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
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
48 #define VLC_STATIC_SIZE 512
49 #endif
50
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.
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
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).
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
118 /// Number of PCM samples decoded so far in this frame.
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
128
131
132 /// Current access unit being read has a major sync.
134
135 /// Set if a valid major sync block has been read. Otherwise no decoding is possible.
137
138 /// Number of substreams contained within this stream.
140
141 /// Index of the last substream to decode - further substreams are skipped.
143
144 /// Stream needs channel reordering to comply with FFmpeg's channel order
146
147 /// number of PCM samples contained in each frame
149 /// next power of two above the number of samples in each frame
151
153
156
160
163
178 };
179
182 {
183 int i;
184
186 return 0;
187
191 return 0;
192 }
193
195
196 /** Initialize static data, constant between all invocations of the codec. */
197
199 {
200 if (!huff_vlc[0].
bits) {
210 }
211
213 }
214
216 unsigned int substr, unsigned int ch)
217 {
223
225 sign_huff_offset -= 7 << lsb_bits;
226
227 if (sign_shift >= 0)
228 sign_huff_offset -= 1 << sign_shift;
229
230 return sign_huff_offset;
231 }
232
233 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
234 * and plain LSBs. */
235
237 unsigned int substr, unsigned int pos)
238 {
240 unsigned int mat, channel;
241
245
246 for (channel = s->
min_channel; channel <= s->max_channel; channel++) {
250 int lsb_bits = cp->
huff_lsbs - quant_step_size;
251 int result = 0;
252
253 if (codebook > 0)
256
257 if (result < 0)
259
260 if (lsb_bits > 0)
261 result = (result << lsb_bits) +
get_bits(gbp, lsb_bits);
262
264 result <<= quant_step_size;
265
267 }
268
269 return 0;
270 }
271
273 {
275 int substr;
276
282
283 return 0;
284 }
285
286 /** Read a major sync info header - contains high level information about
287 * the stream - sample rate, channel arrangement etc. Most of this
288 * information is not actually necessary for decoding, only for playback.
289 */
290
292 {
295
298
302 }
305 "Channel group 2 cannot have more bits per sample than group 1.\n");
307 }
308
311 "Channel groups with differing sample rates are not currently supported.\n");
313 }
314
318 }
321 "Sampling rate %d is greater than the supported maximum (%d).\n",
324 }
327 "Block size %d is greater than the supported maximum (%d).\n",
330 }
333 "Block size pow2 %d is greater than the supported maximum (%d).\n",
336 }
337
343 }
346 "%d substreams (more than the "
347 "maximum supported by the decoder)",
350 }
351
354
357
360
364 else
370
374
375 /* Set the layout for each substream. When there's more than one, the first
376 * substream is Stereo. Subsequent substreams' layouts are indicated in the
377 * major sync. */
381 "unexpected stream_type %X in MLP",
384 }
388 } else {
391 "unexpected stream_type %X in !MLP",
394 }
400 else
403
409 }
410 }
411
413
414 /* Parse the TrueHD decoder channel modifiers and set each substream's
415 * AVMatrixEncoding accordingly.
416 *
417 * The meaning of the modifiers depends on the channel layout:
418 *
419 * - THD_CH_MODIFIER_LTRT, THD_CH_MODIFIER_LBINRBIN only apply to 2-channel
420 *
421 * - THD_CH_MODIFIER_MONO applies to 1-channel or 2-channel (dual mono)
422 *
423 * - THD_CH_MODIFIER_SURROUNDEX, THD_CH_MODIFIER_NOTSURROUNDEX only apply to
424 * layouts with an Ls/Rs channel pair
425 */
434
440
445 break;
448 break;
449 default:
450 break;
451 }
452 }
453
454 return 0;
455 }
456
457 /** Read a restart header from a block in a substream. This contains parameters
458 * required to decode the audio that do not change very often. Generally
459 * (always) present only in blocks following a major sync. */
460
463 {
465 unsigned int ch;
466 int sync_word, tmp;
470 int min_channel, max_channel, max_matrix_channel;
474
476
477 if (sync_word != 0x31ea >> 1) {
479 "restart header sync incorrect (got 0x%04x)\n", sync_word);
481 }
482
484
488 }
489
490 skip_bits(gbp, 16);
/* Output timestamp */
491
494 max_matrix_channel =
get_bits(gbp, 4);
495
496 if (max_matrix_channel > std_max_matrix_channel) {
498 "Max matrix channel cannot be greater than %d.\n",
499 std_max_matrix_channel);
501 }
502
503 if (max_channel != max_matrix_channel) {
505 "Max channel must be equal max matrix channel.\n");
507 }
508
509 /* This should happen for TrueHD streams with >6 channels and MLP's noise
510 * type. It is not yet known if this is allowed. */
513 "%d channels (more than the "
514 "maximum supported by the decoder)",
515 max_channel + 2);
517 }
518
519 if (min_channel > max_channel) {
521 "Substream min channel cannot be greater than max channel.\n");
523 }
524
528
529 #if FF_API_REQUEST_CHANNELS
535 "Extracting %d-channel downmix from substream %d. "
536 "Further substreams will be skipped.\n",
540 } else
541 #endif
545 "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
546 "Further substreams will be skipped.\n",
549 }
550
553
555
561 if (tmp != lossless_check)
563 "Lossless check failed - expected %02x, calculated %02x.\n",
564 lossless_check, tmp);
565 }
566
568
570
575 ch_assign);
577 channel);
578 }
581 "Assignment of matrix channel %d to invalid output channel %d",
582 ch, ch_assign);
584 }
586 }
587
589
592
593 /* Set default decoding parameters. */
598
601
602 for (ch = s->
min_channel; ch <= s->max_channel; ch++) {
608
609 /* Default audio coding is 24-bit raw PCM. */
614 }
615
623
634 }
635 }
636
637 }
638
639 return 0;
640 }
641
642 /** Read parameters for one of the prediction filters. */
643
645 unsigned int substr, unsigned int channel,
647 {
651 const char fchar = filter ? 'I' : 'F';
652 int i, order;
653
654 // Filter is 0 for FIR, 1 for IIR.
656
660 }
661
663 if (order > max_order) {
665 "%cIR filter order %d is greater than maximum %d.\n",
666 fchar, order, max_order);
668 }
670
671 if (order > 0) {
673 int coeff_bits, coeff_shift;
674
676
679 if (coeff_bits < 1 || coeff_bits > 16) {
681 "%cIR filter coeff_bits must be between 1 and 16.\n",
682 fchar);
684 }
685 if (coeff_bits + coeff_shift > 16) {
687 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
688 fchar);
690 }
691
692 for (i = 0; i < order; i++)
693 fcoeff[i] =
get_sbits(gbp, coeff_bits) << coeff_shift;
694
696 int state_bits, state_shift;
697
700 "FIR filter has state data specified.\n");
702 }
703
706
707 /* TODO: Check validity of state data. */
708
709 for (i = 0; i < order; i++)
710 fp->
state[i] = state_bits ?
get_sbits(gbp, state_bits) << state_shift : 0;
711 }
712 }
713
714 return 0;
715 }
716
717 /** Read parameters for primitive matrices. */
718
720 {
722 unsigned int mat, ch;
726
730 }
731
733
736 "Number of primitive matrices cannot be greater than %d.\n",
737 max_primitive_matrices);
739 }
740
742 int frac_bits, max_chan;
746
749 "Invalid channel %d specified as output from matrix.\n",
752 }
753 if (frac_bits > 14) {
755 "Too many fractional bits specified.\n");
757 }
758
761 max_chan+=2;
762
763 for (ch = 0; ch <= max_chan; ch++) {
764 int coeff_val = 0;
766 coeff_val =
get_sbits(gbp, frac_bits + 2);
767
768 s->
matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
769 }
770
773 else
775 }
776
777 return 0;
778 }
779
780 /** Read channel parameters. */
781
784 {
790
795
800
804 }
805
809 "FIR and IIR filters must use the same precision.\n");
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. */
819
823
826
831 }
832
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 {
845 unsigned int ch;
847
851
859 }
860 }
861
866
876 }
877
882
884
886 }
887
888 for (ch = s->
min_channel; ch <= s->max_channel; ch++)
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
902 unsigned int channel)
903 {
911 unsigned int filter_shift = fir->
shift;
913
916
921
924 }
925
926 /** Read a block of PCM residual data (or actual if no filtering active). */
927
929 unsigned int substr)
930 {
932 unsigned int i, ch, expected_stream_pos = 0;
934
937 expected_stream_pos +=
get_bits(gbp, 16);
939 "Substreams with VLC block size check info");
940 }
941
945 }
946
949
953
954 for (ch = s->
min_channel; ch <= s->max_channel; ch++)
956
958
963 }
964
965 return 0;
966 }
967
968 /** Data table used for TrueHD noise generation function. */
969
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
1000 {
1002 unsigned int i;
1005
1006 for (i = 0; i < s->
blockpos; i++) {
1007 uint16_t seed_shr7 = seed >> 7;
1010
1011 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
1012 }
1013
1015 }
1016
1017 /** Generate a block of noise, used when restart sync word == 0x31eb. */
1018
1020 {
1022 unsigned int i;
1024
1026 uint8_t seed_shr15 = seed >> 15;
1028 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
1029 }
1030
1032 }
1033
1034
1035 /** Apply the channel matrices in turn to reconstruct the original audio
1036 * samples. */
1037
1039 {
1041 unsigned int mat;
1042 unsigned int maxchan;
1043
1047 maxchan += 2;
1048 } else {
1050 }
1051
1059 dest_ch,
1061 maxchan,
1065 }
1066 }
1067
1068 /** Write the audio data into the output buffer. */
1069
1072 {
1077
1081 }
1082
1086 }
1087
1088 /* get output buffer */
1099 is32);
1100
1101 /* Update matrix encoding side data */
1104
1105 *got_frame_ptr = 1;
1106
1107 return 0;
1108 }
1109
1110 /** Read an access unit from the stream.
1111 * @return negative on error, 0 if not enough data is present in the input stream,
1112 * otherwise the number of bytes consumed. */
1113
1115 int *got_frame_ptr,
AVPacket *avpkt)
1116 {
1118 int buf_size = avpkt->
size;
1121 unsigned int length, substr;
1122 unsigned int substream_start;
1123 unsigned int header_size = 4;
1124 unsigned int substr_header_size = 0;
1129
1130 if (buf_size < 4)
1132
1133 length = (
AV_RB16(buf) & 0xfff) * 2;
1134
1137
1139
1143 goto error;
1145 header_size += 28;
1146 }
1147
1150 "Stream parameters not seen; skipping frame.\n");
1151 *got_frame_ptr = 0;
1153 }
1154
1155 substream_start = 0;
1156
1158 int extraword_present, checkdata_present,
end, nonrestart_substr;
1159
1164
1166
1167 substr_header_size += 2;
1168
1169 if (extraword_present) {
1172 goto error;
1173 }
1175 substr_header_size += 2;
1176 }
1177
1180 goto error;
1181 }
1182
1183 if (end + header_size + substr_header_size > length) {
1185 "Indicated length of substream %d data goes off end of "
1186 "packet.\n", substr);
1187 end = length - header_size - substr_header_size;
1188 }
1189
1190 if (end < substream_start) {
1192 "Indicated end offset of substream %d data "
1193 "is smaller than calculated start offset.\n",
1194 substr);
1195 goto error;
1196 }
1197
1199 continue;
1200
1201 substream_parity_present[substr] = checkdata_present;
1202 substream_data_len[substr] = end - substream_start;
1203 substream_start =
end;
1204 }
1205
1208
1209 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1211 goto error;
1212 }
1213
1214 buf += header_size + substr_header_size;
1215
1219
1222
1224 do {
1227 /* A restart header should be present. */
1229 goto next_substr;
1231 }
1232
1234 goto next_substr;
1236 goto next_substr;
1237 }
1238
1240 goto next_substr;
1241
1244
1246 goto substream_length_mismatch;
1247
1249
1251
1252 if (substream_data_len[substr] * 8 -
get_bits_count(&gb) >= 32) {
1253 int shorten_by;
1254
1257
1263
1266 }
1267
1268 if (substream_parity_present[substr]) {
1270
1272 goto substream_length_mismatch;
1273
1276
1277 if ((
get_bits(&gb, 8) ^ parity) != 0xa9 )
1281 }
1282
1284 goto substream_length_mismatch;
1285
1286 next_substr:
1289 "No restart header present in substream %d.\n", substr);
1290
1291 buf += substream_data_len[substr];
1292 }
1293
1295
1298
1300
1301 substream_length_mismatch:
1304
1305 error:
1308 }
1309
1310 #if CONFIG_MLP_DECODER
1320 };
1321 #endif
1322 #if CONFIG_TRUEHD_DECODER
1332 };
1333 #endif /* CONFIG_TRUEHD_DECODER */