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Rename audioresample files and types to legacyresample

Browse source 
Finish the move/rename of audioresample to legacyresample
to prevent any confusion.
This commit is contained in:
Sebastian Dröge 2009-01-23 12:46:28 +01:00
parent 6fec8619b5
commit e4e3b44e04
96 changed files with 1054 additions and 486 deletions
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4
configure.ac
View file

@ -239,7 +239,7 @@ dnl these are all the gst plug-ins, compilable without additional libs
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)
@ -1391,7 +1391,7 @@ gst/Makefile
gst/aacparse/Makefile
gst/aiffparse/Makefile
gst/amrparse/Makefile
gst/audioresample/Makefile
gst/legacyresample/Makefile
gst/bayer/Makefile
gst/cdxaparse/Makefile
gst/dccp/Makefile

2
docs/plugins/Makefile.am
View file

@ -115,7 +115,7 @@ EXTRA_HFILES = \
$(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 \

16
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -695,15 +695,15 @@ gst_stereo_get_type
<SECTION>
<FILE>element-legacyresample</FILE>
<TITLE>legacyresample</TITLE>
GstAudioresample
GstLegacyresample
<SUBSECTION Standard>
GstAudioresampleClass
GST_AUDIORESAMPLE
GST_AUDIORESAMPLE_CLASS
GST_IS_AUDIORESAMPLE
GST_IS_AUDIORESAMPLE_CLASS
GST_TYPE_AUDIORESAMPLE
gst_audioresample_get_type
GstLegacyresampleClass
GST_LEGACYRESAMPLE
GST_LEGACYRESAMPLE_CLASS
GST_IS_LEGACYRESAMPLE
GST_IS_LEGACYRESAMPLE_CLASS
GST_TYPE_LEGACYRESAMPLE
gst_legacyresample_get_type
</SECTION>
<SECTION>

52
docs/plugins/gst-plugins-bad-plugins.args
View file

@ -41,7 +41,7 @@
<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>
@ -91,7 +91,7 @@
<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>
@ -121,7 +121,7 @@
<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>
@ -151,7 +151,7 @@
<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>
@ -211,7 +211,7 @@
<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>
@ -221,7 +221,7 @@
<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>
@ -281,7 +281,7 @@
<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>
@ -291,7 +291,7 @@
<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>
@ -421,7 +421,7 @@
<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>
@ -1681,7 +1681,7 @@
<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>
@ -17455,7 +17455,7 @@
<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>
@ -17495,7 +17495,7 @@
<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>
@ -18431,7 +18431,7 @@
<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>
@ -22186,7 +22186,7 @@
<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>
@ -22336,7 +22336,7 @@
<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>
@ -22376,7 +22376,7 @@
<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>
@ -22436,7 +22436,7 @@
<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>
@ -22496,7 +22496,7 @@
<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>
@ -22556,7 +22556,7 @@
<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>
@ -22616,7 +22616,7 @@
<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>
@ -22636,7 +22636,7 @@
<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>
@ -22923,3 +22923,13 @@
<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>

164
docs/plugins/gst-plugins-bad-plugins.hierarchy
View file

@ -12,144 +12,139 @@ GObject
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
GstY4mEncode
GstSpeed
GstInterleave
GstDeinterleave
GstFreeze
GstDVDSpu
AIFFParse
GstAacBaseParse
GstAacParse
GstCDXAParse
GstVcdParse
GstNsfDec
GstTtaParse
GstTtaDec
GstNuvDemux
GstFLVDemux
GstFlvMux
GstMpegPSDemux
GstMpegTSDemux
MpegTSParse
GstDeinterlace2
GstModPlug
GstY4mEncode
GstFreeze
GstVMncDec
AIFFParse
GstSrtEnc
GstFestival
MpegTsMux
GstDVDSpu
GstMXFDemux
GstRealVideoDec
GstRealAudioDec
GstAmrBaseParse
GstAmrParse
GstPcapParse
GstBus
GstTask
GstClock
@ -162,8 +157,6 @@ GObject
GstJackAudioSinkRingBuffer
GstSignalObject
GstColorBalanceChannel
GstMixerTrack
GstMixerOptions
RTPSession
FluTsPatInfo
FluTsPmtInfo
@ -171,10 +164,11 @@ GInterface
GTypePlugin
GstChildProxy
GstURIHandler
GstTagSetter
GstImplementsInterface
GstNavigation
GstColorBalance
GstXOverlay
GstTagSetter
GstMixer
GstPropertyProbe
MXFDescriptiveMetadataFrameworkInterface

15
docs/plugins/gst-plugins-bad-plugins.interfaces
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@ -2,25 +2,24 @@ GstBin GstChildProxy
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

3
docs/plugins/gst-plugins-bad-plugins.prerequisites
View file

@ -1,6 +1,7 @@
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

4
docs/plugins/inspect/plugin-aacparse.xml
View file

@ -3,10 +3,10 @@
<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>

34
docs/plugins/inspect/plugin-aiffparse.xml Normal file
View file

@ -0,0 +1,34 @@
<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 &lt;songbird@songbirdnest.com&gt;</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>

4
docs/plugins/inspect/plugin-alsaspdif.xml
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@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-amrparse.xml
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@ -3,10 +3,10 @@
<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>

2
docs/plugins/inspect/plugin-amrwb.xml
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@ -3,7 +3,7 @@
<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>

28
docs/plugins/inspect/plugin-apex.xml Normal file
View file

@ -0,0 +1,28 @@
<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] &lt;gremimail@gmail.com&gt;</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>

4
docs/plugins/inspect/plugin-bayer.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-bz2.xml
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@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-cdaudio.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-cdxaparse.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-celt.xml
View file

@ -3,10 +3,10 @@
<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>

28
docs/plugins/inspect/plugin-dc1394.xml Normal file
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@ -0,0 +1,28 @@
<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 &lt;hexa00@gmail.com&gt;</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>

2
docs/plugins/inspect/plugin-dccp.xml
View file

@ -3,7 +3,7 @@
<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>

34
docs/plugins/inspect/plugin-deinterlace2.xml Normal file
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@ -0,0 +1,34 @@
<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 &lt;meiker@upb.de&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</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>

4
docs/plugins/inspect/plugin-dfbvideosink.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-dtsdec.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-dvb.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-dvdspu.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-faac.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-faad.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-fbdevsink.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-festival.xml
View file

@ -3,10 +3,10 @@
<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>

67
docs/plugins/inspect/plugin-flv.xml Normal file
View file

@ -0,0 +1,67 @@
<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 &lt;julien@moutte.net&gt;</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 &lt;sebastian.droege@collabora.co.uk&gt;</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>

4
docs/plugins/inspect/plugin-freeze.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-gsm.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-gstinterlace.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-gstrtpmanager.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-h264parse.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-jack.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-ladspa.xml
View file

@ -3,10 +3,10 @@
<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>

6
docs/plugins/inspect/plugin-legacyresample.xml
View file

@ -1,12 +1,12 @@
<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>

4
docs/plugins/inspect/plugin-metadata.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-mms.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-modplug.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-mpeg4videoparse.xml
View file

@ -3,10 +3,10 @@
<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>

107
docs/plugins/inspect/plugin-mpegdemux2.xml Normal file
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@ -0,0 +1,107 @@
<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 &lt;wim@fluendo.com&gt;</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 &lt;wim@fluendo.com&gt;</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 &lt;alessandro@nnva.org&gt;
Zaheer Abbas Merali &lt;zaheerabbas at merali dot org&gt;</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>

34
docs/plugins/inspect/plugin-mpegtsmux.xml Normal file
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@ -0,0 +1,34 @@
<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 &lt;contact@fluendo.com&gt;</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>

4
docs/plugins/inspect/plugin-mpegvideoparse.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-musepack.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-musicbrainz.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-mve.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-mxf.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-mythtv.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-nas.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-neon.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-nsfdec.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-nuvdemux.xml
View file

@ -3,10 +3,10 @@
<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>

34
docs/plugins/inspect/plugin-ofa.xml Normal file
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@ -0,0 +1,34 @@
<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 &lt;internalerror@gmail.com&gt;, Eric Buehl &lt;eric.buehl@gmail.com&gt;</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>

4
docs/plugins/inspect/plugin-oss4.xml
View file

@ -3,10 +3,10 @@
<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>

34
docs/plugins/inspect/plugin-pcapparse.xml Normal file
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@ -0,0 +1,34 @@
<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 &lt;ole.andre.ravnas@tandberg.com&gt;</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>

121
docs/plugins/inspect/plugin-qtmux.xml Normal file
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@ -0,0 +1,121 @@
<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 &lt;thiagoss@embedded.ufcg.edu.br&gt;</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 &lt;thiagoss@embedded.ufcg.edu.br&gt;</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 &lt;thiagoss@embedded.ufcg.edu.br&gt;</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 &lt;thiagoss@embedded.ufcg.edu.br&gt;</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>

4
docs/plugins/inspect/plugin-rawparse.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-real.xml
View file

@ -3,10 +3,10 @@
<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>

40
docs/plugins/inspect/plugin-resindvd.xml Normal file
View file

@ -0,0 +1,40 @@
<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 &lt;thaytan@noraisin.net&gt;</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>

4
docs/plugins/inspect/plugin-rfbsrc.xml
View file

@ -3,10 +3,10 @@
<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>

2
docs/plugins/inspect/plugin-scaletempo.xml
View file

@ -3,7 +3,7 @@
<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>

4
docs/plugins/inspect/plugin-sdl.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-sdp.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-selector.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-sndfile.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-soundtouch.xml
View file

@ -3,10 +3,10 @@
<description>Audio Pitch Controller &amp; 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>

4
docs/plugins/inspect/plugin-speed.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-stereo.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-subenc.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-tta.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-twolame.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-vcdsrc.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-videosignal.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-vmnc.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-wildmidi.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-x264.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-xvid.xml
View file

@ -3,10 +3,10 @@
<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>

4
docs/plugins/inspect/plugin-y4menc.xml
View file

@ -3,10 +3,10 @@
<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>

4
gst/audioresample/Makefile.am → gst/legacyresample/Makefile.am
View file

@ -10,12 +10,12 @@ resample_SOURCES = \
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)

0
gst/audioresample/buffer.c → gst/legacyresample/buffer.c
View file

0
gst/audioresample/buffer.h → gst/legacyresample/buffer.h
View file

0
gst/audioresample/debug.c → gst/legacyresample/debug.c
View file

0
gst/audioresample/debug.h → gst/legacyresample/debug.h
View file

0
gst/audioresample/functable.c → gst/legacyresample/functable.c
View file

0
gst/audioresample/functable.h → gst/legacyresample/functable.h
View file

357
gst/audioresample/gstaudioresample.c → gst/legacyresample/gstlegacyresample.c
View file

@ -44,15 +44,15 @@
#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",
@ -94,70 +94,71 @@ GST_STATIC_CAPS ( \
"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",
@ -165,82 +166,82 @@ gst_audioresample_class_init (GstAudioresampleClass * klass)
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;
@ -261,7 +262,7 @@ audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
}
static GstCaps *
audioresample_transform_caps (GstBaseTransform * base,
legacyresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *res;
@ -278,7 +279,7 @@ audioresample_transform_caps (GstBaseTransform * base,
/* 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;
@ -387,11 +388,11 @@ no_out_rate:
}
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 */
@ -409,15 +410,15 @@ audioresample_transform_size (GstBaseTransform * base,
/* 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);
}
@ -442,64 +443,64 @@ audioresample_transform_size (GstBaseTransform * base,
}
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;
@ -508,57 +509,59 @@ audioresample_event (GstBaseTransform * base, GstEvent * event)
}
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 */
@ -566,28 +569,28 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
/* 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)),
@ -599,22 +602,22 @@ audioresample_do_output (GstAudioresample * audioresample, GstBuffer * 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;
}
@ -624,23 +627,23 @@ audioresample_check_discont (GstAudioresample * audioresample,
}
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),
@ -648,16 +651,16 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
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)) {
@ -666,29 +669,29 @@ audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
/* 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;
@ -696,25 +699,25 @@ audioresample_pushthrough (GstAudioresample * audioresample)
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;
@ -725,11 +728,11 @@ 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)) {
@ -739,8 +742,8 @@ audioresample_query (GstPad * pad, GstQuery * 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;
@ -780,12 +783,12 @@ audioresample_query (GstPad * pad, GstQuery * query)
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,
@ -796,23 +799,23 @@ audioresample_query_type (GstPad * pad)
}
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:
@ -822,16 +825,16 @@ gst_audioresample_set_property (GObject * object, guint prop_id,
}
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);
@ -846,7 +849,7 @@ plugin_init (GstPlugin * plugin)
resample_init ();
if (!gst_element_register (plugin, "legacyresample", GST_RANK_MARGINAL,
GST_TYPE_AUDIORESAMPLE)) {
GST_TYPE_LEGACYRESAMPLE)) {
return FALSE;
}

38
gst/audioresample/gstaudioresample.h → gst/legacyresample/gstlegacyresample.h
View file

@ -18,8 +18,8 @@
*/
#ifndef __AUDIORESAMPLE_H__
#define __AUDIORESAMPLE_H__
#ifndef __LEGACYRESAMPLE_H__
#define __LEGACYRESAMPLE_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
@ -28,26 +28,26 @@
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;
@ -68,12 +68,12 @@ struct _GstAudioresample {
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__ */

0
gst/audioresample/resample.c → gst/legacyresample/resample.c
View file

0
gst/audioresample/resample.h → gst/legacyresample/resample.h
View file

0
gst/audioresample/resample_chunk.c → gst/legacyresample/resample_chunk.c
View file

0
gst/audioresample/resample_functable.c → gst/legacyresample/resample_functable.c
View file

0
gst/audioresample/resample_ref.c → gst/legacyresample/resample_ref.c
View file

2
tests/check/Makefile.am
View file

@ -89,7 +89,7 @@ check_PROGRAMS = \
$(check_x264enc) \
elements/aacparse \
elements/amrparse \
elements/audioresample \
elements/legacyresample \
elements/qtmux \
elements/selector \
elements/mxfdemux \

78
tests/check/elements/audioresample.c → tests/check/elements/legacyresample.c
View file

@ -1,6 +1,6 @@
/* 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>
@ -52,14 +52,14 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
);
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);
@ -67,11 +67,11 @@ setup_audioresample (int channels, int inrate, int outrate)
"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);
@ -81,7 +81,7 @@ setup_audioresample (int channels, int inrate, int outrate)
"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);
@ -90,22 +90,22 @@ setup_audioresample (int channels, int inrate, int outrate)
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
@ -145,7 +145,7 @@ 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;
@ -153,11 +153,11 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
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");
@ -188,7 +188,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
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);
@ -197,7 +197,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
/* cleanup */
gst_caps_unref (caps);
cleanup_audioresample (audioresample);
cleanup_legacyresample (legacyresample);
}
@ -229,7 +229,7 @@ static void
test_discont_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
GstElement *audioresample;
GstElement *legacyresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
GstClockTime ints;
@ -240,11 +240,11 @@ test_discont_stream_instance (int inrate, int outrate, int samples,
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");
@ -295,7 +295,7 @@ test_discont_stream_instance (int inrate, int outrate, int samples,
/* cleanup */
gst_caps_unref (caps);
cleanup_audioresample (audioresample);
cleanup_legacyresample (legacyresample);
}
GST_START_TEST (test_discont_stream)
@ -321,16 +321,16 @@ GST_END_TEST;
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");
@ -351,10 +351,10 @@ GST_START_TEST (test_reuse)
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");
@ -371,12 +371,12 @@ GST_START_TEST (test_reuse)
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);
}
@ -388,7 +388,7 @@ GST_START_TEST (test_shutdown)
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");
@ -512,11 +512,11 @@ live_switch_push (int rate, GstCaps * caps)
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
@ -528,7 +528,7 @@ GST_START_TEST (test_live_switch)
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");
@ -545,14 +545,14 @@ GST_START_TEST (test_live_switch)
/* 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);
@ -565,4 +565,4 @@ audioresample_suite (void)
return s;
}
GST_CHECK_MAIN (audioresample);
GST_CHECK_MAIN (legacyresample);