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 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
159 &ff_mlp_huffman_tables[0][0][1], 2, 1,
160 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
161 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
162 &ff_mlp_huffman_tables[1][0][1], 2, 1,
163 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
164 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
165 &ff_mlp_huffman_tables[2][0][1], 2, 1,
166 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
171 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
172 unsigned int substr, unsigned int ch)
174 ChannelParams *cp = &m->channel_params[ch];
175 SubStream *s = &m->substream[substr];
176 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
177 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
178 int32_t sign_huff_offset = cp->huff_offset;
180 if (cp->codebook > 0)
181 sign_huff_offset -= 7 << lsb_bits;
184 sign_huff_offset -= 1 << sign_shift;
186 return sign_huff_offset;
189 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
192 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
193 unsigned int substr, unsigned int pos)
195 SubStream *s = &m->substream[substr];
196 unsigned int mat, channel;
198 for (mat = 0; mat < s->num_primitive_matrices; mat++)
199 if (s->lsb_bypass[mat])
200 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
202 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
203 ChannelParams *cp = &m->channel_params[channel];
204 int codebook = cp->codebook;
205 int quant_step_size = s->quant_step_size[channel];
206 int lsb_bits = cp->huff_lsbs - quant_step_size;
210 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
211 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
217 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
219 result += cp->sign_huff_offset;
220 result <<= quant_step_size;
222 m->sample_buffer[pos + s->blockpos][channel] = result;
228 static av_cold int mlp_decode_init(AVCodecContext *avctx)
230 MLPDecodeContext *m = avctx->priv_data;
235 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
236 m->substream[substr].lossless_check_data = 0xffffffff;
237 dsputil_init(&m->dsp, avctx);
242 /** Read a major sync info header - contains high level information about
243 * the stream - sample rate, channel arrangement etc. Most of this
244 * information is not actually necessary for decoding, only for playback.
247 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
252 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
255 if (mh.group1_bits == 0) {
256 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
259 if (mh.group2_bits > mh.group1_bits) {
260 av_log(m->avctx, AV_LOG_ERROR,
261 "Channel group 2 cannot have more bits per sample than group 1.\n");
265 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
266 av_log(m->avctx, AV_LOG_ERROR,
267 "Channel groups with differing sample rates are not currently supported.\n");
271 if (mh.group1_samplerate == 0) {
272 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
275 if (mh.group1_samplerate > MAX_SAMPLERATE) {
276 av_log(m->avctx, AV_LOG_ERROR,
277 "Sampling rate %d is greater than the supported maximum (%d).\n",
278 mh.group1_samplerate, MAX_SAMPLERATE);
281 if (mh.access_unit_size > MAX_BLOCKSIZE) {
282 av_log(m->avctx, AV_LOG_ERROR,
283 "Block size %d is greater than the supported maximum (%d).\n",
284 mh.access_unit_size, MAX_BLOCKSIZE);
287 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
288 av_log(m->avctx, AV_LOG_ERROR,
289 "Block size pow2 %d is greater than the supported maximum (%d).\n",
290 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
294 if (mh.num_substreams == 0)
296 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
297 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
300 if (mh.num_substreams > MAX_SUBSTREAMS) {
301 av_log(m->avctx, AV_LOG_ERROR,
302 "Number of substreams %d is larger than the maximum supported "
303 "by the decoder. %s\n", mh.num_substreams, sample_message);
307 m->access_unit_size = mh.access_unit_size;
308 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
310 m->num_substreams = mh.num_substreams;
311 m->max_decoded_substream = m->num_substreams - 1;
313 m->avctx->sample_rate = mh.group1_samplerate;
314 m->avctx->frame_size = mh.access_unit_size;
316 m->avctx->bits_per_raw_sample = mh.group1_bits;
317 if (mh.group1_bits > 16)
318 m->avctx->sample_fmt = SAMPLE_FMT_S32;
320 m->avctx->sample_fmt = SAMPLE_FMT_S16;
323 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
324 m->substream[substr].restart_seen = 0;
329 /** Read a restart header from a block in a substream. This contains parameters
330 * required to decode the audio that do not change very often. Generally
331 * (always) present only in blocks following a major sync. */
333 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
334 const uint8_t *buf, unsigned int substr)
336 SubStream *s = &m->substream[substr];
340 uint8_t lossless_check;
341 int start_count = get_bits_count(gbp);
342 const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
343 ? MAX_MATRIX_CHANNEL_MLP
344 : MAX_MATRIX_CHANNEL_TRUEHD;
346 sync_word = get_bits(gbp, 13);
347 s->noise_type = get_bits1(gbp);
349 if ((m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) ||
350 sync_word != 0x31ea >> 1) {
351 av_log(m->avctx, AV_LOG_ERROR,
352 "restart header sync incorrect (got 0x%04x)\n", sync_word);
356 skip_bits(gbp, 16); /* Output timestamp */
358 s->min_channel = get_bits(gbp, 4);
359 s->max_channel = get_bits(gbp, 4);
360 s->max_matrix_channel = get_bits(gbp, 4);
362 if (s->max_matrix_channel > max_matrix_channel) {
363 av_log(m->avctx, AV_LOG_ERROR,
364 "Max matrix channel cannot be greater than %d.\n",
369 if (s->max_channel != s->max_matrix_channel) {
370 av_log(m->avctx, AV_LOG_ERROR,
371 "Max channel must be equal max matrix channel.\n");
375 if (s->min_channel > s->max_channel) {
376 av_log(m->avctx, AV_LOG_ERROR,
377 "Substream min channel cannot be greater than max channel.\n");
381 if (m->avctx->request_channels > 0
382 && s->max_channel + 1 >= m->avctx->request_channels
383 && substr < m->max_decoded_substream) {
384 av_log(m->avctx, AV_LOG_INFO,
385 "Extracting %d channel downmix from substream %d. "
386 "Further substreams will be skipped.\n",
387 s->max_channel + 1, substr);
388 m->max_decoded_substream = substr;
391 s->noise_shift = get_bits(gbp, 4);
392 s->noisegen_seed = get_bits(gbp, 23);
396 s->data_check_present = get_bits1(gbp);
397 lossless_check = get_bits(gbp, 8);
398 if (substr == m->max_decoded_substream
399 && s->lossless_check_data != 0xffffffff) {
400 tmp = xor_32_to_8(s->lossless_check_data);
401 if (tmp != lossless_check)
402 av_log(m->avctx, AV_LOG_WARNING,
403 "Lossless check failed - expected %02x, calculated %02x.\n",
404 lossless_check, tmp);
409 memset(s->ch_assign, 0, sizeof(s->ch_assign));
411 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
412 int ch_assign = get_bits(gbp, 6);
413 if (ch_assign > s->max_matrix_channel) {
414 av_log(m->avctx, AV_LOG_ERROR,
415 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
416 ch, ch_assign, sample_message);
419 s->ch_assign[ch_assign] = ch;
422 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
424 if (checksum != get_bits(gbp, 8))
425 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
427 /* Set default decoding parameters. */
428 s->param_presence_flags = 0xff;
429 s->num_primitive_matrices = 0;
431 s->lossless_check_data = 0;
433 memset(s->output_shift , 0, sizeof(s->output_shift ));
434 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
436 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
437 ChannelParams *cp = &m->channel_params[ch];
438 cp->filter_params[FIR].order = 0;
439 cp->filter_params[IIR].order = 0;
440 cp->filter_params[FIR].shift = 0;
441 cp->filter_params[IIR].shift = 0;
443 /* Default audio coding is 24-bit raw PCM. */
445 cp->sign_huff_offset = (-1) << 23;
450 if (substr == m->max_decoded_substream)
451 m->avctx->channels = s->max_matrix_channel + 1;
456 /** Read parameters for one of the prediction filters. */
458 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
459 unsigned int channel, unsigned int filter)
461 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
462 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
463 const char fchar = filter ? 'I' : 'F';
466 // Filter is 0 for FIR, 1 for IIR.
469 if (m->filter_changed[channel][filter]++ > 1) {
470 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
474 order = get_bits(gbp, 4);
475 if (order > max_order) {
476 av_log(m->avctx, AV_LOG_ERROR,
477 "%cIR filter order %d is greater than maximum %d.\n",
478 fchar, order, max_order);
484 int coeff_bits, coeff_shift;
486 fp->shift = get_bits(gbp, 4);
488 coeff_bits = get_bits(gbp, 5);
489 coeff_shift = get_bits(gbp, 3);
490 if (coeff_bits < 1 || coeff_bits > 16) {
491 av_log(m->avctx, AV_LOG_ERROR,
492 "%cIR filter coeff_bits must be between 1 and 16.\n",
496 if (coeff_bits + coeff_shift > 16) {
497 av_log(m->avctx, AV_LOG_ERROR,
498 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
503 for (i = 0; i < order; i++)
504 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
506 if (get_bits1(gbp)) {
507 int state_bits, state_shift;
510 av_log(m->avctx, AV_LOG_ERROR,
511 "FIR filter has state data specified.\n");
515 state_bits = get_bits(gbp, 4);
516 state_shift = get_bits(gbp, 4);
518 /* TODO: Check validity of state data. */
520 for (i = 0; i < order; i++)
521 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
528 /** Read parameters for primitive matrices. */
530 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
532 SubStream *s = &m->substream[substr];
533 unsigned int mat, ch;
534 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
536 : MAX_MATRICES_TRUEHD;
538 if (m->matrix_changed++ > 1) {
539 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
543 s->num_primitive_matrices = get_bits(gbp, 4);
545 if (s->num_primitive_matrices > max_primitive_matrices) {
546 av_log(m->avctx, AV_LOG_ERROR,
547 "Number of primitive matrices cannot be greater than %d.\n",
548 max_primitive_matrices);
552 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
553 int frac_bits, max_chan;
554 s->matrix_out_ch[mat] = get_bits(gbp, 4);
555 frac_bits = get_bits(gbp, 4);
556 s->lsb_bypass [mat] = get_bits1(gbp);
558 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
559 av_log(m->avctx, AV_LOG_ERROR,
560 "Invalid channel %d specified as output from matrix.\n",
561 s->matrix_out_ch[mat]);
564 if (frac_bits > 14) {
565 av_log(m->avctx, AV_LOG_ERROR,
566 "Too many fractional bits specified.\n");
570 max_chan = s->max_matrix_channel;
574 for (ch = 0; ch <= max_chan; ch++) {
577 coeff_val = get_sbits(gbp, frac_bits + 2);
579 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
583 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
585 s->matrix_noise_shift[mat] = 0;
591 /** Read channel parameters. */
593 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
594 GetBitContext *gbp, unsigned int ch)
596 ChannelParams *cp = &m->channel_params[ch];
597 FilterParams *fir = &cp->filter_params[FIR];
598 FilterParams *iir = &cp->filter_params[IIR];
599 SubStream *s = &m->substream[substr];
601 if (s->param_presence_flags & PARAM_FIR)
603 if (read_filter_params(m, gbp, ch, FIR) < 0)
606 if (s->param_presence_flags & PARAM_IIR)
608 if (read_filter_params(m, gbp, ch, IIR) < 0)
611 if (fir->order + iir->order > 8) {
612 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
616 if (fir->order && iir->order &&
617 fir->shift != iir->shift) {
618 av_log(m->avctx, AV_LOG_ERROR,
619 "FIR and IIR filters must use the same precision.\n");
622 /* The FIR and IIR filters must have the same precision.
623 * To simplify the filtering code, only the precision of the
624 * FIR filter is considered. If only the IIR filter is employed,
625 * the FIR filter precision is set to that of the IIR filter, so
626 * that the filtering code can use it. */
627 if (!fir->order && iir->order)
628 fir->shift = iir->shift;
630 if (s->param_presence_flags & PARAM_HUFFOFFSET)
632 cp->huff_offset = get_sbits(gbp, 15);
634 cp->codebook = get_bits(gbp, 2);
635 cp->huff_lsbs = get_bits(gbp, 5);
637 if (cp->huff_lsbs > 24) {
638 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
642 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
647 /** Read decoding parameters that change more often than those in the restart
650 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
653 SubStream *s = &m->substream[substr];
656 if (s->param_presence_flags & PARAM_PRESENCE)
658 s->param_presence_flags = get_bits(gbp, 8);
660 if (s->param_presence_flags & PARAM_BLOCKSIZE)
661 if (get_bits1(gbp)) {
662 s->blocksize = get_bits(gbp, 9);
663 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
664 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
670 if (s->param_presence_flags & PARAM_MATRIX)
672 if (read_matrix_params(m, substr, gbp) < 0)
675 if (s->param_presence_flags & PARAM_OUTSHIFT)
677 for (ch = 0; ch <= s->max_matrix_channel; ch++)
678 s->output_shift[ch] = get_sbits(gbp, 4);
680 if (s->param_presence_flags & PARAM_QUANTSTEP)
682 for (ch = 0; ch <= s->max_channel; ch++) {
683 ChannelParams *cp = &m->channel_params[ch];
685 s->quant_step_size[ch] = get_bits(gbp, 4);
687 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
690 for (ch = s->min_channel; ch <= s->max_channel; ch++)
692 if (read_channel_params(m, substr, gbp, ch) < 0)
698 #define MSB_MASK(bits) (-1u << bits)
700 /** Generate PCM samples using the prediction filters and residual values
701 * read from the data stream, and update the filter state. */
703 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
704 unsigned int channel)
706 SubStream *s = &m->substream[substr];
707 int32_t fir_state_buffer[MAX_BLOCKSIZE + MAX_FIR_ORDER];
708 int32_t iir_state_buffer[MAX_BLOCKSIZE + MAX_IIR_ORDER];
709 int32_t *firbuf = fir_state_buffer + MAX_BLOCKSIZE;
710 int32_t *iirbuf = iir_state_buffer + MAX_BLOCKSIZE;
711 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
712 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
713 unsigned int filter_shift = fir->shift;
714 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
716 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
717 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
719 m->dsp.mlp_filter_channel(firbuf, fir->coeff, fir->order,
720 iirbuf, iir->coeff, iir->order,
721 filter_shift, mask, s->blocksize,
722 &m->sample_buffer[s->blockpos][channel]);
724 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
725 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
728 /** Read a block of PCM residual data (or actual if no filtering active). */
730 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
733 SubStream *s = &m->substream[substr];
734 unsigned int i, ch, expected_stream_pos = 0;
736 if (s->data_check_present) {
737 expected_stream_pos = get_bits_count(gbp);
738 expected_stream_pos += get_bits(gbp, 16);
739 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
740 "we have not tested yet. %s\n", sample_message);
743 if (s->blockpos + s->blocksize > m->access_unit_size) {
744 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
748 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
749 s->blocksize * sizeof(m->bypassed_lsbs[0]));
751 for (i = 0; i < s->blocksize; i++)
752 if (read_huff_channels(m, gbp, substr, i) < 0)
755 for (ch = s->min_channel; ch <= s->max_channel; ch++)
756 filter_channel(m, substr, ch);
758 s->blockpos += s->blocksize;
760 if (s->data_check_present) {
761 if (get_bits_count(gbp) != expected_stream_pos)
762 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
769 /** Data table used for TrueHD noise generation function. */
771 static const int8_t noise_table[256] = {
772 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
773 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
774 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
775 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
776 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
777 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
778 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
779 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
780 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
781 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
782 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
783 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
784 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
785 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
786 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
787 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
790 /** Noise generation functions.
791 * I'm not sure what these are for - they seem to be some kind of pseudorandom
792 * sequence generators, used to generate noise data which is used when the
793 * channels are rematrixed. I'm not sure if they provide a practical benefit
794 * to compression, or just obfuscate the decoder. Are they for some kind of
797 /** Generate two channels of noise, used in the matrix when
798 * restart sync word == 0x31ea. */
800 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
802 SubStream *s = &m->substream[substr];
804 uint32_t seed = s->noisegen_seed;
805 unsigned int maxchan = s->max_matrix_channel;
807 for (i = 0; i < s->blockpos; i++) {
808 uint16_t seed_shr7 = seed >> 7;
809 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
810 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
812 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
815 s->noisegen_seed = seed;
818 /** Generate a block of noise, used when restart sync word == 0x31eb. */
820 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
822 SubStream *s = &m->substream[substr];
824 uint32_t seed = s->noisegen_seed;
826 for (i = 0; i < m->access_unit_size_pow2; i++) {
827 uint8_t seed_shr15 = seed >> 15;
828 m->noise_buffer[i] = noise_table[seed_shr15];
829 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
832 s->noisegen_seed = seed;
836 /** Apply the channel matrices in turn to reconstruct the original audio
839 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
841 SubStream *s = &m->substream[substr];
842 unsigned int mat, src_ch, i;
843 unsigned int maxchan;
845 maxchan = s->max_matrix_channel;
846 if (!s->noise_type) {
847 generate_2_noise_channels(m, substr);
850 fill_noise_buffer(m, substr);
853 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
854 int matrix_noise_shift = s->matrix_noise_shift[mat];
855 unsigned int dest_ch = s->matrix_out_ch[mat];
856 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
857 int32_t *coeffs = s->matrix_coeff[mat];
858 int index = s->num_primitive_matrices - mat;
859 int index2 = 2 * index + 1;
861 /* TODO: DSPContext? */
863 for (i = 0; i < s->blockpos; i++) {
864 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
865 int32_t *samples = m->sample_buffer[i];
868 for (src_ch = 0; src_ch <= maxchan; src_ch++)
869 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
871 if (matrix_noise_shift) {
872 index &= m->access_unit_size_pow2 - 1;
873 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
877 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
882 /** Write the audio data into the output buffer. */
884 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
885 uint8_t *data, unsigned int *data_size, int is32)
887 SubStream *s = &m->substream[substr];
888 unsigned int i, out_ch = 0;
889 int32_t *data_32 = (int32_t*) data;
890 int16_t *data_16 = (int16_t*) data;
892 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
895 for (i = 0; i < s->blockpos; i++) {
896 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
897 int mat_ch = s->ch_assign[out_ch];
898 int32_t sample = m->sample_buffer[i][mat_ch]
899 << s->output_shift[mat_ch];
900 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
901 if (is32) *data_32++ = sample << 8;
902 else *data_16++ = sample >> 8;
906 *data_size = i * out_ch * (is32 ? 4 : 2);
911 static int output_data(MLPDecodeContext *m, unsigned int substr,
912 uint8_t *data, unsigned int *data_size)
914 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
915 return output_data_internal(m, substr, data, data_size, 1);
917 return output_data_internal(m, substr, data, data_size, 0);
921 /** Read an access unit from the stream.
922 * Returns < 0 on error, 0 if not enough data is present in the input stream
923 * otherwise returns the number of bytes consumed. */
925 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
928 const uint8_t *buf = avpkt->data;
929 int buf_size = avpkt->size;
930 MLPDecodeContext *m = avctx->priv_data;
932 unsigned int length, substr;
933 unsigned int substream_start;
934 unsigned int header_size = 4;
935 unsigned int substr_header_size = 0;
936 uint8_t substream_parity_present[MAX_SUBSTREAMS];
937 uint16_t substream_data_len[MAX_SUBSTREAMS];
943 length = (AV_RB16(buf) & 0xfff) * 2;
945 if (length > buf_size)
948 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
950 m->is_major_sync_unit = 0;
951 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
952 if (read_major_sync(m, &gb) < 0)
954 m->is_major_sync_unit = 1;
958 if (!m->params_valid) {
959 av_log(m->avctx, AV_LOG_WARNING,
960 "Stream parameters not seen; skipping frame.\n");
967 for (substr = 0; substr < m->num_substreams; substr++) {
968 int extraword_present, checkdata_present, end, nonrestart_substr;
970 extraword_present = get_bits1(&gb);
971 nonrestart_substr = get_bits1(&gb);
972 checkdata_present = get_bits1(&gb);
975 end = get_bits(&gb, 12) * 2;
977 substr_header_size += 2;
979 if (extraword_present) {
980 if (m->avctx->codec_id == CODEC_ID_MLP) {
981 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
985 substr_header_size += 2;
988 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
989 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
993 if (end + header_size + substr_header_size > length) {
994 av_log(m->avctx, AV_LOG_ERROR,
995 "Indicated length of substream %d data goes off end of "
996 "packet.\n", substr);
997 end = length - header_size - substr_header_size;
1000 if (end < substream_start) {
1001 av_log(avctx, AV_LOG_ERROR,
1002 "Indicated end offset of substream %d data "
1003 "is smaller than calculated start offset.\n",
1008 if (substr > m->max_decoded_substream)
1011 substream_parity_present[substr] = checkdata_present;
1012 substream_data_len[substr] = end - substream_start;
1013 substream_start = end;
1016 parity_bits = ff_mlp_calculate_parity(buf, 4);
1017 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1019 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1020 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1024 buf += header_size + substr_header_size;
1026 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1027 SubStream *s = &m->substream[substr];
1028 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1030 m->matrix_changed = 0;
1031 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1035 if (get_bits1(&gb)) {
1036 if (get_bits1(&gb)) {
1037 /* A restart header should be present. */
1038 if (read_restart_header(m, &gb, buf, substr) < 0)
1040 s->restart_seen = 1;
1043 if (!s->restart_seen)
1045 if (read_decoding_params(m, &gb, substr) < 0)
1049 if (!s->restart_seen)
1052 if (read_block_data(m, &gb, substr) < 0)
1055 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1056 goto substream_length_mismatch;
1058 } while (!get_bits1(&gb));
1060 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1062 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1065 if (get_bits(&gb, 16) != 0xD234)
1068 shorten_by = get_bits(&gb, 16);
1069 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1070 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1071 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1074 if (substr == m->max_decoded_substream)
1075 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1078 if (substream_parity_present[substr]) {
1079 uint8_t parity, checksum;
1081 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1082 goto substream_length_mismatch;
1084 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1085 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1087 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1088 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1089 if ( get_bits(&gb, 8) != checksum)
1090 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1093 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1094 goto substream_length_mismatch;
1097 if (!s->restart_seen)
1098 av_log(m->avctx, AV_LOG_ERROR,
1099 "No restart header present in substream %d.\n", substr);
1101 buf += substream_data_len[substr];
1104 rematrix_channels(m, m->max_decoded_substream);
1106 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1111 substream_length_mismatch:
1112 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1116 m->params_valid = 0;
1120 #if CONFIG_MLP_DECODER
1121 AVCodec mlp_decoder = {
1125 sizeof(MLPDecodeContext),
1130 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1132 #endif /* CONFIG_MLP_DECODER */
1134 #if CONFIG_TRUEHD_DECODER
1135 AVCodec truehd_decoder = {
1139 sizeof(MLPDecodeContext),
1144 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1146 #endif /* CONFIG_TRUEHD_DECODER */