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mlpdec.c
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1 /*
2  * MLP decoder
3  * Copyright (c) 2007-2008 Ian Caulfield
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * MLP decoder
25  */
26 
27 #include <stdint.h>
28 
29 #include "avcodec.h"
30 #include "libavutil/internal.h"
31 #include "libavutil/intreadwrite.h"
33 #include "get_bits.h"
34 #include "internal.h"
35 #include "libavutil/crc.h"
36 #include "parser.h"
37 #include "mlp_parser.h"
38 #include "mlpdsp.h"
39 #include "mlp.h"
40 #include "config.h"
41 
42 /** number of bits used for VLC lookup - longest Huffman code is 9 */
43 #if ARCH_ARM
44 #define VLC_BITS 5
45 #define VLC_STATIC_SIZE 64
46 #else
47 #define VLC_BITS 9
48 #define VLC_STATIC_SIZE 512
49 #endif
50 
51 typedef struct SubStream {
52  /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
54 
55  //@{
56  /** restart header data */
57  /// The type of noise to be used in the rematrix stage.
58  uint16_t noise_type;
59 
60  /// The index of the first channel coded in this substream.
62  /// The index of the last channel coded in this substream.
64  /// The number of channels input into the rematrix stage.
66  /// For each channel output by the matrix, the output channel to map it to
68  /// The channel layout for this substream
69  uint64_t mask;
70  /// The matrix encoding mode for this substream
72 
73  /// Channel coding parameters for channels in the substream
75 
76  /// The left shift applied to random noise in 0x31ea substreams.
78  /// The current seed value for the pseudorandom noise generator(s).
79  uint32_t noisegen_seed;
80 
81  /// Set if the substream contains extra info to check the size of VLC blocks.
83 
84  /// Bitmask of which parameter sets are conveyed in a decoding parameter block.
86 #define PARAM_BLOCKSIZE (1 << 7)
87 #define PARAM_MATRIX (1 << 6)
88 #define PARAM_OUTSHIFT (1 << 5)
89 #define PARAM_QUANTSTEP (1 << 4)
90 #define PARAM_FIR (1 << 3)
91 #define PARAM_IIR (1 << 2)
92 #define PARAM_HUFFOFFSET (1 << 1)
93 #define PARAM_PRESENCE (1 << 0)
94  //@}
95 
96  //@{
97  /** matrix data */
98 
99  /// Number of matrices to be applied.
101 
102  /// matrix output channel
104 
105  /// Whether the LSBs of the matrix output are encoded in the bitstream.
107  /// Matrix coefficients, stored as 2.14 fixed point.
109  /// Left shift to apply to noise values in 0x31eb substreams.
111  //@}
112 
113  /// Left shift to apply to Huffman-decoded residuals.
115 
116  /// number of PCM samples in current audio block
117  uint16_t blocksize;
118  /// Number of PCM samples decoded so far in this frame.
119  uint16_t blockpos;
120 
121  /// Left shift to apply to decoded PCM values to get final 24-bit output.
123 
124  /// Running XOR of all output samples.
126 
127 } SubStream;
128 
129 typedef struct MLPDecodeContext {
131 
132  /// Current access unit being read has a major sync.
134 
135  /// Size of the major sync unit, in bytes
137 
138  /// Set if a valid major sync block has been read. Otherwise no decoding is possible.
140 
141  /// Number of substreams contained within this stream.
143 
144  /// Index of the last substream to decode - further substreams are skipped.
146 
147  /// Stream needs channel reordering to comply with FFmpeg's channel order
149 
150  /// number of PCM samples contained in each frame
152  /// next power of two above the number of samples in each frame
154 
156 
159 
163 
166 
167 static const uint64_t thd_channel_order[] = {
169  AV_CH_FRONT_CENTER, // C
170  AV_CH_LOW_FREQUENCY, // LFE
175  AV_CH_BACK_CENTER, // Cs
176  AV_CH_TOP_CENTER, // Ts
179  AV_CH_TOP_FRONT_CENTER, // Cvh
180  AV_CH_LOW_FREQUENCY_2, // LFE2
181 };
182 
183 static int mlp_channel_layout_subset(uint64_t channel_layout, uint64_t mask)
184 {
185  return channel_layout && ((channel_layout & mask) == channel_layout);
186 }
187 
188 static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout,
189  int index)
190 {
191  int i;
192 
193  if (av_get_channel_layout_nb_channels(channel_layout) <= index)
194  return 0;
195 
196  for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++)
197  if (channel_layout & thd_channel_order[i] && !index--)
198  return thd_channel_order[i];
199  return 0;
200 }
201 
202 static VLC huff_vlc[3];
203 
204 /** Initialize static data, constant between all invocations of the codec. */
205 
206 static av_cold void init_static(void)
207 {
208  if (!huff_vlc[0].bits) {
209  INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
210  &ff_mlp_huffman_tables[0][0][1], 2, 1,
211  &ff_mlp_huffman_tables[0][0][0], 2, 1, VLC_STATIC_SIZE);
212  INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
213  &ff_mlp_huffman_tables[1][0][1], 2, 1,
214  &ff_mlp_huffman_tables[1][0][0], 2, 1, VLC_STATIC_SIZE);
215  INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
216  &ff_mlp_huffman_tables[2][0][1], 2, 1,
217  &ff_mlp_huffman_tables[2][0][0], 2, 1, VLC_STATIC_SIZE);
218  }
219 
220  ff_mlp_init_crc();
221 }
222 
224  unsigned int substr, unsigned int ch)
225 {
226  SubStream *s = &m->substream[substr];
227  ChannelParams *cp = &s->channel_params[ch];
228  int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
229  int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
230  int32_t sign_huff_offset = cp->huff_offset;
231 
232  if (cp->codebook > 0)
233  sign_huff_offset -= 7 << lsb_bits;
234 
235  if (sign_shift >= 0)
236  sign_huff_offset -= 1 << sign_shift;
237 
238  return sign_huff_offset;
239 }
240 
241 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
242  * and plain LSBs. */
243 
245  unsigned int substr, unsigned int pos)
246 {
247  SubStream *s = &m->substream[substr];
248  unsigned int mat, channel;
249 
250  for (mat = 0; mat < s->num_primitive_matrices; mat++)
251  if (s->lsb_bypass[mat])
252  m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
253 
254  for (channel = s->min_channel; channel <= s->max_channel; channel++) {
256  int codebook = cp->codebook;
257  int quant_step_size = s->quant_step_size[channel];
258  int lsb_bits = cp->huff_lsbs - quant_step_size;
259  int result = 0;
260 
261  if (codebook > 0)
262  result = get_vlc2(gbp, huff_vlc[codebook-1].table,
263  VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
264 
265  if (result < 0)
266  return AVERROR_INVALIDDATA;
267 
268  if (lsb_bits > 0)
269  result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
270 
271  result += cp->sign_huff_offset;
272  result *= 1 << quant_step_size;
273 
274  m->sample_buffer[pos + s->blockpos][channel] = result;
275  }
276 
277  return 0;
278 }
279 
281 {
282  MLPDecodeContext *m = avctx->priv_data;
283  int substr;
284 
285  init_static();
286  m->avctx = avctx;
287  for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
288  m->substream[substr].lossless_check_data = 0xffffffff;
289  ff_mlpdsp_init(&m->dsp);
290 
291  return 0;
292 }
293 
294 /** Read a major sync info header - contains high level information about
295  * the stream - sample rate, channel arrangement etc. Most of this
296  * information is not actually necessary for decoding, only for playback.
297  */
298 
300 {
302  int substr, ret;
303 
304  if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
305  return ret;
306 
307  if (mh.group1_bits == 0) {
308  av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
309  return AVERROR_INVALIDDATA;
310  }
311  if (mh.group2_bits > mh.group1_bits) {
313  "Channel group 2 cannot have more bits per sample than group 1.\n");
314  return AVERROR_INVALIDDATA;
315  }
316 
319  "Channel groups with differing sample rates are not currently supported.\n");
320  return AVERROR_INVALIDDATA;
321  }
322 
323  if (mh.group1_samplerate == 0) {
324  av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
325  return AVERROR_INVALIDDATA;
326  }
329  "Sampling rate %d is greater than the supported maximum (%d).\n",
331  return AVERROR_INVALIDDATA;
332  }
333  if (mh.access_unit_size > MAX_BLOCKSIZE) {
335  "Block size %d is greater than the supported maximum (%d).\n",
337  return AVERROR_INVALIDDATA;
338  }
341  "Block size pow2 %d is greater than the supported maximum (%d).\n",
343  return AVERROR_INVALIDDATA;
344  }
345 
346  if (mh.num_substreams == 0)
347  return AVERROR_INVALIDDATA;
348  if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) {
349  av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
350  return AVERROR_INVALIDDATA;
351  }
352  if (mh.num_substreams > MAX_SUBSTREAMS) {
354  "%d substreams (more than the "
355  "maximum supported by the decoder)",
356  mh.num_substreams);
357  return AVERROR_PATCHWELCOME;
358  }
359 
361 
364 
366 
367  /* limit to decoding 3 substreams, as the 4th is used by Dolby Atmos for non-audio data */
369 
372 
374  if (mh.group1_bits > 16)
376  else
382 
383  m->params_valid = 1;
384  for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
385  m->substream[substr].restart_seen = 0;
386 
387  /* Set the layout for each substream. When there's more than one, the first
388  * substream is Stereo. Subsequent substreams' layouts are indicated in the
389  * major sync. */
390  if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
391  if (mh.stream_type != 0xbb) {
393  "unexpected stream_type %X in MLP",
394  mh.stream_type);
395  return AVERROR_PATCHWELCOME;
396  }
397  if ((substr = (mh.num_substreams > 1)))
399  m->substream[substr].mask = mh.channel_layout_mlp;
400  } else {
401  if (mh.stream_type != 0xba) {
403  "unexpected stream_type %X in !MLP",
404  mh.stream_type);
405  return AVERROR_PATCHWELCOME;
406  }
407  if ((substr = (mh.num_substreams > 1)))
409  if (mh.num_substreams > 2)
412  else
414  m->substream[substr].mask = mh.channel_layout_thd_stream1;
415 
416  if (m->avctx->channels<=2 && m->substream[substr].mask == AV_CH_LAYOUT_MONO && m->max_decoded_substream == 1) {
417  av_log(m->avctx, AV_LOG_DEBUG, "Mono stream with 2 substreams, ignoring 2nd\n");
418  m->max_decoded_substream = 0;
419  if (m->avctx->channels==2)
421  }
422  }
423 
424  m->needs_reordering = mh.channel_arrangement >= 18 && mh.channel_arrangement <= 20;
425 
426  /* Parse the TrueHD decoder channel modifiers and set each substream's
427  * AVMatrixEncoding accordingly.
428  *
429  * The meaning of the modifiers depends on the channel layout:
430  *
431  * - THD_CH_MODIFIER_LTRT, THD_CH_MODIFIER_LBINRBIN only apply to 2-channel
432  *
433  * - THD_CH_MODIFIER_MONO applies to 1-channel or 2-channel (dual mono)
434  *
435  * - THD_CH_MODIFIER_SURROUNDEX, THD_CH_MODIFIER_NOTSURROUNDEX only apply to
436  * layouts with an Ls/Rs channel pair
437  */
438  for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
441  if (mh.num_substreams > 2 &&
446 
447  if (mh.num_substreams > 1 &&
452 
453  if (mh.num_substreams > 0)
454  switch (mh.channel_modifier_thd_stream0) {
457  break;
460  break;
461  default:
462  break;
463  }
464  }
465 
466  return 0;
467 }
468 
469 /** Read a restart header from a block in a substream. This contains parameters
470  * required to decode the audio that do not change very often. Generally
471  * (always) present only in blocks following a major sync. */
472 
474  const uint8_t *buf, unsigned int substr)
475 {
476  SubStream *s = &m->substream[substr];
477  unsigned int ch;
478  int sync_word, tmp;
480  uint8_t lossless_check;
481  int start_count = get_bits_count(gbp);
482  int min_channel, max_channel, max_matrix_channel, noise_type;
483  const int std_max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP
486 
487  sync_word = get_bits(gbp, 13);
488 
489  if (sync_word != 0x31ea >> 1) {
491  "restart header sync incorrect (got 0x%04x)\n", sync_word);
492  return AVERROR_INVALIDDATA;
493  }
494 
495  noise_type = get_bits1(gbp);
496 
497  if (m->avctx->codec_id == AV_CODEC_ID_MLP && noise_type) {
498  av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
499  return AVERROR_INVALIDDATA;
500  }
501 
502  skip_bits(gbp, 16); /* Output timestamp */
503 
504  min_channel = get_bits(gbp, 4);
505  max_channel = get_bits(gbp, 4);
506  max_matrix_channel = get_bits(gbp, 4);
507 
508  if (max_matrix_channel > std_max_matrix_channel) {
510  "Max matrix channel cannot be greater than %d.\n",
511  std_max_matrix_channel);
512  return AVERROR_INVALIDDATA;
513  }
514 
515  if (max_channel != max_matrix_channel) {
517  "Max channel must be equal max matrix channel.\n");
518  return AVERROR_INVALIDDATA;
519  }
520 
521  /* This should happen for TrueHD streams with >6 channels and MLP's noise
522  * type. It is not yet known if this is allowed. */
523  if (max_channel > MAX_MATRIX_CHANNEL_MLP && !noise_type) {
525  "%d channels (more than the "
526  "maximum supported by the decoder)",
527  max_channel + 2);
528  return AVERROR_PATCHWELCOME;
529  }
530 
531  if (min_channel > max_channel) {
533  "Substream min channel cannot be greater than max channel.\n");
534  return AVERROR_INVALIDDATA;
535  }
536 
537  s->min_channel = min_channel;
538  s->max_channel = max_channel;
539  s->max_matrix_channel = max_matrix_channel;
540  s->noise_type = noise_type;
541 
543  m->max_decoded_substream > substr) {
545  "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
546  "Further substreams will be skipped.\n",
547  s->max_channel + 1, s->mask, substr);
548  m->max_decoded_substream = substr;
549  }
550 
551  s->noise_shift = get_bits(gbp, 4);
552  s->noisegen_seed = get_bits(gbp, 23);
553 
554  skip_bits(gbp, 19);
555 
556  s->data_check_present = get_bits1(gbp);
557  lossless_check = get_bits(gbp, 8);
558  if (substr == m->max_decoded_substream
559  && s->lossless_check_data != 0xffffffff) {
561  if (tmp != lossless_check)
563  "Lossless check failed - expected %02x, calculated %02x.\n",
564  lossless_check, tmp);
565  }
566 
567  skip_bits(gbp, 16);
568 
569  memset(s->ch_assign, 0, sizeof(s->ch_assign));
570 
571  for (ch = 0; ch <= s->max_matrix_channel; ch++) {
572  int ch_assign = get_bits(gbp, 6);
573  if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
575  ch_assign);
577  channel);
578  }
579  if (ch_assign < 0 || ch_assign > s->max_matrix_channel) {
581  "Assignment of matrix channel %d to invalid output channel %d",
582  ch, ch_assign);
583  return AVERROR_PATCHWELCOME;
584  }
585  s->ch_assign[ch_assign] = ch;
586  }
587 
588  checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
589 
590  if (checksum != get_bits(gbp, 8))
591  av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
592 
593  /* Set default decoding parameters. */
594  s->param_presence_flags = 0xff;
595  s->num_primitive_matrices = 0;
596  s->blocksize = 8;
597  s->lossless_check_data = 0;
598 
599  memset(s->output_shift , 0, sizeof(s->output_shift ));
600  memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
601 
602  for (ch = s->min_channel; ch <= s->max_channel; ch++) {
603  ChannelParams *cp = &s->channel_params[ch];
604  cp->filter_params[FIR].order = 0;
605  cp->filter_params[IIR].order = 0;
606  cp->filter_params[FIR].shift = 0;
607  cp->filter_params[IIR].shift = 0;
608 
609  /* Default audio coding is 24-bit raw PCM. */
610  cp->huff_offset = 0;
611  cp->sign_huff_offset = -(1 << 23);
612  cp->codebook = 0;
613  cp->huff_lsbs = 24;
614  }
615 
616  if (substr == m->max_decoded_substream) {
617  m->avctx->channels = s->max_matrix_channel + 1;
618  m->avctx->channel_layout = s->mask;
620  s->output_shift,
623 
624  if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) {
627  int i = s->ch_assign[4];
628  s->ch_assign[4] = s->ch_assign[3];
629  s->ch_assign[3] = s->ch_assign[2];
630  s->ch_assign[2] = i;
631  } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
632  FFSWAP(int, s->ch_assign[2], s->ch_assign[4]);
633  FFSWAP(int, s->ch_assign[3], s->ch_assign[5]);
634  }
635  }
636 
637  }
638 
639  return 0;
640 }
641 
642 /** Read parameters for one of the prediction filters. */
643 
645  unsigned int substr, unsigned int channel,
646  unsigned int filter)
647 {
648  SubStream *s = &m->substream[substr];
650  const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
651  const char fchar = filter ? 'I' : 'F';
652  int i, order;
653 
654  // Filter is 0 for FIR, 1 for IIR.
655  av_assert0(filter < 2);
656 
657  if (m->filter_changed[channel][filter]++ > 1) {
658  av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
659  return AVERROR_INVALIDDATA;
660  }
661 
662  order = get_bits(gbp, 4);
663  if (order > max_order) {
665  "%cIR filter order %d is greater than maximum %d.\n",
666  fchar, order, max_order);
667  return AVERROR_INVALIDDATA;
668  }
669  fp->order = order;
670 
671  if (order > 0) {
672  int32_t *fcoeff = s->channel_params[channel].coeff[filter];
673  int coeff_bits, coeff_shift;
674 
675  fp->shift = get_bits(gbp, 4);
676 
677  coeff_bits = get_bits(gbp, 5);
678  coeff_shift = get_bits(gbp, 3);
679  if (coeff_bits < 1 || coeff_bits > 16) {
681  "%cIR filter coeff_bits must be between 1 and 16.\n",
682  fchar);
683  return AVERROR_INVALIDDATA;
684  }
685  if (coeff_bits + coeff_shift > 16) {
687  "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
688  fchar);
689  return AVERROR_INVALIDDATA;
690  }
691 
692  for (i = 0; i < order; i++)
693  fcoeff[i] = get_sbits(gbp, coeff_bits) * (1 << coeff_shift);
694 
695  if (get_bits1(gbp)) {
696  int state_bits, state_shift;
697 
698  if (filter == FIR) {
700  "FIR filter has state data specified.\n");
701  return AVERROR_INVALIDDATA;
702  }
703 
704  state_bits = get_bits(gbp, 4);
705  state_shift = get_bits(gbp, 4);
706 
707  /* TODO: Check validity of state data. */
708 
709  for (i = 0; i < order; i++)
710  fp->state[i] = state_bits ? get_sbits(gbp, state_bits) * (1 << state_shift) : 0;
711  }
712  }
713 
714  return 0;
715 }
716 
717 /** Read parameters for primitive matrices. */
718 
719 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
720 {
721  SubStream *s = &m->substream[substr];
722  unsigned int mat, ch;
723  const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP
726 
727  if (m->matrix_changed++ > 1) {
728  av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
729  return AVERROR_INVALIDDATA;
730  }
731 
732  s->num_primitive_matrices = get_bits(gbp, 4);
733 
734  if (s->num_primitive_matrices > max_primitive_matrices) {
736  "Number of primitive matrices cannot be greater than %d.\n",
737  max_primitive_matrices);
738  goto error;
739  }
740 
741  for (mat = 0; mat < s->num_primitive_matrices; mat++) {
742  int frac_bits, max_chan;
743  s->matrix_out_ch[mat] = get_bits(gbp, 4);
744  frac_bits = get_bits(gbp, 4);
745  s->lsb_bypass [mat] = get_bits1(gbp);
746 
747  if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
749  "Invalid channel %d specified as output from matrix.\n",
750  s->matrix_out_ch[mat]);
751  goto error;
752  }
753  if (frac_bits > 14) {
755  "Too many fractional bits specified.\n");
756  goto error;
757  }
758 
759  max_chan = s->max_matrix_channel;
760  if (!s->noise_type)
761  max_chan+=2;
762 
763  for (ch = 0; ch <= max_chan; ch++) {
764  int coeff_val = 0;
765  if (get_bits1(gbp))
766  coeff_val = get_sbits(gbp, frac_bits + 2);
767 
768  s->matrix_coeff[mat][ch] = coeff_val * (1 << (14 - frac_bits));
769  }
770 
771  if (s->noise_type)
772  s->matrix_noise_shift[mat] = get_bits(gbp, 4);
773  else
774  s->matrix_noise_shift[mat] = 0;
775  }
776 
777  return 0;
778 error:
779  s->num_primitive_matrices = 0;
780  memset(s->matrix_out_ch, 0, sizeof(s->matrix_out_ch));
781 
782  return AVERROR_INVALIDDATA;
783 }
784 
785 /** Read channel parameters. */
786 
787 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
788  GetBitContext *gbp, unsigned int ch)
789 {
790  SubStream *s = &m->substream[substr];
791  ChannelParams *cp = &s->channel_params[ch];
792  FilterParams *fir = &cp->filter_params[FIR];
793  FilterParams *iir = &cp->filter_params[IIR];
794  int ret;
795 
797  if (get_bits1(gbp))
798  if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
799  return ret;
800 
802  if (get_bits1(gbp))
803  if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
804  return ret;
805 
806  if (fir->order + iir->order > 8) {
807  av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
808  return AVERROR_INVALIDDATA;
809  }
810 
811  if (fir->order && iir->order &&
812  fir->shift != iir->shift) {
814  "FIR and IIR filters must use the same precision.\n");
815  return AVERROR_INVALIDDATA;
816  }
817  /* The FIR and IIR filters must have the same precision.
818  * To simplify the filtering code, only the precision of the
819  * FIR filter is considered. If only the IIR filter is employed,
820  * the FIR filter precision is set to that of the IIR filter, so
821  * that the filtering code can use it. */
822  if (!fir->order && iir->order)
823  fir->shift = iir->shift;
824 
826  if (get_bits1(gbp))
827  cp->huff_offset = get_sbits(gbp, 15);
828 
829  cp->codebook = get_bits(gbp, 2);
830  cp->huff_lsbs = get_bits(gbp, 5);
831 
832  if (cp->huff_lsbs > 24) {
833  av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
834  cp->huff_lsbs = 0;
835  return AVERROR_INVALIDDATA;
836  }
837 
838  return 0;
839 }
840 
841 /** Read decoding parameters that change more often than those in the restart
842  * header. */
843 
845  unsigned int substr)
846 {
847  SubStream *s = &m->substream[substr];
848  unsigned int ch;
849  int ret = 0;
850  unsigned recompute_sho = 0;
851 
853  if (get_bits1(gbp))
854  s->param_presence_flags = get_bits(gbp, 8);
855 
857  if (get_bits1(gbp)) {
858  s->blocksize = get_bits(gbp, 9);
859  if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
860  av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n");
861  s->blocksize = 0;
862  return AVERROR_INVALIDDATA;
863  }
864  }
865 
867  if (get_bits1(gbp))
868  if ((ret = read_matrix_params(m, substr, gbp)) < 0)
869  return ret;
870 
872  if (get_bits1(gbp)) {
873  for (ch = 0; ch <= s->max_matrix_channel; ch++) {
874  s->output_shift[ch] = get_sbits(gbp, 4);
875  if (s->output_shift[ch] < 0) {
876  avpriv_request_sample(m->avctx, "Negative output_shift");
877  s->output_shift[ch] = 0;
878  }
879  }
880  if (substr == m->max_decoded_substream)
882  s->output_shift,
885  }
886 
888  if (get_bits1(gbp))
889  for (ch = 0; ch <= s->max_channel; ch++) {
890  s->quant_step_size[ch] = get_bits(gbp, 4);
891 
892  recompute_sho |= 1<<ch;
893  }
894 
895  for (ch = s->min_channel; ch <= s->max_channel; ch++)
896  if (get_bits1(gbp)) {
897  recompute_sho |= 1<<ch;
898  if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
899  goto fail;
900  }
901 
902 
903 fail:
904  for (ch = 0; ch <= s->max_channel; ch++) {
905  if (recompute_sho & (1<<ch)) {
906  ChannelParams *cp = &s->channel_params[ch];
907 
908  if (cp->codebook > 0 && cp->huff_lsbs < s->quant_step_size[ch]) {
909  if (ret >= 0) {
910  av_log(m->avctx, AV_LOG_ERROR, "quant_step_size larger than huff_lsbs\n");
911  ret = AVERROR_INVALIDDATA;
912  }
913  s->quant_step_size[ch] = 0;
914  }
915 
916  cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
917  }
918  }
919  return ret;
920 }
921 
922 #define MSB_MASK(bits) (-1u << (bits))
923 
924 /** Generate PCM samples using the prediction filters and residual values
925  * read from the data stream, and update the filter state. */
926 
927 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
928  unsigned int channel)
929 {
930  SubStream *s = &m->substream[substr];
931  const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
933  int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
934  int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
937  unsigned int filter_shift = fir->shift;
938  int32_t mask = MSB_MASK(s->quant_step_size[channel]);
939 
940  memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
941  memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
942 
943  m->dsp.mlp_filter_channel(firbuf, fircoeff,
944  fir->order, iir->order,
945  filter_shift, mask, s->blocksize,
946  &m->sample_buffer[s->blockpos][channel]);
947 
948  memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
949  memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
950 }
951 
952 /** Read a block of PCM residual data (or actual if no filtering active). */
953 
955  unsigned int substr)
956 {
957  SubStream *s = &m->substream[substr];
958  unsigned int i, ch, expected_stream_pos = 0;
959  int ret;
960 
961  if (s->data_check_present) {
962  expected_stream_pos = get_bits_count(gbp);
963  expected_stream_pos += get_bits(gbp, 16);
965  "Substreams with VLC block size check info");
966  }
967 
968  if (s->blockpos + s->blocksize > m->access_unit_size) {
969  av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
970  return AVERROR_INVALIDDATA;
971  }
972 
973  memset(&m->bypassed_lsbs[s->blockpos][0], 0,
974  s->blocksize * sizeof(m->bypassed_lsbs[0]));
975 
976  for (i = 0; i < s->blocksize; i++)
977  if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
978  return ret;
979 
980  for (ch = s->min_channel; ch <= s->max_channel; ch++)
981  filter_channel(m, substr, ch);
982 
983  s->blockpos += s->blocksize;
984 
985  if (s->data_check_present) {
986  if (get_bits_count(gbp) != expected_stream_pos)
987  av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
988  skip_bits(gbp, 8);
989  }
990 
991  return 0;
992 }
993 
994 /** Data table used for TrueHD noise generation function. */
995 
996 static const int8_t noise_table[256] = {
997  30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
998  52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
999  10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
1000  51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
1001  38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
1002  61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
1003  67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
1004  48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
1005  0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
1006  16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
1007  13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
1008  89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
1009  36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
1010  39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
1011  45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
1012  -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
1013 };
1014 
1015 /** Noise generation functions.
1016  * I'm not sure what these are for - they seem to be some kind of pseudorandom
1017  * sequence generators, used to generate noise data which is used when the
1018  * channels are rematrixed. I'm not sure if they provide a practical benefit
1019  * to compression, or just obfuscate the decoder. Are they for some kind of
1020  * dithering? */
1021 
1022 /** Generate two channels of noise, used in the matrix when
1023  * restart sync word == 0x31ea. */
1024 
1025 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
1026 {
1027  SubStream *s = &m->substream[substr];
1028  unsigned int i;
1029  uint32_t seed = s->noisegen_seed;
1030  unsigned int maxchan = s->max_matrix_channel;
1031 
1032  for (i = 0; i < s->blockpos; i++) {
1033  uint16_t seed_shr7 = seed >> 7;
1034  m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) * (1 << s->noise_shift);
1035  m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) * (1 << s->noise_shift);
1036 
1037  seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
1038  }
1039 
1040  s->noisegen_seed = seed;
1041 }
1042 
1043 /** Generate a block of noise, used when restart sync word == 0x31eb. */
1044 
1045 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
1046 {
1047  SubStream *s = &m->substream[substr];
1048  unsigned int i;
1049  uint32_t seed = s->noisegen_seed;
1050 
1051  for (i = 0; i < m->access_unit_size_pow2; i++) {
1052  uint8_t seed_shr15 = seed >> 15;
1053  m->noise_buffer[i] = noise_table[seed_shr15];
1054  seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
1055  }
1056 
1057  s->noisegen_seed = seed;
1058 }
1059 
1060 /** Write the audio data into the output buffer. */
1061 
1062 static int output_data(MLPDecodeContext *m, unsigned int substr,
1063  AVFrame *frame, int *got_frame_ptr)
1064 {
1065  AVCodecContext *avctx = m->avctx;
1066  SubStream *s = &m->substream[substr];
1067  unsigned int mat;
1068  unsigned int maxchan;
1069  int ret;
1070  int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
1071 
1072  if (m->avctx->channels != s->max_matrix_channel + 1) {
1073  av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
1074  return AVERROR_INVALIDDATA;
1075  }
1076 
1077  if (!s->blockpos) {
1078  av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
1079  return AVERROR_INVALIDDATA;
1080  }
1081 
1082  maxchan = s->max_matrix_channel;
1083  if (!s->noise_type) {
1084  generate_2_noise_channels(m, substr);
1085  maxchan += 2;
1086  } else {
1087  fill_noise_buffer(m, substr);
1088  }
1089 
1090  /* Apply the channel matrices in turn to reconstruct the original audio
1091  * samples. */
1092  for (mat = 0; mat < s->num_primitive_matrices; mat++) {
1093  unsigned int dest_ch = s->matrix_out_ch[mat];
1094  m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0],
1095  s->matrix_coeff[mat],
1096  &m->bypassed_lsbs[0][mat],
1097  m->noise_buffer,
1098  s->num_primitive_matrices - mat,
1099  dest_ch,
1100  s->blockpos,
1101  maxchan,
1102  s->matrix_noise_shift[mat],
1104  MSB_MASK(s->quant_step_size[dest_ch]));
1105  }
1106 
1107  /* get output buffer */
1108  frame->nb_samples = s->blockpos;
1109  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1110  return ret;
1112  s->blockpos,
1113  m->sample_buffer,
1114  frame->data[0],
1115  s->ch_assign,
1116  s->output_shift,
1117  s->max_matrix_channel,
1118  is32);
1119 
1120  /* Update matrix encoding side data */
1121  if ((ret = ff_side_data_update_matrix_encoding(frame, s->matrix_encoding)) < 0)
1122  return ret;
1123 
1124  *got_frame_ptr = 1;
1125 
1126  return 0;
1127 }
1128 
1129 /** Read an access unit from the stream.
1130  * @return negative on error, 0 if not enough data is present in the input stream,
1131  * otherwise the number of bytes consumed. */
1132 
1133 static int read_access_unit(AVCodecContext *avctx, void* data,
1134  int *got_frame_ptr, AVPacket *avpkt)
1135 {
1136  const uint8_t *buf = avpkt->data;
1137  int buf_size = avpkt->size;
1138  MLPDecodeContext *m = avctx->priv_data;
1139  GetBitContext gb;
1140  unsigned int length, substr;
1141  unsigned int substream_start;
1142  unsigned int header_size = 4;
1143  unsigned int substr_header_size = 0;
1144  uint8_t substream_parity_present[MAX_SUBSTREAMS];
1145  uint16_t substream_data_len[MAX_SUBSTREAMS];
1146  uint8_t parity_bits;
1147  int ret;
1148 
1149  if (buf_size < 4)
1150  return AVERROR_INVALIDDATA;
1151 
1152  length = (AV_RB16(buf) & 0xfff) * 2;
1153 
1154  if (length < 4 || length > buf_size)
1155  return AVERROR_INVALIDDATA;
1156 
1157  init_get_bits(&gb, (buf + 4), (length - 4) * 8);
1158 
1159  m->is_major_sync_unit = 0;
1160  if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
1161  if (read_major_sync(m, &gb) < 0)
1162  goto error;
1163  m->is_major_sync_unit = 1;
1164  header_size += m->major_sync_header_size;
1165  }
1166 
1167  if (!m->params_valid) {
1169  "Stream parameters not seen; skipping frame.\n");
1170  *got_frame_ptr = 0;
1171  return length;
1172  }
1173 
1174  substream_start = 0;
1175 
1176  for (substr = 0; substr < m->num_substreams; substr++) {
1177  int extraword_present, checkdata_present, end, nonrestart_substr;
1178 
1179  extraword_present = get_bits1(&gb);
1180  nonrestart_substr = get_bits1(&gb);
1181  checkdata_present = get_bits1(&gb);
1182  skip_bits1(&gb);
1183 
1184  end = get_bits(&gb, 12) * 2;
1185 
1186  substr_header_size += 2;
1187 
1188  if (extraword_present) {
1189  if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
1190  av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
1191  goto error;
1192  }
1193  skip_bits(&gb, 16);
1194  substr_header_size += 2;
1195  }
1196 
1197  if (length < header_size + substr_header_size) {
1198  av_log(m->avctx, AV_LOG_ERROR, "Insuffient data for headers\n");
1199  goto error;
1200  }
1201 
1202  if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1203  av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1204  goto error;
1205  }
1206 
1207  if (end + header_size + substr_header_size > length) {
1209  "Indicated length of substream %d data goes off end of "
1210  "packet.\n", substr);
1211  end = length - header_size - substr_header_size;
1212  }
1213 
1214  if (end < substream_start) {
1215  av_log(avctx, AV_LOG_ERROR,
1216  "Indicated end offset of substream %d data "
1217  "is smaller than calculated start offset.\n",
1218  substr);
1219  goto error;
1220  }
1221 
1222  if (substr > m->max_decoded_substream)
1223  continue;
1224 
1225  substream_parity_present[substr] = checkdata_present;
1226  substream_data_len[substr] = end - substream_start;
1227  substream_start = end;
1228  }
1229 
1230  parity_bits = ff_mlp_calculate_parity(buf, 4);
1231  parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1232 
1233  if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1234  av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1235  goto error;
1236  }
1237 
1238  buf += header_size + substr_header_size;
1239 
1240  for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1241  SubStream *s = &m->substream[substr];
1242  init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1243 
1244  m->matrix_changed = 0;
1245  memset(m->filter_changed, 0, sizeof(m->filter_changed));
1246 
1247  s->blockpos = 0;
1248  do {
1249  if (get_bits1(&gb)) {
1250  if (get_bits1(&gb)) {
1251  /* A restart header should be present. */
1252  if (read_restart_header(m, &gb, buf, substr) < 0)
1253  goto next_substr;
1254  s->restart_seen = 1;
1255  }
1256 
1257  if (!s->restart_seen)
1258  goto next_substr;
1259  if (read_decoding_params(m, &gb, substr) < 0)
1260  goto next_substr;
1261  }
1262 
1263  if (!s->restart_seen)
1264  goto next_substr;
1265 
1266  if ((ret = read_block_data(m, &gb, substr)) < 0)
1267  return ret;
1268 
1269  if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1270  goto substream_length_mismatch;
1271 
1272  } while (!get_bits1(&gb));
1273 
1274  skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1275 
1276  if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1277  int shorten_by;
1278 
1279  if (get_bits(&gb, 16) != 0xD234)
1280  return AVERROR_INVALIDDATA;
1281 
1282  shorten_by = get_bits(&gb, 16);
1283  if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by & 0x2000)
1284  s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1285  else if (m->avctx->codec_id == AV_CODEC_ID_MLP && shorten_by != 0xD234)
1286  return AVERROR_INVALIDDATA;
1287 
1288  if (substr == m->max_decoded_substream)
1289  av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1290  }
1291 
1292  if (substream_parity_present[substr]) {
1294 
1295  if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1296  goto substream_length_mismatch;
1297 
1298  parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1299  checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1300 
1301  if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1302  av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1303  if ( get_bits(&gb, 8) != checksum)
1304  av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1305  }
1306 
1307  if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1308  goto substream_length_mismatch;
1309 
1310 next_substr:
1311  if (!s->restart_seen)
1313  "No restart header present in substream %d.\n", substr);
1314 
1315  buf += substream_data_len[substr];
1316  }
1317 
1318  if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
1319  return ret;
1320 
1321  return length;
1322 
1323 substream_length_mismatch:
1324  av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1325  return AVERROR_INVALIDDATA;
1326 
1327 error:
1328  m->params_valid = 0;
1329  return AVERROR_INVALIDDATA;
1330 }
1331 
1332 #if CONFIG_MLP_DECODER
1334  .name = "mlp",
1335  .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1336  .type = AVMEDIA_TYPE_AUDIO,
1337  .id = AV_CODEC_ID_MLP,
1338  .priv_data_size = sizeof(MLPDecodeContext),
1339  .init = mlp_decode_init,
1341  .capabilities = AV_CODEC_CAP_DR1,
1342 };
1343 #endif
1344 #if CONFIG_TRUEHD_DECODER
1346  .name = "truehd",
1347  .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1348  .type = AVMEDIA_TYPE_AUDIO,
1349  .id = AV_CODEC_ID_TRUEHD,
1350  .priv_data_size = sizeof(MLPDecodeContext),
1351  .init = mlp_decode_init,
1353  .capabilities = AV_CODEC_CAP_DR1,
1354 };
1355 #endif /* CONFIG_TRUEHD_DECODER */
uint8_t shift
Right shift to apply to output of filter.
Definition: mlp.h:76
static unsigned int show_bits_long(GetBitContext *s, int n)
Show 0-32 bits.
Definition: get_bits.h:587
static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout, int index)
Definition: mlpdec.c:188
int major_sync_header_size
Size of the major sync unit, in bytes.
Definition: mlpdec.c:136
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define MAX_IIR_ORDER
Definition: mlp.h:65
FilterParams filter_params[NUM_FILTERS]
Definition: mlp.h:86
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
static av_cold int mlp_decode_init(AVCodecContext *avctx)
Definition: mlpdec.c:280
#define AV_CH_TOP_FRONT_RIGHT
void(* mlp_rematrix_channel)(int32_t *samples, const int32_t *coeffs, const uint8_t *bypassed_lsbs, const int8_t *noise_buffer, int index, unsigned int dest_ch, uint16_t blockpos, unsigned int maxchan, int matrix_noise_shift, int access_unit_size_pow2, int32_t mask)
Definition: mlpdsp.h:54
int8_t noise_buffer[MAX_BLOCKSIZE_POW2]
Definition: mlpdec.c:160
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
uint8_t param_presence_flags
Bitmask of which parameter sets are conveyed in a decoding parameter block.
Definition: mlpdec.c:85
uint8_t params_valid
Set if a valid major sync block has been read. Otherwise no decoding is possible. ...
Definition: mlpdec.c:139
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:381
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
uint64_t mask
The channel layout for this substream.
Definition: mlpdec.c:69
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: vlc.h:75
#define AV_CH_TOP_FRONT_LEFT
static int mlp_channel_layout_subset(uint64_t channel_layout, uint64_t mask)
Definition: mlpdec.c:183
int num_substreams
Number of substreams within stream.
Definition: mlp_parser.h:62
#define AV_CH_TOP_FRONT_CENTER
int size
Definition: avcodec.h:1445
#define AV_CH_LOW_FREQUENCY_2
const uint8_t ff_mlp_huffman_tables[3][18][2]
Tables defining the Huffman codes.
Definition: mlp.c:28
#define MAX_BLOCKSIZE_POW2
next power of two greater than MAX_BLOCKSIZE
Definition: mlp.h:58
enum AVMatrixEncoding matrix_encoding
The matrix encoding mode for this substream.
Definition: mlpdec.c:71
static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, const uint8_t *buf, unsigned int substr)
Read a restart header from a block in a substream.
Definition: mlpdec.c:473
#define MAX_SAMPLERATE
maximum sample frequency seen in files
Definition: mlp.h:53
uint64_t channel_layout_mlp
Channel layout for MLP streams.
Definition: mlp_parser.h:52
int8_t output_shift[MAX_CHANNELS]
Left shift to apply to decoded PCM values to get final 24-bit output.
Definition: mlpdec.c:122
#define AV_CH_SURROUND_DIRECT_RIGHT
#define AV_CH_LAYOUT_STEREO
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:2756
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
AVCodec.
Definition: avcodec.h:3423
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:361
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
int access_unit_size
Number of samples per coded frame.
Definition: mlp_parser.h:56
static VLC huff_vlc[3]
Definition: mlpdec.c:202
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS]
Matrix coefficients, stored as 2.14 fixed point.
Definition: mlpdec.c:108
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AVCodec ff_mlp_decoder
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
static void filter(int16_t *output, ptrdiff_t out_stride, int16_t *low, ptrdiff_t low_stride, int16_t *high, ptrdiff_t high_stride, int len, int clip)
Definition: cfhd.c:153
int matrix_changed
Definition: mlpdec.c:157
#define AV_CH_WIDE_LEFT
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2196
uint8_t
#define av_cold
Definition: attributes.h:82
#define PARAM_BLOCKSIZE
Definition: mlpdec.c:86
MLPDSPContext dsp
Definition: mlpdec.c:164
static uint8_t xor_32_to_8(uint32_t value)
XOR four bytes into one.
Definition: mlp.h:160
int channel_modifier_thd_stream0
Channel modifier for substream 0 of TrueHD streams ("2-channel presentation")
Definition: mlp_parser.h:45
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
#define MAX_FIR_ORDER
The maximum number of taps in IIR and FIR filters.
Definition: mlp.h:64
uint8_t ch_assign[MAX_CHANNELS]
For each channel output by the matrix, the output channel to map it to.
Definition: mlpdec.c:67
#define AV_CH_WIDE_RIGHT
#define AV_CH_LOW_FREQUENCY
static AVFrame * frame
Public header for CRC hash function implementation.
static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
Generate a block of noise, used when restart sync word == 0x31eb.
Definition: mlpdec.c:1045
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:112
uint8_t * data
Definition: avcodec.h:1444
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
uint8_t restart_seen
Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
Definition: mlpdec.c:53
#define PARAM_HUFFOFFSET
Definition: mlpdec.c:92
bitstream reader API header.
#define AV_CH_BACK_LEFT
static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
Noise generation functions.
Definition: mlpdec.c:1025
int channel_arrangement
Definition: mlp_parser.h:43
uint8_t min_channel
The index of the first channel coded in this substream.
Definition: mlpdec.c:61
static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr, unsigned int channel, unsigned int filter)
Read parameters for one of the prediction filters.
Definition: mlpdec.c:644
#define PARAM_PRESENCE
Definition: mlpdec.c:93
signed 32 bits
Definition: samplefmt.h:62
#define av_log(a,...)
int16_t huff_offset
Offset to apply to residual values.
Definition: mlp.h:89
#define PARAM_OUTSHIFT
Definition: mlpdec.c:88
static int read_channel_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp, unsigned int ch)
Read channel parameters.
Definition: mlpdec.c:787
#define VLC_BITS
number of bits used for VLC lookup - longest Huffman code is 9
Definition: mlpdec.c:47
#define NUM_FILTERS
number of allowed filters
Definition: mlp.h:61
uint8_t max_channel
The index of the last channel coded in this substream.
Definition: mlpdec.c:63
uint8_t ff_mlp_calculate_parity(const uint8_t *buf, unsigned int buf_size)
XOR together all the bytes of a buffer.
Definition: mlp.c:120
#define MAX_MATRICES
Definition: mlp.h:43
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
ChannelParams channel_params[MAX_CHANNELS]
Channel coding parameters for channels in the substream.
Definition: mlpdec.c:74
#define MAX_MATRIX_CHANNEL_TRUEHD
Definition: mlp.h:31
int channel_modifier_thd_stream2
Channel modifier for substream 2 of TrueHD streams ("8-channel presentation")
Definition: mlp_parser.h:47
static const uint16_t mask[17]
Definition: lzw.c:38
uint8_t needs_reordering
Stream needs channel reordering to comply with FFmpeg's channel order.
Definition: mlpdec.c:148
int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]
Definition: mlpdec.c:161
uint8_t quant_step_size[MAX_CHANNELS]
Left shift to apply to Huffman-decoded residuals.
Definition: mlpdec.c:114
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
static const uint64_t thd_channel_order[]
Definition: mlpdec.c:167
#define AV_CH_LAYOUT_QUAD
static int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr, unsigned int pos)
Read a sample, consisting of either, both or neither of entropy-coded MSBs and plain LSBs...
Definition: mlpdec.c:244
GLsizei GLsizei * length
Definition: opengl_enc.c:115
const char * name
Name of the codec implementation.
Definition: avcodec.h:3430
uint8_t num_substreams
Number of substreams contained within this stream.
Definition: mlpdec.c:142
static int read_access_unit(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Read an access unit from the stream.
Definition: mlpdec.c:1133
#define fail()
Definition: checkasm.h:117
Definition: vlc.h:26
uint8_t max_matrix_channel
The number of channels input into the rematrix stage.
Definition: mlpdec.c:65
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2239
#define MAX_BLOCKSIZE
Definition: diracdec.c:54
#define VLC_STATIC_SIZE
Definition: mlpdec.c:48
common internal API header
static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
Read parameters for primitive matrices.
Definition: mlpdec.c:719
static void filter_channel(MLPDecodeContext *m, unsigned int substr, unsigned int channel)
Generate PCM samples using the prediction filters and residual values read from the data stream...
Definition: mlpdec.c:927
#define AV_CH_TOP_CENTER
audio channel layout utility functions
#define MAX_MATRIX_CHANNEL_MLP
Last possible matrix channel for each codec.
Definition: mlp.h:30
uint8_t ff_mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
Calculate an 8-bit checksum over a restart header – a non-multiple-of-8 number of bits...
Definition: mlp.c:101
static int output_data(MLPDecodeContext *m, unsigned int substr, AVFrame *frame, int *got_frame_ptr)
Write the audio data into the output buffer.
Definition: mlpdec.c:1062
#define FFMIN(a, b)
Definition: common.h:96
uint16_t noise_type
restart header data
Definition: mlpdec.c:58
static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
Read a major sync info header - contains high level information about the stream - sample rate...
Definition: mlpdec.c:299
int32_t(*(* mlp_select_pack_output)(uint8_t *ch_assign, int8_t *output_shift, uint8_t max_matrix_channel, int is32))(int32_t
Definition: mlpdsp.h:65
int32_t
int32_t lossless_check_data
Running XOR of all output samples.
Definition: mlpdec.c:125
MLP parser prototypes.
static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr)
Read a block of PCM residual data (or actual if no filtering active).
Definition: mlpdec.c:954
#define s(width, name)
Definition: cbs_vp9.c:257
mcdeint parity
Definition: vf_mcdeint.c:274
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:762
#define AV_CH_FRONT_LEFT_OF_CENTER
static const int8_t noise_table[256]
Data table used for TrueHD noise generation function.
Definition: mlpdec.c:996
#define AV_CH_FRONT_CENTER
int filter_changed[MAX_CHANNELS][NUM_FILTERS]
Definition: mlpdec.c:158
uint8_t lsb_bypass[MAX_MATRICES]
Whether the LSBs of the matrix output are encoded in the bitstream.
Definition: mlpdec.c:106
int32_t coeff[NUM_FILTERS][MAX_FIR_ORDER]
Definition: mlp.h:87
static volatile int checksum
Definition: adler32.c:30
#define AV_CH_LAYOUT_5POINT1_BACK
static void error(const char *err)
int access_unit_size
number of PCM samples contained in each frame
Definition: mlpdec.c:151
#define FF_ARRAY_ELEMS(a)
#define AV_CH_FRONT_RIGHT_OF_CENTER
int ff_side_data_update_matrix_encoding(AVFrame *frame, enum AVMatrixEncoding matrix_encoding)
Add or update AV_FRAME_DATA_MATRIXENCODING side data.
Definition: utils.c:134
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
int stream_type
0xBB for MLP, 0xBA for TrueHD
Definition: mlp_parser.h:34
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2208
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
int access_unit_size_pow2
Next power of two above number of samples per frame.
Definition: mlp_parser.h:57
AVCodec ff_truehd_decoder
uint16_t blocksize
number of PCM samples in current audio block
Definition: mlpdec.c:117
uint8_t codebook
Which VLC codebook to use to read residuals.
Definition: mlp.h:91
#define MAX_MATRICES_TRUEHD
Definition: mlp.h:42
Libavcodec external API header.
uint8_t data_check_present
Set if the substream contains extra info to check the size of VLC blocks.
Definition: mlpdec.c:82
int32_t state[MAX_FIR_ORDER]
Definition: mlp.h:78
enum AVCodecID codec_id
Definition: avcodec.h:1542
int sample_rate
samples per second
Definition: avcodec.h:2188
av_cold void ff_mlpdsp_init(MLPDSPContext *c)
Definition: mlpdsp.c:128
SubStream substream[MAX_SUBSTREAMS]
Definition: mlpdec.c:155
uint8_t order
number of taps in filter
Definition: mlp.h:75
static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, unsigned int substr)
Read decoding parameters that change more often than those in the restart header. ...
Definition: mlpdec.c:844
int channel_modifier_thd_stream1
Channel modifier for substream 1 of TrueHD streams ("6-channel presentation")
Definition: mlp_parser.h:46
main external API structure.
Definition: avcodec.h:1532
#define PARAM_QUANTSTEP
Definition: mlpdec.c:89
#define AV_CH_FRONT_LEFT
int is_major_sync_unit
Current access unit being read has a major sync.
Definition: mlpdec.c:133
static unsigned int seed
Definition: videogen.c:78
#define fp
Definition: regdef.h:44
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1922
int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS]
Definition: mlpdec.c:162
void * buf
Definition: avisynth_c.h:690
filter data
Definition: mlp.h:74
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:487
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:523
static const uint32_t table[]
Definition: prosumer.c:204
#define IIR
Definition: mlp.h:71
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:460
AVCodecContext * avctx
Definition: mlpdec.c:130
#define PARAM_IIR
Definition: mlpdec.c:91
int index
Definition: gxfenc.c:89
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:615
uint8_t num_primitive_matrices
matrix data
Definition: mlpdec.c:100
#define AV_CH_LAYOUT_5POINT0_BACK
uint8_t max_decoded_substream
Index of the last substream to decode - further substreams are skipped.
Definition: mlpdec.c:145
#define MAX_CHANNELS
Definition: aac.h:47
#define FIR
Definition: mlp.h:70
int av_get_channel_layout_channel_index(uint64_t channel_layout, uint64_t channel)
Get the index of a channel in channel_layout.
uint8_t huff_lsbs
Size of residual suffix not encoded using VLC.
Definition: mlp.h:92
static av_cold void init_static(void)
Initialize static data, constant between all invocations of the codec.
Definition: mlpdec.c:206
uint16_t blockpos
Number of PCM samples decoded so far in this frame.
Definition: mlpdec.c:119
int group2_bits
Bit depth of the second substream (MLP only)
Definition: mlp_parser.h:38
#define AV_CH_BACK_CENTER
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:232
#define AV_CH_SIDE_RIGHT
uint32_t noisegen_seed
The current seed value for the pseudorandom noise generator(s).
Definition: mlpdec.c:79
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:279
uint8_t matrix_out_ch[MAX_MATRICES]
matrix output channel
Definition: mlpdec.c:103
signed 16 bits
Definition: samplefmt.h:61
int access_unit_size_pow2
next power of two above the number of samples in each frame
Definition: mlpdec.c:153
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
uint64_t channel_layout_thd_stream1
Channel layout for substream 1 of TrueHD streams ("6-channel presentation")
Definition: mlp_parser.h:53
#define MAX_SUBSTREAMS
Maximum number of substreams that can be decoded.
Definition: mlp.h:48
uint64_t channel_layout_thd_stream2
Channel layout for substream 2 of TrueHD streams ("8-channel presentation")
Definition: mlp_parser.h:54
static int32_t calculate_sign_huff(MLPDecodeContext *m, unsigned int substr, unsigned int ch)
Definition: mlpdec.c:223
int header_size
Size of the major sync header, in bytes.
Definition: mlp_parser.h:35
int ff_mlp_read_major_sync(void *log, MLPHeaderInfo *mh, GetBitContext *gb)
Read a major sync info header - contains high level information about the stream - sample rate...
Definition: mlp_parser.c:145
void * priv_data
Definition: avcodec.h:1559
uint8_t matrix_noise_shift[MAX_MATRICES]
Left shift to apply to noise values in 0x31eb substreams.
Definition: mlpdec.c:110
uint8_t noise_shift
The left shift applied to random noise in 0x31ea substreams.
Definition: mlpdec.c:77
#define PARAM_MATRIX
Definition: mlpdec.c:87
sample data coding information
Definition: mlp.h:85
int channels
number of audio channels
Definition: avcodec.h:2189
int group1_bits
The bit depth of the first substream.
Definition: mlp_parser.h:37
av_cold void ff_mlp_init_crc(void)
Definition: mlp.c:75
#define AV_CH_SURROUND_DIRECT_LEFT
void(* mlp_filter_channel)(int32_t *state, const int32_t *coeff, int firorder, int iirorder, unsigned int filter_shift, int32_t mask, int blocksize, int32_t *sample_buffer)
Definition: mlpdsp.h:50
#define AV_CH_FRONT_RIGHT
#define MAX_MATRICES_MLP
Maximum number of matrices used in decoding; most streams have one matrix per output channel...
Definition: mlp.h:41
#define MSB_MASK(bits)
Definition: mlpdec.c:922
AVMatrixEncoding
#define AV_CH_SIDE_LEFT
#define FFSWAP(type, a, b)
Definition: common.h:99
int group1_samplerate
Sample rate of first substream.
Definition: mlp_parser.h:40
#define AV_CH_LAYOUT_MONO
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
Definition: avcodec.h:2246
int32_t(* mlp_pack_output)(int32_t lossless_check_data, uint16_t blockpos, int32_t(*sample_buffer)[MAX_CHANNELS], void *data, uint8_t *ch_assign, int8_t *output_shift, uint8_t max_matrix_channel, int is32)
Definition: mlpdsp.h:69
This structure stores compressed data.
Definition: avcodec.h:1421
int group2_samplerate
Sample rate of second substream (MLP only)
Definition: mlp_parser.h:41
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:967
#define mh
#define AV_CH_BACK_RIGHT
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
#define PARAM_FIR
Definition: mlpdec.c:90
int32_t sign_huff_offset
sign/rounding-corrected version of huff_offset
Definition: mlp.h:90
static uint8_t tmp[11]
Definition: aes_ctr.c:26
uint8_t ff_mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
MLP uses checksums that seem to be based on the standard CRC algorithm, but are not (in implementatio...
Definition: mlp.c:94