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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.     return 0;

  340. }

  341. 

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

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

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

  345.  */

  346. 

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

  348. {

  349.     MLPHeaderInfo mh;

  350.     int substr;

  351. 

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

  353.         return -1;

  354. 

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

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

  357.         return -1;

  358.     }

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

  360.         av_log(m->avctx, AV_LOG_ERROR,

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

  362.         return -1;

  363.     }

  364. 

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

  366.         av_log(m->avctx, AV_LOG_ERROR,

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

  368.         return -1;

  369.     }

  370. 

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

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

  373.         return -1;

  374.     }

  375.     if (mh.group1_samplerate > MAX_SAMPLERATE) {

  376.         av_log(m->avctx, AV_LOG_ERROR,

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

  378.                mh.group1_samplerate, MAX_SAMPLERATE);

  379.         return -1;

  380.     }

  381.     if (mh.access_unit_size > MAX_BLOCKSIZE) {

  382.         av_log(m->avctx, AV_LOG_ERROR,

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

  384.                mh.access_unit_size, MAX_BLOCKSIZE);

  385.         return -1;

  386.     }

  387.     if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {

  388.         av_log(m->avctx, AV_LOG_ERROR,

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

  390.                mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);

  391.         return -1;

  392.     }

  393. 

  394.     if (mh.num_substreams == 0)

  395.         return -1;

  396.     if (mh.num_substreams > MAX_SUBSTREAMS) {

  397.         av_log(m->avctx, AV_LOG_ERROR,

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

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

  400.         return -1;

  401.     }

  402. 

  403.     m->access_unit_size      = mh.access_unit_size;

  404.     m->access_unit_size_pow2 = mh.access_unit_size_pow2;

  405. 

  406.     m->num_substreams        = mh.num_substreams;

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

  408. 

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

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

  411. 

  412. #ifdef CONFIG_AUDIO_NONSHORT

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

  414.     if (mh.group1_bits > 16) {

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

  416.     }

  417. #endif

  418. 

  419.     m->params_valid = 1;

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

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

  422. 

  423.     return 0;

  424. }

  425. 

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

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

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

  429. 

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

  431.                                const uint8_t *buf, unsigned int substr)

  432. {

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

  434.     unsigned int ch;

  435.     int sync_word, tmp;

  436.     uint8_t checksum;

  437.     uint8_t lossless_check;

  438.     int start_count = get_bits_count(gbp);

  439. 

  440.     sync_word = get_bits(gbp, 13);

  441. 

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

  443.         av_log(m->avctx, AV_LOG_ERROR,

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

  445.         return -1;

  446.     }

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

  448. 

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

  450. 

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

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

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

  454. 

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

  456.         av_log(m->avctx, AV_LOG_ERROR,

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

  458.         return -1;

  459.     }

  460. 

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

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

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

  464.         av_log(m->avctx, AV_LOG_INFO,

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

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

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

  468.         m->max_decoded_substream = substr;

  469.     }

  470. 

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

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

  473. 

  474.     skip_bits(gbp, 19);

  475. 

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

  477.     lossless_check = get_bits(gbp, 8);

  478.     if (substr == m->max_decoded_substream

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

  480.         tmp = s->lossless_check_data;

  481.         tmp ^= tmp >> 16;

  482.         tmp ^= tmp >> 8;

  483.         tmp &= 0xff;

  484.         if (tmp != lossless_check)

  485.             av_log(m->avctx, AV_LOG_WARNING,

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

  487.                    lossless_check, tmp);

  488.         else

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

  490.                     substr, tmp);

  491.     }

  492. 

  493.     skip_bits(gbp, 16);

  494. 

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

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

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

  498.                 ch_assign);

  499.         if (ch_assign != ch) {

  500.             av_log(m->avctx, AV_LOG_ERROR,

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

  502.                    sample_message);

  503.             return -1;

  504.         }

  505.     }

  506. 

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

  508. 

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

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

  511. 

  512.     /* Set default decoding parameters. */

  513.     s->param_presence_flags   = 0xff;

  514.     s->num_primitive_matrices = 0;

  515.     s->blocksize              = 8;

  516.     s->lossless_check_data    = 0;

  517. 

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

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

  520. 

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

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

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

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

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

  526. 

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

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

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

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

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

  532.     }

  533. 

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

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

  536.     }

  537. 

  538.     return 0;

  539. }

  540. 

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

  542. 

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

  544.                               unsigned int channel, unsigned int filter)

  545. {

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

  547.     int i, order;

  548. 

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

  550.     assert(filter < 2);

  551. 

  552.     order = get_bits(gbp, 4);

  553.     if (order > MAX_FILTER_ORDER) {

  554.         av_log(m->avctx, AV_LOG_ERROR,

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

  556.                fchar, order, MAX_FILTER_ORDER);

  557.         return -1;

  558.     }

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

  560. 

  561.     if (order > 0) {

  562.         int coeff_bits, coeff_shift;

  563. 

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

  565. 

  566.         coeff_bits  = get_bits(gbp, 5);

  567.         coeff_shift = get_bits(gbp, 3);

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

  569.             av_log(m->avctx, AV_LOG_ERROR,

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

  571.                    fchar);

  572.             return -1;

  573.         }

  574.         if (coeff_bits + coeff_shift > 16) {

  575.             av_log(m->avctx, AV_LOG_ERROR,

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

  577.                    fchar);

  578.             return -1;

  579.         }

  580. 

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

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

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

  584. 

  585.         if (get_bits1(gbp)) {

  586.             int state_bits, state_shift;

  587. 

  588.             if (filter == FIR) {

  589.                 av_log(m->avctx, AV_LOG_ERROR,

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

  591.                 return -1;

  592.             }

  593. 

  594.             state_bits  = get_bits(gbp, 4);

  595.             state_shift = get_bits(gbp, 4);

  596. 

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

  598. 

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

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

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

  602.         }

  603.     }

  604. 

  605.     return 0;

  606. }

  607. 

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

  609.  *  header. */

  610. 

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

  612.                                 unsigned int substr)

  613. {

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

  615.     unsigned int mat, ch;

  616. 

  617.     if (get_bits1(gbp))

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

  619. 

  620.     if (s->param_presence_flags & PARAM_BLOCKSIZE)

  621.         if (get_bits1(gbp)) {

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

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

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

  625.                 s->blocksize = 0;

  626.                 return -1;

  627.             }

  628.         }

  629. 

  630.     if (s->param_presence_flags & PARAM_MATRIX)

  631.         if (get_bits1(gbp)) {

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

  633. 

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

  635.                 int frac_bits, max_chan;

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

  637.                 frac_bits             = get_bits(gbp, 4);

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

  639. 

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

  641.                     av_log(m->avctx, AV_LOG_ERROR,

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

  643.                            s->matrix_out_ch[mat]);

  644.                     return -1;

  645.                 }

  646.                 if (frac_bits > 14) {

  647.                     av_log(m->avctx, AV_LOG_ERROR,

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

  649.                     return -1;

  650.                 }

  651. 

  652.                 max_chan = s->max_matrix_channel;

  653.                 if (!s->noise_type)

  654.                     max_chan+=2;

  655. 

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

  657.                     int coeff_val = 0;

  658.                     if (get_bits1(gbp))

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

  660. 

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

  662.                 }

  663. 

  664.                 if (s->noise_type)

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

  666.                 else

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

  668.             }

  669.         }

  670. 

  671.     if (s->param_presence_flags & PARAM_OUTSHIFT)

  672.         if (get_bits1(gbp))

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

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

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

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

  677.                 /* TODO: validate */

  678.             }

  679. 

  680.     if (s->param_presence_flags & PARAM_QUANTSTEP)

  681.         if (get_bits1(gbp))

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

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

  684.                 /* TODO: validate */

  685. 

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

  687.             }

  688. 

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

  690.         if (get_bits1(gbp)) {

  691.             if (s->param_presence_flags & PARAM_FIR)

  692.                 if (get_bits1(gbp))

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

  694.                         return -1;

  695. 

  696.             if (s->param_presence_flags & PARAM_IIR)

  697.                 if (get_bits1(gbp))

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

  699.                         return -1;

  700. 

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

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

  703.                 av_log(m->avctx, AV_LOG_ERROR,

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

  705.                 return -1;

  706.             }

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

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

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

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

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

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

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

  714. 

  715.             if (s->param_presence_flags & PARAM_HUFFOFFSET)

  716.                 if (get_bits1(gbp))

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

  718. 

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

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

  721. 

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

  723. 

  724.             /* TODO: validate */

  725.         }

  726. 

  727.     return 0;

  728. }

  729. 

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

  731. 

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

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

  734. 

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

  736.                            unsigned int channel)

  737. {

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

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

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

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

  742.     int index = MAX_BLOCKSIZE;

  743.     int j, i;

  744. 

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

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

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

  748.                MAX_FILTER_ORDER * sizeof(int32_t));

  749.     }

  750. 

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

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

  753.         unsigned int order;

  754.         int64_t accum = 0;

  755.         int32_t result;

  756. 

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

  758. 

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

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

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

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

  763. 

  764.         accum  = accum >> filter_shift;

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

  766. 

  767.         --index;

  768. 

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

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

  771. 

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

  773.     }

  774. 

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

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

  777.                &   filter_state_buffer  [j][index],

  778.                MAX_FILTER_ORDER * sizeof(int32_t));

  779.     }

  780. }

  781. 

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

  783. 

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

  785.                            unsigned int substr)

  786. {

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

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

  789. 

  790.     if (s->data_check_present) {

  791.         expected_stream_pos  = get_bits_count(gbp);

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

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

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

  795.     }

  796. 

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

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

  799.         return -1;

  800.     }

  801. 

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

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

  804. 

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

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

  807.             return -1;

  808.     }

  809. 

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

  811.         filter_channel(m, substr, ch);

  812.     }

  813. 

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

  815. 

  816.     if (s->data_check_present) {

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

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

  819.         skip_bits(gbp, 8);

  820.     }

  821. 

  822.     return 0;

  823. }

  824. 

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

  826. 

  827. static const int8_t noise_table[256] = {

  828.      30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,

  829.      52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,

  830.      10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,

  831.      51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,

  832.      38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,

  833.      61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,

  834.      67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,

  835.      48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,

  836.       0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,

  837.      16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,

  838.      13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,

  839.      89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,

  840.      36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,

  841.      39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,

  842.      45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,

  843.     -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,

  844. };

  845. 

  846. /** Noise generation functions.

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

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

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

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

  851.  *  dithering? */

  852. 

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

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

  855. 

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

  857. {

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

  859.     unsigned int i;

  860.     uint32_t seed = s->noisegen_seed;

  861.     unsigned int maxchan = s->max_matrix_channel;

  862. 

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

  864.         uint16_t seed_shr7 = seed >> 7;

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

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

  867. 

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

  869.     }

  870. 

  871.     s->noisegen_seed = seed;

  872. }

  873. 

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

  875. 

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

  877. {

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

  879.     unsigned int i;

  880.     uint32_t seed = s->noisegen_seed;

  881. 

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

  883.         uint8_t seed_shr15 = seed >> 15;

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

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

  886.     }

  887. 

  888.     s->noisegen_seed = seed;

  889. }

  890. 

  891. 

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

  893.  *  samples. */

  894. 

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

  896. {

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

  898.     unsigned int mat, src_ch, i;

  899.     unsigned int maxchan;

  900. 

  901.     maxchan = s->max_matrix_channel;

  902.     if (!s->noise_type) {

  903.         generate_2_noise_channels(m, substr);

  904.         maxchan += 2;

  905.     } else {

  906.         fill_noise_buffer(m, substr);

  907.     }

  908. 

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

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

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

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

  913. 

  914.         /* TODO: DSPContext? */

  915. 

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

  917.             int64_t accum = 0;

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

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

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

  921.             }

  922.             if (matrix_noise_shift) {

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

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

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

  926.             }

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

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

  929.         }

  930.     }

  931. }

  932. 

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

  934. 

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

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

  937. {

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

  939.     unsigned int i, ch = 0;

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

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

  942. 

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

  944.         return -1;

  945. 

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

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

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

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

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

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

  952.         }

  953.     }

  954. 

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

  956. 

  957.     return 0;

  958. }

  959. 

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

  961.                        uint8_t *data, unsigned int *data_size)

  962. {

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

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

  965.     else

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

  967. }

  968. 

  969. 

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

  971.  *  Does this belong in dspcontext? */

  972. 

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

  974. {

  975.     uint32_t scratch = 0;

  976.     const uint8_t *buf_end = buf + buf_size;

  977. 

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

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

  980. 

  981.     scratch ^= scratch >> 16;

  982.     scratch ^= scratch >> 8;

  983. 

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

  985.         scratch ^= *buf;

  986. 

  987.     return scratch;

  988. }

  989. 

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

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

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

  993. 

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

  995.                             const uint8_t *buf, int buf_size)

  996. {

  997.     MLPDecodeContext *m = avctx->priv_data;

  998.     GetBitContext gb;

  999.     unsigned int length, substr;

  1000.     unsigned int substream_start;

  1001.     unsigned int header_size = 4;

  1002.     unsigned int substr_header_size = 0;

  1003.     uint8_t substream_parity_present[MAX_SUBSTREAMS];

  1004.     uint16_t substream_data_len[MAX_SUBSTREAMS];

  1005.     uint8_t parity_bits;

  1006. 

  1007.     if (buf_size < 4)

  1008.         return 0;

  1009. 

  1010.     length = (AV_RB16(buf) & 0xfff) * 2;

  1011. 

  1012.     if (length > buf_size)

  1013.         return -1;

  1014. 

  1015.     init_get_bits(&gb, (buf + 4), (length - 4) * 8);

  1016. 

  1017.     if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {

  1018.         dprintf(m->avctx, "Found major sync.\n");

  1019.         if (read_major_sync(m, &gb) < 0)

  1020.             goto error;

  1021.         header_size += 28;

  1022.     }

  1023. 

  1024.     if (!m->params_valid) {

  1025.         av_log(m->avctx, AV_LOG_WARNING,

  1026.                "Stream parameters not seen; skipping frame.\n");

  1027.         *data_size = 0;

  1028.         return length;

  1029.     }

  1030. 

  1031.     substream_start = 0;

  1032. 

  1033.     for (substr = 0; substr < m->num_substreams; substr++) {

  1034.         int extraword_present, checkdata_present, end;

  1035. 

  1036.         extraword_present = get_bits1(&gb);

  1037.         skip_bits1(&gb);

  1038.         checkdata_present = get_bits1(&gb);

  1039.         skip_bits1(&gb);

  1040. 

  1041.         end = get_bits(&gb, 12) * 2;

  1042. 

  1043.         substr_header_size += 2;

  1044. 

  1045.         if (extraword_present) {

  1046.             skip_bits(&gb, 16);

  1047.             substr_header_size += 2;

  1048.         }

  1049. 

  1050.         if (end + header_size + substr_header_size > length) {

  1051.             av_log(m->avctx, AV_LOG_ERROR,

  1052.                    "Indicated length of substream %d data goes off end of "

  1053.                    "packet.\n", substr);

  1054.             end = length - header_size - substr_header_size;

  1055.         }

  1056. 

  1057.         if (end < substream_start) {

  1058.             av_log(avctx, AV_LOG_ERROR,

  1059.                    "Indicated end offset of substream %d data "

  1060.                    "is smaller than calculated start offset.\n",

  1061.                    substr);

  1062.             goto error;

  1063.         }

  1064. 

  1065.         if (substr > m->max_decoded_substream)

  1066.             continue;

  1067. 

  1068.         substream_parity_present[substr] = checkdata_present;

  1069.         substream_data_len[substr] = end - substream_start;

  1070.         substream_start = end;

  1071.     }

  1072. 

  1073.     parity_bits  = calculate_parity(buf, 4);

  1074.     parity_bits ^= calculate_parity(buf + header_size, substr_header_size);

  1075. 

  1076.     if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {

  1077.         av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");

  1078.         goto error;

  1079.     }

  1080. 

  1081.     buf += header_size + substr_header_size;

  1082. 

  1083.     for (substr = 0; substr <= m->max_decoded_substream; substr++) {

  1084.         SubStream *s = &m->substream[substr];

  1085.         init_get_bits(&gb, buf, substream_data_len[substr] * 8);

  1086. 

  1087.         s->blockpos = 0;

  1088.         do {

  1089.             if (get_bits1(&gb)) {

  1090.                 if (get_bits1(&gb)) {

  1091.                     /* A restart header should be present. */

  1092.                     if (read_restart_header(m, &gb, buf, substr) < 0)

  1093.                         goto next_substr;

  1094.                     s->restart_seen = 1;

  1095.                 }

  1096. 

  1097.                 if (!s->restart_seen) {

  1098.                     av_log(m->avctx, AV_LOG_ERROR,

  1099.                            "No restart header present in substream %d.\n",

  1100.                            substr);

  1101.                     goto next_substr;

  1102.                 }

  1103. 

  1104.                 if (read_decoding_params(m, &gb, substr) < 0)

  1105.                     goto next_substr;

  1106.             }

  1107. 

  1108.             if (!s->restart_seen) {

  1109.                 av_log(m->avctx, AV_LOG_ERROR,

  1110.                        "No restart header present in substream %d.\n",

  1111.                        substr);

  1112.                 goto next_substr;

  1113.             }

  1114. 

  1115.             if (read_block_data(m, &gb, substr) < 0)

  1116.                 return -1;

  1117. 

  1118.         } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)

  1119.                  && get_bits1(&gb) == 0);

  1120. 

  1121.         skip_bits(&gb, (-get_bits_count(&gb)) & 15);

  1122.         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 48 &&

  1123.             (show_bits_long(&gb, 32) == 0xd234d234 ||

  1124.              show_bits_long(&gb, 20) == 0xd234e)) {

  1125.             skip_bits(&gb, 18);

  1126.             if (substr == m->max_decoded_substream)

  1127.                 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");

  1128. 

  1129.             if (get_bits1(&gb)) {

  1130.                 int shorten_by = get_bits(&gb, 13);

  1131.                 shorten_by = FFMIN(shorten_by, s->blockpos);

  1132.                 s->blockpos -= shorten_by;

  1133.             } else

  1134.                 skip_bits(&gb, 13);

  1135.         }

  1136.         if (substream_parity_present[substr]) {

  1137.             uint8_t parity, checksum;

  1138. 

  1139.             parity = calculate_parity(buf, substream_data_len[substr] - 2);

  1140.             if ((parity ^ get_bits(&gb, 8)) != 0xa9)

  1141.                 av_log(m->avctx, AV_LOG_ERROR,

  1142.                        "Substream %d parity check failed.\n", substr);

  1143. 

  1144.             checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);

  1145.             if (checksum != get_bits(&gb, 8))

  1146.                 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",

  1147.                        substr);

  1148.         }

  1149.         if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {

  1150.             av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",

  1151.                    substr);

  1152.             return -1;

  1153.         }

  1154. 

  1155. next_substr:

  1156.         buf += substream_data_len[substr];

  1157.     }

  1158. 

  1159.     rematrix_channels(m, m->max_decoded_substream);

  1160. 

  1161.     if (output_data(m, m->max_decoded_substream, data, data_size) < 0)

  1162.         return -1;

  1163. 

  1164.     return length;

  1165. 

  1166. error:

  1167.     m->params_valid = 0;

  1168.     return -1;

  1169. }

  1170. 

  1171. AVCodec mlp_decoder = {

  1172.     "mlp",

  1173.     CODEC_TYPE_AUDIO,

  1174.     CODEC_ID_MLP,

  1175.     sizeof(MLPDecodeContext),

  1176.     mlp_decode_init,

  1177.     NULL,

  1178.     NULL,

  1179.     read_access_unit,

  1180.     .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),

  1181. };