cook: move samples_per_frame from COOKSubpacket to where it is used
[libav.git] / libavcodec / cook.c
1 /*
2 * COOK compatible decoder
3 * Copyright (c) 2003 Sascha Sommer
4 * Copyright (c) 2005 Benjamin Larsson
5 *
6 * This file is part of Libav.
7 *
8 * Libav is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * Libav is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with Libav; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23 /**
24 * @file
25 * Cook compatible decoder. Bastardization of the G.722.1 standard.
26 * This decoder handles RealNetworks, RealAudio G2 data.
27 * Cook is identified by the codec name cook in RM files.
28 *
29 * To use this decoder, a calling application must supply the extradata
30 * bytes provided from the RM container; 8+ bytes for mono streams and
31 * 16+ for stereo streams (maybe more).
32 *
33 * Codec technicalities (all this assume a buffer length of 1024):
34 * Cook works with several different techniques to achieve its compression.
35 * In the timedomain the buffer is divided into 8 pieces and quantized. If
36 * two neighboring pieces have different quantization index a smooth
37 * quantization curve is used to get a smooth overlap between the different
38 * pieces.
39 * To get to the transformdomain Cook uses a modulated lapped transform.
40 * The transform domain has 50 subbands with 20 elements each. This
41 * means only a maximum of 50*20=1000 coefficients are used out of the 1024
42 * available.
43 */
44
45 #include "libavutil/lfg.h"
46 #include "avcodec.h"
47 #include "get_bits.h"
48 #include "dsputil.h"
49 #include "bytestream.h"
50 #include "fft.h"
51 #include "libavutil/audioconvert.h"
52 #include "sinewin.h"
53
54 #include "cookdata.h"
55
56 /* the different Cook versions */
57 #define MONO 0x1000001
58 #define STEREO 0x1000002
59 #define JOINT_STEREO 0x1000003
60 #define MC_COOK 0x2000000 // multichannel Cook, not supported
61
62 #define SUBBAND_SIZE 20
63 #define MAX_SUBPACKETS 5
64
65 typedef struct {
66 int *now;
67 int *previous;
68 } cook_gains;
69
70 typedef struct {
71 int ch_idx;
72 int size;
73 int num_channels;
74 int cookversion;
75 int subbands;
76 int js_subband_start;
77 int js_vlc_bits;
78 int samples_per_channel;
79 int log2_numvector_size;
80 unsigned int channel_mask;
81 VLC channel_coupling;
82 int joint_stereo;
83 int bits_per_subpacket;
84 int bits_per_subpdiv;
85 int total_subbands;
86 int numvector_size; // 1 << log2_numvector_size;
87
88 float mono_previous_buffer1[1024];
89 float mono_previous_buffer2[1024];
90
91 cook_gains gains1;
92 cook_gains gains2;
93 int gain_1[9];
94 int gain_2[9];
95 int gain_3[9];
96 int gain_4[9];
97 } COOKSubpacket;
98
99 typedef struct cook {
100 /*
101 * The following 5 functions provide the lowlevel arithmetic on
102 * the internal audio buffers.
103 */
104 void (*scalar_dequant)(struct cook *q, int index, int quant_index,
105 int *subband_coef_index, int *subband_coef_sign,
106 float *mlt_p);
107
108 void (*decouple)(struct cook *q,
109 COOKSubpacket *p,
110 int subband,
111 float f1, float f2,
112 float *decode_buffer,
113 float *mlt_buffer1, float *mlt_buffer2);
114
115 void (*imlt_window)(struct cook *q, float *buffer1,
116 cook_gains *gains_ptr, float *previous_buffer);
117
118 void (*interpolate)(struct cook *q, float *buffer,
119 int gain_index, int gain_index_next);
120
121 void (*saturate_output)(struct cook *q, float *out);
122
123 AVCodecContext* avctx;
124 DSPContext dsp;
125 AVFrame frame;
126 GetBitContext gb;
127 /* stream data */
128 int num_vectors;
129 int samples_per_channel;
130 /* states */
131 AVLFG random_state;
132 int discarded_packets;
133
134 /* transform data */
135 FFTContext mdct_ctx;
136 float* mlt_window;
137
138 /* VLC data */
139 VLC envelope_quant_index[13];
140 VLC sqvh[7]; // scalar quantization
141
142 /* generatable tables and related variables */
143 int gain_size_factor;
144 float gain_table[23];
145
146 /* data buffers */
147
148 uint8_t* decoded_bytes_buffer;
149 DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
150 float decode_buffer_1[1024];
151 float decode_buffer_2[1024];
152 float decode_buffer_0[1060]; /* static allocation for joint decode */
153
154 const float *cplscales[5];
155 int num_subpackets;
156 COOKSubpacket subpacket[MAX_SUBPACKETS];
157 } COOKContext;
158
159 static float pow2tab[127];
160 static float rootpow2tab[127];
161
162 /*************** init functions ***************/
163
164 /* table generator */
165 static av_cold void init_pow2table(void)
166 {
167 int i;
168 for (i = -63; i < 64; i++) {
169 pow2tab[63 + i] = pow(2, i);
170 rootpow2tab[63 + i] = sqrt(pow(2, i));
171 }
172 }
173
174 /* table generator */
175 static av_cold void init_gain_table(COOKContext *q)
176 {
177 int i;
178 q->gain_size_factor = q->samples_per_channel / 8;
179 for (i = 0; i < 23; i++)
180 q->gain_table[i] = pow(pow2tab[i + 52],
181 (1.0 / (double) q->gain_size_factor));
182 }
183
184
185 static av_cold int init_cook_vlc_tables(COOKContext *q)
186 {
187 int i, result;
188
189 result = 0;
190 for (i = 0; i < 13; i++) {
191 result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
192 envelope_quant_index_huffbits[i], 1, 1,
193 envelope_quant_index_huffcodes[i], 2, 2, 0);
194 }
195 av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
196 for (i = 0; i < 7; i++) {
197 result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
198 cvh_huffbits[i], 1, 1,
199 cvh_huffcodes[i], 2, 2, 0);
200 }
201
202 for (i = 0; i < q->num_subpackets; i++) {
203 if (q->subpacket[i].joint_stereo == 1) {
204 result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
205 (1 << q->subpacket[i].js_vlc_bits) - 1,
206 ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
207 ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
208 av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
209 }
210 }
211
212 av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
213 return result;
214 }
215
216 static av_cold int init_cook_mlt(COOKContext *q)
217 {
218 int j, ret;
219 int mlt_size = q->samples_per_channel;
220
221 if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
222 return AVERROR(ENOMEM);
223
224 /* Initialize the MLT window: simple sine window. */
225 ff_sine_window_init(q->mlt_window, mlt_size);
226 for (j = 0; j < mlt_size; j++)
227 q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
228
229 /* Initialize the MDCT. */
230 if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
231 av_free(q->mlt_window);
232 return ret;
233 }
234 av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
235 av_log2(mlt_size) + 1);
236
237 return 0;
238 }
239
240 static av_cold void init_cplscales_table(COOKContext *q)
241 {
242 int i;
243 for (i = 0; i < 5; i++)
244 q->cplscales[i] = cplscales[i];
245 }
246
247 /*************** init functions end ***********/
248
249 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
250 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
251
252 /**
253 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
254 * Why? No idea, some checksum/error detection method maybe.
255 *
256 * Out buffer size: extra bytes are needed to cope with
257 * padding/misalignment.
258 * Subpackets passed to the decoder can contain two, consecutive
259 * half-subpackets, of identical but arbitrary size.
260 * 1234 1234 1234 1234 extraA extraB
261 * Case 1: AAAA BBBB 0 0
262 * Case 2: AAAA ABBB BB-- 3 3
263 * Case 3: AAAA AABB BBBB 2 2
264 * Case 4: AAAA AAAB BBBB BB-- 1 5
265 *
266 * Nice way to waste CPU cycles.
267 *
268 * @param inbuffer pointer to byte array of indata
269 * @param out pointer to byte array of outdata
270 * @param bytes number of bytes
271 */
272 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
273 {
274 static const uint32_t tab[4] = {
275 AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
276 AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
277 };
278 int i, off;
279 uint32_t c;
280 const uint32_t *buf;
281 uint32_t *obuf = (uint32_t *) out;
282 /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
283 * I'm too lazy though, should be something like
284 * for (i = 0; i < bitamount / 64; i++)
285 * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
286 * Buffer alignment needs to be checked. */
287
288 off = (intptr_t) inbuffer & 3;
289 buf = (const uint32_t *) (inbuffer - off);
290 c = tab[off];
291 bytes += 3 + off;
292 for (i = 0; i < bytes / 4; i++)
293 obuf[i] = c ^ buf[i];
294
295 return off;
296 }
297
298 static av_cold int cook_decode_close(AVCodecContext *avctx)
299 {
300 int i;
301 COOKContext *q = avctx->priv_data;
302 av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
303
304 /* Free allocated memory buffers. */
305 av_free(q->mlt_window);
306 av_free(q->decoded_bytes_buffer);
307
308 /* Free the transform. */
309 ff_mdct_end(&q->mdct_ctx);
310
311 /* Free the VLC tables. */
312 for (i = 0; i < 13; i++)
313 ff_free_vlc(&q->envelope_quant_index[i]);
314 for (i = 0; i < 7; i++)
315 ff_free_vlc(&q->sqvh[i]);
316 for (i = 0; i < q->num_subpackets; i++)
317 ff_free_vlc(&q->subpacket[i].channel_coupling);
318
319 av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
320
321 return 0;
322 }
323
324 /**
325 * Fill the gain array for the timedomain quantization.
326 *
327 * @param gb pointer to the GetBitContext
328 * @param gaininfo array[9] of gain indexes
329 */
330 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
331 {
332 int i, n;
333
334 while (get_bits1(gb)) {
335 /* NOTHING */
336 }
337
338 n = get_bits_count(gb) - 1; // amount of elements*2 to update
339
340 i = 0;
341 while (n--) {
342 int index = get_bits(gb, 3);
343 int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
344
345 while (i <= index)
346 gaininfo[i++] = gain;
347 }
348 while (i <= 8)
349 gaininfo[i++] = 0;
350 }
351
352 /**
353 * Create the quant index table needed for the envelope.
354 *
355 * @param q pointer to the COOKContext
356 * @param quant_index_table pointer to the array
357 */
358 static int decode_envelope(COOKContext *q, COOKSubpacket *p,
359 int *quant_index_table)
360 {
361 int i, j, vlc_index;
362
363 quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
364
365 for (i = 1; i < p->total_subbands; i++) {
366 vlc_index = i;
367 if (i >= p->js_subband_start * 2) {
368 vlc_index -= p->js_subband_start;
369 } else {
370 vlc_index /= 2;
371 if (vlc_index < 1)
372 vlc_index = 1;
373 }
374 if (vlc_index > 13)
375 vlc_index = 13; // the VLC tables >13 are identical to No. 13
376
377 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
378 q->envelope_quant_index[vlc_index - 1].bits, 2);
379 quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
380 if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
381 av_log(q->avctx, AV_LOG_ERROR,
382 "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
383 quant_index_table[i], i);
384 return AVERROR_INVALIDDATA;
385 }
386 }
387
388 return 0;
389 }
390
391 /**
392 * Calculate the category and category_index vector.
393 *
394 * @param q pointer to the COOKContext
395 * @param quant_index_table pointer to the array
396 * @param category pointer to the category array
397 * @param category_index pointer to the category_index array
398 */
399 static void categorize(COOKContext *q, COOKSubpacket *p, int *quant_index_table,
400 int *category, int *category_index)
401 {
402 int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
403 int exp_index2[102] = { 0 };
404 int exp_index1[102] = { 0 };
405
406 int tmp_categorize_array[128 * 2] = { 0 };
407 int tmp_categorize_array1_idx = p->numvector_size;
408 int tmp_categorize_array2_idx = p->numvector_size;
409
410 bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
411
412 if (bits_left > q->samples_per_channel)
413 bits_left = q->samples_per_channel +
414 ((bits_left - q->samples_per_channel) * 5) / 8;
415
416 bias = -32;
417
418 /* Estimate bias. */
419 for (i = 32; i > 0; i = i / 2) {
420 num_bits = 0;
421 index = 0;
422 for (j = p->total_subbands; j > 0; j--) {
423 exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
424 index++;
425 num_bits += expbits_tab[exp_idx];
426 }
427 if (num_bits >= bits_left - 32)
428 bias += i;
429 }
430
431 /* Calculate total number of bits. */
432 num_bits = 0;
433 for (i = 0; i < p->total_subbands; i++) {
434 exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
435 num_bits += expbits_tab[exp_idx];
436 exp_index1[i] = exp_idx;
437 exp_index2[i] = exp_idx;
438 }
439 tmpbias1 = tmpbias2 = num_bits;
440
441 for (j = 1; j < p->numvector_size; j++) {
442 if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
443 int max = -999999;
444 index = -1;
445 for (i = 0; i < p->total_subbands; i++) {
446 if (exp_index1[i] < 7) {
447 v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
448 if (v >= max) {
449 max = v;
450 index = i;
451 }
452 }
453 }
454 if (index == -1)
455 break;
456 tmp_categorize_array[tmp_categorize_array1_idx++] = index;
457 tmpbias1 -= expbits_tab[exp_index1[index]] -
458 expbits_tab[exp_index1[index] + 1];
459 ++exp_index1[index];
460 } else { /* <--- */
461 int min = 999999;
462 index = -1;
463 for (i = 0; i < p->total_subbands; i++) {
464 if (exp_index2[i] > 0) {
465 v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
466 if (v < min) {
467 min = v;
468 index = i;
469 }
470 }
471 }
472 if (index == -1)
473 break;
474 tmp_categorize_array[--tmp_categorize_array2_idx] = index;
475 tmpbias2 -= expbits_tab[exp_index2[index]] -
476 expbits_tab[exp_index2[index] - 1];
477 --exp_index2[index];
478 }
479 }
480
481 for (i = 0; i < p->total_subbands; i++)
482 category[i] = exp_index2[i];
483
484 for (i = 0; i < p->numvector_size - 1; i++)
485 category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
486 }
487
488
489 /**
490 * Expand the category vector.
491 *
492 * @param q pointer to the COOKContext
493 * @param category pointer to the category array
494 * @param category_index pointer to the category_index array
495 */
496 static inline void expand_category(COOKContext *q, int *category,
497 int *category_index)
498 {
499 int i;
500 for (i = 0; i < q->num_vectors; i++)
501 {
502 int idx = category_index[i];
503 if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
504 --category[idx];
505 }
506 }
507
508 /**
509 * The real requantization of the mltcoefs
510 *
511 * @param q pointer to the COOKContext
512 * @param index index
513 * @param quant_index quantisation index
514 * @param subband_coef_index array of indexes to quant_centroid_tab
515 * @param subband_coef_sign signs of coefficients
516 * @param mlt_p pointer into the mlt buffer
517 */
518 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
519 int *subband_coef_index, int *subband_coef_sign,
520 float *mlt_p)
521 {
522 int i;
523 float f1;
524
525 for (i = 0; i < SUBBAND_SIZE; i++) {
526 if (subband_coef_index[i]) {
527 f1 = quant_centroid_tab[index][subband_coef_index[i]];
528 if (subband_coef_sign[i])
529 f1 = -f1;
530 } else {
531 /* noise coding if subband_coef_index[i] == 0 */
532 f1 = dither_tab[index];
533 if (av_lfg_get(&q->random_state) < 0x80000000)
534 f1 = -f1;
535 }
536 mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
537 }
538 }
539 /**
540 * Unpack the subband_coef_index and subband_coef_sign vectors.
541 *
542 * @param q pointer to the COOKContext
543 * @param category pointer to the category array
544 * @param subband_coef_index array of indexes to quant_centroid_tab
545 * @param subband_coef_sign signs of coefficients
546 */
547 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
548 int *subband_coef_index, int *subband_coef_sign)
549 {
550 int i, j;
551 int vlc, vd, tmp, result;
552
553 vd = vd_tab[category];
554 result = 0;
555 for (i = 0; i < vpr_tab[category]; i++) {
556 vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
557 if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
558 vlc = 0;
559 result = 1;
560 }
561 for (j = vd - 1; j >= 0; j--) {
562 tmp = (vlc * invradix_tab[category]) / 0x100000;
563 subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
564 vlc = tmp;
565 }
566 for (j = 0; j < vd; j++) {
567 if (subband_coef_index[i * vd + j]) {
568 if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
569 subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
570 } else {
571 result = 1;
572 subband_coef_sign[i * vd + j] = 0;
573 }
574 } else {
575 subband_coef_sign[i * vd + j] = 0;
576 }
577 }
578 }
579 return result;
580 }
581
582
583 /**
584 * Fill the mlt_buffer with mlt coefficients.
585 *
586 * @param q pointer to the COOKContext
587 * @param category pointer to the category array
588 * @param quant_index_table pointer to the array
589 * @param mlt_buffer pointer to mlt coefficients
590 */
591 static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
592 int *quant_index_table, float *mlt_buffer)
593 {
594 /* A zero in this table means that the subband coefficient is
595 random noise coded. */
596 int subband_coef_index[SUBBAND_SIZE];
597 /* A zero in this table means that the subband coefficient is a
598 positive multiplicator. */
599 int subband_coef_sign[SUBBAND_SIZE];
600 int band, j;
601 int index = 0;
602
603 for (band = 0; band < p->total_subbands; band++) {
604 index = category[band];
605 if (category[band] < 7) {
606 if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
607 index = 7;
608 for (j = 0; j < p->total_subbands; j++)
609 category[band + j] = 7;
610 }
611 }
612 if (index >= 7) {
613 memset(subband_coef_index, 0, sizeof(subband_coef_index));
614 memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
615 }
616 q->scalar_dequant(q, index, quant_index_table[band],
617 subband_coef_index, subband_coef_sign,
618 &mlt_buffer[band * SUBBAND_SIZE]);
619 }
620
621 /* FIXME: should this be removed, or moved into loop above? */
622 if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
623 return;
624 }
625
626
627 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
628 {
629 int category_index[128] = { 0 };
630 int category[128] = { 0 };
631 int quant_index_table[102];
632 int res;
633
634 if ((res = decode_envelope(q, p, quant_index_table)) < 0)
635 return res;
636 q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
637 categorize(q, p, quant_index_table, category, category_index);
638 expand_category(q, category, category_index);
639 decode_vectors(q, p, category, quant_index_table, mlt_buffer);
640
641 return 0;
642 }
643
644
645 /**
646 * the actual requantization of the timedomain samples
647 *
648 * @param q pointer to the COOKContext
649 * @param buffer pointer to the timedomain buffer
650 * @param gain_index index for the block multiplier
651 * @param gain_index_next index for the next block multiplier
652 */
653 static void interpolate_float(COOKContext *q, float *buffer,
654 int gain_index, int gain_index_next)
655 {
656 int i;
657 float fc1, fc2;
658 fc1 = pow2tab[gain_index + 63];
659
660 if (gain_index == gain_index_next) { // static gain
661 for (i = 0; i < q->gain_size_factor; i++)
662 buffer[i] *= fc1;
663 } else { // smooth gain
664 fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
665 for (i = 0; i < q->gain_size_factor; i++) {
666 buffer[i] *= fc1;
667 fc1 *= fc2;
668 }
669 }
670 }
671
672 /**
673 * Apply transform window, overlap buffers.
674 *
675 * @param q pointer to the COOKContext
676 * @param inbuffer pointer to the mltcoefficients
677 * @param gains_ptr current and previous gains
678 * @param previous_buffer pointer to the previous buffer to be used for overlapping
679 */
680 static void imlt_window_float(COOKContext *q, float *inbuffer,
681 cook_gains *gains_ptr, float *previous_buffer)
682 {
683 const float fc = pow2tab[gains_ptr->previous[0] + 63];
684 int i;
685 /* The weird thing here, is that the two halves of the time domain
686 * buffer are swapped. Also, the newest data, that we save away for
687 * next frame, has the wrong sign. Hence the subtraction below.
688 * Almost sounds like a complex conjugate/reverse data/FFT effect.
689 */
690
691 /* Apply window and overlap */
692 for (i = 0; i < q->samples_per_channel; i++)
693 inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
694 previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
695 }
696
697 /**
698 * The modulated lapped transform, this takes transform coefficients
699 * and transforms them into timedomain samples.
700 * Apply transform window, overlap buffers, apply gain profile
701 * and buffer management.
702 *
703 * @param q pointer to the COOKContext
704 * @param inbuffer pointer to the mltcoefficients
705 * @param gains_ptr current and previous gains
706 * @param previous_buffer pointer to the previous buffer to be used for overlapping
707 */
708 static void imlt_gain(COOKContext *q, float *inbuffer,
709 cook_gains *gains_ptr, float *previous_buffer)
710 {
711 float *buffer0 = q->mono_mdct_output;
712 float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
713 int i;
714
715 /* Inverse modified discrete cosine transform */
716 q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
717
718 q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
719
720 /* Apply gain profile */
721 for (i = 0; i < 8; i++)
722 if (gains_ptr->now[i] || gains_ptr->now[i + 1])
723 q->interpolate(q, &buffer1[q->gain_size_factor * i],
724 gains_ptr->now[i], gains_ptr->now[i + 1]);
725
726 /* Save away the current to be previous block. */
727 memcpy(previous_buffer, buffer0,
728 q->samples_per_channel * sizeof(*previous_buffer));
729 }
730
731
732 /**
733 * function for getting the jointstereo coupling information
734 *
735 * @param q pointer to the COOKContext
736 * @param decouple_tab decoupling array
737 */
738 static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
739 {
740 int i;
741 int vlc = get_bits1(&q->gb);
742 int start = cplband[p->js_subband_start];
743 int end = cplband[p->subbands - 1];
744 int length = end - start + 1;
745
746 if (start > end)
747 return;
748
749 if (vlc)
750 for (i = 0; i < length; i++)
751 decouple_tab[start + i] = get_vlc2(&q->gb,
752 p->channel_coupling.table,
753 p->channel_coupling.bits, 2);
754 else
755 for (i = 0; i < length; i++)
756 decouple_tab[start + i] = get_bits(&q->gb, p->js_vlc_bits);
757 }
758
759 /*
760 * function decouples a pair of signals from a single signal via multiplication.
761 *
762 * @param q pointer to the COOKContext
763 * @param subband index of the current subband
764 * @param f1 multiplier for channel 1 extraction
765 * @param f2 multiplier for channel 2 extraction
766 * @param decode_buffer input buffer
767 * @param mlt_buffer1 pointer to left channel mlt coefficients
768 * @param mlt_buffer2 pointer to right channel mlt coefficients
769 */
770 static void decouple_float(COOKContext *q,
771 COOKSubpacket *p,
772 int subband,
773 float f1, float f2,
774 float *decode_buffer,
775 float *mlt_buffer1, float *mlt_buffer2)
776 {
777 int j, tmp_idx;
778 for (j = 0; j < SUBBAND_SIZE; j++) {
779 tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
780 mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
781 mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
782 }
783 }
784
785 /**
786 * function for decoding joint stereo data
787 *
788 * @param q pointer to the COOKContext
789 * @param mlt_buffer1 pointer to left channel mlt coefficients
790 * @param mlt_buffer2 pointer to right channel mlt coefficients
791 */
792 static int joint_decode(COOKContext *q, COOKSubpacket *p,
793 float *mlt_buffer_left, float *mlt_buffer_right)
794 {
795 int i, j, res;
796 int decouple_tab[SUBBAND_SIZE] = { 0 };
797 float *decode_buffer = q->decode_buffer_0;
798 int idx, cpl_tmp;
799 float f1, f2;
800 const float *cplscale;
801
802 memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
803
804 /* Make sure the buffers are zeroed out. */
805 memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
806 memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
807 decouple_info(q, p, decouple_tab);
808 if ((res = mono_decode(q, p, decode_buffer)) < 0)
809 return res;
810
811 /* The two channels are stored interleaved in decode_buffer. */
812 for (i = 0; i < p->js_subband_start; i++) {
813 for (j = 0; j < SUBBAND_SIZE; j++) {
814 mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
815 mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
816 }
817 }
818
819 /* When we reach js_subband_start (the higher frequencies)
820 the coefficients are stored in a coupling scheme. */
821 idx = (1 << p->js_vlc_bits) - 1;
822 for (i = p->js_subband_start; i < p->subbands; i++) {
823 cpl_tmp = cplband[i];
824 idx -= decouple_tab[cpl_tmp];
825 cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
826 f1 = cplscale[decouple_tab[cpl_tmp] + 1];
827 f2 = cplscale[idx];
828 q->decouple(q, p, i, f1, f2, decode_buffer,
829 mlt_buffer_left, mlt_buffer_right);
830 idx = (1 << p->js_vlc_bits) - 1;
831 }
832
833 return 0;
834 }
835
836 /**
837 * First part of subpacket decoding:
838 * decode raw stream bytes and read gain info.
839 *
840 * @param q pointer to the COOKContext
841 * @param inbuffer pointer to raw stream data
842 * @param gains_ptr array of current/prev gain pointers
843 */
844 static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
845 const uint8_t *inbuffer,
846 cook_gains *gains_ptr)
847 {
848 int offset;
849
850 offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
851 p->bits_per_subpacket / 8);
852 init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
853 p->bits_per_subpacket);
854 decode_gain_info(&q->gb, gains_ptr->now);
855
856 /* Swap current and previous gains */
857 FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
858 }
859
860 /**
861 * Saturate the output signal and interleave.
862 *
863 * @param q pointer to the COOKContext
864 * @param out pointer to the output vector
865 */
866 static void saturate_output_float(COOKContext *q, float *out)
867 {
868 q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
869 -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
870 }
871
872
873 /**
874 * Final part of subpacket decoding:
875 * Apply modulated lapped transform, gain compensation,
876 * clip and convert to integer.
877 *
878 * @param q pointer to the COOKContext
879 * @param decode_buffer pointer to the mlt coefficients
880 * @param gains_ptr array of current/prev gain pointers
881 * @param previous_buffer pointer to the previous buffer to be used for overlapping
882 * @param out pointer to the output buffer
883 */
884 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
885 cook_gains *gains_ptr, float *previous_buffer,
886 float *out)
887 {
888 imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
889 if (out)
890 q->saturate_output(q, out);
891 }
892
893
894 /**
895 * Cook subpacket decoding. This function returns one decoded subpacket,
896 * usually 1024 samples per channel.
897 *
898 * @param q pointer to the COOKContext
899 * @param inbuffer pointer to the inbuffer
900 * @param outbuffer pointer to the outbuffer
901 */
902 static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
903 const uint8_t *inbuffer, float **outbuffer)
904 {
905 int sub_packet_size = p->size;
906 int res;
907
908 memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
909 decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
910
911 if (p->joint_stereo) {
912 if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
913 return res;
914 } else {
915 if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
916 return res;
917
918 if (p->num_channels == 2) {
919 decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
920 if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
921 return res;
922 }
923 }
924
925 mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
926 p->mono_previous_buffer1,
927 outbuffer ? outbuffer[p->ch_idx] : NULL);
928
929 if (p->num_channels == 2)
930 if (p->joint_stereo)
931 mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
932 p->mono_previous_buffer2,
933 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
934 else
935 mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
936 p->mono_previous_buffer2,
937 outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
938
939 return 0;
940 }
941
942
943 static int cook_decode_frame(AVCodecContext *avctx, void *data,
944 int *got_frame_ptr, AVPacket *avpkt)
945 {
946 const uint8_t *buf = avpkt->data;
947 int buf_size = avpkt->size;
948 COOKContext *q = avctx->priv_data;
949 float **samples = NULL;
950 int i, ret;
951 int offset = 0;
952 int chidx = 0;
953
954 if (buf_size < avctx->block_align)
955 return buf_size;
956
957 /* get output buffer */
958 if (q->discarded_packets >= 2) {
959 q->frame.nb_samples = q->samples_per_channel;
960 if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
961 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
962 return ret;
963 }
964 samples = (float **)q->frame.extended_data;
965 }
966
967 /* estimate subpacket sizes */
968 q->subpacket[0].size = avctx->block_align;
969
970 for (i = 1; i < q->num_subpackets; i++) {
971 q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
972 q->subpacket[0].size -= q->subpacket[i].size + 1;
973 if (q->subpacket[0].size < 0) {
974 av_log(avctx, AV_LOG_DEBUG,
975 "frame subpacket size total > avctx->block_align!\n");
976 return AVERROR_INVALIDDATA;
977 }
978 }
979
980 /* decode supbackets */
981 for (i = 0; i < q->num_subpackets; i++) {
982 q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
983 q->subpacket[i].bits_per_subpdiv;
984 q->subpacket[i].ch_idx = chidx;
985 av_log(avctx, AV_LOG_DEBUG,
986 "subpacket[%i] size %i js %i %i block_align %i\n",
987 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
988 avctx->block_align);
989
990 if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
991 return ret;
992 offset += q->subpacket[i].size;
993 chidx += q->subpacket[i].num_channels;
994 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
995 i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
996 }
997
998 /* Discard the first two frames: no valid audio. */
999 if (q->discarded_packets < 2) {
1000 q->discarded_packets++;
1001 *got_frame_ptr = 0;
1002 return avctx->block_align;
1003 }
1004
1005 *got_frame_ptr = 1;
1006 *(AVFrame *) data = q->frame;
1007
1008 return avctx->block_align;
1009 }
1010
1011 #ifdef DEBUG
1012 static void dump_cook_context(COOKContext *q)
1013 {
1014 //int i=0;
1015 #define PRINT(a, b) av_log(q->avctx, AV_LOG_ERROR, " %s = %d\n", a, b);
1016 av_log(q->avctx, AV_LOG_ERROR, "COOKextradata\n");
1017 av_log(q->avctx, AV_LOG_ERROR, "cookversion=%x\n", q->subpacket[0].cookversion);
1018 if (q->subpacket[0].cookversion > STEREO) {
1019 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1020 PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1021 }
1022 av_log(q->avctx, AV_LOG_ERROR, "COOKContext\n");
1023 PRINT("nb_channels", q->avctx->channels);
1024 PRINT("bit_rate", q->avctx->bit_rate);
1025 PRINT("sample_rate", q->avctx->sample_rate);
1026 PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1027 PRINT("subbands", q->subpacket[0].subbands);
1028 PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1029 PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1030 PRINT("numvector_size", q->subpacket[0].numvector_size);
1031 PRINT("total_subbands", q->subpacket[0].total_subbands);
1032 }
1033 #endif
1034
1035 /**
1036 * Cook initialization
1037 *
1038 * @param avctx pointer to the AVCodecContext
1039 */
1040 static av_cold int cook_decode_init(AVCodecContext *avctx)
1041 {
1042 COOKContext *q = avctx->priv_data;
1043 const uint8_t *edata_ptr = avctx->extradata;
1044 const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
1045 int extradata_size = avctx->extradata_size;
1046 int s = 0;
1047 unsigned int channel_mask = 0;
1048 int samples_per_frame;
1049 int ret;
1050 q->avctx = avctx;
1051
1052 /* Take care of the codec specific extradata. */
1053 if (extradata_size <= 0) {
1054 av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1055 return AVERROR_INVALIDDATA;
1056 }
1057 av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1058
1059 /* Take data from the AVCodecContext (RM container). */
1060 if (!avctx->channels) {
1061 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1062 return AVERROR_INVALIDDATA;
1063 }
1064
1065 /* Initialize RNG. */
1066 av_lfg_init(&q->random_state, 0);
1067
1068 ff_dsputil_init(&q->dsp, avctx);
1069
1070 while (edata_ptr < edata_ptr_end) {
1071 /* 8 for mono, 16 for stereo, ? for multichannel
1072 Swap to right endianness so we don't need to care later on. */
1073 if (extradata_size >= 8) {
1074 q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
1075 samples_per_frame = bytestream_get_be16(&edata_ptr);
1076 q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
1077 extradata_size -= 8;
1078 }
1079 if (extradata_size >= 8) {
1080 bytestream_get_be32(&edata_ptr); // Unknown unused
1081 q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
1082 q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
1083 extradata_size -= 8;
1084 }
1085
1086 /* Initialize extradata related variables. */
1087 q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1088 q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1089
1090 /* Initialize default data states. */
1091 q->subpacket[s].log2_numvector_size = 5;
1092 q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1093 q->subpacket[s].num_channels = 1;
1094
1095 /* Initialize version-dependent variables */
1096
1097 av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1098 q->subpacket[s].cookversion);
1099 q->subpacket[s].joint_stereo = 0;
1100 switch (q->subpacket[s].cookversion) {
1101 case MONO:
1102 if (avctx->channels != 1) {
1103 av_log_ask_for_sample(avctx, "Container channels != 1.\n");
1104 return AVERROR_PATCHWELCOME;
1105 }
1106 av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1107 break;
1108 case STEREO:
1109 if (avctx->channels != 1) {
1110 q->subpacket[s].bits_per_subpdiv = 1;
1111 q->subpacket[s].num_channels = 2;
1112 }
1113 av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1114 break;
1115 case JOINT_STEREO:
1116 if (avctx->channels != 2) {
1117 av_log_ask_for_sample(avctx, "Container channels != 2.\n");
1118 return AVERROR_PATCHWELCOME;
1119 }
1120 av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1121 if (avctx->extradata_size >= 16) {
1122 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1123 q->subpacket[s].js_subband_start;
1124 q->subpacket[s].joint_stereo = 1;
1125 q->subpacket[s].num_channels = 2;
1126 }
1127 if (q->subpacket[s].samples_per_channel > 256) {
1128 q->subpacket[s].log2_numvector_size = 6;
1129 }
1130 if (q->subpacket[s].samples_per_channel > 512) {
1131 q->subpacket[s].log2_numvector_size = 7;
1132 }
1133 break;
1134 case MC_COOK:
1135 av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1136 if (extradata_size >= 4)
1137 channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
1138
1139 if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
1140 q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1141 q->subpacket[s].js_subband_start;
1142 q->subpacket[s].joint_stereo = 1;
1143 q->subpacket[s].num_channels = 2;
1144 q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1145
1146 if (q->subpacket[s].samples_per_channel > 256) {
1147 q->subpacket[s].log2_numvector_size = 6;
1148 }
1149 if (q->subpacket[s].samples_per_channel > 512) {
1150 q->subpacket[s].log2_numvector_size = 7;
1151 }
1152 } else
1153 q->subpacket[s].samples_per_channel = samples_per_frame;
1154
1155 break;
1156 default:
1157 av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
1158 return AVERROR_PATCHWELCOME;
1159 }
1160
1161 if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1162 av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1163 return AVERROR_INVALIDDATA;
1164 } else
1165 q->samples_per_channel = q->subpacket[0].samples_per_channel;
1166
1167
1168 /* Initialize variable relations */
1169 q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1170
1171 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1172 if (q->subpacket[s].total_subbands > 53) {
1173 av_log_ask_for_sample(avctx, "total_subbands > 53\n");
1174 return AVERROR_PATCHWELCOME;
1175 }
1176
1177 if ((q->subpacket[s].js_vlc_bits > 6) ||
1178 (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1179 av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1180 q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1181 return AVERROR_INVALIDDATA;
1182 }
1183
1184 if (q->subpacket[s].subbands > 50) {
1185 av_log_ask_for_sample(avctx, "subbands > 50\n");
1186 return AVERROR_PATCHWELCOME;
1187 }
1188 q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1189 q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1190 q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1191 q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1192
1193 q->num_subpackets++;
1194 s++;
1195 if (s > MAX_SUBPACKETS) {
1196 av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
1197 return AVERROR_PATCHWELCOME;
1198 }
1199 }
1200 /* Generate tables */
1201 init_pow2table();
1202 init_gain_table(q);
1203 init_cplscales_table(q);
1204
1205 if ((ret = init_cook_vlc_tables(q)))
1206 return ret;
1207
1208
1209 if (avctx->block_align >= UINT_MAX / 2)
1210 return AVERROR(EINVAL);
1211
1212 /* Pad the databuffer with:
1213 DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1214 FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1215 q->decoded_bytes_buffer =
1216 av_mallocz(avctx->block_align
1217 + DECODE_BYTES_PAD1(avctx->block_align)
1218 + FF_INPUT_BUFFER_PADDING_SIZE);
1219 if (q->decoded_bytes_buffer == NULL)
1220 return AVERROR(ENOMEM);
1221
1222 /* Initialize transform. */
1223 if ((ret = init_cook_mlt(q)))
1224 return ret;
1225
1226 /* Initialize COOK signal arithmetic handling */
1227 if (1) {
1228 q->scalar_dequant = scalar_dequant_float;
1229 q->decouple = decouple_float;
1230 q->imlt_window = imlt_window_float;
1231 q->interpolate = interpolate_float;
1232 q->saturate_output = saturate_output_float;
1233 }
1234
1235 /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1236 if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1237 q->samples_per_channel != 1024) {
1238 av_log_ask_for_sample(avctx,
1239 "unknown amount of samples_per_channel = %d\n",
1240 q->samples_per_channel);
1241 return AVERROR_PATCHWELCOME;
1242 }
1243
1244 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1245 if (channel_mask)
1246 avctx->channel_layout = channel_mask;
1247 else
1248 avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1249
1250 avcodec_get_frame_defaults(&q->frame);
1251 avctx->coded_frame = &q->frame;
1252
1253 #ifdef DEBUG
1254 dump_cook_context(q);
1255 #endif
1256 return 0;
1257 }
1258
1259 AVCodec ff_cook_decoder = {
1260 .name = "cook",
1261 .type = AVMEDIA_TYPE_AUDIO,
1262 .id = AV_CODEC_ID_COOK,
1263 .priv_data_size = sizeof(COOKContext),
1264 .init = cook_decode_init,
1265 .close = cook_decode_close,
1266 .decode = cook_decode_frame,
1267 .capabilities = CODEC_CAP_DR1,
1268 .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1269 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1270 AV_SAMPLE_FMT_NONE },
1271 };