Add myself as mlp* maintainer.
[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"
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",
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377 lossless_check, tmp);
378 else
9906a2be 379 dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n",
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380 substr, tmp);
381 }
382
383 skip_bits(gbp, 16);
384
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385 memset(s->ch_assign, 0, sizeof(s->ch_assign));
386
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387 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
388 int ch_assign = get_bits(gbp, 6);
389 dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
390 ch_assign);
9731e7f1 391 if (ch_assign > s->max_matrix_channel) {
b517af05 392 av_log(m->avctx, AV_LOG_ERROR,
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393 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
394 ch, ch_assign, sample_message);
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395 return -1;
396 }
9731e7f1 397 s->ch_assign[ch_assign] = ch;
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398 }
399
ce15710f 400 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
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401
402 if (checksum != get_bits(gbp, 8))
9906a2be 403 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
b517af05 404
9906a2be 405 /* Set default decoding parameters. */
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406 s->param_presence_flags = 0xff;
407 s->num_primitive_matrices = 0;
408 s->blocksize = 8;
409 s->lossless_check_data = 0;
410
411 memset(s->output_shift , 0, sizeof(s->output_shift ));
412 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
413
414 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
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415 ChannelParams *cp = &m->channel_params[ch];
416 cp->filter_params[FIR].order = 0;
417 cp->filter_params[IIR].order = 0;
418 cp->filter_params[FIR].shift = 0;
419 cp->filter_params[IIR].shift = 0;
b517af05 420
9906a2be 421 /* Default audio coding is 24-bit raw PCM. */
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422 cp->huff_offset = 0;
423 cp->sign_huff_offset = (-1) << 23;
424 cp->codebook = 0;
425 cp->huff_lsbs = 24;
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426 }
427
428 if (substr == m->max_decoded_substream) {
9731e7f1 429 m->avctx->channels = s->max_matrix_channel + 1;
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430 }
431
432 return 0;
433}
434
435/** Read parameters for one of the prediction filters. */
436
437static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
438 unsigned int channel, unsigned int filter)
439{
f53acb7b 440 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
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441 const char fchar = filter ? 'I' : 'F';
442 int i, order;
443
9906a2be 444 // Filter is 0 for FIR, 1 for IIR.
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445 assert(filter < 2);
446
447 order = get_bits(gbp, 4);
448 if (order > MAX_FILTER_ORDER) {
449 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 450 "%cIR filter order %d is greater than maximum %d.\n",
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451 fchar, order, MAX_FILTER_ORDER);
452 return -1;
453 }
d6aa052d 454 fp->order = order;
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455
456 if (order > 0) {
457 int coeff_bits, coeff_shift;
458
d6aa052d 459 fp->shift = get_bits(gbp, 4);
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460
461 coeff_bits = get_bits(gbp, 5);
462 coeff_shift = get_bits(gbp, 3);
463 if (coeff_bits < 1 || coeff_bits > 16) {
464 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 465 "%cIR filter coeff_bits must be between 1 and 16.\n",
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466 fchar);
467 return -1;
468 }
469 if (coeff_bits + coeff_shift > 16) {
470 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 471 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
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472 fchar);
473 return -1;
474 }
475
476 for (i = 0; i < order; i++)
db01fa13 477 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
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478
479 if (get_bits1(gbp)) {
480 int state_bits, state_shift;
481
482 if (filter == FIR) {
483 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 484 "FIR filter has state data specified.\n");
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485 return -1;
486 }
487
488 state_bits = get_bits(gbp, 4);
489 state_shift = get_bits(gbp, 4);
490
9906a2be 491 /* TODO: Check validity of state data. */
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492
493 for (i = 0; i < order; i++)
db01fa13 494 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
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495 }
496 }
497
498 return 0;
499}
500
501/** Read decoding parameters that change more often than those in the restart
502 * header. */
503
504static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
505 unsigned int substr)
506{
507 SubStream *s = &m->substream[substr];
508 unsigned int mat, ch;
509
cbf3cf19 510 if (s->param_presence_flags & PARAM_PRESENCE)
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511 if (get_bits1(gbp))
512 s->param_presence_flags = get_bits(gbp, 8);
513
514 if (s->param_presence_flags & PARAM_BLOCKSIZE)
515 if (get_bits1(gbp)) {
516 s->blocksize = get_bits(gbp, 9);
517 if (s->blocksize > MAX_BLOCKSIZE) {
9906a2be 518 av_log(m->avctx, AV_LOG_ERROR, "block size too large\n");
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519 s->blocksize = 0;
520 return -1;
521 }
522 }
523
524 if (s->param_presence_flags & PARAM_MATRIX)
525 if (get_bits1(gbp)) {
526 s->num_primitive_matrices = get_bits(gbp, 4);
527
528 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
529 int frac_bits, max_chan;
530 s->matrix_out_ch[mat] = get_bits(gbp, 4);
531 frac_bits = get_bits(gbp, 4);
532 s->lsb_bypass [mat] = get_bits1(gbp);
533
534 if (s->matrix_out_ch[mat] > s->max_channel) {
535 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 536 "Invalid channel %d specified as output from matrix.\n",
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537 s->matrix_out_ch[mat]);
538 return -1;
539 }
540 if (frac_bits > 14) {
541 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 542 "Too many fractional bits specified.\n");
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543 return -1;
544 }
545
546 max_chan = s->max_matrix_channel;
547 if (!s->noise_type)
548 max_chan+=2;
549
550 for (ch = 0; ch <= max_chan; ch++) {
551 int coeff_val = 0;
552 if (get_bits1(gbp))
553 coeff_val = get_sbits(gbp, frac_bits + 2);
554
555 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
556 }
557
558 if (s->noise_type)
559 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
560 else
561 s->matrix_noise_shift[mat] = 0;
562 }
563 }
564
565 if (s->param_presence_flags & PARAM_OUTSHIFT)
566 if (get_bits1(gbp))
567 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
568 s->output_shift[ch] = get_bits(gbp, 4);
569 dprintf(m->avctx, "output shift[%d] = %d\n",
570 ch, s->output_shift[ch]);
571 /* TODO: validate */
572 }
573
574 if (s->param_presence_flags & PARAM_QUANTSTEP)
575 if (get_bits1(gbp))
576 for (ch = 0; ch <= s->max_channel; ch++) {
f53acb7b
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577 ChannelParams *cp = &m->channel_params[ch];
578
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RP
579 s->quant_step_size[ch] = get_bits(gbp, 4);
580 /* TODO: validate */
581
f53acb7b 582 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
b517af05
RP
583 }
584
585 for (ch = s->min_channel; ch <= s->max_channel; ch++)
586 if (get_bits1(gbp)) {
f53acb7b
RP
587 ChannelParams *cp = &m->channel_params[ch];
588 FilterParams *fir = &cp->filter_params[FIR];
589 FilterParams *iir = &cp->filter_params[IIR];
d6aa052d 590
b517af05
RP
591 if (s->param_presence_flags & PARAM_FIR)
592 if (get_bits1(gbp))
593 if (read_filter_params(m, gbp, ch, FIR) < 0)
594 return -1;
595
596 if (s->param_presence_flags & PARAM_IIR)
597 if (get_bits1(gbp))
598 if (read_filter_params(m, gbp, ch, IIR) < 0)
599 return -1;
600
d6aa052d
RP
601 if (fir->order && iir->order &&
602 fir->shift != iir->shift) {
b517af05 603 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 604 "FIR and IIR filters must use the same precision.\n");
b517af05
RP
605 return -1;
606 }
607 /* The FIR and IIR filters must have the same precision.
608 * To simplify the filtering code, only the precision of the
609 * FIR filter is considered. If only the IIR filter is employed,
610 * the FIR filter precision is set to that of the IIR filter, so
611 * that the filtering code can use it. */
d6aa052d
RP
612 if (!fir->order && iir->order)
613 fir->shift = iir->shift;
b517af05
RP
614
615 if (s->param_presence_flags & PARAM_HUFFOFFSET)
616 if (get_bits1(gbp))
f53acb7b 617 cp->huff_offset = get_sbits(gbp, 15);
b517af05 618
f53acb7b
RP
619 cp->codebook = get_bits(gbp, 2);
620 cp->huff_lsbs = get_bits(gbp, 5);
b517af05 621
f53acb7b 622 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
b517af05
RP
623
624 /* TODO: validate */
625 }
626
627 return 0;
628}
629
630#define MSB_MASK(bits) (-1u << bits)
631
632/** Generate PCM samples using the prediction filters and residual values
633 * read from the data stream, and update the filter state. */
634
635static void filter_channel(MLPDecodeContext *m, unsigned int substr,
636 unsigned int channel)
637{
638 SubStream *s = &m->substream[substr];
639 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
f53acb7b
RP
640 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
641 &m->channel_params[channel].filter_params[IIR], };
d6aa052d 642 unsigned int filter_shift = fp[FIR]->shift;
b517af05
RP
643 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
644 int index = MAX_BLOCKSIZE;
645 int j, i;
646
647 for (j = 0; j < NUM_FILTERS; j++) {
db01fa13 648 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
b517af05
RP
649 MAX_FILTER_ORDER * sizeof(int32_t));
650 }
651
652 for (i = 0; i < s->blocksize; i++) {
653 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
654 unsigned int order;
655 int64_t accum = 0;
656 int32_t result;
657
658 /* TODO: Move this code to DSPContext? */
659
660 for (j = 0; j < NUM_FILTERS; j++)
d6aa052d 661 for (order = 0; order < fp[j]->order; order++)
b517af05 662 accum += (int64_t)filter_state_buffer[j][index + order] *
db01fa13 663 fp[j]->coeff[order];
b517af05
RP
664
665 accum = accum >> filter_shift;
666 result = (accum + residual) & mask;
667
668 --index;
669
670 filter_state_buffer[FIR][index] = result;
671 filter_state_buffer[IIR][index] = result - accum;
672
673 m->sample_buffer[i + s->blockpos][channel] = result;
674 }
675
676 for (j = 0; j < NUM_FILTERS; j++) {
db01fa13 677 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
b517af05
RP
678 MAX_FILTER_ORDER * sizeof(int32_t));
679 }
680}
681
682/** Read a block of PCM residual data (or actual if no filtering active). */
683
684static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
685 unsigned int substr)
686{
687 SubStream *s = &m->substream[substr];
688 unsigned int i, ch, expected_stream_pos = 0;
689
690 if (s->data_check_present) {
691 expected_stream_pos = get_bits_count(gbp);
692 expected_stream_pos += get_bits(gbp, 16);
693 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
694 "we have not tested yet. %s\n", sample_message);
695 }
696
697 if (s->blockpos + s->blocksize > m->access_unit_size) {
9906a2be 698 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
b517af05
RP
699 return -1;
700 }
701
702 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
703 s->blocksize * sizeof(m->bypassed_lsbs[0]));
704
705 for (i = 0; i < s->blocksize; i++) {
706 if (read_huff_channels(m, gbp, substr, i) < 0)
707 return -1;
708 }
709
710 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
711 filter_channel(m, substr, ch);
712 }
713
714 s->blockpos += s->blocksize;
715
716 if (s->data_check_present) {
717 if (get_bits_count(gbp) != expected_stream_pos)
9906a2be 718 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
b517af05
RP
719 skip_bits(gbp, 8);
720 }
721
722 return 0;
723}
724
9906a2be 725/** Data table used for TrueHD noise generation function. */
b517af05
RP
726
727static const int8_t noise_table[256] = {
728 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
729 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
730 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
731 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
732 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
733 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
734 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
735 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
736 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
737 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
738 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
739 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
740 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
741 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
742 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
743 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
744};
745
746/** Noise generation functions.
747 * I'm not sure what these are for - they seem to be some kind of pseudorandom
748 * sequence generators, used to generate noise data which is used when the
749 * channels are rematrixed. I'm not sure if they provide a practical benefit
750 * to compression, or just obfuscate the decoder. Are they for some kind of
751 * dithering? */
752
753/** Generate two channels of noise, used in the matrix when
754 * restart sync word == 0x31ea. */
755
756static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
757{
758 SubStream *s = &m->substream[substr];
759 unsigned int i;
760 uint32_t seed = s->noisegen_seed;
761 unsigned int maxchan = s->max_matrix_channel;
762
763 for (i = 0; i < s->blockpos; i++) {
764 uint16_t seed_shr7 = seed >> 7;
765 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
766 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
767
768 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
769 }
770
771 s->noisegen_seed = seed;
772}
773
774/** Generate a block of noise, used when restart sync word == 0x31eb. */
775
776static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
777{
778 SubStream *s = &m->substream[substr];
779 unsigned int i;
780 uint32_t seed = s->noisegen_seed;
781
782 for (i = 0; i < m->access_unit_size_pow2; i++) {
783 uint8_t seed_shr15 = seed >> 15;
784 m->noise_buffer[i] = noise_table[seed_shr15];
785 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
786 }
787
788 s->noisegen_seed = seed;
789}
790
791
792/** Apply the channel matrices in turn to reconstruct the original audio
793 * samples. */
794
795static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
796{
797 SubStream *s = &m->substream[substr];
798 unsigned int mat, src_ch, i;
799 unsigned int maxchan;
800
801 maxchan = s->max_matrix_channel;
802 if (!s->noise_type) {
803 generate_2_noise_channels(m, substr);
804 maxchan += 2;
805 } else {
806 fill_noise_buffer(m, substr);
807 }
808
809 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
810 int matrix_noise_shift = s->matrix_noise_shift[mat];
811 unsigned int dest_ch = s->matrix_out_ch[mat];
812 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
813
814 /* TODO: DSPContext? */
815
816 for (i = 0; i < s->blockpos; i++) {
817 int64_t accum = 0;
818 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
819 accum += (int64_t)m->sample_buffer[i][src_ch]
820 * s->matrix_coeff[mat][src_ch];
821 }
822 if (matrix_noise_shift) {
823 uint32_t index = s->num_primitive_matrices - mat;
824 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
825 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
826 }
827 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
828 + m->bypassed_lsbs[i][mat];
829 }
830 }
831}
832
833/** Write the audio data into the output buffer. */
834
835static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
836 uint8_t *data, unsigned int *data_size, int is32)
837{
838 SubStream *s = &m->substream[substr];
9731e7f1 839 unsigned int i, out_ch = 0;
b517af05
RP
840 int32_t *data_32 = (int32_t*) data;
841 int16_t *data_16 = (int16_t*) data;
842
843 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
844 return -1;
845
846 for (i = 0; i < s->blockpos; i++) {
9731e7f1
RP
847 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
848 int mat_ch = s->ch_assign[out_ch];
849 int32_t sample = m->sample_buffer[i][mat_ch]
850 << s->output_shift[mat_ch];
851 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
b517af05
RP
852 if (is32) *data_32++ = sample << 8;
853 else *data_16++ = sample >> 8;
854 }
855 }
856
9731e7f1 857 *data_size = i * out_ch * (is32 ? 4 : 2);
b517af05
RP
858
859 return 0;
860}
861
862static int output_data(MLPDecodeContext *m, unsigned int substr,
863 uint8_t *data, unsigned int *data_size)
864{
865 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
866 return output_data_internal(m, substr, data, data_size, 1);
867 else
868 return output_data_internal(m, substr, data, data_size, 0);
869}
870
871
b517af05
RP
872/** Read an access unit from the stream.
873 * Returns < 0 on error, 0 if not enough data is present in the input stream
874 * otherwise returns the number of bytes consumed. */
875
876static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
877 const uint8_t *buf, int buf_size)
878{
879 MLPDecodeContext *m = avctx->priv_data;
880 GetBitContext gb;
881 unsigned int length, substr;
882 unsigned int substream_start;
883 unsigned int header_size = 4;
884 unsigned int substr_header_size = 0;
885 uint8_t substream_parity_present[MAX_SUBSTREAMS];
886 uint16_t substream_data_len[MAX_SUBSTREAMS];
887 uint8_t parity_bits;
888
889 if (buf_size < 4)
890 return 0;
891
892 length = (AV_RB16(buf) & 0xfff) * 2;
893
894 if (length > buf_size)
895 return -1;
896
897 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
898
899 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
9906a2be 900 dprintf(m->avctx, "Found major sync.\n");
b517af05
RP
901 if (read_major_sync(m, &gb) < 0)
902 goto error;
903 header_size += 28;
904 }
905
906 if (!m->params_valid) {
907 av_log(m->avctx, AV_LOG_WARNING,
9906a2be 908 "Stream parameters not seen; skipping frame.\n");
b517af05
RP
909 *data_size = 0;
910 return length;
911 }
912
913 substream_start = 0;
914
915 for (substr = 0; substr < m->num_substreams; substr++) {
916 int extraword_present, checkdata_present, end;
917
918 extraword_present = get_bits1(&gb);
919 skip_bits1(&gb);
920 checkdata_present = get_bits1(&gb);
921 skip_bits1(&gb);
922
923 end = get_bits(&gb, 12) * 2;
924
925 substr_header_size += 2;
926
927 if (extraword_present) {
928 skip_bits(&gb, 16);
929 substr_header_size += 2;
930 }
931
932 if (end + header_size + substr_header_size > length) {
933 av_log(m->avctx, AV_LOG_ERROR,
934 "Indicated length of substream %d data goes off end of "
935 "packet.\n", substr);
936 end = length - header_size - substr_header_size;
937 }
938
939 if (end < substream_start) {
940 av_log(avctx, AV_LOG_ERROR,
941 "Indicated end offset of substream %d data "
942 "is smaller than calculated start offset.\n",
943 substr);
944 goto error;
945 }
946
947 if (substr > m->max_decoded_substream)
948 continue;
949
950 substream_parity_present[substr] = checkdata_present;
951 substream_data_len[substr] = end - substream_start;
952 substream_start = end;
953 }
954
ce15710f
RP
955 parity_bits = ff_mlp_calculate_parity(buf, 4);
956 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
b517af05
RP
957
958 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
959 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
960 goto error;
961 }
962
963 buf += header_size + substr_header_size;
964
965 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
966 SubStream *s = &m->substream[substr];
967 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
968
969 s->blockpos = 0;
970 do {
971 if (get_bits1(&gb)) {
972 if (get_bits1(&gb)) {
9906a2be 973 /* A restart header should be present. */
b517af05
RP
974 if (read_restart_header(m, &gb, buf, substr) < 0)
975 goto next_substr;
976 s->restart_seen = 1;
977 }
978
979 if (!s->restart_seen) {
980 av_log(m->avctx, AV_LOG_ERROR,
981 "No restart header present in substream %d.\n",
982 substr);
983 goto next_substr;
984 }
985
986 if (read_decoding_params(m, &gb, substr) < 0)
987 goto next_substr;
988 }
989
990 if (!s->restart_seen) {
991 av_log(m->avctx, AV_LOG_ERROR,
992 "No restart header present in substream %d.\n",
993 substr);
994 goto next_substr;
995 }
996
997 if (read_block_data(m, &gb, substr) < 0)
998 return -1;
999
1000 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1001 && get_bits1(&gb) == 0);
1002
1003 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
eb2cd99c 1004 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
1e8bd10c 1005 (show_bits_long(&gb, 32) == END_OF_STREAM ||
b517af05
RP
1006 show_bits_long(&gb, 20) == 0xd234e)) {
1007 skip_bits(&gb, 18);
1008 if (substr == m->max_decoded_substream)
9906a2be 1009 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
b517af05
RP
1010
1011 if (get_bits1(&gb)) {
1012 int shorten_by = get_bits(&gb, 13);
1013 shorten_by = FFMIN(shorten_by, s->blockpos);
1014 s->blockpos -= shorten_by;
1015 } else
1016 skip_bits(&gb, 13);
1017 }
eb2cd99c
RP
1018 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1019 substream_parity_present[substr]) {
b517af05
RP
1020 uint8_t parity, checksum;
1021
ce15710f 1022 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
b517af05
RP
1023 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1024 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 1025 "Substream %d parity check failed.\n", substr);
b517af05 1026
ce15710f 1027 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
b517af05 1028 if (checksum != get_bits(&gb, 8))
9906a2be 1029 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
b517af05
RP
1030 substr);
1031 }
1032 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
9906a2be 1033 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
b517af05
RP
1034 substr);
1035 return -1;
1036 }
1037
1038next_substr:
1039 buf += substream_data_len[substr];
1040 }
1041
1042 rematrix_channels(m, m->max_decoded_substream);
1043
1044 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1045 return -1;
1046
1047 return length;
1048
1049error:
1050 m->params_valid = 0;
1051 return -1;
1052}
1053
9ba4821d 1054#if CONFIG_MLP_DECODER
b517af05
RP
1055AVCodec mlp_decoder = {
1056 "mlp",
1057 CODEC_TYPE_AUDIO,
1058 CODEC_ID_MLP,
1059 sizeof(MLPDecodeContext),
1060 mlp_decode_init,
1061 NULL,
1062 NULL,
1063 read_access_unit,
9ba4821d 1064 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
b517af05 1065};
9ba4821d 1066#endif /* CONFIG_MLP_DECODER */
b517af05 1067
9ba4821d
RP
1068#if CONFIG_TRUEHD_DECODER
1069AVCodec truehd_decoder = {
1070 "truehd",
1071 CODEC_TYPE_AUDIO,
1072 CODEC_ID_TRUEHD,
1073 sizeof(MLPDecodeContext),
1074 mlp_decode_init,
1075 NULL,
1076 NULL,
1077 read_access_unit,
1078 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1079};
1080#endif /* CONFIG_TRUEHD_DECODER */