mirror of https://git.ffmpeg.org/ffmpeg.git
remove dts_internal.h
avoiding code redundance license copy paste fix patch by (Benjamin Zores <ben at geexbox dot org>) Originally committed as revision 3315 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Benjamin Zores
Michael Niedermayer
parent
5f63d108eb
commit
a708785f2e
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@ -1,203 +0,0 @@
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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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#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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struct dts_state_s {
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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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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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/* 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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/* 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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/* VQ encoded high frequency subbands */
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int high_freq_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
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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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/* 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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/* 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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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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sample_t * samples; /* pointer to the internal audio samples buffer */
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int downmixed;
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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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/* 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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/* Current position in DTS frame */
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int current_subframe;
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int current_subsubframe;
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/* Pre-calculated cosine modulation coefs for the QMF */
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double cos_mod[544];
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/* Debug flag */
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int debug_flag;
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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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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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#define ROUND(x) ((int)((x) + ((x) > 0 ? 0.5 : -0.5)))
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#ifndef LIBDTS_FIXED
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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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#else /* LIBDTS_FIXED */
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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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#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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#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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#endif
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@ -4,19 +4,19 @@
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*
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* This file is part of libavcodec.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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* This library 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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* This library 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 GNU
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* Lesser General Public License for more details.
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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 Lesser General Public
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* License along with this library; 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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* 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., 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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#ifdef HAVE_AV_CONFIG_H
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@ -25,12 +25,10 @@
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#include "avcodec.h"
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#include <dts.h>
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#include "dts_internal.h"
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#include <stdlib.h>
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#include <string.h>
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#include <malloc.h>
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#include <math.h>
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#define INBUF_SIZE 4096
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#define BUFFER_SIZE 4096
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@ -44,26 +42,6 @@
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#define CONVERT_BIAS 384
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#endif
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static void
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pre_calc_cosmod (dts_state_t * state)
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{
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int i, j, k;
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for (j=0,k=0;k<16;k++)
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for (i=0;i<16;i++)
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state->cos_mod[j++] = cos((2*i+1)*(2*k+1)*M_PI/64);
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for (k=0;k<16;k++)
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for (i=0;i<16;i++)
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state->cos_mod[j++] = cos((i)*(2*k+1)*M_PI/32);
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for (k=0;k<16;k++)
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state->cos_mod[j++] = 0.25/(2*cos((2*k+1)*M_PI/128));
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for (k=0;k<16;k++)
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state->cos_mod[j++] = -0.25/(2.0*sin((2*k+1)*M_PI/128));
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}
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static inline
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int16_t convert (int32_t i)
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{
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@ -311,23 +289,10 @@ dts_decode_frame (AVCodecContext *avctx, void *data, int *data_size,
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static int
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dts_decode_init (AVCodecContext *avctx)
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{
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dts_state_t * state;
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int i;
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state = avctx->priv_data;
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memset (state, 0, sizeof (dts_state_t));
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state->samples = (sample_t *) memalign (16, 256 * 12 * sizeof (sample_t));
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if (state->samples == NULL)
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avctx->priv_data = dts_init (0);
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if (avctx->priv_data == NULL)
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return 1;
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for (i = 0; i < 256 * 12; i++)
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state->samples[i] = 0;
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/* Pre-calculate cosine modulation coefficients */
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pre_calc_cosmod (state);
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state->downmixed = 1;
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return 0;
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}
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@ -341,7 +306,7 @@ AVCodec dts_decoder = {
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"dts",
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CODEC_TYPE_AUDIO,
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CODEC_ID_DTS,
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sizeof (dts_state_t),
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sizeof (dts_state_t *),
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dts_decode_init,
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NULL,
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dts_decode_end,
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