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);
382 memset(s->ch_assign, 0, sizeof(s->ch_assign));
384 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
385 int ch_assign = get_bits(gbp, 6);
386 if (ch_assign > s->max_matrix_channel) {
387 av_log(m->avctx, AV_LOG_ERROR,
388 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
389 ch, ch_assign, sample_message);
392 s->ch_assign[ch_assign] = ch;
395 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
397 if (checksum != get_bits(gbp, 8))
398 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
400 /* Set default decoding parameters. */
401 s->param_presence_flags = 0xff;
402 s->num_primitive_matrices = 0;
404 s->lossless_check_data = 0;
406 memset(s->output_shift , 0, sizeof(s->output_shift ));
407 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
409 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
410 ChannelParams *cp = &m->channel_params[ch];
411 cp->filter_params[FIR].order = 0;
412 cp->filter_params[IIR].order = 0;
413 cp->filter_params[FIR].shift = 0;
414 cp->filter_params[IIR].shift = 0;
416 /* Default audio coding is 24-bit raw PCM. */
418 cp->sign_huff_offset = (-1) << 23;
423 if (substr == m->max_decoded_substream) {
424 m->avctx->channels = s->max_matrix_channel + 1;
430 /** Read parameters for one of the prediction filters. */
432 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
433 unsigned int channel, unsigned int filter)
435 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
436 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
437 const char fchar = filter ? 'I' : 'F';
440 // Filter is 0 for FIR, 1 for IIR.
443 order = get_bits(gbp, 4);
444 if (order > max_order) {
445 av_log(m->avctx, AV_LOG_ERROR,
446 "%cIR filter order %d is greater than maximum %d.\n",
447 fchar, order, max_order);
453 int coeff_bits, coeff_shift;
455 fp->shift = get_bits(gbp, 4);
457 coeff_bits = get_bits(gbp, 5);
458 coeff_shift = get_bits(gbp, 3);
459 if (coeff_bits < 1 || coeff_bits > 16) {
460 av_log(m->avctx, AV_LOG_ERROR,
461 "%cIR filter coeff_bits must be between 1 and 16.\n",
465 if (coeff_bits + coeff_shift > 16) {
466 av_log(m->avctx, AV_LOG_ERROR,
467 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
472 for (i = 0; i < order; i++)
473 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
475 if (get_bits1(gbp)) {
476 int state_bits, state_shift;
479 av_log(m->avctx, AV_LOG_ERROR,
480 "FIR filter has state data specified.\n");
484 state_bits = get_bits(gbp, 4);
485 state_shift = get_bits(gbp, 4);
487 /* TODO: Check validity of state data. */
489 for (i = 0; i < order; i++)
490 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
497 /** Read parameters for primitive matrices. */
499 static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
501 unsigned int mat, ch;
503 s->num_primitive_matrices = get_bits(gbp, 4);
505 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
506 int frac_bits, max_chan;
507 s->matrix_out_ch[mat] = get_bits(gbp, 4);
508 frac_bits = get_bits(gbp, 4);
509 s->lsb_bypass [mat] = get_bits1(gbp);
511 if (s->matrix_out_ch[mat] > s->max_channel) {
512 av_log(m->avctx, AV_LOG_ERROR,
513 "Invalid channel %d specified as output from matrix.\n",
514 s->matrix_out_ch[mat]);
517 if (frac_bits > 14) {
518 av_log(m->avctx, AV_LOG_ERROR,
519 "Too many fractional bits specified.\n");
523 max_chan = s->max_matrix_channel;
527 for (ch = 0; ch <= max_chan; ch++) {
530 coeff_val = get_sbits(gbp, frac_bits + 2);
532 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
536 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
538 s->matrix_noise_shift[mat] = 0;
544 /** Read channel parameters. */
546 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
547 GetBitContext *gbp, unsigned int ch)
549 ChannelParams *cp = &m->channel_params[ch];
550 FilterParams *fir = &cp->filter_params[FIR];
551 FilterParams *iir = &cp->filter_params[IIR];
552 SubStream *s = &m->substream[substr];
554 if (s->param_presence_flags & PARAM_FIR)
556 if (read_filter_params(m, gbp, ch, FIR) < 0)
559 if (s->param_presence_flags & PARAM_IIR)
561 if (read_filter_params(m, gbp, ch, IIR) < 0)
564 if (fir->order + iir->order > 8) {
565 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
569 if (fir->order && iir->order &&
570 fir->shift != iir->shift) {
571 av_log(m->avctx, AV_LOG_ERROR,
572 "FIR and IIR filters must use the same precision.\n");
575 /* The FIR and IIR filters must have the same precision.
576 * To simplify the filtering code, only the precision of the
577 * FIR filter is considered. If only the IIR filter is employed,
578 * the FIR filter precision is set to that of the IIR filter, so
579 * that the filtering code can use it. */
580 if (!fir->order && iir->order)
581 fir->shift = iir->shift;
583 if (s->param_presence_flags & PARAM_HUFFOFFSET)
585 cp->huff_offset = get_sbits(gbp, 15);
587 cp->codebook = get_bits(gbp, 2);
588 cp->huff_lsbs = get_bits(gbp, 5);
590 if (cp->huff_lsbs > 24) {
591 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
595 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
600 /** Read decoding parameters that change more often than those in the restart
603 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
606 SubStream *s = &m->substream[substr];
609 if (s->param_presence_flags & PARAM_PRESENCE)
611 s->param_presence_flags = get_bits(gbp, 8);
613 if (s->param_presence_flags & PARAM_BLOCKSIZE)
614 if (get_bits1(gbp)) {
615 s->blocksize = get_bits(gbp, 9);
616 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
617 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
623 if (s->param_presence_flags & PARAM_MATRIX)
624 if (get_bits1(gbp)) {
625 if (read_matrix_params(m, s, gbp) < 0)
629 if (s->param_presence_flags & PARAM_OUTSHIFT)
631 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
632 s->output_shift[ch] = get_sbits(gbp, 4);
635 if (s->param_presence_flags & PARAM_QUANTSTEP)
637 for (ch = 0; ch <= s->max_channel; ch++) {
638 ChannelParams *cp = &m->channel_params[ch];
640 s->quant_step_size[ch] = get_bits(gbp, 4);
642 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
645 for (ch = s->min_channel; ch <= s->max_channel; ch++)
646 if (get_bits1(gbp)) {
647 if (read_channel_params(m, substr, gbp, ch) < 0)
654 #define MSB_MASK(bits) (-1u << bits)
656 /** Generate PCM samples using the prediction filters and residual values
657 * read from the data stream, and update the filter state. */
659 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
660 unsigned int channel)
662 SubStream *s = &m->substream[substr];
663 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
664 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
665 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
666 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
667 unsigned int filter_shift = fir->shift;
668 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
669 int index = MAX_BLOCKSIZE;
672 memcpy(&firbuf[index], fir->state, MAX_FIR_ORDER * sizeof(int32_t));
673 memcpy(&iirbuf[index], iir->state, MAX_IIR_ORDER * sizeof(int32_t));
675 for (i = 0; i < s->blocksize; i++) {
676 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
681 /* TODO: Move this code to DSPContext? */
683 for (order = 0; order < fir->order; order++)
684 accum += (int64_t) firbuf[index + order] * fir->coeff[order];
685 for (order = 0; order < iir->order; order++)
686 accum += (int64_t) iirbuf[index + order] * iir->coeff[order];
688 accum = accum >> filter_shift;
689 result = (accum + residual) & mask;
693 firbuf[index] = result;
694 iirbuf[index] = result - accum;
696 m->sample_buffer[i + s->blockpos][channel] = result;
699 memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
700 memcpy(iir->state, &iirbuf[index], MAX_IIR_ORDER * sizeof(int32_t));
703 /** Read a block of PCM residual data (or actual if no filtering active). */
705 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
708 SubStream *s = &m->substream[substr];
709 unsigned int i, ch, expected_stream_pos = 0;
711 if (s->data_check_present) {
712 expected_stream_pos = get_bits_count(gbp);
713 expected_stream_pos += get_bits(gbp, 16);
714 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
715 "we have not tested yet. %s\n", sample_message);
718 if (s->blockpos + s->blocksize > m->access_unit_size) {
719 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
723 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
724 s->blocksize * sizeof(m->bypassed_lsbs[0]));
726 for (i = 0; i < s->blocksize; i++) {
727 if (read_huff_channels(m, gbp, substr, i) < 0)
731 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
732 filter_channel(m, substr, ch);
735 s->blockpos += s->blocksize;
737 if (s->data_check_present) {
738 if (get_bits_count(gbp) != expected_stream_pos)
739 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
746 /** Data table used for TrueHD noise generation function. */
748 static const int8_t noise_table[256] = {
749 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
750 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
751 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
752 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
753 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
754 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
755 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
756 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
757 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
758 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
759 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
760 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
761 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
762 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
763 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
764 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
767 /** Noise generation functions.
768 * I'm not sure what these are for - they seem to be some kind of pseudorandom
769 * sequence generators, used to generate noise data which is used when the
770 * channels are rematrixed. I'm not sure if they provide a practical benefit
771 * to compression, or just obfuscate the decoder. Are they for some kind of
774 /** Generate two channels of noise, used in the matrix when
775 * restart sync word == 0x31ea. */
777 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
779 SubStream *s = &m->substream[substr];
781 uint32_t seed = s->noisegen_seed;
782 unsigned int maxchan = s->max_matrix_channel;
784 for (i = 0; i < s->blockpos; i++) {
785 uint16_t seed_shr7 = seed >> 7;
786 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
787 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
789 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
792 s->noisegen_seed = seed;
795 /** Generate a block of noise, used when restart sync word == 0x31eb. */
797 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
799 SubStream *s = &m->substream[substr];
801 uint32_t seed = s->noisegen_seed;
803 for (i = 0; i < m->access_unit_size_pow2; i++) {
804 uint8_t seed_shr15 = seed >> 15;
805 m->noise_buffer[i] = noise_table[seed_shr15];
806 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
809 s->noisegen_seed = seed;
813 /** Apply the channel matrices in turn to reconstruct the original audio
816 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
818 SubStream *s = &m->substream[substr];
819 unsigned int mat, src_ch, i;
820 unsigned int maxchan;
822 maxchan = s->max_matrix_channel;
823 if (!s->noise_type) {
824 generate_2_noise_channels(m, substr);
827 fill_noise_buffer(m, substr);
830 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
831 int matrix_noise_shift = s->matrix_noise_shift[mat];
832 unsigned int dest_ch = s->matrix_out_ch[mat];
833 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
835 /* TODO: DSPContext? */
837 for (i = 0; i < s->blockpos; i++) {
839 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
840 accum += (int64_t)m->sample_buffer[i][src_ch]
841 * s->matrix_coeff[mat][src_ch];
843 if (matrix_noise_shift) {
844 uint32_t index = s->num_primitive_matrices - mat;
845 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
846 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
848 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
849 + m->bypassed_lsbs[i][mat];
854 /** Write the audio data into the output buffer. */
856 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
857 uint8_t *data, unsigned int *data_size, int is32)
859 SubStream *s = &m->substream[substr];
860 unsigned int i, out_ch = 0;
861 int32_t *data_32 = (int32_t*) data;
862 int16_t *data_16 = (int16_t*) data;
864 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
867 for (i = 0; i < s->blockpos; i++) {
868 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
869 int mat_ch = s->ch_assign[out_ch];
870 int32_t sample = m->sample_buffer[i][mat_ch]
871 << s->output_shift[mat_ch];
872 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
873 if (is32) *data_32++ = sample << 8;
874 else *data_16++ = sample >> 8;
878 *data_size = i * out_ch * (is32 ? 4 : 2);
883 static int output_data(MLPDecodeContext *m, unsigned int substr,
884 uint8_t *data, unsigned int *data_size)
886 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
887 return output_data_internal(m, substr, data, data_size, 1);
889 return output_data_internal(m, substr, data, data_size, 0);
893 /** Read an access unit from the stream.
894 * Returns < 0 on error, 0 if not enough data is present in the input stream
895 * otherwise returns the number of bytes consumed. */
897 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
898 const uint8_t *buf, int buf_size)
900 MLPDecodeContext *m = avctx->priv_data;
902 unsigned int length, substr;
903 unsigned int substream_start;
904 unsigned int header_size = 4;
905 unsigned int substr_header_size = 0;
906 uint8_t substream_parity_present[MAX_SUBSTREAMS];
907 uint16_t substream_data_len[MAX_SUBSTREAMS];
913 length = (AV_RB16(buf) & 0xfff) * 2;
915 if (length > buf_size)
918 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
920 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
921 if (read_major_sync(m, &gb) < 0)
926 if (!m->params_valid) {
927 av_log(m->avctx, AV_LOG_WARNING,
928 "Stream parameters not seen; skipping frame.\n");
935 for (substr = 0; substr < m->num_substreams; substr++) {
936 int extraword_present, checkdata_present, end;
938 extraword_present = get_bits1(&gb);
940 checkdata_present = get_bits1(&gb);
943 end = get_bits(&gb, 12) * 2;
945 substr_header_size += 2;
947 if (extraword_present) {
949 substr_header_size += 2;
952 if (end + header_size + substr_header_size > length) {
953 av_log(m->avctx, AV_LOG_ERROR,
954 "Indicated length of substream %d data goes off end of "
955 "packet.\n", substr);
956 end = length - header_size - substr_header_size;
959 if (end < substream_start) {
960 av_log(avctx, AV_LOG_ERROR,
961 "Indicated end offset of substream %d data "
962 "is smaller than calculated start offset.\n",
967 if (substr > m->max_decoded_substream)
970 substream_parity_present[substr] = checkdata_present;
971 substream_data_len[substr] = end - substream_start;
972 substream_start = end;
975 parity_bits = ff_mlp_calculate_parity(buf, 4);
976 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
978 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
979 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
983 buf += header_size + substr_header_size;
985 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
986 SubStream *s = &m->substream[substr];
987 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
991 if (get_bits1(&gb)) {
992 if (get_bits1(&gb)) {
993 /* A restart header should be present. */
994 if (read_restart_header(m, &gb, buf, substr) < 0)
999 if (!s->restart_seen) {
1000 av_log(m->avctx, AV_LOG_ERROR,
1001 "No restart header present in substream %d.\n",
1006 if (read_decoding_params(m, &gb, substr) < 0)
1010 if (!s->restart_seen) {
1011 av_log(m->avctx, AV_LOG_ERROR,
1012 "No restart header present in substream %d.\n",
1017 if (read_block_data(m, &gb, substr) < 0)
1020 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1021 && get_bits1(&gb) == 0);
1023 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1024 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1027 if (get_bits(&gb, 16) != 0xD234)
1030 shorten_by = get_bits(&gb, 16);
1031 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1032 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1033 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1036 if (substr == m->max_decoded_substream)
1037 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1039 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1040 substream_parity_present[substr]) {
1041 uint8_t parity, checksum;
1043 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1044 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1045 av_log(m->avctx, AV_LOG_ERROR,
1046 "Substream %d parity check failed.\n", substr);
1048 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
1049 if (checksum != get_bits(&gb, 8))
1050 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
1053 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1054 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
1060 buf += substream_data_len[substr];
1063 rematrix_channels(m, m->max_decoded_substream);
1065 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1071 m->params_valid = 0;
1075 #if CONFIG_MLP_DECODER
1076 AVCodec mlp_decoder = {
1080 sizeof(MLPDecodeContext),
1085 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1087 #endif /* CONFIG_MLP_DECODER */
1089 #if CONFIG_TRUEHD_DECODER
1090 AVCodec truehd_decoder = {
1094 sizeof(MLPDecodeContext),
1099 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1101 #endif /* CONFIG_TRUEHD_DECODER */