mlpdec: Split read_matrix_params() into its own function.
[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
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501/** Read parameters for primitive matrices. */
502
503static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
504{
505 unsigned int mat, ch;
506
507 s->num_primitive_matrices = get_bits(gbp, 4);
508
509 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
510 int frac_bits, max_chan;
511 s->matrix_out_ch[mat] = get_bits(gbp, 4);
512 frac_bits = get_bits(gbp, 4);
513 s->lsb_bypass [mat] = get_bits1(gbp);
514
515 if (s->matrix_out_ch[mat] > s->max_channel) {
516 av_log(m->avctx, AV_LOG_ERROR,
517 "Invalid channel %d specified as output from matrix.\n",
518 s->matrix_out_ch[mat]);
519 return -1;
520 }
521 if (frac_bits > 14) {
522 av_log(m->avctx, AV_LOG_ERROR,
523 "Too many fractional bits specified.\n");
524 return -1;
525 }
526
527 max_chan = s->max_matrix_channel;
528 if (!s->noise_type)
529 max_chan+=2;
530
531 for (ch = 0; ch <= max_chan; ch++) {
532 int coeff_val = 0;
533 if (get_bits1(gbp))
534 coeff_val = get_sbits(gbp, frac_bits + 2);
535
536 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
537 }
538
539 if (s->noise_type)
540 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
541 else
542 s->matrix_noise_shift[mat] = 0;
543 }
544
545 return 0;
546}
547
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548/** Read decoding parameters that change more often than those in the restart
549 * header. */
550
551static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
552 unsigned int substr)
553{
554 SubStream *s = &m->substream[substr];
f8e6293b 555 unsigned int ch;
b517af05 556
cbf3cf19 557 if (s->param_presence_flags & PARAM_PRESENCE)
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558 if (get_bits1(gbp))
559 s->param_presence_flags = get_bits(gbp, 8);
560
561 if (s->param_presence_flags & PARAM_BLOCKSIZE)
562 if (get_bits1(gbp)) {
563 s->blocksize = get_bits(gbp, 9);
564 if (s->blocksize > MAX_BLOCKSIZE) {
9906a2be 565 av_log(m->avctx, AV_LOG_ERROR, "block size too large\n");
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566 s->blocksize = 0;
567 return -1;
568 }
569 }
570
571 if (s->param_presence_flags & PARAM_MATRIX)
572 if (get_bits1(gbp)) {
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573 if (read_matrix_params(m, s, gbp) < 0)
574 return -1;
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575 }
576
577 if (s->param_presence_flags & PARAM_OUTSHIFT)
578 if (get_bits1(gbp))
579 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
580 s->output_shift[ch] = get_bits(gbp, 4);
581 dprintf(m->avctx, "output shift[%d] = %d\n",
582 ch, s->output_shift[ch]);
583 /* TODO: validate */
584 }
585
586 if (s->param_presence_flags & PARAM_QUANTSTEP)
587 if (get_bits1(gbp))
588 for (ch = 0; ch <= s->max_channel; ch++) {
f53acb7b
RP
589 ChannelParams *cp = &m->channel_params[ch];
590
b517af05
RP
591 s->quant_step_size[ch] = get_bits(gbp, 4);
592 /* TODO: validate */
593
f53acb7b 594 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
b517af05
RP
595 }
596
597 for (ch = s->min_channel; ch <= s->max_channel; ch++)
598 if (get_bits1(gbp)) {
f53acb7b
RP
599 ChannelParams *cp = &m->channel_params[ch];
600 FilterParams *fir = &cp->filter_params[FIR];
601 FilterParams *iir = &cp->filter_params[IIR];
d6aa052d 602
b517af05
RP
603 if (s->param_presence_flags & PARAM_FIR)
604 if (get_bits1(gbp))
605 if (read_filter_params(m, gbp, ch, FIR) < 0)
606 return -1;
607
608 if (s->param_presence_flags & PARAM_IIR)
609 if (get_bits1(gbp))
610 if (read_filter_params(m, gbp, ch, IIR) < 0)
611 return -1;
612
d6aa052d
RP
613 if (fir->order && iir->order &&
614 fir->shift != iir->shift) {
b517af05 615 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 616 "FIR and IIR filters must use the same precision.\n");
b517af05
RP
617 return -1;
618 }
619 /* The FIR and IIR filters must have the same precision.
620 * To simplify the filtering code, only the precision of the
621 * FIR filter is considered. If only the IIR filter is employed,
622 * the FIR filter precision is set to that of the IIR filter, so
623 * that the filtering code can use it. */
d6aa052d
RP
624 if (!fir->order && iir->order)
625 fir->shift = iir->shift;
b517af05
RP
626
627 if (s->param_presence_flags & PARAM_HUFFOFFSET)
628 if (get_bits1(gbp))
f53acb7b 629 cp->huff_offset = get_sbits(gbp, 15);
b517af05 630
f53acb7b
RP
631 cp->codebook = get_bits(gbp, 2);
632 cp->huff_lsbs = get_bits(gbp, 5);
b517af05 633
f53acb7b 634 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
b517af05
RP
635
636 /* TODO: validate */
637 }
638
639 return 0;
640}
641
642#define MSB_MASK(bits) (-1u << bits)
643
644/** Generate PCM samples using the prediction filters and residual values
645 * read from the data stream, and update the filter state. */
646
647static void filter_channel(MLPDecodeContext *m, unsigned int substr,
648 unsigned int channel)
649{
650 SubStream *s = &m->substream[substr];
651 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
f53acb7b
RP
652 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
653 &m->channel_params[channel].filter_params[IIR], };
d6aa052d 654 unsigned int filter_shift = fp[FIR]->shift;
b517af05
RP
655 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
656 int index = MAX_BLOCKSIZE;
657 int j, i;
658
659 for (j = 0; j < NUM_FILTERS; j++) {
db01fa13 660 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
b517af05
RP
661 MAX_FILTER_ORDER * sizeof(int32_t));
662 }
663
664 for (i = 0; i < s->blocksize; i++) {
665 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
666 unsigned int order;
667 int64_t accum = 0;
668 int32_t result;
669
670 /* TODO: Move this code to DSPContext? */
671
672 for (j = 0; j < NUM_FILTERS; j++)
d6aa052d 673 for (order = 0; order < fp[j]->order; order++)
b517af05 674 accum += (int64_t)filter_state_buffer[j][index + order] *
db01fa13 675 fp[j]->coeff[order];
b517af05
RP
676
677 accum = accum >> filter_shift;
678 result = (accum + residual) & mask;
679
680 --index;
681
682 filter_state_buffer[FIR][index] = result;
683 filter_state_buffer[IIR][index] = result - accum;
684
685 m->sample_buffer[i + s->blockpos][channel] = result;
686 }
687
688 for (j = 0; j < NUM_FILTERS; j++) {
db01fa13 689 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
b517af05
RP
690 MAX_FILTER_ORDER * sizeof(int32_t));
691 }
692}
693
694/** Read a block of PCM residual data (or actual if no filtering active). */
695
696static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
697 unsigned int substr)
698{
699 SubStream *s = &m->substream[substr];
700 unsigned int i, ch, expected_stream_pos = 0;
701
702 if (s->data_check_present) {
703 expected_stream_pos = get_bits_count(gbp);
704 expected_stream_pos += get_bits(gbp, 16);
705 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
706 "we have not tested yet. %s\n", sample_message);
707 }
708
709 if (s->blockpos + s->blocksize > m->access_unit_size) {
9906a2be 710 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
b517af05
RP
711 return -1;
712 }
713
714 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
715 s->blocksize * sizeof(m->bypassed_lsbs[0]));
716
717 for (i = 0; i < s->blocksize; i++) {
718 if (read_huff_channels(m, gbp, substr, i) < 0)
719 return -1;
720 }
721
722 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
723 filter_channel(m, substr, ch);
724 }
725
726 s->blockpos += s->blocksize;
727
728 if (s->data_check_present) {
729 if (get_bits_count(gbp) != expected_stream_pos)
9906a2be 730 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
b517af05
RP
731 skip_bits(gbp, 8);
732 }
733
734 return 0;
735}
736
9906a2be 737/** Data table used for TrueHD noise generation function. */
b517af05
RP
738
739static const int8_t noise_table[256] = {
740 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
741 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
742 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
743 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
744 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
745 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
746 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
747 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
748 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
749 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
750 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
751 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
752 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
753 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
754 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
755 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
756};
757
758/** Noise generation functions.
759 * I'm not sure what these are for - they seem to be some kind of pseudorandom
760 * sequence generators, used to generate noise data which is used when the
761 * channels are rematrixed. I'm not sure if they provide a practical benefit
762 * to compression, or just obfuscate the decoder. Are they for some kind of
763 * dithering? */
764
765/** Generate two channels of noise, used in the matrix when
766 * restart sync word == 0x31ea. */
767
768static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
769{
770 SubStream *s = &m->substream[substr];
771 unsigned int i;
772 uint32_t seed = s->noisegen_seed;
773 unsigned int maxchan = s->max_matrix_channel;
774
775 for (i = 0; i < s->blockpos; i++) {
776 uint16_t seed_shr7 = seed >> 7;
777 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
778 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
779
780 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
781 }
782
783 s->noisegen_seed = seed;
784}
785
786/** Generate a block of noise, used when restart sync word == 0x31eb. */
787
788static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
789{
790 SubStream *s = &m->substream[substr];
791 unsigned int i;
792 uint32_t seed = s->noisegen_seed;
793
794 for (i = 0; i < m->access_unit_size_pow2; i++) {
795 uint8_t seed_shr15 = seed >> 15;
796 m->noise_buffer[i] = noise_table[seed_shr15];
797 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
798 }
799
800 s->noisegen_seed = seed;
801}
802
803
804/** Apply the channel matrices in turn to reconstruct the original audio
805 * samples. */
806
807static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
808{
809 SubStream *s = &m->substream[substr];
810 unsigned int mat, src_ch, i;
811 unsigned int maxchan;
812
813 maxchan = s->max_matrix_channel;
814 if (!s->noise_type) {
815 generate_2_noise_channels(m, substr);
816 maxchan += 2;
817 } else {
818 fill_noise_buffer(m, substr);
819 }
820
821 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
822 int matrix_noise_shift = s->matrix_noise_shift[mat];
823 unsigned int dest_ch = s->matrix_out_ch[mat];
824 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
825
826 /* TODO: DSPContext? */
827
828 for (i = 0; i < s->blockpos; i++) {
829 int64_t accum = 0;
830 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
831 accum += (int64_t)m->sample_buffer[i][src_ch]
832 * s->matrix_coeff[mat][src_ch];
833 }
834 if (matrix_noise_shift) {
835 uint32_t index = s->num_primitive_matrices - mat;
836 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
837 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
838 }
839 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
840 + m->bypassed_lsbs[i][mat];
841 }
842 }
843}
844
845/** Write the audio data into the output buffer. */
846
847static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
848 uint8_t *data, unsigned int *data_size, int is32)
849{
850 SubStream *s = &m->substream[substr];
9731e7f1 851 unsigned int i, out_ch = 0;
b517af05
RP
852 int32_t *data_32 = (int32_t*) data;
853 int16_t *data_16 = (int16_t*) data;
854
855 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
856 return -1;
857
858 for (i = 0; i < s->blockpos; i++) {
9731e7f1
RP
859 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
860 int mat_ch = s->ch_assign[out_ch];
861 int32_t sample = m->sample_buffer[i][mat_ch]
862 << s->output_shift[mat_ch];
863 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
b517af05
RP
864 if (is32) *data_32++ = sample << 8;
865 else *data_16++ = sample >> 8;
866 }
867 }
868
9731e7f1 869 *data_size = i * out_ch * (is32 ? 4 : 2);
b517af05
RP
870
871 return 0;
872}
873
874static int output_data(MLPDecodeContext *m, unsigned int substr,
875 uint8_t *data, unsigned int *data_size)
876{
877 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
878 return output_data_internal(m, substr, data, data_size, 1);
879 else
880 return output_data_internal(m, substr, data, data_size, 0);
881}
882
883
b517af05
RP
884/** Read an access unit from the stream.
885 * Returns < 0 on error, 0 if not enough data is present in the input stream
886 * otherwise returns the number of bytes consumed. */
887
888static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
889 const uint8_t *buf, int buf_size)
890{
891 MLPDecodeContext *m = avctx->priv_data;
892 GetBitContext gb;
893 unsigned int length, substr;
894 unsigned int substream_start;
895 unsigned int header_size = 4;
896 unsigned int substr_header_size = 0;
897 uint8_t substream_parity_present[MAX_SUBSTREAMS];
898 uint16_t substream_data_len[MAX_SUBSTREAMS];
899 uint8_t parity_bits;
900
901 if (buf_size < 4)
902 return 0;
903
904 length = (AV_RB16(buf) & 0xfff) * 2;
905
906 if (length > buf_size)
907 return -1;
908
909 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
910
911 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
9906a2be 912 dprintf(m->avctx, "Found major sync.\n");
b517af05
RP
913 if (read_major_sync(m, &gb) < 0)
914 goto error;
915 header_size += 28;
916 }
917
918 if (!m->params_valid) {
919 av_log(m->avctx, AV_LOG_WARNING,
9906a2be 920 "Stream parameters not seen; skipping frame.\n");
b517af05
RP
921 *data_size = 0;
922 return length;
923 }
924
925 substream_start = 0;
926
927 for (substr = 0; substr < m->num_substreams; substr++) {
928 int extraword_present, checkdata_present, end;
929
930 extraword_present = get_bits1(&gb);
931 skip_bits1(&gb);
932 checkdata_present = get_bits1(&gb);
933 skip_bits1(&gb);
934
935 end = get_bits(&gb, 12) * 2;
936
937 substr_header_size += 2;
938
939 if (extraword_present) {
940 skip_bits(&gb, 16);
941 substr_header_size += 2;
942 }
943
944 if (end + header_size + substr_header_size > length) {
945 av_log(m->avctx, AV_LOG_ERROR,
946 "Indicated length of substream %d data goes off end of "
947 "packet.\n", substr);
948 end = length - header_size - substr_header_size;
949 }
950
951 if (end < substream_start) {
952 av_log(avctx, AV_LOG_ERROR,
953 "Indicated end offset of substream %d data "
954 "is smaller than calculated start offset.\n",
955 substr);
956 goto error;
957 }
958
959 if (substr > m->max_decoded_substream)
960 continue;
961
962 substream_parity_present[substr] = checkdata_present;
963 substream_data_len[substr] = end - substream_start;
964 substream_start = end;
965 }
966
ce15710f
RP
967 parity_bits = ff_mlp_calculate_parity(buf, 4);
968 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
b517af05
RP
969
970 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
971 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
972 goto error;
973 }
974
975 buf += header_size + substr_header_size;
976
977 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
978 SubStream *s = &m->substream[substr];
979 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
980
981 s->blockpos = 0;
982 do {
983 if (get_bits1(&gb)) {
984 if (get_bits1(&gb)) {
9906a2be 985 /* A restart header should be present. */
b517af05
RP
986 if (read_restart_header(m, &gb, buf, substr) < 0)
987 goto next_substr;
988 s->restart_seen = 1;
989 }
990
991 if (!s->restart_seen) {
992 av_log(m->avctx, AV_LOG_ERROR,
993 "No restart header present in substream %d.\n",
994 substr);
995 goto next_substr;
996 }
997
998 if (read_decoding_params(m, &gb, substr) < 0)
999 goto next_substr;
1000 }
1001
1002 if (!s->restart_seen) {
1003 av_log(m->avctx, AV_LOG_ERROR,
1004 "No restart header present in substream %d.\n",
1005 substr);
1006 goto next_substr;
1007 }
1008
1009 if (read_block_data(m, &gb, substr) < 0)
1010 return -1;
1011
1012 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1013 && get_bits1(&gb) == 0);
1014
1015 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
eb2cd99c 1016 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
1e8bd10c 1017 (show_bits_long(&gb, 32) == END_OF_STREAM ||
b517af05
RP
1018 show_bits_long(&gb, 20) == 0xd234e)) {
1019 skip_bits(&gb, 18);
1020 if (substr == m->max_decoded_substream)
9906a2be 1021 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
b517af05
RP
1022
1023 if (get_bits1(&gb)) {
1024 int shorten_by = get_bits(&gb, 13);
1025 shorten_by = FFMIN(shorten_by, s->blockpos);
1026 s->blockpos -= shorten_by;
1027 } else
1028 skip_bits(&gb, 13);
1029 }
eb2cd99c
RP
1030 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1031 substream_parity_present[substr]) {
b517af05
RP
1032 uint8_t parity, checksum;
1033
ce15710f 1034 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
b517af05
RP
1035 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1036 av_log(m->avctx, AV_LOG_ERROR,
9906a2be 1037 "Substream %d parity check failed.\n", substr);
b517af05 1038
ce15710f 1039 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
b517af05 1040 if (checksum != get_bits(&gb, 8))
9906a2be 1041 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
b517af05
RP
1042 substr);
1043 }
1044 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
9906a2be 1045 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
b517af05
RP
1046 substr);
1047 return -1;
1048 }
1049
1050next_substr:
1051 buf += substream_data_len[substr];
1052 }
1053
1054 rematrix_channels(m, m->max_decoded_substream);
1055
1056 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1057 return -1;
1058
1059 return length;
1060
1061error:
1062 m->params_valid = 0;
1063 return -1;
1064}
1065
9ba4821d 1066#if CONFIG_MLP_DECODER
b517af05
RP
1067AVCodec mlp_decoder = {
1068 "mlp",
1069 CODEC_TYPE_AUDIO,
1070 CODEC_ID_MLP,
1071 sizeof(MLPDecodeContext),
1072 mlp_decode_init,
1073 NULL,
1074 NULL,
1075 read_access_unit,
9ba4821d 1076 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
b517af05 1077};
9ba4821d 1078#endif /* CONFIG_MLP_DECODER */
b517af05 1079
9ba4821d
RP
1080#if CONFIG_TRUEHD_DECODER
1081AVCodec truehd_decoder = {
1082 "truehd",
1083 CODEC_TYPE_AUDIO,
1084 CODEC_ID_TRUEHD,
1085 sizeof(MLPDecodeContext),
1086 mlp_decode_init,
1087 NULL,
1088 NULL,
1089 read_access_unit,
1090 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1091};
1092#endif /* CONFIG_TRUEHD_DECODER */