mirror of https://github.com/FFmpeg/FFmpeg.git
Originally committed as revision 3310 to svn://svn.ffmpeg.org/ffmpeg/trunkpull/126/head
Michael Niedermayer
21 years ago
parent
eb507b21c4
commit
23c9925329
14 changed files with 635 additions and 6 deletions
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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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@ -0,0 +1,349 @@ |
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/*
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* dtsdec.c : free DTS Coherent Acoustics stream decoder. |
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* Copyright (C) 2004 Benjamin Zores <ben@geexbox.org> |
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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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* |
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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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* |
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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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*/ |
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#ifdef HAVE_AV_CONFIG_H |
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#undef HAVE_AV_CONFIG_H |
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#endif |
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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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#define HEADER_SIZE 14 |
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#ifdef LIBDTS_FIXED |
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#define CONVERT_LEVEL (1 << 26) |
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#define CONVERT_BIAS 0 |
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#else |
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#define CONVERT_LEVEL 1 |
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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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#ifdef LIBDTS_FIXED |
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i >>= 15; |
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#else |
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i -= 0x43c00000; |
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#endif |
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return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i); |
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} |
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void |
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convert2s16_2 (sample_t * _f, int16_t * s16) |
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{ |
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int i; |
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int32_t * f = (int32_t *) _f; |
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for (i = 0; i < 256; i++) |
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{ |
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s16[2*i] = convert (f[i]); |
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s16[2*i+1] = convert (f[i+256]); |
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} |
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} |
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void |
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convert2s16_4 (sample_t * _f, int16_t * s16) |
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{ |
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int i; |
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int32_t * f = (int32_t *) _f; |
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for (i = 0; i < 256; i++) |
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{ |
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s16[4*i] = convert (f[i]); |
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s16[4*i+1] = convert (f[i+256]); |
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s16[4*i+2] = convert (f[i+512]); |
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s16[4*i+3] = convert (f[i+768]); |
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} |
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} |
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void |
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convert2s16_5 (sample_t * _f, int16_t * s16) |
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{ |
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int i; |
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int32_t * f = (int32_t *) _f; |
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for (i = 0; i < 256; i++) |
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{ |
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s16[5*i] = convert (f[i]); |
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s16[5*i+1] = convert (f[i+256]); |
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s16[5*i+2] = convert (f[i+512]); |
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s16[5*i+3] = convert (f[i+768]); |
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s16[5*i+4] = convert (f[i+1024]); |
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} |
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} |
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static void |
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convert2s16_multi (sample_t * _f, int16_t * s16, int flags) |
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{ |
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int i; |
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int32_t * f = (int32_t *) _f; |
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switch (flags) |
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{ |
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case DTS_MONO: |
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for (i = 0; i < 256; i++) |
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{ |
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s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0; |
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s16[5*i+4] = convert (f[i]); |
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} |
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break; |
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case DTS_CHANNEL: |
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case DTS_STEREO: |
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case DTS_DOLBY: |
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convert2s16_2 (_f, s16); |
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break; |
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case DTS_3F: |
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for (i = 0; i < 256; i++) |
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{ |
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s16[5*i] = convert (f[i]); |
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s16[5*i+1] = convert (f[i+512]); |
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s16[5*i+2] = s16[5*i+3] = 0; |
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s16[5*i+4] = convert (f[i+256]); |
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} |
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break; |
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case DTS_2F2R: |
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convert2s16_4 (_f, s16); |
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break; |
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case DTS_3F2R: |
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convert2s16_5 (_f, s16); |
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break; |
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case DTS_MONO | DTS_LFE: |
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for (i = 0; i < 256; i++) |
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{ |
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s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0; |
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s16[6*i+4] = convert (f[i+256]); |
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s16[6*i+5] = convert (f[i]); |
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} |
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break; |
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case DTS_CHANNEL | DTS_LFE: |
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case DTS_STEREO | DTS_LFE: |
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case DTS_DOLBY | DTS_LFE: |
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for (i = 0; i < 256; i++) |
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{ |
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s16[6*i] = convert (f[i+256]); |
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s16[6*i+1] = convert (f[i+512]); |
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s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0; |
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s16[6*i+5] = convert (f[i]); |
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} |
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break; |
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case DTS_3F | DTS_LFE: |
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for (i = 0; i < 256; i++) |
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{ |
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s16[6*i] = convert (f[i+256]); |
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s16[6*i+1] = convert (f[i+768]); |
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s16[6*i+2] = s16[6*i+3] = 0; |
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s16[6*i+4] = convert (f[i+512]); |
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s16[6*i+5] = convert (f[i]); |
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} |
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break; |
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case DTS_2F2R | DTS_LFE: |
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for (i = 0; i < 256; i++) |
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{ |
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s16[6*i] = convert (f[i+256]); |
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s16[6*i+1] = convert (f[i+512]); |
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s16[6*i+2] = convert (f[i+768]); |
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s16[6*i+3] = convert (f[i+1024]); |
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s16[6*i+4] = 0; |
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s16[6*i+5] = convert (f[i]); |
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} |
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break; |
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case DTS_3F2R | DTS_LFE: |
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for (i = 0; i < 256; i++) |
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{ |
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s16[6*i] = convert (f[i+256]); |
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s16[6*i+1] = convert (f[i+768]); |
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s16[6*i+2] = convert (f[i+1024]); |
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s16[6*i+3] = convert (f[i+1280]); |
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s16[6*i+4] = convert (f[i+512]); |
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s16[6*i+5] = convert (f[i]); |
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} |
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break; |
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} |
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} |
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static int |
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channels_multi (int flags) |
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{ |
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if (flags & DTS_LFE) |
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return 6; |
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else if (flags & 1) /* center channel */ |
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return 5; |
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else if ((flags & DTS_CHANNEL_MASK) == DTS_2F2R) |
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return 4; |
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else |
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return 2; |
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} |
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static int |
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dts_decode_frame (AVCodecContext *avctx, void *data, int *data_size, |
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uint8_t *buff, int buff_size) |
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{ |
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uint8_t * start = buff; |
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uint8_t * end = buff + buff_size; |
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*data_size = 0; |
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static uint8_t buf[BUFFER_SIZE]; |
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static uint8_t * bufptr = buf; |
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static uint8_t * bufpos = buf + HEADER_SIZE; |
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static int sample_rate; |
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static int frame_length; |
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static int flags; |
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int bit_rate; |
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int len; |
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dts_state_t *state = avctx->priv_data; |
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while (1) |
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{ |
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len = end - start; |
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if (!len) |
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break; |
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if (len > bufpos - bufptr) |
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len = bufpos - bufptr; |
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memcpy (bufptr, start, len); |
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bufptr += len; |
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start += len; |
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if (bufptr == bufpos) |
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{ |
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if (bufpos == buf + HEADER_SIZE) |
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{ |
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int length; |
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length = dts_syncinfo (state, buf, &flags, &sample_rate, |
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&bit_rate, &frame_length); |
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if (!length) |
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{ |
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av_log (NULL, AV_LOG_INFO, "skip\n"); |
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for (bufptr = buf; bufptr < buf + HEADER_SIZE-1; bufptr++) |
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bufptr[0] = bufptr[1]; |
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continue; |
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} |
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bufpos = buf + length; |
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} |
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else |
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{ |
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level_t level; |
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sample_t bias; |
||||
int i; |
||||
|
||||
flags = 2; /* ???????????? */ |
||||
level = CONVERT_LEVEL; |
||||
bias = CONVERT_BIAS; |
||||
|
||||
flags |= DTS_ADJUST_LEVEL; |
||||
if (dts_frame (state, buf, &flags, &level, bias)) |
||||
goto error; |
||||
for (i = 0; i < dts_blocks_num (state); i++) |
||||
{ |
||||
if (dts_block (state)) |
||||
goto error; |
||||
{ |
||||
int chans; |
||||
chans = channels_multi (flags); |
||||
convert2s16_multi (dts_samples (state), data, |
||||
flags & (DTS_CHANNEL_MASK | DTS_LFE)); |
||||
|
||||
data += 256 * sizeof (int16_t) * chans; |
||||
*data_size += 256 * sizeof (int16_t) * chans; |
||||
} |
||||
} |
||||
bufptr = buf; |
||||
bufpos = buf + HEADER_SIZE; |
||||
continue; |
||||
error: |
||||
av_log (NULL, AV_LOG_ERROR, "error\n"); |
||||
bufptr = buf; |
||||
bufpos = buf + HEADER_SIZE; |
||||
} |
||||
} |
||||
} |
||||
|
||||
return buff_size; |
||||
} |
||||
|
||||
static int |
||||
dts_decode_init (AVCodecContext *avctx) |
||||
{ |
||||
dts_state_t * state; |
||||
int i; |
||||
|
||||
state = avctx->priv_data; |
||||
memset (state, 0, sizeof (dts_state_t)); |
||||
|
||||
state->samples = (sample_t *) memalign (16, 256 * 12 * sizeof (sample_t)); |
||||
if (state->samples == NULL) |
||||
return 1; |
||||
|
||||
for (i = 0; i < 256 * 12; i++) |
||||
state->samples[i] = 0; |
||||
|
||||
/* Pre-calculate cosine modulation coefficients */ |
||||
pre_calc_cosmod (state); |
||||
state->downmixed = 1; |
||||
|
||||
return 0; |
||||
} |
||||
|
||||
static int |
||||
dts_decode_end (AVCodecContext *s) |
||||
{ |
||||
return 0; |
||||
} |
||||
|
||||
AVCodec dts_decoder = { |
||||
"dts",
|
||||
CODEC_TYPE_AUDIO, |
||||
CODEC_ID_DTS, |
||||
sizeof (dts_state_t), |
||||
dts_decode_init, |
||||
NULL, |
||||
dts_decode_end, |
||||
dts_decode_frame, |
||||
}; |
||||
Loading…
Reference in new issue