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/*
* dtsdec.c : free DTS Coherent Acoustics stream decoder.
* Copyright (C) 2004 Benjamin Zores <ben@geexbox.org>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2.1 of the License, or
* (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif
#include "avcodec.h"
#include <dts.h>
#include <stdlib.h>
#include <string.h>
#ifdef HAVE_MALLOC_H
#include <malloc.h>
#endif
#define BUFFER_SIZE 18726
#define HEADER_SIZE 14
#ifdef LIBDTS_FIXED
#define CONVERT_LEVEL (1 << 26)
#define CONVERT_BIAS 0
#else
#define CONVERT_LEVEL 1
#define CONVERT_BIAS 384
#endif
static inline
int16_t convert (int32_t i)
{
#ifdef LIBDTS_FIXED
i >>= 15;
#else
i -= 0x43c00000;
#endif
return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
}
void
convert2s16_2 (sample_t * _f, int16_t * s16)
{
int i;
int32_t * f = (int32_t *) _f;
for (i = 0; i < 256; i++)
{
s16[2*i] = convert (f[i]);
s16[2*i+1] = convert (f[i+256]);
}
}
void
convert2s16_4 (sample_t * _f, int16_t * s16)
{
int i;
int32_t * f = (int32_t *) _f;
for (i = 0; i < 256; i++)
{
s16[4*i] = convert (f[i]);
s16[4*i+1] = convert (f[i+256]);
s16[4*i+2] = convert (f[i+512]);
s16[4*i+3] = convert (f[i+768]);
}
}
void
convert2s16_5 (sample_t * _f, int16_t * s16)
{
int i;
int32_t * f = (int32_t *) _f;
for (i = 0; i < 256; i++)
{
s16[5*i] = convert (f[i]);
s16[5*i+1] = convert (f[i+256]);
s16[5*i+2] = convert (f[i+512]);
s16[5*i+3] = convert (f[i+768]);
s16[5*i+4] = convert (f[i+1024]);
}
}
static void
convert2s16_multi (sample_t * _f, int16_t * s16, int flags)
{
int i;
int32_t * f = (int32_t *) _f;
switch (flags)
{
case DTS_MONO:
for (i = 0; i < 256; i++)
{
s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0;
s16[5*i+4] = convert (f[i]);
}
break;
case DTS_CHANNEL:
case DTS_STEREO:
case DTS_DOLBY:
convert2s16_2 (_f, s16);
break;
case DTS_3F:
for (i = 0; i < 256; i++)
{
s16[5*i] = convert (f[i]);
s16[5*i+1] = convert (f[i+512]);
s16[5*i+2] = s16[5*i+3] = 0;
s16[5*i+4] = convert (f[i+256]);
}
break;
case DTS_2F2R:
convert2s16_4 (_f, s16);
break;
case DTS_3F2R:
convert2s16_5 (_f, s16);
break;
case DTS_MONO | DTS_LFE:
for (i = 0; i < 256; i++)
{
s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0;
s16[6*i+4] = convert (f[i+256]);
s16[6*i+5] = convert (f[i]);
}
break;
case DTS_CHANNEL | DTS_LFE:
case DTS_STEREO | DTS_LFE:
case DTS_DOLBY | DTS_LFE:
for (i = 0; i < 256; i++)
{
s16[6*i] = convert (f[i+256]);
s16[6*i+1] = convert (f[i+512]);
s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
s16[6*i+5] = convert (f[i]);
}
break;
case DTS_3F | DTS_LFE:
for (i = 0; i < 256; i++)
{
s16[6*i] = convert (f[i+256]);
s16[6*i+1] = convert (f[i+768]);
s16[6*i+2] = s16[6*i+3] = 0;
s16[6*i+4] = convert (f[i+512]);
s16[6*i+5] = convert (f[i]);
}
break;
case DTS_2F2R | DTS_LFE:
for (i = 0; i < 256; i++)
{
s16[6*i] = convert (f[i+256]);
s16[6*i+1] = convert (f[i+512]);
s16[6*i+2] = convert (f[i+768]);
s16[6*i+3] = convert (f[i+1024]);
s16[6*i+4] = 0;
s16[6*i+5] = convert (f[i]);
}
break;
case DTS_3F2R | DTS_LFE:
for (i = 0; i < 256; i++)
{
s16[6*i] = convert (f[i+256]);
s16[6*i+1] = convert (f[i+768]);
s16[6*i+2] = convert (f[i+1024]);
s16[6*i+3] = convert (f[i+1280]);
s16[6*i+4] = convert (f[i+512]);
s16[6*i+5] = convert (f[i]);
}
break;
}
}
static int
channels_multi (int flags)
{
if (flags & DTS_LFE)
return 6;
else if (flags & 1) /* center channel */
return 5;
else if ((flags & DTS_CHANNEL_MASK) == DTS_2F2R)
return 4;
else
return 2;
}
static int
dts_decode_frame (AVCodecContext *avctx, void *data, int *data_size,
uint8_t *buff, int buff_size)
{
uint8_t * start = buff;
uint8_t * end = buff + buff_size;
static uint8_t buf[BUFFER_SIZE];
static uint8_t * bufptr = buf;
static uint8_t * bufpos = buf + HEADER_SIZE;
static int sample_rate;
static int frame_length;
static int flags;
int bit_rate;
int len;
dts_state_t *state = avctx->priv_data;
*data_size = 0;
while (1)
{
len = end - start;
if (!len)
break;
if (len > bufpos - bufptr)
len = bufpos - bufptr;
memcpy (bufptr, start, len);
bufptr += len;
start += len;
if (bufptr != bufpos)
return start - buff;
if (bufpos != buf + HEADER_SIZE)
break;
{
int length;
length = dts_syncinfo (state, buf, &flags, &sample_rate,
&bit_rate, &frame_length);
if (!length)
{
av_log (NULL, AV_LOG_INFO, "skip\n");
for (bufptr = buf; bufptr < buf + HEADER_SIZE-1; bufptr++)
bufptr[0] = bufptr[1];
continue;
}
bufpos = buf + length;
}
}
{
level_t level;
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;
avctx->sample_rate = sample_rate;
avctx->channels = channels_multi (flags);
avctx->bit_rate = bit_rate;
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;
return start-buff;
error:
av_log (NULL, AV_LOG_ERROR, "error\n");
bufptr = buf;
bufpos = buf + HEADER_SIZE;
}
return start-buff;
}
static int
dts_decode_init (AVCodecContext *avctx)
{
avctx->priv_data = dts_init (0);
if (avctx->priv_data == NULL)
return -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,
};
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