FFmpeg
wavpack.c
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1 /*
2  * WavPack lossless audio decoder
3  * Copyright (c) 2006,2011 Konstantin Shishkov
4  * Copyright (c) 2020 David Bryant
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #include "libavutil/buffer.h"
25 
26 #define BITSTREAM_READER_LE
27 #include "avcodec.h"
28 #include "bytestream.h"
29 #include "get_bits.h"
30 #include "internal.h"
31 #include "thread.h"
32 #include "unary.h"
33 #include "wavpack.h"
34 #include "dsd.h"
35 
36 /**
37  * @file
38  * WavPack lossless audio decoder
39  */
40 
41 #define DSD_BYTE_READY(low,high) (!(((low) ^ (high)) & 0xff000000))
42 
43 #define PTABLE_BITS 8
44 #define PTABLE_BINS (1<<PTABLE_BITS)
45 #define PTABLE_MASK (PTABLE_BINS-1)
46 
47 #define UP 0x010000fe
48 #define DOWN 0x00010000
49 #define DECAY 8
50 
51 #define PRECISION 20
52 #define VALUE_ONE (1 << PRECISION)
53 #define PRECISION_USE 12
54 
55 #define RATE_S 20
56 
57 #define MAX_HISTORY_BITS 5
58 #define MAX_HISTORY_BINS (1 << MAX_HISTORY_BITS)
59 #define MAX_BIN_BYTES 1280 // for value_lookup, per bin (2k - 512 - 256)
60 
61 typedef enum {
62  MODULATION_PCM, // pulse code modulation
63  MODULATION_DSD // pulse density modulation (aka DSD)
64 } Modulation;
65 
66 typedef struct WavpackFrameContext {
70  int joint;
71  uint32_t CRC;
74  uint32_t crc_extra_bits;
76  int samples;
77  int terms;
79  int zero, one, zeroes;
81  int and, or, shift;
89 
97 
98 #define WV_MAX_FRAME_DECODERS 14
99 
100 typedef struct WavpackContext {
102 
104  int fdec_num;
105 
106  int block;
107  int samples;
109 
113 
118 
119 #define LEVEL_DECAY(a) (((a) + 0x80) >> 8)
120 
121 static av_always_inline unsigned get_tail(GetBitContext *gb, int k)
122 {
123  int p, e, res;
124 
125  if (k < 1)
126  return 0;
127  p = av_log2(k);
128  e = (1 << (p + 1)) - k - 1;
129  res = get_bitsz(gb, p);
130  if (res >= e)
131  res = (res << 1) - e + get_bits1(gb);
132  return res;
133 }
134 
136 {
137  int i, br[2], sl[2];
138 
139  for (i = 0; i <= ctx->stereo_in; i++) {
140  if (ctx->ch[i].bitrate_acc > UINT_MAX - ctx->ch[i].bitrate_delta)
141  return AVERROR_INVALIDDATA;
142  ctx->ch[i].bitrate_acc += ctx->ch[i].bitrate_delta;
143  br[i] = ctx->ch[i].bitrate_acc >> 16;
144  sl[i] = LEVEL_DECAY(ctx->ch[i].slow_level);
145  }
146  if (ctx->stereo_in && ctx->hybrid_bitrate) {
147  int balance = (sl[1] - sl[0] + br[1] + 1) >> 1;
148  if (balance > br[0]) {
149  br[1] = br[0] * 2;
150  br[0] = 0;
151  } else if (-balance > br[0]) {
152  br[0] *= 2;
153  br[1] = 0;
154  } else {
155  br[1] = br[0] + balance;
156  br[0] = br[0] - balance;
157  }
158  }
159  for (i = 0; i <= ctx->stereo_in; i++) {
160  if (ctx->hybrid_bitrate) {
161  if (sl[i] - br[i] > -0x100)
162  ctx->ch[i].error_limit = wp_exp2(sl[i] - br[i] + 0x100);
163  else
164  ctx->ch[i].error_limit = 0;
165  } else {
166  ctx->ch[i].error_limit = wp_exp2(br[i]);
167  }
168  }
169 
170  return 0;
171 }
172 
174  int channel, int *last)
175 {
176  int t, t2;
177  int sign, base, add, ret;
178  WvChannel *c = &ctx->ch[channel];
179 
180  *last = 0;
181 
182  if ((ctx->ch[0].median[0] < 2U) && (ctx->ch[1].median[0] < 2U) &&
183  !ctx->zero && !ctx->one) {
184  if (ctx->zeroes) {
185  ctx->zeroes--;
186  if (ctx->zeroes) {
187  c->slow_level -= LEVEL_DECAY(c->slow_level);
188  return 0;
189  }
190  } else {
191  t = get_unary_0_33(gb);
192  if (t >= 2) {
193  if (t >= 32 || get_bits_left(gb) < t - 1)
194  goto error;
195  t = get_bits_long(gb, t - 1) | (1 << (t - 1));
196  } else {
197  if (get_bits_left(gb) < 0)
198  goto error;
199  }
200  ctx->zeroes = t;
201  if (ctx->zeroes) {
202  memset(ctx->ch[0].median, 0, sizeof(ctx->ch[0].median));
203  memset(ctx->ch[1].median, 0, sizeof(ctx->ch[1].median));
204  c->slow_level -= LEVEL_DECAY(c->slow_level);
205  return 0;
206  }
207  }
208  }
209 
210  if (ctx->zero) {
211  t = 0;
212  ctx->zero = 0;
213  } else {
214  t = get_unary_0_33(gb);
215  if (get_bits_left(gb) < 0)
216  goto error;
217  if (t == 16) {
218  t2 = get_unary_0_33(gb);
219  if (t2 < 2) {
220  if (get_bits_left(gb) < 0)
221  goto error;
222  t += t2;
223  } else {
224  if (t2 >= 32 || get_bits_left(gb) < t2 - 1)
225  goto error;
226  t += get_bits_long(gb, t2 - 1) | (1 << (t2 - 1));
227  }
228  }
229 
230  if (ctx->one) {
231  ctx->one = t & 1;
232  t = (t >> 1) + 1;
233  } else {
234  ctx->one = t & 1;
235  t >>= 1;
236  }
237  ctx->zero = !ctx->one;
238  }
239 
240  if (ctx->hybrid && !channel) {
241  if (update_error_limit(ctx) < 0)
242  goto error;
243  }
244 
245  if (!t) {
246  base = 0;
247  add = GET_MED(0) - 1;
248  DEC_MED(0);
249  } else if (t == 1) {
250  base = GET_MED(0);
251  add = GET_MED(1) - 1;
252  INC_MED(0);
253  DEC_MED(1);
254  } else if (t == 2) {
255  base = GET_MED(0) + GET_MED(1);
256  add = GET_MED(2) - 1;
257  INC_MED(0);
258  INC_MED(1);
259  DEC_MED(2);
260  } else {
261  base = GET_MED(0) + GET_MED(1) + GET_MED(2) * (t - 2U);
262  add = GET_MED(2) - 1;
263  INC_MED(0);
264  INC_MED(1);
265  INC_MED(2);
266  }
267  if (!c->error_limit) {
268  if (add >= 0x2000000U) {
269  av_log(ctx->avctx, AV_LOG_ERROR, "k %d is too large\n", add);
270  goto error;
271  }
272  ret = base + get_tail(gb, add);
273  if (get_bits_left(gb) <= 0)
274  goto error;
275  } else {
276  int mid = (base * 2U + add + 1) >> 1;
277  while (add > c->error_limit) {
278  if (get_bits_left(gb) <= 0)
279  goto error;
280  if (get_bits1(gb)) {
281  add -= (mid - (unsigned)base);
282  base = mid;
283  } else
284  add = mid - (unsigned)base - 1;
285  mid = (base * 2U + add + 1) >> 1;
286  }
287  ret = mid;
288  }
289  sign = get_bits1(gb);
290  if (ctx->hybrid_bitrate)
291  c->slow_level += wp_log2(ret) - LEVEL_DECAY(c->slow_level);
292  return sign ? ~ret : ret;
293 
294 error:
295  ret = get_bits_left(gb);
296  if (ret <= 0) {
297  av_log(ctx->avctx, AV_LOG_ERROR, "Too few bits (%d) left\n", ret);
298  }
299  *last = 1;
300  return 0;
301 }
302 
303 static inline int wv_get_value_integer(WavpackFrameContext *s, uint32_t *crc,
304  unsigned S)
305 {
306  unsigned bit;
307 
308  if (s->extra_bits) {
309  S *= 1 << s->extra_bits;
310 
311  if (s->got_extra_bits &&
312  get_bits_left(&s->gb_extra_bits) >= s->extra_bits) {
313  S |= get_bits_long(&s->gb_extra_bits, s->extra_bits);
314  *crc = *crc * 9 + (S & 0xffff) * 3 + ((unsigned)S >> 16);
315  }
316  }
317 
318  bit = (S & s->and) | s->or;
319  bit = ((S + bit) << s->shift) - bit;
320 
321  if (s->hybrid)
322  bit = av_clip(bit, s->hybrid_minclip, s->hybrid_maxclip);
323 
324  return bit << s->post_shift;
325 }
326 
327 static float wv_get_value_float(WavpackFrameContext *s, uint32_t *crc, int S)
328 {
329  union {
330  float f;
331  uint32_t u;
332  } value;
333 
334  unsigned int sign;
335  int exp = s->float_max_exp;
336 
337  if (s->got_extra_bits) {
338  const int max_bits = 1 + 23 + 8 + 1;
339  const int left_bits = get_bits_left(&s->gb_extra_bits);
340 
341  if (left_bits + 8 * AV_INPUT_BUFFER_PADDING_SIZE < max_bits)
342  return 0.0;
343  }
344 
345  if (S) {
346  S *= 1U << s->float_shift;
347  sign = S < 0;
348  if (sign)
349  S = -(unsigned)S;
350  if (S >= 0x1000000U) {
351  if (s->got_extra_bits && get_bits1(&s->gb_extra_bits))
352  S = get_bits(&s->gb_extra_bits, 23);
353  else
354  S = 0;
355  exp = 255;
356  } else if (exp) {
357  int shift = 23 - av_log2(S);
358  exp = s->float_max_exp;
359  if (exp <= shift)
360  shift = --exp;
361  exp -= shift;
362 
363  if (shift) {
364  S <<= shift;
365  if ((s->float_flag & WV_FLT_SHIFT_ONES) ||
366  (s->got_extra_bits &&
367  (s->float_flag & WV_FLT_SHIFT_SAME) &&
368  get_bits1(&s->gb_extra_bits))) {
369  S |= (1 << shift) - 1;
370  } else if (s->got_extra_bits &&
371  (s->float_flag & WV_FLT_SHIFT_SENT)) {
372  S |= get_bits(&s->gb_extra_bits, shift);
373  }
374  }
375  } else {
376  exp = s->float_max_exp;
377  }
378  S &= 0x7fffff;
379  } else {
380  sign = 0;
381  exp = 0;
382  if (s->got_extra_bits && (s->float_flag & WV_FLT_ZERO_SENT)) {
383  if (get_bits1(&s->gb_extra_bits)) {
384  S = get_bits(&s->gb_extra_bits, 23);
385  if (s->float_max_exp >= 25)
386  exp = get_bits(&s->gb_extra_bits, 8);
387  sign = get_bits1(&s->gb_extra_bits);
388  } else {
389  if (s->float_flag & WV_FLT_ZERO_SIGN)
390  sign = get_bits1(&s->gb_extra_bits);
391  }
392  }
393  }
394 
395  *crc = *crc * 27 + S * 9 + exp * 3 + sign;
396 
397  value.u = (sign << 31) | (exp << 23) | S;
398  return value.f;
399 }
400 
401 static inline int wv_check_crc(WavpackFrameContext *s, uint32_t crc,
402  uint32_t crc_extra_bits)
403 {
404  if (crc != s->CRC) {
405  av_log(s->avctx, AV_LOG_ERROR, "CRC error\n");
406  return AVERROR_INVALIDDATA;
407  }
408  if (s->got_extra_bits && crc_extra_bits != s->crc_extra_bits) {
409  av_log(s->avctx, AV_LOG_ERROR, "Extra bits CRC error\n");
410  return AVERROR_INVALIDDATA;
411  }
412 
413  return 0;
414 }
415 
416 static void init_ptable(int *table, int rate_i, int rate_s)
417 {
418  int value = 0x808000, rate = rate_i << 8;
419 
420  for (int c = (rate + 128) >> 8; c--;)
421  value += (DOWN - value) >> DECAY;
422 
423  for (int i = 0; i < PTABLE_BINS/2; i++) {
424  table[i] = value;
425  table[PTABLE_BINS-1-i] = 0x100ffff - value;
426 
427  if (value > 0x010000) {
428  rate += (rate * rate_s + 128) >> 8;
429 
430  for (int c = (rate + 64) >> 7; c--;)
431  value += (DOWN - value) >> DECAY;
432  }
433  }
434 }
435 
436 typedef struct {
437  int32_t value, fltr0, fltr1, fltr2, fltr3, fltr4, fltr5, fltr6, factor;
438  unsigned int byte;
439 } DSDfilters;
440 
441 static int wv_unpack_dsd_high(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
442 {
443  uint32_t checksum = 0xFFFFFFFF;
444  uint8_t *dst_l = dst_left, *dst_r = dst_right;
445  int total_samples = s->samples, stereo = dst_r ? 1 : 0;
446  DSDfilters filters[2], *sp = filters;
447  int rate_i, rate_s;
448  uint32_t low, high, value;
449 
450  if (bytestream2_get_bytes_left(&s->gbyte) < (stereo ? 20 : 13))
451  return AVERROR_INVALIDDATA;
452 
453  rate_i = bytestream2_get_byte(&s->gbyte);
454  rate_s = bytestream2_get_byte(&s->gbyte);
455 
456  if (rate_s != RATE_S)
457  return AVERROR_INVALIDDATA;
458 
459  init_ptable(s->ptable, rate_i, rate_s);
460 
461  for (int channel = 0; channel < stereo + 1; channel++) {
463 
464  sp->fltr1 = bytestream2_get_byte(&s->gbyte) << (PRECISION - 8);
465  sp->fltr2 = bytestream2_get_byte(&s->gbyte) << (PRECISION - 8);
466  sp->fltr3 = bytestream2_get_byte(&s->gbyte) << (PRECISION - 8);
467  sp->fltr4 = bytestream2_get_byte(&s->gbyte) << (PRECISION - 8);
468  sp->fltr5 = bytestream2_get_byte(&s->gbyte) << (PRECISION - 8);
469  sp->fltr6 = 0;
470  sp->factor = bytestream2_get_byte(&s->gbyte) & 0xff;
471  sp->factor |= (bytestream2_get_byte(&s->gbyte) << 8) & 0xff00;
472  sp->factor = (int32_t)((uint32_t)sp->factor << 16) >> 16;
473  }
474 
475  value = bytestream2_get_be32(&s->gbyte);
476  high = 0xffffffff;
477  low = 0x0;
478 
479  while (total_samples--) {
480  int bitcount = 8;
481 
482  sp[0].value = sp[0].fltr1 - sp[0].fltr5 + ((sp[0].fltr6 * sp[0].factor) >> 2);
483 
484  if (stereo)
485  sp[1].value = sp[1].fltr1 - sp[1].fltr5 + ((sp[1].fltr6 * sp[1].factor) >> 2);
486 
487  while (bitcount--) {
488  int32_t *pp = s->ptable + ((sp[0].value >> (PRECISION - PRECISION_USE)) & PTABLE_MASK);
489  uint32_t split = low + ((high - low) >> 8) * (*pp >> 16);
490 
491  if (value <= split) {
492  high = split;
493  *pp += (UP - *pp) >> DECAY;
494  sp[0].fltr0 = -1;
495  } else {
496  low = split + 1;
497  *pp += (DOWN - *pp) >> DECAY;
498  sp[0].fltr0 = 0;
499  }
500 
501  while (DSD_BYTE_READY(high, low) && bytestream2_get_bytes_left(&s->gbyte)) {
502  value = (value << 8) | bytestream2_get_byte(&s->gbyte);
503  high = (high << 8) | 0xff;
504  low <<= 8;
505  }
506 
507  sp[0].value += sp[0].fltr6 * 8;
508  sp[0].byte = (sp[0].byte << 1) | (sp[0].fltr0 & 1);
509  sp[0].factor += (((sp[0].value ^ sp[0].fltr0) >> 31) | 1) &
510  ((sp[0].value ^ (sp[0].value - (sp[0].fltr6 * 16))) >> 31);
511  sp[0].fltr1 += ((sp[0].fltr0 & VALUE_ONE) - sp[0].fltr1) >> 6;
512  sp[0].fltr2 += ((sp[0].fltr0 & VALUE_ONE) - sp[0].fltr2) >> 4;
513  sp[0].fltr3 += (sp[0].fltr2 - sp[0].fltr3) >> 4;
514  sp[0].fltr4 += (sp[0].fltr3 - sp[0].fltr4) >> 4;
515  sp[0].value = (sp[0].fltr4 - sp[0].fltr5) >> 4;
516  sp[0].fltr5 += sp[0].value;
517  sp[0].fltr6 += (sp[0].value - sp[0].fltr6) >> 3;
518  sp[0].value = sp[0].fltr1 - sp[0].fltr5 + ((sp[0].fltr6 * sp[0].factor) >> 2);
519 
520  if (!stereo)
521  continue;
522 
523  pp = s->ptable + ((sp[1].value >> (PRECISION - PRECISION_USE)) & PTABLE_MASK);
524  split = low + ((high - low) >> 8) * (*pp >> 16);
525 
526  if (value <= split) {
527  high = split;
528  *pp += (UP - *pp) >> DECAY;
529  sp[1].fltr0 = -1;
530  } else {
531  low = split + 1;
532  *pp += (DOWN - *pp) >> DECAY;
533  sp[1].fltr0 = 0;
534  }
535 
536  while (DSD_BYTE_READY(high, low) && bytestream2_get_bytes_left(&s->gbyte)) {
537  value = (value << 8) | bytestream2_get_byte(&s->gbyte);
538  high = (high << 8) | 0xff;
539  low <<= 8;
540  }
541 
542  sp[1].value += sp[1].fltr6 * 8;
543  sp[1].byte = (sp[1].byte << 1) | (sp[1].fltr0 & 1);
544  sp[1].factor += (((sp[1].value ^ sp[1].fltr0) >> 31) | 1) &
545  ((sp[1].value ^ (sp[1].value - (sp[1].fltr6 * 16))) >> 31);
546  sp[1].fltr1 += ((sp[1].fltr0 & VALUE_ONE) - sp[1].fltr1) >> 6;
547  sp[1].fltr2 += ((sp[1].fltr0 & VALUE_ONE) - sp[1].fltr2) >> 4;
548  sp[1].fltr3 += (sp[1].fltr2 - sp[1].fltr3) >> 4;
549  sp[1].fltr4 += (sp[1].fltr3 - sp[1].fltr4) >> 4;
550  sp[1].value = (sp[1].fltr4 - sp[1].fltr5) >> 4;
551  sp[1].fltr5 += sp[1].value;
552  sp[1].fltr6 += (sp[1].value - sp[1].fltr6) >> 3;
553  sp[1].value = sp[1].fltr1 - sp[1].fltr5 + ((sp[1].fltr6 * sp[1].factor) >> 2);
554  }
555 
556  checksum += (checksum << 1) + (*dst_l = sp[0].byte & 0xff);
557  sp[0].factor -= (sp[0].factor + 512) >> 10;
558  dst_l += 4;
559 
560  if (stereo) {
561  checksum += (checksum << 1) + (*dst_r = filters[1].byte & 0xff);
562  filters[1].factor -= (filters[1].factor + 512) >> 10;
563  dst_r += 4;
564  }
565  }
566 
567  if (wv_check_crc(s, checksum, 0)) {
568  if (s->avctx->err_recognition & AV_EF_CRCCHECK)
569  return AVERROR_INVALIDDATA;
570 
571  memset(dst_left, 0x69, s->samples * 4);
572 
573  if (dst_r)
574  memset(dst_right, 0x69, s->samples * 4);
575  }
576 
577  return 0;
578 }
579 
580 static int wv_unpack_dsd_fast(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
581 {
582  uint8_t *dst_l = dst_left, *dst_r = dst_right;
583  uint8_t history_bits, max_probability;
584  int total_summed_probabilities = 0;
585  int total_samples = s->samples;
586  uint8_t *vlb = s->value_lookup_buffer;
587  int history_bins, p0, p1, chan;
588  uint32_t checksum = 0xFFFFFFFF;
589  uint32_t low, high, value;
590 
591  if (!bytestream2_get_bytes_left(&s->gbyte))
592  return AVERROR_INVALIDDATA;
593 
594  history_bits = bytestream2_get_byte(&s->gbyte);
595 
596  if (!bytestream2_get_bytes_left(&s->gbyte) || history_bits > MAX_HISTORY_BITS)
597  return AVERROR_INVALIDDATA;
598 
599  history_bins = 1 << history_bits;
600  max_probability = bytestream2_get_byte(&s->gbyte);
601 
602  if (max_probability < 0xff) {
603  uint8_t *outptr = (uint8_t *)s->probabilities;
604  uint8_t *outend = outptr + sizeof(*s->probabilities) * history_bins;
605 
606  while (outptr < outend && bytestream2_get_bytes_left(&s->gbyte)) {
607  int code = bytestream2_get_byte(&s->gbyte);
608 
609  if (code > max_probability) {
610  int zcount = code - max_probability;
611 
612  while (outptr < outend && zcount--)
613  *outptr++ = 0;
614  } else if (code) {
615  *outptr++ = code;
616  }
617  else {
618  break;
619  }
620  }
621 
622  if (outptr < outend ||
623  (bytestream2_get_bytes_left(&s->gbyte) && bytestream2_get_byte(&s->gbyte)))
624  return AVERROR_INVALIDDATA;
625  } else if (bytestream2_get_bytes_left(&s->gbyte) > (int)sizeof(*s->probabilities) * history_bins) {
626  bytestream2_get_buffer(&s->gbyte, (uint8_t *)s->probabilities,
627  sizeof(*s->probabilities) * history_bins);
628  } else {
629  return AVERROR_INVALIDDATA;
630  }
631 
632  for (p0 = 0; p0 < history_bins; p0++) {
633  int32_t sum_values = 0;
634 
635  for (int i = 0; i < 256; i++)
636  s->summed_probabilities[p0][i] = sum_values += s->probabilities[p0][i];
637 
638  if (sum_values) {
639  total_summed_probabilities += sum_values;
640 
641  if (total_summed_probabilities > history_bins * MAX_BIN_BYTES)
642  return AVERROR_INVALIDDATA;
643 
644  s->value_lookup[p0] = vlb;
645 
646  for (int i = 0; i < 256; i++) {
647  int c = s->probabilities[p0][i];
648 
649  while (c--)
650  *vlb++ = i;
651  }
652  }
653  }
654 
655  if (bytestream2_get_bytes_left(&s->gbyte) < 4)
656  return AVERROR_INVALIDDATA;
657 
658  chan = p0 = p1 = 0;
659  low = 0; high = 0xffffffff;
660  value = bytestream2_get_be32(&s->gbyte);
661 
662  if (dst_r)
663  total_samples *= 2;
664 
665  while (total_samples--) {
666  unsigned int mult, index, code;
667 
668  if (!s->summed_probabilities[p0][255])
669  return AVERROR_INVALIDDATA;
670 
671  mult = (high - low) / s->summed_probabilities[p0][255];
672 
673  if (!mult) {
674  if (bytestream2_get_bytes_left(&s->gbyte) >= 4)
675  value = bytestream2_get_be32(&s->gbyte);
676 
677  low = 0;
678  high = 0xffffffff;
679  mult = high / s->summed_probabilities[p0][255];
680 
681  if (!mult)
682  return AVERROR_INVALIDDATA;
683  }
684 
685  index = (value - low) / mult;
686 
687  if (index >= s->summed_probabilities[p0][255])
688  return AVERROR_INVALIDDATA;
689 
690  if (!dst_r) {
691  if ((*dst_l = code = s->value_lookup[p0][index]))
692  low += s->summed_probabilities[p0][code-1] * mult;
693 
694  dst_l += 4;
695  } else {
696  if ((code = s->value_lookup[p0][index]))
697  low += s->summed_probabilities[p0][code-1] * mult;
698 
699  if (chan) {
700  *dst_r = code;
701  dst_r += 4;
702  }
703  else {
704  *dst_l = code;
705  dst_l += 4;
706  }
707 
708  chan ^= 1;
709  }
710 
711  high = low + s->probabilities[p0][code] * mult - 1;
712  checksum += (checksum << 1) + code;
713 
714  if (!dst_r) {
715  p0 = code & (history_bins-1);
716  } else {
717  p0 = p1;
718  p1 = code & (history_bins-1);
719  }
720 
721  while (DSD_BYTE_READY(high, low) && bytestream2_get_bytes_left(&s->gbyte)) {
722  value = (value << 8) | bytestream2_get_byte(&s->gbyte);
723  high = (high << 8) | 0xff;
724  low <<= 8;
725  }
726  }
727 
728  if (wv_check_crc(s, checksum, 0)) {
729  if (s->avctx->err_recognition & AV_EF_CRCCHECK)
730  return AVERROR_INVALIDDATA;
731 
732  memset(dst_left, 0x69, s->samples * 4);
733 
734  if (dst_r)
735  memset(dst_right, 0x69, s->samples * 4);
736  }
737 
738  return 0;
739 }
740 
741 static int wv_unpack_dsd_copy(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
742 {
743  uint8_t *dst_l = dst_left, *dst_r = dst_right;
744  int total_samples = s->samples;
745  uint32_t checksum = 0xFFFFFFFF;
746 
747  if (bytestream2_get_bytes_left(&s->gbyte) != total_samples * (dst_r ? 2 : 1))
748  return AVERROR_INVALIDDATA;
749 
750  while (total_samples--) {
751  checksum += (checksum << 1) + (*dst_l = bytestream2_get_byte(&s->gbyte));
752  dst_l += 4;
753 
754  if (dst_r) {
755  checksum += (checksum << 1) + (*dst_r = bytestream2_get_byte(&s->gbyte));
756  dst_r += 4;
757  }
758  }
759 
760  if (wv_check_crc(s, checksum, 0)) {
761  if (s->avctx->err_recognition & AV_EF_CRCCHECK)
762  return AVERROR_INVALIDDATA;
763 
764  memset(dst_left, 0x69, s->samples * 4);
765 
766  if (dst_r)
767  memset(dst_right, 0x69, s->samples * 4);
768  }
769 
770  return 0;
771 }
772 
774  void *dst_l, void *dst_r, const int type)
775 {
776  int i, j, count = 0;
777  int last, t;
778  int A, B, L, L2, R, R2;
779  int pos = 0;
780  uint32_t crc = 0xFFFFFFFF;
781  uint32_t crc_extra_bits = 0xFFFFFFFF;
782  int16_t *dst16_l = dst_l;
783  int16_t *dst16_r = dst_r;
784  int32_t *dst32_l = dst_l;
785  int32_t *dst32_r = dst_r;
786  float *dstfl_l = dst_l;
787  float *dstfl_r = dst_r;
788 
789  s->one = s->zero = s->zeroes = 0;
790  do {
791  L = wv_get_value(s, gb, 0, &last);
792  if (last)
793  break;
794  R = wv_get_value(s, gb, 1, &last);
795  if (last)
796  break;
797  for (i = 0; i < s->terms; i++) {
798  t = s->decorr[i].value;
799  if (t > 0) {
800  if (t > 8) {
801  if (t & 1) {
802  A = 2U * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1];
803  B = 2U * s->decorr[i].samplesB[0] - s->decorr[i].samplesB[1];
804  } else {
805  A = (int)(3U * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1]) >> 1;
806  B = (int)(3U * s->decorr[i].samplesB[0] - s->decorr[i].samplesB[1]) >> 1;
807  }
808  s->decorr[i].samplesA[1] = s->decorr[i].samplesA[0];
809  s->decorr[i].samplesB[1] = s->decorr[i].samplesB[0];
810  j = 0;
811  } else {
812  A = s->decorr[i].samplesA[pos];
813  B = s->decorr[i].samplesB[pos];
814  j = (pos + t) & 7;
815  }
816  if (type != AV_SAMPLE_FMT_S16P) {
817  L2 = L + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
818  R2 = R + ((s->decorr[i].weightB * (int64_t)B + 512) >> 10);
819  } else {
820  L2 = L + (unsigned)((int)(s->decorr[i].weightA * (unsigned)A + 512) >> 10);
821  R2 = R + (unsigned)((int)(s->decorr[i].weightB * (unsigned)B + 512) >> 10);
822  }
823  if (A && L)
824  s->decorr[i].weightA -= ((((L ^ A) >> 30) & 2) - 1) * s->decorr[i].delta;
825  if (B && R)
826  s->decorr[i].weightB -= ((((R ^ B) >> 30) & 2) - 1) * s->decorr[i].delta;
827  s->decorr[i].samplesA[j] = L = L2;
828  s->decorr[i].samplesB[j] = R = R2;
829  } else if (t == -1) {
830  if (type != AV_SAMPLE_FMT_S16P)
831  L2 = L + ((s->decorr[i].weightA * (int64_t)s->decorr[i].samplesA[0] + 512) >> 10);
832  else
833  L2 = L + (unsigned)((int)(s->decorr[i].weightA * (unsigned)s->decorr[i].samplesA[0] + 512) >> 10);
834  UPDATE_WEIGHT_CLIP(s->decorr[i].weightA, s->decorr[i].delta, s->decorr[i].samplesA[0], L);
835  L = L2;
836  if (type != AV_SAMPLE_FMT_S16P)
837  R2 = R + ((s->decorr[i].weightB * (int64_t)L2 + 512) >> 10);
838  else
839  R2 = R + (unsigned)((int)(s->decorr[i].weightB * (unsigned)L2 + 512) >> 10);
840  UPDATE_WEIGHT_CLIP(s->decorr[i].weightB, s->decorr[i].delta, L2, R);
841  R = R2;
842  s->decorr[i].samplesA[0] = R;
843  } else {
844  if (type != AV_SAMPLE_FMT_S16P)
845  R2 = R + ((s->decorr[i].weightB * (int64_t)s->decorr[i].samplesB[0] + 512) >> 10);
846  else
847  R2 = R + (unsigned)((int)(s->decorr[i].weightB * (unsigned)s->decorr[i].samplesB[0] + 512) >> 10);
848  UPDATE_WEIGHT_CLIP(s->decorr[i].weightB, s->decorr[i].delta, s->decorr[i].samplesB[0], R);
849  R = R2;
850 
851  if (t == -3) {
852  R2 = s->decorr[i].samplesA[0];
853  s->decorr[i].samplesA[0] = R;
854  }
855 
856  if (type != AV_SAMPLE_FMT_S16P)
857  L2 = L + ((s->decorr[i].weightA * (int64_t)R2 + 512) >> 10);
858  else
859  L2 = L + (unsigned)((int)(s->decorr[i].weightA * (unsigned)R2 + 512) >> 10);
860  UPDATE_WEIGHT_CLIP(s->decorr[i].weightA, s->decorr[i].delta, R2, L);
861  L = L2;
862  s->decorr[i].samplesB[0] = L;
863  }
864  }
865 
866  if (type == AV_SAMPLE_FMT_S16P) {
867  if (FFABS((int64_t)L) + FFABS((int64_t)R) > (1<<19)) {
868  av_log(s->avctx, AV_LOG_ERROR, "sample %d %d too large\n", L, R);
869  return AVERROR_INVALIDDATA;
870  }
871  }
872 
873  pos = (pos + 1) & 7;
874  if (s->joint)
875  L += (unsigned)(R -= (unsigned)(L >> 1));
876  crc = (crc * 3 + L) * 3 + R;
877 
878  if (type == AV_SAMPLE_FMT_FLTP) {
879  *dstfl_l++ = wv_get_value_float(s, &crc_extra_bits, L);
880  *dstfl_r++ = wv_get_value_float(s, &crc_extra_bits, R);
881  } else if (type == AV_SAMPLE_FMT_S32P) {
882  *dst32_l++ = wv_get_value_integer(s, &crc_extra_bits, L);
883  *dst32_r++ = wv_get_value_integer(s, &crc_extra_bits, R);
884  } else {
885  *dst16_l++ = wv_get_value_integer(s, &crc_extra_bits, L);
886  *dst16_r++ = wv_get_value_integer(s, &crc_extra_bits, R);
887  }
888  count++;
889  } while (!last && count < s->samples);
890 
891  if (last && count < s->samples) {
893  memset((uint8_t*)dst_l + count*size, 0, (s->samples-count)*size);
894  memset((uint8_t*)dst_r + count*size, 0, (s->samples-count)*size);
895  }
896 
897  if ((s->avctx->err_recognition & AV_EF_CRCCHECK) &&
898  wv_check_crc(s, crc, crc_extra_bits))
899  return AVERROR_INVALIDDATA;
900 
901  return 0;
902 }
903 
905  void *dst, const int type)
906 {
907  int i, j, count = 0;
908  int last, t;
909  int A, S, T;
910  int pos = 0;
911  uint32_t crc = 0xFFFFFFFF;
912  uint32_t crc_extra_bits = 0xFFFFFFFF;
913  int16_t *dst16 = dst;
914  int32_t *dst32 = dst;
915  float *dstfl = dst;
916 
917  s->one = s->zero = s->zeroes = 0;
918  do {
919  T = wv_get_value(s, gb, 0, &last);
920  S = 0;
921  if (last)
922  break;
923  for (i = 0; i < s->terms; i++) {
924  t = s->decorr[i].value;
925  if (t > 8) {
926  if (t & 1)
927  A = 2U * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1];
928  else
929  A = (int)(3U * s->decorr[i].samplesA[0] - s->decorr[i].samplesA[1]) >> 1;
930  s->decorr[i].samplesA[1] = s->decorr[i].samplesA[0];
931  j = 0;
932  } else {
933  A = s->decorr[i].samplesA[pos];
934  j = (pos + t) & 7;
935  }
936  if (type != AV_SAMPLE_FMT_S16P)
937  S = T + ((s->decorr[i].weightA * (int64_t)A + 512) >> 10);
938  else
939  S = T + (unsigned)((int)(s->decorr[i].weightA * (unsigned)A + 512) >> 10);
940  if (A && T)
941  s->decorr[i].weightA -= ((((T ^ A) >> 30) & 2) - 1) * s->decorr[i].delta;
942  s->decorr[i].samplesA[j] = T = S;
943  }
944  pos = (pos + 1) & 7;
945  crc = crc * 3 + S;
946 
947  if (type == AV_SAMPLE_FMT_FLTP) {
948  *dstfl++ = wv_get_value_float(s, &crc_extra_bits, S);
949  } else if (type == AV_SAMPLE_FMT_S32P) {
950  *dst32++ = wv_get_value_integer(s, &crc_extra_bits, S);
951  } else {
952  *dst16++ = wv_get_value_integer(s, &crc_extra_bits, S);
953  }
954  count++;
955  } while (!last && count < s->samples);
956 
957  if (last && count < s->samples) {
959  memset((uint8_t*)dst + count*size, 0, (s->samples-count)*size);
960  }
961 
962  if (s->avctx->err_recognition & AV_EF_CRCCHECK) {
963  int ret = wv_check_crc(s, crc, crc_extra_bits);
964  if (ret < 0 && s->avctx->err_recognition & AV_EF_EXPLODE)
965  return ret;
966  }
967 
968  return 0;
969 }
970 
972 {
973  if (c->fdec_num == WV_MAX_FRAME_DECODERS)
974  return -1;
975 
976  c->fdec[c->fdec_num] = av_mallocz(sizeof(**c->fdec));
977  if (!c->fdec[c->fdec_num])
978  return -1;
979  c->fdec_num++;
980  c->fdec[c->fdec_num - 1]->avctx = c->avctx;
981 
982  return 0;
983 }
984 
986 {
987  int i;
988 
989  s->dsdctx = NULL;
990  s->dsd_channels = 0;
991  av_buffer_unref(&s->dsd_ref);
992 
993  if (!channels)
994  return 0;
995 
996  if (channels > INT_MAX / sizeof(*s->dsdctx))
997  return AVERROR(EINVAL);
998 
999  s->dsd_ref = av_buffer_allocz(channels * sizeof(*s->dsdctx));
1000  if (!s->dsd_ref)
1001  return AVERROR(ENOMEM);
1002  s->dsdctx = (DSDContext*)s->dsd_ref->data;
1003  s->dsd_channels = channels;
1004 
1005  for (i = 0; i < channels; i++)
1006  memset(s->dsdctx[i].buf, 0x69, sizeof(s->dsdctx[i].buf));
1007 
1008  return 0;
1009 }
1010 
1011 #if HAVE_THREADS
1012 static int update_thread_context(AVCodecContext *dst, const AVCodecContext *src)
1013 {
1014  WavpackContext *fsrc = src->priv_data;
1015  WavpackContext *fdst = dst->priv_data;
1016  int ret;
1017 
1018  if (dst == src)
1019  return 0;
1020 
1021  ff_thread_release_buffer(dst, &fdst->curr_frame);
1022  if (fsrc->curr_frame.f->data[0]) {
1023  if ((ret = ff_thread_ref_frame(&fdst->curr_frame, &fsrc->curr_frame)) < 0)
1024  return ret;
1025  }
1026 
1027  av_buffer_unref(&fdst->dsd_ref);
1028  fdst->dsdctx = NULL;
1029  fdst->dsd_channels = 0;
1030  if (fsrc->dsd_ref) {
1031  fdst->dsd_ref = av_buffer_ref(fsrc->dsd_ref);
1032  if (!fdst->dsd_ref)
1033  return AVERROR(ENOMEM);
1034  fdst->dsdctx = (DSDContext*)fdst->dsd_ref->data;
1035  fdst->dsd_channels = fsrc->dsd_channels;
1036  }
1037 
1038  return 0;
1039 }
1040 #endif
1041 
1043 {
1044  WavpackContext *s = avctx->priv_data;
1045 
1046  s->avctx = avctx;
1047 
1048  s->fdec_num = 0;
1049 
1050  s->curr_frame.f = av_frame_alloc();
1051  s->prev_frame.f = av_frame_alloc();
1052 
1053  if (!s->curr_frame.f || !s->prev_frame.f)
1054  return AVERROR(ENOMEM);
1055 
1056  ff_init_dsd_data();
1057 
1058  return 0;
1059 }
1060 
1062 {
1063  WavpackContext *s = avctx->priv_data;
1064 
1065  for (int i = 0; i < s->fdec_num; i++)
1066  av_freep(&s->fdec[i]);
1067  s->fdec_num = 0;
1068 
1069  ff_thread_release_buffer(avctx, &s->curr_frame);
1070  av_frame_free(&s->curr_frame.f);
1071 
1072  ff_thread_release_buffer(avctx, &s->prev_frame);
1073  av_frame_free(&s->prev_frame.f);
1074 
1075  av_buffer_unref(&s->dsd_ref);
1076 
1077  return 0;
1078 }
1079 
1080 static int wavpack_decode_block(AVCodecContext *avctx, int block_no,
1081  const uint8_t *buf, int buf_size)
1082 {
1083  WavpackContext *wc = avctx->priv_data;
1085  GetByteContext gb;
1086  enum AVSampleFormat sample_fmt;
1087  void *samples_l = NULL, *samples_r = NULL;
1088  int ret;
1089  int got_terms = 0, got_weights = 0, got_samples = 0,
1090  got_entropy = 0, got_pcm = 0, got_float = 0, got_hybrid = 0;
1091  int got_dsd = 0;
1092  int i, j, id, size, ssize, weights, t;
1093  int bpp, chan = 0, orig_bpp, sample_rate = 0, rate_x = 1, dsd_mode = 0;
1094  int multiblock;
1095  uint64_t chmask = 0;
1096 
1097  if (block_no >= wc->fdec_num && wv_alloc_frame_context(wc) < 0) {
1098  av_log(avctx, AV_LOG_ERROR, "Error creating frame decode context\n");
1099  return AVERROR_INVALIDDATA;
1100  }
1101 
1102  s = wc->fdec[block_no];
1103  if (!s) {
1104  av_log(avctx, AV_LOG_ERROR, "Context for block %d is not present\n",
1105  block_no);
1106  return AVERROR_INVALIDDATA;
1107  }
1108 
1109  memset(s->decorr, 0, MAX_TERMS * sizeof(Decorr));
1110  memset(s->ch, 0, sizeof(s->ch));
1111  s->extra_bits = 0;
1112  s->and = s->or = s->shift = 0;
1113  s->got_extra_bits = 0;
1114 
1115  bytestream2_init(&gb, buf, buf_size);
1116 
1117  s->samples = bytestream2_get_le32(&gb);
1118  if (s->samples != wc->samples) {
1119  av_log(avctx, AV_LOG_ERROR, "Mismatching number of samples in "
1120  "a sequence: %d and %d\n", wc->samples, s->samples);
1121  return AVERROR_INVALIDDATA;
1122  }
1123  s->frame_flags = bytestream2_get_le32(&gb);
1124 
1125  if (s->frame_flags & (WV_FLOAT_DATA | WV_DSD_DATA))
1126  sample_fmt = AV_SAMPLE_FMT_FLTP;
1127  else if ((s->frame_flags & 0x03) <= 1)
1128  sample_fmt = AV_SAMPLE_FMT_S16P;
1129  else
1130  sample_fmt = AV_SAMPLE_FMT_S32P;
1131 
1132  if (wc->ch_offset && avctx->sample_fmt != sample_fmt)
1133  return AVERROR_INVALIDDATA;
1134 
1135  bpp = av_get_bytes_per_sample(sample_fmt);
1136  orig_bpp = ((s->frame_flags & 0x03) + 1) << 3;
1137  multiblock = (s->frame_flags & WV_SINGLE_BLOCK) != WV_SINGLE_BLOCK;
1138 
1139  s->stereo = !(s->frame_flags & WV_MONO);
1140  s->stereo_in = (s->frame_flags & WV_FALSE_STEREO) ? 0 : s->stereo;
1141  s->joint = s->frame_flags & WV_JOINT_STEREO;
1142  s->hybrid = s->frame_flags & WV_HYBRID_MODE;
1143  s->hybrid_bitrate = s->frame_flags & WV_HYBRID_BITRATE;
1144  s->post_shift = bpp * 8 - orig_bpp + ((s->frame_flags >> 13) & 0x1f);
1145  if (s->post_shift < 0 || s->post_shift > 31) {
1146  return AVERROR_INVALIDDATA;
1147  }
1148  s->hybrid_maxclip = ((1LL << (orig_bpp - 1)) - 1);
1149  s->hybrid_minclip = ((-1UL << (orig_bpp - 1)));
1150  s->CRC = bytestream2_get_le32(&gb);
1151 
1152  // parse metadata blocks
1153  while (bytestream2_get_bytes_left(&gb)) {
1154  id = bytestream2_get_byte(&gb);
1155  size = bytestream2_get_byte(&gb);
1156  if (id & WP_IDF_LONG)
1157  size |= (bytestream2_get_le16u(&gb)) << 8;
1158  size <<= 1; // size is specified in words
1159  ssize = size;
1160  if (id & WP_IDF_ODD)
1161  size--;
1162  if (size < 0) {
1163  av_log(avctx, AV_LOG_ERROR,
1164  "Got incorrect block %02X with size %i\n", id, size);
1165  break;
1166  }
1167  if (bytestream2_get_bytes_left(&gb) < ssize) {
1168  av_log(avctx, AV_LOG_ERROR,
1169  "Block size %i is out of bounds\n", size);
1170  break;
1171  }
1172  switch (id & WP_IDF_MASK) {
1173  case WP_ID_DECTERMS:
1174  if (size > MAX_TERMS) {
1175  av_log(avctx, AV_LOG_ERROR, "Too many decorrelation terms\n");
1176  s->terms = 0;
1177  bytestream2_skip(&gb, ssize);
1178  continue;
1179  }
1180  s->terms = size;
1181  for (i = 0; i < s->terms; i++) {
1182  uint8_t val = bytestream2_get_byte(&gb);
1183  s->decorr[s->terms - i - 1].value = (val & 0x1F) - 5;
1184  s->decorr[s->terms - i - 1].delta = val >> 5;
1185  }
1186  got_terms = 1;
1187  break;
1188  case WP_ID_DECWEIGHTS:
1189  if (!got_terms) {
1190  av_log(avctx, AV_LOG_ERROR, "No decorrelation terms met\n");
1191  continue;
1192  }
1193  weights = size >> s->stereo_in;
1194  if (weights > MAX_TERMS || weights > s->terms) {
1195  av_log(avctx, AV_LOG_ERROR, "Too many decorrelation weights\n");
1196  bytestream2_skip(&gb, ssize);
1197  continue;
1198  }
1199  for (i = 0; i < weights; i++) {
1200  t = (int8_t)bytestream2_get_byte(&gb);
1201  s->decorr[s->terms - i - 1].weightA = t * (1 << 3);
1202  if (s->decorr[s->terms - i - 1].weightA > 0)
1203  s->decorr[s->terms - i - 1].weightA +=
1204  (s->decorr[s->terms - i - 1].weightA + 64) >> 7;
1205  if (s->stereo_in) {
1206  t = (int8_t)bytestream2_get_byte(&gb);
1207  s->decorr[s->terms - i - 1].weightB = t * (1 << 3);
1208  if (s->decorr[s->terms - i - 1].weightB > 0)
1209  s->decorr[s->terms - i - 1].weightB +=
1210  (s->decorr[s->terms - i - 1].weightB + 64) >> 7;
1211  }
1212  }
1213  got_weights = 1;
1214  break;
1215  case WP_ID_DECSAMPLES:
1216  if (!got_terms) {
1217  av_log(avctx, AV_LOG_ERROR, "No decorrelation terms met\n");
1218  continue;
1219  }
1220  t = 0;
1221  for (i = s->terms - 1; (i >= 0) && (t < size); i--) {
1222  if (s->decorr[i].value > 8) {
1223  s->decorr[i].samplesA[0] =
1224  wp_exp2(bytestream2_get_le16(&gb));
1225  s->decorr[i].samplesA[1] =
1226  wp_exp2(bytestream2_get_le16(&gb));
1227 
1228  if (s->stereo_in) {
1229  s->decorr[i].samplesB[0] =
1230  wp_exp2(bytestream2_get_le16(&gb));
1231  s->decorr[i].samplesB[1] =
1232  wp_exp2(bytestream2_get_le16(&gb));
1233  t += 4;
1234  }
1235  t += 4;
1236  } else if (s->decorr[i].value < 0) {
1237  s->decorr[i].samplesA[0] =
1238  wp_exp2(bytestream2_get_le16(&gb));
1239  s->decorr[i].samplesB[0] =
1240  wp_exp2(bytestream2_get_le16(&gb));
1241  t += 4;
1242  } else {
1243  for (j = 0; j < s->decorr[i].value; j++) {
1244  s->decorr[i].samplesA[j] =
1245  wp_exp2(bytestream2_get_le16(&gb));
1246  if (s->stereo_in) {
1247  s->decorr[i].samplesB[j] =
1248  wp_exp2(bytestream2_get_le16(&gb));
1249  }
1250  }
1251  t += s->decorr[i].value * 2 * (s->stereo_in + 1);
1252  }
1253  }
1254  got_samples = 1;
1255  break;
1256  case WP_ID_ENTROPY:
1257  if (size != 6 * (s->stereo_in + 1)) {
1258  av_log(avctx, AV_LOG_ERROR,
1259  "Entropy vars size should be %i, got %i.\n",
1260  6 * (s->stereo_in + 1), size);
1261  bytestream2_skip(&gb, ssize);
1262  continue;
1263  }
1264  for (j = 0; j <= s->stereo_in; j++)
1265  for (i = 0; i < 3; i++) {
1266  s->ch[j].median[i] = wp_exp2(bytestream2_get_le16(&gb));
1267  }
1268  got_entropy = 1;
1269  break;
1270  case WP_ID_HYBRID:
1271  if (s->hybrid_bitrate) {
1272  for (i = 0; i <= s->stereo_in; i++) {
1273  s->ch[i].slow_level = wp_exp2(bytestream2_get_le16(&gb));
1274  size -= 2;
1275  }
1276  }
1277  for (i = 0; i < (s->stereo_in + 1); i++) {
1278  s->ch[i].bitrate_acc = bytestream2_get_le16(&gb) << 16;
1279  size -= 2;
1280  }
1281  if (size > 0) {
1282  for (i = 0; i < (s->stereo_in + 1); i++) {
1283  s->ch[i].bitrate_delta =
1284  wp_exp2((int16_t)bytestream2_get_le16(&gb));
1285  }
1286  } else {
1287  for (i = 0; i < (s->stereo_in + 1); i++)
1288  s->ch[i].bitrate_delta = 0;
1289  }
1290  got_hybrid = 1;
1291  break;
1292  case WP_ID_INT32INFO: {
1293  uint8_t val[4];
1294  if (size != 4) {
1295  av_log(avctx, AV_LOG_ERROR,
1296  "Invalid INT32INFO, size = %i\n",
1297  size);
1298  bytestream2_skip(&gb, ssize - 4);
1299  continue;
1300  }
1301  bytestream2_get_buffer(&gb, val, 4);
1302  if (val[0] > 30) {
1303  av_log(avctx, AV_LOG_ERROR,
1304  "Invalid INT32INFO, extra_bits = %d (> 30)\n", val[0]);
1305  continue;
1306  } else if (val[0]) {
1307  s->extra_bits = val[0];
1308  } else if (val[1]) {
1309  s->shift = val[1];
1310  } else if (val[2]) {
1311  s->and = s->or = 1;
1312  s->shift = val[2];
1313  } else if (val[3]) {
1314  s->and = 1;
1315  s->shift = val[3];
1316  }
1317  if (s->shift > 31) {
1318  av_log(avctx, AV_LOG_ERROR,
1319  "Invalid INT32INFO, shift = %d (> 31)\n", s->shift);
1320  s->and = s->or = s->shift = 0;
1321  continue;
1322  }
1323  /* original WavPack decoder forces 32-bit lossy sound to be treated
1324  * as 24-bit one in order to have proper clipping */
1325  if (s->hybrid && bpp == 4 && s->post_shift < 8 && s->shift > 8) {
1326  s->post_shift += 8;
1327  s->shift -= 8;
1328  s->hybrid_maxclip >>= 8;
1329  s->hybrid_minclip >>= 8;
1330  }
1331  break;
1332  }
1333  case WP_ID_FLOATINFO:
1334  if (size != 4) {
1335  av_log(avctx, AV_LOG_ERROR,
1336  "Invalid FLOATINFO, size = %i\n", size);
1337  bytestream2_skip(&gb, ssize);
1338  continue;
1339  }
1340  s->float_flag = bytestream2_get_byte(&gb);
1341  s->float_shift = bytestream2_get_byte(&gb);
1342  s->float_max_exp = bytestream2_get_byte(&gb);
1343  if (s->float_shift > 31) {
1344  av_log(avctx, AV_LOG_ERROR,
1345  "Invalid FLOATINFO, shift = %d (> 31)\n", s->float_shift);
1346  s->float_shift = 0;
1347  continue;
1348  }
1349  got_float = 1;
1350  bytestream2_skip(&gb, 1);
1351  break;
1352  case WP_ID_DATA:
1353  if ((ret = init_get_bits8(&s->gb, gb.buffer, size)) < 0)
1354  return ret;
1355  bytestream2_skip(&gb, size);
1356  got_pcm = 1;
1357  break;
1358  case WP_ID_DSD_DATA:
1359  if (size < 2) {
1360  av_log(avctx, AV_LOG_ERROR, "Invalid DSD_DATA, size = %i\n",
1361  size);
1362  bytestream2_skip(&gb, ssize);
1363  continue;
1364  }
1365  rate_x = bytestream2_get_byte(&gb);
1366  if (rate_x > 30)
1367  return AVERROR_INVALIDDATA;
1368  rate_x = 1 << rate_x;
1369  dsd_mode = bytestream2_get_byte(&gb);
1370  if (dsd_mode && dsd_mode != 1 && dsd_mode != 3) {
1371  av_log(avctx, AV_LOG_ERROR, "Invalid DSD encoding mode: %d\n",
1372  dsd_mode);
1373  return AVERROR_INVALIDDATA;
1374  }
1375  bytestream2_init(&s->gbyte, gb.buffer, size-2);
1376  bytestream2_skip(&gb, size-2);
1377  got_dsd = 1;
1378  break;
1379  case WP_ID_EXTRABITS:
1380  if (size <= 4) {
1381  av_log(avctx, AV_LOG_ERROR, "Invalid EXTRABITS, size = %i\n",
1382  size);
1383  bytestream2_skip(&gb, size);
1384  continue;
1385  }
1386  if ((ret = init_get_bits8(&s->gb_extra_bits, gb.buffer, size)) < 0)
1387  return ret;
1388  s->crc_extra_bits = get_bits_long(&s->gb_extra_bits, 32);
1389  bytestream2_skip(&gb, size);
1390  s->got_extra_bits = 1;
1391  break;
1392  case WP_ID_CHANINFO:
1393  if (size <= 1) {
1394  av_log(avctx, AV_LOG_ERROR,
1395  "Insufficient channel information\n");
1396  return AVERROR_INVALIDDATA;
1397  }
1398  chan = bytestream2_get_byte(&gb);
1399  switch (size - 2) {
1400  case 0:
1401  chmask = bytestream2_get_byte(&gb);
1402  break;
1403  case 1:
1404  chmask = bytestream2_get_le16(&gb);
1405  break;
1406  case 2:
1407  chmask = bytestream2_get_le24(&gb);
1408  break;
1409  case 3:
1410  chmask = bytestream2_get_le32(&gb);
1411  break;
1412  case 4:
1413  size = bytestream2_get_byte(&gb);
1414  chan |= (bytestream2_get_byte(&gb) & 0xF) << 8;
1415  chan += 1;
1416  if (avctx->channels != chan)
1417  av_log(avctx, AV_LOG_WARNING, "%i channels signalled"
1418  " instead of %i.\n", chan, avctx->channels);
1419  chmask = bytestream2_get_le24(&gb);
1420  break;
1421  case 5:
1422  size = bytestream2_get_byte(&gb);
1423  chan |= (bytestream2_get_byte(&gb) & 0xF) << 8;
1424  chan += 1;
1425  if (avctx->channels != chan)
1426  av_log(avctx, AV_LOG_WARNING, "%i channels signalled"
1427  " instead of %i.\n", chan, avctx->channels);
1428  chmask = bytestream2_get_le32(&gb);
1429  break;
1430  default:
1431  av_log(avctx, AV_LOG_ERROR, "Invalid channel info size %d\n",
1432  size);
1433  chan = avctx->channels;
1434  chmask = avctx->channel_layout;
1435  }
1436  break;
1437  case WP_ID_SAMPLE_RATE:
1438  if (size != 3) {
1439  av_log(avctx, AV_LOG_ERROR, "Invalid custom sample rate.\n");
1440  return AVERROR_INVALIDDATA;
1441  }
1442  sample_rate = bytestream2_get_le24(&gb);
1443  break;
1444  default:
1445  bytestream2_skip(&gb, size);
1446  }
1447  if (id & WP_IDF_ODD)
1448  bytestream2_skip(&gb, 1);
1449  }
1450 
1451  if (got_pcm) {
1452  if (!got_terms) {
1453  av_log(avctx, AV_LOG_ERROR, "No block with decorrelation terms\n");
1454  return AVERROR_INVALIDDATA;
1455  }
1456  if (!got_weights) {
1457  av_log(avctx, AV_LOG_ERROR, "No block with decorrelation weights\n");
1458  return AVERROR_INVALIDDATA;
1459  }
1460  if (!got_samples) {
1461  av_log(avctx, AV_LOG_ERROR, "No block with decorrelation samples\n");
1462  return AVERROR_INVALIDDATA;
1463  }
1464  if (!got_entropy) {
1465  av_log(avctx, AV_LOG_ERROR, "No block with entropy info\n");
1466  return AVERROR_INVALIDDATA;
1467  }
1468  if (s->hybrid && !got_hybrid) {
1469  av_log(avctx, AV_LOG_ERROR, "Hybrid config not found\n");
1470  return AVERROR_INVALIDDATA;
1471  }
1472  if (!got_float && sample_fmt == AV_SAMPLE_FMT_FLTP) {
1473  av_log(avctx, AV_LOG_ERROR, "Float information not found\n");
1474  return AVERROR_INVALIDDATA;
1475  }
1476  if (s->got_extra_bits && sample_fmt != AV_SAMPLE_FMT_FLTP) {
1477  const int size = get_bits_left(&s->gb_extra_bits);
1478  const int wanted = s->samples * s->extra_bits << s->stereo_in;
1479  if (size < wanted) {
1480  av_log(avctx, AV_LOG_ERROR, "Too small EXTRABITS\n");
1481  s->got_extra_bits = 0;
1482  }
1483  }
1484  }
1485 
1486  if (!got_pcm && !got_dsd) {
1487  av_log(avctx, AV_LOG_ERROR, "Packed samples not found\n");
1488  return AVERROR_INVALIDDATA;
1489  }
1490 
1491  if ((got_pcm && wc->modulation != MODULATION_PCM) ||
1492  (got_dsd && wc->modulation != MODULATION_DSD)) {
1493  av_log(avctx, AV_LOG_ERROR, "Invalid PCM/DSD mix encountered\n");
1494  return AVERROR_INVALIDDATA;
1495  }
1496 
1497  if (!wc->ch_offset) {
1498  int new_channels = avctx->channels;
1499  uint64_t new_chmask = avctx->channel_layout;
1500  int new_samplerate;
1501  int sr = (s->frame_flags >> 23) & 0xf;
1502  if (sr == 0xf) {
1503  if (!sample_rate) {
1504  av_log(avctx, AV_LOG_ERROR, "Custom sample rate missing.\n");
1505  return AVERROR_INVALIDDATA;
1506  }
1507  new_samplerate = sample_rate;
1508  } else
1509  new_samplerate = wv_rates[sr];
1510 
1511  if (new_samplerate * (uint64_t)rate_x > INT_MAX)
1512  return AVERROR_INVALIDDATA;
1513  new_samplerate *= rate_x;
1514 
1515  if (multiblock) {
1516  if (chan)
1517  new_channels = chan;
1518  if (chmask)
1519  new_chmask = chmask;
1520  } else {
1521  new_channels = s->stereo ? 2 : 1;
1522  new_chmask = s->stereo ? AV_CH_LAYOUT_STEREO :
1524  }
1525 
1526  if (new_chmask &&
1527  av_get_channel_layout_nb_channels(new_chmask) != new_channels) {
1528  av_log(avctx, AV_LOG_ERROR, "Channel mask does not match the channel count\n");
1529  return AVERROR_INVALIDDATA;
1530  }
1531 
1532  /* clear DSD state if stream properties change */
1533  if (new_channels != wc->dsd_channels ||
1534  new_chmask != avctx->channel_layout ||
1535  new_samplerate != avctx->sample_rate ||
1536  !!got_dsd != !!wc->dsdctx) {
1537  ret = wv_dsd_reset(wc, got_dsd ? new_channels : 0);
1538  if (ret < 0) {
1539  av_log(avctx, AV_LOG_ERROR, "Error reinitializing the DSD context\n");
1540  return ret;
1541  }
1542  ff_thread_release_buffer(avctx, &wc->curr_frame);
1543  }
1544  avctx->channels = new_channels;
1545  avctx->channel_layout = new_chmask;
1546  avctx->sample_rate = new_samplerate;
1547  avctx->sample_fmt = sample_fmt;
1548  avctx->bits_per_raw_sample = orig_bpp;
1549 
1550  ff_thread_release_buffer(avctx, &wc->prev_frame);
1552 
1553  /* get output buffer */
1554  wc->curr_frame.f->nb_samples = s->samples;
1555  if ((ret = ff_thread_get_buffer(avctx, &wc->curr_frame, AV_GET_BUFFER_FLAG_REF)) < 0)
1556  return ret;
1557 
1558  wc->frame = wc->curr_frame.f;
1559  ff_thread_finish_setup(avctx);
1560  }
1561 
1562  if (wc->ch_offset + s->stereo >= avctx->channels) {
1563  av_log(avctx, AV_LOG_WARNING, "Too many channels coded in a packet.\n");
1564  return ((avctx->err_recognition & AV_EF_EXPLODE) || !wc->ch_offset) ? AVERROR_INVALIDDATA : 0;
1565  }
1566 
1567  samples_l = wc->frame->extended_data[wc->ch_offset];
1568  if (s->stereo)
1569  samples_r = wc->frame->extended_data[wc->ch_offset + 1];
1570 
1571  wc->ch_offset += 1 + s->stereo;
1572 
1573  if (s->stereo_in) {
1574  if (got_dsd) {
1575  if (dsd_mode == 3) {
1576  ret = wv_unpack_dsd_high(s, samples_l, samples_r);
1577  } else if (dsd_mode == 1) {
1578  ret = wv_unpack_dsd_fast(s, samples_l, samples_r);
1579  } else {
1580  ret = wv_unpack_dsd_copy(s, samples_l, samples_r);
1581  }
1582  } else {
1583  ret = wv_unpack_stereo(s, &s->gb, samples_l, samples_r, avctx->sample_fmt);
1584  }
1585  if (ret < 0)
1586  return ret;
1587  } else {
1588  if (got_dsd) {
1589  if (dsd_mode == 3) {
1590  ret = wv_unpack_dsd_high(s, samples_l, NULL);
1591  } else if (dsd_mode == 1) {
1592  ret = wv_unpack_dsd_fast(s, samples_l, NULL);
1593  } else {
1594  ret = wv_unpack_dsd_copy(s, samples_l, NULL);
1595  }
1596  } else {
1597  ret = wv_unpack_mono(s, &s->gb, samples_l, avctx->sample_fmt);
1598  }
1599  if (ret < 0)
1600  return ret;
1601 
1602  if (s->stereo)
1603  memcpy(samples_r, samples_l, bpp * s->samples);
1604  }
1605 
1606  return 0;
1607 }
1608 
1610 {
1611  WavpackContext *s = avctx->priv_data;
1612 
1613  wv_dsd_reset(s, 0);
1614 }
1615 
1616 static int dsd_channel(AVCodecContext *avctx, void *frmptr, int jobnr, int threadnr)
1617 {
1618  WavpackContext *s = avctx->priv_data;
1619  AVFrame *frame = frmptr;
1620 
1621  ff_dsd2pcm_translate (&s->dsdctx [jobnr], s->samples, 0,
1622  (uint8_t *)frame->extended_data[jobnr], 4,
1623  (float *)frame->extended_data[jobnr], 1);
1624 
1625  return 0;
1626 }
1627 
1628 static int wavpack_decode_frame(AVCodecContext *avctx, void *data,
1629  int *got_frame_ptr, AVPacket *avpkt)
1630 {
1631  WavpackContext *s = avctx->priv_data;
1632  const uint8_t *buf = avpkt->data;
1633  int buf_size = avpkt->size;
1634  int frame_size, ret, frame_flags;
1635 
1636  if (avpkt->size <= WV_HEADER_SIZE)
1637  return AVERROR_INVALIDDATA;
1638 
1639  s->frame = NULL;
1640  s->block = 0;
1641  s->ch_offset = 0;
1642 
1643  /* determine number of samples */
1644  s->samples = AV_RL32(buf + 20);
1645  frame_flags = AV_RL32(buf + 24);
1646  if (s->samples <= 0 || s->samples > WV_MAX_SAMPLES) {
1647  av_log(avctx, AV_LOG_ERROR, "Invalid number of samples: %d\n",
1648  s->samples);
1649  return AVERROR_INVALIDDATA;
1650  }
1651 
1652  s->modulation = (frame_flags & WV_DSD_DATA) ? MODULATION_DSD : MODULATION_PCM;
1653 
1654  while (buf_size > WV_HEADER_SIZE) {
1655  frame_size = AV_RL32(buf + 4) - 12;
1656  buf += 20;
1657  buf_size -= 20;
1658  if (frame_size <= 0 || frame_size > buf_size) {
1659  av_log(avctx, AV_LOG_ERROR,
1660  "Block %d has invalid size (size %d vs. %d bytes left)\n",
1661  s->block, frame_size, buf_size);
1663  goto error;
1664  }
1665  if ((ret = wavpack_decode_block(avctx, s->block, buf, frame_size)) < 0)
1666  goto error;
1667  s->block++;
1668  buf += frame_size;
1669  buf_size -= frame_size;
1670  }
1671 
1672  if (s->ch_offset != avctx->channels) {
1673  av_log(avctx, AV_LOG_ERROR, "Not enough channels coded in a packet.\n");
1675  goto error;
1676  }
1677 
1678  ff_thread_await_progress(&s->prev_frame, INT_MAX, 0);
1679  ff_thread_release_buffer(avctx, &s->prev_frame);
1680 
1681  if (s->modulation == MODULATION_DSD)
1682  avctx->execute2(avctx, dsd_channel, s->frame, NULL, avctx->channels);
1683 
1684  ff_thread_report_progress(&s->curr_frame, INT_MAX, 0);
1685 
1686  if ((ret = av_frame_ref(data, s->frame)) < 0)
1687  return ret;
1688 
1689  *got_frame_ptr = 1;
1690 
1691  return avpkt->size;
1692 
1693 error:
1694  if (s->frame) {
1695  ff_thread_await_progress(&s->prev_frame, INT_MAX, 0);
1696  ff_thread_release_buffer(avctx, &s->prev_frame);
1697  ff_thread_report_progress(&s->curr_frame, INT_MAX, 0);
1698  }
1699 
1700  return ret;
1701 }
1702 
1704  .name = "wavpack",
1705  .long_name = NULL_IF_CONFIG_SMALL("WavPack"),
1706  .type = AVMEDIA_TYPE_AUDIO,
1707  .id = AV_CODEC_ID_WAVPACK,
1708  .priv_data_size = sizeof(WavpackContext),
1710  .close = wavpack_decode_end,
1714  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_FRAME_THREADS |
1717 };
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:29
WavpackFrameContext::joint
int joint
Definition: wavpack.c:70
WV_HYBRID_BITRATE
#define WV_HYBRID_BITRATE
Definition: wavpack.h:42
WavpackFrameContext::decorr
Decorr decorr[MAX_TERMS]
Definition: wavpack.c:78
AVCodec
AVCodec.
Definition: codec.h:190
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
WavpackContext::avctx
AVCodecContext * avctx
Definition: wavpack.c:101
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
L2
F H1 F F H1 F F F F H1<-F-------F-------F v v v H2 H3 H2 ^ ^ ^ F-------F-------F-> H1<-F-------F-------F|||||||||F H1 F|||||||||F H1 Funavailable fullpel samples(outside the picture for example) shall be equalto the closest available fullpel sampleSmaller pel interpolation:--------------------------if diag_mc is set then points which lie on a line between 2 vertically, horizontally or diagonally adjacent halfpel points shall be interpolatedlinearly with rounding to nearest and halfway values rounded up.points which lie on 2 diagonals at the same time should only use the onediagonal not containing the fullpel point F--> O q O<--h1-> O q O<--F v \/v \/v O O O O O O O|/|\|q q q q q|/|\|O O O O O O O ^/\ ^/\ ^ h2--> O q O<--h3-> O q O<--h2 v \/v \/v O O O O O O O|\|/|q q q q q|\|/|O O O O O O O ^/\ ^/\ ^ F--> O q O<--h1-> O q O<--Fthe remaining points shall be bilinearly interpolated from theup to 4 surrounding halfpel and fullpel points, again rounding should be tonearest and halfway values rounded upcompliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chromainterpolation at leastOverlapped block motion compensation:-------------------------------------FIXMELL band prediction:===================Each sample in the LL0 subband is predicted by the median of the left, top andleft+top-topleft samples, samples outside the subband shall be considered tobe 0. To reverse this prediction in the decoder apply the following.for(y=0;y< height;y++){ for(x=0;x< width;x++){ sample[y][x]+=median(sample[y-1][x], sample[y][x-1], sample[y-1][x]+sample[y][x-1]-sample[y-1][x-1]);}}sample[-1][ *]=sample[ *][-1]=0;width, height here are the width and height of the LL0 subband not of the finalvideoDequantization:===============FIXMEWavelet Transform:==================Snow supports 2 wavelet transforms, the symmetric biorthogonal 5/3 integertransform and an integer approximation of the symmetric biorthogonal 9/7daubechies wavelet.2D IDWT(inverse discrete wavelet transform) --------------------------------------------The 2D IDWT applies a 2D filter recursively, each time combining the4 lowest frequency subbands into a single subband until only 1 subbandremains.The 2D filter is done by first applying a 1D filter in the vertical directionand then applying it in the horizontal one. --------------- --------------- --------------- ---------------|LL0|HL0|||||||||||||---+---|HL1||L0|H0|HL1||LL1|HL1|||||LH0|HH0|||||||||||||-------+-------|-> L1 H1 LH1 HH1 LH1 HH1 LH1 HH1 L2
Definition: snow.txt:554
WP_ID_DSD_DATA
@ WP_ID_DSD_DATA
Definition: wavpack.h:81
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
WP_ID_HYBRID
@ WP_ID_HYBRID
Definition: wavpack.h:73
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1186
FFSWAP
#define FFSWAP(type, a, b)
Definition: common.h:99
WP_ID_FLOATINFO
@ WP_ID_FLOATINFO
Definition: wavpack.h:75
GetByteContext
Definition: bytestream.h:33
u
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:262
AVBufferRef::data
uint8_t * data
The data buffer.
Definition: buffer.h:89
WV_MAX_SAMPLES
#define WV_MAX_SAMPLES
Definition: wavpack.h:57
filters
static const struct PPFilter filters[]
Definition: postprocess.c:134
MAX_TERMS
#define MAX_TERMS
Definition: wavpack.h:27
wv_unpack_dsd_high
static int wv_unpack_dsd_high(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
Definition: wavpack.c:441
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
WavpackFrameContext::CRC
uint32_t CRC
Definition: wavpack.c:71
AVCodecContext::err_recognition
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:1655
wavpack_decode_flush
static void wavpack_decode_flush(AVCodecContext *avctx)
Definition: wavpack.c:1609
get_bits_long
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:546
DECAY
#define DECAY
Definition: wavpack.c:49
MODULATION_PCM
@ MODULATION_PCM
Definition: wavpack.c:62
WP_ID_DECWEIGHTS
@ WP_ID_DECWEIGHTS
Definition: wavpack.h:70
WV_FLT_SHIFT_ONES
#define WV_FLT_SHIFT_ONES
Definition: wavpack.h:51
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
WavpackContext::prev_frame
ThreadFrame prev_frame
Definition: wavpack.c:111
WP_IDF_LONG
@ WP_IDF_LONG
Definition: wavpack.h:63
R
#define R
Definition: huffyuvdsp.h:34
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:355
WavpackFrameContext::post_shift
int post_shift
Definition: wavpack.c:82
WV_MONO
@ WV_MONO
Definition: wvdec.c:32
table
static const uint16_t table[]
Definition: prosumer.c:206
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
data
const char data[16]
Definition: mxf.c:91
WV_FLT_ZERO_SIGN
#define WV_FLT_ZERO_SIGN
Definition: wavpack.h:55
av_buffer_allocz
AVBufferRef * av_buffer_allocz(int size)
Same as av_buffer_alloc(), except the returned buffer will be initialized to zero.
Definition: buffer.c:83
WavpackFrameContext::stereo
int stereo
Definition: wavpack.c:69
base
uint8_t base
Definition: vp3data.h:202
PRECISION_USE
#define PRECISION_USE
Definition: wavpack.c:53
WvChannel
Definition: wavpack.h:96
dsd.h
WV_FLOAT_DATA
#define WV_FLOAT_DATA
Definition: wavpack.h:35
thread.h
ff_thread_await_progress
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before ff_thread_await_progress() has been called on them. reget_buffer() and buffer age optimizations no longer work. *The contents of buffers must not be written to after ff_thread_report_progress() has been called on them. This includes draw_edges(). Porting codecs to frame threading
sample_rate
sample_rate
Definition: ffmpeg_filter.c:192
ThreadFrame::f
AVFrame * f
Definition: thread.h:35
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:314
MODULATION_DSD
@ MODULATION_DSD
Definition: wavpack.c:63
WavpackContext::frame
AVFrame * frame
Definition: wavpack.c:110
WavpackContext::fdec_num
int fdec_num
Definition: wavpack.c:104
bit
#define bit(string, value)
Definition: cbs_mpeg2.c:58
S
#define S(s, c, i)
Definition: flacdsp_template.c:46
A
#define A(x)
Definition: vp56_arith.h:28
wv_unpack_dsd_copy
static int wv_unpack_dsd_copy(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
Definition: wavpack.c:741
WavpackContext
Definition: wavpack.c:100
MAX_HISTORY_BINS
#define MAX_HISTORY_BINS
Definition: wavpack.c:58
bytestream2_skip
static av_always_inline void bytestream2_skip(GetByteContext *g, unsigned int size)
Definition: bytestream.h:164
WP_ID_DATA
@ WP_ID_DATA
Definition: wavpack.h:77
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
update_error_limit
static int update_error_limit(WavpackFrameContext *ctx)
Definition: wavpack.c:135
Decorr
Definition: wavpack.h:85
WP_ID_SAMPLE_RATE
@ WP_ID_SAMPLE_RATE
Definition: wavpack.h:82
U
#define U(x)
Definition: vp56_arith.h:37
WavpackFrameContext::got_extra_bits
int got_extra_bits
Definition: wavpack.c:73
WP_ID_ENTROPY
@ WP_ID_ENTROPY
Definition: wavpack.h:72
WavpackFrameContext::ch
WvChannel ch[2]
Definition: wavpack.c:88
GetBitContext
Definition: get_bits.h:61
ff_thread_get_buffer
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have so the codec calls ff_thread_report set FF_CODEC_CAP_ALLOCATE_PROGRESS in AVCodec caps_internal and use ff_thread_get_buffer() to allocate frames. The frames must then be freed with ff_thread_release_buffer(). Otherwise decode directly into the user-supplied frames. Call ff_thread_report_progress() after some part of the current picture has decoded. A good place to put this is where draw_horiz_band() is called - add this if it isn 't called anywhere
Modulation
Modulation
Definition: wavpack.c:61
val
static double val(void *priv, double ch)
Definition: aeval.c:76
type
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type
Definition: writing_filters.txt:86
wavpack_decode_block
static int wavpack_decode_block(AVCodecContext *avctx, int block_no, const uint8_t *buf, int buf_size)
Definition: wavpack.c:1080
WavpackContext::curr_frame
ThreadFrame curr_frame
Definition: wavpack.c:111
MAX_BIN_BYTES
#define MAX_BIN_BYTES
Definition: wavpack.c:59
ff_wavpack_decoder
AVCodec ff_wavpack_decoder
Definition: wavpack.c:1703
WV_DSD_DATA
#define WV_DSD_DATA
Definition: wavpack.h:38
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:86
wp_log2
static av_always_inline int wp_log2(uint32_t val)
Definition: wavpack.h:182
MAX_HISTORY_BITS
#define MAX_HISTORY_BITS
Definition: wavpack.c:57
av_frame_alloc
AVFrame * av_frame_alloc(void)
Allocate an AVFrame and set its fields to default values.
Definition: frame.c:190
mult
static int16_t mult(Float11 *f1, Float11 *f2)
Definition: g726.c:55
WavpackFrameContext::shift
int shift
Definition: wavpack.c:81
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold
#define av_cold
Definition: attributes.h:90
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
ff_thread_report_progress
void ff_thread_report_progress(ThreadFrame *f, int n, int field)
Notify later decoding threads when part of their reference picture is ready.
Definition: pthread_frame.c:568
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
PRECISION
#define PRECISION
Definition: wavpack.c:51
s
#define s(width, name)
Definition: cbs_vp9.c:257
AV_GET_BUFFER_FLAG_REF
#define AV_GET_BUFFER_FLAG_REF
The decoder will keep a reference to the frame and may reuse it later.
Definition: avcodec.h:509
frame_size
int frame_size
Definition: mxfenc.c:2137
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
WP_ID_EXTRABITS
@ WP_ID_EXTRABITS
Definition: wavpack.h:79
wv_rates
static const int wv_rates[16]
Definition: wavpack.h:121
GetByteContext::buffer
const uint8_t * buffer
Definition: bytestream.h:34
wp_exp2
static av_always_inline int wp_exp2(int16_t val)
Definition: wavpack.h:165
WavpackFrameContext::summed_probabilities
uint16_t summed_probabilities[MAX_HISTORY_BINS][256]
Definition: wavpack.c:93
AVCodecContext::bits_per_raw_sample
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:1757
ctx
AVFormatContext * ctx
Definition: movenc.c:48
channels
channels
Definition: aptx.h:33
get_bits.h
WavpackFrameContext::hybrid
int hybrid
Definition: wavpack.c:83
f
#define f(width, name)
Definition: cbs_vp9.c:255
int32_t
int32_t
Definition: audio_convert.c:194
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
if
if(ret)
Definition: filter_design.txt:179
WavpackFrameContext::gb
GetBitContext gb
Definition: wavpack.c:72
AV_CODEC_CAP_FRAME_THREADS
#define AV_CODEC_CAP_FRAME_THREADS
Codec supports frame-level multithreading.
Definition: codec.h:106
WV_FALSE_STEREO
#define WV_FALSE_STEREO
Definition: wavpack.h:37
flush
static void flush(AVCodecContext *avctx)
Definition: aacdec_template.c:500
NULL
#define NULL
Definition: coverity.c:32
WavpackFrameContext::hybrid_minclip
int hybrid_minclip
Definition: wavpack.c:84
av_buffer_unref
void av_buffer_unref(AVBufferRef **buf)
Free a given reference and automatically free the buffer if there are no more references to it.
Definition: buffer.c:125
DSD_BYTE_READY
#define DSD_BYTE_READY(low, high)
Definition: wavpack.c:41
WavpackContext::dsdctx
DSDContext * dsdctx
Definition: wavpack.c:115
WV_HEADER_SIZE
#define WV_HEADER_SIZE
Definition: wavpack.h:30
WV_SINGLE_BLOCK
#define WV_SINGLE_BLOCK
Definition: wavpack.h:49
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
src
#define src
Definition: vp8dsp.c:254
VALUE_ONE
#define VALUE_ONE
Definition: wavpack.c:52
bytestream2_get_buffer
static av_always_inline unsigned int bytestream2_get_buffer(GetByteContext *g, uint8_t *dst, unsigned int size)
Definition: bytestream.h:263
WavpackContext::ch_offset
int ch_offset
Definition: wavpack.c:108
ONLY_IF_THREADS_ENABLED
#define ONLY_IF_THREADS_ENABLED(x)
Define a function with only the non-default version specified.
Definition: internal.h:227
AV_EF_EXPLODE
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:1666
get_unary_0_33
static int get_unary_0_33(GetBitContext *gb)
Get unary code terminated by a 0 with a maximum length of 33.
Definition: unary.h:59
exp
int8_t exp
Definition: eval.c:72
T
#define T(x)
Definition: vp56_arith.h:29
WavpackFrameContext::one
int one
Definition: wavpack.c:79
LEVEL_DECAY
#define LEVEL_DECAY(a)
Definition: wavpack.c:119
index
int index
Definition: gxfenc.c:89
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
av_get_channel_layout_nb_channels
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
Definition: channel_layout.c:220
wv_get_value_integer
static int wv_get_value_integer(WavpackFrameContext *s, uint32_t *crc, unsigned S)
Definition: wavpack.c:303
bytestream2_get_bytes_left
static av_always_inline int bytestream2_get_bytes_left(GetByteContext *g)
Definition: bytestream.h:154
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
WavpackFrameContext::or
int or
Definition: wavpack.c:81
DSDfilters::value
int32_t value
Definition: wavpack.c:437
ff_thread_release_buffer
void ff_thread_release_buffer(AVCodecContext *avctx, ThreadFrame *f)
Wrapper around release_buffer() frame-for multithreaded codecs.
Definition: pthread_frame.c:1006
WavpackFrameContext::gbyte
GetByteContext gbyte
Definition: wavpack.c:90
ff_init_dsd_data
av_cold void ff_init_dsd_data(void)
Definition: dsd.c:46
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
AVPacket::size
int size
Definition: packet.h:356
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:188
id
enum AVCodecID id
Definition: extract_extradata_bsf.c:332
av_frame_ref
int av_frame_ref(AVFrame *dst, const AVFrame *src)
Set up a new reference to the data described by the source frame.
Definition: frame.c:444
wv_check_crc
static int wv_check_crc(WavpackFrameContext *s, uint32_t crc, uint32_t crc_extra_bits)
Definition: wavpack.c:401
sp
#define sp
Definition: regdef.h:63
wavpack_decode_frame
static int wavpack_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: wavpack.c:1628
WavpackFrameContext::float_flag
int float_flag
Definition: wavpack.c:85
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
R2
#define R2
Definition: simple_idct.c:173
size
int size
Definition: twinvq_data.h:11134
WavpackFrameContext::crc_extra_bits
uint32_t crc_extra_bits
Definition: wavpack.c:74
DSDfilters
Definition: wavpack.c:436
WP_IDF_ODD
@ WP_IDF_ODD
Definition: wavpack.h:62
WavpackContext::block
int block
Definition: wavpack.c:106
DSDfilters::byte
unsigned int byte
Definition: wavpack.c:438
WavpackFrameContext::hybrid_maxclip
int hybrid_maxclip
Definition: wavpack.c:84
buffer.h
split
static char * split(char *message, char delim)
Definition: af_channelmap.c:81
RATE_S
#define RATE_S
Definition: wavpack.c:55
init_ptable
static void init_ptable(int *table, int rate_i, int rate_s)
Definition: wavpack.c:416
AV_CODEC_CAP_SLICE_THREADS
#define AV_CODEC_CAP_SLICE_THREADS
Codec supports slice-based (or partition-based) multithreading.
Definition: codec.h:110
WP_ID_DECTERMS
@ WP_ID_DECTERMS
Definition: wavpack.h:69
WV_FLT_SHIFT_SENT
#define WV_FLT_SHIFT_SENT
Definition: wavpack.h:53
unary.h
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
WavpackContext::fdec
WavpackFrameContext * fdec[WV_MAX_FRAME_DECODERS]
Definition: wavpack.c:103
wv_unpack_stereo
static int wv_unpack_stereo(WavpackFrameContext *s, GetBitContext *gb, void *dst_l, void *dst_r, const int type)
Definition: wavpack.c:773
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
wv_unpack_mono
static int wv_unpack_mono(WavpackFrameContext *s, GetBitContext *gb, void *dst, const int type)
Definition: wavpack.c:904
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
WavpackContext::dsd_ref
AVBufferRef * dsd_ref
Definition: wavpack.c:114
WavpackFrameContext::float_max_exp
int float_max_exp
Definition: wavpack.c:87
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
code
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
Definition: filter_design.txt:178
wavpack_decode_end
static av_cold int wavpack_decode_end(AVCodecContext *avctx)
Definition: wavpack.c:1061
WavpackContext::samples
int samples
Definition: wavpack.c:107
av_get_bytes_per_sample
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
FF_CODEC_CAP_INIT_CLEANUP
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:48
WavpackFrameContext::samples
int samples
Definition: wavpack.c:76
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:347
WavpackFrameContext::and
int and
Definition: wavpack.c:81
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
xf
#define xf(width, name, var, range_min, range_max, subs,...)
Definition: cbs_av1.c:668
av_always_inline
#define av_always_inline
Definition: attributes.h:49
value
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default value
Definition: writing_filters.txt:86
PTABLE_BINS
#define PTABLE_BINS
Definition: wavpack.c:44
wavpack.h
WP_ID_CHANINFO
@ WP_ID_CHANINFO
Definition: wavpack.h:80
WV_FLT_SHIFT_SAME
#define WV_FLT_SHIFT_SAME
Definition: wavpack.h:52
uint8_t
uint8_t
Definition: audio_convert.c:194
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
get_tail
static av_always_inline unsigned get_tail(GetBitContext *gb, int k)
Definition: wavpack.c:121
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:197
update_thread_context
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call have update_thread_context() run it in the next thread. Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little speed gain at this point but it should work. If there are inter-frame dependencies
WavpackFrameContext::zero
int zero
Definition: wavpack.c:79
UP
#define UP
Definition: wavpack.c:47
DSDContext
Per-channel buffer.
Definition: dsd.h:42
avcodec.h
ff_dsd2pcm_translate
void ff_dsd2pcm_translate(DSDContext *s, size_t samples, int lsbf, const uint8_t *src, ptrdiff_t src_stride, float *dst, ptrdiff_t dst_stride)
Definition: dsd.c:55
WavpackFrameContext::value_lookup_buffer
uint8_t value_lookup_buffer[MAX_HISTORY_BINS *MAX_BIN_BYTES]
Definition: wavpack.c:92
WP_ID_DECSAMPLES
@ WP_ID_DECSAMPLES
Definition: wavpack.h:71
ret
ret
Definition: filter_design.txt:187
wv_unpack_dsd_fast
static int wv_unpack_dsd_fast(WavpackFrameContext *s, uint8_t *dst_left, uint8_t *dst_right)
Definition: wavpack.c:580
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
WavpackFrameContext::zeroes
int zeroes
Definition: wavpack.c:79
AV_EF_CRCCHECK
#define AV_EF_CRCCHECK
Verify checksums embedded in the bitstream (could be of either encoded or decoded data,...
Definition: avcodec.h:1663
GET_MED
#define GET_MED(n)
Definition: wavpack.h:103
pos
unsigned int pos
Definition: spdifenc.c:412
checksum
static volatile int checksum
Definition: adler32.c:30
DOWN
#define DOWN
Definition: wavpack.c:48
ff_thread_finish_setup
the pkt_dts and pkt_pts fields in AVFrame will work as usual Restrictions on codec whose streams don t reset across will not work because their bitstreams cannot be decoded in parallel *The contents of buffers must not be read before as well as code calling up to before the decode process starts Call ff_thread_finish_setup() afterwards. If some code can 't be moved
WV_MAX_FRAME_DECODERS
#define WV_MAX_FRAME_DECODERS
Definition: wavpack.c:98
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Definition: avcodec.h:215
AV_RL32
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
Definition: bytestream.h:88
L
#define L(x)
Definition: vp56_arith.h:36
WV_FLT_ZERO_SENT
#define WV_FLT_ZERO_SENT
Definition: wavpack.h:54
WavpackFrameContext::gb_extra_bits
GetBitContext gb_extra_bits
Definition: wavpack.c:75
B
#define B
Definition: huffyuvdsp.h:32
WavpackFrameContext::terms
int terms
Definition: wavpack.c:77
WavpackContext::modulation
Modulation modulation
Definition: wavpack.c:112
WavpackFrameContext
Definition: wavpack.c:66
wavpack_decode_init
static av_cold int wavpack_decode_init(AVCodecContext *avctx)
Definition: wavpack.c:1042
AVCodecContext
main external API structure.
Definition: avcodec.h:526
ThreadFrame
Definition: thread.h:34
WavpackFrameContext::stereo_in
int stereo_in
Definition: wavpack.c:69
channel_layout.h
t2
#define t2
Definition: regdef.h:30
UPDATE_WEIGHT_CLIP
#define UPDATE_WEIGHT_CLIP(weight, delta, samples, in)
Definition: wavpack.h:108
wv_get_value
static int wv_get_value(WavpackFrameContext *ctx, GetBitContext *gb, int channel, int *last)
Definition: wavpack.c:173
WavpackFrameContext::ptable
int ptable[PTABLE_BINS]
Definition: wavpack.c:91
WavpackFrameContext::hybrid_bitrate
int hybrid_bitrate
Definition: wavpack.c:83
wv_get_value_float
static float wv_get_value_float(WavpackFrameContext *s, uint32_t *crc, int S)
Definition: wavpack.c:327
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
av_buffer_ref
AVBufferRef * av_buffer_ref(AVBufferRef *buf)
Create a new reference to an AVBuffer.
Definition: buffer.c:93
factor
static const int factor[16]
Definition: vf_pp7.c:75
ff_thread_ref_frame
int ff_thread_ref_frame(ThreadFrame *dst, ThreadFrame *src)
Definition: utils.c:1885
shift
static int shift(int a, int b)
Definition: sonic.c:82
WV_HYBRID_MODE
#define WV_HYBRID_MODE
Definition: wavpack.h:40
INC_MED
#define INC_MED(n)
Definition: wavpack.h:105
wv_alloc_frame_context
static av_cold int wv_alloc_frame_context(WavpackContext *c)
Definition: wavpack.c:971
AVBufferRef
A reference to a data buffer.
Definition: buffer.h:81
get_bitsz
static av_always_inline int get_bitsz(GetBitContext *s, int n)
Read 0-25 bits.
Definition: get_bits.h:415
FF_CODEC_CAP_ALLOCATE_PROGRESS
#define FF_CODEC_CAP_ALLOCATE_PROGRESS
Definition: internal.h:75
DEC_MED
#define DEC_MED(n)
Definition: wavpack.h:104
PTABLE_MASK
#define PTABLE_MASK
Definition: wavpack.c:45
WavpackFrameContext::probabilities
uint8_t probabilities[MAX_HISTORY_BINS][256]
Definition: wavpack.c:94
WV_JOINT_STEREO
#define WV_JOINT_STEREO
Definition: wavpack.h:33
WavpackFrameContext::frame_flags
int frame_flags
Definition: wavpack.c:68
AVPacket
This structure stores compressed data.
Definition: packet.h:332
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
WavpackFrameContext::extra_bits
int extra_bits
Definition: wavpack.c:80
bytestream.h
bytestream2_init
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:133
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
WavpackFrameContext::avctx
AVCodecContext * avctx
Definition: wavpack.c:67
AV_CODEC_ID_WAVPACK
@ AV_CODEC_ID_WAVPACK
Definition: codec_id.h:435
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
WP_ID_INT32INFO
@ WP_ID_INT32INFO
Definition: wavpack.h:76
wv_dsd_reset
static int wv_dsd_reset(WavpackContext *s, int channels)
Definition: wavpack.c:985
WavpackFrameContext::value_lookup
uint8_t * value_lookup[MAX_HISTORY_BINS]
Definition: wavpack.c:95
WavpackFrameContext::float_shift
int float_shift
Definition: wavpack.c:86
int
int
Definition: ffmpeg_filter.c:192
AVCodecContext::execute2
int(* execute2)(struct AVCodecContext *c, int(*func)(struct AVCodecContext *c2, void *arg, int jobnr, int threadnr), void *arg2, int *ret, int count)
The codec may call this to execute several independent things.
Definition: avcodec.h:1845
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
channel
channel
Definition: ebur128.h:39
WavpackContext::dsd_channels
int dsd_channels
Definition: wavpack.c:116
WP_IDF_MASK
@ WP_IDF_MASK
Definition: wavpack.h:60
dsd_channel
static int dsd_channel(AVCodecContext *avctx, void *frmptr, int jobnr, int threadnr)
Definition: wavpack.c:1616