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Rename files and types from speexresample to audioresample

Browse source 
Rename files and types from speexresample to audioresample
to finish the move and to prevent any confusion.
This commit is contained in:
Sebastian Dröge 2009-01-23 12:31:06 +01:00
parent e327bbcd7d
commit 5dfcb63252
73 changed files with 2774 additions and 6090 deletions
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4
configure.ac
View file

@ -305,7 +305,7 @@ AG_GST_CHECK_PLUGIN(audiotestsrc)
AG_GST_CHECK_PLUGIN(ffmpegcolorspace) AG_GST_CHECK_PLUGIN(ffmpegcolorspace)
AG_GST_CHECK_PLUGIN(gdp) AG_GST_CHECK_PLUGIN(gdp)
AG_GST_CHECK_PLUGIN(playback) AG_GST_CHECK_PLUGIN(playback)
AG_GST_CHECK_PLUGIN(speexresample) AG_GST_CHECK_PLUGIN(audioresample)
AG_GST_CHECK_PLUGIN(subparse) AG_GST_CHECK_PLUGIN(subparse)
AG_GST_CHECK_PLUGIN(tcp) AG_GST_CHECK_PLUGIN(tcp)
AG_GST_CHECK_PLUGIN(typefind) AG_GST_CHECK_PLUGIN(typefind)
@ -690,7 +690,7 @@ gst/audiotestsrc/Makefile
gst/ffmpegcolorspace/Makefile gst/ffmpegcolorspace/Makefile
gst/gdp/Makefile gst/gdp/Makefile
gst/playback/Makefile gst/playback/Makefile
gst/speexresample/Makefile gst/audioresample/Makefile
gst/subparse/Makefile gst/subparse/Makefile
gst/tcp/Makefile gst/tcp/Makefile
gst/typefind/Makefile gst/typefind/Makefile

2
docs/plugins/Makefile.am
View file

@ -106,7 +106,7 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/gdp/gstgdpdepay.h \ $(top_srcdir)/gst/gdp/gstgdpdepay.h \
$(top_srcdir)/gst/gdp/gstgdppay.h \ $(top_srcdir)/gst/gdp/gstgdppay.h \
$(top_srcdir)/gst/playback/gstplay-enum.h \ $(top_srcdir)/gst/playback/gstplay-enum.h \
$(top_srcdir)/gst/speexresample/gstspeexresample.h \ $(top_srcdir)/gst/audioresample/gstaudioresample.h \
$(top_srcdir)/gst/tcp/gstmultifdsink.h \ $(top_srcdir)/gst/tcp/gstmultifdsink.h \
$(top_srcdir)/gst/tcp/gsttcpclientsrc.h \ $(top_srcdir)/gst/tcp/gsttcpclientsrc.h \
$(top_srcdir)/gst/tcp/gsttcpclientsink.h \ $(top_srcdir)/gst/tcp/gsttcpclientsink.h \

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

@ -107,15 +107,15 @@ audio_convert_prepare_context
<SECTION> <SECTION>
<FILE>element-audioresample</FILE> <FILE>element-audioresample</FILE>
<TITLE>audioresample</TITLE> <TITLE>audioresample</TITLE>
GstSpeexResample GstAudioResample
<SUBSECTION Standard> <SUBSECTION Standard>
GST_SPEEX_RESAMPLE GST_AUDIO_RESAMPLE
GST_IS_SPEEX_RESAMPLE GST_IS_AUDIO_RESAMPLE
GST_TYPE_SPEEX_RESAMPLE GST_TYPE_AUDIO_RESAMPLE
gst_speex_resample_get_type gst_audio_resample_get_type
GST_SPEEX_RESAMPLE_CLASS GST_AUDIO_RESAMPLE_CLASS
GST_IS_SPEEX_RESAMPLE_CLASS GST_IS_AUDIO_RESAMPLE_CLASS
GstSpeexResampleClass GstAudioResampleClass
</SECTION> </SECTION>
<SECTION> <SECTION>

66
docs/plugins/gst-plugins-base-plugins.args
View file

@ -641,7 +641,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::buffers-max</NAME> <NAME>GstMultiFdSink::buffers-max</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffers max</NICK> <NICK>Buffers max</NICK>
<BLURB>max number of buffers to queue for a client (-1 = no limit).</BLURB> <BLURB>max number of buffers to queue for a client (-1 = no limit).</BLURB>
@ -661,7 +661,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::buffers-soft-max</NAME> <NAME>GstMultiFdSink::buffers-soft-max</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffers soft max</NICK> <NICK>Buffers soft max</NICK>
<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB> <BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
@ -751,7 +751,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::buffers-min</NAME> <NAME>GstMultiFdSink::buffers-min</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffers min</NICK> <NICK>Buffers min</NICK>
<BLURB>min number of buffers to queue (-1 = as few as possible).</BLURB> <BLURB>min number of buffers to queue (-1 = as few as possible).</BLURB>
@ -781,7 +781,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::bytes-min</NAME> <NAME>GstMultiFdSink::bytes-min</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Bytes min</NICK> <NICK>Bytes min</NICK>
<BLURB>min number of bytes to queue (-1 = as little as possible).</BLURB> <BLURB>min number of bytes to queue (-1 = as little as possible).</BLURB>
@ -791,7 +791,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::time-min</NAME> <NAME>GstMultiFdSink::time-min</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Time min</NICK> <NICK>Time min</NICK>
<BLURB>min number of time to queue (-1 = as little as possible).</BLURB> <BLURB>min number of time to queue (-1 = as little as possible).</BLURB>
@ -811,7 +811,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::units-max</NAME> <NAME>GstMultiFdSink::units-max</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Units max</NICK> <NICK>Units max</NICK>
<BLURB>max number of units to queue (-1 = no limit).</BLURB> <BLURB>max number of units to queue (-1 = no limit).</BLURB>
@ -821,7 +821,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::units-soft-max</NAME> <NAME>GstMultiFdSink::units-soft-max</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Units soft max</NICK> <NICK>Units soft max</NICK>
<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB> <BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
@ -831,7 +831,7 @@
<ARG> <ARG>
<NAME>GstMultiFdSink::qos-dscp</NAME> <NAME>GstMultiFdSink::qos-dscp</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[-1,63]</RANGE> <RANGE>[G_MAXULONG,63]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>QoS diff srv code point</NICK> <NICK>QoS diff srv code point</NICK>
<BLURB>Quality of Service, differentiated services code point (-1 default).</BLURB> <BLURB>Quality of Service, differentiated services code point (-1 default).</BLURB>
@ -981,7 +981,7 @@
<ARG> <ARG>
<NAME>GstVorbisEnc::bitrate</NAME> <NAME>GstVorbisEnc::bitrate</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[-1,250001]</RANGE> <RANGE>[G_MAXULONG,250001]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Target Bitrate</NICK> <NICK>Target Bitrate</NICK>
<BLURB>Attempt to encode at a bitrate averaging this (in bps). This uses the bitrate management engine, and is not recommended for most users. Quality is a better alternative. (-1 == disabled).</BLURB> <BLURB>Attempt to encode at a bitrate averaging this (in bps). This uses the bitrate management engine, and is not recommended for most users. Quality is a better alternative. (-1 == disabled).</BLURB>
@ -1011,7 +1011,7 @@
<ARG> <ARG>
<NAME>GstVorbisEnc::max-bitrate</NAME> <NAME>GstVorbisEnc::max-bitrate</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[-1,250001]</RANGE> <RANGE>[G_MAXULONG,250001]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Maximum Bitrate</NICK> <NICK>Maximum Bitrate</NICK>
<BLURB>Specify a maximum bitrate (in bps). Useful for streaming applications. (-1 == disabled).</BLURB> <BLURB>Specify a maximum bitrate (in bps). Useful for streaming applications. (-1 == disabled).</BLURB>
@ -1021,7 +1021,7 @@
<ARG> <ARG>
<NAME>GstVorbisEnc::min-bitrate</NAME> <NAME>GstVorbisEnc::min-bitrate</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[-1,250001]</RANGE> <RANGE>[G_MAXULONG,250001]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Minimum Bitrate</NICK> <NICK>Minimum Bitrate</NICK>
<BLURB>Specify a minimum bitrate (in bps). Useful for encoding for a fixed-size channel. (-1 == disabled).</BLURB> <BLURB>Specify a minimum bitrate (in bps). Useful for encoding for a fixed-size channel. (-1 == disabled).</BLURB>
@ -1861,7 +1861,7 @@
<ARG> <ARG>
<NAME>GstCdParanoiaSrc::read-speed</NAME> <NAME>GstCdParanoiaSrc::read-speed</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Read speed</NICK> <NICK>Read speed</NICK>
<BLURB>Read from device at specified speed.</BLURB> <BLURB>Read from device at specified speed.</BLURB>
@ -1871,7 +1871,7 @@
<ARG> <ARG>
<NAME>GstCdParanoiaSrc::search-overlap</NAME> <NAME>GstCdParanoiaSrc::search-overlap</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[-1,75]</RANGE> <RANGE>[G_MAXULONG,75]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Search overlap</NICK> <NICK>Search overlap</NICK>
<BLURB>Force minimum overlap search during verification to n sectors.</BLURB> <BLURB>Force minimum overlap search during verification to n sectors.</BLURB>
@ -2011,7 +2011,7 @@
<ARG> <ARG>
<NAME>GstURIDecodeBin::buffer-duration</NAME> <NAME>GstURIDecodeBin::buffer-duration</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffer duration (ns)</NICK> <NICK>Buffer duration (ns)</NICK>
<BLURB>Buffer duration when buffering network streams.</BLURB> <BLURB>Buffer duration when buffering network streams.</BLURB>
@ -2021,7 +2021,7 @@
<ARG> <ARG>
<NAME>GstURIDecodeBin::buffer-size</NAME> <NAME>GstURIDecodeBin::buffer-size</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffer size (bytes)</NICK> <NICK>Buffer size (bytes)</NICK>
<BLURB>Buffer size when buffering network streams.</BLURB> <BLURB>Buffer size when buffering network streams.</BLURB>
@ -2181,7 +2181,7 @@
<ARG> <ARG>
<NAME>GstPlayBin2::current-audio</NAME> <NAME>GstPlayBin2::current-audio</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Current audio</NICK> <NICK>Current audio</NICK>
<BLURB>Currently playing audio stream (-1 = auto).</BLURB> <BLURB>Currently playing audio stream (-1 = auto).</BLURB>
@ -2191,7 +2191,7 @@
<ARG> <ARG>
<NAME>GstPlayBin2::current-text</NAME> <NAME>GstPlayBin2::current-text</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Current Text</NICK> <NICK>Current Text</NICK>
<BLURB>Currently playing text stream (-1 = auto).</BLURB> <BLURB>Currently playing text stream (-1 = auto).</BLURB>
@ -2201,7 +2201,7 @@
<ARG> <ARG>
<NAME>GstPlayBin2::current-video</NAME> <NAME>GstPlayBin2::current-video</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Current Video</NICK> <NICK>Current Video</NICK>
<BLURB>Currently playing video stream (-1 = auto).</BLURB> <BLURB>Currently playing video stream (-1 = auto).</BLURB>
@ -2361,7 +2361,7 @@
<ARG> <ARG>
<NAME>GstPlayBin2::buffer-duration</NAME> <NAME>GstPlayBin2::buffer-duration</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffer duration (ns)</NICK> <NICK>Buffer duration (ns)</NICK>
<BLURB>Buffer duration when buffering network streams.</BLURB> <BLURB>Buffer duration when buffering network streams.</BLURB>
@ -2371,7 +2371,7 @@
<ARG> <ARG>
<NAME>GstPlayBin2::buffer-size</NAME> <NAME>GstPlayBin2::buffer-size</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffer size (bytes)</NICK> <NICK>Buffer size (bytes)</NICK>
<BLURB>Buffer size when buffering network streams.</BLURB> <BLURB>Buffer size when buffering network streams.</BLURB>
@ -2711,7 +2711,7 @@
<ARG> <ARG>
<NAME>GstAppSrc::max-latency</NAME> <NAME>GstAppSrc::max-latency</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Max Latency</NICK> <NICK>Max Latency</NICK>
<BLURB>The maximum latency (-1 = unlimited).</BLURB> <BLURB>The maximum latency (-1 = unlimited).</BLURB>
@ -2721,7 +2721,7 @@
<ARG> <ARG>
<NAME>GstAppSrc::min-latency</NAME> <NAME>GstAppSrc::min-latency</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Min Latency</NICK> <NICK>Min Latency</NICK>
<BLURB>The minimum latency (-1 = default).</BLURB> <BLURB>The minimum latency (-1 = default).</BLURB>
@ -2731,7 +2731,7 @@
<ARG> <ARG>
<NAME>GstAppSrc::size</NAME> <NAME>GstAppSrc::size</NAME>
<TYPE>gint64</TYPE> <TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE> <RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Size</NICK> <NICK>Size</NICK>
<BLURB>The size of the data stream in bytes (-1 if unknown).</BLURB> <BLURB>The size of the data stream in bytes (-1 if unknown).</BLURB>
@ -2748,3 +2748,23 @@
<DEFAULT>Stream</DEFAULT> <DEFAULT>Stream</DEFAULT>
</ARG> </ARG>
<ARG>
<NAME>GstAudioResample::filter-length</NAME>
<TYPE>gint</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Filter length</NICK>
<BLURB>DEPRECATED, DON'T USE THIS! Length of the resample filter.</BLURB>
<DEFAULT>64</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioResample::quality</NAME>
<TYPE>gint</TYPE>
<RANGE>[0,10]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Quality</NICK>
<BLURB>Resample quality with 0 being the lowest and 10 being the best.</BLURB>
<DEFAULT>4</DEFAULT>
</ARG>

142
docs/plugins/gst-plugins-base-plugins.hierarchy
View file

@ -12,28 +12,9 @@ GObject
GstPlayBaseBin GstPlayBaseBin
GstPlayBin GstPlayBin
GstPlayBin2 GstPlayBin2
GstDecodeBin
GstDecodeBin2 GstDecodeBin2
GstURIDecodeBin GstURIDecodeBin
GstDecodeBin
GstBaseSrc
GstPushSrc
GstCddaBaseSrc
GstCdParanoiaSrc
GstBaseAudioSrc
GstAudioSrc
GstAlsaSrc
GstV4lElement
GstV4lSrc
GstTCPClientSrc
GstTCPServerSrc
GstVideoTestSrc
GstGnomeVFSSrc
GstAppSrc
GstAudioTestSrc
GstVorbisEnc
GstVorbisDec
GstVorbisParse
GstVorbisTag
GstOggDemux GstOggDemux
GstOggMux GstOggMux
GstOgmParse GstOgmParse
@ -43,49 +24,74 @@ GObject
GstOggParse GstOggParse
GstOggAviParse GstOggAviParse
GstBaseSink GstBaseSink
GstGnomeVFSSink GstGioBaseSink
GstGioSink
GstGioStreamSink
GstBaseAudioSink GstBaseAudioSink
GstAudioSink GstAudioSink
GstAlsaSink GstAlsaSink
GstGnomeVFSSink
GstVideoSink GstVideoSink
GstXvImageSink GstXvImageSink
GstXImageSink GstXImageSink
GstAppSink
GstTCPClientSink GstTCPClientSink
GstMultiFdSink GstMultiFdSink
GstTCPServerSink GstTCPServerSink
GstAppSink GstBaseSrc
GstVisual GstGioBaseSrc
GstVisualbumpscope GstGioSrc
GstVisualcorona GstGioStreamSrc
GstVisualinfinite GstPushSrc
GstVisualjakdaw GstBaseAudioSrc
GstVisualjess GstAudioSrc
GstVisuallv_analyzer GstAlsaSrc
GstVisuallv_scope GstCddaBaseSrc
GstVisualoinksie GstCdParanoiaSrc
GstTheoraDec GstV4lElement
GstTheoraEnc GstV4lSrc
GstTheoraParse GstTCPClientSrc
GstTCPServerSrc
GstVideoTestSrc
GstGnomeVFSSrc
GstAudioTestSrc
GstAppSrc
GstVorbisEnc
GstVorbisDec
GstVorbisParse
GstVorbisTag
GstTextOverlay GstTextOverlay
GstTimeOverlay GstTimeOverlay
GstClockOverlay GstClockOverlay
GstTextRender GstTextRender
GstTheoraDec
GstTheoraEnc
GstTheoraParse
GstAlsaMixerElement GstAlsaMixerElement
GstGDPDepay GstVisual
GstGDPPay GstVisualjess
GstBaseTransform GstVisualbumpscope
GstFFMpegCsp GstVisualcorona
GstVideoScale GstVisualinfinite
GstAudioFilter GstVisualjakdaw
GstVolume GstVisuallv_analyzer
GstSpeexResample GstVisuallv_scope
GstAudioConvert GstVisualoinksie
GstSubParse GstSubParse
GstSsaParse GstSsaParse
GstAudioRate
GstBaseTransform
GstAudioConvert
GstFFMpegCsp
GstAudioFilter
GstVolume
GstAudioResample
GstVideoScale
GstAdder GstAdder
GstGDPDepay
GstGDPPay
GstStreamSelector GstStreamSelector
GstQueue2 GstQueue2
GstAudioRate
GstVideoRate GstVideoRate
GstBus GstBus
GstTask GstTask
@ -96,26 +102,62 @@ GObject
GstRegistry GstRegistry
GstRingBuffer GstRingBuffer
GstSignalObject GstSignalObject
GConfClient GFileMonitor
GLocalDirectoryMonitor
GFamDirectoryMonitor
GInotifyDirectoryMonitor
GLocalFileMonitor
GFamFileMonitor
GInotifyFileMonitor
GVolumeMonitor
GNativeVolumeMonitor
GProxyVolumeMonitor
GProxyVolumeMonitorHal
GProxyVolumeMonitorGPhoto2
GUnixVolumeMonitor
GDaemonVolumeMonitor
GVfs
GDaemonVfs
GLocalVfs
GTypeModule
GIOModule
GVfsUriMapper
GVfsUriMapperSmb
GVfsUriMapperHttp
GAppLookupGConf
GProxyDrive
GProxyMount
GProxyVolume
GOutputStream
GInputStream
PangoFontMap PangoFontMap
PangoFcFontMap PangoFcFontMap
PangoFT2FontMap PangoFT2FontMap
PangoContext PangoContext
GstMixerTrack LinkConnection
GstMixerOptions GIOPConnection
LinkServer
GIOPServer
GConfClient
GstColorBalanceChannel
GstTunerNorm GstTunerNorm
GstTunerChannel GstTunerChannel
GstColorBalanceChannel GstMixerTrack
GstStreamInfo GstStreamInfo
GInterface GInterface
GTypePlugin GTypePlugin
GstChildProxy GstChildProxy
GstURIHandler GstURIHandler
GFile
GDesktopAppInfoLookup
GDrive
GMount
GVolume
GstTagSetter GstTagSetter
GstImplementsInterface GstImplementsInterface
GstMixer GstMixer
GstPropertyProbe GstPropertyProbe
GstTuner GstNavigation
GstXOverlay GstXOverlay
GstColorBalance GstColorBalance
GstNavigation GstTuner

24
docs/plugins/gst-plugins-base-plugins.interfaces
View file

@ -3,21 +3,29 @@ GstPipeline GstChildProxy
GstPlayBaseBin GstChildProxy GstPlayBaseBin GstChildProxy
GstPlayBin GstChildProxy GstPlayBin GstChildProxy
GstPlayBin2 GstChildProxy GstPlayBin2 GstChildProxy
GstDecodeBin GstChildProxy
GstDecodeBin2 GstChildProxy GstDecodeBin2 GstChildProxy
GstURIDecodeBin GstChildProxy GstURIDecodeBin GstChildProxy
GstDecodeBin GstChildProxy GstGioSink GstURIHandler
GstAlsaSink GstPropertyProbe
GstGnomeVFSSink GstURIHandler
GstXvImageSink GstImplementsInterface GstPropertyProbe GstNavigation GstXOverlay GstColorBalance
GstXImageSink GstImplementsInterface GstNavigation GstXOverlay
GstGioSrc GstURIHandler
GstAlsaSrc GstImplementsInterface GstMixer GstPropertyProbe
GstCddaBaseSrc GstURIHandler GstCddaBaseSrc GstURIHandler
GstCdParanoiaSrc GstURIHandler GstCdParanoiaSrc GstURIHandler
GstAlsaSrc GstImplementsInterface GstMixer GstPropertyProbe GstV4lElement GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstTuner
GstV4lElement GstImplementsInterface GstPropertyProbe GstTuner GstXOverlay GstColorBalance GstV4lSrc GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstTuner
GstV4lSrc GstImplementsInterface GstPropertyProbe GstTuner GstXOverlay GstColorBalance
GstGnomeVFSSrc GstURIHandler GstGnomeVFSSrc GstURIHandler
GstAppSrc GstURIHandler GstAppSrc GstURIHandler
GstVorbisEnc GstTagSetter GstVorbisEnc GstTagSetter
GstVorbisTag GstTagSetter GstVorbisTag GstTagSetter
GstGnomeVFSSink GstURIHandler
GstAlsaSink GstPropertyProbe
GstXvImageSink GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstNavigation
GstXImageSink GstImplementsInterface GstXOverlay GstNavigation
GstAlsaMixerElement GstImplementsInterface GstMixer GstPropertyProbe GstAlsaMixerElement GstImplementsInterface GstMixer GstPropertyProbe
GstVolume GstImplementsInterface GstMixer GstVolume GstImplementsInterface GstMixer
GTypeModule GTypePlugin
GIOModule GTypePlugin
GAppLookupGConf GDesktopAppInfoLookup
GProxyDrive GDrive
GProxyMount GMount
GProxyVolume GVolume

7
docs/plugins/gst-plugins-base-plugins.prerequisites
View file

@ -1,7 +1,12 @@
GstChildProxy GstObject GstChildProxy GstObject
GFile GObject
GDesktopAppInfoLookup GObject
GDrive GObject
GMount GObject
GVolume GObject
GstTagSetter GstObject GstElement GstTagSetter GstObject GstElement
GstImplementsInterface GstObject GstElement GstImplementsInterface GstObject GstElement
GstMixer GstObject GstImplementsInterface GstElement GstMixer GstObject GstImplementsInterface GstElement
GstTuner GstObject GstImplementsInterface GstElement
GstXOverlay GstObject GstImplementsInterface GstElement GstXOverlay GstObject GstImplementsInterface GstElement
GstColorBalance GstObject GstImplementsInterface GstElement GstColorBalance GstObject GstImplementsInterface GstElement
GstTuner GstObject GstImplementsInterface GstElement

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

@ -3,10 +3,10 @@
<description>Adds multiple streams</description> <description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename> <filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename> <basename>libgstadder.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>ALSA plugin library</description> <description>ALSA plugin library</description>
<filename>../../ext/alsa/.libs/libgstalsa.so</filename> <filename>../../ext/alsa/.libs/libgstalsa.so</filename>
<basename>libgstalsa.so</basename> <basename>libgstalsa.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Elements used to communicate with applications</description> <description>Elements used to communicate with applications</description>
<filename>../../gst/app/.libs/libgstapp.so</filename> <filename>../../gst/app/.libs/libgstapp.so</filename>
<basename>libgstapp.so</basename> <basename>libgstapp.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Convert audio to different formats</description> <description>Convert audio to different formats</description>
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename> <filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
<basename>libgstaudioconvert.so</basename> <basename>libgstaudioconvert.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Adjusts audio frames</description> <description>Adjusts audio frames</description>
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename> <filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
<basename>libgstaudiorate.so</basename> <basename>libgstaudiorate.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -1,12 +1,12 @@
<plugin> <plugin>
<name>audioresample</name> <name>audioresample</name>
<description>Resamples audio</description> <description>Resamples audio</description>
<filename>../../gst/speexresample/.libs/libgstaudioresample.so</filename> <filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
<basename>libgstaudioresample.so</basename> <basename>libgstaudioresample.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Creates audio test signals of given frequency and volume</description> <description>Creates audio test signals of given frequency and volume</description>
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename> <filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
<basename>libgstaudiotestsrc.so</basename> <basename>libgstaudiotestsrc.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Read audio from CD in paranoid mode</description> <description>Read audio from CD in paranoid mode</description>
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename> <filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
<basename>libgstcdparanoia.so</basename> <basename>libgstcdparanoia.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>GPL</license> <license>GPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>decoder bin</description> <description>decoder bin</description>
<filename>../../gst/playback/.libs/libgstdecodebin.so</filename> <filename>../../gst/playback/.libs/libgstdecodebin.so</filename>
<basename>libgstdecodebin.so</basename> <basename>libgstdecodebin.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,7 +3,7 @@
<description>colorspace conversion copied from FFMpeg 0.4.9-pre1</description> <description>colorspace conversion copied from FFMpeg 0.4.9-pre1</description>
<filename>../../gst/ffmpegcolorspace/.libs/libgstffmpegcolorspace.so</filename> <filename>../../gst/ffmpegcolorspace/.libs/libgstffmpegcolorspace.so</filename>
<basename>libgstffmpegcolorspace.so</basename> <basename>libgstffmpegcolorspace.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>FFMpeg</package> <package>FFMpeg</package>

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

@ -3,10 +3,10 @@
<description>Payload/depayload GDP packets</description> <description>Payload/depayload GDP packets</description>
<filename>../../gst/gdp/.libs/libgstgdp.so</filename> <filename>../../gst/gdp/.libs/libgstgdp.so</filename>
<basename>libgstgdp.so</basename> <basename>libgstgdp.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,7 +3,7 @@
<description>GIO elements</description> <description>GIO elements</description>
<filename>../../ext/gio/.libs/libgstgio.so</filename> <filename>../../ext/gio/.libs/libgstgio.so</filename>
<basename>libgstgio.so</basename> <basename>libgstgio.so</basename>
<version>0.10.21.1</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins CVS/prerelease</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>

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

@ -3,10 +3,10 @@
<description>elements to read from and write to Gnome-VFS uri's</description> <description>elements to read from and write to Gnome-VFS uri's</description>
<filename>../../ext/gnomevfs/.libs/libgstgnomevfs.so</filename> <filename>../../ext/gnomevfs/.libs/libgstgnomevfs.so</filename>
<basename>libgstgnomevfs.so</basename> <basename>libgstgnomevfs.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>libvisual visualization plugins</description> <description>libvisual visualization plugins</description>
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename> <filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
<basename>libgstlibvisual.so</basename> <basename>libgstlibvisual.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description> <description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
<filename>../../ext/ogg/.libs/libgstogg.so</filename> <filename>../../ext/ogg/.libs/libgstogg.so</filename>
<basename>libgstogg.so</basename> <basename>libgstogg.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Pango-based text rendering and overlay</description> <description>Pango-based text rendering and overlay</description>
<filename>../../ext/pango/.libs/libgstpango.so</filename> <filename>../../ext/pango/.libs/libgstpango.so</filename>
<basename>libgstpango.so</basename> <basename>libgstpango.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>various playback elements</description> <description>various playback elements</description>
<filename>../../gst/playback/.libs/libgstplaybin.so</filename> <filename>../../gst/playback/.libs/libgstplaybin.so</filename>
<basename>libgstplaybin.so</basename> <basename>libgstplaybin.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

4
docs/plugins/inspect/plugin-queue2.xml
View file

@ -3,10 +3,10 @@
<description>Queue newer version</description> <description>Queue newer version</description>
<filename>../../gst/playback/.libs/libgstqueue2.so</filename> <filename>../../gst/playback/.libs/libgstqueue2.so</filename>
<basename>libgstqueue2.so</basename> <basename>libgstqueue2.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Subtitle parsing</description> <description>Subtitle parsing</description>
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename> <filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
<basename>libgstsubparse.so</basename> <basename>libgstsubparse.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>transfer data over the network via TCP</description> <description>transfer data over the network via TCP</description>
<filename>../../gst/tcp/.libs/libgsttcp.so</filename> <filename>../../gst/tcp/.libs/libgsttcp.so</filename>
<basename>libgsttcp.so</basename> <basename>libgsttcp.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Theora plugin library</description> <description>Theora plugin library</description>
<filename>../../ext/theora/.libs/libgsttheora.so</filename> <filename>../../ext/theora/.libs/libgsttheora.so</filename>
<basename>libgsttheora.so</basename> <basename>libgsttheora.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>default typefind functions</description> <description>default typefind functions</description>
<filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename> <filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
<basename>libgsttypefindfunctions.so</basename> <basename>libgsttypefindfunctions.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>

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

@ -3,10 +3,10 @@
<description>URI Decoder bin</description> <description>URI Decoder bin</description>
<filename>../../gst/playback/.libs/libgstdecodebin2.so</filename> <filename>../../gst/playback/.libs/libgstdecodebin2.so</filename>
<basename>libgstdecodebin2.so</basename> <basename>libgstdecodebin2.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

4
docs/plugins/inspect/plugin-video4linux.xml
View file

@ -3,10 +3,10 @@
<description>elements for Video 4 Linux</description> <description>elements for Video 4 Linux</description>
<filename>../../sys/v4l/.libs/libgstvideo4linux.so</filename> <filename>../../sys/v4l/.libs/libgstvideo4linux.so</filename>
<basename>libgstvideo4linux.so</basename> <basename>libgstvideo4linux.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Adjusts video frames</description> <description>Adjusts video frames</description>
<filename>../../gst/videorate/.libs/libgstvideorate.so</filename> <filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
<basename>libgstvideorate.so</basename> <basename>libgstvideorate.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Resizes video</description> <description>Resizes video</description>
<filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename> <filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
<basename>libgstvideoscale.so</basename> <basename>libgstvideoscale.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Creates a test video stream</description> <description>Creates a test video stream</description>
<filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename> <filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
<basename>libgstvideotestsrc.so</basename> <basename>libgstvideotestsrc.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>plugin for controlling audio volume</description> <description>plugin for controlling audio volume</description>
<filename>../../gst/volume/.libs/libgstvolume.so</filename> <filename>../../gst/volume/.libs/libgstvolume.so</filename>
<basename>libgstvolume.so</basename> <basename>libgstvolume.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>Vorbis plugin library</description> <description>Vorbis plugin library</description>
<filename>../../ext/vorbis/.libs/libgstvorbis.so</filename> <filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
<basename>libgstvorbis.so</basename> <basename>libgstvorbis.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>X11 video output element based on standard Xlib calls</description> <description>X11 video output element based on standard Xlib calls</description>
<filename>../../sys/ximage/.libs/libgstximagesink.so</filename> <filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
<basename>libgstximagesink.so</basename> <basename>libgstximagesink.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

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

@ -3,10 +3,10 @@
<description>XFree86 video output plugin using Xv extension</description> <description>XFree86 video output plugin using Xv extension</description>
<filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename> <filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
<basename>libgstxvimagesink.so</basename> <basename>libgstxvimagesink.so</basename>
<version>0.10.22</version> <version>0.10.22.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-base</source> <source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package> <package>GStreamer Base Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

46
gst/audioresample/Makefile.am
View file

@ -1,21 +1,37 @@
plugin_LTLIBRARIES = libgstaudioresample.la plugin_LTLIBRARIES = libgstaudioresample.la
resample_SOURCES = \ libgstaudioresample_la_SOURCES = \
functable.c \ gstaudioresample.c \
resample.c \ speex_resampler_int.c \
resample_functable.c \ speex_resampler_float.c \
resample_ref.c \ speex_resampler_double.c
resample_chunk.c \
resample.h \ libgstaudioresample_la_CFLAGS = \
buffer.c $(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) \
$(GST_CFLAGS) \
$(LIBOIL_CFLAGS)
libgstaudioresample_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) \
$(GST_BASE_LIBS) \
$(GST_LIBS) \
$(LIBOIL_LIBS) \
$(LIBM)
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudioresample_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = \ noinst_HEADERS = \
arch.h \
fixed_arm4.h \
fixed_arm5e.h \
fixed_bfin.h \
fixed_debug.h \
fixed_generic.h \
gstaudioresample.h \ gstaudioresample.h \
functable.h \ resample.c \
debug.h \ resample_sse.h \
buffer.h speex_resampler.h \
speex_resampler_wrapper.h
libgstaudioresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
libgstaudioresample_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS)
libgstaudioresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS)
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)

0
gst/speexresample/README → gst/audioresample/README
View file

0
gst/speexresample/arch.h → gst/audioresample/arch.h
View file

253
gst/audioresample/buffer.c
View file

@ -1,253 +0,0 @@
#ifndef HAVE_CONFIG_H
#include "config.h"
#endif
#include <glib.h>
#include <string.h>
#include "buffer.h"
#include "debug.h"
static void audioresample_buffer_free_mem (AudioresampleBuffer * buffer,
void *);
static void audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer,
void *priv);
AudioresampleBuffer *
audioresample_buffer_new (void)
{
AudioresampleBuffer *buffer;
buffer = g_new0 (AudioresampleBuffer, 1);
buffer->ref_count = 1;
return buffer;
}
AudioresampleBuffer *
audioresample_buffer_new_and_alloc (int size)
{
AudioresampleBuffer *buffer = audioresample_buffer_new ();
buffer->data = g_malloc (size);
buffer->length = size;
buffer->free = audioresample_buffer_free_mem;
return buffer;
}
AudioresampleBuffer *
audioresample_buffer_new_with_data (void *data, int size)
{
AudioresampleBuffer *buffer = audioresample_buffer_new ();
buffer->data = data;
buffer->length = size;
buffer->free = audioresample_buffer_free_mem;
return buffer;
}
AudioresampleBuffer *
audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
int length)
{
AudioresampleBuffer *subbuffer = audioresample_buffer_new ();
if (buffer->parent) {
audioresample_buffer_ref (buffer->parent);
subbuffer->parent = buffer->parent;
} else {
audioresample_buffer_ref (buffer);
subbuffer->parent = buffer;
}
subbuffer->data = buffer->data + offset;
subbuffer->length = length;
subbuffer->free = audioresample_buffer_free_subbuffer;
return subbuffer;
}
void
audioresample_buffer_ref (AudioresampleBuffer * buffer)
{
buffer->ref_count++;
}
void
audioresample_buffer_unref (AudioresampleBuffer * buffer)
{
buffer->ref_count--;
if (buffer->ref_count == 0) {
if (buffer->free)
buffer->free (buffer, buffer->priv);
g_free (buffer);
}
}
static void
audioresample_buffer_free_mem (AudioresampleBuffer * buffer, void *priv)
{
g_free (buffer->data);
}
static void
audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer, void *priv)
{
audioresample_buffer_unref (buffer->parent);
}
AudioresampleBufferQueue *
audioresample_buffer_queue_new (void)
{
return g_new0 (AudioresampleBufferQueue, 1);
}
int
audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue)
{
return queue->depth;
}
int
audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue)
{
return queue->offset;
}
void
audioresample_buffer_queue_free (AudioresampleBufferQueue * queue)
{
GList *g;
for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
}
g_list_free (queue->buffers);
g_free (queue);
}
void
audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer)
{
queue->buffers = g_list_append (queue->buffers, buffer);
queue->depth += buffer->length;
}
AudioresampleBuffer *
audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int length)
{
GList *g;
AudioresampleBuffer *newbuffer;
AudioresampleBuffer *buffer;
AudioresampleBuffer *subbuffer;
g_return_val_if_fail (length > 0, NULL);
if (queue->depth < length) {
return NULL;
}
RESAMPLE_LOG ("pulling %d, %d available", length, queue->depth);
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length) {
newbuffer = audioresample_buffer_new_subbuffer (buffer, 0, length);
subbuffer = audioresample_buffer_new_subbuffer (buffer, length,
buffer->length - length);
g->data = subbuffer;
audioresample_buffer_unref (buffer);
} else {
int offset = 0;
newbuffer = audioresample_buffer_new_and_alloc (length);
while (offset < length) {
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length - offset) {
int n = length - offset;
memcpy (newbuffer->data + offset, buffer->data, n);
subbuffer =
audioresample_buffer_new_subbuffer (buffer, n, buffer->length - n);
g->data = subbuffer;
audioresample_buffer_unref (buffer);
offset += n;
} else {
memcpy (newbuffer->data + offset, buffer->data, buffer->length);
queue->buffers = g_list_delete_link (queue->buffers, g);
offset += buffer->length;
audioresample_buffer_unref (buffer);
}
}
}
queue->depth -= length;
queue->offset += length;
return newbuffer;
}
AudioresampleBuffer *
audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int length)
{
GList *g;
AudioresampleBuffer *newbuffer;
AudioresampleBuffer *buffer;
int offset = 0;
g_return_val_if_fail (length > 0, NULL);
if (queue->depth < length) {
return NULL;
}
RESAMPLE_LOG ("peeking %d, %d available", length, queue->depth);
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length) {
newbuffer = audioresample_buffer_new_subbuffer (buffer, 0, length);
} else {
newbuffer = audioresample_buffer_new_and_alloc (length);
while (offset < length) {
buffer = g->data;
if (buffer->length > length - offset) {
int n = length - offset;
memcpy (newbuffer->data + offset, buffer->data, n);
offset += n;
} else {
memcpy (newbuffer->data + offset, buffer->data, buffer->length);
offset += buffer->length;
}
g = g_list_next (g);
}
}
return newbuffer;
}
void
audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue)
{
GList *g;
for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
}
g_list_free (queue->buffers);
queue->buffers = NULL;
queue->depth = 0;
queue->offset = 0;
}

51
gst/audioresample/buffer.h
View file

@ -1,51 +0,0 @@
#ifndef __AUDIORESAMPLE_BUFFER_H__
#define __AUDIORESAMPLE_BUFFER_H__
#include <glib.h>
typedef struct _AudioresampleBuffer AudioresampleBuffer;
typedef struct _AudioresampleBufferQueue AudioresampleBufferQueue;
struct _AudioresampleBuffer
{
unsigned char *data;
int length;
int ref_count;
AudioresampleBuffer *parent;
void (*free) (AudioresampleBuffer *, void *);
void *priv;
void *priv2;
};
struct _AudioresampleBufferQueue
{
GList *buffers;
int depth;
int offset;
};
AudioresampleBuffer * audioresample_buffer_new (void);
AudioresampleBuffer * audioresample_buffer_new_and_alloc (int size);
AudioresampleBuffer * audioresample_buffer_new_with_data (void *data, int size);
AudioresampleBuffer * audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer,
int offset,
int length);
void audioresample_buffer_ref (AudioresampleBuffer * buffer);
void audioresample_buffer_unref (AudioresampleBuffer * buffer);
AudioresampleBufferQueue *
audioresample_buffer_queue_new (void);
void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer);
AudioresampleBuffer * audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
AudioresampleBuffer * audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
void audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue);
#endif

65
gst/audioresample/debug.c
View file

@ -1,65 +0,0 @@
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <glib.h>
#include <stdio.h>
#include <debug.h>
static const char *resample_debug_level_names[] = {
"NONE",
"ERROR",
"WARNING",
"INFO",
"DEBUG",
"LOG"
};
static int resample_debug_level = RESAMPLE_LEVEL_ERROR;
void
resample_debug_log (int level, const char *file, const char *function,
int line, const char *format, ...)
{
#ifndef GLIB_COMPAT
va_list varargs;
char *s;
if (level > resample_debug_level)
return;
va_start (varargs, format);
s = g_strdup_vprintf (format, varargs);
va_end (varargs);
fprintf (stderr, "RESAMPLE: %s: %s(%d): %s: %s\n",
resample_debug_level_names[level], file, line, function, s);
g_free (s);
#else
va_list varargs;
char s[1000];
if (level > resample_debug_level)
return;
va_start (varargs, format);
vsnprintf (s, 999, format, varargs);
va_end (varargs);
fprintf (stderr, "RESAMPLE: %s: %s(%d): %s: %s\n",
resample_debug_level_names[level], file, line, function, s);
#endif
}
void
resample_debug_set_level (int level)
{
resample_debug_level = level;
}
int
resample_debug_get_level (void)
{
return resample_debug_level;
}

51
gst/audioresample/debug.h
View file

@ -1,51 +0,0 @@
#ifndef __RESAMPLE_DEBUG_H__
#define __RESAMPLE_DEBUG_H__
#if 0
enum
{
RESAMPLE_LEVEL_NONE = 0,
RESAMPLE_LEVEL_ERROR,
RESAMPLE_LEVEL_WARNING,
RESAMPLE_LEVEL_INFO,
RESAMPLE_LEVEL_DEBUG,
RESAMPLE_LEVEL_LOG
};
#define RESAMPLE_ERROR(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_ERROR, __VA_ARGS__)
#define RESAMPLE_WARNING(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_WARNING, __VA_ARGS__)
#define RESAMPLE_INFO(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_INFO, __VA_ARGS__)
#define RESAMPLE_DEBUG(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_DEBUG, __VA_ARGS__)
#define RESAMPLE_LOG(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_LOG, __VA_ARGS__)
#define RESAMPLE_DEBUG_LEVEL(level,...) \
resample_debug_log ((level), __FILE__, __FUNCTION__, __LINE__, __VA_ARGS__)
void resample_debug_log (int level, const char *file, const char *function,
int line, const char *format, ...);
void resample_debug_set_level (int level);
int resample_debug_get_level (void);
#else
#include <gst/gst.h>
GST_DEBUG_CATEGORY_EXTERN (libaudioresample_debug);
#define GST_CAT_DEFAULT libaudioresample_debug
#define RESAMPLE_ERROR GST_ERROR
#define RESAMPLE_WARNING GST_WARNING
#define RESAMPLE_INFO GST_INFO
#define RESAMPLE_DEBUG GST_DEBUG
#define RESAMPLE_LOG GST_LOG
#define resample_debug_set_level(x) do { } while (0)
#endif
#endif

0
gst/speexresample/fixed_arm4.h → gst/audioresample/fixed_arm4.h
View file

0
gst/speexresample/fixed_arm5e.h → gst/audioresample/fixed_arm5e.h
View file

0
gst/speexresample/fixed_bfin.h → gst/audioresample/fixed_bfin.h
View file

0
gst/speexresample/fixed_debug.h → gst/audioresample/fixed_debug.h
View file

0
gst/speexresample/fixed_generic.h → gst/audioresample/fixed_generic.h
View file

254
gst/audioresample/functable.c
View file

@ -1,254 +0,0 @@
/* Resampling library
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include "functable.h"
#include "debug.h"
void
functable_func_sinc (double *fx, double *dfx, double x, void *closure)
{
if (x == 0) {
*fx = 1;
*dfx = 0;
return;
}
*fx = sin (x) / x;
*dfx = (cos (x) - sin (x) / x) / x;
}
void
functable_func_boxcar (double *fx, double *dfx, double x, void *closure)
{
double width = *(double *) closure;
if (x < width && x > -width) {
*fx = 1;
} else {
*fx = 0;
}
*dfx = 0;
}
void
functable_func_hanning (double *fx, double *dfx, double x, void *closure)
{
double width = *(double *) closure;
if (x < width && x > -width) {
x /= width;
*fx = (1 - x * x) * (1 - x * x);
*dfx = -2 * 2 * x / width * (1 - x * x);
} else {
*fx = 0;
*dfx = 0;
}
}
Functable *
functable_new (void)
{
Functable *ft;
ft = malloc (sizeof (Functable));
memset (ft, 0, sizeof (Functable));
return ft;
}
void
functable_free (Functable * ft)
{
free (ft);
}
void
functable_set_length (Functable * t, int length)
{
t->length = length;
}
void
functable_set_offset (Functable * t, double offset)
{
t->offset = offset;
}
void
functable_set_multiplier (Functable * t, double multiplier)
{
t->multiplier = multiplier;
}
void
functable_calculate (Functable * t, FunctableFunc func, void *closure)
{
int i;
double x;
if (t->fx)
free (t->fx);
if (t->dfx)
free (t->dfx);
t->fx = malloc (sizeof (double) * (t->length + 1));
t->dfx = malloc (sizeof (double) * (t->length + 1));
t->inv_multiplier = 1.0 / t->multiplier;
for (i = 0; i < t->length + 1; i++) {
x = t->offset + t->multiplier * i;
func (&t->fx[i], &t->dfx[i], x, closure);
}
}
void
functable_calculate_multiply (Functable * t, FunctableFunc func, void *closure)
{
int i;
double x;
for (i = 0; i < t->length + 1; i++) {
double afx, adfx, bfx, bdfx;
afx = t->fx[i];
adfx = t->dfx[i];
x = t->offset + t->multiplier * i;
func (&bfx, &bdfx, x, closure);
t->fx[i] = afx * bfx;
t->dfx[i] = afx * bdfx + adfx * bfx;
}
}
double
functable_evaluate (Functable * t, double x)
{
int i;
double f0, f1, w0, w1;
double x2, x3;
double w;
if (x < t->offset || x > (t->offset + t->length * t->multiplier)) {
RESAMPLE_DEBUG ("x out of range %g", x);
}
x -= t->offset;
x *= t->inv_multiplier;
i = floor (x);
x -= i;
x2 = x * x;
x3 = x2 * x;
f1 = 3 * x2 - 2 * x3;
f0 = 1 - f1;
w0 = (x - 2 * x2 + x3) * t->multiplier;
w1 = (-x2 + x3) * t->multiplier;
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
/*w = t->fx[i] * (1-x) + t->fx[i+1] * x; */
return w;
}
double
functable_fir (Functable * t, double x, int n, double *data, int len)
{
int i, j;
double f0, f1, w0, w1;
double x2, x3;
double w;
double sum;
x -= t->offset;
x /= t->multiplier;
i = floor (x);
x -= i;
x2 = x * x;
x3 = x2 * x;
f1 = 3 * x2 - 2 * x3;
f0 = 1 - f1;
w0 = (x - 2 * x2 + x3) * t->multiplier;
w1 = (-x2 + x3) * t->multiplier;
sum = 0;
for (j = 0; j < len; j++) {
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
sum += data[j * 2] * w;
i += n;
}
return sum;
}
void
functable_fir2 (Functable * t, double *r0, double *r1, double x,
int n, double *data, int len)
{
int i, j;
double f0, f1, w0, w1;
double x2, x3;
double w;
double sum0, sum1;
double floor_x;
x -= t->offset;
x *= t->inv_multiplier;
floor_x = floor (x);
i = floor_x;
x -= floor_x;
x2 = x * x;
x3 = x2 * x;
f1 = 3 * x2 - 2 * x3;
f0 = 1 - f1;
w0 = (x - 2 * x2 + x3) * t->multiplier;
w1 = (-x2 + x3) * t->multiplier;
sum0 = 0;
sum1 = 0;
for (j = 0; j < len; j++) {
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
sum0 += data[j * 2] * w;
sum1 += data[j * 2 + 1] * w;
i += n;
}
*r0 = sum0;
*r1 = sum1;
}

61
gst/audioresample/functable.h
View file

@ -1,61 +0,0 @@
/* Resampling library
* Copyright (C) <2001> David Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __FUNCTABLE_H__
#define __FUNCTABLE_H__
typedef void FunctableFunc (double *fx, double *dfx, double x, void *closure);
typedef struct _Functable Functable;
struct _Functable {
int length;
double offset;
double multiplier;
double inv_multiplier;
double *fx;
double *dfx;
};
Functable *functable_new (void);
void functable_setup (Functable *t);
void functable_free (Functable *t);
void functable_set_length (Functable *t, int length);
void functable_set_offset (Functable *t, double offset);
void functable_set_multiplier (Functable *t, double multiplier);
void functable_calculate (Functable *t, FunctableFunc func, void *closure);
void functable_calculate_multiply (Functable *t, FunctableFunc func, void *closure);
double functable_evaluate (Functable *t,double x);
double functable_fir(Functable *t,double x0,int n,double *data,int len);
void functable_fir2(Functable *t,double *r0, double *r1, double x0,
int n,double *data,int len);
void functable_func_sinc(double *fx, double *dfx, double x, void *closure);
void functable_func_boxcar(double *fx, double *dfx, double x, void *closure);
void functable_func_hanning(double *fx, double *dfx, double x, void *closure);
#endif /* __PRIVATE_H__ */

1419
gst/audioresample/gstaudioresample.c
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69
gst/audioresample/gstaudioresample.h
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@ -1,5 +1,6 @@
/* GStreamer /* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2007-2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
* *
* This library is free software; you can redistribute it and/or * This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public * modify it under the terms of the GNU Library General Public
@ -18,62 +19,72 @@
*/ */
#ifndef __AUDIORESAMPLE_H__ #ifndef __AUDIO_RESAMPLE_H__
#define __AUDIORESAMPLE_H__ #define __AUDIO_RESAMPLE_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/base/gstbasetransform.h> #include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include "resample.h" #include "speex_resampler_wrapper.h"
G_BEGIN_DECLS G_BEGIN_DECLS
#define GST_TYPE_AUDIORESAMPLE \ #define GST_TYPE_AUDIO_RESAMPLE \
(gst_audioresample_get_type()) (gst_audio_resample_get_type())
#define GST_AUDIORESAMPLE(obj) \ #define GST_AUDIO_RESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,GstAudioresample)) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RESAMPLE,GstAudioResample))
#define GST_AUDIORESAMPLE_CLASS(klass) \ #define GST_AUDIO_RESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,GstAudioresampleClass)) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RESAMPLE,GstAudioResampleClass))
#define GST_IS_AUDIORESAMPLE(obj) \ #define GST_IS_AUDIO_RESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE)) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RESAMPLE))
#define GST_IS_AUDIORESAMPLE_CLASS(klass) \ #define GST_IS_AUDIO_RESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE)) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RESAMPLE))
typedef struct _GstAudioresample GstAudioresample; typedef struct _GstAudioResample GstAudioResample;
typedef struct _GstAudioresampleClass GstAudioresampleClass; typedef struct _GstAudioResampleClass GstAudioResampleClass;
/** /**
* GstAudioresample: * GstAudioResample:
* *
* Opaque data structure. * Opaque data structure.
*/ */
struct _GstAudioresample { struct _GstAudioResample {
GstBaseTransform element; GstBaseTransform element;
/* <private> */
GstCaps *srccaps, *sinkcaps; GstCaps *srccaps, *sinkcaps;
gboolean passthru;
gboolean need_discont; gboolean need_discont;
guint64 offset; guint64 next_offset;
guint64 ts_offset;
GstClockTime next_ts; GstClockTime next_ts;
GstClockTime prev_ts, prev_duration; GstClockTime next_upstream_ts;
int channels;
int i_rate; gint channels;
int o_rate; gint inrate;
int filter_length; gint outrate;
gint quality;
gint width;
gboolean fp;
ResampleState * resample; guint8 *tmp_in;
guint tmp_in_size;
guint8 *tmp_out;
guint tmp_out_size;
SpeexResamplerState *state;
const SpeexResampleFuncs *funcs;
}; };
struct _GstAudioresampleClass { struct _GstAudioResampleClass {
GstBaseTransformClass parent_class; GstBaseTransformClass parent_class;
}; };
GType gst_audioresample_get_type(void); GType gst_audio_resample_get_type(void);
G_END_DECLS G_END_DECLS
#endif /* __AUDIORESAMPLE_H__ */ #endif /* __AUDIO_RESAMPLE_H__ */

1607
gst/audioresample/resample.c
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128
gst/audioresample/resample.h
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@ -1,128 +0,0 @@
/* Resampling library
* Copyright (C) <2001> David Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __RESAMPLE_H__
#define __RESAMPLE_H__
#include "functable.h"
#include "buffer.h"
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
#ifdef WIN32
#define rint(x) (floor((x)+0.5))
#endif
typedef enum {
RESAMPLE_FORMAT_S16 = 0,
RESAMPLE_FORMAT_S32,
RESAMPLE_FORMAT_F32,
RESAMPLE_FORMAT_F64
} ResampleFormat;
typedef void (*ResampleCallback) (void *);
typedef struct _ResampleState ResampleState;
struct _ResampleState {
/* parameters */
int n_channels;
ResampleFormat format;
int filter_length;
double i_rate;
double o_rate;
int method;
/* internal parameters */
int need_reinit;
double halftaps;
/* filter state */
unsigned char *o_buf;
int o_size;
AudioresampleBufferQueue *queue;
int eos;
int started;
int sample_size;
unsigned char *buffer;
int buffer_len;
int buffer_filled;
double i_start;
double o_start;
double i_inc;
double o_inc;
double sinc_scale;
double i_end;
double o_end;
int i_samples;
int o_samples;
//void *i_buf;
Functable *ft;
double *out_tmp;
};
void resample_init (void);
void resample_cleanup (void);
ResampleState *resample_new (void);
void resample_free (ResampleState *state);
void resample_add_input_data (ResampleState * r, void *data, int size,
ResampleCallback free_func, void *closure);
void resample_input_eos (ResampleState *r);
void resample_input_flush (ResampleState *r);
void resample_input_pushthrough (ResampleState *r);
int resample_get_output_size_for_input (ResampleState * r, int size);
int resample_get_input_size_for_output (ResampleState * r, int size);
int resample_get_output_size (ResampleState *r);
int resample_get_output_data (ResampleState *r, void *data, int size);
void resample_set_filter_length (ResampleState *r, int length);
void resample_set_input_rate (ResampleState *r, double rate);
void resample_set_output_rate (ResampleState *r, double rate);
void resample_set_n_channels (ResampleState *r, int n_channels);
void resample_set_format (ResampleState *r, ResampleFormat format);
void resample_set_method (ResampleState *r, int method);
int resample_format_size (ResampleFormat format);
#endif /* __RESAMPLE_H__ */

209
gst/audioresample/resample_chunk.c
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@ -1,209 +0,0 @@
/* Resampling library
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <limits.h>
#include <liboil/liboil.h>
#include "resample.h"
#include "buffer.h"
#include "debug.h"
static double
resample_sinc_window (double x, double halfwidth, double scale)
{
double y;
if (x == 0)
return 1.0;
if (x < -halfwidth || x > halfwidth)
return 0.0;
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
x /= halfwidth;
y *= (1 - x * x) * (1 - x * x);
return y;
}
void
resample_scale_chunk (ResampleState * r)
{
if (r->need_reinit) {
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
if (r->buffer)
free (r->buffer);
r->buffer_len = r->sample_size * 1000;
r->buffer = malloc (r->buffer_len);
memset (r->buffer, 0, r->buffer_len);
r->i_inc = r->o_rate / r->i_rate;
r->o_inc = r->i_rate / r->o_rate;
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
r->i_start = -r->i_inc * r->filter_length;
r->need_reinit = 0;
#if 0
if (r->i_inc < 1.0) {
r->sinc_scale = r->i_inc;
if (r->sinc_scale == 0.5) {
/* strange things happen at integer multiples */
r->sinc_scale = 1.0;
}
} else {
r->sinc_scale = 1.0;
}
#else
r->sinc_scale = 1.0;
#endif
}
while (r->o_size > 0) {
double midpoint;
int i;
int j;
RESAMPLE_DEBUG ("i_start %g", r->i_start);
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
if (midpoint > 0.5 * r->i_inc) {
RESAMPLE_ERROR ("inconsistent state");
}
while (midpoint < -0.5 * r->i_inc) {
AudioresampleBuffer *buffer;
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
if (buffer == NULL) {
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
return;
}
r->i_start += r->i_inc;
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
midpoint += r->i_inc;
memmove (r->buffer, r->buffer + r->sample_size,
r->buffer_len - r->sample_size);
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
r->sample_size);
audioresample_buffer_unref (buffer);
}
switch (r->format) {
case RESAMPLE_FORMAT_S16:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
if (acc < -32768.0)
acc = -32768.0;
if (acc > 32767.0)
acc = 32767.0;
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_S32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
if (acc < -2147483648.0)
acc = -2147483648.0;
if (acc > 2147483647.0)
acc = 2147483647.0;
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_F32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(float *) (r->buffer + i * sizeof (float) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
*(float *) (r->o_buf + i * sizeof (float)) = acc;
}
break;
case RESAMPLE_FORMAT_F64:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(double *) (r->buffer + i * sizeof (double) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
*(double *) (r->o_buf + i * sizeof (double)) = acc;
}
break;
}
r->i_start -= 1.0;
r->o_buf += r->sample_size;
r->o_size -= r->sample_size;
}
}

271
gst/audioresample/resample_functable.c
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@ -1,271 +0,0 @@
/* Resampling library
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <limits.h>
#include <liboil/liboil.h>
#include "resample.h"
#include "buffer.h"
#include "debug.h"
static void
func_sinc (double *fx, double *dfx, double x, void *closure)
{
//double scale = *(double *)closure;
double scale = M_PI;
if (x == 0) {
*fx = 1;
*dfx = 0;
return;
}
x *= scale;
*fx = sin (x) / x;
*dfx = scale * (cos (x) - sin (x) / x) / x;
}
static void
func_hanning (double *fx, double *dfx, double x, void *closure)
{
double width = *(double *) closure;
if (x < width && x > -width) {
x /= width;
*fx = (1 - x * x) * (1 - x * x);
*dfx = -2 * 2 * x / width * (1 - x * x);
} else {
*fx = 0;
*dfx = 0;
}
}
#if 0
static double
resample_sinc_window (double x, double halfwidth, double scale)
{
double y;
if (x == 0)
return 1.0;
if (x < -halfwidth || x > halfwidth)
return 0.0;
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
x /= halfwidth;
y *= (1 - x * x) * (1 - x * x);
return y;
}
#endif
#if 0
static void
functable_test (Functable * ft, double halfwidth)
{
int i;
double x;
for (i = 0; i < 100; i++) {
x = i * 0.1;
printf ("%d %g %g\n", i, resample_sinc_window (x, halfwidth, 1.0),
functable_evaluate (ft, x));
}
exit (0);
}
#endif
void
resample_scale_functable (ResampleState * r)
{
if (r->need_reinit) {
double hanning_width;
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
if (r->buffer)
free (r->buffer);
r->buffer_len = r->sample_size * r->filter_length;
r->buffer = malloc (r->buffer_len);
memset (r->buffer, 0, r->buffer_len);
r->i_inc = r->o_rate / r->i_rate;
r->o_inc = r->i_rate / r->o_rate;
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
r->i_start = -r->i_inc * r->filter_length;
if (r->ft) {
functable_free (r->ft);
}
r->ft = functable_new ();
functable_set_length (r->ft, r->filter_length * 16);
functable_set_offset (r->ft, -r->filter_length / 2);
functable_set_multiplier (r->ft, 1 / 16.0);
hanning_width = r->filter_length / 2;
functable_calculate (r->ft, func_sinc, NULL);
functable_calculate_multiply (r->ft, func_hanning, &hanning_width);
//functable_test(r->ft, 0.5 * r->filter_length);
#if 0
if (r->i_inc < 1.0) {
r->sinc_scale = r->i_inc;
if (r->sinc_scale == 0.5) {
/* strange things happen at integer multiples */
r->sinc_scale = 1.0;
}
} else {
r->sinc_scale = 1.0;
}
#else
r->sinc_scale = 1.0;
#endif
r->need_reinit = 0;
}
while (r->o_size > 0) {
double midpoint;
int i;
int j;
RESAMPLE_DEBUG ("i_start %g", r->i_start);
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
if (midpoint > 0.5 * r->i_inc) {
RESAMPLE_ERROR ("inconsistent state");
}
while (midpoint < -0.5 * r->i_inc) {
AudioresampleBuffer *buffer;
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
if (buffer == NULL) {
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
return;
}
r->i_start += r->i_inc;
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
midpoint += r->i_inc;
memmove (r->buffer, r->buffer + r->sample_size,
r->buffer_len - r->sample_size);
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
r->sample_size);
audioresample_buffer_unref (buffer);
}
switch (r->format) {
case RESAMPLE_FORMAT_S16:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
j * r->sample_size);
acc += functable_evaluate (r->ft, offset) * x;
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
}
if (acc < -32768.0)
acc = -32768.0;
if (acc > 32767.0)
acc = 32767.0;
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_S32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
j * r->sample_size);
acc += functable_evaluate (r->ft, offset) * x;
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
}
if (acc < -2147483648.0)
acc = -2147483648.0;
if (acc > 2147483647.0)
acc = 2147483647.0;
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_F32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(float *) (r->buffer + i * sizeof (float) +
j * r->sample_size);
acc += functable_evaluate (r->ft, offset) * x;
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
}
*(float *) (r->o_buf + i * sizeof (float)) = acc;
}
break;
case RESAMPLE_FORMAT_F64:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(double *) (r->buffer + i * sizeof (double) +
j * r->sample_size);
acc += functable_evaluate (r->ft, offset) * x;
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
}
*(double *) (r->o_buf + i * sizeof (double)) = acc;
}
break;
}
r->i_start -= 1.0;
r->o_buf += r->sample_size;
r->o_size -= r->sample_size;
}
}

223
gst/audioresample/resample_ref.c
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@ -1,223 +0,0 @@
/* Resampling library
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <limits.h>
#include <liboil/liboil.h>
#include "resample.h"
#include "buffer.h"
#include "debug.h"
static double
resample_sinc_window (double x, double halfwidth, double scale)
{
double y;
if (x == 0)
return 1.0;
if (x < -halfwidth || x > halfwidth)
return 0.0;
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
x /= halfwidth;
y *= (1 - x * x) * (1 - x * x);
return y;
}
void
resample_scale_ref (ResampleState * r)
{
if (r->need_reinit) {
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
if (r->buffer)
free (r->buffer);
r->buffer_len = r->sample_size * r->filter_length;
r->buffer = malloc (r->buffer_len);
memset (r->buffer, 0, r->buffer_len);
r->buffer_filled = 0;
r->i_inc = r->o_rate / r->i_rate;
r->o_inc = r->i_rate / r->o_rate;
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
r->i_start = -r->i_inc * r->filter_length;
r->need_reinit = 0;
#if 0
if (r->i_inc < 1.0) {
r->sinc_scale = r->i_inc;
if (r->sinc_scale == 0.5) {
/* strange things happen at integer multiples */
r->sinc_scale = 1.0;
}
} else {
r->sinc_scale = 1.0;
}
#else
r->sinc_scale = 1.0;
#endif
}
RESAMPLE_DEBUG ("asked to resample %d bytes", r->o_size);
RESAMPLE_DEBUG ("%d bytes in queue",
audioresample_buffer_queue_get_depth (r->queue));
while (r->o_size >= r->sample_size) {
double midpoint;
int i;
int j;
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
RESAMPLE_DEBUG
("still need to output %d bytes, %d input left, i_start %g, midpoint %f",
r->o_size, audioresample_buffer_queue_get_depth (r->queue), r->i_start,
midpoint);
if (midpoint > 0.5 * r->i_inc) {
RESAMPLE_ERROR ("inconsistent state");
}
while (midpoint < -0.5 * r->i_inc) {
AudioresampleBuffer *buffer;
RESAMPLE_DEBUG ("midpoint %f < %f, r->i_inc %f", midpoint,
-0.5 * r->i_inc, r->i_inc);
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
if (buffer == NULL) {
/* FIXME: for the first buffer, this isn't necessarily an error,
* since because of the filter length we'll output less buffers.
* deal with that so we don't print to console */
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
return;
}
r->i_start += r->i_inc;
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
midpoint += r->i_inc;
memmove (r->buffer, r->buffer + r->sample_size,
r->buffer_len - r->sample_size);
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
r->sample_size);
r->buffer_filled = MIN (r->buffer_filled + r->sample_size, r->buffer_len);
audioresample_buffer_unref (buffer);
}
switch (r->format) {
case RESAMPLE_FORMAT_S16:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
if (acc < -32768.0)
acc = -32768.0;
if (acc > 32767.0)
acc = 32767.0;
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_S32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
if (acc < -2147483648.0)
acc = -2147483648.0;
if (acc > 2147483647.0)
acc = 2147483647.0;
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_F32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(float *) (r->buffer + i * sizeof (float) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
*(float *) (r->o_buf + i * sizeof (float)) = acc;
}
break;
case RESAMPLE_FORMAT_F64:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(double *) (r->buffer + i * sizeof (double) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
*(double *) (r->o_buf + i * sizeof (double)) = acc;
}
break;
}
r->i_start -= 1.0;
r->o_buf += r->sample_size;
r->o_size -= r->sample_size;
}
}

0
gst/speexresample/resample_sse.h → gst/audioresample/resample_sse.h
View file

0
gst/speexresample/speex_resampler.h → gst/audioresample/speex_resampler.h
View file

0
gst/speexresample/speex_resampler_double.c → gst/audioresample/speex_resampler_double.c
View file

0
gst/speexresample/speex_resampler_float.c → gst/audioresample/speex_resampler_float.c
View file

0
gst/speexresample/speex_resampler_int.c → gst/audioresample/speex_resampler_int.c
View file

0
gst/speexresample/speex_resampler_wrapper.h → gst/audioresample/speex_resampler_wrapper.h
View file

37
gst/speexresample/Makefile.am
View file

@ -1,37 +0,0 @@
plugin_LTLIBRARIES = libgstaudioresample.la
libgstaudioresample_la_SOURCES = \
gstspeexresample.c \
speex_resampler_int.c \
speex_resampler_float.c \
speex_resampler_double.c
libgstaudioresample_la_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) \
$(GST_CFLAGS) \
$(LIBOIL_CFLAGS)
libgstaudioresample_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) \
$(GST_BASE_LIBS) \
$(GST_LIBS) \
$(LIBOIL_LIBS) \
$(LIBM)
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudioresample_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = \
arch.h \
fixed_arm4.h \
fixed_arm5e.h \
fixed_bfin.h \
fixed_debug.h \
fixed_generic.h \
gstspeexresample.h \
resample.c \
resample_sse.h \
speex_resampler.h \
speex_resampler_wrapper.h

1404
gst/speexresample/gstspeexresample.c
View file

File diff suppressed because it is too large Load diff

90
gst/speexresample/gstspeexresample.h
View file

@ -1,90 +0,0 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2007-2008> Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __SPEEX_RESAMPLE_H__
#define __SPEEX_RESAMPLE_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include "speex_resampler_wrapper.h"
G_BEGIN_DECLS
#define GST_TYPE_SPEEX_RESAMPLE \
(gst_speex_resample_get_type())
#define GST_SPEEX_RESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResample))
#define GST_SPEEX_RESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SPEEX_RESAMPLE,GstSpeexResampleClass))
#define GST_IS_SPEEX_RESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SPEEX_RESAMPLE))
#define GST_IS_SPEEX_RESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SPEEX_RESAMPLE))
typedef struct _GstSpeexResample GstSpeexResample;
typedef struct _GstSpeexResampleClass GstSpeexResampleClass;
/**
* GstSpeexResample:
*
* Opaque data structure.
*/
struct _GstSpeexResample {
GstBaseTransform element;
/* <private> */
GstCaps *srccaps, *sinkcaps;
gboolean need_discont;
guint64 next_offset;
GstClockTime next_ts;
GstClockTime next_upstream_ts;
gint channels;
gint inrate;
gint outrate;
gint quality;
gint width;
gboolean fp;
guint8 *tmp_in;
guint tmp_in_size;
guint8 *tmp_out;
guint tmp_out_size;
SpeexResamplerState *state;
const SpeexResampleFuncs *funcs;
};
struct _GstSpeexResampleClass {
GstBaseTransformClass parent_class;
};
GType gst_speex_resample_get_type(void);
G_END_DECLS
#endif /* __SPEEX_RESAMPLE_H__ */

1340
gst/speexresample/resample.c
View file

File diff suppressed because it is too large Load diff

33
gst/typefind/gsttypefindfunctions.c
View file

@ -412,6 +412,7 @@ mid_type_find (GstTypeFind * tf, gpointer unused)
&& data[3] == 'd') && data[3] == 'd')
gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MID_CAPS); gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MID_CAPS);
} }
/*** audio/mobile-xmf ***/ /*** audio/mobile-xmf ***/
static GstStaticCaps mxmf_caps = GST_STATIC_CAPS ("audio/mobile-xmf"); static GstStaticCaps mxmf_caps = GST_STATIC_CAPS ("audio/mobile-xmf");
@ -422,21 +423,21 @@ mxmf_type_find (GstTypeFind * tf, gpointer unused)
{ {
guint8 *data = NULL; guint8 *data = NULL;
/* Search FileId "XMF_" 4 bytes */ /* Search FileId "XMF_" 4 bytes */
data = gst_type_find_peek (tf, 0, 4); data = gst_type_find_peek (tf, 0, 4);
if (data && data[0] == 'X' && data[1] == 'M' && data[2] == 'F' if (data && data[0] == 'X' && data[1] == 'M' && data[2] == 'F'
&& data[3] == '_') { && data[3] == '_') {
/* Search Format version "2.00" 4 bytes */ /* Search Format version "2.00" 4 bytes */
data = gst_type_find_peek (tf, 4, 4); data = gst_type_find_peek (tf, 4, 4);
if (data && data[0] == '2' && data[1] == '.' && data[2] == '0' if (data && data[0] == '2' && data[1] == '.' && data[2] == '0'
&& data[3] == '0') { && data[3] == '0') {
/* Search TypeId 2 1 byte */ /* Search TypeId 2 1 byte */
data = gst_type_find_peek (tf, 11, 1); data = gst_type_find_peek (tf, 11, 1);
if (data && data[0] == 2 ) { if (data && data[0] == 2) {
gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MXMF_CAPS); gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MXMF_CAPS);
} }
} }
} }
} }
@ -3208,8 +3209,8 @@ plugin_init (GstPlugin * plugin)
#endif #endif
TYPE_FIND_REGISTER (plugin, "audio/midi", GST_RANK_PRIMARY, mid_type_find, TYPE_FIND_REGISTER (plugin, "audio/midi", GST_RANK_PRIMARY, mid_type_find,
mid_exts, MID_CAPS, NULL, NULL); mid_exts, MID_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "audio/mobile-xmf", GST_RANK_PRIMARY, mxmf_type_find, TYPE_FIND_REGISTER (plugin, "audio/mobile-xmf", GST_RANK_PRIMARY,
mxmf_exts, MXMF_CAPS, NULL, NULL); mxmf_type_find, mxmf_exts, MXMF_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "video/x-fli", GST_RANK_MARGINAL, flx_type_find, TYPE_FIND_REGISTER (plugin, "video/x-fli", GST_RANK_MARGINAL, flx_type_find,
flx_exts, FLX_CAPS, NULL, NULL); flx_exts, FLX_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "application/x-id3v2", GST_RANK_PRIMARY + 103, TYPE_FIND_REGISTER (plugin, "application/x-id3v2", GST_RANK_PRIMARY + 103,
@ -3241,8 +3242,8 @@ plugin_init (GstPlugin * plugin)
mpeg4_video_type_find, m4v_exts, MPEG_VIDEO_CAPS, NULL, NULL); mpeg4_video_type_find, m4v_exts, MPEG_VIDEO_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "video/x-h264", GST_RANK_PRIMARY, TYPE_FIND_REGISTER (plugin, "video/x-h264", GST_RANK_PRIMARY,
h264_video_type_find, h264_exts, MPEG_VIDEO_CAPS, NULL, NULL); h264_video_type_find, h264_exts, MPEG_VIDEO_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "video/x-nuv", GST_RANK_SECONDARY, TYPE_FIND_REGISTER (plugin, "video/x-nuv", GST_RANK_SECONDARY, nuv_type_find,
nuv_type_find, nuv_exts, NUV_CAPS, NULL, NULL); nuv_exts, NUV_CAPS, NULL, NULL);
/* ISO formats */ /* ISO formats */
TYPE_FIND_REGISTER (plugin, "audio/x-m4a", GST_RANK_PRIMARY, m4a_type_find, TYPE_FIND_REGISTER (plugin, "audio/x-m4a", GST_RANK_PRIMARY, m4a_type_find,

6
tests/check/Makefile.am
View file

@ -91,6 +91,7 @@ check_PROGRAMS = \
elements/adder \ elements/adder \
elements/audioconvert \ elements/audioconvert \
elements/audiorate \ elements/audiorate \
elements/audioresample \
elements/audiotestsrc \ elements/audiotestsrc \
elements/decodebin \ elements/decodebin \
elements/ffmpegcolorspace \ elements/ffmpegcolorspace \
@ -99,7 +100,6 @@ check_PROGRAMS = \
elements/multifdsink \ elements/multifdsink \
elements/playbin \ elements/playbin \
$(check_subparse) \ $(check_subparse) \
elements/speexresample \
elements/videorate \ elements/videorate \
elements/videotestsrc \ elements/videotestsrc \
elements/volume \ elements/volume \
@ -241,12 +241,12 @@ elements_decodebin_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_subparse_LDADD = $(LDADD) elements_subparse_LDADD = $(LDADD)
elements_subparse_CFLAGS = $(CFLAGS) $(AM_CFLAGS) elements_subparse_CFLAGS = $(CFLAGS) $(AM_CFLAGS)
elements_speexresample_CFLAGS = \ elements_audioresample_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \ $(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) \ $(GST_BASE_CFLAGS) \
$(AM_CFLAGS) $(AM_CFLAGS)
elements_speexresample_LDADD = \ elements_audioresample_LDADD = \
$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \ $(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
$(top_builddir)/gst-libs/gst/interfaces/libgstinterfaces-@GST_MAJORMINOR@.la \ $(top_builddir)/gst-libs/gst/interfaces/libgstinterfaces-@GST_MAJORMINOR@.la \
$(GST_BASE_LIBS) \ $(GST_BASE_LIBS) \

158
tests/check/elements/audioresample.c
View file

@ -1,6 +1,6 @@
/* GStreamer /* GStreamer
* *
* unit test for audioresample * unit test for audioresample, based on the audioresample unit test
* *
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org> * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net> * Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
@ -25,13 +25,21 @@
#include <gst/check/gstcheck.h> #include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
/* For ease of programming we use globals to keep refs for our floating /* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad, * src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */ * get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad; static GstPad *mysrcpad, *mysinkpad;
#define RESAMPLE_CAPS_FLOAT \
"audio/x-raw-float, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) { 32, 64 }"
#define RESAMPLE_CAPS_TEMPLATE_STRING \ #define RESAMPLE_CAPS_INT \
"audio/x-raw-int, " \ "audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \ "channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \ "rate = (int) [ 1, MAX ], " \
@ -40,6 +48,10 @@ static GstPad *mysrcpad, *mysinkpad;
"depth = (int) 16, " \ "depth = (int) 16, " \
"signed = (bool) TRUE" "signed = (bool) TRUE"
#define RESAMPLE_CAPS_TEMPLATE_STRING \
RESAMPLE_CAPS_FLOAT " ; " \
RESAMPLE_CAPS_INT
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_SINK,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
@ -52,7 +64,8 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
); );
static GstElement * static GstElement *
setup_audioresample (int channels, int inrate, int outrate) setup_audioresample (int channels, int inrate, int outrate, int width,
gboolean fp)
{ {
GstElement *audioresample; GstElement *audioresample;
GstCaps *caps; GstCaps *caps;
@ -61,10 +74,15 @@ setup_audioresample (int channels, int inrate, int outrate)
GST_DEBUG ("setup_audioresample"); GST_DEBUG ("setup_audioresample");
audioresample = gst_check_setup_element ("audioresample"); audioresample = gst_check_setup_element ("audioresample");
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING); if (fp)
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
else
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0); structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels, gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, inrate, NULL); "rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
if (!fp)
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample, fail_unless (gst_element_set_state (audioresample,
@ -75,10 +93,15 @@ setup_audioresample (int channels, int inrate, int outrate)
gst_pad_set_caps (mysrcpad, caps); gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps); gst_caps_unref (caps);
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING); if (fp)
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
else
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0); structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels, gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, outrate, NULL); "rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
if (!fp)
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_caps_is_fixed (caps));
mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps); mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
@ -90,6 +113,8 @@ setup_audioresample (int channels, int inrate, int outrate)
gst_pad_set_active (mysinkpad, TRUE); gst_pad_set_active (mysinkpad, TRUE);
gst_pad_set_active (mysrcpad, TRUE); gst_pad_set_active (mysrcpad, TRUE);
gst_caps_unref (caps);
return audioresample; return audioresample;
} }
@ -153,7 +178,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
int i, j; int i, j;
gint16 *p; gint16 *p;
audioresample = setup_audioresample (2, inrate, outrate); audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad); caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_caps_is_fixed (caps));
@ -164,7 +189,7 @@ test_perfect_stream_instance (int inrate, int outrate, int samples,
for (j = 1; j <= numbuffers; ++j) { for (j = 1; j <= numbuffers; ++j) {
inbuffer = gst_buffer_new_and_alloc (samples * 4); inbuffer = gst_buffer_new_and_alloc (samples * 4);
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate; GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1); GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
GST_BUFFER_OFFSET (inbuffer) = offset; GST_BUFFER_OFFSET (inbuffer) = offset;
offset += samples; offset += samples;
@ -240,7 +265,7 @@ test_discont_stream_instance (int inrate, int outrate, int samples,
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d", GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
inrate, outrate, samples, numbuffers); inrate, outrate, samples, numbuffers);
audioresample = setup_audioresample (2, inrate, outrate); audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad); caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_caps_is_fixed (caps));
@ -326,7 +351,7 @@ GST_START_TEST (test_reuse)
GstBuffer *inbuffer; GstBuffer *inbuffer;
GstCaps *caps; GstCaps *caps;
audioresample = setup_audioresample (1, 9343, 48000); audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad); caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_caps_is_fixed (caps));
@ -477,6 +502,7 @@ live_switch_push (int rate, GstCaps * caps)
desired = gst_caps_copy (caps); desired = gst_caps_copy (caps);
gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL); gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
gst_pad_set_caps (mysrcpad, desired);
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad, fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK); GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
@ -485,6 +511,8 @@ live_switch_push (int rate, GstCaps * caps)
* returns a buffer with exactly the same caps as we requested so the actual * returns a buffer with exactly the same caps as we requested so the actual
* renegotiation (if needed) will be done in the _chain*/ * renegotiation (if needed) will be done in the _chain*/
fail_unless (inbuffer != NULL); fail_unless (inbuffer != NULL);
GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
desired, GST_BUFFER_CAPS (inbuffer));
fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer))); fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer)); memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
@ -516,7 +544,7 @@ GST_START_TEST (test_live_switch)
GstEvent *newseg; GstEvent *newseg;
GstCaps *caps; GstCaps *caps;
audioresample = setup_audioresample (4, 48000, 48000); audioresample = setup_audioresample (4, 48000, 48000, 16, FALSE);
/* Let the sinkpad act like something that can only handle things of /* 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 * rate 48000- and can only allocate buffers for that rate, but if someone
@ -525,6 +553,8 @@ GST_START_TEST (test_live_switch)
gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000); gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps); gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
gst_pad_use_fixed_caps (mysrcpad);
caps = gst_pad_get_negotiated_caps (mysrcpad); caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps)); fail_unless (gst_caps_is_fixed (caps));
@ -549,7 +579,104 @@ GST_START_TEST (test_live_switch)
gst_caps_unref (caps); gst_caps_unref (caps);
} }
GST_END_TEST static Suite * GST_END_TEST;
#ifndef GST_DISABLE_PARSE
static GMainLoop *loop;
static gint messages = 0;
static void
element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
gchar *s;
s = gst_structure_to_string (gst_message_get_structure (message));
GST_DEBUG ("Received message: %s", s);
g_free (s);
messages++;
}
static void
eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
GST_DEBUG ("Received eos");
g_main_loop_quit (loop);
}
static void
test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
{
GstElement *pipeline;
GstBus *bus;
GError *error = NULL;
gchar *pipe_str;
pipe_str =
g_strdup_printf
("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
(fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
outrate, width);
pipeline = gst_parse_launch (pipe_str, &error);
fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
error ? error->message : "(invalid error)");
g_free (pipe_str);
bus = gst_element_get_bus (pipeline);
fail_if (bus == NULL);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
NULL);
g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* run until we receive EOS */
loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (loop);
g_main_loop_unref (loop);
loop = NULL;
gst_element_set_state (pipeline, GST_STATE_NULL);
fail_if (messages > 0, "Received imperfect timestamp messages");
gst_object_unref (pipeline);
}
GST_START_TEST (test_pipelines)
{
gint quality;
/* Test qualities 0, 5 and 10 */
for (quality = 0; quality < 11; quality += 5) {
test_pipeline (8, FALSE, 44100, 48000, quality);
test_pipeline (8, FALSE, 48000, 44100, quality);
test_pipeline (16, FALSE, 44100, 48000, quality);
test_pipeline (16, FALSE, 48000, 44100, quality);
test_pipeline (24, FALSE, 44100, 48000, quality);
test_pipeline (24, FALSE, 48000, 44100, quality);
test_pipeline (32, FALSE, 44100, 48000, quality);
test_pipeline (32, FALSE, 48000, 44100, quality);
test_pipeline (32, TRUE, 44100, 48000, quality);
test_pipeline (32, TRUE, 48000, 44100, quality);
test_pipeline (64, TRUE, 44100, 48000, quality);
test_pipeline (64, TRUE, 48000, 44100, quality);
}
}
GST_END_TEST;
#endif
static Suite *
audioresample_suite (void) audioresample_suite (void)
{ {
Suite *s = suite_create ("audioresample"); Suite *s = suite_create ("audioresample");
@ -562,6 +689,11 @@ audioresample_suite (void)
tcase_add_test (tc_chain, test_shutdown); tcase_add_test (tc_chain, test_shutdown);
tcase_add_test (tc_chain, test_live_switch); tcase_add_test (tc_chain, test_live_switch);
#ifndef GST_DISABLE_PARSE
tcase_set_timeout (tc_chain, 360);
tcase_add_test (tc_chain, test_pipelines);
#endif
return s; return s;
} }

700
tests/check/elements/speexresample.c
View file

@ -1,700 +0,0 @@
/* GStreamer
*
* unit test for speexresample, based on the audioresample unit test
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
#define RESAMPLE_CAPS_FLOAT \
"audio/x-raw-float, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) { 32, 64 }"
#define RESAMPLE_CAPS_INT \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (bool) TRUE"
#define RESAMPLE_CAPS_TEMPLATE_STRING \
RESAMPLE_CAPS_FLOAT " ; " \
RESAMPLE_CAPS_INT
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
);
static GstElement *
setup_speexresample (int channels, int inrate, int outrate, int width,
gboolean fp)
{
GstElement *speexresample;
GstCaps *caps;
GstStructure *structure;
GST_DEBUG ("setup_speexresample");
speexresample = gst_check_setup_element ("audioresample");
if (fp)
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
else
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, inrate, "width", G_TYPE_INT, width, NULL);
if (!fp)
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (speexresample,
GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
"could not set to paused");
mysrcpad = gst_check_setup_src_pad (speexresample, &srctemplate, caps);
gst_pad_set_caps (mysrcpad, caps);
gst_caps_unref (caps);
if (fp)
caps = gst_caps_from_string (RESAMPLE_CAPS_FLOAT);
else
caps = gst_caps_from_string (RESAMPLE_CAPS_INT);
structure = gst_caps_get_structure (caps, 0);
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, outrate, "width", G_TYPE_INT, width, NULL);
if (!fp)
gst_structure_set (structure, "depth", G_TYPE_INT, width, NULL);
fail_unless (gst_caps_is_fixed (caps));
mysinkpad = gst_check_setup_sink_pad (speexresample, &sinktemplate, caps);
/* this installs a getcaps func that will always return the caps we set
* later */
gst_pad_set_caps (mysinkpad, caps);
gst_pad_use_fixed_caps (mysinkpad);
gst_pad_set_active (mysinkpad, TRUE);
gst_pad_set_active (mysrcpad, TRUE);
gst_caps_unref (caps);
return speexresample;
}
static void
cleanup_speexresample (GstElement * speexresample)
{
GST_DEBUG ("cleanup_speexresample");
fail_unless (gst_element_set_state (speexresample,
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 (speexresample);
gst_check_teardown_sink_pad (speexresample);
gst_check_teardown_element (speexresample);
}
static void
fail_unless_perfect_stream (void)
{
guint64 timestamp = 0L, duration = 0L;
guint64 offset = 0L, offset_end = 0L;
GList *l;
GstBuffer *buffer;
for (l = buffers; l; l = l->next) {
buffer = GST_BUFFER (l->data);
ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
G_GUINT64_FORMAT,
GST_BUFFER_TIMESTAMP (buffer),
GST_BUFFER_DURATION (buffer),
GST_BUFFER_OFFSET (buffer), GST_BUFFER_OFFSET_END (buffer));
fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
duration = GST_BUFFER_DURATION (buffer);
offset_end = GST_BUFFER_OFFSET_END (buffer);
timestamp += duration;
offset = offset_end;
gst_buffer_unref (buffer);
}
g_list_free (buffers);
buffers = NULL;
}
/* this tests that the output is a perfect stream if the input is */
static void
test_perfect_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
GstElement *speexresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
guint64 offset = 0;
int i, j;
gint16 *p;
speexresample = setup_speexresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (speexresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
for (j = 1; j <= numbuffers; ++j) {
inbuffer = gst_buffer_new_and_alloc (samples * 4);
GST_BUFFER_DURATION (inbuffer) = GST_FRAMES_TO_CLOCK_TIME (samples, inrate);
GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
GST_BUFFER_OFFSET (inbuffer) = offset;
offset += samples;
GST_BUFFER_OFFSET_END (inbuffer) = offset;
gst_buffer_set_caps (inbuffer, caps);
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
/* create a 16 bit signed ramp */
for (i = 0; i < samples; ++i) {
*p = -32767 + i * (65535 / samples);
++p;
*p = -32767 + i * (65535 / samples);
++p;
}
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), j);
}
/* FIXME: we should make speexresample 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);
fail_unless_perfect_stream ();
/* cleanup */
gst_caps_unref (caps);
cleanup_speexresample (speexresample);
}
/* make sure that outgoing buffers are contiguous in timestamp/duration and
* offset/offsetend
*/
GST_START_TEST (test_perfect_stream)
{
/* integral scalings */
test_perfect_stream_instance (48000, 24000, 500, 20);
test_perfect_stream_instance (48000, 12000, 500, 20);
test_perfect_stream_instance (12000, 24000, 500, 20);
test_perfect_stream_instance (12000, 48000, 500, 20);
/* non-integral scalings */
test_perfect_stream_instance (44100, 8000, 500, 20);
test_perfect_stream_instance (8000, 44100, 500, 20);
/* wacky scalings */
test_perfect_stream_instance (12345, 54321, 500, 20);
test_perfect_stream_instance (101, 99, 500, 20);
}
GST_END_TEST;
/* this tests that the output is a correct discontinuous stream
* if the input is; ie input drops in time come out the same way */
static void
test_discont_stream_instance (int inrate, int outrate, int samples,
int numbuffers)
{
GstElement *speexresample;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
GstClockTime ints;
int i, j;
gint16 *p;
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
inrate, outrate, samples, numbuffers);
speexresample = setup_speexresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (speexresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
for (j = 1; j <= numbuffers; ++j) {
inbuffer = gst_buffer_new_and_alloc (samples * 4);
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
/* "drop" half the buffers */
ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
GST_BUFFER_TIMESTAMP (inbuffer) = ints;
GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
gst_buffer_set_caps (inbuffer, caps);
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
/* create a 16 bit signed ramp */
for (i = 0; i < samples; ++i) {
*p = -32767 + i * (65535 / samples);
++p;
*p = -32767 + i * (65535 / samples);
++p;
}
GST_DEBUG ("Sending Buffer time:%" G_GUINT64_FORMAT " duration:%"
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (inbuffer),
GST_BUFFER_DURATION (inbuffer), GST_BUFFER_IS_DISCONT (inbuffer),
GST_BUFFER_OFFSET (inbuffer), GST_BUFFER_OFFSET_END (inbuffer));
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* check if the timestamp of the pushed buffer matches the incoming one */
outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
fail_if (outbuffer == NULL);
fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
GST_DEBUG ("Got Buffer time:%" G_GUINT64_FORMAT " duration:%"
G_GINT64_FORMAT " discont:%d offset:%" G_GUINT64_FORMAT " offset_end:%"
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (outbuffer),
GST_BUFFER_DURATION (outbuffer), GST_BUFFER_IS_DISCONT (outbuffer),
GST_BUFFER_OFFSET (outbuffer), GST_BUFFER_OFFSET_END (outbuffer));
if (j > 1) {
fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
"expected discont for buffer #%d", j);
}
}
/* cleanup */
gst_caps_unref (caps);
cleanup_speexresample (speexresample);
}
GST_START_TEST (test_discont_stream)
{
/* integral scalings */
test_discont_stream_instance (48000, 24000, 500, 20);
test_discont_stream_instance (48000, 12000, 500, 20);
test_discont_stream_instance (12000, 24000, 500, 20);
test_discont_stream_instance (12000, 48000, 500, 20);
/* non-integral scalings */
test_discont_stream_instance (44100, 8000, 500, 20);
test_discont_stream_instance (8000, 44100, 500, 20);
/* wacky scalings */
test_discont_stream_instance (12345, 54321, 500, 20);
test_discont_stream_instance (101, 99, 500, 20);
}
GST_END_TEST;
GST_START_TEST (test_reuse)
{
GstElement *speexresample;
GstEvent *newseg;
GstBuffer *inbuffer;
GstCaps *caps;
speexresample = setup_speexresample (1, 9343, 48000, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (speexresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
gst_buffer_set_caps (inbuffer, caps);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
/* now reset and try again ... */
fail_unless (gst_element_set_state (speexresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
fail_unless (gst_element_set_state (speexresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
gst_buffer_set_caps (inbuffer, caps);
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 speexresample 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_speexresample (speexresample);
gst_caps_unref (caps);
}
GST_END_TEST;
GST_START_TEST (test_shutdown)
{
GstElement *pipeline, *src, *cf1, *ar, *cf2, *sink;
GstCaps *caps;
guint i;
/* create pipeline, force speexresample to actually resample */
pipeline = gst_pipeline_new (NULL);
src = gst_check_setup_element ("audiotestsrc");
cf1 = gst_check_setup_element ("capsfilter");
ar = gst_check_setup_element ("audioresample");
cf2 = gst_check_setup_element ("capsfilter");
g_object_set (cf2, "name", "capsfilter2", NULL);
sink = gst_check_setup_element ("fakesink");
caps =
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 11025, NULL);
g_object_set (cf1, "caps", caps, NULL);
gst_caps_unref (caps);
caps =
gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT, 48000, NULL);
g_object_set (cf2, "caps", caps, NULL);
gst_caps_unref (caps);
/* don't want to sync against the clock, the more throughput the better */
g_object_set (src, "is-live", FALSE, NULL);
g_object_set (sink, "sync", FALSE, NULL);
gst_bin_add_many (GST_BIN (pipeline), src, cf1, ar, cf2, sink, NULL);
fail_if (!gst_element_link_many (src, cf1, ar, cf2, sink, NULL));
/* now, wait until pipeline is running and then shut it down again; repeat */
for (i = 0; i < 20; ++i) {
gst_element_set_state (pipeline, GST_STATE_PAUSED);
gst_element_get_state (pipeline, NULL, NULL, -1);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_usleep (100);
gst_element_set_state (pipeline, GST_STATE_NULL);
}
gst_object_unref (pipeline);
}
GST_END_TEST;
static GstFlowReturn
live_switch_alloc_only_48000 (GstPad * pad, guint64 offset,
guint size, GstCaps * caps, GstBuffer ** buf)
{
GstStructure *structure;
gint rate;
gint channels;
GstCaps *desired;
structure = gst_caps_get_structure (caps, 0);
fail_unless (gst_structure_get_int (structure, "rate", &rate));
fail_unless (gst_structure_get_int (structure, "channels", &channels));
if (rate < 48000)
return GST_FLOW_NOT_NEGOTIATED;
desired = gst_caps_copy (caps);
gst_caps_set_simple (desired, "rate", G_TYPE_INT, 48000, NULL);
*buf = gst_buffer_new_and_alloc (channels * 48000);
gst_buffer_set_caps (*buf, desired);
gst_caps_unref (desired);
return GST_FLOW_OK;
}
static GstCaps *
live_switch_get_sink_caps (GstPad * pad)
{
GstCaps *result;
result = gst_caps_copy (GST_PAD_CAPS (pad));
gst_caps_set_simple (result,
"rate", GST_TYPE_INT_RANGE, 48000, G_MAXINT, NULL);
return result;
}
static void
live_switch_push (int rate, GstCaps * caps)
{
GstBuffer *inbuffer;
GstCaps *desired;
GList *l;
desired = gst_caps_copy (caps);
gst_caps_set_simple (desired, "rate", G_TYPE_INT, rate, NULL);
gst_pad_set_caps (mysrcpad, desired);
fail_unless (gst_pad_alloc_buffer_and_set_caps (mysrcpad,
GST_BUFFER_OFFSET_NONE, rate * 4, desired, &inbuffer) == GST_FLOW_OK);
/* When the basetransform hits the non-configured case it always
* returns a buffer with exactly the same caps as we requested so the actual
* renegotiation (if needed) will be done in the _chain*/
fail_unless (inbuffer != NULL);
GST_DEBUG ("desired: %" GST_PTR_FORMAT ".... got: %" GST_PTR_FORMAT,
desired, GST_BUFFER_CAPS (inbuffer));
fail_unless (gst_caps_is_equal (desired, GST_BUFFER_CAPS (inbuffer)));
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
GST_BUFFER_OFFSET (inbuffer) = 0;
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
for (l = buffers; l; l = l->next) {
GstBuffer *buffer = GST_BUFFER (l->data);
gst_buffer_unref (buffer);
}
g_list_free (buffers);
buffers = NULL;
gst_caps_unref (desired);
}
GST_START_TEST (test_live_switch)
{
GstElement *speexresample;
GstEvent *newseg;
GstCaps *caps;
speexresample = setup_speexresample (4, 48000, 48000, 16, FALSE);
/* 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
* tries to get a buffer with a rate higher then 48000 tries to renegotiate
* */
gst_pad_set_bufferalloc_function (mysinkpad, live_switch_alloc_only_48000);
gst_pad_set_getcaps_function (mysinkpad, live_switch_get_sink_caps);
gst_pad_use_fixed_caps (mysrcpad);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (speexresample,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
/* downstream can provide the requested rate, a buffer alloc will be passed
* on */
live_switch_push (48000, caps);
/* Downstream can never accept this rate, buffer alloc isn't passed on */
live_switch_push (40000, caps);
/* Downstream can provide the requested rate but will re-negotiate */
live_switch_push (50000, caps);
cleanup_speexresample (speexresample);
gst_caps_unref (caps);
}
GST_END_TEST;
#ifndef GST_DISABLE_PARSE
static GMainLoop *loop;
static gint messages = 0;
static void
element_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
gchar *s;
s = gst_structure_to_string (gst_message_get_structure (message));
GST_DEBUG ("Received message: %s", s);
g_free (s);
messages++;
}
static void
eos_message_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
GST_DEBUG ("Received eos");
g_main_loop_quit (loop);
}
static void
test_pipeline (gint width, gboolean fp, gint inrate, gint outrate, gint quality)
{
GstElement *pipeline;
GstBus *bus;
GError *error = NULL;
gchar *pipe_str;
pipe_str =
g_strdup_printf
("audiotestsrc num-buffers=10 ! audioconvert ! audio/x-raw-%s,rate=%d,width=%d,channels=2 ! audioresample quality=%d ! audio/x-raw-%s,rate=%d,width=%d ! identity check-imperfect-timestamp=TRUE ! fakesink",
(fp) ? "float" : "int", inrate, width, quality, (fp) ? "float" : "int",
outrate, width);
pipeline = gst_parse_launch (pipe_str, &error);
fail_unless (pipeline != NULL, "Error parsing pipeline: %s",
error ? error->message : "(invalid error)");
g_free (pipe_str);
bus = gst_element_get_bus (pipeline);
fail_if (bus == NULL);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message::element", (GCallback) element_message_cb,
NULL);
g_signal_connect (bus, "message::eos", (GCallback) eos_message_cb, NULL);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* run until we receive EOS */
loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (loop);
g_main_loop_unref (loop);
loop = NULL;
gst_element_set_state (pipeline, GST_STATE_NULL);
fail_if (messages > 0, "Received imperfect timestamp messages");
gst_object_unref (pipeline);
}
GST_START_TEST (test_pipelines)
{
gint quality;
/* Test qualities 0, 5 and 10 */
for (quality = 0; quality < 11; quality += 5) {
test_pipeline (8, FALSE, 44100, 48000, quality);
test_pipeline (8, FALSE, 48000, 44100, quality);
test_pipeline (16, FALSE, 44100, 48000, quality);
test_pipeline (16, FALSE, 48000, 44100, quality);
test_pipeline (24, FALSE, 44100, 48000, quality);
test_pipeline (24, FALSE, 48000, 44100, quality);
test_pipeline (32, FALSE, 44100, 48000, quality);
test_pipeline (32, FALSE, 48000, 44100, quality);
test_pipeline (32, TRUE, 44100, 48000, quality);
test_pipeline (32, TRUE, 48000, 44100, quality);
test_pipeline (64, TRUE, 44100, 48000, quality);
test_pipeline (64, TRUE, 48000, 44100, quality);
}
}
GST_END_TEST;
#endif
static Suite *
speexresample_suite (void)
{
Suite *s = suite_create ("speexresample");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_perfect_stream);
tcase_add_test (tc_chain, test_discont_stream);
tcase_add_test (tc_chain, test_reuse);
tcase_add_test (tc_chain, test_shutdown);
tcase_add_test (tc_chain, test_live_switch);
#ifndef GST_DISABLE_PARSE
tcase_set_timeout (tc_chain, 360);
tcase_add_test (tc_chain, test_pipelines);
#endif
return s;
}
GST_CHECK_MAIN (speexresample);