Libav
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 Libav.
7  *
8  * Libav is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * Libav is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with Libav; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
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 
127  /* stream data */
130  /* states */
133 
134  /* transform data */
136  float* mlt_window;
137 
138  /* VLC data */
139  VLC envelope_quant_index[13];
140  VLC sqvh[7]; // scalar quantization
141 
142  /* generatable tables and related variables */
144  float gain_table[23];
145 
146  /* data buffers */
147 
149  DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
150  float decode_buffer_1[1024];
151  float decode_buffer_2[1024];
152  float decode_buffer_0[1060]; /* static allocation for joint decode */
153 
154  const float *cplscales[5];
157 } COOKContext;
158 
159 static float pow2tab[127];
160 static float rootpow2tab[127];
161 
162 /*************** init functions ***************/
163 
164 /* table generator */
165 static av_cold void init_pow2table(void)
166 {
167  int i;
168  for (i = -63; i < 64; i++) {
169  pow2tab[63 + i] = pow(2, i);
170  rootpow2tab[63 + i] = sqrt(pow(2, i));
171  }
172 }
173 
174 /* table generator */
176 {
177  int i;
179  for (i = 0; i < 23; i++)
180  q->gain_table[i] = pow(pow2tab[i + 52],
181  (1.0 / (double) q->gain_size_factor));
182 }
183 
184 
186 {
187  int i, result;
188 
189  result = 0;
190  for (i = 0; i < 13; i++) {
191  result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
193  envelope_quant_index_huffcodes[i], 2, 2, 0);
194  }
195  av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
196  for (i = 0; i < 7; i++) {
197  result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
198  cvh_huffbits[i], 1, 1,
199  cvh_huffcodes[i], 2, 2, 0);
200  }
201 
202  for (i = 0; i < q->num_subpackets; i++) {
203  if (q->subpacket[i].joint_stereo == 1) {
204  result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
205  (1 << q->subpacket[i].js_vlc_bits) - 1,
206  ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
207  ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
208  av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
209  }
210  }
211 
212  av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
213  return result;
214 }
215 
217 {
218  int j, ret;
219  int mlt_size = q->samples_per_channel;
220 
221  if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
222  return AVERROR(ENOMEM);
223 
224  /* Initialize the MLT window: simple sine window. */
225  ff_sine_window_init(q->mlt_window, mlt_size);
226  for (j = 0; j < mlt_size; j++)
227  q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
228 
229  /* Initialize the MDCT. */
230  if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
231  av_free(q->mlt_window);
232  return ret;
233  }
234  av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
235  av_log2(mlt_size) + 1);
236 
237  return 0;
238 }
239 
241 {
242  int i;
243  for (i = 0; i < 5; i++)
244  q->cplscales[i] = cplscales[i];
245 }
246 
247 /*************** init functions end ***********/
248 
249 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
250 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
251 
272 static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
273 {
274  static const uint32_t tab[4] = {
275  AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
276  AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
277  };
278  int i, off;
279  uint32_t c;
280  const uint32_t *buf;
281  uint32_t *obuf = (uint32_t *) out;
282  /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
283  * I'm too lazy though, should be something like
284  * for (i = 0; i < bitamount / 64; i++)
285  * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
286  * Buffer alignment needs to be checked. */
287 
288  off = (intptr_t) inbuffer & 3;
289  buf = (const uint32_t *) (inbuffer - off);
290  c = tab[off];
291  bytes += 3 + off;
292  for (i = 0; i < bytes / 4; i++)
293  obuf[i] = c ^ buf[i];
294 
295  return off;
296 }
297 
299 {
300  int i;
301  COOKContext *q = avctx->priv_data;
302  av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
303 
304  /* Free allocated memory buffers. */
305  av_free(q->mlt_window);
307 
308  /* Free the transform. */
309  ff_mdct_end(&q->mdct_ctx);
310 
311  /* Free the VLC tables. */
312  for (i = 0; i < 13; i++)
314  for (i = 0; i < 7; i++)
315  ff_free_vlc(&q->sqvh[i]);
316  for (i = 0; i < q->num_subpackets; i++)
318 
319  av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
320 
321  return 0;
322 }
323 
330 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
331 {
332  int i, n;
333 
334  while (get_bits1(gb)) {
335  /* NOTHING */
336  }
337 
338  n = get_bits_count(gb) - 1; // amount of elements*2 to update
339 
340  i = 0;
341  while (n--) {
342  int index = get_bits(gb, 3);
343  int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
344 
345  while (i <= index)
346  gaininfo[i++] = gain;
347  }
348  while (i <= 8)
349  gaininfo[i++] = 0;
350 }
351 
359  int *quant_index_table)
360 {
361  int i, j, vlc_index;
362 
363  quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
364 
365  for (i = 1; i < p->total_subbands; i++) {
366  vlc_index = i;
367  if (i >= p->js_subband_start * 2) {
368  vlc_index -= p->js_subband_start;
369  } else {
370  vlc_index /= 2;
371  if (vlc_index < 1)
372  vlc_index = 1;
373  }
374  if (vlc_index > 13)
375  vlc_index = 13; // the VLC tables >13 are identical to No. 13
376 
377  j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
378  q->envelope_quant_index[vlc_index - 1].bits, 2);
379  quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
380  if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
382  "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
383  quant_index_table[i], i);
384  return AVERROR_INVALIDDATA;
385  }
386  }
387 
388  return 0;
389 }
390 
399 static void categorize(COOKContext *q, COOKSubpacket *p, int *quant_index_table,
400  int *category, int *category_index)
401 {
402  int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
403  int exp_index2[102] = { 0 };
404  int exp_index1[102] = { 0 };
405 
406  int tmp_categorize_array[128 * 2] = { 0 };
407  int tmp_categorize_array1_idx = p->numvector_size;
408  int tmp_categorize_array2_idx = p->numvector_size;
409 
410  bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
411 
412  if (bits_left > q->samples_per_channel)
413  bits_left = q->samples_per_channel +
414  ((bits_left - q->samples_per_channel) * 5) / 8;
415 
416  bias = -32;
417 
418  /* Estimate bias. */
419  for (i = 32; i > 0; i = i / 2) {
420  num_bits = 0;
421  index = 0;
422  for (j = p->total_subbands; j > 0; j--) {
423  exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
424  index++;
425  num_bits += expbits_tab[exp_idx];
426  }
427  if (num_bits >= bits_left - 32)
428  bias += i;
429  }
430 
431  /* Calculate total number of bits. */
432  num_bits = 0;
433  for (i = 0; i < p->total_subbands; i++) {
434  exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
435  num_bits += expbits_tab[exp_idx];
436  exp_index1[i] = exp_idx;
437  exp_index2[i] = exp_idx;
438  }
439  tmpbias1 = tmpbias2 = num_bits;
440 
441  for (j = 1; j < p->numvector_size; j++) {
442  if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */
443  int max = -999999;
444  index = -1;
445  for (i = 0; i < p->total_subbands; i++) {
446  if (exp_index1[i] < 7) {
447  v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
448  if (v >= max) {
449  max = v;
450  index = i;
451  }
452  }
453  }
454  if (index == -1)
455  break;
456  tmp_categorize_array[tmp_categorize_array1_idx++] = index;
457  tmpbias1 -= expbits_tab[exp_index1[index]] -
458  expbits_tab[exp_index1[index] + 1];
459  ++exp_index1[index];
460  } else { /* <--- */
461  int min = 999999;
462  index = -1;
463  for (i = 0; i < p->total_subbands; i++) {
464  if (exp_index2[i] > 0) {
465  v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
466  if (v < min) {
467  min = v;
468  index = i;
469  }
470  }
471  }
472  if (index == -1)
473  break;
474  tmp_categorize_array[--tmp_categorize_array2_idx] = index;
475  tmpbias2 -= expbits_tab[exp_index2[index]] -
476  expbits_tab[exp_index2[index] - 1];
477  --exp_index2[index];
478  }
479  }
480 
481  for (i = 0; i < p->total_subbands; i++)
482  category[i] = exp_index2[i];
483 
484  for (i = 0; i < p->numvector_size - 1; i++)
485  category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
486 }
487 
488 
496 static inline void expand_category(COOKContext *q, int *category,
497  int *category_index)
498 {
499  int i;
500  for (i = 0; i < q->num_vectors; i++)
501  {
502  int idx = category_index[i];
503  if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
504  --category[idx];
505  }
506 }
507 
518 static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
519  int *subband_coef_index, int *subband_coef_sign,
520  float *mlt_p)
521 {
522  int i;
523  float f1;
524 
525  for (i = 0; i < SUBBAND_SIZE; i++) {
526  if (subband_coef_index[i]) {
527  f1 = quant_centroid_tab[index][subband_coef_index[i]];
528  if (subband_coef_sign[i])
529  f1 = -f1;
530  } else {
531  /* noise coding if subband_coef_index[i] == 0 */
532  f1 = dither_tab[index];
533  if (av_lfg_get(&q->random_state) < 0x80000000)
534  f1 = -f1;
535  }
536  mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
537  }
538 }
547 static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
548  int *subband_coef_index, int *subband_coef_sign)
549 {
550  int i, j;
551  int vlc, vd, tmp, result;
552 
553  vd = vd_tab[category];
554  result = 0;
555  for (i = 0; i < vpr_tab[category]; i++) {
556  vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
557  if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
558  vlc = 0;
559  result = 1;
560  }
561  for (j = vd - 1; j >= 0; j--) {
562  tmp = (vlc * invradix_tab[category]) / 0x100000;
563  subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
564  vlc = tmp;
565  }
566  for (j = 0; j < vd; j++) {
567  if (subband_coef_index[i * vd + j]) {
568  if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
569  subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
570  } else {
571  result = 1;
572  subband_coef_sign[i * vd + j] = 0;
573  }
574  } else {
575  subband_coef_sign[i * vd + j] = 0;
576  }
577  }
578  }
579  return result;
580 }
581 
582 
591 static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
592  int *quant_index_table, float *mlt_buffer)
593 {
594  /* A zero in this table means that the subband coefficient is
595  random noise coded. */
596  int subband_coef_index[SUBBAND_SIZE];
597  /* A zero in this table means that the subband coefficient is a
598  positive multiplicator. */
599  int subband_coef_sign[SUBBAND_SIZE];
600  int band, j;
601  int index = 0;
602 
603  for (band = 0; band < p->total_subbands; band++) {
604  index = category[band];
605  if (category[band] < 7) {
606  if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
607  index = 7;
608  for (j = 0; j < p->total_subbands; j++)
609  category[band + j] = 7;
610  }
611  }
612  if (index >= 7) {
613  memset(subband_coef_index, 0, sizeof(subband_coef_index));
614  memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
615  }
616  q->scalar_dequant(q, index, quant_index_table[band],
617  subband_coef_index, subband_coef_sign,
618  &mlt_buffer[band * SUBBAND_SIZE]);
619  }
620 
621  /* FIXME: should this be removed, or moved into loop above? */
623  return;
624 }
625 
626 
627 static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
628 {
629  int category_index[128] = { 0 };
630  int category[128] = { 0 };
631  int quant_index_table[102];
632  int res;
633 
634  if ((res = decode_envelope(q, p, quant_index_table)) < 0)
635  return res;
637  categorize(q, p, quant_index_table, category, category_index);
638  expand_category(q, category, category_index);
639  decode_vectors(q, p, category, quant_index_table, mlt_buffer);
640 
641  return 0;
642 }
643 
644 
653 static void interpolate_float(COOKContext *q, float *buffer,
654  int gain_index, int gain_index_next)
655 {
656  int i;
657  float fc1, fc2;
658  fc1 = pow2tab[gain_index + 63];
659 
660  if (gain_index == gain_index_next) { // static gain
661  for (i = 0; i < q->gain_size_factor; i++)
662  buffer[i] *= fc1;
663  } else { // smooth gain
664  fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
665  for (i = 0; i < q->gain_size_factor; i++) {
666  buffer[i] *= fc1;
667  fc1 *= fc2;
668  }
669  }
670 }
671 
680 static void imlt_window_float(COOKContext *q, float *inbuffer,
681  cook_gains *gains_ptr, float *previous_buffer)
682 {
683  const float fc = pow2tab[gains_ptr->previous[0] + 63];
684  int i;
685  /* The weird thing here, is that the two halves of the time domain
686  * buffer are swapped. Also, the newest data, that we save away for
687  * next frame, has the wrong sign. Hence the subtraction below.
688  * Almost sounds like a complex conjugate/reverse data/FFT effect.
689  */
690 
691  /* Apply window and overlap */
692  for (i = 0; i < q->samples_per_channel; i++)
693  inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
694  previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
695 }
696 
708 static void imlt_gain(COOKContext *q, float *inbuffer,
709  cook_gains *gains_ptr, float *previous_buffer)
710 {
711  float *buffer0 = q->mono_mdct_output;
712  float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
713  int i;
714 
715  /* Inverse modified discrete cosine transform */
716  q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
717 
718  q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
719 
720  /* Apply gain profile */
721  for (i = 0; i < 8; i++)
722  if (gains_ptr->now[i] || gains_ptr->now[i + 1])
723  q->interpolate(q, &buffer1[q->gain_size_factor * i],
724  gains_ptr->now[i], gains_ptr->now[i + 1]);
725 
726  /* Save away the current to be previous block. */
727  memcpy(previous_buffer, buffer0,
728  q->samples_per_channel * sizeof(*previous_buffer));
729 }
730 
731 
738 static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
739 {
740  int i;
741  int vlc = get_bits1(&q->gb);
742  int start = cplband[p->js_subband_start];
743  int end = cplband[p->subbands - 1];
744  int length = end - start + 1;
745 
746  if (start > end)
747  return;
748 
749  if (vlc)
750  for (i = 0; i < length; i++)
751  decouple_tab[start + i] = get_vlc2(&q->gb,
753  p->channel_coupling.bits, 2);
754  else
755  for (i = 0; i < length; i++)
756  decouple_tab[start + i] = get_bits(&q->gb, p->js_vlc_bits);
757 }
758 
759 /*
760  * function decouples a pair of signals from a single signal via multiplication.
761  *
762  * @param q pointer to the COOKContext
763  * @param subband index of the current subband
764  * @param f1 multiplier for channel 1 extraction
765  * @param f2 multiplier for channel 2 extraction
766  * @param decode_buffer input buffer
767  * @param mlt_buffer1 pointer to left channel mlt coefficients
768  * @param mlt_buffer2 pointer to right channel mlt coefficients
769  */
771  COOKSubpacket *p,
772  int subband,
773  float f1, float f2,
774  float *decode_buffer,
775  float *mlt_buffer1, float *mlt_buffer2)
776 {
777  int j, tmp_idx;
778  for (j = 0; j < SUBBAND_SIZE; j++) {
779  tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
780  mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
781  mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
782  }
783 }
784 
793  float *mlt_buffer_left, float *mlt_buffer_right)
794 {
795  int i, j, res;
796  int decouple_tab[SUBBAND_SIZE] = { 0 };
797  float *decode_buffer = q->decode_buffer_0;
798  int idx, cpl_tmp;
799  float f1, f2;
800  const float *cplscale;
801 
802  memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
803 
804  /* Make sure the buffers are zeroed out. */
805  memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
806  memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
807  decouple_info(q, p, decouple_tab);
808  if ((res = mono_decode(q, p, decode_buffer)) < 0)
809  return res;
810 
811  /* The two channels are stored interleaved in decode_buffer. */
812  for (i = 0; i < p->js_subband_start; i++) {
813  for (j = 0; j < SUBBAND_SIZE; j++) {
814  mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
815  mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
816  }
817  }
818 
819  /* When we reach js_subband_start (the higher frequencies)
820  the coefficients are stored in a coupling scheme. */
821  idx = (1 << p->js_vlc_bits) - 1;
822  for (i = p->js_subband_start; i < p->subbands; i++) {
823  cpl_tmp = cplband[i];
824  idx -= decouple_tab[cpl_tmp];
825  cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
826  f1 = cplscale[decouple_tab[cpl_tmp] + 1];
827  f2 = cplscale[idx];
828  q->decouple(q, p, i, f1, f2, decode_buffer,
829  mlt_buffer_left, mlt_buffer_right);
830  idx = (1 << p->js_vlc_bits) - 1;
831  }
832 
833  return 0;
834 }
835 
845  const uint8_t *inbuffer,
846  cook_gains *gains_ptr)
847 {
848  int offset;
849 
850  offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
851  p->bits_per_subpacket / 8);
852  init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
853  p->bits_per_subpacket);
854  decode_gain_info(&q->gb, gains_ptr->now);
855 
856  /* Swap current and previous gains */
857  FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
858 }
859 
866 static void saturate_output_float(COOKContext *q, float *out)
867 {
869  -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
870 }
871 
872 
884 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
885  cook_gains *gains_ptr, float *previous_buffer,
886  float *out)
887 {
888  imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
889  if (out)
890  q->saturate_output(q, out);
891 }
892 
893 
903  const uint8_t *inbuffer, float **outbuffer)
904 {
905  int sub_packet_size = p->size;
906  int res;
907 
908  memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
909  decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
910 
911  if (p->joint_stereo) {
912  if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
913  return res;
914  } else {
915  if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
916  return res;
917 
918  if (p->num_channels == 2) {
919  decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
920  if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
921  return res;
922  }
923  }
924 
927  outbuffer ? outbuffer[p->ch_idx] : NULL);
928 
929  if (p->num_channels == 2)
930  if (p->joint_stereo)
933  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
934  else
937  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
938 
939  return 0;
940 }
941 
942 
943 static int cook_decode_frame(AVCodecContext *avctx, void *data,
944  int *got_frame_ptr, AVPacket *avpkt)
945 {
946  AVFrame *frame = data;
947  const uint8_t *buf = avpkt->data;
948  int buf_size = avpkt->size;
949  COOKContext *q = avctx->priv_data;
950  float **samples = NULL;
951  int i, ret;
952  int offset = 0;
953  int chidx = 0;
954 
955  if (buf_size < avctx->block_align)
956  return buf_size;
957 
958  /* get output buffer */
959  if (q->discarded_packets >= 2) {
960  frame->nb_samples = q->samples_per_channel;
961  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
962  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
963  return ret;
964  }
965  samples = (float **)frame->extended_data;
966  }
967 
968  /* estimate subpacket sizes */
969  q->subpacket[0].size = avctx->block_align;
970 
971  for (i = 1; i < q->num_subpackets; i++) {
972  q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
973  q->subpacket[0].size -= q->subpacket[i].size + 1;
974  if (q->subpacket[0].size < 0) {
975  av_log(avctx, AV_LOG_DEBUG,
976  "frame subpacket size total > avctx->block_align!\n");
977  return AVERROR_INVALIDDATA;
978  }
979  }
980 
981  /* decode supbackets */
982  for (i = 0; i < q->num_subpackets; i++) {
983  q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
985  q->subpacket[i].ch_idx = chidx;
986  av_log(avctx, AV_LOG_DEBUG,
987  "subpacket[%i] size %i js %i %i block_align %i\n",
988  i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
989  avctx->block_align);
990 
991  if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
992  return ret;
993  offset += q->subpacket[i].size;
994  chidx += q->subpacket[i].num_channels;
995  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
996  i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
997  }
998 
999  /* Discard the first two frames: no valid audio. */
1000  if (q->discarded_packets < 2) {
1001  q->discarded_packets++;
1002  *got_frame_ptr = 0;
1003  return avctx->block_align;
1004  }
1005 
1006  *got_frame_ptr = 1;
1007 
1008  return avctx->block_align;
1009 }
1010 
1011 #ifdef DEBUG
1012 static void dump_cook_context(COOKContext *q)
1013 {
1014  //int i=0;
1015 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
1016  av_dlog(q->avctx, "COOKextradata\n");
1017  av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1018  if (q->subpacket[0].cookversion > STEREO) {
1019  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1020  PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1021  }
1022  av_dlog(q->avctx, "COOKContext\n");
1023  PRINT("nb_channels", q->avctx->channels);
1024  PRINT("bit_rate", q->avctx->bit_rate);
1025  PRINT("sample_rate", q->avctx->sample_rate);
1026  PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1027  PRINT("subbands", q->subpacket[0].subbands);
1028  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1029  PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1030  PRINT("numvector_size", q->subpacket[0].numvector_size);
1031  PRINT("total_subbands", q->subpacket[0].total_subbands);
1032 }
1033 #endif
1034 
1041 {
1042  COOKContext *q = avctx->priv_data;
1043  const uint8_t *edata_ptr = avctx->extradata;
1044  const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
1045  int extradata_size = avctx->extradata_size;
1046  int s = 0;
1047  unsigned int channel_mask = 0;
1048  int samples_per_frame;
1049  int ret;
1050  q->avctx = avctx;
1051 
1052  /* Take care of the codec specific extradata. */
1053  if (extradata_size <= 0) {
1054  av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1055  return AVERROR_INVALIDDATA;
1056  }
1057  av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1058 
1059  /* Take data from the AVCodecContext (RM container). */
1060  if (!avctx->channels) {
1061  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1062  return AVERROR_INVALIDDATA;
1063  }
1064 
1065  /* Initialize RNG. */
1066  av_lfg_init(&q->random_state, 0);
1067 
1068  ff_dsputil_init(&q->dsp, avctx);
1069 
1070  while (edata_ptr < edata_ptr_end) {
1071  /* 8 for mono, 16 for stereo, ? for multichannel
1072  Swap to right endianness so we don't need to care later on. */
1073  if (extradata_size >= 8) {
1074  q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
1075  samples_per_frame = bytestream_get_be16(&edata_ptr);
1076  q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
1077  extradata_size -= 8;
1078  }
1079  if (extradata_size >= 8) {
1080  bytestream_get_be32(&edata_ptr); // Unknown unused
1081  q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
1082  q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
1083  extradata_size -= 8;
1084  }
1085 
1086  /* Initialize extradata related variables. */
1087  q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1088  q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1089 
1090  /* Initialize default data states. */
1091  q->subpacket[s].log2_numvector_size = 5;
1093  q->subpacket[s].num_channels = 1;
1094 
1095  /* Initialize version-dependent variables */
1096 
1097  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1098  q->subpacket[s].cookversion);
1099  q->subpacket[s].joint_stereo = 0;
1100  switch (q->subpacket[s].cookversion) {
1101  case MONO:
1102  if (avctx->channels != 1) {
1103  avpriv_request_sample(avctx, "Container channels != 1");
1104  return AVERROR_PATCHWELCOME;
1105  }
1106  av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1107  break;
1108  case STEREO:
1109  if (avctx->channels != 1) {
1110  q->subpacket[s].bits_per_subpdiv = 1;
1111  q->subpacket[s].num_channels = 2;
1112  }
1113  av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1114  break;
1115  case JOINT_STEREO:
1116  if (avctx->channels != 2) {
1117  avpriv_request_sample(avctx, "Container channels != 2");
1118  return AVERROR_PATCHWELCOME;
1119  }
1120  av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1121  if (avctx->extradata_size >= 16) {
1122  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1124  q->subpacket[s].joint_stereo = 1;
1125  q->subpacket[s].num_channels = 2;
1126  }
1127  if (q->subpacket[s].samples_per_channel > 256) {
1128  q->subpacket[s].log2_numvector_size = 6;
1129  }
1130  if (q->subpacket[s].samples_per_channel > 512) {
1131  q->subpacket[s].log2_numvector_size = 7;
1132  }
1133  break;
1134  case MC_COOK:
1135  av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1136  if (extradata_size >= 4)
1137  channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
1138 
1140  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1142  q->subpacket[s].joint_stereo = 1;
1143  q->subpacket[s].num_channels = 2;
1144  q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1145 
1146  if (q->subpacket[s].samples_per_channel > 256) {
1147  q->subpacket[s].log2_numvector_size = 6;
1148  }
1149  if (q->subpacket[s].samples_per_channel > 512) {
1150  q->subpacket[s].log2_numvector_size = 7;
1151  }
1152  } else
1153  q->subpacket[s].samples_per_channel = samples_per_frame;
1154 
1155  break;
1156  default:
1157  avpriv_request_sample(avctx, "Cook version %d",
1158  q->subpacket[s].cookversion);
1159  return AVERROR_PATCHWELCOME;
1160  }
1161 
1162  if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1163  av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1164  return AVERROR_INVALIDDATA;
1165  } else
1167 
1168 
1169  /* Initialize variable relations */
1171 
1172  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1173  if (q->subpacket[s].total_subbands > 53) {
1174  avpriv_request_sample(avctx, "total_subbands > 53");
1175  return AVERROR_PATCHWELCOME;
1176  }
1177 
1178  if ((q->subpacket[s].js_vlc_bits > 6) ||
1179  (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1180  av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1181  q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1182  return AVERROR_INVALIDDATA;
1183  }
1184 
1185  if (q->subpacket[s].subbands > 50) {
1186  avpriv_request_sample(avctx, "subbands > 50");
1187  return AVERROR_PATCHWELCOME;
1188  }
1189  q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1190  q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1191  q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1192  q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1193 
1194  q->num_subpackets++;
1195  s++;
1196  if (s > MAX_SUBPACKETS) {
1197  avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS);
1198  return AVERROR_PATCHWELCOME;
1199  }
1200  }
1201  /* Generate tables */
1202  init_pow2table();
1203  init_gain_table(q);
1205 
1206  if ((ret = init_cook_vlc_tables(q)))
1207  return ret;
1208 
1209 
1210  if (avctx->block_align >= UINT_MAX / 2)
1211  return AVERROR(EINVAL);
1212 
1213  /* Pad the databuffer with:
1214  DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1215  FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1217  av_mallocz(avctx->block_align
1218  + DECODE_BYTES_PAD1(avctx->block_align)
1220  if (q->decoded_bytes_buffer == NULL)
1221  return AVERROR(ENOMEM);
1222 
1223  /* Initialize transform. */
1224  if ((ret = init_cook_mlt(q)))
1225  return ret;
1226 
1227  /* Initialize COOK signal arithmetic handling */
1228  if (1) {
1230  q->decouple = decouple_float;
1234  }
1235 
1236  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1237  if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1238  q->samples_per_channel != 1024) {
1239  avpriv_request_sample(avctx, "samples_per_channel = %d",
1240  q->samples_per_channel);
1241  return AVERROR_PATCHWELCOME;
1242  }
1243 
1244  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1245  if (channel_mask)
1246  avctx->channel_layout = channel_mask;
1247  else
1248  avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1249 
1250 #ifdef DEBUG
1251  dump_cook_context(q);
1252 #endif
1253  return 0;
1254 }
1255 
1257  .name = "cook",
1258  .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1259  .type = AVMEDIA_TYPE_AUDIO,
1260  .id = AV_CODEC_ID_COOK,
1261  .priv_data_size = sizeof(COOKContext),
1265  .capabilities = CODEC_CAP_DR1,
1266  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1268 };
int joint_stereo
Definition: cook.c:83
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:884
Definition: lfg.h:25
static av_cold void init_cplscales_table(COOKContext *q)
Definition: cook.c:240
static const int cplband[51]
Definition: cookdata.h:504
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:62
av_cold void ff_dsputil_init(DSPContext *c, AVCodecContext *avctx)
Definition: dsputil.c:2440
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:54
static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
function for getting the jointstereo coupling information
Definition: cook.c:738
This structure describes decoded (raw) audio or video data.
Definition: frame.h:107
static void categorize(COOKContext *q, COOKSubpacket *p, int *quant_index_table, int *category, int *category_index)
Calculate the category and category_index vector.
Definition: cook.c:399
DSPContext dsp
Definition: cook.c:125
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:105
VLC channel_coupling
Definition: cook.c:82
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:240
int * previous
Definition: cook.c:68
float decode_buffer_1[1024]
Definition: cook.c:150
int gain_1[9]
Definition: cook.c:94
static const int kmax_tab[7]
Definition: cookdata.h:57
float gain_table[23]
Definition: cook.c:144
static const int expbits_tab[8]
Definition: cookdata.h:35
int size
Definition: avcodec.h:974
static const float *const cplscales[5]
Definition: cookdata.h:576
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:58
int subbands
Definition: cook.c:76
static av_cold void init_pow2table(void)
Definition: cook.c:165
static int16_t * samples
Definition: output.c:53
#define FF_ARRAY_ELEMS(a)
av_dlog(ac->avr,"%d samples - audio_convert: %s to %s (%s)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt), use_generic?ac->func_descr_generic:ac->func_descr)
#define AV_CH_LAYOUT_STEREO
VLC envelope_quant_index[13]
Definition: cook.c:139
int num_vectors
Definition: cook.c:128
AVCodec.
Definition: avcodec.h:2755
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1816
int samples_per_channel
Definition: cook.c:79
#define FFALIGN(x, a)
Definition: common.h:62
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:116
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:708
AVCodec ff_cook_decoder
Definition: cook.c:1256
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
Definition: mimic.c:269
static av_cold void init_gain_table(COOKContext *q)
Definition: cook.c:175
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:87
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1787
uint8_t
int total_subbands
Definition: cook.c:86
#define av_cold
Definition: attributes.h:66
int js_subband_start
Definition: cook.c:77
uint8_t * decoded_bytes_buffer
Definition: cook.c:148
float mono_previous_buffer1[1024]
Definition: cook.c:89
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:591
static void expand_category(COOKContext *q, int *category, int *category_index)
Expand the category vector.
Definition: cook.c:496
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1162
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
int bits_per_subpdiv
Definition: cook.c:85
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:711
static void interpolate(float *out, float v1, float v2, int size)
Definition: twinvq.c:84
const char data[16]
Definition: mxf.c:66
cook_gains gains1
Definition: cook.c:92
uint8_t * data
Definition: avcodec.h:973
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:194
bitstream reader API header.
const float * cplscales[5]
Definition: cook.c:154
static int decode_subpacket(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, float **outbuffer)
Cook subpacket decoding.
Definition: cook.c:902
static int cook_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: cook.c:943
float, planar
Definition: samplefmt.h:60
AVLFG random_state
Definition: cook.c:131
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:844
#define DECODE_BYTES_PAD1(bytes)
Definition: cook.c:249
GetBitContext gb
Definition: cook.c:126
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
static const int vd_tab[7]
Definition: cookdata.h:61
void av_free(void *ptr)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc(). ...
Definition: mem.c:186
VLC sqvh[7]
Definition: cook.c:140
#define AVERROR(e)
Definition: error.h:43
static const float dither_tab[9]
Definition: cookdata.h:39
sample_fmts
Definition: avconv_filter.c:68
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:142
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:144
#define JOINT_STEREO
Definition: cook.c:60
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
#define AV_BE2NE32C(x)
Definition: bswap.h:105
static const uint16_t *const ccpl_huffcodes[5]
Definition: cookdata.h:491
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:148
float mono_previous_buffer2[1024]
Definition: cook.c:90
const char * name
Name of the codec implementation.
Definition: avcodec.h:2762
static int decode_envelope(COOKContext *q, COOKSubpacket *p, int *quant_index_table)
Create the quant index table needed for the envelope.
Definition: cook.c:358
#define ff_mdct_init
Definition: fft.h:144
int gain_2[9]
Definition: cook.c:95
Definition: get_bits.h:64
int off
Definition: dsputil_bfin.c:29
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1840
static void saturate_output_float(COOKContext *q, float *out)
Saturate the output signal and interleave.
Definition: cook.c:866
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:81
static const int vhvlcsize_tab[7]
Definition: cookdata.h:77
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:509
int gain_3[9]
Definition: cook.c:96
int discarded_packets
Definition: cook.c:132
int log2_numvector_size
Definition: cook.c:80
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:547
Definition: fft.h:62
int bit_rate
the average bitrate
Definition: avcodec.h:1112
static av_cold int init_cook_mlt(COOKContext *q)
Definition: cook.c:216
audio channel layout utility functions
int gain_4[9]
Definition: cook.c:97
static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
Definition: cook.c:627
static av_cold int cook_decode_init(AVCodecContext *avctx)
Cook initialization.
Definition: cook.c:1040
cook_gains gains2
Definition: cook.c:93
static char buffer[20]
Definition: seek-test.c:31
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:522
static const uint16_t *const cvh_huffcodes[7]
Definition: cookdata.h:425
int bits
Definition: get_bits.h:65
void ff_sine_window_init(float *window, int n)
Generate a sine window.
#define AVERROR_PATCHWELCOME
Not yet implemented in Libav, patches welcome.
Definition: error.h:57
NULL
Definition: eval.c:55
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:653
int num_subpackets
Definition: cook.c:155
Libavcodec external API header.
int samples_per_channel
Definition: cook.c:129
void(* interpolate)(struct cook *q, float *buffer, int gain_index, int gain_index_next)
Definition: cook.c:119
static av_cold int init_cook_vlc_tables(COOKContext *q)
Definition: cook.c:185
FFTContext mdct_ctx
Definition: cook.c:135
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:68
int sample_rate
samples per second
Definition: avcodec.h:1779
void(* vector_clipf)(float *dst, const float *src, float min, float max, int len)
Definition: dsputil.h:206
main external API structure.
Definition: avcodec.h:1054
static void(WINAPI *cond_broadcast)(pthread_cond_t *cond)
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:489
float mono_mdct_output[2048]
Definition: cook.c:149
float * mlt_window
Definition: cook.c:136
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:575
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
Definition: get_bits.h:424
static unsigned int av_lfg_get(AVLFG *c)
Get the next random unsigned 32-bit number using an ALFG.
Definition: lfg.h:38
int * now
Definition: cook.c:67
int extradata_size
Definition: avcodec.h:1163
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:271
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:109
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:792
#define SUBBAND_SIZE
Definition: cook.c:63
static av_cold int cook_decode_close(AVCodecContext *avctx)
Definition: cook.c:298
int index
Definition: gxfenc.c:72
#define MONO
Definition: cook.c:58
static float pow2tab[127]
Definition: cook.c:159
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:375
float decode_buffer_0[1060]
Definition: cook.c:152
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:30
COOKSubpacket subpacket[MAX_SUBPACKETS]
Definition: cook.c:156
float decode_buffer_2[1024]
Definition: cook.c:151
static float rootpow2tab[127]
Definition: cook.c:160
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:72
int bits_per_subpacket
Definition: cook.c:84
#define PRINT(a, b)
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:272
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:518
int num_channels
Definition: cook.c:74
common internal api header.
#define STEREO
Definition: cook.c:59
#define ff_mdct_end
Definition: fft.h:145
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:49
AVCodecContext * avctx
Definition: cook.c:124
static av_cold int init(AVCodecParserContext *s)
Definition: h264_parser.c:498
int gain_size_factor
Definition: cook.c:143
DSP utils.
#define MAX_SUBPACKETS
Definition: cook.c:64
void * priv_data
Definition: avcodec.h:1090
static const int invradix_tab[7]
Definition: cookdata.h:53
int channels
number of audio channels
Definition: avcodec.h:1780
#define av_log2
Definition: intmath.h:85
#define MC_COOK
Definition: cook.c:61
int js_vlc_bits
Definition: cook.c:78
VLC_TYPE(* table)[2]
code, bits
Definition: get_bits.h:66
static const struct twinvq_data tab
static void decode_gain_info(GetBitContext *gb, int *gaininfo)
Fill the gain array for the timedomain quantization.
Definition: cook.c:330
#define FFSWAP(type, a, b)
Definition: common.h:60
static void imlt_window_float(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer)
Apply transform window, overlap buffers.
Definition: cook.c:680
int cookversion
Definition: cook.c:75
static void decouple_float(COOKContext *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2)
Definition: cook.c:770
static const int vpr_tab[7]
Definition: cookdata.h:65
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:141
#define AV_CH_LAYOUT_MONO
float min
This structure stores compressed data.
Definition: avcodec.h:950
void ff_free_vlc(VLC *vlc)
Definition: bitstream.c:333
int size
Definition: cook.c:73
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:151
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:205
unsigned int channel_mask
Definition: cook.c:81
DSPContext.
Definition: dsputil.h:124
Cook AKA RealAudio G2 compatible decoderdata.
void(* saturate_output)(struct cook *q, float *out)
Definition: cook.c:122