv;
1000 q->subpacket[i].ch_idx = chidx;
1001 av_log(avctx, AV_LOG_DEBUG,
1002 "subpacket[%i] size %i js %i %i block_align %i\n",
1003 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
1004 avctx->block_align);
1005
1006 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1007 return ret;
1008 offset += q->subpacket[i].size;
1009 chidx += q->subpacket[i].num_channels;
1010 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1011 i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1012 }
1013
1014 /* Discard the first two frames: no valid audio. */
1015 if (q->discarded_packets < 2) {
1016 q->discarded_packets++;
1017 *got_frame_ptr = 0;
1018 return avctx->block_align;
1019 }
1020
1021 *got_frame_ptr = 1;
1022
1023 return avctx->block_align;
1024}
1025
1026static void dump_cook_context(COOKContext *q)
1027{
1028 //int i=0;
1029#define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1030 ff_dlog(q->avctx, "COOKextradata\n");
1031 ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1032 if (q->subpacket[0].cookversion > STEREO) {
1033 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1034 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1035 }
1036 ff_dlog(q->avctx, "COOKContext\n");
1037 PRINT("nb_channels", q->avctx->channels);
1038 PRINT("bit_rate", (int)q->avctx->bit_rate);
1039 PRINT("sample_rate", q->avctx->sample_rate);
1040 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1041 PRINT("subbands", q->subpacket[0].subbands);
1042 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1043 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1044 PRINT("numvector_size", q->subpacket[0].numvector_size);
1045 PRINT("total_subbands", q->subpacket[0].total_subbands);
1046}
1047
1048/**
1049 * Cook initialization
1050 *
1051 * @param avctx pointer to the AVCodecContext
1052 */
1053static av_cold int cook_decode_init(AVCodecContext *avctx)
1054{
1055 COOKContext *q = avctx->priv_data;
1056 GetByteContext gb;
1057 int s = 0;
1058 unsigned int channel_mask = 0;
1059 int samples_per_frame = 0;
1060 int ret;
1061 q->avctx = avctx;
1062
1063 /* Take care of the codec specific extradata. */
1064 if (avctx->extradata_size < 8) {
1065 av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1066 return AVERROR_INVALIDDATA;
1067 }
1068 av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1069
1070 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1071
1072 /* Take data from the AVCodecContext (RM container). */
1073 if (!avctx->channels) {
1074 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1075 return AVERROR_INVALIDDATA;
1076 }
1077
1078 /* Initialize RNG. */
1079 av_lfg_init(&q->random_state, 0);
1080
1081 ff_audiodsp_init(&q->adsp);
1082
1083 while (bytestream2_get_bytes_left(&gb)) {
1084 /* 8 for mono, 16 for stereo, ? for multichannel
1085 Swap to right endianness so we don't need to care later on. */
1086 q->subpacket[s].cookversion = bytestream2_get_be32(&gb);
1087 samples_per_frame = bytestream2_get_be16(&gb);
1088 q->subpacket[s].subbands = bytestream2_get_be16(&gb);
1089 bytestream2_get_be32(&gb); // Unknown unused
1090 q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
1091 if (q->subpacket[s].js_subband_start >= 51) {
1092 av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
1093 return AVERROR_INVALIDDATA;
1094 }
1095 q->subpacket[s].js_vlc_bits = bytestream2_get_be16(&gb);
1096
1097 /* Initialize extradata related variables. */
1098 q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1099 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1100
1101 /* Initialize default data states. */
1102 q->subpacket[s].log2_numvector_size = 5;
1103 q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1104 q->subpacket[s].num_channels = 1;
1105
1106 /* Initialize version-dependent variables */
1107
1108 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1109 q->subpacket[s].cookversion);
1110 q->subpacket[s].joint_stereo = 0;
1111 switch (q->subpacket[s].cookversion) {
1112 case MONO:
1113 if (avctx->channels != 1) {
1114 avpriv_request_sample(avctx, "Container channels != 1");
1115 return AVERROR_PATCHWELCOME;
1116 }
1117 av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1118 break;
1119 case STEREO:
1120 if (avctx->channels != 1) {
1121 q->subpacket[s].bits_per_subpdiv = 1;
1122 q->subpacket[s].num_channels = 2;
1123 }
1124 av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1125 break;
1126 case JOINT_STEREO:
1127 if (avctx->channels != 2) {
1128 avpriv_request_sample(avctx, "Container channels != 2");
1129 return AVERROR_PATCHWELCOME;
1130 }
1131 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1132 if (avctx->extradata_size >= 16) {
1133 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1134 q->subpacket[s].js_subband_start;
1135 q->subpacket[s].joint_stereo = 1;
1136 q->subpacket[s].num_channels = 2;
1137 }
1138 if (q->subpacket[s].samples_per_channel > 256) {
1139 q->subpacket[s].log2_numvector_size = 6;
1140 }
1141 if (q->subpacket[s].samples_per_channel > 512) {
1142 q->subpacket[s].log2_numvector_size = 7;
1143 }
1144 break;
1145 case MC_COOK:
1146 av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1147 channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
1148
1149 if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
1150 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1151 q->subpacket[s].js_subband_start;
1152 q->subpacket[s].joint_stereo = 1;
1153 q->subpacket[s].num_channels = 2;
1154 q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1155
1156 if (q->subpacket[s].samples_per_channel > 256) {
1157 q->subpacket[s].log2_numvector_size = 6;
1158 }
1159 if (q->subpacket[s].samples_per_channel > 512) {
1160 q->subpacket[s].log2_numvector_size = 7;
1161 }
1162 } else
1163 q->subpacket[s].samples_per_channel = samples_per_frame;
1164
1165 break;
1166 default:
1167 avpriv_request_sample(avctx, "Cook version %d",
1168 q->subpacket[s].cookversion);
1169 return AVERROR_PATCHWELCOME;
1170 }
1171
1172 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1173 av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1174 return AVERROR_INVALIDDATA;
1175 } else
1176 q->samples_per_channel = q->subpacket[0].samples_per_channel;
1177
1178
1179 /* Initialize variable relations */
1180 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1181
1182 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1183 if (q->subpacket[s].total_subbands > 53) {
1184 avpriv_request_sample(avctx, "total_subbands > 53");
1185 return AVERROR_PATCHWELCOME;
1186 }
1187
1188 if ((q->subpacket[s].js_vlc_bits > 6) ||
1189 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1190 av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1191 q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1192 return AVERROR_INVALIDDATA;
1193 }
1194
1195 if (q->subpacket[s].subbands > 50) {
1196 avpriv_request_sample(avctx, "subbands > 50");
1197 return AVERROR_PATCHWELCOME;
1198 }
1199 if (q->subpacket[s].subbands == 0) {
1200 avpriv_request_sample(avctx, "subbands = 0");
1201 return AVERROR_PATCHWELCOME;
1202 }
1203 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1204 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1205 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1206 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1207
1208 if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
1209 av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
1210 return AVERROR_INVALIDDATA;
1211 }
1212
1213 q->num_subpackets++;
1214 s++;
1215 if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
1216 avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1217 return AVERROR_PATCHWELCOME;
1218 }
1219 }
1220 /* Generate tables */
1221 init_pow2table();
1222 init_gain_table(q);
1223 init_cplscales_table(q);
1224
1225 if ((ret = init_cook_vlc_tables(q)))
1226 return ret;
1227
1228
1229 if (avctx->block_align >= UINT_MAX / 2)
1230 return AVERROR(EINVAL);
1231
1232 /* Pad the databuffer with:
1233 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1234 AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1235 q->decoded_bytes_buffer =
1236 av_mallocz(avctx->block_align
1237 + DECODE_BYTES_PAD1(avctx->block_align)
1238 + AV_INPUT_BUFFER_PADDING_SIZE);
1239 if (!q->decoded_bytes_buffer)
1240 return AVERROR(ENOMEM);
1241
1242 /* Initialize transform. */
1243 if ((ret = init_cook_mlt(q)))
1244 return ret;
1245
1246 /* Initialize COOK signal arithmetic handling */
1247 if (1) {
1248 q->scalar_dequant = scalar_dequant_float;
1249 q->decouple = decouple_float;
1250 q->imlt_window = imlt_window_float;
1251 q->interpolate = interpolate_float;
1252 q->saturate_output = saturate_output_float;
1253 }
1254
1255 /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1256 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1257 q->samples_per_channel != 1024) {
1258 avpriv_request_sample(avctx, "samples_per_channel = %d",
1259 q->samples_per_channel);
1260 return AVERROR_PATCHWELCOME;
1261 }
1262
1263 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1264 if (channel_mask)
1265 avctx->channel_layout = channel_mask;
1266 else
1267 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1268
1269
1270 dump_cook_context(q);
1271
1272 return 0;
1273}
1274
1275AVCodec ff_cook_decoder = {
1276 .name = "cook",
1277 .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1278 .type = AVMEDIA_TYPE_AUDIO,
1279 .id = AV_CODEC_ID_COOK,
1280 .priv_data_size = sizeof(COOKContext),
1281 .init = cook_decode_init,
1282 .close = cook_decode_close,
1283 .decode = cook_decode_frame,
1284 .capabilities = AV_CODEC_CAP_DR1,
1285 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1286 AV_SAMPLE_FMT_NONE },
1287};
1288
generated by cgit at 2024-05-17 08:25:28 (GMT)