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| author | Sebastian Dröge <sebastian.droege@collabora.co.uk> | 2009-01-23 12:46:28 +0100 |
|---|---|---|
| committer | Sebastian Dröge <sebastian.droege@collabora.co.uk> | 2009-01-23 12:47:19 +0100 |
| commit | e4e3b44e048ddc1d7499c6108175a5f89c6273d9 (patch) | |
| tree | f88b685f1b6baf849649494ec557d2ef0ef13a88 | |
| parent | 6fec8619b597f5cc9c58d268ddd9f64ea0a94277 (diff) | |
| download | gst-plugins-bad-e4e3b44e048ddc1d7499c6108175a5f89c6273d9.tar.gz gst-plugins-bad-e4e3b44e048ddc1d7499c6108175a5f89c6273d9.tar.bz2 gst-plugins-bad-e4e3b44e048ddc1d7499c6108175a5f89c6273d9.zip | |
Rename audioresample files and types to legacyresample
Finish the move/rename of audioresample to legacyresample
to prevent any confusion.
96 files changed, 1054 insertions, 486 deletions
diff --git a/configure.ac b/configure.ac index ec8e9a57..33d780c1 100644 --- a/configure.ac +++ b/configure.ac @@ -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 diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index f132618d..4c058d5d 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -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 \ diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt index 4c52bf49..c396f5d3 100644 --- a/docs/plugins/gst-plugins-bad-plugins-sections.txt +++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt @@ -695,15 +695,15 @@ gst_stereo_get_type <SECTION> <FILE>element-legacyresample</FILE> <TITLE>legacyresample</TITLE> -GstAudioresample -<SUBSECTION Standard> -GstAudioresampleClass -GST_AUDIORESAMPLE -GST_AUDIORESAMPLE_CLASS -GST_IS_AUDIORESAMPLE -GST_IS_AUDIORESAMPLE_CLASS -GST_TYPE_AUDIORESAMPLE -gst_audioresample_get_type +GstLegacyresample +<SUBSECTION Standard> +GstLegacyresampleClass +GST_LEGACYRESAMPLE +GST_LEGACYRESAMPLE_CLASS +GST_IS_LEGACYRESAMPLE +GST_IS_LEGACYRESAMPLE_CLASS +GST_TYPE_LEGACYRESAMPLE +gst_legacyresample_get_type </SECTION> <SECTION> diff --git a/docs/plugins/gst-plugins-bad-plugins.args b/docs/plugins/gst-plugins-bad-plugins.args index 7e68fee6..77142e00 100644 --- a/docs/plugins/gst-plugins-bad-plugins.args +++ b/docs/plugins/gst-plugins-bad-plugins.args @@ -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> + diff --git a/docs/plugins/gst-plugins-bad-plugins.hierarchy b/docs/plugins/gst-plugins-bad-plugins.hierarchy index e4b44632..78502bee 100644 --- a/docs/plugins/gst-plugins-bad-plugins.hierarchy +++ b/docs/plugins/gst-plugins-bad-plugins.hierarchy @@ -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 + GstAacBaseParse + GstAacParse + GstCDXAParse + GstVcdParse + GstNsfDec + GstTtaParse + GstTtaDec + GstModPlug GstY4mEncode - GstSpeed - GstInterleave - GstDeinterleave GstFreeze - GstDVDSpu + GstVMncDec AIFFParse - GstTtaParse - GstTtaDec - GstNuvDemux - GstFLVDemux - GstFlvMux - GstMpegPSDemux - GstMpegTSDemux - MpegTSParse - GstDeinterlace2 + GstSrtEnc + GstFestival + MpegTsMux + GstDVDSpu + GstMXFDemux + GstRealVideoDec + GstRealAudioDec + GstAmrBaseParse + GstAmrParse + GstPcapParse GstBus GstTask GstClock @@ -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 diff --git a/docs/plugins/gst-plugins-bad-plugins.interfaces b/docs/plugins/gst-plugins-bad-plugins.interfaces index c155afcf..17592fe2 100644 --- a/docs/plugins/gst-plugins-bad-plugins.interfaces +++ b/docs/plugins/gst-plugins-bad-plugins.interfaces @@ -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 diff --git a/docs/plugins/gst-plugins-bad-plugins.prerequisites b/docs/plugins/gst-plugins-bad-plugins.prerequisites index 81371d2d..fb1c089f 100644 --- a/docs/plugins/gst-plugins-bad-plugins.prerequisites +++ b/docs/plugins/gst-plugins-bad-plugins.prerequisites @@ -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 diff --git a/docs/plugins/inspect/plugin-aacparse.xml b/docs/plugins/inspect/plugin-aacparse.xml index 5191407d..b67f222e 100644 --- a/docs/plugins/inspect/plugin-aacparse.xml +++ b/docs/plugins/inspect/plugin-aacparse.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-aiffparse.xml b/docs/plugins/inspect/plugin-aiffparse.xml new file mode 100644 index 00000000..4d53c2d4 --- /dev/null +++ b/docs/plugins/inspect/plugin-aiffparse.xml @@ -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 <songbird@songbirdnest.com></author> + <pads> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int){ 1234, 4321 }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], signed=(boolean){ true, false }</details> + </caps> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>audio/x-aiff</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-alsaspdif.xml b/docs/plugins/inspect/plugin-alsaspdif.xml index f601afe2..e3cb19ed 100644 --- a/docs/plugins/inspect/plugin-alsaspdif.xml +++ b/docs/plugins/inspect/plugin-alsaspdif.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-amrparse.xml b/docs/plugins/inspect/plugin-amrparse.xml index 6edad58f..439d023c 100644 --- a/docs/plugins/inspect/plugin-amrparse.xml +++ b/docs/plugins/inspect/plugin-amrparse.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-amrwb.xml b/docs/plugins/inspect/plugin-amrwb.xml index 26900583..ff4fb7b0 100644 --- a/docs/plugins/inspect/plugin-amrwb.xml +++ b/docs/plugins/inspect/plugin-amrwb.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-apex.xml b/docs/plugins/inspect/plugin-apex.xml new file mode 100644 index 00000000..9a639c9b --- /dev/null +++ b/docs/plugins/inspect/plugin-apex.xml @@ -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] <gremimail@gmail.com></author> + <pads> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, channels=(int)2, rate=(int)44100, signed=(boolean)true</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-bayer.xml b/docs/plugins/inspect/plugin-bayer.xml index bb7d6c37..87cf4a06 100644 --- a/docs/plugins/inspect/plugin-bayer.xml +++ b/docs/plugins/inspect/plugin-bayer.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-bz2.xml b/docs/plugins/inspect/plugin-bz2.xml index 6332dd1d..a36b935b 100644 --- a/docs/plugins/inspect/plugin-bz2.xml +++ b/docs/plugins/inspect/plugin-bz2.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-cdaudio.xml b/docs/plugins/inspect/plugin-cdaudio.xml index eff01e12..d15f5b54 100644 --- a/docs/plugins/inspect/plugin-cdaudio.xml +++ b/docs/plugins/inspect/plugin-cdaudio.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-cdxaparse.xml b/docs/plugins/inspect/plugin-cdxaparse.xml index 92090815..f57d1fbc 100644 --- a/docs/plugins/inspect/plugin-cdxaparse.xml +++ b/docs/plugins/inspect/plugin-cdxaparse.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-celt.xml b/docs/plugins/inspect/plugin-celt.xml index 8336eef7..59d7cf6b 100644 --- a/docs/plugins/inspect/plugin-celt.xml +++ b/docs/plugins/inspect/plugin-celt.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-dc1394.xml b/docs/plugins/inspect/plugin-dc1394.xml new file mode 100644 index 00000000..842184b6 --- /dev/null +++ b/docs/plugins/inspect/plugin-dc1394.xml @@ -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 <hexa00@gmail.com></author> + <pads> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>video/x-raw-yuv, format=(fourcc)IYU2, bpp=(int)16, width=(int)160, height=(int)120, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)64; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)320, height=(int)240, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)65; video/x-raw-yuv, format=(fourcc)IYU1, bpp=(int)12, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)66; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)67; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)68; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)69; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)640, height=(int)480, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)70; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)71; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)72; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)73; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)74; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)75; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)76; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)800, height=(int)600, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)77; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)1024, height=(int)768, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)78; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)79; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)80; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)81; video/x-raw-yuv, format=(fourcc)UYVY, bpp=(int)16, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)82; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)83; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)84; video/x-raw-gray, bpp=(int)16, depth=(int)16, width=(int)1280, height=(int)960, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)85; video/x-raw-gray, bpp=(int)8, depth=(int)8, width=(int)1600, height=(int)1200, framerate=(fraction)[ 0/1, 2147483647/1 ], vmode=(int)86; video/x-raw-gray, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], bpp=(int)8, depth=(int)8; video/x-raw-yuv, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], format=(fourcc)IYU1, bpp=(int)12; video/x-raw-yuv, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], format=(fourcc)UYVY, bpp=(int)16; video/x-raw-yuv, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], format=(fourcc)IYU2, bpp=(int)16; video/x-raw-rgb, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255; video/x-raw-gray, vmode=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], bpp=(int)16, depth=(int)16</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-dccp.xml b/docs/plugins/inspect/plugin-dccp.xml index 56223206..fecf307e 100644 --- a/docs/plugins/inspect/plugin-dccp.xml +++ b/docs/plugins/inspect/plugin-dccp.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-deinterlace2.xml b/docs/plugins/inspect/plugin-deinterlace2.xml new file mode 100644 index 00000000..3c04fdbb --- /dev/null +++ b/docs/plugins/inspect/plugin-deinterlace2.xml @@ -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 <meiker@upb.de>, Sebastian Dröge <slomo@circular-chaos.org></author> + <pads> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details> + </caps> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-dfbvideosink.xml b/docs/plugins/inspect/plugin-dfbvideosink.xml index 3837f8f4..1610b78f 100644 --- a/docs/plugins/inspect/plugin-dfbvideosink.xml +++ b/docs/plugins/inspect/plugin-dfbvideosink.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-dtsdec.xml b/docs/plugins/inspect/plugin-dtsdec.xml index 353ccef6..3e110449 100644 --- a/docs/plugins/inspect/plugin-dtsdec.xml +++ b/docs/plugins/inspect/plugin-dtsdec.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-dvb.xml b/docs/plugins/inspect/plugin-dvb.xml index d1672d68..bb9342d1 100644 --- a/docs/plugins/inspect/plugin-dvb.xml +++ b/docs/plugins/inspect/plugin-dvb.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-dvdspu.xml b/docs/plugins/inspect/plugin-dvdspu.xml index 32747b52..4fd0900f 100644 --- a/docs/plugins/inspect/plugin-dvdspu.xml +++ b/docs/plugins/inspect/plugin-dvdspu.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-faac.xml b/docs/plugins/inspect/plugin-faac.xml index 37bfa0e5..65f84348 100644 --- a/docs/plugins/inspect/plugin-faac.xml +++ b/docs/plugins/inspect/plugin-faac.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-faad.xml b/docs/plugins/inspect/plugin-faad.xml index 1039d796..2d4d328e 100644 --- a/docs/plugins/inspect/plugin-faad.xml +++ b/docs/plugins/inspect/plugin-faad.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-fbdevsink.xml b/docs/plugins/inspect/plugin-fbdevsink.xml index f26c4c58..df50d869 100644 --- a/docs/plugins/inspect/plugin-fbdevsink.xml +++ b/docs/plugins/inspect/plugin-fbdevsink.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-festival.xml b/docs/plugins/inspect/plugin-festival.xml index dc4e5f1e..d0bf58cc 100644 --- a/docs/plugins/inspect/plugin-festival.xml +++ b/docs/plugins/inspect/plugin-festival.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-flv.xml b/docs/plugins/inspect/plugin-flv.xml new file mode 100644 index 00000000..99f8ec2d --- /dev/null +++ b/docs/plugins/inspect/plugin-flv.xml @@ -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 <julien@moutte.net></author> + <pads> + <caps> + <name>video</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>ANY</details> + </caps> + <caps> + <name>audio</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>ANY</details> + </caps> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>video/x-flv</details> + </caps> + </pads> + </element> + <element> + <name>flvmux</name> + <longname>FLV muxer</longname> + <class>Codec/Muxer</class> + <description>Muxes video/audio streams into a FLV stream</description> + <author>Sebastian Dröge <sebastian.droege@collabora.co.uk></author> + <pads> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>video/x-flv</details> + </caps> + <caps> + <name>audio</name> + <direction>sink</direction> + <presence>request</presence> + <details>audio/x-adpcm, layout=(string)swf, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 22050, 44100 }; audio/mpeg, mpegversion=(int)4; audio/x-nellymoser, channels=(int){ 1, 2 }, rate=(int){ 5512, 8000, 11025, 16000, 22050, 44100 }; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)8, depth=(int)8, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)false; audio/x-raw-int, endianness=(int)1234, channels=(int){ 1, 2 }, width=(int)16, depth=(int)16, rate=(int){ 5512, 11025, 22050, 44100 }, signed=(boolean)true; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }; audio/x-speex, channels=(int){ 1, 2 }, rate=(int){ 5512, 11025, 22050, 44100 }</details> + </caps> + <caps> + <name>video</name> + <direction>sink</direction> + <presence>request</presence> + <details>video/x-flash-video; video/x-flash-screen; video/x-vp6-flash; video/x-vp6-alpha; video/x-h264</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-freeze.xml b/docs/plugins/inspect/plugin-freeze.xml index 1051e0ac..5d73eaa5 100644 --- a/docs/plugins/inspect/plugin-freeze.xml +++ b/docs/plugins/inspect/plugin-freeze.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-gsm.xml b/docs/plugins/inspect/plugin-gsm.xml index b019cf82..e769aafc 100644 --- a/docs/plugins/inspect/plugin-gsm.xml +++ b/docs/plugins/inspect/plugin-gsm.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-gstinterlace.xml b/docs/plugins/inspect/plugin-gstinterlace.xml index 221cb078..9685be55 100644 --- a/docs/plugins/inspect/plugin-gstinterlace.xml +++ b/docs/plugins/inspect/plugin-gstinterlace.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml index 473e9209..5737a0bc 100644 --- a/docs/plugins/inspect/plugin-gstrtpmanager.xml +++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-h264parse.xml b/docs/plugins/inspect/plugin-h264parse.xml index da713917..dc69a91f 100644 --- a/docs/plugins/inspect/plugin-h264parse.xml +++ b/docs/plugins/inspect/plugin-h264parse.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-jack.xml b/docs/plugins/inspect/plugin-jack.xml index cdd3a278..0f18f4e3 100644 --- a/docs/plugins/inspect/plugin-jack.xml +++ b/docs/plugins/inspect/plugin-jack.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-ladspa.xml b/docs/plugins/inspect/plugin-ladspa.xml index ea45d62d..e6f7b9c0 100644 --- a/docs/plugins/inspect/plugin-ladspa.xml +++ b/docs/plugins/inspect/plugin-ladspa.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-legacyresample.xml b/docs/plugins/inspect/plugin-legacyresample.xml index 1d2e48ae..d5641242 100644 --- a/docs/plugins/inspect/plugin-legacyresample.xml +++ b/docs/plugins/inspect/plugin-legacyresample.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-metadata.xml b/docs/plugins/inspect/plugin-metadata.xml index 3415e3b9..bbfe1fee 100644 --- a/docs/plugins/inspect/plugin-metadata.xml +++ b/docs/plugins/inspect/plugin-metadata.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-mms.xml b/docs/plugins/inspect/plugin-mms.xml index 036d02a5..058392c8 100644 --- a/docs/plugins/inspect/plugin-mms.xml +++ b/docs/plugins/inspect/plugin-mms.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-modplug.xml b/docs/plugins/inspect/plugin-modplug.xml index 65a955e6..22a986b3 100644 --- a/docs/plugins/inspect/plugin-modplug.xml +++ b/docs/plugins/inspect/plugin-modplug.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-mpeg4videoparse.xml b/docs/plugins/inspect/plugin-mpeg4videoparse.xml index d7b1e384..e62d3522 100644 --- a/docs/plugins/inspect/plugin-mpeg4videoparse.xml +++ b/docs/plugins/inspect/plugin-mpeg4videoparse.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-mpegdemux2.xml b/docs/plugins/inspect/plugin-mpegdemux2.xml new file mode 100644 index 00000000..2c2fd45c --- /dev/null +++ b/docs/plugins/inspect/plugin-mpegdemux2.xml @@ -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 <wim@fluendo.com></author> + <pads> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>video/mpeg, mpegversion=(int){ 1, 2 }, systemstream=(boolean)true; video/x-cdxa</details> + </caps> + <caps> + <name>private_%d</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>ANY</details> + </caps> + <caps> + <name>audio_%02x</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>audio/mpeg, mpegversion=(int)1; audio/x-private1-lpcm; audio/x-private1-ac3; audio/x-private1-dts; audio/ac3</details> + </caps> + <caps> + <name>video_%02x</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false; video/x-h264</details> + </caps> + </pads> + </element> + <element> + <name>mpegtsdemux</name> + <longname>The Fluendo MPEG Transport stream demuxer</longname> + <class>Codec/Demuxer</class> + <description>Demultiplexes MPEG2 Transport Streams</description> + <author>Wim Taymans <wim@fluendo.com></author> + <pads> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>video/mpegts</details> + </caps> + <caps> + <name>private_%04x</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>ANY</details> + </caps> + <caps> + <name>audio_%04x</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>audio/mpeg, mpegversion=(int){ 1, 4 }; audio/x-lpcm, width=(int){ 16, 20, 24 }, rate=(int){ 48000, 96000 }, channels=(int)[ 1, 8 ], dynamic_range=(int)[ 0, 255 ], emphasis=(boolean){ false, true }, mute=(boolean){ false, true }; audio/x-ac3; audio/x-dts</details> + </caps> + <caps> + <name>video_%04x</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false; video/x-h264; video/x-dirac</details> + </caps> + </pads> + </element> + <element> + <name>mpegtsparse</name> + <longname>MPEG transport stream parser</longname> + <class>Codec/Parser</class> + <description>Parses MPEG2 transport streams</description> + <author>Alessandro Decina <alessandro@nnva.org> + Zaheer Abbas Merali <zaheerabbas at merali dot org></author> + <pads> + <caps> + <name>program_%d</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>video/mpegts, systemstream=(boolean)true</details> + </caps> + <caps> + <name>src%d</name> + <direction>source</direction> + <presence>request</presence> + <details>video/mpegts, systemstream=(boolean)true</details> + </caps> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>video/mpegts, systemstream=(boolean)true</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-mpegtsmux.xml b/docs/plugins/inspect/plugin-mpegtsmux.xml new file mode 100644 index 00000000..ac2a0649 --- /dev/null +++ b/docs/plugins/inspect/plugin-mpegtsmux.xml @@ -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 <contact@fluendo.com></author> + <pads> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>video/mpegts, systemstream=(boolean)true, packetsize=(int){ 188, 192 }</details> + </caps> + <caps> + <name>sink_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>video/mpeg, mpegversion=(int){ 1, 2, 4 }, systemstream=(boolean)false; video/x-dirac; video/x-h264; audio/mpeg, mpegversion=(int){ 1, 2, 4 }; audio/x-lpcm, width=(int){ 16, 20, 24 }, rate=(int){ 48000, 96000 }, channels=(int)[ 1, 8 ], dynamic_range=(int)[ 0, 255 ], emphasis=(boolean){ false, true }, mute=(boolean){ false, true }; audio/x-ac3; audio/x-dts</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-mpegvideoparse.xml b/docs/plugins/inspect/plugin-mpegvideoparse.xml index 80001d10..995ef241 100644 --- a/docs/plugins/inspect/plugin-mpegvideoparse.xml +++ b/docs/plugins/inspect/plugin-mpegvideoparse.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-musepack.xml b/docs/plugins/inspect/plugin-musepack.xml index 0813992d..2d628ef1 100644 --- a/docs/plugins/inspect/plugin-musepack.xml +++ b/docs/plugins/inspect/plugin-musepack.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-musicbrainz.xml b/docs/plugins/inspect/plugin-musicbrainz.xml index 6a90fb5f..94c5f2d7 100644 --- a/docs/plugins/inspect/plugin-musicbrainz.xml +++ b/docs/plugins/inspect/plugin-musicbrainz.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-mve.xml b/docs/plugins/inspect/plugin-mve.xml index e5d510ea..bc0a77fe 100644 --- a/docs/plugins/inspect/plugin-mve.xml +++ b/docs/plugins/inspect/plugin-mve.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-mxf.xml b/docs/plugins/inspect/plugin-mxf.xml index cda6aa0b..ed01db6e 100644 --- a/docs/plugins/inspect/plugin-mxf.xml +++ b/docs/plugins/inspect/plugin-mxf.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-mythtv.xml b/docs/plugins/inspect/plugin-mythtv.xml index 2f7885bd..cc553da8 100644 --- a/docs/plugins/inspect/plugin-mythtv.xml +++ b/docs/plugins/inspect/plugin-mythtv.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-nas.xml b/docs/plugins/inspect/plugin-nas.xml index a258b20b..8fc7fc01 100644 --- a/docs/plugins/inspect/plugin-nas.xml +++ b/docs/plugins/inspect/plugin-nas.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-neon.xml b/docs/plugins/inspect/plugin-neon.xml index b2a365fb..cd71f15b 100644 --- a/docs/plugins/inspect/plugin-neon.xml +++ b/docs/plugins/inspect/plugin-neon.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-nsfdec.xml b/docs/plugins/inspect/plugin-nsfdec.xml index 129cdc65..32eff1fb 100644 --- a/docs/plugins/inspect/plugin-nsfdec.xml +++ b/docs/plugins/inspect/plugin-nsfdec.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-nuvdemux.xml b/docs/plugins/inspect/plugin-nuvdemux.xml index 130530f3..ef232827 100644 --- a/docs/plugins/inspect/plugin-nuvdemux.xml +++ b/docs/plugins/inspect/plugin-nuvdemux.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-ofa.xml b/docs/plugins/inspect/plugin-ofa.xml new file mode 100644 index 00000000..88d00b0b --- /dev/null +++ b/docs/plugins/inspect/plugin-ofa.xml @@ -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 <internalerror@gmail.com>, Eric Buehl <eric.buehl@gmail.com></author> + <pads> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int){ 1234, 4321 }, width=(int){ 16 }, depth=(int){ 16 }, signed=(boolean)true</details> + </caps> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int){ 1234, 4321 }, width=(int){ 16 }, depth=(int){ 16 }, signed=(boolean)true</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-oss4.xml b/docs/plugins/inspect/plugin-oss4.xml index 0e0280ce..f0ab2ada 100644 --- a/docs/plugins/inspect/plugin-oss4.xml +++ b/docs/plugins/inspect/plugin-oss4.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-pcapparse.xml b/docs/plugins/inspect/plugin-pcapparse.xml new file mode 100644 index 00000000..ce679f7d --- /dev/null +++ b/docs/plugins/inspect/plugin-pcapparse.xml @@ -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 <ole.andre.ravnas@tandberg.com></author> + <pads> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>ANY</details> + </caps> + <caps> + <name>sink</name> + <direction>sink</direction> + <presence>always</presence> + <details>raw/x-pcap</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-qtmux.xml b/docs/plugins/inspect/plugin-qtmux.xml new file mode 100644 index 00000000..fd58973c --- /dev/null +++ b/docs/plugins/inspect/plugin-qtmux.xml @@ -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 <thiagoss@embedded.ufcg.edu.br></author> + <pads> + <caps> + <name>video_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>video/x-h264, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details> + </caps> + <caps> + <name>audio_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>audio/AMR, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]</details> + </caps> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>application/x-3gp</details> + </caps> + </pads> + </element> + <element> + <name>mj2mux</name> + <longname>MJ2 Muxer</longname> + <class>Codec/Muxer</class> + <description>Multiplex audio and video into a MJ2 file</description> + <author>Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br></author> + <pads> + <caps> + <name>video_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>image/x-j2c, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details> + </caps> + <caps> + <name>audio_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>audio/x-raw-int, width=(int)8, depth=(int)8, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean){ true, false }; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true</details> + </caps> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>video/mj2</details> + </caps> + </pads> + </element> + <element> + <name>mp4mux</name> + <longname>MP4 Muxer</longname> + <class>Codec/Muxer</class> + <description>Multiplex audio and video into a MP4 file</description> + <author>Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br></author> + <pads> + <caps> + <name>video_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-mp4-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details> + </caps> + <caps> + <name>audio_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]</details> + </caps> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>video/quicktime</details> + </caps> + </pads> + </element> + <element> + <name>qtmux</name> + <longname>QuickTime Muxer</longname> + <class>Codec/Muxer</class> + <description>Multiplex audio and video into a QuickTime file</description> + <author>Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br></author> + <pads> + <caps> + <name>video_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>video/x-raw-rgb, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-yuv, format=(fourcc)UYVY, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h263, h263version=(string)h263, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/mpeg, mpegversion=(int)4, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-divx, divxversion=(int)5, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-h264, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-dv, systemstream=(boolean)false, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-qt-part, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details> + </caps> + <caps> + <name>audio_%d</name> + <direction>sink</direction> + <presence>request</presence> + <details>audio/x-raw-int, width=(int)8, depth=(int)8, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean){ true, false }; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ], signed=(boolean)true; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-mulaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]</details> + </caps> + <caps> + <name>src</name> + <direction>source</direction> + <presence>always</presence> + <details>video/quicktime</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-rawparse.xml b/docs/plugins/inspect/plugin-rawparse.xml index cecce239..67b907f5 100644 --- a/docs/plugins/inspect/plugin-rawparse.xml +++ b/docs/plugins/inspect/plugin-rawparse.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-real.xml b/docs/plugins/inspect/plugin-real.xml index 490a632e..5eed48fc 100644 --- a/docs/plugins/inspect/plugin-real.xml +++ b/docs/plugins/inspect/plugin-real.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-resindvd.xml b/docs/plugins/inspect/plugin-resindvd.xml new file mode 100644 index 00000000..b6b2393c --- /dev/null +++ b/docs/plugins/inspect/plugin-resindvd.xml @@ -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 <thaytan@noraisin.net></author> + <pads> + <caps> + <name>subpicture</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>video/x-dvd-subpicture</details> + </caps> + <caps> + <name>audio</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>audio/x-raw-int; audio/x-raw-float</details> + </caps> + <caps> + <name>video</name> + <direction>source</direction> + <presence>sometimes</presence> + <details>video/mpeg, mpegversion=(int){ 1, 2 }, systemstream=(boolean)false</details> + </caps> + </pads> + </element> + </elements> +</plugin>
\ No newline at end of file diff --git a/docs/plugins/inspect/plugin-rfbsrc.xml b/docs/plugins/inspect/plugin-rfbsrc.xml index 016c7294..2488cf07 100644 --- a/docs/plugins/inspect/plugin-rfbsrc.xml +++ b/docs/plugins/inspect/plugin-rfbsrc.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-scaletempo.xml b/docs/plugins/inspect/plugin-scaletempo.xml index 896ad98a..f2bc9da9 100644 --- a/docs/plugins/inspect/plugin-scaletempo.xml +++ b/docs/plugins/inspect/plugin-scaletempo.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-sdl.xml b/docs/plugins/inspect/plugin-sdl.xml index f2d9a6b5..2b2090d5 100644 --- a/docs/plugins/inspect/plugin-sdl.xml +++ b/docs/plugins/inspect/plugin-sdl.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-sdp.xml b/docs/plugins/inspect/plugin-sdp.xml index 809c1d09..0add3049 100644 --- a/docs/plugins/inspect/plugin-sdp.xml +++ b/docs/plugins/inspect/plugin-sdp.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-selector.xml b/docs/plugins/inspect/plugin-selector.xml index 26029515..596a45fc 100644 --- a/docs/plugins/inspect/plugin-selector.xml +++ b/docs/plugins/inspect/plugin-selector.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-sndfile.xml b/docs/plugins/inspect/plugin-sndfile.xml index 89dfa504..4a3668bc 100644 --- a/docs/plugins/inspect/plugin-sndfile.xml +++ b/docs/plugins/inspect/plugin-sndfile.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-soundtouch.xml b/docs/plugins/inspect/plugin-soundtouch.xml index e000752c..0e567f64 100644 --- a/docs/plugins/inspect/plugin-soundtouch.xml +++ b/docs/plugins/inspect/plugin-soundtouch.xml @@ -3,10 +3,10 @@ <description>Audio Pitch Controller & BPM Detection</description> <filename>../../ext/soundtouch/.libs/libgstsoundtouch.so</filename> <basename>libgstsoundtouch.so</basename> - <version>0.10.10</version> + <version>0.10.10.1</version> <license>LGPL</license> <source>gst-plugins-bad</source> - <package>GStreamer Bad Plug-ins source release</package> + <package>GStreamer Bad Plug-ins CVS/prerelease</package> <origin>Unknown package origin</origin> <elements> <element> diff --git a/docs/plugins/inspect/plugin-speed.xml b/docs/plugins/inspect/plugin-speed.xml index 2c60338e..871aa169 100644 --- a/docs/plugins/inspect/plugin-speed.xml +++ b/docs/plugins/inspect/plugin-speed.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-stereo.xml b/docs/plugins/inspect/plugin-stereo.xml index 1f4cd741..e713b4d0 100644 --- a/docs/plugins/inspect/plugin-stereo.xml +++ b/docs/plugins/inspect/plugin-stereo.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-subenc.xml b/docs/plugins/inspect/plugin-subenc.xml index 72725e17..444ad2a7 100644 --- a/docs/plugins/inspect/plugin-subenc.xml +++ b/docs/plugins/inspect/plugin-subenc.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-tta.xml b/docs/plugins/inspect/plugin-tta.xml index 624fc34e..23e1e3b2 100644 --- a/docs/plugins/inspect/plugin-tta.xml +++ b/docs/plugins/inspect/plugin-tta.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-twolame.xml b/docs/plugins/inspect/plugin-twolame.xml index 5695861f..b7e96967 100644 --- a/docs/plugins/inspect/plugin-twolame.xml +++ b/docs/plugins/inspect/plugin-twolame.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-vcdsrc.xml b/docs/plugins/inspect/plugin-vcdsrc.xml index d9e742d5..92cc7a00 100644 --- a/docs/plugins/inspect/plugin-vcdsrc.xml +++ b/docs/plugins/inspect/plugin-vcdsrc.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-videosignal.xml b/docs/plugins/inspect/plugin-videosignal.xml index 4d4e34f8..7a580e8c 100644 --- a/docs/plugins/inspect/plugin-videosignal.xml +++ b/docs/plugins/inspect/plugin-videosignal.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-vmnc.xml b/docs/plugins/inspect/plugin-vmnc.xml index 1c261390..b1004c1d 100644 --- a/docs/plugins/inspect/plugin-vmnc.xml +++ b/docs/plugins/inspect/plugin-vmnc.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-wildmidi.xml b/docs/plugins/inspect/plugin-wildmidi.xml index 741f039e..3d917be5 100644 --- a/docs/plugins/inspect/plugin-wildmidi.xml +++ b/docs/plugins/inspect/plugin-wildmidi.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-x264.xml b/docs/plugins/inspect/plugin-x264.xml index c62d8c26..8dc4ae31 100644 --- a/docs/plugins/inspect/plugin-x264.xml +++ b/docs/plugins/inspect/plugin-x264.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-xvid.xml b/docs/plugins/inspect/plugin-xvid.xml index e2dd49fa..6d0bcb83 100644 --- a/docs/plugins/inspect/plugin-xvid.xml +++ b/docs/plugins/inspect/plugin-xvid.xml @@ -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> diff --git a/docs/plugins/inspect/plugin-y4menc.xml b/docs/plugins/inspect/plugin-y4menc.xml index 1bee89c0..6f2eaa05 100644 --- a/docs/plugins/inspect/plugin-y4menc.xml +++ b/docs/plugins/inspect/plugin-y4menc.xml @@ -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> diff --git a/gst/audioresample/Makefile.am b/gst/legacyresample/Makefile.am index c08ab262..41abdc67 100644 --- a/gst/audioresample/Makefile.am +++ b/gst/legacyresample/Makefile.am @@ -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) diff --git a/gst/audioresample/buffer.c b/gst/legacyresample/buffer.c index 442b4f8c..442b4f8c 100644 --- a/gst/audioresample/buffer.c +++ b/gst/legacyresample/buffer.c diff --git a/gst/audioresample/buffer.h b/gst/legacyresample/buffer.h index 4cf1fd94..4cf1fd94 100644 --- a/gst/audioresample/buffer.h +++ b/gst/legacyresample/buffer.h diff --git a/gst/audioresample/debug.c b/gst/legacyresample/debug.c index 27877277..27877277 100644 --- a/gst/audioresample/debug.c +++ b/gst/legacyresample/debug.c diff --git a/gst/audioresample/debug.h b/gst/legacyresample/debug.h index ff7deafb..ff7deafb 100644 --- a/gst/audioresample/debug.h +++ b/gst/legacyresample/debug.h diff --git a/gst/audioresample/functable.c b/gst/legacyresample/functable.c index d627361f..d627361f 100644 --- a/gst/audioresample/functable.c +++ b/gst/legacyresample/functable.c diff --git a/gst/audioresample/functable.h b/gst/legacyresample/functable.h index 5f56e2bd..5f56e2bd 100644 --- a/gst/audioresample/functable.h +++ b/gst/legacyresample/functable.h diff --git a/gst/audioresample/gstaudioresample.c b/gst/legacyresample/gstlegacyresample.c index 4f6f85e0..908b6ad9 100644 --- a/gst/audioresample/gstaudioresample.c +++ b/gst/legacyresample/gstlegacyresample.c @@ -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; } diff --git a/gst/audioresample/gstaudioresample.h b/gst/legacyresample/gstlegacyresample.h index c969ccdb..59babc17 100644 --- a/gst/audioresample/gstaudioresample.h +++ b/gst/legacyresample/gstlegacyresample.h @@ -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__ */ diff --git a/gst/audioresample/resample.c b/gst/legacyresample/resample.c index c464adf8..c464adf8 100644 --- a/gst/audioresample/resample.c +++ b/gst/legacyresample/resample.c diff --git a/gst/audioresample/resample.h b/gst/legacyresample/resample.h index 84bf8f09..84bf8f09 100644 --- a/gst/audioresample/resample.h +++ b/gst/legacyresample/resample.h diff --git a/gst/audioresample/resample_chunk.c b/gst/legacyresample/resample_chunk.c index 1cf9f09f..1cf9f09f 100644 --- a/gst/audioresample/resample_chunk.c +++ b/gst/legacyresample/resample_chunk.c diff --git a/gst/audioresample/resample_functable.c b/gst/legacyresample/resample_functable.c index af124276..af124276 100644 --- a/gst/audioresample/resample_functable.c +++ b/gst/legacyresample/resample_functable.c diff --git a/gst/audioresample/resample_ref.c b/gst/legacyresample/resample_ref.c index bb8d2411..bb8d2411 100644 --- a/gst/audioresample/resample_ref.c +++ b/gst/legacyresample/resample_ref.c diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 9d20a05c..45a5bc57 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -89,7 +89,7 @@ check_PROGRAMS = \ $(check_x264enc) \ elements/aacparse \ elements/amrparse \ - elements/audioresample \ + elements/legacyresample \ elements/qtmux \ elements/selector \ elements/mxfdemux \ diff --git a/tests/check/elements/audioresample.c b/tests/check/elements/legacyresample.c index 598e4b12..4e804e3e 100644 --- a/tests/check/elements/audioresample.c +++ b/tests/check/elements/legacyresample.c @@ -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); |