3 * Copyright (c) 2007-2008 Ian Caulfield
5 * This file is part of FFmpeg.
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.
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.
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
23 * @file libavcodec/mlpdec.c
30 #include "libavutil/intreadwrite.h"
31 #include "bitstream.h"
32 #include "libavutil/crc.h"
34 #include "mlp_parser.h"
37 /** number of bits used for VLC lookup - longest Huffman code is 9 */
41 static const char* sample_message =
42 "Please file a bug report following the instructions at "
43 "http://ffmpeg.org/bugreports.html and include "
44 "a sample of this file.";
46 typedef struct SubStream {
47 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
51 /** restart header data */
52 //! The type of noise to be used in the rematrix stage.
55 //! The index of the first channel coded in this substream.
57 //! The index of the last channel coded in this substream.
59 //! The number of channels input into the rematrix stage.
60 uint8_t max_matrix_channel;
61 //! For each channel output by the matrix, the output channel to map it to
62 uint8_t ch_assign[MAX_CHANNELS];
64 //! The left shift applied to random noise in 0x31ea substreams.
66 //! The current seed value for the pseudorandom noise generator(s).
67 uint32_t noisegen_seed;
69 //! Set if the substream contains extra info to check the size of VLC blocks.
70 uint8_t data_check_present;
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)
81 #define PARAM_PRESENCE (1 << 0)
87 //! Number of matrices to be applied.
88 uint8_t num_primitive_matrices;
90 //! matrix output channel
91 uint8_t matrix_out_ch[MAX_MATRICES];
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];
101 //! Left shift to apply to Huffman-decoded residuals.
102 uint8_t quant_step_size[MAX_CHANNELS];
104 //! number of PCM samples in current audio block
106 //! Number of PCM samples decoded so far in this frame.
109 //! Left shift to apply to decoded PCM values to get final 24-bit output.
110 int8_t output_shift[MAX_CHANNELS];
112 //! Running XOR of all output samples.
113 int32_t lossless_check_data;
117 typedef struct MLPDecodeContext {
118 AVCodecContext *avctx;
120 //! Current access unit being read has a major sync.
121 int is_major_sync_unit;
123 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
124 uint8_t params_valid;
126 //! Number of substreams contained within this stream.
127 uint8_t num_substreams;
129 //! Index of the last substream to decode - further substreams are skipped.
130 uint8_t max_decoded_substream;
132 //! number of PCM samples contained in each frame
133 int access_unit_size;
134 //! next power of two above the number of samples in each frame
135 int access_unit_size_pow2;
137 SubStream substream[MAX_SUBSTREAMS];
139 ChannelParams channel_params[MAX_CHANNELS];
141 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
142 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
143 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
146 static VLC huff_vlc[3];
148 /** Initialize static data, constant between all invocations of the codec. */
150 static av_cold void init_static(void)
152 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
153 &ff_mlp_huffman_tables[0][0][1], 2, 1,
154 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
155 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
156 &ff_mlp_huffman_tables[1][0][1], 2, 1,
157 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
158 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
159 &ff_mlp_huffman_tables[2][0][1], 2, 1,
160 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
165 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
166 unsigned int substr, unsigned int ch)
168 ChannelParams *cp = &m->channel_params[ch];
169 SubStream *s = &m->substream[substr];
170 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
171 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
172 int32_t sign_huff_offset = cp->huff_offset;
174 if (cp->codebook > 0)
175 sign_huff_offset -= 7 << lsb_bits;
178 sign_huff_offset -= 1 << sign_shift;
180 return sign_huff_offset;
183 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
186 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
187 unsigned int substr, unsigned int pos)
189 SubStream *s = &m->substream[substr];
190 unsigned int mat, channel;
192 for (mat = 0; mat < s->num_primitive_matrices; mat++)
193 if (s->lsb_bypass[mat])
194 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
196 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
197 ChannelParams *cp = &m->channel_params[channel];
198 int codebook = cp->codebook;
199 int quant_step_size = s->quant_step_size[channel];
200 int lsb_bits = cp->huff_lsbs - quant_step_size;
204 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
205 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
211 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
213 result += cp->sign_huff_offset;
214 result <<= quant_step_size;
216 m->sample_buffer[pos + s->blockpos][channel] = result;
222 static av_cold int mlp_decode_init(AVCodecContext *avctx)
224 MLPDecodeContext *m = avctx->priv_data;
229 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
230 m->substream[substr].lossless_check_data = 0xffffffff;
235 /** Read a major sync info header - contains high level information about
236 * the stream - sample rate, channel arrangement etc. Most of this
237 * information is not actually necessary for decoding, only for playback.
240 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
245 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
248 if (mh.group1_bits == 0) {
249 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
252 if (mh.group2_bits > mh.group1_bits) {
253 av_log(m->avctx, AV_LOG_ERROR,
254 "Channel group 2 cannot have more bits per sample than group 1.\n");
258 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
259 av_log(m->avctx, AV_LOG_ERROR,
260 "Channel groups with differing sample rates are not currently supported.\n");
264 if (mh.group1_samplerate == 0) {
265 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
268 if (mh.group1_samplerate > MAX_SAMPLERATE) {
269 av_log(m->avctx, AV_LOG_ERROR,
270 "Sampling rate %d is greater than the supported maximum (%d).\n",
271 mh.group1_samplerate, MAX_SAMPLERATE);
274 if (mh.access_unit_size > MAX_BLOCKSIZE) {
275 av_log(m->avctx, AV_LOG_ERROR,
276 "Block size %d is greater than the supported maximum (%d).\n",
277 mh.access_unit_size, MAX_BLOCKSIZE);
280 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
281 av_log(m->avctx, AV_LOG_ERROR,
282 "Block size pow2 %d is greater than the supported maximum (%d).\n",
283 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
287 if (mh.num_substreams == 0)
289 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
290 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
293 if (mh.num_substreams > MAX_SUBSTREAMS) {
294 av_log(m->avctx, AV_LOG_ERROR,
295 "Number of substreams %d is larger than the maximum supported "
296 "by the decoder. %s\n", mh.num_substreams, sample_message);
300 m->access_unit_size = mh.access_unit_size;
301 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
303 m->num_substreams = mh.num_substreams;
304 m->max_decoded_substream = m->num_substreams - 1;
306 m->avctx->sample_rate = mh.group1_samplerate;
307 m->avctx->frame_size = mh.access_unit_size;
309 m->avctx->bits_per_raw_sample = mh.group1_bits;
310 if (mh.group1_bits > 16)
311 m->avctx->sample_fmt = SAMPLE_FMT_S32;
313 m->avctx->sample_fmt = SAMPLE_FMT_S16;
316 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
317 m->substream[substr].restart_seen = 0;
322 /** Read a restart header from a block in a substream. This contains parameters
323 * required to decode the audio that do not change very often. Generally
324 * (always) present only in blocks following a major sync. */
326 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
327 const uint8_t *buf, unsigned int substr)
329 SubStream *s = &m->substream[substr];
333 uint8_t lossless_check;
334 int start_count = get_bits_count(gbp);
336 sync_word = get_bits(gbp, 13);
338 if (sync_word != 0x31ea >> 1) {
339 av_log(m->avctx, AV_LOG_ERROR,
340 "restart header sync incorrect (got 0x%04x)\n", sync_word);
343 s->noise_type = get_bits1(gbp);
345 skip_bits(gbp, 16); /* Output timestamp */
347 s->min_channel = get_bits(gbp, 4);
348 s->max_channel = get_bits(gbp, 4);
349 s->max_matrix_channel = get_bits(gbp, 4);
351 if (s->min_channel > s->max_channel) {
352 av_log(m->avctx, AV_LOG_ERROR,
353 "Substream min channel cannot be greater than max channel.\n");
357 if (m->avctx->request_channels > 0
358 && s->max_channel + 1 >= m->avctx->request_channels
359 && substr < m->max_decoded_substream) {
360 av_log(m->avctx, AV_LOG_INFO,
361 "Extracting %d channel downmix from substream %d. "
362 "Further substreams will be skipped.\n",
363 s->max_channel + 1, substr);
364 m->max_decoded_substream = substr;
367 s->noise_shift = get_bits(gbp, 4);
368 s->noisegen_seed = get_bits(gbp, 23);
372 s->data_check_present = get_bits1(gbp);
373 lossless_check = get_bits(gbp, 8);
374 if (substr == m->max_decoded_substream
375 && s->lossless_check_data != 0xffffffff) {
376 tmp = xor_32_to_8(s->lossless_check_data);
377 if (tmp != lossless_check)
378 av_log(m->avctx, AV_LOG_WARNING,
379 "Lossless check failed - expected %02x, calculated %02x.\n",
380 lossless_check, tmp);
385 memset(s->ch_assign, 0, sizeof(s->ch_assign));
387 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
388 int ch_assign = get_bits(gbp, 6);
389 if (ch_assign > s->max_matrix_channel) {
390 av_log(m->avctx, AV_LOG_ERROR,
391 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
392 ch, ch_assign, sample_message);
395 s->ch_assign[ch_assign] = ch;
398 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
400 if (checksum != get_bits(gbp, 8))
401 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
403 /* Set default decoding parameters. */
404 s->param_presence_flags = 0xff;
405 s->num_primitive_matrices = 0;
407 s->lossless_check_data = 0;
409 memset(s->output_shift , 0, sizeof(s->output_shift ));
410 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
412 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
413 ChannelParams *cp = &m->channel_params[ch];
414 cp->filter_params[FIR].order = 0;
415 cp->filter_params[IIR].order = 0;
416 cp->filter_params[FIR].shift = 0;
417 cp->filter_params[IIR].shift = 0;
419 /* Default audio coding is 24-bit raw PCM. */
421 cp->sign_huff_offset = (-1) << 23;
426 if (substr == m->max_decoded_substream) {
427 m->avctx->channels = s->max_matrix_channel + 1;
433 /** Read parameters for one of the prediction filters. */
435 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
436 unsigned int channel, unsigned int filter)
438 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
439 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
440 const char fchar = filter ? 'I' : 'F';
443 // Filter is 0 for FIR, 1 for IIR.
446 order = get_bits(gbp, 4);
447 if (order > max_order) {
448 av_log(m->avctx, AV_LOG_ERROR,
449 "%cIR filter order %d is greater than maximum %d.\n",
450 fchar, order, max_order);
456 int coeff_bits, coeff_shift;
458 fp->shift = get_bits(gbp, 4);
460 coeff_bits = get_bits(gbp, 5);
461 coeff_shift = get_bits(gbp, 3);
462 if (coeff_bits < 1 || coeff_bits > 16) {
463 av_log(m->avctx, AV_LOG_ERROR,
464 "%cIR filter coeff_bits must be between 1 and 16.\n",
468 if (coeff_bits + coeff_shift > 16) {
469 av_log(m->avctx, AV_LOG_ERROR,
470 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
475 for (i = 0; i < order; i++)
476 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
478 if (get_bits1(gbp)) {
479 int state_bits, state_shift;
482 av_log(m->avctx, AV_LOG_ERROR,
483 "FIR filter has state data specified.\n");
487 state_bits = get_bits(gbp, 4);
488 state_shift = get_bits(gbp, 4);
490 /* TODO: Check validity of state data. */
492 for (i = 0; i < order; i++)
493 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
500 /** Read parameters for primitive matrices. */
502 static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
504 unsigned int mat, ch;
506 s->num_primitive_matrices = get_bits(gbp, 4);
508 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
509 int frac_bits, max_chan;
510 s->matrix_out_ch[mat] = get_bits(gbp, 4);
511 frac_bits = get_bits(gbp, 4);
512 s->lsb_bypass [mat] = get_bits1(gbp);
514 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
515 av_log(m->avctx, AV_LOG_ERROR,
516 "Invalid channel %d specified as output from matrix.\n",
517 s->matrix_out_ch[mat]);
520 if (frac_bits > 14) {
521 av_log(m->avctx, AV_LOG_ERROR,
522 "Too many fractional bits specified.\n");
526 max_chan = s->max_matrix_channel;
530 for (ch = 0; ch <= max_chan; ch++) {
533 coeff_val = get_sbits(gbp, frac_bits + 2);
535 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
539 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
541 s->matrix_noise_shift[mat] = 0;
547 /** Read channel parameters. */
549 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
550 GetBitContext *gbp, unsigned int ch)
552 ChannelParams *cp = &m->channel_params[ch];
553 FilterParams *fir = &cp->filter_params[FIR];
554 FilterParams *iir = &cp->filter_params[IIR];
555 SubStream *s = &m->substream[substr];
557 if (s->param_presence_flags & PARAM_FIR)
559 if (read_filter_params(m, gbp, ch, FIR) < 0)
562 if (s->param_presence_flags & PARAM_IIR)
564 if (read_filter_params(m, gbp, ch, IIR) < 0)
567 if (fir->order + iir->order > 8) {
568 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
572 if (fir->order && iir->order &&
573 fir->shift != iir->shift) {
574 av_log(m->avctx, AV_LOG_ERROR,
575 "FIR and IIR filters must use the same precision.\n");
578 /* The FIR and IIR filters must have the same precision.
579 * To simplify the filtering code, only the precision of the
580 * FIR filter is considered. If only the IIR filter is employed,
581 * the FIR filter precision is set to that of the IIR filter, so
582 * that the filtering code can use it. */
583 if (!fir->order && iir->order)
584 fir->shift = iir->shift;
586 if (s->param_presence_flags & PARAM_HUFFOFFSET)
588 cp->huff_offset = get_sbits(gbp, 15);
590 cp->codebook = get_bits(gbp, 2);
591 cp->huff_lsbs = get_bits(gbp, 5);
593 if (cp->huff_lsbs > 24) {
594 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
598 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
603 /** Read decoding parameters that change more often than those in the restart
606 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
609 SubStream *s = &m->substream[substr];
612 if (s->param_presence_flags & PARAM_PRESENCE)
614 s->param_presence_flags = get_bits(gbp, 8);
616 if (s->param_presence_flags & PARAM_BLOCKSIZE)
617 if (get_bits1(gbp)) {
618 s->blocksize = get_bits(gbp, 9);
619 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
620 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
626 if (s->param_presence_flags & PARAM_MATRIX)
627 if (get_bits1(gbp)) {
628 if (read_matrix_params(m, s, gbp) < 0)
632 if (s->param_presence_flags & PARAM_OUTSHIFT)
634 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
635 s->output_shift[ch] = get_sbits(gbp, 4);
638 if (s->param_presence_flags & PARAM_QUANTSTEP)
640 for (ch = 0; ch <= s->max_channel; ch++) {
641 ChannelParams *cp = &m->channel_params[ch];
643 s->quant_step_size[ch] = get_bits(gbp, 4);
645 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
648 for (ch = s->min_channel; ch <= s->max_channel; ch++)
649 if (get_bits1(gbp)) {
650 if (read_channel_params(m, substr, gbp, ch) < 0)
657 #define MSB_MASK(bits) (-1u << bits)
659 /** Generate PCM samples using the prediction filters and residual values
660 * read from the data stream, and update the filter state. */
662 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
663 unsigned int channel)
665 SubStream *s = &m->substream[substr];
666 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
667 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
668 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
669 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
670 unsigned int filter_shift = fir->shift;
671 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
672 int index = MAX_BLOCKSIZE;
675 memcpy(&firbuf[index], fir->state, MAX_FIR_ORDER * sizeof(int32_t));
676 memcpy(&iirbuf[index], iir->state, MAX_IIR_ORDER * sizeof(int32_t));
678 for (i = 0; i < s->blocksize; i++) {
679 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
684 /* TODO: Move this code to DSPContext? */
686 for (order = 0; order < fir->order; order++)
687 accum += (int64_t) firbuf[index + order] * fir->coeff[order];
688 for (order = 0; order < iir->order; order++)
689 accum += (int64_t) iirbuf[index + order] * iir->coeff[order];
691 accum = accum >> filter_shift;
692 result = (accum + residual) & mask;
696 firbuf[index] = result;
697 iirbuf[index] = result - accum;
699 m->sample_buffer[i + s->blockpos][channel] = result;
702 memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
703 memcpy(iir->state, &iirbuf[index], MAX_IIR_ORDER * sizeof(int32_t));
706 /** Read a block of PCM residual data (or actual if no filtering active). */
708 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
711 SubStream *s = &m->substream[substr];
712 unsigned int i, ch, expected_stream_pos = 0;
714 if (s->data_check_present) {
715 expected_stream_pos = get_bits_count(gbp);
716 expected_stream_pos += get_bits(gbp, 16);
717 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
718 "we have not tested yet. %s\n", sample_message);
721 if (s->blockpos + s->blocksize > m->access_unit_size) {
722 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
726 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
727 s->blocksize * sizeof(m->bypassed_lsbs[0]));
729 for (i = 0; i < s->blocksize; i++) {
730 if (read_huff_channels(m, gbp, substr, i) < 0)
734 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
735 filter_channel(m, substr, ch);
738 s->blockpos += s->blocksize;
740 if (s->data_check_present) {
741 if (get_bits_count(gbp) != expected_stream_pos)
742 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
749 /** Data table used for TrueHD noise generation function. */
751 static const int8_t noise_table[256] = {
752 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
753 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
754 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
755 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
756 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
757 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
758 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
759 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
760 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
761 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
762 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
763 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
764 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
765 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
766 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
767 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
770 /** Noise generation functions.
771 * I'm not sure what these are for - they seem to be some kind of pseudorandom
772 * sequence generators, used to generate noise data which is used when the
773 * channels are rematrixed. I'm not sure if they provide a practical benefit
774 * to compression, or just obfuscate the decoder. Are they for some kind of
777 /** Generate two channels of noise, used in the matrix when
778 * restart sync word == 0x31ea. */
780 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
782 SubStream *s = &m->substream[substr];
784 uint32_t seed = s->noisegen_seed;
785 unsigned int maxchan = s->max_matrix_channel;
787 for (i = 0; i < s->blockpos; i++) {
788 uint16_t seed_shr7 = seed >> 7;
789 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
790 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
792 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
795 s->noisegen_seed = seed;
798 /** Generate a block of noise, used when restart sync word == 0x31eb. */
800 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
802 SubStream *s = &m->substream[substr];
804 uint32_t seed = s->noisegen_seed;
806 for (i = 0; i < m->access_unit_size_pow2; i++) {
807 uint8_t seed_shr15 = seed >> 15;
808 m->noise_buffer[i] = noise_table[seed_shr15];
809 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
812 s->noisegen_seed = seed;
816 /** Apply the channel matrices in turn to reconstruct the original audio
819 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
821 SubStream *s = &m->substream[substr];
822 unsigned int mat, src_ch, i;
823 unsigned int maxchan;
825 maxchan = s->max_matrix_channel;
826 if (!s->noise_type) {
827 generate_2_noise_channels(m, substr);
830 fill_noise_buffer(m, substr);
833 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
834 int matrix_noise_shift = s->matrix_noise_shift[mat];
835 unsigned int dest_ch = s->matrix_out_ch[mat];
836 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
838 /* TODO: DSPContext? */
840 for (i = 0; i < s->blockpos; i++) {
842 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
843 accum += (int64_t)m->sample_buffer[i][src_ch]
844 * s->matrix_coeff[mat][src_ch];
846 if (matrix_noise_shift) {
847 uint32_t index = s->num_primitive_matrices - mat;
848 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
849 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
851 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
852 + m->bypassed_lsbs[i][mat];
857 /** Write the audio data into the output buffer. */
859 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
860 uint8_t *data, unsigned int *data_size, int is32)
862 SubStream *s = &m->substream[substr];
863 unsigned int i, out_ch = 0;
864 int32_t *data_32 = (int32_t*) data;
865 int16_t *data_16 = (int16_t*) data;
867 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
870 for (i = 0; i < s->blockpos; i++) {
871 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
872 int mat_ch = s->ch_assign[out_ch];
873 int32_t sample = m->sample_buffer[i][mat_ch]
874 << s->output_shift[mat_ch];
875 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
876 if (is32) *data_32++ = sample << 8;
877 else *data_16++ = sample >> 8;
881 *data_size = i * out_ch * (is32 ? 4 : 2);
886 static int output_data(MLPDecodeContext *m, unsigned int substr,
887 uint8_t *data, unsigned int *data_size)
889 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
890 return output_data_internal(m, substr, data, data_size, 1);
892 return output_data_internal(m, substr, data, data_size, 0);
896 /** Read an access unit from the stream.
897 * Returns < 0 on error, 0 if not enough data is present in the input stream
898 * otherwise returns the number of bytes consumed. */
900 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
901 const uint8_t *buf, int buf_size)
903 MLPDecodeContext *m = avctx->priv_data;
905 unsigned int length, substr;
906 unsigned int substream_start;
907 unsigned int header_size = 4;
908 unsigned int substr_header_size = 0;
909 uint8_t substream_parity_present[MAX_SUBSTREAMS];
910 uint16_t substream_data_len[MAX_SUBSTREAMS];
916 length = (AV_RB16(buf) & 0xfff) * 2;
918 if (length > buf_size)
921 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
923 m->is_major_sync_unit = 0;
924 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
925 if (read_major_sync(m, &gb) < 0)
927 m->is_major_sync_unit = 1;
931 if (!m->params_valid) {
932 av_log(m->avctx, AV_LOG_WARNING,
933 "Stream parameters not seen; skipping frame.\n");
940 for (substr = 0; substr < m->num_substreams; substr++) {
941 int extraword_present, checkdata_present, end, nonrestart_substr;
943 extraword_present = get_bits1(&gb);
944 nonrestart_substr = get_bits1(&gb);
945 checkdata_present = get_bits1(&gb);
948 end = get_bits(&gb, 12) * 2;
950 substr_header_size += 2;
952 if (extraword_present) {
954 substr_header_size += 2;
957 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
958 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
962 if (end + header_size + substr_header_size > length) {
963 av_log(m->avctx, AV_LOG_ERROR,
964 "Indicated length of substream %d data goes off end of "
965 "packet.\n", substr);
966 end = length - header_size - substr_header_size;
969 if (end < substream_start) {
970 av_log(avctx, AV_LOG_ERROR,
971 "Indicated end offset of substream %d data "
972 "is smaller than calculated start offset.\n",
977 if (substr > m->max_decoded_substream)
980 substream_parity_present[substr] = checkdata_present;
981 substream_data_len[substr] = end - substream_start;
982 substream_start = end;
985 parity_bits = ff_mlp_calculate_parity(buf, 4);
986 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
988 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
989 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
993 buf += header_size + substr_header_size;
995 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
996 SubStream *s = &m->substream[substr];
997 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1001 if (get_bits1(&gb)) {
1002 if (get_bits1(&gb)) {
1003 /* A restart header should be present. */
1004 if (read_restart_header(m, &gb, buf, substr) < 0)
1006 s->restart_seen = 1;
1009 if (!s->restart_seen) {
1013 if (read_decoding_params(m, &gb, substr) < 0)
1017 if (!s->restart_seen) {
1021 if (read_block_data(m, &gb, substr) < 0)
1024 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1025 goto substream_length_mismatch;
1027 } while (!get_bits1(&gb));
1029 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1030 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1033 if (get_bits(&gb, 16) != 0xD234)
1036 shorten_by = get_bits(&gb, 16);
1037 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1038 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1039 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1042 if (substr == m->max_decoded_substream)
1043 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1045 if (substream_parity_present[substr]) {
1046 uint8_t parity, checksum;
1048 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1049 goto substream_length_mismatch;
1051 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1052 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1054 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1055 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1056 if ( get_bits(&gb, 8) != checksum)
1057 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1059 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1060 goto substream_length_mismatch;
1064 if (!s->restart_seen) {
1065 av_log(m->avctx, AV_LOG_ERROR,
1066 "No restart header present in substream %d.\n", substr);
1069 buf += substream_data_len[substr];
1072 rematrix_channels(m, m->max_decoded_substream);
1074 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1079 substream_length_mismatch:
1080 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1084 m->params_valid = 0;
1088 #if CONFIG_MLP_DECODER
1089 AVCodec mlp_decoder = {
1093 sizeof(MLPDecodeContext),
1098 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1100 #endif /* CONFIG_MLP_DECODER */
1102 #if CONFIG_TRUEHD_DECODER
1103 AVCodec truehd_decoder = {
1107 sizeof(MLPDecodeContext),
1112 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1114 #endif /* CONFIG_TRUEHD_DECODER */