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Merging GST_FFMPEG_NO_MIRROR branch to trunk
Original commit message from CVS: Merging GST_FFMPEG_NO_MIRROR branch to trunk
This commit is contained in:
Edward Hervey
parent
22e3b86e43
commit
f6186655c5
17 changed files with 705 additions and 89 deletions
101
ChangeLog
101
ChangeLog
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@ -1,3 +1,104 @@
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2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
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* configure.ac:
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Minor typo in disabling cavs decoder. Now compiles AND works on x86 32
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and 64 bits ! Time to merge :)
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2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
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* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
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(gst_ffmpeg_caps_to_codecid):
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* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_open):
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Handle VC-1 properly , which is handled differently from WMV3.
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2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
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* autogen.sh:
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Fix call to ffmpegrev
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* configure.ac:
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Re-apply -fPIC extra-cflag for ffmpeg while removing support for the
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flac encoder which is currently broken for x86/32bits with -fPIC.
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* ffmpegrev:
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Switch to latest upstream revision so we can have the split-up for
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flac mmx optimizations.
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2007-12-15 Sebastian Dröge <slomo@circular-chaos.org>
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Based on a patch by:
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Hans de Goede <j dot w dot r degoede at hhs dot nl>
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* ext/ffmpeg/gstffmpegcfg.c: (gst_ffmpeg_pre_me_get_type),
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(gst_ffmpeg_pred_method_get_type):
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NULL-terminate the GEnumValue arrays, otherwise they will cause
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crashes. Fixes #503733.
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2007-12-09 Sebastian Dröge <slomo@circular-chaos.org>
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* configure.ac:
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Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
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2007-11-28 Edward Hervey <bilboed@bilboed.com>
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* configure.ac:
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Temporarily remove fPIC option for building ffmpeg
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* ffmpegrev:
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Switch to latest ffmpeg checkout so people can enjoy all the niceties
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that have just landed (mmx optimisations for vc1 and h263, bugfixes,..)
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* ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_collected),
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(gst_ffmpegmux_change_state):
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Adjust code to ffmpeg API changes.
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2007-11-22 Edward Hervey <bilboed@bilboed.com>
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* ext/ffmpeg/gstffmpegaudioresample.c:
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(gst_ffmpegaudioresample_get_unit_size),
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(gst_ffmpegaudioresample_transform):
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Correct the output size of the buffer.
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2007-11-15 Edward Hervey <bilboed@bilboed.com>
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* ext/ffmpeg/Makefile.am:
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* ext/ffmpeg/gstffmpeg.c: (plugin_init):
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* ext/ffmpeg/gstffmpeg.h:
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* ext/ffmpeg/gstffmpegaudioresample.c:
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(gst_ffmpegaudioresample_base_init),
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(gst_ffmpegaudioresample_class_init),
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(gst_ffmpegaudioresample_init), (gst_ffmpegaudioresample_finalize),
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(gst_ffmpegaudioresample_transform_caps),
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(gst_ffmpegaudioresample_transform_size),
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(gst_ffmpegaudioresample_get_unit_size),
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(gst_ffmpegaudioresample_set_caps),
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(gst_ffmpegaudioresample_transform),
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(gst_ffmpegaudioresample_register):
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Added new ffaudioresample element using the ffmpeg resampling code.
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It's (way) faster than audioresample, doesn't introduce latency, but
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might cause a little bit of 'clicking'.
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2007-11-15 Edward Hervey <bilboed@bilboed.com>
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* Makefile.am:
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* autogen.sh:
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* configure.ac:
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* ext/ffmpeg/Makefile.am:
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* ffmpegrev:
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* gst-libs/ext/Makefile.am:
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Initial patch of the new mirror-less build-system for gst-ffmpeg using
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specific revisions of ffmpeg svn instead.
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Might still have some issues, we need people to try this.
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Help by : Dejan Sakelšak <sakdean at gmail dot com>
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* ext/ffmpeg/gstffmpeg.c: (plugin_init):
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* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
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(gst_ffmpeg_caps_with_codecid), (gst_ffmpeg_caps_to_codecid),
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(gst_ffmpeg_get_codecid_longname):
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* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_base_init),
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(gst_ffmpegdec_get_buffer), (gst_ffmpegdec_audio_frame),
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(gst_ffmpegdec_register):
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* ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_averror),
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(gst_ffmpegdemux_register):
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* ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_register):
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* ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_register):
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Update code for new ffmpeg API.
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2007-12-04 Edward Hervey <bilboed@bilboed.com>
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* configure.ac:
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@ -8,7 +8,7 @@ debug:
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echo $(win32)
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EXTRA_DIST = \
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gst-ffmpeg.spec depcomp \
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gst-ffmpeg.spec depcomp ffmpegrev \
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AUTHORS COPYING NEWS README ChangeLog gst-ffmpeg.doap \
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$(win32)
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45
autogen.sh
45
autogen.sh
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@ -5,6 +5,9 @@ DIE=0
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package=gst-ffmpeg
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srcfile=configure.ac
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# FFMPEG specific properties
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. ./ffmpegrev
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# a quick cvs co if necessary to alleviate the pain - may remove this
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# when developers get a clue ;)
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if test ! -d common;
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@ -13,11 +16,16 @@ then
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cvs co common
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fi
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if test ! -f gst-libs/ext/ffmpeg/autogen.sh
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if test ! -f $FFMPEG_CO_DIR/configure
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then
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rm -rf gst-libs/ext/ffmpeg
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echo "+ getting ffmpeg from cvs"
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cvs co mirror-ffmpeg
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# checkout ffmpeg from its repository
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rm -rf $FFMPEG_CO_DIR
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echo "+ getting ffmpeg from svn"
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svn -r $FFMPEG_REVISION co $FFMPEG_SVN $FFMPEG_CO_DIR
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else
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# update ffmpeg from its repository
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echo "+ updating ffmpeg checkout"
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svn -r $FFMPEG_REVISION up $FFMPEG_CO_DIR
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fi
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@ -30,7 +38,23 @@ then
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fi
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. common/gst-autogen.sh
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CONFIGURE_DEF_OPT='--enable-maintainer-mode --enable-gtk-doc'
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# Let's check if we can disable the building of the ffmpeg binary
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can_disable=`$FFMPEG_CO_DIR/configure --help | grep 'disable-ffmpeg' | wc -l`
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if [ $can_disable != "0" ]
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then
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CONFIGURE_DEF_OPT="--disable-ffmpeg"
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fi
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# Let's clear the 'exit 1' command when we post an Unknown option
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echo "Patching ffmpeg ./configure"
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sed -e '/Unknown option/ {
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N
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N
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s/exit 1/#/
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}' $FFMPEG_CO_DIR/configure > $FFMPEG_CO_DIR/configure.tmp
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mv $FFMPEG_CO_DIR/configure.tmp $FFMPEG_CO_DIR/configure
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chmod +x $FFMPEG_CO_DIR/configure
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autogen_options $@
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@ -88,17 +112,6 @@ if test -f disable; then
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done
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fi
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# remove ffmpeg's configure, it's going to get created anyway and it probably
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# conflicted before this too
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rm -f gst-libs/ext/ffmpeg/configure
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# now, run ffmpeg's autogen
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echo "+ running autogen.sh in gst-libs/ext/ffmpeg"
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cd gst-libs/ext/ffmpeg
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chmod +x autogen.sh
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./autogen.sh || exit 1
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cd ../../..
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test -n "$NOCONFIGURE" && {
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echo "+ skipping configure stage for package $package, as requested."
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echo "+ autogen.sh done."
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2
common
2
common
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@ -1 +1 @@
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Subproject commit 423e2ea96b5f79281f4dd20d734bd968b3d95e89
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Subproject commit a00d4c1966aab517c2694c61d580489ebcbce448
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41
configure.ac
41
configure.ac
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@ -113,11 +113,14 @@ AM_CONDITIONAL(HAVE_GST_CHECK, test "x$HAVE_GST_CHECK" = "xyes")
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AC_MSG_NOTICE(Using GStreamer Core Plugins in $GST_PLUGINS_DIR)
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AC_MSG_NOTICE(Using GStreamer Base Plugins in $GSTPB_PLUGINS_DIR)
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dnl liboil is required
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dnl liboil is required for cpu detection for libpostproc
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dnl FIXME : In theory we should be able to compile libpostproc with cpudetect
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dnl capabilities, which would enable us to get rid of this
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PKG_CHECK_MODULES(LIBOIL, liboil-$LIBOIL_MAJORMINOR >= $LIBOIL_REQ, HAVE_LIBOIL=yes, HAVE_LIBOIL=no)
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if test "x$HAVE_LIBOIL" != "xyes"
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then
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AC_ERROR([liboil-$LIBOIL_REQ or later is required])
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AC_MSG_ERROR([liboil-$LIBOIL_REQ or later is required])
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AC_ERROR
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fi
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AC_SUBST(LIBOIL_CFLAGS)
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@ -150,7 +153,11 @@ if test "x$USE_DEBUG" = xyes; then
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fi
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AC_SUBST(PROFILE_CFLAGS)
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if test "x$GST_CVS" = "xyes"; then
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DEPRECATED_CFLAGS="-DGST_DISABLE_DEPRECATED"
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else
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DEPRECATED_CFLAGS=""
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fi
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AC_SUBST(DEPRECATED_CFLAGS)
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dnl every flag in GST_OPTION_CFLAGS can be overridden at make time
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@ -235,25 +242,47 @@ if test "x$with_system_ffmpeg" = "xyes"; then
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dnl No, this is not too extreme, we want people to see and read the above
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sleep 15
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else
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source ./ffmpegrev
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AC_MSG_NOTICE([Using ffmpeg revision $FFMPEG_REVISION])
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dnl libgstffmpeg.la: include dirs
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FFMPEG_CFLAGS="-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavformat \
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-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavutil \
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-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavcodec"
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FFMPEG_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libavformat/libavformat.la"
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-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavcodec -Wno-deprecated-declarations"
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dnl libgstffmpeg.la: libs to statically link to
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FFMPEG_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libavformat/libavformat.a \
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\$(top_builddir)/gst-libs/ext/ffmpeg/libavcodec/libavcodec.a \
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\$(top_builddir)/gst-libs/ext/ffmpeg/libavutil/libavutil.a \
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-lz"
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dnl
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POSTPROC_CFLAGS="-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libpostproc \
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-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavformat \
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-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavutil \
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-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavcodec"
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POSTPROC_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libavcodec/libavcodec.la"
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dnl libgstpostproc.la: libs to statically link to
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POSTPROC_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libpostproc/libpostproc.a \
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\$(top_builddir)/gst-libs/ext/ffmpeg/libavutil/libavutil.a"
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FFMPEG_SUBDIRS=gst-libs
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AC_DEFINE(HAVE_AVI_H)
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AC_DEFINE([FFMPEG_SOURCE], ["local snapshot"], [Describes where the FFmpeg libraries come from.])
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ac_configure_args="$ac_configure_args --disable-v4l --disable-audio-oss --disable-dv1394 --disable-vhook --disable-ffmpeg --disable-ffserver --disable-ffplay --disable-sdltest --enable-pp"
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ac_configure_args="$ac_configure_args --disable-vhook --disable-ffserver --disable-ffplay --enable-pp --enable-gpl --enable-static --disable-shared --disable-encoder=flac --disable-decoder=cavs --extra-cflags=-fPIC"
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AC_SUBST(FFMPEG_CO_DIR)
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AC_SUBST(FFMPEG_SVN)
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AC_SUBST(FFMPEG_REVISION)
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AC_CONFIG_SUBDIRS(gst-libs/ext/ffmpeg)
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AC_MSG_NOTICE([Using included FFMpeg code])
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fi
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AC_SUBST(FFMPEG_CFLAGS)
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AC_SUBST(FFMPEG_LIBS)
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AC_SUBST(FFMPEG_SUBDIRS)
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AC_SUBST(POSTPROC_CFLAGS)
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AC_SUBST(POSTPROC_LIBS)
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AC_DEFINE_UNQUOTED(HAVE_FFMPEG_UNINSTALLED, $HAVE_FFMPEG_UNINSTALLED,
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[Defined if building against uninstalled FFmpeg source])
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@ -9,7 +9,9 @@ libgstffmpeg_la_SOURCES = gstffmpeg.c \
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gstffmpegdemux.c \
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gstffmpegmux.c \
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gstffmpegdeinterlace.c \
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gstffmpegscale.c
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gstffmpegaudioresample.c
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# \
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# gstffmpegscale.c
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libgstffmpeg_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) \
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$(FFMPEG_CFLAGS)
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|
|
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@ -143,10 +143,13 @@ plugin_init (GstPlugin * plugin)
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gst_ffmpegdemux_register (plugin);
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gst_ffmpegmux_register (plugin);
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gst_ffmpegdeinterlace_register (plugin);
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#if 0
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gst_ffmpegscale_register (plugin);
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#endif
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#if 0
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gst_ffmpegcsp_register (plugin);
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#endif
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gst_ffmpegaudioresample_register (plugin);
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register_protocol (&gstreamer_protocol);
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|
|
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|
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@ -48,7 +48,10 @@ extern gboolean gst_ffmpegdec_register (GstPlugin * plugin);
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extern gboolean gst_ffmpegenc_register (GstPlugin * plugin);
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extern gboolean gst_ffmpegmux_register (GstPlugin * plugin);
|
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extern gboolean gst_ffmpegcsp_register (GstPlugin * plugin);
|
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#if 0
|
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extern gboolean gst_ffmpegscale_register (GstPlugin * plugin);
|
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#endif
|
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extern gboolean gst_ffmpegaudioresample_register (GstPlugin * plugin);
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extern gboolean gst_ffmpegdeinterlace_register (GstPlugin * plugin);
|
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|
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int gst_ffmpeg_avcodec_open (AVCodecContext *avctx, AVCodec *codec);
|
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|
|
|
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294
ext/ffmpeg/gstffmpegaudioresample.c
Normal file
294
ext/ffmpeg/gstffmpegaudioresample.c
Normal file
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|
@ -0,0 +1,294 @@
|
|||
/* GStreamer
|
||||
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
|
||||
* This file:
|
||||
* Copyright (C) 2005 Luca Ognibene <luogni@tin.it>
|
||||
* Copyright (C) 2006 Martin Zlomek <martin.zlomek@itonis.tv>
|
||||
*
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library 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
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library 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_CONFIG_H
|
||||
#include "config.h"
|
||||
#endif
|
||||
|
||||
#ifdef HAVE_FFMPEG_UNINSTALLED
|
||||
#include <avcodec.h>
|
||||
#else
|
||||
#include <ffmpeg/avcodec.h>
|
||||
#endif
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/base/gstbasetransform.h>
|
||||
#include <gst/video/video.h>
|
||||
|
||||
#include "gstffmpeg.h"
|
||||
#include "gstffmpegcodecmap.h"
|
||||
|
||||
typedef struct _GstFFMpegAudioResample
|
||||
{
|
||||
GstBaseTransform element;
|
||||
|
||||
GstPad *sinkpad, *srcpad;
|
||||
|
||||
gint in_rate, out_rate;
|
||||
gint in_channels, out_channels;
|
||||
|
||||
ReSampleContext *res;
|
||||
} GstFFMpegAudioResample;
|
||||
|
||||
typedef struct _GstFFMpegAudioResampleClass
|
||||
{
|
||||
GstBaseTransformClass parent_class;
|
||||
} GstFFMpegAudioResampleClass;
|
||||
|
||||
#define GST_TYPE_FFMPEGAUDIORESAMPLE \
|
||||
(gst_ffmpegaudioresample_get_type())
|
||||
#define GST_FFMPEGAUDIORESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_FFMPEGAUDIORESAMPLE,GstFFMpegAudioResample))
|
||||
#define GST_FFMPEGAUDIORESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_FFMPEGAUDIORESAMPLE,GstFFMpegAudioResampleClass))
|
||||
#define GST_IS_FFMPEGAUDIORESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_FFMPEGAUDIORESAMPLE))
|
||||
#define GST_IS_FFMPEGAUDIORESAMPLE_CLASS(klass) \
|
||||
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_FFMPEGAUDIORESAMPLE))
|
||||
|
||||
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
|
||||
GST_PAD_SRC,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS ("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) { 1 , 2 }, rate = (int) [1, MAX ]")
|
||||
);
|
||||
|
||||
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
|
||||
GST_PAD_SINK,
|
||||
GST_PAD_ALWAYS,
|
||||
GST_STATIC_CAPS ("audio/x-raw-int, endianness = (int) BYTE_ORDER, signed = (boolean) true, width = (int) 16, depth = (int) 16, channels = (int) [ 1 , 6 ], rate = (int) [1, MAX ]")
|
||||
);
|
||||
|
||||
GST_BOILERPLATE (GstFFMpegAudioResample, gst_ffmpegaudioresample, GstBaseTransform,
|
||||
GST_TYPE_BASE_TRANSFORM);
|
||||
|
||||
static void gst_ffmpegaudioresample_finalize (GObject * object);
|
||||
|
||||
static GstCaps *gst_ffmpegaudioresample_transform_caps (GstBaseTransform * trans,
|
||||
GstPadDirection direction, GstCaps * caps);
|
||||
static gboolean gst_ffmpegaudioresample_transform_size (GstBaseTransform * trans,
|
||||
GstPadDirection direction, GstCaps * caps, guint size, GstCaps *othercaps,
|
||||
guint * othersize);
|
||||
static gboolean gst_ffmpegaudioresample_get_unit_size (GstBaseTransform * trans,
|
||||
GstCaps * caps, guint * size);
|
||||
static gboolean gst_ffmpegaudioresample_set_caps (GstBaseTransform * trans,
|
||||
GstCaps * incaps, GstCaps * outcaps);
|
||||
static GstFlowReturn gst_ffmpegaudioresample_transform (GstBaseTransform * trans,
|
||||
GstBuffer * inbuf, GstBuffer * outbuf);
|
||||
|
||||
static void
|
||||
gst_ffmpegaudioresample_base_init (gpointer g_class)
|
||||
{
|
||||
static GstElementDetails plugin_details = {
|
||||
"FFMPEG Audio resampling element",
|
||||
"Filter/Converter/Audio",
|
||||
"Converts audio from one samplerate to another",
|
||||
"Edward Hervey <bilboed@bilboed.com>",
|
||||
};
|
||||
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
||||
|
||||
gst_element_class_add_pad_template (element_class,
|
||||
gst_static_pad_template_get (&src_factory));
|
||||
gst_element_class_add_pad_template (element_class,
|
||||
gst_static_pad_template_get (&sink_factory));
|
||||
gst_element_class_set_details (element_class, &plugin_details);
|
||||
}
|
||||
|
||||
static void
|
||||
gst_ffmpegaudioresample_class_init (GstFFMpegAudioResampleClass * klass)
|
||||
{
|
||||
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
|
||||
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
|
||||
|
||||
gobject_class->finalize = gst_ffmpegaudioresample_finalize;
|
||||
|
||||
trans_class->transform_caps =
|
||||
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform_caps);
|
||||
trans_class->get_unit_size =
|
||||
GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_get_unit_size);
|
||||
trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_set_caps);
|
||||
trans_class->transform = GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform);
|
||||
trans_class->transform_size = GST_DEBUG_FUNCPTR (gst_ffmpegaudioresample_transform_size);
|
||||
|
||||
trans_class->passthrough_on_same_caps = TRUE;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_ffmpegaudioresample_init (GstFFMpegAudioResample * resample, GstFFMpegAudioResampleClass * klass)
|
||||
{
|
||||
GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
|
||||
|
||||
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
|
||||
|
||||
resample->res = NULL;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_ffmpegaudioresample_finalize (GObject * object)
|
||||
{
|
||||
GstFFMpegAudioResample *resample = GST_FFMPEGAUDIORESAMPLE (object);
|
||||
|
||||
if (resample->res != NULL)
|
||||
audio_resample_close (resample->res);
|
||||
|
||||
G_OBJECT_CLASS (parent_class)->finalize (object);
|
||||
}
|
||||
|
||||
static GstCaps *
|
||||
gst_ffmpegaudioresample_transform_caps (GstBaseTransform * trans,
|
||||
GstPadDirection direction, GstCaps * caps)
|
||||
{
|
||||
GstCaps *retcaps;
|
||||
GstStructure * struc;
|
||||
|
||||
retcaps = gst_caps_copy (caps);
|
||||
struc = gst_caps_get_structure (retcaps, 0);
|
||||
gst_structure_set (struc, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
|
||||
|
||||
GST_LOG_OBJECT (trans, "returning caps %" GST_PTR_FORMAT,
|
||||
retcaps);
|
||||
|
||||
return retcaps;
|
||||
}
|
||||
|
||||
static gboolean gst_ffmpegaudioresample_transform_size (GstBaseTransform * trans,
|
||||
GstPadDirection direction, GstCaps * caps, guint size, GstCaps *othercaps,
|
||||
guint * othersize)
|
||||
{
|
||||
gint inrate, outrate;
|
||||
gint inchanns, outchanns;
|
||||
GstStructure *ins, *outs;
|
||||
gboolean ret;
|
||||
guint64 conv;
|
||||
|
||||
ins = gst_caps_get_structure (caps, 0);
|
||||
outs = gst_caps_get_structure (othercaps, 0);
|
||||
|
||||
/* Get input/output sample rate and channels */
|
||||
ret = gst_structure_get_int (ins, "rate", &inrate);
|
||||
ret &= gst_structure_get_int (ins, "channels", &inchanns);
|
||||
ret &= gst_structure_get_int (outs, "rate", &outrate);
|
||||
ret &= gst_structure_get_int (outs, "channels", &outchanns);
|
||||
|
||||
if (!ret)
|
||||
return FALSE;
|
||||
|
||||
conv = gst_util_uint64_scale(size, outrate * outchanns,
|
||||
inrate * inchanns);
|
||||
*othersize = (guint) conv;
|
||||
|
||||
GST_DEBUG_OBJECT (trans, "Transformed size from %d to %d",
|
||||
size, *othersize);
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_ffmpegaudioresample_get_unit_size (GstBaseTransform * trans, GstCaps * caps,
|
||||
guint * size)
|
||||
{
|
||||
gint channels;
|
||||
GstStructure * structure;
|
||||
gboolean ret;
|
||||
|
||||
g_assert (size);
|
||||
|
||||
structure = gst_caps_get_structure (caps, 0);
|
||||
ret = gst_structure_get_int (structure, "channels", &channels);
|
||||
g_return_val_if_fail (ret, FALSE);
|
||||
|
||||
*size = 2 * channels;
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_ffmpegaudioresample_set_caps (GstBaseTransform * trans, GstCaps * incaps,
|
||||
GstCaps * outcaps)
|
||||
{
|
||||
GstFFMpegAudioResample *resample = GST_FFMPEGAUDIORESAMPLE (trans);
|
||||
GstStructure *instructure = gst_caps_get_structure (incaps, 0);
|
||||
GstStructure *outstructure = gst_caps_get_structure (outcaps, 0);
|
||||
|
||||
GST_LOG_OBJECT (resample, "incaps:%"GST_PTR_FORMAT,
|
||||
incaps);
|
||||
|
||||
GST_LOG_OBJECT (resample, "outcaps:%"GST_PTR_FORMAT,
|
||||
outcaps);
|
||||
|
||||
if (!gst_structure_get_int (instructure, "channels", &resample->in_channels))
|
||||
return FALSE;
|
||||
if (!gst_structure_get_int (instructure, "rate", &resample->in_rate))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_structure_get_int (outstructure, "channels", &resample->out_channels))
|
||||
return FALSE;
|
||||
if (!gst_structure_get_int (outstructure, "rate", &resample->out_rate))
|
||||
return FALSE;
|
||||
|
||||
resample->res = audio_resample_init (resample->out_channels, resample->in_channels,
|
||||
resample->out_rate, resample->in_rate);
|
||||
if (resample->res == NULL)
|
||||
return FALSE;
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_ffmpegaudioresample_transform (GstBaseTransform * trans, GstBuffer * inbuf,
|
||||
GstBuffer * outbuf)
|
||||
{
|
||||
GstFFMpegAudioResample *resample = GST_FFMPEGAUDIORESAMPLE (trans);
|
||||
gint nbsamples;
|
||||
gint ret;
|
||||
|
||||
gst_buffer_copy_metadata (outbuf, inbuf, GST_BUFFER_COPY_TIMESTAMPS);
|
||||
nbsamples = GST_BUFFER_SIZE (inbuf) / (2 * resample->in_channels);
|
||||
|
||||
GST_LOG_OBJECT (resample, "input buffer duration:%"GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
|
||||
|
||||
GST_DEBUG_OBJECT (resample, "audio_resample(ctx, output:%p [size:%d], input:%p [size:%d], nbsamples:%d",
|
||||
GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf),
|
||||
GST_BUFFER_DATA (inbuf), GST_BUFFER_SIZE (inbuf),
|
||||
nbsamples);
|
||||
|
||||
ret = audio_resample (resample->res, (short *) GST_BUFFER_DATA(outbuf),
|
||||
(short *) GST_BUFFER_DATA (inbuf), nbsamples);
|
||||
|
||||
GST_DEBUG_OBJECT (resample, "audio_resample returned %d", ret);
|
||||
|
||||
GST_BUFFER_DURATION(outbuf) = gst_util_uint64_scale (ret, GST_SECOND,
|
||||
resample->out_rate);
|
||||
GST_BUFFER_SIZE (outbuf) = ret * 2 * resample->out_channels;
|
||||
|
||||
GST_LOG_OBJECT (resample, "Output buffer duration:%"GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
|
||||
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
gboolean
|
||||
gst_ffmpegaudioresample_register (GstPlugin * plugin)
|
||||
{
|
||||
return gst_element_register (plugin, "ffaudioresample",
|
||||
GST_RANK_NONE, GST_TYPE_FFMPEGAUDIORESAMPLE);
|
||||
}
|
||||
|
|
@ -223,6 +223,7 @@ gst_ffmpeg_pre_me_get_type (void)
|
|||
{0, "Disabled", "off"},
|
||||
{1, "Only after I-frames", "key"},
|
||||
{2, "Always", "all"},
|
||||
{0, NULL, NULL}
|
||||
};
|
||||
|
||||
ffmpeg_pre_me_type =
|
||||
|
|
@ -243,6 +244,7 @@ gst_ffmpeg_pred_method_get_type (void)
|
|||
{FF_PRED_LEFT, "Left", "left"},
|
||||
{FF_PRED_PLANE, "Plane", "plane"},
|
||||
{FF_PRED_MEDIAN, "Median", "median"},
|
||||
{0, NULL, NULL}
|
||||
};
|
||||
|
||||
ffmpeg_pred_method =
|
||||
|
|
|
|||
|
|
@ -193,13 +193,10 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
|
||||
switch (codec_id) {
|
||||
case CODEC_ID_MPEG1VIDEO:
|
||||
/* For decoding, CODEC_ID_MPEG2VIDEO is preferred... So omit here */
|
||||
if (encode) {
|
||||
/* FIXME: bitrate */
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "video/mpeg",
|
||||
"mpegversion", G_TYPE_INT, 1,
|
||||
"systemstream", G_TYPE_BOOLEAN, FALSE, NULL);
|
||||
}
|
||||
break;
|
||||
|
||||
case CODEC_ID_MPEG2VIDEO:
|
||||
|
|
@ -288,12 +285,9 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
break;
|
||||
|
||||
case CODEC_ID_MP2:
|
||||
/* we use CODEC_ID_MP3 for decoding */
|
||||
if (encode) {
|
||||
/* FIXME: bitrate */
|
||||
caps = gst_ff_aud_caps_new (context, codec_id, "audio/mpeg",
|
||||
"mpegversion", G_TYPE_INT, 1, "layer", G_TYPE_INT, 2, NULL);
|
||||
}
|
||||
break;
|
||||
|
||||
case CODEC_ID_MP3:
|
||||
|
|
@ -310,26 +304,15 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
}
|
||||
break;
|
||||
|
||||
case CODEC_ID_VORBIS:
|
||||
/* This one is disabled for several reasons:
|
||||
* - GStreamer already has perfect Ogg and Vorbis support
|
||||
* - The ffmpeg implementation depends on libvorbis/libogg,
|
||||
* which are not included in the ffmpeg that GStreamer ships.
|
||||
* - The ffmpeg implementation depends on shared objects between
|
||||
* the ogg demuxer and vorbis decoder, which GStreamer doesn't.
|
||||
*/
|
||||
break;
|
||||
|
||||
case CODEC_ID_AC3:
|
||||
/* Decoding is disabled, because:
|
||||
* - it depends on liba52, which we don't ship in ffmpeg.
|
||||
* - we already have a liba52 plugin ourselves.
|
||||
*/
|
||||
if (encode) {
|
||||
/* FIXME: bitrate */
|
||||
caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-ac3", NULL);
|
||||
}
|
||||
break;
|
||||
|
||||
case CODEC_ID_ATRAC3:
|
||||
caps = gst_ff_aud_caps_new (context, codec_id, "audio/atrac3", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_DTS:
|
||||
caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-dts", NULL);
|
||||
break;
|
||||
|
|
@ -554,12 +537,15 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
caps = gst_ff_vid_caps_new (context, codec_id, "video/x-vp6-flash", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_VP6A:
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "video/x-vp6-alpha", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_THEORA:
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "video/x-theora", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_AAC:
|
||||
case CODEC_ID_MPEG4AAC:
|
||||
caps = gst_ff_aud_caps_new (context, codec_id, "audio/mpeg",
|
||||
"mpegversion", G_TYPE_INT, 4, NULL);
|
||||
break;
|
||||
|
|
@ -645,10 +631,14 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
break;
|
||||
|
||||
case CODEC_ID_WMV3:
|
||||
case CODEC_ID_VC1:
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "video/x-wmv",
|
||||
"wmvversion", G_TYPE_INT, 3, NULL);
|
||||
break;
|
||||
case CODEC_ID_VC1:
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "video/x-wmv",
|
||||
"wmvversion", G_TYPE_INT, 3, "fourcc", GST_TYPE_FOURCC,
|
||||
GST_MAKE_FOURCC('W', 'V', 'C', '1'), NULL);
|
||||
break;
|
||||
case CODEC_ID_QDM2:
|
||||
caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-qdm2", NULL);
|
||||
break;
|
||||
|
|
@ -685,6 +675,14 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
}
|
||||
break;
|
||||
|
||||
case CODEC_ID_KMVC:
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "video/x-kmvc", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_NUV:
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "video/x-nuv", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_PNG:
|
||||
caps = gst_ff_vid_caps_new (context, codec_id, "image/png", NULL);
|
||||
break;
|
||||
|
|
@ -811,6 +809,7 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
case CODEC_ID_ADPCM_IMA_DK4:
|
||||
case CODEC_ID_ADPCM_IMA_WS:
|
||||
case CODEC_ID_ADPCM_IMA_SMJPEG:
|
||||
case CODEC_ID_ADPCM_IMA_AMV:
|
||||
case CODEC_ID_ADPCM_MS:
|
||||
case CODEC_ID_ADPCM_4XM:
|
||||
case CODEC_ID_ADPCM_XA:
|
||||
|
|
@ -823,6 +822,10 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
case CODEC_ID_ADPCM_SBPRO_2:
|
||||
case CODEC_ID_ADPCM_SBPRO_3:
|
||||
case CODEC_ID_ADPCM_SBPRO_4:
|
||||
case CODEC_ID_ADPCM_EA_R1:
|
||||
case CODEC_ID_ADPCM_EA_R2:
|
||||
case CODEC_ID_ADPCM_EA_R3:
|
||||
case CODEC_ID_ADPCM_THP:
|
||||
{
|
||||
gchar *layout = NULL;
|
||||
|
||||
|
|
@ -845,6 +848,9 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
case CODEC_ID_ADPCM_IMA_SMJPEG:
|
||||
layout = "smjpeg";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_IMA_AMV:
|
||||
layout = "amv";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_MS:
|
||||
layout = "microsoft";
|
||||
break;
|
||||
|
|
@ -881,6 +887,18 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
case CODEC_ID_ADPCM_SBPRO_4:
|
||||
layout = "sbpro4";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_EA_R1:
|
||||
layout = "ea-r1";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_EA_R2:
|
||||
layout = "ea-r3";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_EA_R3:
|
||||
layout = "ea-r3";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_THP:
|
||||
layout = "thp";
|
||||
break;
|
||||
default:
|
||||
g_assert (0); /* don't worry, we never get here */
|
||||
break;
|
||||
|
|
@ -905,6 +923,11 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
|
|||
caps = gst_ff_aud_caps_new (context, codec_id, "audio/AMR-WB", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_NELLYMOSER:
|
||||
caps =
|
||||
gst_ff_aud_caps_new (context, codec_id, "audio/x-nellymoser", NULL);
|
||||
break;
|
||||
|
||||
case CODEC_ID_RA_144:
|
||||
case CODEC_ID_RA_288:
|
||||
case CODEC_ID_COOK:
|
||||
|
|
@ -1571,7 +1594,7 @@ gst_ffmpeg_caps_with_codecid (enum CodecID codec_id,
|
|||
gst_structure_get_int (str, "unknown_svq3_flag",
|
||||
&unknown_svq3_flag)) {
|
||||
context->extradata = (guint8 *) av_mallocz (0x64);
|
||||
g_stpcpy (context->extradata, "SVQ3");
|
||||
g_stpcpy ((gchar *) context->extradata, "SVQ3");
|
||||
flags = 1 << 3;
|
||||
flags |= low_delay;
|
||||
flags = flags << 2;
|
||||
|
|
@ -2032,7 +2055,14 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
|
|||
id = CODEC_ID_WMV2;
|
||||
break;
|
||||
case 3:
|
||||
{
|
||||
guint32 fourcc;
|
||||
if (gst_structure_get_fourcc (structure, "fourcc", &fourcc)) {
|
||||
if (fourcc == GST_MAKE_FOURCC ('W', 'V', 'C', '1'))
|
||||
id = CODEC_ID_VC1;
|
||||
} else
|
||||
id = CODEC_ID_WMV3;
|
||||
}
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
|
@ -2088,6 +2118,9 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
|
|||
} else if (!strcmp (mimetype, "audio/x-ac3")) {
|
||||
id = CODEC_ID_AC3;
|
||||
audio = TRUE;
|
||||
} else if (!strcmp (mimetype, "audio/atrac3")) {
|
||||
id = CODEC_ID_ATRAC3;
|
||||
audio = TRUE;
|
||||
} else if (!strcmp (mimetype, "audio/x-dts")) {
|
||||
id = CODEC_ID_DTS;
|
||||
audio = TRUE;
|
||||
|
|
@ -2157,6 +2190,9 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
|
|||
} else if (!strcmp (mimetype, "video/x-vp6-flash")) {
|
||||
id = CODEC_ID_VP6F;
|
||||
video = TRUE;
|
||||
} else if (!strcmp (mimetype, "video/x-vp6-alpha")) {
|
||||
id = CODEC_ID_VP6A;
|
||||
video = TRUE;
|
||||
} else if (!strcmp (mimetype, "video/x-flash-screen")) {
|
||||
id = CODEC_ID_FLASHSV;
|
||||
video = TRUE;
|
||||
|
|
@ -2367,6 +2403,9 @@ gst_ffmpeg_caps_to_codecid (const GstCaps * caps, AVCodecContext * context)
|
|||
}
|
||||
}
|
||||
|
||||
} else if (!strcmp (mimetype, "audio/x-nellymoser")) {
|
||||
id = CODEC_ID_NELLYMOSER;
|
||||
audio = TRUE;
|
||||
} else if (!strncmp (mimetype, "audio/x-gst_ff-", 15)) {
|
||||
gchar ext[16];
|
||||
AVCodec *codec;
|
||||
|
|
@ -2465,6 +2504,9 @@ gst_ffmpeg_get_codecid_longname (enum CodecID codec_id)
|
|||
case CODEC_ID_AC3:
|
||||
name = "AC-3 audio";
|
||||
break;
|
||||
case CODEC_ID_ATRAC3:
|
||||
name = "Sony ATRAC-3";
|
||||
break;
|
||||
case CODEC_ID_DTS:
|
||||
name = "DTS Audio";
|
||||
break;
|
||||
|
|
@ -2564,6 +2606,9 @@ gst_ffmpeg_get_codecid_longname (enum CodecID codec_id)
|
|||
case CODEC_ID_VP6F:
|
||||
name = "VP6 Flash video";
|
||||
break;
|
||||
case CODEC_ID_VP6A:
|
||||
name = "VP6 Alpha video";
|
||||
break;
|
||||
case CODEC_ID_FLASHSV:
|
||||
name = "Flash Screen Video";
|
||||
break;
|
||||
|
|
@ -2571,7 +2616,6 @@ gst_ffmpeg_get_codecid_longname (enum CodecID codec_id)
|
|||
name = "Theora video";
|
||||
break;
|
||||
case CODEC_ID_AAC:
|
||||
case CODEC_ID_MPEG4AAC:
|
||||
name = "MPEG-2/4 AAC audio";
|
||||
break;
|
||||
case CODEC_ID_ASV1:
|
||||
|
|
@ -2745,6 +2789,12 @@ gst_ffmpeg_get_codecid_longname (enum CodecID codec_id)
|
|||
case CODEC_ID_ADPCM_IMA_SMJPEG:
|
||||
name = "IMA/SMJPEG ADPCM audio";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_IMA_AMV:
|
||||
name = "IMA/AMV ADPCM audio";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_THP:
|
||||
name = "Nintendo THP ADPCM audio";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_MS:
|
||||
name = "Microsoft ADPCM audio";
|
||||
break;
|
||||
|
|
@ -2781,6 +2831,15 @@ gst_ffmpeg_get_codecid_longname (enum CodecID codec_id)
|
|||
case CODEC_ID_ADPCM_YAMAHA:
|
||||
name = "Yamaha ADPCM";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_EA_R1:
|
||||
name = "EA ADPCM R1";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_EA_R2:
|
||||
name = "EA ADPCM R2";
|
||||
break;
|
||||
case CODEC_ID_ADPCM_EA_R3:
|
||||
name = "EA ADPCM R3";
|
||||
break;
|
||||
case CODEC_ID_RA_144:
|
||||
name = "Realaudio 14k4bps";
|
||||
break;
|
||||
|
|
@ -2850,6 +2909,15 @@ gst_ffmpeg_get_codecid_longname (enum CodecID codec_id)
|
|||
case CODEC_ID_AMR_WB:
|
||||
name = "3GPP AMR WideBand speech audio codec";
|
||||
break;
|
||||
case CODEC_ID_KMVC:
|
||||
name = "Karl Morton's video Codec";
|
||||
break;
|
||||
case CODEC_ID_NUV:
|
||||
name = "NuppelVideo codec";
|
||||
break;
|
||||
case CODEC_ID_NELLYMOSER:
|
||||
name = "Nellymoser ASAO audio codec";
|
||||
break;
|
||||
default:
|
||||
GST_LOG ("Unknown codecID 0x%x", codec_id);
|
||||
break;
|
||||
|
|
|
|||
|
|
@ -230,7 +230,8 @@ gst_ffmpegdec_base_init (GstFFMpegDecClass * klass)
|
|||
details.description = g_strdup_printf ("FFMPEG %s decoder",
|
||||
params->in_plugin->name);
|
||||
details.author = "Wim Taymans <wim@fluendo.com>, "
|
||||
"Ronald Bultje <rbultje@ronald.bitfreak.net>";
|
||||
"Ronald Bultje <rbultje@ronald.bitfreak.net>, "
|
||||
"Edward Hervey <bilboed@bilboed.com>";
|
||||
gst_element_class_set_details (element_class, &details);
|
||||
g_free (details.longname);
|
||||
g_free (details.klass);
|
||||
|
|
@ -497,6 +498,7 @@ gst_ffmpegdec_open (GstFFMpegDec * ffmpegdec)
|
|||
case CODEC_ID_MPEG4:
|
||||
case CODEC_ID_MJPEG:
|
||||
case CODEC_ID_MP3:
|
||||
case CODEC_ID_VC1:
|
||||
GST_LOG_OBJECT (ffmpegdec, "not using parser, blacklisted codec");
|
||||
ffmpegdec->pctx = NULL;
|
||||
break;
|
||||
|
|
@ -693,6 +695,8 @@ gst_ffmpegdec_get_buffer (AVCodecContext * context, AVFrame * picture)
|
|||
|
||||
switch (context->codec_type) {
|
||||
case CODEC_TYPE_VIDEO:
|
||||
/* some ffmpeg video plugins don't see the point in setting codec_type ... */
|
||||
case CODEC_TYPE_UNKNOWN:
|
||||
avcodec_align_dimensions (context, &width, &height);
|
||||
|
||||
bufsize = avpicture_get_size (context->pix_fmt, width, height);
|
||||
|
|
@ -1586,7 +1590,7 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
|
|||
GstBuffer ** outbuf, GstFlowReturn * ret)
|
||||
{
|
||||
gint len = -1;
|
||||
gint have_data;
|
||||
gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
|
||||
|
||||
GST_DEBUG_OBJECT (ffmpegdec,
|
||||
"size:%d, ts:%" GST_TIME_FORMAT ", dur:%" GST_TIME_FORMAT
|
||||
|
|
@ -1602,7 +1606,7 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
|
|||
ffmpegdec->last_buffer = NULL;
|
||||
}
|
||||
|
||||
len = avcodec_decode_audio (ffmpegdec->context,
|
||||
len = avcodec_decode_audio2 (ffmpegdec->context,
|
||||
(int16_t *) GST_BUFFER_DATA (*outbuf), &have_data, data, size);
|
||||
GST_DEBUG_OBJECT (ffmpegdec,
|
||||
"Decode audio: len=%d, have_data=%d", len, have_data);
|
||||
|
|
@ -2242,10 +2246,18 @@ gst_ffmpegdec_register (GstPlugin * plugin)
|
|||
|
||||
in_plugin = first_avcodec;
|
||||
|
||||
GST_LOG ("Registering decoders");
|
||||
|
||||
while (in_plugin) {
|
||||
GstFFMpegDecClassParams *params;
|
||||
GstCaps *srccaps = NULL, *sinkcaps = NULL;
|
||||
gchar *type_name;
|
||||
gchar *plugin_name;
|
||||
|
||||
/* only decoders */
|
||||
if (!in_plugin->decode) {
|
||||
goto next;
|
||||
}
|
||||
|
||||
/* no quasi-codecs, please */
|
||||
if (in_plugin->id == CODEC_ID_RAWVIDEO ||
|
||||
|
|
@ -2254,30 +2266,46 @@ gst_ffmpegdec_register (GstPlugin * plugin)
|
|||
goto next;
|
||||
}
|
||||
|
||||
/* only decoders */
|
||||
if (!in_plugin->decode) {
|
||||
/* no codecs for which we're GUARANTEED to have better alternatives */
|
||||
/* MPEG1VIDEO : the mpeg2video decoder is preferred */
|
||||
/* MP2 : Use MP3 for decoding */
|
||||
if (!strcmp (in_plugin->name, "gif") ||
|
||||
!strcmp (in_plugin->name, "vorbis") ||
|
||||
!strcmp (in_plugin->name, "mpeg1video") ||
|
||||
!strcmp (in_plugin->name, "mp2")) {
|
||||
GST_LOG ("Ignoring decoder %s", in_plugin->name);
|
||||
goto next;
|
||||
}
|
||||
|
||||
/* name */
|
||||
if (!gst_ffmpeg_get_codecid_longname (in_plugin->id)) {
|
||||
GST_INFO ("Add decoder %s (%d) please", in_plugin->name, in_plugin->id);
|
||||
GST_WARNING ("Add a longname mapping for decoder %s (%d) please",
|
||||
in_plugin->name, in_plugin->id);
|
||||
goto next;
|
||||
}
|
||||
|
||||
/* first make sure we've got a supported type */
|
||||
sinkcaps = gst_ffmpeg_codecid_to_caps (in_plugin->id, NULL, FALSE);
|
||||
if (!sinkcaps) {
|
||||
GST_WARNING ("Couldn't get input caps for decoder '%s'", in_plugin->name);
|
||||
}
|
||||
if (in_plugin->type == CODEC_TYPE_VIDEO) {
|
||||
srccaps = gst_caps_from_string ("video/x-raw-rgb; video/x-raw-yuv");
|
||||
} else {
|
||||
srccaps =
|
||||
gst_ffmpeg_codectype_to_caps (in_plugin->type, NULL, in_plugin->id);
|
||||
}
|
||||
if (!sinkcaps || !srccaps)
|
||||
if (!sinkcaps || !srccaps) {
|
||||
GST_WARNING ("Couldn't get source or sink caps for decoder %s",
|
||||
in_plugin->name);
|
||||
goto next;
|
||||
}
|
||||
|
||||
/* construct the type */
|
||||
type_name = g_strdup_printf ("ffdec_%s", in_plugin->name);
|
||||
plugin_name = g_strdup ((gchar *) in_plugin->name);
|
||||
g_strdelimit (plugin_name, NULL, '_');
|
||||
type_name = g_strdup_printf ("ffdec_%s", plugin_name);
|
||||
g_free (plugin_name);
|
||||
|
||||
/* if it's already registered, drop it */
|
||||
if (g_type_from_name (type_name)) {
|
||||
|
|
@ -2338,5 +2366,7 @@ gst_ffmpegdec_register (GstPlugin * plugin)
|
|||
in_plugin = in_plugin->next;
|
||||
}
|
||||
|
||||
GST_LOG ("Finished Registering decoders");
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -143,9 +143,6 @@ gst_ffmpegdemux_averror (gint av_errno)
|
|||
case AVERROR_NUMEXPECTED:
|
||||
message = "Number syntax expected in filename";
|
||||
break;
|
||||
case AVERROR_INVALIDDATA:
|
||||
message = "Invalid data found";
|
||||
break;
|
||||
case AVERROR_NOMEM:
|
||||
message = "Not enough memory";
|
||||
break;
|
||||
|
|
@ -1458,6 +1455,8 @@ gst_ffmpegdemux_register (GstPlugin * plugin)
|
|||
|
||||
in_plugin = first_iformat;
|
||||
|
||||
GST_LOG ("Registering demuxers");
|
||||
|
||||
while (in_plugin) {
|
||||
gchar *type_name, *typefind_name;
|
||||
gchar *p, *name = NULL;
|
||||
|
|
@ -1465,6 +1464,9 @@ gst_ffmpegdemux_register (GstPlugin * plugin)
|
|||
gint rank;
|
||||
gboolean register_typefind_func = TRUE;
|
||||
|
||||
GST_LOG ("Attempting to handle ffmpeg demuxer plugin %s [%s]",
|
||||
in_plugin->name, in_plugin->long_name);
|
||||
|
||||
/* no emulators */
|
||||
if (!strncmp (in_plugin->long_name, "raw ", 4) ||
|
||||
!strncmp (in_plugin->long_name, "pcm ", 4) ||
|
||||
|
|
@ -1546,6 +1548,7 @@ gst_ffmpegdemux_register (GstPlugin * plugin)
|
|||
/* Try to find the caps that belongs here */
|
||||
sinkcaps = gst_ffmpeg_formatid_to_caps (name);
|
||||
if (!sinkcaps) {
|
||||
GST_WARNING ("Couldn't get sinkcaps for demuxer %s", in_plugin->name);
|
||||
goto next;
|
||||
}
|
||||
/* This is a bit ugly, but we just take all formats
|
||||
|
|
@ -1604,5 +1607,7 @@ gst_ffmpegdemux_register (GstPlugin * plugin)
|
|||
in_plugin = in_plugin->next;
|
||||
}
|
||||
|
||||
GST_LOG ("Finished registering demuxers");
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -946,6 +946,8 @@ gst_ffmpegenc_register (GstPlugin * plugin)
|
|||
|
||||
in_plugin = first_avcodec;
|
||||
|
||||
GST_LOG ("Registering encoders");
|
||||
|
||||
/* build global ffmpeg param/property info */
|
||||
gst_ffmpeg_cfg_init ();
|
||||
|
||||
|
|
@ -967,9 +969,17 @@ gst_ffmpegenc_register (GstPlugin * plugin)
|
|||
goto next;
|
||||
}
|
||||
|
||||
/* no codecs for which we're GUARANTEED to have better alternatives */
|
||||
if (!strcmp (in_plugin->name, "vorbis") ||
|
||||
!strcmp (in_plugin->name, "gif") || !strcmp (in_plugin->name, "flac")) {
|
||||
GST_LOG ("Ignoring encoder %s", in_plugin->name);
|
||||
goto next;
|
||||
}
|
||||
|
||||
/* name */
|
||||
if (!gst_ffmpeg_get_codecid_longname (in_plugin->id)) {
|
||||
GST_INFO ("Add encoder %s (%d) please", in_plugin->name, in_plugin->id);
|
||||
GST_WARNING ("Add a longname mapping for encoder %s (%d) please",
|
||||
in_plugin->name, in_plugin->id);
|
||||
goto next;
|
||||
}
|
||||
|
||||
|
|
@ -982,9 +992,11 @@ gst_ffmpegenc_register (GstPlugin * plugin)
|
|||
sinkcaps =
|
||||
gst_ffmpeg_codectype_to_caps (in_plugin->type, NULL, in_plugin->id);
|
||||
}
|
||||
if (!sinkcaps || !srccaps)
|
||||
if (!sinkcaps || !srccaps) {
|
||||
GST_WARNING ("Couldn't get either source/sink caps for encoder %s",
|
||||
in_plugin->name);
|
||||
goto next;
|
||||
|
||||
}
|
||||
/* construct the type */
|
||||
type_name = g_strdup_printf ("ffenc_%s", in_plugin->name);
|
||||
|
||||
|
|
@ -1018,5 +1030,7 @@ gst_ffmpegenc_register (GstPlugin * plugin)
|
|||
in_plugin = in_plugin->next;
|
||||
}
|
||||
|
||||
GST_LOG ("Finished registering encoders");
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -481,7 +481,7 @@ gst_ffmpegmux_collected (GstCollectPads * pads, gpointer user_data)
|
|||
ffmpegmux->opened = TRUE;
|
||||
|
||||
/* flush the header so it will be used as streamheader */
|
||||
put_flush_packet (&ffmpegmux->context->pb);
|
||||
put_flush_packet (ffmpegmux->context->pb);
|
||||
}
|
||||
|
||||
/* take the one with earliest timestamp,
|
||||
|
|
@ -556,8 +556,8 @@ gst_ffmpegmux_collected (GstCollectPads * pads, gpointer user_data)
|
|||
/* close down */
|
||||
av_write_trailer (ffmpegmux->context);
|
||||
ffmpegmux->opened = FALSE;
|
||||
put_flush_packet (&ffmpegmux->context->pb);
|
||||
url_fclose (&ffmpegmux->context->pb);
|
||||
put_flush_packet (ffmpegmux->context->pb);
|
||||
url_fclose (ffmpegmux->context->pb);
|
||||
gst_pad_push_event (ffmpegmux->srcpad, gst_event_new_eos ());
|
||||
return GST_FLOW_UNEXPECTED;
|
||||
}
|
||||
|
|
@ -598,7 +598,7 @@ gst_ffmpegmux_change_state (GstElement * element, GstStateChange transition)
|
|||
}
|
||||
if (ffmpegmux->opened) {
|
||||
ffmpegmux->opened = FALSE;
|
||||
url_fclose (&ffmpegmux->context->pb);
|
||||
url_fclose (ffmpegmux->context->pb);
|
||||
}
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
|
|
@ -673,6 +673,8 @@ gst_ffmpegmux_register (GstPlugin * plugin)
|
|||
|
||||
in_plugin = first_oformat;
|
||||
|
||||
GST_LOG ("Registering muxers");
|
||||
|
||||
while (in_plugin) {
|
||||
gchar *type_name;
|
||||
gchar *p;
|
||||
|
|
@ -682,11 +684,15 @@ gst_ffmpegmux_register (GstPlugin * plugin)
|
|||
/* Try to find the caps that belongs here */
|
||||
srccaps = gst_ffmpeg_formatid_to_caps (in_plugin->name);
|
||||
if (!srccaps) {
|
||||
GST_WARNING ("Couldn't get source caps for muxer %s", in_plugin->name);
|
||||
goto next;
|
||||
}
|
||||
if (!gst_ffmpeg_formatid_get_codecids (in_plugin->name,
|
||||
&video_ids, &audio_ids)) {
|
||||
gst_caps_unref (srccaps);
|
||||
GST_WARNING
|
||||
("Couldn't get sink caps for muxer %s, mapping maybe missing ?",
|
||||
in_plugin->name);
|
||||
goto next;
|
||||
}
|
||||
videosinkcaps = video_ids ? gst_ffmpegmux_get_id_caps (video_ids) : NULL;
|
||||
|
|
@ -749,5 +755,7 @@ gst_ffmpegmux_register (GstPlugin * plugin)
|
|||
in_plugin = in_plugin->next;
|
||||
}
|
||||
|
||||
GST_LOG ("Finished registering muxers");
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
|
|
|||
4
ffmpegrev
Normal file
4
ffmpegrev
Normal file
|
|
@ -0,0 +1,4 @@
|
|||
# 6315, 10844, 10876, 10910, 10932, 10939
|
||||
FFMPEG_REVISION=11247
|
||||
FFMPEG_CO_DIR=gst-libs/ext/ffmpeg
|
||||
FFMPEG_SVN=svn://svn.mplayerhq.hu/ffmpeg/trunk
|
||||
|
|
@ -10,6 +10,46 @@
|
|||
|
||||
SUBDIRS =
|
||||
DIST_SUBDIRS = ffmpeg
|
||||
TMP_DIST_DIR=ffmpeg-dist
|
||||
DIST_DIR=$(TMP_DIST_DIR)/.ffmpeg
|
||||
|
||||
all-local:
|
||||
cd ffmpeg && $(MAKE)
|
||||
|
||||
clean-local:
|
||||
cd ffmpeg && $(MAKE) distclean
|
||||
|
||||
dist-clean:
|
||||
rm -rf $(TMP_DIST_DIR)
|
||||
|
||||
dist-local: dist-clean
|
||||
svn -r $(FFMPEG_REVISION) co $(FFMPEG_SVN) $(TMP_DIST_DIR)
|
||||
mkdir $(DIST_DIR)
|
||||
pwd
|
||||
cp $(TMP_DIST_DIR)/*.c $(TMP_DIST_DIR)/*.h $(TMP_DIST_DIR)/Makefile $(TMP_DIST_DIR)/configure $(TMP_DIST_DIR)/version.sh $(DIST_DIR)
|
||||
cp $(TMP_DIST_DIR)/common.mak $(TMP_DIST_DIR)/Changelog $(TMP_DIST_DIR)/COPYING.* $(TMP_DIST_DIR)/INSTALL $(DIST_DIR)
|
||||
cp $(TMP_DIST_DIR)/Doxyfile $(TMP_DIST_DIR)/ffinstall.nsi $(TMP_DIST_DIR)/CREDITS $(TMP_DIST_DIR)/MAINTAINERS $(DIST_DIR)
|
||||
@for d in `cd $(TMP_DIST_DIR) && ls -d */`; \
|
||||
do mkdir $(DIST_DIR)/$$d; \
|
||||
cp $(TMP_DIST_DIR)/$$d* $(DIST_DIR)/$$d; \
|
||||
if [ `ls -d $(TMP_DIST_DIR)/$$d*/ | wc -w` != "" ]; \
|
||||
then for id in `cd $(TMP_DIST_DIR)/$$d && ls -d */`; \
|
||||
do mkdir $(DIST_DIR)/$$d$$id; \
|
||||
cp $(TMP_DIST_DIR)/$$d$$id/*.c $(TMP_DIST_DIR)/$$d$$id/*.h $(DIST_DIR)/$$d$$id; \
|
||||
done \
|
||||
fi \
|
||||
done
|
||||
rm -rf ffmpeg
|
||||
mv $(DIST_DIR) ffmpeg
|
||||
echo "Patching ffmpeg ./configure"
|
||||
sed -e '/Unknown option/ {N;N;s/exit 1//; }' ffmpeg/configure > ffmpeg/configure.tmp
|
||||
mv ffmpeg/configure.tmp ffmpeg/configure
|
||||
chmod +x ffmpeg/configure
|
||||
rm -rf $(TMP_DIST_DIR)
|
||||
|
||||
distdir: dist-local
|
||||
cp -r ffmpeg Makefile* ${distdir}
|
||||
|
||||
dist: dist-local
|
||||
tar -czf ffmpeg.tar.gz ffmpeg
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue