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"
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];
142 int filter_changed[MAX_CHANNELS][NUM_FILTERS];
144 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
145 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
146 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
149 static VLC huff_vlc[3];
151 /** Initialize static data, constant between all invocations of the codec. */
153 static av_cold void init_static(void)
155 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
156 &ff_mlp_huffman_tables[0][0][1], 2, 1,
157 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
158 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
159 &ff_mlp_huffman_tables[1][0][1], 2, 1,
160 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
161 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
162 &ff_mlp_huffman_tables[2][0][1], 2, 1,
163 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
168 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
169 unsigned int substr, unsigned int ch)
171 ChannelParams *cp = &m->channel_params[ch];
172 SubStream *s = &m->substream[substr];
173 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
174 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
175 int32_t sign_huff_offset = cp->huff_offset;
177 if (cp->codebook > 0)
178 sign_huff_offset -= 7 << lsb_bits;
181 sign_huff_offset -= 1 << sign_shift;
183 return sign_huff_offset;
186 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
189 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
190 unsigned int substr, unsigned int pos)
192 SubStream *s = &m->substream[substr];
193 unsigned int mat, channel;
195 for (mat = 0; mat < s->num_primitive_matrices; mat++)
196 if (s->lsb_bypass[mat])
197 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
199 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
200 ChannelParams *cp = &m->channel_params[channel];
201 int codebook = cp->codebook;
202 int quant_step_size = s->quant_step_size[channel];
203 int lsb_bits = cp->huff_lsbs - quant_step_size;
207 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
208 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
214 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
216 result += cp->sign_huff_offset;
217 result <<= quant_step_size;
219 m->sample_buffer[pos + s->blockpos][channel] = result;
225 static av_cold int mlp_decode_init(AVCodecContext *avctx)
227 MLPDecodeContext *m = avctx->priv_data;
232 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
233 m->substream[substr].lossless_check_data = 0xffffffff;
238 /** Read a major sync info header - contains high level information about
239 * the stream - sample rate, channel arrangement etc. Most of this
240 * information is not actually necessary for decoding, only for playback.
243 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
248 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
251 if (mh.group1_bits == 0) {
252 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
255 if (mh.group2_bits > mh.group1_bits) {
256 av_log(m->avctx, AV_LOG_ERROR,
257 "Channel group 2 cannot have more bits per sample than group 1.\n");
261 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
262 av_log(m->avctx, AV_LOG_ERROR,
263 "Channel groups with differing sample rates are not currently supported.\n");
267 if (mh.group1_samplerate == 0) {
268 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
271 if (mh.group1_samplerate > MAX_SAMPLERATE) {
272 av_log(m->avctx, AV_LOG_ERROR,
273 "Sampling rate %d is greater than the supported maximum (%d).\n",
274 mh.group1_samplerate, MAX_SAMPLERATE);
277 if (mh.access_unit_size > MAX_BLOCKSIZE) {
278 av_log(m->avctx, AV_LOG_ERROR,
279 "Block size %d is greater than the supported maximum (%d).\n",
280 mh.access_unit_size, MAX_BLOCKSIZE);
283 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
284 av_log(m->avctx, AV_LOG_ERROR,
285 "Block size pow2 %d is greater than the supported maximum (%d).\n",
286 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
290 if (mh.num_substreams == 0)
292 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
293 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
296 if (mh.num_substreams > MAX_SUBSTREAMS) {
297 av_log(m->avctx, AV_LOG_ERROR,
298 "Number of substreams %d is larger than the maximum supported "
299 "by the decoder. %s\n", mh.num_substreams, sample_message);
303 m->access_unit_size = mh.access_unit_size;
304 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
306 m->num_substreams = mh.num_substreams;
307 m->max_decoded_substream = m->num_substreams - 1;
309 m->avctx->sample_rate = mh.group1_samplerate;
310 m->avctx->frame_size = mh.access_unit_size;
312 m->avctx->bits_per_raw_sample = mh.group1_bits;
313 if (mh.group1_bits > 16)
314 m->avctx->sample_fmt = SAMPLE_FMT_S32;
316 m->avctx->sample_fmt = SAMPLE_FMT_S16;
319 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
320 m->substream[substr].restart_seen = 0;
325 /** Read a restart header from a block in a substream. This contains parameters
326 * required to decode the audio that do not change very often. Generally
327 * (always) present only in blocks following a major sync. */
329 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
330 const uint8_t *buf, unsigned int substr)
332 SubStream *s = &m->substream[substr];
336 uint8_t lossless_check;
337 int start_count = get_bits_count(gbp);
339 sync_word = get_bits(gbp, 13);
341 if (sync_word != 0x31ea >> 1) {
342 av_log(m->avctx, AV_LOG_ERROR,
343 "restart header sync incorrect (got 0x%04x)\n", sync_word);
346 s->noise_type = get_bits1(gbp);
348 skip_bits(gbp, 16); /* Output timestamp */
350 s->min_channel = get_bits(gbp, 4);
351 s->max_channel = get_bits(gbp, 4);
352 s->max_matrix_channel = get_bits(gbp, 4);
354 if (s->min_channel > s->max_channel) {
355 av_log(m->avctx, AV_LOG_ERROR,
356 "Substream min channel cannot be greater than max channel.\n");
360 if (m->avctx->request_channels > 0
361 && s->max_channel + 1 >= m->avctx->request_channels
362 && substr < m->max_decoded_substream) {
363 av_log(m->avctx, AV_LOG_INFO,
364 "Extracting %d channel downmix from substream %d. "
365 "Further substreams will be skipped.\n",
366 s->max_channel + 1, substr);
367 m->max_decoded_substream = substr;
370 s->noise_shift = get_bits(gbp, 4);
371 s->noisegen_seed = get_bits(gbp, 23);
375 s->data_check_present = get_bits1(gbp);
376 lossless_check = get_bits(gbp, 8);
377 if (substr == m->max_decoded_substream
378 && s->lossless_check_data != 0xffffffff) {
379 tmp = xor_32_to_8(s->lossless_check_data);
380 if (tmp != lossless_check)
381 av_log(m->avctx, AV_LOG_WARNING,
382 "Lossless check failed - expected %02x, calculated %02x.\n",
383 lossless_check, tmp);
388 memset(s->ch_assign, 0, sizeof(s->ch_assign));
390 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
391 int ch_assign = get_bits(gbp, 6);
392 if (ch_assign > s->max_matrix_channel) {
393 av_log(m->avctx, AV_LOG_ERROR,
394 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
395 ch, ch_assign, sample_message);
398 s->ch_assign[ch_assign] = ch;
401 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
403 if (checksum != get_bits(gbp, 8))
404 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
406 /* Set default decoding parameters. */
407 s->param_presence_flags = 0xff;
408 s->num_primitive_matrices = 0;
410 s->lossless_check_data = 0;
412 memset(s->output_shift , 0, sizeof(s->output_shift ));
413 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
415 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
416 ChannelParams *cp = &m->channel_params[ch];
417 cp->filter_params[FIR].order = 0;
418 cp->filter_params[IIR].order = 0;
419 cp->filter_params[FIR].shift = 0;
420 cp->filter_params[IIR].shift = 0;
422 /* Default audio coding is 24-bit raw PCM. */
424 cp->sign_huff_offset = (-1) << 23;
429 if (substr == m->max_decoded_substream) {
430 m->avctx->channels = s->max_matrix_channel + 1;
436 /** Read parameters for one of the prediction filters. */
438 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
439 unsigned int channel, unsigned int filter)
441 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
442 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
443 const char fchar = filter ? 'I' : 'F';
446 // Filter is 0 for FIR, 1 for IIR.
449 if (m->filter_changed[channel][filter]++ > 1) {
450 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
454 order = get_bits(gbp, 4);
455 if (order > max_order) {
456 av_log(m->avctx, AV_LOG_ERROR,
457 "%cIR filter order %d is greater than maximum %d.\n",
458 fchar, order, max_order);
464 int coeff_bits, coeff_shift;
466 fp->shift = get_bits(gbp, 4);
468 coeff_bits = get_bits(gbp, 5);
469 coeff_shift = get_bits(gbp, 3);
470 if (coeff_bits < 1 || coeff_bits > 16) {
471 av_log(m->avctx, AV_LOG_ERROR,
472 "%cIR filter coeff_bits must be between 1 and 16.\n",
476 if (coeff_bits + coeff_shift > 16) {
477 av_log(m->avctx, AV_LOG_ERROR,
478 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
483 for (i = 0; i < order; i++)
484 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
486 if (get_bits1(gbp)) {
487 int state_bits, state_shift;
490 av_log(m->avctx, AV_LOG_ERROR,
491 "FIR filter has state data specified.\n");
495 state_bits = get_bits(gbp, 4);
496 state_shift = get_bits(gbp, 4);
498 /* TODO: Check validity of state data. */
500 for (i = 0; i < order; i++)
501 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
508 /** Read parameters for primitive matrices. */
510 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
512 SubStream *s = &m->substream[substr];
513 unsigned int mat, ch;
515 if (m->matrix_changed++ > 1) {
516 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
520 s->num_primitive_matrices = get_bits(gbp, 4);
522 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
523 int frac_bits, max_chan;
524 s->matrix_out_ch[mat] = get_bits(gbp, 4);
525 frac_bits = get_bits(gbp, 4);
526 s->lsb_bypass [mat] = get_bits1(gbp);
528 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
529 av_log(m->avctx, AV_LOG_ERROR,
530 "Invalid channel %d specified as output from matrix.\n",
531 s->matrix_out_ch[mat]);
534 if (frac_bits > 14) {
535 av_log(m->avctx, AV_LOG_ERROR,
536 "Too many fractional bits specified.\n");
540 max_chan = s->max_matrix_channel;
544 for (ch = 0; ch <= max_chan; ch++) {
547 coeff_val = get_sbits(gbp, frac_bits + 2);
549 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
553 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
555 s->matrix_noise_shift[mat] = 0;
561 /** Read channel parameters. */
563 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
564 GetBitContext *gbp, unsigned int ch)
566 ChannelParams *cp = &m->channel_params[ch];
567 FilterParams *fir = &cp->filter_params[FIR];
568 FilterParams *iir = &cp->filter_params[IIR];
569 SubStream *s = &m->substream[substr];
571 if (s->param_presence_flags & PARAM_FIR)
573 if (read_filter_params(m, gbp, ch, FIR) < 0)
576 if (s->param_presence_flags & PARAM_IIR)
578 if (read_filter_params(m, gbp, ch, IIR) < 0)
581 if (fir->order + iir->order > 8) {
582 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
586 if (fir->order && iir->order &&
587 fir->shift != iir->shift) {
588 av_log(m->avctx, AV_LOG_ERROR,
589 "FIR and IIR filters must use the same precision.\n");
592 /* The FIR and IIR filters must have the same precision.
593 * To simplify the filtering code, only the precision of the
594 * FIR filter is considered. If only the IIR filter is employed,
595 * the FIR filter precision is set to that of the IIR filter, so
596 * that the filtering code can use it. */
597 if (!fir->order && iir->order)
598 fir->shift = iir->shift;
600 if (s->param_presence_flags & PARAM_HUFFOFFSET)
602 cp->huff_offset = get_sbits(gbp, 15);
604 cp->codebook = get_bits(gbp, 2);
605 cp->huff_lsbs = get_bits(gbp, 5);
607 if (cp->huff_lsbs > 24) {
608 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
612 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
617 /** Read decoding parameters that change more often than those in the restart
620 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
623 SubStream *s = &m->substream[substr];
626 if (s->param_presence_flags & PARAM_PRESENCE)
628 s->param_presence_flags = get_bits(gbp, 8);
630 if (s->param_presence_flags & PARAM_BLOCKSIZE)
631 if (get_bits1(gbp)) {
632 s->blocksize = get_bits(gbp, 9);
633 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
634 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
640 if (s->param_presence_flags & PARAM_MATRIX)
641 if (get_bits1(gbp)) {
642 if (read_matrix_params(m, substr, gbp) < 0)
646 if (s->param_presence_flags & PARAM_OUTSHIFT)
648 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
649 s->output_shift[ch] = get_sbits(gbp, 4);
652 if (s->param_presence_flags & PARAM_QUANTSTEP)
654 for (ch = 0; ch <= s->max_channel; ch++) {
655 ChannelParams *cp = &m->channel_params[ch];
657 s->quant_step_size[ch] = get_bits(gbp, 4);
659 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
662 for (ch = s->min_channel; ch <= s->max_channel; ch++)
663 if (get_bits1(gbp)) {
664 if (read_channel_params(m, substr, gbp, ch) < 0)
671 #define MSB_MASK(bits) (-1u << bits)
673 /** Generate PCM samples using the prediction filters and residual values
674 * read from the data stream, and update the filter state. */
676 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
677 unsigned int channel)
679 SubStream *s = &m->substream[substr];
680 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
681 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
682 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
683 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
684 unsigned int filter_shift = fir->shift;
685 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
686 int index = MAX_BLOCKSIZE;
689 memcpy(&firbuf[index], fir->state, MAX_FIR_ORDER * sizeof(int32_t));
690 memcpy(&iirbuf[index], iir->state, MAX_IIR_ORDER * sizeof(int32_t));
692 for (i = 0; i < s->blocksize; i++) {
693 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
698 /* TODO: Move this code to DSPContext? */
700 for (order = 0; order < fir->order; order++)
701 accum += (int64_t) firbuf[index + order] * fir->coeff[order];
702 for (order = 0; order < iir->order; order++)
703 accum += (int64_t) iirbuf[index + order] * iir->coeff[order];
705 accum = accum >> filter_shift;
706 result = (accum + residual) & mask;
710 firbuf[index] = result;
711 iirbuf[index] = result - accum;
713 m->sample_buffer[i + s->blockpos][channel] = result;
716 memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
717 memcpy(iir->state, &iirbuf[index], MAX_IIR_ORDER * sizeof(int32_t));
720 /** Read a block of PCM residual data (or actual if no filtering active). */
722 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
725 SubStream *s = &m->substream[substr];
726 unsigned int i, ch, expected_stream_pos = 0;
728 if (s->data_check_present) {
729 expected_stream_pos = get_bits_count(gbp);
730 expected_stream_pos += get_bits(gbp, 16);
731 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
732 "we have not tested yet. %s\n", sample_message);
735 if (s->blockpos + s->blocksize > m->access_unit_size) {
736 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
740 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
741 s->blocksize * sizeof(m->bypassed_lsbs[0]));
743 for (i = 0; i < s->blocksize; i++) {
744 if (read_huff_channels(m, gbp, substr, i) < 0)
748 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
749 filter_channel(m, substr, ch);
752 s->blockpos += s->blocksize;
754 if (s->data_check_present) {
755 if (get_bits_count(gbp) != expected_stream_pos)
756 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
763 /** Data table used for TrueHD noise generation function. */
765 static const int8_t noise_table[256] = {
766 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
767 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
768 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
769 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
770 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
771 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
772 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
773 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
774 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
775 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
776 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
777 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
778 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
779 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
780 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
781 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
784 /** Noise generation functions.
785 * I'm not sure what these are for - they seem to be some kind of pseudorandom
786 * sequence generators, used to generate noise data which is used when the
787 * channels are rematrixed. I'm not sure if they provide a practical benefit
788 * to compression, or just obfuscate the decoder. Are they for some kind of
791 /** Generate two channels of noise, used in the matrix when
792 * restart sync word == 0x31ea. */
794 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
796 SubStream *s = &m->substream[substr];
798 uint32_t seed = s->noisegen_seed;
799 unsigned int maxchan = s->max_matrix_channel;
801 for (i = 0; i < s->blockpos; i++) {
802 uint16_t seed_shr7 = seed >> 7;
803 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
804 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
806 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
809 s->noisegen_seed = seed;
812 /** Generate a block of noise, used when restart sync word == 0x31eb. */
814 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
816 SubStream *s = &m->substream[substr];
818 uint32_t seed = s->noisegen_seed;
820 for (i = 0; i < m->access_unit_size_pow2; i++) {
821 uint8_t seed_shr15 = seed >> 15;
822 m->noise_buffer[i] = noise_table[seed_shr15];
823 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
826 s->noisegen_seed = seed;
830 /** Apply the channel matrices in turn to reconstruct the original audio
833 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
835 SubStream *s = &m->substream[substr];
836 unsigned int mat, src_ch, i;
837 unsigned int maxchan;
839 maxchan = s->max_matrix_channel;
840 if (!s->noise_type) {
841 generate_2_noise_channels(m, substr);
844 fill_noise_buffer(m, substr);
847 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
848 int matrix_noise_shift = s->matrix_noise_shift[mat];
849 unsigned int dest_ch = s->matrix_out_ch[mat];
850 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
852 /* TODO: DSPContext? */
854 for (i = 0; i < s->blockpos; i++) {
856 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
857 accum += (int64_t)m->sample_buffer[i][src_ch]
858 * s->matrix_coeff[mat][src_ch];
860 if (matrix_noise_shift) {
861 uint32_t index = s->num_primitive_matrices - mat;
862 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
863 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
865 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
866 + m->bypassed_lsbs[i][mat];
871 /** Write the audio data into the output buffer. */
873 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
874 uint8_t *data, unsigned int *data_size, int is32)
876 SubStream *s = &m->substream[substr];
877 unsigned int i, out_ch = 0;
878 int32_t *data_32 = (int32_t*) data;
879 int16_t *data_16 = (int16_t*) data;
881 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
884 for (i = 0; i < s->blockpos; i++) {
885 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
886 int mat_ch = s->ch_assign[out_ch];
887 int32_t sample = m->sample_buffer[i][mat_ch]
888 << s->output_shift[mat_ch];
889 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
890 if (is32) *data_32++ = sample << 8;
891 else *data_16++ = sample >> 8;
895 *data_size = i * out_ch * (is32 ? 4 : 2);
900 static int output_data(MLPDecodeContext *m, unsigned int substr,
901 uint8_t *data, unsigned int *data_size)
903 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
904 return output_data_internal(m, substr, data, data_size, 1);
906 return output_data_internal(m, substr, data, data_size, 0);
910 /** Read an access unit from the stream.
911 * Returns < 0 on error, 0 if not enough data is present in the input stream
912 * otherwise returns the number of bytes consumed. */
914 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
917 const uint8_t *buf = avpkt->data;
918 int buf_size = avpkt->size;
919 MLPDecodeContext *m = avctx->priv_data;
921 unsigned int length, substr;
922 unsigned int substream_start;
923 unsigned int header_size = 4;
924 unsigned int substr_header_size = 0;
925 uint8_t substream_parity_present[MAX_SUBSTREAMS];
926 uint16_t substream_data_len[MAX_SUBSTREAMS];
932 length = (AV_RB16(buf) & 0xfff) * 2;
934 if (length > buf_size)
937 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
939 m->is_major_sync_unit = 0;
940 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
941 if (read_major_sync(m, &gb) < 0)
943 m->is_major_sync_unit = 1;
947 if (!m->params_valid) {
948 av_log(m->avctx, AV_LOG_WARNING,
949 "Stream parameters not seen; skipping frame.\n");
956 for (substr = 0; substr < m->num_substreams; substr++) {
957 int extraword_present, checkdata_present, end, nonrestart_substr;
959 extraword_present = get_bits1(&gb);
960 nonrestart_substr = get_bits1(&gb);
961 checkdata_present = get_bits1(&gb);
964 end = get_bits(&gb, 12) * 2;
966 substr_header_size += 2;
968 if (extraword_present) {
969 if (m->avctx->codec_id == CODEC_ID_MLP) {
970 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
974 substr_header_size += 2;
977 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
978 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
982 if (end + header_size + substr_header_size > length) {
983 av_log(m->avctx, AV_LOG_ERROR,
984 "Indicated length of substream %d data goes off end of "
985 "packet.\n", substr);
986 end = length - header_size - substr_header_size;
989 if (end < substream_start) {
990 av_log(avctx, AV_LOG_ERROR,
991 "Indicated end offset of substream %d data "
992 "is smaller than calculated start offset.\n",
997 if (substr > m->max_decoded_substream)
1000 substream_parity_present[substr] = checkdata_present;
1001 substream_data_len[substr] = end - substream_start;
1002 substream_start = end;
1005 parity_bits = ff_mlp_calculate_parity(buf, 4);
1006 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1008 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1009 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1013 buf += header_size + substr_header_size;
1015 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1016 SubStream *s = &m->substream[substr];
1017 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1019 m->matrix_changed = 0;
1020 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1024 if (get_bits1(&gb)) {
1025 if (get_bits1(&gb)) {
1026 /* A restart header should be present. */
1027 if (read_restart_header(m, &gb, buf, substr) < 0)
1029 s->restart_seen = 1;
1032 if (!s->restart_seen) {
1036 if (read_decoding_params(m, &gb, substr) < 0)
1040 if (!s->restart_seen) {
1044 if (read_block_data(m, &gb, substr) < 0)
1047 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1048 goto substream_length_mismatch;
1050 } while (!get_bits1(&gb));
1052 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1053 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1056 if (get_bits(&gb, 16) != 0xD234)
1059 shorten_by = get_bits(&gb, 16);
1060 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1061 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1062 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1065 if (substr == m->max_decoded_substream)
1066 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1068 if (substream_parity_present[substr]) {
1069 uint8_t parity, checksum;
1071 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1072 goto substream_length_mismatch;
1074 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1075 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1077 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1078 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1079 if ( get_bits(&gb, 8) != checksum)
1080 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1082 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1083 goto substream_length_mismatch;
1087 if (!s->restart_seen) {
1088 av_log(m->avctx, AV_LOG_ERROR,
1089 "No restart header present in substream %d.\n", substr);
1092 buf += substream_data_len[substr];
1095 rematrix_channels(m, m->max_decoded_substream);
1097 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1102 substream_length_mismatch:
1103 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1107 m->params_valid = 0;
1111 #if CONFIG_MLP_DECODER
1112 AVCodec mlp_decoder = {
1116 sizeof(MLPDecodeContext),
1121 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1123 #endif /* CONFIG_MLP_DECODER */
1125 #if CONFIG_TRUEHD_DECODER
1126 AVCodec truehd_decoder = {
1130 sizeof(MLPDecodeContext),
1135 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1137 #endif /* CONFIG_TRUEHD_DECODER */