AG_GST_CHECK_PLUGIN(aacparse)
AG_GST_CHECK_PLUGIN(aiffparse)
AG_GST_CHECK_PLUGIN(amrparse)
-AG_GST_CHECK_PLUGIN(audioresample)
+AG_GST_CHECK_PLUGIN(legacyresample)
AG_GST_CHECK_PLUGIN(bayer)
AG_GST_CHECK_PLUGIN(cdxaparse)
AG_GST_CHECK_PLUGIN(dccp)
gst/aacparse/Makefile
gst/aiffparse/Makefile
gst/amrparse/Makefile
-gst/audioresample/Makefile
+gst/legacyresample/Makefile
gst/bayer/Makefile
gst/cdxaparse/Makefile
gst/dccp/Makefile
$(top_srcdir)/ext/x264/gstx264enc.h \
$(top_srcdir)/gst/aacparse/gstaacparse.h \
$(top_srcdir)/gst/amrparse/gstamrparse.h \
- $(top_srcdir)/gst/audioresample/gstaudioresample.h \
+ $(top_srcdir)/gst/legacyresample/gstlegacyresample.h \
$(top_srcdir)/gst/deinterlace/gstdeinterlace.h \
$(top_srcdir)/gst/dccp/gstdccpclientsink.h \
$(top_srcdir)/gst/dccp/gstdccpclientsrc.h \
<SECTION>
<FILE>element-legacyresample</FILE>
<TITLE>legacyresample</TITLE>
-GstAudioresample
-<SUBSECTION Standard>
-GstAudioresampleClass
-GST_AUDIORESAMPLE
-GST_AUDIORESAMPLE_CLASS
-GST_IS_AUDIORESAMPLE
-GST_IS_AUDIORESAMPLE_CLASS
-GST_TYPE_AUDIORESAMPLE
-gst_audioresample_get_type
+GstLegacyresample
+<SUBSECTION Standard>
+GstLegacyresampleClass
+GST_LEGACYRESAMPLE
+GST_LEGACYRESAMPLE_CLASS
+GST_IS_LEGACYRESAMPLE
+GST_IS_LEGACYRESAMPLE_CLASS
+GST_TYPE_LEGACYRESAMPLE
+gst_legacyresample_get_type
</SECTION>
<SECTION>
<ARG>
<NAME>GstXvidEnc::averaging-period</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Averaging Period</NICK>
<BLURB>[CBR] Number of frames for which XviD averages bitrate.</BLURB>
<ARG>
<NAME>GstXvidEnc::buffer</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer Size</NICK>
<BLURB>[CBR] Size of the video buffers.</BLURB>
<ARG>
<NAME>GstXvidEnc::container-frame-overhead</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Container Frame Overhead</NICK>
<BLURB>[PASS2] Average container overhead per frame.</BLURB>
<ARG>
<NAME>GstXvidEnc::flow-control-strength</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Flow Control Strength</NICK>
<BLURB>[PASS2] Overflow control strength per frame.</BLURB>
<ARG>
<NAME>GstXvidEnc::keyframe-reduction</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Keyframe Reduction</NICK>
<BLURB>[PASS2] Keyframe size reduction in % of those within threshold.</BLURB>
<ARG>
<NAME>GstXvidEnc::keyframe-threshold</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Keyframe Threshold</NICK>
<BLURB>[PASS2] Distance between keyframes not to be subject to reduction.</BLURB>
<ARG>
<NAME>GstXvidEnc::max-overflow-degradation</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Max Overflow Degradation</NICK>
<BLURB>[PASS2] Amount in % that flow control can decrease frame size compared to ideal curve.</BLURB>
<ARG>
<NAME>GstXvidEnc::max-overflow-improvement</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Max Overflow Improvement</NICK>
<BLURB>[PASS2] Amount in % that flow control can increase frame size compared to ideal curve.</BLURB>
<ARG>
<NAME>GstXvidEnc::reaction-delay-factor</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
+<RANGE>[G_MAXULONG,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Reaction Delay Factor</NICK>
<BLURB>[CBR] Reaction delay factor.</BLURB>
<ARG>
<NAME>GstDvbSrc::diseqc-source</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,7]</RANGE>
+<RANGE>[G_MAXULONG,7]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>diseqc source</NICK>
<BLURB>DISEqC selected source (-1 disabled) (DVB-S).</BLURB>
<FLAGS>rw</FLAGS>
<NICK>Path where to search for RealPlayer codecs</NICK>
<BLURB>Path where to search for RealPlayer codecs.</BLURB>
-<DEFAULT>"/usr/lib/win32:/usr/lib/codecs:/usr/local/RealPlayer/codecs:/usr/local/lib/win32:/usr/local/lib/codecs"</DEFAULT>
+<DEFAULT>"/usr/lib64/win32:/usr/lib64/codecs:/usr/local/lib64/win32:/usr/local/lib64/codecs"</DEFAULT>
</ARG>
<ARG>
<FLAGS>rw</FLAGS>
<NICK>Path where to search for RealPlayer codecs</NICK>
<BLURB>Path where to search for RealPlayer codecs.</BLURB>
-<DEFAULT>"/usr/lib/win32:/usr/lib/codecs:/usr/local/RealPlayer/codecs:/usr/local/lib/win32:/usr/local/lib/codecs"</DEFAULT>
+<DEFAULT>"/usr/lib64/win32:/usr/lib64/codecs:/usr/local/lib64/win32:/usr/local/lib64/codecs"</DEFAULT>
</ARG>
<ARG>
<ARG>
<NAME>DvbBaseBin::diseqc-source</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,7]</RANGE>
+<RANGE>[G_MAXULONG,7]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>diseqc source</NICK>
<BLURB>DISEqC selected source (-1 disabled) (DVB-S).</BLURB>
<ARG>
<NAME>GstTwoLame::psymodel</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,4]</RANGE>
+<RANGE>[G_MAXULONG,4]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Psychoacoustic Model</NICK>
<BLURB>Psychoacoustic model used to encode the audio.</BLURB>
<ARG>
<NAME>GstDCCPClientSrc::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket fd</NICK>
<BLURB>The socket file descriptor.</BLURB>
<ARG>
<NAME>GstDCCPServerSink::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket fd</NICK>
<BLURB>The client socket file descriptor.</BLURB>
<ARG>
<NAME>GstDCCPClientSink::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket fd</NICK>
<BLURB>The socket file descriptor.</BLURB>
<ARG>
<NAME>GstDCCPServerSrc::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket fd</NICK>
<BLURB>The client socket file descriptor.</BLURB>
<ARG>
<NAME>GstMpegTSDemux::program-number</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Program Number</NICK>
<BLURB>Program number to demux for (-1 to ignore).</BLURB>
<ARG>
<NAME>GstPcapParse::dst-port</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,65535]</RANGE>
+<RANGE>[G_MAXULONG,65535]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Destination port</NICK>
<BLURB>Destination port to restrict to.</BLURB>
<ARG>
<NAME>GstPcapParse::src-port</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,65535]</RANGE>
+<RANGE>[G_MAXULONG,65535]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Source port</NICK>
<BLURB>Source port to restrict to.</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
+<ARG>
+<NAME>GstLegacyresample::filter-length</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>filter length</NICK>
+<BLURB>Length of the resample filter.</BLURB>
+<DEFAULT>16</DEFAULT>
+</ARG>
+
GstPipeline
RsnDvdBin
DvbBaseBin
- GstRgVolume
GstRtpBin
GstRtpClient
GstSDPDemux
+ GstAmrwbDec
+ GstAmrwbParse
+ GstAmrwbEnc
+ GstBaseMetadata
+ GstMetadataDemux
+ GstMetadataMux
+ GstXvidEnc
+ GstXvidDec
+ GstFaad
GstBz2enc
GstBz2dec
- GstBaseSrc
- GstPushSrc
- GstNeonhttpSrc
- GstMythtvSrc
- GstDc1394
- GstMMS
- GstBaseAudioSrc
- GstJackAudioSrc
- GstAudioSrc
- GstOss4Source
- GstVCDSrc
- GstDvbSrc
- GstDCCPClientSrc
- GstDCCPServerSrc
- GstRfbSrc
- GstSFSrc
GstCDAudio
+ GstX264Enc
GstBaseSink
GstVideoSink
GstDfbVideoSink
GstSDLVideoSink
GstBaseAudioSink
GstAudioSink
+ GstNasSink
GstSDLAudioSink
GstApExSink
- GstNasSink
GstOss4Sink
GstJackAudioSink
- GstSFSink
AlsaSPDIFSink
+ GstSFSink
GstFBDEVSink
GstDCCPServerSink
GstDCCPClientSink
- GstFaad
- GstCeltEnc
- GstCeltDec
- GstSpcDec
- GstWildmidi
+ GstBaseSrc
+ GstPushSrc
+ GstMythtvSrc
+ GstMMS
+ GstDc1394
+ GstBaseAudioSrc
+ GstJackAudioSrc
+ GstAudioSrc
+ GstOss4Source
+ GstNeonhttpSrc
+ GstVCDSrc
+ GstDvbSrc
+ GstRfbSrc
+ GstDCCPClientSrc
+ GstDCCPServerSrc
+ GstSFSrc
GstBaseTransform
GstAudioFilter
GstOFA
GstBPMDetect
GstStereo
GstBayer2RGB
- GstRgAnalysis
- GstRgLimiter
- GstAudioresample
GstScaletempo
- GstDeinterlace
+ GstLegacyresample
GstVideoFilter
GstVideoAnalyse
GstVideoDetect
GstVideoMark
+ GstDeinterlace
GstIIR
+ GstDtsDec
+ GstFaac
+ GstMusepackDec
+ GstGSMEnc
+ GstGSMDec
+ GstWildmidi
GstSignalProcessor
- ladspa-noise-white
- ladspa-delay-5s
ladspa-amp-mono
ladspa-amp-stereo
+ ladspa-lpf
+ ladspa-hpf
+ ladspa-delay-5s
ladspa-sine-faaa
ladspa-sine-faac
ladspa-sine-fcaa
ladspa-sine-fcac
- ladspa-lpf
- ladspa-hpf
- GstXvidEnc
- GstXvidDec
- GstPitch
+ ladspa-noise-white
GstTwoLame
- GstMusepackDec
- GstMpeg2enc
- GstGSMEnc
- GstGSMDec
- GstFaac
- GstDtsDec
- GstDiracEnc
+ GstPitch
+ GstCeltEnc
+ GstCeltDec
GstTRM
- GstX264Enc
- GstBaseMetadata
- GstMetadataDemux
- GstMetadataMux
GstOss4Mixer
- GstAmrBaseParse
- GstAmrParse
- GstFestival
- GstModPlug
GstMveDemux
GstMveMux
- GstSrtEnc
- GstMpeg4VParse
- GstCDXAParse
- GstVcdParse
- GstNsfDec
- MpegTsMux
- GstRealVideoDec
- GstRealAudioDec
- GstRawParse
- GstVideoParse
- GstAudioParse
+ GstDeinterlace2
GstRtpJitterBuffer
GstRtpPtDemux
GstRtpSession
GstRtpSsrcDemux
- GstPcapParse
+ GstMpegPSDemux
+ GstMpegTSDemux
+ MpegTSParse
+ GstH264Parse
+ GstMpeg4VParse
+ MpegVideoParse
+ GstFLVDemux
+ GstFlvMux
+ GstNuvDemux
+ GstRawParse
+ GstVideoParse
+ GstAudioParse
+ GstSpeed
GstInputSelector
GstOutputSelector
- GstAacBaseParse
- GstAacParse
- GstVMncDec
GstQTMux
GstMP4Mux
GstGPPMux
GstMJ2Mux
- MpegVideoParse
- GstH264Parse
- GstMXFDemux
+ GstAacBaseParse
+ GstAacParse
+ GstCDXAParse
+ GstVcdParse
+ GstNsfDec
+ GstTtaParse
+ GstTtaDec
+ GstModPlug
GstY4mEncode
- GstSpeed
- GstInterleave
- GstDeinterleave
GstFreeze
- GstDVDSpu
+ GstVMncDec
AIFFParse
- GstTtaParse
- GstTtaDec
- GstNuvDemux
- GstFLVDemux
- GstFlvMux
- GstMpegPSDemux
- GstMpegTSDemux
- MpegTSParse
- GstDeinterlace2
+ GstSrtEnc
+ GstFestival
+ MpegTsMux
+ GstDVDSpu
+ GstMXFDemux
+ GstRealVideoDec
+ GstRealAudioDec
+ GstAmrBaseParse
+ GstAmrParse
+ GstPcapParse
GstBus
GstTask
GstClock
GstJackAudioSinkRingBuffer
GstSignalObject
GstColorBalanceChannel
- GstMixerTrack
- GstMixerOptions
RTPSession
FluTsPatInfo
FluTsPmtInfo
GTypePlugin
GstChildProxy
GstURIHandler
+ GstTagSetter
GstImplementsInterface
GstNavigation
GstColorBalance
GstXOverlay
- GstTagSetter
GstMixer
GstPropertyProbe
+ MXFDescriptiveMetadataFrameworkInterface
GstPipeline GstChildProxy
RsnDvdBin GstChildProxy GstURIHandler
DvbBaseBin GstChildProxy GstURIHandler
-GstRgVolume GstChildProxy
GstRtpBin GstChildProxy
GstRtpClient GstChildProxy
GstSDPDemux GstChildProxy
-GstNeonhttpSrc GstURIHandler
-GstMythtvSrc GstURIHandler
-GstMMS GstURIHandler
-GstOss4Source GstImplementsInterface GstMixer GstPropertyProbe
-GstVCDSrc GstURIHandler
+GstMetadataMux GstTagSetter
GstCDAudio GstURIHandler
GstDfbVideoSink GstImplementsInterface GstNavigation GstColorBalance
GstSDLVideoSink GstImplementsInterface GstNavigation GstXOverlay
GstApExSink GstImplementsInterface GstMixer
GstOss4Sink GstPropertyProbe
+GstMythtvSrc GstURIHandler
+GstMMS GstURIHandler
+GstOss4Source GstImplementsInterface GstMixer GstPropertyProbe
+GstNeonhttpSrc GstURIHandler
+GstVCDSrc GstURIHandler
GstCeltEnc GstTagSetter
-GstMetadataMux GstTagSetter
GstOss4Mixer GstImplementsInterface GstMixer GstPropertyProbe
+GstDeinterlace2 GstChildProxy
GstQTMux GstTagSetter
GstMP4Mux GstTagSetter
GstGPPMux GstTagSetter
GstMJ2Mux GstTagSetter
-GstDeinterlace2 GstChildProxy
GstChildProxy GstObject
+GstTagSetter GstObject GstElement
GstImplementsInterface GstObject GstElement
GstColorBalance GstObject GstImplementsInterface GstElement
GstXOverlay GstObject GstImplementsInterface GstElement
-GstTagSetter GstObject GstElement
GstMixer GstObject GstImplementsInterface GstElement
+MXFDescriptiveMetadataFrameworkInterface MXFDescriptiveMetadata
<description>Advanced Audio Coding Parser</description>
<filename>../../gst/aacparse/.libs/libgstaacparse.so</filename>
<basename>libgstaacparse.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>unknown</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
--- /dev/null
+<plugin>
+ <name>aiffparse</name>
+ <description>Parse an .aiff file into raw audio</description>
+ <filename>../../gst/aiffparse/.libs/libgstaiffparse.so</filename>
+ <basename>libgstaiffparse.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>aiffparse</name>
+ <longname>AIFF audio demuxer</longname>
+ <class>Codec/Demuxer/Audio</class>
+ <description>Parse a .aiff file into raw audio</description>
+ <author>Pioneers of the Inevitable <songbird@songbirdnest.com></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int){ 1234, 4321 }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], signed=(boolean){ true, false }</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-aiff</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>Alsa plugin for S/PDIF output</description>
<filename>../../ext/alsaspdif/.libs/libgstalsaspdif.so</filename>
<basename>libgstalsaspdif.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Adaptive Multi-Rate Parser</description>
<filename>../../gst/amrparse/.libs/libgstamrparse.so</filename>
<basename>libgstamrparse.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Adaptive Multi-Rate Wide-Band</description>
<filename>../../ext/amrwb/.libs/libgstamrwb.so</filename>
<basename>libgstamrwb.so</basename>
- <version>0.10.9.1</version>
+ <version>0.10.10.1</version>
<license>unknown</license>
<source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins CVS/prerelease</package>
--- /dev/null
+<plugin>
+ <name>apex</name>
+ <description>Apple AirPort Express Plugin</description>
+ <filename>../../ext/apexsink/.libs/libgstapexsink.so</filename>
+ <basename>libgstapexsink.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>apexsink</name>
+ <longname>Apple AirPort Express Audio Sink</longname>
+ <class>Sink/Audio/Wireless</class>
+ <description>Output stream to an AirPort Express</description>
+ <author>Jérémie Bernard [GRemi] <gremimail@gmail.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, channels=(int)2, rate=(int)44100, signed=(boolean)true</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>Elements to convert Bayer images</description>
<filename>../../gst/bayer/.libs/libgstbayer.so</filename>
<basename>libgstbayer.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Compress or decompress streams</description>
<filename>../../ext/bz2/.libs/libgstbz2.so</filename>
<basename>libgstbz2.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Play CD audio through the CD Drive</description>
<filename>../../ext/cdaudio/.libs/libgstcdaudio.so</filename>
<basename>libgstcdaudio.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Parse a .dat file (VCD) into raw mpeg1</description>
<filename>../../gst/cdxaparse/.libs/libgstcdxaparse.so</filename>
<basename>libgstcdxaparse.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>CELT plugin library</description>
<filename>../../ext/celt/.libs/libgstcelt.so</filename>
<basename>libgstcelt.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
--- /dev/null
+<plugin>
+ <name>dc1394</name>
+ <description>1394 IIDC Video Source</description>
+ <filename>../../ext/dc1394/.libs/libgstdc1394.so</filename>
+ <basename>libgstdc1394.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>dc1394src</name>
+ <longname>1394 IIDC Video Source</longname>
+ <class>Source/Video</class>
+ <description>libdc1394 based source, supports 1394 IIDC cameras</description>
+ <author>Antoine Tremblay <hexa00@gmail.com></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/x-raw-yuv, format=(fourcc)IYU2, bpp=(int)16, width=(int)160, height=(int)120, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)64; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)320, height=(int)240, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)65; video/x-raw-yuv, format=(fourcc)IYU1, bpp=(int)12, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)66; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)67; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)68; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)69; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)70; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)71; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)72; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)73; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)74; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)75; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)76; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)77; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)78; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)79; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)80; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)81; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)82; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)83; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)84; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)85; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)86; video/x-raw-gray, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], bpp=(int)8, depth=(int)8; video/x-raw-yuv, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], format=(fourcc)IYU1, bpp=(int)12; video/x-raw-yuv, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], format=(fourcc)UYVY, bpp=(int)16; video/x-raw-yuv, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], format=(fourcc)IYU2, bpp=(int)16; video/x-raw-rgb, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255; video/x-raw-gray, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], bpp=(int)16, depth=(int)16</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>transfer data over the network via DCCP.</description>
<filename>../../gst/dccp/.libs/libgstdccp.so</filename>
<basename>libgstdccp.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
<package>DCCP</package>
--- /dev/null
+<plugin>
+ <name>deinterlace2</name>
+ <description>Deinterlacer</description>
+ <filename>../../gst/deinterlace2/.libs/libgstdeinterlace2.so</filename>
+ <basename>libgstdeinterlace2.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>deinterlace2</name>
+ <longname>Deinterlacer</longname>
+ <class>Filter/Video</class>
+ <description>Deinterlace Methods ported from DScaler/TvTime</description>
+ <author>Martin Eikermann <meiker@upb.de>, Sebastian Dröge <slomo@circular-chaos.org></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>DirectFB video output plugin</description>
<filename>../../ext/directfb/.libs/libgstdfbvideosink.so</filename>
<basename>libgstdfbvideosink.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Decodes DTS audio streams</description>
<filename>../../ext/dts/.libs/libgstdtsdec.so</filename>
<basename>libgstdtsdec.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>GPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>DVB elements</description>
<filename>../../sys/dvb/.libs/libgstdvb.so</filename>
<basename>libgstdvb.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>DVD Sub-picture Overlay element</description>
<filename>../../gst/dvdspu/.libs/libgstdvdspu.so</filename>
<basename>libgstdvdspu.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Free AAC Encoder (FAAC)</description>
<filename>../../ext/faac/.libs/libgstfaac.so</filename>
<basename>libgstfaac.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Free AAC Decoder (FAAD)</description>
<filename>../../ext/faad/.libs/libgstfaad.so</filename>
<basename>libgstfaad.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>GPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>linux framebuffer video sink</description>
<filename>../../sys/fbdev/.libs/libgstfbdevsink.so</filename>
<basename>libgstfbdevsink.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Synthesizes plain text into audio</description>
<filename>../../gst/festival/.libs/libgstfestival.so</filename>
<basename>libgstfestival.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
--- /dev/null
+<plugin>
+ <name>flv</name>
+ <description>FLV muxing and demuxing plugin</description>
+ <filename>../../gst/flv/.libs/libgstflv.so</filename>
+ <basename>libgstflv.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>flvdemux</name>
+ <longname>FLV Demuxer</longname>
+ <class>Codec/Demuxer</class>
+ <description>Demux FLV feeds into digital streams</description>
+ <author>Julien Moutte <julien@moutte.net></author>
+ <pads>
+ <caps>
+ <name>video</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>ANY</details>
+ </caps>
+ <caps>
+ <name>audio</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>ANY</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/x-flv</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>flvmux</name>
+ <longname>FLV muxer</longname>
+ <class>Codec/Muxer</class>
+ <description>Muxes video/audio streams into a FLV stream</description>
+ <author>Sebastian Dröge <sebastian.droege@collabora.co.uk></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/x-flv</details>
+ </caps>
+ <caps>
+ <name>audio</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)4; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)8, depth=(int)8, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)false; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)16, depth=(int)16, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)true; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }</details>
+ </caps>
+ <caps>
+ <name>video</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>video/x-flash-video; video/x-flash-screen; video/x-vp6-flash; video/x-vp6-alpha; video/x-h264</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>Stream freezer</description>
<filename>../../gst/freeze/.libs/libgstfreeze.so</filename>
<basename>libgstfreeze.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>GSM encoder/decoder</description>
<filename>../../ext/gsm/.libs/libgstgsm.so</filename>
<basename>libgstgsm.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Deinterlace video</description>
<filename>../../gst/deinterlace/.libs/libgstdeinterlace.so</filename>
<basename>libgstdeinterlace.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Element parsing raw h264 streams</description>
<filename>../../gst/h264parse/.libs/libgsth264parse.so</filename>
<basename>libgsth264parse.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Jack elements</description>
<filename>../../ext/jack/.libs/libgstjack.so</filename>
<basename>libgstjack.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>All LADSPA plugins</description>
<filename>../../ext/ladspa/.libs/libgstladspa.so</filename>
<basename>libgstladspa.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<plugin>
<name>legacyresample</name>
<description>Resamples audio</description>
- <filename>../../gst/audioresample/.libs/libgstlegacyresample.so</filename>
+ <filename>../../gst/legacyresample/.libs/libgstlegacyresample.so</filename>
<basename>libgstlegacyresample.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Metadata (EXIF, IPTC and XMP) image (JPEG, TIFF) demuxer and muxer</description>
<filename>../../ext/metadata/.libs/libgstmetadata.so</filename>
<basename>libgstmetadata.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Microsoft Multi Media Server streaming protocol support</description>
<filename>../../ext/libmms/.libs/libgstmms.so</filename>
<basename>libgstmms.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>.MOD audio decoding</description>
<filename>../../gst/modplug/.libs/libgstmodplug.so</filename>
<basename>libgstmodplug.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>MPEG-4 video parser</description>
<filename>../../gst/mpeg4videoparse/.libs/libgstmpeg4videoparse.so</filename>
<basename>libgstmpeg4videoparse.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
--- /dev/null
+<plugin>
+ <name>mpegdemux2</name>
+ <description>MPEG demuxers</description>
+ <filename>../../gst/mpegdemux/.libs/libgstmpegdemux.so</filename>
+ <basename>libgstmpegdemux.so</basename>
+ <version>0.10.10.1</version>
+ <license>unknown</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>mpegpsdemux</name>
+ <longname>The Fluendo MPEG Program Stream Demuxer</longname>
+ <class>Codec/Demuxer</class>
+ <description>Demultiplexes MPEG Program Streams</description>
+ <author>Wim Taymans <wim@fluendo.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/mpeg, mpegversion=(int){ 1, 2 }, systemstream=(boolean)true; video/x-cdxa</details>
+ </caps>
+ <caps>
+ <name>private_%d</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>ANY</details>
+ </caps>
+ <caps>
+ <name>audio_%02x</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>audio/mpeg, mpegversion=(int)1; audio/x-private1-lpcm; audio/x-private1-ac3; audio/x-private1-dts; audio/ac3</details>
+ </caps>
+ <caps>
+ <name>video_%02x</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false; video/x-h264</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>mpegtsdemux</name>
+ <longname>The Fluendo MPEG Transport stream demuxer</longname>
+ <class>Codec/Demuxer</class>
+ <description>Demultiplexes MPEG2 Transport Streams</description>
+ <author>Wim Taymans <wim@fluendo.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/mpegts</details>
+ </caps>
+ <caps>
+ <name>private_%04x</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>ANY</details>
+ </caps>
+ <caps>
+ <name>audio_%04x</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>audio/mpeg, mpegversion=(int){ 1, 4 }; audio/x-lpcm, width=(int){ 16, 20, 24 }, rate=(int){ 48000, 96000 }, channels=(int)[ 1, 8 ], dynamic_range=(int)[ 0, 255 ], emphasis=(boolean){ false, true }, mute=(boolean){ false, true }; audio/x-ac3; audio/x-dts</details>
+ </caps>
+ <caps>
+ <name>video_%04x</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false; video/x-h264; video/x-dirac</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>mpegtsparse</name>
+ <longname>MPEG transport stream parser</longname>
+ <class>Codec/Parser</class>
+ <description>Parses MPEG2 transport streams</description>
+ <author>Alessandro Decina <alessandro@nnva.org>
+ Zaheer Abbas Merali <zaheerabbas at merali dot org></author>
+ <pads>
+ <caps>
+ <name>program_%d</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>video/mpegts, systemstream=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>src%d</name>
+ <direction>source</direction>
+ <presence>request</presence>
+ <details>video/mpegts, systemstream=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/mpegts, systemstream=(boolean)true</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
--- /dev/null
+<plugin>
+ <name>mpegtsmux</name>
+ <description>MPEG-TS muxer</description>
+ <filename>../../gst/mpegtsmux/.libs/libgstmpegtsmux.so</filename>
+ <basename>libgstmpegtsmux.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>mpegtsmux</name>
+ <longname>MPEG Transport Stream Muxer</longname>
+ <class>Codec/Muxer</class>
+ <description>Multiplexes media streams into an MPEG Transport Stream</description>
+ <author>Fluendo <contact@fluendo.com></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/mpegts, systemstream=(boolean)true, packetsize=(int){ 188, 192 }</details>
+ </caps>
+ <caps>
+ <name>sink_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false; video/x-dirac; video/x-h264; audio/mpeg, mpegversion=(int){ 1, 2, 4 }; audio/x-lpcm, width=(int){ 16, 20, 24 }, rate=(int){ 48000, 96000 }, channels=(int)[ 1, 8 ], dynamic_range=(int)[ 0, 255 ], emphasis=(boolean){ false, true }, mute=(boolean){ false, true }; audio/x-ac3; audio/x-dts</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>MPEG-1 and MPEG-2 video parser</description>
<filename>../../gst/mpegvideoparse/.libs/libgstmpegvideoparse.so</filename>
<basename>libgstmpegvideoparse.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Musepack decoder</description>
<filename>../../ext/musepack/.libs/libgstmusepack.so</filename>
<basename>libgstmusepack.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>A TRM signature producer based on libmusicbrainz</description>
<filename>../../ext/musicbrainz/.libs/libgsttrm.so</filename>
<basename>libgsttrm.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Interplay MVE movie format manipulation</description>
<filename>../../gst/mve/.libs/libgstmve.so</filename>
<basename>libgstmve.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>MXF plugin library</description>
<filename>../../gst/mxf/.libs/libgstmxf.so</filename>
<basename>libgstmxf.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>lib MythTV src</description>
<filename>../../ext/mythtv/.libs/libgstmythtvsrc.so</filename>
<basename>libgstmythtvsrc.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>NAS (Network Audio System) support for GStreamer</description>
<filename>../../ext/nas/.libs/libgstnassink.so</filename>
<basename>libgstnassink.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>lib neon http client src</description>
<filename>../../ext/neon/.libs/libgstneonhttpsrc.so</filename>
<basename>libgstneonhttpsrc.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Uses nosefart to decode .nsf files</description>
<filename>../../gst/nsf/.libs/libgstnsf.so</filename>
<basename>libgstnsf.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>GPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Demuxes and muxes audio and video</description>
<filename>../../gst/nuvdemux/.libs/libgstnuvdemux.so</filename>
<basename>libgstnuvdemux.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
--- /dev/null
+<plugin>
+ <name>ofa</name>
+ <description>Calculate MusicIP fingerprint from audio files</description>
+ <filename>../../ext/ofa/.libs/libgstofa.so</filename>
+ <basename>libgstofa.so</basename>
+ <version>0.10.10.1</version>
+ <license>GPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>ofa</name>
+ <longname>OFA</longname>
+ <class>MusicIP Fingerprinting element</class>
+ <description>Find a music fingerprint using MusicIP's libofa</description>
+ <author>Milosz Derezynski <internalerror@gmail.com>, Eric Buehl <eric.buehl@gmail.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int){ 1234, 4321 }, width=(int){ 16 }, depth=(int){ 16 }, signed=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int){ 1234, 4321 }, width=(int){ 16 }, depth=(int){ 16 }, signed=(boolean)true</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>Open Sound System (OSS) version 4 support for GStreamer</description>
<filename>../../sys/oss4/.libs/libgstoss4audio.so</filename>
<basename>libgstoss4audio.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
--- /dev/null
+<plugin>
+ <name>pcapparse</name>
+ <description>Element parsing raw pcap streams</description>
+ <filename>../../gst/pcapparse/.libs/libgstpcapparse.so</filename>
+ <basename>libgstpcapparse.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer</package>
+ <origin>http://gstreamer.net/</origin>
+ <elements>
+ <element>
+ <name>pcapparse</name>
+ <longname>PCapParse</longname>
+ <class>Raw/Parser</class>
+ <description>Parses a raw pcap stream</description>
+ <author>Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>ANY</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>raw/x-pcap</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
--- /dev/null
+<plugin>
+ <name>qtmux</name>
+ <description>Quicktime Muxer plugin</description>
+ <filename>../../gst/qtmux/.libs/libgstqtmux.so</filename>
+ <basename>libgstqtmux.so</basename>
+ <version>0.10.10.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-bad</source>
+ <package>gsoc2008 package</package>
+ <origin>embedded.ufcg.edu.br</origin>
+ <elements>
+ <element>
+ <name>gppmux</name>
+ <longname>3GPP Muxer</longname>
+ <class>Codec/Muxer</class>
+ <description>Multiplex audio and video into a 3GPP file</description>
+ <author>Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br></author>
+ <pads>
+ <caps>
+ <name>video_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>video/x-h264, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
+ <caps>
+ <name>audio_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>audio/AMR, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>application/x-3gp</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>mj2mux</name>
+ <longname>MJ2 Muxer</longname>
+ <class>Codec/Muxer</class>
+ <description>Multiplex audio and video into a MJ2 file</description>
+ <author>Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br></author>
+ <pads>
+ <caps>
+ <name>video_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>image/x-j2c, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
+ <caps>
+ <name>audio_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>audio/x-raw-int, width=(int)8, depth=(int)8, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean){ true, false }; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/mj2</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>mp4mux</name>
+ <longname>MP4 Muxer</longname>
+ <class>Codec/Muxer</class>
+ <description>Multiplex audio and video into a MP4 file</description>
+ <author>Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br></author>
+ <pads>
+ <caps>
+ <name>video_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-mp4-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
+ <caps>
+ <name>audio_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/quicktime</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>qtmux</name>
+ <longname>QuickTime Muxer</longname>
+ <class>Codec/Muxer</class>
+ <description>Multiplex audio and video into a QuickTime file</description>
+ <author>Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br></author>
+ <pads>
+ <caps>
+ <name>video_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>video/x-raw-rgb, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-yuv, format=(fourcc)UYVY, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h263, h263version=(string)h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-dv, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-qt-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
+ <caps>
+ <name>audio_%d</name>
+ <direction>sink</direction>
+ <presence>request</presence>
+ <details>audio/x-raw-int, width=(int)8, depth=(int)8, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean){ true, false }; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-mulaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/quicktime</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>Parses byte streams into raw frames</description>
<filename>../../gst/rawparse/.libs/libgstrawparse.so</filename>
<basename>libgstrawparse.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Decode REAL streams</description>
<filename>../../gst/real/.libs/libgstreal.so</filename>
<basename>libgstreal.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
--- /dev/null
+<plugin>
+ <name>resindvd</name>
+ <description>Resin DVD playback elements</description>
+ <filename>../../ext/resindvd/.libs/libresindvd.so</filename>
+ <basename>libresindvd.so</basename>
+ <version>0.10.10.1</version>
+ <license>GPL</license>
+ <source>gst-plugins-bad</source>
+ <package>GStreamer</package>
+ <origin>http://gstreamer.net/</origin>
+ <elements>
+ <element>
+ <name>rsndvdbin</name>
+ <longname>rsndvdbin</longname>
+ <class>Generic/Bin/Player</class>
+ <description>DVD playback element</description>
+ <author>Jan Schmidt <thaytan@noraisin.net></author>
+ <pads>
+ <caps>
+ <name>subpicture</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>video/x-dvd-subpicture</details>
+ </caps>
+ <caps>
+ <name>audio</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>audio/x-raw-int; audio/x-raw-float</details>
+ </caps>
+ <caps>
+ <name>video</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>video/mpeg, mpegversion=(int){ 1, 2 }, systemstream=(boolean)false</details>
+ </caps>
+ </pads>
+ </element>
+ </elements>
+</plugin>
\ No newline at end of file
<description>Connects to a VNC server and decodes RFB stream</description>
<filename>../../gst/librfb/.libs/libgstrfbsrc.so</filename>
<basename>libgstrfbsrc.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Scale audio tempo in sync with playback rate</description>
<filename>../../gst/scaletempo/.libs/libgstscaletempoplugin.so</filename>
<basename>libgstscaletempoplugin.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
<package>GStreamer</package>
<description>SDL (Simple DirectMedia Layer) support for GStreamer</description>
<filename>../../ext/sdl/.libs/libgstsdl.so</filename>
<basename>libgstsdl.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>configure streaming sessions using SDP</description>
<filename>../../gst/sdp/.libs/libgstsdpelem.so</filename>
<basename>libgstsdpelem.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>input/output stream selector elements</description>
<filename>../../gst/selector/.libs/libgstselector.so</filename>
<basename>libgstselector.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>use libsndfile to read and write audio from and to files</description>
<filename>../../ext/sndfile/.libs/libgstsndfile.so</filename>
<basename>libgstsndfile.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Audio Pitch Controller & BPM Detection</description>
<filename>../../ext/soundtouch/.libs/libgstsoundtouch.so</filename>
<basename>libgstsoundtouch.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Set speed/pitch on audio/raw streams (resampler)</description>
<filename>../../gst/speed/.libs/libgstspeed.so</filename>
<basename>libgstspeed.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Muck with the stereo signal, enhance it's 'stereo-ness'</description>
<filename>../../gst/stereo/.libs/libgststereo.so</filename>
<basename>libgststereo.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>subtitle encoders</description>
<filename>../../gst/subenc/.libs/libgstsubenc.so</filename>
<basename>libgstsubenc.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>TTA lossless audio format handling</description>
<filename>../../gst/tta/.libs/libgsttta.so</filename>
<basename>libgsttta.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Encode MP2s with TwoLAME</description>
<filename>../../ext/twolame/.libs/libgsttwolame.so</filename>
<basename>libgsttwolame.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Asynchronous read from VCD disk</description>
<filename>../../sys/vcd/.libs/libgstvcdsrc.so</filename>
<basename>libgstvcdsrc.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Various video signal analysers</description>
<filename>../../gst/videosignal/.libs/libgstvideosignal.so</filename>
<basename>libgstvideosignal.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>VMnc video plugin library</description>
<filename>../../gst/vmnc/.libs/libgstvmnc.so</filename>
<basename>libgstvmnc.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Wildmidi Plugin</description>
<filename>../../ext/timidity/.libs/libgstwildmidi.so</filename>
<basename>libgstwildmidi.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>GPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>libx264-based H264 plugins</description>
<filename>../../ext/x264/.libs/libgstx264.so</filename>
<basename>libgstx264.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>GPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>XviD plugin library</description>
<filename>../../ext/xvid/.libs/libgstxvid.so</filename>
<basename>libgstxvid.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>GPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Encodes a YUV frame into the yuv4mpeg format (mjpegtools)</description>
<filename>../../gst/y4m/.libs/libgsty4menc.so</filename>
<basename>libgsty4menc.so</basename>
- <version>0.10.10</version>
+ <version>0.10.10.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins source release</package>
+ <package>GStreamer Bad Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
buffer.c
noinst_HEADERS = \
- gstaudioresample.h \
+ gstlegacyresample.h \
functable.h \
debug.h \
buffer.h
-libgstlegacyresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
+libgstlegacyresample_la_SOURCES = gstlegacyresample.c $(resample_SOURCES)
libgstlegacyresample_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS)
libgstlegacyresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS)
libgstlegacyresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
#include <math.h>
/*#define DEBUG_ENABLED */
-#include "gstaudioresample.h"
+#include "gstlegacyresample.h"
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
-GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
-#define GST_CAT_DEFAULT audioresample_debug
+GST_DEBUG_CATEGORY_STATIC (legacyresample_debug);
+#define GST_CAT_DEFAULT legacyresample_debug
/* elementfactory information */
-static const GstElementDetails gst_audioresample_details =
+static const GstElementDetails gst_legacyresample_details =
GST_ELEMENT_DETAILS ("Audio scaler",
"Filter/Converter/Audio",
"Resample audio",
"width = (int) 64" \
)
-static GstStaticPadTemplate gst_audioresample_sink_template =
+static GstStaticPadTemplate gst_legacyresample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static GstStaticPadTemplate gst_audioresample_src_template =
+static GstStaticPadTemplate gst_legacyresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static void gst_audioresample_set_property (GObject * object,
+static void gst_legacyresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_audioresample_get_property (GObject * object,
+static void gst_legacyresample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
-static gboolean audioresample_get_unit_size (GstBaseTransform * base,
+static gboolean legacyresample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, guint * size);
-static GstCaps *audioresample_transform_caps (GstBaseTransform * base,
+static GstCaps *legacyresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps);
-static void audioresample_fixate_caps (GstBaseTransform * base,
+static void legacyresample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
-static gboolean audioresample_transform_size (GstBaseTransform * trans,
+static gboolean legacyresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * incaps, guint insize,
GstCaps * outcaps, guint * outsize);
-static gboolean audioresample_set_caps (GstBaseTransform * base,
+static gboolean legacyresample_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
-static GstFlowReturn audioresample_pushthrough (GstAudioresample *
- audioresample);
-static GstFlowReturn audioresample_transform (GstBaseTransform * base,
+static GstFlowReturn legacyresample_pushthrough (GstLegacyresample *
+ legacyresample);
+static GstFlowReturn legacyresample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
-static gboolean audioresample_start (GstBaseTransform * base);
-static gboolean audioresample_stop (GstBaseTransform * base);
+static gboolean legacyresample_event (GstBaseTransform * base,
+ GstEvent * event);
+static gboolean legacyresample_start (GstBaseTransform * base);
+static gboolean legacyresample_stop (GstBaseTransform * base);
-static gboolean audioresample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audioresample_query_type (GstPad * pad);
+static gboolean legacyresample_query (GstPad * pad, GstQuery * query);
+static const GstQueryType *legacyresample_query_type (GstPad * pad);
#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (audioresample_debug, "legacyresample", 0, "audio resampling element");
+ GST_DEBUG_CATEGORY_INIT (legacyresample_debug, "legacyresample", 0, "audio resampling element");
-GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
+GST_BOILERPLATE_FULL (GstLegacyresample, gst_legacyresample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
static void
-gst_audioresample_base_init (gpointer g_class)
+gst_legacyresample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_src_template));
+ gst_static_pad_template_get (&gst_legacyresample_src_template));
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_sink_template));
+ gst_static_pad_template_get (&gst_legacyresample_sink_template));
- gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
+ gst_element_class_set_details (gstelement_class, &gst_legacyresample_details);
}
static void
-gst_audioresample_class_init (GstAudioresampleClass * klass)
+gst_legacyresample_class_init (GstLegacyresampleClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
- gobject_class->set_property = gst_audioresample_set_property;
- gobject_class->get_property = gst_audioresample_get_property;
+ gobject_class->set_property = gst_legacyresample_set_property;
+ gobject_class->get_property = gst_legacyresample_get_property;
g_object_class_install_property (gobject_class, PROP_FILTERLEN,
g_param_spec_int ("filter-length", "filter length",
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
GST_BASE_TRANSFORM_CLASS (klass)->start =
- GST_DEBUG_FUNCPTR (audioresample_start);
+ GST_DEBUG_FUNCPTR (legacyresample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
- GST_DEBUG_FUNCPTR (audioresample_stop);
+ GST_DEBUG_FUNCPTR (legacyresample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
- GST_DEBUG_FUNCPTR (audioresample_transform_size);
+ GST_DEBUG_FUNCPTR (legacyresample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
- GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
+ GST_DEBUG_FUNCPTR (legacyresample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
- GST_DEBUG_FUNCPTR (audioresample_transform_caps);
+ GST_DEBUG_FUNCPTR (legacyresample_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
- GST_DEBUG_FUNCPTR (audioresample_fixate_caps);
+ GST_DEBUG_FUNCPTR (legacyresample_fixate_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
- GST_DEBUG_FUNCPTR (audioresample_set_caps);
+ GST_DEBUG_FUNCPTR (legacyresample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
- GST_DEBUG_FUNCPTR (audioresample_transform);
+ GST_DEBUG_FUNCPTR (legacyresample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (audioresample_event);
+ GST_DEBUG_FUNCPTR (legacyresample_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void
-gst_audioresample_init (GstAudioresample * audioresample,
- GstAudioresampleClass * klass)
+gst_legacyresample_init (GstLegacyresample * legacyresample,
+ GstLegacyresampleClass * klass)
{
GstBaseTransform *trans;
- trans = GST_BASE_TRANSFORM (audioresample);
+ trans = GST_BASE_TRANSFORM (legacyresample);
/* buffer alloc passthrough is too impossible. FIXME, it
* is trivial in the passthrough case. */
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
- audioresample->filter_length = DEFAULT_FILTERLEN;
+ legacyresample->filter_length = DEFAULT_FILTERLEN;
- audioresample->need_discont = FALSE;
+ legacyresample->need_discont = FALSE;
- gst_pad_set_query_function (trans->srcpad, audioresample_query);
- gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);
+ gst_pad_set_query_function (trans->srcpad, legacyresample_query);
+ gst_pad_set_query_type_function (trans->srcpad, legacyresample_query_type);
}
/* vmethods */
static gboolean
-audioresample_start (GstBaseTransform * base)
+legacyresample_start (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (base);
- audioresample->resample = resample_new ();
- audioresample->ts_offset = -1;
- audioresample->offset = -1;
- audioresample->next_ts = -1;
+ legacyresample->resample = resample_new ();
+ legacyresample->ts_offset = -1;
+ legacyresample->offset = -1;
+ legacyresample->next_ts = -1;
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
+ resample_set_filter_length (legacyresample->resample,
+ legacyresample->filter_length);
return TRUE;
}
static gboolean
-audioresample_stop (GstBaseTransform * base)
+legacyresample_stop (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (base);
- if (audioresample->resample) {
- resample_free (audioresample->resample);
- audioresample->resample = NULL;
+ if (legacyresample->resample) {
+ resample_free (legacyresample->resample);
+ legacyresample->resample = NULL;
}
- gst_caps_replace (&audioresample->sinkcaps, NULL);
- gst_caps_replace (&audioresample->srccaps, NULL);
+ gst_caps_replace (&legacyresample->sinkcaps, NULL);
+ gst_caps_replace (&legacyresample->srccaps, NULL);
return TRUE;
}
static gboolean
-audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
+legacyresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size)
{
gint width, channels;
}
static GstCaps *
-audioresample_transform_caps (GstBaseTransform * base,
+legacyresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *res;
/* Fixate rate to the allowed rate that has the smallest difference */
static void
-audioresample_fixate_caps (GstBaseTransform * base,
+legacyresample_fixate_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
{
GstStructure *s;
}
static gboolean
-audioresample_transform_size (GstBaseTransform * base,
+legacyresample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (base);
ResampleState *state;
GstCaps *srccaps, *sinkcaps;
gboolean use_internal = FALSE; /* whether we use the internal state */
/* if the caps are the ones that _set_caps got called with; we can use
* our own state; otherwise we'll have to create a state */
- if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&
- gst_caps_is_equal (srccaps, audioresample->srccaps)) {
+ if (gst_caps_is_equal (sinkcaps, legacyresample->sinkcaps) &&
+ gst_caps_is_equal (srccaps, legacyresample->srccaps)) {
use_internal = TRUE;
- state = audioresample->resample;
+ state = legacyresample->resample;
} else {
- GST_DEBUG_OBJECT (audioresample,
+ GST_DEBUG_OBJECT (legacyresample,
"caps are not the set caps, creating state");
state = resample_new ();
- resample_set_filter_length (state, audioresample->filter_length);
+ resample_set_filter_length (state, legacyresample->filter_length);
resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
}
}
static gboolean
-audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
+legacyresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
gint inrate, outrate;
int channels;
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstLegacyresample *legacyresample = GST_LEGACYRESAMPLE (base);
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
- ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,
+ ret = resample_set_state_from_caps (legacyresample->resample, incaps, outcaps,
&channels, &inrate, &outrate);
g_return_val_if_fail (ret, FALSE);
- audioresample->channels = channels;
- GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);
- audioresample->i_rate = inrate;
- GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);
- audioresample->o_rate = outrate;
- GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);
+ legacyresample->channels = channels;
+ GST_DEBUG_OBJECT (legacyresample, "set channels to %d", channels);
+ legacyresample->i_rate = inrate;
+ GST_DEBUG_OBJECT (legacyresample, "set i_rate to %d", inrate);
+ legacyresample->o_rate = outrate;
+ GST_DEBUG_OBJECT (legacyresample, "set o_rate to %d", outrate);
/* save caps so we can short-circuit in the size_transform if the caps
* are the same */
- gst_caps_replace (&audioresample->sinkcaps, incaps);
- gst_caps_replace (&audioresample->srccaps, outcaps);
+ gst_caps_replace (&legacyresample->sinkcaps, incaps);
+ gst_caps_replace (&legacyresample->srccaps, outcaps);
return TRUE;
}
static gboolean
-audioresample_event (GstBaseTransform * base, GstEvent * event)
+legacyresample_event (GstBaseTransform * base, GstEvent * event)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
- audioresample = GST_AUDIORESAMPLE (base);
+ legacyresample = GST_LEGACYRESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
- if (audioresample->resample)
- resample_input_flush (audioresample->resample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
+ if (legacyresample->resample)
+ resample_input_flush (legacyresample->resample);
+ legacyresample->ts_offset = -1;
+ legacyresample->next_ts = -1;
+ legacyresample->offset = -1;
break;
case GST_EVENT_NEWSEGMENT:
- resample_input_pushthrough (audioresample->resample);
- audioresample_pushthrough (audioresample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
+ resample_input_pushthrough (legacyresample->resample);
+ legacyresample_pushthrough (legacyresample);
+ legacyresample->ts_offset = -1;
+ legacyresample->next_ts = -1;
+ legacyresample->offset = -1;
break;
case GST_EVENT_EOS:
- resample_input_eos (audioresample->resample);
- audioresample_pushthrough (audioresample);
+ resample_input_eos (legacyresample->resample);
+ legacyresample_pushthrough (legacyresample);
break;
default:
break;
}
static GstFlowReturn
-audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
+legacyresample_do_output (GstLegacyresample * legacyresample,
+ GstBuffer * outbuf)
{
int outsize;
int outsamples;
ResampleState *r;
- r = audioresample->resample;
+ r = legacyresample->resample;
outsize = resample_get_output_size (r);
- GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
+ GST_LOG_OBJECT (legacyresample, "legacyresample can give me %d bytes",
+ outsize);
/* protect against mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
- GST_WARNING_OBJECT (audioresample,
- "overriding audioresample's outsize %d with outbuffer's size %d",
+ GST_WARNING_OBJECT (legacyresample,
+ "overriding legacyresample's outsize %d with outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
outsize = GST_BUFFER_SIZE (outbuf);
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's outsize %d too far from outbuffer's size %d",
+ GST_WARNING_OBJECT (legacyresample,
+ "legacyresample's outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
outsamples = outsize / r->sample_size;
- GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
+ GST_LOG_OBJECT (legacyresample, "resample gave me %d bytes or %d samples",
outsize, outsamples);
- GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
- GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
+ GST_BUFFER_OFFSET (outbuf) = legacyresample->offset;
+ GST_BUFFER_TIMESTAMP (outbuf) = legacyresample->next_ts;
- if (audioresample->ts_offset != -1) {
- audioresample->offset += outsamples;
- audioresample->ts_offset += outsamples;
- audioresample->next_ts =
- gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
- audioresample->o_rate);
- GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
+ if (legacyresample->ts_offset != -1) {
+ legacyresample->offset += outsamples;
+ legacyresample->ts_offset += outsamples;
+ legacyresample->next_ts =
+ gst_util_uint64_scale_int (legacyresample->ts_offset, GST_SECOND,
+ legacyresample->o_rate);
+ GST_BUFFER_OFFSET_END (outbuf) = legacyresample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
- GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
+ GST_BUFFER_DURATION (outbuf) = legacyresample->next_ts -
GST_BUFFER_TIMESTAMP (outbuf);
} else {
/* no valid offset know, we can still sortof calculate the duration though */
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale_int (outsamples, GST_SECOND,
- audioresample->o_rate);
+ legacyresample->o_rate);
}
/* check for possible mem corruption */
/* this is an error that when it happens, would need fixing in the
* resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
* and it gave us more ! */
- GST_WARNING_OBJECT (audioresample,
- "audioresample, you memory corrupting bastard. "
+ GST_WARNING_OBJECT (legacyresample,
+ "legacyresample, you memory corrupting bastard. "
"you gave me outsize %d while my buffer was size %d",
outsize, GST_BUFFER_SIZE (outbuf));
return GST_FLOW_ERROR;
}
/* catch possibly wrong size differences */
if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's written outsize %d too far from outbuffer's size %d",
+ GST_WARNING_OBJECT (legacyresample,
+ "legacyresample's written outsize %d too far from outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
}
GST_BUFFER_SIZE (outbuf) = outsize;
- if (G_UNLIKELY (audioresample->need_discont)) {
- GST_DEBUG_OBJECT (audioresample,
+ if (G_UNLIKELY (legacyresample->need_discont)) {
+ GST_DEBUG_OBJECT (legacyresample,
"marking this buffer with the DISCONT flag");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- audioresample->need_discont = FALSE;
+ legacyresample->need_discont = FALSE;
}
- GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
+ GST_LOG_OBJECT (legacyresample, "transformed to buffer of %d bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
}
static gboolean
-audioresample_check_discont (GstAudioresample * audioresample,
+legacyresample_check_discont (GstLegacyresample * legacyresample,
GstClockTime timestamp)
{
if (timestamp != GST_CLOCK_TIME_NONE &&
- audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
- audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
- timestamp != audioresample->prev_ts + audioresample->prev_duration) {
+ legacyresample->prev_ts != GST_CLOCK_TIME_NONE &&
+ legacyresample->prev_duration != GST_CLOCK_TIME_NONE &&
+ timestamp != legacyresample->prev_ts + legacyresample->prev_duration) {
/* Potentially a discontinuous buffer. However, it turns out that many
* elements generate imperfect streams due to rounding errors, so we permit
* a small error (up to one sample) without triggering a filter
* flush/restart (if triggered incorrectly, this will be audible) */
GstClockTimeDiff diff = timestamp -
- (audioresample->prev_ts + audioresample->prev_duration);
+ (legacyresample->prev_ts + legacyresample->prev_duration);
- if (ABS (diff) > GST_SECOND / audioresample->i_rate) {
- GST_WARNING_OBJECT (audioresample,
+ if (ABS (diff) > GST_SECOND / legacyresample->i_rate) {
+ GST_WARNING_OBJECT (legacyresample,
"encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
return TRUE;
}
}
static GstFlowReturn
-audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+legacyresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
ResampleState *r;
guchar *data, *datacopy;
gulong size;
GstClockTime timestamp;
- audioresample = GST_AUDIORESAMPLE (base);
- r = audioresample->resample;
+ legacyresample = GST_LEGACYRESAMPLE (base);
+ r = legacyresample->resample;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
- GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
+ GST_LOG_OBJECT (legacyresample, "transforming buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
size, GST_TIME_ARGS (timestamp),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
/* check for timestamp discontinuities and flush/reset if needed */
- if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
+ if (G_UNLIKELY (legacyresample_check_discont (legacyresample, timestamp))) {
/* Flush internal samples */
- audioresample_pushthrough (audioresample);
+ legacyresample_pushthrough (legacyresample);
/* Inform downstream element about discontinuity */
- audioresample->need_discont = TRUE;
+ legacyresample->need_discont = TRUE;
/* We want to recalculate the offset */
- audioresample->ts_offset = -1;
+ legacyresample->ts_offset = -1;
}
- if (audioresample->ts_offset == -1) {
+ if (legacyresample->ts_offset == -1) {
/* if we don't know the initial offset yet, calculate it based on the
* input timestamp. */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* offset used to calculate the timestamps. We use the sample offset for
* this to make it more accurate. We want the first buffer to have the
* same timestamp as the incoming timestamp. */
- audioresample->next_ts = timestamp;
- audioresample->ts_offset =
+ legacyresample->next_ts = timestamp;
+ legacyresample->ts_offset =
gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
/* offset used to set as the buffer offset, this offset is always
* relative to the stream time, note that timestamp is not... */
stime = (timestamp - base->segment.start) + base->segment.time;
- audioresample->offset =
+ legacyresample->offset =
gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
}
}
- audioresample->prev_ts = timestamp;
- audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
+ legacyresample->prev_ts = timestamp;
+ legacyresample->prev_duration = GST_BUFFER_DURATION (inbuf);
/* need to memdup, resample takes ownership. */
datacopy = g_memdup (data, size);
resample_add_input_data (r, datacopy, size, g_free, datacopy);
- return audioresample_do_output (audioresample, outbuf);
+ return legacyresample_do_output (legacyresample, outbuf);
}
/* push remaining data in the buffers out */
static GstFlowReturn
-audioresample_pushthrough (GstAudioresample * audioresample)
+legacyresample_pushthrough (GstLegacyresample * legacyresample)
{
int outsize;
ResampleState *r;
GstFlowReturn res = GST_FLOW_OK;
GstBaseTransform *trans;
- r = audioresample->resample;
+ r = legacyresample->resample;
outsize = resample_get_output_size (r);
if (outsize == 0) {
- GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
+ GST_DEBUG_OBJECT (legacyresample, "no internal buffers needing flush");
goto done;
}
- trans = GST_BASE_TRANSFORM (audioresample);
+ trans = GST_BASE_TRANSFORM (legacyresample);
res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (trans->srcpad), &outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
+ GST_WARNING_OBJECT (legacyresample, "failed allocating buffer of %d bytes",
outsize);
goto done;
}
- res = audioresample_do_output (audioresample, outbuf);
+ res = legacyresample_do_output (legacyresample, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK))
goto done;
}
static gboolean
-audioresample_query (GstPad * pad, GstQuery * query)
+legacyresample_query (GstPad * pad, GstQuery * query)
{
- GstAudioresample *audioresample =
- GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
- GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample);
+ GstLegacyresample *legacyresample =
+ GST_LEGACYRESAMPLE (gst_pad_get_parent (pad));
+ GstBaseTransform *trans = GST_BASE_TRANSFORM (legacyresample);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
gboolean live;
guint64 latency;
GstPad *peer;
- gint rate = audioresample->i_rate;
- gint resampler_latency = audioresample->filter_length / 2;
+ gint rate = legacyresample->i_rate;
+ gint resampler_latency = legacyresample->filter_length / 2;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
res = gst_pad_query_default (pad, query);
break;
}
- gst_object_unref (audioresample);
+ gst_object_unref (legacyresample);
return res;
}
static const GstQueryType *
-audioresample_query_type (GstPad * pad)
+legacyresample_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
}
static void
-gst_audioresample_set_property (GObject * object, guint prop_id,
+gst_legacyresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
- audioresample = GST_AUDIORESAMPLE (object);
+ legacyresample = GST_LEGACYRESAMPLE (object);
switch (prop_id) {
case PROP_FILTERLEN:
- audioresample->filter_length = g_value_get_int (value);
- GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
- audioresample->filter_length);
- if (audioresample->resample) {
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
- gst_element_post_message (GST_ELEMENT (audioresample),
- gst_message_new_latency (GST_OBJECT (audioresample)));
+ legacyresample->filter_length = g_value_get_int (value);
+ GST_DEBUG_OBJECT (GST_ELEMENT (legacyresample), "new filter length %d",
+ legacyresample->filter_length);
+ if (legacyresample->resample) {
+ resample_set_filter_length (legacyresample->resample,
+ legacyresample->filter_length);
+ gst_element_post_message (GST_ELEMENT (legacyresample),
+ gst_message_new_latency (GST_OBJECT (legacyresample)));
}
break;
default:
}
static void
-gst_audioresample_get_property (GObject * object, guint prop_id,
+gst_legacyresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstLegacyresample *legacyresample;
- audioresample = GST_AUDIORESAMPLE (object);
+ legacyresample = GST_LEGACYRESAMPLE (object);
switch (prop_id) {
case PROP_FILTERLEN:
- g_value_set_int (value, audioresample->filter_length);
+ g_value_set_int (value, legacyresample->filter_length);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
resample_init ();
if (!gst_element_register (plugin, "legacyresample", GST_RANK_MARGINAL,
- GST_TYPE_AUDIORESAMPLE)) {
+ GST_TYPE_LEGACYRESAMPLE)) {
return FALSE;
}
*/
-#ifndef __AUDIORESAMPLE_H__
-#define __AUDIORESAMPLE_H__
+#ifndef __LEGACYRESAMPLE_H__
+#define __LEGACYRESAMPLE_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
G_BEGIN_DECLS
-#define GST_TYPE_AUDIORESAMPLE \
- (gst_audioresample_get_type())
-#define GST_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,GstAudioresample))
-#define GST_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,GstAudioresampleClass))
-#define GST_IS_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
-#define GST_IS_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
+#define GST_TYPE_LEGACYRESAMPLE \
+ (gst_legacyresample_get_type())
+#define GST_LEGACYRESAMPLE(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_LEGACYRESAMPLE,GstLegacyresample))
+#define GST_LEGACYRESAMPLE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_LEGACYRESAMPLE,GstLegacyresampleClass))
+#define GST_IS_LEGACYRESAMPLE(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_LEGACYRESAMPLE))
+#define GST_IS_LEGACYRESAMPLE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_LEGACYRESAMPLE))
-typedef struct _GstAudioresample GstAudioresample;
-typedef struct _GstAudioresampleClass GstAudioresampleClass;
+typedef struct _GstLegacyresample GstLegacyresample;
+typedef struct _GstLegacyresampleClass GstLegacyresampleClass;
/**
- * GstAudioresample:
+ * GstLegacyresample:
*
* Opaque data structure.
*/
-struct _GstAudioresample {
+struct _GstLegacyresample {
GstBaseTransform element;
GstCaps *srccaps, *sinkcaps;
ResampleState * resample;
};
-struct _GstAudioresampleClass {
+struct _GstLegacyresampleClass {
GstBaseTransformClass parent_class;
};
-GType gst_audioresample_get_type(void);
+GType gst_legacyresample_get_type(void);
G_END_DECLS
-#endif /* __AUDIORESAMPLE_H__ */
+#endif /* __LEGACYRESAMPLE_H__ */
$(check_x264enc) \
elements/aacparse \
elements/amrparse \
- elements/audioresample \
+ elements/legacyresample \
elements/qtmux \
elements/selector \
elements/mxfdemux \
/* GStreamer
*
- * unit test for audioresample
+ * unit test for legacyresample
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
);
static GstElement *
-setup_audioresample (int channels, int inrate, int outrate)
+setup_legacyresample (int channels, int inrate, int outrate)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstCaps *caps;
GstStructure *structure;
- GST_DEBUG ("setup_audioresample");
- audioresample = gst_check_setup_element ("legacyresample");
+ GST_DEBUG ("setup_legacyresample");
+ legacyresample = gst_check_setup_element ("legacyresample");
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
structure = gst_caps_get_structure (caps, 0);
"rate", G_TYPE_INT, inrate, NULL);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
"could not set to paused");
- mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
+ mysrcpad = gst_check_setup_src_pad (legacyresample, &srctemplate, caps);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
"rate", G_TYPE_INT, outrate, NULL);
fail_unless (gst_caps_is_fixed (caps));
- mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
+ mysinkpad = gst_check_setup_sink_pad (legacyresample, &sinktemplate, caps);
/* this installs a getcaps func that will always return the caps we set
* later */
gst_pad_set_caps (mysinkpad, caps);
gst_pad_set_active (mysinkpad, TRUE);
gst_pad_set_active (mysrcpad, TRUE);
- return audioresample;
+ return legacyresample;
}
static void
-cleanup_audioresample (GstElement * audioresample)
+cleanup_legacyresample (GstElement * legacyresample)
{
- GST_DEBUG ("cleanup_audioresample");
+ GST_DEBUG ("cleanup_legacyresample");
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
- gst_check_teardown_src_pad (audioresample);
- gst_check_teardown_sink_pad (audioresample);
- gst_check_teardown_element (audioresample);
+ gst_check_teardown_src_pad (legacyresample);
+ gst_check_teardown_sink_pad (legacyresample);
+ gst_check_teardown_element (legacyresample);
}
static void
test_perfect_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
guint64 offset = 0;
int i, j;
gint16 *p;
- audioresample = setup_audioresample (2, inrate, outrate);
+ legacyresample = setup_legacyresample (2, inrate, outrate);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
fail_unless_equals_int (g_list_length (buffers), j);
}
- /* FIXME: we should make audioresample handle eos by flushing out the last
+ /* FIXME: we should make legacyresample handle eos by flushing out the last
* samples, which will give us one more, small, buffer */
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
/* cleanup */
gst_caps_unref (caps);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
}
test_discont_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
GstClockTime ints;
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
inrate, outrate, samples, numbuffers);
- audioresample = setup_audioresample (2, inrate, outrate);
+ legacyresample = setup_legacyresample (2, inrate, outrate);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
/* cleanup */
gst_caps_unref (caps);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
}
GST_START_TEST (test_discont_stream)
GST_START_TEST (test_reuse)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstEvent *newseg;
GstBuffer *inbuffer;
GstCaps *caps;
- audioresample = setup_audioresample (1, 9343, 48000);
+ legacyresample = setup_legacyresample (1, 9343, 48000);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
fail_unless_equals_int (g_list_length (buffers), 1);
/* now reset and try again ... */
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... it also ends up being collected on the global buffer list. If we
- * now have more than 2 buffers, then audioresample probably didn't clean
+ * now have more than 2 buffers, then legacyresample probably didn't clean
* up its internal buffer properly and tried to push the remaining samples
* when it got the second NEWSEGMENT event */
fail_unless_equals_int (g_list_length (buffers), 2);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
gst_caps_unref (caps);
}
GstCaps *caps;
guint i;
- /* create pipeline, force audioresample to actually resample */
+ /* create pipeline, force legacyresample to actually resample */
pipeline = gst_pipeline_new (NULL);
src = gst_check_setup_element ("audiotestsrc");
GST_START_TEST (test_live_switch)
{
- GstElement *audioresample;
+ GstElement *legacyresample;
GstEvent *newseg;
GstCaps *caps;
- audioresample = setup_audioresample (4, 48000, 48000);
+ legacyresample = setup_legacyresample (4, 48000, 48000);
/* Let the sinkpad act like something that can only handle things of
* rate 48000- and can only allocate buffers for that rate, but if someone
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
- fail_unless (gst_element_set_state (audioresample,
+ fail_unless (gst_element_set_state (legacyresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
/* Downstream can provide the requested rate but will re-negotiate */
live_switch_push (50000, caps);
- cleanup_audioresample (audioresample);
+ cleanup_legacyresample (legacyresample);
gst_caps_unref (caps);
}
GST_END_TEST static Suite *
-audioresample_suite (void)
+legacyresample_suite (void)
{
- Suite *s = suite_create ("audioresample");
+ Suite *s = suite_create ("legacyresample");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
return s;
}
-GST_CHECK_MAIN (audioresample);
+GST_CHECK_MAIN (legacyresample);
projects / platform / upstream / gstreamer.git / commitdiff