FFmpeg/libavcodec/dts_internal.h
Michael Niedermayer 23c9925329 libdts support by (Benjamin Zores <ben at geexbox dot org>)
Originally committed as revision 3310 to svn://svn.ffmpeg.org/ffmpeg/trunk
2004-07-14 01:32:14 +00:00

204 lines
8.0 KiB
C

/*
* 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