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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. #define FIR 0

  143. #define IIR 1

  144. 

  145. /** filter data */

  146. typedef struct {

  147.     uint8_t     order; ///< number of taps in filter

  148.     uint8_t     shift; ///< Right shift to apply to output of filter.

  149. 

  150.     int32_t     coeff[MAX_FILTER_ORDER];

  151.     int32_t     state[MAX_FILTER_ORDER];

  152. } FilterParams;

  153. 

  154. /** sample data coding information */

  155. typedef struct {

  156.     FilterParams filter_params[NUM_FILTERS];

  157. 

  158.     int16_t     huff_offset;      ///< Offset to apply to residual values.

  159.     int32_t     sign_huff_offset; ///< sign/rounding-corrected version of huff_offset

  160.     uint8_t     codebook;         ///< Which VLC codebook to use to read residuals.

  161.     uint8_t     huff_lsbs;        ///< Size of residual suffix not encoded using VLC.

  162. } ChannelParams;

  163. 

  164. typedef struct MLPDecodeContext {

  165.     AVCodecContext *avctx;

  166. 

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

  168.     uint8_t     params_valid;

  169. 

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

  171.     uint8_t     num_substreams;

  172. 

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

  174.     uint8_t     max_decoded_substream;

  175. 

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

  177.     int         access_unit_size;

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

  179.     int         access_unit_size_pow2;

  180. 

  181.     SubStream   substream[MAX_SUBSTREAMS];

  182. 

  183.     ChannelParams channel_params[MAX_CHANNELS];

  184. 

  185.     int8_t      noise_buffer[MAX_BLOCKSIZE_POW2];

  186.     int8_t      bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];

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

  188. } MLPDecodeContext;

  189. 

  190. /** Tables defining the Huffman codes.

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

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

  193.  *  00 or 01, but have different codes starting with 1. */

  194. 

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

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

  197.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  198.         {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},

  199.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

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

  201.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  202.         {0x02, 2}, {0x03, 2},

  203.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

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

  205.         {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},

  206.         {0x01, 1},

  207.         {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},

  208.     }

  209. };

  210. 

  211. static VLC huff_vlc[3];

  212. 

  213. static int crc_init = 0;

  214. static AVCRC crc_63[1024];

  215. static AVCRC crc_1D[1024];

  216. 

  217. 

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

  219. 

  220. static av_cold void init_static()

  221. {

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

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

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

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

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

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

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

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

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

  231. 

  232.     if (!crc_init) {

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

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

  235.         crc_init = 1;

  236.     }

  237. }

  238. 

  239. 

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

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

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

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

  244. 

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

  246. {

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

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

  249.     return checksum;

  250. }

  251. 

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

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

  254. 

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

  256. {

  257.     int i;

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

  259. 

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

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

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

  263. 

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

  265.         crc <<= 1;

  266.         if (crc & 0x100)

  267.             crc ^= 0x11D;

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

  269.     }

  270. 

  271.     return crc;

  272. }

  273. 

  274. static inline int32_t calculate_sign_huff(MLPDecodeContext *m,

  275.                                           unsigned int substr, unsigned int ch)

  276. {

  277.     ChannelParams *cp = &m->channel_params[ch];

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

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

  280.     int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);

  281.     int32_t sign_huff_offset = cp->huff_offset;

  282. 

  283.     if (cp->codebook > 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.         ChannelParams *cp = &m->channel_params[channel];

  307.         int codebook = cp->codebook;

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

  309.         int lsb_bits = cp->huff_lsbs - quant_step_size;

  310.         int result = 0;

  311. 

  312.         if (codebook > 0)

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

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

  315. 

  316.         if (result < 0)

  317.             return -1;

  318. 

  319.         if (lsb_bits > 0)

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

  321. 

  322.         result  += cp->sign_huff_offset;

  323.         result <<= quant_step_size;

  324. 

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

  326.     }

  327. 

  328.     return 0;

  329. }

  330. 

  331. static av_cold int mlp_decode_init(AVCodecContext *avctx)

  332. {

  333.     MLPDecodeContext *m = avctx->priv_data;

  334.     int substr;

  335. 

  336.     init_static();

  337.     m->avctx = avctx;

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

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

  340.     avctx->sample_fmt = SAMPLE_FMT_S16;

  341.     return 0;

  342. }

  343. 

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

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

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

  347.  */

  348. 

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

  350. {

  351.     MLPHeaderInfo mh;

  352.     int substr;

  353. 

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

  355.         return -1;

  356. 

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

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

  359.         return -1;

  360.     }

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

  362.         av_log(m->avctx, AV_LOG_ERROR,

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

  364.         return -1;

  365.     }

  366. 

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

  368.         av_log(m->avctx, AV_LOG_ERROR,

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

  370.         return -1;

  371.     }

  372. 

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

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

  375.         return -1;

  376.     }

  377.     if (mh.group1_samplerate > MAX_SAMPLERATE) {

  378.         av_log(m->avctx, AV_LOG_ERROR,

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

  380.                mh.group1_samplerate, MAX_SAMPLERATE);

  381.         return -1;

  382.     }

  383.     if (mh.access_unit_size > MAX_BLOCKSIZE) {

  384.         av_log(m->avctx, AV_LOG_ERROR,

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

  386.                mh.access_unit_size, MAX_BLOCKSIZE);

  387.         return -1;

  388.     }

  389.     if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {

  390.         av_log(m->avctx, AV_LOG_ERROR,

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

  392.                mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);

  393.         return -1;

  394.     }

  395. 

  396.     if (mh.num_substreams == 0)

  397.         return -1;

  398.     if (mh.num_substreams > MAX_SUBSTREAMS) {

  399.         av_log(m->avctx, AV_LOG_ERROR,

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

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

  402.         return -1;

  403.     }

  404. 

  405.     m->access_unit_size      = mh.access_unit_size;

  406.     m->access_unit_size_pow2 = mh.access_unit_size_pow2;

  407. 

  408.     m->num_substreams        = mh.num_substreams;

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

  410. 

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

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

  413. 

  414. #ifdef CONFIG_AUDIO_NONSHORT

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

  416.     if (mh.group1_bits > 16) {

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

  418.     }

  419. #endif

  420. 

  421.     m->params_valid = 1;

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

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

  424. 

  425.     return 0;

  426. }

  427. 

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

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

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

  431. 

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

  433.                                const uint8_t *buf, unsigned int substr)

  434. {

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

  436.     unsigned int ch;

  437.     int sync_word, tmp;

  438.     uint8_t checksum;

  439.     uint8_t lossless_check;

  440.     int start_count = get_bits_count(gbp);

  441. 

  442.     sync_word = get_bits(gbp, 13);

  443. 

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

  445.         av_log(m->avctx, AV_LOG_ERROR,

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

  447.         return -1;

  448.     }

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

  450. 

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

  452. 

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

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

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

  456. 

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

  458.         av_log(m->avctx, AV_LOG_ERROR,

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

  460.         return -1;

  461.     }

  462. 

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

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

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

  466.         av_log(m->avctx, AV_LOG_INFO,

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

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

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

  470.         m->max_decoded_substream = substr;

  471.     }

  472. 

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

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

  475. 

  476.     skip_bits(gbp, 19);

  477. 

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

  479.     lossless_check = get_bits(gbp, 8);

  480.     if (substr == m->max_decoded_substream

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

  482.         tmp = s->lossless_check_data;

  483.         tmp ^= tmp >> 16;

  484.         tmp ^= tmp >> 8;

  485.         tmp &= 0xff;

  486.         if (tmp != lossless_check)

  487.             av_log(m->avctx, AV_LOG_WARNING,

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

  489.                    lossless_check, tmp);

  490.         else

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

  492.                     substr, tmp);

  493.     }

  494. 

  495.     skip_bits(gbp, 16);

  496. 

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

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

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

  500.                 ch_assign);

  501.         if (ch_assign != ch) {

  502.             av_log(m->avctx, AV_LOG_ERROR,

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

  504.                    sample_message);

  505.             return -1;

  506.         }

  507.     }

  508. 

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

  510. 

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

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

  513. 

  514.     /* Set default decoding parameters. */

  515.     s->param_presence_flags   = 0xff;

  516.     s->num_primitive_matrices = 0;

  517.     s->blocksize              = 8;

  518.     s->lossless_check_data    = 0;

  519. 

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

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

  522. 

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

  524.         ChannelParams *cp = &m->channel_params[ch];

  525.         cp->filter_params[FIR].order = 0;

  526.         cp->filter_params[IIR].order = 0;

  527.         cp->filter_params[FIR].shift = 0;

  528.         cp->filter_params[IIR].shift = 0;

  529. 

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

  531.         cp->huff_offset      = 0;

  532.         cp->sign_huff_offset = (-1) << 23;

  533.         cp->codebook         = 0;

  534.         cp->huff_lsbs        = 24;

  535.     }

  536. 

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

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

  539.     }

  540. 

  541.     return 0;

  542. }

  543. 

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

  545. 

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

  547.                               unsigned int channel, unsigned int filter)

  548. {

  549.     FilterParams *fp = &m->channel_params[channel].filter_params[filter];

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

  551.     int i, order;

  552. 

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

  554.     assert(filter < 2);

  555. 

  556.     order = get_bits(gbp, 4);

  557.     if (order > MAX_FILTER_ORDER) {

  558.         av_log(m->avctx, AV_LOG_ERROR,

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

  560.                fchar, order, MAX_FILTER_ORDER);

  561.         return -1;

  562.     }

  563.     fp->order = order;

  564. 

  565.     if (order > 0) {

  566.         int coeff_bits, coeff_shift;

  567. 

  568.         fp->shift = get_bits(gbp, 4);

  569. 

  570.         coeff_bits  = get_bits(gbp, 5);

  571.         coeff_shift = get_bits(gbp, 3);

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

  573.             av_log(m->avctx, AV_LOG_ERROR,

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

  575.                    fchar);

  576.             return -1;

  577.         }

  578.         if (coeff_bits + coeff_shift > 16) {

  579.             av_log(m->avctx, AV_LOG_ERROR,

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

  581.                    fchar);

  582.             return -1;

  583.         }

  584. 

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

  586.             fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;

  587. 

  588.         if (get_bits1(gbp)) {

  589.             int state_bits, state_shift;

  590. 

  591.             if (filter == FIR) {

  592.                 av_log(m->avctx, AV_LOG_ERROR,

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

  594.                 return -1;

  595.             }

  596. 

  597.             state_bits  = get_bits(gbp, 4);

  598.             state_shift = get_bits(gbp, 4);

  599. 

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

  601. 

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

  603.                 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;

  604.         }

  605.     }

  606. 

  607.     return 0;

  608. }

  609. 

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

  611.  *  header. */

  612. 

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

  614.                                 unsigned int substr)

  615. {

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

  617.     unsigned int mat, ch;

  618. 

  619.     if (get_bits1(gbp))

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

  621. 

  622.     if (s->param_presence_flags & PARAM_BLOCKSIZE)

  623.         if (get_bits1(gbp)) {

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

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

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

  627.                 s->blocksize = 0;

  628.                 return -1;

  629.             }

  630.         }

  631. 

  632.     if (s->param_presence_flags & PARAM_MATRIX)

  633.         if (get_bits1(gbp)) {

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

  635. 

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

  637.                 int frac_bits, max_chan;

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

  639.                 frac_bits             = get_bits(gbp, 4);

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

  641. 

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

  643.                     av_log(m->avctx, AV_LOG_ERROR,

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

  645.                            s->matrix_out_ch[mat]);

  646.                     return -1;

  647.                 }

  648.                 if (frac_bits > 14) {

  649.                     av_log(m->avctx, AV_LOG_ERROR,

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

  651.                     return -1;

  652.                 }

  653. 

  654.                 max_chan = s->max_matrix_channel;

  655.                 if (!s->noise_type)

  656.                     max_chan+=2;

  657. 

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

  659.                     int coeff_val = 0;

  660.                     if (get_bits1(gbp))

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

  662. 

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

  664.                 }

  665. 

  666.                 if (s->noise_type)

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

  668.                 else

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

  670.             }

  671.         }

  672. 

  673.     if (s->param_presence_flags & PARAM_OUTSHIFT)

  674.         if (get_bits1(gbp))

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

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

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

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

  679.                 /* TODO: validate */

  680.             }

  681. 

  682.     if (s->param_presence_flags & PARAM_QUANTSTEP)

  683.         if (get_bits1(gbp))

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

  685.                 ChannelParams *cp = &m->channel_params[ch];

  686. 

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

  688.                 /* TODO: validate */

  689. 

  690.                 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);

  691.             }

  692. 

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

  694.         if (get_bits1(gbp)) {

  695.             ChannelParams *cp = &m->channel_params[ch];

  696.             FilterParams *fir = &cp->filter_params[FIR];

  697.             FilterParams *iir = &cp->filter_params[IIR];

  698. 

  699.             if (s->param_presence_flags & PARAM_FIR)

  700.                 if (get_bits1(gbp))

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

  702.                         return -1;

  703. 

  704.             if (s->param_presence_flags & PARAM_IIR)

  705.                 if (get_bits1(gbp))

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

  707.                         return -1;

  708. 

  709.             if (fir->order && iir->order &&

  710.                 fir->shift != iir->shift) {

  711.                 av_log(m->avctx, AV_LOG_ERROR,

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

  713.                 return -1;

  714.             }

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

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

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

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

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

  720.             if (!fir->order && iir->order)

  721.                 fir->shift = iir->shift;

  722. 

  723.             if (s->param_presence_flags & PARAM_HUFFOFFSET)

  724.                 if (get_bits1(gbp))

  725.                     cp->huff_offset = get_sbits(gbp, 15);

  726. 

  727.             cp->codebook  = get_bits(gbp, 2);

  728.             cp->huff_lsbs = get_bits(gbp, 5);

  729. 

  730.             cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);

  731. 

  732.             /* TODO: validate */

  733.         }

  734. 

  735.     return 0;

  736. }

  737. 

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

  739. 

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

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

  742. 

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

  744.                            unsigned int channel)

  745. {

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

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

  748.     FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],

  749.                                       &m->channel_params[channel].filter_params[IIR], };

  750.     unsigned int filter_shift = fp[FIR]->shift;

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

  752.     int index = MAX_BLOCKSIZE;

  753.     int j, i;

  754. 

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

  756.         memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],

  757.                MAX_FILTER_ORDER * sizeof(int32_t));

  758.     }

  759. 

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

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

  762.         unsigned int order;

  763.         int64_t accum = 0;

  764.         int32_t result;

  765. 

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

  767. 

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

  769.             for (order = 0; order < fp[j]->order; order++)

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

  771.                                   fp[j]->coeff[order];

  772. 

  773.         accum  = accum >> filter_shift;

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

  775. 

  776.         --index;

  777. 

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

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

  780. 

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

  782.     }

  783. 

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

  785.         memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],

  786.                MAX_FILTER_ORDER * sizeof(int32_t));

  787.     }

  788. }

  789. 

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

  791. 

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

  793.                            unsigned int substr)

  794. {

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

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

  797. 

  798.     if (s->data_check_present) {

  799.         expected_stream_pos  = get_bits_count(gbp);

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

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

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

  803.     }

  804. 

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

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

  807.         return -1;

  808.     }

  809. 

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

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

  812. 

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

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

  815.             return -1;

  816.     }

  817. 

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

  819.         filter_channel(m, substr, ch);

  820.     }

  821. 

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

  823. 

  824.     if (s->data_check_present) {

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

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

  827.         skip_bits(gbp, 8);

  828.     }

  829. 

  830.     return 0;

  831. }

  832. 

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

  834. 

  835. static const int8_t noise_table[256] = {

  836.      30,  51,  22,  54,   3,   7,  -4,  38,  14,  55,  46,  81,  22,  58,  -3,   2,

  837.      52,  31,  -7,  51,  15,  44,  74,  30,  85, -17,  10,  33,  18,  80,  28,  62,

  838.      10,  32,  23,  69,  72,  26,  35,  17,  73,  60,   8,  56,   2,   6,  -2,  -5,

  839.      51,   4,  11,  50,  66,  76,  21,  44,  33,  47,   1,  26,  64,  48,  57,  40,

  840.      38,  16, -10, -28,  92,  22, -18,  29, -10,   5, -13,  49,  19,  24,  70,  34,

  841.      61,  48,  30,  14,  -6,  25,  58,  33,  42,  60,  67,  17,  54,  17,  22,  30,

  842.      67,  44,  -9,  50, -11,  43,  40,  32,  59,  82,  13,  49, -14,  55,  60,  36,

  843.      48,  49,  31,  47,  15,  12,   4,  65,   1,  23,  29,  39,  45,  -2,  84,  69,

  844.       0,  72,  37,  57,  27,  41, -15, -16,  35,  31,  14,  61,  24,   0,  27,  24,

  845.      16,  41,  55,  34,  53,   9,  56,  12,  25,  29,  53,   5,  20, -20,  -8,  20,

  846.      13,  28,  -3,  78,  38,  16,  11,  62,  46,  29,  21,  24,  46,  65,  43, -23,

  847.      89,  18,  74,  21,  38, -12,  19,  12, -19,   8,  15,  33,   4,  57,   9,  -8,

  848.      36,  35,  26,  28,   7,  83,  63,  79,  75,  11,   3,  87,  37,  47,  34,  40,

  849.      39,  19,  20,  42,  27,  34,  39,  77,  13,  42,  59,  64,  45,  -1,  32,  37,

  850.      45,  -5,  53,  -6,   7,  36,  50,  23,   6,  32,   9, -21,  18,  71,  27,  52,

  851.     -25,  31,  35,  42,  -1,  68,  63,  52,  26,  43,  66,  37,  41,  25,  40,  70,

  852. };

  853. 

  854. /** Noise generation functions.

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

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

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

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

  859.  *  dithering? */

  860. 

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

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

  863. 

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

  865. {

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

  867.     unsigned int i;

  868.     uint32_t seed = s->noisegen_seed;

  869.     unsigned int maxchan = s->max_matrix_channel;

  870. 

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

  872.         uint16_t seed_shr7 = seed >> 7;

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

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

  875. 

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

  877.     }

  878. 

  879.     s->noisegen_seed = seed;

  880. }

  881. 

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

  883. 

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

  885. {

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

  887.     unsigned int i;

  888.     uint32_t seed = s->noisegen_seed;

  889. 

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

  891.         uint8_t seed_shr15 = seed >> 15;

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

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

  894.     }

  895. 

  896.     s->noisegen_seed = seed;

  897. }

  898. 

  899. 

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

  901.  *  samples. */

  902. 

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

  904. {

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

  906.     unsigned int mat, src_ch, i;

  907.     unsigned int maxchan;

  908. 

  909.     maxchan = s->max_matrix_channel;

  910.     if (!s->noise_type) {

  911.         generate_2_noise_channels(m, substr);

  912.         maxchan += 2;

  913.     } else {

  914.         fill_noise_buffer(m, substr);

  915.     }

  916. 

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

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

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

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

  921. 

  922.         /* TODO: DSPContext? */

  923. 

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

  925.             int64_t accum = 0;

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

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

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

  929.             }

  930.             if (matrix_noise_shift) {

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

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

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

  934.             }

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

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

  937.         }

  938.     }

  939. }

  940. 

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

  942. 

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

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

  945. {

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

  947.     unsigned int i, ch = 0;

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

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

  950. 

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

  952.         return -1;

  953. 

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

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

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

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

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

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

  960.         }

  961.     }

  962. 

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

  964. 

  965.     return 0;

  966. }

  967. 

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

  969.                        uint8_t *data, unsigned int *data_size)

  970. {

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

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

  973.     else

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

  975. }

  976. 

  977. 

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

  979.  *  Does this belong in dspcontext? */

  980. 

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

  982. {

  983.     uint32_t scratch = 0;

  984.     const uint8_t *buf_end = buf + buf_size;

  985. 

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

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

  988. 

  989.     scratch ^= scratch >> 16;

  990.     scratch ^= scratch >> 8;

  991. 

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

  993.         scratch ^= *buf;

  994. 

  995.     return scratch;

  996. }

  997. 

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

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

  1000.  *  otherwise returns the number of bytes consumed. */

  1001. 

  1002. static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,

  1003.                             const uint8_t *buf, int buf_size)

  1004. {

  1005.     MLPDecodeContext *m = avctx->priv_data;

  1006.     GetBitContext gb;

  1007.     unsigned int length, substr;

  1008.     unsigned int substream_start;

  1009.     unsigned int header_size = 4;

  1010.     unsigned int substr_header_size = 0;

  1011.     uint8_t substream_parity_present[MAX_SUBSTREAMS];

  1012.     uint16_t substream_data_len[MAX_SUBSTREAMS];

  1013.     uint8_t parity_bits;

  1014. 

  1015.     if (buf_size < 4)

  1016.         return 0;

  1017. 

  1018.     length = (AV_RB16(buf) & 0xfff) * 2;

  1019. 

  1020.     if (length > buf_size)

  1021.         return -1;

  1022. 

  1023.     init_get_bits(&gb, (buf + 4), (length - 4) * 8);

  1024. 

  1025.     if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {

  1026.         dprintf(m->avctx, "Found major sync.\n");

  1027.         if (read_major_sync(m, &gb) < 0)

  1028.             goto error;

  1029.         header_size += 28;

  1030.     }

  1031. 

  1032.     if (!m->params_valid) {

  1033.         av_log(m->avctx, AV_LOG_WARNING,

  1034.                "Stream parameters not seen; skipping frame.\n");

  1035.         *data_size = 0;

  1036.         return length;

  1037.     }

  1038. 

  1039.     substream_start = 0;

  1040. 

  1041.     for (substr = 0; substr < m->num_substreams; substr++) {

  1042.         int extraword_present, checkdata_present, end;

  1043. 

  1044.         extraword_present = get_bits1(&gb);

  1045.         skip_bits1(&gb);

  1046.         checkdata_present = get_bits1(&gb);

  1047.         skip_bits1(&gb);

  1048. 

  1049.         end = get_bits(&gb, 12) * 2;

  1050. 

  1051.         substr_header_size += 2;

  1052. 

  1053.         if (extraword_present) {

  1054.             skip_bits(&gb, 16);

  1055.             substr_header_size += 2;

  1056.         }

  1057. 

  1058.         if (end + header_size + substr_header_size > length) {

  1059.             av_log(m->avctx, AV_LOG_ERROR,

  1060.                    "Indicated length of substream %d data goes off end of "

  1061.                    "packet.\n", substr);

  1062.             end = length - header_size - substr_header_size;

  1063.         }

  1064. 

  1065.         if (end < substream_start) {

  1066.             av_log(avctx, AV_LOG_ERROR,

  1067.                    "Indicated end offset of substream %d data "

  1068.                    "is smaller than calculated start offset.\n",

  1069.                    substr);

  1070.             goto error;

  1071.         }

  1072. 

  1073.         if (substr > m->max_decoded_substream)

  1074.             continue;

  1075. 

  1076.         substream_parity_present[substr] = checkdata_present;

  1077.         substream_data_len[substr] = end - substream_start;

  1078.         substream_start = end;

  1079.     }

  1080. 

  1081.     parity_bits  = calculate_parity(buf, 4);

  1082.     parity_bits ^= calculate_parity(buf + header_size, substr_header_size);

  1083. 

  1084.     if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {

  1085.         av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");

  1086.         goto error;

  1087.     }

  1088. 

  1089.     buf += header_size + substr_header_size;

  1090. 

  1091.     for (substr = 0; substr <= m->max_decoded_substream; substr++) {

  1092.         SubStream *s = &m->substream[substr];

  1093.         init_get_bits(&gb, buf, substream_data_len[substr] * 8);

  1094. 

  1095.         s->blockpos = 0;

  1096.         do {

  1097.             if (get_bits1(&gb)) {

  1098.                 if (get_bits1(&gb)) {

  1099.                     /* A restart header should be present. */

  1100.                     if (read_restart_header(m, &gb, buf, substr) < 0)

  1101.                         goto next_substr;

  1102.                     s->restart_seen = 1;

  1103.                 }

  1104. 

  1105.                 if (!s->restart_seen) {

  1106.                     av_log(m->avctx, AV_LOG_ERROR,

  1107.                            "No restart header present in substream %d.\n",

  1108.                            substr);

  1109.                     goto next_substr;

  1110.                 }

  1111. 

  1112.                 if (read_decoding_params(m, &gb, substr) < 0)

  1113.                     goto next_substr;

  1114.             }

  1115. 

  1116.             if (!s->restart_seen) {

  1117.                 av_log(m->avctx, AV_LOG_ERROR,

  1118.                        "No restart header present in substream %d.\n",

  1119.                        substr);

  1120.                 goto next_substr;

  1121.             }

  1122. 

  1123.             if (read_block_data(m, &gb, substr) < 0)

  1124.                 return -1;

  1125. 

  1126.         } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)

  1127.                  && get_bits1(&gb) == 0);

  1128. 

  1129.         skip_bits(&gb, (-get_bits_count(&gb)) & 15);

  1130.         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&

  1131.             (show_bits_long(&gb, 32) == 0xd234d234 ||

  1132.              show_bits_long(&gb, 20) == 0xd234e)) {

  1133.             skip_bits(&gb, 18);

  1134.             if (substr == m->max_decoded_substream)

  1135.                 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");

  1136. 

  1137.             if (get_bits1(&gb)) {

  1138.                 int shorten_by = get_bits(&gb, 13);

  1139.                 shorten_by = FFMIN(shorten_by, s->blockpos);

  1140.                 s->blockpos -= shorten_by;

  1141.             } else

  1142.                 skip_bits(&gb, 13);

  1143.         }

  1144.         if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&

  1145.             substream_parity_present[substr]) {

  1146.             uint8_t parity, checksum;

  1147. 

  1148.             parity = calculate_parity(buf, substream_data_len[substr] - 2);

  1149.             if ((parity ^ get_bits(&gb, 8)) != 0xa9)

  1150.                 av_log(m->avctx, AV_LOG_ERROR,

  1151.                        "Substream %d parity check failed.\n", substr);

  1152. 

  1153.             checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);

  1154.             if (checksum != get_bits(&gb, 8))

  1155.                 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",

  1156.                        substr);

  1157.         }

  1158.         if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {

  1159.             av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",

  1160.                    substr);

  1161.             return -1;

  1162.         }

  1163. 

  1164. next_substr:

  1165.         buf += substream_data_len[substr];

  1166.     }

  1167. 

  1168.     rematrix_channels(m, m->max_decoded_substream);

  1169. 

  1170.     if (output_data(m, m->max_decoded_substream, data, data_size) < 0)

  1171.         return -1;

  1172. 

  1173.     return length;

  1174. 

  1175. error:

  1176.     m->params_valid = 0;

  1177.     return -1;

  1178. }

  1179. 

  1180. AVCodec mlp_decoder = {

  1181.     "mlp",

  1182.     CODEC_TYPE_AUDIO,

  1183.     CODEC_ID_MLP,

  1184.     sizeof(MLPDecodeContext),

  1185.     mlp_decode_init,

  1186.     NULL,

  1187.     NULL,

  1188.     read_access_unit,

  1189.     .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),

  1190. };