3 * Copyright (c) 2007-2008 Ian Caulfield
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30 #include "libavutil/internal.h"
31 #include "libavutil/intreadwrite.h"
32 #include "libavutil/channel_layout.h"
35 #include "libavutil/crc.h"
37 #include "mlp_parser.h"
42 /** number of bits used for VLC lookup - longest Huffman code is 9 */
45 #define VLC_STATIC_SIZE 64
48 #define VLC_STATIC_SIZE 512
51 typedef struct SubStream {
52 /// Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
56 /** restart header data */
57 /// The type of noise to be used in the rematrix stage.
60 /// The index of the first channel coded in this substream.
62 /// The index of the last channel coded in this substream.
64 /// The number of channels input into the rematrix stage.
65 uint8_t max_matrix_channel;
66 /// For each channel output by the matrix, the output channel to map it to
67 uint8_t ch_assign[MAX_CHANNELS];
68 /// The channel layout for this substream
70 /// The matrix encoding mode for this substream
71 enum AVMatrixEncoding matrix_encoding;
73 /// Channel coding parameters for channels in the substream
74 ChannelParams channel_params[MAX_CHANNELS];
76 /// The left shift applied to random noise in 0x31ea substreams.
78 /// The current seed value for the pseudorandom noise generator(s).
79 uint32_t noisegen_seed;
81 /// Set if the substream contains extra info to check the size of VLC blocks.
82 uint8_t data_check_present;
84 /// Bitmask of which parameter sets are conveyed in a decoding parameter block.
85 uint8_t param_presence_flags;
86 #define PARAM_BLOCKSIZE (1 << 7)
87 #define PARAM_MATRIX (1 << 6)
88 #define PARAM_OUTSHIFT (1 << 5)
89 #define PARAM_QUANTSTEP (1 << 4)
90 #define PARAM_FIR (1 << 3)
91 #define PARAM_IIR (1 << 2)
92 #define PARAM_HUFFOFFSET (1 << 1)
93 #define PARAM_PRESENCE (1 << 0)
99 /// Number of matrices to be applied.
100 uint8_t num_primitive_matrices;
102 /// matrix output channel
103 uint8_t matrix_out_ch[MAX_MATRICES];
105 /// Whether the LSBs of the matrix output are encoded in the bitstream.
106 uint8_t lsb_bypass[MAX_MATRICES];
107 /// Matrix coefficients, stored as 2.14 fixed point.
108 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
109 /// Left shift to apply to noise values in 0x31eb substreams.
110 uint8_t matrix_noise_shift[MAX_MATRICES];
113 /// Left shift to apply to Huffman-decoded residuals.
114 uint8_t quant_step_size[MAX_CHANNELS];
116 /// number of PCM samples in current audio block
118 /// Number of PCM samples decoded so far in this frame.
121 /// Left shift to apply to decoded PCM values to get final 24-bit output.
122 int8_t output_shift[MAX_CHANNELS];
124 /// Running XOR of all output samples.
125 int32_t lossless_check_data;
129 typedef struct MLPDecodeContext {
130 AVCodecContext *avctx;
132 /// Current access unit being read has a major sync.
133 int is_major_sync_unit;
135 /// Size of the major sync unit, in bytes
136 int major_sync_header_size;
138 /// Set if a valid major sync block has been read. Otherwise no decoding is possible.
139 uint8_t params_valid;
141 /// Number of substreams contained within this stream.
142 uint8_t num_substreams;
144 /// Index of the last substream to decode - further substreams are skipped.
145 uint8_t max_decoded_substream;
147 /// number of PCM samples contained in each frame
148 int access_unit_size;
149 /// next power of two above the number of samples in each frame
150 int access_unit_size_pow2;
152 SubStream substream[MAX_SUBSTREAMS];
155 int filter_changed[MAX_CHANNELS][NUM_FILTERS];
157 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
158 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
159 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
164 static const uint64_t thd_channel_order[] = {
165 AV_CH_FRONT_LEFT, AV_CH_FRONT_RIGHT, // LR
166 AV_CH_FRONT_CENTER, // C
167 AV_CH_LOW_FREQUENCY, // LFE
168 AV_CH_SIDE_LEFT, AV_CH_SIDE_RIGHT, // LRs
169 AV_CH_TOP_FRONT_LEFT, AV_CH_TOP_FRONT_RIGHT, // LRvh
170 AV_CH_FRONT_LEFT_OF_CENTER, AV_CH_FRONT_RIGHT_OF_CENTER, // LRc
171 AV_CH_BACK_LEFT, AV_CH_BACK_RIGHT, // LRrs
172 AV_CH_BACK_CENTER, // Cs
173 AV_CH_TOP_CENTER, // Ts
174 AV_CH_SURROUND_DIRECT_LEFT, AV_CH_SURROUND_DIRECT_RIGHT, // LRsd
175 AV_CH_WIDE_LEFT, AV_CH_WIDE_RIGHT, // LRw
176 AV_CH_TOP_FRONT_CENTER, // Cvh
177 AV_CH_LOW_FREQUENCY_2, // LFE2
180 static uint64_t thd_channel_layout_extract_channel(uint64_t channel_layout,
185 if (av_get_channel_layout_nb_channels(channel_layout) <= index)
188 for (i = 0; i < FF_ARRAY_ELEMS(thd_channel_order); i++)
189 if (channel_layout & thd_channel_order[i] && !index--)
190 return thd_channel_order[i];
194 static VLC huff_vlc[3];
196 /** Initialize static data, constant between all invocations of the codec. */
198 static av_cold void init_static(void)
200 if (!huff_vlc[0].bits) {
201 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
202 &ff_mlp_huffman_tables[0][0][1], 2, 1,
203 &ff_mlp_huffman_tables[0][0][0], 2, 1, VLC_STATIC_SIZE);
204 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
205 &ff_mlp_huffman_tables[1][0][1], 2, 1,
206 &ff_mlp_huffman_tables[1][0][0], 2, 1, VLC_STATIC_SIZE);
207 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
208 &ff_mlp_huffman_tables[2][0][1], 2, 1,
209 &ff_mlp_huffman_tables[2][0][0], 2, 1, VLC_STATIC_SIZE);
215 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
216 unsigned int substr, unsigned int ch)
218 SubStream *s = &m->substream[substr];
219 ChannelParams *cp = &s->channel_params[ch];
220 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
221 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
222 int32_t sign_huff_offset = cp->huff_offset;
224 if (cp->codebook > 0)
225 sign_huff_offset -= 7 << lsb_bits;
228 sign_huff_offset -= 1 << sign_shift;
230 return sign_huff_offset;
233 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
236 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
237 unsigned int substr, unsigned int pos)
239 SubStream *s = &m->substream[substr];
240 unsigned int mat, channel;
242 for (mat = 0; mat < s->num_primitive_matrices; mat++)
243 if (s->lsb_bypass[mat])
244 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
246 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
247 ChannelParams *cp = &s->channel_params[channel];
248 int codebook = cp->codebook;
249 int quant_step_size = s->quant_step_size[channel];
250 int lsb_bits = cp->huff_lsbs - quant_step_size;
254 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
255 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
258 return AVERROR_INVALIDDATA;
261 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
263 result += cp->sign_huff_offset;
264 result <<= quant_step_size;
266 m->sample_buffer[pos + s->blockpos][channel] = result;
272 static av_cold int mlp_decode_init(AVCodecContext *avctx)
274 MLPDecodeContext *m = avctx->priv_data;
279 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
280 m->substream[substr].lossless_check_data = 0xffffffff;
281 ff_mlpdsp_init(&m->dsp);
286 /** Read a major sync info header - contains high level information about
287 * the stream - sample rate, channel arrangement etc. Most of this
288 * information is not actually necessary for decoding, only for playback.
291 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
296 if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
299 if (mh.group1_bits == 0) {
300 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
301 return AVERROR_INVALIDDATA;
303 if (mh.group2_bits > mh.group1_bits) {
304 av_log(m->avctx, AV_LOG_ERROR,
305 "Channel group 2 cannot have more bits per sample than group 1.\n");
306 return AVERROR_INVALIDDATA;
309 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
310 av_log(m->avctx, AV_LOG_ERROR,
311 "Channel groups with differing sample rates are not currently supported.\n");
312 return AVERROR_INVALIDDATA;
315 if (mh.group1_samplerate == 0) {
316 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
317 return AVERROR_INVALIDDATA;
319 if (mh.group1_samplerate > MAX_SAMPLERATE) {
320 av_log(m->avctx, AV_LOG_ERROR,
321 "Sampling rate %d is greater than the supported maximum (%d).\n",
322 mh.group1_samplerate, MAX_SAMPLERATE);
323 return AVERROR_INVALIDDATA;
325 if (mh.access_unit_size > MAX_BLOCKSIZE) {
326 av_log(m->avctx, AV_LOG_ERROR,
327 "Block size %d is greater than the supported maximum (%d).\n",
328 mh.access_unit_size, MAX_BLOCKSIZE);
329 return AVERROR_INVALIDDATA;
331 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
332 av_log(m->avctx, AV_LOG_ERROR,
333 "Block size pow2 %d is greater than the supported maximum (%d).\n",
334 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
335 return AVERROR_INVALIDDATA;
338 if (mh.num_substreams == 0)
339 return AVERROR_INVALIDDATA;
340 if (m->avctx->codec_id == AV_CODEC_ID_MLP && mh.num_substreams > 2) {
341 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
342 return AVERROR_INVALIDDATA;
344 if (mh.num_substreams > MAX_SUBSTREAMS) {
345 avpriv_request_sample(m->avctx,
346 "%d substreams (more than the "
347 "maximum supported by the decoder)",
349 return AVERROR_PATCHWELCOME;
352 m->major_sync_header_size = mh.header_size;
354 m->access_unit_size = mh.access_unit_size;
355 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
357 m->num_substreams = mh.num_substreams;
359 /* limit to decoding 3 substreams, as the 4th is used by Dolby Atmos for non-audio data */
360 m->max_decoded_substream = FFMIN(m->num_substreams - 1, 2);
362 m->avctx->sample_rate = mh.group1_samplerate;
363 m->avctx->frame_size = mh.access_unit_size;
365 m->avctx->bits_per_raw_sample = mh.group1_bits;
366 if (mh.group1_bits > 16)
367 m->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
369 m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
370 m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(m->substream[m->max_decoded_substream].ch_assign,
371 m->substream[m->max_decoded_substream].output_shift,
372 m->substream[m->max_decoded_substream].max_matrix_channel,
373 m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
376 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
377 m->substream[substr].restart_seen = 0;
379 /* Set the layout for each substream. When there's more than one, the first
380 * substream is Stereo. Subsequent substreams' layouts are indicated in the
382 if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
383 if ((substr = (mh.num_substreams > 1)))
384 m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
385 m->substream[substr].ch_layout = mh.channel_layout_mlp;
387 if ((substr = (mh.num_substreams > 1)))
388 m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
389 if (mh.num_substreams > 2)
390 if (mh.channel_layout_thd_stream2)
391 m->substream[2].ch_layout = mh.channel_layout_thd_stream2;
393 m->substream[2].ch_layout = mh.channel_layout_thd_stream1;
394 m->substream[substr].ch_layout = mh.channel_layout_thd_stream1;
397 /* Parse the TrueHD decoder channel modifiers and set each substream's
398 * AVMatrixEncoding accordingly.
400 * The meaning of the modifiers depends on the channel layout:
402 * - THD_CH_MODIFIER_LTRT, THD_CH_MODIFIER_LBINRBIN only apply to 2-channel
404 * - THD_CH_MODIFIER_MONO applies to 1-channel or 2-channel (dual mono)
406 * - THD_CH_MODIFIER_SURROUNDEX, THD_CH_MODIFIER_NOTSURROUNDEX only apply to
407 * layouts with an Ls/Rs channel pair
409 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
410 m->substream[substr].matrix_encoding = AV_MATRIX_ENCODING_NONE;
411 if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
412 if (mh.num_substreams > 2 &&
413 mh.channel_layout_thd_stream2 & AV_CH_SIDE_LEFT &&
414 mh.channel_layout_thd_stream2 & AV_CH_SIDE_RIGHT &&
415 mh.channel_modifier_thd_stream2 == THD_CH_MODIFIER_SURROUNDEX)
416 m->substream[2].matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX;
418 if (mh.num_substreams > 1 &&
419 mh.channel_layout_thd_stream1 & AV_CH_SIDE_LEFT &&
420 mh.channel_layout_thd_stream1 & AV_CH_SIDE_RIGHT &&
421 mh.channel_modifier_thd_stream1 == THD_CH_MODIFIER_SURROUNDEX)
422 m->substream[1].matrix_encoding = AV_MATRIX_ENCODING_DOLBYEX;
424 if (mh.num_substreams > 0)
425 switch (mh.channel_modifier_thd_stream0) {
426 case THD_CH_MODIFIER_LTRT:
427 m->substream[0].matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
429 case THD_CH_MODIFIER_LBINRBIN:
430 m->substream[0].matrix_encoding = AV_MATRIX_ENCODING_DOLBYHEADPHONE;
440 /** Read a restart header from a block in a substream. This contains parameters
441 * required to decode the audio that do not change very often. Generally
442 * (always) present only in blocks following a major sync. */
444 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
445 const uint8_t *buf, unsigned int substr)
447 SubStream *s = &m->substream[substr];
451 uint8_t lossless_check;
452 int start_count = get_bits_count(gbp);
453 int min_channel, max_channel, max_matrix_channel;
454 const int std_max_matrix_channel = m->avctx->codec_id == AV_CODEC_ID_MLP
455 ? MAX_MATRIX_CHANNEL_MLP
456 : MAX_MATRIX_CHANNEL_TRUEHD;
458 sync_word = get_bits(gbp, 13);
460 if (sync_word != 0x31ea >> 1) {
461 av_log(m->avctx, AV_LOG_ERROR,
462 "restart header sync incorrect (got 0x%04x)\n", sync_word);
463 return AVERROR_INVALIDDATA;
466 s->noise_type = get_bits1(gbp);
468 if (m->avctx->codec_id == AV_CODEC_ID_MLP && s->noise_type) {
469 av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
470 return AVERROR_INVALIDDATA;
473 skip_bits(gbp, 16); /* Output timestamp */
475 min_channel = get_bits(gbp, 4);
476 max_channel = get_bits(gbp, 4);
477 max_matrix_channel = get_bits(gbp, 4);
479 if (max_matrix_channel > std_max_matrix_channel) {
480 av_log(m->avctx, AV_LOG_ERROR,
481 "Max matrix channel cannot be greater than %d.\n",
483 return AVERROR_INVALIDDATA;
486 if (max_channel != max_matrix_channel) {
487 av_log(m->avctx, AV_LOG_ERROR,
488 "Max channel must be equal max matrix channel.\n");
489 return AVERROR_INVALIDDATA;
492 /* This should happen for TrueHD streams with >6 channels and MLP's noise
493 * type. It is not yet known if this is allowed. */
494 if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
495 avpriv_request_sample(m->avctx,
496 "%d channels (more than the "
497 "maximum supported by the decoder)",
499 return AVERROR_PATCHWELCOME;
502 if (min_channel > max_channel) {
503 av_log(m->avctx, AV_LOG_ERROR,
504 "Substream min channel cannot be greater than max channel.\n");
505 return AVERROR_INVALIDDATA;
508 s->min_channel = min_channel;
509 s->max_channel = max_channel;
510 s->max_matrix_channel = max_matrix_channel;
512 if (m->avctx->request_channel_layout && (s->ch_layout & m->avctx->request_channel_layout) ==
513 m->avctx->request_channel_layout && m->max_decoded_substream > substr) {
514 av_log(m->avctx, AV_LOG_DEBUG,
515 "Extracting %d-channel downmix (0x%"PRIx64") from substream %d. "
516 "Further substreams will be skipped.\n",
517 s->max_channel + 1, s->ch_layout, substr);
518 m->max_decoded_substream = substr;
521 s->noise_shift = get_bits(gbp, 4);
522 s->noisegen_seed = get_bits(gbp, 23);
526 s->data_check_present = get_bits1(gbp);
527 lossless_check = get_bits(gbp, 8);
528 if (substr == m->max_decoded_substream
529 && s->lossless_check_data != 0xffffffff) {
530 tmp = xor_32_to_8(s->lossless_check_data);
531 if (tmp != lossless_check)
532 av_log(m->avctx, AV_LOG_WARNING,
533 "Lossless check failed - expected %02x, calculated %02x.\n",
534 lossless_check, tmp);
539 memset(s->ch_assign, 0, sizeof(s->ch_assign));
541 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
542 int ch_assign = get_bits(gbp, 6);
543 if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD) {
544 uint64_t channel = thd_channel_layout_extract_channel(s->ch_layout,
546 ch_assign = av_get_channel_layout_channel_index(s->ch_layout,
549 if (ch_assign < 0 || ch_assign > s->max_matrix_channel) {
550 avpriv_request_sample(m->avctx,
551 "Assignment of matrix channel %d to invalid output channel %d",
553 return AVERROR_PATCHWELCOME;
555 s->ch_assign[ch_assign] = ch;
558 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
560 if (checksum != get_bits(gbp, 8))
561 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
563 /* Set default decoding parameters. */
564 s->param_presence_flags = 0xff;
565 s->num_primitive_matrices = 0;
567 s->lossless_check_data = 0;
569 memset(s->output_shift , 0, sizeof(s->output_shift ));
570 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
572 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
573 ChannelParams *cp = &s->channel_params[ch];
574 cp->filter_params[FIR].order = 0;
575 cp->filter_params[IIR].order = 0;
576 cp->filter_params[FIR].shift = 0;
577 cp->filter_params[IIR].shift = 0;
579 /* Default audio coding is 24-bit raw PCM. */
581 cp->sign_huff_offset = (-1) << 23;
586 if (substr == m->max_decoded_substream) {
587 m->avctx->channels = s->max_matrix_channel + 1;
588 m->avctx->channel_layout = s->ch_layout;
589 m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(s->ch_assign,
591 s->max_matrix_channel,
592 m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
598 /** Read parameters for one of the prediction filters. */
600 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
601 unsigned int substr, unsigned int channel,
604 SubStream *s = &m->substream[substr];
605 FilterParams *fp = &s->channel_params[channel].filter_params[filter];
606 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
607 const char fchar = filter ? 'I' : 'F';
610 // Filter is 0 for FIR, 1 for IIR.
613 if (m->filter_changed[channel][filter]++ > 1) {
614 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
615 return AVERROR_INVALIDDATA;
618 order = get_bits(gbp, 4);
619 if (order > max_order) {
620 av_log(m->avctx, AV_LOG_ERROR,
621 "%cIR filter order %d is greater than maximum %d.\n",
622 fchar, order, max_order);
623 return AVERROR_INVALIDDATA;
628 int32_t *fcoeff = s->channel_params[channel].coeff[filter];
629 int coeff_bits, coeff_shift;
631 fp->shift = get_bits(gbp, 4);
633 coeff_bits = get_bits(gbp, 5);
634 coeff_shift = get_bits(gbp, 3);
635 if (coeff_bits < 1 || coeff_bits > 16) {
636 av_log(m->avctx, AV_LOG_ERROR,
637 "%cIR filter coeff_bits must be between 1 and 16.\n",
639 return AVERROR_INVALIDDATA;
641 if (coeff_bits + coeff_shift > 16) {
642 av_log(m->avctx, AV_LOG_ERROR,
643 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
645 return AVERROR_INVALIDDATA;
648 for (i = 0; i < order; i++)
649 fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
651 if (get_bits1(gbp)) {
652 int state_bits, state_shift;
655 av_log(m->avctx, AV_LOG_ERROR,
656 "FIR filter has state data specified.\n");
657 return AVERROR_INVALIDDATA;
660 state_bits = get_bits(gbp, 4);
661 state_shift = get_bits(gbp, 4);
663 /* TODO: Check validity of state data. */
665 for (i = 0; i < order; i++)
666 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
673 /** Read parameters for primitive matrices. */
675 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
677 SubStream *s = &m->substream[substr];
678 unsigned int mat, ch;
679 const int max_primitive_matrices = m->avctx->codec_id == AV_CODEC_ID_MLP
681 : MAX_MATRICES_TRUEHD;
683 if (m->matrix_changed++ > 1) {
684 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
685 return AVERROR_INVALIDDATA;
688 s->num_primitive_matrices = get_bits(gbp, 4);
690 if (s->num_primitive_matrices > max_primitive_matrices) {
691 av_log(m->avctx, AV_LOG_ERROR,
692 "Number of primitive matrices cannot be greater than %d.\n",
693 max_primitive_matrices);
694 return AVERROR_INVALIDDATA;
697 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
698 int frac_bits, max_chan;
699 s->matrix_out_ch[mat] = get_bits(gbp, 4);
700 frac_bits = get_bits(gbp, 4);
701 s->lsb_bypass [mat] = get_bits1(gbp);
703 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
704 av_log(m->avctx, AV_LOG_ERROR,
705 "Invalid channel %d specified as output from matrix.\n",
706 s->matrix_out_ch[mat]);
707 return AVERROR_INVALIDDATA;
709 if (frac_bits > 14) {
710 av_log(m->avctx, AV_LOG_ERROR,
711 "Too many fractional bits specified.\n");
712 return AVERROR_INVALIDDATA;
715 max_chan = s->max_matrix_channel;
719 for (ch = 0; ch <= max_chan; ch++) {
722 coeff_val = get_sbits(gbp, frac_bits + 2);
724 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
728 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
730 s->matrix_noise_shift[mat] = 0;
736 /** Read channel parameters. */
738 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
739 GetBitContext *gbp, unsigned int ch)
741 SubStream *s = &m->substream[substr];
742 ChannelParams *cp = &s->channel_params[ch];
743 FilterParams *fir = &cp->filter_params[FIR];
744 FilterParams *iir = &cp->filter_params[IIR];
747 if (s->param_presence_flags & PARAM_FIR)
749 if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
752 if (s->param_presence_flags & PARAM_IIR)
754 if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
757 if (fir->order + iir->order > 8) {
758 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
759 return AVERROR_INVALIDDATA;
762 if (fir->order && iir->order &&
763 fir->shift != iir->shift) {
764 av_log(m->avctx, AV_LOG_ERROR,
765 "FIR and IIR filters must use the same precision.\n");
766 return AVERROR_INVALIDDATA;
768 /* The FIR and IIR filters must have the same precision.
769 * To simplify the filtering code, only the precision of the
770 * FIR filter is considered. If only the IIR filter is employed,
771 * the FIR filter precision is set to that of the IIR filter, so
772 * that the filtering code can use it. */
773 if (!fir->order && iir->order)
774 fir->shift = iir->shift;
776 if (s->param_presence_flags & PARAM_HUFFOFFSET)
778 cp->huff_offset = get_sbits(gbp, 15);
780 cp->codebook = get_bits(gbp, 2);
781 cp->huff_lsbs = get_bits(gbp, 5);
783 if (cp->huff_lsbs > 24) {
784 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
785 return AVERROR_INVALIDDATA;
788 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
793 /** Read decoding parameters that change more often than those in the restart
796 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
799 SubStream *s = &m->substream[substr];
803 if (s->param_presence_flags & PARAM_PRESENCE)
805 s->param_presence_flags = get_bits(gbp, 8);
807 if (s->param_presence_flags & PARAM_BLOCKSIZE)
808 if (get_bits1(gbp)) {
809 s->blocksize = get_bits(gbp, 9);
810 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
811 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
813 return AVERROR_INVALIDDATA;
817 if (s->param_presence_flags & PARAM_MATRIX)
819 if ((ret = read_matrix_params(m, substr, gbp)) < 0)
822 if (s->param_presence_flags & PARAM_OUTSHIFT)
823 if (get_bits1(gbp)) {
824 for (ch = 0; ch <= s->max_matrix_channel; ch++)
825 s->output_shift[ch] = get_sbits(gbp, 4);
826 if (substr == m->max_decoded_substream)
827 m->dsp.mlp_pack_output = m->dsp.mlp_select_pack_output(s->ch_assign,
829 s->max_matrix_channel,
830 m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
833 if (s->param_presence_flags & PARAM_QUANTSTEP)
835 for (ch = 0; ch <= s->max_channel; ch++) {
836 ChannelParams *cp = &s->channel_params[ch];
838 s->quant_step_size[ch] = get_bits(gbp, 4);
840 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
843 for (ch = s->min_channel; ch <= s->max_channel; ch++)
845 if ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
851 #define MSB_MASK(bits) (-1u << bits)
853 /** Generate PCM samples using the prediction filters and residual values
854 * read from the data stream, and update the filter state. */
856 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
857 unsigned int channel)
859 SubStream *s = &m->substream[substr];
860 const int32_t *fircoeff = s->channel_params[channel].coeff[FIR];
861 int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
862 int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
863 int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
864 FilterParams *fir = &s->channel_params[channel].filter_params[FIR];
865 FilterParams *iir = &s->channel_params[channel].filter_params[IIR];
866 unsigned int filter_shift = fir->shift;
867 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
869 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
870 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
872 m->dsp.mlp_filter_channel(firbuf, fircoeff,
873 fir->order, iir->order,
874 filter_shift, mask, s->blocksize,
875 &m->sample_buffer[s->blockpos][channel]);
877 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
878 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
881 /** Read a block of PCM residual data (or actual if no filtering active). */
883 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
886 SubStream *s = &m->substream[substr];
887 unsigned int i, ch, expected_stream_pos = 0;
890 if (s->data_check_present) {
891 expected_stream_pos = get_bits_count(gbp);
892 expected_stream_pos += get_bits(gbp, 16);
893 avpriv_request_sample(m->avctx,
894 "Substreams with VLC block size check info");
897 if (s->blockpos + s->blocksize > m->access_unit_size) {
898 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
899 return AVERROR_INVALIDDATA;
902 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
903 s->blocksize * sizeof(m->bypassed_lsbs[0]));
905 for (i = 0; i < s->blocksize; i++)
906 if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
909 for (ch = s->min_channel; ch <= s->max_channel; ch++)
910 filter_channel(m, substr, ch);
912 s->blockpos += s->blocksize;
914 if (s->data_check_present) {
915 if (get_bits_count(gbp) != expected_stream_pos)
916 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
923 /** Data table used for TrueHD noise generation function. */
925 static const int8_t noise_table[256] = {
926 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
927 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
928 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
929 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
930 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
931 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
932 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
933 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
934 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
935 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
936 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
937 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
938 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
939 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
940 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
941 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
944 /** Noise generation functions.
945 * I'm not sure what these are for - they seem to be some kind of pseudorandom
946 * sequence generators, used to generate noise data which is used when the
947 * channels are rematrixed. I'm not sure if they provide a practical benefit
948 * to compression, or just obfuscate the decoder. Are they for some kind of
951 /** Generate two channels of noise, used in the matrix when
952 * restart sync word == 0x31ea. */
954 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
956 SubStream *s = &m->substream[substr];
958 uint32_t seed = s->noisegen_seed;
959 unsigned int maxchan = s->max_matrix_channel;
961 for (i = 0; i < s->blockpos; i++) {
962 uint16_t seed_shr7 = seed >> 7;
963 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
964 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
966 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
969 s->noisegen_seed = seed;
972 /** Generate a block of noise, used when restart sync word == 0x31eb. */
974 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
976 SubStream *s = &m->substream[substr];
978 uint32_t seed = s->noisegen_seed;
980 for (i = 0; i < m->access_unit_size_pow2; i++) {
981 uint8_t seed_shr15 = seed >> 15;
982 m->noise_buffer[i] = noise_table[seed_shr15];
983 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
986 s->noisegen_seed = seed;
990 /** Apply the channel matrices in turn to reconstruct the original audio
993 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
995 SubStream *s = &m->substream[substr];
997 unsigned int maxchan;
999 maxchan = s->max_matrix_channel;
1000 if (!s->noise_type) {
1001 generate_2_noise_channels(m, substr);
1004 fill_noise_buffer(m, substr);
1007 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
1008 unsigned int dest_ch = s->matrix_out_ch[mat];
1009 m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0],
1010 s->matrix_coeff[mat],
1011 &m->bypassed_lsbs[0][mat],
1013 s->num_primitive_matrices - mat,
1017 s->matrix_noise_shift[mat],
1018 m->access_unit_size_pow2,
1019 MSB_MASK(s->quant_step_size[dest_ch]));
1023 /** Write the audio data into the output buffer. */
1025 static int output_data(MLPDecodeContext *m, unsigned int substr,
1026 AVFrame *frame, int *got_frame_ptr)
1028 AVCodecContext *avctx = m->avctx;
1029 SubStream *s = &m->substream[substr];
1031 int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
1033 if (m->avctx->channels != s->max_matrix_channel + 1) {
1034 av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
1035 return AVERROR_INVALIDDATA;
1039 av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
1040 return AVERROR_INVALIDDATA;
1043 /* get output buffer */
1044 frame->nb_samples = s->blockpos;
1045 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
1046 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1049 s->lossless_check_data = m->dsp.mlp_pack_output(s->lossless_check_data,
1055 s->max_matrix_channel,
1058 /* Update matrix encoding side data */
1059 if ((ret = ff_side_data_update_matrix_encoding(frame, s->matrix_encoding)) < 0)
1067 /** Read an access unit from the stream.
1068 * @return negative on error, 0 if not enough data is present in the input stream,
1069 * otherwise the number of bytes consumed. */
1071 static int read_access_unit(AVCodecContext *avctx, void* data,
1072 int *got_frame_ptr, AVPacket *avpkt)
1074 const uint8_t *buf = avpkt->data;
1075 int buf_size = avpkt->size;
1076 MLPDecodeContext *m = avctx->priv_data;
1078 unsigned int length, substr;
1079 unsigned int substream_start;
1080 unsigned int header_size = 4;
1081 unsigned int substr_header_size = 0;
1082 uint8_t substream_parity_present[MAX_SUBSTREAMS];
1083 uint16_t substream_data_len[MAX_SUBSTREAMS];
1084 uint8_t parity_bits;
1090 length = (AV_RB16(buf) & 0xfff) * 2;
1092 if (length < 4 || length > buf_size)
1093 return AVERROR_INVALIDDATA;
1095 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
1097 m->is_major_sync_unit = 0;
1098 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
1099 if (read_major_sync(m, &gb) < 0)
1101 m->is_major_sync_unit = 1;
1102 header_size += m->major_sync_header_size;
1105 if (!m->params_valid) {
1106 av_log(m->avctx, AV_LOG_WARNING,
1107 "Stream parameters not seen; skipping frame.\n");
1112 substream_start = 0;
1114 for (substr = 0; substr < m->num_substreams; substr++) {
1115 int extraword_present, checkdata_present, end, nonrestart_substr;
1117 extraword_present = get_bits1(&gb);
1118 nonrestart_substr = get_bits1(&gb);
1119 checkdata_present = get_bits1(&gb);
1122 end = get_bits(&gb, 12) * 2;
1124 substr_header_size += 2;
1126 if (extraword_present) {
1127 if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
1128 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
1132 substr_header_size += 2;
1135 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1136 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1140 if (end + header_size + substr_header_size > length) {
1141 av_log(m->avctx, AV_LOG_ERROR,
1142 "Indicated length of substream %d data goes off end of "
1143 "packet.\n", substr);
1144 end = length - header_size - substr_header_size;
1147 if (end < substream_start) {
1148 av_log(avctx, AV_LOG_ERROR,
1149 "Indicated end offset of substream %d data "
1150 "is smaller than calculated start offset.\n",
1155 if (substr > m->max_decoded_substream)
1158 substream_parity_present[substr] = checkdata_present;
1159 substream_data_len[substr] = end - substream_start;
1160 substream_start = end;
1163 parity_bits = ff_mlp_calculate_parity(buf, 4);
1164 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1166 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1167 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1171 buf += header_size + substr_header_size;
1173 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1174 SubStream *s = &m->substream[substr];
1175 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1177 m->matrix_changed = 0;
1178 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1182 if (get_bits1(&gb)) {
1183 if (get_bits1(&gb)) {
1184 /* A restart header should be present. */
1185 if (read_restart_header(m, &gb, buf, substr) < 0)
1187 s->restart_seen = 1;
1190 if (!s->restart_seen)
1192 if (read_decoding_params(m, &gb, substr) < 0)
1196 if (!s->restart_seen)
1199 if ((ret = read_block_data(m, &gb, substr)) < 0)
1202 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1203 goto substream_length_mismatch;
1205 } while (!get_bits1(&gb));
1207 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1209 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1212 if (get_bits(&gb, 16) != 0xD234)
1213 return AVERROR_INVALIDDATA;
1215 shorten_by = get_bits(&gb, 16);
1216 if (m->avctx->codec_id == AV_CODEC_ID_TRUEHD && shorten_by & 0x2000)
1217 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1218 else if (m->avctx->codec_id == AV_CODEC_ID_MLP && shorten_by != 0xD234)
1219 return AVERROR_INVALIDDATA;
1221 if (substr == m->max_decoded_substream)
1222 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1225 if (substream_parity_present[substr]) {
1226 uint8_t parity, checksum;
1228 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1229 goto substream_length_mismatch;
1231 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1232 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1234 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1235 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1236 if ( get_bits(&gb, 8) != checksum)
1237 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1240 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1241 goto substream_length_mismatch;
1244 if (!s->restart_seen)
1245 av_log(m->avctx, AV_LOG_ERROR,
1246 "No restart header present in substream %d.\n", substr);
1248 buf += substream_data_len[substr];
1251 rematrix_channels(m, m->max_decoded_substream);
1253 if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
1258 substream_length_mismatch:
1259 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1260 return AVERROR_INVALIDDATA;
1263 m->params_valid = 0;
1264 return AVERROR_INVALIDDATA;
1267 AVCodec ff_mlp_decoder = {
1269 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1270 .type = AVMEDIA_TYPE_AUDIO,
1271 .id = AV_CODEC_ID_MLP,
1272 .priv_data_size = sizeof(MLPDecodeContext),
1273 .init = mlp_decode_init,
1274 .decode = read_access_unit,
1275 .capabilities = AV_CODEC_CAP_DR1,
1278 #if CONFIG_TRUEHD_DECODER
1279 AVCodec ff_truehd_decoder = {
1281 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1282 .type = AVMEDIA_TYPE_AUDIO,
1283 .id = AV_CODEC_ID_TRUEHD,
1284 .priv_data_size = sizeof(MLPDecodeContext),
1285 .init = mlp_decode_init,
1286 .decode = read_access_unit,
1287 .capabilities = AV_CODEC_CAP_DR1,
1289 #endif /* CONFIG_TRUEHD_DECODER */