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 m->filter_changed[channel][filter]++;
451 order = get_bits(gbp, 4);
452 if (order > max_order) {
453 av_log(m->avctx, AV_LOG_ERROR,
454 "%cIR filter order %d is greater than maximum %d.\n",
455 fchar, order, max_order);
461 int coeff_bits, coeff_shift;
463 fp->shift = get_bits(gbp, 4);
465 coeff_bits = get_bits(gbp, 5);
466 coeff_shift = get_bits(gbp, 3);
467 if (coeff_bits < 1 || coeff_bits > 16) {
468 av_log(m->avctx, AV_LOG_ERROR,
469 "%cIR filter coeff_bits must be between 1 and 16.\n",
473 if (coeff_bits + coeff_shift > 16) {
474 av_log(m->avctx, AV_LOG_ERROR,
475 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
480 for (i = 0; i < order; i++)
481 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
483 if (get_bits1(gbp)) {
484 int state_bits, state_shift;
487 av_log(m->avctx, AV_LOG_ERROR,
488 "FIR filter has state data specified.\n");
492 state_bits = get_bits(gbp, 4);
493 state_shift = get_bits(gbp, 4);
495 /* TODO: Check validity of state data. */
497 for (i = 0; i < order; i++)
498 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
505 /** Read parameters for primitive matrices. */
507 static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
509 unsigned int mat, ch;
511 s->num_primitive_matrices = get_bits(gbp, 4);
514 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
515 int frac_bits, max_chan;
516 s->matrix_out_ch[mat] = get_bits(gbp, 4);
517 frac_bits = get_bits(gbp, 4);
518 s->lsb_bypass [mat] = get_bits1(gbp);
520 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
521 av_log(m->avctx, AV_LOG_ERROR,
522 "Invalid channel %d specified as output from matrix.\n",
523 s->matrix_out_ch[mat]);
526 if (frac_bits > 14) {
527 av_log(m->avctx, AV_LOG_ERROR,
528 "Too many fractional bits specified.\n");
532 max_chan = s->max_matrix_channel;
536 for (ch = 0; ch <= max_chan; ch++) {
539 coeff_val = get_sbits(gbp, frac_bits + 2);
541 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
545 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
547 s->matrix_noise_shift[mat] = 0;
553 /** Read channel parameters. */
555 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
556 GetBitContext *gbp, unsigned int ch)
558 ChannelParams *cp = &m->channel_params[ch];
559 FilterParams *fir = &cp->filter_params[FIR];
560 FilterParams *iir = &cp->filter_params[IIR];
561 SubStream *s = &m->substream[substr];
563 if (s->param_presence_flags & PARAM_FIR)
565 if (read_filter_params(m, gbp, ch, FIR) < 0)
568 if (s->param_presence_flags & PARAM_IIR)
570 if (read_filter_params(m, gbp, ch, IIR) < 0)
573 if (fir->order + iir->order > 8) {
574 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
578 if (fir->order && iir->order &&
579 fir->shift != iir->shift) {
580 av_log(m->avctx, AV_LOG_ERROR,
581 "FIR and IIR filters must use the same precision.\n");
584 /* The FIR and IIR filters must have the same precision.
585 * To simplify the filtering code, only the precision of the
586 * FIR filter is considered. If only the IIR filter is employed,
587 * the FIR filter precision is set to that of the IIR filter, so
588 * that the filtering code can use it. */
589 if (!fir->order && iir->order)
590 fir->shift = iir->shift;
592 if (s->param_presence_flags & PARAM_HUFFOFFSET)
594 cp->huff_offset = get_sbits(gbp, 15);
596 cp->codebook = get_bits(gbp, 2);
597 cp->huff_lsbs = get_bits(gbp, 5);
599 if (cp->huff_lsbs > 24) {
600 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
604 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
609 /** Read decoding parameters that change more often than those in the restart
612 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
615 SubStream *s = &m->substream[substr];
618 if (s->param_presence_flags & PARAM_PRESENCE)
620 s->param_presence_flags = get_bits(gbp, 8);
622 if (s->param_presence_flags & PARAM_BLOCKSIZE)
623 if (get_bits1(gbp)) {
624 s->blocksize = get_bits(gbp, 9);
625 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
626 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
632 if (s->param_presence_flags & PARAM_MATRIX)
633 if (get_bits1(gbp)) {
634 if (read_matrix_params(m, s, gbp) < 0)
638 if (s->param_presence_flags & PARAM_OUTSHIFT)
640 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
641 s->output_shift[ch] = get_sbits(gbp, 4);
644 if (s->param_presence_flags & PARAM_QUANTSTEP)
646 for (ch = 0; ch <= s->max_channel; ch++) {
647 ChannelParams *cp = &m->channel_params[ch];
649 s->quant_step_size[ch] = get_bits(gbp, 4);
651 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
654 for (ch = s->min_channel; ch <= s->max_channel; ch++)
655 if (get_bits1(gbp)) {
656 if (read_channel_params(m, substr, gbp, ch) < 0)
663 #define MSB_MASK(bits) (-1u << bits)
665 /** Generate PCM samples using the prediction filters and residual values
666 * read from the data stream, and update the filter state. */
668 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
669 unsigned int channel)
671 SubStream *s = &m->substream[substr];
672 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
673 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
674 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
675 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
676 unsigned int filter_shift = fir->shift;
677 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
678 int index = MAX_BLOCKSIZE;
681 memcpy(&firbuf[index], fir->state, MAX_FIR_ORDER * sizeof(int32_t));
682 memcpy(&iirbuf[index], iir->state, MAX_IIR_ORDER * sizeof(int32_t));
684 for (i = 0; i < s->blocksize; i++) {
685 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
690 /* TODO: Move this code to DSPContext? */
692 for (order = 0; order < fir->order; order++)
693 accum += (int64_t) firbuf[index + order] * fir->coeff[order];
694 for (order = 0; order < iir->order; order++)
695 accum += (int64_t) iirbuf[index + order] * iir->coeff[order];
697 accum = accum >> filter_shift;
698 result = (accum + residual) & mask;
702 firbuf[index] = result;
703 iirbuf[index] = result - accum;
705 m->sample_buffer[i + s->blockpos][channel] = result;
708 memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
709 memcpy(iir->state, &iirbuf[index], MAX_IIR_ORDER * sizeof(int32_t));
712 /** Read a block of PCM residual data (or actual if no filtering active). */
714 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
717 SubStream *s = &m->substream[substr];
718 unsigned int i, ch, expected_stream_pos = 0;
720 if (s->data_check_present) {
721 expected_stream_pos = get_bits_count(gbp);
722 expected_stream_pos += get_bits(gbp, 16);
723 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
724 "we have not tested yet. %s\n", sample_message);
727 if (s->blockpos + s->blocksize > m->access_unit_size) {
728 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
732 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
733 s->blocksize * sizeof(m->bypassed_lsbs[0]));
735 for (i = 0; i < s->blocksize; i++) {
736 if (read_huff_channels(m, gbp, substr, i) < 0)
740 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
741 filter_channel(m, substr, ch);
744 s->blockpos += s->blocksize;
746 if (s->data_check_present) {
747 if (get_bits_count(gbp) != expected_stream_pos)
748 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
755 /** Data table used for TrueHD noise generation function. */
757 static const int8_t noise_table[256] = {
758 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
759 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
760 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
761 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
762 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
763 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
764 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
765 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
766 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
767 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
768 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
769 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
770 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
771 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
772 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
773 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
776 /** Noise generation functions.
777 * I'm not sure what these are for - they seem to be some kind of pseudorandom
778 * sequence generators, used to generate noise data which is used when the
779 * channels are rematrixed. I'm not sure if they provide a practical benefit
780 * to compression, or just obfuscate the decoder. Are they for some kind of
783 /** Generate two channels of noise, used in the matrix when
784 * restart sync word == 0x31ea. */
786 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
788 SubStream *s = &m->substream[substr];
790 uint32_t seed = s->noisegen_seed;
791 unsigned int maxchan = s->max_matrix_channel;
793 for (i = 0; i < s->blockpos; i++) {
794 uint16_t seed_shr7 = seed >> 7;
795 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
796 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
798 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
801 s->noisegen_seed = seed;
804 /** Generate a block of noise, used when restart sync word == 0x31eb. */
806 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
808 SubStream *s = &m->substream[substr];
810 uint32_t seed = s->noisegen_seed;
812 for (i = 0; i < m->access_unit_size_pow2; i++) {
813 uint8_t seed_shr15 = seed >> 15;
814 m->noise_buffer[i] = noise_table[seed_shr15];
815 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
818 s->noisegen_seed = seed;
822 /** Apply the channel matrices in turn to reconstruct the original audio
825 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
827 SubStream *s = &m->substream[substr];
828 unsigned int mat, src_ch, i;
829 unsigned int maxchan;
831 maxchan = s->max_matrix_channel;
832 if (!s->noise_type) {
833 generate_2_noise_channels(m, substr);
836 fill_noise_buffer(m, substr);
839 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
840 int matrix_noise_shift = s->matrix_noise_shift[mat];
841 unsigned int dest_ch = s->matrix_out_ch[mat];
842 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
844 /* TODO: DSPContext? */
846 for (i = 0; i < s->blockpos; i++) {
848 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
849 accum += (int64_t)m->sample_buffer[i][src_ch]
850 * s->matrix_coeff[mat][src_ch];
852 if (matrix_noise_shift) {
853 uint32_t index = s->num_primitive_matrices - mat;
854 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
855 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
857 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
858 + m->bypassed_lsbs[i][mat];
863 /** Write the audio data into the output buffer. */
865 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
866 uint8_t *data, unsigned int *data_size, int is32)
868 SubStream *s = &m->substream[substr];
869 unsigned int i, out_ch = 0;
870 int32_t *data_32 = (int32_t*) data;
871 int16_t *data_16 = (int16_t*) data;
873 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
876 for (i = 0; i < s->blockpos; i++) {
877 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
878 int mat_ch = s->ch_assign[out_ch];
879 int32_t sample = m->sample_buffer[i][mat_ch]
880 << s->output_shift[mat_ch];
881 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
882 if (is32) *data_32++ = sample << 8;
883 else *data_16++ = sample >> 8;
887 *data_size = i * out_ch * (is32 ? 4 : 2);
892 static int output_data(MLPDecodeContext *m, unsigned int substr,
893 uint8_t *data, unsigned int *data_size)
895 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
896 return output_data_internal(m, substr, data, data_size, 1);
898 return output_data_internal(m, substr, data, data_size, 0);
902 /** Read an access unit from the stream.
903 * Returns < 0 on error, 0 if not enough data is present in the input stream
904 * otherwise returns the number of bytes consumed. */
906 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
909 const uint8_t *buf = avpkt->data;
910 int buf_size = avpkt->size;
911 MLPDecodeContext *m = avctx->priv_data;
913 unsigned int length, substr;
914 unsigned int substream_start;
915 unsigned int header_size = 4;
916 unsigned int substr_header_size = 0;
917 uint8_t substream_parity_present[MAX_SUBSTREAMS];
918 uint16_t substream_data_len[MAX_SUBSTREAMS];
924 length = (AV_RB16(buf) & 0xfff) * 2;
926 if (length > buf_size)
929 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
931 m->is_major_sync_unit = 0;
932 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
933 if (read_major_sync(m, &gb) < 0)
935 m->is_major_sync_unit = 1;
939 if (!m->params_valid) {
940 av_log(m->avctx, AV_LOG_WARNING,
941 "Stream parameters not seen; skipping frame.\n");
948 for (substr = 0; substr < m->num_substreams; substr++) {
949 int extraword_present, checkdata_present, end, nonrestart_substr;
951 extraword_present = get_bits1(&gb);
952 nonrestart_substr = get_bits1(&gb);
953 checkdata_present = get_bits1(&gb);
956 end = get_bits(&gb, 12) * 2;
958 substr_header_size += 2;
960 if (extraword_present) {
961 if (m->avctx->codec_id == CODEC_ID_MLP) {
962 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
966 substr_header_size += 2;
969 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
970 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
974 if (end + header_size + substr_header_size > length) {
975 av_log(m->avctx, AV_LOG_ERROR,
976 "Indicated length of substream %d data goes off end of "
977 "packet.\n", substr);
978 end = length - header_size - substr_header_size;
981 if (end < substream_start) {
982 av_log(avctx, AV_LOG_ERROR,
983 "Indicated end offset of substream %d data "
984 "is smaller than calculated start offset.\n",
989 if (substr > m->max_decoded_substream)
992 substream_parity_present[substr] = checkdata_present;
993 substream_data_len[substr] = end - substream_start;
994 substream_start = end;
997 parity_bits = ff_mlp_calculate_parity(buf, 4);
998 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1000 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1001 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1005 buf += header_size + substr_header_size;
1007 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1008 SubStream *s = &m->substream[substr];
1009 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1011 m->matrix_changed = 0;
1012 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1018 if (get_bits1(&gb)) {
1019 if (get_bits1(&gb)) {
1020 /* A restart header should be present. */
1021 if (read_restart_header(m, &gb, buf, substr) < 0)
1023 s->restart_seen = 1;
1026 if (!s->restart_seen) {
1030 if (read_decoding_params(m, &gb, substr) < 0)
1034 if (m->matrix_changed > 1) {
1035 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
1038 for (ch = 0; ch < s->max_channel; ch++)
1039 if (m->filter_changed[ch][FIR] > 1 ||
1040 m->filter_changed[ch][IIR] > 1) {
1041 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
1045 if (!s->restart_seen) {
1049 if (read_block_data(m, &gb, substr) < 0)
1052 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1053 goto substream_length_mismatch;
1055 } while (!get_bits1(&gb));
1057 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1058 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1061 if (get_bits(&gb, 16) != 0xD234)
1064 shorten_by = get_bits(&gb, 16);
1065 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1066 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1067 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1070 if (substr == m->max_decoded_substream)
1071 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1073 if (substream_parity_present[substr]) {
1074 uint8_t parity, checksum;
1076 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1077 goto substream_length_mismatch;
1079 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1080 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1082 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1083 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1084 if ( get_bits(&gb, 8) != checksum)
1085 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1087 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1088 goto substream_length_mismatch;
1092 if (!s->restart_seen) {
1093 av_log(m->avctx, AV_LOG_ERROR,
1094 "No restart header present in substream %d.\n", substr);
1097 buf += substream_data_len[substr];
1100 rematrix_channels(m, m->max_decoded_substream);
1102 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1107 substream_length_mismatch:
1108 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1112 m->params_valid = 0;
1116 #if CONFIG_MLP_DECODER
1117 AVCodec mlp_decoder = {
1121 sizeof(MLPDecodeContext),
1126 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1128 #endif /* CONFIG_MLP_DECODER */
1130 #if CONFIG_TRUEHD_DECODER
1131 AVCodec truehd_decoder = {
1135 sizeof(MLPDecodeContext),
1140 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1142 #endif /* CONFIG_TRUEHD_DECODER */