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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 ch_layout;
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 uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout,
184  int index)
185 {
186  int i;
187 
188  if (av_get_channel_layout_nb_channels(channel_layout) <= index)
189  return 0;
190 
191  for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++)
192  if (channel_layout & thd_channel_order[i] && !index--)
193  return thd_channel_order[i];
194  return 0;
195 }
196 
197 static VLC huff_vlc[3];
198 
199 /** Initialize static data, constant between all invocations of the codec. */
200 
201 static av_cold void init_static(void)
202 {
203  if (!huff_vlc[0].bits) {
204  INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
205  &ff_mlp_huffman_tables[0][0][1], 2, 1,
206  &ff_mlp_huffman_tables[0][0][0], 2, 1, VLC_STATIC_SIZE);
207  INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
208  &ff_mlp_huffman_tables[1][0][1], 2, 1,
209  &ff_mlp_huffman_tables[1][0][0], 2, 1, VLC_STATIC_SIZE);
210  INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
211  &ff_mlp_huffman_tables[2][0][1], 2, 1,
212  &ff_mlp_huffman_tables[2][0][0], 2, 1, VLC_STATIC_SIZE);
213  }
214 
215  ff_mlp_init_crc();
216 }
217 
219  unsigned int substr, unsigned int ch)
220 {
221  SubStream *s = &m->substream[substr];
222  ChannelParams *cp = &s->channel_params[ch];
223  int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
224  int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
225  int32_t sign_huff_offset = cp->huff_offset;
226 
227  if (cp->codebook > 0)
228  sign_huff_offset -= 7 << lsb_bits;
229 
230  if (sign_shift >= 0)
231  sign_huff_offset -= 1 << sign_shift;
232 
233  return sign_huff_offset;
234 }
235 
236 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
237  * and plain LSBs. */
238 
240  unsigned int substr, unsigned int pos)
241 {
242  SubStream *s = &m->substream[substr];
243  unsigned int mat, channel;
244 
245  for (mat = 0; mat < s->num_primitive_matrices; mat++)
246  if (s->lsb_bypass[mat])
247  m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
248 
249  for (channel = s->min_channel; channel <= s->max_channel; channel++) {
250  ChannelParams *cp = &s->channel_params[channel];
251  int codebook = cp->codebook;
252  int quant_step_size = s->quant_step_size[channel];
253  int lsb_bits = cp->huff_lsbs - quant_step_size;
254  int result = 0;
255 
256  if (codebook > 0)
257  result = get_vlc2(gbp, huff_vlc[codebook-1].table,
258  VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
259 
260  if (result < 0)
261  return AVERROR_INVALIDDATA;
262 
263  if (lsb_bits > 0)
264  result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
265 
266  result += cp->sign_huff_offset;
267  result <<= quant_step_size;
268 
269  m->sample_buffer[pos + s->blockpos][channel] = result;
270  }
271 
272  return 0;
273 }
274 
276 {
277  MLPDecodeContext *m = avctx->priv_data;
278  int substr;
279 
280  init_static();
281  m->avctx = avctx;
282  for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
283  m->substream[substr].lossless_check_data = 0xffffffff;
284  ff_mlpdsp_init(&m->dsp);
285 
286  return 0;
287 }
288 
289 /** Read a major sync info header - contains high level information about
290  * the stream - sample rate, channel arrangement etc. Most of this
291  * information is not actually necessary for decoding, only for playback.
292  */
293 
295 {
297  int substr, ret;
298 
299  if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
300  return ret;
301 
302  if (mh.group1_bits == 0) {
303  av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
304  return AVERROR_INVALIDDATA;
305  }
306  if (mh.group2_bits > mh.group1_bits) {
308  "Channel group 2 cannot have more bits per sample than group 1.\n");
309  return AVERROR_INVALIDDATA;
310  }
311 
314  "Channel groups with differing sample rates are not currently supported.\n");
315  return AVERROR_INVALIDDATA;
316  }
317 
318  if (mh.group1_samplerate == 0) {
319  av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
320  return AVERROR_INVALIDDATA;
321  }
324  "Sampling rate %d is greater than the supported maximum (%d).\n",
326  return AVERROR_INVALIDDATA;
327  }
328  if (mh.access_unit_size > MAX_BLOCKSIZE) {
330  "Block size %d is greater than the supported maximum (%d).\n",
332  return AVERROR_INVALIDDATA;
333  }
336  "Block size pow2 %d is greater than the supported maximum (%d).\n",
338  return AVERROR_INVALIDDATA;
339  }
340 
341  if (mh.num_substreams == 0)
342  return AVERROR_INVALIDDATA;
343  if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) {
344  av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
345  return AVERROR_INVALIDDATA;
346  }
347  if (mh.num_substreams > MAX_SUBSTREAMS) {
349  "%d substreams (more than the "
350  "maximum supported by the decoder)",
351  mh.num_substreams);
352  return AVERROR_PATCHWELCOME;
353  }
354 
356 
359 
361 
362  /* limit to decoding 3 substreams, as the 4th is used by Dolby Atmos for non-audio data */
364 
367 
369  if (mh.group1_bits > 16)
371  else
377 
378  m->params_valid = 1;
379  for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
380  m->substream[substr].restart_seen = 0;
381 
382  /* Set the layout for each substream. When there's more than one, the first
383  * substream is Stereo. Subsequent substreams' layouts are indicated in the
384  * major sync. */
385  if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
386  if (mh.stream_type != 0xbb) {
388  "unexpected stream_type %X in MLP",
389  mh.stream_type);
390  return AVERROR_PATCHWELCOME;
391  }
392  if ((substr = (mh.num_substreams > 1)))
394  m->substream[substr].ch_layout = mh.channel_layout_mlp;
395  } else {
396  if (mh.stream_type != 0xba) {
398  "unexpected stream_type %X in !MLP",
399  mh.stream_type);
400  return AVERROR_PATCHWELCOME;
401  }
402  if ((substr = (mh.num_substreams > 1)))
404  if (mh.num_substreams > 2)
407  else
410 
411  if (m->avctx->channels<=2 && m->substream[substr].ch_layout == AV_CH_LAYOUT_MONO && m->max_decoded_substream == 1) {
412  av_log(m->avctx, AV_LOG_DEBUG, "Mono stream with 2 substreams, ignoring 2nd\n");
413  m->max_decoded_substream = 0;
414  if (m->avctx->channels==2)
416  }
417  }
418 
419  m->needs_reordering = mh.channel_arrangement >= 18 && mh.channel_arrangement <= 20;
420 
421  /* Parse the TrueHD decoder channel modifiers and set each substream's
422  * AVMatrixEncoding accordingly.
423  *
424  * The meaning of the modifiers depends on the channel layout:
425  *
426  * - THD_CH_MODIFIER_LTRT, THD_CH_MODIFIER_LBINRBIN only apply to 2-channel
427  *
428  * - THD_CH_MODIFIER_MONO applies to 1-channel or 2-channel (dual mono)
429  *
430  * - THD_CH_MODIFIER_SURROUNDEX, THD_CH_MODIFIER_NOTSURROUNDEX only apply to
431  * layouts with an Ls/Rs channel pair
432  */
433  for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
436  if (mh.num_substreams > 2 &&
441 
442  if (mh.num_substreams > 1 &&
447 
448  if (mh.num_substreams > 0)
449  switch (mh.channel_modifier_thd_stream0) {
452  break;
455  break;
456  default:
457  break;
458  }
459  }
460 
461  return 0;
462 }
463 
464 /** Read a restart header from a block in a substream. This contains parameters
465  * required to decode the audio that do not change very often. Generally
466  * (always) present only in blocks following a major sync. */
467 
469  const uint8_t *buf, unsigned int substr)
470 {
471  SubStream *s = &m->substream[substr];
472  unsigned int ch;
473  int sync_word, tmp;
474  uint8_t checksum;
475  uint8_t lossless_check;
476  int start_count = get_bits_count(gbp);
477  int min_channel, max_channel, max_matrix_channel;
478  const int std_max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP
481 
482  sync_word = get_bits(gbp, 13);
483 
484  if (sync_word != 0x31ea >> 1) {
486  "restart header sync incorrect (got 0x%04x)\n", sync_word);
487  return AVERROR_INVALIDDATA;
488  }
489 
490  s->noise_type = get_bits1(gbp);
491 
492  if (m->avctx->codec_id == AV_CODEC_ID_MLP && s->noise_type) {
493  av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
494  return AVERROR_INVALIDDATA;
495  }
496 
497  skip_bits(gbp, 16); /* Output timestamp */
498 
499  min_channel = get_bits(gbp, 4);
500  max_channel = get_bits(gbp, 4);
501  max_matrix_channel = get_bits(gbp, 4);
502 
503  if (max_matrix_channel > std_max_matrix_channel) {
505  "Max matrix channel cannot be greater than %d.\n",
506  std_max_matrix_channel);
507  return AVERROR_INVALIDDATA;
508  }
509 
510  if (max_channel != max_matrix_channel) {
512  "Max channel must be equal max matrix channel.\n");
513  return AVERROR_INVALIDDATA;
514  }
515 
516  /* This should happen for TrueHD streams with >6 channels and MLP's noise
517  * type. It is not yet known if this is allowed. */
518  if (max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
520  "%d channels (more than the "
521  "maximum supported by the decoder)",
522  max_channel + 2);
523  return AVERROR_PATCHWELCOME;
524  }
525 
526  if (min_channel > max_channel) {
528  "Substream min channel cannot be greater than max channel.\n");
529  return AVERROR_INVALIDDATA;
530  }
531 
532  s->min_channel = min_channel;
533  s->max_channel = max_channel;
534  s->max_matrix_channel = max_matrix_channel;
535 
539  "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
540  "Further substreams will be skipped.\n",
541  s->max_channel + 1, s->ch_layout, substr);
542  m->max_decoded_substream = substr;
543  }
544 
545  s->noise_shift = get_bits(gbp, 4);
546  s->noisegen_seed = get_bits(gbp, 23);
547 
548  skip_bits(gbp, 19);
549 
550  s->data_check_present = get_bits1(gbp);
551  lossless_check = get_bits(gbp, 8);
552  if (substr == m->max_decoded_substream
553  && s->lossless_check_data != 0xffffffff) {
555  if (tmp != lossless_check)
557  "Lossless check failed - expected %02x, calculated %02x.\n",
558  lossless_check, tmp);
559  }
560 
561  skip_bits(gbp, 16);
562 
563  memset(s->ch_assign, 0, sizeof(s->ch_assign));
564 
565  for (ch = 0; ch <= s->max_matrix_channel; ch++) {
566  int ch_assign = get_bits(gbp, 6);
567  if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
568  uint64_t channel = thd_channel_layout_extract_channel(s->ch_layout,
569  ch_assign);
571  channel);
572  }
573  if (ch_assign < 0 || ch_assign > s->max_matrix_channel) {
575  "Assignment of matrix channel %d to invalid output channel %d",
576  ch, ch_assign);
577  return AVERROR_PATCHWELCOME;
578  }
579  s->ch_assign[ch_assign] = ch;
580  }
581 
582  checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
583 
584  if (checksum != get_bits(gbp, 8))
585  av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
586 
587  /* Set default decoding parameters. */
588  s->param_presence_flags = 0xff;
589  s->num_primitive_matrices = 0;
590  s->blocksize = 8;
591  s->lossless_check_data = 0;
592 
593  memset(s->output_shift , 0, sizeof(s->output_shift ));
594  memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
595 
596  for (ch = s->min_channel; ch <= s->max_channel; ch++) {
597  ChannelParams *cp = &s->channel_params[ch];
598  cp->filter_params[FIR].order = 0;
599  cp->filter_params[IIR].order = 0;
600  cp->filter_params[FIR].shift = 0;
601  cp->filter_params[IIR].shift = 0;
602 
603  /* Default audio coding is 24-bit raw PCM. */
604  cp->huff_offset = 0;
605  cp->sign_huff_offset = -(1 << 23);
606  cp->codebook = 0;
607  cp->huff_lsbs = 24;
608  }
609 
610  if (substr == m->max_decoded_substream) {
611  m->avctx->channels = s->max_matrix_channel + 1;
612  m->avctx->channel_layout = s->ch_layout;
614  s->output_shift,
617 
618  if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) {
621  int i = s->ch_assign[4];
622  s->ch_assign[4] = s->ch_assign[3];
623  s->ch_assign[3] = s->ch_assign[2];
624  s->ch_assign[2] = i;
625  } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
626  FFSWAP(int, s->ch_assign[2], s->ch_assign[4]);
627  FFSWAP(int, s->ch_assign[3], s->ch_assign[5]);
628  }
629  }
630 
631  }
632 
633  return 0;
634 }
635 
636 /** Read parameters for one of the prediction filters. */
637 
639  unsigned int substr, unsigned int channel,
640  unsigned int filter)
641 {
642  SubStream *s = &m->substream[substr];
644  const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
645  const char fchar = filter ? 'I' : 'F';
646  int i, order;
647 
648  // Filter is 0 for FIR, 1 for IIR.
649  av_assert0(filter < 2);
650 
651  if (m->filter_changed[channel][filter]++ > 1) {
652  av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
653  return AVERROR_INVALIDDATA;
654  }
655 
656  order = get_bits(gbp, 4);
657  if (order > max_order) {
659  "%cIR filter order %d is greater than maximum %d.\n",
660  fchar, order, max_order);
661  return AVERROR_INVALIDDATA;
662  }
663  fp->order = order;
664 
665  if (order > 0) {
666  int32_t *fcoeff = s->channel_params[channel].coeff[filter];
667  int coeff_bits, coeff_shift;
668 
669  fp->shift = get_bits(gbp, 4);
670 
671  coeff_bits = get_bits(gbp, 5);
672  coeff_shift = get_bits(gbp, 3);
673  if (coeff_bits < 1 || coeff_bits > 16) {
675  "%cIR filter coeff_bits must be between 1 and 16.\n",
676  fchar);
677  return AVERROR_INVALIDDATA;
678  }
679  if (coeff_bits + coeff_shift > 16) {
681  "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
682  fchar);
683  return AVERROR_INVALIDDATA;
684  }
685 
686  for (i = 0; i < order; i++)
687  fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
688 
689  if (get_bits1(gbp)) {
690  int state_bits, state_shift;
691 
692  if (filter == FIR) {
694  "FIR filter has state data specified.\n");
695  return AVERROR_INVALIDDATA;
696  }
697 
698  state_bits = get_bits(gbp, 4);
699  state_shift = get_bits(gbp, 4);
700 
701  /* TODO: Check validity of state data. */
702 
703  for (i = 0; i < order; i++)
704  fp->state[i] = state_bits ? get_sbits(gbp, state_bits) << state_shift : 0;
705  }
706  }
707 
708  return 0;
709 }
710 
711 /** Read parameters for primitive matrices. */
712 
713 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
714 {
715  SubStream *s = &m->substream[substr];
716  unsigned int mat, ch;
717  const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP
720 
721  if (m->matrix_changed++ > 1) {
722  av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
723  return AVERROR_INVALIDDATA;
724  }
725 
726  s->num_primitive_matrices = get_bits(gbp, 4);
727 
728  if (s->num_primitive_matrices > max_primitive_matrices) {
730  "Number of primitive matrices cannot be greater than %d.\n",
731  max_primitive_matrices);
732  return AVERROR_INVALIDDATA;
733  }
734 
735  for (mat = 0; mat < s->num_primitive_matrices; mat++) {
736  int frac_bits, max_chan;
737  s->matrix_out_ch[mat] = get_bits(gbp, 4);
738  frac_bits = get_bits(gbp, 4);
739  s->lsb_bypass [mat] = get_bits1(gbp);
740 
741  if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
743  "Invalid channel %d specified as output from matrix.\n",
744  s->matrix_out_ch[mat]);
745  return AVERROR_INVALIDDATA;
746  }
747  if (frac_bits > 14) {
749  "Too many fractional bits specified.\n");
750  return AVERROR_INVALIDDATA;
751  }
752 
753  max_chan = s->max_matrix_channel;
754  if (!s->noise_type)
755  max_chan+=2;
756 
757  for (ch = 0; ch <= max_chan; ch++) {
758  int coeff_val = 0;
759  if (get_bits1(gbp))
760  coeff_val = get_sbits(gbp, frac_bits + 2);
761 
762  s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
763  }
764 
765  if (s->noise_type)
766  s->matrix_noise_shift[mat] = get_bits(gbp, 4);
767  else
768  s->matrix_noise_shift[mat] = 0;
769  }
770 
771  return 0;
772 }
773 
774 /** Read channel parameters. */
775 
776 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
777  GetBitContext *gbp, unsigned int ch)
778 {
779  SubStream *s = &m->substream[substr];
780  ChannelParams *cp = &s->channel_params[ch];
781  FilterParams *fir = &cp->filter_params[FIR];
782  FilterParams *iir = &cp->filter_params[IIR];
783  int ret;
784 
786  if (get_bits1(gbp))
787  if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
788  return ret;
789 
791  if (get_bits1(gbp))
792  if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
793  return ret;
794 
795  if (fir->order + iir->order > 8) {
796  av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
797  return AVERROR_INVALIDDATA;
798  }
799 
800  if (fir->order && iir->order &&
801  fir->shift != iir->shift) {
803  "FIR and IIR filters must use the same precision.\n");
804  return AVERROR_INVALIDDATA;
805  }
806  /* The FIR and IIR filters must have the same precision.
807  * To simplify the filtering code, only the precision of the
808  * FIR filter is considered. If only the IIR filter is employed,
809  * the FIR filter precision is set to that of the IIR filter, so
810  * that the filtering code can use it. */
811  if (!fir->order && iir->order)
812  fir->shift = iir->shift;
813 
815  if (get_bits1(gbp))
816  cp->huff_offset = get_sbits(gbp, 15);
817 
818  cp->codebook = get_bits(gbp, 2);
819  cp->huff_lsbs = get_bits(gbp, 5);
820 
821  if (cp->huff_lsbs > 24) {
822  av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
823  cp->huff_lsbs = 0;
824  return AVERROR_INVALIDDATA;
825  }
826 
827  cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
828 
829  return 0;
830 }
831 
832 /** Read decoding parameters that change more often than those in the restart
833  * header. */
834 
836  unsigned int substr)
837 {
838  SubStream *s = &m->substream[substr];
839  unsigned int ch;
840  int ret;
841 
843  if (get_bits1(gbp))
844  s->param_presence_flags = get_bits(gbp, 8);
845 
847  if (get_bits1(gbp)) {
848  s->blocksize = get_bits(gbp, 9);
849  if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
850  av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n");
851  s->blocksize = 0;
852  return AVERROR_INVALIDDATA;
853  }
854  }
855 
857  if (get_bits1(gbp))
858  if ((ret = read_matrix_params(m, substr, gbp)) < 0)
859  return ret;
860 
862  if (get_bits1(gbp)) {
863  for (ch = 0; ch <= s->max_matrix_channel; ch++)
864  s->output_shift[ch] = get_sbits(gbp, 4);
865  if (substr == m->max_decoded_substream)
867  s->output_shift,
870  }
871 
873  if (get_bits1(gbp))
874  for (ch = 0; ch <= s->max_channel; ch++) {
875  ChannelParams *cp = &s->channel_params[ch];
876 
877  s->quant_step_size[ch] = get_bits(gbp, 4);
878 
879  cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
880  }
881 
882  for (ch = s->min_channel; ch <= s->max_channel; ch++)
883  if (get_bits1(gbp))
884  if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
885  return ret;
886 
887  return 0;
888 }
889 
890 #define MSB_MASK(bits) (-1u << (bits))
891 
892 /** Generate PCM samples using the prediction filters and residual values
893  * read from the data stream, and update the filter state. */
894 
895 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
896  unsigned int channel)
897 {
898  SubStream *s = &m->substream[substr];
899  const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
901  int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
902  int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
903  FilterParams *fir = &s->channel_params[channel].filter_params[FIR];
904  FilterParams *iir = &s->channel_params[channel].filter_params[IIR];
905  unsigned int filter_shift = fir->shift;
906  int32_t mask = MSB_MASK(s->quant_step_size[channel]);
907 
908  memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
909  memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
910 
911  m->dsp.mlp_filter_channel(firbuf, fircoeff,
912  fir->order, iir->order,
913  filter_shift, mask, s->blocksize,
914  &m->sample_buffer[s->blockpos][channel]);
915 
916  memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
917  memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
918 }
919 
920 /** Read a block of PCM residual data (or actual if no filtering active). */
921 
923  unsigned int substr)
924 {
925  SubStream *s = &m->substream[substr];
926  unsigned int i, ch, expected_stream_pos = 0;
927  int ret;
928 
929  if (s->data_check_present) {
930  expected_stream_pos = get_bits_count(gbp);
931  expected_stream_pos += get_bits(gbp, 16);
933  "Substreams with VLC block size check info");
934  }
935 
936  if (s->blockpos + s->blocksize > m->access_unit_size) {
937  av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
938  return AVERROR_INVALIDDATA;
939  }
940 
941  memset(&m->bypassed_lsbs[s->blockpos][0], 0,
942  s->blocksize * sizeof(m->bypassed_lsbs[0]));
943 
944  for (i = 0; i < s->blocksize; i++)
945  if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
946  return ret;
947 
948  for (ch = s->min_channel; ch <= s->max_channel; ch++)
949  filter_channel(m, substr, ch);
950 
951  s->blockpos += s->blocksize;
952 
953  if (s->data_check_present) {
954  if (get_bits_count(gbp) != expected_stream_pos)
955  av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
956  skip_bits(gbp, 8);
957  }
958 
959  return 0;
960 }
961 
962 /** Data table used for TrueHD noise generation function. */
963 
964 static const int8_t noise_table[256] = {
965  30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
966  52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
967  10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
968  51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
969  38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
970  61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
971  67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
972  48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
973  0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
974  16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
975  13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
976  89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
977  36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
978  39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
979  45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
980  -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
981 };
982 
983 /** Noise generation functions.
984  * I'm not sure what these are for - they seem to be some kind of pseudorandom
985  * sequence generators, used to generate noise data which is used when the
986  * channels are rematrixed. I'm not sure if they provide a practical benefit
987  * to compression, or just obfuscate the decoder. Are they for some kind of
988  * dithering? */
989 
990 /** Generate two channels of noise, used in the matrix when
991  * restart sync word == 0x31ea. */
992 
993 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
994 {
995  SubStream *s = &m->substream[substr];
996  unsigned int i;
997  uint32_t seed = s->noisegen_seed;
998  unsigned int maxchan = s->max_matrix_channel;
999 
1000  for (i = 0; i < s->blockpos; i++) {
1001  uint16_t seed_shr7 = seed >> 7;
1002  m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
1003  m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
1004 
1005  seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
1006  }
1007 
1008  s->noisegen_seed = seed;
1009 }
1010 
1011 /** Generate a block of noise, used when restart sync word == 0x31eb. */
1012 
1013 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
1014 {
1015  SubStream *s = &m->substream[substr];
1016  unsigned int i;
1017  uint32_t seed = s->noisegen_seed;
1018 
1019  for (i = 0; i < m->access_unit_size_pow2; i++) {
1020  uint8_t seed_shr15 = seed >> 15;
1021  m->noise_buffer[i] = noise_table[seed_shr15];
1022  seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
1023  }
1024 
1025  s->noisegen_seed = seed;
1026 }
1027 
1028 /** Write the audio data into the output buffer. */
1029 
1030 static int output_data(MLPDecodeContext *m, unsigned int substr,
1031  AVFrame *frame, int *got_frame_ptr)
1032 {
1033  AVCodecContext *avctx = m->avctx;
1034  SubStream *s = &m->substream[substr];
1035  unsigned int mat;
1036  unsigned int maxchan;
1037  int ret;
1038  int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
1039 
1040  if (m->avctx->channels != s->max_matrix_channel + 1) {
1041  av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
1042  return AVERROR_INVALIDDATA;
1043  }
1044 
1045  if (!s->blockpos) {
1046  av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
1047  return AVERROR_INVALIDDATA;
1048  }
1049 
1050  maxchan = s->max_matrix_channel;
1051  if (!s->noise_type) {
1052  generate_2_noise_channels(m, substr);
1053  maxchan += 2;
1054  } else {
1055  fill_noise_buffer(m, substr);
1056  }
1057 
1058  /* Apply the channel matrices in turn to reconstruct the original audio
1059  * samples. */
1060  for (mat = 0; mat < s->num_primitive_matrices; mat++) {
1061  unsigned int dest_ch = s->matrix_out_ch[mat];
1062  m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0],
1063  s->matrix_coeff[mat],
1064  &m->bypassed_lsbs[0][mat],
1065  m->noise_buffer,
1066  s->num_primitive_matrices - mat,
1067  dest_ch,
1068  s->blockpos,
1069  maxchan,
1070  s->matrix_noise_shift[mat],
1072  MSB_MASK(s->quant_step_size[dest_ch]));
1073  }
1074 
1075  /* get output buffer */
1076  frame->nb_samples = s->blockpos;
1077  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1078  return ret;
1080  s->blockpos,
1081  m->sample_buffer,
1082  frame->data[0],
1083  s->ch_assign,
1084  s->output_shift,
1085  s->max_matrix_channel,
1086  is32);
1087 
1088  /* Update matrix encoding side data */
1089  if ((ret = ff_side_data_update_matrix_encoding(frame, s->matrix_encoding)) < 0)
1090  return ret;
1091 
1092  *got_frame_ptr = 1;
1093 
1094  return 0;
1095 }
1096 
1097 /** Read an access unit from the stream.
1098  * @return negative on error, 0 if not enough data is present in the input stream,
1099  * otherwise the number of bytes consumed. */
1100 
1101 static int read_access_unit(AVCodecContext *avctx, void* data,
1102  int *got_frame_ptr, AVPacket *avpkt)
1103 {
1104  const uint8_t *buf = avpkt->data;
1105  int buf_size = avpkt->size;
1106  MLPDecodeContext *m = avctx->priv_data;
1107  GetBitContext gb;
1108  unsigned int length, substr;
1109  unsigned int substream_start;
1110  unsigned int header_size = 4;
1111  unsigned int substr_header_size = 0;
1112  uint8_t substream_parity_present[MAX_SUBSTREAMS];
1113  uint16_t substream_data_len[MAX_SUBSTREAMS];
1114  uint8_t parity_bits;
1115  int ret;
1116 
1117  if (buf_size < 4)
1118  return AVERROR_INVALIDDATA;
1119 
1120  length = (AV_RB16(buf) & 0xfff) * 2;
1121 
1122  if (length < 4 || length > buf_size)
1123  return AVERROR_INVALIDDATA;
1124 
1125  init_get_bits(&gb, (buf + 4), (length - 4) * 8);
1126 
1127  m->is_major_sync_unit = 0;
1128  if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
1129  if (read_major_sync(m, &gb) < 0)
1130  goto error;
1131  m->is_major_sync_unit = 1;
1132  header_size += m->major_sync_header_size;
1133  }
1134 
1135  if (!m->params_valid) {
1137  "Stream parameters not seen; skipping frame.\n");
1138  *got_frame_ptr = 0;
1139  return length;
1140  }
1141 
1142  substream_start = 0;
1143 
1144  for (substr = 0; substr < m->num_substreams; substr++) {
1145  int extraword_present, checkdata_present, end, nonrestart_substr;
1146 
1147  extraword_present = get_bits1(&gb);
1148  nonrestart_substr = get_bits1(&gb);
1149  checkdata_present = get_bits1(&gb);
1150  skip_bits1(&gb);
1151 
1152  end = get_bits(&gb, 12) * 2;
1153 
1154  substr_header_size += 2;
1155 
1156  if (extraword_present) {
1157  if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
1158  av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
1159  goto error;
1160  }
1161  skip_bits(&gb, 16);
1162  substr_header_size += 2;
1163  }
1164 
1165  if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1166  av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1167  goto error;
1168  }
1169 
1170  if (end + header_size + substr_header_size > length) {
1172  "Indicated length of substream %d data goes off end of "
1173  "packet.\n", substr);
1174  end = length - header_size - substr_header_size;
1175  }
1176 
1177  if (end < substream_start) {
1178  av_log(avctx, AV_LOG_ERROR,
1179  "Indicated end offset of substream %d data "
1180  "is smaller than calculated start offset.\n",
1181  substr);
1182  goto error;
1183  }
1184 
1185  if (substr > m->max_decoded_substream)
1186  continue;
1187 
1188  substream_parity_present[substr] = checkdata_present;
1189  substream_data_len[substr] = end - substream_start;
1190  substream_start = end;
1191  }
1192 
1193  parity_bits = ff_mlp_calculate_parity(buf, 4);
1194  parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1195 
1196  if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1197  av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1198  goto error;
1199  }
1200 
1201  buf += header_size + substr_header_size;
1202 
1203  for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1204  SubStream *s = &m->substream[substr];
1205  init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1206 
1207  m->matrix_changed = 0;
1208  memset(m->filter_changed, 0, sizeof(m->filter_changed));
1209 
1210  s->blockpos = 0;
1211  do {
1212  if (get_bits1(&gb)) {
1213  if (get_bits1(&gb)) {
1214  /* A restart header should be present. */
1215  if (read_restart_header(m, &gb, buf, substr) < 0)
1216  goto next_substr;
1217  s->restart_seen = 1;
1218  }
1219 
1220  if (!s->restart_seen)
1221  goto next_substr;
1222  if (read_decoding_params(m, &gb, substr) < 0)
1223  goto next_substr;
1224  }
1225 
1226  if (!s->restart_seen)
1227  goto next_substr;
1228 
1229  if ((ret = read_block_data(m, &gb, substr)) < 0)
1230  return ret;
1231 
1232  if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1233  goto substream_length_mismatch;
1234 
1235  } while (!get_bits1(&gb));
1236 
1237  skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1238 
1239  if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1240  int shorten_by;
1241 
1242  if (get_bits(&gb, 16) != 0xD234)
1243  return AVERROR_INVALIDDATA;
1244 
1245  shorten_by = get_bits(&gb, 16);
1246  if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by & 0x2000)
1247  s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1248  else if (m->avctx->codec_id == AV_CODEC_ID_MLP && shorten_by != 0xD234)
1249  return AVERROR_INVALIDDATA;
1250 
1251  if (substr == m->max_decoded_substream)
1252  av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1253  }
1254 
1255  if (substream_parity_present[substr]) {
1256  uint8_t parity, checksum;
1257 
1258  if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1259  goto substream_length_mismatch;
1260 
1261  parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1262  checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1263 
1264  if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1265  av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1266  if ( get_bits(&gb, 8) != checksum)
1267  av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1268  }
1269 
1270  if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1271  goto substream_length_mismatch;
1272 
1273 next_substr:
1274  if (!s->restart_seen)
1276  "No restart header present in substream %d.\n", substr);
1277 
1278  buf += substream_data_len[substr];
1279  }
1280 
1281  if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
1282  return ret;
1283 
1284  return length;
1285 
1286 substream_length_mismatch:
1287  av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1288  return AVERROR_INVALIDDATA;
1289 
1290 error:
1291  m->params_valid = 0;
1292  return AVERROR_INVALIDDATA;
1293 }
1294 
1295 #if CONFIG_MLP_DECODER
1296 AVCodec ff_mlp_decoder = {
1297  .name = "mlp",
1298  .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1299  .type = AVMEDIA_TYPE_AUDIO,
1300  .id = AV_CODEC_ID_MLP,
1301  .priv_data_size = sizeof(MLPDecodeContext),
1302  .init = mlp_decode_init,
1304  .capabilities = AV_CODEC_CAP_DR1,
1305 };
1306 #endif
1307 #if CONFIG_TRUEHD_DECODER
1308 AVCodec ff_truehd_decoder = {
1309  .name = "truehd",
1310  .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1311  .type = AVMEDIA_TYPE_AUDIO,
1312  .id = AV_CODEC_ID_TRUEHD,
1313  .priv_data_size = sizeof(MLPDecodeContext),
1314  .init = mlp_decode_init,
1316  .capabilities = AV_CODEC_CAP_DR1,
1317 };
1318 #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:391
static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout, int index)
Definition: mlpdec.c:183
const char * s
Definition: avisynth_c.h:631
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:83
This structure describes decoded (raw) audio or video data.
Definition: frame.h:181
static av_cold int mlp_decode_init(AVCodecContext *avctx)
Definition: mlpdec.c:275
#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:260
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
#define AV_CH_TOP_FRONT_LEFT
int num_substreams
Number of substreams within stream.
Definition: mlp_parser.h:62
#define AV_CH_TOP_FRONT_CENTER
int size
Definition: avcodec.h:1468
#define AV_CH_LOW_FREQUENCY_2
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:53
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:468
#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:2924
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:3392
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:245
int access_unit_size
Number of samples per coded frame.
Definition: mlp_parser.h:56
static VLC huff_vlc[3]
Definition: mlpdec.c:197
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
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
int matrix_changed
Definition: mlpdec.c:157
#define AV_CH_WIDE_LEFT
uint8_t bits
Definition: crc.c:295
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2295
uint8_t
#define av_cold
Definition: attributes.h:82
#define PARAM_BLOCKSIZE
Definition: mlpdec.c:86
MLPDSPContext dsp
Definition: mlpdec.c:164
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: get_bits.h:484
static uint8_t xor_32_to_8(uint32_t value)
XOR four bytes into one.
Definition: mlp.h:120
int channel_modifier_thd_stream0
Channel modifier for substream 0 of TrueHD sreams ("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
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, uint8_t clip)
Definition: cfhd.c:82
#define AV_CH_LOW_FREQUENCY
static AVFrame * frame
static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
Generate a block of noise, used when restart sync word == 0x31eb.
Definition: mlpdec.c:1013
uint8_t * data
Definition: avcodec.h:1467
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:212
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:993
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:638
#define PARAM_PRESENCE
Definition: mlpdec.c:93
signed 32 bits
Definition: samplefmt.h:63
#define av_log(a,...)
unsigned m
Definition: audioconvert.c:187
int16_t huff_offset
Offset to apply to residual values.
Definition: mlp.h:86
#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:776
#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:99
#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 sreams ("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
static const struct endianess table[]
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:176
#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:239
GLsizei GLsizei * length
Definition: opengl_enc.c:115
const char * name
Name of the codec implementation.
Definition: avcodec.h:3399
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:1101
Definition: get_bits.h:63
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:2338
#define MAX_BLOCKSIZE
Definition: diracdec.c:51
#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:713
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:895
#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:80
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:1030
#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:294
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:922
mcdeint parity
Definition: vf_mcdeint.c:277
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:574
#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:964
#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:84
#define AV_CH_LAYOUT_5POINT1_BACK
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:239
#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:2307
#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
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:88
#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:1549
int sample_rate
samples per second
Definition: avcodec.h:2287
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:835
int channel_modifier_thd_stream1
Channel modifier for substream 1 of TrueHD sreams ("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
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:894
#define fp
Definition: regdef.h:44
int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS]
Definition: mlpdec.c:162
void * buf
Definition: avisynth_c.h:553
filter data
Definition: mlp.h:74
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:312
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:337
#define IIR
Definition: mlp.h:71
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:305
AVCodecContext * avctx
Definition: mlpdec.c:130
#define PARAM_IIR
Definition: mlpdec.c:91
int index
Definition: gxfenc.c:89
uint64_t ch_layout
The channel layout for this substream.
Definition: mlpdec.c:69
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:418
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:89
static av_cold void init_static(void)
Initialize static data, constant between all invocations of the codec.
Definition: mlpdec.c:201
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:192
#define AV_CH_SIDE_RIGHT
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:572
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:282
uint8_t matrix_out_ch[MAX_MATRICES]
matrix output channel
Definition: mlpdec.c:103
signed 16 bits
Definition: samplefmt.h:62
int access_unit_size_pow2
next power of two above the number of samples in each frame
Definition: mlpdec.c:153
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:218
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:1574
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:82
int channels
number of audio channels
Definition: avcodec.h:2288
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:54
#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:890
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:2345
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:1444
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:235
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:856
#define mh
#define AV_CH_BACK_RIGHT
#define PARAM_FIR
Definition: mlpdec.c:90
int32_t sign_huff_offset
sign/rounding-corrected version of huff_offset
Definition: mlp.h:87
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:73