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 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
121 uint8_t params_valid;
123 //! Number of substreams contained within this stream.
124 uint8_t num_substreams;
126 //! Index of the last substream to decode - further substreams are skipped.
127 uint8_t max_decoded_substream;
129 //! number of PCM samples contained in each frame
130 int access_unit_size;
131 //! next power of two above the number of samples in each frame
132 int access_unit_size_pow2;
134 SubStream substream[MAX_SUBSTREAMS];
136 ChannelParams channel_params[MAX_CHANNELS];
138 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
139 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
140 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
143 static VLC huff_vlc[3];
145 /** Initialize static data, constant between all invocations of the codec. */
147 static av_cold void init_static(void)
149 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
150 &ff_mlp_huffman_tables[0][0][1], 2, 1,
151 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
152 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
153 &ff_mlp_huffman_tables[1][0][1], 2, 1,
154 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
155 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
156 &ff_mlp_huffman_tables[2][0][1], 2, 1,
157 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
162 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
163 unsigned int substr, unsigned int ch)
165 ChannelParams *cp = &m->channel_params[ch];
166 SubStream *s = &m->substream[substr];
167 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
168 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
169 int32_t sign_huff_offset = cp->huff_offset;
171 if (cp->codebook > 0)
172 sign_huff_offset -= 7 << lsb_bits;
175 sign_huff_offset -= 1 << sign_shift;
177 return sign_huff_offset;
180 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
183 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
184 unsigned int substr, unsigned int pos)
186 SubStream *s = &m->substream[substr];
187 unsigned int mat, channel;
189 for (mat = 0; mat < s->num_primitive_matrices; mat++)
190 if (s->lsb_bypass[mat])
191 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
193 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
194 ChannelParams *cp = &m->channel_params[channel];
195 int codebook = cp->codebook;
196 int quant_step_size = s->quant_step_size[channel];
197 int lsb_bits = cp->huff_lsbs - quant_step_size;
201 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
202 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
208 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
210 result += cp->sign_huff_offset;
211 result <<= quant_step_size;
213 m->sample_buffer[pos + s->blockpos][channel] = result;
219 static av_cold int mlp_decode_init(AVCodecContext *avctx)
221 MLPDecodeContext *m = avctx->priv_data;
226 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
227 m->substream[substr].lossless_check_data = 0xffffffff;
232 /** Read a major sync info header - contains high level information about
233 * the stream - sample rate, channel arrangement etc. Most of this
234 * information is not actually necessary for decoding, only for playback.
237 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
242 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
245 if (mh.group1_bits == 0) {
246 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
249 if (mh.group2_bits > mh.group1_bits) {
250 av_log(m->avctx, AV_LOG_ERROR,
251 "Channel group 2 cannot have more bits per sample than group 1.\n");
255 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
256 av_log(m->avctx, AV_LOG_ERROR,
257 "Channel groups with differing sample rates are not currently supported.\n");
261 if (mh.group1_samplerate == 0) {
262 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
265 if (mh.group1_samplerate > MAX_SAMPLERATE) {
266 av_log(m->avctx, AV_LOG_ERROR,
267 "Sampling rate %d is greater than the supported maximum (%d).\n",
268 mh.group1_samplerate, MAX_SAMPLERATE);
271 if (mh.access_unit_size > MAX_BLOCKSIZE) {
272 av_log(m->avctx, AV_LOG_ERROR,
273 "Block size %d is greater than the supported maximum (%d).\n",
274 mh.access_unit_size, MAX_BLOCKSIZE);
277 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
278 av_log(m->avctx, AV_LOG_ERROR,
279 "Block size pow2 %d is greater than the supported maximum (%d).\n",
280 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
284 if (mh.num_substreams == 0)
286 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
287 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
290 if (mh.num_substreams > MAX_SUBSTREAMS) {
291 av_log(m->avctx, AV_LOG_ERROR,
292 "Number of substreams %d is larger than the maximum supported "
293 "by the decoder. %s\n", mh.num_substreams, sample_message);
297 m->access_unit_size = mh.access_unit_size;
298 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
300 m->num_substreams = mh.num_substreams;
301 m->max_decoded_substream = m->num_substreams - 1;
303 m->avctx->sample_rate = mh.group1_samplerate;
304 m->avctx->frame_size = mh.access_unit_size;
306 m->avctx->bits_per_raw_sample = mh.group1_bits;
307 if (mh.group1_bits > 16)
308 m->avctx->sample_fmt = SAMPLE_FMT_S32;
310 m->avctx->sample_fmt = SAMPLE_FMT_S16;
313 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
314 m->substream[substr].restart_seen = 0;
319 /** Read a restart header from a block in a substream. This contains parameters
320 * required to decode the audio that do not change very often. Generally
321 * (always) present only in blocks following a major sync. */
323 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
324 const uint8_t *buf, unsigned int substr)
326 SubStream *s = &m->substream[substr];
330 uint8_t lossless_check;
331 int start_count = get_bits_count(gbp);
333 sync_word = get_bits(gbp, 13);
335 if (sync_word != 0x31ea >> 1) {
336 av_log(m->avctx, AV_LOG_ERROR,
337 "restart header sync incorrect (got 0x%04x)\n", sync_word);
340 s->noise_type = get_bits1(gbp);
342 skip_bits(gbp, 16); /* Output timestamp */
344 s->min_channel = get_bits(gbp, 4);
345 s->max_channel = get_bits(gbp, 4);
346 s->max_matrix_channel = get_bits(gbp, 4);
348 if (s->min_channel > s->max_channel) {
349 av_log(m->avctx, AV_LOG_ERROR,
350 "Substream min channel cannot be greater than max channel.\n");
354 if (m->avctx->request_channels > 0
355 && s->max_channel + 1 >= m->avctx->request_channels
356 && substr < m->max_decoded_substream) {
357 av_log(m->avctx, AV_LOG_INFO,
358 "Extracting %d channel downmix from substream %d. "
359 "Further substreams will be skipped.\n",
360 s->max_channel + 1, substr);
361 m->max_decoded_substream = substr;
364 s->noise_shift = get_bits(gbp, 4);
365 s->noisegen_seed = get_bits(gbp, 23);
369 s->data_check_present = get_bits1(gbp);
370 lossless_check = get_bits(gbp, 8);
371 if (substr == m->max_decoded_substream
372 && s->lossless_check_data != 0xffffffff) {
373 tmp = xor_32_to_8(s->lossless_check_data);
374 if (tmp != lossless_check)
375 av_log(m->avctx, AV_LOG_WARNING,
376 "Lossless check failed - expected %02x, calculated %02x.\n",
377 lossless_check, tmp);
379 dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n",
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 dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
391 if (ch_assign > s->max_matrix_channel) {
392 av_log(m->avctx, AV_LOG_ERROR,
393 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
394 ch, ch_assign, sample_message);
397 s->ch_assign[ch_assign] = ch;
400 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
402 if (checksum != get_bits(gbp, 8))
403 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
405 /* Set default decoding parameters. */
406 s->param_presence_flags = 0xff;
407 s->num_primitive_matrices = 0;
409 s->lossless_check_data = 0;
411 memset(s->output_shift , 0, sizeof(s->output_shift ));
412 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
414 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
415 ChannelParams *cp = &m->channel_params[ch];
416 cp->filter_params[FIR].order = 0;
417 cp->filter_params[IIR].order = 0;
418 cp->filter_params[FIR].shift = 0;
419 cp->filter_params[IIR].shift = 0;
421 /* Default audio coding is 24-bit raw PCM. */
423 cp->sign_huff_offset = (-1) << 23;
428 if (substr == m->max_decoded_substream) {
429 m->avctx->channels = s->max_matrix_channel + 1;
435 /** Read parameters for one of the prediction filters. */
437 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
438 unsigned int channel, unsigned int filter)
440 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
441 const char fchar = filter ? 'I' : 'F';
444 // Filter is 0 for FIR, 1 for IIR.
447 order = get_bits(gbp, 4);
448 if (order > MAX_FILTER_ORDER) {
449 av_log(m->avctx, AV_LOG_ERROR,
450 "%cIR filter order %d is greater than maximum %d.\n",
451 fchar, order, MAX_FILTER_ORDER);
457 int coeff_bits, coeff_shift;
459 fp->shift = get_bits(gbp, 4);
461 coeff_bits = get_bits(gbp, 5);
462 coeff_shift = get_bits(gbp, 3);
463 if (coeff_bits < 1 || coeff_bits > 16) {
464 av_log(m->avctx, AV_LOG_ERROR,
465 "%cIR filter coeff_bits must be between 1 and 16.\n",
469 if (coeff_bits + coeff_shift > 16) {
470 av_log(m->avctx, AV_LOG_ERROR,
471 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
476 for (i = 0; i < order; i++)
477 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
479 if (get_bits1(gbp)) {
480 int state_bits, state_shift;
483 av_log(m->avctx, AV_LOG_ERROR,
484 "FIR filter has state data specified.\n");
488 state_bits = get_bits(gbp, 4);
489 state_shift = get_bits(gbp, 4);
491 /* TODO: Check validity of state data. */
493 for (i = 0; i < order; i++)
494 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
501 /** Read parameters for primitive matrices. */
503 static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
505 unsigned int mat, ch;
507 s->num_primitive_matrices = get_bits(gbp, 4);
509 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
510 int frac_bits, max_chan;
511 s->matrix_out_ch[mat] = get_bits(gbp, 4);
512 frac_bits = get_bits(gbp, 4);
513 s->lsb_bypass [mat] = get_bits1(gbp);
515 if (s->matrix_out_ch[mat] > s->max_channel) {
516 av_log(m->avctx, AV_LOG_ERROR,
517 "Invalid channel %d specified as output from matrix.\n",
518 s->matrix_out_ch[mat]);
521 if (frac_bits > 14) {
522 av_log(m->avctx, AV_LOG_ERROR,
523 "Too many fractional bits specified.\n");
527 max_chan = s->max_matrix_channel;
531 for (ch = 0; ch <= max_chan; ch++) {
534 coeff_val = get_sbits(gbp, frac_bits + 2);
536 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
540 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
542 s->matrix_noise_shift[mat] = 0;
548 /** Read channel parameters. */
550 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
551 GetBitContext *gbp, unsigned int ch)
553 ChannelParams *cp = &m->channel_params[ch];
554 FilterParams *fir = &cp->filter_params[FIR];
555 FilterParams *iir = &cp->filter_params[IIR];
556 SubStream *s = &m->substream[substr];
558 if (s->param_presence_flags & PARAM_FIR)
560 if (read_filter_params(m, gbp, ch, FIR) < 0)
563 if (s->param_presence_flags & PARAM_IIR)
565 if (read_filter_params(m, gbp, ch, IIR) < 0)
568 if (fir->order && iir->order &&
569 fir->shift != iir->shift) {
570 av_log(m->avctx, AV_LOG_ERROR,
571 "FIR and IIR filters must use the same precision.\n");
574 /* The FIR and IIR filters must have the same precision.
575 * To simplify the filtering code, only the precision of the
576 * FIR filter is considered. If only the IIR filter is employed,
577 * the FIR filter precision is set to that of the IIR filter, so
578 * that the filtering code can use it. */
579 if (!fir->order && iir->order)
580 fir->shift = iir->shift;
582 if (s->param_presence_flags & PARAM_HUFFOFFSET)
584 cp->huff_offset = get_sbits(gbp, 15);
586 cp->codebook = get_bits(gbp, 2);
587 cp->huff_lsbs = get_bits(gbp, 5);
589 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
596 /** Read decoding parameters that change more often than those in the restart
599 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
602 SubStream *s = &m->substream[substr];
605 if (s->param_presence_flags & PARAM_PRESENCE)
607 s->param_presence_flags = get_bits(gbp, 8);
609 if (s->param_presence_flags & PARAM_BLOCKSIZE)
610 if (get_bits1(gbp)) {
611 s->blocksize = get_bits(gbp, 9);
612 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
613 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
619 if (s->param_presence_flags & PARAM_MATRIX)
620 if (get_bits1(gbp)) {
621 if (read_matrix_params(m, s, gbp) < 0)
625 if (s->param_presence_flags & PARAM_OUTSHIFT)
627 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
628 s->output_shift[ch] = get_sbits(gbp, 4);
629 dprintf(m->avctx, "output shift[%d] = %d\n",
630 ch, s->output_shift[ch]);
633 if (s->param_presence_flags & PARAM_QUANTSTEP)
635 for (ch = 0; ch <= s->max_channel; ch++) {
636 ChannelParams *cp = &m->channel_params[ch];
638 s->quant_step_size[ch] = get_bits(gbp, 4);
640 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
643 for (ch = s->min_channel; ch <= s->max_channel; ch++)
644 if (get_bits1(gbp)) {
645 if (read_channel_params(m, substr, gbp, ch) < 0)
652 #define MSB_MASK(bits) (-1u << bits)
654 /** Generate PCM samples using the prediction filters and residual values
655 * read from the data stream, and update the filter state. */
657 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
658 unsigned int channel)
660 SubStream *s = &m->substream[substr];
661 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
662 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
663 &m->channel_params[channel].filter_params[IIR], };
664 unsigned int filter_shift = fp[FIR]->shift;
665 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
666 int index = MAX_BLOCKSIZE;
669 for (j = 0; j < NUM_FILTERS; j++) {
670 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
671 MAX_FILTER_ORDER * sizeof(int32_t));
674 for (i = 0; i < s->blocksize; i++) {
675 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
680 /* TODO: Move this code to DSPContext? */
682 for (j = 0; j < NUM_FILTERS; j++)
683 for (order = 0; order < fp[j]->order; order++)
684 accum += (int64_t)filter_state_buffer[j][index + order] *
687 accum = accum >> filter_shift;
688 result = (accum + residual) & mask;
692 filter_state_buffer[FIR][index] = result;
693 filter_state_buffer[IIR][index] = result - accum;
695 m->sample_buffer[i + s->blockpos][channel] = result;
698 for (j = 0; j < NUM_FILTERS; j++) {
699 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
700 MAX_FILTER_ORDER * sizeof(int32_t));
704 /** Read a block of PCM residual data (or actual if no filtering active). */
706 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
709 SubStream *s = &m->substream[substr];
710 unsigned int i, ch, expected_stream_pos = 0;
712 if (s->data_check_present) {
713 expected_stream_pos = get_bits_count(gbp);
714 expected_stream_pos += get_bits(gbp, 16);
715 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
716 "we have not tested yet. %s\n", sample_message);
719 if (s->blockpos + s->blocksize > m->access_unit_size) {
720 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
724 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
725 s->blocksize * sizeof(m->bypassed_lsbs[0]));
727 for (i = 0; i < s->blocksize; i++) {
728 if (read_huff_channels(m, gbp, substr, i) < 0)
732 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
733 filter_channel(m, substr, ch);
736 s->blockpos += s->blocksize;
738 if (s->data_check_present) {
739 if (get_bits_count(gbp) != expected_stream_pos)
740 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
747 /** Data table used for TrueHD noise generation function. */
749 static const int8_t noise_table[256] = {
750 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
751 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
752 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
753 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
754 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
755 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
756 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
757 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
758 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
759 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
760 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
761 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
762 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
763 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
764 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
765 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
768 /** Noise generation functions.
769 * I'm not sure what these are for - they seem to be some kind of pseudorandom
770 * sequence generators, used to generate noise data which is used when the
771 * channels are rematrixed. I'm not sure if they provide a practical benefit
772 * to compression, or just obfuscate the decoder. Are they for some kind of
775 /** Generate two channels of noise, used in the matrix when
776 * restart sync word == 0x31ea. */
778 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
780 SubStream *s = &m->substream[substr];
782 uint32_t seed = s->noisegen_seed;
783 unsigned int maxchan = s->max_matrix_channel;
785 for (i = 0; i < s->blockpos; i++) {
786 uint16_t seed_shr7 = seed >> 7;
787 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
788 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
790 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
793 s->noisegen_seed = seed;
796 /** Generate a block of noise, used when restart sync word == 0x31eb. */
798 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
800 SubStream *s = &m->substream[substr];
802 uint32_t seed = s->noisegen_seed;
804 for (i = 0; i < m->access_unit_size_pow2; i++) {
805 uint8_t seed_shr15 = seed >> 15;
806 m->noise_buffer[i] = noise_table[seed_shr15];
807 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
810 s->noisegen_seed = seed;
814 /** Apply the channel matrices in turn to reconstruct the original audio
817 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
819 SubStream *s = &m->substream[substr];
820 unsigned int mat, src_ch, i;
821 unsigned int maxchan;
823 maxchan = s->max_matrix_channel;
824 if (!s->noise_type) {
825 generate_2_noise_channels(m, substr);
828 fill_noise_buffer(m, substr);
831 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
832 int matrix_noise_shift = s->matrix_noise_shift[mat];
833 unsigned int dest_ch = s->matrix_out_ch[mat];
834 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
836 /* TODO: DSPContext? */
838 for (i = 0; i < s->blockpos; i++) {
840 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
841 accum += (int64_t)m->sample_buffer[i][src_ch]
842 * s->matrix_coeff[mat][src_ch];
844 if (matrix_noise_shift) {
845 uint32_t index = s->num_primitive_matrices - mat;
846 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
847 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
849 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
850 + m->bypassed_lsbs[i][mat];
855 /** Write the audio data into the output buffer. */
857 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
858 uint8_t *data, unsigned int *data_size, int is32)
860 SubStream *s = &m->substream[substr];
861 unsigned int i, out_ch = 0;
862 int32_t *data_32 = (int32_t*) data;
863 int16_t *data_16 = (int16_t*) data;
865 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
868 for (i = 0; i < s->blockpos; i++) {
869 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
870 int mat_ch = s->ch_assign[out_ch];
871 int32_t sample = m->sample_buffer[i][mat_ch]
872 << s->output_shift[mat_ch];
873 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
874 if (is32) *data_32++ = sample << 8;
875 else *data_16++ = sample >> 8;
879 *data_size = i * out_ch * (is32 ? 4 : 2);
884 static int output_data(MLPDecodeContext *m, unsigned int substr,
885 uint8_t *data, unsigned int *data_size)
887 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
888 return output_data_internal(m, substr, data, data_size, 1);
890 return output_data_internal(m, substr, data, data_size, 0);
894 /** Read an access unit from the stream.
895 * Returns < 0 on error, 0 if not enough data is present in the input stream
896 * otherwise returns the number of bytes consumed. */
898 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
899 const uint8_t *buf, int buf_size)
901 MLPDecodeContext *m = avctx->priv_data;
903 unsigned int length, substr;
904 unsigned int substream_start;
905 unsigned int header_size = 4;
906 unsigned int substr_header_size = 0;
907 uint8_t substream_parity_present[MAX_SUBSTREAMS];
908 uint16_t substream_data_len[MAX_SUBSTREAMS];
914 length = (AV_RB16(buf) & 0xfff) * 2;
916 if (length > buf_size)
919 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
921 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
922 dprintf(m->avctx, "Found major sync.\n");
923 if (read_major_sync(m, &gb) < 0)
928 if (!m->params_valid) {
929 av_log(m->avctx, AV_LOG_WARNING,
930 "Stream parameters not seen; skipping frame.\n");
937 for (substr = 0; substr < m->num_substreams; substr++) {
938 int extraword_present, checkdata_present, end;
940 extraword_present = get_bits1(&gb);
942 checkdata_present = get_bits1(&gb);
945 end = get_bits(&gb, 12) * 2;
947 substr_header_size += 2;
949 if (extraword_present) {
951 substr_header_size += 2;
954 if (end + header_size + substr_header_size > length) {
955 av_log(m->avctx, AV_LOG_ERROR,
956 "Indicated length of substream %d data goes off end of "
957 "packet.\n", substr);
958 end = length - header_size - substr_header_size;
961 if (end < substream_start) {
962 av_log(avctx, AV_LOG_ERROR,
963 "Indicated end offset of substream %d data "
964 "is smaller than calculated start offset.\n",
969 if (substr > m->max_decoded_substream)
972 substream_parity_present[substr] = checkdata_present;
973 substream_data_len[substr] = end - substream_start;
974 substream_start = end;
977 parity_bits = ff_mlp_calculate_parity(buf, 4);
978 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
980 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
981 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
985 buf += header_size + substr_header_size;
987 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
988 SubStream *s = &m->substream[substr];
989 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
993 if (get_bits1(&gb)) {
994 if (get_bits1(&gb)) {
995 /* A restart header should be present. */
996 if (read_restart_header(m, &gb, buf, substr) < 0)
1001 if (!s->restart_seen) {
1002 av_log(m->avctx, AV_LOG_ERROR,
1003 "No restart header present in substream %d.\n",
1008 if (read_decoding_params(m, &gb, substr) < 0)
1012 if (!s->restart_seen) {
1013 av_log(m->avctx, AV_LOG_ERROR,
1014 "No restart header present in substream %d.\n",
1019 if (read_block_data(m, &gb, substr) < 0)
1022 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1023 && get_bits1(&gb) == 0);
1025 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1026 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
1027 (show_bits_long(&gb, 32) == END_OF_STREAM ||
1028 show_bits_long(&gb, 20) == 0xd234e)) {
1030 if (substr == m->max_decoded_substream)
1031 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1033 if (get_bits1(&gb)) {
1034 int shorten_by = get_bits(&gb, 13);
1035 shorten_by = FFMIN(shorten_by, s->blockpos);
1036 s->blockpos -= shorten_by;
1040 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1041 substream_parity_present[substr]) {
1042 uint8_t parity, checksum;
1044 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1045 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1046 av_log(m->avctx, AV_LOG_ERROR,
1047 "Substream %d parity check failed.\n", substr);
1049 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
1050 if (checksum != get_bits(&gb, 8))
1051 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
1054 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1055 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
1061 buf += substream_data_len[substr];
1064 rematrix_channels(m, m->max_decoded_substream);
1066 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1072 m->params_valid = 0;
1076 #if CONFIG_MLP_DECODER
1077 AVCodec mlp_decoder = {
1081 sizeof(MLPDecodeContext),
1086 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1088 #endif /* CONFIG_MLP_DECODER */
1090 #if CONFIG_TRUEHD_DECODER
1091 AVCodec truehd_decoder = {
1095 sizeof(MLPDecodeContext),
1100 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1102 #endif /* CONFIG_TRUEHD_DECODER */