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];
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];
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);
338 const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
339 ? MAX_MATRIX_CHANNEL_MLP
340 : MAX_MATRIX_CHANNEL_TRUEHD;
342 sync_word = get_bits(gbp, 13);
343 s->noise_type = get_bits1(gbp);
345 if ((m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) ||
346 sync_word != 0x31ea >> 1) {
347 av_log(m->avctx, AV_LOG_ERROR,
348 "restart header sync incorrect (got 0x%04x)\n", sync_word);
352 skip_bits(gbp, 16); /* Output timestamp */
354 s->min_channel = get_bits(gbp, 4);
355 s->max_channel = get_bits(gbp, 4);
356 s->max_matrix_channel = get_bits(gbp, 4);
358 if (s->max_matrix_channel > max_matrix_channel) {
359 av_log(m->avctx, AV_LOG_ERROR,
360 "Max matrix channel cannot be greater than %d.\n",
365 if (s->max_channel != s->max_matrix_channel) {
366 av_log(m->avctx, AV_LOG_ERROR,
367 "Max channel must be equal max matrix channel.\n");
371 if (s->min_channel > s->max_channel) {
372 av_log(m->avctx, AV_LOG_ERROR,
373 "Substream min channel cannot be greater than max channel.\n");
377 if (m->avctx->request_channels > 0
378 && s->max_channel + 1 >= m->avctx->request_channels
379 && substr < m->max_decoded_substream) {
380 av_log(m->avctx, AV_LOG_INFO,
381 "Extracting %d channel downmix from substream %d. "
382 "Further substreams will be skipped.\n",
383 s->max_channel + 1, substr);
384 m->max_decoded_substream = substr;
387 s->noise_shift = get_bits(gbp, 4);
388 s->noisegen_seed = get_bits(gbp, 23);
392 s->data_check_present = get_bits1(gbp);
393 lossless_check = get_bits(gbp, 8);
394 if (substr == m->max_decoded_substream
395 && s->lossless_check_data != 0xffffffff) {
396 tmp = xor_32_to_8(s->lossless_check_data);
397 if (tmp != lossless_check)
398 av_log(m->avctx, AV_LOG_WARNING,
399 "Lossless check failed - expected %02x, calculated %02x.\n",
400 lossless_check, tmp);
405 memset(s->ch_assign, 0, sizeof(s->ch_assign));
407 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
408 int ch_assign = get_bits(gbp, 6);
409 if (ch_assign > s->max_matrix_channel) {
410 av_log(m->avctx, AV_LOG_ERROR,
411 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
412 ch, ch_assign, sample_message);
415 s->ch_assign[ch_assign] = ch;
418 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
420 if (checksum != get_bits(gbp, 8))
421 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
423 /* Set default decoding parameters. */
424 s->param_presence_flags = 0xff;
425 s->num_primitive_matrices = 0;
427 s->lossless_check_data = 0;
429 memset(s->output_shift , 0, sizeof(s->output_shift ));
430 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
432 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
433 ChannelParams *cp = &m->channel_params[ch];
434 cp->filter_params[FIR].order = 0;
435 cp->filter_params[IIR].order = 0;
436 cp->filter_params[FIR].shift = 0;
437 cp->filter_params[IIR].shift = 0;
439 /* Default audio coding is 24-bit raw PCM. */
441 cp->sign_huff_offset = (-1) << 23;
446 if (substr == m->max_decoded_substream)
447 m->avctx->channels = s->max_matrix_channel + 1;
452 /** Read parameters for one of the prediction filters. */
454 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
455 unsigned int channel, unsigned int filter)
457 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
458 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
459 const char fchar = filter ? 'I' : 'F';
462 // Filter is 0 for FIR, 1 for IIR.
465 if (m->filter_changed[channel][filter]++ > 1) {
466 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
470 order = get_bits(gbp, 4);
471 if (order > max_order) {
472 av_log(m->avctx, AV_LOG_ERROR,
473 "%cIR filter order %d is greater than maximum %d.\n",
474 fchar, order, max_order);
480 int coeff_bits, coeff_shift;
482 fp->shift = get_bits(gbp, 4);
484 coeff_bits = get_bits(gbp, 5);
485 coeff_shift = get_bits(gbp, 3);
486 if (coeff_bits < 1 || coeff_bits > 16) {
487 av_log(m->avctx, AV_LOG_ERROR,
488 "%cIR filter coeff_bits must be between 1 and 16.\n",
492 if (coeff_bits + coeff_shift > 16) {
493 av_log(m->avctx, AV_LOG_ERROR,
494 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
499 for (i = 0; i < order; i++)
500 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
502 if (get_bits1(gbp)) {
503 int state_bits, state_shift;
506 av_log(m->avctx, AV_LOG_ERROR,
507 "FIR filter has state data specified.\n");
511 state_bits = get_bits(gbp, 4);
512 state_shift = get_bits(gbp, 4);
514 /* TODO: Check validity of state data. */
516 for (i = 0; i < order; i++)
517 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
524 /** Read parameters for primitive matrices. */
526 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
528 SubStream *s = &m->substream[substr];
529 unsigned int mat, ch;
530 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
532 : MAX_MATRICES_TRUEHD;
534 if (m->matrix_changed++ > 1) {
535 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
539 s->num_primitive_matrices = get_bits(gbp, 4);
541 if (s->num_primitive_matrices > max_primitive_matrices) {
542 av_log(m->avctx, AV_LOG_ERROR,
543 "Number of primitive matrices cannot be greater than %d.\n",
544 max_primitive_matrices);
548 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
549 int frac_bits, max_chan;
550 s->matrix_out_ch[mat] = get_bits(gbp, 4);
551 frac_bits = get_bits(gbp, 4);
552 s->lsb_bypass [mat] = get_bits1(gbp);
554 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
555 av_log(m->avctx, AV_LOG_ERROR,
556 "Invalid channel %d specified as output from matrix.\n",
557 s->matrix_out_ch[mat]);
560 if (frac_bits > 14) {
561 av_log(m->avctx, AV_LOG_ERROR,
562 "Too many fractional bits specified.\n");
566 max_chan = s->max_matrix_channel;
570 for (ch = 0; ch <= max_chan; ch++) {
573 coeff_val = get_sbits(gbp, frac_bits + 2);
575 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
579 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
581 s->matrix_noise_shift[mat] = 0;
587 /** Read channel parameters. */
589 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
590 GetBitContext *gbp, unsigned int ch)
592 ChannelParams *cp = &m->channel_params[ch];
593 FilterParams *fir = &cp->filter_params[FIR];
594 FilterParams *iir = &cp->filter_params[IIR];
595 SubStream *s = &m->substream[substr];
597 if (s->param_presence_flags & PARAM_FIR)
599 if (read_filter_params(m, gbp, ch, FIR) < 0)
602 if (s->param_presence_flags & PARAM_IIR)
604 if (read_filter_params(m, gbp, ch, IIR) < 0)
607 if (fir->order + iir->order > 8) {
608 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
612 if (fir->order && iir->order &&
613 fir->shift != iir->shift) {
614 av_log(m->avctx, AV_LOG_ERROR,
615 "FIR and IIR filters must use the same precision.\n");
618 /* The FIR and IIR filters must have the same precision.
619 * To simplify the filtering code, only the precision of the
620 * FIR filter is considered. If only the IIR filter is employed,
621 * the FIR filter precision is set to that of the IIR filter, so
622 * that the filtering code can use it. */
623 if (!fir->order && iir->order)
624 fir->shift = iir->shift;
626 if (s->param_presence_flags & PARAM_HUFFOFFSET)
628 cp->huff_offset = get_sbits(gbp, 15);
630 cp->codebook = get_bits(gbp, 2);
631 cp->huff_lsbs = get_bits(gbp, 5);
633 if (cp->huff_lsbs > 24) {
634 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
638 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
643 /** Read decoding parameters that change more often than those in the restart
646 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
649 SubStream *s = &m->substream[substr];
652 if (s->param_presence_flags & PARAM_PRESENCE)
654 s->param_presence_flags = get_bits(gbp, 8);
656 if (s->param_presence_flags & PARAM_BLOCKSIZE)
657 if (get_bits1(gbp)) {
658 s->blocksize = get_bits(gbp, 9);
659 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
660 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
666 if (s->param_presence_flags & PARAM_MATRIX)
668 if (read_matrix_params(m, substr, gbp) < 0)
671 if (s->param_presence_flags & PARAM_OUTSHIFT)
673 for (ch = 0; ch <= s->max_matrix_channel; ch++)
674 s->output_shift[ch] = get_sbits(gbp, 4);
676 if (s->param_presence_flags & PARAM_QUANTSTEP)
678 for (ch = 0; ch <= s->max_channel; ch++) {
679 ChannelParams *cp = &m->channel_params[ch];
681 s->quant_step_size[ch] = get_bits(gbp, 4);
683 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
686 for (ch = s->min_channel; ch <= s->max_channel; ch++)
688 if (read_channel_params(m, substr, gbp, ch) < 0)
694 #define MSB_MASK(bits) (-1u << bits)
696 /** Generate PCM samples using the prediction filters and residual values
697 * read from the data stream, and update the filter state. */
699 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
700 unsigned int channel)
702 SubStream *s = &m->substream[substr];
703 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
704 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
705 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
706 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
707 unsigned int filter_shift = fir->shift;
708 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
709 int index = MAX_BLOCKSIZE;
712 memcpy(&firbuf[index], fir->state, MAX_FIR_ORDER * sizeof(int32_t));
713 memcpy(&iirbuf[index], iir->state, MAX_IIR_ORDER * sizeof(int32_t));
715 for (i = 0; i < s->blocksize; i++) {
716 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
721 /* TODO: Move this code to DSPContext? */
723 for (order = 0; order < fir->order; order++)
724 accum += (int64_t) firbuf[index + order] * fir->coeff[order];
725 for (order = 0; order < iir->order; order++)
726 accum += (int64_t) iirbuf[index + order] * iir->coeff[order];
728 accum = accum >> filter_shift;
729 result = (accum + residual) & mask;
733 firbuf[index] = result;
734 iirbuf[index] = result - accum;
736 m->sample_buffer[i + s->blockpos][channel] = result;
739 memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
740 memcpy(iir->state, &iirbuf[index], MAX_IIR_ORDER * sizeof(int32_t));
743 /** Read a block of PCM residual data (or actual if no filtering active). */
745 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
748 SubStream *s = &m->substream[substr];
749 unsigned int i, ch, expected_stream_pos = 0;
751 if (s->data_check_present) {
752 expected_stream_pos = get_bits_count(gbp);
753 expected_stream_pos += get_bits(gbp, 16);
754 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
755 "we have not tested yet. %s\n", sample_message);
758 if (s->blockpos + s->blocksize > m->access_unit_size) {
759 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
763 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
764 s->blocksize * sizeof(m->bypassed_lsbs[0]));
766 for (i = 0; i < s->blocksize; i++)
767 if (read_huff_channels(m, gbp, substr, i) < 0)
770 for (ch = s->min_channel; ch <= s->max_channel; ch++)
771 filter_channel(m, substr, ch);
773 s->blockpos += s->blocksize;
775 if (s->data_check_present) {
776 if (get_bits_count(gbp) != expected_stream_pos)
777 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
784 /** Data table used for TrueHD noise generation function. */
786 static const int8_t noise_table[256] = {
787 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
788 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
789 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
790 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
791 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
792 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
793 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
794 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
795 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
796 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
797 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
798 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
799 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
800 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
801 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
802 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
805 /** Noise generation functions.
806 * I'm not sure what these are for - they seem to be some kind of pseudorandom
807 * sequence generators, used to generate noise data which is used when the
808 * channels are rematrixed. I'm not sure if they provide a practical benefit
809 * to compression, or just obfuscate the decoder. Are they for some kind of
812 /** Generate two channels of noise, used in the matrix when
813 * restart sync word == 0x31ea. */
815 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
817 SubStream *s = &m->substream[substr];
819 uint32_t seed = s->noisegen_seed;
820 unsigned int maxchan = s->max_matrix_channel;
822 for (i = 0; i < s->blockpos; i++) {
823 uint16_t seed_shr7 = seed >> 7;
824 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
825 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
827 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
830 s->noisegen_seed = seed;
833 /** Generate a block of noise, used when restart sync word == 0x31eb. */
835 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
837 SubStream *s = &m->substream[substr];
839 uint32_t seed = s->noisegen_seed;
841 for (i = 0; i < m->access_unit_size_pow2; i++) {
842 uint8_t seed_shr15 = seed >> 15;
843 m->noise_buffer[i] = noise_table[seed_shr15];
844 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
847 s->noisegen_seed = seed;
851 /** Apply the channel matrices in turn to reconstruct the original audio
854 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
856 SubStream *s = &m->substream[substr];
857 unsigned int mat, src_ch, i;
858 unsigned int maxchan;
860 maxchan = s->max_matrix_channel;
861 if (!s->noise_type) {
862 generate_2_noise_channels(m, substr);
865 fill_noise_buffer(m, substr);
868 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
869 int matrix_noise_shift = s->matrix_noise_shift[mat];
870 unsigned int dest_ch = s->matrix_out_ch[mat];
871 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
872 int32_t *coeffs = s->matrix_coeff[mat];
873 int index = s->num_primitive_matrices - mat;
874 int index2 = 2 * index + 1;
876 /* TODO: DSPContext? */
878 for (i = 0; i < s->blockpos; i++) {
879 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
880 int32_t *samples = m->sample_buffer[i];
883 for (src_ch = 0; src_ch <= maxchan; src_ch++)
884 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
886 if (matrix_noise_shift) {
887 index &= m->access_unit_size_pow2 - 1;
888 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
892 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
897 /** Write the audio data into the output buffer. */
899 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
900 uint8_t *data, unsigned int *data_size, int is32)
902 SubStream *s = &m->substream[substr];
903 unsigned int i, out_ch = 0;
904 int32_t *data_32 = (int32_t*) data;
905 int16_t *data_16 = (int16_t*) data;
907 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
910 for (i = 0; i < s->blockpos; i++) {
911 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
912 int mat_ch = s->ch_assign[out_ch];
913 int32_t sample = m->sample_buffer[i][mat_ch]
914 << s->output_shift[mat_ch];
915 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
916 if (is32) *data_32++ = sample << 8;
917 else *data_16++ = sample >> 8;
921 *data_size = i * out_ch * (is32 ? 4 : 2);
926 static int output_data(MLPDecodeContext *m, unsigned int substr,
927 uint8_t *data, unsigned int *data_size)
929 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
930 return output_data_internal(m, substr, data, data_size, 1);
932 return output_data_internal(m, substr, data, data_size, 0);
936 /** Read an access unit from the stream.
937 * Returns < 0 on error, 0 if not enough data is present in the input stream
938 * otherwise returns the number of bytes consumed. */
940 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
943 const uint8_t *buf = avpkt->data;
944 int buf_size = avpkt->size;
945 MLPDecodeContext *m = avctx->priv_data;
947 unsigned int length, substr;
948 unsigned int substream_start;
949 unsigned int header_size = 4;
950 unsigned int substr_header_size = 0;
951 uint8_t substream_parity_present[MAX_SUBSTREAMS];
952 uint16_t substream_data_len[MAX_SUBSTREAMS];
958 length = (AV_RB16(buf) & 0xfff) * 2;
960 if (length > buf_size)
963 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
965 m->is_major_sync_unit = 0;
966 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
967 if (read_major_sync(m, &gb) < 0)
969 m->is_major_sync_unit = 1;
973 if (!m->params_valid) {
974 av_log(m->avctx, AV_LOG_WARNING,
975 "Stream parameters not seen; skipping frame.\n");
982 for (substr = 0; substr < m->num_substreams; substr++) {
983 int extraword_present, checkdata_present, end, nonrestart_substr;
985 extraword_present = get_bits1(&gb);
986 nonrestart_substr = get_bits1(&gb);
987 checkdata_present = get_bits1(&gb);
990 end = get_bits(&gb, 12) * 2;
992 substr_header_size += 2;
994 if (extraword_present) {
995 if (m->avctx->codec_id == CODEC_ID_MLP) {
996 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
1000 substr_header_size += 2;
1003 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1004 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1008 if (end + header_size + substr_header_size > length) {
1009 av_log(m->avctx, AV_LOG_ERROR,
1010 "Indicated length of substream %d data goes off end of "
1011 "packet.\n", substr);
1012 end = length - header_size - substr_header_size;
1015 if (end < substream_start) {
1016 av_log(avctx, AV_LOG_ERROR,
1017 "Indicated end offset of substream %d data "
1018 "is smaller than calculated start offset.\n",
1023 if (substr > m->max_decoded_substream)
1026 substream_parity_present[substr] = checkdata_present;
1027 substream_data_len[substr] = end - substream_start;
1028 substream_start = end;
1031 parity_bits = ff_mlp_calculate_parity(buf, 4);
1032 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1034 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1035 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1039 buf += header_size + substr_header_size;
1041 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1042 SubStream *s = &m->substream[substr];
1043 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1045 m->matrix_changed = 0;
1046 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1050 if (get_bits1(&gb)) {
1051 if (get_bits1(&gb)) {
1052 /* A restart header should be present. */
1053 if (read_restart_header(m, &gb, buf, substr) < 0)
1055 s->restart_seen = 1;
1058 if (!s->restart_seen)
1060 if (read_decoding_params(m, &gb, substr) < 0)
1064 if (!s->restart_seen)
1067 if (read_block_data(m, &gb, substr) < 0)
1070 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1071 goto substream_length_mismatch;
1073 } while (!get_bits1(&gb));
1075 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1077 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1080 if (get_bits(&gb, 16) != 0xD234)
1083 shorten_by = get_bits(&gb, 16);
1084 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1085 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1086 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1089 if (substr == m->max_decoded_substream)
1090 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1093 if (substream_parity_present[substr]) {
1094 uint8_t parity, checksum;
1096 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1097 goto substream_length_mismatch;
1099 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1100 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1102 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1103 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1104 if ( get_bits(&gb, 8) != checksum)
1105 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1108 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1109 goto substream_length_mismatch;
1112 if (!s->restart_seen)
1113 av_log(m->avctx, AV_LOG_ERROR,
1114 "No restart header present in substream %d.\n", substr);
1116 buf += substream_data_len[substr];
1119 rematrix_channels(m, m->max_decoded_substream);
1121 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1126 substream_length_mismatch:
1127 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1131 m->params_valid = 0;
1135 #if CONFIG_MLP_DECODER
1136 AVCodec mlp_decoder = {
1140 sizeof(MLPDecodeContext),
1145 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1147 #endif /* CONFIG_MLP_DECODER */
1149 #if CONFIG_TRUEHD_DECODER
1150 AVCodec truehd_decoder = {
1154 sizeof(MLPDecodeContext),
1159 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1161 #endif /* CONFIG_TRUEHD_DECODER */