mlpdec: Simplify &foo[0] to foo and use index (which has just been initialized
[libav.git] / libavcodec / mlpdec.c
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1/*
2 * MLP decoder
3 * Copyright (c) 2007-2008 Ian Caulfield
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
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"
31#include "bitstream.h"
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
120 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
121 uint8_t params_valid;
122
123 //! Number of substreams contained within this stream.
124 uint8_t num_substreams;
125
126 //! Index of the last substream to decode - further substreams are skipped.
127 uint8_t max_decoded_substream;
128
9906a2be 129 //! number of PCM samples contained in each frame
b517af05 130 int access_unit_size;
9906a2be 131 //! next power of two above the number of samples in each frame
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132 int access_unit_size_pow2;
133
134 SubStream substream[MAX_SUBSTREAMS];
135
f53acb7b 136 ChannelParams channel_params[MAX_CHANNELS];
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137
138 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
139 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
140 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
141} MLPDecodeContext;
142
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143static VLC huff_vlc[3];
144
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145/** Initialize static data, constant between all invocations of the codec. */
146
dc8a7c93 147static av_cold void init_static(void)
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148{
149 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
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150 &ff_mlp_huffman_tables[0][0][1], 2, 1,
151 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
b517af05 152 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
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153 &ff_mlp_huffman_tables[1][0][1], 2, 1,
154 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
b517af05 155 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
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156 &ff_mlp_huffman_tables[2][0][1], 2, 1,
157 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
b517af05 158
ce15710f 159 ff_mlp_init_crc();
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160}
161
162static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
163 unsigned int substr, unsigned int ch)
164{
f53acb7b 165 ChannelParams *cp = &m->channel_params[ch];
b517af05 166 SubStream *s = &m->substream[substr];
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167 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
168 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
169 int32_t sign_huff_offset = cp->huff_offset;
b517af05 170
f53acb7b 171 if (cp->codebook > 0)
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172 sign_huff_offset -= 7 << lsb_bits;
173
174 if (sign_shift >= 0)
175 sign_huff_offset -= 1 << sign_shift;
176
177 return sign_huff_offset;
178}
179
180/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
181 * and plain LSBs. */
182
183static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
184 unsigned int substr, unsigned int pos)
185{
186 SubStream *s = &m->substream[substr];
187 unsigned int mat, channel;
188
189 for (mat = 0; mat < s->num_primitive_matrices; mat++)
190 if (s->lsb_bypass[mat])
191 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
192
193 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
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194 ChannelParams *cp = &m->channel_params[channel];
195 int codebook = cp->codebook;
b517af05 196 int quant_step_size = s->quant_step_size[channel];
f53acb7b 197 int lsb_bits = cp->huff_lsbs - quant_step_size;
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198 int result = 0;
199
200 if (codebook > 0)
201 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
202 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
203
204 if (result < 0)
205 return -1;
206
207 if (lsb_bits > 0)
208 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
209
f53acb7b 210 result += cp->sign_huff_offset;
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211 result <<= quant_step_size;
212
213 m->sample_buffer[pos + s->blockpos][channel] = result;
214 }
215
216 return 0;
217}
218
219static av_cold int mlp_decode_init(AVCodecContext *avctx)
220{
221 MLPDecodeContext *m = avctx->priv_data;
222 int substr;
223
224 init_static();
225 m->avctx = avctx;
226 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
227 m->substream[substr].lossless_check_data = 0xffffffff;
ee1a8f62 228
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229 return 0;
230}
231
232/** Read a major sync info header - contains high level information about
233 * the stream - sample rate, channel arrangement etc. Most of this
234 * information is not actually necessary for decoding, only for playback.
235 */
236
237static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
238{
239 MLPHeaderInfo mh;
240 int substr;
241
242 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
243 return -1;
244
245 if (mh.group1_bits == 0) {
9906a2be 246 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
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247 return -1;
248 }
249 if (mh.group2_bits > mh.group1_bits) {
250 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 251 "Channel group 2 cannot have more bits per sample than group 1.\n");
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252 return -1;
253 }
254
255 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
256 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 257 "Channel groups with differing sample rates are not currently supported.\n");
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258 return -1;
259 }
260
261 if (mh.group1_samplerate == 0) {
9906a2be 262 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
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263 return -1;
264 }
265 if (mh.group1_samplerate > MAX_SAMPLERATE) {
266 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 267 "Sampling rate %d is greater than the supported maximum (%d).\n",
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268 mh.group1_samplerate, MAX_SAMPLERATE);
269 return -1;
270 }
271 if (mh.access_unit_size > MAX_BLOCKSIZE) {
272 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 273 "Block size %d is greater than the supported maximum (%d).\n",
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274 mh.access_unit_size, MAX_BLOCKSIZE);
275 return -1;
276 }
277 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
278 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 279 "Block size pow2 %d is greater than the supported maximum (%d).\n",
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280 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
281 return -1;
282 }
283
284 if (mh.num_substreams == 0)
285 return -1;
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286 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
287 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
288 return -1;
289 }
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290 if (mh.num_substreams > MAX_SUBSTREAMS) {
291 av_log(m->avctx, AV_LOG_ERROR,
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292 "Number of substreams %d is larger than the maximum supported "
293 "by the decoder. %s\n", mh.num_substreams, sample_message);
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294 return -1;
295 }
296
297 m->access_unit_size = mh.access_unit_size;
298 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
299
300 m->num_substreams = mh.num_substreams;
301 m->max_decoded_substream = m->num_substreams - 1;
302
303 m->avctx->sample_rate = mh.group1_samplerate;
304 m->avctx->frame_size = mh.access_unit_size;
305
97679e6e 306 m->avctx->bits_per_raw_sample = mh.group1_bits;
c687643c 307 if (mh.group1_bits > 16)
b517af05 308 m->avctx->sample_fmt = SAMPLE_FMT_S32;
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309 else
310 m->avctx->sample_fmt = SAMPLE_FMT_S16;
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311
312 m->params_valid = 1;
313 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
314 m->substream[substr].restart_seen = 0;
315
316 return 0;
317}
318
319/** Read a restart header from a block in a substream. This contains parameters
320 * required to decode the audio that do not change very often. Generally
321 * (always) present only in blocks following a major sync. */
322
323static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
324 const uint8_t *buf, unsigned int substr)
325{
326 SubStream *s = &m->substream[substr];
327 unsigned int ch;
328 int sync_word, tmp;
329 uint8_t checksum;
330 uint8_t lossless_check;
331 int start_count = get_bits_count(gbp);
332
333 sync_word = get_bits(gbp, 13);
334
335 if (sync_word != 0x31ea >> 1) {
336 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 337 "restart header sync incorrect (got 0x%04x)\n", sync_word);
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338 return -1;
339 }
340 s->noise_type = get_bits1(gbp);
341
342 skip_bits(gbp, 16); /* Output timestamp */
343
344 s->min_channel = get_bits(gbp, 4);
345 s->max_channel = get_bits(gbp, 4);
346 s->max_matrix_channel = get_bits(gbp, 4);
347
348 if (s->min_channel > s->max_channel) {
349 av_log(m->avctx, AV_LOG_ERROR,
350 "Substream min channel cannot be greater than max channel.\n");
351 return -1;
352 }
353
354 if (m->avctx->request_channels > 0
355 && s->max_channel + 1 >= m->avctx->request_channels
356 && substr < m->max_decoded_substream) {
357 av_log(m->avctx, AV_LOG_INFO,
358 "Extracting %d channel downmix from substream %d. "
359 "Further substreams will be skipped.\n",
360 s->max_channel + 1, substr);
361 m->max_decoded_substream = substr;
362 }
363
364 s->noise_shift = get_bits(gbp, 4);
365 s->noisegen_seed = get_bits(gbp, 23);
366
367 skip_bits(gbp, 19);
368
369 s->data_check_present = get_bits1(gbp);
370 lossless_check = get_bits(gbp, 8);
371 if (substr == m->max_decoded_substream
372 && s->lossless_check_data != 0xffffffff) {
a7cc783d 373 tmp = xor_32_to_8(s->lossless_check_data);
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374 if (tmp != lossless_check)
375 av_log(m->avctx, AV_LOG_WARNING,
9906a2be 376 "Lossless check failed - expected %02x, calculated %02x.\n",
b517af05 377 lossless_check, tmp);
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378 }
379
380 skip_bits(gbp, 16);
381
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382 memset(s->ch_assign, 0, sizeof(s->ch_assign));
383
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384 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
385 int ch_assign = get_bits(gbp, 6);
9731e7f1 386 if (ch_assign > s->max_matrix_channel) {
b517af05 387 av_log(m->avctx, AV_LOG_ERROR,
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388 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
389 ch, ch_assign, sample_message);
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390 return -1;
391 }
9731e7f1 392 s->ch_assign[ch_assign] = ch;
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393 }
394
ce15710f 395 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
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396
397 if (checksum != get_bits(gbp, 8))
9906a2be 398 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
b517af05 399
9906a2be 400 /* Set default decoding parameters. */
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401 s->param_presence_flags = 0xff;
402 s->num_primitive_matrices = 0;
403 s->blocksize = 8;
404 s->lossless_check_data = 0;
405
406 memset(s->output_shift , 0, sizeof(s->output_shift ));
407 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
408
409 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
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410 ChannelParams *cp = &m->channel_params[ch];
411 cp->filter_params[FIR].order = 0;
412 cp->filter_params[IIR].order = 0;
413 cp->filter_params[FIR].shift = 0;
414 cp->filter_params[IIR].shift = 0;
b517af05 415
9906a2be 416 /* Default audio coding is 24-bit raw PCM. */
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417 cp->huff_offset = 0;
418 cp->sign_huff_offset = (-1) << 23;
419 cp->codebook = 0;
420 cp->huff_lsbs = 24;
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421 }
422
423 if (substr == m->max_decoded_substream) {
9731e7f1 424 m->avctx->channels = s->max_matrix_channel + 1;
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425 }
426
427 return 0;
428}
429
430/** Read parameters for one of the prediction filters. */
431
432static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
433 unsigned int channel, unsigned int filter)
434{
f53acb7b 435 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
0c5670a0 436 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
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437 const char fchar = filter ? 'I' : 'F';
438 int i, order;
439
9906a2be 440 // Filter is 0 for FIR, 1 for IIR.
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441 assert(filter < 2);
442
443 order = get_bits(gbp, 4);
0c5670a0 444 if (order > max_order) {
b517af05 445 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 446 "%cIR filter order %d is greater than maximum %d.\n",
0c5670a0 447 fchar, order, max_order);
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448 return -1;
449 }
d6aa052d 450 fp->order = order;
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451
452 if (order > 0) {
453 int coeff_bits, coeff_shift;
454
d6aa052d 455 fp->shift = get_bits(gbp, 4);
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456
457 coeff_bits = get_bits(gbp, 5);
458 coeff_shift = get_bits(gbp, 3);
459 if (coeff_bits < 1 || coeff_bits > 16) {
460 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 461 "%cIR filter coeff_bits must be between 1 and 16.\n",
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462 fchar);
463 return -1;
464 }
465 if (coeff_bits + coeff_shift > 16) {
466 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 467 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
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468 fchar);
469 return -1;
470 }
471
472 for (i = 0; i < order; i++)
db01fa13 473 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
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474
475 if (get_bits1(gbp)) {
476 int state_bits, state_shift;
477
478 if (filter == FIR) {
479 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 480 "FIR filter has state data specified.\n");
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481 return -1;
482 }
483
484 state_bits = get_bits(gbp, 4);
485 state_shift = get_bits(gbp, 4);
486
9906a2be 487 /* TODO: Check validity of state data. */
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488
489 for (i = 0; i < order; i++)
db01fa13 490 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
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491 }
492 }
493
494 return 0;
495}
496
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497/** Read parameters for primitive matrices. */
498
499static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
500{
501 unsigned int mat, ch;
502
503 s->num_primitive_matrices = get_bits(gbp, 4);
504
505 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
506 int frac_bits, max_chan;
507 s->matrix_out_ch[mat] = get_bits(gbp, 4);
508 frac_bits = get_bits(gbp, 4);
509 s->lsb_bypass [mat] = get_bits1(gbp);
510
511 if (s->matrix_out_ch[mat] > s->max_channel) {
512 av_log(m->avctx, AV_LOG_ERROR,
513 "Invalid channel %d specified as output from matrix.\n",
514 s->matrix_out_ch[mat]);
515 return -1;
516 }
517 if (frac_bits > 14) {
518 av_log(m->avctx, AV_LOG_ERROR,
519 "Too many fractional bits specified.\n");
520 return -1;
521 }
522
523 max_chan = s->max_matrix_channel;
524 if (!s->noise_type)
525 max_chan+=2;
526
527 for (ch = 0; ch <= max_chan; ch++) {
528 int coeff_val = 0;
529 if (get_bits1(gbp))
530 coeff_val = get_sbits(gbp, frac_bits + 2);
531
532 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
533 }
534
535 if (s->noise_type)
536 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
537 else
538 s->matrix_noise_shift[mat] = 0;
539 }
540
541 return 0;
542}
543
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544/** Read channel parameters. */
545
546static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
547 GetBitContext *gbp, unsigned int ch)
548{
549 ChannelParams *cp = &m->channel_params[ch];
550 FilterParams *fir = &cp->filter_params[FIR];
551 FilterParams *iir = &cp->filter_params[IIR];
552 SubStream *s = &m->substream[substr];
553
554 if (s->param_presence_flags & PARAM_FIR)
555 if (get_bits1(gbp))
556 if (read_filter_params(m, gbp, ch, FIR) < 0)
557 return -1;
558
559 if (s->param_presence_flags & PARAM_IIR)
560 if (get_bits1(gbp))
561 if (read_filter_params(m, gbp, ch, IIR) < 0)
562 return -1;
563
564 if (fir->order && iir->order &&
565 fir->shift != iir->shift) {
566 av_log(m->avctx, AV_LOG_ERROR,
567 "FIR and IIR filters must use the same precision.\n");
568 return -1;
569 }
570 /* The FIR and IIR filters must have the same precision.
571 * To simplify the filtering code, only the precision of the
572 * FIR filter is considered. If only the IIR filter is employed,
573 * the FIR filter precision is set to that of the IIR filter, so
574 * that the filtering code can use it. */
575 if (!fir->order && iir->order)
576 fir->shift = iir->shift;
577
578 if (s->param_presence_flags & PARAM_HUFFOFFSET)
579 if (get_bits1(gbp))
580 cp->huff_offset = get_sbits(gbp, 15);
581
582 cp->codebook = get_bits(gbp, 2);
583 cp->huff_lsbs = get_bits(gbp, 5);
584
585 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
586
587 /* TODO: validate */
588
589 return 0;
590}
591
b517af05
RP
592/** Read decoding parameters that change more often than those in the restart
593 * header. */
594
595static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
596 unsigned int substr)
597{
598 SubStream *s = &m->substream[substr];
f8e6293b 599 unsigned int ch;
b517af05 600
cbf3cf19 601 if (s->param_presence_flags & PARAM_PRESENCE)
b517af05
RP
602 if (get_bits1(gbp))
603 s->param_presence_flags = get_bits(gbp, 8);
604
605 if (s->param_presence_flags & PARAM_BLOCKSIZE)
606 if (get_bits1(gbp)) {
607 s->blocksize = get_bits(gbp, 9);
b864098c
RP
608 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
609 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
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610 s->blocksize = 0;
611 return -1;
612 }
613 }
614
615 if (s->param_presence_flags & PARAM_MATRIX)
616 if (get_bits1(gbp)) {
f8e6293b
RP
617 if (read_matrix_params(m, s, gbp) < 0)
618 return -1;
b517af05
RP
619 }
620
621 if (s->param_presence_flags & PARAM_OUTSHIFT)
622 if (get_bits1(gbp))
623 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
af048026 624 s->output_shift[ch] = get_sbits(gbp, 4);
b517af05
RP
625 }
626
627 if (s->param_presence_flags & PARAM_QUANTSTEP)
628 if (get_bits1(gbp))
629 for (ch = 0; ch <= s->max_channel; ch++) {
f53acb7b
RP
630 ChannelParams *cp = &m->channel_params[ch];
631
b517af05 632 s->quant_step_size[ch] = get_bits(gbp, 4);
b517af05 633
f53acb7b 634 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
b517af05
RP
635 }
636
637 for (ch = s->min_channel; ch <= s->max_channel; ch++)
638 if (get_bits1(gbp)) {
43ee5fe0 639 if (read_channel_params(m, substr, gbp, ch) < 0)
b517af05 640 return -1;
b517af05
RP
641 }
642
643 return 0;
644}
645
646#define MSB_MASK(bits) (-1u << bits)
647
648/** Generate PCM samples using the prediction filters and residual values
649 * read from the data stream, and update the filter state. */
650
651static void filter_channel(MLPDecodeContext *m, unsigned int substr,
652 unsigned int channel)
653{
654 SubStream *s = &m->substream[substr];
e71365f4
RP
655 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
656 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
657 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
658 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
659 unsigned int filter_shift = fir->shift;
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RP
660 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
661 int index = MAX_BLOCKSIZE;
0ae59bb0 662 int i;
b517af05 663
a44b9f95 664 memcpy(&firbuf[index], fir->state,
0ae59bb0 665 MAX_FIR_ORDER * sizeof(int32_t));
a44b9f95 666 memcpy(&iirbuf[index], iir->state,
0ae59bb0 667 MAX_IIR_ORDER * sizeof(int32_t));
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RP
668
669 for (i = 0; i < s->blocksize; i++) {
670 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
671 unsigned int order;
672 int64_t accum = 0;
673 int32_t result;
674
675 /* TODO: Move this code to DSPContext? */
676
e71365f4
RP
677 for (order = 0; order < fir->order; order++)
678 accum += (int64_t)firbuf[index + order] *
679 fir->coeff[order];
680 for (order = 0; order < iir->order; order++)
681 accum += (int64_t)iirbuf[index + order] *
682 iir->coeff[order];
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RP
683
684 accum = accum >> filter_shift;
685 result = (accum + residual) & mask;
686
687 --index;
688
e71365f4
RP
689 firbuf[index] = result;
690 iirbuf[index] = result - accum;
b517af05
RP
691
692 m->sample_buffer[i + s->blockpos][channel] = result;
693 }
694
a44b9f95 695 memcpy(fir->state, &firbuf[index],
0ae59bb0 696 MAX_FIR_ORDER * sizeof(int32_t));
a44b9f95 697 memcpy(iir->state, &iirbuf[index],
0ae59bb0 698 MAX_IIR_ORDER * sizeof(int32_t));
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RP
699}
700
701/** Read a block of PCM residual data (or actual if no filtering active). */
702
703static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
704 unsigned int substr)
705{
706 SubStream *s = &m->substream[substr];
707 unsigned int i, ch, expected_stream_pos = 0;
708
709 if (s->data_check_present) {
710 expected_stream_pos = get_bits_count(gbp);
711 expected_stream_pos += get_bits(gbp, 16);
712 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
713 "we have not tested yet. %s\n", sample_message);
714 }
715
716 if (s->blockpos + s->blocksize > m->access_unit_size) {
9906a2be 717 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
b517af05
RP
718 return -1;
719 }
720
721 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
722 s->blocksize * sizeof(m->bypassed_lsbs[0]));
723
724 for (i = 0; i < s->blocksize; i++) {
725 if (read_huff_channels(m, gbp, substr, i) < 0)
726 return -1;
727 }
728
729 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
730 filter_channel(m, substr, ch);
731 }
732
733 s->blockpos += s->blocksize;
734
735 if (s->data_check_present) {
736 if (get_bits_count(gbp) != expected_stream_pos)
9906a2be 737 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
b517af05
RP
738 skip_bits(gbp, 8);
739 }
740
741 return 0;
742}
743
9906a2be 744/** Data table used for TrueHD noise generation function. */
b517af05
RP
745
746static const int8_t noise_table[256] = {
747 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
748 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
749 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
750 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
751 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
752 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
753 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
754 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
755 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
756 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
757 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
758 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
759 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
760 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
761 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
762 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
763};
764
765/** Noise generation functions.
766 * I'm not sure what these are for - they seem to be some kind of pseudorandom
767 * sequence generators, used to generate noise data which is used when the
768 * channels are rematrixed. I'm not sure if they provide a practical benefit
769 * to compression, or just obfuscate the decoder. Are they for some kind of
770 * dithering? */
771
772/** Generate two channels of noise, used in the matrix when
773 * restart sync word == 0x31ea. */
774
775static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
776{
777 SubStream *s = &m->substream[substr];
778 unsigned int i;
779 uint32_t seed = s->noisegen_seed;
780 unsigned int maxchan = s->max_matrix_channel;
781
782 for (i = 0; i < s->blockpos; i++) {
783 uint16_t seed_shr7 = seed >> 7;
784 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
785 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
786
787 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
788 }
789
790 s->noisegen_seed = seed;
791}
792
793/** Generate a block of noise, used when restart sync word == 0x31eb. */
794
795static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
796{
797 SubStream *s = &m->substream[substr];
798 unsigned int i;
799 uint32_t seed = s->noisegen_seed;
800
801 for (i = 0; i < m->access_unit_size_pow2; i++) {
802 uint8_t seed_shr15 = seed >> 15;
803 m->noise_buffer[i] = noise_table[seed_shr15];
804 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
805 }
806
807 s->noisegen_seed = seed;
808}
809
810
811/** Apply the channel matrices in turn to reconstruct the original audio
812 * samples. */
813
814static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
815{
816 SubStream *s = &m->substream[substr];
817 unsigned int mat, src_ch, i;
818 unsigned int maxchan;
819
820 maxchan = s->max_matrix_channel;
821 if (!s->noise_type) {
822 generate_2_noise_channels(m, substr);
823 maxchan += 2;
824 } else {
825 fill_noise_buffer(m, substr);
826 }
827
828 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
829 int matrix_noise_shift = s->matrix_noise_shift[mat];
830 unsigned int dest_ch = s->matrix_out_ch[mat];
831 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
832
833 /* TODO: DSPContext? */
834
835 for (i = 0; i < s->blockpos; i++) {
836 int64_t accum = 0;
837 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
838 accum += (int64_t)m->sample_buffer[i][src_ch]
839 * s->matrix_coeff[mat][src_ch];
840 }
841 if (matrix_noise_shift) {
842 uint32_t index = s->num_primitive_matrices - mat;
843 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
844 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
845 }
846 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
847 + m->bypassed_lsbs[i][mat];
848 }
849 }
850}
851
852/** Write the audio data into the output buffer. */
853
854static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
855 uint8_t *data, unsigned int *data_size, int is32)
856{
857 SubStream *s = &m->substream[substr];
9731e7f1 858 unsigned int i, out_ch = 0;
b517af05
RP
859 int32_t *data_32 = (int32_t*) data;
860 int16_t *data_16 = (int16_t*) data;
861
862 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
863 return -1;
864
865 for (i = 0; i < s->blockpos; i++) {
9731e7f1
RP
866 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
867 int mat_ch = s->ch_assign[out_ch];
868 int32_t sample = m->sample_buffer[i][mat_ch]
869 << s->output_shift[mat_ch];
870 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
b517af05
RP
871 if (is32) *data_32++ = sample << 8;
872 else *data_16++ = sample >> 8;
873 }
874 }
875
9731e7f1 876 *data_size = i * out_ch * (is32 ? 4 : 2);
b517af05
RP
877
878 return 0;
879}
880
881static int output_data(MLPDecodeContext *m, unsigned int substr,
882 uint8_t *data, unsigned int *data_size)
883{
884 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
885 return output_data_internal(m, substr, data, data_size, 1);
886 else
887 return output_data_internal(m, substr, data, data_size, 0);
888}
889
890
b517af05
RP
891/** Read an access unit from the stream.
892 * Returns < 0 on error, 0 if not enough data is present in the input stream
893 * otherwise returns the number of bytes consumed. */
894
895static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
896 const uint8_t *buf, int buf_size)
897{
898 MLPDecodeContext *m = avctx->priv_data;
899 GetBitContext gb;
900 unsigned int length, substr;
901 unsigned int substream_start;
902 unsigned int header_size = 4;
903 unsigned int substr_header_size = 0;
904 uint8_t substream_parity_present[MAX_SUBSTREAMS];
905 uint16_t substream_data_len[MAX_SUBSTREAMS];
906 uint8_t parity_bits;
907
908 if (buf_size < 4)
909 return 0;
910
911 length = (AV_RB16(buf) & 0xfff) * 2;
912
913 if (length > buf_size)
914 return -1;
915
916 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
917
918 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
b517af05
RP
919 if (read_major_sync(m, &gb) < 0)
920 goto error;
921 header_size += 28;
922 }
923
924 if (!m->params_valid) {
925 av_log(m->avctx, AV_LOG_WARNING,
9906a2be 926 "Stream parameters not seen; skipping frame.\n");
b517af05
RP
927 *data_size = 0;
928 return length;
929 }
930
931 substream_start = 0;
932
933 for (substr = 0; substr < m->num_substreams; substr++) {
934 int extraword_present, checkdata_present, end;
935
936 extraword_present = get_bits1(&gb);
937 skip_bits1(&gb);
938 checkdata_present = get_bits1(&gb);
939 skip_bits1(&gb);
940
941 end = get_bits(&gb, 12) * 2;
942
943 substr_header_size += 2;
944
945 if (extraword_present) {
946 skip_bits(&gb, 16);
947 substr_header_size += 2;
948 }
949
950 if (end + header_size + substr_header_size > length) {
951 av_log(m->avctx, AV_LOG_ERROR,
952 "Indicated length of substream %d data goes off end of "
953 "packet.\n", substr);
954 end = length - header_size - substr_header_size;
955 }
956
957 if (end < substream_start) {
958 av_log(avctx, AV_LOG_ERROR,
959 "Indicated end offset of substream %d data "
960 "is smaller than calculated start offset.\n",
961 substr);
962 goto error;
963 }
964
965 if (substr > m->max_decoded_substream)
966 continue;
967
968 substream_parity_present[substr] = checkdata_present;
969 substream_data_len[substr] = end - substream_start;
970 substream_start = end;
971 }
972
ce15710f
RP
973 parity_bits = ff_mlp_calculate_parity(buf, 4);
974 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
b517af05
RP
975
976 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
977 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
978 goto error;
979 }
980
981 buf += header_size + substr_header_size;
982
983 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
984 SubStream *s = &m->substream[substr];
985 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
986
987 s->blockpos = 0;
988 do {
989 if (get_bits1(&gb)) {
990 if (get_bits1(&gb)) {
9906a2be 991 /* A restart header should be present. */
b517af05
RP
992 if (read_restart_header(m, &gb, buf, substr) < 0)
993 goto next_substr;
994 s->restart_seen = 1;
995 }
996
997 if (!s->restart_seen) {
998 av_log(m->avctx, AV_LOG_ERROR,
999 "No restart header present in substream %d.\n",
1000 substr);
1001 goto next_substr;
1002 }
1003
1004 if (read_decoding_params(m, &gb, substr) < 0)
1005 goto next_substr;
1006 }
1007
1008 if (!s->restart_seen) {
1009 av_log(m->avctx, AV_LOG_ERROR,
1010 "No restart header present in substream %d.\n",
1011 substr);
1012 goto next_substr;
1013 }
1014
1015 if (read_block_data(m, &gb, substr) < 0)
1016 return -1;
1017
1018 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1019 && get_bits1(&gb) == 0);
1020
1021 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
eb2cd99c 1022 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
1e8bd10c 1023 (show_bits_long(&gb, 32) == END_OF_STREAM ||
b517af05
RP
1024 show_bits_long(&gb, 20) == 0xd234e)) {
1025 skip_bits(&gb, 18);
1026 if (substr == m->max_decoded_substream)
9906a2be 1027 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
b517af05
RP
1028
1029 if (get_bits1(&gb)) {
1030 int shorten_by = get_bits(&gb, 13);
1031 shorten_by = FFMIN(shorten_by, s->blockpos);
1032 s->blockpos -= shorten_by;
1033 } else
1034 skip_bits(&gb, 13);
1035 }
eb2cd99c
RP
1036 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1037 substream_parity_present[substr]) {
b517af05
RP
1038 uint8_t parity, checksum;
1039
ce15710f 1040 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
b517af05
RP
1041 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1042 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 1043 "Substream %d parity check failed.\n", substr);
b517af05 1044
ce15710f 1045 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
b517af05 1046 if (checksum != get_bits(&gb, 8))
9906a2be 1047 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
b517af05
RP
1048 substr);
1049 }
1050 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
9906a2be 1051 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
b517af05
RP
1052 substr);
1053 return -1;
1054 }
1055
1056next_substr:
1057 buf += substream_data_len[substr];
1058 }
1059
1060 rematrix_channels(m, m->max_decoded_substream);
1061
1062 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1063 return -1;
1064
1065 return length;
1066
1067error:
1068 m->params_valid = 0;
1069 return -1;
1070}
1071
9ba4821d 1072#if CONFIG_MLP_DECODER
b517af05
RP
1073AVCodec mlp_decoder = {
1074 "mlp",
1075 CODEC_TYPE_AUDIO,
1076 CODEC_ID_MLP,
1077 sizeof(MLPDecodeContext),
1078 mlp_decode_init,
1079 NULL,
1080 NULL,
1081 read_access_unit,
9ba4821d 1082 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
b517af05 1083};
9ba4821d 1084#endif /* CONFIG_MLP_DECODER */
b517af05 1085
9ba4821d
RP
1086#if CONFIG_TRUEHD_DECODER
1087AVCodec truehd_decoder = {
1088 "truehd",
1089 CODEC_TYPE_AUDIO,
1090 CODEC_ID_TRUEHD,
1091 sizeof(MLPDecodeContext),
1092 mlp_decode_init,
1093 NULL,
1094 NULL,
1095 read_access_unit,
1096 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1097};
1098#endif /* CONFIG_TRUEHD_DECODER */