mirror of
https://github.com/FFmpeg/FFmpeg.git
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321 lines
7.6 KiB
C
321 lines
7.6 KiB
C
/*
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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 FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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#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 <stdlib.h>
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#include <string.h>
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#ifdef HAVE_MALLOC_H
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#include <malloc.h>
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#endif
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#define BUFFER_SIZE 18726
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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 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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static 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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static 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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static 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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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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*data_size = 0;
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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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return start - buff;
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if (bufpos != buf + HEADER_SIZE)
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break;
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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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}
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{
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level_t level;
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sample_t bias;
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int i;
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flags = 2; /* ???????????? */
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level = CONVERT_LEVEL;
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bias = CONVERT_BIAS;
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flags |= DTS_ADJUST_LEVEL;
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if (dts_frame (state, buf, &flags, &level, bias))
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goto error;
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avctx->sample_rate = sample_rate;
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avctx->channels = channels_multi (flags);
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avctx->bit_rate = bit_rate;
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for (i = 0; i < dts_blocks_num (state); i++)
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{
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if (dts_block (state))
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goto error;
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{
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int chans;
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chans = channels_multi (flags);
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convert2s16_multi (dts_samples (state), data,
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flags & (DTS_CHANNEL_MASK | DTS_LFE));
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data += 256 * sizeof (int16_t) * chans;
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*data_size += 256 * sizeof (int16_t) * chans;
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}
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}
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bufptr = buf;
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bufpos = buf + HEADER_SIZE;
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return start-buff;
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error:
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av_log (NULL, AV_LOG_ERROR, "error\n");
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bufptr = buf;
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bufpos = buf + HEADER_SIZE;
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}
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return start-buff;
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}
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static int
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dts_decode_init (AVCodecContext *avctx)
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{
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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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return 0;
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}
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static int
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dts_decode_end (AVCodecContext *s)
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{
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return 0;
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}
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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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dts_decode_init,
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NULL,
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dts_decode_end,
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dts_decode_frame,
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};
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