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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 #include "avcodec.h"
48 #include "get_bits.h"
49 #include "dsputil.h"
50 #include "bytestream.h"
51 #include "fft.h"
52 #include "internal.h"
53 #include "sinewin.h"
54 
55 #include "cookdata.h"
56 
57 /* the different Cook versions */
58 #define MONO 0x1000001
59 #define STEREO 0x1000002
60 #define JOINT_STEREO 0x1000003
61 #define MC_COOK 0x2000000 // multichannel Cook, not supported
62 
63 #define SUBBAND_SIZE 20
64 #define MAX_SUBPACKETS 5
65 
66 typedef struct {
67  int *now;
68  int *previous;
69 } cook_gains;
70 
71 typedef struct {
72  int ch_idx;
73  int size;
76  int subbands;
81  unsigned int channel_mask;
87  int numvector_size; // 1 << log2_numvector_size;
88 
89  float mono_previous_buffer1[1024];
90  float mono_previous_buffer2[1024];
91 
94  int gain_1[9];
95  int gain_2[9];
96  int gain_3[9];
97  int gain_4[9];
99 
100 typedef struct cook {
101  /*
102  * The following 5 functions provide the lowlevel arithmetic on
103  * the internal audio buffers.
104  */
105  void (*scalar_dequant)(struct cook *q, int index, int quant_index,
106  int *subband_coef_index, int *subband_coef_sign,
107  float *mlt_p);
108 
109  void (*decouple)(struct cook *q,
110  COOKSubpacket *p,
111  int subband,
112  float f1, float f2,
113  float *decode_buffer,
114  float *mlt_buffer1, float *mlt_buffer2);
115 
116  void (*imlt_window)(struct cook *q, float *buffer1,
117  cook_gains *gains_ptr, float *previous_buffer);
118 
119  void (*interpolate)(struct cook *q, float *buffer,
120  int gain_index, int gain_index_next);
121 
122  void (*saturate_output)(struct cook *q, float *out);
123 
128  /* stream data */
131  /* states */
134 
135  /* transform data */
137  float* mlt_window;
138 
139  /* VLC data */
140  VLC envelope_quant_index[13];
141  VLC sqvh[7]; // scalar quantization
142 
143  /* generatable tables and related variables */
145  float gain_table[23];
146 
147  /* data buffers */
148 
150  DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
151  float decode_buffer_1[1024];
152  float decode_buffer_2[1024];
153  float decode_buffer_0[1060]; /* static allocation for joint decode */
154 
155  const float *cplscales[5];
158 } COOKContext;
159 
160 static float pow2tab[127];
161 static float rootpow2tab[127];
162 
163 /*************** init functions ***************/
164 
165 /* table generator */
166 static av_cold void init_pow2table(void)
167 {
168  int i;
169  for (i = -63; i < 64; i++) {
170  pow2tab[63 + i] = pow(2, i);
171  rootpow2tab[63 + i] = sqrt(pow(2, i));
172  }
173 }
174 
175 /* table generator */
177 {
178  int i;
180  for (i = 0; i < 23; i++)
181  q->gain_table[i] = pow(pow2tab[i + 52],
182  (1.0 / (double) q->gain_size_factor));
183 }
184 
185 
187 {
188  int i, result;
189 
190  result = 0;
191  for (i = 0; i < 13; i++) {
192  result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
194  envelope_quant_index_huffcodes[i], 2, 2, 0);
195  }
196  av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
197  for (i = 0; i < 7; i++) {
198  result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
199  cvh_huffbits[i], 1, 1,
200  cvh_huffcodes[i], 2, 2, 0);
201  }
202 
203  for (i = 0; i < q->num_subpackets; i++) {
204  if (q->subpacket[i].joint_stereo == 1) {
205  result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
206  (1 << q->subpacket[i].js_vlc_bits) - 1,
207  ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
208  ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
209  av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
210  }
211  }
212 
213  av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
214  return result;
215 }
216 
218 {
219  int j, ret;
220  int mlt_size = q->samples_per_channel;
221 
222  if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
223  return AVERROR(ENOMEM);
224 
225  /* Initialize the MLT window: simple sine window. */
226  ff_sine_window_init(q->mlt_window, mlt_size);
227  for (j = 0; j < mlt_size; j++)
228  q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
229 
230  /* Initialize the MDCT. */
231  if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
232  av_free(q->mlt_window);
233  return ret;
234  }
235  av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
236  av_log2(mlt_size) + 1);
237 
238  return 0;
239 }
240 
242 {
243  int i;
244  for (i = 0; i < 5; i++)
245  q->cplscales[i] = cplscales[i];
246 }
247 
248 /*************** init functions end ***********/
249 
250 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
251 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
252 
253 /**
254  * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
255  * Why? No idea, some checksum/error detection method maybe.
256  *
257  * Out buffer size: extra bytes are needed to cope with
258  * padding/misalignment.
259  * Subpackets passed to the decoder can contain two, consecutive
260  * half-subpackets, of identical but arbitrary size.
261  * 1234 1234 1234 1234 extraA extraB
262  * Case 1: AAAA BBBB 0 0
263  * Case 2: AAAA ABBB BB-- 3 3
264  * Case 3: AAAA AABB BBBB 2 2
265  * Case 4: AAAA AAAB BBBB BB-- 1 5
266  *
267  * Nice way to waste CPU cycles.
268  *
269  * @param inbuffer pointer to byte array of indata
270  * @param out pointer to byte array of outdata
271  * @param bytes number of bytes
272  */
273 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
274 {
275  static const uint32_t tab[4] = {
276  AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
277  AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
278  };
279  int i, off;
280  uint32_t c;
281  const uint32_t *buf;
282  uint32_t *obuf = (uint32_t *) out;
283  /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
284  * I'm too lazy though, should be something like
285  * for (i = 0; i < bitamount / 64; i++)
286  * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
287  * Buffer alignment needs to be checked. */
288 
289  off = (intptr_t) inbuffer & 3;
290  buf = (const uint32_t *) (inbuffer - off);
291  c = tab[off];
292  bytes += 3 + off;
293  for (i = 0; i < bytes / 4; i++)
294  obuf[i] = c ^ buf[i];
295 
296  return off;
297 }
298 
300 {
301  int i;
302  COOKContext *q = avctx->priv_data;
303  av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
304 
305  /* Free allocated memory buffers. */
306  av_free(q->mlt_window);
308 
309  /* Free the transform. */
310  ff_mdct_end(&q->mdct_ctx);
311 
312  /* Free the VLC tables. */
313  for (i = 0; i < 13; i++)
315  for (i = 0; i < 7; i++)
316  ff_free_vlc(&q->sqvh[i]);
317  for (i = 0; i < q->num_subpackets; i++)
319 
320  av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
321 
322  return 0;
323 }
324 
325 /**
326  * Fill the gain array for the timedomain quantization.
327  *
328  * @param gb pointer to the GetBitContext
329  * @param gaininfo array[9] of gain indexes
330  */
331 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
332 {
333  int i, n;
334 
335  while (get_bits1(gb)) {
336  /* NOTHING */
337  }
338 
339  n = get_bits_count(gb) - 1; // amount of elements*2 to update
340 
341  i = 0;
342  while (n--) {
343  int index = get_bits(gb, 3);
344  int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
345 
346  while (i <= index)
347  gaininfo[i++] = gain;
348  }
349  while (i <= 8)
350  gaininfo[i++] = 0;
351 }
352 
353 /**
354  * Create the quant index table needed for the envelope.
355  *
356  * @param q pointer to the COOKContext
357  * @param quant_index_table pointer to the array
358  */
360  int *quant_index_table)
361 {
362  int i, j, vlc_index;
363 
364  quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
365 
366  for (i = 1; i < p->total_subbands; i++) {
367  vlc_index = i;
368  if (i >= p->js_subband_start * 2) {
369  vlc_index -= p->js_subband_start;
370  } else {
371  vlc_index /= 2;
372  if (vlc_index < 1)
373  vlc_index = 1;
374  }
375  if (vlc_index > 13)
376  vlc_index = 13; // the VLC tables >13 are identical to No. 13
377 
378  j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
379  q->envelope_quant_index[vlc_index - 1].bits, 2);
380  quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
381  if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
383  "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
384  quant_index_table[i], i);
385  return AVERROR_INVALIDDATA;
386  }
387  }
388 
389  return 0;
390 }
391 
392 /**
393  * Calculate the category and category_index vector.
394  *
395  * @param q pointer to the COOKContext
396  * @param quant_index_table pointer to the array
397  * @param category pointer to the category array
398  * @param category_index pointer to the category_index array
399  */
400 static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
401  int *category, int *category_index)
402 {
403  int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
404  int exp_index2[102] = { 0 };
405  int exp_index1[102] = { 0 };
406 
407  int tmp_categorize_array[128 * 2] = { 0 };
408  int tmp_categorize_array1_idx = p->numvector_size;
409  int tmp_categorize_array2_idx = p->numvector_size;
410 
411  bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
412 
413  if (bits_left > q->samples_per_channel)
414  bits_left = q->samples_per_channel +
415  ((bits_left - q->samples_per_channel) * 5) / 8;
416 
417  bias = -32;
418 
419  /* Estimate bias. */
420  for (i = 32; i > 0; i = i / 2) {
421  num_bits = 0;
422  index = 0;
423  for (j = p->total_subbands; j > 0; j--) {
424  exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
425  index++;
426  num_bits += expbits_tab[exp_idx];
427  }
428  if (num_bits >= bits_left - 32)
429  bias += i;
430  }
431 
432  /* Calculate total number of bits. */
433  num_bits = 0;
434  for (i = 0; i < p->total_subbands; i++) {
435  exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
436  num_bits += expbits_tab[exp_idx];
437  exp_index1[i] = exp_idx;
438  exp_index2[i] = exp_idx;
439  }
440  tmpbias1 = tmpbias2 = num_bits;
441 
442  for (j = 1; j < p->numvector_size; j++) {
443  if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
444  int max = -999999;
445  index = -1;
446  for (i = 0; i < p->total_subbands; i++) {
447  if (exp_index1[i] < 7) {
448  v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
449  if (v >= max) {
450  max = v;
451  index = i;
452  }
453  }
454  }
455  if (index == -1)
456  break;
457  tmp_categorize_array[tmp_categorize_array1_idx++] = index;
458  tmpbias1 -= expbits_tab[exp_index1[index]] -
459  expbits_tab[exp_index1[index] + 1];
460  ++exp_index1[index];
461  } else { /* <--- */
462  int min = 999999;
463  index = -1;
464  for (i = 0; i < p->total_subbands; i++) {
465  if (exp_index2[i] > 0) {
466  v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
467  if (v < min) {
468  min = v;
469  index = i;
470  }
471  }
472  }
473  if (index == -1)
474  break;
475  tmp_categorize_array[--tmp_categorize_array2_idx] = index;
476  tmpbias2 -= expbits_tab[exp_index2[index]] -
477  expbits_tab[exp_index2[index] - 1];
478  --exp_index2[index];
479  }
480  }
481 
482  for (i = 0; i < p->total_subbands; i++)
483  category[i] = exp_index2[i];
484 
485  for (i = 0; i < p->numvector_size - 1; i++)
486  category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
487 }
488 
489 
490 /**
491  * Expand the category vector.
492  *
493  * @param q pointer to the COOKContext
494  * @param category pointer to the category array
495  * @param category_index pointer to the category_index array
496  */
497 static inline void expand_category(COOKContext *q, int *category,
498  int *category_index)
499 {
500  int i;
501  for (i = 0; i < q->num_vectors; i++)
502  {
503  int idx = category_index[i];
504  if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
505  --category[idx];
506  }
507 }
508 
509 /**
510  * The real requantization of the mltcoefs
511  *
512  * @param q pointer to the COOKContext
513  * @param index index
514  * @param quant_index quantisation index
515  * @param subband_coef_index array of indexes to quant_centroid_tab
516  * @param subband_coef_sign signs of coefficients
517  * @param mlt_p pointer into the mlt buffer
518  */
519 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
520  int *subband_coef_index, int *subband_coef_sign,
521  float *mlt_p)
522 {
523  int i;
524  float f1;
525 
526  for (i = 0; i < SUBBAND_SIZE; i++) {
527  if (subband_coef_index[i]) {
528  f1 = quant_centroid_tab[index][subband_coef_index[i]];
529  if (subband_coef_sign[i])
530  f1 = -f1;
531  } else {
532  /* noise coding if subband_coef_index[i] == 0 */
533  f1 = dither_tab[index];
534  if (av_lfg_get(&q->random_state) < 0x80000000)
535  f1 = -f1;
536  }
537  mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
538  }
539 }
540 /**
541  * Unpack the subband_coef_index and subband_coef_sign vectors.
542  *
543  * @param q pointer to the COOKContext
544  * @param category pointer to the category array
545  * @param subband_coef_index array of indexes to quant_centroid_tab
546  * @param subband_coef_sign signs of coefficients
547  */
548 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
549  int *subband_coef_index, int *subband_coef_sign)
550 {
551  int i, j;
552  int vlc, vd, tmp, result;
553 
554  vd = vd_tab[category];
555  result = 0;
556  for (i = 0; i < vpr_tab[category]; i++) {
557  vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
558  if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
559  vlc = 0;
560  result = 1;
561  }
562  for (j = vd - 1; j >= 0; j--) {
563  tmp = (vlc * invradix_tab[category]) / 0x100000;
564  subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
565  vlc = tmp;
566  }
567  for (j = 0; j < vd; j++) {
568  if (subband_coef_index[i * vd + j]) {
569  if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
570  subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
571  } else {
572  result = 1;
573  subband_coef_sign[i * vd + j] = 0;
574  }
575  } else {
576  subband_coef_sign[i * vd + j] = 0;
577  }
578  }
579  }
580  return result;
581 }
582 
583 
584 /**
585  * Fill the mlt_buffer with mlt coefficients.
586  *
587  * @param q pointer to the COOKContext
588  * @param category pointer to the category array
589  * @param quant_index_table pointer to the array
590  * @param mlt_buffer pointer to mlt coefficients
591  */
592 static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
593  int *quant_index_table, float *mlt_buffer)
594 {
595  /* A zero in this table means that the subband coefficient is
596  random noise coded. */
597  int subband_coef_index[SUBBAND_SIZE];
598  /* A zero in this table means that the subband coefficient is a
599  positive multiplicator. */
600  int subband_coef_sign[SUBBAND_SIZE];
601  int band, j;
602  int index = 0;
603 
604  for (band = 0; band < p->total_subbands; band++) {
605  index = category[band];
606  if (category[band] < 7) {
607  if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
608  index = 7;
609  for (j = 0; j < p->total_subbands; j++)
610  category[band + j] = 7;
611  }
612  }
613  if (index >= 7) {
614  memset(subband_coef_index, 0, sizeof(subband_coef_index));
615  memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
616  }
617  q->scalar_dequant(q, index, quant_index_table[band],
618  subband_coef_index, subband_coef_sign,
619  &mlt_buffer[band * SUBBAND_SIZE]);
620  }
621 
622  /* FIXME: should this be removed, or moved into loop above? */
624  return;
625 }
626 
627 
628 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
629 {
630  int category_index[128] = { 0 };
631  int category[128] = { 0 };
632  int quant_index_table[102];
633  int res, i;
634 
635  if ((res = decode_envelope(q, p, quant_index_table)) < 0)
636  return res;
638  categorize(q, p, quant_index_table, category, category_index);
639  expand_category(q, category, category_index);
640  for (i=0; i<p->total_subbands; i++) {
641  if (category[i] > 7)
642  return AVERROR_INVALIDDATA;
643  }
644  decode_vectors(q, p, category, quant_index_table, mlt_buffer);
645 
646  return 0;
647 }
648 
649 
650 /**
651  * the actual requantization of the timedomain samples
652  *
653  * @param q pointer to the COOKContext
654  * @param buffer pointer to the timedomain buffer
655  * @param gain_index index for the block multiplier
656  * @param gain_index_next index for the next block multiplier
657  */
658 static void interpolate_float(COOKContext *q, float *buffer,
659  int gain_index, int gain_index_next)
660 {
661  int i;
662  float fc1, fc2;
663  fc1 = pow2tab[gain_index + 63];
664 
665  if (gain_index == gain_index_next) { // static gain
666  for (i = 0; i < q->gain_size_factor; i++)
667  buffer[i] *= fc1;
668  } else { // smooth gain
669  fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
670  for (i = 0; i < q->gain_size_factor; i++) {
671  buffer[i] *= fc1;
672  fc1 *= fc2;
673  }
674  }
675 }
676 
677 /**
678  * Apply transform window, overlap buffers.
679  *
680  * @param q pointer to the COOKContext
681  * @param inbuffer pointer to the mltcoefficients
682  * @param gains_ptr current and previous gains
683  * @param previous_buffer pointer to the previous buffer to be used for overlapping
684  */
685 static void imlt_window_float(COOKContext *q, float *inbuffer,
686  cook_gains *gains_ptr, float *previous_buffer)
687 {
688  const float fc = pow2tab[gains_ptr->previous[0] + 63];
689  int i;
690  /* The weird thing here, is that the two halves of the time domain
691  * buffer are swapped. Also, the newest data, that we save away for
692  * next frame, has the wrong sign. Hence the subtraction below.
693  * Almost sounds like a complex conjugate/reverse data/FFT effect.
694  */
695 
696  /* Apply window and overlap */
697  for (i = 0; i < q->samples_per_channel; i++)
698  inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
699  previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
700 }
701 
702 /**
703  * The modulated lapped transform, this takes transform coefficients
704  * and transforms them into timedomain samples.
705  * Apply transform window, overlap buffers, apply gain profile
706  * and buffer management.
707  *
708  * @param q pointer to the COOKContext
709  * @param inbuffer pointer to the mltcoefficients
710  * @param gains_ptr current and previous gains
711  * @param previous_buffer pointer to the previous buffer to be used for overlapping
712  */
713 static void imlt_gain(COOKContext *q, float *inbuffer,
714  cook_gains *gains_ptr, float *previous_buffer)
715 {
716  float *buffer0 = q->mono_mdct_output;
717  float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
718  int i;
719 
720  /* Inverse modified discrete cosine transform */
721  q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
722 
723  q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
724 
725  /* Apply gain profile */
726  for (i = 0; i < 8; i++)
727  if (gains_ptr->now[i] || gains_ptr->now[i + 1])
728  q->interpolate(q, &buffer1[q->gain_size_factor * i],
729  gains_ptr->now[i], gains_ptr->now[i + 1]);
730 
731  /* Save away the current to be previous block. */
732  memcpy(previous_buffer, buffer0,
733  q->samples_per_channel * sizeof(*previous_buffer));
734 }
735 
736 
737 /**
738  * function for getting the jointstereo coupling information
739  *
740  * @param q pointer to the COOKContext
741  * @param decouple_tab decoupling array
742  */
743 static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
744 {
745  int i;
746  int vlc = get_bits1(&q->gb);
747  int start = cplband[p->js_subband_start];
748  int end = cplband[p->subbands - 1];
749  int length = end - start + 1;
750 
751  if (start > end)
752  return 0;
753 
754  if (vlc)
755  for (i = 0; i < length; i++)
756  decouple_tab[start + i] = get_vlc2(&q->gb,
758  p->channel_coupling.bits, 2);
759  else
760  for (i = 0; i < length; i++) {
761  int v = get_bits(&q->gb, p->js_vlc_bits);
762  if (v == (1<<p->js_vlc_bits)-1) {
763  av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
764  return AVERROR_INVALIDDATA;
765  }
766  decouple_tab[start + i] = v;
767  }
768  return 0;
769 }
770 
771 /**
772  * function decouples a pair of signals from a single signal via multiplication.
773  *
774  * @param q pointer to the COOKContext
775  * @param subband index of the current subband
776  * @param f1 multiplier for channel 1 extraction
777  * @param f2 multiplier for channel 2 extraction
778  * @param decode_buffer input buffer
779  * @param mlt_buffer1 pointer to left channel mlt coefficients
780  * @param mlt_buffer2 pointer to right channel mlt coefficients
781  */
783  COOKSubpacket *p,
784  int subband,
785  float f1, float f2,
786  float *decode_buffer,
787  float *mlt_buffer1, float *mlt_buffer2)
788 {
789  int j, tmp_idx;
790  for (j = 0; j < SUBBAND_SIZE; j++) {
791  tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
792  mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
793  mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
794  }
795 }
796 
797 /**
798  * function for decoding joint stereo data
799  *
800  * @param q pointer to the COOKContext
801  * @param mlt_buffer1 pointer to left channel mlt coefficients
802  * @param mlt_buffer2 pointer to right channel mlt coefficients
803  */
805  float *mlt_buffer_left, float *mlt_buffer_right)
806 {
807  int i, j, res;
808  int decouple_tab[SUBBAND_SIZE] = { 0 };
809  float *decode_buffer = q->decode_buffer_0;
810  int idx, cpl_tmp;
811  float f1, f2;
812  const float *cplscale;
813 
814  memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
815 
816  /* Make sure the buffers are zeroed out. */
817  memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
818  memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
819  if ((res = decouple_info(q, p, decouple_tab)) < 0)
820  return res;
821  if ((res = mono_decode(q, p, decode_buffer)) < 0)
822  return res;
823  /* The two channels are stored interleaved in decode_buffer. */
824  for (i = 0; i < p->js_subband_start; i++) {
825  for (j = 0; j < SUBBAND_SIZE; j++) {
826  mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
827  mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
828  }
829  }
830 
831  /* When we reach js_subband_start (the higher frequencies)
832  the coefficients are stored in a coupling scheme. */
833  idx = (1 << p->js_vlc_bits) - 1;
834  for (i = p->js_subband_start; i < p->subbands; i++) {
835  cpl_tmp = cplband[i];
836  idx -= decouple_tab[cpl_tmp];
837  cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
838  f1 = cplscale[decouple_tab[cpl_tmp] + 1];
839  f2 = cplscale[idx];
840  q->decouple(q, p, i, f1, f2, decode_buffer,
841  mlt_buffer_left, mlt_buffer_right);
842  idx = (1 << p->js_vlc_bits) - 1;
843  }
844 
845  return 0;
846 }
847 
848 /**
849  * First part of subpacket decoding:
850  * decode raw stream bytes and read gain info.
851  *
852  * @param q pointer to the COOKContext
853  * @param inbuffer pointer to raw stream data
854  * @param gains_ptr array of current/prev gain pointers
855  */
857  const uint8_t *inbuffer,
858  cook_gains *gains_ptr)
859 {
860  int offset;
861 
862  offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
863  p->bits_per_subpacket / 8);
864  init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
865  p->bits_per_subpacket);
866  decode_gain_info(&q->gb, gains_ptr->now);
867 
868  /* Swap current and previous gains */
869  FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
870 }
871 
872 /**
873  * Saturate the output signal and interleave.
874  *
875  * @param q pointer to the COOKContext
876  * @param out pointer to the output vector
877  */
878 static void saturate_output_float(COOKContext *q, float *out)
879 {
881  -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
882 }
883 
884 
885 /**
886  * Final part of subpacket decoding:
887  * Apply modulated lapped transform, gain compensation,
888  * clip and convert to integer.
889  *
890  * @param q pointer to the COOKContext
891  * @param decode_buffer pointer to the mlt coefficients
892  * @param gains_ptr array of current/prev gain pointers
893  * @param previous_buffer pointer to the previous buffer to be used for overlapping
894  * @param out pointer to the output buffer
895  */
896 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
897  cook_gains *gains_ptr, float *previous_buffer,
898  float *out)
899 {
900  imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
901  if (out)
902  q->saturate_output(q, out);
903 }
904 
905 
906 /**
907  * Cook subpacket decoding. This function returns one decoded subpacket,
908  * usually 1024 samples per channel.
909  *
910  * @param q pointer to the COOKContext
911  * @param inbuffer pointer to the inbuffer
912  * @param outbuffer pointer to the outbuffer
913  */
915  const uint8_t *inbuffer, float **outbuffer)
916 {
917  int sub_packet_size = p->size;
918  int res;
919 
920  memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
921  decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
922 
923  if (p->joint_stereo) {
924  if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
925  return res;
926  } else {
927  if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
928  return res;
929 
930  if (p->num_channels == 2) {
931  decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
932  if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
933  return res;
934  }
935  }
936 
939  outbuffer ? outbuffer[p->ch_idx] : NULL);
940 
941  if (p->num_channels == 2) {
942  if (p->joint_stereo)
945  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
946  else
949  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
950  }
951 
952  return 0;
953 }
954 
955 
956 static int cook_decode_frame(AVCodecContext *avctx, void *data,
957  int *got_frame_ptr, AVPacket *avpkt)
958 {
959  const uint8_t *buf = avpkt->data;
960  int buf_size = avpkt->size;
961  COOKContext *q = avctx->priv_data;
962  float **samples = NULL;
963  int i, ret;
964  int offset = 0;
965  int chidx = 0;
966 
967  if (buf_size < avctx->block_align)
968  return buf_size;
969 
970  /* get output buffer */
971  if (q->discarded_packets >= 2) {
973  if ((ret = ff_get_buffer(avctx, &q->frame)) < 0) {
974  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
975  return ret;
976  }
977  samples = (float **)q->frame.extended_data;
978  }
979 
980  /* estimate subpacket sizes */
981  q->subpacket[0].size = avctx->block_align;
982 
983  for (i = 1; i < q->num_subpackets; i++) {
984  q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
985  q->subpacket[0].size -= q->subpacket[i].size + 1;
986  if (q->subpacket[0].size < 0) {
987  av_log(avctx, AV_LOG_DEBUG,
988  "frame subpacket size total > avctx->block_align!\n");
989  return AVERROR_INVALIDDATA;
990  }
991  }
992 
993  /* decode supbackets */
994  for (i = 0; i < q->num_subpackets; i++) {
995  q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
997  q->subpacket[i].ch_idx = chidx;
998  av_log(avctx, AV_LOG_DEBUG,
999  "subpacket[%i] size %i js %i %i block_align %i\n",
1000  i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
1001  avctx->block_align);
1002 
1003  if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1004  return ret;
1005  offset += q->subpacket[i].size;
1006  chidx += q->subpacket[i].num_channels;
1007  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1008  i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1009  }
1010 
1011  /* Discard the first two frames: no valid audio. */
1012  if (q->discarded_packets < 2) {
1013  q->discarded_packets++;
1014  *got_frame_ptr = 0;
1015  return avctx->block_align;
1016  }
1017 
1018  *got_frame_ptr = 1;
1019  *(AVFrame *) data = q->frame;
1020 
1021  return avctx->block_align;
1022 }
1023 
1024 #ifdef DEBUG
1025 static void dump_cook_context(COOKContext *q)
1026 {
1027  //int i=0;
1028 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
1029  av_dlog(q->avctx, "COOKextradata\n");
1030  av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1031  if (q->subpacket[0].cookversion > STEREO) {
1032  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1033  PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1034  }
1035  av_dlog(q->avctx, "COOKContext\n");
1036  PRINT("nb_channels", q->avctx->channels);
1037  PRINT("bit_rate", q->avctx->bit_rate);
1038  PRINT("sample_rate", q->avctx->sample_rate);
1039  PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1040  PRINT("subbands", q->subpacket[0].subbands);
1041  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1042  PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1043  PRINT("numvector_size", q->subpacket[0].numvector_size);
1044  PRINT("total_subbands", q->subpacket[0].total_subbands);
1045 }
1046 #endif
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 <= 0) {
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_dsputil_init(&q->dsp, avctx);
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  q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
1096  extradata_size -= 8;
1097  }
1098 
1099  /* Initialize extradata related variables. */
1100  q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1101  q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1102 
1103  /* Initialize default data states. */
1104  q->subpacket[s].log2_numvector_size = 5;
1106  q->subpacket[s].num_channels = 1;
1107 
1108  /* Initialize version-dependent variables */
1109 
1110  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1111  q->subpacket[s].cookversion);
1112  q->subpacket[s].joint_stereo = 0;
1113  switch (q->subpacket[s].cookversion) {
1114  case MONO:
1115  if (avctx->channels != 1) {
1116  av_log_ask_for_sample(avctx, "Container channels != 1.\n");
1117  return AVERROR_PATCHWELCOME;
1118  }
1119  av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1120  break;
1121  case STEREO:
1122  if (avctx->channels != 1) {
1123  q->subpacket[s].bits_per_subpdiv = 1;
1124  q->subpacket[s].num_channels = 2;
1125  }
1126  av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1127  break;
1128  case JOINT_STEREO:
1129  if (avctx->channels != 2) {
1130  av_log_ask_for_sample(avctx, "Container channels != 2.\n");
1131  return AVERROR_PATCHWELCOME;
1132  }
1133  av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1134  if (avctx->extradata_size >= 16) {
1135  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1137  q->subpacket[s].joint_stereo = 1;
1138  q->subpacket[s].num_channels = 2;
1139  }
1140  if (q->subpacket[s].samples_per_channel > 256) {
1141  q->subpacket[s].log2_numvector_size = 6;
1142  }
1143  if (q->subpacket[s].samples_per_channel > 512) {
1144  q->subpacket[s].log2_numvector_size = 7;
1145  }
1146  break;
1147  case MC_COOK:
1148  av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1149  if (extradata_size >= 4)
1150  channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
1151 
1153  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1155  q->subpacket[s].joint_stereo = 1;
1156  q->subpacket[s].num_channels = 2;
1157  q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1158 
1159  if (q->subpacket[s].samples_per_channel > 256) {
1160  q->subpacket[s].log2_numvector_size = 6;
1161  }
1162  if (q->subpacket[s].samples_per_channel > 512) {
1163  q->subpacket[s].log2_numvector_size = 7;
1164  }
1165  } else
1166  q->subpacket[s].samples_per_channel = samples_per_frame;
1167 
1168  break;
1169  default:
1170  av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
1171  return AVERROR_PATCHWELCOME;
1172  }
1173 
1174  if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1175  av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1176  return AVERROR_INVALIDDATA;
1177  } else
1179 
1180 
1181  /* Initialize variable relations */
1183 
1184  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1185  if (q->subpacket[s].total_subbands > 53) {
1186  av_log_ask_for_sample(avctx, "total_subbands > 53\n");
1187  return AVERROR_PATCHWELCOME;
1188  }
1189 
1190  if ((q->subpacket[s].js_vlc_bits > 6) ||
1191  (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1192  av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1193  q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1194  return AVERROR_INVALIDDATA;
1195  }
1196 
1197  if (q->subpacket[s].subbands > 50) {
1198  av_log_ask_for_sample(avctx, "subbands > 50\n");
1199  return AVERROR_PATCHWELCOME;
1200  }
1201  if (q->subpacket[s].subbands == 0) {
1202  av_log_ask_for_sample(avctx, "subbands is 0\n");
1203  return AVERROR_PATCHWELCOME;
1204  }
1205  q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1206  q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1207  q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1208  q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1209 
1210  if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
1211  av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
1212  return AVERROR_INVALIDDATA;
1213  }
1214 
1215  q->num_subpackets++;
1216  s++;
1217  if (s > MAX_SUBPACKETS) {
1218  av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
1219  return AVERROR_PATCHWELCOME;
1220  }
1221  }
1222  /* Generate tables */
1223  init_pow2table();
1224  init_gain_table(q);
1226 
1227  if ((ret = init_cook_vlc_tables(q)))
1228  return ret;
1229 
1230 
1231  if (avctx->block_align >= UINT_MAX / 2)
1232  return AVERROR(EINVAL);
1233 
1234  /* Pad the databuffer with:
1235  DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1236  FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1238  av_mallocz(avctx->block_align
1239  + DECODE_BYTES_PAD1(avctx->block_align)
1241  if (q->decoded_bytes_buffer == NULL)
1242  return AVERROR(ENOMEM);
1243 
1244  /* Initialize transform. */
1245  if ((ret = init_cook_mlt(q)))
1246  return ret;
1247 
1248  /* Initialize COOK signal arithmetic handling */
1249  if (1) {
1251  q->decouple = decouple_float;
1255  }
1256 
1257  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1258  if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1259  q->samples_per_channel != 1024) {
1260  av_log_ask_for_sample(avctx,
1261  "unknown amount of samples_per_channel = %d\n",
1262  q->samples_per_channel);
1263  return AVERROR_PATCHWELCOME;
1264  }
1265 
1266  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1267  if (channel_mask)
1268  avctx->channel_layout = channel_mask;
1269  else
1270  avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1271 
1273  avctx->coded_frame = &q->frame;
1274 
1275 #ifdef DEBUG
1276  dump_cook_context(q);
1277 #endif
1278  return 0;
1279 }
1280 
1282  .name = "cook",
1283  .type = AVMEDIA_TYPE_AUDIO,
1284  .id = AV_CODEC_ID_COOK,
1285  .priv_data_size = sizeof(COOKContext),
1289  .capabilities = CODEC_CAP_DR1,
1290  .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1291  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1293 };