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FFmpeg/libavcodec/mlpdec.c

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  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 <stdint.h>

  28. 

  29. #include "avcodec.h"

  30. #include "libavutil/intreadwrite.h"

  31. #include "bitstream.h"

  32. #include "libavutil/crc.h"

  33. #include "parser.h"

  34. #include "mlp_parser.h"

  35. 

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

  37. #define MAX_CHANNELS        16

  38. 

  39. /** Maximum number of matrices used in decoding; most streams have one matrix

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

  41.  */

  42. 

  43. #define MAX_MATRICES        15

  44. 

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

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

  47. #define MAX_SUBSTREAMS      2

  48. 

  49. /** maximum sample frequency seen in files */

  50. #define MAX_SAMPLERATE      192000

  51. 

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

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

  54. /** next power of two greater than MAX_BLOCKSIZE */

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

  56. 

  57. /** number of allowed filters */

  58. #define NUM_FILTERS         2

  59. 

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

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

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

  63. #define MAX_FILTER_ORDER    8

  64. 

  65. /** number of bits used for VLC lookup - longest Huffman code is 9 */

  66. #define VLC_BITS            9

  67. 

  68. 

  69. static const char* sample_message =

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

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

  72.     "a sample of this file.";

  73. 

  74. typedef struct SubStream {

  75.     //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.

  76.     uint8_t     restart_seen;

  77. 

  78.     //@{

  79.     /** restart header data */

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

  81.     uint16_t    noise_type;

  82. 

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

  84.     uint8_t     min_channel;

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

  86.     uint8_t     max_channel;

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

  88.     uint8_t     max_matrix_channel;

  89. 

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

  91.     uint8_t     noise_shift;

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

  93.     uint32_t    noisegen_seed;

  94. 

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

  96.     uint8_t     data_check_present;

  97. 

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

  99.     uint8_t     param_presence_flags;

  100. #define PARAM_BLOCKSIZE     (1 << 7)

  101. #define PARAM_MATRIX        (1 << 6)

  102. #define PARAM_OUTSHIFT      (1 << 5)

  103. #define PARAM_QUANTSTEP     (1 << 4)

  104. #define PARAM_FIR           (1 << 3)

  105. #define PARAM_IIR           (1 << 2)

  106. #define PARAM_HUFFOFFSET    (1 << 1)

  107.     //@}

  108. 

  109.     //@{

  110.     /** matrix data */

  111. 

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

  113.     uint8_t     num_primitive_matrices;

  114. 

  115.     //! matrix output channel

  116.     uint8_t     matrix_out_ch[MAX_MATRICES];

  117. 

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

  119.     uint8_t     lsb_bypass[MAX_MATRICES];

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

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

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

  123.     uint8_t     matrix_noise_shift[MAX_MATRICES];

  124.     //@}

  125. 

  126.     //! Left shift to apply to Huffman-decoded residuals.

  127.     uint8_t     quant_step_size[MAX_CHANNELS];

  128. 

  129.     //! number of PCM samples in current audio block

  130.     uint16_t    blocksize;

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

  132.     uint16_t    blockpos;

  133. 

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

  135.     int8_t      output_shift[MAX_CHANNELS];

  136. 

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

  138.     int32_t     lossless_check_data;

  139. 

  140. } SubStream;

  141. 

  142. typedef struct MLPDecodeContext {

  143.     AVCodecContext *avctx;

  144. 

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

  146.     uint8_t     params_valid;

  147. 

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

  149.     uint8_t     num_substreams;

  150. 

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

  152.     uint8_t     max_decoded_substream;

  153. 

  154.     //! number of PCM samples contained in each frame

  155.     int         access_unit_size;

  156.     //! next power of two above the number of samples in each frame

  157.     int         access_unit_size_pow2;

  158. 

  159.     SubStream   substream[MAX_SUBSTREAMS];

  160. 

  161.     //@{

  162.     /** filter data */

  163. #define FIR 0

  164. #define IIR 1

  165.     //! number of taps in filter

  166.     uint8_t     filter_order[MAX_CHANNELS][NUM_FILTERS];

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

  168.     uint8_t     filter_shift[MAX_CHANNELS][NUM_FILTERS];

  169. 

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

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

  172.     //@}

  173. 

  174.     //@{

  175.     /** sample data coding information */

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

  177.     int16_t     huff_offset[MAX_CHANNELS];

  178.     //! sign/rounding-corrected version of huff_offset

  179.     int32_t     sign_huff_offset[MAX_CHANNELS];

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

  181.     uint8_t     codebook[MAX_CHANNELS];

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

  183.     uint8_t     huff_lsbs[MAX_CHANNELS];

  184.     //@}

  185. 

  186.     int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];

  187.     int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];

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

  189. } MLPDecodeContext;

  190. 

  191. /** Tables defining the Huffman codes.

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

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

  194.  *  00 or 01, but have different codes starting with 1. */

  195. 

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

  197.     {    /* Huffman table 0, -7 - +10 */

  198.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  199.         {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},

  200.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

  201.     }, { /* Huffman table 1, -7 - +8 */

  202.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  203.         {0x02, 2}, {0x03, 2},

  204.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

  205.     }, { /* Huffman table 2, -7 - +7 */

  206.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  207.         {0x01, 1},

  208.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

  209.     }

  210. };

  211. 

  212. static VLC huff_vlc[3];

  213. 

  214. static int crc_init = 0;

  215. static AVCRC crc_63[1024];

  216. static AVCRC crc_1D[1024];

  217. 

  218. 

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

  220. 

  221. static av_cold void init_static()

  222. {

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

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

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

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

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

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

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

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

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

  232. 

  233.     if (!crc_init) {

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

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

  236.         crc_init = 1;

  237.     }

  238. }

  239. 

  240. 

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

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

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

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

  245. 

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

  247. {

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

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

  250.     return checksum;

  251. }

  252. 

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

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

  255. 

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

  257. {

  258.     int i;

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

  260. 

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

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

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

  264. 

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

  266.         crc <<= 1;

  267.         if (crc & 0x100)

  268.             crc ^= 0x11D;

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

  270.     }

  271. 

  272.     return crc;

  273. }

  274. 

  275. static inline int32_t calculate_sign_huff(MLPDecodeContext *m,

  276.                                           unsigned int substr, unsigned int ch)

  277. {

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

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

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

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

  282. 

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

  284.         sign_huff_offset -= 7 << lsb_bits;

  285. 

  286.     if (sign_shift >= 0)

  287.         sign_huff_offset -= 1 << sign_shift;

  288. 

  289.     return sign_huff_offset;

  290. }

  291. 

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

  293.  *  and plain LSBs. */

  294. 

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

  296.                                      unsigned int substr, unsigned int pos)

  297. {

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

  299.     unsigned int mat, channel;

  300. 

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

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

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

  304. 

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

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

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

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

  309.         int result = 0;

  310. 

  311.         if (codebook > 0)

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

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

  314. 

  315.         if (result < 0)

  316.             return -1;

  317. 

  318.         if (lsb_bits > 0)

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

  320. 

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

  322.         result <<= quant_step_size;

  323. 

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

  325.     }

  326. 

  327.     return 0;

  328. }

  329. 

  330. static av_cold int mlp_decode_init(AVCodecContext *avctx)

  331. {

  332.     MLPDecodeContext *m = avctx->priv_data;

  333.     int substr;

  334. 

  335.     init_static();

  336.     m->avctx = avctx;

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

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

  339.     avctx->sample_fmt = SAMPLE_FMT_S16;

  340.     return 0;

  341. }

  342. 

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

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

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

  346.  */

  347. 

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

  349. {

  350.     MLPHeaderInfo mh;

  351.     int substr;

  352. 

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

  354.         return -1;

  355. 

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

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

  358.         return -1;

  359.     }

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

  361.         av_log(m->avctx, AV_LOG_ERROR,

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

  363.         return -1;

  364.     }

  365. 

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

  367.         av_log(m->avctx, AV_LOG_ERROR,

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

  369.         return -1;

  370.     }

  371. 

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

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

  374.         return -1;

  375.     }

  376.     if (mh.group1_samplerate > MAX_SAMPLERATE) {

  377.         av_log(m->avctx, AV_LOG_ERROR,

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

  379.                mh.group1_samplerate, MAX_SAMPLERATE);

  380.         return -1;

  381.     }

  382.     if (mh.access_unit_size > MAX_BLOCKSIZE) {

  383.         av_log(m->avctx, AV_LOG_ERROR,

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

  385.                mh.access_unit_size, MAX_BLOCKSIZE);

  386.         return -1;

  387.     }

  388.     if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {

  389.         av_log(m->avctx, AV_LOG_ERROR,

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

  391.                mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);

  392.         return -1;

  393.     }

  394. 

  395.     if (mh.num_substreams == 0)

  396.         return -1;

  397.     if (mh.num_substreams > MAX_SUBSTREAMS) {

  398.         av_log(m->avctx, AV_LOG_ERROR,

  399.                "Number of substreams %d is larger than the maximum supported "

  400.                "by the decoder. %s\n", mh.num_substreams, sample_message);

  401.         return -1;

  402.     }

  403. 

  404.     m->access_unit_size      = mh.access_unit_size;

  405.     m->access_unit_size_pow2 = mh.access_unit_size_pow2;

  406. 

  407.     m->num_substreams        = mh.num_substreams;

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

  409. 

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

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

  412. 

  413. #ifdef CONFIG_AUDIO_NONSHORT

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

  415.     if (mh.group1_bits > 16) {

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

  417.     }

  418. #endif

  419. 

  420.     m->params_valid = 1;

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

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

  423. 

  424.     return 0;

  425. }

  426. 

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

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

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

  430. 

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

  432.                                const uint8_t *buf, unsigned int substr)

  433. {

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

  435.     unsigned int ch;

  436.     int sync_word, tmp;

  437.     uint8_t checksum;

  438.     uint8_t lossless_check;

  439.     int start_count = get_bits_count(gbp);

  440. 

  441.     sync_word = get_bits(gbp, 13);

  442. 

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

  444.         av_log(m->avctx, AV_LOG_ERROR,

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

  446.         return -1;

  447.     }

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

  449. 

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

  451. 

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

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

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

  455. 

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

  457.         av_log(m->avctx, AV_LOG_ERROR,

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

  459.         return -1;

  460.     }

  461. 

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

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

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

  465.         av_log(m->avctx, AV_LOG_INFO,

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

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

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

  469.         m->max_decoded_substream = substr;

  470.     }

  471. 

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

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

  474. 

  475.     skip_bits(gbp, 19);

  476. 

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

  478.     lossless_check = get_bits(gbp, 8);

  479.     if (substr == m->max_decoded_substream

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

  481.         tmp = s->lossless_check_data;

  482.         tmp ^= tmp >> 16;

  483.         tmp ^= tmp >> 8;

  484.         tmp &= 0xff;

  485.         if (tmp != lossless_check)

  486.             av_log(m->avctx, AV_LOG_WARNING,

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

  488.                    lossless_check, tmp);

  489.         else

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

  491.                     substr, tmp);

  492.     }

  493. 

  494.     skip_bits(gbp, 16);

  495. 

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

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

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

  499.                 ch_assign);

  500.         if (ch_assign != ch) {

  501.             av_log(m->avctx, AV_LOG_ERROR,

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

  503.                    sample_message);

  504.             return -1;

  505.         }

  506.     }

  507. 

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

  509. 

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

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

  512. 

  513.     /* Set default decoding parameters. */

  514.     s->param_presence_flags   = 0xff;

  515.     s->num_primitive_matrices = 0;

  516.     s->blocksize              = 8;

  517.     s->lossless_check_data    = 0;

  518. 

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

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

  521. 

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

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

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

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

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

  527. 

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

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

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

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

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

  533.     }

  534. 

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

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

  537.     }

  538. 

  539.     return 0;

  540. }

  541. 

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

  543. 

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

  545.                               unsigned int channel, unsigned int filter)

  546. {

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

  548.     int i, order;

  549. 

  550.     // Filter is 0 for FIR, 1 for IIR.

  551.     assert(filter < 2);

  552. 

  553.     order = get_bits(gbp, 4);

  554.     if (order > MAX_FILTER_ORDER) {

  555.         av_log(m->avctx, AV_LOG_ERROR,

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

  557.                fchar, order, MAX_FILTER_ORDER);

  558.         return -1;

  559.     }

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

  561. 

  562.     if (order > 0) {

  563.         int coeff_bits, coeff_shift;

  564. 

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

  566. 

  567.         coeff_bits  = get_bits(gbp, 5);

  568.         coeff_shift = get_bits(gbp, 3);

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

  570.             av_log(m->avctx, AV_LOG_ERROR,

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

  572.                    fchar);

  573.             return -1;

  574.         }

  575.         if (coeff_bits + coeff_shift > 16) {

  576.             av_log(m->avctx, AV_LOG_ERROR,

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

  578.                    fchar);

  579.             return -1;

  580.         }

  581. 

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

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

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

  585. 

  586.         if (get_bits1(gbp)) {

  587.             int state_bits, state_shift;

  588. 

  589.             if (filter == FIR) {

  590.                 av_log(m->avctx, AV_LOG_ERROR,

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

  592.                 return -1;

  593.             }

  594. 

  595.             state_bits  = get_bits(gbp, 4);

  596.             state_shift = get_bits(gbp, 4);

  597. 

  598.             /* TODO: Check validity of state data. */

  599. 

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

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

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

  603.         }

  604.     }

  605. 

  606.     return 0;

  607. }

  608. 

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

  610.  *  header. */

  611. 

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

  613.                                 unsigned int substr)

  614. {

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

  616.     unsigned int mat, ch;

  617. 

  618.     if (get_bits1(gbp))

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

  620. 

  621.     if (s->param_presence_flags & PARAM_BLOCKSIZE)

  622.         if (get_bits1(gbp)) {

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

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

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

  626.                 s->blocksize = 0;

  627.                 return -1;

  628.             }

  629.         }

  630. 

  631.     if (s->param_presence_flags & PARAM_MATRIX)

  632.         if (get_bits1(gbp)) {

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

  634. 

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

  636.                 int frac_bits, max_chan;

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

  638.                 frac_bits             = get_bits(gbp, 4);

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

  640. 

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

  642.                     av_log(m->avctx, AV_LOG_ERROR,

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

  644.                            s->matrix_out_ch[mat]);

  645.                     return -1;

  646.                 }

  647.                 if (frac_bits > 14) {

  648.                     av_log(m->avctx, AV_LOG_ERROR,

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

  650.                     return -1;

  651.                 }

  652. 

  653.                 max_chan = s->max_matrix_channel;

  654.                 if (!s->noise_type)

  655.                     max_chan+=2;

  656. 

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

  658.                     int coeff_val = 0;

  659.                     if (get_bits1(gbp))

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

  661. 

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

  663.                 }

  664. 

  665.                 if (s->noise_type)

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

  667.                 else

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

  669.             }

  670.         }

  671. 

  672.     if (s->param_presence_flags & PARAM_OUTSHIFT)

  673.         if (get_bits1(gbp))

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

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

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

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

  678.                 /* TODO: validate */

  679.             }

  680. 

  681.     if (s->param_presence_flags & PARAM_QUANTSTEP)

  682.         if (get_bits1(gbp))

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

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

  685.                 /* TODO: validate */

  686. 

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

  688.             }

  689. 

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

  691.         if (get_bits1(gbp)) {

  692.             if (s->param_presence_flags & PARAM_FIR)

  693.                 if (get_bits1(gbp))

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

  695.                         return -1;

  696. 

  697.             if (s->param_presence_flags & PARAM_IIR)

  698.                 if (get_bits1(gbp))

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

  700.                         return -1;

  701. 

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

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

  704.                 av_log(m->avctx, AV_LOG_ERROR,

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

  706.                 return -1;

  707.             }

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

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

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

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

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

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

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

  715. 

  716.             if (s->param_presence_flags & PARAM_HUFFOFFSET)

  717.                 if (get_bits1(gbp))

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

  719. 

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

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

  722. 

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

  724. 

  725.             /* TODO: validate */

  726.         }

  727. 

  728.     return 0;

  729. }

  730. 

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

  732. 

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

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

  735. 

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

  737.                            unsigned int channel)

  738. {

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

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

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

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

  743.     int index = MAX_BLOCKSIZE;

  744.     int j, i;

  745. 

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

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

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

  749.                MAX_FILTER_ORDER * sizeof(int32_t));

  750.     }

  751. 

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

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

  754.         unsigned int order;

  755.         int64_t accum = 0;

  756.         int32_t result;

  757. 

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

  759. 

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

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

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

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

  764. 

  765.         accum  = accum >> filter_shift;

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

  767. 

  768.         --index;

  769. 

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

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

  772. 

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

  774.     }

  775. 

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

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

  778.                &   filter_state_buffer  [j][index],

  779.                MAX_FILTER_ORDER * sizeof(int32_t));

  780.     }

  781. }

  782. 

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

  784. 

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

  786.                            unsigned int substr)

  787. {

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

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

  790. 

  791.     if (s->data_check_present) {

  792.         expected_stream_pos  = get_bits_count(gbp);

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

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

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

  796.     }

  797. 

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

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

  800.         return -1;

  801.     }

  802. 

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

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

  805. 

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

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

  808.             return -1;

  809.     }

  810. 

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

  812.         filter_channel(m, substr, ch);

  813.     }

  814. 

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

  816. 

  817.     if (s->data_check_present) {

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

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

  820.         skip_bits(gbp, 8);

  821.     }

  822. 

  823.     return 0;

  824. }

  825. 

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

  827. 

  828. static const int8_t noise_table[256] = {

  829.      30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,

  830.      52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,

  831.      10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,

  832.      51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,

  833.      38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,

  834.      61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,

  835.      67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,

  836.      48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,

  837.       0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,

  838.      16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,

  839.      13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,

  840.      89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,

  841.      36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,

  842.      39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,

  843.      45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,

  844.     -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,

  845. };

  846. 

  847. /** Noise generation functions.

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

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

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

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

  852.  *  dithering? */

  853. 

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

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

  856. 

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

  858. {

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

  860.     unsigned int i;

  861.     uint32_t seed = s->noisegen_seed;

  862.     unsigned int maxchan = s->max_matrix_channel;

  863. 

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

  865.         uint16_t seed_shr7 = seed >> 7;

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

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

  868. 

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

  870.     }

  871. 

  872.     s->noisegen_seed = seed;

  873. }

  874. 

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

  876. 

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

  878. {

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

  880.     unsigned int i;

  881.     uint32_t seed = s->noisegen_seed;

  882. 

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

  884.         uint8_t seed_shr15 = seed >> 15;

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

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

  887.     }

  888. 

  889.     s->noisegen_seed = seed;

  890. }

  891. 

  892. 

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

  894.  *  samples. */

  895. 

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

  897. {

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

  899.     unsigned int mat, src_ch, i;

  900.     unsigned int maxchan;

  901. 

  902.     maxchan = s->max_matrix_channel;

  903.     if (!s->noise_type) {

  904.         generate_2_noise_channels(m, substr);

  905.         maxchan += 2;

  906.     } else {

  907.         fill_noise_buffer(m, substr);

  908.     }

  909. 

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

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

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

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

  914. 

  915.         /* TODO: DSPContext? */

  916. 

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

  918.             int64_t accum = 0;

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

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

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

  922.             }

  923.             if (matrix_noise_shift) {

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

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

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

  927.             }

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

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

  930.         }

  931.     }

  932. }

  933. 

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

  935. 

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

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

  938. {

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

  940.     unsigned int i, ch = 0;

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

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

  943. 

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

  945.         return -1;

  946. 

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

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

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

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

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

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

  953.         }

  954.     }

  955. 

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

  957. 

  958.     return 0;

  959. }

  960. 

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

  962.                        uint8_t *data, unsigned int *data_size)

  963. {

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

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

  966.     else

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

  968. }

  969. 

  970. 

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

  972.  *  Does this belong in dspcontext? */

  973. 

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

  975. {

  976.     uint32_t scratch = 0;

  977.     const uint8_t *buf_end = buf + buf_size;

  978. 

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

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

  981. 

  982.     scratch ^= scratch >> 16;

  983.     scratch ^= scratch >> 8;

  984. 

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

  986.         scratch ^= *buf;

  987. 

  988.     return scratch;

  989. }

  990. 

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

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

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

  994. 

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

  996.                             const uint8_t *buf, int buf_size)

  997. {

  998.     MLPDecodeContext *m = avctx->priv_data;

  999.     GetBitContext gb;

  1000.     unsigned int length, substr;

  1001.     unsigned int substream_start;

  1002.     unsigned int header_size = 4;

  1003.     unsigned int substr_header_size = 0;

  1004.     uint8_t substream_parity_present[MAX_SUBSTREAMS];

  1005.     uint16_t substream_data_len[MAX_SUBSTREAMS];

  1006.     uint8_t parity_bits;

  1007. 

  1008.     if (buf_size < 4)

  1009.         return 0;

  1010. 

  1011.     length = (AV_RB16(buf) & 0xfff) * 2;

  1012. 

  1013.     if (length > buf_size)

  1014.         return -1;

  1015. 

  1016.     init_get_bits(&gb, (buf + 4), (length - 4) * 8);

  1017. 

  1018.     if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {

  1019.         dprintf(m->avctx, "Found major sync.\n");

  1020.         if (read_major_sync(m, &gb) < 0)

  1021.             goto error;

  1022.         header_size += 28;

  1023.     }

  1024. 

  1025.     if (!m->params_valid) {

  1026.         av_log(m->avctx, AV_LOG_WARNING,

  1027.                "Stream parameters not seen; skipping frame.\n");

  1028.         *data_size = 0;

  1029.         return length;

  1030.     }

  1031. 

  1032.     substream_start = 0;

  1033. 

  1034.     for (substr = 0; substr < m->num_substreams; substr++) {

  1035.         int extraword_present, checkdata_present, end;

  1036. 

  1037.         extraword_present = get_bits1(&gb);

  1038.         skip_bits1(&gb);

  1039.         checkdata_present = get_bits1(&gb);

  1040.         skip_bits1(&gb);

  1041. 

  1042.         end = get_bits(&gb, 12) * 2;

  1043. 

  1044.         substr_header_size += 2;

  1045. 

  1046.         if (extraword_present) {

  1047.             skip_bits(&gb, 16);

  1048.             substr_header_size += 2;

  1049.         }

  1050. 

  1051.         if (end + header_size + substr_header_size > length) {

  1052.             av_log(m->avctx, AV_LOG_ERROR,

  1053.                    "Indicated length of substream %d data goes off end of "

  1054.                    "packet.\n", substr);

  1055.             end = length - header_size - substr_header_size;

  1056.         }

  1057. 

  1058.         if (end < substream_start) {

  1059.             av_log(avctx, AV_LOG_ERROR,

  1060.                    "Indicated end offset of substream %d data "

  1061.                    "is smaller than calculated start offset.\n",

  1062.                    substr);

  1063.             goto error;

  1064.         }

  1065. 

  1066.         if (substr > m->max_decoded_substream)

  1067.             continue;

  1068. 

  1069.         substream_parity_present[substr] = checkdata_present;

  1070.         substream_data_len[substr] = end - substream_start;

  1071.         substream_start = end;

  1072.     }

  1073. 

  1074.     parity_bits  = calculate_parity(buf, 4);

  1075.     parity_bits ^= calculate_parity(buf + header_size, substr_header_size);

  1076. 

  1077.     if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {

  1078.         av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");

  1079.         goto error;

  1080.     }

  1081. 

  1082.     buf += header_size + substr_header_size;

  1083. 

  1084.     for (substr = 0; substr <= m->max_decoded_substream; substr++) {

  1085.         SubStream *s = &m->substream[substr];

  1086.         init_get_bits(&gb, buf, substream_data_len[substr] * 8);

  1087. 

  1088.         s->blockpos = 0;

  1089.         do {

  1090.             if (get_bits1(&gb)) {

  1091.                 if (get_bits1(&gb)) {

  1092.                     /* A restart header should be present. */

  1093.                     if (read_restart_header(m, &gb, buf, substr) < 0)

  1094.                         goto next_substr;

  1095.                     s->restart_seen = 1;

  1096.                 }

  1097. 

  1098.                 if (!s->restart_seen) {

  1099.                     av_log(m->avctx, AV_LOG_ERROR,

  1100.                            "No restart header present in substream %d.\n",

  1101.                            substr);

  1102.                     goto next_substr;

  1103.                 }

  1104. 

  1105.                 if (read_decoding_params(m, &gb, substr) < 0)

  1106.                     goto next_substr;

  1107.             }

  1108. 

  1109.             if (!s->restart_seen) {

  1110.                 av_log(m->avctx, AV_LOG_ERROR,

  1111.                        "No restart header present in substream %d.\n",

  1112.                        substr);

  1113.                 goto next_substr;

  1114.             }

  1115. 

  1116.             if (read_block_data(m, &gb, substr) < 0)

  1117.                 return -1;

  1118. 

  1119.         } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)

  1120.                  && get_bits1(&gb) == 0);

  1121. 

  1122.         skip_bits(&gb, (-get_bits_count(&gb)) & 15);

  1123.         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&

  1124.             (show_bits_long(&gb, 32) == 0xd234d234 ||

  1125.              show_bits_long(&gb, 20) == 0xd234e)) {

  1126.             skip_bits(&gb, 18);

  1127.             if (substr == m->max_decoded_substream)

  1128.                 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");

  1129. 

  1130.             if (get_bits1(&gb)) {

  1131.                 int shorten_by = get_bits(&gb, 13);

  1132.                 shorten_by = FFMIN(shorten_by, s->blockpos);

  1133.                 s->blockpos -= shorten_by;

  1134.             } else

  1135.                 skip_bits(&gb, 13);

  1136.         }

  1137.         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&

  1138.             substream_parity_present[substr]) {

  1139.             uint8_t parity, checksum;

  1140. 

  1141.             parity = calculate_parity(buf, substream_data_len[substr] - 2);

  1142.             if ((parity ^ get_bits(&gb, 8)) != 0xa9)

  1143.                 av_log(m->avctx, AV_LOG_ERROR,

  1144.                        "Substream %d parity check failed.\n", substr);

  1145. 

  1146.             checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);

  1147.             if (checksum != get_bits(&gb, 8))

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

  1149.                        substr);

  1150.         }

  1151.         if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {

  1152.             av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",

  1153.                    substr);

  1154.             return -1;

  1155.         }

  1156. 

  1157. next_substr:

  1158.         buf += substream_data_len[substr];

  1159.     }

  1160. 

  1161.     rematrix_channels(m, m->max_decoded_substream);

  1162. 

  1163.     if (output_data(m, m->max_decoded_substream, data, data_size) < 0)

  1164.         return -1;

  1165. 

  1166.     return length;

  1167. 

  1168. error:

  1169.     m->params_valid = 0;

  1170.     return -1;

  1171. }

  1172. 

  1173. AVCodec mlp_decoder = {

  1174.     "mlp",

  1175.     CODEC_TYPE_AUDIO,

  1176.     CODEC_ID_MLP,

  1177.     sizeof(MLPDecodeContext),

  1178.     mlp_decode_init,

  1179.     NULL,

  1180.     NULL,

  1181.     read_access_unit,

  1182.     .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),

  1183. };