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cook.c
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1 /*
2  * COOK compatible decoder
3  * Copyright (c) 2003 Sascha Sommer
4  * Copyright (c) 2005 Benjamin Larsson
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 /**
24  * @file
25  * Cook compatible decoder. Bastardization of the G.722.1 standard.
26  * This decoder handles RealNetworks, RealAudio G2 data.
27  * Cook is identified by the codec name cook in RM files.
28  *
29  * To use this decoder, a calling application must supply the extradata
30  * bytes provided from the RM container; 8+ bytes for mono streams and
31  * 16+ for stereo streams (maybe more).
32  *
33  * Codec technicalities (all this assume a buffer length of 1024):
34  * Cook works with several different techniques to achieve its compression.
35  * In the timedomain the buffer is divided into 8 pieces and quantized. If
36  * two neighboring pieces have different quantization index a smooth
37  * quantization curve is used to get a smooth overlap between the different
38  * pieces.
39  * To get to the transformdomain Cook uses a modulated lapped transform.
40  * The transform domain has 50 subbands with 20 elements each. This
41  * means only a maximum of 50*20=1000 coefficients are used out of the 1024
42  * available.
43  */
44 
46 #include "libavutil/lfg.h"
47 
48 #include "audiodsp.h"
49 #include "avcodec.h"
50 #include "get_bits.h"
51 #include "bytestream.h"
52 #include "fft.h"
53 #include "internal.h"
54 #include "sinewin.h"
55 #include "unary.h"
56 
57 #include "cookdata.h"
58 
59 /* the different Cook versions */
60 #define MONO 0x1000001
61 #define STEREO 0x1000002
62 #define JOINT_STEREO 0x1000003
63 #define MC_COOK 0x2000000 // multichannel Cook, not supported
64 
65 #define SUBBAND_SIZE 20
66 #define MAX_SUBPACKETS 5
67 
68 typedef struct cook_gains {
69  int *now;
70  int *previous;
71 } cook_gains;
72 
73 typedef struct COOKSubpacket {
74  int ch_idx;
75  int size;
78  int subbands;
83  unsigned int channel_mask;
89  int numvector_size; // 1 << log2_numvector_size;
90 
91  float mono_previous_buffer1[1024];
92  float mono_previous_buffer2[1024];
93 
96  int gain_1[9];
97  int gain_2[9];
98  int gain_3[9];
99  int gain_4[9];
100 } COOKSubpacket;
101 
102 typedef struct cook {
103  /*
104  * The following 5 functions provide the lowlevel arithmetic on
105  * the internal audio buffers.
106  */
107  void (*scalar_dequant)(struct cook *q, int index, int quant_index,
108  int *subband_coef_index, int *subband_coef_sign,
109  float *mlt_p);
110 
111  void (*decouple)(struct cook *q,
112  COOKSubpacket *p,
113  int subband,
114  float f1, float f2,
115  float *decode_buffer,
116  float *mlt_buffer1, float *mlt_buffer2);
117 
118  void (*imlt_window)(struct cook *q, float *buffer1,
119  cook_gains *gains_ptr, float *previous_buffer);
120 
121  void (*interpolate)(struct cook *q, float *buffer,
122  int gain_index, int gain_index_next);
123 
124  void (*saturate_output)(struct cook *q, float *out);
125 
129  /* stream data */
132  /* states */
135 
136  /* transform data */
138  float* mlt_window;
139 
140  /* VLC data */
141  VLC envelope_quant_index[13];
142  VLC sqvh[7]; // scalar quantization
143 
144  /* generatable tables and related variables */
146  float gain_table[23];
147 
148  /* data buffers */
149 
151  DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
152  float decode_buffer_1[1024];
153  float decode_buffer_2[1024];
154  float decode_buffer_0[1060]; /* static allocation for joint decode */
155 
156  const float *cplscales[5];
159 } COOKContext;
160 
161 static float pow2tab[127];
162 static float rootpow2tab[127];
163 
164 /*************** init functions ***************/
165 
166 /* table generator */
167 static av_cold void init_pow2table(void)
168 {
169  /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
170  int i;
171  static const float exp2_tab[2] = {1, M_SQRT2};
172  float exp2_val = powf(2, -63);
173  float root_val = powf(2, -32);
174  for (i = -63; i < 64; i++) {
175  if (!(i & 1))
176  root_val *= 2;
177  pow2tab[63 + i] = exp2_val;
178  rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
179  exp2_val *= 2;
180  }
181 }
182 
183 /* table generator */
185 {
186  int i;
188  for (i = 0; i < 23; i++)
189  q->gain_table[i] = pow(pow2tab[i + 52],
190  (1.0 / (double) q->gain_size_factor));
191 }
192 
193 
195 {
196  int i, result;
197 
198  result = 0;
199  for (i = 0; i < 13; i++) {
200  result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
202  envelope_quant_index_huffcodes[i], 2, 2, 0);
203  }
204  av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
205  for (i = 0; i < 7; i++) {
206  result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
207  cvh_huffbits[i], 1, 1,
208  cvh_huffcodes[i], 2, 2, 0);
209  }
210 
211  for (i = 0; i < q->num_subpackets; i++) {
212  if (q->subpacket[i].joint_stereo == 1) {
213  result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
214  (1 << q->subpacket[i].js_vlc_bits) - 1,
215  ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
216  ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
217  av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
218  }
219  }
220 
221  av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
222  return result;
223 }
224 
226 {
227  int j, ret;
228  int mlt_size = q->samples_per_channel;
229 
230  if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
231  return AVERROR(ENOMEM);
232 
233  /* Initialize the MLT window: simple sine window. */
234  ff_sine_window_init(q->mlt_window, mlt_size);
235  for (j = 0; j < mlt_size; j++)
236  q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
237 
238  /* Initialize the MDCT. */
239  if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
240  av_freep(&q->mlt_window);
241  return ret;
242  }
243  av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
244  av_log2(mlt_size) + 1);
245 
246  return 0;
247 }
248 
250 {
251  int i;
252  for (i = 0; i < 5; i++)
253  q->cplscales[i] = cplscales[i];
254 }
255 
256 /*************** init functions end ***********/
257 
258 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
259 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
260 
261 /**
262  * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
263  * Why? No idea, some checksum/error detection method maybe.
264  *
265  * Out buffer size: extra bytes are needed to cope with
266  * padding/misalignment.
267  * Subpackets passed to the decoder can contain two, consecutive
268  * half-subpackets, of identical but arbitrary size.
269  * 1234 1234 1234 1234 extraA extraB
270  * Case 1: AAAA BBBB 0 0
271  * Case 2: AAAA ABBB BB-- 3 3
272  * Case 3: AAAA AABB BBBB 2 2
273  * Case 4: AAAA AAAB BBBB BB-- 1 5
274  *
275  * Nice way to waste CPU cycles.
276  *
277  * @param inbuffer pointer to byte array of indata
278  * @param out pointer to byte array of outdata
279  * @param bytes number of bytes
280  */
281 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
282 {
283  static const uint32_t tab[4] = {
284  AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
285  AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
286  };
287  int i, off;
288  uint32_t c;
289  const uint32_t *buf;
290  uint32_t *obuf = (uint32_t *) out;
291  /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
292  * I'm too lazy though, should be something like
293  * for (i = 0; i < bitamount / 64; i++)
294  * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
295  * Buffer alignment needs to be checked. */
296 
297  off = (intptr_t) inbuffer & 3;
298  buf = (const uint32_t *) (inbuffer - off);
299  c = tab[off];
300  bytes += 3 + off;
301  for (i = 0; i < bytes / 4; i++)
302  obuf[i] = c ^ buf[i];
303 
304  return off;
305 }
306 
308 {
309  int i;
310  COOKContext *q = avctx->priv_data;
311  av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
312 
313  /* Free allocated memory buffers. */
314  av_freep(&q->mlt_window);
316 
317  /* Free the transform. */
318  ff_mdct_end(&q->mdct_ctx);
319 
320  /* Free the VLC tables. */
321  for (i = 0; i < 13; i++)
323  for (i = 0; i < 7; i++)
324  ff_free_vlc(&q->sqvh[i]);
325  for (i = 0; i < q->num_subpackets; i++)
327 
328  av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
329 
330  return 0;
331 }
332 
333 /**
334  * Fill the gain array for the timedomain quantization.
335  *
336  * @param gb pointer to the GetBitContext
337  * @param gaininfo array[9] of gain indexes
338  */
339 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
340 {
341  int i, n;
342 
343  n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update
344 
345  i = 0;
346  while (n--) {
347  int index = get_bits(gb, 3);
348  int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
349 
350  while (i <= index)
351  gaininfo[i++] = gain;
352  }
353  while (i <= 8)
354  gaininfo[i++] = 0;
355 }
356 
357 /**
358  * Create the quant index table needed for the envelope.
359  *
360  * @param q pointer to the COOKContext
361  * @param quant_index_table pointer to the array
362  */
364  int *quant_index_table)
365 {
366  int i, j, vlc_index;
367 
368  quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
369 
370  for (i = 1; i < p->total_subbands; i++) {
371  vlc_index = i;
372  if (i >= p->js_subband_start * 2) {
373  vlc_index -= p->js_subband_start;
374  } else {
375  vlc_index /= 2;
376  if (vlc_index < 1)
377  vlc_index = 1;
378  }
379  if (vlc_index > 13)
380  vlc_index = 13; // the VLC tables >13 are identical to No. 13
381 
382  j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
383  q->envelope_quant_index[vlc_index - 1].bits, 2);
384  quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
385  if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
387  "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
388  quant_index_table[i], i);
389  return AVERROR_INVALIDDATA;
390  }
391  }
392 
393  return 0;
394 }
395 
396 /**
397  * Calculate the category and category_index vector.
398  *
399  * @param q pointer to the COOKContext
400  * @param quant_index_table pointer to the array
401  * @param category pointer to the category array
402  * @param category_index pointer to the category_index array
403  */
404 static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
405  int *category, int *category_index)
406 {
407  int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
408  int exp_index2[102] = { 0 };
409  int exp_index1[102] = { 0 };
410 
411  int tmp_categorize_array[128 * 2] = { 0 };
412  int tmp_categorize_array1_idx = p->numvector_size;
413  int tmp_categorize_array2_idx = p->numvector_size;
414 
415  bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
416 
417  if (bits_left > q->samples_per_channel)
418  bits_left = q->samples_per_channel +
419  ((bits_left - q->samples_per_channel) * 5) / 8;
420 
421  bias = -32;
422 
423  /* Estimate bias. */
424  for (i = 32; i > 0; i = i / 2) {
425  num_bits = 0;
426  index = 0;
427  for (j = p->total_subbands; j > 0; j--) {
428  exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
429  index++;
430  num_bits += expbits_tab[exp_idx];
431  }
432  if (num_bits >= bits_left - 32)
433  bias += i;
434  }
435 
436  /* Calculate total number of bits. */
437  num_bits = 0;
438  for (i = 0; i < p->total_subbands; i++) {
439  exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
440  num_bits += expbits_tab[exp_idx];
441  exp_index1[i] = exp_idx;
442  exp_index2[i] = exp_idx;
443  }
444  tmpbias1 = tmpbias2 = num_bits;
445 
446  for (j = 1; j < p->numvector_size; j++) {
447  if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
448  int max = -999999;
449  index = -1;
450  for (i = 0; i < p->total_subbands; i++) {
451  if (exp_index1[i] < 7) {
452  v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
453  if (v >= max) {
454  max = v;
455  index = i;
456  }
457  }
458  }
459  if (index == -1)
460  break;
461  tmp_categorize_array[tmp_categorize_array1_idx++] = index;
462  tmpbias1 -= expbits_tab[exp_index1[index]] -
463  expbits_tab[exp_index1[index] + 1];
464  ++exp_index1[index];
465  } else { /* <--- */
466  int min = 999999;
467  index = -1;
468  for (i = 0; i < p->total_subbands; i++) {
469  if (exp_index2[i] > 0) {
470  v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
471  if (v < min) {
472  min = v;
473  index = i;
474  }
475  }
476  }
477  if (index == -1)
478  break;
479  tmp_categorize_array[--tmp_categorize_array2_idx] = index;
480  tmpbias2 -= expbits_tab[exp_index2[index]] -
481  expbits_tab[exp_index2[index] - 1];
482  --exp_index2[index];
483  }
484  }
485 
486  for (i = 0; i < p->total_subbands; i++)
487  category[i] = exp_index2[i];
488 
489  for (i = 0; i < p->numvector_size - 1; i++)
490  category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
491 }
492 
493 
494 /**
495  * Expand the category vector.
496  *
497  * @param q pointer to the COOKContext
498  * @param category pointer to the category array
499  * @param category_index pointer to the category_index array
500  */
501 static inline void expand_category(COOKContext *q, int *category,
502  int *category_index)
503 {
504  int i;
505  for (i = 0; i < q->num_vectors; i++)
506  {
507  int idx = category_index[i];
508  if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
509  --category[idx];
510  }
511 }
512 
513 /**
514  * The real requantization of the mltcoefs
515  *
516  * @param q pointer to the COOKContext
517  * @param index index
518  * @param quant_index quantisation index
519  * @param subband_coef_index array of indexes to quant_centroid_tab
520  * @param subband_coef_sign signs of coefficients
521  * @param mlt_p pointer into the mlt buffer
522  */
523 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
524  int *subband_coef_index, int *subband_coef_sign,
525  float *mlt_p)
526 {
527  int i;
528  float f1;
529 
530  for (i = 0; i < SUBBAND_SIZE; i++) {
531  if (subband_coef_index[i]) {
532  f1 = quant_centroid_tab[index][subband_coef_index[i]];
533  if (subband_coef_sign[i])
534  f1 = -f1;
535  } else {
536  /* noise coding if subband_coef_index[i] == 0 */
537  f1 = dither_tab[index];
538  if (av_lfg_get(&q->random_state) < 0x80000000)
539  f1 = -f1;
540  }
541  mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
542  }
543 }
544 /**
545  * Unpack the subband_coef_index and subband_coef_sign vectors.
546  *
547  * @param q pointer to the COOKContext
548  * @param category pointer to the category array
549  * @param subband_coef_index array of indexes to quant_centroid_tab
550  * @param subband_coef_sign signs of coefficients
551  */
553  int *subband_coef_index, int *subband_coef_sign)
554 {
555  int i, j;
556  int vlc, vd, tmp, result;
557 
558  vd = vd_tab[category];
559  result = 0;
560  for (i = 0; i < vpr_tab[category]; i++) {
561  vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
562  if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
563  vlc = 0;
564  result = 1;
565  }
566  for (j = vd - 1; j >= 0; j--) {
567  tmp = (vlc * invradix_tab[category]) / 0x100000;
568  subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
569  vlc = tmp;
570  }
571  for (j = 0; j < vd; j++) {
572  if (subband_coef_index[i * vd + j]) {
573  if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
574  subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
575  } else {
576  result = 1;
577  subband_coef_sign[i * vd + j] = 0;
578  }
579  } else {
580  subband_coef_sign[i * vd + j] = 0;
581  }
582  }
583  }
584  return result;
585 }
586 
587 
588 /**
589  * Fill the mlt_buffer with mlt coefficients.
590  *
591  * @param q pointer to the COOKContext
592  * @param category pointer to the category array
593  * @param quant_index_table pointer to the array
594  * @param mlt_buffer pointer to mlt coefficients
595  */
597  int *quant_index_table, float *mlt_buffer)
598 {
599  /* A zero in this table means that the subband coefficient is
600  random noise coded. */
601  int subband_coef_index[SUBBAND_SIZE];
602  /* A zero in this table means that the subband coefficient is a
603  positive multiplicator. */
604  int subband_coef_sign[SUBBAND_SIZE];
605  int band, j;
606  int index = 0;
607 
608  for (band = 0; band < p->total_subbands; band++) {
609  index = category[band];
610  if (category[band] < 7) {
611  if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
612  index = 7;
613  for (j = 0; j < p->total_subbands; j++)
614  category[band + j] = 7;
615  }
616  }
617  if (index >= 7) {
618  memset(subband_coef_index, 0, sizeof(subband_coef_index));
619  memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
620  }
621  q->scalar_dequant(q, index, quant_index_table[band],
622  subband_coef_index, subband_coef_sign,
623  &mlt_buffer[band * SUBBAND_SIZE]);
624  }
625 
626  /* FIXME: should this be removed, or moved into loop above? */
628  return;
629 }
630 
631 
632 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
633 {
634  int category_index[128] = { 0 };
635  int category[128] = { 0 };
636  int quant_index_table[102];
637  int res, i;
638 
639  if ((res = decode_envelope(q, p, quant_index_table)) < 0)
640  return res;
642  categorize(q, p, quant_index_table, category, category_index);
643  expand_category(q, category, category_index);
644  for (i=0; i<p->total_subbands; i++) {
645  if (category[i] > 7)
646  return AVERROR_INVALIDDATA;
647  }
648  decode_vectors(q, p, category, quant_index_table, mlt_buffer);
649 
650  return 0;
651 }
652 
653 
654 /**
655  * the actual requantization of the timedomain samples
656  *
657  * @param q pointer to the COOKContext
658  * @param buffer pointer to the timedomain buffer
659  * @param gain_index index for the block multiplier
660  * @param gain_index_next index for the next block multiplier
661  */
662 static void interpolate_float(COOKContext *q, float *buffer,
663  int gain_index, int gain_index_next)
664 {
665  int i;
666  float fc1, fc2;
667  fc1 = pow2tab[gain_index + 63];
668 
669  if (gain_index == gain_index_next) { // static gain
670  for (i = 0; i < q->gain_size_factor; i++)
671  buffer[i] *= fc1;
672  } else { // smooth gain
673  fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
674  for (i = 0; i < q->gain_size_factor; i++) {
675  buffer[i] *= fc1;
676  fc1 *= fc2;
677  }
678  }
679 }
680 
681 /**
682  * Apply transform window, overlap buffers.
683  *
684  * @param q pointer to the COOKContext
685  * @param inbuffer pointer to the mltcoefficients
686  * @param gains_ptr current and previous gains
687  * @param previous_buffer pointer to the previous buffer to be used for overlapping
688  */
689 static void imlt_window_float(COOKContext *q, float *inbuffer,
690  cook_gains *gains_ptr, float *previous_buffer)
691 {
692  const float fc = pow2tab[gains_ptr->previous[0] + 63];
693  int i;
694  /* The weird thing here, is that the two halves of the time domain
695  * buffer are swapped. Also, the newest data, that we save away for
696  * next frame, has the wrong sign. Hence the subtraction below.
697  * Almost sounds like a complex conjugate/reverse data/FFT effect.
698  */
699 
700  /* Apply window and overlap */
701  for (i = 0; i < q->samples_per_channel; i++)
702  inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
703  previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
704 }
705 
706 /**
707  * The modulated lapped transform, this takes transform coefficients
708  * and transforms them into timedomain samples.
709  * Apply transform window, overlap buffers, apply gain profile
710  * and buffer management.
711  *
712  * @param q pointer to the COOKContext
713  * @param inbuffer pointer to the mltcoefficients
714  * @param gains_ptr current and previous gains
715  * @param previous_buffer pointer to the previous buffer to be used for overlapping
716  */
717 static void imlt_gain(COOKContext *q, float *inbuffer,
718  cook_gains *gains_ptr, float *previous_buffer)
719 {
720  float *buffer0 = q->mono_mdct_output;
721  float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
722  int i;
723 
724  /* Inverse modified discrete cosine transform */
725  q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
726 
727  q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
728 
729  /* Apply gain profile */
730  for (i = 0; i < 8; i++)
731  if (gains_ptr->now[i] || gains_ptr->now[i + 1])
732  q->interpolate(q, &buffer1[q->gain_size_factor * i],
733  gains_ptr->now[i], gains_ptr->now[i + 1]);
734 
735  /* Save away the current to be previous block. */
736  memcpy(previous_buffer, buffer0,
737  q->samples_per_channel * sizeof(*previous_buffer));
738 }
739 
740 
741 /**
742  * function for getting the jointstereo coupling information
743  *
744  * @param q pointer to the COOKContext
745  * @param decouple_tab decoupling array
746  */
747 static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
748 {
749  int i;
750  int vlc = get_bits1(&q->gb);
751  int start = cplband[p->js_subband_start];
752  int end = cplband[p->subbands - 1];
753  int length = end - start + 1;
754 
755  if (start > end)
756  return 0;
757 
758  if (vlc)
759  for (i = 0; i < length; i++)
760  decouple_tab[start + i] = get_vlc2(&q->gb,
762  p->channel_coupling.bits, 2);
763  else
764  for (i = 0; i < length; i++) {
765  int v = get_bits(&q->gb, p->js_vlc_bits);
766  if (v == (1<<p->js_vlc_bits)-1) {
767  av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
768  return AVERROR_INVALIDDATA;
769  }
770  decouple_tab[start + i] = v;
771  }
772  return 0;
773 }
774 
775 /**
776  * function decouples a pair of signals from a single signal via multiplication.
777  *
778  * @param q pointer to the COOKContext
779  * @param subband index of the current subband
780  * @param f1 multiplier for channel 1 extraction
781  * @param f2 multiplier for channel 2 extraction
782  * @param decode_buffer input buffer
783  * @param mlt_buffer1 pointer to left channel mlt coefficients
784  * @param mlt_buffer2 pointer to right channel mlt coefficients
785  */
787  COOKSubpacket *p,
788  int subband,
789  float f1, float f2,
790  float *decode_buffer,
791  float *mlt_buffer1, float *mlt_buffer2)
792 {
793  int j, tmp_idx;
794  for (j = 0; j < SUBBAND_SIZE; j++) {
795  tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
796  mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
797  mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
798  }
799 }
800 
801 /**
802  * function for decoding joint stereo data
803  *
804  * @param q pointer to the COOKContext
805  * @param mlt_buffer1 pointer to left channel mlt coefficients
806  * @param mlt_buffer2 pointer to right channel mlt coefficients
807  */
809  float *mlt_buffer_left, float *mlt_buffer_right)
810 {
811  int i, j, res;
812  int decouple_tab[SUBBAND_SIZE] = { 0 };
813  float *decode_buffer = q->decode_buffer_0;
814  int idx, cpl_tmp;
815  float f1, f2;
816  const float *cplscale;
817 
818  memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
819 
820  /* Make sure the buffers are zeroed out. */
821  memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
822  memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
823  if ((res = decouple_info(q, p, decouple_tab)) < 0)
824  return res;
825  if ((res = mono_decode(q, p, decode_buffer)) < 0)
826  return res;
827  /* The two channels are stored interleaved in decode_buffer. */
828  for (i = 0; i < p->js_subband_start; i++) {
829  for (j = 0; j < SUBBAND_SIZE; j++) {
830  mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
831  mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
832  }
833  }
834 
835  /* When we reach js_subband_start (the higher frequencies)
836  the coefficients are stored in a coupling scheme. */
837  idx = (1 << p->js_vlc_bits) - 1;
838  for (i = p->js_subband_start; i < p->subbands; i++) {
839  cpl_tmp = cplband[i];
840  idx -= decouple_tab[cpl_tmp];
841  cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
842  f1 = cplscale[decouple_tab[cpl_tmp] + 1];
843  f2 = cplscale[idx];
844  q->decouple(q, p, i, f1, f2, decode_buffer,
845  mlt_buffer_left, mlt_buffer_right);
846  idx = (1 << p->js_vlc_bits) - 1;
847  }
848 
849  return 0;
850 }
851 
852 /**
853  * First part of subpacket decoding:
854  * decode raw stream bytes and read gain info.
855  *
856  * @param q pointer to the COOKContext
857  * @param inbuffer pointer to raw stream data
858  * @param gains_ptr array of current/prev gain pointers
859  */
861  const uint8_t *inbuffer,
862  cook_gains *gains_ptr)
863 {
864  int offset;
865 
866  offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
867  p->bits_per_subpacket / 8);
868  init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
869  p->bits_per_subpacket);
870  decode_gain_info(&q->gb, gains_ptr->now);
871 
872  /* Swap current and previous gains */
873  FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
874 }
875 
876 /**
877  * Saturate the output signal and interleave.
878  *
879  * @param q pointer to the COOKContext
880  * @param out pointer to the output vector
881  */
882 static void saturate_output_float(COOKContext *q, float *out)
883 {
885  -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
886 }
887 
888 
889 /**
890  * Final part of subpacket decoding:
891  * Apply modulated lapped transform, gain compensation,
892  * clip and convert to integer.
893  *
894  * @param q pointer to the COOKContext
895  * @param decode_buffer pointer to the mlt coefficients
896  * @param gains_ptr array of current/prev gain pointers
897  * @param previous_buffer pointer to the previous buffer to be used for overlapping
898  * @param out pointer to the output buffer
899  */
900 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
901  cook_gains *gains_ptr, float *previous_buffer,
902  float *out)
903 {
904  imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
905  if (out)
906  q->saturate_output(q, out);
907 }
908 
909 
910 /**
911  * Cook subpacket decoding. This function returns one decoded subpacket,
912  * usually 1024 samples per channel.
913  *
914  * @param q pointer to the COOKContext
915  * @param inbuffer pointer to the inbuffer
916  * @param outbuffer pointer to the outbuffer
917  */
919  const uint8_t *inbuffer, float **outbuffer)
920 {
921  int sub_packet_size = p->size;
922  int res;
923 
924  memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
925  decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
926 
927  if (p->joint_stereo) {
928  if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
929  return res;
930  } else {
931  if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
932  return res;
933 
934  if (p->num_channels == 2) {
935  decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
936  if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
937  return res;
938  }
939  }
940 
943  outbuffer ? outbuffer[p->ch_idx] : NULL);
944 
945  if (p->num_channels == 2) {
946  if (p->joint_stereo)
949  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
950  else
953  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
954  }
955 
956  return 0;
957 }
958 
959 
960 static int cook_decode_frame(AVCodecContext *avctx, void *data,
961  int *got_frame_ptr, AVPacket *avpkt)
962 {
963  AVFrame *frame = data;
964  const uint8_t *buf = avpkt->data;
965  int buf_size = avpkt->size;
966  COOKContext *q = avctx->priv_data;
967  float **samples = NULL;
968  int i, ret;
969  int offset = 0;
970  int chidx = 0;
971 
972  if (buf_size < avctx->block_align)
973  return buf_size;
974 
975  /* get output buffer */
976  if (q->discarded_packets >= 2) {
977  frame->nb_samples = q->samples_per_channel;
978  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
979  return ret;
980  samples = (float **)frame->extended_data;
981  }
982 
983  /* estimate subpacket sizes */
984  q->subpacket[0].size = avctx->block_align;
985 
986  for (i = 1; i < q->num_subpackets; i++) {
987  q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
988  q->subpacket[0].size -= q->subpacket[i].size + 1;
989  if (q->subpacket[0].size < 0) {
990  av_log(avctx, AV_LOG_DEBUG,
991  "frame subpacket size total > avctx->block_align!\n");
992  return AVERROR_INVALIDDATA;
993  }
994  }
995 
996  /* decode supbackets */
997  for (i = 0; i < q->num_subpackets; i++) {
998  q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
1000  q->subpacket[i].ch_idx = chidx;
1001  av_log(avctx, AV_LOG_DEBUG,
1002  "subpacket[%i] size %i js %i %i block_align %i\n",
1003  i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
1004  avctx->block_align);
1005 
1006  if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1007  return ret;
1008  offset += q->subpacket[i].size;
1009  chidx += q->subpacket[i].num_channels;
1010  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1011  i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1012  }
1013 
1014  /* Discard the first two frames: no valid audio. */
1015  if (q->discarded_packets < 2) {
1016  q->discarded_packets++;
1017  *got_frame_ptr = 0;
1018  return avctx->block_align;
1019  }
1020 
1021  *got_frame_ptr = 1;
1022 
1023  return avctx->block_align;
1024 }
1025 
1027 {
1028  //int i=0;
1029 #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1030  ff_dlog(q->avctx, "COOKextradata\n");
1031  ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1032  if (q->subpacket[0].cookversion > STEREO) {
1033  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1034  PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1035  }
1036  ff_dlog(q->avctx, "COOKContext\n");
1037  PRINT("nb_channels", q->avctx->channels);
1038  PRINT("bit_rate", (int)q->avctx->bit_rate);
1039  PRINT("sample_rate", q->avctx->sample_rate);
1040  PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1041  PRINT("subbands", q->subpacket[0].subbands);
1042  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1043  PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1044  PRINT("numvector_size", q->subpacket[0].numvector_size);
1045  PRINT("total_subbands", q->subpacket[0].total_subbands);
1046 }
1047 
1048 /**
1049  * Cook initialization
1050  *
1051  * @param avctx pointer to the AVCodecContext
1052  */
1054 {
1055  COOKContext *q = avctx->priv_data;
1056  const uint8_t *edata_ptr = avctx->extradata;
1057  const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
1058  int extradata_size = avctx->extradata_size;
1059  int s = 0;
1060  unsigned int channel_mask = 0;
1061  int samples_per_frame = 0;
1062  int ret;
1063  q->avctx = avctx;
1064 
1065  /* Take care of the codec specific extradata. */
1066  if (extradata_size < 8) {
1067  av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1068  return AVERROR_INVALIDDATA;
1069  }
1070  av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1071 
1072  /* Take data from the AVCodecContext (RM container). */
1073  if (!avctx->channels) {
1074  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1075  return AVERROR_INVALIDDATA;
1076  }
1077 
1078  /* Initialize RNG. */
1079  av_lfg_init(&q->random_state, 0);
1080 
1081  ff_audiodsp_init(&q->adsp);
1082 
1083  while (edata_ptr < edata_ptr_end) {
1084  /* 8 for mono, 16 for stereo, ? for multichannel
1085  Swap to right endianness so we don't need to care later on. */
1086  if (extradata_size >= 8) {
1087  q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
1088  samples_per_frame = bytestream_get_be16(&edata_ptr);
1089  q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
1090  extradata_size -= 8;
1091  }
1092  if (extradata_size >= 8) {
1093  bytestream_get_be32(&edata_ptr); // Unknown unused
1094  q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
1095  if (q->subpacket[s].js_subband_start >= 51) {
1096  av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
1097  return AVERROR_INVALIDDATA;
1098  }
1099 
1100  q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
1101  extradata_size -= 8;
1102  }
1103 
1104  /* Initialize extradata related variables. */
1105  q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1106  q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1107 
1108  /* Initialize default data states. */
1111  q->subpacket[s].num_channels = 1;
1112 
1113  /* Initialize version-dependent variables */
1114 
1115  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1116  q->subpacket[s].cookversion);
1117  q->subpacket[s].joint_stereo = 0;
1118  switch (q->subpacket[s].cookversion) {
1119  case MONO:
1120  if (avctx->channels != 1) {
1121  avpriv_request_sample(avctx, "Container channels != 1");
1122  return AVERROR_PATCHWELCOME;
1123  }
1124  av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1125  break;
1126  case STEREO:
1127  if (avctx->channels != 1) {
1128  q->subpacket[s].bits_per_subpdiv = 1;
1129  q->subpacket[s].num_channels = 2;
1130  }
1131  av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1132  break;
1133  case JOINT_STEREO:
1134  if (avctx->channels != 2) {
1135  avpriv_request_sample(avctx, "Container channels != 2");
1136  return AVERROR_PATCHWELCOME;
1137  }
1138  av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1139  if (avctx->extradata_size >= 16) {
1142  q->subpacket[s].joint_stereo = 1;
1143  q->subpacket[s].num_channels = 2;
1144  }
1145  if (q->subpacket[s].samples_per_channel > 256) {
1147  }
1148  if (q->subpacket[s].samples_per_channel > 512) {
1150  }
1151  break;
1152  case MC_COOK:
1153  av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1154  if (extradata_size >= 4)
1155  channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
1156 
1160  q->subpacket[s].joint_stereo = 1;
1161  q->subpacket[s].num_channels = 2;
1162  q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1163 
1164  if (q->subpacket[s].samples_per_channel > 256) {
1166  }
1167  if (q->subpacket[s].samples_per_channel > 512) {
1169  }
1170  } else
1171  q->subpacket[s].samples_per_channel = samples_per_frame;
1172 
1173  break;
1174  default:
1175  avpriv_request_sample(avctx, "Cook version %d",
1176  q->subpacket[s].cookversion);
1177  return AVERROR_PATCHWELCOME;
1178  }
1179 
1180  if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1181  av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1182  return AVERROR_INVALIDDATA;
1183  } else
1185 
1186 
1187  /* Initialize variable relations */
1189 
1190  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1191  if (q->subpacket[s].total_subbands > 53) {
1192  avpriv_request_sample(avctx, "total_subbands > 53");
1193  return AVERROR_PATCHWELCOME;
1194  }
1195 
1196  if ((q->subpacket[s].js_vlc_bits > 6) ||
1197  (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1198  av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1199  q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1200  return AVERROR_INVALIDDATA;
1201  }
1202 
1203  if (q->subpacket[s].subbands > 50) {
1204  avpriv_request_sample(avctx, "subbands > 50");
1205  return AVERROR_PATCHWELCOME;
1206  }
1207  if (q->subpacket[s].subbands == 0) {
1208  avpriv_request_sample(avctx, "subbands = 0");
1209  return AVERROR_PATCHWELCOME;
1210  }
1211  q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1213  q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1215 
1216  if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
1217  av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
1218  return AVERROR_INVALIDDATA;
1219  }
1220 
1221  q->num_subpackets++;
1222  s++;
1223  if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
1224  avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1225  return AVERROR_PATCHWELCOME;
1226  }
1227  }
1228  /* Generate tables */
1229  init_pow2table();
1230  init_gain_table(q);
1232 
1233  if ((ret = init_cook_vlc_tables(q)))
1234  return ret;
1235 
1236 
1237  if (avctx->block_align >= UINT_MAX / 2)
1238  return AVERROR(EINVAL);
1239 
1240  /* Pad the databuffer with:
1241  DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1242  AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1244  av_mallocz(avctx->block_align
1245  + DECODE_BYTES_PAD1(avctx->block_align)
1247  if (!q->decoded_bytes_buffer)
1248  return AVERROR(ENOMEM);
1249 
1250  /* Initialize transform. */
1251  if ((ret = init_cook_mlt(q)))
1252  return ret;
1253 
1254  /* Initialize COOK signal arithmetic handling */
1255  if (1) {
1257  q->decouple = decouple_float;
1261  }
1262 
1263  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1264  if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1265  q->samples_per_channel != 1024) {
1266  avpriv_request_sample(avctx, "samples_per_channel = %d",
1267  q->samples_per_channel);
1268  return AVERROR_PATCHWELCOME;
1269  }
1270 
1271  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1272  if (channel_mask)
1273  avctx->channel_layout = channel_mask;
1274  else
1275  avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1276 
1277 
1278  dump_cook_context(q);
1279 
1280  return 0;
1281 }
1282 
1284  .name = "cook",
1285  .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1286  .type = AVMEDIA_TYPE_AUDIO,
1287  .id = AV_CODEC_ID_COOK,
1288  .priv_data_size = sizeof(COOKContext),
1290  .close = cook_decode_close,
1292  .capabilities = AV_CODEC_CAP_DR1,
1293  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1295 };
category
Definition: openal-dec.c:241
int joint_stereo
Definition: cook.c:85
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, clip and convert to integer.
Definition: cook.c:900
Definition: lfg.h:25
static av_cold void init_cplscales_table(COOKContext *q)
Definition: cook.c:249
static const int cplband[51]
Definition: cookdata.h:504
float, planar
Definition: samplefmt.h:69
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:631
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:180
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:107
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
VLC channel_coupling
Definition: cook.c:84
#define PRINT(a, b)
static const uint16_t envelope_quant_index_huffcodes[13][24]
Definition: cookdata.h:97
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:260
int * previous
Definition: cook.c:70
float decode_buffer_1[1024]
Definition: cook.c:152
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1707
int gain_1[9]
Definition: cook.c:96
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static const int kmax_tab[7]
Definition: cookdata.h:57
float gain_table[23]
Definition: cook.c:146
static const int expbits_tab[8]
Definition: cookdata.h:35
int size
Definition: avcodec.h:1578
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:404
static const float *const cplscales[5]
Definition: cookdata.h:576
int av_log2(unsigned v)
Definition: intmath.c:26
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:53
int subbands
Definition: cook.c:78
av_cold void ff_audiodsp_init(AudioDSPContext *c)
Definition: audiodsp.c:106
static av_cold void init_pow2table(void)
Definition: cook.c:167
#define AV_CH_LAYOUT_STEREO
VLC envelope_quant_index[13]
Definition: cook.c:141
int num_vectors
Definition: cook.c:130
AVCodec.
Definition: avcodec.h:3535
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2440
int samples_per_channel
Definition: cook.c:81
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
static const uint8_t *const ccpl_huffbits[5]
Definition: cookdata.h:496
static const int vhsize_tab[7]
Definition: cookdata.h:73
static const float quant_centroid_tab[7][14]
Definition: cookdata.h:43
void(* imlt_window)(struct cook *q, float *buffer1, cook_gains *gains_ptr, float *previous_buffer)
Definition: cook.c:118
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:717
AVCodec ff_cook_decoder
Definition: cook.c:1283
static av_cold void init_gain_table(COOKContext *q)
Definition: cook.c:184
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
int numvector_size
Definition: cook.c:89
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2411
uint8_t
int total_subbands
Definition: cook.c:88
#define av_cold
Definition: attributes.h:82
int js_subband_start
Definition: cook.c:79
uint8_t * decoded_bytes_buffer
Definition: cook.c:150
static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
function for getting the jointstereo coupling information
Definition: cook.c:747
float mono_previous_buffer1[1024]
Definition: cook.c:91
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:596
static void expand_category(COOKContext *q, int *category, int *category_index)
Expand the category vector.
Definition: cook.c:501
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1757
int bits_per_subpdiv
Definition: cook.c:87
static AVFrame * frame
static void interpolate(float *out, float v1, float v2, int size)
Definition: twinvq.c:84
cook_gains gains1
Definition: cook.c:94
uint8_t * data
Definition: avcodec.h:1577
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:212
#define ff_dlog(a,...)
bitstream reader API header.
const float * cplscales[5]
Definition: cook.c:156
#define FFALIGN(x, a)
Definition: macros.h:48
static int decode_subpacket(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, float **outbuffer)
Cook subpacket decoding.
Definition: cook.c:918
#define av_log(a,...)
static int cook_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: cook.c:960
AVLFG random_state
Definition: cook.c:133
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:860
#define DECODE_BYTES_PAD1(bytes)
Definition: cook.c:258
GetBitContext gb
Definition: cook.c:128
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:607
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static const int vd_tab[7]
Definition: cookdata.h:61
VLC sqvh[7]
Definition: cook.c:142
#define AVERROR(e)
Definition: error.h:43
static const float dither_tab[9]
Definition: cookdata.h:39
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
#define JOINT_STEREO
Definition: cook.c:62
#define AV_BE2NE32C(x)
Definition: bswap.h:103
static const uint16_t *const ccpl_huffcodes[5]
Definition: cookdata.h:491
GLsizei GLsizei * length
Definition: opengl_enc.c:115
float mono_previous_buffer2[1024]
Definition: cook.c:92
const char * name
Name of the codec implementation.
Definition: avcodec.h:3542
static int decode_envelope(COOKContext *q, COOKSubpacket *p, int *quant_index_table)
Create the quant index table needed for the envelope.
Definition: cook.c:363
#define ff_mdct_init
Definition: fft.h:169
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
int gain_2[9]
Definition: cook.c:97
Definition: get_bits.h:63
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2454
static void saturate_output_float(COOKContext *q, float *out)
Saturate the output signal and interleave.
Definition: cook.c:882
#define powf(x, y)
Definition: libm.h:50
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:107
static const int vhvlcsize_tab[7]
Definition: cookdata.h:77
int gain_3[9]
Definition: cook.c:98
int discarded_packets
Definition: cook.c:134
static const uint16_t fc[]
Definition: dcaenc.h:41
int log2_numvector_size
Definition: cook.c:82
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:552
Definition: fft.h:88
static av_cold int init_cook_mlt(COOKContext *q)
Definition: cook.c:225
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
int gain_4[9]
Definition: cook.c:99
static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
Definition: cook.c:632
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
static uint8_t tmp[8]
Definition: des-test.c:38
static av_cold int cook_decode_init(AVCodecContext *avctx)
Cook initialization.
Definition: cook.c:1053
cook_gains gains2
Definition: cook.c:95
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:574
int n
Definition: avisynth_c.h:547
#define FF_ARRAY_ELEMS(a)
static const uint16_t *const cvh_huffcodes[7]
Definition: cookdata.h:425
int bits
Definition: get_bits.h:64
FILE * out
Definition: movenc-test.c:54
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
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:662
int num_subpackets
Definition: cook.c:157
Libavcodec external API header.
void(* vector_clipf)(float *dst, const float *src, float min, float max, int len)
Definition: audiodsp.h:49
int samples_per_channel
Definition: cook.c:131
void(* interpolate)(struct cook *q, float *buffer, int gain_index, int gain_index_next)
Definition: cook.c:121
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
static av_cold int init_cook_vlc_tables(COOKContext *q)
Definition: cook.c:194
FFTContext mdct_ctx
Definition: cook.c:137
int sample_rate
samples per second
Definition: avcodec.h:2403
void AAC_RENAME() ff_sine_window_init(INTFLOAT *window, int n)
Generate a sine window.
main external API structure.
Definition: avcodec.h:1642
float mono_mdct_output[2048]
Definition: cook.c:151
float * mlt_window
Definition: cook.c:138
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:928
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
Definition: get_bits.h:465
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:38
void * buf
Definition: avisynth_c.h:553
int * now
Definition: cook.c:69
int extradata_size
Definition: avcodec.h:1758
static void dump_cook_context(COOKContext *q)
Definition: cook.c:1026
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:312
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:111
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:808
#define SUBBAND_SIZE
Definition: cook.c:65
static av_cold int cook_decode_close(AVCodecContext *avctx)
Definition: cook.c:307
int index
Definition: gxfenc.c:89
#define MONO
Definition: cook.c:60
static float pow2tab[127]
Definition: cook.c:161
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:418
float decode_buffer_0[1060]
Definition: cook.c:154
AudioDSPContext adsp
Definition: cook.c:127
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:30
COOKSubpacket subpacket[MAX_SUBPACKETS]
Definition: cook.c:158
float decode_buffer_2[1024]
Definition: cook.c:153
static float rootpow2tab[127]
Definition: cook.c:162
static const uint8_t envelope_quant_index_huffbits[13][24]
Definition: cookdata.h:81
static const uint8_t *const cvh_huffbits[7]
Definition: cookdata.h:430
int ch_idx
Definition: cook.c:74
int bits_per_subpacket
Definition: cook.c:86
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:281
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:523
#define M_SQRT2
Definition: mathematics.h:55
int num_channels
Definition: cook.c:76
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:722
common internal api header.
#define STEREO
Definition: cook.c:61
static int get_unary(GetBitContext *gb, int stop, int len)
Get unary code of limited length.
Definition: unary.h:33
#define ff_mdct_end
Definition: fft.h:170
static double c[64]
AVCodecContext * avctx
Definition: cook.c:126
int gain_size_factor
Definition: cook.c:145
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:728
#define MAX_SUBPACKETS
Definition: cook.c:66
void * priv_data
Definition: avcodec.h:1684
static const int invradix_tab[7]
Definition: cookdata.h:53
int channels
number of audio channels
Definition: avcodec.h:2404
#define MC_COOK
Definition: cook.c:63
int js_vlc_bits
Definition: cook.c:80
VLC_TYPE(* table)[2]
code, bits
Definition: get_bits.h:65
static const struct twinvq_data tab
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_freep(p)
static void decode_gain_info(GetBitContext *gb, int *gaininfo)
Fill the gain array for the timedomain quantization.
Definition: cook.c:339
void INT64 start
Definition: avisynth_c.h:553
#define av_malloc_array(a, b)
#define FFSWAP(type, a, b)
Definition: common.h:99
static void imlt_window_float(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer)
Apply transform window, overlap buffers.
Definition: cook.c:689
int cookversion
Definition: cook.c:77
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:786
static const int vpr_tab[7]
Definition: cookdata.h:65
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:227
#define AV_CH_LAYOUT_MONO
float min
This structure stores compressed data.
Definition: avcodec.h:1554
void ff_free_vlc(VLC *vlc)
Definition: bitstream.c:360
int size
Definition: cook.c:75
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:237
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:252
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:953
GLuint buffer
Definition: opengl_enc.c:102
unsigned int channel_mask
Definition: cook.c:83
Cook AKA RealAudio G2 compatible decoderdata.
void(* saturate_output)(struct cook *q, float *out)
Definition: cook.c:124