FFmpeg  3.3.9
mlpdec.c
Go to the documentation of this file.
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 
158  int filter_changed[MAX_CHANNELS][NUM_FILTERS];
159 
160  int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
161  int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
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++) {
251  int codebook = cp->codebook;
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 *= 1 << 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;
475  uint8_t lossless_check;
476  int start_count = get_bits_count(gbp);
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  noise_type = get_bits1(gbp);
491 
492  if (m->avctx->codec_id == AV_CODEC_ID_MLP && 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 && !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 
535  s->noise_type = noise_type;
536 
540  "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
541  "Further substreams will be skipped.\n",
542  s->max_channel + 1, s->ch_layout, substr);
543  m->max_decoded_substream = substr;
544  }
545 
546  s->noise_shift = get_bits(gbp, 4);
547  s->noisegen_seed = get_bits(gbp, 23);
548 
549  skip_bits(gbp, 19);
550 
551  s->data_check_present = get_bits1(gbp);
552  lossless_check = get_bits(gbp, 8);
553  if (substr == m->max_decoded_substream
554  && s->lossless_check_data != 0xffffffff) {
556  if (tmp != lossless_check)
558  "Lossless check failed - expected %02x, calculated %02x.\n",
559  lossless_check, tmp);
560  }
561 
562  skip_bits(gbp, 16);
563 
564  memset(s->ch_assign, 0, sizeof(s->ch_assign));
565 
566  for (ch = 0; ch <= s->max_matrix_channel; ch++) {
567  int ch_assign = get_bits(gbp, 6);
568  if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
570  ch_assign);
572  channel);
573  }
574  if (ch_assign < 0 || ch_assign > s->max_matrix_channel) {
576  "Assignment of matrix channel %d to invalid output channel %d",
577  ch, ch_assign);
578  return AVERROR_PATCHWELCOME;
579  }
580  s->ch_assign[ch_assign] = ch;
581  }
582 
583  checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
584 
585  if (checksum != get_bits(gbp, 8))
586  av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
587 
588  /* Set default decoding parameters. */
589  s->param_presence_flags = 0xff;
590  s->num_primitive_matrices = 0;
591  s->blocksize = 8;
592  s->lossless_check_data = 0;
593 
594  memset(s->output_shift , 0, sizeof(s->output_shift ));
595  memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
596 
597  for (ch = s->min_channel; ch <= s->max_channel; ch++) {
598  ChannelParams *cp = &s->channel_params[ch];
599  cp->filter_params[FIR].order = 0;
600  cp->filter_params[IIR].order = 0;
601  cp->filter_params[FIR].shift = 0;
602  cp->filter_params[IIR].shift = 0;
603 
604  /* Default audio coding is 24-bit raw PCM. */
605  cp->huff_offset = 0;
606  cp->sign_huff_offset = -(1 << 23);
607  cp->codebook = 0;
608  cp->huff_lsbs = 24;
609  }
610 
611  if (substr == m->max_decoded_substream) {
612  m->avctx->channels = s->max_matrix_channel + 1;
613  m->avctx->channel_layout = s->ch_layout;
615  s->output_shift,
618 
619  if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) {
622  int i = s->ch_assign[4];
623  s->ch_assign[4] = s->ch_assign[3];
624  s->ch_assign[3] = s->ch_assign[2];
625  s->ch_assign[2] = i;
626  } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
627  FFSWAP(int, s->ch_assign[2], s->ch_assign[4]);
628  FFSWAP(int, s->ch_assign[3], s->ch_assign[5]);
629  }
630  }
631 
632  }
633 
634  return 0;
635 }
636 
637 /** Read parameters for one of the prediction filters. */
638 
640  unsigned int substr, unsigned int channel,
641  unsigned int filter)
642 {
643  SubStream *s = &m->substream[substr];
645  const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
646  const char fchar = filter ? 'I' : 'F';
647  int i, order;
648 
649  // Filter is 0 for FIR, 1 for IIR.
650  av_assert0(filter < 2);
651 
652  if (m->filter_changed[channel][filter]++ > 1) {
653  av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
654  return AVERROR_INVALIDDATA;
655  }
656 
657  order = get_bits(gbp, 4);
658  if (order > max_order) {
660  "%cIR filter order %d is greater than maximum %d.\n",
661  fchar, order, max_order);
662  return AVERROR_INVALIDDATA;
663  }
664  fp->order = order;
665 
666  if (order > 0) {
667  int32_t *fcoeff = s->channel_params[channel].coeff[filter];
668  int coeff_bits, coeff_shift;
669 
670  fp->shift = get_bits(gbp, 4);
671 
672  coeff_bits = get_bits(gbp, 5);
673  coeff_shift = get_bits(gbp, 3);
674  if (coeff_bits < 1 || coeff_bits > 16) {
676  "%cIR filter coeff_bits must be between 1 and 16.\n",
677  fchar);
678  return AVERROR_INVALIDDATA;
679  }
680  if (coeff_bits + coeff_shift > 16) {
682  "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
683  fchar);
684  return AVERROR_INVALIDDATA;
685  }
686 
687  for (i = 0; i < order; i++)
688  fcoeff[i] = get_sbits(gbp, coeff_bits) * (1 << coeff_shift);
689 
690  if (get_bits1(gbp)) {
691  int state_bits, state_shift;
692 
693  if (filter == FIR) {
695  "FIR filter has state data specified.\n");
696  return AVERROR_INVALIDDATA;
697  }
698 
699  state_bits = get_bits(gbp, 4);
700  state_shift = get_bits(gbp, 4);
701 
702  /* TODO: Check validity of state data. */
703 
704  for (i = 0; i < order; i++)
705  fp->state[i] = state_bits ? get_sbits(gbp, state_bits) * (1 << state_shift) : 0;
706  }
707  }
708 
709  return 0;
710 }
711 
712 /** Read parameters for primitive matrices. */
713 
714 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
715 {
716  SubStream *s = &m->substream[substr];
717  unsigned int mat, ch;
718  const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP
721 
722  if (m->matrix_changed++ > 1) {
723  av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
724  return AVERROR_INVALIDDATA;
725  }
726 
727  s->num_primitive_matrices = get_bits(gbp, 4);
728 
729  if (s->num_primitive_matrices > max_primitive_matrices) {
731  "Number of primitive matrices cannot be greater than %d.\n",
732  max_primitive_matrices);
733  goto error;
734  }
735 
736  for (mat = 0; mat < s->num_primitive_matrices; mat++) {
737  int frac_bits, max_chan;
738  s->matrix_out_ch[mat] = get_bits(gbp, 4);
739  frac_bits = get_bits(gbp, 4);
740  s->lsb_bypass [mat] = get_bits1(gbp);
741 
742  if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
744  "Invalid channel %d specified as output from matrix.\n",
745  s->matrix_out_ch[mat]);
746  goto error;
747  }
748  if (frac_bits > 14) {
750  "Too many fractional bits specified.\n");
751  goto error;
752  }
753 
754  max_chan = s->max_matrix_channel;
755  if (!s->noise_type)
756  max_chan+=2;
757 
758  for (ch = 0; ch <= max_chan; ch++) {
759  int coeff_val = 0;
760  if (get_bits1(gbp))
761  coeff_val = get_sbits(gbp, frac_bits + 2);
762 
763  s->matrix_coeff[mat][ch] = coeff_val * (1 << (14 - frac_bits));
764  }
765 
766  if (s->noise_type)
767  s->matrix_noise_shift[mat] = get_bits(gbp, 4);
768  else
769  s->matrix_noise_shift[mat] = 0;
770  }
771 
772  return 0;
773 error:
774  s->num_primitive_matrices = 0;
775  memset(s->matrix_out_ch, 0, sizeof(s->matrix_out_ch));
776 
777  return AVERROR_INVALIDDATA;
778 }
779 
780 /** Read channel parameters. */
781 
782 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
783  GetBitContext *gbp, unsigned int ch)
784 {
785  SubStream *s = &m->substream[substr];
786  ChannelParams *cp = &s->channel_params[ch];
787  FilterParams *fir = &cp->filter_params[FIR];
788  FilterParams *iir = &cp->filter_params[IIR];
789  int ret;
790 
792  if (get_bits1(gbp))
793  if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
794  return ret;
795 
797  if (get_bits1(gbp))
798  if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
799  return ret;
800 
801  if (fir->order + iir->order > 8) {
802  av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
803  return AVERROR_INVALIDDATA;
804  }
805 
806  if (fir->order && iir->order &&
807  fir->shift != iir->shift) {
809  "FIR and IIR filters must use the same precision.\n");
810  return AVERROR_INVALIDDATA;
811  }
812  /* The FIR and IIR filters must have the same precision.
813  * To simplify the filtering code, only the precision of the
814  * FIR filter is considered. If only the IIR filter is employed,
815  * the FIR filter precision is set to that of the IIR filter, so
816  * that the filtering code can use it. */
817  if (!fir->order && iir->order)
818  fir->shift = iir->shift;
819 
821  if (get_bits1(gbp))
822  cp->huff_offset = get_sbits(gbp, 15);
823 
824  cp->codebook = get_bits(gbp, 2);
825  cp->huff_lsbs = get_bits(gbp, 5);
826 
827  if (cp->huff_lsbs > 24) {
828  av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
829  cp->huff_lsbs = 0;
830  return AVERROR_INVALIDDATA;
831  }
832 
833  cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
834 
835  return 0;
836 }
837 
838 /** Read decoding parameters that change more often than those in the restart
839  * header. */
840 
842  unsigned int substr)
843 {
844  SubStream *s = &m->substream[substr];
845  unsigned int ch;
846  int ret;
847 
849  if (get_bits1(gbp))
850  s->param_presence_flags = get_bits(gbp, 8);
851 
853  if (get_bits1(gbp)) {
854  s->blocksize = get_bits(gbp, 9);
855  if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
856  av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n");
857  s->blocksize = 0;
858  return AVERROR_INVALIDDATA;
859  }
860  }
861 
863  if (get_bits1(gbp))
864  if ((ret = read_matrix_params(m, substr, gbp)) < 0)
865  return ret;
866 
868  if (get_bits1(gbp)) {
869  for (ch = 0; ch <= s->max_matrix_channel; ch++)
870  s->output_shift[ch] = get_sbits(gbp, 4);
871  if (substr == m->max_decoded_substream)
873  s->output_shift,
876  }
877 
879  if (get_bits1(gbp))
880  for (ch = 0; ch <= s->max_channel; ch++) {
881  ChannelParams *cp = &s->channel_params[ch];
882 
883  s->quant_step_size[ch] = get_bits(gbp, 4);
884 
885  cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
886  }
887 
888  for (ch = s->min_channel; ch <= s->max_channel; ch++)
889  if (get_bits1(gbp))
890  if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
891  return ret;
892 
893  return 0;
894 }
895 
896 #define MSB_MASK(bits) (-1u << (bits))
897 
898 /** Generate PCM samples using the prediction filters and residual values
899  * read from the data stream, and update the filter state. */
900 
901 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
902  unsigned int channel)
903 {
904  SubStream *s = &m->substream[substr];
905  const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
907  int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
908  int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
911  unsigned int filter_shift = fir->shift;
912  int32_t mask = MSB_MASK(s->quant_step_size[channel]);
913 
914  memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
915  memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
916 
917  m->dsp.mlp_filter_channel(firbuf, fircoeff,
918  fir->order, iir->order,
919  filter_shift, mask, s->blocksize,
920  &m->sample_buffer[s->blockpos][channel]);
921 
922  memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
923  memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
924 }
925 
926 /** Read a block of PCM residual data (or actual if no filtering active). */
927 
929  unsigned int substr)
930 {
931  SubStream *s = &m->substream[substr];
932  unsigned int i, ch, expected_stream_pos = 0;
933  int ret;
934 
935  if (s->data_check_present) {
936  expected_stream_pos = get_bits_count(gbp);
937  expected_stream_pos += get_bits(gbp, 16);
939  "Substreams with VLC block size check info");
940  }
941 
942  if (s->blockpos + s->blocksize > m->access_unit_size) {
943  av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
944  return AVERROR_INVALIDDATA;
945  }
946 
947  memset(&m->bypassed_lsbs[s->blockpos][0], 0,
948  s->blocksize * sizeof(m->bypassed_lsbs[0]));
949 
950  for (i = 0; i < s->blocksize; i++)
951  if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
952  return ret;
953 
954  for (ch = s->min_channel; ch <= s->max_channel; ch++)
955  filter_channel(m, substr, ch);
956 
957  s->blockpos += s->blocksize;
958 
959  if (s->data_check_present) {
960  if (get_bits_count(gbp) != expected_stream_pos)
961  av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
962  skip_bits(gbp, 8);
963  }
964 
965  return 0;
966 }
967 
968 /** Data table used for TrueHD noise generation function. */
969 
970 static const int8_t noise_table[256] = {
971  30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
972  52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
973  10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
974  51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
975  38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
976  61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
977  67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
978  48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
979  0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
980  16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
981  13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
982  89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
983  36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
984  39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
985  45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
986  -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
987 };
988 
989 /** Noise generation functions.
990  * I'm not sure what these are for - they seem to be some kind of pseudorandom
991  * sequence generators, used to generate noise data which is used when the
992  * channels are rematrixed. I'm not sure if they provide a practical benefit
993  * to compression, or just obfuscate the decoder. Are they for some kind of
994  * dithering? */
995 
996 /** Generate two channels of noise, used in the matrix when
997  * restart sync word == 0x31ea. */
998 
999 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
1000 {
1001  SubStream *s = &m->substream[substr];
1002  unsigned int i;
1003  uint32_t seed = s->noisegen_seed;
1004  unsigned int maxchan = s->max_matrix_channel;
1005 
1006  for (i = 0; i < s->blockpos; i++) {
1007  uint16_t seed_shr7 = seed >> 7;
1008  m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) * (1 << s->noise_shift);
1009  m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) * (1 << s->noise_shift);
1010 
1011  seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
1012  }
1013 
1014  s->noisegen_seed = seed;
1015 }
1016 
1017 /** Generate a block of noise, used when restart sync word == 0x31eb. */
1018 
1019 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
1020 {
1021  SubStream *s = &m->substream[substr];
1022  unsigned int i;
1023  uint32_t seed = s->noisegen_seed;
1024 
1025  for (i = 0; i < m->access_unit_size_pow2; i++) {
1026  uint8_t seed_shr15 = seed >> 15;
1027  m->noise_buffer[i] = noise_table[seed_shr15];
1028  seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
1029  }
1030 
1031  s->noisegen_seed = seed;
1032 }
1033 
1034 /** Write the audio data into the output buffer. */
1035 
1036 static int output_data(MLPDecodeContext *m, unsigned int substr,
1037  AVFrame *frame, int *got_frame_ptr)
1038 {
1039  AVCodecContext *avctx = m->avctx;
1040  SubStream *s = &m->substream[substr];
1041  unsigned int mat;
1042  unsigned int maxchan;
1043  int ret;
1044  int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
1045 
1046  if (m->avctx->channels != s->max_matrix_channel + 1) {
1047  av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
1048  return AVERROR_INVALIDDATA;
1049  }
1050 
1051  if (!s->blockpos) {
1052  av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
1053  return AVERROR_INVALIDDATA;
1054  }
1055 
1056  maxchan = s->max_matrix_channel;
1057  if (!s->noise_type) {
1058  generate_2_noise_channels(m, substr);
1059  maxchan += 2;
1060  } else {
1061  fill_noise_buffer(m, substr);
1062  }
1063 
1064  /* Apply the channel matrices in turn to reconstruct the original audio
1065  * samples. */
1066  for (mat = 0; mat < s->num_primitive_matrices; mat++) {
1067  unsigned int dest_ch = s->matrix_out_ch[mat];
1068  m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0],
1069  s->matrix_coeff[mat],
1070  &m->bypassed_lsbs[0][mat],
1071  m->noise_buffer,
1072  s->num_primitive_matrices - mat,
1073  dest_ch,
1074  s->blockpos,
1075  maxchan,
1076  s->matrix_noise_shift[mat],
1078  MSB_MASK(s->quant_step_size[dest_ch]));
1079  }
1080 
1081  /* get output buffer */
1082  frame->nb_samples = s->blockpos;
1083  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1084  return ret;
1086  s->blockpos,
1087  m->sample_buffer,
1088  frame->data[0],
1089  s->ch_assign,
1090  s->output_shift,
1091  s->max_matrix_channel,
1092  is32);
1093 
1094  /* Update matrix encoding side data */
1095  if ((ret = ff_side_data_update_matrix_encoding(frame, s->matrix_encoding)) < 0)
1096  return ret;
1097 
1098  *got_frame_ptr = 1;
1099 
1100  return 0;
1101 }
1102 
1103 /** Read an access unit from the stream.
1104  * @return negative on error, 0 if not enough data is present in the input stream,
1105  * otherwise the number of bytes consumed. */
1106 
1107 static int read_access_unit(AVCodecContext *avctx, void* data,
1108  int *got_frame_ptr, AVPacket *avpkt)
1109 {
1110  const uint8_t *buf = avpkt->data;
1111  int buf_size = avpkt->size;
1112  MLPDecodeContext *m = avctx->priv_data;
1113  GetBitContext gb;
1114  unsigned int length, substr;
1115  unsigned int substream_start;
1116  unsigned int header_size = 4;
1117  unsigned int substr_header_size = 0;
1118  uint8_t substream_parity_present[MAX_SUBSTREAMS];
1119  uint16_t substream_data_len[MAX_SUBSTREAMS];
1120  uint8_t parity_bits;
1121  int ret;
1122 
1123  if (buf_size < 4)
1124  return AVERROR_INVALIDDATA;
1125 
1126  length = (AV_RB16(buf) & 0xfff) * 2;
1127 
1128  if (length < 4 || length > buf_size)
1129  return AVERROR_INVALIDDATA;
1130 
1131  init_get_bits(&gb, (buf + 4), (length - 4) * 8);
1132 
1133  m->is_major_sync_unit = 0;
1134  if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
1135  if (read_major_sync(m, &gb) < 0)
1136  goto error;
1137  m->is_major_sync_unit = 1;
1138  header_size += m->major_sync_header_size;
1139  }
1140 
1141  if (!m->params_valid) {
1143  "Stream parameters not seen; skipping frame.\n");
1144  *got_frame_ptr = 0;
1145  return length;
1146  }
1147 
1148  substream_start = 0;
1149 
1150  for (substr = 0; substr < m->num_substreams; substr++) {
1151  int extraword_present, checkdata_present, end, nonrestart_substr;
1152 
1153  extraword_present = get_bits1(&gb);
1154  nonrestart_substr = get_bits1(&gb);
1155  checkdata_present = get_bits1(&gb);
1156  skip_bits1(&gb);
1157 
1158  end = get_bits(&gb, 12) * 2;
1159 
1160  substr_header_size += 2;
1161 
1162  if (extraword_present) {
1163  if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
1164  av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
1165  goto error;
1166  }
1167  skip_bits(&gb, 16);
1168  substr_header_size += 2;
1169  }
1170 
1171  if (length < header_size + substr_header_size) {
1172  av_log(m->avctx, AV_LOG_ERROR, "Insuffient data for headers\n");
1173  goto error;
1174  }
1175 
1176  if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1177  av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1178  goto error;
1179  }
1180 
1181  if (end + header_size + substr_header_size > length) {
1183  "Indicated length of substream %d data goes off end of "
1184  "packet.\n", substr);
1185  end = length - header_size - substr_header_size;
1186  }
1187 
1188  if (end < substream_start) {
1189  av_log(avctx, AV_LOG_ERROR,
1190  "Indicated end offset of substream %d data "
1191  "is smaller than calculated start offset.\n",
1192  substr);
1193  goto error;
1194  }
1195 
1196  if (substr > m->max_decoded_substream)
1197  continue;
1198 
1199  substream_parity_present[substr] = checkdata_present;
1200  substream_data_len[substr] = end - substream_start;
1201  substream_start = end;
1202  }
1203 
1204  parity_bits = ff_mlp_calculate_parity(buf, 4);
1205  parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1206 
1207  if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1208  av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1209  goto error;
1210  }
1211 
1212  buf += header_size + substr_header_size;
1213 
1214  for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1215  SubStream *s = &m->substream[substr];
1216  init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1217 
1218  m->matrix_changed = 0;
1219  memset(m->filter_changed, 0, sizeof(m->filter_changed));
1220 
1221  s->blockpos = 0;
1222  do {
1223  if (get_bits1(&gb)) {
1224  if (get_bits1(&gb)) {
1225  /* A restart header should be present. */
1226  if (read_restart_header(m, &gb, buf, substr) < 0)
1227  goto next_substr;
1228  s->restart_seen = 1;
1229  }
1230 
1231  if (!s->restart_seen)
1232  goto next_substr;
1233  if (read_decoding_params(m, &gb, substr) < 0)
1234  goto next_substr;
1235  }
1236 
1237  if (!s->restart_seen)
1238  goto next_substr;
1239 
1240  if ((ret = read_block_data(m, &gb, substr)) < 0)
1241  return ret;
1242 
1243  if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1244  goto substream_length_mismatch;
1245 
1246  } while (!get_bits1(&gb));
1247 
1248  skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1249 
1250  if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1251  int shorten_by;
1252 
1253  if (get_bits(&gb, 16) != 0xD234)
1254  return AVERROR_INVALIDDATA;
1255 
1256  shorten_by = get_bits(&gb, 16);
1257  if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by & 0x2000)
1258  s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1259  else if (m->avctx->codec_id == AV_CODEC_ID_MLP && shorten_by != 0xD234)
1260  return AVERROR_INVALIDDATA;
1261 
1262  if (substr == m->max_decoded_substream)
1263  av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1264  }
1265 
1266  if (substream_parity_present[substr]) {
1268 
1269  if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1270  goto substream_length_mismatch;
1271 
1272  parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1273  checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1274 
1275  if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1276  av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1277  if ( get_bits(&gb, 8) != checksum)
1278  av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1279  }
1280 
1281  if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1282  goto substream_length_mismatch;
1283 
1284 next_substr:
1285  if (!s->restart_seen)
1287  "No restart header present in substream %d.\n", substr);
1288 
1289  buf += substream_data_len[substr];
1290  }
1291 
1292  if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
1293  return ret;
1294 
1295  return length;
1296 
1297 substream_length_mismatch:
1298  av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1299  return AVERROR_INVALIDDATA;
1300 
1301 error:
1302  m->params_valid = 0;
1303  return AVERROR_INVALIDDATA;
1304 }
1305 
1306 #if CONFIG_MLP_DECODER
1307 AVCodec ff_mlp_decoder = {
1308  .name = "mlp",
1309  .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1310  .type = AVMEDIA_TYPE_AUDIO,
1311  .id = AV_CODEC_ID_MLP,
1312  .priv_data_size = sizeof(MLPDecodeContext),
1313  .init = mlp_decode_init,
1315  .capabilities = AV_CODEC_CAP_DR1,
1316 };
1317 #endif
1318 #if CONFIG_TRUEHD_DECODER
1319 AVCodec ff_truehd_decoder = {
1320  .name = "truehd",
1321  .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1322  .type = AVMEDIA_TYPE_AUDIO,
1323  .id = AV_CODEC_ID_TRUEHD,
1324  .priv_data_size = sizeof(MLPDecodeContext),
1325  .init = mlp_decode_init,
1327  .capabilities = AV_CODEC_CAP_DR1,
1328 };
1329 #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:398
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:768
int major_sync_header_size
Size of the major sync unit, in bytes.
Definition: mlpdec.c:136
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
#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:187
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
static av_cold int mlp_decode_init(AVCodecContext *avctx)
Definition: mlpdec.c:275
#define AV_CH_TOP_FRONT_RIGHT
int8_t noise_buffer[MAX_BLOCKSIZE_POW2]
Definition: mlpdec.c:160
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:262
#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 INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: vlc.h:75
#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:1658
#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: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:3133
AVCodec.
Definition: avcodec.h:3681
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:247
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:296
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2502
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
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:80
#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:1019
const char data[16]
Definition: mxf.c:90
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:104
uint8_t * data
Definition: avcodec.h:1657
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:200
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:999
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:639
#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:782
#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
#define AV_RB16
Definition: intreadwrite.h:53
uint8_t needs_reordering
Stream needs channel reordering to comply with FFmpeg&#39;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:179
#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
const char * name
Name of the codec implementation.
Definition: avcodec.h:3688
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
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:1107
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:2545
#define MAX_BLOCKSIZE
Definition: diracdec.c:53
#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:714
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:901
#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:1036
#define FFMIN(a, b)
Definition: common.h:96
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
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
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:928
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:556
#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:970
int filter_changed[MAX_CHANNELS][NUM_FILTERS]
Definition: mlpdec.c:158
#define AV_CH_FRONT_CENTER
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:28
#define AV_CH_LAYOUT_5POINT1_BACK
if(ret< 0)
Definition: vf_mcdeint.c:282
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:240
#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:2514
#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: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:1749
int sample_rate
samples per second
Definition: avcodec.h:2494
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:841
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:1732
#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:956
#define fp
Definition: regdef.h:44
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:314
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:339
#define IIR
Definition: mlp.h:71
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:307
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:426
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: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:201
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { 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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_YASM &&HAVE_MMX) 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);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=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) { int planes=out->planar ? out->ch_count :1;unsigned m=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){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
#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.
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: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:1774
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:2495
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
#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
static int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt)
Definition: ffmpeg.c:2261
#define MSB_MASK(bits)
Definition: mlpdec.c:896
AVMatrixEncoding
#define AV_CH_SIDE_LEFT
#define FFSWAP(type, a, b)
Definition: common.h:99
const char int length
Definition: avisynth_c.h:768
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:2552
This structure stores compressed data.
Definition: avcodec.h:1634
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:244
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:994
#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: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