falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
14612 Commits
32 Branches
369MB
Tree: bbdfa06d43
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from 'bbdfa06d43'
${ noResults }
FFmpeg/libavcodec/mlpdec.c

1181 lines
39KB
Raw Blame History 

  1. /*

  2.  * MLP decoder

  3.  * Copyright (c) 2007-2008 Ian Caulfield

  4.  *

  5.  * This file is part of FFmpeg.

  6.  *

  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.

  11.  *

  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.

  16.  *

  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

  20.  */

  21. 

  22. /**

  23.  * @file mlpdec.c

  24.  * MLP decoder

  25.  */

  26. 

  27. #include "avcodec.h"

  28. #include "libavutil/intreadwrite.h"

  29. #include "bitstream.h"

  30. #include "libavutil/crc.h"

  31. #include "parser.h"

  32. #include "mlp_parser.h"

  33. 

  34. /** Maximum number of channels that can be decoded. */

  35. #define MAX_CHANNELS        16

  36. 

  37. /** Maximum number of matrices used in decoding. Most streams have one matrix

  38.  *  per output channel, but some rematrix a channel (usually 0) more than once.

  39.  */

  40. 

  41. #define MAX_MATRICES        15

  42. 

  43. /** Maximum number of substreams that can be decoded. This could also be set

  44.  *  higher, but again I haven't seen any examples with more than two. */

  45. #define MAX_SUBSTREAMS      2

  46. 

  47. /** Maximum sample frequency seen in files. */

  48. #define MAX_SAMPLERATE      192000

  49. 

  50. /** The maximum number of audio samples within one access unit. */

  51. #define MAX_BLOCKSIZE       (40 * (MAX_SAMPLERATE / 48000))

  52. /** The next power of two greater than MAX_BLOCKSIZE. */

  53. #define MAX_BLOCKSIZE_POW2  (64 * (MAX_SAMPLERATE / 48000))

  54. 

  55. /** Number of allowed filters. */

  56. #define NUM_FILTERS         2

  57. 

  58. /** The maximum number of taps in either the IIR or FIR filter.

  59.  *  I believe MLP actually specifies the maximum order for IIR filters as four,

  60.  *  and that the sum of the orders of both filters must be <= 8. */

  61. #define MAX_FILTER_ORDER    8

  62. 

  63. /** Number of bits used for VLC lookup - longest huffman code is 9. */

  64. #define VLC_BITS            9

  65. 

  66. 

  67. static const char* sample_message =

  68.     "Please file a bug report following the instructions at "

  69.     "http://ffmpeg.mplayerhq.hu/bugreports.html and include "

  70.     "a sample of this file.";

  71. 

  72. typedef struct SubStream {

  73.     //! Set if a valid restart header has been read. Otherwise the substream can not be decoded.

  74.     uint8_t     restart_seen;

  75. 

  76.     //@{

  77.     /** restart header data */

  78.     //! The type of noise to be used in the rematrix stage.

  79.     uint16_t    noise_type;

  80. 

  81.     //! The index of the first channel coded in this substream.

  82.     uint8_t     min_channel;

  83.     //! The index of the last channel coded in this substream.

  84.     uint8_t     max_channel;

  85.     //! The number of channels input into the rematrix stage.

  86.     uint8_t     max_matrix_channel;

  87. 

  88.     //! The left shift applied to random noise in 0x31ea substreams.

  89.     uint8_t     noise_shift;

  90.     //! The current seed value for the pseudorandom noise generator(s).

  91.     uint32_t    noisegen_seed;

  92. 

  93.     //! Set if the substream contains extra info to check the size of VLC blocks.

  94.     uint8_t     data_check_present;

  95. 

  96.     //! Bitmask of which parameter sets are conveyed in a decoding parameter block.

  97.     uint8_t     param_presence_flags;

  98. #define PARAM_BLOCKSIZE     (1 << 7)

  99. #define PARAM_MATRIX        (1 << 6)

  100. #define PARAM_OUTSHIFT      (1 << 5)

  101. #define PARAM_QUANTSTEP     (1 << 4)

  102. #define PARAM_FIR           (1 << 3)

  103. #define PARAM_IIR           (1 << 2)

  104. #define PARAM_HUFFOFFSET    (1 << 1)

  105.     //@}

  106. 

  107.     //@{

  108.     /** matrix data */

  109. 

  110.     //! Number of matrices to be applied.

  111.     uint8_t     num_primitive_matrices;

  112. 

  113.     //! matrix output channel

  114.     uint8_t     matrix_out_ch[MAX_MATRICES];

  115. 

  116.     //! Whether the LSBs of the matrix output are encoded in the bitstream.

  117.     uint8_t     lsb_bypass[MAX_MATRICES];

  118.     //! Matrix coefficients, stored as 2.14 fixed point.

  119.     int32_t     matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];

  120.     //! Left shift to apply to noise values in 0x31eb substreams.

  121.     uint8_t     matrix_noise_shift[MAX_MATRICES];

  122.     //@}

  123. 

  124.     //! Left shift to apply to huffman-decoded residuals.

  125.     uint8_t     quant_step_size[MAX_CHANNELS];

  126. 

  127.     //! Number of PCM samples in current audio block.

  128.     uint16_t    blocksize;

  129.     //! Number of PCM samples decoded so far in this frame.

  130.     uint16_t    blockpos;

  131. 

  132.     //! Left shift to apply to decoded PCM values to get final 24-bit output.

  133.     int8_t      output_shift[MAX_CHANNELS];

  134. 

  135.     //! Running XOR of all output samples.

  136.     int32_t     lossless_check_data;

  137. 

  138. } SubStream;

  139. 

  140. typedef struct MLPDecodeContext {

  141.     AVCodecContext *avctx;

  142. 

  143.     //! Set if a valid major sync block has been read. Otherwise no decoding is possible.

  144.     uint8_t     params_valid;

  145. 

  146.     //! Number of substreams contained within this stream.

  147.     uint8_t     num_substreams;

  148. 

  149.     //! Index of the last substream to decode - further substreams are skipped.

  150.     uint8_t     max_decoded_substream;

  151. 

  152.     //! Number of PCM samples contained in each frame.

  153.     int         access_unit_size;

  154.     //! Next power of two above the number of samples in each frame.

  155.     int         access_unit_size_pow2;

  156. 

  157.     SubStream   substream[MAX_SUBSTREAMS];

  158. 

  159.     //@{

  160.     /** filter data */

  161. #define FIR 0

  162. #define IIR 1

  163.     //! Number of taps in filter.

  164.     uint8_t     filter_order[MAX_CHANNELS][NUM_FILTERS];

  165.     //! Right shift to apply to output of filter.

  166.     uint8_t     filter_shift[MAX_CHANNELS][NUM_FILTERS];

  167. 

  168.     int32_t     filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];

  169.     int32_t     filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];

  170.     //@}

  171. 

  172.     //@{

  173.     /** sample data coding infomation */

  174.     //! Offset to apply to residual values.

  175.     int16_t     huff_offset[MAX_CHANNELS];

  176.     //! Sign/rounding corrected version of huff_offset.

  177.     int32_t     sign_huff_offset[MAX_CHANNELS];

  178.     //! Which VLC codebook to use to read residuals.

  179.     uint8_t     codebook[MAX_CHANNELS];

  180.     //! Size of residual suffix not encoded using VLC.

  181.     uint8_t     huff_lsbs[MAX_CHANNELS];

  182.     //@}

  183. 

  184.     int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];

  185.     int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];

  186.     int32_t     sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];

  187. } MLPDecodeContext;

  188. 

  189. /** Tables defining the huffman codes.

  190.  *  There are three entropy coding methods used in MLP (four if you count

  191.  *  "none" as a method). These use the same sequences for codes starting with

  192.  *  00 or 01, but have different codes starting with 1. */

  193. 

  194. static const uint8_t huffman_tables[3][18][2] = {

  195.     {    /* huffman table 0, -7 - +10 */

  196.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  197.         {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},

  198.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

  199.     }, { /* huffman table 1, -7 - +8 */

  200.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  201.         {0x02, 2}, {0x03, 2},

  202.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

  203.     }, { /* huffman table 2, -7 - +7 */

  204.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  205.         {0x01, 1},

  206.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

  207.     }

  208. };

  209. 

  210. static VLC huff_vlc[3];

  211. 

  212. static int crc_init = 0;

  213. static AVCRC crc_63[1024];

  214. static AVCRC crc_1D[1024];

  215. 

  216. 

  217. /** Initialize static data, constant between all invocations of the codec. */

  218. 

  219. static av_cold void init_static()

  220. {

  221.     INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,

  222.                 &huffman_tables[0][0][1], 2, 1,

  223.                 &huffman_tables[0][0][0], 2, 1, 512);

  224.     INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,

  225.                 &huffman_tables[1][0][1], 2, 1,

  226.                 &huffman_tables[1][0][0], 2, 1, 512);

  227.     INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,

  228.                 &huffman_tables[2][0][1], 2, 1,

  229.                 &huffman_tables[2][0][0], 2, 1, 512);

  230. 

  231.     if (!crc_init) {

  232.         av_crc_init(crc_63, 0,  8,   0x63, sizeof(crc_63));

  233.         av_crc_init(crc_1D, 0,  8,   0x1D, sizeof(crc_1D));

  234.         crc_init = 1;

  235.     }

  236. }

  237. 

  238. 

  239. /** MLP uses checksums that seem to be based on the standard CRC algorithm,

  240.  *  but not (in implementation terms, the table lookup and XOR are reversed).

  241.  *  We can implement this behavior using a standard av_crc on all but the

  242.  *  last element, then XOR that with the last element. */

  243. 

  244. static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)

  245. {

  246.     uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c

  247.     checksum ^= buf[buf_size-1];

  248.     return checksum;

  249. }

  250. 

  251. /** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8

  252.  *  number of bits, starting two bits into the first byte of buf. */

  253. 

  254. static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)

  255. {

  256.     int i;

  257.     int num_bytes = (bit_size + 2) / 8;

  258. 

  259.     int crc = crc_1D[buf[0] & 0x3f];

  260.     crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2);

  261.     crc ^= buf[num_bytes - 1];

  262. 

  263.     for (i = 0; i < ((bit_size + 2) & 7); i++) {

  264.         crc <<= 1;

  265.         if (crc & 0x100)

  266.             crc ^= 0x11D;

  267.         crc ^= (buf[num_bytes] >> (7 - i)) & 1;

  268.     }

  269. 

  270.     return crc;

  271. }

  272. 

  273. static inline int32_t calculate_sign_huff(MLPDecodeContext *m,

  274.                                           unsigned int substr, unsigned int ch)

  275. {

  276.     SubStream *s = &m->substream[substr];

  277.     int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch];

  278.     int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1);

  279.     int32_t sign_huff_offset = m->huff_offset[ch];

  280. 

  281.     if (m->codebook[ch] > 0)

  282.         sign_huff_offset -= 7 << lsb_bits;

  283. 

  284.     if (sign_shift >= 0)

  285.         sign_huff_offset -= 1 << sign_shift;

  286. 

  287.     return sign_huff_offset;

  288. }

  289. 

  290. /** Read a sample, consisting of either, both or neither of entropy-coded MSBs

  291.  *  and plain LSBs. */

  292. 

  293. static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,

  294.                                      unsigned int substr, unsigned int pos)

  295. {

  296.     SubStream *s = &m->substream[substr];

  297.     unsigned int mat, channel;

  298. 

  299.     for (mat = 0; mat < s->num_primitive_matrices; mat++)

  300.         if (s->lsb_bypass[mat])

  301.             m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);

  302. 

  303.     for (channel = s->min_channel; channel <= s->max_channel; channel++) {

  304.         int codebook = m->codebook[channel];

  305.         int quant_step_size = s->quant_step_size[channel];

  306.         int lsb_bits = m->huff_lsbs[channel] - quant_step_size;

  307.         int result = 0;

  308. 

  309.         if (codebook > 0)

  310.             result = get_vlc2(gbp, huff_vlc[codebook-1].table,

  311.                             VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);

  312. 

  313.         if (result < 0)

  314.             return -1;

  315. 

  316.         if (lsb_bits > 0)

  317.             result = (result << lsb_bits) + get_bits(gbp, lsb_bits);

  318. 

  319.         result  += m->sign_huff_offset[channel];

  320.         result <<= quant_step_size;

  321. 

  322.         m->sample_buffer[pos + s->blockpos][channel] = result;

  323.     }

  324. 

  325.     return 0;

  326. }

  327. 

  328. static av_cold int mlp_decode_init(AVCodecContext *avctx)

  329. {

  330.     MLPDecodeContext *m = avctx->priv_data;

  331.     int substr;

  332. 

  333.     init_static();

  334.     m->avctx = avctx;

  335.     for (substr = 0; substr < MAX_SUBSTREAMS; substr++)

  336.         m->substream[substr].lossless_check_data = 0xffffffff;

  337.     return 0;

  338. }

  339. 

  340. /** Read a major sync info header - contains high level information about

  341.  *  the stream - sample rate, channel arrangement etc. Most of this

  342.  *  information is not actually necessary for decoding, only for playback.

  343.  */

  344. 

  345. static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)

  346. {

  347.     MLPHeaderInfo mh;

  348.     int substr;

  349. 

  350.     if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)

  351.         return -1;

  352. 

  353.     if (mh.group1_bits == 0) {

  354.         av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown bits per sample\n");

  355.         return -1;

  356.     }

  357.     if (mh.group2_bits > mh.group1_bits) {

  358.         av_log(m->avctx, AV_LOG_ERROR,

  359.                "Channel group 2 cannot have more bits per sample than group 1\n");

  360.         return -1;

  361.     }

  362. 

  363.     if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {

  364.         av_log(m->avctx, AV_LOG_ERROR,

  365.                "Channel groups with differing sample rates not currently supported\n");

  366.         return -1;

  367.     }

  368. 

  369.     if (mh.group1_samplerate == 0) {

  370.         av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown sampling rate\n");

  371.         return -1;

  372.     }

  373.     if (mh.group1_samplerate > MAX_SAMPLERATE) {

  374.         av_log(m->avctx, AV_LOG_ERROR,

  375.                "Sampling rate %d is greater than maximum supported (%d)\n",

  376.                mh.group1_samplerate, MAX_SAMPLERATE);

  377.         return -1;

  378.     }

  379.     if (mh.access_unit_size > MAX_BLOCKSIZE) {

  380.         av_log(m->avctx, AV_LOG_ERROR,

  381.                "Block size %d is greater than maximum supported (%d)\n",

  382.                mh.access_unit_size, MAX_BLOCKSIZE);

  383.         return -1;

  384.     }

  385.     if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {

  386.         av_log(m->avctx, AV_LOG_ERROR,

  387.                "Block size pow2 %d is greater than maximum supported (%d)\n",

  388.                mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);

  389.         return -1;

  390.     }

  391. 

  392.     if (mh.num_substreams == 0)

  393.         return -1;

  394.     if (mh.num_substreams > MAX_SUBSTREAMS) {

  395.         av_log(m->avctx, AV_LOG_ERROR,

  396.                "Number of substreams %d is more than maximum supported by "

  397.                "decoder. %s\n", mh.num_substreams, sample_message);

  398.         return -1;

  399.     }

  400. 

  401.     m->access_unit_size      = mh.access_unit_size;

  402.     m->access_unit_size_pow2 = mh.access_unit_size_pow2;

  403. 

  404.     m->num_substreams        = mh.num_substreams;

  405.     m->max_decoded_substream = m->num_substreams - 1;

  406. 

  407.     m->avctx->sample_rate    = mh.group1_samplerate;

  408.     m->avctx->frame_size     = mh.access_unit_size;

  409. 

  410. #ifdef CONFIG_AUDIO_NONSHORT

  411.     m->avctx->bits_per_sample = mh.group1_bits;

  412.     if (mh.group1_bits > 16) {

  413.         m->avctx->sample_fmt = SAMPLE_FMT_S32;

  414.     }

  415. #endif

  416. 

  417.     m->params_valid = 1;

  418.     for (substr = 0; substr < MAX_SUBSTREAMS; substr++)

  419.         m->substream[substr].restart_seen = 0;

  420. 

  421.     return 0;

  422. }

  423. 

  424. /** Read a restart header from a block in a substream. This contains parameters

  425.  *  required to decode the audio that do not change very often. Generally

  426.  *  (always) present only in blocks following a major sync. */

  427. 

  428. static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,

  429.                                const uint8_t *buf, unsigned int substr)

  430. {

  431.     SubStream *s = &m->substream[substr];

  432.     unsigned int ch;

  433.     int sync_word, tmp;

  434.     uint8_t checksum;

  435.     uint8_t lossless_check;

  436.     int start_count = get_bits_count(gbp);

  437. 

  438.     sync_word = get_bits(gbp, 13);

  439. 

  440.     if (sync_word != 0x31ea >> 1) {

  441.         av_log(m->avctx, AV_LOG_ERROR,

  442.                "Restart header sync incorrect (got 0x%04x)\n", sync_word);

  443.         return -1;

  444.     }

  445.     s->noise_type = get_bits1(gbp);

  446. 

  447.     skip_bits(gbp, 16); /* Output timestamp */

  448. 

  449.     s->min_channel        = get_bits(gbp, 4);

  450.     s->max_channel        = get_bits(gbp, 4);

  451.     s->max_matrix_channel = get_bits(gbp, 4);

  452. 

  453.     if (s->min_channel > s->max_channel) {

  454.         av_log(m->avctx, AV_LOG_ERROR,

  455.                "Substream min channel cannot be greater than max channel.\n");

  456.         return -1;

  457.     }

  458. 

  459.     if (m->avctx->request_channels > 0

  460.         && s->max_channel + 1 >= m->avctx->request_channels

  461.         && substr < m->max_decoded_substream) {

  462.         av_log(m->avctx, AV_LOG_INFO,

  463.                "Extracting %d channel downmix from substream %d. "

  464.                "Further substreams will be skipped.\n",

  465.                s->max_channel + 1, substr);

  466.         m->max_decoded_substream = substr;

  467.     }

  468. 

  469.     s->noise_shift   = get_bits(gbp,  4);

  470.     s->noisegen_seed = get_bits(gbp, 23);

  471. 

  472.     skip_bits(gbp, 19);

  473. 

  474.     s->data_check_present = get_bits1(gbp);

  475.     lossless_check = get_bits(gbp, 8);

  476.     if (substr == m->max_decoded_substream

  477.         && s->lossless_check_data != 0xffffffff) {

  478.         tmp = s->lossless_check_data;

  479.         tmp ^= tmp >> 16;

  480.         tmp ^= tmp >> 8;

  481.         tmp &= 0xff;

  482.         if (tmp != lossless_check)

  483.             av_log(m->avctx, AV_LOG_WARNING,

  484.                    "Lossless check failed - expected %02x, calculated %02x\n",

  485.                    lossless_check, tmp);

  486.         else

  487.             dprintf(m->avctx, "Lossless check passed for substream %d (%x)\n",

  488.                     substr, tmp);

  489.     }

  490. 

  491.     skip_bits(gbp, 16);

  492. 

  493.     for (ch = 0; ch <= s->max_matrix_channel; ch++) {

  494.         int ch_assign = get_bits(gbp, 6);

  495.         dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,

  496.                 ch_assign);

  497.         if (ch_assign != ch) {

  498.             av_log(m->avctx, AV_LOG_ERROR,

  499.                    "Non 1:1 channel assignments are used in this stream. %s\n",

  500.                    sample_message);

  501.             return -1;

  502.         }

  503.     }

  504. 

  505.     checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);

  506. 

  507.     if (checksum != get_bits(gbp, 8))

  508.         av_log(m->avctx, AV_LOG_ERROR, "Restart header checksum error\n");

  509. 

  510.     /* Set default decoding parameters */

  511.     s->param_presence_flags   = 0xff;

  512.     s->num_primitive_matrices = 0;

  513.     s->blocksize              = 8;

  514.     s->lossless_check_data    = 0;

  515. 

  516.     memset(s->output_shift   , 0, sizeof(s->output_shift   ));

  517.     memset(s->quant_step_size, 0, sizeof(s->quant_step_size));

  518. 

  519.     for (ch = s->min_channel; ch <= s->max_channel; ch++) {

  520.         m->filter_order[ch][FIR] = 0;

  521.         m->filter_order[ch][IIR] = 0;

  522.         m->filter_shift[ch][FIR] = 0;

  523.         m->filter_shift[ch][IIR] = 0;

  524. 

  525.         /* Default audio coding is 24-bit raw PCM */

  526.         m->huff_offset     [ch] = 0;

  527.         m->sign_huff_offset[ch] = (-1) << 23;

  528.         m->codebook        [ch] = 0;

  529.         m->huff_lsbs       [ch] = 24;

  530.     }

  531. 

  532.     if (substr == m->max_decoded_substream) {

  533.         m->avctx->channels = s->max_channel + 1;

  534.     }

  535. 

  536.     return 0;

  537. }

  538. 

  539. /** Read parameters for one of the prediction filters. */

  540. 

  541. static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,

  542.                               unsigned int channel, unsigned int filter)

  543. {

  544.     const char fchar = filter ? 'I' : 'F';

  545.     int i, order;

  546. 

  547.     // filter is 0 for FIR, 1 for IIR

  548.     assert(filter < 2);

  549. 

  550.     order = get_bits(gbp, 4);

  551.     if (order > MAX_FILTER_ORDER) {

  552.         av_log(m->avctx, AV_LOG_ERROR,

  553.                "%cIR filter order %d is greater than maximum %d\n",

  554.                fchar, order, MAX_FILTER_ORDER);

  555.         return -1;

  556.     }

  557.     m->filter_order[channel][filter] = order;

  558. 

  559.     if (order > 0) {

  560.         int coeff_bits, coeff_shift;

  561. 

  562.         m->filter_shift[channel][filter] = get_bits(gbp, 4);

  563. 

  564.         coeff_bits  = get_bits(gbp, 5);

  565.         coeff_shift = get_bits(gbp, 3);

  566.         if (coeff_bits < 1 || coeff_bits > 16) {

  567.             av_log(m->avctx, AV_LOG_ERROR,

  568.                    "%cIR filter coeff_bits must be between 1 and 16\n",

  569.                    fchar);

  570.             return -1;

  571.         }

  572.         if (coeff_bits + coeff_shift > 16) {

  573.             av_log(m->avctx, AV_LOG_ERROR,

  574.                    "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less\n",

  575.                    fchar);

  576.             return -1;

  577.         }

  578. 

  579.         for (i = 0; i < order; i++)

  580.             m->filter_coeff[channel][filter][i] =

  581.                     get_sbits(gbp, coeff_bits) << coeff_shift;

  582. 

  583.         if (get_bits1(gbp)) {

  584.             int state_bits, state_shift;

  585. 

  586.             if (filter == FIR) {

  587.                 av_log(m->avctx, AV_LOG_ERROR,

  588.                        "FIR filter has state data specified\n");

  589.                 return -1;

  590.             }

  591. 

  592.             state_bits  = get_bits(gbp, 4);

  593.             state_shift = get_bits(gbp, 4);

  594. 

  595.             /* TODO: check validity of state data */

  596. 

  597.             for (i = 0; i < order; i++)

  598.                 m->filter_state[channel][filter][i] =

  599.                     get_sbits(gbp, state_bits) << state_shift;

  600.         }

  601.     }

  602. 

  603.     return 0;

  604. }

  605. 

  606. /** Read decoding parameters that change more often than those in the restart

  607.  *  header. */

  608. 

  609. static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,

  610.                                 unsigned int substr)

  611. {

  612.     SubStream *s = &m->substream[substr];

  613.     unsigned int mat, ch;

  614. 

  615.     if (get_bits1(gbp))

  616.         s->param_presence_flags = get_bits(gbp, 8);

  617. 

  618.     if (s->param_presence_flags & PARAM_BLOCKSIZE)

  619.         if (get_bits1(gbp)) {

  620.             s->blocksize = get_bits(gbp, 9);

  621.             if (s->blocksize > MAX_BLOCKSIZE) {

  622.                 av_log(m->avctx, AV_LOG_ERROR, "Block size too large\n");

  623.                 s->blocksize = 0;

  624.                 return -1;

  625.             }

  626.         }

  627. 

  628.     if (s->param_presence_flags & PARAM_MATRIX)

  629.         if (get_bits1(gbp)) {

  630.             s->num_primitive_matrices = get_bits(gbp, 4);

  631. 

  632.             for (mat = 0; mat < s->num_primitive_matrices; mat++) {

  633.                 int frac_bits, max_chan;

  634.                 s->matrix_out_ch[mat] = get_bits(gbp, 4);

  635.                 frac_bits             = get_bits(gbp, 4);

  636.                 s->lsb_bypass   [mat] = get_bits1(gbp);

  637. 

  638.                 if (s->matrix_out_ch[mat] > s->max_channel) {

  639.                     av_log(m->avctx, AV_LOG_ERROR,

  640.                            "Invalid channel %d specified as output from matrix\n",

  641.                            s->matrix_out_ch[mat]);

  642.                     return -1;

  643.                 }

  644.                 if (frac_bits > 14) {

  645.                     av_log(m->avctx, AV_LOG_ERROR,

  646.                            "Too many fractional bits specified\n");

  647.                     return -1;

  648.                 }

  649. 

  650.                 max_chan = s->max_matrix_channel;

  651.                 if (!s->noise_type)

  652.                     max_chan+=2;

  653. 

  654.                 for (ch = 0; ch <= max_chan; ch++) {

  655.                     int coeff_val = 0;

  656.                     if (get_bits1(gbp))

  657.                         coeff_val = get_sbits(gbp, frac_bits + 2);

  658. 

  659.                     s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);

  660.                 }

  661. 

  662.                 if (s->noise_type)

  663.                     s->matrix_noise_shift[mat] = get_bits(gbp, 4);

  664.                 else

  665.                     s->matrix_noise_shift[mat] = 0;

  666.             }

  667.         }

  668. 

  669.     if (s->param_presence_flags & PARAM_OUTSHIFT)

  670.         if (get_bits1(gbp))

  671.             for (ch = 0; ch <= s->max_matrix_channel; ch++) {

  672.                 s->output_shift[ch] = get_bits(gbp, 4);

  673.                 dprintf(m->avctx, "output shift[%d] = %d\n",

  674.                         ch, s->output_shift[ch]);

  675.                 /* TODO: validate */

  676.             }

  677. 

  678.     if (s->param_presence_flags & PARAM_QUANTSTEP)

  679.         if (get_bits1(gbp))

  680.             for (ch = 0; ch <= s->max_channel; ch++) {

  681.                 s->quant_step_size[ch] = get_bits(gbp, 4);

  682.                 /* TODO: validate */

  683. 

  684.                 m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);

  685.             }

  686. 

  687.     for (ch = s->min_channel; ch <= s->max_channel; ch++)

  688.         if (get_bits1(gbp)) {

  689.             if (s->param_presence_flags & PARAM_FIR)

  690.                 if (get_bits1(gbp))

  691.                     if (read_filter_params(m, gbp, ch, FIR) < 0)

  692.                         return -1;

  693. 

  694.             if (s->param_presence_flags & PARAM_IIR)

  695.                 if (get_bits1(gbp))

  696.                     if (read_filter_params(m, gbp, ch, IIR) < 0)

  697.                         return -1;

  698. 

  699.             if (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] &&

  700.                 m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) {

  701.                 av_log(m->avctx, AV_LOG_ERROR,

  702.                        "FIR and IIR filters must use same precision\n");

  703.                 return -1;

  704.             }

  705.             /* The FIR and IIR filters must have the same precision.

  706.              * To simplify the filtering code, only the precision of the

  707.              * FIR filter is considered. If only the IIR filter is employed,

  708.              * the FIR filter precision is set to that of the IIR filter, so

  709.              * that the filtering code can use it. */

  710.             if (!m->filter_order[ch][FIR] && m->filter_order[ch][IIR])

  711.                 m->filter_shift[ch][FIR] = m->filter_shift[ch][IIR];

  712. 

  713.             if (s->param_presence_flags & PARAM_HUFFOFFSET)

  714.                 if (get_bits1(gbp))

  715.                     m->huff_offset[ch] = get_sbits(gbp, 15);

  716. 

  717.             m->codebook [ch] = get_bits(gbp, 2);

  718.             m->huff_lsbs[ch] = get_bits(gbp, 5);

  719. 

  720.             m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);

  721. 

  722.             /* TODO: validate */

  723.         }

  724. 

  725.     return 0;

  726. }

  727. 

  728. #define MSB_MASK(bits)  (-1u << bits)

  729. 

  730. /** Generate PCM samples using the prediction filters and residual values

  731.  *  read from the data stream, and update the filter state. */

  732. 

  733. static void filter_channel(MLPDecodeContext *m, unsigned int substr,

  734.                            unsigned int channel)

  735. {

  736.     SubStream *s = &m->substream[substr];

  737.     int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];

  738.     unsigned int filter_shift = m->filter_shift[channel][FIR];

  739.     int32_t mask = MSB_MASK(s->quant_step_size[channel]);

  740.     int index = MAX_BLOCKSIZE;

  741.     int j, i;

  742. 

  743.     for (j = 0; j < NUM_FILTERS; j++) {

  744.         memcpy(&   filter_state_buffer  [j][MAX_BLOCKSIZE],

  745.                &m->filter_state[channel][j][0],

  746.                MAX_FILTER_ORDER * sizeof(int32_t));

  747.     }

  748. 

  749.     for (i = 0; i < s->blocksize; i++) {

  750.         int32_t residual = m->sample_buffer[i + s->blockpos][channel];

  751.         unsigned int order;

  752.         int64_t accum = 0;

  753.         int32_t result;

  754. 

  755.         /* TODO: Move this code to DSPContext? */

  756. 

  757.         for (j = 0; j < NUM_FILTERS; j++)

  758.             for (order = 0; order < m->filter_order[channel][j]; order++)

  759.                 accum += (int64_t)filter_state_buffer[j][index + order] *

  760.                         m->filter_coeff[channel][j][order];

  761. 

  762.         accum  = accum >> filter_shift;

  763.         result = (accum + residual) & mask;

  764. 

  765.         --index;

  766. 

  767.         filter_state_buffer[FIR][index] = result;

  768.         filter_state_buffer[IIR][index] = result - accum;

  769. 

  770.         m->sample_buffer[i + s->blockpos][channel] = result;

  771.     }

  772. 

  773.     for (j = 0; j < NUM_FILTERS; j++) {

  774.         memcpy(&m->filter_state[channel][j][0],

  775.                &   filter_state_buffer  [j][index],

  776.                MAX_FILTER_ORDER * sizeof(int32_t));

  777.     }

  778. }

  779. 

  780. /** Read a block of PCM residual data (or actual if no filtering active). */

  781. 

  782. static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,

  783.                            unsigned int substr)

  784. {

  785.     SubStream *s = &m->substream[substr];

  786.     unsigned int i, ch, expected_stream_pos = 0;

  787. 

  788.     if (s->data_check_present) {

  789.         expected_stream_pos  = get_bits_count(gbp);

  790.         expected_stream_pos += get_bits(gbp, 16);

  791.         av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "

  792.                "we have not tested yet. %s\n", sample_message);

  793.     }

  794. 

  795.     if (s->blockpos + s->blocksize > m->access_unit_size) {

  796.         av_log(m->avctx, AV_LOG_ERROR, "Too many audio samples in frame\n");

  797.         return -1;

  798.     }

  799. 

  800.     memset(&m->bypassed_lsbs[s->blockpos][0], 0,

  801.            s->blocksize * sizeof(m->bypassed_lsbs[0]));

  802. 

  803.     for (i = 0; i < s->blocksize; i++) {

  804.         if (read_huff_channels(m, gbp, substr, i) < 0)

  805.             return -1;

  806.     }

  807. 

  808.     for (ch = s->min_channel; ch <= s->max_channel; ch++) {

  809.         filter_channel(m, substr, ch);

  810.     }

  811. 

  812.     s->blockpos += s->blocksize;

  813. 

  814.     if (s->data_check_present) {

  815.         if (get_bits_count(gbp) != expected_stream_pos)

  816.             av_log(m->avctx, AV_LOG_ERROR, "Block data length mismatch\n");

  817.         skip_bits(gbp, 8);

  818.     }

  819. 

  820.     return 0;

  821. }

  822. 

  823. /** Data table used for TrueHD noise generation function */

  824. 

  825. static const int8_t noise_table[256] = {

  826.      30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,

  827.      52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,

  828.      10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,

  829.      51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,

  830.      38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,

  831.      61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,

  832.      67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,

  833.      48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,

  834.       0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,

  835.      16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,

  836.      13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,

  837.      89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,

  838.      36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,

  839.      39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,

  840.      45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,

  841.     -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,

  842. };

  843. 

  844. /** Noise generation functions.

  845.  *  I'm not sure what these are for - they seem to be some kind of pseudorandom

  846.  *  sequence generators, used to generate noise data which is used when the

  847.  *  channels are rematrixed. I'm not sure if they provide a practical benefit

  848.  *  to compression, or just obfuscate the decoder. Are they for some kind of

  849.  *  dithering? */

  850. 

  851. /** Generate two channels of noise, used in the matrix when

  852.  *  restart sync word == 0x31ea. */

  853. 

  854. static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)

  855. {

  856.     SubStream *s = &m->substream[substr];

  857.     unsigned int i;

  858.     uint32_t seed = s->noisegen_seed;

  859.     unsigned int maxchan = s->max_matrix_channel;

  860. 

  861.     for (i = 0; i < s->blockpos; i++) {

  862.         uint16_t seed_shr7 = seed >> 7;

  863.         m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;

  864.         m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7)   << s->noise_shift;

  865. 

  866.         seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);

  867.     }

  868. 

  869.     s->noisegen_seed = seed;

  870. }

  871. 

  872. /** Generate a block of noise, used when restart sync word == 0x31eb. */

  873. 

  874. static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)

  875. {

  876.     SubStream *s = &m->substream[substr];

  877.     unsigned int i;

  878.     uint32_t seed = s->noisegen_seed;

  879. 

  880.     for (i = 0; i < m->access_unit_size_pow2; i++) {

  881.         uint8_t seed_shr15 = seed >> 15;

  882.         m->noise_buffer[i] = noise_table[seed_shr15];

  883.         seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);

  884.     }

  885. 

  886.     s->noisegen_seed = seed;

  887. }

  888. 

  889. 

  890. /** Apply the channel matrices in turn to reconstruct the original audio

  891.  *  samples. */

  892. 

  893. static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)

  894. {

  895.     SubStream *s = &m->substream[substr];

  896.     unsigned int mat, src_ch, i;

  897.     unsigned int maxchan;

  898. 

  899.     maxchan = s->max_matrix_channel;

  900.     if (!s->noise_type) {

  901.         generate_2_noise_channels(m, substr);

  902.         maxchan += 2;

  903.     } else {

  904.         fill_noise_buffer(m, substr);

  905.     }

  906. 

  907.     for (mat = 0; mat < s->num_primitive_matrices; mat++) {

  908.         int matrix_noise_shift = s->matrix_noise_shift[mat];

  909.         unsigned int dest_ch = s->matrix_out_ch[mat];

  910.         int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);

  911. 

  912.         /* TODO: DSPContext? */

  913. 

  914.         for (i = 0; i < s->blockpos; i++) {

  915.             int64_t accum = 0;

  916.             for (src_ch = 0; src_ch <= maxchan; src_ch++) {

  917.                 accum += (int64_t)m->sample_buffer[i][src_ch]

  918.                                   * s->matrix_coeff[mat][src_ch];

  919.             }

  920.             if (matrix_noise_shift) {

  921.                 uint32_t index = s->num_primitive_matrices - mat;

  922.                 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);

  923.                 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);

  924.             }

  925.             m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)

  926.                                              + m->bypassed_lsbs[i][mat];

  927.         }

  928.     }

  929. }

  930. 

  931. /** Write the audio data into the output buffer. */

  932. 

  933. static int output_data_internal(MLPDecodeContext *m, unsigned int substr,

  934.                                 uint8_t *data, unsigned int *data_size, int is32)

  935. {

  936.     SubStream *s = &m->substream[substr];

  937.     unsigned int i, ch = 0;

  938.     int32_t *data_32 = (int32_t*) data;

  939.     int16_t *data_16 = (int16_t*) data;

  940. 

  941.     if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))

  942.         return -1;

  943. 

  944.     for (i = 0; i < s->blockpos; i++) {

  945.         for (ch = 0; ch <= s->max_channel; ch++) {

  946.             int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch];

  947.             s->lossless_check_data ^= (sample & 0xffffff) << ch;

  948.             if (is32) *data_32++ = sample << 8;

  949.             else      *data_16++ = sample >> 8;

  950.         }

  951.     }

  952. 

  953.     *data_size = i * ch * (is32 ? 4 : 2);

  954. 

  955.     return 0;

  956. }

  957. 

  958. static int output_data(MLPDecodeContext *m, unsigned int substr,

  959.                        uint8_t *data, unsigned int *data_size)

  960. {

  961.     if (m->avctx->sample_fmt == SAMPLE_FMT_S32)

  962.         return output_data_internal(m, substr, data, data_size, 1);

  963.     else

  964.         return output_data_internal(m, substr, data, data_size, 0);

  965. }

  966. 

  967. 

  968. /** XOR together all the bytes of a buffer.

  969.  *  Does this belong in dspcontext? */

  970. 

  971. static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size)

  972. {

  973.     uint32_t scratch = 0;

  974.     const uint8_t *buf_end = buf + buf_size;

  975. 

  976.     for (; buf < buf_end - 3; buf += 4)

  977.         scratch ^= *((const uint32_t*)buf);

  978. 

  979.     scratch ^= scratch >> 16;

  980.     scratch ^= scratch >> 8;

  981. 

  982.     for (; buf < buf_end; buf++)

  983.         scratch ^= *buf;

  984. 

  985.     return scratch;

  986. }

  987. 

  988. /** Read an access unit from the stream.

  989.  *  Returns < 0 on error, 0 if not enough data is present in the input stream

  990.  *  otherwise returns the number of bytes consumed. */

  991. 

  992. static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,

  993.                             const uint8_t *buf, int buf_size)

  994. {

  995.     MLPDecodeContext *m = avctx->priv_data;

  996.     GetBitContext gb;

  997.     unsigned int length, substr;

  998.     unsigned int substream_start;

  999.     unsigned int header_size = 4;

  1000.     unsigned int substr_header_size = 0;

  1001.     uint8_t substream_parity_present[MAX_SUBSTREAMS];

  1002.     uint16_t substream_data_len[MAX_SUBSTREAMS];

  1003.     uint8_t parity_bits;

  1004. 

  1005.     if (buf_size < 4)

  1006.         return 0;

  1007. 

  1008.     length = (AV_RB16(buf) & 0xfff) * 2;

  1009. 

  1010.     if (length > buf_size)

  1011.         return -1;

  1012. 

  1013.     init_get_bits(&gb, (buf + 4), (length - 4) * 8);

  1014. 

  1015.     if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {

  1016.         dprintf(m->avctx, "Found major sync\n");

  1017.         if (read_major_sync(m, &gb) < 0)

  1018.             goto error;

  1019.         header_size += 28;

  1020.     }

  1021. 

  1022.     if (!m->params_valid) {

  1023.         av_log(m->avctx, AV_LOG_WARNING,

  1024.                "Stream parameters not seen; skipping frame\n");

  1025.         *data_size = 0;

  1026.         return length;

  1027.     }

  1028. 

  1029.     substream_start = 0;

  1030. 

  1031.     for (substr = 0; substr < m->num_substreams; substr++) {

  1032.         int extraword_present, checkdata_present, end;

  1033. 

  1034.         extraword_present = get_bits1(&gb);

  1035.         skip_bits1(&gb);

  1036.         checkdata_present = get_bits1(&gb);

  1037.         skip_bits1(&gb);

  1038. 

  1039.         end = get_bits(&gb, 12) * 2;

  1040. 

  1041.         substr_header_size += 2;

  1042. 

  1043.         if (extraword_present) {

  1044.             skip_bits(&gb, 16);

  1045.             substr_header_size += 2;

  1046.         }

  1047. 

  1048.         if (end + header_size + substr_header_size > length) {

  1049.             av_log(m->avctx, AV_LOG_ERROR,

  1050.                    "Indicated length of substream %d data goes off end of "

  1051.                    "packet.\n", substr);

  1052.             end = length - header_size - substr_header_size;

  1053.         }

  1054. 

  1055.         if (end < substream_start) {

  1056.             av_log(avctx, AV_LOG_ERROR,

  1057.                    "Indicated end offset of substream %d data "

  1058.                    "is smaller than calculated start offset.\n",

  1059.                    substr);

  1060.             goto error;

  1061.         }

  1062. 

  1063.         if (substr > m->max_decoded_substream)

  1064.             continue;

  1065. 

  1066.         substream_parity_present[substr] = checkdata_present;

  1067.         substream_data_len[substr] = end - substream_start;

  1068.         substream_start = end;

  1069.     }

  1070. 

  1071.     parity_bits  = calculate_parity(buf, 4);

  1072.     parity_bits ^= calculate_parity(buf + header_size, substr_header_size);

  1073. 

  1074.     if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {

  1075.         av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");

  1076.         goto error;

  1077.     }

  1078. 

  1079.     buf += header_size + substr_header_size;

  1080. 

  1081.     for (substr = 0; substr <= m->max_decoded_substream; substr++) {

  1082.         SubStream *s = &m->substream[substr];

  1083.         init_get_bits(&gb, buf, substream_data_len[substr] * 8);

  1084. 

  1085.         s->blockpos = 0;

  1086.         do {

  1087.             if (get_bits1(&gb)) {

  1088.                 if (get_bits1(&gb)) {

  1089.                     /* A restart header should be present */

  1090.                     if (read_restart_header(m, &gb, buf, substr) < 0)

  1091.                         goto next_substr;

  1092.                     s->restart_seen = 1;

  1093.                 }

  1094. 

  1095.                 if (!s->restart_seen) {

  1096.                     av_log(m->avctx, AV_LOG_ERROR,

  1097.                            "No restart header present in substream %d.\n",

  1098.                            substr);

  1099.                     goto next_substr;

  1100.                 }

  1101. 

  1102.                 if (read_decoding_params(m, &gb, substr) < 0)

  1103.                     goto next_substr;

  1104.             }

  1105. 

  1106.             if (!s->restart_seen) {

  1107.                 av_log(m->avctx, AV_LOG_ERROR,

  1108.                        "No restart header present in substream %d.\n",

  1109.                        substr);

  1110.                 goto next_substr;

  1111.             }

  1112. 

  1113.             if (read_block_data(m, &gb, substr) < 0)

  1114.                 return -1;

  1115. 

  1116.         } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)

  1117.                  && get_bits1(&gb) == 0);

  1118. 

  1119.         skip_bits(&gb, (-get_bits_count(&gb)) & 15);

  1120.         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 48 &&

  1121.             (show_bits_long(&gb, 32) == 0xd234d234 ||

  1122.              show_bits_long(&gb, 20) == 0xd234e)) {

  1123.             skip_bits(&gb, 18);

  1124.             if (substr == m->max_decoded_substream)

  1125.                 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated\n");

  1126. 

  1127.             if (get_bits1(&gb)) {

  1128.                 int shorten_by = get_bits(&gb, 13);

  1129.                 shorten_by = FFMIN(shorten_by, s->blockpos);

  1130.                 s->blockpos -= shorten_by;

  1131.             } else

  1132.                 skip_bits(&gb, 13);

  1133.         }

  1134.         if (substream_parity_present[substr]) {

  1135.             uint8_t parity, checksum;

  1136. 

  1137.             parity = calculate_parity(buf, substream_data_len[substr] - 2);

  1138.             if ((parity ^ get_bits(&gb, 8)) != 0xa9)

  1139.                 av_log(m->avctx, AV_LOG_ERROR,

  1140.                        "Substream %d parity check failed\n", substr);

  1141. 

  1142.             checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);

  1143.             if (checksum != get_bits(&gb, 8))

  1144.                 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed\n",

  1145.                        substr);

  1146.         }

  1147.         if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {

  1148.             av_log(m->avctx, AV_LOG_ERROR, "Substream %d length mismatch.\n",

  1149.                    substr);

  1150.             return -1;

  1151.         }

  1152. 

  1153. next_substr:

  1154.         buf += substream_data_len[substr];

  1155.     }

  1156. 

  1157.     rematrix_channels(m, m->max_decoded_substream);

  1158. 

  1159.     if (output_data(m, m->max_decoded_substream, data, data_size) < 0)

  1160.         return -1;

  1161. 

  1162.     return length;

  1163. 

  1164. error:

  1165.     m->params_valid = 0;

  1166.     return -1;

  1167. }

  1168. 

  1169. AVCodec mlp_decoder = {

  1170.     "mlp",

  1171.     CODEC_TYPE_AUDIO,

  1172.     CODEC_ID_MLP,

  1173.     sizeof(MLPDecodeContext),

  1174.     mlp_decode_init,

  1175.     NULL,

  1176.     NULL,

  1177.     read_access_unit,

  1178.     .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),

  1179. };