3 * Copyright (c) 2007-2008 Ian Caulfield
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 * @file libavcodec/mlpdec.c
30 #include "libavutil/intreadwrite.h"
32 #include "libavutil/crc.h"
34 #include "mlp_parser.h"
37 /** number of bits used for VLC lookup - longest Huffman code is 9 */
41 static const char* sample_message
=
42 "Please file a bug report following the instructions at "
43 "http://ffmpeg.org/bugreports.html and include "
44 "a sample of this file.";
46 typedef struct SubStream
{
47 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
51 /** restart header data */
52 //! The type of noise to be used in the rematrix stage.
55 //! The index of the first channel coded in this substream.
57 //! The index of the last channel coded in this substream.
59 //! The number of channels input into the rematrix stage.
60 uint8_t max_matrix_channel
;
61 //! For each channel output by the matrix, the output channel to map it to
62 uint8_t ch_assign
[MAX_CHANNELS
];
64 //! The left shift applied to random noise in 0x31ea substreams.
66 //! The current seed value for the pseudorandom noise generator(s).
67 uint32_t noisegen_seed
;
69 //! Set if the substream contains extra info to check the size of VLC blocks.
70 uint8_t data_check_present
;
72 //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
73 uint8_t param_presence_flags
;
74 #define PARAM_BLOCKSIZE (1 << 7)
75 #define PARAM_MATRIX (1 << 6)
76 #define PARAM_OUTSHIFT (1 << 5)
77 #define PARAM_QUANTSTEP (1 << 4)
78 #define PARAM_FIR (1 << 3)
79 #define PARAM_IIR (1 << 2)
80 #define PARAM_HUFFOFFSET (1 << 1)
81 #define PARAM_PRESENCE (1 << 0)
87 //! Number of matrices to be applied.
88 uint8_t num_primitive_matrices
;
90 //! matrix output channel
91 uint8_t matrix_out_ch
[MAX_MATRICES
];
93 //! Whether the LSBs of the matrix output are encoded in the bitstream.
94 uint8_t lsb_bypass
[MAX_MATRICES
];
95 //! Matrix coefficients, stored as 2.14 fixed point.
96 int32_t matrix_coeff
[MAX_MATRICES
][MAX_CHANNELS
];
97 //! Left shift to apply to noise values in 0x31eb substreams.
98 uint8_t matrix_noise_shift
[MAX_MATRICES
];
101 //! Left shift to apply to Huffman-decoded residuals.
102 uint8_t quant_step_size
[MAX_CHANNELS
];
104 //! number of PCM samples in current audio block
106 //! Number of PCM samples decoded so far in this frame.
109 //! Left shift to apply to decoded PCM values to get final 24-bit output.
110 int8_t output_shift
[MAX_CHANNELS
];
112 //! Running XOR of all output samples.
113 int32_t lossless_check_data
;
117 typedef struct MLPDecodeContext
{
118 AVCodecContext
*avctx
;
120 //! Current access unit being read has a major sync.
121 int is_major_sync_unit
;
123 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
124 uint8_t params_valid
;
126 //! Number of substreams contained within this stream.
127 uint8_t num_substreams
;
129 //! Index of the last substream to decode - further substreams are skipped.
130 uint8_t max_decoded_substream
;
132 //! number of PCM samples contained in each frame
133 int access_unit_size
;
134 //! next power of two above the number of samples in each frame
135 int access_unit_size_pow2
;
137 SubStream substream
[MAX_SUBSTREAMS
];
139 ChannelParams channel_params
[MAX_CHANNELS
];
142 int filter_changed
[MAX_CHANNELS
][NUM_FILTERS
];
144 int8_t noise_buffer
[MAX_BLOCKSIZE_POW2
];
145 int8_t bypassed_lsbs
[MAX_BLOCKSIZE
][MAX_CHANNELS
];
146 int32_t sample_buffer
[MAX_BLOCKSIZE
][MAX_CHANNELS
];
149 static VLC huff_vlc
[3];
151 /** Initialize static data, constant between all invocations of the codec. */
153 static av_cold
void init_static(void)
155 INIT_VLC_STATIC(&huff_vlc
[0], VLC_BITS
, 18,
156 &ff_mlp_huffman_tables
[0][0][1], 2, 1,
157 &ff_mlp_huffman_tables
[0][0][0], 2, 1, 512);
158 INIT_VLC_STATIC(&huff_vlc
[1], VLC_BITS
, 16,
159 &ff_mlp_huffman_tables
[1][0][1], 2, 1,
160 &ff_mlp_huffman_tables
[1][0][0], 2, 1, 512);
161 INIT_VLC_STATIC(&huff_vlc
[2], VLC_BITS
, 15,
162 &ff_mlp_huffman_tables
[2][0][1], 2, 1,
163 &ff_mlp_huffman_tables
[2][0][0], 2, 1, 512);
168 static inline int32_t calculate_sign_huff(MLPDecodeContext
*m
,
169 unsigned int substr
, unsigned int ch
)
171 ChannelParams
*cp
= &m
->channel_params
[ch
];
172 SubStream
*s
= &m
->substream
[substr
];
173 int lsb_bits
= cp
->huff_lsbs
- s
->quant_step_size
[ch
];
174 int sign_shift
= lsb_bits
+ (cp
->codebook
? 2 - cp
->codebook
: -1);
175 int32_t sign_huff_offset
= cp
->huff_offset
;
177 if (cp
->codebook
> 0)
178 sign_huff_offset
-= 7 << lsb_bits
;
181 sign_huff_offset
-= 1 << sign_shift
;
183 return sign_huff_offset
;
186 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
189 static inline int read_huff_channels(MLPDecodeContext
*m
, GetBitContext
*gbp
,
190 unsigned int substr
, unsigned int pos
)
192 SubStream
*s
= &m
->substream
[substr
];
193 unsigned int mat
, channel
;
195 for (mat
= 0; mat
< s
->num_primitive_matrices
; mat
++)
196 if (s
->lsb_bypass
[mat
])
197 m
->bypassed_lsbs
[pos
+ s
->blockpos
][mat
] = get_bits1(gbp
);
199 for (channel
= s
->min_channel
; channel
<= s
->max_channel
; channel
++) {
200 ChannelParams
*cp
= &m
->channel_params
[channel
];
201 int codebook
= cp
->codebook
;
202 int quant_step_size
= s
->quant_step_size
[channel
];
203 int lsb_bits
= cp
->huff_lsbs
- quant_step_size
;
207 result
= get_vlc2(gbp
, huff_vlc
[codebook
-1].table
,
208 VLC_BITS
, (9 + VLC_BITS
- 1) / VLC_BITS
);
214 result
= (result
<< lsb_bits
) + get_bits(gbp
, lsb_bits
);
216 result
+= cp
->sign_huff_offset
;
217 result
<<= quant_step_size
;
219 m
->sample_buffer
[pos
+ s
->blockpos
][channel
] = result
;
225 static av_cold
int mlp_decode_init(AVCodecContext
*avctx
)
227 MLPDecodeContext
*m
= avctx
->priv_data
;
232 for (substr
= 0; substr
< MAX_SUBSTREAMS
; substr
++)
233 m
->substream
[substr
].lossless_check_data
= 0xffffffff;
238 /** Read a major sync info header - contains high level information about
239 * the stream - sample rate, channel arrangement etc. Most of this
240 * information is not actually necessary for decoding, only for playback.
243 static int read_major_sync(MLPDecodeContext
*m
, GetBitContext
*gb
)
248 if (ff_mlp_read_major_sync(m
->avctx
, &mh
, gb
) != 0)
251 if (mh
.group1_bits
== 0) {
252 av_log(m
->avctx
, AV_LOG_ERROR
, "invalid/unknown bits per sample\n");
255 if (mh
.group2_bits
> mh
.group1_bits
) {
256 av_log(m
->avctx
, AV_LOG_ERROR
,
257 "Channel group 2 cannot have more bits per sample than group 1.\n");
261 if (mh
.group2_samplerate
&& mh
.group2_samplerate
!= mh
.group1_samplerate
) {
262 av_log(m
->avctx
, AV_LOG_ERROR
,
263 "Channel groups with differing sample rates are not currently supported.\n");
267 if (mh
.group1_samplerate
== 0) {
268 av_log(m
->avctx
, AV_LOG_ERROR
, "invalid/unknown sampling rate\n");
271 if (mh
.group1_samplerate
> MAX_SAMPLERATE
) {
272 av_log(m
->avctx
, AV_LOG_ERROR
,
273 "Sampling rate %d is greater than the supported maximum (%d).\n",
274 mh
.group1_samplerate
, MAX_SAMPLERATE
);
277 if (mh
.access_unit_size
> MAX_BLOCKSIZE
) {
278 av_log(m
->avctx
, AV_LOG_ERROR
,
279 "Block size %d is greater than the supported maximum (%d).\n",
280 mh
.access_unit_size
, MAX_BLOCKSIZE
);
283 if (mh
.access_unit_size_pow2
> MAX_BLOCKSIZE_POW2
) {
284 av_log(m
->avctx
, AV_LOG_ERROR
,
285 "Block size pow2 %d is greater than the supported maximum (%d).\n",
286 mh
.access_unit_size_pow2
, MAX_BLOCKSIZE_POW2
);
290 if (mh
.num_substreams
== 0)
292 if (m
->avctx
->codec_id
== CODEC_ID_MLP
&& mh
.num_substreams
> 2) {
293 av_log(m
->avctx
, AV_LOG_ERROR
, "MLP only supports up to 2 substreams.\n");
296 if (mh
.num_substreams
> MAX_SUBSTREAMS
) {
297 av_log(m
->avctx
, AV_LOG_ERROR
,
298 "Number of substreams %d is larger than the maximum supported "
299 "by the decoder. %s\n", mh
.num_substreams
, sample_message
);
303 m
->access_unit_size
= mh
.access_unit_size
;
304 m
->access_unit_size_pow2
= mh
.access_unit_size_pow2
;
306 m
->num_substreams
= mh
.num_substreams
;
307 m
->max_decoded_substream
= m
->num_substreams
- 1;
309 m
->avctx
->sample_rate
= mh
.group1_samplerate
;
310 m
->avctx
->frame_size
= mh
.access_unit_size
;
312 m
->avctx
->bits_per_raw_sample
= mh
.group1_bits
;
313 if (mh
.group1_bits
> 16)
314 m
->avctx
->sample_fmt
= SAMPLE_FMT_S32
;
316 m
->avctx
->sample_fmt
= SAMPLE_FMT_S16
;
319 for (substr
= 0; substr
< MAX_SUBSTREAMS
; substr
++)
320 m
->substream
[substr
].restart_seen
= 0;
325 /** Read a restart header from a block in a substream. This contains parameters
326 * required to decode the audio that do not change very often. Generally
327 * (always) present only in blocks following a major sync. */
329 static int read_restart_header(MLPDecodeContext
*m
, GetBitContext
*gbp
,
330 const uint8_t *buf
, unsigned int substr
)
332 SubStream
*s
= &m
->substream
[substr
];
336 uint8_t lossless_check
;
337 int start_count
= get_bits_count(gbp
);
338 const int max_matrix_channel
= m
->avctx
->codec_id
== CODEC_ID_MLP
339 ? MAX_MATRIX_CHANNEL_MLP
340 : MAX_MATRIX_CHANNEL_TRUEHD
;
342 sync_word
= get_bits(gbp
, 13);
343 s
->noise_type
= get_bits1(gbp
);
345 if ((m
->avctx
->codec_id
== CODEC_ID_MLP
&& s
->noise_type
) ||
346 sync_word
!= 0x31ea >> 1) {
347 av_log(m
->avctx
, AV_LOG_ERROR
,
348 "restart header sync incorrect (got 0x%04x)\n", sync_word
);
352 skip_bits(gbp
, 16); /* Output timestamp */
354 s
->min_channel
= get_bits(gbp
, 4);
355 s
->max_channel
= get_bits(gbp
, 4);
356 s
->max_matrix_channel
= get_bits(gbp
, 4);
358 if (s
->max_matrix_channel
> max_matrix_channel
) {
359 av_log(m
->avctx
, AV_LOG_ERROR
,
360 "Max matrix channel cannot be greater than %d.\n",
365 if (s
->max_channel
!= s
->max_matrix_channel
) {
366 av_log(m
->avctx
, AV_LOG_ERROR
,
367 "Max channel must be equal max matrix channel.\n");
371 if (s
->min_channel
> s
->max_channel
) {
372 av_log(m
->avctx
, AV_LOG_ERROR
,
373 "Substream min channel cannot be greater than max channel.\n");
377 if (m
->avctx
->request_channels
> 0
378 && s
->max_channel
+ 1 >= m
->avctx
->request_channels
379 && substr
< m
->max_decoded_substream
) {
380 av_log(m
->avctx
, AV_LOG_INFO
,
381 "Extracting %d channel downmix from substream %d. "
382 "Further substreams will be skipped.\n",
383 s
->max_channel
+ 1, substr
);
384 m
->max_decoded_substream
= substr
;
387 s
->noise_shift
= get_bits(gbp
, 4);
388 s
->noisegen_seed
= get_bits(gbp
, 23);
392 s
->data_check_present
= get_bits1(gbp
);
393 lossless_check
= get_bits(gbp
, 8);
394 if (substr
== m
->max_decoded_substream
395 && s
->lossless_check_data
!= 0xffffffff) {
396 tmp
= xor_32_to_8(s
->lossless_check_data
);
397 if (tmp
!= lossless_check
)
398 av_log(m
->avctx
, AV_LOG_WARNING
,
399 "Lossless check failed - expected %02x, calculated %02x.\n",
400 lossless_check
, tmp
);
405 memset(s
->ch_assign
, 0, sizeof(s
->ch_assign
));
407 for (ch
= 0; ch
<= s
->max_matrix_channel
; ch
++) {
408 int ch_assign
= get_bits(gbp
, 6);
409 if (ch_assign
> s
->max_matrix_channel
) {
410 av_log(m
->avctx
, AV_LOG_ERROR
,
411 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
412 ch
, ch_assign
, sample_message
);
415 s
->ch_assign
[ch_assign
] = ch
;
418 checksum
= ff_mlp_restart_checksum(buf
, get_bits_count(gbp
) - start_count
);
420 if (checksum
!= get_bits(gbp
, 8))
421 av_log(m
->avctx
, AV_LOG_ERROR
, "restart header checksum error\n");
423 /* Set default decoding parameters. */
424 s
->param_presence_flags
= 0xff;
425 s
->num_primitive_matrices
= 0;
427 s
->lossless_check_data
= 0;
429 memset(s
->output_shift
, 0, sizeof(s
->output_shift
));
430 memset(s
->quant_step_size
, 0, sizeof(s
->quant_step_size
));
432 for (ch
= s
->min_channel
; ch
<= s
->max_channel
; ch
++) {
433 ChannelParams
*cp
= &m
->channel_params
[ch
];
434 cp
->filter_params
[FIR
].order
= 0;
435 cp
->filter_params
[IIR
].order
= 0;
436 cp
->filter_params
[FIR
].shift
= 0;
437 cp
->filter_params
[IIR
].shift
= 0;
439 /* Default audio coding is 24-bit raw PCM. */
441 cp
->sign_huff_offset
= (-1) << 23;
446 if (substr
== m
->max_decoded_substream
)
447 m
->avctx
->channels
= s
->max_matrix_channel
+ 1;
452 /** Read parameters for one of the prediction filters. */
454 static int read_filter_params(MLPDecodeContext
*m
, GetBitContext
*gbp
,
455 unsigned int channel
, unsigned int filter
)
457 FilterParams
*fp
= &m
->channel_params
[channel
].filter_params
[filter
];
458 const int max_order
= filter
? MAX_IIR_ORDER
: MAX_FIR_ORDER
;
459 const char fchar
= filter
? 'I' : 'F';
462 // Filter is 0 for FIR, 1 for IIR.
465 if (m
->filter_changed
[channel
][filter
]++ > 1) {
466 av_log(m
->avctx
, AV_LOG_ERROR
, "Filters may change only once per access unit.\n");
470 order
= get_bits(gbp
, 4);
471 if (order
> max_order
) {
472 av_log(m
->avctx
, AV_LOG_ERROR
,
473 "%cIR filter order %d is greater than maximum %d.\n",
474 fchar
, order
, max_order
);
480 int coeff_bits
, coeff_shift
;
482 fp
->shift
= get_bits(gbp
, 4);
484 coeff_bits
= get_bits(gbp
, 5);
485 coeff_shift
= get_bits(gbp
, 3);
486 if (coeff_bits
< 1 || coeff_bits
> 16) {
487 av_log(m
->avctx
, AV_LOG_ERROR
,
488 "%cIR filter coeff_bits must be between 1 and 16.\n",
492 if (coeff_bits
+ coeff_shift
> 16) {
493 av_log(m
->avctx
, AV_LOG_ERROR
,
494 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
499 for (i
= 0; i
< order
; i
++)
500 fp
->coeff
[i
] = get_sbits(gbp
, coeff_bits
) << coeff_shift
;
502 if (get_bits1(gbp
)) {
503 int state_bits
, state_shift
;
506 av_log(m
->avctx
, AV_LOG_ERROR
,
507 "FIR filter has state data specified.\n");
511 state_bits
= get_bits(gbp
, 4);
512 state_shift
= get_bits(gbp
, 4);
514 /* TODO: Check validity of state data. */
516 for (i
= 0; i
< order
; i
++)
517 fp
->state
[i
] = get_sbits(gbp
, state_bits
) << state_shift
;
524 /** Read parameters for primitive matrices. */
526 static int read_matrix_params(MLPDecodeContext
*m
, unsigned int substr
, GetBitContext
*gbp
)
528 SubStream
*s
= &m
->substream
[substr
];
529 unsigned int mat
, ch
;
530 const int max_primitive_matrices
= m
->avctx
->codec_id
== CODEC_ID_MLP
532 : MAX_MATRICES_TRUEHD
;
534 if (m
->matrix_changed
++ > 1) {
535 av_log(m
->avctx
, AV_LOG_ERROR
, "Matrices may change only once per access unit.\n");
539 s
->num_primitive_matrices
= get_bits(gbp
, 4);
541 if (s
->num_primitive_matrices
> max_primitive_matrices
) {
542 av_log(m
->avctx
, AV_LOG_ERROR
,
543 "Number of primitive matrices cannot be greater than %d.\n",
544 max_primitive_matrices
);
548 for (mat
= 0; mat
< s
->num_primitive_matrices
; mat
++) {
549 int frac_bits
, max_chan
;
550 s
->matrix_out_ch
[mat
] = get_bits(gbp
, 4);
551 frac_bits
= get_bits(gbp
, 4);
552 s
->lsb_bypass
[mat
] = get_bits1(gbp
);
554 if (s
->matrix_out_ch
[mat
] > s
->max_matrix_channel
) {
555 av_log(m
->avctx
, AV_LOG_ERROR
,
556 "Invalid channel %d specified as output from matrix.\n",
557 s
->matrix_out_ch
[mat
]);
560 if (frac_bits
> 14) {
561 av_log(m
->avctx
, AV_LOG_ERROR
,
562 "Too many fractional bits specified.\n");
566 max_chan
= s
->max_matrix_channel
;
570 for (ch
= 0; ch
<= max_chan
; ch
++) {
573 coeff_val
= get_sbits(gbp
, frac_bits
+ 2);
575 s
->matrix_coeff
[mat
][ch
] = coeff_val
<< (14 - frac_bits
);
579 s
->matrix_noise_shift
[mat
] = get_bits(gbp
, 4);
581 s
->matrix_noise_shift
[mat
] = 0;
587 /** Read channel parameters. */
589 static int read_channel_params(MLPDecodeContext
*m
, unsigned int substr
,
590 GetBitContext
*gbp
, unsigned int ch
)
592 ChannelParams
*cp
= &m
->channel_params
[ch
];
593 FilterParams
*fir
= &cp
->filter_params
[FIR
];
594 FilterParams
*iir
= &cp
->filter_params
[IIR
];
595 SubStream
*s
= &m
->substream
[substr
];
597 if (s
->param_presence_flags
& PARAM_FIR
)
599 if (read_filter_params(m
, gbp
, ch
, FIR
) < 0)
602 if (s
->param_presence_flags
& PARAM_IIR
)
604 if (read_filter_params(m
, gbp
, ch
, IIR
) < 0)
607 if (fir
->order
+ iir
->order
> 8) {
608 av_log(m
->avctx
, AV_LOG_ERROR
, "Total filter orders too high.\n");
612 if (fir
->order
&& iir
->order
&&
613 fir
->shift
!= iir
->shift
) {
614 av_log(m
->avctx
, AV_LOG_ERROR
,
615 "FIR and IIR filters must use the same precision.\n");
618 /* The FIR and IIR filters must have the same precision.
619 * To simplify the filtering code, only the precision of the
620 * FIR filter is considered. If only the IIR filter is employed,
621 * the FIR filter precision is set to that of the IIR filter, so
622 * that the filtering code can use it. */
623 if (!fir
->order
&& iir
->order
)
624 fir
->shift
= iir
->shift
;
626 if (s
->param_presence_flags
& PARAM_HUFFOFFSET
)
628 cp
->huff_offset
= get_sbits(gbp
, 15);
630 cp
->codebook
= get_bits(gbp
, 2);
631 cp
->huff_lsbs
= get_bits(gbp
, 5);
633 if (cp
->huff_lsbs
> 24) {
634 av_log(m
->avctx
, AV_LOG_ERROR
, "Invalid huff_lsbs.\n");
638 cp
->sign_huff_offset
= calculate_sign_huff(m
, substr
, ch
);
643 /** Read decoding parameters that change more often than those in the restart
646 static int read_decoding_params(MLPDecodeContext
*m
, GetBitContext
*gbp
,
649 SubStream
*s
= &m
->substream
[substr
];
652 if (s
->param_presence_flags
& PARAM_PRESENCE
)
654 s
->param_presence_flags
= get_bits(gbp
, 8);
656 if (s
->param_presence_flags
& PARAM_BLOCKSIZE
)
657 if (get_bits1(gbp
)) {
658 s
->blocksize
= get_bits(gbp
, 9);
659 if (s
->blocksize
< 8 || s
->blocksize
> m
->access_unit_size
) {
660 av_log(m
->avctx
, AV_LOG_ERROR
, "Invalid blocksize.");
666 if (s
->param_presence_flags
& PARAM_MATRIX
)
668 if (read_matrix_params(m
, substr
, gbp
) < 0)
671 if (s
->param_presence_flags
& PARAM_OUTSHIFT
)
673 for (ch
= 0; ch
<= s
->max_matrix_channel
; ch
++)
674 s
->output_shift
[ch
] = get_sbits(gbp
, 4);
676 if (s
->param_presence_flags
& PARAM_QUANTSTEP
)
678 for (ch
= 0; ch
<= s
->max_channel
; ch
++) {
679 ChannelParams
*cp
= &m
->channel_params
[ch
];
681 s
->quant_step_size
[ch
] = get_bits(gbp
, 4);
683 cp
->sign_huff_offset
= calculate_sign_huff(m
, substr
, ch
);
686 for (ch
= s
->min_channel
; ch
<= s
->max_channel
; ch
++)
688 if (read_channel_params(m
, substr
, gbp
, ch
) < 0)
694 #define MSB_MASK(bits) (-1u << bits)
696 /** Generate PCM samples using the prediction filters and residual values
697 * read from the data stream, and update the filter state. */
699 static void filter_channel(MLPDecodeContext
*m
, unsigned int substr
,
700 unsigned int channel
)
702 SubStream
*s
= &m
->substream
[substr
];
703 int32_t fir_state_buffer
[MAX_BLOCKSIZE
+ MAX_FIR_ORDER
];
704 int32_t iir_state_buffer
[MAX_BLOCKSIZE
+ MAX_IIR_ORDER
];
705 int32_t *firbuf
= fir_state_buffer
+ MAX_BLOCKSIZE
;
706 int32_t *iirbuf
= iir_state_buffer
+ MAX_BLOCKSIZE
;
707 FilterParams
*fir
= &m
->channel_params
[channel
].filter_params
[FIR
];
708 FilterParams
*iir
= &m
->channel_params
[channel
].filter_params
[IIR
];
709 unsigned int filter_shift
= fir
->shift
;
710 int32_t mask
= MSB_MASK(s
->quant_step_size
[channel
]);
713 memcpy(firbuf
, fir
->state
, MAX_FIR_ORDER
* sizeof(int32_t));
714 memcpy(iirbuf
, iir
->state
, MAX_IIR_ORDER
* sizeof(int32_t));
716 for (i
= 0; i
< s
->blocksize
; i
++) {
717 int32_t residual
= m
->sample_buffer
[i
+ s
->blockpos
][channel
];
722 /* TODO: Move this code to DSPContext? */
724 for (order
= 0; order
< fir
->order
; order
++)
725 accum
+= (int64_t) firbuf
[order
] * fir
->coeff
[order
];
726 for (order
= 0; order
< iir
->order
; order
++)
727 accum
+= (int64_t) iirbuf
[order
] * iir
->coeff
[order
];
729 accum
= accum
>> filter_shift
;
730 result
= (accum
+ residual
) & mask
;
733 *--iirbuf
= result
- accum
;
735 m
->sample_buffer
[i
+ s
->blockpos
][channel
] = result
;
738 memcpy(fir
->state
, firbuf
, MAX_FIR_ORDER
* sizeof(int32_t));
739 memcpy(iir
->state
, iirbuf
, MAX_IIR_ORDER
* sizeof(int32_t));
742 /** Read a block of PCM residual data (or actual if no filtering active). */
744 static int read_block_data(MLPDecodeContext
*m
, GetBitContext
*gbp
,
747 SubStream
*s
= &m
->substream
[substr
];
748 unsigned int i
, ch
, expected_stream_pos
= 0;
750 if (s
->data_check_present
) {
751 expected_stream_pos
= get_bits_count(gbp
);
752 expected_stream_pos
+= get_bits(gbp
, 16);
753 av_log(m
->avctx
, AV_LOG_WARNING
, "This file contains some features "
754 "we have not tested yet. %s\n", sample_message
);
757 if (s
->blockpos
+ s
->blocksize
> m
->access_unit_size
) {
758 av_log(m
->avctx
, AV_LOG_ERROR
, "too many audio samples in frame\n");
762 memset(&m
->bypassed_lsbs
[s
->blockpos
][0], 0,
763 s
->blocksize
* sizeof(m
->bypassed_lsbs
[0]));
765 for (i
= 0; i
< s
->blocksize
; i
++)
766 if (read_huff_channels(m
, gbp
, substr
, i
) < 0)
769 for (ch
= s
->min_channel
; ch
<= s
->max_channel
; ch
++)
770 filter_channel(m
, substr
, ch
);
772 s
->blockpos
+= s
->blocksize
;
774 if (s
->data_check_present
) {
775 if (get_bits_count(gbp
) != expected_stream_pos
)
776 av_log(m
->avctx
, AV_LOG_ERROR
, "block data length mismatch\n");
783 /** Data table used for TrueHD noise generation function. */
785 static const int8_t noise_table
[256] = {
786 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
787 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
788 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
789 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
790 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
791 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
792 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
793 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
794 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
795 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
796 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
797 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
798 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
799 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
800 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
801 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
804 /** Noise generation functions.
805 * I'm not sure what these are for - they seem to be some kind of pseudorandom
806 * sequence generators, used to generate noise data which is used when the
807 * channels are rematrixed. I'm not sure if they provide a practical benefit
808 * to compression, or just obfuscate the decoder. Are they for some kind of
811 /** Generate two channels of noise, used in the matrix when
812 * restart sync word == 0x31ea. */
814 static void generate_2_noise_channels(MLPDecodeContext
*m
, unsigned int substr
)
816 SubStream
*s
= &m
->substream
[substr
];
818 uint32_t seed
= s
->noisegen_seed
;
819 unsigned int maxchan
= s
->max_matrix_channel
;
821 for (i
= 0; i
< s
->blockpos
; i
++) {
822 uint16_t seed_shr7
= seed
>> 7;
823 m
->sample_buffer
[i
][maxchan
+1] = ((int8_t)(seed
>> 15)) << s
->noise_shift
;
824 m
->sample_buffer
[i
][maxchan
+2] = ((int8_t) seed_shr7
) << s
->noise_shift
;
826 seed
= (seed
<< 16) ^ seed_shr7
^ (seed_shr7
<< 5);
829 s
->noisegen_seed
= seed
;
832 /** Generate a block of noise, used when restart sync word == 0x31eb. */
834 static void fill_noise_buffer(MLPDecodeContext
*m
, unsigned int substr
)
836 SubStream
*s
= &m
->substream
[substr
];
838 uint32_t seed
= s
->noisegen_seed
;
840 for (i
= 0; i
< m
->access_unit_size_pow2
; i
++) {
841 uint8_t seed_shr15
= seed
>> 15;
842 m
->noise_buffer
[i
] = noise_table
[seed_shr15
];
843 seed
= (seed
<< 8) ^ seed_shr15
^ (seed_shr15
<< 5);
846 s
->noisegen_seed
= seed
;
850 /** Apply the channel matrices in turn to reconstruct the original audio
853 static void rematrix_channels(MLPDecodeContext
*m
, unsigned int substr
)
855 SubStream
*s
= &m
->substream
[substr
];
856 unsigned int mat
, src_ch
, i
;
857 unsigned int maxchan
;
859 maxchan
= s
->max_matrix_channel
;
860 if (!s
->noise_type
) {
861 generate_2_noise_channels(m
, substr
);
864 fill_noise_buffer(m
, substr
);
867 for (mat
= 0; mat
< s
->num_primitive_matrices
; mat
++) {
868 int matrix_noise_shift
= s
->matrix_noise_shift
[mat
];
869 unsigned int dest_ch
= s
->matrix_out_ch
[mat
];
870 int32_t mask
= MSB_MASK(s
->quant_step_size
[dest_ch
]);
871 int32_t *coeffs
= s
->matrix_coeff
[mat
];
872 int index
= s
->num_primitive_matrices
- mat
;
873 int index2
= 2 * index
+ 1;
875 /* TODO: DSPContext? */
877 for (i
= 0; i
< s
->blockpos
; i
++) {
878 int32_t bypassed_lsb
= m
->bypassed_lsbs
[i
][mat
];
879 int32_t *samples
= m
->sample_buffer
[i
];
882 for (src_ch
= 0; src_ch
<= maxchan
; src_ch
++)
883 accum
+= (int64_t) samples
[src_ch
] * coeffs
[src_ch
];
885 if (matrix_noise_shift
) {
886 index
&= m
->access_unit_size_pow2
- 1;
887 accum
+= m
->noise_buffer
[index
] << (matrix_noise_shift
+ 7);
891 samples
[dest_ch
] = ((accum
>> 14) & mask
) + bypassed_lsb
;
896 /** Write the audio data into the output buffer. */
898 static int output_data_internal(MLPDecodeContext
*m
, unsigned int substr
,
899 uint8_t *data
, unsigned int *data_size
, int is32
)
901 SubStream
*s
= &m
->substream
[substr
];
902 unsigned int i
, out_ch
= 0;
903 int32_t *data_32
= (int32_t*) data
;
904 int16_t *data_16
= (int16_t*) data
;
906 if (*data_size
< (s
->max_channel
+ 1) * s
->blockpos
* (is32
? 4 : 2))
909 for (i
= 0; i
< s
->blockpos
; i
++) {
910 for (out_ch
= 0; out_ch
<= s
->max_matrix_channel
; out_ch
++) {
911 int mat_ch
= s
->ch_assign
[out_ch
];
912 int32_t sample
= m
->sample_buffer
[i
][mat_ch
]
913 << s
->output_shift
[mat_ch
];
914 s
->lossless_check_data
^= (sample
& 0xffffff) << mat_ch
;
915 if (is32
) *data_32
++ = sample
<< 8;
916 else *data_16
++ = sample
>> 8;
920 *data_size
= i
* out_ch
* (is32
? 4 : 2);
925 static int output_data(MLPDecodeContext
*m
, unsigned int substr
,
926 uint8_t *data
, unsigned int *data_size
)
928 if (m
->avctx
->sample_fmt
== SAMPLE_FMT_S32
)
929 return output_data_internal(m
, substr
, data
, data_size
, 1);
931 return output_data_internal(m
, substr
, data
, data_size
, 0);
935 /** Read an access unit from the stream.
936 * Returns < 0 on error, 0 if not enough data is present in the input stream
937 * otherwise returns the number of bytes consumed. */
939 static int read_access_unit(AVCodecContext
*avctx
, void* data
, int *data_size
,
942 const uint8_t *buf
= avpkt
->data
;
943 int buf_size
= avpkt
->size
;
944 MLPDecodeContext
*m
= avctx
->priv_data
;
946 unsigned int length
, substr
;
947 unsigned int substream_start
;
948 unsigned int header_size
= 4;
949 unsigned int substr_header_size
= 0;
950 uint8_t substream_parity_present
[MAX_SUBSTREAMS
];
951 uint16_t substream_data_len
[MAX_SUBSTREAMS
];
957 length
= (AV_RB16(buf
) & 0xfff) * 2;
959 if (length
> buf_size
)
962 init_get_bits(&gb
, (buf
+ 4), (length
- 4) * 8);
964 m
->is_major_sync_unit
= 0;
965 if (show_bits_long(&gb
, 31) == (0xf8726fba >> 1)) {
966 if (read_major_sync(m
, &gb
) < 0)
968 m
->is_major_sync_unit
= 1;
972 if (!m
->params_valid
) {
973 av_log(m
->avctx
, AV_LOG_WARNING
,
974 "Stream parameters not seen; skipping frame.\n");
981 for (substr
= 0; substr
< m
->num_substreams
; substr
++) {
982 int extraword_present
, checkdata_present
, end
, nonrestart_substr
;
984 extraword_present
= get_bits1(&gb
);
985 nonrestart_substr
= get_bits1(&gb
);
986 checkdata_present
= get_bits1(&gb
);
989 end
= get_bits(&gb
, 12) * 2;
991 substr_header_size
+= 2;
993 if (extraword_present
) {
994 if (m
->avctx
->codec_id
== CODEC_ID_MLP
) {
995 av_log(m
->avctx
, AV_LOG_ERROR
, "There must be no extraword for MLP.\n");
999 substr_header_size
+= 2;
1002 if (!(nonrestart_substr
^ m
->is_major_sync_unit
)) {
1003 av_log(m
->avctx
, AV_LOG_ERROR
, "Invalid nonrestart_substr.\n");
1007 if (end
+ header_size
+ substr_header_size
> length
) {
1008 av_log(m
->avctx
, AV_LOG_ERROR
,
1009 "Indicated length of substream %d data goes off end of "
1010 "packet.\n", substr
);
1011 end
= length
- header_size
- substr_header_size
;
1014 if (end
< substream_start
) {
1015 av_log(avctx
, AV_LOG_ERROR
,
1016 "Indicated end offset of substream %d data "
1017 "is smaller than calculated start offset.\n",
1022 if (substr
> m
->max_decoded_substream
)
1025 substream_parity_present
[substr
] = checkdata_present
;
1026 substream_data_len
[substr
] = end
- substream_start
;
1027 substream_start
= end
;
1030 parity_bits
= ff_mlp_calculate_parity(buf
, 4);
1031 parity_bits
^= ff_mlp_calculate_parity(buf
+ header_size
, substr_header_size
);
1033 if ((((parity_bits
>> 4) ^ parity_bits
) & 0xF) != 0xF) {
1034 av_log(avctx
, AV_LOG_ERROR
, "Parity check failed.\n");
1038 buf
+= header_size
+ substr_header_size
;
1040 for (substr
= 0; substr
<= m
->max_decoded_substream
; substr
++) {
1041 SubStream
*s
= &m
->substream
[substr
];
1042 init_get_bits(&gb
, buf
, substream_data_len
[substr
] * 8);
1044 m
->matrix_changed
= 0;
1045 memset(m
->filter_changed
, 0, sizeof(m
->filter_changed
));
1049 if (get_bits1(&gb
)) {
1050 if (get_bits1(&gb
)) {
1051 /* A restart header should be present. */
1052 if (read_restart_header(m
, &gb
, buf
, substr
) < 0)
1054 s
->restart_seen
= 1;
1057 if (!s
->restart_seen
)
1059 if (read_decoding_params(m
, &gb
, substr
) < 0)
1063 if (!s
->restart_seen
)
1066 if (read_block_data(m
, &gb
, substr
) < 0)
1069 if (get_bits_count(&gb
) >= substream_data_len
[substr
] * 8)
1070 goto substream_length_mismatch
;
1072 } while (!get_bits1(&gb
));
1074 skip_bits(&gb
, (-get_bits_count(&gb
)) & 15);
1076 if (substream_data_len
[substr
] * 8 - get_bits_count(&gb
) >= 32) {
1079 if (get_bits(&gb
, 16) != 0xD234)
1082 shorten_by
= get_bits(&gb
, 16);
1083 if (m
->avctx
->codec_id
== CODEC_ID_TRUEHD
&& shorten_by
& 0x2000)
1084 s
->blockpos
-= FFMIN(shorten_by
& 0x1FFF, s
->blockpos
);
1085 else if (m
->avctx
->codec_id
== CODEC_ID_MLP
&& shorten_by
!= 0xD234)
1088 if (substr
== m
->max_decoded_substream
)
1089 av_log(m
->avctx
, AV_LOG_INFO
, "End of stream indicated.\n");
1092 if (substream_parity_present
[substr
]) {
1093 uint8_t parity
, checksum
;
1095 if (substream_data_len
[substr
] * 8 - get_bits_count(&gb
) != 16)
1096 goto substream_length_mismatch
;
1098 parity
= ff_mlp_calculate_parity(buf
, substream_data_len
[substr
] - 2);
1099 checksum
= ff_mlp_checksum8 (buf
, substream_data_len
[substr
] - 2);
1101 if ((get_bits(&gb
, 8) ^ parity
) != 0xa9 )
1102 av_log(m
->avctx
, AV_LOG_ERROR
, "Substream %d parity check failed.\n", substr
);
1103 if ( get_bits(&gb
, 8) != checksum
)
1104 av_log(m
->avctx
, AV_LOG_ERROR
, "Substream %d checksum failed.\n" , substr
);
1107 if (substream_data_len
[substr
] * 8 != get_bits_count(&gb
))
1108 goto substream_length_mismatch
;
1111 if (!s
->restart_seen
)
1112 av_log(m
->avctx
, AV_LOG_ERROR
,
1113 "No restart header present in substream %d.\n", substr
);
1115 buf
+= substream_data_len
[substr
];
1118 rematrix_channels(m
, m
->max_decoded_substream
);
1120 if (output_data(m
, m
->max_decoded_substream
, data
, data_size
) < 0)
1125 substream_length_mismatch
:
1126 av_log(m
->avctx
, AV_LOG_ERROR
, "substream %d length mismatch\n", substr
);
1130 m
->params_valid
= 0;
1134 #if CONFIG_MLP_DECODER
1135 AVCodec mlp_decoder
= {
1139 sizeof(MLPDecodeContext
),
1144 .long_name
= NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1146 #endif /* CONFIG_MLP_DECODER */
1148 #if CONFIG_TRUEHD_DECODER
1149 AVCodec truehd_decoder
= {
1153 sizeof(MLPDecodeContext
),
1158 .long_name
= NULL_IF_CONFIG_SMALL("TrueHD"),
1160 #endif /* CONFIG_TRUEHD_DECODER */