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 
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 
56 #include "cookdata.h"
57 
58 /* the different Cook versions */
59 #define MONO 0x1000001
60 #define STEREO 0x1000002
61 #define JOINT_STEREO 0x1000003
62 #define MC_COOK 0x2000000 // multichannel Cook, not supported
63 
64 #define SUBBAND_SIZE 20
65 #define MAX_SUBPACKETS 5
66 
67 typedef struct {
68  int *now;
69  int *previous;
70 } cook_gains;
71 
72 typedef struct {
73  int ch_idx;
74  int size;
77  int subbands;
82  unsigned int channel_mask;
88  int numvector_size; // 1 << log2_numvector_size;
89 
90  float mono_previous_buffer1[1024];
91  float mono_previous_buffer2[1024];
92 
95  int gain_1[9];
96  int gain_2[9];
97  int gain_3[9];
98  int gain_4[9];
100 
101 typedef struct cook {
102  /*
103  * The following 5 functions provide the lowlevel arithmetic on
104  * the internal audio buffers.
105  */
106  void (*scalar_dequant)(struct cook *q, int index, int quant_index,
107  int *subband_coef_index, int *subband_coef_sign,
108  float *mlt_p);
109 
110  void (*decouple)(struct cook *q,
111  COOKSubpacket *p,
112  int subband,
113  float f1, float f2,
114  float *decode_buffer,
115  float *mlt_buffer1, float *mlt_buffer2);
116 
117  void (*imlt_window)(struct cook *q, float *buffer1,
118  cook_gains *gains_ptr, float *previous_buffer);
119 
120  void (*interpolate)(struct cook *q, float *buffer,
121  int gain_index, int gain_index_next);
122 
123  void (*saturate_output)(struct cook *q, float *out);
124 
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 
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 
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 
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 
400 static void categorize(COOKContext *q, COOKSubpacket *p, 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 
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 
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 }
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 
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;
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  decode_vectors(q, p, category, quant_index_table, mlt_buffer);
641 
642  return 0;
643 }
644 
645 
654 static void interpolate_float(COOKContext *q, float *buffer,
655  int gain_index, int gain_index_next)
656 {
657  int i;
658  float fc1, fc2;
659  fc1 = pow2tab[gain_index + 63];
660 
661  if (gain_index == gain_index_next) { // static gain
662  for (i = 0; i < q->gain_size_factor; i++)
663  buffer[i] *= fc1;
664  } else { // smooth gain
665  fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
666  for (i = 0; i < q->gain_size_factor; i++) {
667  buffer[i] *= fc1;
668  fc1 *= fc2;
669  }
670  }
671 }
672 
681 static void imlt_window_float(COOKContext *q, float *inbuffer,
682  cook_gains *gains_ptr, float *previous_buffer)
683 {
684  const float fc = pow2tab[gains_ptr->previous[0] + 63];
685  int i;
686  /* The weird thing here, is that the two halves of the time domain
687  * buffer are swapped. Also, the newest data, that we save away for
688  * next frame, has the wrong sign. Hence the subtraction below.
689  * Almost sounds like a complex conjugate/reverse data/FFT effect.
690  */
691 
692  /* Apply window and overlap */
693  for (i = 0; i < q->samples_per_channel; i++)
694  inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
695  previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
696 }
697 
709 static void imlt_gain(COOKContext *q, float *inbuffer,
710  cook_gains *gains_ptr, float *previous_buffer)
711 {
712  float *buffer0 = q->mono_mdct_output;
713  float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
714  int i;
715 
716  /* Inverse modified discrete cosine transform */
717  q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
718 
719  q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
720 
721  /* Apply gain profile */
722  for (i = 0; i < 8; i++)
723  if (gains_ptr->now[i] || gains_ptr->now[i + 1])
724  q->interpolate(q, &buffer1[q->gain_size_factor * i],
725  gains_ptr->now[i], gains_ptr->now[i + 1]);
726 
727  /* Save away the current to be previous block. */
728  memcpy(previous_buffer, buffer0,
729  q->samples_per_channel * sizeof(*previous_buffer));
730 }
731 
732 
739 static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
740 {
741  int i;
742  int vlc = get_bits1(&q->gb);
743  int start = cplband[p->js_subband_start];
744  int end = cplband[p->subbands - 1];
745  int length = end - start + 1;
746 
747  if (start > end)
748  return;
749 
750  if (vlc)
751  for (i = 0; i < length; i++)
752  decouple_tab[start + i] = get_vlc2(&q->gb,
754  p->channel_coupling.bits, 2);
755  else
756  for (i = 0; i < length; i++)
757  decouple_tab[start + i] = get_bits(&q->gb, p->js_vlc_bits);
758 }
759 
760 /*
761  * function decouples a pair of signals from a single signal via multiplication.
762  *
763  * @param q pointer to the COOKContext
764  * @param subband index of the current subband
765  * @param f1 multiplier for channel 1 extraction
766  * @param f2 multiplier for channel 2 extraction
767  * @param decode_buffer input buffer
768  * @param mlt_buffer1 pointer to left channel mlt coefficients
769  * @param mlt_buffer2 pointer to right channel mlt coefficients
770  */
772  COOKSubpacket *p,
773  int subband,
774  float f1, float f2,
775  float *decode_buffer,
776  float *mlt_buffer1, float *mlt_buffer2)
777 {
778  int j, tmp_idx;
779  for (j = 0; j < SUBBAND_SIZE; j++) {
780  tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
781  mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
782  mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
783  }
784 }
785 
794  float *mlt_buffer_left, float *mlt_buffer_right)
795 {
796  int i, j, res;
797  int decouple_tab[SUBBAND_SIZE] = { 0 };
798  float *decode_buffer = q->decode_buffer_0;
799  int idx, cpl_tmp;
800  float f1, f2;
801  const float *cplscale;
802 
803  memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
804 
805  /* Make sure the buffers are zeroed out. */
806  memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left));
807  memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
808  decouple_info(q, p, decouple_tab);
809  if ((res = mono_decode(q, p, decode_buffer)) < 0)
810  return res;
811 
812  /* The two channels are stored interleaved in decode_buffer. */
813  for (i = 0; i < p->js_subband_start; i++) {
814  for (j = 0; j < SUBBAND_SIZE; j++) {
815  mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j];
816  mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
817  }
818  }
819 
820  /* When we reach js_subband_start (the higher frequencies)
821  the coefficients are stored in a coupling scheme. */
822  idx = (1 << p->js_vlc_bits) - 1;
823  for (i = p->js_subband_start; i < p->subbands; i++) {
824  cpl_tmp = cplband[i];
825  idx -= decouple_tab[cpl_tmp];
826  cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table
827  f1 = cplscale[decouple_tab[cpl_tmp] + 1];
828  f2 = cplscale[idx];
829  q->decouple(q, p, i, f1, f2, decode_buffer,
830  mlt_buffer_left, mlt_buffer_right);
831  idx = (1 << p->js_vlc_bits) - 1;
832  }
833 
834  return 0;
835 }
836 
846  const uint8_t *inbuffer,
847  cook_gains *gains_ptr)
848 {
849  int offset;
850 
851  offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
852  p->bits_per_subpacket / 8);
853  init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
854  p->bits_per_subpacket);
855  decode_gain_info(&q->gb, gains_ptr->now);
856 
857  /* Swap current and previous gains */
858  FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
859 }
860 
867 static void saturate_output_float(COOKContext *q, float *out)
868 {
870  -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
871 }
872 
873 
885 static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
886  cook_gains *gains_ptr, float *previous_buffer,
887  float *out)
888 {
889  imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
890  if (out)
891  q->saturate_output(q, out);
892 }
893 
894 
904  const uint8_t *inbuffer, float **outbuffer)
905 {
906  int sub_packet_size = p->size;
907  int res;
908 
909  memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
910  decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
911 
912  if (p->joint_stereo) {
913  if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
914  return res;
915  } else {
916  if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
917  return res;
918 
919  if (p->num_channels == 2) {
920  decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
921  if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
922  return res;
923  }
924  }
925 
928  outbuffer ? outbuffer[p->ch_idx] : NULL);
929 
930  if (p->num_channels == 2)
931  if (p->joint_stereo)
934  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
935  else
938  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
939 
940  return 0;
941 }
942 
943 
944 static int cook_decode_frame(AVCodecContext *avctx, void *data,
945  int *got_frame_ptr, AVPacket *avpkt)
946 {
947  AVFrame *frame = data;
948  const uint8_t *buf = avpkt->data;
949  int buf_size = avpkt->size;
950  COOKContext *q = avctx->priv_data;
951  float **samples = NULL;
952  int i, ret;
953  int offset = 0;
954  int chidx = 0;
955 
956  if (buf_size < avctx->block_align)
957  return buf_size;
958 
959  /* get output buffer */
960  if (q->discarded_packets >= 2) {
961  frame->nb_samples = q->samples_per_channel;
962  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
963  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
964  return ret;
965  }
966  samples = (float **)frame->extended_data;
967  }
968 
969  /* estimate subpacket sizes */
970  q->subpacket[0].size = avctx->block_align;
971 
972  for (i = 1; i < q->num_subpackets; i++) {
973  q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
974  q->subpacket[0].size -= q->subpacket[i].size + 1;
975  if (q->subpacket[0].size < 0) {
976  av_log(avctx, AV_LOG_DEBUG,
977  "frame subpacket size total > avctx->block_align!\n");
978  return AVERROR_INVALIDDATA;
979  }
980  }
981 
982  /* decode supbackets */
983  for (i = 0; i < q->num_subpackets; i++) {
984  q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
986  q->subpacket[i].ch_idx = chidx;
987  av_log(avctx, AV_LOG_DEBUG,
988  "subpacket[%i] size %i js %i %i block_align %i\n",
989  i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
990  avctx->block_align);
991 
992  if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
993  return ret;
994  offset += q->subpacket[i].size;
995  chidx += q->subpacket[i].num_channels;
996  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
997  i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
998  }
999 
1000  /* Discard the first two frames: no valid audio. */
1001  if (q->discarded_packets < 2) {
1002  q->discarded_packets++;
1003  *got_frame_ptr = 0;
1004  return avctx->block_align;
1005  }
1006 
1007  *got_frame_ptr = 1;
1008 
1009  return avctx->block_align;
1010 }
1011 
1012 #ifdef DEBUG
1013 static void dump_cook_context(COOKContext *q)
1014 {
1015  //int i=0;
1016 #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
1017  av_dlog(q->avctx, "COOKextradata\n");
1018  av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1019  if (q->subpacket[0].cookversion > STEREO) {
1020  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1021  PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1022  }
1023  av_dlog(q->avctx, "COOKContext\n");
1024  PRINT("nb_channels", q->avctx->channels);
1025  PRINT("bit_rate", q->avctx->bit_rate);
1026  PRINT("sample_rate", q->avctx->sample_rate);
1027  PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1028  PRINT("subbands", q->subpacket[0].subbands);
1029  PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1030  PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1031  PRINT("numvector_size", q->subpacket[0].numvector_size);
1032  PRINT("total_subbands", q->subpacket[0].total_subbands);
1033 }
1034 #endif
1035 
1042 {
1043  COOKContext *q = avctx->priv_data;
1044  const uint8_t *edata_ptr = avctx->extradata;
1045  const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
1046  int extradata_size = avctx->extradata_size;
1047  int s = 0;
1048  unsigned int channel_mask = 0;
1049  int samples_per_frame;
1050  int ret;
1051  q->avctx = avctx;
1052 
1053  /* Take care of the codec specific extradata. */
1054  if (extradata_size <= 0) {
1055  av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1056  return AVERROR_INVALIDDATA;
1057  }
1058  av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1059 
1060  /* Take data from the AVCodecContext (RM container). */
1061  if (!avctx->channels) {
1062  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1063  return AVERROR_INVALIDDATA;
1064  }
1065 
1066  /* Initialize RNG. */
1067  av_lfg_init(&q->random_state, 0);
1068 
1069  ff_audiodsp_init(&q->adsp);
1070 
1071  while (edata_ptr < edata_ptr_end) {
1072  /* 8 for mono, 16 for stereo, ? for multichannel
1073  Swap to right endianness so we don't need to care later on. */
1074  if (extradata_size >= 8) {
1075  q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
1076  samples_per_frame = bytestream_get_be16(&edata_ptr);
1077  q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
1078  extradata_size -= 8;
1079  }
1080  if (extradata_size >= 8) {
1081  bytestream_get_be32(&edata_ptr); // Unknown unused
1082  q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
1083  q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
1084  extradata_size -= 8;
1085  }
1086 
1087  /* Initialize extradata related variables. */
1088  q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1089  q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1090 
1091  /* Initialize default data states. */
1092  q->subpacket[s].log2_numvector_size = 5;
1094  q->subpacket[s].num_channels = 1;
1095 
1096  /* Initialize version-dependent variables */
1097 
1098  av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1099  q->subpacket[s].cookversion);
1100  q->subpacket[s].joint_stereo = 0;
1101  switch (q->subpacket[s].cookversion) {
1102  case MONO:
1103  if (avctx->channels != 1) {
1104  avpriv_request_sample(avctx, "Container channels != 1");
1105  return AVERROR_PATCHWELCOME;
1106  }
1107  av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1108  break;
1109  case STEREO:
1110  if (avctx->channels != 1) {
1111  q->subpacket[s].bits_per_subpdiv = 1;
1112  q->subpacket[s].num_channels = 2;
1113  }
1114  av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1115  break;
1116  case JOINT_STEREO:
1117  if (avctx->channels != 2) {
1118  avpriv_request_sample(avctx, "Container channels != 2");
1119  return AVERROR_PATCHWELCOME;
1120  }
1121  av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1122  if (avctx->extradata_size >= 16) {
1123  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1125  q->subpacket[s].joint_stereo = 1;
1126  q->subpacket[s].num_channels = 2;
1127  }
1128  if (q->subpacket[s].samples_per_channel > 256) {
1129  q->subpacket[s].log2_numvector_size = 6;
1130  }
1131  if (q->subpacket[s].samples_per_channel > 512) {
1132  q->subpacket[s].log2_numvector_size = 7;
1133  }
1134  break;
1135  case MC_COOK:
1136  av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1137  if (extradata_size >= 4)
1138  channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
1139 
1141  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1143  q->subpacket[s].joint_stereo = 1;
1144  q->subpacket[s].num_channels = 2;
1145  q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1146 
1147  if (q->subpacket[s].samples_per_channel > 256) {
1148  q->subpacket[s].log2_numvector_size = 6;
1149  }
1150  if (q->subpacket[s].samples_per_channel > 512) {
1151  q->subpacket[s].log2_numvector_size = 7;
1152  }
1153  } else
1154  q->subpacket[s].samples_per_channel = samples_per_frame;
1155 
1156  break;
1157  default:
1158  avpriv_request_sample(avctx, "Cook version %d",
1159  q->subpacket[s].cookversion);
1160  return AVERROR_PATCHWELCOME;
1161  }
1162 
1163  if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1164  av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1165  return AVERROR_INVALIDDATA;
1166  } else
1168 
1169 
1170  /* Initialize variable relations */
1172 
1173  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1174  if (q->subpacket[s].total_subbands > 53) {
1175  avpriv_request_sample(avctx, "total_subbands > 53");
1176  return AVERROR_PATCHWELCOME;
1177  }
1178 
1179  if ((q->subpacket[s].js_vlc_bits > 6) ||
1180  (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1181  av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1182  q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1183  return AVERROR_INVALIDDATA;
1184  }
1185 
1186  if (q->subpacket[s].subbands > 50) {
1187  avpriv_request_sample(avctx, "subbands > 50");
1188  return AVERROR_PATCHWELCOME;
1189  }
1190  q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
1191  q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1192  q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
1193  q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1194 
1195  q->num_subpackets++;
1196  s++;
1197  if (s > MAX_SUBPACKETS) {
1198  avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS);
1199  return AVERROR_PATCHWELCOME;
1200  }
1201  }
1202  /* Generate tables */
1203  init_pow2table();
1204  init_gain_table(q);
1206 
1207  if ((ret = init_cook_vlc_tables(q)))
1208  return ret;
1209 
1210 
1211  if (avctx->block_align >= UINT_MAX / 2)
1212  return AVERROR(EINVAL);
1213 
1214  /* Pad the databuffer with:
1215  DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1216  FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1218  av_mallocz(avctx->block_align
1219  + DECODE_BYTES_PAD1(avctx->block_align)
1221  if (!q->decoded_bytes_buffer)
1222  return AVERROR(ENOMEM);
1223 
1224  /* Initialize transform. */
1225  if ((ret = init_cook_mlt(q)))
1226  return ret;
1227 
1228  /* Initialize COOK signal arithmetic handling */
1229  if (1) {
1231  q->decouple = decouple_float;
1235  }
1236 
1237  /* Try to catch some obviously faulty streams, othervise it might be exploitable */
1238  if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1239  q->samples_per_channel != 1024) {
1240  avpriv_request_sample(avctx, "samples_per_channel = %d",
1241  q->samples_per_channel);
1242  return AVERROR_PATCHWELCOME;
1243  }
1244 
1245  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1246  if (channel_mask)
1247  avctx->channel_layout = channel_mask;
1248  else
1249  avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1250 
1251 #ifdef DEBUG
1252  dump_cook_context(q);
1253 #endif
1254  return 0;
1255 }
1256 
1258  .name = "cook",
1259  .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1260  .type = AVMEDIA_TYPE_AUDIO,
1261  .id = AV_CODEC_ID_COOK,
1262  .priv_data_size = sizeof(COOKContext),
1266  .capabilities = CODEC_CAP_DR1,
1267  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1269 };
int joint_stereo
Definition: cook.c:84
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:885
Definition: lfg.h:25
static av_cold void init_cplscales_table(COOKContext *q)
Definition: cook.c:241
static const int cplband[51]
Definition: cookdata.h:504
float, planar
Definition: samplefmt.h:72
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
#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:739
This structure describes decoded (raw) audio or video data.
Definition: frame.h:135
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:400
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:106
VLC channel_coupling
Definition: cook.c:83
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:69
float decode_buffer_1[1024]
Definition: cook.c:151
int gain_1[9]
Definition: cook.c:95
static const int kmax_tab[7]
Definition: cookdata.h:57
float gain_table[23]
Definition: cook.c:145
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:77
av_cold void ff_audiodsp_init(AudioDSPContext *c)
Definition: audiodsp.c:106
static av_cold void init_pow2table(void)
Definition: cook.c:166
#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:140
int num_vectors
Definition: cook.c:129
AVCodec.
Definition: avcodec.h:2796
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1828
int samples_per_channel
Definition: cook.c:80
#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:117
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:709
AVCodec ff_cook_decoder
Definition: cook.c:1257
static int decode(MimicContext *ctx, int quality, int num_coeffs, int is_iframe)
Definition: mimic.c:275
static av_cold void init_gain_table(COOKContext *q)
Definition: cook.c:176
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:88
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1799
uint8_t
int total_subbands
Definition: cook.c:87
#define av_cold
Definition: attributes.h:66
int js_subband_start
Definition: cook.c:78
uint8_t * decoded_bytes_buffer
Definition: cook.c:149
float mono_previous_buffer1[1024]
Definition: cook.c:90
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:592
static void expand_category(COOKContext *q, int *category, int *category_index)
Expand the category vector.
Definition: cook.c:497
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1164
int bits_per_subpdiv
Definition: cook.c:86
#define CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:684
static void interpolate(float *out, float v1, float v2, int size)
Definition: twinvq.c:84
const char data[16]
Definition: mxf.c:70
cook_gains gains1
Definition: cook.c:93
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:155
static int decode_subpacket(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, float **outbuffer)
Cook subpacket decoding.
Definition: cook.c:903
static int cook_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: cook.c:944
AVLFG random_state
Definition: cook.c:132
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:845
#define DECODE_BYTES_PAD1(bytes)
Definition: cook.c:250
GetBitContext gb
Definition: cook.c:127
#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:141
#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:150
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:144
#define JOINT_STEREO
Definition: cook.c:61
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:168
float mono_previous_buffer2[1024]
Definition: cook.c:91
const char * name
Name of the codec implementation.
Definition: avcodec.h:2803
static int decode_envelope(COOKContext *q, COOKSubpacket *p, int *quant_index_table)
Create the quant index table needed for the envelope.
Definition: cook.c:359
#define ff_mdct_init
Definition: fft.h:151
int gain_2[9]
Definition: cook.c:96
Definition: get_bits.h:64
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1852
static void saturate_output_float(COOKContext *q, float *out)
Saturate the output signal and interleave.
Definition: cook.c:867
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:92
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:531
int gain_3[9]
Definition: cook.c:97
int discarded_packets
Definition: cook.c:133
int log2_numvector_size
Definition: cook.c:81
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:548
Definition: fft.h:73
int bit_rate
the average bitrate
Definition: avcodec.h:1114
static av_cold int init_cook_mlt(COOKContext *q)
Definition: cook.c:217
audio channel layout utility functions
int gain_4[9]
Definition: cook.c:98
static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
Definition: cook.c:628
static av_cold int cook_decode_init(AVCodecContext *avctx)
Cook initialization.
Definition: cook.c:1041
cook_gains gains2
Definition: cook.c:94
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:654
int num_subpackets
Definition: cook.c:156
Libavcodec external API header.
void(* vector_clipf)(float *dst, const float *src, float min, float max, int len)
Definition: audiodsp.h:49
int samples_per_channel
Definition: cook.c:130
void(* interpolate)(struct cook *q, float *buffer, int gain_index, int gain_index_next)
Definition: cook.c:120
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:61
static av_cold int init_cook_vlc_tables(COOKContext *q)
Definition: cook.c:186
FFTContext mdct_ctx
Definition: cook.c:136
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:68
int sample_rate
samples per second
Definition: avcodec.h:1791
main external API structure.
Definition: avcodec.h:1050
static void(WINAPI *cond_broadcast)(pthread_cond_t *cond)
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:490
float mono_mdct_output[2048]
Definition: cook.c:150
float * mlt_window
Definition: cook.c:137
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:612
#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:68
int extradata_size
Definition: avcodec.h:1165
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:110
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:793
#define SUBBAND_SIZE
Definition: cook.c:64
static av_cold int cook_decode_close(AVCodecContext *avctx)
Definition: cook.c:299
int index
Definition: gxfenc.c:72
#define MONO
Definition: cook.c:59
static float pow2tab[127]
Definition: cook.c:160
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:153
AudioDSPContext adsp
Definition: cook.c:126
av_cold void av_lfg_init(AVLFG *c, unsigned int seed)
Definition: lfg.c:30
COOKSubpacket subpacket[MAX_SUBPACKETS]
Definition: cook.c:157
float decode_buffer_2[1024]
Definition: cook.c:152
static float rootpow2tab[127]
Definition: cook.c:161
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:73
int bits_per_subpacket
Definition: cook.c:85
#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:273
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:519
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=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);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_AARCH64) ff_audio_convert_init_aarch64(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 num_channels
Definition: cook.c:75
common internal api header.
#define STEREO
Definition: cook.c:60
#define ff_mdct_end
Definition: fft.h:152
AVCodecContext * avctx
Definition: cook.c:125
static av_cold int init(AVCodecParserContext *s)
Definition: h264_parser.c:499
int gain_size_factor
Definition: cook.c:144
#define MAX_SUBPACKETS
Definition: cook.c:65
void * priv_data
Definition: avcodec.h:1092
static const int invradix_tab[7]
Definition: cookdata.h:53
int channels
number of audio channels
Definition: avcodec.h:1792
#define av_log2
Definition: intmath.h:85
#define MC_COOK
Definition: cook.c:62
int js_vlc_bits
Definition: cook.c:79
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:331
#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:681
int cookversion
Definition: cook.c:76
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:771
static const int vpr_tab[7]
Definition: cookdata.h:65
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:169
#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:74
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:179
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:82
Cook AKA RealAudio G2 compatible decoderdata.
void(* saturate_output)(struct cook *q, float *out)
Definition: cook.c:123