libdts support by (Benjamin Zores <ben at geexbox dot org>)
Originally committed as revision 3310 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
parent
eb507b21c4
commit
23c9925329
16
configure
vendored
16
configure
vendored
@ -22,6 +22,7 @@ echo " --enable-faac enable faac support via libfaac [default=no]"
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echo " --enable-mingw32 enable mingw32 native/cross windows compile"
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echo " --enable-mingw32 enable mingw32 native/cross windows compile"
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echo " --enable-a52 enable GPL'ed A52 support [default=no]"
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echo " --enable-a52 enable GPL'ed A52 support [default=no]"
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echo " --enable-a52bin open liba52.so.0 at runtime [default=no]"
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echo " --enable-a52bin open liba52.so.0 at runtime [default=no]"
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echo " --enable-dts enable GPL'ed DTS support [default=no]"
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echo " --enable-pp enable GPL'ed post processing support [default=no]"
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echo " --enable-pp enable GPL'ed post processing support [default=no]"
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echo " --enable-shared-pp use libpostproc.so [default=no]"
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echo " --enable-shared-pp use libpostproc.so [default=no]"
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echo " --enable-shared build shared libraries [default=no]"
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echo " --enable-shared build shared libraries [default=no]"
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@ -143,6 +144,7 @@ faadbin="no"
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faac="no"
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faac="no"
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a52="no"
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a52="no"
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a52bin="no"
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a52bin="no"
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dts="no"
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pp="no"
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pp="no"
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shared_pp="no"
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shared_pp="no"
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mingw32="no"
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mingw32="no"
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@ -381,6 +383,8 @@ for opt do
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;;
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;;
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--enable-a52bin) a52bin="yes" ; extralibs="$ldl $extralibs"
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--enable-a52bin) a52bin="yes" ; extralibs="$ldl $extralibs"
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;;
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;;
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--enable-dts) dts="yes" ; extralibs="$extralibs -ldts"
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;;
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--enable-pp) pp="yes"
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--enable-pp) pp="yes"
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;;
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;;
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--enable-shared-pp) shared_pp="yes"
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--enable-shared-pp) shared_pp="yes"
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@ -444,6 +448,11 @@ if test "$gpl" != "yes"; then
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echo "liba52 is under GPL and --enable-gpl is not specified"
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echo "liba52 is under GPL and --enable-gpl is not specified"
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fail="yes"
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fail="yes"
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fi
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fi
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if test "$dts" != "no"; then
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echo "libdts is under GPL and --enable-gpl is not specified"
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fail="yes"
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fi
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if test "$faad" != "no" -o "$faadbin" != "no"; then
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if test "$faad" != "no" -o "$faadbin" != "no"; then
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cat > $TMPC << EOF
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cat > $TMPC << EOF
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@ -973,6 +982,7 @@ echo "faadbin enabled $faadbin"
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echo "faac enabled $faac"
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echo "faac enabled $faac"
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echo "a52 support $a52"
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echo "a52 support $a52"
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echo "a52 dlopened $a52bin"
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echo "a52 dlopened $a52bin"
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echo "dts support $dts"
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echo "pp support $pp"
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echo "pp support $pp"
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echo "debug symbols $debug"
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echo "debug symbols $debug"
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echo "optimize $optimize"
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echo "optimize $optimize"
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@ -1169,6 +1179,12 @@ if test "$a52" = "yes" ; then
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fi
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fi
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fi
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fi
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# DTS
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if test "$dts" = "yes" ; then
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echo "#define CONFIG_DTS 1" >> $TMPH
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echo "CONFIG_DTS=yes" >> config.mak
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fi
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# PP
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# PP
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if test "$pp" = "yes" ; then
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if test "$pp" = "yes" ; then
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echo "#define CONFIG_PP 1" >> $TMPH
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echo "#define CONFIG_PP 1" >> $TMPH
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10
ffmpeg.c
10
ffmpeg.c
@ -1502,8 +1502,9 @@ static int av_encode(AVFormatContext **output_files,
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ost->audio_resample = 0;
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ost->audio_resample = 0;
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} else {
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} else {
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if (codec->channels != icodec->channels &&
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if (codec->channels != icodec->channels &&
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icodec->codec_id == CODEC_ID_AC3) {
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(icodec->codec_id == CODEC_ID_AC3 ||
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/* Special case for 5:1 AC3 input */
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icodec->codec_id == CODEC_ID_DTS)) {
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/* Special case for 5:1 AC3 and DTS input */
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/* and mono or stereo output */
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/* and mono or stereo output */
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/* Request specific number of channels */
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/* Request specific number of channels */
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icodec->channels = codec->channels;
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icodec->channels = codec->channels;
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@ -3144,9 +3145,10 @@ static void opt_output_file(const char *filename)
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audio_enc->bit_rate = audio_bit_rate;
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audio_enc->bit_rate = audio_bit_rate;
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audio_enc->strict_std_compliance = strict;
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audio_enc->strict_std_compliance = strict;
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audio_enc->thread_count = thread_count;
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audio_enc->thread_count = thread_count;
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/* For audio codecs other than AC3 we limit */
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/* For audio codecs other than AC3 or DTS we limit */
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/* the number of coded channels to stereo */
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/* the number of coded channels to stereo */
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if (audio_channels > 2 && codec_id != CODEC_ID_AC3) {
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if (audio_channels > 2 && codec_id != CODEC_ID_AC3
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&& codec_id != CODEC_ID_DTS) {
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audio_enc->channels = 2;
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audio_enc->channels = 2;
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} else
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} else
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audio_enc->channels = audio_channels;
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audio_enc->channels = audio_channels;
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@ -73,6 +73,11 @@ OBJS+= liba52/bit_allocate.o liba52/bitstream.o liba52/downmix.o \
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endif
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endif
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endif
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endif
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# currently using libdts for dts decoding
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ifeq ($(CONFIG_DTS),yes)
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OBJS+= dtsdec.o
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endif
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ifeq ($(CONFIG_FAAD),yes)
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ifeq ($(CONFIG_FAAD),yes)
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OBJS+= faad.o
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OBJS+= faad.o
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ifeq ($(CONFIG_FAADBIN),yes)
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ifeq ($(CONFIG_FAADBIN),yes)
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@ -150,6 +150,9 @@ void avcodec_register_all(void)
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register_avcodec(&zlib_decoder);
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register_avcodec(&zlib_decoder);
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#ifdef CONFIG_AC3
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#ifdef CONFIG_AC3
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register_avcodec(&ac3_decoder);
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register_avcodec(&ac3_decoder);
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#endif
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#ifdef CONFIG_DTS
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register_avcodec(&dts_decoder);
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#endif
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#endif
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register_avcodec(&ra_144_decoder);
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register_avcodec(&ra_144_decoder);
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register_avcodec(&ra_288_decoder);
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register_avcodec(&ra_288_decoder);
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@ -38,6 +38,7 @@ static AVCodec* avcodec_find_by_fcc(uint32_t fcc)
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{ CODEC_ID_MJPEG, { MKTAG('M', 'J', 'P', 'G'), 0 } },
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{ CODEC_ID_MJPEG, { MKTAG('M', 'J', 'P', 'G'), 0 } },
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{ CODEC_ID_MPEG1VIDEO, { MKTAG('P', 'I', 'M', '1'), 0 } },
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{ CODEC_ID_MPEG1VIDEO, { MKTAG('P', 'I', 'M', '1'), 0 } },
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{ CODEC_ID_AC3, { 0x2000, 0 } },
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{ CODEC_ID_AC3, { 0x2000, 0 } },
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{ CODEC_ID_DTS, { 0x10, 0 } },
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{ CODEC_ID_MP2, { 0x50, 0x55, 0 } },
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{ CODEC_ID_MP2, { 0x50, 0x55, 0 } },
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{ CODEC_ID_FLV1, { MKTAG('F', 'L', 'V', '1'), 0 } },
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{ CODEC_ID_FLV1, { MKTAG('F', 'L', 'V', '1'), 0 } },
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@ -140,6 +140,8 @@ enum CodecID {
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CODEC_ID_MPEG2TS, /* _FAKE_ codec to indicate a raw MPEG2 transport
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CODEC_ID_MPEG2TS, /* _FAKE_ codec to indicate a raw MPEG2 transport
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stream (only used by libavformat) */
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stream (only used by libavformat) */
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CODEC_ID_DTS,
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};
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};
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/* CODEC_ID_MP3LAME is absolete */
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/* CODEC_ID_MP3LAME is absolete */
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@ -1858,6 +1860,7 @@ extern AVCodec rawvideo_decoder;
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/* the following codecs use external GPL libs */
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/* the following codecs use external GPL libs */
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extern AVCodec ac3_decoder;
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extern AVCodec ac3_decoder;
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extern AVCodec dts_decoder;
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/* resample.c */
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/* resample.c */
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203
libavcodec/dts_internal.h
Normal file
203
libavcodec/dts_internal.h
Normal file
@ -0,0 +1,203 @@
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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))
|
||||||
|
#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
libavcodec/dtsdec.c
Normal file
349
libavcodec/dtsdec.c
Normal 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,
|
||||||
|
};
|
@ -2228,6 +2228,9 @@ matroska_read_header (AVFormatContext *s,
|
|||||||
else if (!strcmp(track->codec_id,
|
else if (!strcmp(track->codec_id,
|
||||||
MATROSKA_CODEC_ID_AUDIO_AC3))
|
MATROSKA_CODEC_ID_AUDIO_AC3))
|
||||||
codec_id = CODEC_ID_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. */
|
/* No such codec id so far. */
|
||||||
/* else if (!strcmp(track->codec_id, */
|
/* else if (!strcmp(track->codec_id, */
|
||||||
/* MATROSKA_CODEC_ID_AUDIO_DTS)) */
|
/* MATROSKA_CODEC_ID_AUDIO_DTS)) */
|
||||||
|
@ -77,6 +77,7 @@ typedef struct {
|
|||||||
#define AUDIO_ID 0xc0
|
#define AUDIO_ID 0xc0
|
||||||
#define VIDEO_ID 0xe0
|
#define VIDEO_ID 0xe0
|
||||||
#define AC3_ID 0x80
|
#define AC3_ID 0x80
|
||||||
|
#define DTS_ID 0x8a
|
||||||
#define LPCM_ID 0xa0
|
#define LPCM_ID 0xa0
|
||||||
|
|
||||||
static const int lpcm_freq_tab[4] = { 48000, 96000, 44100, 32000 };
|
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)
|
static int mpeg_mux_init(AVFormatContext *ctx)
|
||||||
{
|
{
|
||||||
MpegMuxContext *s = ctx->priv_data;
|
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;
|
AVStream *st;
|
||||||
StreamInfo *stream;
|
StreamInfo *stream;
|
||||||
int audio_bitrate;
|
int audio_bitrate;
|
||||||
@ -258,6 +259,7 @@ static int mpeg_mux_init(AVFormatContext *ctx)
|
|||||||
s->video_bound = 0;
|
s->video_bound = 0;
|
||||||
mpa_id = AUDIO_ID;
|
mpa_id = AUDIO_ID;
|
||||||
ac3_id = AC3_ID;
|
ac3_id = AC3_ID;
|
||||||
|
dts_id = DTS_ID;
|
||||||
mpv_id = VIDEO_ID;
|
mpv_id = VIDEO_ID;
|
||||||
lpcm_id = LPCM_ID;
|
lpcm_id = LPCM_ID;
|
||||||
s->scr_stream_index = -1;
|
s->scr_stream_index = -1;
|
||||||
@ -272,6 +274,8 @@ static int mpeg_mux_init(AVFormatContext *ctx)
|
|||||||
case CODEC_TYPE_AUDIO:
|
case CODEC_TYPE_AUDIO:
|
||||||
if (st->codec.codec_id == CODEC_ID_AC3) {
|
if (st->codec.codec_id == CODEC_ID_AC3) {
|
||||||
stream->id = ac3_id++;
|
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) {
|
} else if (st->codec.codec_id == CODEC_ID_PCM_S16BE) {
|
||||||
stream->id = lpcm_id++;
|
stream->id = lpcm_id++;
|
||||||
for(j = 0; j < 4; j++) {
|
for(j = 0; j < 4; j++) {
|
||||||
@ -1304,9 +1308,12 @@ static int mpegps_read_packet(AVFormatContext *s,
|
|||||||
} else if (startcode >= 0x1c0 && startcode <= 0x1df) {
|
} else if (startcode >= 0x1c0 && startcode <= 0x1df) {
|
||||||
type = CODEC_TYPE_AUDIO;
|
type = CODEC_TYPE_AUDIO;
|
||||||
codec_id = CODEC_ID_MP2;
|
codec_id = CODEC_ID_MP2;
|
||||||
} else if (startcode >= 0x80 && startcode <= 0x9f) {
|
} else if (startcode >= 0x80 && startcode <= 0x89) {
|
||||||
type = CODEC_TYPE_AUDIO;
|
type = CODEC_TYPE_AUDIO;
|
||||||
codec_id = CODEC_ID_AC3;
|
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) {
|
} else if (startcode >= 0xa0 && startcode <= 0xbf) {
|
||||||
type = CODEC_TYPE_AUDIO;
|
type = CODEC_TYPE_AUDIO;
|
||||||
codec_id = CODEC_ID_PCM_S16BE;
|
codec_id = CODEC_ID_PCM_S16BE;
|
||||||
|
@ -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_VIDEO_H264:
|
||||||
case STREAM_TYPE_AUDIO_AAC:
|
case STREAM_TYPE_AUDIO_AAC:
|
||||||
case STREAM_TYPE_AUDIO_AC3:
|
case STREAM_TYPE_AUDIO_AC3:
|
||||||
|
case STREAM_TYPE_AUDIO_DTS:
|
||||||
add_pes_stream(ts, pid, stream_type);
|
add_pes_stream(ts, pid, stream_type);
|
||||||
break;
|
break;
|
||||||
default:
|
default:
|
||||||
@ -753,6 +754,10 @@ static void mpegts_push_data(void *opaque,
|
|||||||
codec_type = CODEC_TYPE_AUDIO;
|
codec_type = CODEC_TYPE_AUDIO;
|
||||||
codec_id = CODEC_ID_AC3;
|
codec_id = CODEC_ID_AC3;
|
||||||
break;
|
break;
|
||||||
|
case STREAM_TYPE_AUDIO_DTS:
|
||||||
|
codec_type = CODEC_TYPE_AUDIO;
|
||||||
|
codec_id = CODEC_ID_DTS;
|
||||||
|
break;
|
||||||
default:
|
default:
|
||||||
if (code >= 0x1c0 && code <= 0x1df) {
|
if (code >= 0x1c0 && code <= 0x1df) {
|
||||||
codec_type = CODEC_TYPE_AUDIO;
|
codec_type = CODEC_TYPE_AUDIO;
|
||||||
|
@ -42,6 +42,7 @@
|
|||||||
#define STREAM_TYPE_VIDEO_H264 0x1b
|
#define STREAM_TYPE_VIDEO_H264 0x1b
|
||||||
|
|
||||||
#define STREAM_TYPE_AUDIO_AC3 0x81
|
#define STREAM_TYPE_AUDIO_AC3 0x81
|
||||||
|
#define STREAM_TYPE_AUDIO_DTS 0x8a
|
||||||
|
|
||||||
unsigned int mpegts_crc32(const uint8_t *data, int len);
|
unsigned int mpegts_crc32(const uint8_t *data, int len);
|
||||||
extern AVOutputFormat mpegts_mux;
|
extern AVOutputFormat mpegts_mux;
|
||||||
|
@ -184,6 +184,23 @@ static int ac3_read_header(AVFormatContext *s,
|
|||||||
return 0;
|
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 */
|
/* mpeg1/h263 input */
|
||||||
static int video_read_header(AVFormatContext *s,
|
static int video_read_header(AVFormatContext *s,
|
||||||
AVFormatParameters *ap)
|
AVFormatParameters *ap)
|
||||||
@ -300,6 +317,17 @@ AVOutputFormat ac3_oformat = {
|
|||||||
};
|
};
|
||||||
#endif //CONFIG_ENCODERS
|
#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 = {
|
AVInputFormat h261_iformat = {
|
||||||
"h261",
|
"h261",
|
||||||
"raw h261",
|
"raw h261",
|
||||||
@ -613,6 +641,8 @@ int raw_init(void)
|
|||||||
av_register_input_format(&ac3_iformat);
|
av_register_input_format(&ac3_iformat);
|
||||||
av_register_output_format(&ac3_oformat);
|
av_register_output_format(&ac3_oformat);
|
||||||
|
|
||||||
|
av_register_input_format(&dts_iformat);
|
||||||
|
|
||||||
av_register_input_format(&h261_iformat);
|
av_register_input_format(&h261_iformat);
|
||||||
|
|
||||||
av_register_input_format(&h263_iformat);
|
av_register_input_format(&h263_iformat);
|
||||||
|
@ -348,6 +348,7 @@ static int wav_read_seek(AVFormatContext *s,
|
|||||||
case CODEC_ID_MP2:
|
case CODEC_ID_MP2:
|
||||||
case CODEC_ID_MP3:
|
case CODEC_ID_MP3:
|
||||||
case CODEC_ID_AC3:
|
case CODEC_ID_AC3:
|
||||||
|
case CODEC_ID_DTS:
|
||||||
/* use generic seeking with dynamically generated indexes */
|
/* use generic seeking with dynamically generated indexes */
|
||||||
return -1;
|
return -1;
|
||||||
default:
|
default:
|
||||||
|
Loading…
Reference in New Issue
Block a user