FFmpeg  4.4
cook.c
Go to the documentation of this file.
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 
46 #include "libavutil/lfg.h"
47 #include "libavutil/mem_internal.h"
48 
49 #include "audiodsp.h"
50 #include "avcodec.h"
51 #include "get_bits.h"
52 #include "bytestream.h"
53 #include "fft.h"
54 #include "internal.h"
55 #include "sinewin.h"
56 #include "unary.h"
57 
58 #include "cookdata.h"
59 
60 /* the different Cook versions */
61 #define MONO 0x1000001
62 #define STEREO 0x1000002
63 #define JOINT_STEREO 0x1000003
64 #define MC_COOK 0x2000000
65 
66 #define SUBBAND_SIZE 20
67 #define MAX_SUBPACKETS 5
68 
69 #define QUANT_VLC_BITS 9
70 #define COUPLING_VLC_BITS 6
71 
72 typedef struct cook_gains {
73  int *now;
74  int *previous;
75 } cook_gains;
76 
77 typedef struct COOKSubpacket {
78  int ch_idx;
79  int size;
82  int subbands;
87  unsigned int channel_mask;
93  int numvector_size; // 1 << log2_numvector_size;
94 
95  float mono_previous_buffer1[1024];
96  float mono_previous_buffer2[1024];
97 
100  int gain_1[9];
101  int gain_2[9];
102  int gain_3[9];
103  int gain_4[9];
104 } COOKSubpacket;
105 
106 typedef struct cook {
107  /*
108  * The following 5 functions provide the lowlevel arithmetic on
109  * the internal audio buffers.
110  */
111  void (*scalar_dequant)(struct cook *q, int index, int quant_index,
112  int *subband_coef_index, int *subband_coef_sign,
113  float *mlt_p);
114 
115  void (*decouple)(struct cook *q,
116  COOKSubpacket *p,
117  int subband,
118  float f1, float f2,
119  float *decode_buffer,
120  float *mlt_buffer1, float *mlt_buffer2);
121 
122  void (*imlt_window)(struct cook *q, float *buffer1,
123  cook_gains *gains_ptr, float *previous_buffer);
124 
125  void (*interpolate)(struct cook *q, float *buffer,
126  int gain_index, int gain_index_next);
127 
128  void (*saturate_output)(struct cook *q, float *out);
129 
133  /* stream data */
136  /* states */
139 
140  /* transform data */
142  float* mlt_window;
143 
144  /* VLC data */
145  VLC envelope_quant_index[13];
146  VLC sqvh[7]; // scalar quantization
147 
148  /* generate tables and related variables */
150  float gain_table[31];
151 
152  /* data buffers */
153 
155  DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
156  float decode_buffer_1[1024];
157  float decode_buffer_2[1024];
158  float decode_buffer_0[1060]; /* static allocation for joint decode */
159 
160  const float *cplscales[5];
163 } COOKContext;
164 
165 static float pow2tab[127];
166 static float rootpow2tab[127];
167 
168 /*************** init functions ***************/
169 
170 /* table generator */
171 static av_cold void init_pow2table(void)
172 {
173  /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
174  int i;
175  static const float exp2_tab[2] = {1, M_SQRT2};
176  float exp2_val = powf(2, -63);
177  float root_val = powf(2, -32);
178  for (i = -63; i < 64; i++) {
179  if (!(i & 1))
180  root_val *= 2;
181  pow2tab[63 + i] = exp2_val;
182  rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
183  exp2_val *= 2;
184  }
185 }
186 
187 /* table generator */
189 {
190  int i;
192  for (i = 0; i < 31; i++)
193  q->gain_table[i] = pow(pow2tab[i + 48],
194  (1.0 / (double) q->gain_size_factor));
195 }
196 
197 static av_cold int build_vlc(VLC *vlc, int nb_bits, const uint8_t counts[16],
198  const void *syms, int symbol_size, int offset,
199  void *logctx)
200 {
202  unsigned num = 0;
203 
204  for (int i = 0; i < 16; i++)
205  for (unsigned count = num + counts[i]; num < count; num++)
206  lens[num] = i + 1;
207 
208  return ff_init_vlc_from_lengths(vlc, nb_bits, num, lens, 1,
209  syms, symbol_size, symbol_size,
210  offset, 0, logctx);
211 }
212 
214 {
215  int i, result;
216 
217  result = 0;
218  for (i = 0; i < 13; i++) {
221  envelope_quant_index_huffsyms[i], 1, -12, q->avctx);
222  }
223  av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
224  for (i = 0; i < 7; i++) {
225  int sym_size = 1 + (i == 3);
226  result |= build_vlc(&q->sqvh[i], vhvlcsize_tab[i],
227  cvh_huffcounts[i],
228  cvh_huffsyms[i], sym_size, 0, q->avctx);
229  }
230 
231  for (i = 0; i < q->num_subpackets; i++) {
232  if (q->subpacket[i].joint_stereo == 1) {
235  ccpl_huffsyms[q->subpacket[i].js_vlc_bits - 2], 1,
236  0, q->avctx);
237  av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
238  }
239  }
240 
241  av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
242  return result;
243 }
244 
246 {
247  int j, ret;
248  int mlt_size = q->samples_per_channel;
249 
250  if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
251  return AVERROR(ENOMEM);
252 
253  /* Initialize the MLT window: simple sine window. */
254  ff_sine_window_init(q->mlt_window, mlt_size);
255  for (j = 0; j < mlt_size; j++)
256  q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
257 
258  /* Initialize the MDCT. */
259  if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
260  av_freep(&q->mlt_window);
261  return ret;
262  }
263  av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
264  av_log2(mlt_size) + 1);
265 
266  return 0;
267 }
268 
270 {
271  int i;
272  for (i = 0; i < 5; i++)
273  q->cplscales[i] = cplscales[i];
274 }
275 
276 /*************** init functions end ***********/
277 
278 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
279 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
280 
281 /**
282  * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
283  * Why? No idea, some checksum/error detection method maybe.
284  *
285  * Out buffer size: extra bytes are needed to cope with
286  * padding/misalignment.
287  * Subpackets passed to the decoder can contain two, consecutive
288  * half-subpackets, of identical but arbitrary size.
289  * 1234 1234 1234 1234 extraA extraB
290  * Case 1: AAAA BBBB 0 0
291  * Case 2: AAAA ABBB BB-- 3 3
292  * Case 3: AAAA AABB BBBB 2 2
293  * Case 4: AAAA AAAB BBBB BB-- 1 5
294  *
295  * Nice way to waste CPU cycles.
296  *
297  * @param inbuffer pointer to byte array of indata
298  * @param out pointer to byte array of outdata
299  * @param bytes number of bytes
300  */
301 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
302 {
303  static const uint32_t tab[4] = {
304  AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
305  AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
306  };
307  int i, off;
308  uint32_t c;
309  const uint32_t *buf;
310  uint32_t *obuf = (uint32_t *) out;
311  /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
312  * I'm too lazy though, should be something like
313  * for (i = 0; i < bitamount / 64; i++)
314  * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
315  * Buffer alignment needs to be checked. */
316 
317  off = (intptr_t) inbuffer & 3;
318  buf = (const uint32_t *) (inbuffer - off);
319  c = tab[off];
320  bytes += 3 + off;
321  for (i = 0; i < bytes / 4; i++)
322  obuf[i] = c ^ buf[i];
323 
324  return off;
325 }
326 
328 {
329  int i;
330  COOKContext *q = avctx->priv_data;
331  av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
332 
333  /* Free allocated memory buffers. */
334  av_freep(&q->mlt_window);
336 
337  /* Free the transform. */
338  ff_mdct_end(&q->mdct_ctx);
339 
340  /* Free the VLC tables. */
341  for (i = 0; i < 13; i++)
343  for (i = 0; i < 7; i++)
344  ff_free_vlc(&q->sqvh[i]);
345  for (i = 0; i < q->num_subpackets; i++)
347 
348  av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
349 
350  return 0;
351 }
352 
353 /**
354  * Fill the gain array for the timedomain quantization.
355  *
356  * @param gb pointer to the GetBitContext
357  * @param gaininfo array[9] of gain indexes
358  */
359 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
360 {
361  int i, n;
362 
363  n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update
364 
365  i = 0;
366  while (n--) {
367  int index = get_bits(gb, 3);
368  int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
369 
370  while (i <= index)
371  gaininfo[i++] = gain;
372  }
373  while (i <= 8)
374  gaininfo[i++] = 0;
375 }
376 
377 /**
378  * Create the quant index table needed for the envelope.
379  *
380  * @param q pointer to the COOKContext
381  * @param quant_index_table pointer to the array
382  */
384  int *quant_index_table)
385 {
386  int i, j, vlc_index;
387 
388  quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
389 
390  for (i = 1; i < p->total_subbands; i++) {
391  vlc_index = i;
392  if (i >= p->js_subband_start * 2) {
393  vlc_index -= p->js_subband_start;
394  } else {
395  vlc_index /= 2;
396  if (vlc_index < 1)
397  vlc_index = 1;
398  }
399  if (vlc_index > 13)
400  vlc_index = 13; // the VLC tables >13 are identical to No. 13
401 
402  j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
403  QUANT_VLC_BITS, 2);
404  quant_index_table[i] = quant_index_table[i - 1] + j; // differential encoding
405  if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
407  "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
408  quant_index_table[i], i);
409  return AVERROR_INVALIDDATA;
410  }
411  }
412 
413  return 0;
414 }
415 
416 /**
417  * Calculate the category and category_index vector.
418  *
419  * @param q pointer to the COOKContext
420  * @param quant_index_table pointer to the array
421  * @param category pointer to the category array
422  * @param category_index pointer to the category_index array
423  */
424 static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
425  int *category, int *category_index)
426 {
427  int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
428  int exp_index2[102] = { 0 };
429  int exp_index1[102] = { 0 };
430 
431  int tmp_categorize_array[128 * 2] = { 0 };
432  int tmp_categorize_array1_idx = p->numvector_size;
433  int tmp_categorize_array2_idx = p->numvector_size;
434 
435  bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
436 
437  if (bits_left > q->samples_per_channel)
438  bits_left = q->samples_per_channel +
439  ((bits_left - q->samples_per_channel) * 5) / 8;
440 
441  bias = -32;
442 
443  /* Estimate bias. */
444  for (i = 32; i > 0; i = i / 2) {
445  num_bits = 0;
446  index = 0;
447  for (j = p->total_subbands; j > 0; j--) {
448  exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
449  index++;
450  num_bits += expbits_tab[exp_idx];
451  }
452  if (num_bits >= bits_left - 32)
453  bias += i;
454  }
455 
456  /* Calculate total number of bits. */
457  num_bits = 0;
458  for (i = 0; i < p->total_subbands; i++) {
459  exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
460  num_bits += expbits_tab[exp_idx];
461  exp_index1[i] = exp_idx;
462  exp_index2[i] = exp_idx;
463  }
464  tmpbias1 = tmpbias2 = num_bits;
465 
466  for (j = 1; j < p->numvector_size; j++) {
467  if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
468  int max = -999999;
469  index = -1;
470  for (i = 0; i < p->total_subbands; i++) {
471  if (exp_index1[i] < 7) {
472  v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
473  if (v >= max) {
474  max = v;
475  index = i;
476  }
477  }
478  }
479  if (index == -1)
480  break;
481  tmp_categorize_array[tmp_categorize_array1_idx++] = index;
482  tmpbias1 -= expbits_tab[exp_index1[index]] -
483  expbits_tab[exp_index1[index] + 1];
484  ++exp_index1[index];
485  } else { /* <--- */
486  int min = 999999;
487  index = -1;
488  for (i = 0; i < p->total_subbands; i++) {
489  if (exp_index2[i] > 0) {
490  v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
491  if (v < min) {
492  min = v;
493  index = i;
494  }
495  }
496  }
497  if (index == -1)
498  break;
499  tmp_categorize_array[--tmp_categorize_array2_idx] = index;
500  tmpbias2 -= expbits_tab[exp_index2[index]] -
501  expbits_tab[exp_index2[index] - 1];
502  --exp_index2[index];
503  }
504  }
505 
506  for (i = 0; i < p->total_subbands; i++)
507  category[i] = exp_index2[i];
508 
509  for (i = 0; i < p->numvector_size - 1; i++)
510  category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
511 }
512 
513 
514 /**
515  * Expand the category vector.
516  *
517  * @param q pointer to the COOKContext
518  * @param category pointer to the category array
519  * @param category_index pointer to the category_index array
520  */
521 static inline void expand_category(COOKContext *q, int *category,
522  int *category_index)
523 {
524  int i;
525  for (i = 0; i < q->num_vectors; i++)
526  {
527  int idx = category_index[i];
528  if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
529  --category[idx];
530  }
531 }
532 
533 /**
534  * The real requantization of the mltcoefs
535  *
536  * @param q pointer to the COOKContext
537  * @param index index
538  * @param quant_index quantisation index
539  * @param subband_coef_index array of indexes to quant_centroid_tab
540  * @param subband_coef_sign signs of coefficients
541  * @param mlt_p pointer into the mlt buffer
542  */
543 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
544  int *subband_coef_index, int *subband_coef_sign,
545  float *mlt_p)
546 {
547  int i;
548  float f1;
549 
550  for (i = 0; i < SUBBAND_SIZE; i++) {
551  if (subband_coef_index[i]) {
552  f1 = quant_centroid_tab[index][subband_coef_index[i]];
553  if (subband_coef_sign[i])
554  f1 = -f1;
555  } else {
556  /* noise coding if subband_coef_index[i] == 0 */
557  f1 = dither_tab[index];
558  if (av_lfg_get(&q->random_state) < 0x80000000)
559  f1 = -f1;
560  }
561  mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
562  }
563 }
564 /**
565  * Unpack the subband_coef_index and subband_coef_sign vectors.
566  *
567  * @param q pointer to the COOKContext
568  * @param category pointer to the category array
569  * @param subband_coef_index array of indexes to quant_centroid_tab
570  * @param subband_coef_sign signs of coefficients
571  */
573  int *subband_coef_index, int *subband_coef_sign)
574 {
575  int i, j;
576  int vlc, vd, tmp, result;
577 
578  vd = vd_tab[category];
579  result = 0;
580  for (i = 0; i < vpr_tab[category]; i++) {
581  vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
582  if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
583  vlc = 0;
584  result = 1;
585  }
586  for (j = vd - 1; j >= 0; j--) {
587  tmp = (vlc * invradix_tab[category]) / 0x100000;
588  subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
589  vlc = tmp;
590  }
591  for (j = 0; j < vd; j++) {
592  if (subband_coef_index[i * vd + j]) {
593  if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
594  subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
595  } else {
596  result = 1;
597  subband_coef_sign[i * vd + j] = 0;
598  }
599  } else {
600  subband_coef_sign[i * vd + j] = 0;
601  }
602  }
603  }
604  return result;
605 }
606 
607 
608 /**
609  * Fill the mlt_buffer with mlt coefficients.
610  *
611  * @param q pointer to the COOKContext
612  * @param category pointer to the category array
613  * @param quant_index_table pointer to the array
614  * @param mlt_buffer pointer to mlt coefficients
615  */
617  int *quant_index_table, float *mlt_buffer)
618 {
619  /* A zero in this table means that the subband coefficient is
620  random noise coded. */
621  int subband_coef_index[SUBBAND_SIZE];
622  /* A zero in this table means that the subband coefficient is a
623  positive multiplicator. */
624  int subband_coef_sign[SUBBAND_SIZE];
625  int band, j;
626  int index = 0;
627 
628  for (band = 0; band < p->total_subbands; band++) {
629  index = category[band];
630  if (category[band] < 7) {
631  if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
632  index = 7;
633  for (j = 0; j < p->total_subbands; j++)
634  category[band + j] = 7;
635  }
636  }
637  if (index >= 7) {
638  memset(subband_coef_index, 0, sizeof(subband_coef_index));
639  memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
640  }
641  q->scalar_dequant(q, index, quant_index_table[band],
642  subband_coef_index, subband_coef_sign,
643  &mlt_buffer[band * SUBBAND_SIZE]);
644  }
645 
646  /* FIXME: should this be removed, or moved into loop above? */
648  return;
649 }
650 
651 
652 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
653 {
654  int category_index[128] = { 0 };
655  int category[128] = { 0 };
656  int quant_index_table[102];
657  int res, i;
658 
659  if ((res = decode_envelope(q, p, quant_index_table)) < 0)
660  return res;
662  categorize(q, p, quant_index_table, category, category_index);
663  expand_category(q, category, category_index);
664  for (i=0; i<p->total_subbands; i++) {
665  if (category[i] > 7)
666  return AVERROR_INVALIDDATA;
667  }
668  decode_vectors(q, p, category, quant_index_table, mlt_buffer);
669 
670  return 0;
671 }
672 
673 
674 /**
675  * the actual requantization of the timedomain samples
676  *
677  * @param q pointer to the COOKContext
678  * @param buffer pointer to the timedomain buffer
679  * @param gain_index index for the block multiplier
680  * @param gain_index_next index for the next block multiplier
681  */
682 static void interpolate_float(COOKContext *q, float *buffer,
683  int gain_index, int gain_index_next)
684 {
685  int i;
686  float fc1, fc2;
687  fc1 = pow2tab[gain_index + 63];
688 
689  if (gain_index == gain_index_next) { // static gain
690  for (i = 0; i < q->gain_size_factor; i++)
691  buffer[i] *= fc1;
692  } else { // smooth gain
693  fc2 = q->gain_table[15 + (gain_index_next - gain_index)];
694  for (i = 0; i < q->gain_size_factor; i++) {
695  buffer[i] *= fc1;
696  fc1 *= fc2;
697  }
698  }
699 }
700 
701 /**
702  * Apply transform window, overlap buffers.
703  *
704  * @param q pointer to the COOKContext
705  * @param inbuffer pointer to the mltcoefficients
706  * @param gains_ptr current and previous gains
707  * @param previous_buffer pointer to the previous buffer to be used for overlapping
708  */
709 static void imlt_window_float(COOKContext *q, float *inbuffer,
710  cook_gains *gains_ptr, float *previous_buffer)
711 {
712  const float fc = pow2tab[gains_ptr->previous[0] + 63];
713  int i;
714  /* The weird thing here, is that the two halves of the time domain
715  * buffer are swapped. Also, the newest data, that we save away for
716  * next frame, has the wrong sign. Hence the subtraction below.
717  * Almost sounds like a complex conjugate/reverse data/FFT effect.
718  */
719 
720  /* Apply window and overlap */
721  for (i = 0; i < q->samples_per_channel; i++)
722  inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
723  previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
724 }
725 
726 /**
727  * The modulated lapped transform, this takes transform coefficients
728  * and transforms them into timedomain samples.
729  * Apply transform window, overlap buffers, apply gain profile
730  * and buffer management.
731  *
732  * @param q pointer to the COOKContext
733  * @param inbuffer pointer to the mltcoefficients
734  * @param gains_ptr current and previous gains
735  * @param previous_buffer pointer to the previous buffer to be used for overlapping
736  */
737 static void imlt_gain(COOKContext *q, float *inbuffer,
738  cook_gains *gains_ptr, float *previous_buffer)
739 {
740  float *buffer0 = q->mono_mdct_output;
741  float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
742  int i;
743 
744  /* Inverse modified discrete cosine transform */
745  q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
746 
747  q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
748 
749  /* Apply gain profile */
750  for (i = 0; i < 8; i++)
751  if (gains_ptr->now[i] || gains_ptr->now[i + 1])
752  q->interpolate(q, &buffer1[q->gain_size_factor * i],
753  gains_ptr->now[i], gains_ptr->now[i + 1]);
754 
755  /* Save away the current to be previous block. */
756  memcpy(previous_buffer, buffer0,
757  q->samples_per_channel * sizeof(*previous_buffer));
758 }
759 
760 
761 /**
762  * function for getting the jointstereo coupling information
763  *
764  * @param q pointer to the COOKContext
765  * @param decouple_tab decoupling array
766  */
767 static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
768 {
769  int i;
770  int vlc = get_bits1(&q->gb);
771  int start = cplband[p->js_subband_start];
772  int end = cplband[p->subbands - 1];
773  int length = end - start + 1;
774 
775  if (start > end)
776  return 0;
777 
778  if (vlc)
779  for (i = 0; i < length; i++)
780  decouple_tab[start + i] = get_vlc2(&q->gb,
782  COUPLING_VLC_BITS, 3);
783  else
784  for (i = 0; i < length; i++) {
785  int v = get_bits(&q->gb, p->js_vlc_bits);
786  if (v == (1<<p->js_vlc_bits)-1) {
787  av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
788  return AVERROR_INVALIDDATA;
789  }
790  decouple_tab[start + i] = v;
791  }
792  return 0;
793 }
794 
795 /**
796  * function decouples a pair of signals from a single signal via multiplication.
797  *
798  * @param q pointer to the COOKContext
799  * @param subband index of the current subband
800  * @param f1 multiplier for channel 1 extraction
801  * @param f2 multiplier for channel 2 extraction
802  * @param decode_buffer input buffer
803  * @param mlt_buffer1 pointer to left channel mlt coefficients
804  * @param mlt_buffer2 pointer to right channel mlt coefficients
805  */
807  COOKSubpacket *p,
808  int subband,
809  float f1, float f2,
810  float *decode_buffer,
811  float *mlt_buffer1, float *mlt_buffer2)
812 {
813  int j, tmp_idx;
814  for (j = 0; j < SUBBAND_SIZE; j++) {
815  tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
816  mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
817  mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
818  }
819 }
820 
821 /**
822  * function for decoding joint stereo data
823  *
824  * @param q pointer to the COOKContext
825  * @param mlt_buffer1 pointer to left channel mlt coefficients
826  * @param mlt_buffer2 pointer to right channel mlt coefficients
827  */
829  float *mlt_buffer_left, float *mlt_buffer_right)
830 {
831  int i, j, res;
832  int decouple_tab[SUBBAND_SIZE] = { 0 };
833  float *decode_buffer = q->decode_buffer_0;
834  int idx, cpl_tmp;
835  float f1, f2;
836  const float *cplscale;
837 
838  memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
839 
840  /* Make sure the buffers are zeroed out. */
841  memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
842  memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
843  if ((res = decouple_info(q, p, decouple_tab)) < 0)
844  return res;
845  if ((res = mono_decode(q, p, decode_buffer)) < 0)
846  return res;
847  /* The two channels are stored interleaved in decode_buffer. */
848  for (i = 0; i < p->js_subband_start; i++) {
849  for (j = 0; j < SUBBAND_SIZE; j++) {
850  mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
851  mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
852  }
853  }
854 
855  /* When we reach js_subband_start (the higher frequencies)
856  the coefficients are stored in a coupling scheme. */
857  idx = (1 << p->js_vlc_bits) - 1;
858  for (i = p->js_subband_start; i < p->subbands; i++) {
859  cpl_tmp = cplband[i];
860  idx -= decouple_tab[cpl_tmp];
861  cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
862  f1 = cplscale[decouple_tab[cpl_tmp] + 1];
863  f2 = cplscale[idx];
864  q->decouple(q, p, i, f1, f2, decode_buffer,
865  mlt_buffer_left, mlt_buffer_right);
866  idx = (1 << p->js_vlc_bits) - 1;
867  }
868 
869  return 0;
870 }
871 
872 /**
873  * First part of subpacket decoding:
874  * decode raw stream bytes and read gain info.
875  *
876  * @param q pointer to the COOKContext
877  * @param inbuffer pointer to raw stream data
878  * @param gains_ptr array of current/prev gain pointers
879  */
881  const uint8_t *inbuffer,
882  cook_gains *gains_ptr)
883 {
884  int offset;
885 
887  p->bits_per_subpacket / 8);
889  p->bits_per_subpacket);
890  decode_gain_info(&q->gb, gains_ptr->now);
891 
892  /* Swap current and previous gains */
893  FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
894 }
895 
896 /**
897  * Saturate the output signal and interleave.
898  *
899  * @param q pointer to the COOKContext
900  * @param out pointer to the output vector
901  */
902 static void saturate_output_float(COOKContext *q, float *out)
903 {
905  FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
906 }
907 
908 
909 /**
910  * Final part of subpacket decoding:
911  * Apply modulated lapped transform, gain compensation,
912  * clip and convert to integer.
913  *
914  * @param q pointer to the COOKContext
915  * @param decode_buffer pointer to the mlt coefficients
916  * @param gains_ptr array of current/prev gain pointers
917  * @param previous_buffer pointer to the previous buffer to be used for overlapping
918  * @param out pointer to the output buffer
919  */
920 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
921  cook_gains *gains_ptr, float *previous_buffer,
922  float *out)
923 {
924  imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
925  if (out)
926  q->saturate_output(q, out);
927 }
928 
929 
930 /**
931  * Cook subpacket decoding. This function returns one decoded subpacket,
932  * usually 1024 samples per channel.
933  *
934  * @param q pointer to the COOKContext
935  * @param inbuffer pointer to the inbuffer
936  * @param outbuffer pointer to the outbuffer
937  */
939  const uint8_t *inbuffer, float **outbuffer)
940 {
941  int sub_packet_size = p->size;
942  int res;
943 
944  memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
945  decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
946 
947  if (p->joint_stereo) {
948  if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
949  return res;
950  } else {
951  if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
952  return res;
953 
954  if (p->num_channels == 2) {
955  decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
956  if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
957  return res;
958  }
959  }
960 
963  outbuffer ? outbuffer[p->ch_idx] : NULL);
964 
965  if (p->num_channels == 2) {
966  if (p->joint_stereo)
969  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
970  else
973  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
974  }
975 
976  return 0;
977 }
978 
979 
980 static int cook_decode_frame(AVCodecContext *avctx, void *data,
981  int *got_frame_ptr, AVPacket *avpkt)
982 {
983  AVFrame *frame = data;
984  const uint8_t *buf = avpkt->data;
985  int buf_size = avpkt->size;
986  COOKContext *q = avctx->priv_data;
987  float **samples = NULL;
988  int i, ret;
989  int offset = 0;
990  int chidx = 0;
991 
992  if (buf_size < avctx->block_align)
993  return buf_size;
994 
995  /* get output buffer */
996  if (q->discarded_packets >= 2) {
998  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
999  return ret;
1000  samples = (float **)frame->extended_data;
1001  }
1002 
1003  /* estimate subpacket sizes */
1004  q->subpacket[0].size = avctx->block_align;
1005 
1006  for (i = 1; i < q->num_subpackets; i++) {
1007  q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
1008  q->subpacket[0].size -= q->subpacket[i].size + 1;
1009  if (q->subpacket[0].size < 0) {
1010  av_log(avctx, AV_LOG_DEBUG,
1011  "frame subpacket size total > avctx->block_align!\n");
1012  return AVERROR_INVALIDDATA;
1013  }
1014  }
1015 
1016  /* decode supbackets */
1017  for (i = 0; i < q->num_subpackets; i++) {
1018  q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
1020  q->subpacket[i].ch_idx = chidx;
1021  av_log(avctx, AV_LOG_DEBUG,
1022  "subpacket[%i] size %i js %i %i block_align %i\n",
1024  avctx->block_align);
1025 
1026  if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1027  return ret;
1028  offset += q->subpacket[i].size;
1029  chidx += q->subpacket[i].num_channels;
1030  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1031  i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1032  }
1033 
1034  /* Discard the first two frames: no valid audio. */
1035  if (q->discarded_packets < 2) {
1036  q->discarded_packets++;
1037  *got_frame_ptr = 0;
1038  return avctx->block_align;
1039  }
1040 
1041  *got_frame_ptr = 1;
1042 
1043  return avctx->block_align;
1044 }
1045 
1047 {
1048  //int i=0;
1049 #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1050  ff_dlog(q->avctx, "COOKextradata\n");
1051  ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1052  if (q->subpacket[0].cookversion > STEREO) {
1053  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1054  PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1055  }
1056  ff_dlog(q->avctx, "COOKContext\n");
1057  PRINT("nb_channels", q->avctx->channels);
1058  PRINT("bit_rate", (int)q->avctx->bit_rate);
1059  PRINT("sample_rate", q->avctx->sample_rate);
1060  PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1061  PRINT("subbands", q->subpacket[0].subbands);
1062  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1063  PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1064  PRINT("numvector_size", q->subpacket[0].numvector_size);
1065  PRINT("total_subbands", q->subpacket[0].total_subbands);
1066 }
1067 
1068 /**
1069  * Cook initialization
1070  *
1071  * @param avctx pointer to the AVCodecContext
1072  */
1074 {
1075  COOKContext *q = avctx->priv_data;
1076  GetByteContext gb;
1077  int s = 0;
1078  unsigned int channel_mask = 0;
1079  int samples_per_frame = 0;
1080  int ret;
1081  q->avctx = avctx;
1082 
1083  /* Take care of the codec specific extradata. */
1084  if (avctx->extradata_size < 8) {
1085  av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1086  return AVERROR_INVALIDDATA;
1087  }
1088  av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1089 
1090  bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1091 
1092  /* Take data from the AVCodecContext (RM container). */
1093  if (!avctx->channels) {
1094  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1095  return AVERROR_INVALIDDATA;
1096  }
1097 
1098  if (avctx->block_align >= INT_MAX / 8)
1099  return AVERROR(EINVAL);
1100 
1101  /* Initialize RNG. */
1102  av_lfg_init(&q->random_state, 0);
1103 
1104  ff_audiodsp_init(&q->adsp);
1105 
1106  while (bytestream2_get_bytes_left(&gb)) {
1107  if (s >= FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
1108  avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1109  return AVERROR_PATCHWELCOME;
1110  }
1111  /* 8 for mono, 16 for stereo, ? for multichannel
1112  Swap to right endianness so we don't need to care later on. */
1113  q->subpacket[s].cookversion = bytestream2_get_be32(&gb);
1114  samples_per_frame = bytestream2_get_be16(&gb);
1115  q->subpacket[s].subbands = bytestream2_get_be16(&gb);
1116  bytestream2_get_be32(&gb); // Unknown unused
1117  q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
1118  if (q->subpacket[s].js_subband_start >= 51) {
1119  av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
1120  return AVERROR_INVALIDDATA;
1121  }
1122  q->subpacket[s].js_vlc_bits = bytestream2_get_be16(&gb);
1123 
1124  /* Initialize extradata related variables. */
1125  q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1126  q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1127 
1128  /* Initialize default data states. */
1131  q->subpacket[s].num_channels = 1;
1132 
1133  /* Initialize version-dependent variables */
1134 
1135  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1136  q->subpacket[s].cookversion);
1137  q->subpacket[s].joint_stereo = 0;
1138  switch (q->subpacket[s].cookversion) {
1139  case MONO:
1140  if (avctx->channels != 1) {
1141  avpriv_request_sample(avctx, "Container channels != 1");
1142  return AVERROR_PATCHWELCOME;
1143  }
1144  av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1145  break;
1146  case STEREO:
1147  if (avctx->channels != 1) {
1148  q->subpacket[s].bits_per_subpdiv = 1;
1149  q->subpacket[s].num_channels = 2;
1150  }
1151  av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1152  break;
1153  case JOINT_STEREO:
1154  if (avctx->channels != 2) {
1155  avpriv_request_sample(avctx, "Container channels != 2");
1156  return AVERROR_PATCHWELCOME;
1157  }
1158  av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1159  if (avctx->extradata_size >= 16) {
1162  q->subpacket[s].joint_stereo = 1;
1163  q->subpacket[s].num_channels = 2;
1164  }
1165  if (q->subpacket[s].samples_per_channel > 256) {
1167  }
1168  if (q->subpacket[s].samples_per_channel > 512) {
1170  }
1171  break;
1172  case MC_COOK:
1173  av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1174  channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
1175 
1179  q->subpacket[s].joint_stereo = 1;
1180  q->subpacket[s].num_channels = 2;
1181  q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1182 
1183  if (q->subpacket[s].samples_per_channel > 256) {
1185  }
1186  if (q->subpacket[s].samples_per_channel > 512) {
1188  }
1189  } else
1190  q->subpacket[s].samples_per_channel = samples_per_frame;
1191 
1192  break;
1193  default:
1194  avpriv_request_sample(avctx, "Cook version %d",
1195  q->subpacket[s].cookversion);
1196  return AVERROR_PATCHWELCOME;
1197  }
1198 
1199  if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1200  av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1201  return AVERROR_INVALIDDATA;
1202  } else
1204 
1205 
1206  /* Initialize variable relations */
1208 
1209  /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1210  if (q->subpacket[s].total_subbands > 53) {
1211  avpriv_request_sample(avctx, "total_subbands > 53");
1212  return AVERROR_PATCHWELCOME;
1213  }
1214 
1215  if ((q->subpacket[s].js_vlc_bits > 6) ||
1216  (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1217  av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1219  return AVERROR_INVALIDDATA;
1220  }
1221 
1222  if (q->subpacket[s].subbands > 50) {
1223  avpriv_request_sample(avctx, "subbands > 50");
1224  return AVERROR_PATCHWELCOME;
1225  }
1226  if (q->subpacket[s].subbands == 0) {
1227  avpriv_request_sample(avctx, "subbands = 0");
1228  return AVERROR_PATCHWELCOME;
1229  }
1230  q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1232  q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1234 
1235  if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
1236  av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
1237  return AVERROR_INVALIDDATA;
1238  }
1239 
1240  q->num_subpackets++;
1241  s++;
1242  }
1243 
1244  /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1245  if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1246  q->samples_per_channel != 1024) {
1247  avpriv_request_sample(avctx, "samples_per_channel = %d",
1248  q->samples_per_channel);
1249  return AVERROR_PATCHWELCOME;
1250  }
1251 
1252  /* Generate tables */
1253  init_pow2table();
1254  init_gain_table(q);
1256 
1257  if ((ret = init_cook_vlc_tables(q)))
1258  return ret;
1259 
1260  /* Pad the databuffer with:
1261  DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1262  AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1264  av_mallocz(avctx->block_align
1265  + DECODE_BYTES_PAD1(avctx->block_align)
1267  if (!q->decoded_bytes_buffer)
1268  return AVERROR(ENOMEM);
1269 
1270  /* Initialize transform. */
1271  if ((ret = init_cook_mlt(q)))
1272  return ret;
1273 
1274  /* Initialize COOK signal arithmetic handling */
1275  if (1) {
1277  q->decouple = decouple_float;
1281  }
1282 
1283  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1284  if (channel_mask)
1285  avctx->channel_layout = channel_mask;
1286  else
1287  avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1288 
1289 
1290  dump_cook_context(q);
1291 
1292  return 0;
1293 }
1294 
1296  .name = "cook",
1297  .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1298  .type = AVMEDIA_TYPE_AUDIO,
1299  .id = AV_CODEC_ID_COOK,
1300  .priv_data_size = sizeof(COOKContext),
1302  .close = cook_decode_close,
1304  .capabilities = AV_CODEC_CAP_DR1,
1305  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1306  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1308 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
subbands
Definition: aptx.h:39
#define av_cold
Definition: attributes.h:88
uint8_t
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
void ff_free_vlc(VLC *vlc)
Definition: bitstream.c:431
int ff_init_vlc_from_lengths(VLC *vlc_arg, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags, void *logctx)
Build VLC decoding tables suitable for use with get_vlc2()
Definition: bitstream.c:381
#define AV_BE2NE32C(x)
Definition: bswap.h:103
static av_always_inline int bytestream2_get_bytes_left(GetByteContext *g)
Definition: bytestream.h:158
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:137
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:551
#define s(width, name)
Definition: cbs_vp9.c:257
audio channel layout utility functions
#define FFSWAP(type, a, b)
Definition: common.h:108
#define FFMIN(a, b)
Definition: common.h:105
#define av_clip_uintp2
Definition: common.h:146
static int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
Definition: cook.c:301
static av_cold int build_vlc(VLC *vlc, int nb_bits, const uint8_t counts[16], const void *syms, int symbol_size, int offset, void *logctx)
Definition: cook.c:197
static void imlt_gain(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer)
The modulated lapped transform, this takes transform coefficients and transforms them into timedomain...
Definition: cook.c:737
static int decode_envelope(COOKContext *q, COOKSubpacket *p, int *quant_index_table)
Create the quant index table needed for the envelope.
Definition: cook.c:383
static void saturate_output_float(COOKContext *q, float *out)
Saturate the output signal and interleave.
Definition: cook.c:902
static int joint_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer_left, float *mlt_buffer_right)
function for decoding joint stereo data
Definition: cook.c:828
static void dump_cook_context(COOKContext *q)
Definition: cook.c:1046
#define QUANT_VLC_BITS
Definition: cook.c:69
#define SUBBAND_SIZE
Definition: cook.c:66
static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table, int *category, int *category_index)
Calculate the category and category_index vector.
Definition: cook.c:424
static void interpolate_float(COOKContext *q, float *buffer, int gain_index, int gain_index_next)
the actual requantization of the timedomain samples
Definition: cook.c:682
#define JOINT_STEREO
Definition: cook.c:63
static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
function for getting the jointstereo coupling information
Definition: cook.c:767
#define STEREO
Definition: cook.c:62
static int cook_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: cook.c:980
static av_cold void init_pow2table(void)
Definition: cook.c:171
#define COUPLING_VLC_BITS
Definition: cook.c:70
static void decouple_float(COOKContext *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2)
function decouples a pair of signals from a single signal via multiplication.
Definition: cook.c:806
static void imlt_window_float(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer)
Apply transform window, overlap buffers.
Definition: cook.c:709
static float rootpow2tab[127]
Definition: cook.c:166
AVCodec ff_cook_decoder
Definition: cook.c:1295
static av_cold void init_cplscales_table(COOKContext *q)
Definition: cook.c:269
#define PRINT(a, b)
#define MC_COOK
Definition: cook.c:64
static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, int *subband_coef_index, int *subband_coef_sign)
Unpack the subband_coef_index and subband_coef_sign vectors.
Definition: cook.c:572
static int decode_subpacket(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, float **outbuffer)
Cook subpacket decoding.
Definition: cook.c:938
#define DECODE_BYTES_PAD1(bytes)
Definition: cook.c:278
static void expand_category(COOKContext *q, int *category, int *category_index)
Expand the category vector.
Definition: cook.c:521
static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
Definition: cook.c:652
#define MONO
Definition: cook.c:61
static void scalar_dequant_float(COOKContext *q, int index, int quant_index, int *subband_coef_index, int *subband_coef_sign, float *mlt_p)
The real requantization of the mltcoefs.
Definition: cook.c:543
static av_cold void init_gain_table(COOKContext *q)
Definition: cook.c:188
static void decode_gain_info(GetBitContext *gb, int *gaininfo)
Fill the gain array for the timedomain quantization.
Definition: cook.c:359
static av_cold int init_cook_mlt(COOKContext *q)
Definition: cook.c:245
#define MAX_SUBPACKETS
Definition: cook.c:67
static void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, cook_gains *gains_ptr)
First part of subpacket decoding: decode raw stream bytes and read gain info.
Definition: cook.c:880
static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category, int *quant_index_table, float *mlt_buffer)
Fill the mlt_buffer with mlt coefficients.
Definition: cook.c:616
static void mlt_compensate_output(COOKContext *q, float *decode_buffer, cook_gains *gains_ptr, float *previous_buffer, float *out)
Final part of subpacket decoding: Apply modulated lapped transform, gain compensation,...
Definition: cook.c:920
static av_cold int cook_decode_init(AVCodecContext *avctx)
Cook initialization.
Definition: cook.c:1073
static av_cold int init_cook_vlc_tables(COOKContext *q)
Definition: cook.c:213
static float pow2tab[127]
Definition: cook.c:165
static av_cold int cook_decode_close(AVCodecContext *avctx)
Definition: cook.c:327
Cook AKA RealAudio G2 compatible decoder data.
static const int kmax_tab[7]
Definition: cookdata.h:57
static const uint8_t *const ccpl_huffsyms[5]
Definition: cookdata.h:278
static const void *const cvh_huffsyms[7]
Definition: cookdata.h:241
static const int expbits_tab[8]
Definition: cookdata.h:35
static const float quant_centroid_tab[7][14]
Definition: cookdata.h:43
static const uint8_t ccpl_huffcounts[5][16]
Definition: cookdata.h:270
static const uint8_t cvh_huffcounts[7][16]
Definition: cookdata.h:125
static const int vd_tab[7]
Definition: cookdata.h:61
static const int vpr_tab[7]
Definition: cookdata.h:65
static const uint8_t envelope_quant_index_huffcounts[13][16]
Definition: cookdata.h:79
#define MAX_COOK_VLC_ENTRIES
Definition: cookdata.h:73
static const float dither_tab[9]
Definition: cookdata.h:39
static const uint8_t envelope_quant_index_huffsyms[13][24]
Definition: cookdata.h:95
static const float *const cplscales[5]
Definition: cookdata.h:357
static const int cplband[51]
Definition: cookdata.h:285
static const int invradix_tab[7]
Definition: cookdata.h:53
static const int vhvlcsize_tab[7]
Definition: cookdata.h:75
#define NULL
Definition: coverity.c:32
#define max(a, b)
Definition: cuda_runtime.h:33
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1893
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static AVFrame * frame
#define ff_mdct_init
Definition: fft.h:161
#define ff_mdct_end
Definition: fft.h:162
bitstream reader API header.
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:797
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
@ AV_CODEC_ID_COOK
Definition: codec_id.h:444
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding.
Definition: avcodec.h:215
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:117
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
int index
Definition: gxfenc.c:89
int i
Definition: input.c:407
#define av_log2
Definition: intmath.h:83
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:32
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:53
av_cold void ff_audiodsp_init(AudioDSPContext *c)
Definition: audiodsp.c:106
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:49
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define powf(x, y)
Definition: libm.h:50
#define FFALIGN(x, a)
Definition: macros.h:48
#define M_SQRT2
Definition: mathematics.h:61
const char data[16]
Definition: mxf.c:142
category
Definition: openal-dec.c:248
typedef void(RENAME(mix_any_func_type))
static char buffer[20]
Definition: seek.c:32
void ff_sine_window_init(float *window, int n)
Generate a sine window.
#define FF_ARRAY_ELEMS(a)
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
int sample_rate
samples per second
Definition: avcodec.h:1196
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int channels
number of audio channels
Definition: avcodec.h:1197
int extradata_size
Definition: avcodec.h:638
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1233
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
Context structure for the Lagged Fibonacci PRNG.
Definition: lfg.h:33
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
void(* vector_clipf)(float *dst, const float *src, int len, float min, float max)
Definition: audiodsp.h:49
int samples_per_channel
Definition: cook.c:135
void(* imlt_window)(struct cook *q, float *buffer1, cook_gains *gains_ptr, float *previous_buffer)
Definition: cook.c:122
AudioDSPContext adsp
Definition: cook.c:131
int num_vectors
Definition: cook.c:134
VLC envelope_quant_index[13]
Definition: cook.c:145
int gain_size_factor
Definition: cook.c:149
float gain_table[31]
Definition: cook.c:150
float decode_buffer_0[1060]
Definition: cook.c:158
AVLFG random_state
Definition: cook.c:137
FFTContext mdct_ctx
Definition: cook.c:141
void(* scalar_dequant)(struct cook *q, int index, int quant_index, int *subband_coef_index, int *subband_coef_sign, float *mlt_p)
Definition: cook.c:111
uint8_t * decoded_bytes_buffer
Definition: cook.c:154
int num_subpackets
Definition: cook.c:161
void(* decouple)(struct cook *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2)
Definition: cook.c:115
float * mlt_window
Definition: cook.c:142
int discarded_packets
Definition: cook.c:138
GetBitContext gb
Definition: cook.c:132
void(* interpolate)(struct cook *q, float *buffer, int gain_index, int gain_index_next)
Definition: cook.c:125
float decode_buffer_1[1024]
Definition: cook.c:156
float mono_mdct_output[2048]
Definition: cook.c:155
AVCodecContext * avctx
Definition: cook.c:130
COOKSubpacket subpacket[MAX_SUBPACKETS]
Definition: cook.c:162
VLC sqvh[7]
Definition: cook.c:146
const float * cplscales[5]
Definition: cook.c:160
float decode_buffer_2[1024]
Definition: cook.c:157
void(* saturate_output)(struct cook *q, float *out)
Definition: cook.c:128
int gain_4[9]
Definition: cook.c:103
float mono_previous_buffer1[1024]
Definition: cook.c:95
int subbands
Definition: cook.c:82
int total_subbands
Definition: cook.c:92
int gain_1[9]
Definition: cook.c:100
int size
Definition: cook.c:79
int js_subband_start
Definition: cook.c:83
int samples_per_channel
Definition: cook.c:85
int bits_per_subpdiv
Definition: cook.c:91
int numvector_size
Definition: cook.c:93
int num_channels
Definition: cook.c:80
unsigned int channel_mask
Definition: cook.c:87
int log2_numvector_size
Definition: cook.c:86
int bits_per_subpacket
Definition: cook.c:90
VLC channel_coupling
Definition: cook.c:88
int cookversion
Definition: cook.c:81
cook_gains gains1
Definition: cook.c:98
int joint_stereo
Definition: cook.c:89
int gain_3[9]
Definition: cook.c:102
int gain_2[9]
Definition: cook.c:101
int js_vlc_bits
Definition: cook.c:84
float mono_previous_buffer2[1024]
Definition: cook.c:96
cook_gains gains2
Definition: cook.c:99
int ch_idx
Definition: cook.c:78
Definition: fft.h:83
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:102
Definition: vlc.h:26
int bits
Definition: vlc.h:27
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
int * now
Definition: cook.c:73
int * previous
Definition: cook.c:74
#define av_malloc_array(a, b)
#define ff_dlog(a,...)
#define avpriv_request_sample(...)
#define av_freep(p)
#define av_log(a,...)
static uint8_t tmp[11]
Definition: aes_ctr.c:27
FILE * out
Definition: movenc.c:54
static void interpolate(float *out, float v1, float v2, int size)
Definition: twinvq.c:84
static const struct twinvq_data tab
static int get_unary(GetBitContext *gb, int stop, int len)
Get unary code of limited length.
Definition: unary.h:46
static const uint8_t offset[127][2]
Definition: vf_spp.c:107
float min
static double c[64]