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libdts support by (Benjamin Zores <ben at geexbox dot org>) 

Originally committed as revision 3310 to svn://svn.ffmpeg.org/ffmpeg/trunk
tags/v0.5
Michael Niedermayer 21 years ago
parent
eb507b21c4
commit
23c9925329
14 changed files with 635 additions and 6 deletions
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  1. +16
    -0
      configure
  2. +6
    -4
      ffmpeg.c
  3. +5
    -0
      libavcodec/Makefile
  4. +3
    -0
      libavcodec/allcodecs.c
  5. +1
    -0
      libavcodec/avcodec.c
  6. +3
    -0
      libavcodec/avcodec.h
  7. +203
    -0
      libavcodec/dts_internal.h
  8. +349
    -0
      libavcodec/dtsdec.c
  9. +3
    -0
      libavformat/matroska.c
  10. +9
    -2
      libavformat/mpeg.c
  11. +5
    -0
      libavformat/mpegts.c
  12. +1
    -0
      libavformat/mpegts.h
  13. +30
    -0
      libavformat/raw.c
  14. +1
    -0
      libavformat/wav.c

+ 16
- 0
configure View File

@@ -22,6 +22,7 @@ echo " --enable-faac enable faac support via libfaac [default=no]"
echo " --enable-mingw32 enable mingw32 native/cross windows compile"
echo " --enable-a52 enable GPL'ed A52 support [default=no]"
echo " --enable-a52bin open liba52.so.0 at runtime [default=no]"
echo " --enable-dts enable GPL'ed DTS support [default=no]"
echo " --enable-pp enable GPL'ed post processing support [default=no]"
echo " --enable-shared-pp use libpostproc.so [default=no]"
echo " --enable-shared build shared libraries [default=no]"
@@ -143,6 +144,7 @@ faadbin="no"
faac="no"
a52="no"
a52bin="no"
dts="no"
pp="no"
shared_pp="no"
mingw32="no"
@@ -381,6 +383,8 @@ for opt do
;;
--enable-a52bin) a52bin="yes" ; extralibs="$ldl $extralibs"
;;
--enable-dts) dts="yes" ; extralibs="$extralibs -ldts"
;;
--enable-pp) pp="yes"
;;
--enable-shared-pp) shared_pp="yes"
@@ -444,6 +448,11 @@ if test "$gpl" != "yes"; then
echo "liba52 is under GPL and --enable-gpl is not specified"
fail="yes"
fi

if test "$dts" != "no"; then
echo "libdts is under GPL and --enable-gpl is not specified"
fail="yes"
fi
if test "$faad" != "no" -o "$faadbin" != "no"; then
cat > $TMPC << EOF
@@ -973,6 +982,7 @@ echo "faadbin enabled $faadbin"
echo "faac enabled $faac"
echo "a52 support $a52"
echo "a52 dlopened $a52bin"
echo "dts support $dts"
echo "pp support $pp"
echo "debug symbols $debug"
echo "optimize $optimize"
@@ -1169,6 +1179,12 @@ if test "$a52" = "yes" ; then
fi
fi

# DTS
if test "$dts" = "yes" ; then
echo "#define CONFIG_DTS 1" >> $TMPH
echo "CONFIG_DTS=yes" >> config.mak
fi

# PP
if test "$pp" = "yes" ; then
echo "#define CONFIG_PP 1" >> $TMPH


+ 6
- 4
ffmpeg.c View File

@@ -1502,8 +1502,9 @@ static int av_encode(AVFormatContext **output_files,
ost->audio_resample = 0;
} else {
if (codec->channels != icodec->channels &&
icodec->codec_id == CODEC_ID_AC3) {
/* Special case for 5:1 AC3 input */
(icodec->codec_id == CODEC_ID_AC3 ||
icodec->codec_id == CODEC_ID_DTS)) {
/* Special case for 5:1 AC3 and DTS input */
/* and mono or stereo output */
/* Request specific number of channels */
icodec->channels = codec->channels;
@@ -3144,9 +3145,10 @@ static void opt_output_file(const char *filename)
audio_enc->bit_rate = audio_bit_rate;
audio_enc->strict_std_compliance = strict;
audio_enc->thread_count = thread_count;
/* For audio codecs other than AC3 we limit */
/* For audio codecs other than AC3 or DTS we limit */
/* the number of coded channels to stereo */
if (audio_channels > 2 && codec_id != CODEC_ID_AC3) {
if (audio_channels > 2 && codec_id != CODEC_ID_AC3
&& codec_id != CODEC_ID_DTS) {
audio_enc->channels = 2;
} else
audio_enc->channels = audio_channels;


+ 5
- 0
libavcodec/Makefile View File

@@ -73,6 +73,11 @@ OBJS+= liba52/bit_allocate.o liba52/bitstream.o liba52/downmix.o \
endif
endif

# currently using libdts for dts decoding
ifeq ($(CONFIG_DTS),yes)
OBJS+= dtsdec.o
endif

ifeq ($(CONFIG_FAAD),yes)
OBJS+= faad.o
ifeq ($(CONFIG_FAADBIN),yes)


+ 3
- 0
libavcodec/allcodecs.c View File

@@ -150,6 +150,9 @@ void avcodec_register_all(void)
register_avcodec(&zlib_decoder);
#ifdef CONFIG_AC3
register_avcodec(&ac3_decoder);
#endif
#ifdef CONFIG_DTS
register_avcodec(&dts_decoder);
#endif
register_avcodec(&ra_144_decoder);
register_avcodec(&ra_288_decoder);


+ 1
- 0
libavcodec/avcodec.c View File

@@ -38,6 +38,7 @@ static AVCodec* avcodec_find_by_fcc(uint32_t fcc)
{ CODEC_ID_MJPEG, { MKTAG('M', 'J', 'P', 'G'), 0 } },
{ CODEC_ID_MPEG1VIDEO, { MKTAG('P', 'I', 'M', '1'), 0 } },
{ CODEC_ID_AC3, { 0x2000, 0 } },
{ CODEC_ID_DTS, { 0x10, 0 } },
{ CODEC_ID_MP2, { 0x50, 0x55, 0 } },
{ CODEC_ID_FLV1, { MKTAG('F', 'L', 'V', '1'), 0 } },



+ 3
- 0
libavcodec/avcodec.h View File

@@ -140,6 +140,8 @@ enum CodecID {
CODEC_ID_MPEG2TS, /* _FAKE_ codec to indicate a raw MPEG2 transport
stream (only used by libavformat) */

CODEC_ID_DTS,
};

/* CODEC_ID_MP3LAME is absolete */
@@ -1858,6 +1860,7 @@ extern AVCodec rawvideo_decoder;

/* the following codecs use external GPL libs */
extern AVCodec ac3_decoder;
extern AVCodec dts_decoder;

/* resample.c */



+ 203
- 0
libavcodec/dts_internal.h View File

@@ -0,0 +1,203 @@
/*
* dts_internal.h
* Copyright (C) 2004 Gildas Bazin <gbazin@videolan.org>
* Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
* Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of dtsdec, a free DTS Coherent Acoustics stream decoder.
* See http://www.videolan.org/dtsdec.html for updates.
*
* dtsdec 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 of the License, or
* (at your option) any later version.
*
* dtsdec 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 this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/

#define DTS_SUBFRAMES_MAX (16)
#define DTS_PRIM_CHANNELS_MAX (5)
#define DTS_SUBBANDS (32)
#define DTS_ABITS_MAX (32) /* Should be 28 */
#define DTS_SUBSUBFAMES_MAX (4)
#define DTS_LFE_MAX (3)

struct dts_state_s {

/* Frame header */
int frame_type; /* type of the current frame */
int samples_deficit; /* deficit sample count */
int crc_present; /* crc is present in the bitstream */
int sample_blocks; /* number of PCM sample blocks */
int frame_size; /* primary frame byte size */
int amode; /* audio channels arrangement */
int sample_rate; /* audio sampling rate */
int bit_rate; /* transmission bit rate */

int downmix; /* embedded downmix enabled */
int dynrange; /* embedded dynamic range flag */
int timestamp; /* embedded time stamp flag */
int aux_data; /* auxiliary data flag */
int hdcd; /* source material is mastered in HDCD */
int ext_descr; /* extension audio descriptor flag */
int ext_coding; /* extended coding flag */
int aspf; /* audio sync word insertion flag */
int lfe; /* low frequency effects flag */
int predictor_history; /* predictor history flag */
int header_crc; /* header crc check bytes */
int multirate_inter; /* multirate interpolator switch */
int version; /* encoder software revision */
int copy_history; /* copy history */
int source_pcm_res; /* source pcm resolution */
int front_sum; /* front sum/difference flag */
int surround_sum; /* surround sum/difference flag */
int dialog_norm; /* dialog normalisation parameter */

/* Primary audio coding header */
int subframes; /* number of subframes */
int prim_channels; /* number of primary audio channels */
/* subband activity count */
int subband_activity[DTS_PRIM_CHANNELS_MAX];
/* high frequency vq start subband */
int vq_start_subband[DTS_PRIM_CHANNELS_MAX];
/* joint intensity coding index */
int joint_intensity[DTS_PRIM_CHANNELS_MAX];
/* transient mode code book */
int transient_huffman[DTS_PRIM_CHANNELS_MAX];
/* scale factor code book */
int scalefactor_huffman[DTS_PRIM_CHANNELS_MAX];
/* bit allocation quantizer select */
int bitalloc_huffman[DTS_PRIM_CHANNELS_MAX];
/* quantization index codebook select */
int quant_index_huffman[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];
/* scale factor adjustment */
float scalefactor_adj[DTS_PRIM_CHANNELS_MAX][DTS_ABITS_MAX];

/* Primary audio coding side information */
int subsubframes; /* number of subsubframes */
int partial_samples; /* partial subsubframe samples count */
/* prediction mode (ADPCM used or not) */
int prediction_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* prediction VQ coefs */
int prediction_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* bit allocation index */
int bitalloc[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* transition mode (transients) */
int transition_mode[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* scale factors (2 if transient)*/
int scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][2];
/* joint subband scale factors codebook */
int joint_huff[DTS_PRIM_CHANNELS_MAX];
/* joint subband scale factors */
int joint_scale_factor[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];
/* stereo downmix coefficients */
int downmix_coef[DTS_PRIM_CHANNELS_MAX][2];
/* dynamic range coefficient */
int dynrange_coef;

/* VQ encoded high frequency subbands */
int high_freq_vq[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS];

/* Low frequency effect data */
double lfe_data[2*DTS_SUBSUBFAMES_MAX*DTS_LFE_MAX * 2 /*history*/];
int lfe_scale_factor;

/* Subband samples history (for ADPCM) */
double subband_samples_hist[DTS_PRIM_CHANNELS_MAX][DTS_SUBBANDS][4];
double subband_fir_hist[DTS_PRIM_CHANNELS_MAX][512];
double subband_fir_noidea[DTS_PRIM_CHANNELS_MAX][64];

/* Audio output */
level_t clev; /* centre channel mix level */
level_t slev; /* surround channels mix level */

int output; /* type of output */
level_t level; /* output level */
sample_t bias; /* output bias */

sample_t * samples; /* pointer to the internal audio samples buffer */
int downmixed;

int dynrnge; /* apply dynamic range */
level_t dynrng; /* dynamic range */
void * dynrngdata; /* dynamic range callback funtion and data */
level_t (* dynrngcall) (level_t range, void * dynrngdata);

/* Bitstream handling */
uint32_t * buffer_start;
uint32_t bits_left;
uint32_t current_word;
int word_mode; /* 16/14 bits word format (1 -> 16, 0 -> 14) */
int bigendian_mode; /* endianness (1 -> be, 0 -> le) */

/* Current position in DTS frame */
int current_subframe;
int current_subsubframe;

/* Pre-calculated cosine modulation coefs for the QMF */
double cos_mod[544];

/* Debug flag */
int debug_flag;
};

#define LEVEL_PLUS6DB 2.0
#define LEVEL_PLUS3DB 1.4142135623730951
#define LEVEL_3DB 0.7071067811865476
#define LEVEL_45DB 0.5946035575013605
#define LEVEL_6DB 0.5

int dts_downmix_init (int input, int flags, level_t * level,
level_t clev, level_t slev);
int dts_downmix_coeff (level_t * coeff, int acmod, int output, level_t level,
level_t clev, level_t slev);
void dts_downmix (sample_t * samples, int acmod, int output, sample_t bias,
level_t clev, level_t slev);
void dts_upmix (sample_t * samples, int acmod, int output);

#define ROUND(x) ((int)((x) + ((x) > 0 ? 0.5 : -0.5)))

#ifndef LIBDTS_FIXED

typedef sample_t quantizer_t;
#define SAMPLE(x) (x)
#define LEVEL(x) (x)
#define MUL(a,b) ((a) * (b))
#define MUL_L(a,b) ((a) * (b))
#define MUL_C(a,b) ((a) * (b))
#define DIV(a,b) ((a) / (b))
#define BIAS(x) ((x) + bias)

#else /* LIBDTS_FIXED */

typedef int16_t quantizer_t;
#define SAMPLE(x) (sample_t)((x) * (1 << 30))
#define LEVEL(x) (level_t)((x) * (1 << 26))

#if 0
#define MUL(a,b) ((int)(((int64_t)(a) * (b) + (1 << 29)) >> 30))
#define MUL_L(a,b) ((int)(((int64_t)(a) * (b) + (1 << 25)) >> 26))
#elif 1
#define MUL(a,b) \
({ int32_t _ta=(a), _tb=(b), _tc; \
_tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)(((_tc >> 14))+ (((_ta >> 16)*(_tb >> 16)) << 2 )); })
#define MUL_L(a,b) \
({ int32_t _ta=(a), _tb=(b), _tc; \
_tc=(_ta & 0xffff)*(_tb >> 16)+(_ta >> 16)*(_tb & 0xffff); (int32_t)((_tc >> 10) + (((_ta >> 16)*(_tb >> 16)) << 6)); })
#else
#define MUL(a,b) (((a) >> 15) * ((b) >> 15))
#define MUL_L(a,b) (((a) >> 13) * ((b) >> 13))
#endif

#define MUL_C(a,b) MUL_L (a, LEVEL (b))
#define DIV(a,b) ((((int64_t)LEVEL (a)) << 26) / (b))
#define BIAS(x) (x)

#endif

+ 349
- 0
libavcodec/dtsdec.c View File

@@ -0,0 +1,349 @@
/*
* dtsdec.c : free DTS Coherent Acoustics stream decoder.
* Copyright (C) 2004 Benjamin Zores <ben@geexbox.org>
*
* This file is part of libavcodec.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/

#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif

#include "avcodec.h"
#include <dts.h>
#include "dts_internal.h"

#include <stdlib.h>
#include <string.h>
#include <malloc.h>
#include <math.h>

#define INBUF_SIZE 4096
#define BUFFER_SIZE 4096
#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 void
pre_calc_cosmod (dts_state_t * state)
{
int i, j, k;

for (j=0,k=0;k<16;k++)
for (i=0;i<16;i++)
state->cos_mod[j++] = cos((2*i+1)*(2*k+1)*M_PI/64);

for (k=0;k<16;k++)
for (i=0;i<16;i++)
state->cos_mod[j++] = cos((i)*(2*k+1)*M_PI/32);

for (k=0;k<16;k++)
state->cos_mod[j++] = 0.25/(2*cos((2*k+1)*M_PI/128));

for (k=0;k<16;k++)
state->cos_mod[j++] = -0.25/(2.0*sin((2*k+1)*M_PI/128));
}

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;
*data_size = 0;

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;

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)
{
if (bufpos == buf + HEADER_SIZE)
{
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;
}
else
{
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;
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,
};

+ 3
- 0
libavformat/matroska.c View File

@@ -2228,6 +2228,9 @@ matroska_read_header (AVFormatContext *s,
else if (!strcmp(track->codec_id,
MATROSKA_CODEC_ID_AUDIO_AC3))
codec_id = CODEC_ID_AC3;
else if (!strcmp(track->codec_id,
MATROSKA_CODEC_ID_AUDIO_DTS))
codec_id = CODEC_ID_DTS;
/* No such codec id so far. */
/* else if (!strcmp(track->codec_id, */
/* MATROSKA_CODEC_ID_AUDIO_DTS)) */


+ 9
- 2
libavformat/mpeg.c View File

@@ -77,6 +77,7 @@ typedef struct {
#define AUDIO_ID 0xc0
#define VIDEO_ID 0xe0
#define AC3_ID 0x80
#define DTS_ID 0x8a
#define LPCM_ID 0xa0

static const int lpcm_freq_tab[4] = { 48000, 96000, 44100, 32000 };
@@ -235,7 +236,7 @@ static int get_system_header_size(AVFormatContext *ctx)
static int mpeg_mux_init(AVFormatContext *ctx)
{
MpegMuxContext *s = ctx->priv_data;
int bitrate, i, mpa_id, mpv_id, ac3_id, lpcm_id, j;
int bitrate, i, mpa_id, mpv_id, ac3_id, dts_id, lpcm_id, j;
AVStream *st;
StreamInfo *stream;
int audio_bitrate;
@@ -258,6 +259,7 @@ static int mpeg_mux_init(AVFormatContext *ctx)
s->video_bound = 0;
mpa_id = AUDIO_ID;
ac3_id = AC3_ID;
dts_id = DTS_ID;
mpv_id = VIDEO_ID;
lpcm_id = LPCM_ID;
s->scr_stream_index = -1;
@@ -272,6 +274,8 @@ static int mpeg_mux_init(AVFormatContext *ctx)
case CODEC_TYPE_AUDIO:
if (st->codec.codec_id == CODEC_ID_AC3) {
stream->id = ac3_id++;
} else if (st->codec.codec_id == CODEC_ID_DTS) {
stream->id = dts_id++;
} else if (st->codec.codec_id == CODEC_ID_PCM_S16BE) {
stream->id = lpcm_id++;
for(j = 0; j < 4; j++) {
@@ -1304,9 +1308,12 @@ static int mpegps_read_packet(AVFormatContext *s,
} else if (startcode >= 0x1c0 && startcode <= 0x1df) {
type = CODEC_TYPE_AUDIO;
codec_id = CODEC_ID_MP2;
} else if (startcode >= 0x80 && startcode <= 0x9f) {
} else if (startcode >= 0x80 && startcode <= 0x89) {
type = CODEC_TYPE_AUDIO;
codec_id = CODEC_ID_AC3;
} else if (startcode >= 0x8a && startcode <= 0x9f) {
type = CODEC_TYPE_AUDIO;
codec_id = CODEC_ID_DTS;
} else if (startcode >= 0xa0 && startcode <= 0xbf) {
type = CODEC_TYPE_AUDIO;
codec_id = CODEC_ID_PCM_S16BE;


+ 5
- 0
libavformat/mpegts.c View File

@@ -431,6 +431,7 @@ static void pmt_cb(void *opaque, const uint8_t *section, int section_len)
case STREAM_TYPE_VIDEO_H264:
case STREAM_TYPE_AUDIO_AAC:
case STREAM_TYPE_AUDIO_AC3:
case STREAM_TYPE_AUDIO_DTS:
add_pes_stream(ts, pid, stream_type);
break;
default:
@@ -753,6 +754,10 @@ static void mpegts_push_data(void *opaque,
codec_type = CODEC_TYPE_AUDIO;
codec_id = CODEC_ID_AC3;
break;
case STREAM_TYPE_AUDIO_DTS:
codec_type = CODEC_TYPE_AUDIO;
codec_id = CODEC_ID_DTS;
break;
default:
if (code >= 0x1c0 && code <= 0x1df) {
codec_type = CODEC_TYPE_AUDIO;


+ 1
- 0
libavformat/mpegts.h View File

@@ -42,6 +42,7 @@
#define STREAM_TYPE_VIDEO_H264 0x1b

#define STREAM_TYPE_AUDIO_AC3 0x81
#define STREAM_TYPE_AUDIO_DTS 0x8a

unsigned int mpegts_crc32(const uint8_t *data, int len);
extern AVOutputFormat mpegts_mux;


+ 30
- 0
libavformat/raw.c View File

@@ -184,6 +184,23 @@ static int ac3_read_header(AVFormatContext *s,
return 0;
}

/* dts read */
static int dts_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
AVStream *st;

st = av_new_stream(s, 0);
if (!st)
return AVERROR_NOMEM;

st->codec.codec_type = CODEC_TYPE_AUDIO;
st->codec.codec_id = CODEC_ID_DTS;
st->need_parsing = 1;
/* the parameters will be extracted from the compressed bitstream */
return 0;
}

/* mpeg1/h263 input */
static int video_read_header(AVFormatContext *s,
AVFormatParameters *ap)
@@ -300,6 +317,17 @@ AVOutputFormat ac3_oformat = {
};
#endif //CONFIG_ENCODERS

AVInputFormat dts_iformat = {
"dts",
"raw dts",
0,
NULL,
dts_read_header,
raw_read_partial_packet,
raw_read_close,
.extensions = "dts",
};

AVInputFormat h261_iformat = {
"h261",
"raw h261",
@@ -613,6 +641,8 @@ int raw_init(void)
av_register_input_format(&ac3_iformat);
av_register_output_format(&ac3_oformat);

av_register_input_format(&dts_iformat);

av_register_input_format(&h261_iformat);

av_register_input_format(&h263_iformat);


+ 1
- 0
libavformat/wav.c View File

@@ -348,6 +348,7 @@ static int wav_read_seek(AVFormatContext *s,
case CODEC_ID_MP2:
case CODEC_ID_MP3:
case CODEC_ID_AC3:
case CODEC_ID_DTS:
/* use generic seeking with dynamically generated indexes */
return -1;
default:


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