+2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * configure.ac:
+ Minor typo in disabling cavs decoder. Now compiles AND works on x86 32
+ and 64 bits ! Time to merge :)
+
+2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
+ (gst_ffmpeg_caps_to_codecid):
+ * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_open):
+ Handle VC-1 properly , which is handled differently from WMV3.
+
+2007-12-17 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * autogen.sh:
+ Fix call to ffmpegrev
+ * configure.ac:
+ Re-apply -fPIC extra-cflag for ffmpeg while removing support for the
+ flac encoder which is currently broken for x86/32bits with -fPIC.
+ * ffmpegrev:
+ Switch to latest upstream revision so we can have the split-up for
+ flac mmx optimizations.
+
+2007-12-15 Sebastian Dröge <slomo@circular-chaos.org>
+
+ Based on a patch by:
+ Hans de Goede <j dot w dot r degoede at hhs dot nl>
+
+ * ext/ffmpeg/gstffmpegcfg.c: (gst_ffmpeg_pre_me_get_type),
+ (gst_ffmpeg_pred_method_get_type):
+ NULL-terminate the GEnumValue arrays, otherwise they will cause
+ crashes. Fixes #503733.
+
+2007-12-09 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
+
+2007-11-28 Edward Hervey <bilboed@bilboed.com>
+
+ * configure.ac:
+ Temporarily remove fPIC option for building ffmpeg
+ * ffmpegrev:
+ Switch to latest ffmpeg checkout so people can enjoy all the niceties
+ that have just landed (mmx optimisations for vc1 and h263, bugfixes,..)
+ * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_collected),
+ (gst_ffmpegmux_change_state):
+ Adjust code to ffmpeg API changes.
+
+2007-11-22 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/ffmpeg/gstffmpegaudioresample.c:
+ (gst_ffmpegaudioresample_get_unit_size),
+ (gst_ffmpegaudioresample_transform):
+ Correct the output size of the buffer.
+
+2007-11-15 Edward Hervey <bilboed@bilboed.com>
+
+ * ext/ffmpeg/Makefile.am:
+ * ext/ffmpeg/gstffmpeg.c: (plugin_init):
+ * ext/ffmpeg/gstffmpeg.h:
+ * ext/ffmpeg/gstffmpegaudioresample.c:
+ (gst_ffmpegaudioresample_base_init),
+ (gst_ffmpegaudioresample_class_init),
+ (gst_ffmpegaudioresample_init), (gst_ffmpegaudioresample_finalize),
+ (gst_ffmpegaudioresample_transform_caps),
+ (gst_ffmpegaudioresample_transform_size),
+ (gst_ffmpegaudioresample_get_unit_size),
+ (gst_ffmpegaudioresample_set_caps),
+ (gst_ffmpegaudioresample_transform),
+ (gst_ffmpegaudioresample_register):
+ Added new ffaudioresample element using the ffmpeg resampling code.
+ It's (way) faster than audioresample, doesn't introduce latency, but
+ might cause a little bit of 'clicking'.
+
+2007-11-15 Edward Hervey <bilboed@bilboed.com>
+
+ * Makefile.am:
+ * autogen.sh:
+ * configure.ac:
+ * ext/ffmpeg/Makefile.am:
+ * ffmpegrev:
+ * gst-libs/ext/Makefile.am:
+ Initial patch of the new mirror-less build-system for gst-ffmpeg using
+ specific revisions of ffmpeg svn instead.
+ Might still have some issues, we need people to try this.
+ Help by : Dejan Sakelšak <sakdean at gmail dot com>
+ * ext/ffmpeg/gstffmpeg.c: (plugin_init):
+ * ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_codecid_to_caps),
+ (gst_ffmpeg_caps_with_codecid), (gst_ffmpeg_caps_to_codecid),
+ (gst_ffmpeg_get_codecid_longname):
+ * ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_base_init),
+ (gst_ffmpegdec_get_buffer), (gst_ffmpegdec_audio_frame),
+ (gst_ffmpegdec_register):
+ * ext/ffmpeg/gstffmpegdemux.c: (gst_ffmpegdemux_averror),
+ (gst_ffmpegdemux_register):
+ * ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_register):
+ * ext/ffmpeg/gstffmpegmux.c: (gst_ffmpegmux_register):
+ Update code for new ffmpeg API.
+
2007-12-04 Edward Hervey <bilboed@bilboed.com>
* configure.ac:
echo $(win32)
EXTRA_DIST = \
- gst-ffmpeg.spec depcomp \
+ gst-ffmpeg.spec depcomp ffmpegrev \
AUTHORS COPYING NEWS README ChangeLog gst-ffmpeg.doap \
$(win32)
package=gst-ffmpeg
srcfile=configure.ac
+# FFMPEG specific properties
+. ./ffmpegrev
+
# a quick cvs co if necessary to alleviate the pain - may remove this
# when developers get a clue ;)
if test ! -d common;
cvs co common
fi
-if test ! -f gst-libs/ext/ffmpeg/autogen.sh
-then
- rm -rf gst-libs/ext/ffmpeg
- echo "+ getting ffmpeg from cvs"
- cvs co mirror-ffmpeg
+if test ! -f $FFMPEG_CO_DIR/configure
+then
+ # checkout ffmpeg from its repository
+ rm -rf $FFMPEG_CO_DIR
+ echo "+ getting ffmpeg from svn"
+ svn -r $FFMPEG_REVISION co $FFMPEG_SVN $FFMPEG_CO_DIR
+else
+ # update ffmpeg from its repository
+ echo "+ updating ffmpeg checkout"
+ svn -r $FFMPEG_REVISION up $FFMPEG_CO_DIR
fi
fi
. common/gst-autogen.sh
-CONFIGURE_DEF_OPT='--enable-maintainer-mode --enable-gtk-doc'
+# Let's check if we can disable the building of the ffmpeg binary
+can_disable=`$FFMPEG_CO_DIR/configure --help | grep 'disable-ffmpeg' | wc -l`
+
+if [ $can_disable != "0" ]
+then
+ CONFIGURE_DEF_OPT="--disable-ffmpeg"
+fi
+
+# Let's clear the 'exit 1' command when we post an Unknown option
+echo "Patching ffmpeg ./configure"
+sed -e '/Unknown option/ {
+N
+N
+s/exit 1/#/
+}' $FFMPEG_CO_DIR/configure > $FFMPEG_CO_DIR/configure.tmp
+mv $FFMPEG_CO_DIR/configure.tmp $FFMPEG_CO_DIR/configure
+chmod +x $FFMPEG_CO_DIR/configure
autogen_options $@
done
fi
-# remove ffmpeg's configure, it's going to get created anyway and it probably
-# conflicted before this too
-rm -f gst-libs/ext/ffmpeg/configure
-
-# now, run ffmpeg's autogen
-echo "+ running autogen.sh in gst-libs/ext/ffmpeg"
-cd gst-libs/ext/ffmpeg
-chmod +x autogen.sh
-./autogen.sh || exit 1
-cd ../../..
-
test -n "$NOCONFIGURE" && {
echo "+ skipping configure stage for package $package, as requested."
echo "+ autogen.sh done."
-Subproject commit 423e2ea96b5f79281f4dd20d734bd968b3d95e89
+Subproject commit a00d4c1966aab517c2694c61d580489ebcbce448
AC_MSG_NOTICE(Using GStreamer Core Plugins in $GST_PLUGINS_DIR)
AC_MSG_NOTICE(Using GStreamer Base Plugins in $GSTPB_PLUGINS_DIR)
-dnl liboil is required
+dnl liboil is required for cpu detection for libpostproc
+dnl FIXME : In theory we should be able to compile libpostproc with cpudetect
+dnl capabilities, which would enable us to get rid of this
PKG_CHECK_MODULES(LIBOIL, liboil-$LIBOIL_MAJORMINOR >= $LIBOIL_REQ, HAVE_LIBOIL=yes, HAVE_LIBOIL=no)
if test "x$HAVE_LIBOIL" != "xyes"
then
- AC_ERROR([liboil-$LIBOIL_REQ or later is required])
+ AC_MSG_ERROR([liboil-$LIBOIL_REQ or later is required])
+ AC_ERROR
fi
AC_SUBST(LIBOIL_CFLAGS)
fi
AC_SUBST(PROFILE_CFLAGS)
-DEPRECATED_CFLAGS="-DGST_DISABLE_DEPRECATED"
+if test "x$GST_CVS" = "xyes"; then
+ DEPRECATED_CFLAGS="-DGST_DISABLE_DEPRECATED"
+else
+ DEPRECATED_CFLAGS=""
+fi
AC_SUBST(DEPRECATED_CFLAGS)
dnl every flag in GST_OPTION_CFLAGS can be overridden at make time
dnl No, this is not too extreme, we want people to see and read the above
sleep 15
else
+
+ source ./ffmpegrev
+
+ AC_MSG_NOTICE([Using ffmpeg revision $FFMPEG_REVISION])
+
+ dnl libgstffmpeg.la: include dirs
FFMPEG_CFLAGS="-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavformat \
-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavutil \
- -I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavcodec"
- FFMPEG_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libavformat/libavformat.la"
+ -I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavcodec -Wno-deprecated-declarations"
+
+ dnl libgstffmpeg.la: libs to statically link to
+ FFMPEG_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libavformat/libavformat.a \
+ \$(top_builddir)/gst-libs/ext/ffmpeg/libavcodec/libavcodec.a \
+ \$(top_builddir)/gst-libs/ext/ffmpeg/libavutil/libavutil.a \
+ -lz"
+ dnl
POSTPROC_CFLAGS="-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libpostproc \
-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavformat \
-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavutil \
-I \$(top_srcdir)/gst-libs/ext/ffmpeg/libavcodec"
- POSTPROC_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libavcodec/libavcodec.la"
+
+ dnl libgstpostproc.la: libs to statically link to
+ POSTPROC_LIBS="\$(top_builddir)/gst-libs/ext/ffmpeg/libpostproc/libpostproc.a \
+ \$(top_builddir)/gst-libs/ext/ffmpeg/libavutil/libavutil.a"
+
FFMPEG_SUBDIRS=gst-libs
AC_DEFINE(HAVE_AVI_H)
AC_DEFINE([FFMPEG_SOURCE], ["local snapshot"], [Describes where the FFmpeg libraries come from.])
- ac_configure_args="$ac_configure_args --disable-v4l --disable-audio-oss --disable-dv1394 --disable-vhook --disable-ffmpeg --disable-ffserver --disable-ffplay --disable-sdltest --enable-pp"
+ 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"
+ AC_SUBST(FFMPEG_CO_DIR)
+ AC_SUBST(FFMPEG_SVN)
+ AC_SUBST(FFMPEG_REVISION)
AC_CONFIG_SUBDIRS(gst-libs/ext/ffmpeg)
AC_MSG_NOTICE([Using included FFMpeg code])
fi
+
AC_SUBST(FFMPEG_CFLAGS)
AC_SUBST(FFMPEG_LIBS)
AC_SUBST(FFMPEG_SUBDIRS)
-
+AC_SUBST(POSTPROC_CFLAGS)
+AC_SUBST(POSTPROC_LIBS)
+
AC_DEFINE_UNQUOTED(HAVE_FFMPEG_UNINSTALLED, $HAVE_FFMPEG_UNINSTALLED,
[Defined if building against uninstalled FFmpeg source])
AM_CONDITIONAL(HAVE_FFMPEG_UNINSTALLED, test x$HAVE_FFMPEG_UNINSTALLED = x1)
gstffmpegdemux.c \
gstffmpegmux.c \
gstffmpegdeinterlace.c \
- gstffmpegscale.c
+ gstffmpegaudioresample.c
+# \
+# gstffmpegscale.c
libgstffmpeg_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) \
$(FFMPEG_CFLAGS)
gst_ffmpegdemux_register (plugin);
gst_ffmpegmux_register (plugin);
gst_ffmpegdeinterlace_register (plugin);
+#if 0
gst_ffmpegscale_register (plugin);
+#endif
#if 0
gst_ffmpegcsp_register (plugin);
#endif
+ gst_ffmpegaudioresample_register (plugin);
register_protocol (&gstreamer_protocol);
extern gboolean gst_ffmpegenc_register (GstPlugin * plugin);
extern gboolean gst_ffmpegmux_register (GstPlugin * plugin);
extern gboolean gst_ffmpegcsp_register (GstPlugin * plugin);
+#if 0
extern gboolean gst_ffmpegscale_register (GstPlugin * plugin);
+#endif
+extern gboolean gst_ffmpegaudioresample_register (GstPlugin * plugin);
extern gboolean gst_ffmpegdeinterlace_register (GstPlugin * plugin);
int gst_ffmpeg_avcodec_open (AVCodecContext *avctx, AVCodec *codec);
--- /dev/null
+/* 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);
+}
static const GEnumValue ffmpeg_mb_decisions[] = {
{FF_MB_DECISION_SIMPLE, "Use method set by mb-cmp", "simple"},
{FF_MB_DECISION_BITS,
- "Chooses the one which needs the fewest bits aka vhq mode", "bits"},
+ "Chooses the one which needs the fewest bits aka vhq mode", "bits"},
{FF_MB_DECISION_RD, "Rate Distortion", "rd"},
{0, NULL, NULL},
};
{0, "Disabled", "off"},
{1, "Only after I-frames", "key"},
{2, "Always", "all"},
+ {0, NULL, NULL}
};
ffmpeg_pre_me_type =
{FF_PRED_LEFT, "Left", "left"},
{FF_PRED_PLANE, "Plane", "plane"},
{FF_PRED_MEDIAN, "Median", "median"},
+ {0, NULL, NULL}
};
ffmpeg_pred_method =
{CODEC_FLAG_GMC, "GMC", "gmc"},
{CODEC_FLAG_MV0, "Always try a MB with MV (0,0)", "mv0"},
{CODEC_FLAG_PART,
- "Store MV, DC and AC coefficients in seperate partitions", "part"},
+ "Store MV, DC and AC coefficients in seperate partitions", "part"},
{CODEC_FLAG_GRAY, "Only decode/encode grayscale", "gray"},
{CODEC_FLAG_NORMALIZE_AQP,
- "Normalize Adaptive Quantization (masking, etc)", "aqp"},
+ "Normalize Adaptive Quantization (masking, etc)", "aqp"},
{CODEC_FLAG_TRELLIS_QUANT, "Trellis Quantization", "trellis"},
{CODEC_FLAG_GLOBAL_HEADER,
"Global headers in extradata instead of every keyframe",
- "global-headers"},
+ "global-headers"},
{CODEC_FLAG_AC_PRED, "H263 Advanced Intra Coding / MPEG4 AC prediction",
- "aic"},
+ "aic"},
{CODEC_FLAG_H263P_UMV, "Unlimited Motion Vector", "umv"},
{CODEC_FLAG_CBP_RD, "Rate Distoration Optimization for CBP", "cbp-rd"},
{CODEC_FLAG_QP_RD, "Rate Distoration Optimization for QP selection",
- "qp-rd"},
+ "qp-rd"},
{CODEC_FLAG_SVCD_SCAN_OFFSET,
- "Reserve space for SVCD scan offset user data", "scanoffset"},
+ "Reserve space for SVCD scan offset user data", "scanoffset"},
{CODEC_FLAG_CLOSED_GOP, "Closed GOP", "closedgop"},
{0, NULL, NULL},
};
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);
- }
+ /* 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:
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);
- }
+ /* 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:
}
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.
- */
+ case CODEC_ID_AC3:
+ /* FIXME: bitrate */
+ caps = gst_ff_aud_caps_new (context, codec_id, "audio/x-ac3", NULL);
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);
- }
+ 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;
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;
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;
}
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;
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:
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;
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;
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;
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:
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;
id = CODEC_ID_WMV2;
break;
case 3:
- id = CODEC_ID_WMV3;
+ {
+ 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;
}
}
} 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;
} 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;
}
}
+ } 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;
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;
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;
name = "Theora video";
break;
case CODEC_ID_AAC:
- case CODEC_ID_MPEG4AAC:
name = "MPEG-2/4 AAC audio";
break;
case CODEC_ID_ASV1:
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;
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;
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;
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);
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;
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);
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
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);
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 ||
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)) {
in_plugin = in_plugin->next;
}
+ GST_LOG ("Finished Registering decoders");
+
return TRUE;
}
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;
in_plugin = first_iformat;
+ GST_LOG ("Registering demuxers");
+
while (in_plugin) {
gchar *type_name, *typefind_name;
gchar *p, *name = NULL;
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) ||
/* 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
in_plugin = in_plugin->next;
}
+ GST_LOG ("Finished registering demuxers");
+
return TRUE;
}
in_plugin = first_avcodec;
+ GST_LOG ("Registering encoders");
+
/* build global ffmpeg param/property info */
gst_ffmpeg_cfg_init ();
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;
}
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);
in_plugin = in_plugin->next;
}
+ GST_LOG ("Finished registering encoders");
+
return TRUE;
}
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,
/* 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;
}
}
if (ffmpegmux->opened) {
ffmpegmux->opened = FALSE;
- url_fclose (&ffmpegmux->context->pb);
+ url_fclose (ffmpegmux->context->pb);
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
in_plugin = first_oformat;
+ GST_LOG ("Registering muxers");
+
while (in_plugin) {
gchar *type_name;
gchar *p;
/* 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;
in_plugin = in_plugin->next;
}
+ GST_LOG ("Finished registering muxers");
+
return TRUE;
}
--- /dev/null
+# 6315, 10844, 10876, 10910, 10932, 10939
+FFMPEG_REVISION=11247
+FFMPEG_CO_DIR=gst-libs/ext/ffmpeg
+FFMPEG_SVN=svn://svn.mplayerhq.hu/ffmpeg/trunk
# - add an all-local hook so it does get built
# this also satisfies make distcheck
-SUBDIRS =
+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
+
projects / platform / upstream / gstreamer.git / commitdiff