Add 3 channel layout convenience macros.
[libav.git] / libavcodec / mlpdec.c
CommitLineData
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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/**
bad5537e 23 * @file libavcodec/mlpdec.c
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24 * MLP decoder
25 */
26
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27#include <stdint.h>
28
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29#include "avcodec.h"
30#include "libavutil/intreadwrite.h"
9106a698 31#include "get_bits.h"
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32#include "libavutil/crc.h"
33#include "parser.h"
34#include "mlp_parser.h"
ce15710f 35#include "mlp.h"
b517af05 36
9906a2be 37/** number of bits used for VLC lookup - longest Huffman code is 9 */
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38#define VLC_BITS 9
39
40
41static const char* sample_message =
42 "Please file a bug report following the instructions at "
0fa5f24c 43 "http://ffmpeg.org/bugreports.html and include "
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44 "a sample of this file.";
45
46typedef struct SubStream {
9906a2be 47 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
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48 uint8_t restart_seen;
49
50 //@{
51 /** restart header data */
52 //! The type of noise to be used in the rematrix stage.
53 uint16_t noise_type;
54
55 //! The index of the first channel coded in this substream.
56 uint8_t min_channel;
57 //! The index of the last channel coded in this substream.
58 uint8_t max_channel;
59 //! The number of channels input into the rematrix stage.
60 uint8_t max_matrix_channel;
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61 //! For each channel output by the matrix, the output channel to map it to
62 uint8_t ch_assign[MAX_CHANNELS];
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63
64 //! The left shift applied to random noise in 0x31ea substreams.
65 uint8_t noise_shift;
66 //! The current seed value for the pseudorandom noise generator(s).
67 uint32_t noisegen_seed;
68
69 //! Set if the substream contains extra info to check the size of VLC blocks.
70 uint8_t data_check_present;
71
72 //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
73 uint8_t param_presence_flags;
74#define PARAM_BLOCKSIZE (1 << 7)
75#define PARAM_MATRIX (1 << 6)
76#define PARAM_OUTSHIFT (1 << 5)
77#define PARAM_QUANTSTEP (1 << 4)
78#define PARAM_FIR (1 << 3)
79#define PARAM_IIR (1 << 2)
80#define PARAM_HUFFOFFSET (1 << 1)
cbf3cf19 81#define PARAM_PRESENCE (1 << 0)
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82 //@}
83
84 //@{
85 /** matrix data */
86
87 //! Number of matrices to be applied.
88 uint8_t num_primitive_matrices;
89
90 //! matrix output channel
91 uint8_t matrix_out_ch[MAX_MATRICES];
92
93 //! Whether the LSBs of the matrix output are encoded in the bitstream.
94 uint8_t lsb_bypass[MAX_MATRICES];
95 //! Matrix coefficients, stored as 2.14 fixed point.
96 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];
97 //! Left shift to apply to noise values in 0x31eb substreams.
98 uint8_t matrix_noise_shift[MAX_MATRICES];
99 //@}
100
9906a2be 101 //! Left shift to apply to Huffman-decoded residuals.
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102 uint8_t quant_step_size[MAX_CHANNELS];
103
9906a2be 104 //! number of PCM samples in current audio block
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105 uint16_t blocksize;
106 //! Number of PCM samples decoded so far in this frame.
107 uint16_t blockpos;
108
109 //! Left shift to apply to decoded PCM values to get final 24-bit output.
110 int8_t output_shift[MAX_CHANNELS];
111
112 //! Running XOR of all output samples.
113 int32_t lossless_check_data;
114
115} SubStream;
116
117typedef struct MLPDecodeContext {
118 AVCodecContext *avctx;
119
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120 //! Current access unit being read has a major sync.
121 int is_major_sync_unit;
122
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123 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
124 uint8_t params_valid;
125
126 //! Number of substreams contained within this stream.
127 uint8_t num_substreams;
128
129 //! Index of the last substream to decode - further substreams are skipped.
130 uint8_t max_decoded_substream;
131
9906a2be 132 //! number of PCM samples contained in each frame
b517af05 133 int access_unit_size;
9906a2be 134 //! next power of two above the number of samples in each frame
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135 int access_unit_size_pow2;
136
137 SubStream substream[MAX_SUBSTREAMS];
138
f53acb7b 139 ChannelParams channel_params[MAX_CHANNELS];
b517af05 140
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141 int matrix_changed;
142 int filter_changed[MAX_CHANNELS][NUM_FILTERS];
143
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144 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
145 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
146 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
147} MLPDecodeContext;
148
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149static VLC huff_vlc[3];
150
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151/** Initialize static data, constant between all invocations of the codec. */
152
dc8a7c93 153static av_cold void init_static(void)
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154{
155 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
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156 &ff_mlp_huffman_tables[0][0][1], 2, 1,
157 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
b517af05 158 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
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159 &ff_mlp_huffman_tables[1][0][1], 2, 1,
160 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
b517af05 161 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
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162 &ff_mlp_huffman_tables[2][0][1], 2, 1,
163 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
b517af05 164
ce15710f 165 ff_mlp_init_crc();
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166}
167
168static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
169 unsigned int substr, unsigned int ch)
170{
f53acb7b 171 ChannelParams *cp = &m->channel_params[ch];
b517af05 172 SubStream *s = &m->substream[substr];
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173 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
174 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
175 int32_t sign_huff_offset = cp->huff_offset;
b517af05 176
f53acb7b 177 if (cp->codebook > 0)
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178 sign_huff_offset -= 7 << lsb_bits;
179
180 if (sign_shift >= 0)
181 sign_huff_offset -= 1 << sign_shift;
182
183 return sign_huff_offset;
184}
185
186/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
187 * and plain LSBs. */
188
189static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
190 unsigned int substr, unsigned int pos)
191{
192 SubStream *s = &m->substream[substr];
193 unsigned int mat, channel;
194
195 for (mat = 0; mat < s->num_primitive_matrices; mat++)
196 if (s->lsb_bypass[mat])
197 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
198
199 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
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200 ChannelParams *cp = &m->channel_params[channel];
201 int codebook = cp->codebook;
b517af05 202 int quant_step_size = s->quant_step_size[channel];
f53acb7b 203 int lsb_bits = cp->huff_lsbs - quant_step_size;
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204 int result = 0;
205
206 if (codebook > 0)
207 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
208 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
209
210 if (result < 0)
211 return -1;
212
213 if (lsb_bits > 0)
214 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
215
f53acb7b 216 result += cp->sign_huff_offset;
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217 result <<= quant_step_size;
218
219 m->sample_buffer[pos + s->blockpos][channel] = result;
220 }
221
222 return 0;
223}
224
225static av_cold int mlp_decode_init(AVCodecContext *avctx)
226{
227 MLPDecodeContext *m = avctx->priv_data;
228 int substr;
229
230 init_static();
231 m->avctx = avctx;
232 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
233 m->substream[substr].lossless_check_data = 0xffffffff;
ee1a8f62 234
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235 return 0;
236}
237
238/** Read a major sync info header - contains high level information about
239 * the stream - sample rate, channel arrangement etc. Most of this
240 * information is not actually necessary for decoding, only for playback.
241 */
242
243static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
244{
245 MLPHeaderInfo mh;
246 int substr;
247
248 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
249 return -1;
250
251 if (mh.group1_bits == 0) {
9906a2be 252 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
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253 return -1;
254 }
255 if (mh.group2_bits > mh.group1_bits) {
256 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 257 "Channel group 2 cannot have more bits per sample than group 1.\n");
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258 return -1;
259 }
260
261 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
262 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 263 "Channel groups with differing sample rates are not currently supported.\n");
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264 return -1;
265 }
266
267 if (mh.group1_samplerate == 0) {
9906a2be 268 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
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269 return -1;
270 }
271 if (mh.group1_samplerate > MAX_SAMPLERATE) {
272 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 273 "Sampling rate %d is greater than the supported maximum (%d).\n",
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274 mh.group1_samplerate, MAX_SAMPLERATE);
275 return -1;
276 }
277 if (mh.access_unit_size > MAX_BLOCKSIZE) {
278 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 279 "Block size %d is greater than the supported maximum (%d).\n",
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280 mh.access_unit_size, MAX_BLOCKSIZE);
281 return -1;
282 }
283 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
284 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 285 "Block size pow2 %d is greater than the supported maximum (%d).\n",
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286 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
287 return -1;
288 }
289
290 if (mh.num_substreams == 0)
291 return -1;
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292 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
293 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
294 return -1;
295 }
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296 if (mh.num_substreams > MAX_SUBSTREAMS) {
297 av_log(m->avctx, AV_LOG_ERROR,
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298 "Number of substreams %d is larger than the maximum supported "
299 "by the decoder. %s\n", mh.num_substreams, sample_message);
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300 return -1;
301 }
302
303 m->access_unit_size = mh.access_unit_size;
304 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
305
306 m->num_substreams = mh.num_substreams;
307 m->max_decoded_substream = m->num_substreams - 1;
308
309 m->avctx->sample_rate = mh.group1_samplerate;
310 m->avctx->frame_size = mh.access_unit_size;
311
97679e6e 312 m->avctx->bits_per_raw_sample = mh.group1_bits;
c687643c 313 if (mh.group1_bits > 16)
b517af05 314 m->avctx->sample_fmt = SAMPLE_FMT_S32;
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315 else
316 m->avctx->sample_fmt = SAMPLE_FMT_S16;
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317
318 m->params_valid = 1;
319 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
320 m->substream[substr].restart_seen = 0;
321
322 return 0;
323}
324
325/** Read a restart header from a block in a substream. This contains parameters
326 * required to decode the audio that do not change very often. Generally
327 * (always) present only in blocks following a major sync. */
328
329static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
330 const uint8_t *buf, unsigned int substr)
331{
332 SubStream *s = &m->substream[substr];
333 unsigned int ch;
334 int sync_word, tmp;
335 uint8_t checksum;
336 uint8_t lossless_check;
337 int start_count = get_bits_count(gbp);
338
339 sync_word = get_bits(gbp, 13);
340
341 if (sync_word != 0x31ea >> 1) {
342 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 343 "restart header sync incorrect (got 0x%04x)\n", sync_word);
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344 return -1;
345 }
346 s->noise_type = get_bits1(gbp);
347
348 skip_bits(gbp, 16); /* Output timestamp */
349
350 s->min_channel = get_bits(gbp, 4);
351 s->max_channel = get_bits(gbp, 4);
352 s->max_matrix_channel = get_bits(gbp, 4);
353
354 if (s->min_channel > s->max_channel) {
355 av_log(m->avctx, AV_LOG_ERROR,
356 "Substream min channel cannot be greater than max channel.\n");
357 return -1;
358 }
359
360 if (m->avctx->request_channels > 0
361 && s->max_channel + 1 >= m->avctx->request_channels
362 && substr < m->max_decoded_substream) {
363 av_log(m->avctx, AV_LOG_INFO,
364 "Extracting %d channel downmix from substream %d. "
365 "Further substreams will be skipped.\n",
366 s->max_channel + 1, substr);
367 m->max_decoded_substream = substr;
368 }
369
370 s->noise_shift = get_bits(gbp, 4);
371 s->noisegen_seed = get_bits(gbp, 23);
372
373 skip_bits(gbp, 19);
374
375 s->data_check_present = get_bits1(gbp);
376 lossless_check = get_bits(gbp, 8);
377 if (substr == m->max_decoded_substream
378 && s->lossless_check_data != 0xffffffff) {
a7cc783d 379 tmp = xor_32_to_8(s->lossless_check_data);
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380 if (tmp != lossless_check)
381 av_log(m->avctx, AV_LOG_WARNING,
9906a2be 382 "Lossless check failed - expected %02x, calculated %02x.\n",
b517af05 383 lossless_check, tmp);
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384 }
385
386 skip_bits(gbp, 16);
387
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388 memset(s->ch_assign, 0, sizeof(s->ch_assign));
389
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390 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
391 int ch_assign = get_bits(gbp, 6);
9731e7f1 392 if (ch_assign > s->max_matrix_channel) {
b517af05 393 av_log(m->avctx, AV_LOG_ERROR,
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394 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
395 ch, ch_assign, sample_message);
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396 return -1;
397 }
9731e7f1 398 s->ch_assign[ch_assign] = ch;
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399 }
400
ce15710f 401 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
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402
403 if (checksum != get_bits(gbp, 8))
9906a2be 404 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
b517af05 405
9906a2be 406 /* Set default decoding parameters. */
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407 s->param_presence_flags = 0xff;
408 s->num_primitive_matrices = 0;
409 s->blocksize = 8;
410 s->lossless_check_data = 0;
411
412 memset(s->output_shift , 0, sizeof(s->output_shift ));
413 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
414
415 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
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416 ChannelParams *cp = &m->channel_params[ch];
417 cp->filter_params[FIR].order = 0;
418 cp->filter_params[IIR].order = 0;
419 cp->filter_params[FIR].shift = 0;
420 cp->filter_params[IIR].shift = 0;
b517af05 421
9906a2be 422 /* Default audio coding is 24-bit raw PCM. */
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423 cp->huff_offset = 0;
424 cp->sign_huff_offset = (-1) << 23;
425 cp->codebook = 0;
426 cp->huff_lsbs = 24;
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427 }
428
429 if (substr == m->max_decoded_substream) {
9731e7f1 430 m->avctx->channels = s->max_matrix_channel + 1;
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431 }
432
433 return 0;
434}
435
436/** Read parameters for one of the prediction filters. */
437
438static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
439 unsigned int channel, unsigned int filter)
440{
f53acb7b 441 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
0c5670a0 442 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
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443 const char fchar = filter ? 'I' : 'F';
444 int i, order;
445
9906a2be 446 // Filter is 0 for FIR, 1 for IIR.
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447 assert(filter < 2);
448
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449 m->filter_changed[channel][filter]++;
450
b517af05 451 order = get_bits(gbp, 4);
0c5670a0 452 if (order > max_order) {
b517af05 453 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 454 "%cIR filter order %d is greater than maximum %d.\n",
0c5670a0 455 fchar, order, max_order);
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456 return -1;
457 }
d6aa052d 458 fp->order = order;
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459
460 if (order > 0) {
461 int coeff_bits, coeff_shift;
462
d6aa052d 463 fp->shift = get_bits(gbp, 4);
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464
465 coeff_bits = get_bits(gbp, 5);
466 coeff_shift = get_bits(gbp, 3);
467 if (coeff_bits < 1 || coeff_bits > 16) {
468 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 469 "%cIR filter coeff_bits must be between 1 and 16.\n",
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470 fchar);
471 return -1;
472 }
473 if (coeff_bits + coeff_shift > 16) {
474 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 475 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
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476 fchar);
477 return -1;
478 }
479
480 for (i = 0; i < order; i++)
db01fa13 481 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
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482
483 if (get_bits1(gbp)) {
484 int state_bits, state_shift;
485
486 if (filter == FIR) {
487 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 488 "FIR filter has state data specified.\n");
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489 return -1;
490 }
491
492 state_bits = get_bits(gbp, 4);
493 state_shift = get_bits(gbp, 4);
494
9906a2be 495 /* TODO: Check validity of state data. */
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496
497 for (i = 0; i < order; i++)
db01fa13 498 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
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499 }
500 }
501
502 return 0;
503}
504
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505/** Read parameters for primitive matrices. */
506
507static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
508{
509 unsigned int mat, ch;
510
511 s->num_primitive_matrices = get_bits(gbp, 4);
75428fa4 512 m->matrix_changed++;
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513
514 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
515 int frac_bits, max_chan;
516 s->matrix_out_ch[mat] = get_bits(gbp, 4);
517 frac_bits = get_bits(gbp, 4);
518 s->lsb_bypass [mat] = get_bits1(gbp);
519
0091d8a1 520 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
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521 av_log(m->avctx, AV_LOG_ERROR,
522 "Invalid channel %d specified as output from matrix.\n",
523 s->matrix_out_ch[mat]);
524 return -1;
525 }
526 if (frac_bits > 14) {
527 av_log(m->avctx, AV_LOG_ERROR,
528 "Too many fractional bits specified.\n");
529 return -1;
530 }
531
532 max_chan = s->max_matrix_channel;
533 if (!s->noise_type)
534 max_chan+=2;
535
536 for (ch = 0; ch <= max_chan; ch++) {
537 int coeff_val = 0;
538 if (get_bits1(gbp))
539 coeff_val = get_sbits(gbp, frac_bits + 2);
540
541 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
542 }
543
544 if (s->noise_type)
545 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
546 else
547 s->matrix_noise_shift[mat] = 0;
548 }
549
550 return 0;
551}
552
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553/** Read channel parameters. */
554
555static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
556 GetBitContext *gbp, unsigned int ch)
557{
558 ChannelParams *cp = &m->channel_params[ch];
559 FilterParams *fir = &cp->filter_params[FIR];
560 FilterParams *iir = &cp->filter_params[IIR];
561 SubStream *s = &m->substream[substr];
562
563 if (s->param_presence_flags & PARAM_FIR)
564 if (get_bits1(gbp))
565 if (read_filter_params(m, gbp, ch, FIR) < 0)
566 return -1;
567
568 if (s->param_presence_flags & PARAM_IIR)
569 if (get_bits1(gbp))
570 if (read_filter_params(m, gbp, ch, IIR) < 0)
571 return -1;
572
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573 if (fir->order + iir->order > 8) {
574 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
575 return -1;
576 }
577
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578 if (fir->order && iir->order &&
579 fir->shift != iir->shift) {
580 av_log(m->avctx, AV_LOG_ERROR,
581 "FIR and IIR filters must use the same precision.\n");
582 return -1;
583 }
584 /* The FIR and IIR filters must have the same precision.
585 * To simplify the filtering code, only the precision of the
586 * FIR filter is considered. If only the IIR filter is employed,
587 * the FIR filter precision is set to that of the IIR filter, so
588 * that the filtering code can use it. */
589 if (!fir->order && iir->order)
590 fir->shift = iir->shift;
591
592 if (s->param_presence_flags & PARAM_HUFFOFFSET)
593 if (get_bits1(gbp))
594 cp->huff_offset = get_sbits(gbp, 15);
595
596 cp->codebook = get_bits(gbp, 2);
597 cp->huff_lsbs = get_bits(gbp, 5);
598
125cf771
RP
599 if (cp->huff_lsbs > 24) {
600 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
601 return -1;
602 }
43ee5fe0 603
125cf771 604 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
43ee5fe0
RP
605
606 return 0;
607}
608
b517af05
RP
609/** Read decoding parameters that change more often than those in the restart
610 * header. */
611
612static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
613 unsigned int substr)
614{
615 SubStream *s = &m->substream[substr];
f8e6293b 616 unsigned int ch;
b517af05 617
cbf3cf19 618 if (s->param_presence_flags & PARAM_PRESENCE)
b517af05
RP
619 if (get_bits1(gbp))
620 s->param_presence_flags = get_bits(gbp, 8);
621
622 if (s->param_presence_flags & PARAM_BLOCKSIZE)
623 if (get_bits1(gbp)) {
624 s->blocksize = get_bits(gbp, 9);
b864098c
RP
625 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
626 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
b517af05
RP
627 s->blocksize = 0;
628 return -1;
629 }
630 }
631
632 if (s->param_presence_flags & PARAM_MATRIX)
633 if (get_bits1(gbp)) {
f8e6293b
RP
634 if (read_matrix_params(m, s, gbp) < 0)
635 return -1;
b517af05
RP
636 }
637
638 if (s->param_presence_flags & PARAM_OUTSHIFT)
639 if (get_bits1(gbp))
640 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
af048026 641 s->output_shift[ch] = get_sbits(gbp, 4);
b517af05
RP
642 }
643
644 if (s->param_presence_flags & PARAM_QUANTSTEP)
645 if (get_bits1(gbp))
646 for (ch = 0; ch <= s->max_channel; ch++) {
f53acb7b
RP
647 ChannelParams *cp = &m->channel_params[ch];
648
b517af05 649 s->quant_step_size[ch] = get_bits(gbp, 4);
b517af05 650
f53acb7b 651 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
b517af05
RP
652 }
653
654 for (ch = s->min_channel; ch <= s->max_channel; ch++)
655 if (get_bits1(gbp)) {
43ee5fe0 656 if (read_channel_params(m, substr, gbp, ch) < 0)
b517af05 657 return -1;
b517af05
RP
658 }
659
660 return 0;
661}
662
663#define MSB_MASK(bits) (-1u << bits)
664
665/** Generate PCM samples using the prediction filters and residual values
666 * read from the data stream, and update the filter state. */
667
668static void filter_channel(MLPDecodeContext *m, unsigned int substr,
669 unsigned int channel)
670{
671 SubStream *s = &m->substream[substr];
e71365f4
RP
672 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
673 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
674 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
675 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
676 unsigned int filter_shift = fir->shift;
b517af05
RP
677 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
678 int index = MAX_BLOCKSIZE;
0ae59bb0 679 int i;
b517af05 680
06e34be8
RP
681 memcpy(&firbuf[index], fir->state, MAX_FIR_ORDER * sizeof(int32_t));
682 memcpy(&iirbuf[index], iir->state, MAX_IIR_ORDER * sizeof(int32_t));
b517af05
RP
683
684 for (i = 0; i < s->blocksize; i++) {
685 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
686 unsigned int order;
687 int64_t accum = 0;
688 int32_t result;
689
690 /* TODO: Move this code to DSPContext? */
691
e71365f4 692 for (order = 0; order < fir->order; order++)
06e34be8 693 accum += (int64_t) firbuf[index + order] * fir->coeff[order];
e71365f4 694 for (order = 0; order < iir->order; order++)
06e34be8 695 accum += (int64_t) iirbuf[index + order] * iir->coeff[order];
b517af05
RP
696
697 accum = accum >> filter_shift;
698 result = (accum + residual) & mask;
699
700 --index;
701
e71365f4
RP
702 firbuf[index] = result;
703 iirbuf[index] = result - accum;
b517af05
RP
704
705 m->sample_buffer[i + s->blockpos][channel] = result;
706 }
707
06e34be8
RP
708 memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
709 memcpy(iir->state, &iirbuf[index], MAX_IIR_ORDER * sizeof(int32_t));
b517af05
RP
710}
711
712/** Read a block of PCM residual data (or actual if no filtering active). */
713
714static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
715 unsigned int substr)
716{
717 SubStream *s = &m->substream[substr];
718 unsigned int i, ch, expected_stream_pos = 0;
719
720 if (s->data_check_present) {
721 expected_stream_pos = get_bits_count(gbp);
722 expected_stream_pos += get_bits(gbp, 16);
723 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
724 "we have not tested yet. %s\n", sample_message);
725 }
726
727 if (s->blockpos + s->blocksize > m->access_unit_size) {
9906a2be 728 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
b517af05
RP
729 return -1;
730 }
731
732 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
733 s->blocksize * sizeof(m->bypassed_lsbs[0]));
734
735 for (i = 0; i < s->blocksize; i++) {
736 if (read_huff_channels(m, gbp, substr, i) < 0)
737 return -1;
738 }
739
740 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
741 filter_channel(m, substr, ch);
742 }
743
744 s->blockpos += s->blocksize;
745
746 if (s->data_check_present) {
747 if (get_bits_count(gbp) != expected_stream_pos)
9906a2be 748 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
b517af05
RP
749 skip_bits(gbp, 8);
750 }
751
752 return 0;
753}
754
9906a2be 755/** Data table used for TrueHD noise generation function. */
b517af05
RP
756
757static const int8_t noise_table[256] = {
758 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
759 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
760 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
761 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
762 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
763 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
764 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
765 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
766 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
767 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
768 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
769 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
770 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
771 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
772 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
773 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
774};
775
776/** Noise generation functions.
777 * I'm not sure what these are for - they seem to be some kind of pseudorandom
778 * sequence generators, used to generate noise data which is used when the
779 * channels are rematrixed. I'm not sure if they provide a practical benefit
780 * to compression, or just obfuscate the decoder. Are they for some kind of
781 * dithering? */
782
783/** Generate two channels of noise, used in the matrix when
784 * restart sync word == 0x31ea. */
785
786static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
787{
788 SubStream *s = &m->substream[substr];
789 unsigned int i;
790 uint32_t seed = s->noisegen_seed;
791 unsigned int maxchan = s->max_matrix_channel;
792
793 for (i = 0; i < s->blockpos; i++) {
794 uint16_t seed_shr7 = seed >> 7;
795 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
796 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
797
798 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
799 }
800
801 s->noisegen_seed = seed;
802}
803
804/** Generate a block of noise, used when restart sync word == 0x31eb. */
805
806static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
807{
808 SubStream *s = &m->substream[substr];
809 unsigned int i;
810 uint32_t seed = s->noisegen_seed;
811
812 for (i = 0; i < m->access_unit_size_pow2; i++) {
813 uint8_t seed_shr15 = seed >> 15;
814 m->noise_buffer[i] = noise_table[seed_shr15];
815 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
816 }
817
818 s->noisegen_seed = seed;
819}
820
821
822/** Apply the channel matrices in turn to reconstruct the original audio
823 * samples. */
824
825static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
826{
827 SubStream *s = &m->substream[substr];
828 unsigned int mat, src_ch, i;
829 unsigned int maxchan;
830
831 maxchan = s->max_matrix_channel;
832 if (!s->noise_type) {
833 generate_2_noise_channels(m, substr);
834 maxchan += 2;
835 } else {
836 fill_noise_buffer(m, substr);
837 }
838
839 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
840 int matrix_noise_shift = s->matrix_noise_shift[mat];
841 unsigned int dest_ch = s->matrix_out_ch[mat];
842 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
843
844 /* TODO: DSPContext? */
845
846 for (i = 0; i < s->blockpos; i++) {
847 int64_t accum = 0;
848 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
849 accum += (int64_t)m->sample_buffer[i][src_ch]
850 * s->matrix_coeff[mat][src_ch];
851 }
852 if (matrix_noise_shift) {
853 uint32_t index = s->num_primitive_matrices - mat;
854 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
855 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
856 }
857 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
858 + m->bypassed_lsbs[i][mat];
859 }
860 }
861}
862
863/** Write the audio data into the output buffer. */
864
865static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
866 uint8_t *data, unsigned int *data_size, int is32)
867{
868 SubStream *s = &m->substream[substr];
9731e7f1 869 unsigned int i, out_ch = 0;
b517af05
RP
870 int32_t *data_32 = (int32_t*) data;
871 int16_t *data_16 = (int16_t*) data;
872
873 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
874 return -1;
875
876 for (i = 0; i < s->blockpos; i++) {
9731e7f1
RP
877 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
878 int mat_ch = s->ch_assign[out_ch];
879 int32_t sample = m->sample_buffer[i][mat_ch]
880 << s->output_shift[mat_ch];
881 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
b517af05
RP
882 if (is32) *data_32++ = sample << 8;
883 else *data_16++ = sample >> 8;
884 }
885 }
886
9731e7f1 887 *data_size = i * out_ch * (is32 ? 4 : 2);
b517af05
RP
888
889 return 0;
890}
891
892static int output_data(MLPDecodeContext *m, unsigned int substr,
893 uint8_t *data, unsigned int *data_size)
894{
895 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
896 return output_data_internal(m, substr, data, data_size, 1);
897 else
898 return output_data_internal(m, substr, data, data_size, 0);
899}
900
901
b517af05
RP
902/** Read an access unit from the stream.
903 * Returns < 0 on error, 0 if not enough data is present in the input stream
904 * otherwise returns the number of bytes consumed. */
905
906static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
7a00bbad 907 AVPacket *avpkt)
b517af05 908{
7a00bbad
TB
909 const uint8_t *buf = avpkt->data;
910 int buf_size = avpkt->size;
b517af05
RP
911 MLPDecodeContext *m = avctx->priv_data;
912 GetBitContext gb;
913 unsigned int length, substr;
914 unsigned int substream_start;
915 unsigned int header_size = 4;
916 unsigned int substr_header_size = 0;
917 uint8_t substream_parity_present[MAX_SUBSTREAMS];
918 uint16_t substream_data_len[MAX_SUBSTREAMS];
919 uint8_t parity_bits;
920
921 if (buf_size < 4)
922 return 0;
923
924 length = (AV_RB16(buf) & 0xfff) * 2;
925
926 if (length > buf_size)
927 return -1;
928
929 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
930
cc9c5126 931 m->is_major_sync_unit = 0;
b517af05 932 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
b517af05
RP
933 if (read_major_sync(m, &gb) < 0)
934 goto error;
cc9c5126 935 m->is_major_sync_unit = 1;
b517af05
RP
936 header_size += 28;
937 }
938
939 if (!m->params_valid) {
940 av_log(m->avctx, AV_LOG_WARNING,
9906a2be 941 "Stream parameters not seen; skipping frame.\n");
b517af05
RP
942 *data_size = 0;
943 return length;
944 }
945
946 substream_start = 0;
947
948 for (substr = 0; substr < m->num_substreams; substr++) {
cc9c5126 949 int extraword_present, checkdata_present, end, nonrestart_substr;
b517af05
RP
950
951 extraword_present = get_bits1(&gb);
cc9c5126 952 nonrestart_substr = get_bits1(&gb);
b517af05
RP
953 checkdata_present = get_bits1(&gb);
954 skip_bits1(&gb);
955
956 end = get_bits(&gb, 12) * 2;
957
958 substr_header_size += 2;
959
960 if (extraword_present) {
89e39be3
RP
961 if (m->avctx->codec_id == CODEC_ID_MLP) {
962 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
963 goto error;
964 }
b517af05
RP
965 skip_bits(&gb, 16);
966 substr_header_size += 2;
967 }
968
cc9c5126
RP
969 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
970 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
971 goto error;
972 }
973
b517af05
RP
974 if (end + header_size + substr_header_size > length) {
975 av_log(m->avctx, AV_LOG_ERROR,
976 "Indicated length of substream %d data goes off end of "
977 "packet.\n", substr);
978 end = length - header_size - substr_header_size;
979 }
980
981 if (end < substream_start) {
982 av_log(avctx, AV_LOG_ERROR,
983 "Indicated end offset of substream %d data "
984 "is smaller than calculated start offset.\n",
985 substr);
986 goto error;
987 }
988
989 if (substr > m->max_decoded_substream)
990 continue;
991
992 substream_parity_present[substr] = checkdata_present;
993 substream_data_len[substr] = end - substream_start;
994 substream_start = end;
995 }
996
ce15710f
RP
997 parity_bits = ff_mlp_calculate_parity(buf, 4);
998 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
b517af05
RP
999
1000 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1001 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1002 goto error;
1003 }
1004
1005 buf += header_size + substr_header_size;
1006
1007 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1008 SubStream *s = &m->substream[substr];
1009 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1010
75428fa4
RP
1011 m->matrix_changed = 0;
1012 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1013
b517af05
RP
1014 s->blockpos = 0;
1015 do {
75428fa4
RP
1016 unsigned int ch;
1017
b517af05
RP
1018 if (get_bits1(&gb)) {
1019 if (get_bits1(&gb)) {
9906a2be 1020 /* A restart header should be present. */
b517af05
RP
1021 if (read_restart_header(m, &gb, buf, substr) < 0)
1022 goto next_substr;
1023 s->restart_seen = 1;
1024 }
1025
1026 if (!s->restart_seen) {
b517af05
RP
1027 goto next_substr;
1028 }
1029
1030 if (read_decoding_params(m, &gb, substr) < 0)
1031 goto next_substr;
1032 }
1033
75428fa4
RP
1034 if (m->matrix_changed > 1) {
1035 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
1036 goto next_substr;
1037 }
1038 for (ch = 0; ch < s->max_channel; ch++)
1039 if (m->filter_changed[ch][FIR] > 1 ||
1040 m->filter_changed[ch][IIR] > 1) {
1041 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
1042 goto next_substr;
1043 }
1044
b517af05 1045 if (!s->restart_seen) {
b517af05
RP
1046 goto next_substr;
1047 }
1048
1049 if (read_block_data(m, &gb, substr) < 0)
1050 return -1;
1051
d7952be3
RP
1052 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1053 goto substream_length_mismatch;
1054
1055 } while (!get_bits1(&gb));
b517af05
RP
1056
1057 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
7b18e13a
RP
1058 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1059 int shorten_by;
1060
1061 if (get_bits(&gb, 16) != 0xD234)
1062 return -1;
1063
1064 shorten_by = get_bits(&gb, 16);
1065 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1066 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1067 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1068 return -1;
1069
b517af05 1070 if (substr == m->max_decoded_substream)
9906a2be 1071 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
b517af05 1072 }
ab79fa44 1073 if (substream_parity_present[substr]) {
b517af05
RP
1074 uint8_t parity, checksum;
1075
ab79fa44
RP
1076 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1077 goto substream_length_mismatch;
1078
d544dcdf
RP
1079 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1080 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
b517af05 1081
d544dcdf
RP
1082 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1083 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1084 if ( get_bits(&gb, 8) != checksum)
1085 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
b517af05
RP
1086 }
1087 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
d7952be3 1088 goto substream_length_mismatch;
b517af05
RP
1089 }
1090
1091next_substr:
01aaf092
RP
1092 if (!s->restart_seen) {
1093 av_log(m->avctx, AV_LOG_ERROR,
1094 "No restart header present in substream %d.\n", substr);
1095 }
1096
b517af05
RP
1097 buf += substream_data_len[substr];
1098 }
1099
1100 rematrix_channels(m, m->max_decoded_substream);
1101
1102 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1103 return -1;
1104
1105 return length;
1106
d7952be3
RP
1107substream_length_mismatch:
1108 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1109 return -1;
1110
b517af05
RP
1111error:
1112 m->params_valid = 0;
1113 return -1;
1114}
1115
9ba4821d 1116#if CONFIG_MLP_DECODER
b517af05
RP
1117AVCodec mlp_decoder = {
1118 "mlp",
1119 CODEC_TYPE_AUDIO,
1120 CODEC_ID_MLP,
1121 sizeof(MLPDecodeContext),
1122 mlp_decode_init,
1123 NULL,
1124 NULL,
1125 read_access_unit,
9ba4821d 1126 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
b517af05 1127};
9ba4821d 1128#endif /* CONFIG_MLP_DECODER */
b517af05 1129
9ba4821d
RP
1130#if CONFIG_TRUEHD_DECODER
1131AVCodec truehd_decoder = {
1132 "truehd",
1133 CODEC_TYPE_AUDIO,
1134 CODEC_ID_TRUEHD,
1135 sizeof(MLPDecodeContext),
1136 mlp_decode_init,
1137 NULL,
1138 NULL,
1139 read_access_unit,
1140 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1141};
1142#endif /* CONFIG_TRUEHD_DECODER */