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+/*
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+ * dts_internal.h
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+ * Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
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+ * Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
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+ * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
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+ *
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+ * This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
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+ * See http://www.videolan.org/dtsdec.html for updates.
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+ *
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+ * dtsdec is free software; you can redistribute it and/or modify
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+ * it under the terms of the GNU General Public License as published by
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+ * the Free Software Foundation; either version 2 of the License, or
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+ * (at your option) any later version.
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+ *
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+ * dtsdec is distributed in the hope that it will be useful,
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+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
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+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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+ * GNU General Public License for more details.
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+ *
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+ * You should have received a copy of the GNU General Public License
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+ * along with this program; if not, write to the Free Software
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+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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+ */
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+
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+#define DTS_SUBFRAMES_MAX (16)
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+#define DTS_PRIM_CHANNELS_MAX (5)
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+#define DTS_SUBBANDS (32)
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+#define DTS_ABITS_MAX (32) /* Should be 28 */
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+#define DTS_SUBSUBFAMES_MAX (4)
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+#define DTS_LFE_MAX (3)
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+
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+struct dts_state_s {
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+
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+ /* Frame header */
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+ int frame_type; /* type of the current frame */
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+ int samples_deficit; /* deficit sample count */
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+ int crc_present; /* crc is present in the bitstream */
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+ int sample_blocks; /* number of PCM sample blocks */
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+ int frame_size; /* primary frame byte size */
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+ int amode; /* audio channels arrangement */
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+ int sample_rate; /* audio sampling rate */
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+ int bit_rate; /* transmission bit rate */
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+
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+ int downmix; /* embedded downmix enabled */
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+ int dynrange; /* embedded dynamic range flag */
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+ int timestamp; /* embedded time stamp flag */
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+ int aux_data; /* auxiliary data flag */
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+ int hdcd; /* source material is mastered in HDCD */
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+ int ext_descr; /* extension audio descriptor flag */
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+ int ext_coding; /* extended coding flag */
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+ int aspf; /* audio sync word insertion flag */
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+ int lfe; /* low frequency effects flag */
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+ int predictor_history; /* predictor history flag */
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+ int header_crc; /* header crc check bytes */
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+ int multirate_inter; /* multirate interpolator switch */
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+ int version; /* encoder software revision */
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+ int copy_history; /* copy history */
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+ int source_pcm_res; /* source pcm resolution */
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+ int front_sum; /* front sum/difference flag */
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+ int surround_sum; /* surround sum/difference flag */
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+ int dialog_norm; /* dialog normalisation parameter */
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+
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+ /* Primary audio coding header */
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+ int subframes; /* number of subframes */
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+ int prim_channels; /* number of primary audio channels */
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+ /* subband activity count */
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+ int subband_activity[DTS_PRIM_CHANNELS_MAX];
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+ /* high frequency vq start subband */
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+ int vq_start_subband[DTS_PRIM_CHANNELS_MAX];
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+ /* joint intensity coding index */
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+ int joint_intensity[DTS_PRIM_CHANNELS_MAX];
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+ /* transient mode code book */
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+ int transient_huffman[DTS_PRIM_CHANNELS_MAX];
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+ /* scale factor code book */
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+ int scalefactor_huffman[DTS_PRIM_CHANNELS_MAX];
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+ /* bit allocation quantizer select */
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+ int bitalloc_huffman[DTS_PRIM_CHANNELS_MAX];
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+ /* quantization index codebook select */
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+ int quant_index_huffman[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];
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+ /* scale factor adjustment */
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+ float scalefactor_adj[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];
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+
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+ /* Primary audio coding side information */
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+ int subsubframes; /* number of subsubframes */
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+ int partial_samples; /* partial subsubframe samples count */
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+ /* prediction mode (ADPCM used or not) */
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+ int prediction_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
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+ /* prediction VQ coefs */
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+ int prediction_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
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+ /* bit allocation index */
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+ int bitalloc[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
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+ /* transition mode (transients) */
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+ int transition_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
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+ /* scale factors (2 if transient)*/
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+ int scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][2];
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+ /* joint subband scale factors codebook */
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+ int joint_huff[DTS_PRIM_CHANNELS_MAX];
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+ /* joint subband scale factors */
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+ int joint_scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
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+ /* stereo downmix coefficients */
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+ int downmix_coef[DTS_PRIM_CHANNELS_MAX][2];
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+ /* dynamic range coefficient */
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+ int dynrange_coef;
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+
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+ /* VQ encoded high frequency subbands */
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+ int high_freq_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
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+
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+ /* Low frequency effect data */
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+ double lfe_data[2*DTS_SUBSUBFAMES_MAX*DTS_LFE_MAX * 2 /*history*/];
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+ int lfe_scale_factor;
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+
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+ /* Subband samples history (for ADPCM) */
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+ double subband_samples_hist[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][4];
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+ double subband_fir_hist[DTS_PRIM_CHANNELS_MAX][512];
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+ double subband_fir_noidea[DTS_PRIM_CHANNELS_MAX][64];
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+
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+ /* Audio output */
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+ level_t clev; /* centre channel mix level */
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+ level_t slev; /* surround channels mix level */
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+
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+ int output; /* type of output */
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+ level_t level; /* output level */
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+ sample_t bias; /* output bias */
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+
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+ sample_t * samples; /* pointer to the internal audio samples buffer */
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+ int downmixed;
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+
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+ int dynrnge; /* apply dynamic range */
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+ level_t dynrng; /* dynamic range */
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+ void * dynrngdata; /* dynamic range callback funtion and data */
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+ level_t (* dynrngcall) (level_t range, void * dynrngdata);
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+
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+ /* Bitstream handling */
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+ uint32_t * buffer_start;
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+ uint32_t bits_left;
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+ uint32_t current_word;
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+ int word_mode; /* 16/14 bits word format (1 -> 16, 0 -> 14) */
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+ int bigendian_mode; /* endianness (1 -> be, 0 -> le) */
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+
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+ /* Current position in DTS frame */
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+ int current_subframe;
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+ int current_subsubframe;
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+
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+ /* Pre-calculated cosine modulation coefs for the QMF */
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+ double cos_mod[544];
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+
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+ /* Debug flag */
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+ int debug_flag;
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+};
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+
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+#define LEVEL_PLUS6DB 2.0
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+#define LEVEL_PLUS3DB 1.4142135623730951
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+#define LEVEL_3DB 0.7071067811865476
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+#define LEVEL_45DB 0.5946035575013605
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+#define LEVEL_6DB 0.5
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+
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+int dts_downmix_init (int input, int flags, level_t * level,
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+ level_t clev, level_t slev);
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+int dts_downmix_coeff (level_t * coeff, int acmod, int output, level_t level,
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+ level_t clev, level_t slev);
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+void dts_downmix (sample_t * samples, int acmod, int output, sample_t bias,
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+ level_t clev, level_t slev);
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+void dts_upmix (sample_t * samples, int acmod, int output);
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+
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+#define ROUND(x) ((int)((x) + ((x) > 0 ? 0.5 : -0.5)))
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+
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+#ifndef LIBDTS_FIXED
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+
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+typedef sample_t quantizer_t;
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+#define SAMPLE(x) (x)
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+#define LEVEL(x) (x)
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+#define MUL(a,b) ((a) * (b))
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+#define MUL_L(a,b) ((a) * (b))
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+#define MUL_C(a,b) ((a) * (b))
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+#define DIV(a,b) ((a) / (b))
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+#define BIAS(x) ((x) + bias)
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+
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+#else /* LIBDTS_FIXED */
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+
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+typedef int16_t quantizer_t;
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+#define SAMPLE(x) (sample_t)((x) * (1 << 30))
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+#define LEVEL(x) (level_t)((x) * (1 << 26))
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+
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+#if 0
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+#define MUL(a,b) ((int)(((int64_t)(a) * (b) + (1 << 29)) >> 30))
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+#define MUL_L(a,b) ((int)(((int64_t)(a) * (b) + (1 << 25)) >> 26))
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+#elif 1
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+#define MUL(a,b) \
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+({ int32_t _ta=(a), _tb=(b), _tc; \
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+ _tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)(((_tc >> 14))+ (((_ta >> 16)*(_tb >> 16)) << 2 )); })
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+#define MUL_L(a,b) \
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+({ int32_t _ta=(a), _tb=(b), _tc; \
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+ _tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)((_tc >> 10) + (((_ta >> 16)*(_tb >> 16)) << 6)); })
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+#else
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+#define MUL(a,b) (((a) >> 15) * ((b) >> 15))
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+#define MUL_L(a,b) (((a) >> 13) * ((b) >> 13))
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+#endif
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+
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+#define MUL_C(a,b) MUL_L (a, LEVEL (b))
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+#define DIV(a,b) ((((int64_t)LEVEL (a)) << 26) / (b))
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+#define BIAS(x) (x)
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+
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+#endif
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Michael Niedermayer