vdr/xine-lib-vdr/src/libdts Makefile.am Makefile.in bitstream.c bitstream.h downmix.c dts.h dts_internal.h parse.c tables.h tables_adpcm.h tables_fir.h tables_huffman.h tables_quantization.h tables_vq.h xine_decoder.c
Darren Salt
pkg-vdr-dvb-changes@lists.alioth.debian.org
Mon, 04 Apr 2005 22:29:50 +0000
Update of /cvsroot/pkg-vdr-dvb/vdr/xine-lib-vdr/src/libdts
In directory haydn:/tmp/cvs-serv2129/src/libdts
Added Files:
Makefile.am Makefile.in bitstream.c bitstream.h downmix.c
dts.h dts_internal.h parse.c tables.h tables_adpcm.h
tables_fir.h tables_huffman.h tables_quantization.h
tables_vq.h xine_decoder.c
Log Message:
Import of VDR-patched xine-lib.
--- NEW FILE: dts_internal.h ---
/*
* dts_internal.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
* Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
* Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#define DTS_SUBFRAMES_MAX (16)
#define DTS_PRIM_CHANNELS_MAX (5)
#define DTS_SUBBANDS (32)
#define DTS_ABITS_MAX (32) /* Should be 28 */
#define DTS_SUBSUBFAMES_MAX (4)
#define DTS_LFE_MAX (3)
struct dts_state_s {
/* Frame header */
int frame_type; /* type of the current frame */
int samples_deficit; /* deficit sample count */
int crc_present; /* crc is present in the bitstream */
int sample_blocks; /* number of PCM sample blocks */
int frame_size; /* primary frame byte size */
int amode; /* audio channels arrangement */
int sample_rate; /* audio sampling rate */
int bit_rate; /* transmission bit rate */
int downmix; /* embedded downmix enabled */
int dynrange; /* embedded dynamic range flag */
int timestamp; /* embedded time stamp flag */
int aux_data; /* auxiliary data flag */
int hdcd; /* source material is mastered in HDCD */
int ext_descr; /* extension audio descriptor flag */
int ext_coding; /* extended coding flag */
int aspf; /* audio sync word insertion flag */
int lfe; /* low frequency effects flag */
int predictor_history; /* predictor history flag */
int header_crc; /* header crc check bytes */
int multirate_inter; /* multirate interpolator switch */
int version; /* encoder software revision */
int copy_history; /* copy history */
int source_pcm_res; /* source pcm resolution */
int front_sum; /* front sum/difference flag */
int surround_sum; /* surround sum/difference flag */
int dialog_norm; /* dialog normalisation parameter */
/* Primary audio coding header */
int subframes; /* number of subframes */
int prim_channels; /* number of primary audio channels */
/* subband activity count */
int subband_activity[DTS_PRIM_CHANNELS_MAX];
/* high frequency vq start subband */
int vq_start_subband[DTS_PRIM_CHANNELS_MAX];
/* joint intensity coding index */
int joint_intensity[DTS_PRIM_CHANNELS_MAX];
/* transient mode code book */
int transient_huffman[DTS_PRIM_CHANNELS_MAX];
/* scale factor code book */
int scalefactor_huffman[DTS_PRIM_CHANNELS_MAX];
/* bit allocation quantizer select */
int bitalloc_huffman[DTS_PRIM_CHANNELS_MAX];
/* quantization index codebook select */
int quant_index_huffman[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];
/* scale factor adjustment */
float scalefactor_adj[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];
/* Primary audio coding side information */
int subsubframes; /* number of subsubframes */
int partial_samples; /* partial subsubframe samples count */
/* prediction mode (ADPCM used or not) */
int prediction_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* prediction VQ coefs */
int prediction_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* bit allocation index */
int bitalloc[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* transition mode (transients) */
int transition_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* scale factors (2 if transient)*/
int scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][2];
/* joint subband scale factors codebook */
int joint_huff[DTS_PRIM_CHANNELS_MAX];
/* joint subband scale factors */
int joint_scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* stereo downmix coefficients */
int downmix_coef[DTS_PRIM_CHANNELS_MAX][2];
/* dynamic range coefficient */
int dynrange_coef;
/* VQ encoded high frequency subbands */
int high_freq_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* Low frequency effect data */
double lfe_data[2*DTS_SUBSUBFAMES_MAX*DTS_LFE_MAX * 2 /*history*/];
int lfe_scale_factor;
/* Subband samples history (for ADPCM) */
double subband_samples_hist[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][4];
double subband_fir_hist[DTS_PRIM_CHANNELS_MAX][512];
double subband_fir_noidea[DTS_PRIM_CHANNELS_MAX][64];
/* Audio output */
level_t clev; /* centre channel mix level */
level_t slev; /* surround channels mix level */
int output; /* type of output */
level_t level; /* output level */
sample_t bias; /* output bias */
sample_t * samples; /* pointer to the internal audio samples buffer */
int downmixed;
int dynrnge; /* apply dynamic range */
level_t dynrng; /* dynamic range */
void * dynrngdata; /* dynamic range callback funtion and data */
level_t (* dynrngcall) (level_t range, void * dynrngdata);
/* Bitstream handling */
uint32_t * buffer_start;
uint32_t bits_left;
uint32_t current_word;
int word_mode; /* 16/14 bits word format (1 -> 16, 0 -> 14) */
int bigendian_mode; /* endianness (1 -> be, 0 -> le) */
/* Current position in DTS frame */
int current_subframe;
int current_subsubframe;
/* Pre-calculated cosine modulation coefs for the QMF */
double cos_mod[544];
/* Debug flag */
int debug_flag;
};
#define LEVEL_PLUS6DB 2.0
#define LEVEL_PLUS3DB 1.4142135623730951
#define LEVEL_3DB 0.7071067811865476
#define LEVEL_45DB 0.5946035575013605
#define LEVEL_6DB 0.5
int dts_downmix_init (int input, int flags, level_t * level,
level_t clev, level_t slev);
int dts_downmix_coeff (level_t * coeff, int acmod, int output, level_t level,
level_t clev, level_t slev);
void dts_downmix (sample_t * samples, int acmod, int output, sample_t bias,
level_t clev, level_t slev);
void dts_upmix (sample_t * samples, int acmod, int output);
#define ROUND(x) ((int)((x) + ((x) > 0 ? 0.5 : -0.5)))
#ifndef LIBDTS_FIXED
typedef sample_t quantizer_t;
#define SAMPLE(x) (x)
#define LEVEL(x) (x)
#define MUL(a,b) ((a) * (b))
#define MUL_L(a,b) ((a) * (b))
#define MUL_C(a,b) ((a) * (b))
#define DIV(a,b) ((a) / (b))
#define BIAS(x) ((x) + bias)
#else /* LIBDTS_FIXED */
typedef int16_t quantizer_t;
#define SAMPLE(x) (sample_t)((x) * (1 << 30))
#define LEVEL(x) (level_t)((x) * (1 << 26))
#if 0
#define MUL(a,b) ((int)(((int64_t)(a) * (b) + (1 << 29)) >> 30))
#define MUL_L(a,b) ((int)(((int64_t)(a) * (b) + (1 << 25)) >> 26))
#elif 1
#define MUL(a,b) \
({ int32_t _ta=(a), _tb=(b), _tc; \
_tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)(((_tc >> 14))+ (((_ta >> 16)*(_tb >> 16)) << 2 )); })
#define MUL_L(a,b) \
({ int32_t _ta=(a), _tb=(b), _tc; \
_tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)((_tc >> 10) + (((_ta >> 16)*(_tb >> 16)) << 6)); })
#else
#define MUL(a,b) (((a) >> 15) * ((b) >> 15))
#define MUL_L(a,b) (((a) >> 13) * ((b) >> 13))
#endif
#define MUL_C(a,b) MUL_L (a, LEVEL (b))
#define DIV(a,b) ((((int64_t)LEVEL (a)) << 26) / (b))
#define BIAS(x) (x)
#endif
--- NEW FILE: tables_vq.h ---
/*
* tables_vq.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
[...2036 lines suppressed...]
6, -5, 6, -4, -7, 7, -21, 8, 1, 8, -9, -4, -3, 11, 25, -13 },
{ 4, 4, -1, -6, 4, 9, -8, 1, -3, -10, -2, 0, 15, -9, -16, 11,
1, 1, 6, 3, -9, -5, 16, 26, 1, -14, 1, -3, -14, 7, 15, -9 },
{ -12, -2, -9, -13, 2, 6, 14, 0, 1, 0, -1, -13, 0, 10, -1, 6,
9, -7, 8, 8, 19, 6, -1, 9, 10, -4, 1, -7, -22, -2, 29, -7 },
{ 2, 4, 13, -12, -8, -4, -5, 13, 12, -5, -3, -3, -4, 1, -1, 10,
15, -6, -1, -11, -30, 4, 15, -1, 9, -7, 0, -2, -7, 10, 25, -16 },
{ 7, -15, -7, -7, -1, -5, -5, -11, -20, 10, 3, -10, -3, 5, 20, -4,
0, -2, -2, 17, 2, 0, -3, 3, 6, 5, -1, -12, -3, 15, 22, -16 },
{ 4, -1, 3, 4, -5, 0, -1, -5, -24, -29, 4, -9, 1, -3, 0, 0,
0, -4, 7, -4, -4, -4, 3, 1, -6, 5, -3, -5, -10, 3, 25, -10 },
{ -2, -1, -1, 4, 4, -1, 2, 0, -4, -4, 2, -1, -3, -1, -2, -2,
1, -3, -5, -1, 2, -3, -4, -4, -3, 5, -9, 1, -11, 7, 46, -46 },
{ 0, -9, 3, 4, 4, 3, -5, -6, 5, -4, 4, -2, 1, 7, -4, -10,
13, 1, 3, -6, 4, -4, 7, 2, -19, -25, -3, -16, -12, 16, 20, -1 },
{ 18, 6, 4, -12, 0, -14, 9, -6, -1, -4, -5, 2, 1, 12, 4, 2,
7, 0, 2, 5, -11, -5, -2, 2, -4, 10, 0, -9, -7, 9, 25, -8 },
{ 5, 0, -6, 5, 6, 3, 3, -10, -5, 1, -1, 4, 3, -11, -8, 5,
4, -5, 5, -5, -7, -5, 11, 5, 20, -8, -16, 21, -4, 27, 23, -5 }
};
--- NEW FILE: tables_adpcm.h ---
/*
* tables_adpcm.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
[...4083 lines suppressed...]
{ 13441, -11509, 6676, -2434 },
{ 7912, -9398, 6663, -4048 },
{ 11723, -13745, 8131, -4148 },
{ 6065, -10257, 5005, -6327 },
{ 11618, -12417, 5336, -1894 },
{ 8891, -13924, 8407, -6131 },
{ 9622, -12563, 7908, -5109 },
{ 11479, -10315, 8349, -3991 },
{ 11676, -14103, 6611, -2330 },
{ 11951, -8953, 3829, -1550 },
{ 10486, -8044, 10493, -5920 },
{ 11801, -10769, 9763, -5305 },
{ 6109, -8676, 5827, -1346 },
{ 7030, -9611, 5624, -5761 },
{ 12808, -12886, 8683, -4148 },
{ 13213, -10464, 6381, -3189 },
{ 11796, -13681, 10703, -6075 },
{ 9639, -7949, 9625, -3944 },
{ 8538, -6997, 5309, 453 }
};
--- NEW FILE: parse.c ---
/*
* parse.c
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
[...1251 lines suppressed...]
}
}
}
void dts_dynrng (dts_state_t * state,
level_t (* call) (level_t, void *), void * data)
{
state->dynrange = 0;
if (call) {
state->dynrange = 1;
state->dynrngcall = call;
state->dynrngdata = data;
}
}
void dts_free (dts_state_t * state)
{
free (state->samples);
free (state);
}
--- NEW FILE: tables_quantization.h ---
/*
* tables_quantization.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
int scale_factor_quant6[] =
{
1, 2, 2, 3, 3, 4, 6, 7,
10, 12, 16, 20, 26, 34, 44, 56,
72, 93, 120, 155, 200, 257, 331, 427,
550, 708, 912, 1175, 1514, 1950, 2512, 3236,
4169, 5370, 6918, 8913, 11482, 14791, 19055, 24547,
31623, 40738, 52481, 67608, 87096, 112202, 144544, 186209,
239883, 309030, 398107, 512861, 660693, 851138, 1096478, 1412538,
1819701, 2344229, 3019952, 3890451, 5011872, 6456542, 8317638, 0
};
int scale_factor_quant7[] =
{
1, 1, 2, 2, 2, 2, 3, 3,
3, 4, 4, 5, 6, 7, 7, 8,
10, 11, 12, 14, 16, 18, 20, 23,
26, 30, 34, 38, 44, 50, 56, 64,
72, 82, 93, 106, 120, 136, 155, 176,
200, 226, 257, 292, 331, 376, 427, 484,
550, 624, 708, 804, 912, 1035, 1175, 1334,
1514, 1718, 1950, 2213, 2512, 2851, 3236, 3673,
4169, 4732, 5370, 6095, 6918, 7852, 8913, 10116,
11482, 13032, 14791, 16788, 19055, 21627, 24547, 27861,
31623, 35892, 40738, 46238, 52481, 59566, 67608, 76736,
87096, 98855, 112202, 127350, 144544, 164059, 186209, 211349,
239883, 272270, 309030, 350752, 398107, 451856, 512861, 582103,
660693, 749894, 851138, 966051, 1096478, 1244515, 1412538, 1603245,
1819701, 2065380, 2344229, 2660725, 3019952, 3427678, 3890451, 4415704,
5011872, 5688529, 6456542, 7328245, 8317638, 0, 0, 0
};
/* 20bits unsigned fractional binary codes */
int lossy_quant[] =
{
0, 6710886, 4194304, 3355443, 2474639, 2097152, 1761608, 1426063,
796918, 461373, 251658, 146801, 79692, 46137, 27263, 16777,
10486, 5872, 3355, 1887, 1258, 713, 336, 168,
84, 42, 21, 0, 0, 0, 0, 0
};
double lossy_quant_d[] =
{
0, 1.6, 1.0, 0.8, 0.59, 0.50, 0.42, 0.34,
0.19, 0.11, 0.06, 0.035, 0.019, 0.011, 0.0065, 0.0040,
0.0025, 0.0014, 0.0008, 0.00045, 0.00030, 0.00017, 0.00008, 0.00004,
0.00002, 0.00001, 0.000005, 0, 0, 0, 0, 0
};
/* 20bits unsigned fractional binary codes */
int lossless_quant[] =
{
0, 4194304, 2097152, 1384120, 1048576, 696254, 524288, 348127,
262144, 131072, 65431, 33026, 16450, 8208, 4100, 2049,
1024, 512, 256, 128, 64, 32, 16, 8,
4, 2, 1, 0, 0, 0, 0, 0
};
double lossless_quant_d[] =
{
0, 1.0, 0.5, 0.33, 0.25, 0.166, 0.125,
0.083, 0.0625, 0.03125, 0.0156, 7.874E-3, 3.922E-3, 1.957E-3,
9.775E-4, 4.885E-4, 2.442E-4, 1.221E-4, 6.104E-5, 3.052E-5, 1.526E-5,
7.629E-6, 3.815E-6, 1.907E-6, 9.537E-7, 4.768E-7, 2.384E-7, 0,
0, 0, 0, 0
};
--- NEW FILE: bitstream.h ---
/*
* bitstream.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
* Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
* Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#ifdef WORDS_BIGENDIAN
# define swab32(x) (x)
#else
# define swab32(x)\
((((uint8_t*)&x)[0] << 24) | (((uint8_t*)&x)[1] << 16) | \
(((uint8_t*)&x)[2] << 8) | (((uint8_t*)&x)[3]))
#endif
#ifdef WORDS_BIGENDIAN
# define swable32(x)\
((((uint8_t*)&x)[0] << 16) | (((uint8_t*)&x)[1] << 24) | \
(((uint8_t*)&x)[2]) | (((uint8_t*)&x)[3] << 8))
#else
# define swable32(x)\
((((uint16_t*)&x)[0] << 16) | (((uint16_t*)&x)[1]))
#endif
void dts_bitstream_init (dts_state_t * state, uint8_t * buf, int word_mode,
int endian_mode);
uint32_t dts_bitstream_get_bh (dts_state_t * state, uint32_t num_bits);
static inline uint32_t bitstream_get (dts_state_t * state, uint32_t num_bits)
{
uint32_t result;
if (num_bits < state->bits_left) {
result = (state->current_word << (32 - state->bits_left))
>> (32 - num_bits);
state->bits_left -= num_bits;
return result;
}
return dts_bitstream_get_bh (state, num_bits);
}
--- NEW FILE: tables_fir.h ---
/*
* fir.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
[...2046 lines suppressed...]
0.00105048984,
0.00096255314,
0.00088037323,
0.00080365466,
0.00073179678,
0.00066567765,
0.00060277141,
0.00054522208,
0.00049204525,
0.00044236859,
0.00039634691,
0.00035398375,
0.00031519096,
0.00027949660,
0.00024667382,
0.00021643363,
0.00018878609,
0.00016358691,
0.00053168571
};
--- NEW FILE: tables.h ---
/*
* tables.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
static const int dts_sample_rates[] =
{
0, 8000, 16000, 32000, 0, 0, 11025, 22050, 44100, 0, 0,
12000, 24000, 48000, 96000, 192000
};
static const int dts_bit_rates[] =
{
32000, 56000, 64000, 96000, 112000, 128000,
192000, 224000, 256000, 320000, 384000,
448000, 512000, 576000, 640000, 768000,
896000, 1024000, 1152000, 1280000, 1344000,
1408000, 1411200, 1472000, 1536000, 1920000,
2048000, 3072000, 3840000, 1/*open*/, 2/*variable*/, 3/*lossless*/
};
static const uint8_t dts_channels[] =
{
1, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 6, 6, 7, 8, 8
};
static const uint8_t dts_bits_per_sample[] =
{
16, 16, 20, 20, 0, 24, 24
};
--- NEW FILE: dts.h ---
/*
* dts.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#ifndef DTS_H
#define DTS_H
/* x86 accelerations */
#define MM_ACCEL_X86_MMX 0x80000000
#define MM_ACCEL_X86_3DNOW 0x40000000
#define MM_ACCEL_X86_MMXEXT 0x20000000
uint32_t mm_accel (void);
#if defined(LIBDTS_FIXED)
typedef int32_t sample_t;
typedef int32_t level_t;
#elif defined(LIBDTS_DOUBLE)
typedef double sample_t;
typedef double level_t;
#else
typedef float sample_t;
typedef float level_t;
#endif
typedef struct dts_state_s dts_state_t;
#define DTS_MONO 0
#define DTS_CHANNEL 1
#define DTS_STEREO 2
#define DTS_STEREO_SUMDIFF 3
#define DTS_STEREO_TOTAL 4
#define DTS_3F 5
#define DTS_2F1R 6
#define DTS_3F1R 7
#define DTS_2F2R 8
#define DTS_3F2R 9
#define DTS_4F2R 10
#define DTS_DOLBY 101 /* FIXME */
#define DTS_CHANNEL_MAX DTS_3F2R /* We don't handle anything above that */
#define DTS_CHANNEL_BITS 6
#define DTS_CHANNEL_MASK 0x3F
#define DTS_LFE 0x80
#define DTS_ADJUST_LEVEL 0x100
dts_state_t * dts_init (uint32_t mm_accel);
int dts_syncinfo (dts_state_t *state, uint8_t * buf, int * flags,
int * sample_rate, int * bit_rate, int *frame_length);
int dts_frame (dts_state_t * state, uint8_t * buf, int * flags,
level_t * level, sample_t bias);
void dts_dynrng (dts_state_t * state,
level_t (* call) (level_t, void *), void * data);
int dts_blocks_num (dts_state_t * state);
int dts_block (dts_state_t * state);
sample_t * dts_samples (dts_state_t * state);
void dts_free (dts_state_t * state);
#endif /* DTS_H */
--- NEW FILE: bitstream.c ---
/*
* bitstream.c
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
* Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
* Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "config.h"
#include <inttypes.h>
#include "dts.h"
#include "dts_internal.h"
#include "bitstream.h"
#define BUFFER_SIZE 4096
void dts_bitstream_init (dts_state_t * state, uint8_t * buf, int word_mode,
int bigendian_mode)
{
intptr_t align;
align = (uintptr_t)buf & 3;
state->buffer_start = (uint32_t *) (buf - align);
state->bits_left = 0;
state->current_word = 0;
state->word_mode = word_mode;
state->bigendian_mode = bigendian_mode;
bitstream_get (state, align * 8);
}
#include<stdio.h>
static inline void bitstream_fill_current (dts_state_t * state)
{
uint32_t tmp;
tmp = *(state->buffer_start++);
if (state->bigendian_mode)
state->current_word = swab32 (tmp);
else
state->current_word = swable32 (tmp);
if (!state->word_mode)
{
state->current_word = (state->current_word & 0x00003FFF) |
((state->current_word & 0x3FFF0000 ) >> 2);
}
}
/*
* The fast paths for _get is in the
* bitstream.h header file so it can be inlined.
*
* The "bottom half" of this routine is suffixed _bh
*
* -ah
*/
uint32_t dts_bitstream_get_bh (dts_state_t * state, uint32_t num_bits)
{
uint32_t result;
num_bits -= state->bits_left;
result = ((state->current_word << (32 - state->bits_left)) >>
(32 - state->bits_left));
if ( !state->word_mode && num_bits > 28 ) {
bitstream_fill_current (state);
result = (result << 28) | state->current_word;
num_bits -= 28;
}
bitstream_fill_current (state);
if ( state->word_mode )
{
if (num_bits != 0)
result = (result << num_bits) |
(state->current_word >> (32 - num_bits));
state->bits_left = 32 - num_bits;
}
else
{
if (num_bits != 0)
result = (result << num_bits) |
(state->current_word >> (28 - num_bits));
state->bits_left = 28 - num_bits;
}
return result;
}
--- NEW FILE: downmix.c ---
/*
* downmix.c
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
* Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
* Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "config.h"
#include <string.h>
#include <inttypes.h>
#include "dts.h"
#include "dts_internal.h"
#define CONVERT(acmod,output) (((output) << DTS_CHANNEL_BITS) + (acmod))
int dts_downmix_init (int input, int flags, level_t * level,
level_t clev, level_t slev)
{
static uint8_t table[11][10] = {
/* DTS_MONO */
{DTS_MONO, DTS_MONO, DTS_MONO, DTS_MONO,
DTS_MONO, DTS_MONO, DTS_MONO, DTS_MONO,
DTS_MONO, DTS_MONO},
/* DTS_CHANNEL */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO},
/* DTS_STEREO */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO},
/* DTS_STEREO_SUMDIFF */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO},
/* DTS_STEREO_TOTAL */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO},
/* DTS_3F */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_3F, DTS_3F, DTS_3F,
DTS_3F, DTS_3F},
/* DTS_2F1R */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_2F1R, DTS_2F1R, DTS_2F1R,
DTS_2F1R, DTS_2F1R},
/* DTS_3F1R */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_3F, DTS_3F1R, DTS_3F1R,
DTS_3F1R, DTS_3F1R},
/* DTS_2F2R */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_STEREO, DTS_2F2R, DTS_2F2R,
DTS_2F2R, DTS_2F2R},
/* DTS_3F2R */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_3F, DTS_3F2R, DTS_3F2R,
DTS_3F2R, DTS_3F2R},
/* DTS_4F2R */
{DTS_MONO, DTS_CHANNEL, DTS_STEREO, DTS_STEREO,
DTS_STEREO, DTS_4F2R, DTS_4F2R, DTS_4F2R,
DTS_4F2R, DTS_4F2R},
};
int output;
output = flags & DTS_CHANNEL_MASK;
if (output > DTS_CHANNEL_MAX)
return -1;
output = table[output][input];
if (output == DTS_STEREO &&
(input == DTS_DOLBY || (input == DTS_3F && clev == LEVEL (LEVEL_3DB))))
output = DTS_DOLBY;
if (flags & DTS_ADJUST_LEVEL) {
level_t adjust;
switch (CONVERT (input & 7, output)) {
case CONVERT (DTS_3F, DTS_MONO):
adjust = DIV (LEVEL_3DB, LEVEL (1) + clev);
break;
case CONVERT (DTS_STEREO, DTS_MONO):
case CONVERT (DTS_2F2R, DTS_2F1R):
case CONVERT (DTS_3F2R, DTS_3F1R):
level_3db:
adjust = LEVEL (LEVEL_3DB);
break;
case CONVERT (DTS_3F2R, DTS_2F1R):
if (clev < LEVEL (LEVEL_PLUS3DB - 1))
goto level_3db;
/* break thru */
case CONVERT (DTS_3F, DTS_STEREO):
case CONVERT (DTS_3F1R, DTS_2F1R):
case CONVERT (DTS_3F1R, DTS_2F2R):
case CONVERT (DTS_3F2R, DTS_2F2R):
adjust = DIV (1, LEVEL (1) + clev);
break;
case CONVERT (DTS_2F1R, DTS_MONO):
adjust = DIV (LEVEL_PLUS3DB, LEVEL (2) + slev);
break;
case CONVERT (DTS_2F1R, DTS_STEREO):
case CONVERT (DTS_3F1R, DTS_3F):
adjust = DIV (1, LEVEL (1) + MUL_C (slev, LEVEL_3DB));
break;
case CONVERT (DTS_3F1R, DTS_MONO):
adjust = DIV (LEVEL_3DB, LEVEL (1) + clev + MUL_C (slev, 0.5));
break;
case CONVERT (DTS_3F1R, DTS_STEREO):
adjust = DIV (1, LEVEL (1) + clev + MUL_C (slev, LEVEL_3DB));
break;
case CONVERT (DTS_2F2R, DTS_MONO):
adjust = DIV (LEVEL_3DB, LEVEL (1) + slev);
break;
case CONVERT (DTS_2F2R, DTS_STEREO):
case CONVERT (DTS_3F2R, DTS_3F):
adjust = DIV (1, LEVEL (1) + slev);
break;
case CONVERT (DTS_3F2R, DTS_MONO):
adjust = DIV (LEVEL_3DB, LEVEL (1) + clev + slev);
break;
case CONVERT (DTS_3F2R, DTS_STEREO):
adjust = DIV (1, LEVEL (1) + clev + slev);
break;
case CONVERT (DTS_MONO, DTS_DOLBY):
adjust = LEVEL (LEVEL_PLUS3DB);
break;
case CONVERT (DTS_3F, DTS_DOLBY):
case CONVERT (DTS_2F1R, DTS_DOLBY):
adjust = LEVEL (1 / (1 + LEVEL_3DB));
break;
case CONVERT (DTS_3F1R, DTS_DOLBY):
case CONVERT (DTS_2F2R, DTS_DOLBY):
adjust = LEVEL (1 / (1 + 2 * LEVEL_3DB));
break;
case CONVERT (DTS_3F2R, DTS_DOLBY):
adjust = LEVEL (1 / (1 + 3 * LEVEL_3DB));
break;
default:
return output;
}
*level = MUL_L (*level, adjust);
}
return output;
}
int dts_downmix_coeff (level_t * coeff, int acmod, int output, level_t level,
level_t clev, level_t slev)
{
level_t level_3db;
level_3db = MUL_C (level, LEVEL_3DB);
switch (CONVERT (acmod, output & DTS_CHANNEL_MASK)) {
case CONVERT (DTS_CHANNEL, DTS_CHANNEL):
case CONVERT (DTS_MONO, DTS_MONO):
case CONVERT (DTS_STEREO, DTS_STEREO):
case CONVERT (DTS_3F, DTS_3F):
case CONVERT (DTS_2F1R, DTS_2F1R):
case CONVERT (DTS_3F1R, DTS_3F1R):
case CONVERT (DTS_2F2R, DTS_2F2R):
case CONVERT (DTS_3F2R, DTS_3F2R):
case CONVERT (DTS_STEREO, DTS_DOLBY):
coeff[0] = coeff[1] = coeff[2] = coeff[3] = coeff[4] = level;
return 0;
case CONVERT (DTS_CHANNEL, DTS_MONO):
coeff[0] = coeff[1] = MUL_C (level, LEVEL_6DB);
return 3;
case CONVERT (DTS_STEREO, DTS_MONO):
coeff[0] = coeff[1] = level_3db;
return 3;
case CONVERT (DTS_3F, DTS_MONO):
coeff[0] = coeff[2] = level_3db;
coeff[1] = MUL_C (MUL_L (level_3db, clev), LEVEL_PLUS6DB);
return 7;
case CONVERT (DTS_2F1R, DTS_MONO):
coeff[0] = coeff[1] = level_3db;
coeff[2] = MUL_L (level_3db, slev);
return 7;
case CONVERT (DTS_2F2R, DTS_MONO):
coeff[0] = coeff[1] = level_3db;
coeff[2] = coeff[3] = MUL_L (level_3db, slev);
return 15;
case CONVERT (DTS_3F1R, DTS_MONO):
coeff[0] = coeff[2] = level_3db;
coeff[1] = MUL_C (MUL_L (level_3db, clev), LEVEL_PLUS6DB);
coeff[3] = MUL_L (level_3db, slev);
return 15;
case CONVERT (DTS_3F2R, DTS_MONO):
coeff[0] = coeff[2] = level_3db;
coeff[1] = MUL_C (MUL_L (level_3db, clev), LEVEL_PLUS6DB);
coeff[3] = coeff[4] = MUL_L (level_3db, slev);
return 31;
case CONVERT (DTS_MONO, DTS_DOLBY):
coeff[0] = level_3db;
return 0;
case CONVERT (DTS_3F, DTS_DOLBY):
coeff[0] = coeff[2] = coeff[3] = coeff[4] = level;
coeff[1] = level_3db;
return 7;
case CONVERT (DTS_3F, DTS_STEREO):
case CONVERT (DTS_3F1R, DTS_2F1R):
case CONVERT (DTS_3F2R, DTS_2F2R):
coeff[0] = coeff[2] = coeff[3] = coeff[4] = level;
coeff[1] = MUL_L (level, clev);
return 7;
case CONVERT (DTS_2F1R, DTS_DOLBY):
coeff[0] = coeff[1] = level;
coeff[2] = level_3db;
return 7;
case CONVERT (DTS_2F1R, DTS_STEREO):
coeff[0] = coeff[1] = level;
coeff[2] = MUL_L (level_3db, slev);
return 7;
case CONVERT (DTS_3F1R, DTS_DOLBY):
coeff[0] = coeff[2] = level;
coeff[1] = coeff[3] = level_3db;
return 15;
case CONVERT (DTS_3F1R, DTS_STEREO):
coeff[0] = coeff[2] = level;
coeff[1] = MUL_L (level, clev);
coeff[3] = MUL_L (level_3db, slev);
return 15;
case CONVERT (DTS_2F2R, DTS_DOLBY):
coeff[0] = coeff[1] = level;
coeff[2] = coeff[3] = level_3db;
return 15;
case CONVERT (DTS_2F2R, DTS_STEREO):
coeff[0] = coeff[1] = level;
coeff[2] = coeff[3] = MUL_L (level, slev);
return 15;
case CONVERT (DTS_3F2R, DTS_DOLBY):
coeff[0] = coeff[2] = level;
coeff[1] = coeff[3] = coeff[4] = level_3db;
return 31;
case CONVERT (DTS_3F2R, DTS_2F1R):
coeff[0] = coeff[2] = level;
coeff[1] = MUL_L (level, clev);
coeff[3] = coeff[4] = level_3db;
return 31;
case CONVERT (DTS_3F2R, DTS_STEREO):
coeff[0] = coeff[2] = level;
coeff[1] = MUL_L (level, clev);
coeff[3] = coeff[4] = MUL_L (level, slev);
return 31;
case CONVERT (DTS_3F1R, DTS_3F):
coeff[0] = coeff[1] = coeff[2] = level;
coeff[3] = MUL_L (level_3db, slev);
return 13;
case CONVERT (DTS_3F2R, DTS_3F):
coeff[0] = coeff[1] = coeff[2] = level;
coeff[3] = coeff[4] = MUL_L (level, slev);
return 29;
case CONVERT (DTS_2F2R, DTS_2F1R):
coeff[0] = coeff[1] = level;
coeff[2] = coeff[3] = level_3db;
return 12;
case CONVERT (DTS_3F2R, DTS_3F1R):
coeff[0] = coeff[1] = coeff[2] = level;
coeff[3] = coeff[4] = level_3db;
return 24;
case CONVERT (DTS_2F1R, DTS_2F2R):
coeff[0] = coeff[1] = level;
coeff[2] = level_3db;
return 0;
case CONVERT (DTS_3F1R, DTS_2F2R):
coeff[0] = coeff[2] = level;
coeff[1] = MUL_L (level, clev);
coeff[3] = level_3db;
return 7;
case CONVERT (DTS_3F1R, DTS_3F2R):
coeff[0] = coeff[1] = coeff[2] = level;
coeff[3] = level_3db;
return 0;
}
return -1; /* NOTREACHED */
}
static void mix2to1 (sample_t * dest, sample_t * src, sample_t bias)
{
int i;
for (i = 0; i < 256; i++)
dest[i] += BIAS (src[i]);
}
static void mix3to1 (sample_t * samples, sample_t bias)
{
int i;
for (i = 0; i < 256; i++)
samples[i] += BIAS (samples[i + 256] + samples[i + 512]);
}
static void mix4to1 (sample_t * samples, sample_t bias)
{
int i;
for (i = 0; i < 256; i++)
samples[i] += BIAS (samples[i + 256] + samples[i + 512] +
samples[i + 768]);
}
static void mix5to1 (sample_t * samples, sample_t bias)
{
int i;
for (i = 0; i < 256; i++)
samples[i] += BIAS (samples[i + 256] + samples[i + 512] +
samples[i + 768] + samples[i + 1024]);
}
static void mix3to2 (sample_t * samples, sample_t bias)
{
int i;
sample_t common;
for (i = 0; i < 256; i++) {
common = BIAS (samples[i]);
samples[i] = samples[i + 256] + common;
samples[i + 256] = samples[i + 512] + common;
}
}
static void mix21to2 (sample_t * left, sample_t * right, sample_t bias)
{
int i;
sample_t common;
for (i = 0; i < 256; i++) {
common = BIAS (right[i + 256]);
left[i] += common;
right[i] += common;
}
}
static void mix21toS (sample_t * samples, sample_t bias)
{
int i;
sample_t surround;
for (i = 0; i < 256; i++) {
surround = samples[i + 512];
samples[i] += BIAS (-surround);
samples[i + 256] += BIAS (surround);
}
}
static void mix31to2 (sample_t * samples, sample_t bias)
{
int i;
sample_t common;
for (i = 0; i < 256; i++) {
common = BIAS (samples[i] + samples[i + 768]);
samples[i] = samples[i + 256] + common;
samples[i + 256] = samples[i + 512] + common;
}
}
static void mix31toS (sample_t * samples, sample_t bias)
{
int i;
sample_t common, surround;
for (i = 0; i < 256; i++) {
common = BIAS (samples[i]);
surround = samples[i + 768];
samples[i] = samples[i + 256] + common - surround;
samples[i + 256] = samples[i + 512] + common + surround;
}
}
static void mix22toS (sample_t * samples, sample_t bias)
{
int i;
sample_t surround;
for (i = 0; i < 256; i++) {
surround = samples[i + 512] + samples[i + 768];
samples[i] += BIAS (-surround);
samples[i + 256] += BIAS (surround);
}
}
static void mix32to2 (sample_t * samples, sample_t bias)
{
int i;
sample_t common;
for (i = 0; i < 256; i++) {
common = BIAS (samples[i]);
samples[i] = common + samples[i + 256] + samples[i + 768];
samples[i + 256] = common + samples[i + 512] + samples[i + 1024];
}
}
static void mix32toS (sample_t * samples, sample_t bias)
{
int i;
sample_t common, surround;
for (i = 0; i < 256; i++) {
common = BIAS (samples[i]);
surround = samples[i + 768] + samples[i + 1024];
samples[i] = samples[i + 256] + common - surround;
samples[i + 256] = samples[i + 512] + common + surround;
}
}
static void move2to1 (sample_t * src, sample_t * dest, sample_t bias)
{
int i;
for (i = 0; i < 256; i++)
dest[i] = BIAS (src[i] + src[i + 256]);
}
static void zero (sample_t * samples)
{
int i;
for (i = 0; i < 256; i++)
samples[i] = 0;
}
void dts_downmix (sample_t * samples, int acmod, int output, sample_t bias,
level_t clev, level_t slev)
{
switch (CONVERT (acmod, output & DTS_CHANNEL_MASK)) {
case CONVERT (DTS_CHANNEL, DTS_MONO):
case CONVERT (DTS_STEREO, DTS_MONO):
mix_2to1:
mix2to1 (samples, samples + 256, bias);
break;
case CONVERT (DTS_2F1R, DTS_MONO):
if (slev == 0)
goto mix_2to1;
case CONVERT (DTS_3F, DTS_MONO):
mix_3to1:
mix3to1 (samples, bias);
break;
case CONVERT (DTS_3F1R, DTS_MONO):
if (slev == 0)
goto mix_3to1;
case CONVERT (DTS_2F2R, DTS_MONO):
if (slev == 0)
goto mix_2to1;
mix4to1 (samples, bias);
break;
case CONVERT (DTS_3F2R, DTS_MONO):
if (slev == 0)
goto mix_3to1;
mix5to1 (samples, bias);
break;
case CONVERT (DTS_MONO, DTS_DOLBY):
memcpy (samples + 256, samples, 256 * sizeof (sample_t));
break;
case CONVERT (DTS_3F, DTS_STEREO):
case CONVERT (DTS_3F, DTS_DOLBY):
mix_3to2:
mix3to2 (samples, bias);
break;
case CONVERT (DTS_2F1R, DTS_STEREO):
if (slev == 0)
break;
mix21to2 (samples, samples + 256, bias);
break;
case CONVERT (DTS_2F1R, DTS_DOLBY):
mix21toS (samples, bias);
break;
case CONVERT (DTS_3F1R, DTS_STEREO):
if (slev == 0)
goto mix_3to2;
mix31to2 (samples, bias);
break;
case CONVERT (DTS_3F1R, DTS_DOLBY):
mix31toS (samples, bias);
break;
case CONVERT (DTS_2F2R, DTS_STEREO):
if (slev == 0)
break;
mix2to1 (samples, samples + 512, bias);
mix2to1 (samples + 256, samples + 768, bias);
break;
case CONVERT (DTS_2F2R, DTS_DOLBY):
mix22toS (samples, bias);
break;
case CONVERT (DTS_3F2R, DTS_STEREO):
if (slev == 0)
goto mix_3to2;
mix32to2 (samples, bias);
break;
case CONVERT (DTS_3F2R, DTS_DOLBY):
mix32toS (samples, bias);
break;
case CONVERT (DTS_3F1R, DTS_3F):
if (slev == 0)
break;
mix21to2 (samples, samples + 512, bias);
break;
case CONVERT (DTS_3F2R, DTS_3F):
if (slev == 0)
break;
mix2to1 (samples, samples + 768, bias);
mix2to1 (samples + 512, samples + 1024, bias);
break;
case CONVERT (DTS_3F1R, DTS_2F1R):
mix3to2 (samples, bias);
memcpy (samples + 512, samples + 768, 256 * sizeof (sample_t));
break;
case CONVERT (DTS_2F2R, DTS_2F1R):
mix2to1 (samples + 512, samples + 768, bias);
break;
case CONVERT (DTS_3F2R, DTS_2F1R):
mix3to2 (samples, bias);
move2to1 (samples + 768, samples + 512, bias);
break;
case CONVERT (DTS_3F2R, DTS_3F1R):
mix2to1 (samples + 768, samples + 1024, bias);
break;
case CONVERT (DTS_2F1R, DTS_2F2R):
memcpy (samples + 768, samples + 512, 256 * sizeof (sample_t));
break;
case CONVERT (DTS_3F1R, DTS_2F2R):
mix3to2 (samples, bias);
memcpy (samples + 512, samples + 768, 256 * sizeof (sample_t));
break;
case CONVERT (DTS_3F2R, DTS_2F2R):
mix3to2 (samples, bias);
memcpy (samples + 512, samples + 768, 256 * sizeof (sample_t));
memcpy (samples + 768, samples + 1024, 256 * sizeof (sample_t));
break;
case CONVERT (DTS_3F1R, DTS_3F2R):
memcpy (samples + 1024, samples + 768, 256 * sizeof (sample_t));
break;
}
}
void dts_upmix (sample_t * samples, int acmod, int output)
{
switch (CONVERT (acmod, output & DTS_CHANNEL_MASK)) {
case CONVERT (DTS_3F2R, DTS_MONO):
zero (samples + 1024);
case CONVERT (DTS_3F1R, DTS_MONO):
case CONVERT (DTS_2F2R, DTS_MONO):
zero (samples + 768);
case CONVERT (DTS_3F, DTS_MONO):
case CONVERT (DTS_2F1R, DTS_MONO):
zero (samples + 512);
case CONVERT (DTS_CHANNEL, DTS_MONO):
case CONVERT (DTS_STEREO, DTS_MONO):
zero (samples + 256);
break;
case CONVERT (DTS_3F2R, DTS_STEREO):
case CONVERT (DTS_3F2R, DTS_DOLBY):
zero (samples + 1024);
case CONVERT (DTS_3F1R, DTS_STEREO):
case CONVERT (DTS_3F1R, DTS_DOLBY):
zero (samples + 768);
case CONVERT (DTS_3F, DTS_STEREO):
case CONVERT (DTS_3F, DTS_DOLBY):
mix_3to2:
memcpy (samples + 512, samples + 256, 256 * sizeof (sample_t));
zero (samples + 256);
break;
case CONVERT (DTS_2F2R, DTS_STEREO):
case CONVERT (DTS_2F2R, DTS_DOLBY):
zero (samples + 768);
case CONVERT (DTS_2F1R, DTS_STEREO):
case CONVERT (DTS_2F1R, DTS_DOLBY):
zero (samples + 512);
break;
case CONVERT (DTS_3F2R, DTS_3F):
zero (samples + 1024);
case CONVERT (DTS_3F1R, DTS_3F):
case CONVERT (DTS_2F2R, DTS_2F1R):
zero (samples + 768);
break;
case CONVERT (DTS_3F2R, DTS_3F1R):
zero (samples + 1024);
break;
case CONVERT (DTS_3F2R, DTS_2F1R):
zero (samples + 1024);
case CONVERT (DTS_3F1R, DTS_2F1R):
mix_31to21:
memcpy (samples + 768, samples + 512, 256 * sizeof (sample_t));
goto mix_3to2;
case CONVERT (DTS_3F2R, DTS_2F2R):
memcpy (samples + 1024, samples + 768, 256 * sizeof (sample_t));
goto mix_31to21;
}
}
--- NEW FILE: Makefile.am ---
include $(top_srcdir)/misc/Makefile.common
libdir = $(XINE_PLUGINDIR)
lib_LTLIBRARIES = xineplug_decode_dts.la
xineplug_decode_dts_la_SOURCES = bitstream.c downmix.c parse.c xine_decoder.c
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noinst_HEADERS = bitstream.h dts.h dts_internal.h tables.h tables_adpcm.h \
tables_fir.h tables_huffman.h tables_quantization.h tables_vq.h
--- NEW FILE: Makefile.in ---
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# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
.NOEXPORT:
--- NEW FILE: xine_decoder.c ---
/*
* Copyright (C) 2000-2004 the xine project
*
* This file is part of xine, a unix video player.
*
* xine is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* xine is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA
*
* $Id: xine_decoder.c,v 1.1 2005/04/04 22:29:47 dsalt-guest Exp $
*
* 04-09-2001 DTS passtrough (C) Joachim Koenig
* 09-12-2001 DTS passthrough inprovements (C) James Courtier-Dutton
*
*/
#ifndef __sun
/* required for swab() */
#define _XOPEN_SOURCE 500
#endif
/* avoid compiler warnings */
#define _BSD_SOURCE 1
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <assert.h>
#define LOG_MODULE "libdts"
#define LOG_VERBOSE
/*
#define LOG
*/
#include "xine_internal.h"
#include "xineutils.h"
#include "audio_out.h"
#include "buffer.h"
#include "dts.h"
#define MAX_AC5_FRAME 4096
typedef struct {
audio_decoder_class_t decoder_class;
} dts_class_t;
typedef struct {
audio_decoder_t audio_decoder;
xine_stream_t *stream;
audio_decoder_class_t *class;
dts_state_t *dts_state;
int64_t pts;
int audio_caps;
uint32_t rate;
uint32_t bits_per_sample;
uint32_t number_of_channels;
int sync_state;
int ac5_length, ac5_pcm_length, frame_todo;
uint32_t syncdword;
uint8_t frame_buffer[MAX_AC5_FRAME];
uint8_t *frame_ptr;
int output_open;
int bypass_mode;
int dts_flags;
int dts_sample_rate;
int dts_bit_rate;
int dts_flags_map[11]; /* Convert from stream dts_flags to the dts_flags we want from the dts downmixer */
int ao_flags_map[11]; /* Convert from the xine AO_CAP's to dts_flags. */
int have_lfe;
} dts_decoder_t;
static void dts_reset (audio_decoder_t *this_gen);
static void dts_discontinuity (audio_decoder_t *this_gen);
static void dts_reset (audio_decoder_t *this_gen) {
/* dts_decoder_t *this = (dts_decoder_t *) this_gen; */
}
static void dts_discontinuity (audio_decoder_t *this_gen) {
}
#if 0
static inline int16_t blah (int32_t i) {
if (i > 0x43c07fff)
return 32767;
else if (i < 0x43bf8000)
return -32768;
else
return i - 0x43c00000;
}
static inline void float_to_int (float * _f, int16_t * s16, int num_channels) {
int i;
int32_t * f = (int32_t *) _f; /* XXX assumes IEEE float format */
for (i = 0; i < 256; i++) {
s16[num_channels*i] = blah (f[i]);
}
}
#endif
static inline void float_to_int (float * _f, int16_t * s16, int num_channels) {
int i;
float f;
for (i = 0; i < 256; i++) {
f = _f[i] * 32767;
if (f > INT16_MAX) f = INT16_MAX;
if (f < INT16_MIN) f = INT16_MIN;
s16[num_channels*i] = f;
/* printf("samples[%d] = %f, %d\n", i, _f[i], s16[num_channels*i]); */
}
}
static inline void mute_channel (int16_t * s16, int num_channels) {
int i;
for (i = 0; i < 256; i++) {
s16[num_channels*i] = 0;
}
}
static void dts_decode_frame (dts_decoder_t *this, int64_t pts, int preview_mode) {
audio_buffer_t *audio_buffer;
uint32_t ac5_spdif_type=0;
int output_mode = AO_CAP_MODE_STEREO;
uint8_t *data_out;
uint8_t *data_in = this->frame_buffer;
lprintf("decode_frame\n");
audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out);
audio_buffer->vpts = pts;
if(this->bypass_mode) {
/* SPDIF digital output */
if (!this->output_open) {
this->output_open = (this->stream->audio_out->open (this->stream->audio_out, this->stream,
this->bits_per_sample,
this->rate,
AO_CAP_MODE_AC5));
}
if (!this->output_open)
return;
data_out=(uint8_t *) audio_buffer->mem;
if (this->ac5_length > 8191) {
xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG, "libdts: ac5_length too long\n");
this->ac5_pcm_length = 0;
}
switch (this->ac5_pcm_length) {
case 512:
ac5_spdif_type = 0x0b; /* DTS-1 (512-sample bursts) */
break;
case 1024:
ac5_spdif_type = 0x0c; /* DTS-1 (1024-sample bursts) */
break;
case 2048:
ac5_spdif_type = 0x0d; /* DTS-1 (2048-sample bursts) */
break;
default:
xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG,
"libdts: DTS %i-sample bursts not supported\n", this->ac5_pcm_length);
return;
}
#ifdef LOG_DEBUG
{
int i;
printf("libdts: DTS frame type=%d\n",data_in[4] >> 7);
printf("libdts: DTS deficit frame count=%d\n",(data_in[4] & 0x7f) >> 2);
printf("libdts: DTS AC5 PCM samples=%d\n",ac5_pcm_samples);
printf("libdts: DTS AC5 length=%d\n",this->ac5_length);
printf("libdts: DTS AC5 bitrate=%d\n",((data_in[8] & 0x03) << 4) | (data_in[8] >> 4));
printf("libdts: DTS AC5 spdif type=%d\n", ac5_spdif_type);
printf("libdts: ");
for(i=2000;i<2048;i++) {
printf("%02x ",data_in[i]);
}
printf("\n");
}
#endif
lprintf("length=%d pts=%lld\n",this->ac5_pcm_length,audio_buffer->vpts);
audio_buffer->num_frames = this->ac5_pcm_length;
data_out[0] = 0x72; data_out[1] = 0xf8; /* spdif syncword */
data_out[2] = 0x1f; data_out[3] = 0x4e; /* .............. */
data_out[4] = ac5_spdif_type; /* DTS data */
data_out[5] = 0; /* Unknown */
data_out[6] = (this->ac5_length << 3) & 0xff; /* ac5_length * 8 */
data_out[7] = ((this->ac5_length ) >> 5) & 0xff;
if( this->ac5_pcm_length ) {
if( this->ac5_pcm_length % 2) {
swab(data_in, &data_out[8], this->ac5_length );
} else {
swab(data_in, &data_out[8], this->ac5_length + 1);
}
}
} else {
/* Software decode */
int i, dts_flags;
int16_t *int_samples = audio_buffer->mem;
int number_of_dts_blocks;
level_t level = 1.0;
sample_t *samples;
this->have_lfe = this->dts_flags & DTS_LFE;
dts_flags = this->dts_flags_map[this->dts_flags & DTS_CHANNEL_MASK];
if (this->have_lfe)
if (this->audio_caps & AO_CAP_MODE_5_1CHANNEL) {
output_mode = AO_CAP_MODE_5_1CHANNEL;
dts_flags |= DTS_LFE;
} else if (this->audio_caps & AO_CAP_MODE_4_1CHANNEL) {
output_mode = AO_CAP_MODE_4_1CHANNEL;
dts_flags |= DTS_LFE;
} else {
xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG, "libdts: WHAT DO I DO!!!\n");
output_mode = this->ao_flags_map[dts_flags & DTS_CHANNEL_MASK];
}
else
output_mode = this->ao_flags_map[dts_flags & DTS_CHANNEL_MASK];
if(dts_frame(this->dts_state, data_in, &dts_flags, &level, 0)) {
xprintf (this->stream->xine, XINE_VERBOSITY_DEBUG, "libdts: dts_frame error\n");
return;
}
if (!this->output_open) {
this->output_open = (this->stream->audio_out->open (this->stream->audio_out, this->stream,
this->bits_per_sample,
this->rate,
output_mode));
}
if (!this->output_open)
return;
number_of_dts_blocks = dts_blocks_num (this->dts_state);
audio_buffer->num_frames = 256*number_of_dts_blocks;
for(i = 0; i < number_of_dts_blocks; i++) {
if(dts_block(this->dts_state)) {
xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG,
"libdts: dts_block error on audio channel %d\n", i);
audio_buffer->num_frames = 0;
break;
}
samples = dts_samples(this->dts_state);
switch (output_mode) {
case AO_CAP_MODE_MONO:
float_to_int (&samples[0], int_samples+(i*256), 1);
break;
case AO_CAP_MODE_STEREO:
/* Tested, working. */
float_to_int (&samples[0*256], int_samples+(i*256*2), 2); /* L */
float_to_int (&samples[1*256], int_samples+(i*256*2)+1, 2); /* R */
break;
case AO_CAP_MODE_4CHANNEL:
/* Tested, working */
float_to_int (&samples[0*256], int_samples+(i*256*4), 4); /* L */
float_to_int (&samples[1*256], int_samples+(i*256*4)+1, 4); /* R */
float_to_int (&samples[2*256], int_samples+(i*256*4)+2, 4); /* RL */
float_to_int (&samples[3*256], int_samples+(i*256*4)+3, 4); /* RR */
break;
case AO_CAP_MODE_4_1CHANNEL:
/* Tested, working */
float_to_int (&samples[0*256], int_samples+(i*256*6)+0, 6); /* L */
float_to_int (&samples[1*256], int_samples+(i*256*6)+1, 6); /* R */
float_to_int (&samples[2*256], int_samples+(i*256*6)+2, 6); /* RL */
float_to_int (&samples[3*256], int_samples+(i*256*6)+3, 6); /* RR */
float_to_int (&samples[4*256], int_samples+(i*256*6)+5, 6); /* LFE */
mute_channel ( int_samples+(i*256*6)+4, 6); /* C */
break;
case AO_CAP_MODE_5CHANNEL:
/* Tested, working */
float_to_int (&samples[0*256], int_samples+(i*256*6)+4, 6); /* C */
float_to_int (&samples[1*256], int_samples+(i*256*6)+0, 6); /* L */
float_to_int (&samples[2*256], int_samples+(i*256*6)+1, 6); /* R */
float_to_int (&samples[3*256], int_samples+(i*256*6)+2, 6); /* RL */
float_to_int (&samples[4*256], int_samples+(i*256*6)+3, 6); /* RR */
mute_channel ( int_samples+(i*256*6)+5, 6); /* LFE */
break;
case AO_CAP_MODE_5_1CHANNEL:
float_to_int (&samples[0*256], int_samples+(i*256*6)+4, 6); /* C */
float_to_int (&samples[1*256], int_samples+(i*256*6)+0, 6); /* L */
float_to_int (&samples[2*256], int_samples+(i*256*6)+1, 6); /* R */
float_to_int (&samples[3*256], int_samples+(i*256*6)+2, 6); /* RL */
float_to_int (&samples[4*256], int_samples+(i*256*6)+3, 6); /* RR */
float_to_int (&samples[5*256], int_samples+(i*256*6)+5, 6); /* LFE */ /* Not working yet */
break;
default:
xprintf (this->stream->xine, XINE_VERBOSITY_DEBUG, "libdts: help - unsupported mode %08x\n", output_mode);
}
}
}
this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream);
}
static void dts_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) {
dts_decoder_t *this = (dts_decoder_t *) this_gen;
uint8_t *current = (uint8_t *)buf->content;
uint8_t *sync_start=current + 1;
uint8_t *end = buf->content + buf->size;
lprintf("decode_data\n");
if (buf->decoder_flags & BUF_FLAG_PREVIEW)
return;
if (buf->decoder_flags & BUF_FLAG_STDHEADER)
return;
lprintf ("processing...state %d\n", this->sync_state);
while (current < end) {
switch (this->sync_state) {
case 0: /* Looking for sync header */
this->syncdword = (this->syncdword << 8) | *current++;
/*
if ((this->syncdword == 0xff1f00e8) ||
(this->syncdword == 0x1fffe800) ||
(this->syncdword == 0xfe7f0180) ||
(this->syncdword == 0x7ffe8001) ) {
*/
if ((this->syncdword == 0x7ffe8001)) {
lprintf ("sync found: syncdword=0x%x\n", this->syncdword);
this->frame_buffer[0] = 0x7f;
this->frame_buffer[1] = 0xfe;
this->frame_buffer[2] = 0x80;
this->frame_buffer[3] = 0x01;
this->sync_state = 1;
this->frame_ptr = this->frame_buffer+4;
this->pts = buf->pts;
break;
}
if ((this->syncdword == 0xff1f00e8)) {
lprintf ("sync found: syncdword=0x%x\n", this->syncdword);
this->frame_buffer[0] = 0xff;
this->frame_buffer[1] = 0x1f;
this->frame_buffer[2] = 0x00;
this->frame_buffer[3] = 0xe8;
this->sync_state = 1;
this->frame_ptr = this->frame_buffer+4;
this->pts = buf->pts;
break;
}
break;
case 1: /* Looking for enough bytes for sync_info. */
sync_start = current - 1;
*this->frame_ptr++ = *current++;
if ((this->frame_ptr - this->frame_buffer) > 19) {
int old_dts_flags = this->dts_flags;
int old_dts_sample_rate = this->dts_sample_rate;
int old_dts_bit_rate = this->dts_bit_rate;
this->ac5_length = dts_syncinfo (this->dts_state, this->frame_buffer,
&this->dts_flags,
&this->dts_sample_rate,
&this->dts_bit_rate, &(this->ac5_pcm_length));
lprintf("ac5_length=%d\n",this->ac5_length);
lprintf("dts_sample_rate=%d\n",this->dts_sample_rate);
if ( (this->ac5_length < 80) || (this->ac5_length > MAX_AC5_FRAME) ) { /* Invalid dts ac5_pcm_length */
this->syncdword = 0;
current = sync_start;
this->sync_state = 0;
break;
}
lprintf("Frame length = %d\n",this->ac5_pcm_length);
this->frame_todo = this->ac5_length - 20;
this->sync_state = 2;
if (!_x_meta_info_get(this->stream, XINE_META_INFO_AUDIOCODEC) ||
old_dts_flags != this->dts_flags ||
old_dts_sample_rate != this->dts_sample_rate ||
old_dts_bit_rate != this->dts_bit_rate) {
this->rate = this->dts_sample_rate;
if (((this->dts_flags & DTS_CHANNEL_MASK) == DTS_3F2R) && (this->dts_flags & DTS_LFE))
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS 5.1");
else if ((((this->dts_flags & DTS_CHANNEL_MASK) == DTS_2F2R) && (this->dts_flags & DTS_LFE)) ||
(((this->dts_flags & DTS_CHANNEL_MASK) == DTS_3F1R) && (this->dts_flags & DTS_LFE)))
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS 4.1");
else if ((this->dts_flags & DTS_CHANNEL_MASK) == DTS_3F2R)
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS 5.0");
else if (((this->dts_flags & DTS_CHANNEL_MASK) == DTS_2F2R) ||
((this->dts_flags & DTS_CHANNEL_MASK) == DTS_3F1R))
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS 4.0");
else if (((this->dts_flags & DTS_CHANNEL_MASK) == DTS_2F1R) ||
((this->dts_flags & DTS_CHANNEL_MASK) == DTS_3F))
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS 3.0");
else if ((this->dts_flags & DTS_CHANNEL_MASK) == DTS_STEREO)
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS 2.0 (stereo)");
else if ((this->dts_flags & DTS_CHANNEL_MASK) == DTS_MONO)
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS 1.0");
else
_x_meta_info_set_utf8(this->stream, XINE_META_INFO_AUDIOCODEC, "DTS");
_x_stream_info_set(this->stream, XINE_STREAM_INFO_AUDIO_BITRATE, this->dts_bit_rate);
_x_stream_info_set(this->stream, XINE_STREAM_INFO_AUDIO_SAMPLERATE, this->dts_sample_rate);
}
}
break;
case 2: /* Filling frame_buffer with sync_info bytes */
*this->frame_ptr++ = *current++;
this->frame_todo--;
if (this->frame_todo < 1) {
this->sync_state = 3;
} else break;
case 3: /* Ready for decode */
#if 0
dtsdec_decode_frame (this, this->pts_list[0], buf->decoder_flags & BUF_FLAG_PREVIEW);
#else
dts_decode_frame (this, this->pts, buf->decoder_flags & BUF_FLAG_PREVIEW);
#endif
case 4: /* Clear up ready for next frame */
this->pts = 0;
this->syncdword = 0;
this->sync_state = 0;
break;
default: /* No come here */
break;
}
}
}
static void dts_dispose (audio_decoder_t *this_gen) {
dts_decoder_t *this = (dts_decoder_t *) this_gen;
if (this->output_open)
this->stream->audio_out->close (this->stream->audio_out, this->stream);
this->output_open = 0;
free (this);
}
static audio_decoder_t *open_plugin (audio_decoder_class_t *class_gen, xine_stream_t *stream) {
dts_decoder_t *this ;
lprintf("open_plugin\n");
this = (dts_decoder_t *) xine_xmalloc (sizeof (dts_decoder_t));
this->audio_decoder.decode_data = dts_decode_data;
this->audio_decoder.reset = dts_reset;
this->audio_decoder.discontinuity = dts_discontinuity;
this->audio_decoder.dispose = dts_dispose;
this->dts_state = dts_init(0);
this->audio_caps = stream->audio_out->get_capabilities(stream->audio_out);
if(this->audio_caps & AO_CAP_MODE_AC5)
this->bypass_mode = 1;
else {
this->bypass_mode = 0;
/* FIXME: Leave "DOLBY pro logic" downmix out for now. */
this->dts_flags_map[DTS_MONO] = DTS_MONO;
this->dts_flags_map[DTS_STEREO] = DTS_STEREO;
this->dts_flags_map[DTS_3F] = DTS_STEREO;
this->dts_flags_map[DTS_2F1R] = DTS_STEREO;
this->dts_flags_map[DTS_3F1R] = DTS_STEREO;
this->dts_flags_map[DTS_2F2R] = DTS_STEREO;
this->dts_flags_map[DTS_3F2R] = DTS_STEREO;
this->ao_flags_map[DTS_MONO] = AO_CAP_MODE_MONO;
this->ao_flags_map[DTS_STEREO] = AO_CAP_MODE_STEREO;
this->ao_flags_map[DTS_3F] = AO_CAP_MODE_STEREO;
this->ao_flags_map[DTS_2F1R] = AO_CAP_MODE_STEREO;
this->ao_flags_map[DTS_3F1R] = AO_CAP_MODE_STEREO;
this->ao_flags_map[DTS_2F2R] = AO_CAP_MODE_STEREO;
this->ao_flags_map[DTS_3F2R] = AO_CAP_MODE_STEREO;
/* find best mode */
if (this->audio_caps & AO_CAP_MODE_5_1CHANNEL) {
this->dts_flags_map[DTS_2F2R] = DTS_2F2R;
this->dts_flags_map[DTS_3F2R] = DTS_3F2R | DTS_LFE;
this->ao_flags_map[DTS_2F2R] = AO_CAP_MODE_4CHANNEL;
this->ao_flags_map[DTS_3F2R] = AO_CAP_MODE_5CHANNEL;
} else if (this->audio_caps & AO_CAP_MODE_5CHANNEL) {
this->dts_flags_map[DTS_2F2R] = DTS_2F2R;
this->dts_flags_map[DTS_3F2R] = DTS_3F2R;
this->ao_flags_map[DTS_2F2R] = AO_CAP_MODE_4CHANNEL;
this->ao_flags_map[DTS_3F2R] = AO_CAP_MODE_5CHANNEL;
} else if (this->audio_caps & AO_CAP_MODE_4_1CHANNEL) {
this->dts_flags_map[DTS_2F2R] = DTS_2F2R;
this->dts_flags_map[DTS_3F2R] = DTS_2F2R | DTS_LFE;
this->ao_flags_map[DTS_2F2R] = AO_CAP_MODE_4CHANNEL;
this->ao_flags_map[DTS_3F2R] = AO_CAP_MODE_4CHANNEL;
} else if (this->audio_caps & AO_CAP_MODE_4CHANNEL) {
this->dts_flags_map[DTS_2F2R] = DTS_2F2R;
this->dts_flags_map[DTS_3F2R] = DTS_2F2R;
this->ao_flags_map[DTS_2F2R] = AO_CAP_MODE_4CHANNEL;
this->ao_flags_map[DTS_3F2R] = AO_CAP_MODE_4CHANNEL;
/* else if (this->audio_caps & AO_CAP_MODE_STEREO)
defaults are ok */
} else if (!(this->audio_caps & AO_CAP_MODE_STEREO)) {
xprintf (this->stream->xine, XINE_VERBOSITY_LOG, _("HELP! a mono-only audio driver?!\n"));
this->dts_flags_map[DTS_MONO] = DTS_MONO;
this->dts_flags_map[DTS_STEREO] = DTS_MONO;
this->dts_flags_map[DTS_3F] = DTS_MONO;
this->dts_flags_map[DTS_2F1R] = DTS_MONO;
this->dts_flags_map[DTS_3F1R] = DTS_MONO;
this->dts_flags_map[DTS_2F2R] = DTS_MONO;
this->dts_flags_map[DTS_3F2R] = DTS_MONO;
this->ao_flags_map[DTS_MONO] = AO_CAP_MODE_MONO;
this->ao_flags_map[DTS_STEREO] = AO_CAP_MODE_MONO;
this->ao_flags_map[DTS_3F] = AO_CAP_MODE_MONO;
this->ao_flags_map[DTS_2F1R] = AO_CAP_MODE_MONO;
this->ao_flags_map[DTS_3F1R] = AO_CAP_MODE_MONO;
this->ao_flags_map[DTS_2F2R] = AO_CAP_MODE_MONO;
this->ao_flags_map[DTS_3F2R] = AO_CAP_MODE_MONO;
}
}
this->stream = stream;
this->class = class_gen;
this->output_open = 0;
this->rate = 48000;
this->bits_per_sample=16;
this->number_of_channels=2;
return &this->audio_decoder;
}
static char *get_identifier (audio_decoder_class_t *this) {
return "DTS";
}
static char *get_description (audio_decoder_class_t *this) {
return "DTS passthru audio format decoder plugin";
}
static void dispose_class (audio_decoder_class_t *this) {
lprintf("dispose_class\n");
free (this);
}
static void *init_plugin (xine_t *xine, void *data) {
dts_class_t *this ;
lprintf("init_plugin\n");
this = (dts_class_t *) xine_xmalloc (sizeof (dts_class_t));
this->decoder_class.open_plugin = open_plugin;
this->decoder_class.get_identifier = get_identifier;
this->decoder_class.get_description = get_description;
this->decoder_class.dispose = dispose_class;
return this;
}
static uint32_t audio_types[] = {
BUF_AUDIO_DTS, 0
};
static decoder_info_t dec_info_audio = {
audio_types, /* supported types */
1 /* priority */
};
plugin_info_t xine_plugin_info[] = {
/* type, API, "name", version, special_info, init_function */
{ PLUGIN_AUDIO_DECODER, 15, "dts", XINE_VERSION_CODE, &dec_info_audio, init_plugin },
{ PLUGIN_NONE, 0, "", 0, NULL, NULL }
};
--- NEW FILE: tables_huffman.h ---
/*
* huffman_tables.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* dtsdec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
[...1031 lines suppressed...]
int length = 0, j;
while( 1 )
{
length++;
value <<= 1;
value |= bitstream_get (state, 1);
for( j = 0; huff[j].length != 0 && huff[j].length < length; j++ );
if( huff[j].length == 0 ) break;
for( ; huff[j].length == length; j++ )
{
if( huff[j].code == value ) return huff[j].value;
}
}
return 0;
}