3 * Copyright (c) 2007-2008 Ian Caulfield
5 * This file is part of Libav.
7 * Libav 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.
12 * Libav 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.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
31 #include "libavutil/intreadwrite.h"
33 #include "libavutil/crc.h"
35 #include "mlp_parser.h"
38 /** number of bits used for VLC lookup - longest Huffman code is 9 */
42 static const char* sample_message =
43 "Please file a bug report following the instructions at "
44 "http://libav.org/bugreports.html and include "
45 "a sample of this file.";
47 typedef struct SubStream {
48 /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
52 /** restart header data */
53 /// The type of noise to be used in the rematrix stage.
56 /// The index of the first channel coded in this substream.
58 /// The index of the last channel coded in this substream.
60 /// The number of channels input into the rematrix stage.
61 uint8_t max_matrix_channel;
62 /// For each channel output by the matrix, the output channel to map it to
63 uint8_t ch_assign[MAX_CHANNELS];
65 /// Channel coding parameters for channels in the substream
66 ChannelParams channel_params[MAX_CHANNELS];
68 /// The left shift applied to random noise in 0x31ea substreams.
70 /// The current seed value for the pseudorandom noise generator(s).
71 uint32_t noisegen_seed;
73 /// Set if the substream contains extra info to check the size of VLC blocks.
74 uint8_t data_check_present;
76 /// Bitmask of which parameter sets are conveyed in a decoding parameter block.
77 uint8_t param_presence_flags;
78 #define PARAM_BLOCKSIZE (1 << 7)
79 #define PARAM_MATRIX (1 << 6)
80 #define PARAM_OUTSHIFT (1 << 5)
81 #define PARAM_QUANTSTEP (1 << 4)
82 #define PARAM_FIR (1 << 3)
83 #define PARAM_IIR (1 << 2)
84 #define PARAM_HUFFOFFSET (1 << 1)
85 #define PARAM_PRESENCE (1 << 0)
91 /// Number of matrices to be applied.
92 uint8_t num_primitive_matrices;
94 /// matrix output channel
95 uint8_t matrix_out_ch[MAX_MATRICES];
97 /// Whether the LSBs of the matrix output are encoded in the bitstream.
98 uint8_t lsb_bypass[MAX_MATRICES];
99 /// Matrix coefficients, stored as 2.14 fixed point.
100 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
101 /// Left shift to apply to noise values in 0x31eb substreams.
102 uint8_t matrix_noise_shift[MAX_MATRICES];
105 /// Left shift to apply to Huffman-decoded residuals.
106 uint8_t quant_step_size[MAX_CHANNELS];
108 /// number of PCM samples in current audio block
110 /// Number of PCM samples decoded so far in this frame.
113 /// Left shift to apply to decoded PCM values to get final 24-bit output.
114 int8_t output_shift[MAX_CHANNELS];
116 /// Running XOR of all output samples.
117 int32_t lossless_check_data;
121 typedef struct MLPDecodeContext {
122 AVCodecContext *avctx;
125 /// Current access unit being read has a major sync.
126 int is_major_sync_unit;
128 /// Set if a valid major sync block has been read. Otherwise no decoding is possible.
129 uint8_t params_valid;
131 /// Number of substreams contained within this stream.
132 uint8_t num_substreams;
134 /// Index of the last substream to decode - further substreams are skipped.
135 uint8_t max_decoded_substream;
137 /// number of PCM samples contained in each frame
138 int access_unit_size;
139 /// next power of two above the number of samples in each frame
140 int access_unit_size_pow2;
142 SubStream substream[MAX_SUBSTREAMS];
145 int filter_changed[MAX_CHANNELS][NUM_FILTERS];
147 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
148 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
149 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
154 static VLC huff_vlc[3];
156 /** Initialize static data, constant between all invocations of the codec. */
158 static av_cold void init_static(void)
160 if (!huff_vlc[0].bits) {
161 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
162 &ff_mlp_huffman_tables[0][0][1], 2, 1,
163 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
164 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
165 &ff_mlp_huffman_tables[1][0][1], 2, 1,
166 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
167 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
168 &ff_mlp_huffman_tables[2][0][1], 2, 1,
169 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
175 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
176 unsigned int substr, unsigned int ch)
178 SubStream *s = &m->substream[substr];
179 ChannelParams *cp = &s->channel_params[ch];
180 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
181 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
182 int32_t sign_huff_offset = cp->huff_offset;
184 if (cp->codebook > 0)
185 sign_huff_offset -= 7 << lsb_bits;
188 sign_huff_offset -= 1 << sign_shift;
190 return sign_huff_offset;
193 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
196 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
197 unsigned int substr, unsigned int pos)
199 SubStream *s = &m->substream[substr];
200 unsigned int mat, channel;
202 for (mat = 0; mat < s->num_primitive_matrices; mat++)
203 if (s->lsb_bypass[mat])
204 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
206 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
207 ChannelParams *cp = &s->channel_params[channel];
208 int codebook = cp->codebook;
209 int quant_step_size = s->quant_step_size[channel];
210 int lsb_bits = cp->huff_lsbs - quant_step_size;
214 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
215 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
218 return AVERROR_INVALIDDATA;
221 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
223 result += cp->sign_huff_offset;
224 result <<= quant_step_size;
226 m->sample_buffer[pos + s->blockpos][channel] = result;
232 static av_cold int mlp_decode_init(AVCodecContext *avctx)
234 MLPDecodeContext *m = avctx->priv_data;
239 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
240 m->substream[substr].lossless_check_data = 0xffffffff;
241 ff_dsputil_init(&m->dsp, avctx);
243 avcodec_get_frame_defaults(&m->frame);
244 avctx->coded_frame = &m->frame;
249 /** Read a major sync info header - contains high level information about
250 * the stream - sample rate, channel arrangement etc. Most of this
251 * information is not actually necessary for decoding, only for playback.
254 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
259 if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
262 if (mh.group1_bits == 0) {
263 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
264 return AVERROR_INVALIDDATA;
266 if (mh.group2_bits > mh.group1_bits) {
267 av_log(m->avctx, AV_LOG_ERROR,
268 "Channel group 2 cannot have more bits per sample than group 1.\n");
269 return AVERROR_INVALIDDATA;
272 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
273 av_log(m->avctx, AV_LOG_ERROR,
274 "Channel groups with differing sample rates are not currently supported.\n");
275 return AVERROR_INVALIDDATA;
278 if (mh.group1_samplerate == 0) {
279 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
280 return AVERROR_INVALIDDATA;
282 if (mh.group1_samplerate > MAX_SAMPLERATE) {
283 av_log(m->avctx, AV_LOG_ERROR,
284 "Sampling rate %d is greater than the supported maximum (%d).\n",
285 mh.group1_samplerate, MAX_SAMPLERATE);
286 return AVERROR_INVALIDDATA;
288 if (mh.access_unit_size > MAX_BLOCKSIZE) {
289 av_log(m->avctx, AV_LOG_ERROR,
290 "Block size %d is greater than the supported maximum (%d).\n",
291 mh.access_unit_size, MAX_BLOCKSIZE);
292 return AVERROR_INVALIDDATA;
294 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
295 av_log(m->avctx, AV_LOG_ERROR,
296 "Block size pow2 %d is greater than the supported maximum (%d).\n",
297 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
298 return AVERROR_INVALIDDATA;
301 if (mh.num_substreams == 0)
302 return AVERROR_INVALIDDATA;
303 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
304 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
305 return AVERROR_INVALIDDATA;
307 if (mh.num_substreams > MAX_SUBSTREAMS) {
308 av_log(m->avctx, AV_LOG_ERROR,
309 "Number of substreams %d is larger than the maximum supported "
310 "by the decoder. %s\n", mh.num_substreams, sample_message);
311 return AVERROR_INVALIDDATA;
314 m->access_unit_size = mh.access_unit_size;
315 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
317 m->num_substreams = mh.num_substreams;
318 m->max_decoded_substream = m->num_substreams - 1;
320 m->avctx->sample_rate = mh.group1_samplerate;
321 m->avctx->frame_size = mh.access_unit_size;
323 m->avctx->bits_per_raw_sample = mh.group1_bits;
324 if (mh.group1_bits > 16)
325 m->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
327 m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
330 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
331 m->substream[substr].restart_seen = 0;
336 /** Read a restart header from a block in a substream. This contains parameters
337 * required to decode the audio that do not change very often. Generally
338 * (always) present only in blocks following a major sync. */
340 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
341 const uint8_t *buf, unsigned int substr)
343 SubStream *s = &m->substream[substr];
347 uint8_t lossless_check;
348 int start_count = get_bits_count(gbp);
349 const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
350 ? MAX_MATRIX_CHANNEL_MLP
351 : MAX_MATRIX_CHANNEL_TRUEHD;
353 sync_word = get_bits(gbp, 13);
355 if (sync_word != 0x31ea >> 1) {
356 av_log(m->avctx, AV_LOG_ERROR,
357 "restart header sync incorrect (got 0x%04x)\n", sync_word);
358 return AVERROR_INVALIDDATA;
361 s->noise_type = get_bits1(gbp);
363 if (m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) {
364 av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
365 return AVERROR_INVALIDDATA;
368 skip_bits(gbp, 16); /* Output timestamp */
370 s->min_channel = get_bits(gbp, 4);
371 s->max_channel = get_bits(gbp, 4);
372 s->max_matrix_channel = get_bits(gbp, 4);
374 if (s->max_matrix_channel > max_matrix_channel) {
375 av_log(m->avctx, AV_LOG_ERROR,
376 "Max matrix channel cannot be greater than %d.\n",
378 return AVERROR_INVALIDDATA;
381 if (s->max_channel != s->max_matrix_channel) {
382 av_log(m->avctx, AV_LOG_ERROR,
383 "Max channel must be equal max matrix channel.\n");
384 return AVERROR_INVALIDDATA;
387 /* This should happen for TrueHD streams with >6 channels and MLP's noise
388 * type. It is not yet known if this is allowed. */
389 if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
390 av_log(m->avctx, AV_LOG_ERROR,
391 "Number of channels %d is larger than the maximum supported "
392 "by the decoder. %s\n", s->max_channel+2, sample_message);
393 return AVERROR_INVALIDDATA;
396 if (s->min_channel > s->max_channel) {
397 av_log(m->avctx, AV_LOG_ERROR,
398 "Substream min channel cannot be greater than max channel.\n");
399 return AVERROR_INVALIDDATA;
402 if (m->avctx->request_channels > 0
403 && s->max_channel + 1 >= m->avctx->request_channels
404 && substr < m->max_decoded_substream) {
405 av_log(m->avctx, AV_LOG_DEBUG,
406 "Extracting %d channel downmix from substream %d. "
407 "Further substreams will be skipped.\n",
408 s->max_channel + 1, substr);
409 m->max_decoded_substream = substr;
412 s->noise_shift = get_bits(gbp, 4);
413 s->noisegen_seed = get_bits(gbp, 23);
417 s->data_check_present = get_bits1(gbp);
418 lossless_check = get_bits(gbp, 8);
419 if (substr == m->max_decoded_substream
420 && s->lossless_check_data != 0xffffffff) {
421 tmp = xor_32_to_8(s->lossless_check_data);
422 if (tmp != lossless_check)
423 av_log(m->avctx, AV_LOG_WARNING,
424 "Lossless check failed - expected %02x, calculated %02x.\n",
425 lossless_check, tmp);
430 memset(s->ch_assign, 0, sizeof(s->ch_assign));
432 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
433 int ch_assign = get_bits(gbp, 6);
434 if (ch_assign > s->max_matrix_channel) {
435 av_log(m->avctx, AV_LOG_ERROR,
436 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
437 ch, ch_assign, sample_message);
438 return AVERROR_INVALIDDATA;
440 s->ch_assign[ch_assign] = ch;
443 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
445 if (checksum != get_bits(gbp, 8))
446 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
448 /* Set default decoding parameters. */
449 s->param_presence_flags = 0xff;
450 s->num_primitive_matrices = 0;
452 s->lossless_check_data = 0;
454 memset(s->output_shift , 0, sizeof(s->output_shift ));
455 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
457 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
458 ChannelParams *cp = &s->channel_params[ch];
459 cp->filter_params[FIR].order = 0;
460 cp->filter_params[IIR].order = 0;
461 cp->filter_params[FIR].shift = 0;
462 cp->filter_params[IIR].shift = 0;
464 /* Default audio coding is 24-bit raw PCM. */
466 cp->sign_huff_offset = (-1) << 23;
471 if (substr == m->max_decoded_substream)
472 m->avctx->channels = s->max_matrix_channel + 1;
477 /** Read parameters for one of the prediction filters. */
479 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
480 unsigned int substr, unsigned int channel,
483 SubStream *s = &m->substream[substr];
484 FilterParams *fp = &s->channel_params[channel].filter_params[filter];
485 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
486 const char fchar = filter ? 'I' : 'F';
489 // Filter is 0 for FIR, 1 for IIR.
492 if (m->filter_changed[channel][filter]++ > 1) {
493 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
494 return AVERROR_INVALIDDATA;
497 order = get_bits(gbp, 4);
498 if (order > max_order) {
499 av_log(m->avctx, AV_LOG_ERROR,
500 "%cIR filter order %d is greater than maximum %d.\n",
501 fchar, order, max_order);
502 return AVERROR_INVALIDDATA;
507 int32_t *fcoeff = s->channel_params[channel].coeff[filter];
508 int coeff_bits, coeff_shift;
510 fp->shift = get_bits(gbp, 4);
512 coeff_bits = get_bits(gbp, 5);
513 coeff_shift = get_bits(gbp, 3);
514 if (coeff_bits < 1 || coeff_bits > 16) {
515 av_log(m->avctx, AV_LOG_ERROR,
516 "%cIR filter coeff_bits must be between 1 and 16.\n",
518 return AVERROR_INVALIDDATA;
520 if (coeff_bits + coeff_shift > 16) {
521 av_log(m->avctx, AV_LOG_ERROR,
522 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
524 return AVERROR_INVALIDDATA;
527 for (i = 0; i < order; i++)
528 fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
530 if (get_bits1(gbp)) {
531 int state_bits, state_shift;
534 av_log(m->avctx, AV_LOG_ERROR,
535 "FIR filter has state data specified.\n");
536 return AVERROR_INVALIDDATA;
539 state_bits = get_bits(gbp, 4);
540 state_shift = get_bits(gbp, 4);
542 /* TODO: Check validity of state data. */
544 for (i = 0; i < order; i++)
545 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
552 /** Read parameters for primitive matrices. */
554 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
556 SubStream *s = &m->substream[substr];
557 unsigned int mat, ch;
558 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
560 : MAX_MATRICES_TRUEHD;
562 if (m->matrix_changed++ > 1) {
563 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
564 return AVERROR_INVALIDDATA;
567 s->num_primitive_matrices = get_bits(gbp, 4);
569 if (s->num_primitive_matrices > max_primitive_matrices) {
570 av_log(m->avctx, AV_LOG_ERROR,
571 "Number of primitive matrices cannot be greater than %d.\n",
572 max_primitive_matrices);
573 return AVERROR_INVALIDDATA;
576 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
577 int frac_bits, max_chan;
578 s->matrix_out_ch[mat] = get_bits(gbp, 4);
579 frac_bits = get_bits(gbp, 4);
580 s->lsb_bypass [mat] = get_bits1(gbp);
582 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
583 av_log(m->avctx, AV_LOG_ERROR,
584 "Invalid channel %d specified as output from matrix.\n",
585 s->matrix_out_ch[mat]);
586 return AVERROR_INVALIDDATA;
588 if (frac_bits > 14) {
589 av_log(m->avctx, AV_LOG_ERROR,
590 "Too many fractional bits specified.\n");
591 return AVERROR_INVALIDDATA;
594 max_chan = s->max_matrix_channel;
598 for (ch = 0; ch <= max_chan; ch++) {
601 coeff_val = get_sbits(gbp, frac_bits + 2);
603 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
607 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
609 s->matrix_noise_shift[mat] = 0;
615 /** Read channel parameters. */
617 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
618 GetBitContext *gbp, unsigned int ch)
620 SubStream *s = &m->substream[substr];
621 ChannelParams *cp = &s->channel_params[ch];
622 FilterParams *fir = &cp->filter_params[FIR];
623 FilterParams *iir = &cp->filter_params[IIR];
626 if (s->param_presence_flags & PARAM_FIR)
628 if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
631 if (s->param_presence_flags & PARAM_IIR)
633 if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
636 if (fir->order + iir->order > 8) {
637 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
638 return AVERROR_INVALIDDATA;
641 if (fir->order && iir->order &&
642 fir->shift != iir->shift) {
643 av_log(m->avctx, AV_LOG_ERROR,
644 "FIR and IIR filters must use the same precision.\n");
645 return AVERROR_INVALIDDATA;
647 /* The FIR and IIR filters must have the same precision.
648 * To simplify the filtering code, only the precision of the
649 * FIR filter is considered. If only the IIR filter is employed,
650 * the FIR filter precision is set to that of the IIR filter, so
651 * that the filtering code can use it. */
652 if (!fir->order && iir->order)
653 fir->shift = iir->shift;
655 if (s->param_presence_flags & PARAM_HUFFOFFSET)
657 cp->huff_offset = get_sbits(gbp, 15);
659 cp->codebook = get_bits(gbp, 2);
660 cp->huff_lsbs = get_bits(gbp, 5);
662 if (cp->huff_lsbs > 24) {
663 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
664 return AVERROR_INVALIDDATA;
667 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
672 /** Read decoding parameters that change more often than those in the restart
675 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
678 SubStream *s = &m->substream[substr];
682 if (s->param_presence_flags & PARAM_PRESENCE)
684 s->param_presence_flags = get_bits(gbp, 8);
686 if (s->param_presence_flags & PARAM_BLOCKSIZE)
687 if (get_bits1(gbp)) {
688 s->blocksize = get_bits(gbp, 9);
689 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
690 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
692 return AVERROR_INVALIDDATA;
696 if (s->param_presence_flags & PARAM_MATRIX)
698 if ((ret = read_matrix_params(m, substr, gbp)) < 0)
701 if (s->param_presence_flags & PARAM_OUTSHIFT)
703 for (ch = 0; ch <= s->max_matrix_channel; ch++)
704 s->output_shift[ch] = get_sbits(gbp, 4);
706 if (s->param_presence_flags & PARAM_QUANTSTEP)
708 for (ch = 0; ch <= s->max_channel; ch++) {
709 ChannelParams *cp = &s->channel_params[ch];
711 s->quant_step_size[ch] = get_bits(gbp, 4);
713 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
716 for (ch = s->min_channel; ch <= s->max_channel; ch++)
718 if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
724 #define MSB_MASK(bits) (-1u << bits)
726 /** Generate PCM samples using the prediction filters and residual values
727 * read from the data stream, and update the filter state. */
729 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
730 unsigned int channel)
732 SubStream *s = &m->substream[substr];
733 const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
734 int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
735 int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
736 int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
737 FilterParams *fir = &s->channel_params[channel].filter_params[FIR];
738 FilterParams *iir = &s->channel_params[channel].filter_params[IIR];
739 unsigned int filter_shift = fir->shift;
740 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
742 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
743 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
745 m->dsp.mlp_filter_channel(firbuf, fircoeff,
746 fir->order, iir->order,
747 filter_shift, mask, s->blocksize,
748 &m->sample_buffer[s->blockpos][channel]);
750 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
751 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
754 /** Read a block of PCM residual data (or actual if no filtering active). */
756 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
759 SubStream *s = &m->substream[substr];
760 unsigned int i, ch, expected_stream_pos = 0;
763 if (s->data_check_present) {
764 expected_stream_pos = get_bits_count(gbp);
765 expected_stream_pos += get_bits(gbp, 16);
766 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
767 "we have not tested yet. %s\n", sample_message);
770 if (s->blockpos + s->blocksize > m->access_unit_size) {
771 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
772 return AVERROR_INVALIDDATA;
775 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
776 s->blocksize * sizeof(m->bypassed_lsbs[0]));
778 for (i = 0; i < s->blocksize; i++)
779 if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
782 for (ch = s->min_channel; ch <= s->max_channel; ch++)
783 filter_channel(m, substr, ch);
785 s->blockpos += s->blocksize;
787 if (s->data_check_present) {
788 if (get_bits_count(gbp) != expected_stream_pos)
789 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
796 /** Data table used for TrueHD noise generation function. */
798 static const int8_t noise_table[256] = {
799 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
800 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
801 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
802 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
803 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
804 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
805 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
806 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
807 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
808 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
809 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
810 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
811 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
812 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
813 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
814 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
817 /** Noise generation functions.
818 * I'm not sure what these are for - they seem to be some kind of pseudorandom
819 * sequence generators, used to generate noise data which is used when the
820 * channels are rematrixed. I'm not sure if they provide a practical benefit
821 * to compression, or just obfuscate the decoder. Are they for some kind of
824 /** Generate two channels of noise, used in the matrix when
825 * restart sync word == 0x31ea. */
827 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
829 SubStream *s = &m->substream[substr];
831 uint32_t seed = s->noisegen_seed;
832 unsigned int maxchan = s->max_matrix_channel;
834 for (i = 0; i < s->blockpos; i++) {
835 uint16_t seed_shr7 = seed >> 7;
836 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
837 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
839 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
842 s->noisegen_seed = seed;
845 /** Generate a block of noise, used when restart sync word == 0x31eb. */
847 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
849 SubStream *s = &m->substream[substr];
851 uint32_t seed = s->noisegen_seed;
853 for (i = 0; i < m->access_unit_size_pow2; i++) {
854 uint8_t seed_shr15 = seed >> 15;
855 m->noise_buffer[i] = noise_table[seed_shr15];
856 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
859 s->noisegen_seed = seed;
863 /** Apply the channel matrices in turn to reconstruct the original audio
866 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
868 SubStream *s = &m->substream[substr];
869 unsigned int mat, src_ch, i;
870 unsigned int maxchan;
872 maxchan = s->max_matrix_channel;
873 if (!s->noise_type) {
874 generate_2_noise_channels(m, substr);
877 fill_noise_buffer(m, substr);
880 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
881 int matrix_noise_shift = s->matrix_noise_shift[mat];
882 unsigned int dest_ch = s->matrix_out_ch[mat];
883 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
884 int32_t *coeffs = s->matrix_coeff[mat];
885 int index = s->num_primitive_matrices - mat;
886 int index2 = 2 * index + 1;
888 /* TODO: DSPContext? */
890 for (i = 0; i < s->blockpos; i++) {
891 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
892 int32_t *samples = m->sample_buffer[i];
895 for (src_ch = 0; src_ch <= maxchan; src_ch++)
896 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
898 if (matrix_noise_shift) {
899 index &= m->access_unit_size_pow2 - 1;
900 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
904 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
909 /** Write the audio data into the output buffer. */
911 static int output_data(MLPDecodeContext *m, unsigned int substr,
912 void *data, int *got_frame_ptr)
914 AVCodecContext *avctx = m->avctx;
915 SubStream *s = &m->substream[substr];
916 unsigned int i, out_ch = 0;
920 int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
922 if (m->avctx->channels != s->max_matrix_channel + 1) {
923 av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
924 return AVERROR_INVALIDDATA;
927 /* get output buffer */
928 m->frame.nb_samples = s->blockpos;
929 if ((ret = avctx->get_buffer(avctx, &m->frame)) < 0) {
930 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
933 data_32 = (int32_t *)m->frame.data[0];
934 data_16 = (int16_t *)m->frame.data[0];
936 for (i = 0; i < s->blockpos; i++) {
937 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
938 int mat_ch = s->ch_assign[out_ch];
939 int32_t sample = m->sample_buffer[i][mat_ch]
940 << s->output_shift[mat_ch];
941 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
942 if (is32) *data_32++ = sample << 8;
943 else *data_16++ = sample >> 8;
948 *(AVFrame *)data = m->frame;
953 /** Read an access unit from the stream.
954 * @return negative on error, 0 if not enough data is present in the input stream,
955 * otherwise the number of bytes consumed. */
957 static int read_access_unit(AVCodecContext *avctx, void* data,
958 int *got_frame_ptr, AVPacket *avpkt)
960 const uint8_t *buf = avpkt->data;
961 int buf_size = avpkt->size;
962 MLPDecodeContext *m = avctx->priv_data;
964 unsigned int length, substr;
965 unsigned int substream_start;
966 unsigned int header_size = 4;
967 unsigned int substr_header_size = 0;
968 uint8_t substream_parity_present[MAX_SUBSTREAMS];
969 uint16_t substream_data_len[MAX_SUBSTREAMS];
976 length = (AV_RB16(buf) & 0xfff) * 2;
978 if (length < 4 || length > buf_size)
979 return AVERROR_INVALIDDATA;
981 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
983 m->is_major_sync_unit = 0;
984 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
985 if (read_major_sync(m, &gb) < 0)
987 m->is_major_sync_unit = 1;
991 if (!m->params_valid) {
992 av_log(m->avctx, AV_LOG_WARNING,
993 "Stream parameters not seen; skipping frame.\n");
1000 for (substr = 0; substr < m->num_substreams; substr++) {
1001 int extraword_present, checkdata_present, end, nonrestart_substr;
1003 extraword_present = get_bits1(&gb);
1004 nonrestart_substr = get_bits1(&gb);
1005 checkdata_present = get_bits1(&gb);
1008 end = get_bits(&gb, 12) * 2;
1010 substr_header_size += 2;
1012 if (extraword_present) {
1013 if (m->avctx->codec_id == CODEC_ID_MLP) {
1014 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
1018 substr_header_size += 2;
1021 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1022 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1026 if (end + header_size + substr_header_size > length) {
1027 av_log(m->avctx, AV_LOG_ERROR,
1028 "Indicated length of substream %d data goes off end of "
1029 "packet.\n", substr);
1030 end = length - header_size - substr_header_size;
1033 if (end < substream_start) {
1034 av_log(avctx, AV_LOG_ERROR,
1035 "Indicated end offset of substream %d data "
1036 "is smaller than calculated start offset.\n",
1041 if (substr > m->max_decoded_substream)
1044 substream_parity_present[substr] = checkdata_present;
1045 substream_data_len[substr] = end - substream_start;
1046 substream_start = end;
1049 parity_bits = ff_mlp_calculate_parity(buf, 4);
1050 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1052 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1053 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1057 buf += header_size + substr_header_size;
1059 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1060 SubStream *s = &m->substream[substr];
1061 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1063 m->matrix_changed = 0;
1064 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1068 if (get_bits1(&gb)) {
1069 if (get_bits1(&gb)) {
1070 /* A restart header should be present. */
1071 if (read_restart_header(m, &gb, buf, substr) < 0)
1073 s->restart_seen = 1;
1076 if (!s->restart_seen)
1078 if (read_decoding_params(m, &gb, substr) < 0)
1082 if (!s->restart_seen)
1085 if ((ret = read_block_data(m, &gb, substr)) < 0)
1088 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1089 goto substream_length_mismatch;
1091 } while (!get_bits1(&gb));
1093 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1095 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1098 if (get_bits(&gb, 16) != 0xD234)
1099 return AVERROR_INVALIDDATA;
1101 shorten_by = get_bits(&gb, 16);
1102 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1103 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1104 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1105 return AVERROR_INVALIDDATA;
1107 if (substr == m->max_decoded_substream)
1108 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1111 if (substream_parity_present[substr]) {
1112 uint8_t parity, checksum;
1114 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1115 goto substream_length_mismatch;
1117 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1118 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1120 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1121 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1122 if ( get_bits(&gb, 8) != checksum)
1123 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1126 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1127 goto substream_length_mismatch;
1130 if (!s->restart_seen)
1131 av_log(m->avctx, AV_LOG_ERROR,
1132 "No restart header present in substream %d.\n", substr);
1134 buf += substream_data_len[substr];
1137 rematrix_channels(m, m->max_decoded_substream);
1139 if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
1144 substream_length_mismatch:
1145 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1146 return AVERROR_INVALIDDATA;
1149 m->params_valid = 0;
1150 return AVERROR_INVALIDDATA;
1153 AVCodec ff_mlp_decoder = {
1155 .type = AVMEDIA_TYPE_AUDIO,
1157 .priv_data_size = sizeof(MLPDecodeContext),
1158 .init = mlp_decode_init,
1159 .decode = read_access_unit,
1160 .capabilities = CODEC_CAP_DR1,
1161 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1164 #if CONFIG_TRUEHD_DECODER
1165 AVCodec ff_truehd_decoder = {
1167 .type = AVMEDIA_TYPE_AUDIO,
1168 .id = CODEC_ID_TRUEHD,
1169 .priv_data_size = sizeof(MLPDecodeContext),
1170 .init = mlp_decode_init,
1171 .decode = read_access_unit,
1172 .capabilities = CODEC_CAP_DR1,
1173 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1175 #endif /* CONFIG_TRUEHD_DECODER */