2007-06-12 Jan Schmidt <thaytan@mad.scientist.com>
+
+ * configure.ac:
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-qtdemux.xml:
+ * docs/plugins/inspect/plugin-quicktime.xml:
+ * win32/MANIFEST:
+ Move qtdemux from -bad.
+
+ * gst-plugins-good.spec.in:
+ Update spec file to reflect moving of qtdemux and wavpack
+
+2007-06-12 Jan Schmidt <thaytan@mad.scientist.com>
* win32/MANIFEST:
* docs/plugins/Makefile.am:
matroska \
monoscope \
multipart \
+ qtdemux \
rtp \
rtsp \
smpte \
AC_SUBST(WAVPACK_LIBS)
])
-dnl *** id3demux prefers to have zlib ***
+dnl *** qtdemux & id3demux prefer to have zlib ***
translit(dnm, m, l) AM_CONDITIONAL(USE_ZLIB, true)
-AG_GST_CHECK_FEATURE(ZLIB, [zlib support for id3demux],, [
+AG_GST_CHECK_FEATURE(ZLIB, [zlib support for id3demux/qtdemux],, [
AG_GST_CHECK_LIBHEADER(ZLIB,
z, uncompress,, zlib.h, [
HAVE_ZLIB="yes"
gst/matroska/Makefile
gst/monoscope/Makefile
gst/multipart/Makefile
+gst/qtdemux/Makefile
gst/rtp/Makefile
gst/rtsp/Makefile
gst/smpte/Makefile
$(top_srcdir)/gst/level/gstlevel.h \
$(top_srcdir)/gst/multipart/multipartdemux.c \
$(top_srcdir)/gst/multipart/multipartmux.c \
+ $(top_srcdir)/gst/qtdemux/qtdemux.h \
$(top_srcdir)/gst/rtsp/gstrtpdec.h \
$(top_srcdir)/gst/rtsp/gstrtspsrc.h \
$(top_srcdir)/gst/udp/gstmultiudpsink.h \
<xi:include href="xml/element-osxaudiosink.xml" />
<xi:include href="xml/element-osxaudiosrc.xml" />
<xi:include href="xml/element-progressreport.xml" />
+ <xi:include href="xml/element-qtdemux.xml" />
<xi:include href="xml/element-rtspsrc.xml" />
<xi:include href="xml/element-rtpdec.xml" />
<xi:include href="xml/element-smokedec.xml" />
<xi:include href="xml/plugin-ossaudio.xml" />
<xi:include href="xml/plugin-osxaudio.xml" />
<xi:include href="xml/plugin-png.xml" />
+ <xi:include href="xml/plugin-quicktime.xml" />
<xi:include href="xml/plugin-rtp.xml" />
<xi:include href="xml/plugin-rtsp.xml" />
<xi:include href="xml/plugin-shout2send.xml" />
</SECTION>
<SECTION>
+<FILE>element-qtdemux</FILE>
+GstQTDemux
+<TITLE>qtdemux</TITLE>
+<SUBSECTION Standard>
+GstQTDemuxClass
+</SECTION>
+
+<SECTION>
<FILE>element-rtspsrc</FILE>
RTSPLowerTrans
GstRTSPSrc
<ARG>
<NAME>GstUDPSrc::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket to use for UDP reception. (-1 == allocate).</BLURB>
<ARG>
<NAME>GstDV1394Src::port</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,16]</RANGE>
+<RANGE>[G_MAXULONG,16]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>Port number (-1 automatic).</BLURB>
<ARG>
<NAME>GstTest::allowed-timestamp-deviation</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>allowed timestamp deviation</NICK>
<BLURB>allowed average difference in usec between timestamp of next buffer and expected timestamp from analyzing last buffer.</BLURB>
<ARG>
<NAME>GstTest::buffer-count</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>buffer count</NICK>
<BLURB>number of buffers in stream.</BLURB>
<ARG>
<NAME>GstTest::expected-buffer-count</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>expected buffer count</NICK>
<BLURB>expected number of buffers in stream.</BLURB>
<ARG>
<NAME>GstTest::expected-length</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>expected length</NICK>
<BLURB>expected length of stream.</BLURB>
<ARG>
<NAME>GstTest::length</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>length</NICK>
<BLURB>length of stream.</BLURB>
<ARG>
<NAME>GstTest::timestamp-deviation</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>timestamp deviation</NICK>
<BLURB>average difference in usec between timestamp of next buffer and expected timestamp from analyzing last buffer.</BLURB>
<ARG>
<NAME>GstBreakMyData::set-to</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,255]</RANGE>
+<RANGE>[G_MAXULONG,255]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>set-to</NICK>
<BLURB>set changed bytes to this value (-1 means random value.</BLURB>
<ARG>
<NAME>GstDynUDPSink::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,32767]</RANGE>
+<RANGE>[G_MAXULONG,32767]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>socket handle</NICK>
<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
<ARG>
<NAME>GstCdioCddaSrc::read-speed</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Read speed</NICK>
<BLURB>Read from device at the specified speed (-1 = default).</BLURB>
<plugin>
- <name>qtdemux</name>
- <description>Quicktime stream demuxer</description>
+ <name>quicktime</name>
+ <description>Quicktime support</description>
<filename>../../gst/qtdemux/.libs/libgstqtdemux.so</filename>
<basename>libgstqtdemux.so</basename>
- <version>0.10.4</version>
+ <version>0.10.5.1</version>
<license>LGPL</license>
- <source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <source>gst-plugins-good</source>
+ <package>GStreamer Good Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Demultiplex a QuickTime file into audio and video streams</description>
<author>David Schleef <ds@schleef.org>, Wim Taymans <wim@fluendo.com></author>
</element>
+ <element>
+ <name>rtpxqtdepay</name>
+ <longname>RTP packet depayloader</longname>
+ <class>Codec/Depayloader/Network</class>
+ <description>Extracts Quicktime audio/video from RTP packets</description>
+ <author>Wim Taymans <wim@fluendo.com></author>
+ </element>
</elements>
</plugin>
\ No newline at end of file
%{_libdir}/gstreamer-%{majorminor}/libgstlevel.so
%{_libdir}/gstreamer-%{majorminor}/libgstefence.so
%{_libdir}/gstreamer-%{majorminor}/libgstmulaw.so
+%{_libdir}/gstreamer-%{majorminor}/libgstqtdemux.so
%{_libdir}/gstreamer-%{majorminor}/libgstrtp.so
%{_libdir}/gstreamer-%{majorminor}/libgstrtsp.so
%{_libdir}/gstreamer-%{majorminor}/libgstsmpte.so
@USE_AALIB_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstaasink.so
@USE_LIBDV_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstdv.so
@USE_DV1394_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgst1394.so
+@USE_WAVPACK_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstwavpack.so
# schema files
@USE_GCONF_TRUE@%{_sysconfdir}/gconf/schemas/gstreamer-%{majorminor}.schemas
%changelog
+* Tue Jun 12 2007 Jan Schmidt <jan at fluendo dot com>
+- wavpack and qtdemux have moved from bad
+
* Fri Sep 02 2005 Thomas Vander Stichele <thomas at apestaart dot org>
- clean up for splitup
win32/vs6/libgstmonoscope.dsp
win32/vs6/libgstmulaw.dsp
win32/vs6/libgstmultipart.dsp
+win32/vs6/libgstqtdemux.dsp
win32/vs6/libgstrtp.dsp
win32/vs6/libgstrtsp.dsp
win32/vs6/libgstsmpte.dsp