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
31 #include "libavutil/intreadwrite.h"
33 #include "libavutil/crc.h"
35 #include "mlp_parser.h"
38 /** number of bits used for VLC lookup - longest Huffman code is 9 */
42 static const char* sample_message =
43 "Please file a bug report following the instructions at "
44 "http://ffmpeg.org/bugreports.html and include "
45 "a sample of this file.";
47 typedef struct SubStream {
48 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
52 /** restart header data */
53 //! The type of noise to be used in the rematrix stage.
56 //! The index of the first channel coded in this substream.
58 //! The index of the last channel coded in this substream.
60 //! The number of channels input into the rematrix stage.
61 uint8_t max_matrix_channel;
62 //! For each channel output by the matrix, the output channel to map it to
63 uint8_t ch_assign[MAX_CHANNELS];
65 //! The left shift applied to random noise in 0x31ea substreams.
67 //! The current seed value for the pseudorandom noise generator(s).
68 uint32_t noisegen_seed;
70 //! Set if the substream contains extra info to check the size of VLC blocks.
71 uint8_t data_check_present;
73 //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
74 uint8_t param_presence_flags;
75 #define PARAM_BLOCKSIZE (1 << 7)
76 #define PARAM_MATRIX (1 << 6)
77 #define PARAM_OUTSHIFT (1 << 5)
78 #define PARAM_QUANTSTEP (1 << 4)
79 #define PARAM_FIR (1 << 3)
80 #define PARAM_IIR (1 << 2)
81 #define PARAM_HUFFOFFSET (1 << 1)
82 #define PARAM_PRESENCE (1 << 0)
88 //! Number of matrices to be applied.
89 uint8_t num_primitive_matrices;
91 //! matrix output channel
92 uint8_t matrix_out_ch[MAX_MATRICES];
94 //! Whether the LSBs of the matrix output are encoded in the bitstream.
95 uint8_t lsb_bypass[MAX_MATRICES];
96 //! Matrix coefficients, stored as 2.14 fixed point.
97 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
98 //! Left shift to apply to noise values in 0x31eb substreams.
99 uint8_t matrix_noise_shift[MAX_MATRICES];
102 //! Left shift to apply to Huffman-decoded residuals.
103 uint8_t quant_step_size[MAX_CHANNELS];
105 //! number of PCM samples in current audio block
107 //! Number of PCM samples decoded so far in this frame.
110 //! Left shift to apply to decoded PCM values to get final 24-bit output.
111 int8_t output_shift[MAX_CHANNELS];
113 //! Running XOR of all output samples.
114 int32_t lossless_check_data;
118 typedef struct MLPDecodeContext {
119 AVCodecContext *avctx;
121 //! Current access unit being read has a major sync.
122 int is_major_sync_unit;
124 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
125 uint8_t params_valid;
127 //! Number of substreams contained within this stream.
128 uint8_t num_substreams;
130 //! Index of the last substream to decode - further substreams are skipped.
131 uint8_t max_decoded_substream;
133 //! number of PCM samples contained in each frame
134 int access_unit_size;
135 //! next power of two above the number of samples in each frame
136 int access_unit_size_pow2;
138 SubStream substream[MAX_SUBSTREAMS];
140 ChannelParams channel_params[MAX_CHANNELS];
143 int filter_changed[MAX_CHANNELS][NUM_FILTERS];
145 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
146 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
147 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
152 static VLC huff_vlc[3];
154 /** Initialize static data, constant between all invocations of the codec. */
156 static av_cold void init_static(void)
158 if (!huff_vlc[0].bits) {
159 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
160 &ff_mlp_huffman_tables[0][0][1], 2, 1,
161 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
162 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
163 &ff_mlp_huffman_tables[1][0][1], 2, 1,
164 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
165 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
166 &ff_mlp_huffman_tables[2][0][1], 2, 1,
167 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
173 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
174 unsigned int substr, unsigned int ch)
176 ChannelParams *cp = &m->channel_params[ch];
177 SubStream *s = &m->substream[substr];
178 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
179 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
180 int32_t sign_huff_offset = cp->huff_offset;
182 if (cp->codebook > 0)
183 sign_huff_offset -= 7 << lsb_bits;
186 sign_huff_offset -= 1 << sign_shift;
188 return sign_huff_offset;
191 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
194 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
195 unsigned int substr, unsigned int pos)
197 SubStream *s = &m->substream[substr];
198 unsigned int mat, channel;
200 for (mat = 0; mat < s->num_primitive_matrices; mat++)
201 if (s->lsb_bypass[mat])
202 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
204 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
205 ChannelParams *cp = &m->channel_params[channel];
206 int codebook = cp->codebook;
207 int quant_step_size = s->quant_step_size[channel];
208 int lsb_bits = cp->huff_lsbs - quant_step_size;
212 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
213 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
219 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
221 result += cp->sign_huff_offset;
222 result <<= quant_step_size;
224 m->sample_buffer[pos + s->blockpos][channel] = result;
230 static av_cold int mlp_decode_init(AVCodecContext *avctx)
232 MLPDecodeContext *m = avctx->priv_data;
237 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
238 m->substream[substr].lossless_check_data = 0xffffffff;
239 dsputil_init(&m->dsp, avctx);
244 /** Read a major sync info header - contains high level information about
245 * the stream - sample rate, channel arrangement etc. Most of this
246 * information is not actually necessary for decoding, only for playback.
249 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
254 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
257 if (mh.group1_bits == 0) {
258 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
261 if (mh.group2_bits > mh.group1_bits) {
262 av_log(m->avctx, AV_LOG_ERROR,
263 "Channel group 2 cannot have more bits per sample than group 1.\n");
267 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
268 av_log(m->avctx, AV_LOG_ERROR,
269 "Channel groups with differing sample rates are not currently supported.\n");
273 if (mh.group1_samplerate == 0) {
274 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
277 if (mh.group1_samplerate > MAX_SAMPLERATE) {
278 av_log(m->avctx, AV_LOG_ERROR,
279 "Sampling rate %d is greater than the supported maximum (%d).\n",
280 mh.group1_samplerate, MAX_SAMPLERATE);
283 if (mh.access_unit_size > MAX_BLOCKSIZE) {
284 av_log(m->avctx, AV_LOG_ERROR,
285 "Block size %d is greater than the supported maximum (%d).\n",
286 mh.access_unit_size, MAX_BLOCKSIZE);
289 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
290 av_log(m->avctx, AV_LOG_ERROR,
291 "Block size pow2 %d is greater than the supported maximum (%d).\n",
292 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
296 if (mh.num_substreams == 0)
298 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
299 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
302 if (mh.num_substreams > MAX_SUBSTREAMS) {
303 av_log(m->avctx, AV_LOG_ERROR,
304 "Number of substreams %d is larger than the maximum supported "
305 "by the decoder. %s\n", mh.num_substreams, sample_message);
309 m->access_unit_size = mh.access_unit_size;
310 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
312 m->num_substreams = mh.num_substreams;
313 m->max_decoded_substream = m->num_substreams - 1;
315 m->avctx->sample_rate = mh.group1_samplerate;
316 m->avctx->frame_size = mh.access_unit_size;
318 m->avctx->bits_per_raw_sample = mh.group1_bits;
319 if (mh.group1_bits > 16)
320 m->avctx->sample_fmt = SAMPLE_FMT_S32;
322 m->avctx->sample_fmt = SAMPLE_FMT_S16;
325 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
326 m->substream[substr].restart_seen = 0;
331 /** Read a restart header from a block in a substream. This contains parameters
332 * required to decode the audio that do not change very often. Generally
333 * (always) present only in blocks following a major sync. */
335 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
336 const uint8_t *buf, unsigned int substr)
338 SubStream *s = &m->substream[substr];
342 uint8_t lossless_check;
343 int start_count = get_bits_count(gbp);
344 const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
345 ? MAX_MATRIX_CHANNEL_MLP
346 : MAX_MATRIX_CHANNEL_TRUEHD;
348 sync_word = get_bits(gbp, 13);
350 if (sync_word != 0x31ea >> 1) {
351 av_log(m->avctx, AV_LOG_ERROR,
352 "restart header sync incorrect (got 0x%04x)\n", sync_word);
356 s->noise_type = get_bits1(gbp);
358 if (m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) {
359 av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
363 skip_bits(gbp, 16); /* Output timestamp */
365 s->min_channel = get_bits(gbp, 4);
366 s->max_channel = get_bits(gbp, 4);
367 s->max_matrix_channel = get_bits(gbp, 4);
369 if (s->max_matrix_channel > max_matrix_channel) {
370 av_log(m->avctx, AV_LOG_ERROR,
371 "Max matrix channel cannot be greater than %d.\n",
376 if (s->max_channel != s->max_matrix_channel) {
377 av_log(m->avctx, AV_LOG_ERROR,
378 "Max channel must be equal max matrix channel.\n");
382 /* This should happen for TrueHD streams with >6 channels and MLP's noise
383 * type. It is not yet known if this is allowed. */
384 if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
385 av_log(m->avctx, AV_LOG_ERROR,
386 "Number of channels %d is larger than the maximum supported "
387 "by the decoder. %s\n", s->max_channel+2, sample_message);
391 if (s->min_channel > s->max_channel) {
392 av_log(m->avctx, AV_LOG_ERROR,
393 "Substream min channel cannot be greater than max channel.\n");
397 if (m->avctx->request_channels > 0
398 && s->max_channel + 1 >= m->avctx->request_channels
399 && substr < m->max_decoded_substream) {
400 av_log(m->avctx, AV_LOG_DEBUG,
401 "Extracting %d channel downmix from substream %d. "
402 "Further substreams will be skipped.\n",
403 s->max_channel + 1, substr);
404 m->max_decoded_substream = substr;
407 s->noise_shift = get_bits(gbp, 4);
408 s->noisegen_seed = get_bits(gbp, 23);
412 s->data_check_present = get_bits1(gbp);
413 lossless_check = get_bits(gbp, 8);
414 if (substr == m->max_decoded_substream
415 && s->lossless_check_data != 0xffffffff) {
416 tmp = xor_32_to_8(s->lossless_check_data);
417 if (tmp != lossless_check)
418 av_log(m->avctx, AV_LOG_WARNING,
419 "Lossless check failed - expected %02x, calculated %02x.\n",
420 lossless_check, tmp);
425 memset(s->ch_assign, 0, sizeof(s->ch_assign));
427 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
428 int ch_assign = get_bits(gbp, 6);
429 if (ch_assign > s->max_matrix_channel) {
430 av_log(m->avctx, AV_LOG_ERROR,
431 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
432 ch, ch_assign, sample_message);
435 s->ch_assign[ch_assign] = ch;
438 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
440 if (checksum != get_bits(gbp, 8))
441 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
443 /* Set default decoding parameters. */
444 s->param_presence_flags = 0xff;
445 s->num_primitive_matrices = 0;
447 s->lossless_check_data = 0;
449 memset(s->output_shift , 0, sizeof(s->output_shift ));
450 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
452 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
453 ChannelParams *cp = &m->channel_params[ch];
454 cp->filter_params[FIR].order = 0;
455 cp->filter_params[IIR].order = 0;
456 cp->filter_params[FIR].shift = 0;
457 cp->filter_params[IIR].shift = 0;
459 /* Default audio coding is 24-bit raw PCM. */
461 cp->sign_huff_offset = (-1) << 23;
466 if (substr == m->max_decoded_substream)
467 m->avctx->channels = s->max_matrix_channel + 1;
472 /** Read parameters for one of the prediction filters. */
474 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
475 unsigned int channel, unsigned int filter)
477 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
478 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
479 const char fchar = filter ? 'I' : 'F';
482 // Filter is 0 for FIR, 1 for IIR.
485 if (m->filter_changed[channel][filter]++ > 1) {
486 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
490 order = get_bits(gbp, 4);
491 if (order > max_order) {
492 av_log(m->avctx, AV_LOG_ERROR,
493 "%cIR filter order %d is greater than maximum %d.\n",
494 fchar, order, max_order);
500 int32_t *fcoeff = m->channel_params[channel].coeff[filter];
501 int coeff_bits, coeff_shift;
503 fp->shift = get_bits(gbp, 4);
505 coeff_bits = get_bits(gbp, 5);
506 coeff_shift = get_bits(gbp, 3);
507 if (coeff_bits < 1 || coeff_bits > 16) {
508 av_log(m->avctx, AV_LOG_ERROR,
509 "%cIR filter coeff_bits must be between 1 and 16.\n",
513 if (coeff_bits + coeff_shift > 16) {
514 av_log(m->avctx, AV_LOG_ERROR,
515 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
520 for (i = 0; i < order; i++)
521 fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
523 if (get_bits1(gbp)) {
524 int state_bits, state_shift;
527 av_log(m->avctx, AV_LOG_ERROR,
528 "FIR filter has state data specified.\n");
532 state_bits = get_bits(gbp, 4);
533 state_shift = get_bits(gbp, 4);
535 /* TODO: Check validity of state data. */
537 for (i = 0; i < order; i++)
538 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
545 /** Read parameters for primitive matrices. */
547 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
549 SubStream *s = &m->substream[substr];
550 unsigned int mat, ch;
551 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
553 : MAX_MATRICES_TRUEHD;
555 if (m->matrix_changed++ > 1) {
556 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
560 s->num_primitive_matrices = get_bits(gbp, 4);
562 if (s->num_primitive_matrices > max_primitive_matrices) {
563 av_log(m->avctx, AV_LOG_ERROR,
564 "Number of primitive matrices cannot be greater than %d.\n",
565 max_primitive_matrices);
569 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
570 int frac_bits, max_chan;
571 s->matrix_out_ch[mat] = get_bits(gbp, 4);
572 frac_bits = get_bits(gbp, 4);
573 s->lsb_bypass [mat] = get_bits1(gbp);
575 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
576 av_log(m->avctx, AV_LOG_ERROR,
577 "Invalid channel %d specified as output from matrix.\n",
578 s->matrix_out_ch[mat]);
581 if (frac_bits > 14) {
582 av_log(m->avctx, AV_LOG_ERROR,
583 "Too many fractional bits specified.\n");
587 max_chan = s->max_matrix_channel;
591 for (ch = 0; ch <= max_chan; ch++) {
594 coeff_val = get_sbits(gbp, frac_bits + 2);
596 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
600 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
602 s->matrix_noise_shift[mat] = 0;
608 /** Read channel parameters. */
610 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
611 GetBitContext *gbp, unsigned int ch)
613 ChannelParams *cp = &m->channel_params[ch];
614 FilterParams *fir = &cp->filter_params[FIR];
615 FilterParams *iir = &cp->filter_params[IIR];
616 SubStream *s = &m->substream[substr];
618 if (s->param_presence_flags & PARAM_FIR)
620 if (read_filter_params(m, gbp, ch, FIR) < 0)
623 if (s->param_presence_flags & PARAM_IIR)
625 if (read_filter_params(m, gbp, ch, IIR) < 0)
628 if (fir->order + iir->order > 8) {
629 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
633 if (fir->order && iir->order &&
634 fir->shift != iir->shift) {
635 av_log(m->avctx, AV_LOG_ERROR,
636 "FIR and IIR filters must use the same precision.\n");
639 /* The FIR and IIR filters must have the same precision.
640 * To simplify the filtering code, only the precision of the
641 * FIR filter is considered. If only the IIR filter is employed,
642 * the FIR filter precision is set to that of the IIR filter, so
643 * that the filtering code can use it. */
644 if (!fir->order && iir->order)
645 fir->shift = iir->shift;
647 if (s->param_presence_flags & PARAM_HUFFOFFSET)
649 cp->huff_offset = get_sbits(gbp, 15);
651 cp->codebook = get_bits(gbp, 2);
652 cp->huff_lsbs = get_bits(gbp, 5);
654 if (cp->huff_lsbs > 24) {
655 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
659 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
664 /** Read decoding parameters that change more often than those in the restart
667 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
670 SubStream *s = &m->substream[substr];
673 if (s->param_presence_flags & PARAM_PRESENCE)
675 s->param_presence_flags = get_bits(gbp, 8);
677 if (s->param_presence_flags & PARAM_BLOCKSIZE)
678 if (get_bits1(gbp)) {
679 s->blocksize = get_bits(gbp, 9);
680 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
681 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
687 if (s->param_presence_flags & PARAM_MATRIX)
689 if (read_matrix_params(m, substr, gbp) < 0)
692 if (s->param_presence_flags & PARAM_OUTSHIFT)
694 for (ch = 0; ch <= s->max_matrix_channel; ch++)
695 s->output_shift[ch] = get_sbits(gbp, 4);
697 if (s->param_presence_flags & PARAM_QUANTSTEP)
699 for (ch = 0; ch <= s->max_channel; ch++) {
700 ChannelParams *cp = &m->channel_params[ch];
702 s->quant_step_size[ch] = get_bits(gbp, 4);
704 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
707 for (ch = s->min_channel; ch <= s->max_channel; ch++)
709 if (read_channel_params(m, substr, gbp, ch) < 0)
715 #define MSB_MASK(bits) (-1u << bits)
717 /** Generate PCM samples using the prediction filters and residual values
718 * read from the data stream, and update the filter state. */
720 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
721 unsigned int channel)
723 SubStream *s = &m->substream[substr];
724 const int32_t *fircoeff = m->channel_params[channel].coeff[FIR];
725 int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
726 int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
727 int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
728 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
729 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
730 unsigned int filter_shift = fir->shift;
731 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
733 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
734 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
736 m->dsp.mlp_filter_channel(firbuf, fircoeff,
737 fir->order, iir->order,
738 filter_shift, mask, s->blocksize,
739 &m->sample_buffer[s->blockpos][channel]);
741 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
742 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
745 /** Read a block of PCM residual data (or actual if no filtering active). */
747 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
750 SubStream *s = &m->substream[substr];
751 unsigned int i, ch, expected_stream_pos = 0;
753 if (s->data_check_present) {
754 expected_stream_pos = get_bits_count(gbp);
755 expected_stream_pos += get_bits(gbp, 16);
756 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
757 "we have not tested yet. %s\n", sample_message);
760 if (s->blockpos + s->blocksize > m->access_unit_size) {
761 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
765 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
766 s->blocksize * sizeof(m->bypassed_lsbs[0]));
768 for (i = 0; i < s->blocksize; i++)
769 if (read_huff_channels(m, gbp, substr, i) < 0)
772 for (ch = s->min_channel; ch <= s->max_channel; ch++)
773 filter_channel(m, substr, ch);
775 s->blockpos += s->blocksize;
777 if (s->data_check_present) {
778 if (get_bits_count(gbp) != expected_stream_pos)
779 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
786 /** Data table used for TrueHD noise generation function. */
788 static const int8_t noise_table[256] = {
789 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
790 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
791 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
792 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
793 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
794 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
795 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
796 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
797 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
798 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
799 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
800 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
801 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
802 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
803 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
804 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
807 /** Noise generation functions.
808 * I'm not sure what these are for - they seem to be some kind of pseudorandom
809 * sequence generators, used to generate noise data which is used when the
810 * channels are rematrixed. I'm not sure if they provide a practical benefit
811 * to compression, or just obfuscate the decoder. Are they for some kind of
814 /** Generate two channels of noise, used in the matrix when
815 * restart sync word == 0x31ea. */
817 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
819 SubStream *s = &m->substream[substr];
821 uint32_t seed = s->noisegen_seed;
822 unsigned int maxchan = s->max_matrix_channel;
824 for (i = 0; i < s->blockpos; i++) {
825 uint16_t seed_shr7 = seed >> 7;
826 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
827 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
829 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
832 s->noisegen_seed = seed;
835 /** Generate a block of noise, used when restart sync word == 0x31eb. */
837 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
839 SubStream *s = &m->substream[substr];
841 uint32_t seed = s->noisegen_seed;
843 for (i = 0; i < m->access_unit_size_pow2; i++) {
844 uint8_t seed_shr15 = seed >> 15;
845 m->noise_buffer[i] = noise_table[seed_shr15];
846 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
849 s->noisegen_seed = seed;
853 /** Apply the channel matrices in turn to reconstruct the original audio
856 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
858 SubStream *s = &m->substream[substr];
859 unsigned int mat, src_ch, i;
860 unsigned int maxchan;
862 maxchan = s->max_matrix_channel;
863 if (!s->noise_type) {
864 generate_2_noise_channels(m, substr);
867 fill_noise_buffer(m, substr);
870 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
871 int matrix_noise_shift = s->matrix_noise_shift[mat];
872 unsigned int dest_ch = s->matrix_out_ch[mat];
873 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
874 int32_t *coeffs = s->matrix_coeff[mat];
875 int index = s->num_primitive_matrices - mat;
876 int index2 = 2 * index + 1;
878 /* TODO: DSPContext? */
880 for (i = 0; i < s->blockpos; i++) {
881 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
882 int32_t *samples = m->sample_buffer[i];
885 for (src_ch = 0; src_ch <= maxchan; src_ch++)
886 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
888 if (matrix_noise_shift) {
889 index &= m->access_unit_size_pow2 - 1;
890 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
894 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
899 /** Write the audio data into the output buffer. */
901 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
902 uint8_t *data, unsigned int *data_size, int is32)
904 SubStream *s = &m->substream[substr];
905 unsigned int i, out_ch = 0;
906 int32_t *data_32 = (int32_t*) data;
907 int16_t *data_16 = (int16_t*) data;
909 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
912 for (i = 0; i < s->blockpos; i++) {
913 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
914 int mat_ch = s->ch_assign[out_ch];
915 int32_t sample = m->sample_buffer[i][mat_ch]
916 << s->output_shift[mat_ch];
917 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
918 if (is32) *data_32++ = sample << 8;
919 else *data_16++ = sample >> 8;
923 *data_size = i * out_ch * (is32 ? 4 : 2);
928 static int output_data(MLPDecodeContext *m, unsigned int substr,
929 uint8_t *data, unsigned int *data_size)
931 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
932 return output_data_internal(m, substr, data, data_size, 1);
934 return output_data_internal(m, substr, data, data_size, 0);
938 /** Read an access unit from the stream.
939 * Returns < 0 on error, 0 if not enough data is present in the input stream
940 * otherwise returns the number of bytes consumed. */
942 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
945 const uint8_t *buf = avpkt->data;
946 int buf_size = avpkt->size;
947 MLPDecodeContext *m = avctx->priv_data;
949 unsigned int length, substr;
950 unsigned int substream_start;
951 unsigned int header_size = 4;
952 unsigned int substr_header_size = 0;
953 uint8_t substream_parity_present[MAX_SUBSTREAMS];
954 uint16_t substream_data_len[MAX_SUBSTREAMS];
960 length = (AV_RB16(buf) & 0xfff) * 2;
962 if (length < 4 || length > buf_size)
965 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
967 m->is_major_sync_unit = 0;
968 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
969 if (read_major_sync(m, &gb) < 0)
971 m->is_major_sync_unit = 1;
975 if (!m->params_valid) {
976 av_log(m->avctx, AV_LOG_WARNING,
977 "Stream parameters not seen; skipping frame.\n");
984 for (substr = 0; substr < m->num_substreams; substr++) {
985 int extraword_present, checkdata_present, end, nonrestart_substr;
987 extraword_present = get_bits1(&gb);
988 nonrestart_substr = get_bits1(&gb);
989 checkdata_present = get_bits1(&gb);
992 end = get_bits(&gb, 12) * 2;
994 substr_header_size += 2;
996 if (extraword_present) {
997 if (m->avctx->codec_id == CODEC_ID_MLP) {
998 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
1002 substr_header_size += 2;
1005 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1006 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1010 if (end + header_size + substr_header_size > length) {
1011 av_log(m->avctx, AV_LOG_ERROR,
1012 "Indicated length of substream %d data goes off end of "
1013 "packet.\n", substr);
1014 end = length - header_size - substr_header_size;
1017 if (end < substream_start) {
1018 av_log(avctx, AV_LOG_ERROR,
1019 "Indicated end offset of substream %d data "
1020 "is smaller than calculated start offset.\n",
1025 if (substr > m->max_decoded_substream)
1028 substream_parity_present[substr] = checkdata_present;
1029 substream_data_len[substr] = end - substream_start;
1030 substream_start = end;
1033 parity_bits = ff_mlp_calculate_parity(buf, 4);
1034 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1036 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1037 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1041 buf += header_size + substr_header_size;
1043 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1044 SubStream *s = &m->substream[substr];
1045 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1047 m->matrix_changed = 0;
1048 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1052 if (get_bits1(&gb)) {
1053 if (get_bits1(&gb)) {
1054 /* A restart header should be present. */
1055 if (read_restart_header(m, &gb, buf, substr) < 0)
1057 s->restart_seen = 1;
1060 if (!s->restart_seen)
1062 if (read_decoding_params(m, &gb, substr) < 0)
1066 if (!s->restart_seen)
1069 if (read_block_data(m, &gb, substr) < 0)
1072 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1073 goto substream_length_mismatch;
1075 } while (!get_bits1(&gb));
1077 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1079 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1082 if (get_bits(&gb, 16) != 0xD234)
1085 shorten_by = get_bits(&gb, 16);
1086 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1087 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1088 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1091 if (substr == m->max_decoded_substream)
1092 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1095 if (substream_parity_present[substr]) {
1096 uint8_t parity, checksum;
1098 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1099 goto substream_length_mismatch;
1101 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1102 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1104 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1105 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1106 if ( get_bits(&gb, 8) != checksum)
1107 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1110 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1111 goto substream_length_mismatch;
1114 if (!s->restart_seen)
1115 av_log(m->avctx, AV_LOG_ERROR,
1116 "No restart header present in substream %d.\n", substr);
1118 buf += substream_data_len[substr];
1121 rematrix_channels(m, m->max_decoded_substream);
1123 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1128 substream_length_mismatch:
1129 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1133 m->params_valid = 0;
1137 AVCodec mlp_decoder = {
1141 sizeof(MLPDecodeContext),
1146 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
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 */