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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 {
69  int *now;
70  int *previous;
71 } cook_gains;
72 
73 typedef struct {
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  int i;
170  for (i = -63; i < 64; i++) {
171  pow2tab[63 + i] = pow(2, i);
172  rootpow2tab[63 + i] = sqrt(pow(2, i));
173  }
174 }
175 
176 /* table generator */
178 {
179  int i;
181  for (i = 0; i < 23; i++)
182  q->gain_table[i] = pow(pow2tab[i + 52],
183  (1.0 / (double) q->gain_size_factor));
184 }
185 
186 
188 {
189  int i, result;
190 
191  result = 0;
192  for (i = 0; i < 13; i++) {
193  result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
195  envelope_quant_index_huffcodes[i], 2, 2, 0);
196  }
197  av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
198  for (i = 0; i < 7; i++) {
199  result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
200  cvh_huffbits[i], 1, 1,
201  cvh_huffcodes[i], 2, 2, 0);
202  }
203 
204  for (i = 0; i < q->num_subpackets; i++) {
205  if (q->subpacket[i].joint_stereo == 1) {
206  result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
207  (1 << q->subpacket[i].js_vlc_bits) - 1,
208  ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
209  ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
210  av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
211  }
212  }
213 
214  av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
215  return result;
216 }
217 
219 {
220  int j, ret;
221  int mlt_size = q->samples_per_channel;
222 
223  if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
224  return AVERROR(ENOMEM);
225 
226  /* Initialize the MLT window: simple sine window. */
227  ff_sine_window_init(q->mlt_window, mlt_size);
228  for (j = 0; j < mlt_size; j++)
229  q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
230 
231  /* Initialize the MDCT. */
232  if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
233  av_freep(&q->mlt_window);
234  return ret;
235  }
236  av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
237  av_log2(mlt_size) + 1);
238 
239  return 0;
240 }
241 
243 {
244  int i;
245  for (i = 0; i < 5; i++)
246  q->cplscales[i] = cplscales[i];
247 }
248 
249 /*************** init functions end ***********/
250 
251 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
252 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
253 
254 /**
255  * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
256  * Why? No idea, some checksum/error detection method maybe.
257  *
258  * Out buffer size: extra bytes are needed to cope with
259  * padding/misalignment.
260  * Subpackets passed to the decoder can contain two, consecutive
261  * half-subpackets, of identical but arbitrary size.
262  * 1234 1234 1234 1234 extraA extraB
263  * Case 1: AAAA BBBB 0 0
264  * Case 2: AAAA ABBB BB-- 3 3
265  * Case 3: AAAA AABB BBBB 2 2
266  * Case 4: AAAA AAAB BBBB BB-- 1 5
267  *
268  * Nice way to waste CPU cycles.
269  *
270  * @param inbuffer pointer to byte array of indata
271  * @param out pointer to byte array of outdata
272  * @param bytes number of bytes
273  */
274 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
275 {
276  static const uint32_t tab[4] = {
277  AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
278  AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
279  };
280  int i, off;
281  uint32_t c;
282  const uint32_t *buf;
283  uint32_t *obuf = (uint32_t *) out;
284  /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
285  * I'm too lazy though, should be something like
286  * for (i = 0; i < bitamount / 64; i++)
287  * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
288  * Buffer alignment needs to be checked. */
289 
290  off = (intptr_t) inbuffer & 3;
291  buf = (const uint32_t *) (inbuffer - off);
292  c = tab[off];
293  bytes += 3 + off;
294  for (i = 0; i < bytes / 4; i++)
295  obuf[i] = c ^ buf[i];
296 
297  return off;
298 }
299 
301 {
302  int i;
303  COOKContext *q = avctx->priv_data;
304  av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
305 
306  /* Free allocated memory buffers. */
307  av_freep(&q->mlt_window);
309 
310  /* Free the transform. */
311  ff_mdct_end(&q->mdct_ctx);
312 
313  /* Free the VLC tables. */
314  for (i = 0; i < 13; i++)
316  for (i = 0; i < 7; i++)
317  ff_free_vlc(&q->sqvh[i]);
318  for (i = 0; i < q->num_subpackets; i++)
320 
321  av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
322 
323  return 0;
324 }
325 
326 /**
327  * Fill the gain array for the timedomain quantization.
328  *
329  * @param gb pointer to the GetBitContext
330  * @param gaininfo array[9] of gain indexes
331  */
332 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
333 {
334  int i, n;
335 
336  n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update
337 
338  i = 0;
339  while (n--) {
340  int index = get_bits(gb, 3);
341  int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
342 
343  while (i <= index)
344  gaininfo[i++] = gain;
345  }
346  while (i <= 8)
347  gaininfo[i++] = 0;
348 }
349 
350 /**
351  * Create the quant index table needed for the envelope.
352  *
353  * @param q pointer to the COOKContext
354  * @param quant_index_table pointer to the array
355  */
357  int *quant_index_table)
358 {
359  int i, j, vlc_index;
360 
361  quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
362 
363  for (i = 1; i < p->total_subbands; i++) {
364  vlc_index = i;
365  if (i >= p->js_subband_start * 2) {
366  vlc_index -= p->js_subband_start;
367  } else {
368  vlc_index /= 2;
369  if (vlc_index < 1)
370  vlc_index = 1;
371  }
372  if (vlc_index > 13)
373  vlc_index = 13; // the VLC tables >13 are identical to No. 13
374 
375  j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
376  q->envelope_quant_index[vlc_index - 1].bits, 2);
377  quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
378  if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
380  "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
381  quant_index_table[i], i);
382  return AVERROR_INVALIDDATA;
383  }
384  }
385 
386  return 0;
387 }
388 
389 /**
390  * Calculate the category and category_index vector.
391  *
392  * @param q pointer to the COOKContext
393  * @param quant_index_table pointer to the array
394  * @param category pointer to the category array
395  * @param category_index pointer to the category_index array
396  */
397 static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
398  int *category, int *category_index)
399 {
400  int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
401  int exp_index2[102] = { 0 };
402  int exp_index1[102] = { 0 };
403 
404  int tmp_categorize_array[128 * 2] = { 0 };
405  int tmp_categorize_array1_idx = p->numvector_size;
406  int tmp_categorize_array2_idx = p->numvector_size;
407 
408  bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
409 
410  if (bits_left > q->samples_per_channel)
411  bits_left = q->samples_per_channel +
412  ((bits_left - q->samples_per_channel) * 5) / 8;
413 
414  bias = -32;
415 
416  /* Estimate bias. */
417  for (i = 32; i > 0; i = i / 2) {
418  num_bits = 0;
419  index = 0;
420  for (j = p->total_subbands; j > 0; j--) {
421  exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
422  index++;
423  num_bits += expbits_tab[exp_idx];
424  }
425  if (num_bits >= bits_left - 32)
426  bias += i;
427  }
428 
429  /* Calculate total number of bits. */
430  num_bits = 0;
431  for (i = 0; i < p->total_subbands; i++) {
432  exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
433  num_bits += expbits_tab[exp_idx];
434  exp_index1[i] = exp_idx;
435  exp_index2[i] = exp_idx;
436  }
437  tmpbias1 = tmpbias2 = num_bits;
438 
439  for (j = 1; j < p->numvector_size; j++) {
440  if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
441  int max = -999999;
442  index = -1;
443  for (i = 0; i < p->total_subbands; i++) {
444  if (exp_index1[i] < 7) {
445  v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
446  if (v >= max) {
447  max = v;
448  index = i;
449  }
450  }
451  }
452  if (index == -1)
453  break;
454  tmp_categorize_array[tmp_categorize_array1_idx++] = index;
455  tmpbias1 -= expbits_tab[exp_index1[index]] -
456  expbits_tab[exp_index1[index] + 1];
457  ++exp_index1[index];
458  } else { /* <--- */
459  int min = 999999;
460  index = -1;
461  for (i = 0; i < p->total_subbands; i++) {
462  if (exp_index2[i] > 0) {
463  v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
464  if (v < min) {
465  min = v;
466  index = i;
467  }
468  }
469  }
470  if (index == -1)
471  break;
472  tmp_categorize_array[--tmp_categorize_array2_idx] = index;
473  tmpbias2 -= expbits_tab[exp_index2[index]] -
474  expbits_tab[exp_index2[index] - 1];
475  --exp_index2[index];
476  }
477  }
478 
479  for (i = 0; i < p->total_subbands; i++)
480  category[i] = exp_index2[i];
481 
482  for (i = 0; i < p->numvector_size - 1; i++)
483  category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
484 }
485 
486 
487 /**
488  * Expand the category vector.
489  *
490  * @param q pointer to the COOKContext
491  * @param category pointer to the category array
492  * @param category_index pointer to the category_index array
493  */
494 static inline void expand_category(COOKContext *q, int *category,
495  int *category_index)
496 {
497  int i;
498  for (i = 0; i < q->num_vectors; i++)
499  {
500  int idx = category_index[i];
501  if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
502  --category[idx];
503  }
504 }
505 
506 /**
507  * The real requantization of the mltcoefs
508  *
509  * @param q pointer to the COOKContext
510  * @param index index
511  * @param quant_index quantisation index
512  * @param subband_coef_index array of indexes to quant_centroid_tab
513  * @param subband_coef_sign signs of coefficients
514  * @param mlt_p pointer into the mlt buffer
515  */
516 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
517  int *subband_coef_index, int *subband_coef_sign,
518  float *mlt_p)
519 {
520  int i;
521  float f1;
522 
523  for (i = 0; i < SUBBAND_SIZE; i++) {
524  if (subband_coef_index[i]) {
525  f1 = quant_centroid_tab[index][subband_coef_index[i]];
526  if (subband_coef_sign[i])
527  f1 = -f1;
528  } else {
529  /* noise coding if subband_coef_index[i] == 0 */
530  f1 = dither_tab[index];
531  if (av_lfg_get(&q->random_state) < 0x80000000)
532  f1 = -f1;
533  }
534  mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
535  }
536 }
537 /**
538  * Unpack the subband_coef_index and subband_coef_sign vectors.
539  *
540  * @param q pointer to the COOKContext
541  * @param category pointer to the category array
542  * @param subband_coef_index array of indexes to quant_centroid_tab
543  * @param subband_coef_sign signs of coefficients
544  */
546  int *subband_coef_index, int *subband_coef_sign)
547 {
548  int i, j;
549  int vlc, vd, tmp, result;
550 
551  vd = vd_tab[category];
552  result = 0;
553  for (i = 0; i < vpr_tab[category]; i++) {
554  vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
555  if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
556  vlc = 0;
557  result = 1;
558  }
559  for (j = vd - 1; j >= 0; j--) {
560  tmp = (vlc * invradix_tab[category]) / 0x100000;
561  subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
562  vlc = tmp;
563  }
564  for (j = 0; j < vd; j++) {
565  if (subband_coef_index[i * vd + j]) {
566  if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
567  subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
568  } else {
569  result = 1;
570  subband_coef_sign[i * vd + j] = 0;
571  }
572  } else {
573  subband_coef_sign[i * vd + j] = 0;
574  }
575  }
576  }
577  return result;
578 }
579 
580 
581 /**
582  * Fill the mlt_buffer with mlt coefficients.
583  *
584  * @param q pointer to the COOKContext
585  * @param category pointer to the category array
586  * @param quant_index_table pointer to the array
587  * @param mlt_buffer pointer to mlt coefficients
588  */
590  int *quant_index_table, float *mlt_buffer)
591 {
592  /* A zero in this table means that the subband coefficient is
593  random noise coded. */
594  int subband_coef_index[SUBBAND_SIZE];
595  /* A zero in this table means that the subband coefficient is a
596  positive multiplicator. */
597  int subband_coef_sign[SUBBAND_SIZE];
598  int band, j;
599  int index = 0;
600 
601  for (band = 0; band < p->total_subbands; band++) {
602  index = category[band];
603  if (category[band] < 7) {
604  if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
605  index = 7;
606  for (j = 0; j < p->total_subbands; j++)
607  category[band + j] = 7;
608  }
609  }
610  if (index >= 7) {
611  memset(subband_coef_index, 0, sizeof(subband_coef_index));
612  memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
613  }
614  q->scalar_dequant(q, index, quant_index_table[band],
615  subband_coef_index, subband_coef_sign,
616  &mlt_buffer[band * SUBBAND_SIZE]);
617  }
618 
619  /* FIXME: should this be removed, or moved into loop above? */
621  return;
622 }
623 
624 
625 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
626 {
627  int category_index[128] = { 0 };
628  int category[128] = { 0 };
629  int quant_index_table[102];
630  int res, i;
631 
632  if ((res = decode_envelope(q, p, quant_index_table)) < 0)
633  return res;
635  categorize(q, p, quant_index_table, category, category_index);
636  expand_category(q, category, category_index);
637  for (i=0; i<p->total_subbands; i++) {
638  if (category[i] > 7)
639  return AVERROR_INVALIDDATA;
640  }
641  decode_vectors(q, p, category, quant_index_table, mlt_buffer);
642 
643  return 0;
644 }
645 
646 
647 /**
648  * the actual requantization of the timedomain samples
649  *
650  * @param q pointer to the COOKContext
651  * @param buffer pointer to the timedomain buffer
652  * @param gain_index index for the block multiplier
653  * @param gain_index_next index for the next block multiplier
654  */
655 static void interpolate_float(COOKContext *q, float *buffer,
656  int gain_index, int gain_index_next)
657 {
658  int i;
659  float fc1, fc2;
660  fc1 = pow2tab[gain_index + 63];
661 
662  if (gain_index == gain_index_next) { // static gain
663  for (i = 0; i < q->gain_size_factor; i++)
664  buffer[i] *= fc1;
665  } else { // smooth gain
666  fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
667  for (i = 0; i < q->gain_size_factor; i++) {
668  buffer[i] *= fc1;
669  fc1 *= fc2;
670  }
671  }
672 }
673 
674 /**
675  * Apply transform window, overlap buffers.
676  *
677  * @param q pointer to the COOKContext
678  * @param inbuffer pointer to the mltcoefficients
679  * @param gains_ptr current and previous gains
680  * @param previous_buffer pointer to the previous buffer to be used for overlapping
681  */
682 static void imlt_window_float(COOKContext *q, float *inbuffer,
683  cook_gains *gains_ptr, float *previous_buffer)
684 {
685  const float fc = pow2tab[gains_ptr->previous[0] + 63];
686  int i;
687  /* The weird thing here, is that the two halves of the time domain
688  * buffer are swapped. Also, the newest data, that we save away for
689  * next frame, has the wrong sign. Hence the subtraction below.
690  * Almost sounds like a complex conjugate/reverse data/FFT effect.
691  */
692 
693  /* Apply window and overlap */
694  for (i = 0; i < q->samples_per_channel; i++)
695  inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
696  previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
697 }
698 
699 /**
700  * The modulated lapped transform, this takes transform coefficients
701  * and transforms them into timedomain samples.
702  * Apply transform window, overlap buffers, apply gain profile
703  * and buffer management.
704  *
705  * @param q pointer to the COOKContext
706  * @param inbuffer pointer to the mltcoefficients
707  * @param gains_ptr current and previous gains
708  * @param previous_buffer pointer to the previous buffer to be used for overlapping
709  */
710 static void imlt_gain(COOKContext *q, float *inbuffer,
711  cook_gains *gains_ptr, float *previous_buffer)
712 {
713  float *buffer0 = q->mono_mdct_output;
714  float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
715  int i;
716 
717  /* Inverse modified discrete cosine transform */
718  q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
719 
720  q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
721 
722  /* Apply gain profile */
723  for (i = 0; i < 8; i++)
724  if (gains_ptr->now[i] || gains_ptr->now[i + 1])
725  q->interpolate(q, &buffer1[q->gain_size_factor * i],
726  gains_ptr->now[i], gains_ptr->now[i + 1]);
727 
728  /* Save away the current to be previous block. */
729  memcpy(previous_buffer, buffer0,
730  q->samples_per_channel * sizeof(*previous_buffer));
731 }
732 
733 
734 /**
735  * function for getting the jointstereo coupling information
736  *
737  * @param q pointer to the COOKContext
738  * @param decouple_tab decoupling array
739  */
740 static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
741 {
742  int i;
743  int vlc = get_bits1(&q->gb);
744  int start = cplband[p->js_subband_start];
745  int end = cplband[p->subbands - 1];
746  int length = end - start + 1;
747 
748  if (start > end)
749  return 0;
750 
751  if (vlc)
752  for (i = 0; i < length; i++)
753  decouple_tab[start + i] = get_vlc2(&q->gb,
755  p->channel_coupling.bits, 2);
756  else
757  for (i = 0; i < length; i++) {
758  int v = get_bits(&q->gb, p->js_vlc_bits);
759  if (v == (1<<p->js_vlc_bits)-1) {
760  av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
761  return AVERROR_INVALIDDATA;
762  }
763  decouple_tab[start + i] = v;
764  }
765  return 0;
766 }
767 
768 /**
769  * function decouples a pair of signals from a single signal via multiplication.
770  *
771  * @param q pointer to the COOKContext
772  * @param subband index of the current subband
773  * @param f1 multiplier for channel 1 extraction
774  * @param f2 multiplier for channel 2 extraction
775  * @param decode_buffer input buffer
776  * @param mlt_buffer1 pointer to left channel mlt coefficients
777  * @param mlt_buffer2 pointer to right channel mlt coefficients
778  */
780  COOKSubpacket *p,
781  int subband,
782  float f1, float f2,
783  float *decode_buffer,
784  float *mlt_buffer1, float *mlt_buffer2)
785 {
786  int j, tmp_idx;
787  for (j = 0; j < SUBBAND_SIZE; j++) {
788  tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
789  mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
790  mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
791  }
792 }
793 
794 /**
795  * function for decoding joint stereo data
796  *
797  * @param q pointer to the COOKContext
798  * @param mlt_buffer1 pointer to left channel mlt coefficients
799  * @param mlt_buffer2 pointer to right channel mlt coefficients
800  */
802  float *mlt_buffer_left, float *mlt_buffer_right)
803 {
804  int i, j, res;
805  int decouple_tab[SUBBAND_SIZE] = { 0 };
806  float *decode_buffer = q->decode_buffer_0;
807  int idx, cpl_tmp;
808  float f1, f2;
809  const float *cplscale;
810 
811  memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
812 
813  /* Make sure the buffers are zeroed out. */
814  memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
815  memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
816  if ((res = decouple_info(q, p, decouple_tab)) < 0)
817  return res;
818  if ((res = mono_decode(q, p, decode_buffer)) < 0)
819  return res;
820  /* The two channels are stored interleaved in decode_buffer. */
821  for (i = 0; i < p->js_subband_start; i++) {
822  for (j = 0; j < SUBBAND_SIZE; j++) {
823  mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
824  mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
825  }
826  }
827 
828  /* When we reach js_subband_start (the higher frequencies)
829  the coefficients are stored in a coupling scheme. */
830  idx = (1 << p->js_vlc_bits) - 1;
831  for (i = p->js_subband_start; i < p->subbands; i++) {
832  cpl_tmp = cplband[i];
833  idx -= decouple_tab[cpl_tmp];
834  cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
835  f1 = cplscale[decouple_tab[cpl_tmp] + 1];
836  f2 = cplscale[idx];
837  q->decouple(q, p, i, f1, f2, decode_buffer,
838  mlt_buffer_left, mlt_buffer_right);
839  idx = (1 << p->js_vlc_bits) - 1;
840  }
841 
842  return 0;
843 }
844 
845 /**
846  * First part of subpacket decoding:
847  * decode raw stream bytes and read gain info.
848  *
849  * @param q pointer to the COOKContext
850  * @param inbuffer pointer to raw stream data
851  * @param gains_ptr array of current/prev gain pointers
852  */
854  const uint8_t *inbuffer,
855  cook_gains *gains_ptr)
856 {
857  int offset;
858 
859  offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
860  p->bits_per_subpacket / 8);
861  init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
862  p->bits_per_subpacket);
863  decode_gain_info(&q->gb, gains_ptr->now);
864 
865  /* Swap current and previous gains */
866  FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
867 }
868 
869 /**
870  * Saturate the output signal and interleave.
871  *
872  * @param q pointer to the COOKContext
873  * @param out pointer to the output vector
874  */
875 static void saturate_output_float(COOKContext *q, float *out)
876 {
878  -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
879 }
880 
881 
882 /**
883  * Final part of subpacket decoding:
884  * Apply modulated lapped transform, gain compensation,
885  * clip and convert to integer.
886  *
887  * @param q pointer to the COOKContext
888  * @param decode_buffer pointer to the mlt coefficients
889  * @param gains_ptr array of current/prev gain pointers
890  * @param previous_buffer pointer to the previous buffer to be used for overlapping
891  * @param out pointer to the output buffer
892  */
893 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
894  cook_gains *gains_ptr, float *previous_buffer,
895  float *out)
896 {
897  imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
898  if (out)
899  q->saturate_output(q, out);
900 }
901 
902 
903 /**
904  * Cook subpacket decoding. This function returns one decoded subpacket,
905  * usually 1024 samples per channel.
906  *
907  * @param q pointer to the COOKContext
908  * @param inbuffer pointer to the inbuffer
909  * @param outbuffer pointer to the outbuffer
910  */
912  const uint8_t *inbuffer, float **outbuffer)
913 {
914  int sub_packet_size = p->size;
915  int res;
916 
917  memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
918  decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
919 
920  if (p->joint_stereo) {
921  if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
922  return res;
923  } else {
924  if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
925  return res;
926 
927  if (p->num_channels == 2) {
928  decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
929  if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
930  return res;
931  }
932  }
933 
936  outbuffer ? outbuffer[p->ch_idx] : NULL);
937 
938  if (p->num_channels == 2) {
939  if (p->joint_stereo)
942  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
943  else
946  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
947  }
948 
949  return 0;
950 }
951 
952 
953 static int cook_decode_frame(AVCodecContext *avctx, void *data,
954  int *got_frame_ptr, AVPacket *avpkt)
955 {
956  AVFrame *frame = data;
957  const uint8_t *buf = avpkt->data;
958  int buf_size = avpkt->size;
959  COOKContext *q = avctx->priv_data;
960  float **samples = NULL;
961  int i, ret;
962  int offset = 0;
963  int chidx = 0;
964 
965  if (buf_size < avctx->block_align)
966  return buf_size;
967 
968  /* get output buffer */
969  if (q->discarded_packets >= 2) {
970  frame->nb_samples = q->samples_per_channel;
971  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
972  return ret;
973  samples = (float **)frame->extended_data;
974  }
975 
976  /* estimate subpacket sizes */
977  q->subpacket[0].size = avctx->block_align;
978 
979  for (i = 1; i < q->num_subpackets; i++) {
980  q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
981  q->subpacket[0].size -= q->subpacket[i].size + 1;
982  if (q->subpacket[0].size < 0) {
983  av_log(avctx, AV_LOG_DEBUG,
984  "frame subpacket size total > avctx->block_align!\n");
985  return AVERROR_INVALIDDATA;
986  }
987  }
988 
989  /* decode supbackets */
990  for (i = 0; i < q->num_subpackets; i++) {
991  q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
993  q->subpacket[i].ch_idx = chidx;
994  av_log(avctx, AV_LOG_DEBUG,
995  "subpacket[%i] size %i js %i %i block_align %i\n",
996  i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
997  avctx->block_align);
998 
999  if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1000  return ret;
1001  offset += q->subpacket[i].size;
1002  chidx += q->subpacket[i].num_channels;
1003  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1004  i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1005  }
1006 
1007  /* Discard the first two frames: no valid audio. */
1008  if (q->discarded_packets < 2) {
1009  q->discarded_packets++;
1010  *got_frame_ptr = 0;
1011  return avctx->block_align;
1012  }
1013 
1014  *got_frame_ptr = 1;
1015 
1016  return avctx->block_align;
1017 }
1018 
1019 #ifdef DEBUG
1020 static void dump_cook_context(COOKContext *q)
1021 {
1022  //int i=0;
1023 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
1024  av_dlog(q->avctx, "COOKextradata\n");
1025  av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1026  if (q->subpacket[0].cookversion > STEREO) {
1027  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1028  PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1029  }
1030  av_dlog(q->avctx, "COOKContext\n");
1031  PRINT("nb_channels", q->avctx->channels);
1032  PRINT("bit_rate", q->avctx->bit_rate);
1033  PRINT("sample_rate", q->avctx->sample_rate);
1034  PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1035  PRINT("subbands", q->subpacket[0].subbands);
1036  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1037  PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1038  PRINT("numvector_size", q->subpacket[0].numvector_size);
1039  PRINT("total_subbands", q->subpacket[0].total_subbands);
1040 }
1041 #endif
1042 
1043 /**
1044  * Cook initialization
1045  *
1046  * @param avctx pointer to the AVCodecContext
1047  */
1049 {
1050  COOKContext *q = avctx->priv_data;
1051  const uint8_t *edata_ptr = avctx->extradata;
1052  const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
1053  int extradata_size = avctx->extradata_size;
1054  int s = 0;
1055  unsigned int channel_mask = 0;
1056  int samples_per_frame = 0;
1057  int ret;
1058  q->avctx = avctx;
1059 
1060  /* Take care of the codec specific extradata. */
1061  if (extradata_size <= 0) {
1062  av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1063  return AVERROR_INVALIDDATA;
1064  }
1065  av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1066 
1067  /* Take data from the AVCodecContext (RM container). */
1068  if (!avctx->channels) {
1069  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1070  return AVERROR_INVALIDDATA;
1071  }
1072 
1073  /* Initialize RNG. */
1074  av_lfg_init(&q->random_state, 0);
1075 
1076  ff_audiodsp_init(&q->adsp);
1077 
1078  while (edata_ptr < edata_ptr_end) {
1079  /* 8 for mono, 16 for stereo, ? for multichannel
1080  Swap to right endianness so we don't need to care later on. */
1081  if (extradata_size >= 8) {
1082  q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
1083  samples_per_frame = bytestream_get_be16(&edata_ptr);
1084  q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
1085  extradata_size -= 8;
1086  }
1087  if (extradata_size >= 8) {
1088  bytestream_get_be32(&edata_ptr); // Unknown unused
1089  q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
1090  if (q->subpacket[s].js_subband_start >= 51) {
1091  av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
1092  return AVERROR_INVALIDDATA;
1093  }
1094 
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. */
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  avpriv_request_sample(avctx, "Container channels != 1");
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  avpriv_request_sample(avctx, "Container channels != 2");
1131  return AVERROR_PATCHWELCOME;
1132  }
1133  av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1134  if (avctx->extradata_size >= 16) {
1137  q->subpacket[s].joint_stereo = 1;
1138  q->subpacket[s].num_channels = 2;
1139  }
1140  if (q->subpacket[s].samples_per_channel > 256) {
1142  }
1143  if (q->subpacket[s].samples_per_channel > 512) {
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 
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) {
1161  }
1162  if (q->subpacket[s].samples_per_channel > 512) {
1164  }
1165  } else
1166  q->subpacket[s].samples_per_channel = samples_per_frame;
1167 
1168  break;
1169  default:
1170  avpriv_request_sample(avctx, "Cook version %d",
1171  q->subpacket[s].cookversion);
1172  return AVERROR_PATCHWELCOME;
1173  }
1174 
1175  if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1176  av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1177  return AVERROR_INVALIDDATA;
1178  } else
1180 
1181 
1182  /* Initialize variable relations */
1184 
1185  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1186  if (q->subpacket[s].total_subbands > 53) {
1187  avpriv_request_sample(avctx, "total_subbands > 53");
1188  return AVERROR_PATCHWELCOME;
1189  }
1190 
1191  if ((q->subpacket[s].js_vlc_bits > 6) ||
1192  (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1193  av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1194  q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1195  return AVERROR_INVALIDDATA;
1196  }
1197 
1198  if (q->subpacket[s].subbands > 50) {
1199  avpriv_request_sample(avctx, "subbands > 50");
1200  return AVERROR_PATCHWELCOME;
1201  }
1202  if (q->subpacket[s].subbands == 0) {
1203  avpriv_request_sample(avctx, "subbands = 0");
1204  return AVERROR_PATCHWELCOME;
1205  }
1206  q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1208  q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1210 
1211  if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
1212  av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
1213  return AVERROR_INVALIDDATA;
1214  }
1215 
1216  q->num_subpackets++;
1217  s++;
1218  if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
1219  avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1220  return AVERROR_PATCHWELCOME;
1221  }
1222  }
1223  /* Generate tables */
1224  init_pow2table();
1225  init_gain_table(q);
1227 
1228  if ((ret = init_cook_vlc_tables(q)))
1229  return ret;
1230 
1231 
1232  if (avctx->block_align >= UINT_MAX / 2)
1233  return AVERROR(EINVAL);
1234 
1235  /* Pad the databuffer with:
1236  DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1237  FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1239  av_mallocz(avctx->block_align
1240  + DECODE_BYTES_PAD1(avctx->block_align)
1242  if (!q->decoded_bytes_buffer)
1243  return AVERROR(ENOMEM);
1244 
1245  /* Initialize transform. */
1246  if ((ret = init_cook_mlt(q)))
1247  return ret;
1248 
1249  /* Initialize COOK signal arithmetic handling */
1250  if (1) {
1252  q->decouple = decouple_float;
1256  }
1257 
1258  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1259  if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1260  q->samples_per_channel != 1024) {
1261  avpriv_request_sample(avctx, "samples_per_channel = %d",
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 
1272 #ifdef DEBUG
1273  dump_cook_context(q);
1274 #endif
1275  return 0;
1276 }
1277 
1279  .name = "cook",
1280  .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1281  .type = AVMEDIA_TYPE_AUDIO,
1282  .id = AV_CODEC_ID_COOK,
1283  .priv_data_size = sizeof(COOKContext),
1287  .capabilities = CODEC_CAP_DR1,
1288  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1290 };