AG_GST_CHECK_PLUGIN(ffmpegcolorspace)
AG_GST_CHECK_PLUGIN(gdp)
AG_GST_CHECK_PLUGIN(playback)
-AG_GST_CHECK_PLUGIN(speexresample)
+AG_GST_CHECK_PLUGIN(audioresample)
AG_GST_CHECK_PLUGIN(subparse)
AG_GST_CHECK_PLUGIN(tcp)
AG_GST_CHECK_PLUGIN(typefind)
gst/ffmpegcolorspace/Makefile
gst/gdp/Makefile
gst/playback/Makefile
-gst/speexresample/Makefile
+gst/audioresample/Makefile
gst/subparse/Makefile
gst/tcp/Makefile
gst/typefind/Makefile
$(top_srcdir)/gst/gdp/gstgdpdepay.h \
$(top_srcdir)/gst/gdp/gstgdppay.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/gsttcpclientsrc.h \
$(top_srcdir)/gst/tcp/gsttcpclientsink.h \
<SECTION>
<FILE>element-audioresample</FILE>
<TITLE>audioresample</TITLE>
-GstSpeexResample
-<SUBSECTION Standard>
-GST_SPEEX_RESAMPLE
-GST_IS_SPEEX_RESAMPLE
-GST_TYPE_SPEEX_RESAMPLE
-gst_speex_resample_get_type
-GST_SPEEX_RESAMPLE_CLASS
-GST_IS_SPEEX_RESAMPLE_CLASS
-GstSpeexResampleClass
+GstAudioResample
+<SUBSECTION Standard>
+GST_AUDIO_RESAMPLE
+GST_IS_AUDIO_RESAMPLE
+GST_TYPE_AUDIO_RESAMPLE
+gst_audio_resample_get_type
+GST_AUDIO_RESAMPLE_CLASS
+GST_IS_AUDIO_RESAMPLE_CLASS
+GstAudioResampleClass
</SECTION>
<SECTION>
<ARG>
<NAME>GstMultiFdSink::buffers-max</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffers max</NICK>
<BLURB>max number of buffers to queue for a client (-1 = no limit).</BLURB>
<ARG>
<NAME>GstMultiFdSink::buffers-soft-max</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffers soft max</NICK>
<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
<ARG>
<NAME>GstMultiFdSink::buffers-min</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffers min</NICK>
<BLURB>min number of buffers to queue (-1 = as few as possible).</BLURB>
<ARG>
<NAME>GstMultiFdSink::bytes-min</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Bytes min</NICK>
<BLURB>min number of bytes to queue (-1 = as little as possible).</BLURB>
<ARG>
<NAME>GstMultiFdSink::time-min</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Time min</NICK>
<BLURB>min number of time to queue (-1 = as little as possible).</BLURB>
<ARG>
<NAME>GstMultiFdSink::units-max</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Units max</NICK>
<BLURB>max number of units to queue (-1 = no limit).</BLURB>
<ARG>
<NAME>GstMultiFdSink::units-soft-max</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Units soft max</NICK>
<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
<ARG>
<NAME>GstMultiFdSink::qos-dscp</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,63]</RANGE>
+<RANGE>[G_MAXULONG,63]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>QoS diff srv code point</NICK>
<BLURB>Quality of Service, differentiated services code point (-1 default).</BLURB>
<ARG>
<NAME>GstVorbisEnc::bitrate</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,250001]</RANGE>
+<RANGE>[G_MAXULONG,250001]</RANGE>
<FLAGS>rw</FLAGS>
<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>
<ARG>
<NAME>GstVorbisEnc::max-bitrate</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,250001]</RANGE>
+<RANGE>[G_MAXULONG,250001]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Maximum Bitrate</NICK>
<BLURB>Specify a maximum bitrate (in bps). Useful for streaming applications. (-1 == disabled).</BLURB>
<ARG>
<NAME>GstVorbisEnc::min-bitrate</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,250001]</RANGE>
+<RANGE>[G_MAXULONG,250001]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Minimum Bitrate</NICK>
<BLURB>Specify a minimum bitrate (in bps). Useful for encoding for a fixed-size channel. (-1 == disabled).</BLURB>
<ARG>
<NAME>GstCdParanoiaSrc::read-speed</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Read speed</NICK>
<BLURB>Read from device at specified speed.</BLURB>
<ARG>
<NAME>GstCdParanoiaSrc::search-overlap</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,75]</RANGE>
+<RANGE>[G_MAXULONG,75]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Search overlap</NICK>
<BLURB>Force minimum overlap search during verification to n sectors.</BLURB>
<ARG>
<NAME>GstURIDecodeBin::buffer-duration</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer duration (ns)</NICK>
<BLURB>Buffer duration when buffering network streams.</BLURB>
<ARG>
<NAME>GstURIDecodeBin::buffer-size</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer size (bytes)</NICK>
<BLURB>Buffer size when buffering network streams.</BLURB>
<ARG>
<NAME>GstPlayBin2::current-audio</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Current audio</NICK>
<BLURB>Currently playing audio stream (-1 = auto).</BLURB>
<ARG>
<NAME>GstPlayBin2::current-text</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Current Text</NICK>
<BLURB>Currently playing text stream (-1 = auto).</BLURB>
<ARG>
<NAME>GstPlayBin2::current-video</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Current Video</NICK>
<BLURB>Currently playing video stream (-1 = auto).</BLURB>
<ARG>
<NAME>GstPlayBin2::buffer-duration</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer duration (ns)</NICK>
<BLURB>Buffer duration when buffering network streams.</BLURB>
<ARG>
<NAME>GstPlayBin2::buffer-size</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer size (bytes)</NICK>
<BLURB>Buffer size when buffering network streams.</BLURB>
<ARG>
<NAME>GstAppSrc::max-latency</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Max Latency</NICK>
<BLURB>The maximum latency (-1 = unlimited).</BLURB>
<ARG>
<NAME>GstAppSrc::min-latency</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Min Latency</NICK>
<BLURB>The minimum latency (-1 = default).</BLURB>
<ARG>
<NAME>GstAppSrc::size</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Size</NICK>
<BLURB>The size of the data stream in bytes (-1 if unknown).</BLURB>
<DEFAULT>Stream</DEFAULT>
</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>
+
GstPlayBaseBin
GstPlayBin
GstPlayBin2
+ GstDecodeBin
GstDecodeBin2
GstURIDecodeBin
- GstDecodeBin
- GstBaseSrc
- GstPushSrc
- GstCddaBaseSrc
- GstCdParanoiaSrc
- GstBaseAudioSrc
- GstAudioSrc
- GstAlsaSrc
- GstV4lElement
- GstV4lSrc
- GstTCPClientSrc
- GstTCPServerSrc
- GstVideoTestSrc
- GstGnomeVFSSrc
- GstAppSrc
- GstAudioTestSrc
- GstVorbisEnc
- GstVorbisDec
- GstVorbisParse
- GstVorbisTag
GstOggDemux
GstOggMux
GstOgmParse
GstOggParse
GstOggAviParse
GstBaseSink
- GstGnomeVFSSink
+ GstGioBaseSink
+ GstGioSink
+ GstGioStreamSink
GstBaseAudioSink
GstAudioSink
GstAlsaSink
+ GstGnomeVFSSink
GstVideoSink
GstXvImageSink
GstXImageSink
+ GstAppSink
GstTCPClientSink
GstMultiFdSink
GstTCPServerSink
- GstAppSink
+ GstBaseSrc
+ GstGioBaseSrc
+ GstGioSrc
+ GstGioStreamSrc
+ GstPushSrc
+ GstBaseAudioSrc
+ GstAudioSrc
+ GstAlsaSrc
+ GstCddaBaseSrc
+ GstCdParanoiaSrc
+ GstV4lElement
+ GstV4lSrc
+ GstTCPClientSrc
+ GstTCPServerSrc
+ GstVideoTestSrc
+ GstGnomeVFSSrc
+ GstAudioTestSrc
+ GstAppSrc
+ GstVorbisEnc
+ GstVorbisDec
+ GstVorbisParse
+ GstVorbisTag
+ GstTextOverlay
+ GstTimeOverlay
+ GstClockOverlay
+ GstTextRender
+ GstTheoraDec
+ GstTheoraEnc
+ GstTheoraParse
+ GstAlsaMixerElement
GstVisual
+ GstVisualjess
GstVisualbumpscope
GstVisualcorona
GstVisualinfinite
GstVisualjakdaw
- GstVisualjess
GstVisuallv_analyzer
GstVisuallv_scope
GstVisualoinksie
- GstTheoraDec
- GstTheoraEnc
- GstTheoraParse
- GstTextOverlay
- GstTimeOverlay
- GstClockOverlay
- GstTextRender
- GstAlsaMixerElement
- GstGDPDepay
- GstGDPPay
+ GstSubParse
+ GstSsaParse
+ GstAudioRate
GstBaseTransform
+ GstAudioConvert
GstFFMpegCsp
- GstVideoScale
GstAudioFilter
GstVolume
- GstSpeexResample
- GstAudioConvert
- GstSubParse
- GstSsaParse
+ GstAudioResample
+ GstVideoScale
GstAdder
+ GstGDPDepay
+ GstGDPPay
GstStreamSelector
GstQueue2
- GstAudioRate
GstVideoRate
GstBus
GstTask
GstRegistry
GstRingBuffer
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
PangoFcFontMap
PangoFT2FontMap
PangoContext
- GstMixerTrack
- GstMixerOptions
+ LinkConnection
+ GIOPConnection
+ LinkServer
+ GIOPServer
+ GConfClient
+ GstColorBalanceChannel
GstTunerNorm
GstTunerChannel
- GstColorBalanceChannel
+ GstMixerTrack
GstStreamInfo
GInterface
GTypePlugin
GstChildProxy
GstURIHandler
+ GFile
+ GDesktopAppInfoLookup
+ GDrive
+ GMount
+ GVolume
GstTagSetter
GstImplementsInterface
GstMixer
GstPropertyProbe
- GstTuner
+ GstNavigation
GstXOverlay
GstColorBalance
- GstNavigation
+ GstTuner
GstPlayBaseBin GstChildProxy
GstPlayBin GstChildProxy
GstPlayBin2 GstChildProxy
+GstDecodeBin GstChildProxy
GstDecodeBin2 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
GstCdParanoiaSrc GstURIHandler
-GstAlsaSrc GstImplementsInterface GstMixer GstPropertyProbe
-GstV4lElement GstImplementsInterface GstPropertyProbe GstTuner GstXOverlay GstColorBalance
-GstV4lSrc GstImplementsInterface GstPropertyProbe GstTuner GstXOverlay GstColorBalance
+GstV4lElement GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstTuner
+GstV4lSrc GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstTuner
GstGnomeVFSSrc GstURIHandler
GstAppSrc GstURIHandler
GstVorbisEnc GstTagSetter
GstVorbisTag GstTagSetter
-GstGnomeVFSSink GstURIHandler
-GstAlsaSink GstPropertyProbe
-GstXvImageSink GstImplementsInterface GstPropertyProbe GstXOverlay GstColorBalance GstNavigation
-GstXImageSink GstImplementsInterface GstXOverlay GstNavigation
GstAlsaMixerElement GstImplementsInterface GstMixer GstPropertyProbe
GstVolume GstImplementsInterface GstMixer
+GTypeModule GTypePlugin
+GIOModule GTypePlugin
+GAppLookupGConf GDesktopAppInfoLookup
+GProxyDrive GDrive
+GProxyMount GMount
+GProxyVolume GVolume
GstChildProxy GstObject
+GFile GObject
+GDesktopAppInfoLookup GObject
+GDrive GObject
+GMount GObject
+GVolume GObject
GstTagSetter GstObject GstElement
GstImplementsInterface GstObject GstElement
GstMixer GstObject GstImplementsInterface GstElement
-GstTuner GstObject GstImplementsInterface GstElement
GstXOverlay GstObject GstImplementsInterface GstElement
GstColorBalance GstObject GstImplementsInterface GstElement
+GstTuner GstObject GstImplementsInterface GstElement
<description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>ALSA plugin library</description>
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
<basename>libgstalsa.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Elements used to communicate with applications</description>
<filename>../../gst/app/.libs/libgstapp.so</filename>
<basename>libgstapp.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Convert audio to different formats</description>
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
<basename>libgstaudioconvert.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Adjusts audio frames</description>
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
<basename>libgstaudiorate.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<plugin>
<name>audioresample</name>
<description>Resamples audio</description>
- <filename>../../gst/speexresample/.libs/libgstaudioresample.so</filename>
+ <filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
<basename>libgstaudioresample.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Creates audio test signals of given frequency and volume</description>
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
<basename>libgstaudiotestsrc.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Read audio from CD in paranoid mode</description>
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
<basename>libgstcdparanoia.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>GPL</license>
<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>
<elements>
<element>
<description>decoder bin</description>
<filename>../../gst/playback/.libs/libgstdecodebin.so</filename>
<basename>libgstdecodebin.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>colorspace conversion copied from FFMpeg 0.4.9-pre1</description>
<filename>../../gst/ffmpegcolorspace/.libs/libgstffmpegcolorspace.so</filename>
<basename>libgstffmpegcolorspace.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>FFMpeg</package>
<description>Payload/depayload GDP packets</description>
<filename>../../gst/gdp/.libs/libgstgdp.so</filename>
<basename>libgstgdp.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>GIO elements</description>
<filename>../../ext/gio/.libs/libgstgio.so</filename>
<basename>libgstgio.so</basename>
- <version>0.10.21.1</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins CVS/prerelease</package>
<description>elements to read from and write to Gnome-VFS uri's</description>
<filename>../../ext/gnomevfs/.libs/libgstgnomevfs.so</filename>
<basename>libgstgnomevfs.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>libvisual visualization plugins</description>
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
<basename>libgstlibvisual.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
<filename>../../ext/ogg/.libs/libgstogg.so</filename>
<basename>libgstogg.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Pango-based text rendering and overlay</description>
<filename>../../ext/pango/.libs/libgstpango.so</filename>
<basename>libgstpango.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>various playback elements</description>
<filename>../../gst/playback/.libs/libgstplaybin.so</filename>
<basename>libgstplaybin.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Queue newer version</description>
<filename>../../gst/playback/.libs/libgstqueue2.so</filename>
<basename>libgstqueue2.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Subtitle parsing</description>
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
<basename>libgstsubparse.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>transfer data over the network via TCP</description>
<filename>../../gst/tcp/.libs/libgsttcp.so</filename>
<basename>libgsttcp.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Theora plugin library</description>
<filename>../../ext/theora/.libs/libgsttheora.so</filename>
<basename>libgsttheora.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>default typefind functions</description>
<filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
<basename>libgsttypefindfunctions.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<description>URI Decoder bin</description>
<filename>../../gst/playback/.libs/libgstdecodebin2.so</filename>
<basename>libgstdecodebin2.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>elements for Video 4 Linux</description>
<filename>../../sys/v4l/.libs/libgstvideo4linux.so</filename>
<basename>libgstvideo4linux.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Adjusts video frames</description>
<filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
<basename>libgstvideorate.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Resizes video</description>
<filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
<basename>libgstvideoscale.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Creates a test video stream</description>
<filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
<basename>libgstvideotestsrc.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>plugin for controlling audio volume</description>
<filename>../../gst/volume/.libs/libgstvolume.so</filename>
<basename>libgstvolume.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>Vorbis plugin library</description>
<filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
<basename>libgstvorbis.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>X11 video output element based on standard Xlib calls</description>
<filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
<basename>libgstximagesink.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
<description>XFree86 video output plugin using Xv extension</description>
<filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
<basename>libgstxvimagesink.so</basename>
- <version>0.10.22</version>
+ <version>0.10.22.1</version>
<license>LGPL</license>
<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>
<elements>
<element>
plugin_LTLIBRARIES = libgstaudioresample.la
-resample_SOURCES = \
- functable.c \
- resample.c \
- resample_functable.c \
- resample_ref.c \
- resample_chunk.c \
- resample.h \
- buffer.c
+libgstaudioresample_la_SOURCES = \
+ gstaudioresample.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 \
gstaudioresample.h \
- functable.h \
- debug.h \
- buffer.h
+ resample.c \
+ resample_sse.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)
+++ /dev/null
-
-#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;
-}
+++ /dev/null
-
-#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
+++ /dev/null
-
-#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;
-}
+++ /dev/null
-
-#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
+++ /dev/null
-/* 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;
-}
+++ /dev/null
-/* 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__ */
-
/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
+ * 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
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
-/* Element-Checklist-Version: 5 */
/**
* SECTION:element-audioresample
*
- * Audioresample resamples raw audio buffers to different sample rates using
+ * audioresample resamples raw audio buffers to different sample rates using
* a configurable windowing function to enhance quality.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw-int, rate=8000 ! alsasink
- * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
+ * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* </refsect2>
- *
- * Last reviewed on 2006-03-02 (0.10.4)
+ */
+
+/* TODO:
+ * - Enable SSE/ARM optimizations and select at runtime
*/
#ifdef HAVE_CONFIG_H
#include <string.h>
#include <math.h>
-/*#define DEBUG_ENABLED */
#include "gstaudioresample.h"
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
-GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
-#define GST_CAT_DEFAULT audioresample_debug
+#define OIL_ENABLE_UNSTABLE_API
+#include <liboil/liboilprofile.h>
+#include <liboil/liboil.h>
-/* elementfactory information */
-static const GstElementDetails gst_audioresample_details =
-GST_ELEMENT_DETAILS ("Audio scaler",
- "Filter/Converter/Audio",
- "Resample audio",
- "David Schleef <ds@schleef.org>");
-
-#define DEFAULT_FILTERLEN 16
+GST_DEBUG_CATEGORY (audio_resample_debug);
+#define GST_CAT_DEFAULT audio_resample_debug
enum
{
PROP_0,
- PROP_FILTERLEN
+ PROP_QUALITY,
+ PROP_FILTER_LENGTH
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS ( \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
+ "audio/x-raw-float, " \
+ "rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
- "width = (int) 16, " \
- "depth = (int) 16, " \
- "signed = (boolean) true;" \
+ "width = (int) { 32, 64 }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"depth = (int) 32, " \
- "signed = (boolean) true;" \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
+ "signed = (boolean) true; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
- "width = (int) 32; " \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
+ "width = (int) 24, " \
+ "depth = (int) 24, " \
+ "signed = (boolean) true; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
- "width = (int) 64" \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (boolean) true; " \
+ "audio/x-raw-int, " \
+ "rate = (int) [ 1, MAX ], " \
+ "channels = (int) [ 1, MAX ], " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 8, " \
+ "depth = (int) 8, " \
+ "signed = (boolean) true" \
)
-static GstStaticPadTemplate gst_audioresample_sink_template =
+/* If TRUE integer arithmetic resampling is faster and will be used if appropiate */
+static gboolean gst_audio_resample_use_int = FALSE;
+
+static GstStaticPadTemplate gst_audio_resample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static GstStaticPadTemplate gst_audioresample_src_template =
+static GstStaticPadTemplate gst_audio_resample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-static void gst_audioresample_set_property (GObject * object,
+static void gst_audio_resample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_audioresample_get_property (GObject * object,
+static void gst_audio_resample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
-static gboolean audioresample_get_unit_size (GstBaseTransform * base,
+static gboolean gst_audio_resample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, guint * size);
-static GstCaps *audioresample_transform_caps (GstBaseTransform * base,
+static GstCaps *gst_audio_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps);
-static gboolean audioresample_transform_size (GstBaseTransform * trans,
+static void gst_audio_resample_fixate_caps (GstBaseTransform * base,
+ GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
+static gboolean gst_audio_resample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * incaps, guint insize,
GstCaps * outcaps, guint * outsize);
-static gboolean audioresample_set_caps (GstBaseTransform * base,
+static gboolean gst_audio_resample_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
-static GstFlowReturn audioresample_pushthrough (GstAudioresample *
- audioresample);
-static GstFlowReturn audioresample_transform (GstBaseTransform * base,
+static GstFlowReturn gst_audio_resample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
-static gboolean audioresample_start (GstBaseTransform * base);
-static gboolean audioresample_stop (GstBaseTransform * base);
-
-static gboolean audioresample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audioresample_query_type (GstPad * pad);
-
-#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
+static gboolean gst_audio_resample_event (GstBaseTransform * base,
+ GstEvent * event);
+static gboolean gst_audio_resample_start (GstBaseTransform * base);
+static gboolean gst_audio_resample_stop (GstBaseTransform * base);
+static gboolean gst_audio_resample_query (GstPad * pad, GstQuery * query);
+static const GstQueryType *gst_audio_resample_query_type (GstPad * pad);
-GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
+GST_BOILERPLATE (GstAudioResample, gst_audio_resample, GstBaseTransform,
+ GST_TYPE_BASE_TRANSFORM);
static void
-gst_audioresample_base_init (gpointer g_class)
+gst_audio_resample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_src_template));
+ gst_static_pad_template_get (&gst_audio_resample_src_template));
gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_audioresample_sink_template));
+ gst_static_pad_template_get (&gst_audio_resample_sink_template));
- gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
+ gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
+ "Filter/Converter/Audio", "Resamples audio",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
-gst_audioresample_class_init (GstAudioresampleClass * klass)
+gst_audio_resample_class_init (GstAudioResampleClass * klass)
{
- GObjectClass *gobject_class;
-
- gobject_class = (GObjectClass *) klass;
-
- gobject_class->set_property = gst_audioresample_set_property;
- gobject_class->get_property = gst_audioresample_get_property;
-
- g_object_class_install_property (gobject_class, PROP_FILTERLEN,
- g_param_spec_int ("filter-length", "filter length",
- "Length of the resample filter", 0, G_MAXINT, DEFAULT_FILTERLEN,
- G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+
+ gobject_class->set_property = gst_audio_resample_set_property;
+ gobject_class->get_property = gst_audio_resample_get_property;
+
+ g_object_class_install_property (gobject_class, PROP_QUALITY,
+ g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
+ "the lowest and 10 being the best",
+ SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX,
+ SPEEX_RESAMPLER_QUALITY_DEFAULT,
+ G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+
+ /* FIXME 0.11: Remove this property, it's just for compatibility
+ * with old audioresample
+ */
+ g_object_class_install_property (gobject_class, PROP_FILTER_LENGTH,
+ g_param_spec_int ("filter-length", "Filter length",
+ "DEPRECATED, DON'T USE THIS! " "Length of the resample filter", 0,
+ G_MAXINT, 64, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
GST_BASE_TRANSFORM_CLASS (klass)->start =
- GST_DEBUG_FUNCPTR (audioresample_start);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
- GST_DEBUG_FUNCPTR (audioresample_stop);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
- GST_DEBUG_FUNCPTR (audioresample_transform_size);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
- GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
- GST_DEBUG_FUNCPTR (audioresample_transform_caps);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_transform_caps);
+ GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
+ GST_DEBUG_FUNCPTR (gst_audio_resample_fixate_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
- GST_DEBUG_FUNCPTR (audioresample_set_caps);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
- GST_DEBUG_FUNCPTR (audioresample_transform);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_transform);
GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (audioresample_event);
+ GST_DEBUG_FUNCPTR (gst_audio_resample_event);
GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
}
static void
-gst_audioresample_init (GstAudioresample * audioresample,
- GstAudioresampleClass * klass)
+gst_audio_resample_init (GstAudioResample * resample,
+ GstAudioResampleClass * klass)
{
- GstBaseTransform *trans;
-
- trans = GST_BASE_TRANSFORM (audioresample);
+ GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
- /* buffer alloc passthrough is too impossible. FIXME, it
- * is trivial in the passthrough case. */
- gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
+ resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
- audioresample->filter_length = DEFAULT_FILTERLEN;
+ resample->need_discont = FALSE;
- audioresample->need_discont = FALSE;
-
- gst_pad_set_query_function (trans->srcpad, audioresample_query);
- gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);
+ gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query);
+ gst_pad_set_query_type_function (trans->srcpad,
+ gst_audio_resample_query_type);
}
/* vmethods */
static gboolean
-audioresample_start (GstBaseTransform * base)
+gst_audio_resample_start (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
-
- audioresample->resample = resample_new ();
- audioresample->ts_offset = -1;
- audioresample->offset = -1;
- audioresample->next_ts = -1;
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
+ resample->next_offset = -1;
+ resample->next_ts = -1;
+ resample->next_upstream_ts = -1;
return TRUE;
}
static gboolean
-audioresample_stop (GstBaseTransform * base)
+gst_audio_resample_stop (GstBaseTransform * base)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- if (audioresample->resample) {
- resample_free (audioresample->resample);
- audioresample->resample = NULL;
+ if (resample->state) {
+ resample->funcs->destroy (resample->state);
+ resample->state = NULL;
}
- gst_caps_replace (&audioresample->sinkcaps, NULL);
- gst_caps_replace (&audioresample->srccaps, NULL);
+ resample->funcs = NULL;
+
+ g_free (resample->tmp_in);
+ resample->tmp_in = NULL;
+ resample->tmp_in_size = 0;
+
+ g_free (resample->tmp_out);
+ resample->tmp_out = NULL;
+ resample->tmp_out_size = 0;
+
+ gst_caps_replace (&resample->sinkcaps, NULL);
+ gst_caps_replace (&resample->srccaps, NULL);
return TRUE;
}
static gboolean
-audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
+gst_audio_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size)
{
gint width, channels;
GstStructure *structure;
gboolean ret;
- g_assert (size);
+ g_return_val_if_fail (size != NULL, FALSE);
/* this works for both float and int */
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "channels", &channels);
- g_return_val_if_fail (ret, FALSE);
- *size = width * channels / 8;
+ if (G_UNLIKELY (!ret))
+ return FALSE;
+
+ *size = (width / 8) * channels;
return TRUE;
}
static GstCaps *
-audioresample_transform_caps (GstBaseTransform * base,
+gst_audio_resample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *res;
return res;
}
+/* Fixate rate to the allowed rate that has the smallest difference */
+static void
+gst_audio_resample_fixate_caps (GstBaseTransform * base,
+ GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
+{
+ GstStructure *s;
+ gint rate;
+
+ s = gst_caps_get_structure (caps, 0);
+ if (G_UNLIKELY (!gst_structure_get_int (s, "rate", &rate)))
+ return;
+
+ s = gst_caps_get_structure (othercaps, 0);
+ gst_structure_fixate_field_nearest_int (s, "rate", rate);
+}
+
+static const SpeexResampleFuncs *
+gst_audio_resample_get_funcs (gint width, gboolean fp)
+{
+ const SpeexResampleFuncs *funcs = NULL;
+
+ if (gst_audio_resample_use_int && (width == 8 || width == 16) && !fp)
+ funcs = &int_funcs;
+ else if ((!gst_audio_resample_use_int && (width == 8 || width == 16) && !fp)
+ || (width == 32 && fp))
+ funcs = &float_funcs;
+ else if ((width == 64 && fp) || ((width == 32 || width == 24) && !fp))
+ funcs = &double_funcs;
+ else
+ g_assert_not_reached ();
+
+ return funcs;
+}
+
+static SpeexResamplerState *
+gst_audio_resample_init_state (GstAudioResample * resample, gint width,
+ gint channels, gint inrate, gint outrate, gint quality, gboolean fp)
+{
+ SpeexResamplerState *ret = NULL;
+ gint err = RESAMPLER_ERR_SUCCESS;
+ const SpeexResampleFuncs *funcs = gst_audio_resample_get_funcs (width, fp);
+
+ ret = funcs->init (channels, inrate, outrate, quality, &err);
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
+ GST_ERROR_OBJECT (resample, "Failed to create resampler state: %s",
+ funcs->strerror (err));
+ return NULL;
+ }
+
+ funcs->skip_zeros (ret);
+
+ return ret;
+}
+
static gboolean
-resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
- GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
+gst_audio_resample_update_state (GstAudioResample * resample, gint width,
+ gint channels, gint inrate, gint outrate, gint quality, gboolean fp)
+{
+ gboolean ret = TRUE;
+ gboolean updated_latency = FALSE;
+
+ updated_latency = (resample->inrate != inrate
+ || quality != resample->quality) && resample->state != NULL;
+
+ if (resample->state == NULL) {
+ ret = TRUE;
+ } else if (resample->channels != channels || fp != resample->fp
+ || width != resample->width) {
+ resample->funcs->destroy (resample->state);
+ resample->state =
+ gst_audio_resample_init_state (resample, width, channels, inrate,
+ outrate, quality, fp);
+
+ resample->funcs = gst_audio_resample_get_funcs (width, fp);
+ ret = (resample->state != NULL);
+ } else if (resample->inrate != inrate || resample->outrate != outrate) {
+ gint err = RESAMPLER_ERR_SUCCESS;
+
+ err = resample->funcs->set_rate (resample->state, inrate, outrate);
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
+ GST_ERROR_OBJECT (resample, "Failed to update rate: %s",
+ resample->funcs->strerror (err));
+
+ ret = (err == RESAMPLER_ERR_SUCCESS);
+ } else if (quality != resample->quality) {
+ gint err = RESAMPLER_ERR_SUCCESS;
+
+ err = resample->funcs->set_quality (resample->state, quality);
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
+ GST_ERROR_OBJECT (resample, "Failed to update quality: %s",
+ resample->funcs->strerror (err));
+
+ ret = (err == RESAMPLER_ERR_SUCCESS);
+ }
+
+ resample->width = width;
+ resample->channels = channels;
+ resample->fp = fp;
+ resample->quality = quality;
+ resample->inrate = inrate;
+ resample->outrate = outrate;
+
+ if (updated_latency)
+ gst_element_post_message (GST_ELEMENT (resample),
+ gst_message_new_latency (GST_OBJECT (resample)));
+
+ return ret;
+}
+
+static void
+gst_audio_resample_reset_state (GstAudioResample * resample)
+{
+ if (resample->state)
+ resample->funcs->reset_mem (resample->state);
+}
+
+static gboolean
+gst_audio_resample_parse_caps (GstCaps * incaps,
+ GstCaps * outcaps, gint * width, gint * channels, gint * inrate,
+ gint * outrate, gboolean * fp)
{
GstStructure *structure;
gboolean ret;
- gint myinrate, myoutrate;
- int mychannels;
- gint width, depth;
- ResampleFormat format;
+ gint mywidth, myinrate, myoutrate, mychannels;
+ gboolean myfp;
GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
structure = gst_caps_get_structure (incaps, 0);
- /* get width */
- ret = gst_structure_get_int (structure, "width", &width);
- if (!ret)
- goto no_width;
-
- /* figure out the format */
- if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
- if (width == 32)
- format = RESAMPLE_FORMAT_F32;
- else if (width == 64)
- format = RESAMPLE_FORMAT_F64;
- else
- goto wrong_depth;
- } else {
- /* for int, depth and width must be the same */
- ret = gst_structure_get_int (structure, "depth", &depth);
- if (!ret || width != depth)
- goto not_equal;
-
- if (width == 16)
- format = RESAMPLE_FORMAT_S16;
- else if (width == 32)
- format = RESAMPLE_FORMAT_S32;
- else
- goto wrong_depth;
- }
+ if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float"))
+ myfp = TRUE;
+ else
+ myfp = FALSE;
+
ret = gst_structure_get_int (structure, "rate", &myinrate);
ret &= gst_structure_get_int (structure, "channels", &mychannels);
- if (!ret)
+ ret &= gst_structure_get_int (structure, "width", &mywidth);
+ if (G_UNLIKELY (!ret))
goto no_in_rate_channels;
structure = gst_caps_get_structure (outcaps, 0);
ret = gst_structure_get_int (structure, "rate", &myoutrate);
- if (!ret)
+ if (G_UNLIKELY (!ret))
goto no_out_rate;
if (channels)
*inrate = myinrate;
if (outrate)
*outrate = myoutrate;
-
- resample_set_format (state, format);
- resample_set_n_channels (state, mychannels);
- resample_set_input_rate (state, myinrate);
- resample_set_output_rate (state, myoutrate);
+ if (width)
+ *width = mywidth;
+ if (fp)
+ *fp = myfp;
return TRUE;
/* ERRORS */
-no_width:
- {
- GST_DEBUG ("failed to get width from caps");
- return FALSE;
- }
-not_equal:
- {
- GST_DEBUG ("width %d and depth %d must be the same", width, depth);
- return FALSE;
- }
-wrong_depth:
- {
- GST_DEBUG ("unknown depth %d found", depth);
- return FALSE;
- }
no_in_rate_channels:
{
GST_DEBUG ("could not get input rate and channels");
}
}
+static gint
+_gcd (gint a, gint b)
+{
+ while (b != 0) {
+ int temp = a;
+
+ a = b;
+ b = temp % b;
+ }
+
+ return ABS (a);
+}
+
static gboolean
-audioresample_transform_size (GstBaseTransform * base,
+gst_audio_resample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
- ResampleState *state;
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
GstCaps *srccaps, *sinkcaps;
- gboolean use_internal = FALSE; /* whether we use the internal state */
gboolean ret = TRUE;
+ guint32 ratio_den, ratio_num;
+ gint inrate, outrate, gcd;
+ gint width;
- GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
+ GST_LOG_OBJECT (resample, "asked to transform size %d in direction %s",
size, direction == GST_PAD_SINK ? "SINK" : "SRC");
if (direction == GST_PAD_SINK) {
sinkcaps = caps;
srccaps = caps;
}
- /* if the caps are the ones that _set_caps got called with; we can use
- * our own state; otherwise we'll have to create a state */
- if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&
- gst_caps_is_equal (srccaps, audioresample->srccaps)) {
- use_internal = TRUE;
- state = audioresample->resample;
- } else {
- GST_DEBUG_OBJECT (audioresample,
- "caps are not the set caps, creating state");
- state = resample_new ();
- resample_set_filter_length (state, audioresample->filter_length);
- resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
+ ret =
+ gst_audio_resample_parse_caps (caps, othercaps, &width, NULL, &inrate,
+ &outrate, NULL);
+ if (G_UNLIKELY (!ret)) {
+ GST_ERROR_OBJECT (resample, "Wrong caps");
+ return FALSE;
}
+ gcd = _gcd (inrate, outrate);
+ ratio_num = inrate / gcd;
+ ratio_den = outrate / gcd;
+
if (direction == GST_PAD_SINK) {
+ gint fac = width / 8;
+
/* asked to convert size of an incoming buffer */
- *othersize = resample_get_output_size_for_input (state, size);
+ size /= fac;
+ *othersize = (size * ratio_den + ratio_num - 1) / ratio_num;
+ *othersize *= fac;
+ size *= fac;
} else {
+ gint fac = width / 8;
+
/* asked to convert size of an outgoing buffer */
- *othersize = resample_get_input_size_for_output (state, size);
+ size /= fac;
+ *othersize = (size * ratio_num + ratio_den - 1) / ratio_den;
+ *othersize *= fac;
+ size *= fac;
}
- g_assert (*othersize % state->sample_size == 0);
- /* we make room for one extra sample, given that the resampling filter
- * can output an extra one for non-integral i_rate/o_rate */
- GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
-
- if (!use_internal) {
- resample_free (state);
- }
+ GST_LOG_OBJECT (resample, "transformed size %d to %d", size, *othersize);
return ret;
}
static gboolean
-audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
+gst_audio_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
- gint inrate, outrate;
- int channels;
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
+ gint width = 0, inrate = 0, outrate = 0, channels = 0;
+ gboolean fp;
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
- GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
+ GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
- ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,
- &channels, &inrate, &outrate);
+ ret = gst_audio_resample_parse_caps (incaps, outcaps,
+ &width, &channels, &inrate, &outrate, &fp);
- g_return_val_if_fail (ret, FALSE);
+ if (G_UNLIKELY (!ret))
+ return FALSE;
- audioresample->channels = channels;
- GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);
- audioresample->i_rate = inrate;
- GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);
- audioresample->o_rate = outrate;
- GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);
+ ret =
+ gst_audio_resample_update_state (resample, width, channels, inrate,
+ outrate, resample->quality, fp);
+
+ if (G_UNLIKELY (!ret))
+ return FALSE;
/* save caps so we can short-circuit in the size_transform if the caps
* are the same */
- gst_caps_replace (&audioresample->sinkcaps, incaps);
- gst_caps_replace (&audioresample->srccaps, outcaps);
+ gst_caps_replace (&resample->sinkcaps, incaps);
+ gst_caps_replace (&resample->srccaps, outcaps);
return TRUE;
}
-static gboolean
-audioresample_event (GstBaseTransform * base, GstEvent * event)
+#define GST_MAXINT24 (8388607)
+#define GST_MININT24 (-8388608)
+
+#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+#define GST_READ_UINT24 GST_READ_UINT24_LE
+#define GST_WRITE_UINT24 GST_WRITE_UINT24_LE
+#else
+#define GST_READ_UINT24 GST_READ_UINT24_BE
+#define GST_WRITE_UINT24 GST_WRITE_UINT24_BE
+#endif
+
+static void
+gst_audio_resample_convert_buffer (GstAudioResample * resample,
+ const guint8 * in, guint8 * out, guint len, gboolean inverse)
+{
+ len *= resample->channels;
+
+ if (inverse) {
+ if (gst_audio_resample_use_int && resample->width == 8 && !resample->fp) {
+ gint8 *o = (gint8 *) out;
+ gint16 *i = (gint16 *) in;
+ gint32 tmp;
+
+ while (len) {
+ tmp = *i + (G_MAXINT8 >> 1);
+ *o = CLAMP (tmp >> 8, G_MININT8, G_MAXINT8);
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 8
+ && !resample->fp) {
+ gint8 *o = (gint8 *) out;
+ gfloat *i = (gfloat *) in;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = (gint8) CLAMP (tmp * G_MAXINT8 + 0.5, G_MININT8, G_MAXINT8);
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 16
+ && !resample->fp) {
+ gint16 *o = (gint16 *) out;
+ gfloat *i = (gfloat *) in;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = (gint16) CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
+ o++;
+ i++;
+ len--;
+ }
+ } else if (resample->width == 24 && !resample->fp) {
+ guint8 *o = (guint8 *) out;
+ gdouble *i = (gdouble *) in;
+ gdouble tmp;
+
+ while (len) {
+ tmp = *i;
+ GST_WRITE_UINT24 (o, (gint32) CLAMP (tmp * GST_MAXINT24 + 0.5,
+ GST_MININT24, GST_MAXINT24));
+ o += 3;
+ i++;
+ len--;
+ }
+ } else if (resample->width == 32 && !resample->fp) {
+ gint32 *o = (gint32 *) out;
+ gdouble *i = (gdouble *) in;
+ gdouble tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = (gint32) CLAMP (tmp * G_MAXINT32 + 0.5, G_MININT32, G_MAXINT32);
+ o++;
+ i++;
+ len--;
+ }
+ } else {
+ g_assert_not_reached ();
+ }
+ } else {
+ if (gst_audio_resample_use_int && resample->width == 8 && !resample->fp) {
+ gint8 *i = (gint8 *) in;
+ gint16 *o = (gint16 *) out;
+ gint32 tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp << 8;
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 8
+ && !resample->fp) {
+ gint8 *i = (gint8 *) in;
+ gfloat *o = (gfloat *) out;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp / G_MAXINT8;
+ o++;
+ i++;
+ len--;
+ }
+ } else if (!gst_audio_resample_use_int && resample->width == 16
+ && !resample->fp) {
+ gint16 *i = (gint16 *) in;
+ gfloat *o = (gfloat *) out;
+ gfloat tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp / G_MAXINT16;
+ o++;
+ i++;
+ len--;
+ }
+ } else if (resample->width == 24 && !resample->fp) {
+ guint8 *i = (guint8 *) in;
+ gdouble *o = (gdouble *) out;
+ gdouble tmp;
+ guint32 tmp2;
+
+ while (len) {
+ tmp2 = GST_READ_UINT24 (i);
+ if (tmp2 & 0x00800000)
+ tmp2 |= 0xff000000;
+ tmp = (gint32) tmp2;
+ *o = tmp / GST_MAXINT24;
+ o++;
+ i += 3;
+ len--;
+ }
+ } else if (resample->width == 32 && !resample->fp) {
+ gint32 *i = (gint32 *) in;
+ gdouble *o = (gdouble *) out;
+ gdouble tmp;
+
+ while (len) {
+ tmp = *i;
+ *o = tmp / G_MAXINT32;
+ o++;
+ i++;
+ len--;
+ }
+ } else {
+ g_assert_not_reached ();
+ }
+ }
+}
+
+static void
+gst_audio_resample_push_drain (GstAudioResample * resample)
{
- GstAudioresample *audioresample;
+ GstBuffer *buf;
+ GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
+ GstFlowReturn res;
+ gint outsize;
+ guint out_len, out_processed;
+ gint err;
+ guint num, den, len;
+ guint8 *outtmp = NULL;
+ gboolean need_convert = FALSE;
+
+ if (!resample->state)
+ return;
+
+ need_convert = (resample->funcs->width != resample->width);
+
+ resample->funcs->get_ratio (resample->state, &num, &den);
+
+ out_len = resample->funcs->get_input_latency (resample->state);
+ out_len = out_processed = (out_len * den + num - 1) / num;
+ outsize = (resample->width / 8) * out_len * resample->channels;
+
+ if (need_convert) {
+ guint outsize_tmp =
+ (resample->funcs->width / 8) * out_len * resample->channels;
+ if (outsize_tmp <= resample->tmp_out_size) {
+ outtmp = resample->tmp_out;
+ } else {
+ resample->tmp_out_size = outsize_tmp;
+ resample->tmp_out = outtmp = g_realloc (resample->tmp_out, outsize_tmp);
+ }
+ }
+
+ res =
+ gst_pad_alloc_buffer_and_set_caps (trans->srcpad, GST_BUFFER_OFFSET_NONE,
+ outsize, GST_PAD_CAPS (trans->srcpad), &buf);
+
+ if (G_UNLIKELY (res != GST_FLOW_OK)) {
+ GST_WARNING_OBJECT (resample, "failed allocating buffer of %d bytes",
+ outsize);
+ return;
+ }
- audioresample = GST_AUDIORESAMPLE (base);
+ len = resample->funcs->get_input_latency (resample->state);
+
+ err =
+ resample->funcs->process (resample->state,
+ NULL, &len, (need_convert) ? outtmp : GST_BUFFER_DATA (buf),
+ &out_processed);
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
+ GST_WARNING_OBJECT (resample, "Failed to process drain: %s",
+ resample->funcs->strerror (err));
+ gst_buffer_unref (buf);
+ return;
+ }
+
+ if (G_UNLIKELY (out_processed == 0)) {
+ GST_WARNING_OBJECT (resample, "Failed to get drain, dropping buffer");
+ gst_buffer_unref (buf);
+ return;
+ }
+
+ /* If we wrote more than allocated something is really wrong now
+ * and we should better abort immediately */
+ g_assert (out_len >= out_processed);
+
+ if (need_convert)
+ gst_audio_resample_convert_buffer (resample, outtmp, GST_BUFFER_DATA (buf),
+ out_processed, TRUE);
+
+ GST_BUFFER_DURATION (buf) =
+ GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
+ GST_BUFFER_SIZE (buf) =
+ out_processed * resample->channels * (resample->width / 8);
+
+ if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
+ GST_BUFFER_OFFSET (buf) = resample->next_offset;
+ GST_BUFFER_OFFSET_END (buf) = resample->next_offset + out_processed;
+ GST_BUFFER_TIMESTAMP (buf) = resample->next_ts;
+
+ resample->next_ts += GST_BUFFER_DURATION (buf);
+ resample->next_offset += out_processed;
+ }
+
+ GST_LOG_OBJECT (resample,
+ "Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT
+ " duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
+ G_GUINT64_FORMAT, GST_BUFFER_SIZE (buf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_OFFSET (buf),
+ GST_BUFFER_OFFSET_END (buf));
+
+ res = gst_pad_push (trans->srcpad, buf);
+
+ if (G_UNLIKELY (res != GST_FLOW_OK))
+ GST_WARNING_OBJECT (resample, "Failed to push drain: %s",
+ gst_flow_get_name (res));
+
+ return;
+}
+
+static gboolean
+gst_audio_resample_event (GstBaseTransform * base, GstEvent * event)
+{
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_FLUSH_STOP:
- resample_input_flush (audioresample->resample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
- break;
+ gst_audio_resample_reset_state (resample);
+ resample->next_offset = -1;
+ resample->next_ts = -1;
+ resample->next_upstream_ts = -1;
case GST_EVENT_NEWSEGMENT:
- resample_input_pushthrough (audioresample->resample);
- audioresample_pushthrough (audioresample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
+ gst_audio_resample_push_drain (resample);
+ gst_audio_resample_reset_state (resample);
+ resample->next_offset = -1;
+ resample->next_ts = -1;
+ resample->next_upstream_ts = -1;
break;
- case GST_EVENT_EOS:
- resample_input_eos (audioresample->resample);
- audioresample_pushthrough (audioresample);
+ case GST_EVENT_EOS:{
+ gst_audio_resample_push_drain (resample);
+ gst_audio_resample_reset_state (resample);
break;
+ }
default:
break;
}
- parent_class->event (base, event);
-
- return TRUE;
-}
-
-static GstFlowReturn
-audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
-{
- int outsize;
- int outsamples;
- ResampleState *r;
-
- r = audioresample->resample;
-
- outsize = resample_get_output_size (r);
- GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
-
- /* protect against mem corruption */
- if (outsize > GST_BUFFER_SIZE (outbuf)) {
- GST_WARNING_OBJECT (audioresample,
- "overriding audioresample's outsize %d with outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- outsize = GST_BUFFER_SIZE (outbuf);
- }
- /* catch possibly wrong size differences */
- if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's outsize %d too far from outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- }
-
- outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
- outsamples = outsize / r->sample_size;
- GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
- outsize, outsamples);
-
- GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
- GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
-
- if (audioresample->ts_offset != -1) {
- audioresample->offset += outsamples;
- audioresample->ts_offset += outsamples;
- audioresample->next_ts =
- gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
- audioresample->o_rate);
- GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
-
- /* we calculate DURATION as the difference between "next" timestamp
- * and current timestamp so we ensure a contiguous stream, instead of
- * having rounding errors. */
- GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
- GST_BUFFER_TIMESTAMP (outbuf);
- } else {
- /* no valid offset know, we can still sortof calculate the duration though */
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale_int (outsamples, GST_SECOND,
- audioresample->o_rate);
- }
-
- /* check for possible mem corruption */
- if (outsize > GST_BUFFER_SIZE (outbuf)) {
- /* this is an error that when it happens, would need fixing in the
- * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
- * and it gave us more ! */
- GST_WARNING_OBJECT (audioresample,
- "audioresample, you memory corrupting bastard. "
- "you gave me outsize %d while my buffer was size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- return GST_FLOW_ERROR;
- }
- /* catch possibly wrong size differences */
- if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's written outsize %d too far from outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- }
- GST_BUFFER_SIZE (outbuf) = outsize;
-
- if (G_UNLIKELY (audioresample->need_discont)) {
- GST_DEBUG_OBJECT (audioresample,
- "marking this buffer with the DISCONT flag");
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- audioresample->need_discont = FALSE;
- }
-
- GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
- G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
- outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
- GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
-
- return GST_FLOW_OK;
+ return parent_class->event (base, event);
}
static gboolean
-audioresample_check_discont (GstAudioresample * audioresample,
+gst_audio_resample_check_discont (GstAudioResample * resample,
GstClockTime timestamp)
{
if (timestamp != GST_CLOCK_TIME_NONE &&
- audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
- audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
- timestamp != audioresample->prev_ts + audioresample->prev_duration) {
+ resample->next_upstream_ts != GST_CLOCK_TIME_NONE &&
+ timestamp != resample->next_upstream_ts) {
/* Potentially a discontinuous buffer. However, it turns out that many
* elements generate imperfect streams due to rounding errors, so we permit
* a small error (up to one sample) without triggering a filter
* flush/restart (if triggered incorrectly, this will be audible) */
- GstClockTimeDiff diff = timestamp -
- (audioresample->prev_ts + audioresample->prev_duration);
+ GstClockTimeDiff diff = timestamp - resample->next_upstream_ts;
- if (ABS (diff) > GST_SECOND / audioresample->i_rate) {
- GST_WARNING_OBJECT (audioresample,
- "encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
+ if (ABS (diff) > (GST_SECOND + resample->inrate - 1) / resample->inrate) {
+ GST_WARNING_OBJECT (resample,
+ "encountered timestamp discontinuity of %s%" GST_TIME_FORMAT,
+ (diff < 0) ? "-" : "", GST_TIME_ARGS ((GstClockTime) ABS (diff)));
return TRUE;
}
}
}
static GstFlowReturn
-audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+gst_audio_resample_process (GstAudioResample * resample, GstBuffer * inbuf,
GstBuffer * outbuf)
{
- GstAudioresample *audioresample;
- ResampleState *r;
- guchar *data, *datacopy;
+ guint32 in_len, in_processed;
+ guint32 out_len, out_processed;
+ gint err = RESAMPLER_ERR_SUCCESS;
+ guint8 *in_tmp = NULL, *out_tmp = NULL;
+ gboolean need_convert = (resample->funcs->width != resample->width);
+
+ in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
+ out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
+
+ in_len /= (resample->width / 8);
+ out_len /= (resample->width / 8);
+
+ in_processed = in_len;
+ out_processed = out_len;
+
+ if (need_convert) {
+ guint in_size_tmp =
+ in_len * resample->channels * (resample->funcs->width / 8);
+ guint out_size_tmp =
+ out_len * resample->channels * (resample->funcs->width / 8);
+
+ if (in_size_tmp <= resample->tmp_in_size) {
+ in_tmp = resample->tmp_in;
+ } else {
+ resample->tmp_in = in_tmp = g_realloc (resample->tmp_in, in_size_tmp);
+ resample->tmp_in_size = in_size_tmp;
+ }
+
+ gst_audio_resample_convert_buffer (resample, GST_BUFFER_DATA (inbuf),
+ in_tmp, in_len, FALSE);
+
+ if (out_size_tmp <= resample->tmp_out_size) {
+ out_tmp = resample->tmp_out;
+ } else {
+ resample->tmp_out = out_tmp = g_realloc (resample->tmp_out, out_size_tmp);
+ resample->tmp_out_size = out_size_tmp;
+ }
+ }
+
+ if (need_convert) {
+ err = resample->funcs->process (resample->state,
+ in_tmp, &in_processed, out_tmp, &out_processed);
+ } else {
+ err = resample->funcs->process (resample->state,
+ (const guint8 *) GST_BUFFER_DATA (inbuf), &in_processed,
+ (guint8 *) GST_BUFFER_DATA (outbuf), &out_processed);
+ }
+
+ if (G_UNLIKELY (in_len != in_processed))
+ GST_WARNING_OBJECT (resample, "Converted %d of %d input samples",
+ in_processed, in_len);
+
+ if (out_len != out_processed) {
+ if (out_processed == 0) {
+ GST_DEBUG_OBJECT (resample, "Converted to 0 samples, buffer dropped");
+
+ return GST_BASE_TRANSFORM_FLOW_DROPPED;
+ }
+
+ /* If we wrote more than allocated something is really wrong now
+ * and we should better abort immediately */
+ g_assert (out_len >= out_processed);
+ }
+
+ if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
+ GST_ERROR_OBJECT (resample, "Failed to convert data: %s",
+ resample->funcs->strerror (err));
+ return GST_FLOW_ERROR;
+ } else {
+
+ if (need_convert)
+ gst_audio_resample_convert_buffer (resample, out_tmp,
+ GST_BUFFER_DATA (outbuf), out_processed, TRUE);
+
+ GST_BUFFER_DURATION (outbuf) =
+ GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
+ GST_BUFFER_SIZE (outbuf) =
+ out_processed * resample->channels * (resample->width / 8);
+
+ if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
+ GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
+ GST_BUFFER_OFFSET (outbuf) = resample->next_offset;
+ GST_BUFFER_OFFSET_END (outbuf) = resample->next_offset + out_processed;
+
+ resample->next_ts += GST_BUFFER_DURATION (outbuf);
+ resample->next_offset += out_processed;
+ }
+
+ GST_LOG_OBJECT (resample,
+ "Converted to buffer of %u bytes with timestamp %" GST_TIME_FORMAT
+ ", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
+ ", offset_end %" G_GUINT64_FORMAT, GST_BUFFER_SIZE (outbuf),
+ GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+ GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
+
+ return GST_FLOW_OK;
+ }
+}
+
+static GstFlowReturn
+gst_audio_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
+ GstBuffer * outbuf)
+{
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (base);
+ guint8 *data;
gulong size;
GstClockTime timestamp;
-
- audioresample = GST_AUDIORESAMPLE (base);
- r = audioresample->resample;
+ guint outsamples, insamples;
+ GstFlowReturn ret;
+
+ if (resample->state == NULL) {
+ if (G_UNLIKELY (!(resample->state =
+ gst_audio_resample_init_state (resample, resample->width,
+ resample->channels, resample->inrate, resample->outrate,
+ resample->quality, resample->fp))))
+ return GST_FLOW_ERROR;
+
+ resample->funcs =
+ gst_audio_resample_get_funcs (resample->width, resample->fp);
+ }
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
- GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
+ GST_LOG_OBJECT (resample, "transforming buffer of %ld bytes, ts %"
GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
size, GST_TIME_ARGS (timestamp),
GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
/* check for timestamp discontinuities and flush/reset if needed */
- if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
+ if (G_UNLIKELY (gst_audio_resample_check_discont (resample, timestamp)
+ || GST_BUFFER_IS_DISCONT (inbuf))) {
/* Flush internal samples */
- audioresample_pushthrough (audioresample);
+ gst_audio_resample_reset_state (resample);
/* Inform downstream element about discontinuity */
- audioresample->need_discont = TRUE;
- /* We want to recalculate the offset */
- audioresample->ts_offset = -1;
- }
-
- if (audioresample->ts_offset == -1) {
- /* if we don't know the initial offset yet, calculate it based on the
- * input timestamp. */
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- GstClockTime stime;
-
- /* offset used to calculate the timestamps. We use the sample offset for
- * this to make it more accurate. We want the first buffer to have the
- * same timestamp as the incoming timestamp. */
- audioresample->next_ts = timestamp;
- audioresample->ts_offset =
- gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
- /* offset used to set as the buffer offset, this offset is always
- * relative to the stream time, note that timestamp is not... */
- stime = (timestamp - base->segment.start) + base->segment.time;
- audioresample->offset =
- gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
- }
+ resample->need_discont = TRUE;
+ /* We want to recalculate the timestamps */
+ resample->next_ts = -1;
+ resample->next_upstream_ts = -1;
+ resample->next_offset = -1;
}
- audioresample->prev_ts = timestamp;
- audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
- /* need to memdup, resample takes ownership. */
- datacopy = g_memdup (data, size);
- resample_add_input_data (r, datacopy, size, g_free, datacopy);
-
- return audioresample_do_output (audioresample, outbuf);
-}
-
-/* push remaining data in the buffers out */
-static GstFlowReturn
-audioresample_pushthrough (GstAudioresample * audioresample)
-{
- int outsize;
- ResampleState *r;
- GstBuffer *outbuf;
- GstFlowReturn res = GST_FLOW_OK;
- GstBaseTransform *trans;
+ insamples = GST_BUFFER_SIZE (inbuf) / resample->channels;
+ insamples /= (resample->width / 8);
- r = audioresample->resample;
+ outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels;
+ outsamples /= (resample->width / 8);
- outsize = resample_get_output_size (r);
- if (outsize == 0) {
- GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
- goto done;
+ if (GST_CLOCK_TIME_IS_VALID (timestamp)
+ && !GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
+ resample->next_ts = timestamp;
+ resample->next_offset =
+ GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
}
- trans = GST_BASE_TRANSFORM (audioresample);
-
- res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
- GST_PAD_CAPS (trans->srcpad), &outbuf);
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
- outsize);
- goto done;
+ if (G_UNLIKELY (resample->need_discont)) {
+ GST_DEBUG_OBJECT (resample, "marking this buffer with the DISCONT flag");
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ resample->need_discont = FALSE;
}
- res = audioresample_do_output (audioresample, outbuf);
- if (G_UNLIKELY (res != GST_FLOW_OK))
- goto done;
+ ret = gst_audio_resample_process (resample, inbuf, outbuf);
+ if (G_UNLIKELY (ret != GST_FLOW_OK))
+ return ret;
- res = gst_pad_push (trans->srcpad, outbuf);
+ if (GST_CLOCK_TIME_IS_VALID (timestamp)
+ && !GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
+ resample->next_upstream_ts = timestamp;
-done:
- return res;
+ if (GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
+ resample->next_upstream_ts +=
+ GST_FRAMES_TO_CLOCK_TIME (insamples, resample->inrate);
+
+ return GST_FLOW_OK;
}
static gboolean
-audioresample_query (GstPad * pad, GstQuery * query)
+gst_audio_resample_query (GstPad * pad, GstQuery * query)
{
- GstAudioresample *audioresample =
- GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
- GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample);
+ GstAudioResample *resample = GST_AUDIO_RESAMPLE (gst_pad_get_parent (pad));
+ GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
gboolean live;
guint64 latency;
GstPad *peer;
- gint rate = audioresample->i_rate;
- gint resampler_latency = audioresample->filter_length / 2;
+ gint rate = resample->inrate;
+ gint resampler_latency;
+
+ if (resample->state)
+ resampler_latency =
+ resample->funcs->get_input_latency (resample->state);
+ else
+ resampler_latency = 0;
if (gst_base_transform_is_passthrough (trans))
resampler_latency = 0;
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
- GST_DEBUG ("Peer latency: min %"
+ GST_DEBUG_OBJECT (resample, "Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
else
latency = 0;
- GST_DEBUG ("Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
+ GST_DEBUG_OBJECT (resample, "Our latency: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
- GST_DEBUG ("Calculated total latency : min %"
+ GST_DEBUG_OBJECT (resample, "Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
res = gst_pad_query_default (pad, query);
break;
}
- gst_object_unref (audioresample);
+ gst_object_unref (resample);
return res;
}
static const GstQueryType *
-audioresample_query_type (GstPad * pad)
+gst_audio_resample_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
}
static void
-gst_audioresample_set_property (GObject * object, guint prop_id,
+gst_audio_resample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstAudioResample *resample;
- audioresample = GST_AUDIORESAMPLE (object);
+ resample = GST_AUDIO_RESAMPLE (object);
switch (prop_id) {
- case PROP_FILTERLEN:
- audioresample->filter_length = g_value_get_int (value);
- GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
- audioresample->filter_length);
- if (audioresample->resample) {
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
- gst_element_post_message (GST_ELEMENT (audioresample),
- gst_message_new_latency (GST_OBJECT (audioresample)));
- }
+ case PROP_QUALITY:
+ resample->quality = g_value_get_int (value);
+ GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
+
+ gst_audio_resample_update_state (resample, resample->width,
+ resample->channels, resample->inrate, resample->outrate,
+ resample->quality, resample->fp);
+ break;
+ case PROP_FILTER_LENGTH:{
+ gint filter_length = g_value_get_int (value);
+
+ if (filter_length <= 8)
+ resample->quality = 0;
+ else if (filter_length <= 16)
+ resample->quality = 1;
+ else if (filter_length <= 32)
+ resample->quality = 2;
+ else if (filter_length <= 48)
+ resample->quality = 3;
+ else if (filter_length <= 64)
+ resample->quality = 4;
+ else if (filter_length <= 80)
+ resample->quality = 5;
+ else if (filter_length <= 96)
+ resample->quality = 6;
+ else if (filter_length <= 128)
+ resample->quality = 7;
+ else if (filter_length <= 160)
+ resample->quality = 8;
+ else if (filter_length <= 192)
+ resample->quality = 9;
+ else
+ resample->quality = 10;
+
+ GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
+
+ gst_audio_resample_update_state (resample, resample->width,
+ resample->channels, resample->inrate, resample->outrate,
+ resample->quality, resample->fp);
break;
+ }
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
static void
-gst_audioresample_get_property (GObject * object, guint prop_id,
+gst_audio_resample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstAudioresample *audioresample;
+ GstAudioResample *resample;
- audioresample = GST_AUDIORESAMPLE (object);
+ resample = GST_AUDIO_RESAMPLE (object);
switch (prop_id) {
- case PROP_FILTERLEN:
- g_value_set_int (value, audioresample->filter_length);
+ case PROP_QUALITY:
+ g_value_set_int (value, resample->quality);
+ break;
+ case PROP_FILTER_LENGTH:
+ switch (resample->quality) {
+ case 0:
+ g_value_set_int (value, 8);
+ break;
+ case 1:
+ g_value_set_int (value, 16);
+ break;
+ case 2:
+ g_value_set_int (value, 32);
+ break;
+ case 3:
+ g_value_set_int (value, 48);
+ break;
+ case 4:
+ g_value_set_int (value, 64);
+ break;
+ case 5:
+ g_value_set_int (value, 80);
+ break;
+ case 6:
+ g_value_set_int (value, 96);
+ break;
+ case 7:
+ g_value_set_int (value, 128);
+ break;
+ case 8:
+ g_value_set_int (value, 160);
+ break;
+ case 9:
+ g_value_set_int (value, 192);
+ break;
+ case 10:
+ g_value_set_int (value, 256);
+ break;
+ }
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
}
+#define BENCHMARK_SIZE 512
+
+static gboolean
+_benchmark_int_float (SpeexResamplerState * st)
+{
+ gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
+ gfloat in_tmp[BENCHMARK_SIZE], out_tmp[BENCHMARK_SIZE / 2];
+ gint i;
+ guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
+
+ for (i = 0; i < BENCHMARK_SIZE; i++) {
+ gfloat tmp = in[i];
+ in_tmp[i] = tmp / G_MAXINT16;
+ }
+
+ resample_float_resampler_process_interleaved_float (st,
+ (const guint8 *) in_tmp, &inlen, (guint8 *) out_tmp, &outlen);
+
+ if (outlen == 0) {
+ GST_ERROR ("Failed to use float resampler");
+ return FALSE;
+ }
+
+ for (i = 0; i < outlen; i++) {
+ gfloat tmp = out_tmp[i];
+ out[i] = CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
+ }
+
+ return TRUE;
+}
+
+static gboolean
+_benchmark_int_int (SpeexResamplerState * st)
+{
+ gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
+ guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
+
+ resample_int_resampler_process_interleaved_int (st, (const guint8 *) in,
+ &inlen, (guint8 *) out, &outlen);
+
+ if (outlen == 0) {
+ GST_ERROR ("Failed to use int resampler");
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+_benchmark_integer_resampling (void)
+{
+ OilProfile a, b;
+ gdouble av, bv;
+ SpeexResamplerState *sta, *stb;
+
+ oil_profile_init (&a);
+ oil_profile_init (&b);
+
+ sta = resample_float_resampler_init (1, 48000, 24000, 4, NULL);
+ if (sta == NULL) {
+ GST_ERROR ("Failed to create float resampler state");
+ return FALSE;
+ }
+
+ stb = resample_int_resampler_init (1, 48000, 24000, 4, NULL);
+ if (stb == NULL) {
+ resample_float_resampler_destroy (sta);
+ GST_ERROR ("Failed to create int resampler state");
+ return FALSE;
+ }
+
+ /* Warm up cache */
+ if (!_benchmark_int_float (sta))
+ goto error;
+ if (!_benchmark_int_float (sta))
+ goto error;
+
+ /* Benchmark */
+ oil_profile_start (&a);
+ if (!_benchmark_int_float (sta))
+ goto error;
+ oil_profile_stop (&a);
+
+ /* Warm up cache */
+ if (!_benchmark_int_int (stb))
+ goto error;
+ if (!_benchmark_int_int (stb))
+ goto error;
+
+ /* Benchmark */
+ oil_profile_start (&b);
+ if (!_benchmark_int_int (stb))
+ goto error;
+ oil_profile_stop (&b);
+
+ /* Handle results */
+ oil_profile_get_ave_std (&a, &av, NULL);
+ oil_profile_get_ave_std (&b, &bv, NULL);
+
+ gst_audio_resample_use_int = (av > bv);
+ resample_float_resampler_destroy (sta);
+ resample_float_resampler_destroy (stb);
+
+ if (av > bv)
+ GST_DEBUG ("Using integer resampler if appropiate: %lf < %lf", bv, av);
+ else
+ GST_DEBUG ("Using float resampler for everything: %lf <= %lf", av, bv);
+
+ return TRUE;
+
+error:
+ resample_float_resampler_destroy (sta);
+ resample_float_resampler_destroy (stb);
+
+ return FALSE;
+}
static gboolean
plugin_init (GstPlugin * plugin)
{
- resample_init ();
+ GST_DEBUG_CATEGORY_INIT (audio_resample_debug, "audioresample", 0,
+ "audio resampling element");
+
+ oil_init ();
+
+ if (!_benchmark_integer_resampling ())
+ return FALSE;
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
- GST_TYPE_AUDIORESAMPLE)) {
+ GST_TYPE_AUDIO_RESAMPLE)) {
return FALSE;
}
/* 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
*/
-#ifndef __AUDIORESAMPLE_H__
-#define __AUDIORESAMPLE_H__
+#ifndef __AUDIO_RESAMPLE_H__
+#define __AUDIO_RESAMPLE_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
-#include "resample.h"
+#include "speex_resampler_wrapper.h"
G_BEGIN_DECLS
-#define GST_TYPE_AUDIORESAMPLE \
- (gst_audioresample_get_type())
-#define GST_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,GstAudioresample))
-#define GST_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,GstAudioresampleClass))
-#define GST_IS_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
-#define GST_IS_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
+#define GST_TYPE_AUDIO_RESAMPLE \
+ (gst_audio_resample_get_type())
+#define GST_AUDIO_RESAMPLE(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RESAMPLE,GstAudioResample))
+#define GST_AUDIO_RESAMPLE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RESAMPLE,GstAudioResampleClass))
+#define GST_IS_AUDIO_RESAMPLE(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RESAMPLE))
+#define GST_IS_AUDIO_RESAMPLE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RESAMPLE))
-typedef struct _GstAudioresample GstAudioresample;
-typedef struct _GstAudioresampleClass GstAudioresampleClass;
+typedef struct _GstAudioResample GstAudioResample;
+typedef struct _GstAudioResampleClass GstAudioResampleClass;
/**
- * GstAudioresample:
+ * GstAudioResample:
*
* Opaque data structure.
*/
-struct _GstAudioresample {
+struct _GstAudioResample {
GstBaseTransform element;
+ /* <private> */
+
GstCaps *srccaps, *sinkcaps;
- gboolean passthru;
gboolean need_discont;
- guint64 offset;
- guint64 ts_offset;
+ guint64 next_offset;
GstClockTime next_ts;
- GstClockTime prev_ts, prev_duration;
- int channels;
-
- int i_rate;
- int o_rate;
- int filter_length;
-
- ResampleState * resample;
+ 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 _GstAudioresampleClass {
+struct _GstAudioResampleClass {
GstBaseTransformClass parent_class;
};
-GType gst_audioresample_get_type(void);
+GType gst_audio_resample_get_type(void);
G_END_DECLS
-#endif /* __AUDIORESAMPLE_H__ */
+#endif /* __AUDIO_RESAMPLE_H__ */
-/* 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.
- */
+/* Copyright (C) 2007-2008 Jean-Marc Valin
+ Copyright (C) 2008 Thorvald Natvig
+
+ File: resample.c
+ Arbitrary resampling code
+
+ Redistribution and use in source and binary forms, with or without
+ modification, are permitted provided that the following conditions are
+ met:
+
+ 1. Redistributions of source code must retain the above copyright notice,
+ this list of conditions and the following disclaimer.
+
+ 2. Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in the
+ documentation and/or other materials provided with the distribution.
+
+ 3. The name of the author may not be used to endorse or promote products
+ derived from this software without specific prior written permission.
+
+ THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
+ IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
+ OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
+ INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+ STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
+ ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/*
+ The design goals of this code are:
+ - Very fast algorithm
+ - SIMD-friendly algorithm
+ - Low memory requirement
+ - Good *perceptual* quality (and not best SNR)
+
+ Warning: This resampler is relatively new. Although I think I got rid of
+ all the major bugs and I don't expect the API to change anymore, there
+ may be something I've missed. So use with caution.
+
+ This algorithm is based on this original resampling algorithm:
+ Smith, Julius O. Digital Audio Resampling Home Page
+ Center for Computer Research in Music and Acoustics (CCRMA),
+ Stanford University, 2007.
+ Web published at http://www-ccrma.stanford.edu/~jos/resample/.
+
+ There is one main difference, though. This resampler uses cubic
+ interpolation instead of linear interpolation in the above paper. This
+ makes the table much smaller and makes it possible to compute that table
+ on a per-stream basis. In turn, being able to tweak the table for each
+ stream makes it possible to both reduce complexity on simple ratios
+ (e.g. 2/3), and get rid of the rounding operations in the inner loop.
+ The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
+*/
#ifdef HAVE_CONFIG_H
-#include <config.h>
+#include "config.h"
#endif
+#ifdef OUTSIDE_SPEEX
+#include <stdlib.h>
+
+#include <glib.h>
+
+#define EXPORT G_GNUC_INTERNAL
+
+static inline void *
+speex_alloc (int size)
+{
+ return g_malloc0 (size);
+}
+
+static inline void *
+speex_realloc (void *ptr, int size)
+{
+ return g_realloc (ptr, size);
+}
+
+static inline void
+speex_free (void *ptr)
+{
+ g_free (ptr);
+}
+
+#include "speex_resampler.h"
+#include "arch.h"
+#else /* OUTSIDE_SPEEX */
+
+#include "../include/speex/speex_resampler.h"
+#include "arch.h"
+#include "os_support.h"
+#endif /* OUTSIDE_SPEEX */
-#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"
+#ifndef M_PI
+#define M_PI 3.14159263
+#endif
+
+#ifdef FIXED_POINT
+#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
+#else
+#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
+#endif
-void resample_scale_ref (ResampleState * r);
-void resample_scale_functable (ResampleState * r);
+#define IMAX(a,b) ((a) > (b) ? (a) : (b))
+#define IMIN(a,b) ((a) < (b) ? (a) : (b))
-GST_DEBUG_CATEGORY (libaudioresample_debug);
+#ifndef NULL
+#define NULL 0
+#endif
-void
-resample_init (void)
+#ifdef _USE_SSE
+#include "resample_sse.h"
+#endif
+
+/* Numer of elements to allocate on the stack */
+#ifdef VAR_ARRAYS
+#define FIXED_STACK_ALLOC 8192
+#else
+#define FIXED_STACK_ALLOC 1024
+#endif
+
+typedef int (*resampler_basic_func) (SpeexResamplerState *, spx_uint32_t,
+ const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
+
+struct SpeexResamplerState_
{
- static int inited = 0;
+ spx_uint32_t in_rate;
+ spx_uint32_t out_rate;
+ spx_uint32_t num_rate;
+ spx_uint32_t den_rate;
+
+ int quality;
+ spx_uint32_t nb_channels;
+ spx_uint32_t filt_len;
+ spx_uint32_t mem_alloc_size;
+ spx_uint32_t buffer_size;
+ int int_advance;
+ int frac_advance;
+ float cutoff;
+ spx_uint32_t oversample;
+ int initialised;
+ int started;
+
+ /* These are per-channel */
+ spx_int32_t *last_sample;
+ spx_uint32_t *samp_frac_num;
+ spx_uint32_t *magic_samples;
+
+ spx_word16_t *mem;
+ spx_word16_t *sinc_table;
+ spx_uint32_t sinc_table_length;
+ resampler_basic_func resampler_ptr;
+
+ int in_stride;
+ int out_stride;
+};
- if (!inited) {
- oil_init ();
- inited = 1;
- GST_DEBUG_CATEGORY_INIT (libaudioresample_debug, "libaudioresample", 0,
- "audio resampling library");
+static double kaiser12_table[68] = {
+ 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
+ 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
+ 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
+ 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
+ 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
+ 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
+ 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
+ 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
+ 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
+ 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
+ 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
+ 0.00001000, 0.00000000
+};
+/*
+static double kaiser12_table[36] = {
+ 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
+ 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
+ 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
+ 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
+ 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
+ 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
+*/
+static double kaiser10_table[36] = {
+ 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
+ 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
+ 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
+ 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
+ 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
+ 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000
+};
+
+static double kaiser8_table[36] = {
+ 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
+ 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
+ 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
+ 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
+ 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
+ 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000
+};
+
+static double kaiser6_table[36] = {
+ 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
+ 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
+ 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
+ 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
+ 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
+ 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000
+};
+
+struct FuncDef
+{
+ double *table;
+ int oversample;
+};
+
+static struct FuncDef _KAISER12 = { kaiser12_table, 64 };
+
+#define KAISER12 (&_KAISER12)
+/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
+#define KAISER12 (&_KAISER12)*/
+static struct FuncDef _KAISER10 = { kaiser10_table, 32 };
+
+#define KAISER10 (&_KAISER10)
+static struct FuncDef _KAISER8 = { kaiser8_table, 32 };
+
+#define KAISER8 (&_KAISER8)
+static struct FuncDef _KAISER6 = { kaiser6_table, 32 };
+
+#define KAISER6 (&_KAISER6)
+
+struct QualityMapping
+{
+ int base_length;
+ int oversample;
+ float downsample_bandwidth;
+ float upsample_bandwidth;
+ struct FuncDef *window_func;
+};
+
+
+/* This table maps conversion quality to internal parameters. There are two
+ reasons that explain why the up-sampling bandwidth is larger than the
+ down-sampling bandwidth:
+ 1) When up-sampling, we can assume that the spectrum is already attenuated
+ close to the Nyquist rate (from an A/D or a previous resampling filter)
+ 2) Any aliasing that occurs very close to the Nyquist rate will be masked
+ by the sinusoids/noise just below the Nyquist rate (guaranteed only for
+ up-sampling).
+*/
+static const struct QualityMapping quality_map[11] = {
+ {8, 4, 0.830f, 0.860f, KAISER6}, /* Q0 */
+ {16, 4, 0.850f, 0.880f, KAISER6}, /* Q1 */
+ {32, 4, 0.882f, 0.910f, KAISER6}, /* Q2 *//* 82.3% cutoff ( ~60 dB stop) 6 */
+ {48, 8, 0.895f, 0.917f, KAISER8}, /* Q3 *//* 84.9% cutoff ( ~80 dB stop) 8 */
+ {64, 8, 0.921f, 0.940f, KAISER8}, /* Q4 *//* 88.7% cutoff ( ~80 dB stop) 8 */
+ {80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 *//* 89.1% cutoff (~100 dB stop) 10 */
+ {96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 *//* 91.5% cutoff (~100 dB stop) 10 */
+ {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 *//* 93.1% cutoff (~100 dB stop) 10 */
+ {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 *//* 94.5% cutoff (~100 dB stop) 10 */
+ {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 *//* 95.5% cutoff (~100 dB stop) 10 */
+ {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 *//* 96.6% cutoff (~100 dB stop) 10 */
+};
+
+/*8,24,40,56,80,104,128,160,200,256,320*/
+#ifdef DOUBLE_PRECISION
+static double
+compute_func (double x, struct FuncDef *func)
+{
+ double y, frac;
+#else
+static double
+compute_func (float x, struct FuncDef *func)
+{
+ float y, frac;
+#endif
+ double interp[4];
+ int ind;
+ y = x * func->oversample;
+ ind = (int) floor (y);
+ frac = (y - ind);
+ /* CSE with handle the repeated powers */
+ interp[3] = -0.1666666667 * frac + 0.1666666667 * (frac * frac * frac);
+ interp[2] = frac + 0.5 * (frac * frac) - 0.5 * (frac * frac * frac);
+ /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
+ interp[0] =
+ -0.3333333333 * frac + 0.5 * (frac * frac) -
+ 0.1666666667 * (frac * frac * frac);
+ /* Just to make sure we don't have rounding problems */
+ interp[1] = 1.f - interp[3] - interp[2] - interp[0];
+
+ /*sum = frac*accum[1] + (1-frac)*accum[2]; */
+ return interp[0] * func->table[ind] + interp[1] * func->table[ind + 1] +
+ interp[2] * func->table[ind + 2] + interp[3] * func->table[ind + 3];
+}
+
+#if 0
+#include <stdio.h>
+int
+main (int argc, char **argv)
+{
+ int i;
+ for (i = 0; i < 256; i++) {
+ printf ("%f\n", compute_func (i / 256., KAISER12));
}
+ return 0;
+}
+#endif
+
+#ifdef FIXED_POINT
+/* The slow way of computing a sinc for the table. Should improve that some day */
+static spx_word16_t
+sinc (float cutoff, float x, int N, struct FuncDef *window_func)
+{
+ /*fprintf (stderr, "%f ", x); */
+ float xx = x * cutoff;
+ if (fabs (x) < 1e-6f)
+ return WORD2INT (32768. * cutoff);
+ else if (fabs (x) > .5f * N)
+ return 0;
+ /*FIXME: Can it really be any slower than this? */
+ return WORD2INT (32768. * cutoff * sin (M_PI * xx) / (M_PI * xx) *
+ compute_func (fabs (2. * x / N), window_func));
+}
+#else
+/* The slow way of computing a sinc for the table. Should improve that some day */
+#ifdef DOUBLE_PRECISION
+static spx_word16_t
+sinc (double cutoff, double x, int N, struct FuncDef *window_func)
+{
+ /*fprintf (stderr, "%f ", x); */
+ double xx = x * cutoff;
+#else
+static spx_word16_t
+sinc (float cutoff, float x, int N, struct FuncDef *window_func)
+{
+ /*fprintf (stderr, "%f ", x); */
+ float xx = x * cutoff;
+#endif
+ if (fabs (x) < 1e-6)
+ return cutoff;
+ else if (fabs (x) > .5 * N)
+ return 0;
+ /*FIXME: Can it really be any slower than this? */
+ return cutoff * sin (M_PI * xx) / (M_PI * xx) * compute_func (fabs (2. * x /
+ N), window_func);
+}
+#endif
+
+#ifdef FIXED_POINT
+static void
+cubic_coef (spx_word16_t x, spx_word16_t interp[4])
+{
+ /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
+ but I know it's MMSE-optimal on a sinc */
+ spx_word16_t x2, x3;
+ x2 = MULT16_16_P15 (x, x);
+ x3 = MULT16_16_P15 (x, x2);
+ interp[0] =
+ PSHR32 (MULT16_16 (QCONST16 (-0.16667f, 15),
+ x) + MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
+ interp[1] =
+ EXTRACT16 (EXTEND32 (x) + SHR32 (SUB32 (EXTEND32 (x2), EXTEND32 (x3)),
+ 1));
+ interp[3] =
+ PSHR32 (MULT16_16 (QCONST16 (-0.33333f, 15),
+ x) + MULT16_16 (QCONST16 (.5f, 15),
+ x2) - MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
+ /* Just to make sure we don't have rounding problems */
+ interp[2] = Q15_ONE - interp[0] - interp[1] - interp[3];
+ if (interp[2] < 32767)
+ interp[2] += 1;
}
+#else
+static void
+cubic_coef (spx_word16_t frac, spx_word16_t interp[4])
+{
+ /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
+ but I know it's MMSE-optimal on a sinc */
+ interp[0] = -0.16667f * frac + 0.16667f * frac * frac * frac;
+ interp[1] = frac + 0.5f * frac * frac - 0.5f * frac * frac * frac;
+ /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
+ interp[3] =
+ -0.33333f * frac + 0.5f * frac * frac - 0.16667f * frac * frac * frac;
+ /* Just to make sure we don't have rounding problems */
+ interp[2] = 1. - interp[0] - interp[1] - interp[3];
+}
+#endif
-ResampleState *
-resample_new (void)
+#ifndef DOUBLE_PRECISION
+static int
+resampler_basic_direct_single (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
+ spx_word16_t * out, spx_uint32_t * out_len)
{
- ResampleState *r;
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const spx_word16_t *sinc_table = st->sinc_table;
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ spx_word32_t sum;
+ int j;
- r = malloc (sizeof (ResampleState));
- memset (r, 0, sizeof (ResampleState));
+ while (!(last_sample >= (spx_int32_t) * in_len
+ || out_sample >= (spx_int32_t) * out_len)) {
+ const spx_word16_t *sinc = &sinc_table[samp_frac_num * N];
+ const spx_word16_t *iptr = &in[last_sample];
- r->filter_length = 16;
+#ifndef OVERRIDE_INNER_PRODUCT_SINGLE
+ float accum[4] = { 0, 0, 0, 0 };
- r->i_start = 0;
- if (r->filter_length & 1) {
- r->o_start = 0;
- } else {
- r->o_start = r->o_inc * 0.5;
+ for (j = 0; j < N; j += 4) {
+ accum[0] += sinc[j] * iptr[j];
+ accum[1] += sinc[j + 1] * iptr[j + 1];
+ accum[2] += sinc[j + 2] * iptr[j + 2];
+ accum[3] += sinc[j + 3] * iptr[j + 3];
+ }
+ sum = accum[0] + accum[1] + accum[2] + accum[3];
+#else
+ sum = inner_product_single (sinc, iptr, N);
+#endif
+
+ out[out_stride * out_sample++] = PSHR32 (sum, 15);
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate) {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
}
- r->queue = audioresample_buffer_queue_new ();
- r->out_tmp = malloc (10000 * sizeof (double));
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+#endif
- r->need_reinit = 1;
+#ifdef FIXED_POINT
+#else
+/* This is the same as the previous function, except with a double-precision accumulator */
+static int
+resampler_basic_direct_double (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
+ spx_word16_t * out, spx_uint32_t * out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const spx_word16_t *sinc_table = st->sinc_table;
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ double sum;
+ int j;
- return r;
+ while (!(last_sample >= (spx_int32_t) * in_len
+ || out_sample >= (spx_int32_t) * out_len)) {
+ const spx_word16_t *sinc = &sinc_table[samp_frac_num * N];
+ const spx_word16_t *iptr = &in[last_sample];
+
+#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
+ double accum[4] = { 0, 0, 0, 0 };
+
+ for (j = 0; j < N; j += 4) {
+ accum[0] += sinc[j] * iptr[j];
+ accum[1] += sinc[j + 1] * iptr[j + 1];
+ accum[2] += sinc[j + 2] * iptr[j + 2];
+ accum[3] += sinc[j + 3] * iptr[j + 3];
+ }
+ sum = accum[0] + accum[1] + accum[2] + accum[3];
+#else
+ sum = inner_product_double (sinc, iptr, N);
+#endif
+
+ out[out_stride * out_sample++] = PSHR32 (sum, 15);
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate) {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
+ }
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
}
+#endif
-void
-resample_free (ResampleState * r)
+#ifndef DOUBLE_PRECISION
+static int
+resampler_basic_interpolate_single (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
+ spx_word16_t * out, spx_uint32_t * out_len)
{
- if (r->buffer) {
- free (r->buffer);
- }
- if (r->ft) {
- functable_free (r->ft);
- }
- if (r->queue) {
- audioresample_buffer_queue_free (r->queue);
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ int j;
+ spx_word32_t sum;
+
+ while (!(last_sample >= (spx_int32_t) * in_len
+ || out_sample >= (spx_int32_t) * out_len)) {
+ const spx_word16_t *iptr = &in[last_sample];
+
+ const int offset = samp_frac_num * st->oversample / st->den_rate;
+#ifdef FIXED_POINT
+ const spx_word16_t frac =
+ PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
+ st->den_rate);
+#else
+ const spx_word16_t frac =
+ ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
+ st->den_rate;
+#endif
+ spx_word16_t interp[4];
+
+
+#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
+ spx_word32_t accum[4] = { 0, 0, 0, 0 };
+
+ for (j = 0; j < N; j++) {
+ const spx_word16_t curr_in = iptr[j];
+ accum[0] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
+ accum[1] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
+ accum[2] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset]);
+ accum[3] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
+ }
+
+ cubic_coef (frac, interp);
+ sum =
+ MULT16_32_Q15 (interp[0], accum[0]) + MULT16_32_Q15 (interp[1],
+ accum[1]) + MULT16_32_Q15 (interp[2],
+ accum[2]) + MULT16_32_Q15 (interp[3], accum[3]);
+#else
+ cubic_coef (frac, interp);
+ sum =
+ interpolate_product_single (iptr,
+ st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample,
+ interp);
+#endif
+
+ out[out_stride * out_sample++] = PSHR32 (sum, 15);
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate) {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
}
- if (r->out_tmp) {
- free (r->out_tmp);
+
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
+}
+#endif
+
+#ifdef FIXED_POINT
+#else
+/* This is the same as the previous function, except with a double-precision accumulator */
+static int
+resampler_basic_interpolate_double (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
+ spx_word16_t * out, spx_uint32_t * out_len)
+{
+ const int N = st->filt_len;
+ int out_sample = 0;
+ int last_sample = st->last_sample[channel_index];
+ spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
+ const int out_stride = st->out_stride;
+ const int int_advance = st->int_advance;
+ const int frac_advance = st->frac_advance;
+ const spx_uint32_t den_rate = st->den_rate;
+ int j;
+ spx_word32_t sum;
+
+ while (!(last_sample >= (spx_int32_t) * in_len
+ || out_sample >= (spx_int32_t) * out_len)) {
+ const spx_word16_t *iptr = &in[last_sample];
+
+ const int offset = samp_frac_num * st->oversample / st->den_rate;
+#ifdef FIXED_POINT
+ const spx_word16_t frac =
+ PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
+ st->den_rate);
+#else
+#ifdef DOUBLE_PRECISION
+ const spx_word16_t frac =
+ ((double) ((samp_frac_num * st->oversample) % st->den_rate)) /
+ st->den_rate;
+#else
+ const spx_word16_t frac =
+ ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
+ st->den_rate;
+#endif
+#endif
+ spx_word16_t interp[4];
+
+
+#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
+ double accum[4] = { 0, 0, 0, 0 };
+
+ for (j = 0; j < N; j++) {
+ const double curr_in = iptr[j];
+ accum[0] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
+ accum[1] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
+ accum[2] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset]);
+ accum[3] +=
+ MULT16_16 (curr_in,
+ st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
+ }
+
+ cubic_coef (frac, interp);
+ sum =
+ MULT16_32_Q15 (interp[0], accum[0]) + MULT16_32_Q15 (interp[1],
+ accum[1]) + MULT16_32_Q15 (interp[2],
+ accum[2]) + MULT16_32_Q15 (interp[3], accum[3]);
+#else
+ cubic_coef (frac, interp);
+ sum =
+ interpolate_product_double (iptr,
+ st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample,
+ interp);
+#endif
+
+ out[out_stride * out_sample++] = PSHR32 (sum, 15);
+ last_sample += int_advance;
+ samp_frac_num += frac_advance;
+ if (samp_frac_num >= den_rate) {
+ samp_frac_num -= den_rate;
+ last_sample++;
+ }
}
- free (r);
+ st->last_sample[channel_index] = last_sample;
+ st->samp_frac_num[channel_index] = samp_frac_num;
+ return out_sample;
}
+#endif
static void
-resample_buffer_free (AudioresampleBuffer * buffer, void *priv)
+update_filter (SpeexResamplerState * st)
{
- if (buffer->priv2) {
- ((void (*)(void *)) buffer->priv2) (buffer->priv);
+ spx_uint32_t old_length;
+
+ old_length = st->filt_len;
+ st->oversample = quality_map[st->quality].oversample;
+ st->filt_len = quality_map[st->quality].base_length;
+
+ if (st->num_rate > st->den_rate) {
+ /* down-sampling */
+ st->cutoff =
+ quality_map[st->quality].downsample_bandwidth * st->den_rate /
+ st->num_rate;
+ /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
+ st->filt_len = st->filt_len * st->num_rate / st->den_rate;
+ /* Round down to make sure we have a multiple of 4 */
+ st->filt_len &= (~0x3);
+ if (2 * st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (4 * st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (8 * st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (16 * st->den_rate < st->num_rate)
+ st->oversample >>= 1;
+ if (st->oversample < 1)
+ st->oversample = 1;
+ } else {
+ /* up-sampling */
+ st->cutoff = quality_map[st->quality].upsample_bandwidth;
}
+
+ /* Choose the resampling type that requires the least amount of memory */
+ if (st->den_rate <= st->oversample) {
+ spx_uint32_t i;
+ if (!st->sinc_table)
+ st->sinc_table =
+ (spx_word16_t *) speex_alloc (st->filt_len * st->den_rate *
+ sizeof (spx_word16_t));
+ else if (st->sinc_table_length < st->filt_len * st->den_rate) {
+ st->sinc_table =
+ (spx_word16_t *) speex_realloc (st->sinc_table,
+ st->filt_len * st->den_rate * sizeof (spx_word16_t));
+ st->sinc_table_length = st->filt_len * st->den_rate;
+ }
+ for (i = 0; i < st->den_rate; i++) {
+ spx_int32_t j;
+ for (j = 0; j < st->filt_len; j++) {
+ st->sinc_table[i * st->filt_len + j] =
+ sinc (st->cutoff, ((j - (spx_int32_t) st->filt_len / 2 + 1) -
+#ifdef DOUBLE_PRECISION
+ ((double) i) / st->den_rate), st->filt_len,
+#else
+ ((float) i) / st->den_rate), st->filt_len,
+#endif
+ quality_map[st->quality].window_func);
+ }
+ }
+#ifdef FIXED_POINT
+ st->resampler_ptr = resampler_basic_direct_single;
+#else
+#ifdef DOUBLE_PRECISION
+ st->resampler_ptr = resampler_basic_direct_double;
+#else
+ if (st->quality > 8)
+ st->resampler_ptr = resampler_basic_direct_double;
+ else
+ st->resampler_ptr = resampler_basic_direct_single;
+#endif
+#endif
+ /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff); */
+ } else {
+ spx_int32_t i;
+ if (!st->sinc_table)
+ st->sinc_table =
+ (spx_word16_t *) speex_alloc ((st->filt_len * st->oversample +
+ 8) * sizeof (spx_word16_t));
+ else if (st->sinc_table_length < st->filt_len * st->oversample + 8) {
+ st->sinc_table =
+ (spx_word16_t *) speex_realloc (st->sinc_table,
+ (st->filt_len * st->oversample + 8) * sizeof (spx_word16_t));
+ st->sinc_table_length = st->filt_len * st->oversample + 8;
+ }
+ for (i = -4; i < (spx_int32_t) (st->oversample * st->filt_len + 4); i++)
+ st->sinc_table[i + 4] =
+#ifdef DOUBLE_PRECISION
+ sinc (st->cutoff, (i / (double) st->oversample - st->filt_len / 2),
+#else
+ sinc (st->cutoff, (i / (float) st->oversample - st->filt_len / 2),
+#endif
+ st->filt_len, quality_map[st->quality].window_func);
+#ifdef FIXED_POINT
+ st->resampler_ptr = resampler_basic_interpolate_single;
+#else
+#ifdef DOUBLE_PRECISION
+ st->resampler_ptr = resampler_basic_interpolate_double;
+#else
+ if (st->quality > 8)
+ st->resampler_ptr = resampler_basic_interpolate_double;
+ else
+ st->resampler_ptr = resampler_basic_interpolate_single;
+#endif
+#endif
+ /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff); */
+ }
+ st->int_advance = st->num_rate / st->den_rate;
+ st->frac_advance = st->num_rate % st->den_rate;
+
+
+ /* Here's the place where we update the filter memory to take into account
+ the change in filter length. It's probably the messiest part of the code
+ due to handling of lots of corner cases. */
+ if (!st->mem) {
+ spx_uint32_t i;
+ st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
+ st->mem =
+ (spx_word16_t *) speex_alloc (st->nb_channels * st->mem_alloc_size *
+ sizeof (spx_word16_t));
+ for (i = 0; i < st->nb_channels * st->mem_alloc_size; i++)
+ st->mem[i] = 0;
+ /*speex_warning("init filter"); */
+ } else if (!st->started) {
+ spx_uint32_t i;
+ st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
+ st->mem =
+ (spx_word16_t *) speex_realloc (st->mem,
+ st->nb_channels * st->mem_alloc_size * sizeof (spx_word16_t));
+ for (i = 0; i < st->nb_channels * st->mem_alloc_size; i++)
+ st->mem[i] = 0;
+ /*speex_warning("reinit filter"); */
+ } else if (st->filt_len > old_length) {
+ spx_int32_t i;
+ /* Increase the filter length */
+ /*speex_warning("increase filter size"); */
+ int old_alloc_size = st->mem_alloc_size;
+ if ((st->filt_len - 1 + st->buffer_size) > st->mem_alloc_size) {
+ st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
+ st->mem =
+ (spx_word16_t *) speex_realloc (st->mem,
+ st->nb_channels * st->mem_alloc_size * sizeof (spx_word16_t));
+ }
+ for (i = st->nb_channels - 1; i >= 0; i--) {
+ spx_int32_t j;
+ spx_uint32_t olen = old_length;
+ /*if (st->magic_samples[i]) */
+ {
+ /* Try and remove the magic samples as if nothing had happened */
+
+ /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
+ olen = old_length + 2 * st->magic_samples[i];
+ for (j = old_length - 2 + st->magic_samples[i]; j >= 0; j--)
+ st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]] =
+ st->mem[i * old_alloc_size + j];
+ for (j = 0; j < st->magic_samples[i]; j++)
+ st->mem[i * st->mem_alloc_size + j] = 0;
+ st->magic_samples[i] = 0;
+ }
+ if (st->filt_len > olen) {
+ /* If the new filter length is still bigger than the "augmented" length */
+ /* Copy data going backward */
+ for (j = 0; j < olen - 1; j++)
+ st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] =
+ st->mem[i * st->mem_alloc_size + (olen - 2 - j)];
+ /* Then put zeros for lack of anything better */
+ for (; j < st->filt_len - 1; j++)
+ st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] = 0;
+ /* Adjust last_sample */
+ st->last_sample[i] += (st->filt_len - olen) / 2;
+ } else {
+ /* Put back some of the magic! */
+ st->magic_samples[i] = (olen - st->filt_len) / 2;
+ for (j = 0; j < st->filt_len - 1 + st->magic_samples[i]; j++)
+ st->mem[i * st->mem_alloc_size + j] =
+ st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]];
+ }
+ }
+ } else if (st->filt_len < old_length) {
+ spx_uint32_t i;
+ /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
+ samples so they can be used directly as input the next time(s) */
+ for (i = 0; i < st->nb_channels; i++) {
+ spx_uint32_t j;
+ spx_uint32_t old_magic = st->magic_samples[i];
+ st->magic_samples[i] = (old_length - st->filt_len) / 2;
+ /* We must copy some of the memory that's no longer used */
+ /* Copy data going backward */
+ for (j = 0; j < st->filt_len - 1 + st->magic_samples[i] + old_magic; j++)
+ st->mem[i * st->mem_alloc_size + j] =
+ st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]];
+ st->magic_samples[i] += old_magic;
+ }
+ }
+
}
-/*
- * free_func: a function that frees the given closure. If NULL, caller is
- * responsible for freeing.
- */
-void
-resample_add_input_data (ResampleState * r, void *data, int size,
- void (*free_func) (void *), void *closure)
+EXPORT SpeexResamplerState *
+speex_resampler_init (spx_uint32_t nb_channels, spx_uint32_t in_rate,
+ spx_uint32_t out_rate, int quality, int *err)
+{
+ return speex_resampler_init_frac (nb_channels, in_rate, out_rate, in_rate,
+ out_rate, quality, err);
+}
+
+EXPORT SpeexResamplerState *
+speex_resampler_init_frac (spx_uint32_t nb_channels, spx_uint32_t ratio_num,
+ spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate,
+ int quality, int *err)
{
- AudioresampleBuffer *buffer;
+ spx_uint32_t i;
+ SpeexResamplerState *st;
+ if (quality > 10 || quality < 0) {
+ if (err)
+ *err = RESAMPLER_ERR_INVALID_ARG;
+ return NULL;
+ }
+ st = (SpeexResamplerState *) speex_alloc (sizeof (SpeexResamplerState));
+ st->initialised = 0;
+ st->started = 0;
+ st->in_rate = 0;
+ st->out_rate = 0;
+ st->num_rate = 0;
+ st->den_rate = 0;
+ st->quality = -1;
+ st->sinc_table_length = 0;
+ st->mem_alloc_size = 0;
+ st->filt_len = 0;
+ st->mem = 0;
+ st->resampler_ptr = 0;
+
+ st->cutoff = 1.f;
+ st->nb_channels = nb_channels;
+ st->in_stride = 1;
+ st->out_stride = 1;
+
+#ifdef FIXED_POINT
+ st->buffer_size = 160;
+#else
+ st->buffer_size = 160;
+#endif
+
+ /* Per channel data */
+ st->last_sample = (spx_int32_t *) speex_alloc (nb_channels * sizeof (int));
+ st->magic_samples = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
+ st->samp_frac_num = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
+ for (i = 0; i < nb_channels; i++) {
+ st->last_sample[i] = 0;
+ st->magic_samples[i] = 0;
+ st->samp_frac_num[i] = 0;
+ }
+
+ speex_resampler_set_quality (st, quality);
+ speex_resampler_set_rate_frac (st, ratio_num, ratio_den, in_rate, out_rate);
+
- RESAMPLE_DEBUG ("data %p size %d", data, size);
+ update_filter (st);
- buffer = audioresample_buffer_new_with_data (data, size);
- buffer->free = resample_buffer_free;
- buffer->priv2 = (void *) free_func;
- buffer->priv = closure;
+ st->initialised = 1;
+ if (err)
+ *err = RESAMPLER_ERR_SUCCESS;
- audioresample_buffer_queue_push (r->queue, buffer);
+ return st;
}
-void
-resample_input_flush (ResampleState * r)
+EXPORT void
+speex_resampler_destroy (SpeexResamplerState * st)
{
- RESAMPLE_DEBUG ("flush");
-
- audioresample_buffer_queue_flush (r->queue);
- r->buffer_filled = 0;
- r->need_reinit = 1;
+ speex_free (st->mem);
+ speex_free (st->sinc_table);
+ speex_free (st->last_sample);
+ speex_free (st->magic_samples);
+ speex_free (st->samp_frac_num);
+ speex_free (st);
}
-void
-resample_input_pushthrough (ResampleState * r)
+static int
+speex_resampler_process_native (SpeexResamplerState * st,
+ spx_uint32_t channel_index, spx_uint32_t * in_len, spx_word16_t * out,
+ spx_uint32_t * out_len)
{
- AudioresampleBuffer *buffer;
- int filter_bytes;
- int buffer_filled;
+ int j = 0;
+ const int N = st->filt_len;
+ int out_sample = 0;
+ spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
+ spx_uint32_t ilen;
+
+ st->started = 1;
- if (r->sample_size == 0)
- return;
+ /* Call the right resampler through the function ptr */
+ out_sample = st->resampler_ptr (st, channel_index, mem, in_len, out, out_len);
- filter_bytes = r->filter_length * r->sample_size;
- buffer_filled = r->buffer_filled;
+ if (st->last_sample[channel_index] < (spx_int32_t) * in_len)
+ *in_len = st->last_sample[channel_index];
+ *out_len = out_sample;
+ st->last_sample[channel_index] -= *in_len;
- RESAMPLE_DEBUG ("pushthrough filter_bytes %d, filled %d",
- filter_bytes, buffer_filled);
+ ilen = *in_len;
- /* if we have no pending samples, we don't need to do anything. */
- if (buffer_filled <= 0)
- return;
+ for (j = 0; j < N - 1; ++j)
+ mem[j] = mem[j + ilen];
- /* send filter_length/2 number of samples so we can get to the
- * last queued samples */
- buffer = audioresample_buffer_new_and_alloc (filter_bytes / 2);
- memset (buffer->data, 0, buffer->length);
+ return RESAMPLER_ERR_SUCCESS;
+}
- RESAMPLE_DEBUG ("pushthrough %u", buffer->length);
+static int
+speex_resampler_magic (SpeexResamplerState * st, spx_uint32_t channel_index,
+ spx_word16_t ** out, spx_uint32_t out_len)
+{
+ spx_uint32_t tmp_in_len = st->magic_samples[channel_index];
+ spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
+ const int N = st->filt_len;
- audioresample_buffer_queue_push (r->queue, buffer);
+ speex_resampler_process_native (st, channel_index, &tmp_in_len, *out,
+ &out_len);
+
+ st->magic_samples[channel_index] -= tmp_in_len;
+
+ /* If we couldn't process all "magic" input samples, save the rest for next time */
+ if (st->magic_samples[channel_index]) {
+ spx_uint32_t i;
+ for (i = 0; i < st->magic_samples[channel_index]; i++)
+ mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len];
+ }
+ *out += out_len * st->out_stride;
+ return out_len;
}
-void
-resample_input_eos (ResampleState * r)
+#ifdef FIXED_POINT
+EXPORT int
+speex_resampler_process_int (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len,
+ spx_int16_t * out, spx_uint32_t * out_len)
+#else
+#ifdef DOUBLE_PRECISION
+EXPORT int
+speex_resampler_process_float (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const double *in, spx_uint32_t * in_len,
+ double *out, spx_uint32_t * out_len)
+#else
+EXPORT int
+speex_resampler_process_float (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len,
+ float *out, spx_uint32_t * out_len)
+#endif
+#endif
{
- RESAMPLE_DEBUG ("EOS");
- resample_input_pushthrough (r);
- r->eos = 1;
+ int j;
+ spx_uint32_t ilen = *in_len;
+ spx_uint32_t olen = *out_len;
+ spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
+ const int filt_offs = st->filt_len - 1;
+ const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
+ const int istride = st->in_stride;
+
+ if (st->magic_samples[channel_index])
+ olen -= speex_resampler_magic (st, channel_index, &out, olen);
+ if (!st->magic_samples[channel_index]) {
+ while (ilen && olen) {
+ spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
+ spx_uint32_t ochunk = olen;
+
+ if (in) {
+ for (j = 0; j < ichunk; ++j)
+ x[j + filt_offs] = in[j * istride];
+ } else {
+ for (j = 0; j < ichunk; ++j)
+ x[j + filt_offs] = 0;
+ }
+ speex_resampler_process_native (st, channel_index, &ichunk, out, &ochunk);
+ ilen -= ichunk;
+ olen -= ochunk;
+ out += ochunk * st->out_stride;
+ if (in)
+ in += ichunk * istride;
+ }
+ }
+ *in_len -= ilen;
+ *out_len -= olen;
+ return RESAMPLER_ERR_SUCCESS;
}
-int
-resample_get_output_size_for_input (ResampleState * r, int size)
+#ifdef FIXED_POINT
+EXPORT int
+speex_resampler_process_float (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len,
+ float *out, spx_uint32_t * out_len)
+#else
+EXPORT int
+speex_resampler_process_int (SpeexResamplerState * st,
+ spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len,
+ spx_int16_t * out, spx_uint32_t * out_len)
+#endif
{
- int outsize;
- double outd;
- int avail;
- int filter_bytes;
- int buffer_filled;
+ int j;
+ const int istride_save = st->in_stride;
+ const int ostride_save = st->out_stride;
+ spx_uint32_t ilen = *in_len;
+ spx_uint32_t olen = *out_len;
+ spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
+ const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
+#ifdef VAR_ARRAYS
+ const unsigned int ylen =
+ (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
+ VARDECL (spx_word16_t * ystack);
+ ALLOC (ystack, ylen, spx_word16_t);
+#else
+ const unsigned int ylen = FIXED_STACK_ALLOC;
+ spx_word16_t ystack[FIXED_STACK_ALLOC];
+#endif
- if (r->sample_size == 0)
- return 0;
+ st->out_stride = 1;
- filter_bytes = r->filter_length * r->sample_size;
- buffer_filled = filter_bytes / 2 - r->buffer_filled / 2;
+ while (ilen && olen) {
+ spx_word16_t *y = ystack;
+ spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
+ spx_uint32_t ochunk = (olen > ylen) ? ylen : olen;
+ spx_uint32_t omagic = 0;
- avail =
- audioresample_buffer_queue_get_depth (r->queue) + size - buffer_filled;
+ if (st->magic_samples[channel_index]) {
+ omagic = speex_resampler_magic (st, channel_index, &y, ochunk);
+ ochunk -= omagic;
+ olen -= omagic;
+ }
+ if (!st->magic_samples[channel_index]) {
+ if (in) {
+ for (j = 0; j < ichunk; ++j)
+#ifdef FIXED_POINT
+ x[j + st->filt_len - 1] = WORD2INT (in[j * istride_save]);
+#else
+ x[j + st->filt_len - 1] = in[j * istride_save];
+#endif
+ } else {
+ for (j = 0; j < ichunk; ++j)
+ x[j + st->filt_len - 1] = 0;
+ }
- RESAMPLE_DEBUG ("avail %d, o_rate %f, i_rate %f, filter_bytes %d, filled %d",
- avail, r->o_rate, r->i_rate, filter_bytes, buffer_filled);
- if (avail <= 0)
- return 0;
+ speex_resampler_process_native (st, channel_index, &ichunk, y, &ochunk);
+ } else {
+ ichunk = 0;
+ ochunk = 0;
+ }
- outd = (double) avail *r->o_rate / r->i_rate;
+ for (j = 0; j < ochunk + omagic; ++j)
+#ifdef FIXED_POINT
+ out[j * ostride_save] = ystack[j];
+#else
+ out[j * ostride_save] = WORD2INT (ystack[j]);
+#endif
- outsize = (int) floor (outd);
+ ilen -= ichunk;
+ olen -= ochunk;
+ out += (ochunk + omagic) * ostride_save;
+ if (in)
+ in += ichunk * istride_save;
+ }
+ st->out_stride = ostride_save;
+ *in_len -= ilen;
+ *out_len -= olen;
- /* round off for sample size */
- outsize -= outsize % r->sample_size;
+ return RESAMPLER_ERR_SUCCESS;
+}
- return outsize;
+#ifdef DOUBLE_PRECISION
+EXPORT int
+speex_resampler_process_interleaved_float (SpeexResamplerState * st,
+ const double *in, spx_uint32_t * in_len, double *out,
+ spx_uint32_t * out_len)
+#else
+EXPORT int
+speex_resampler_process_interleaved_float (SpeexResamplerState * st,
+ const float *in, spx_uint32_t * in_len, float *out, spx_uint32_t * out_len)
+#endif
+{
+ spx_uint32_t i;
+ int istride_save, ostride_save;
+ spx_uint32_t bak_len = *out_len;
+ istride_save = st->in_stride;
+ ostride_save = st->out_stride;
+ st->in_stride = st->out_stride = st->nb_channels;
+ for (i = 0; i < st->nb_channels; i++) {
+ *out_len = bak_len;
+ if (in != NULL)
+ speex_resampler_process_float (st, i, in + i, in_len, out + i, out_len);
+ else
+ speex_resampler_process_float (st, i, NULL, in_len, out + i, out_len);
+ }
+ st->in_stride = istride_save;
+ st->out_stride = ostride_save;
+ return RESAMPLER_ERR_SUCCESS;
}
-int
-resample_get_input_size_for_output (ResampleState * r, int size)
+EXPORT int
+speex_resampler_process_interleaved_int (SpeexResamplerState * st,
+ const spx_int16_t * in, spx_uint32_t * in_len, spx_int16_t * out,
+ spx_uint32_t * out_len)
{
- int outsize;
- double outd;
- int avail;
+ spx_uint32_t i;
+ int istride_save, ostride_save;
+ spx_uint32_t bak_len = *out_len;
+ istride_save = st->in_stride;
+ ostride_save = st->out_stride;
+ st->in_stride = st->out_stride = st->nb_channels;
+ for (i = 0; i < st->nb_channels; i++) {
+ *out_len = bak_len;
+ if (in != NULL)
+ speex_resampler_process_int (st, i, in + i, in_len, out + i, out_len);
+ else
+ speex_resampler_process_int (st, i, NULL, in_len, out + i, out_len);
+ }
+ st->in_stride = istride_save;
+ st->out_stride = ostride_save;
+ return RESAMPLER_ERR_SUCCESS;
+}
- if (r->sample_size == 0)
- return 0;
+EXPORT int
+speex_resampler_set_rate (SpeexResamplerState * st, spx_uint32_t in_rate,
+ spx_uint32_t out_rate)
+{
+ return speex_resampler_set_rate_frac (st, in_rate, out_rate, in_rate,
+ out_rate);
+}
- avail = size;
+EXPORT void
+speex_resampler_get_rate (SpeexResamplerState * st, spx_uint32_t * in_rate,
+ spx_uint32_t * out_rate)
+{
+ *in_rate = st->in_rate;
+ *out_rate = st->out_rate;
+}
- RESAMPLE_DEBUG ("size %d, o_rate %f, i_rate %f", avail, r->o_rate, r->i_rate);
- outd = (double) avail *r->i_rate / r->o_rate;
+EXPORT int
+speex_resampler_set_rate_frac (SpeexResamplerState * st, spx_uint32_t ratio_num,
+ spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
+{
+ spx_uint32_t fact;
+ spx_uint32_t old_den;
+ spx_uint32_t i;
+ if (st->in_rate == in_rate && st->out_rate == out_rate
+ && st->num_rate == ratio_num && st->den_rate == ratio_den)
+ return RESAMPLER_ERR_SUCCESS;
- outsize = (int) ceil (outd);
+ old_den = st->den_rate;
+ st->in_rate = in_rate;
+ st->out_rate = out_rate;
+ st->num_rate = ratio_num;
+ st->den_rate = ratio_den;
+ /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
+ for (fact = 2; fact <= IMIN (st->num_rate, st->den_rate); fact++) {
+ while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0)) {
+ st->num_rate /= fact;
+ st->den_rate /= fact;
+ }
+ }
- /* round off for sample size */
- outsize -= outsize % r->sample_size;
+ if (old_den > 0) {
+ for (i = 0; i < st->nb_channels; i++) {
+ st->samp_frac_num[i] = st->samp_frac_num[i] * st->den_rate / old_den;
+ /* Safety net */
+ if (st->samp_frac_num[i] >= st->den_rate)
+ st->samp_frac_num[i] = st->den_rate - 1;
+ }
+ }
- return outsize;
+ if (st->initialised)
+ update_filter (st);
+ return RESAMPLER_ERR_SUCCESS;
}
-int
-resample_get_output_size (ResampleState * r)
+EXPORT void
+speex_resampler_get_ratio (SpeexResamplerState * st, spx_uint32_t * ratio_num,
+ spx_uint32_t * ratio_den)
{
- return resample_get_output_size_for_input (r, 0);
+ *ratio_num = st->num_rate;
+ *ratio_den = st->den_rate;
}
-int
-resample_get_output_data (ResampleState * r, void *data, int size)
+EXPORT int
+speex_resampler_set_quality (SpeexResamplerState * st, int quality)
{
- r->o_buf = data;
- r->o_size = size;
+ if (quality > 10 || quality < 0)
+ return RESAMPLER_ERR_INVALID_ARG;
+ if (st->quality == quality)
+ return RESAMPLER_ERR_SUCCESS;
+ st->quality = quality;
+ if (st->initialised)
+ update_filter (st);
+ return RESAMPLER_ERR_SUCCESS;
+}
- if (size == 0)
- return 0;
+EXPORT void
+speex_resampler_get_quality (SpeexResamplerState * st, int *quality)
+{
+ *quality = st->quality;
+}
- switch (r->method) {
- case 0:
- resample_scale_ref (r);
- break;
- case 1:
- resample_scale_functable (r);
- break;
- default:
- break;
- }
+EXPORT void
+speex_resampler_set_input_stride (SpeexResamplerState * st, spx_uint32_t stride)
+{
+ st->in_stride = stride;
+}
- return size - r->o_size;
+EXPORT void
+speex_resampler_get_input_stride (SpeexResamplerState * st,
+ spx_uint32_t * stride)
+{
+ *stride = st->in_stride;
}
-void
-resample_set_filter_length (ResampleState * r, int length)
+EXPORT void
+speex_resampler_set_output_stride (SpeexResamplerState * st,
+ spx_uint32_t stride)
{
- r->filter_length = length;
- r->need_reinit = 1;
+ st->out_stride = stride;
}
-void
-resample_set_input_rate (ResampleState * r, double rate)
+EXPORT void
+speex_resampler_get_output_stride (SpeexResamplerState * st,
+ spx_uint32_t * stride)
{
- r->i_rate = rate;
- r->need_reinit = 1;
+ *stride = st->out_stride;
}
-void
-resample_set_output_rate (ResampleState * r, double rate)
+EXPORT int
+speex_resampler_get_input_latency (SpeexResamplerState * st)
{
- r->o_rate = rate;
- r->need_reinit = 1;
+ return st->filt_len / 2;
}
-void
-resample_set_n_channels (ResampleState * r, int n_channels)
+EXPORT int
+speex_resampler_get_output_latency (SpeexResamplerState * st)
{
- r->n_channels = n_channels;
- r->sample_size = r->n_channels * resample_format_size (r->format);
- r->need_reinit = 1;
+ return ((st->filt_len / 2) * st->den_rate +
+ (st->num_rate >> 1)) / st->num_rate;
}
-void
-resample_set_format (ResampleState * r, ResampleFormat format)
+EXPORT int
+speex_resampler_skip_zeros (SpeexResamplerState * st)
{
- r->format = format;
- r->sample_size = r->n_channels * resample_format_size (r->format);
- r->need_reinit = 1;
+ spx_uint32_t i;
+ for (i = 0; i < st->nb_channels; i++)
+ st->last_sample[i] = st->filt_len / 2;
+ return RESAMPLER_ERR_SUCCESS;
}
-void
-resample_set_method (ResampleState * r, int method)
+EXPORT int
+speex_resampler_reset_mem (SpeexResamplerState * st)
{
- r->method = method;
- r->need_reinit = 1;
+ spx_uint32_t i;
+ for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
+ st->mem[i] = 0;
+ return RESAMPLER_ERR_SUCCESS;
}
-int
-resample_format_size (ResampleFormat format)
-{
- switch (format) {
- case RESAMPLE_FORMAT_S16:
- return 2;
- case RESAMPLE_FORMAT_S32:
- case RESAMPLE_FORMAT_F32:
- return 4;
- case RESAMPLE_FORMAT_F64:
- return 8;
+EXPORT const char *
+speex_resampler_strerror (int err)
+{
+ switch (err) {
+ case RESAMPLER_ERR_SUCCESS:
+ return "Success.";
+ case RESAMPLER_ERR_ALLOC_FAILED:
+ return "Memory allocation failed.";
+ case RESAMPLER_ERR_BAD_STATE:
+ return "Bad resampler state.";
+ case RESAMPLER_ERR_INVALID_ARG:
+ return "Invalid argument.";
+ case RESAMPLER_ERR_PTR_OVERLAP:
+ return "Input and output buffers overlap.";
+ default:
+ return "Unknown error. Bad error code or strange version mismatch.";
}
- return 0;
}
+++ /dev/null
-/* 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__ */
-
+++ /dev/null
-/* 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;
- }
-
-}
+++ /dev/null
-/* 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;
- }
-
-}
+++ /dev/null
-/* 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;
- }
-}
+++ /dev/null
-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
-
+++ /dev/null
-/* GStreamer
- * Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
- * Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
- * 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.
- */
-
-/**
- * SECTION:element-audioresample
- *
- * audioresample resamples raw audio buffers to different sample rates using
- * a configurable windowing function to enhance quality.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! audio/x-raw-int, rate=8000 ! alsasink
- * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
- * To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
- * </refsect2>
- */
-
-/* TODO:
- * - Enable SSE/ARM optimizations and select at runtime
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-#include <math.h>
-
-#include "gstspeexresample.h"
-#include <gst/audio/audio.h>
-#include <gst/base/gstbasetransform.h>
-
-#define OIL_ENABLE_UNSTABLE_API
-#include <liboil/liboilprofile.h>
-#include <liboil/liboil.h>
-
-GST_DEBUG_CATEGORY (speex_resample_debug);
-#define GST_CAT_DEFAULT speex_resample_debug
-
-enum
-{
- PROP_0,
- PROP_QUALITY,
- PROP_FILTER_LENGTH
-};
-
-#define SUPPORTED_CAPS \
-GST_STATIC_CAPS ( \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) { 32, 64 }; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 32, " \
- "depth = (int) 32, " \
- "signed = (boolean) true; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 24, " \
- "depth = (int) 24, " \
- "signed = (boolean) true; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 16, " \
- "depth = (int) 16, " \
- "signed = (boolean) true; " \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 8, " \
- "depth = (int) 8, " \
- "signed = (boolean) true" \
-)
-
-/* If TRUE integer arithmetic resampling is faster and will be used if appropiate */
-static gboolean gst_speex_resample_use_int = FALSE;
-
-static GstStaticPadTemplate gst_speex_resample_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-
-static GstStaticPadTemplate gst_speex_resample_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
-
-static void gst_speex_resample_set_property (GObject * object,
- guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_speex_resample_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec);
-
-/* vmethods */
-static gboolean gst_speex_resample_get_unit_size (GstBaseTransform * base,
- GstCaps * caps, guint * size);
-static GstCaps *gst_speex_resample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps);
-static void gst_speex_resample_fixate_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
-static gboolean gst_speex_resample_transform_size (GstBaseTransform * trans,
- GstPadDirection direction, GstCaps * incaps, guint insize,
- GstCaps * outcaps, guint * outsize);
-static gboolean gst_speex_resample_set_caps (GstBaseTransform * base,
- GstCaps * incaps, GstCaps * outcaps);
-static GstFlowReturn gst_speex_resample_transform (GstBaseTransform * base,
- GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean gst_speex_resample_event (GstBaseTransform * base,
- GstEvent * event);
-static gboolean gst_speex_resample_start (GstBaseTransform * base);
-static gboolean gst_speex_resample_stop (GstBaseTransform * base);
-static gboolean gst_speex_resample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *gst_speex_resample_query_type (GstPad * pad);
-
-GST_BOILERPLATE (GstSpeexResample, gst_speex_resample, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM);
-
-static void
-gst_speex_resample_base_init (gpointer g_class)
-{
- GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_speex_resample_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_speex_resample_sink_template));
-
- gst_element_class_set_details_simple (gstelement_class, "Audio resampler",
- "Filter/Converter/Audio", "Resamples audio",
- "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
-
-static void
-gst_speex_resample_class_init (GstSpeexResampleClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
-
- gobject_class->set_property = gst_speex_resample_set_property;
- gobject_class->get_property = gst_speex_resample_get_property;
-
- g_object_class_install_property (gobject_class, PROP_QUALITY,
- g_param_spec_int ("quality", "Quality", "Resample quality with 0 being "
- "the lowest and 10 being the best",
- SPEEX_RESAMPLER_QUALITY_MIN, SPEEX_RESAMPLER_QUALITY_MAX,
- SPEEX_RESAMPLER_QUALITY_DEFAULT,
- G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
-
- /* FIXME 0.11: Remove this property, it's just for compatibility
- * with old audioresample
- */
- g_object_class_install_property (gobject_class, PROP_FILTER_LENGTH,
- g_param_spec_int ("filter-length", "Filter length",
- "DEPRECATED, DON'T USE THIS! " "Length of the resample filter", 0,
- G_MAXINT, 64, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
-
- GST_BASE_TRANSFORM_CLASS (klass)->start =
- GST_DEBUG_FUNCPTR (gst_speex_resample_start);
- GST_BASE_TRANSFORM_CLASS (klass)->stop =
- GST_DEBUG_FUNCPTR (gst_speex_resample_stop);
- GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
- GST_DEBUG_FUNCPTR (gst_speex_resample_transform_size);
- GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
- GST_DEBUG_FUNCPTR (gst_speex_resample_get_unit_size);
- GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
- GST_DEBUG_FUNCPTR (gst_speex_resample_transform_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
- GST_DEBUG_FUNCPTR (gst_speex_resample_fixate_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
- GST_DEBUG_FUNCPTR (gst_speex_resample_set_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->transform =
- GST_DEBUG_FUNCPTR (gst_speex_resample_transform);
- GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (gst_speex_resample_event);
-
- GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
-}
-
-static void
-gst_speex_resample_init (GstSpeexResample * resample,
- GstSpeexResampleClass * klass)
-{
- GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
-
- resample->quality = SPEEX_RESAMPLER_QUALITY_DEFAULT;
-
- resample->need_discont = FALSE;
-
- gst_pad_set_query_function (trans->srcpad, gst_speex_resample_query);
- gst_pad_set_query_type_function (trans->srcpad,
- gst_speex_resample_query_type);
-}
-
-/* vmethods */
-static gboolean
-gst_speex_resample_start (GstBaseTransform * base)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- resample->next_offset = -1;
- resample->next_ts = -1;
- resample->next_upstream_ts = -1;
-
- return TRUE;
-}
-
-static gboolean
-gst_speex_resample_stop (GstBaseTransform * base)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- if (resample->state) {
- resample->funcs->destroy (resample->state);
- resample->state = NULL;
- }
-
- resample->funcs = NULL;
-
- g_free (resample->tmp_in);
- resample->tmp_in = NULL;
- resample->tmp_in_size = 0;
-
- g_free (resample->tmp_out);
- resample->tmp_out = NULL;
- resample->tmp_out_size = 0;
-
- gst_caps_replace (&resample->sinkcaps, NULL);
- gst_caps_replace (&resample->srccaps, NULL);
-
- return TRUE;
-}
-
-static gboolean
-gst_speex_resample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
- guint * size)
-{
- gint width, channels;
- GstStructure *structure;
- gboolean ret;
-
- g_return_val_if_fail (size != NULL, FALSE);
-
- /* this works for both float and int */
- structure = gst_caps_get_structure (caps, 0);
- ret = gst_structure_get_int (structure, "width", &width);
- ret &= gst_structure_get_int (structure, "channels", &channels);
-
- if (G_UNLIKELY (!ret))
- return FALSE;
-
- *size = (width / 8) * channels;
-
- return TRUE;
-}
-
-static GstCaps *
-gst_speex_resample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps)
-{
- GstCaps *res;
- GstStructure *structure;
-
- /* transform caps gives one single caps so we can just replace
- * the rate property with our range. */
- res = gst_caps_copy (caps);
- structure = gst_caps_get_structure (res, 0);
- gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
-
- return res;
-}
-
-/* Fixate rate to the allowed rate that has the smallest difference */
-static void
-gst_speex_resample_fixate_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
-{
- GstStructure *s;
- gint rate;
-
- s = gst_caps_get_structure (caps, 0);
- if (G_UNLIKELY (!gst_structure_get_int (s, "rate", &rate)))
- return;
-
- s = gst_caps_get_structure (othercaps, 0);
- gst_structure_fixate_field_nearest_int (s, "rate", rate);
-}
-
-static const SpeexResampleFuncs *
-gst_speex_resample_get_funcs (gint width, gboolean fp)
-{
- const SpeexResampleFuncs *funcs = NULL;
-
- if (gst_speex_resample_use_int && (width == 8 || width == 16) && !fp)
- funcs = &int_funcs;
- else if ((!gst_speex_resample_use_int && (width == 8 || width == 16) && !fp)
- || (width == 32 && fp))
- funcs = &float_funcs;
- else if ((width == 64 && fp) || ((width == 32 || width == 24) && !fp))
- funcs = &double_funcs;
- else
- g_assert_not_reached ();
-
- return funcs;
-}
-
-static SpeexResamplerState *
-gst_speex_resample_init_state (GstSpeexResample * resample, gint width,
- gint channels, gint inrate, gint outrate, gint quality, gboolean fp)
-{
- SpeexResamplerState *ret = NULL;
- gint err = RESAMPLER_ERR_SUCCESS;
- const SpeexResampleFuncs *funcs = gst_speex_resample_get_funcs (width, fp);
-
- ret = funcs->init (channels, inrate, outrate, quality, &err);
-
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
- GST_ERROR_OBJECT (resample, "Failed to create resampler state: %s",
- funcs->strerror (err));
- return NULL;
- }
-
- funcs->skip_zeros (ret);
-
- return ret;
-}
-
-static gboolean
-gst_speex_resample_update_state (GstSpeexResample * resample, gint width,
- gint channels, gint inrate, gint outrate, gint quality, gboolean fp)
-{
- gboolean ret = TRUE;
- gboolean updated_latency = FALSE;
-
- updated_latency = (resample->inrate != inrate
- || quality != resample->quality) && resample->state != NULL;
-
- if (resample->state == NULL) {
- ret = TRUE;
- } else if (resample->channels != channels || fp != resample->fp
- || width != resample->width) {
- resample->funcs->destroy (resample->state);
- resample->state =
- gst_speex_resample_init_state (resample, width, channels, inrate,
- outrate, quality, fp);
-
- resample->funcs = gst_speex_resample_get_funcs (width, fp);
- ret = (resample->state != NULL);
- } else if (resample->inrate != inrate || resample->outrate != outrate) {
- gint err = RESAMPLER_ERR_SUCCESS;
-
- err = resample->funcs->set_rate (resample->state, inrate, outrate);
-
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
- GST_ERROR_OBJECT (resample, "Failed to update rate: %s",
- resample->funcs->strerror (err));
-
- ret = (err == RESAMPLER_ERR_SUCCESS);
- } else if (quality != resample->quality) {
- gint err = RESAMPLER_ERR_SUCCESS;
-
- err = resample->funcs->set_quality (resample->state, quality);
-
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS))
- GST_ERROR_OBJECT (resample, "Failed to update quality: %s",
- resample->funcs->strerror (err));
-
- ret = (err == RESAMPLER_ERR_SUCCESS);
- }
-
- resample->width = width;
- resample->channels = channels;
- resample->fp = fp;
- resample->quality = quality;
- resample->inrate = inrate;
- resample->outrate = outrate;
-
- if (updated_latency)
- gst_element_post_message (GST_ELEMENT (resample),
- gst_message_new_latency (GST_OBJECT (resample)));
-
- return ret;
-}
-
-static void
-gst_speex_resample_reset_state (GstSpeexResample * resample)
-{
- if (resample->state)
- resample->funcs->reset_mem (resample->state);
-}
-
-static gboolean
-gst_speex_resample_parse_caps (GstCaps * incaps,
- GstCaps * outcaps, gint * width, gint * channels, gint * inrate,
- gint * outrate, gboolean * fp)
-{
- GstStructure *structure;
- gboolean ret;
- gint mywidth, myinrate, myoutrate, mychannels;
- gboolean myfp;
-
- GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
- GST_PTR_FORMAT, incaps, outcaps);
-
- structure = gst_caps_get_structure (incaps, 0);
-
- if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float"))
- myfp = TRUE;
- else
- myfp = FALSE;
-
- ret = gst_structure_get_int (structure, "rate", &myinrate);
- ret &= gst_structure_get_int (structure, "channels", &mychannels);
- ret &= gst_structure_get_int (structure, "width", &mywidth);
- if (G_UNLIKELY (!ret))
- goto no_in_rate_channels;
-
- structure = gst_caps_get_structure (outcaps, 0);
- ret = gst_structure_get_int (structure, "rate", &myoutrate);
- if (G_UNLIKELY (!ret))
- goto no_out_rate;
-
- if (channels)
- *channels = mychannels;
- if (inrate)
- *inrate = myinrate;
- if (outrate)
- *outrate = myoutrate;
- if (width)
- *width = mywidth;
- if (fp)
- *fp = myfp;
-
- return TRUE;
-
- /* ERRORS */
-no_in_rate_channels:
- {
- GST_DEBUG ("could not get input rate and channels");
- return FALSE;
- }
-no_out_rate:
- {
- GST_DEBUG ("could not get output rate");
- return FALSE;
- }
-}
-
-static gint
-_gcd (gint a, gint b)
-{
- while (b != 0) {
- int temp = a;
-
- a = b;
- b = temp % b;
- }
-
- return ABS (a);
-}
-
-static gboolean
-gst_speex_resample_transform_size (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
- guint * othersize)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
- GstCaps *srccaps, *sinkcaps;
- gboolean ret = TRUE;
- guint32 ratio_den, ratio_num;
- gint inrate, outrate, gcd;
- gint width;
-
- GST_LOG_OBJECT (resample, "asked to transform size %d in direction %s",
- size, direction == GST_PAD_SINK ? "SINK" : "SRC");
- if (direction == GST_PAD_SINK) {
- sinkcaps = caps;
- srccaps = othercaps;
- } else {
- sinkcaps = othercaps;
- srccaps = caps;
- }
-
- ret =
- gst_speex_resample_parse_caps (caps, othercaps, &width, NULL, &inrate,
- &outrate, NULL);
- if (G_UNLIKELY (!ret)) {
- GST_ERROR_OBJECT (resample, "Wrong caps");
- return FALSE;
- }
-
- gcd = _gcd (inrate, outrate);
- ratio_num = inrate / gcd;
- ratio_den = outrate / gcd;
-
- if (direction == GST_PAD_SINK) {
- gint fac = width / 8;
-
- /* asked to convert size of an incoming buffer */
- size /= fac;
- *othersize = (size * ratio_den + ratio_num - 1) / ratio_num;
- *othersize *= fac;
- size *= fac;
- } else {
- gint fac = width / 8;
-
- /* asked to convert size of an outgoing buffer */
- size /= fac;
- *othersize = (size * ratio_num + ratio_den - 1) / ratio_den;
- *othersize *= fac;
- size *= fac;
- }
-
- GST_LOG_OBJECT (resample, "transformed size %d to %d", size, *othersize);
-
- return ret;
-}
-
-static gboolean
-gst_speex_resample_set_caps (GstBaseTransform * base, GstCaps * incaps,
- GstCaps * outcaps)
-{
- gboolean ret;
- gint width = 0, inrate = 0, outrate = 0, channels = 0;
- gboolean fp;
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- GST_LOG ("incaps %" GST_PTR_FORMAT ", outcaps %"
- GST_PTR_FORMAT, incaps, outcaps);
-
- ret = gst_speex_resample_parse_caps (incaps, outcaps,
- &width, &channels, &inrate, &outrate, &fp);
-
- if (G_UNLIKELY (!ret))
- return FALSE;
-
- ret =
- gst_speex_resample_update_state (resample, width, channels, inrate,
- outrate, resample->quality, fp);
-
- if (G_UNLIKELY (!ret))
- return FALSE;
-
- /* save caps so we can short-circuit in the size_transform if the caps
- * are the same */
- gst_caps_replace (&resample->sinkcaps, incaps);
- gst_caps_replace (&resample->srccaps, outcaps);
-
- return TRUE;
-}
-
-#define GST_MAXINT24 (8388607)
-#define GST_MININT24 (-8388608)
-
-#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-#define GST_READ_UINT24 GST_READ_UINT24_LE
-#define GST_WRITE_UINT24 GST_WRITE_UINT24_LE
-#else
-#define GST_READ_UINT24 GST_READ_UINT24_BE
-#define GST_WRITE_UINT24 GST_WRITE_UINT24_BE
-#endif
-
-static void
-gst_speex_resample_convert_buffer (GstSpeexResample * resample,
- const guint8 * in, guint8 * out, guint len, gboolean inverse)
-{
- len *= resample->channels;
-
- if (inverse) {
- if (gst_speex_resample_use_int && resample->width == 8 && !resample->fp) {
- gint8 *o = (gint8 *) out;
- gint16 *i = (gint16 *) in;
- gint32 tmp;
-
- while (len) {
- tmp = *i + (G_MAXINT8 >> 1);
- *o = CLAMP (tmp >> 8, G_MININT8, G_MAXINT8);
- o++;
- i++;
- len--;
- }
- } else if (!gst_speex_resample_use_int && resample->width == 8
- && !resample->fp) {
- gint8 *o = (gint8 *) out;
- gfloat *i = (gfloat *) in;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = (gint8) CLAMP (tmp * G_MAXINT8 + 0.5, G_MININT8, G_MAXINT8);
- o++;
- i++;
- len--;
- }
- } else if (!gst_speex_resample_use_int && resample->width == 16
- && !resample->fp) {
- gint16 *o = (gint16 *) out;
- gfloat *i = (gfloat *) in;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = (gint16) CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
- o++;
- i++;
- len--;
- }
- } else if (resample->width == 24 && !resample->fp) {
- guint8 *o = (guint8 *) out;
- gdouble *i = (gdouble *) in;
- gdouble tmp;
-
- while (len) {
- tmp = *i;
- GST_WRITE_UINT24 (o, (gint32) CLAMP (tmp * GST_MAXINT24 + 0.5,
- GST_MININT24, GST_MAXINT24));
- o += 3;
- i++;
- len--;
- }
- } else if (resample->width == 32 && !resample->fp) {
- gint32 *o = (gint32 *) out;
- gdouble *i = (gdouble *) in;
- gdouble tmp;
-
- while (len) {
- tmp = *i;
- *o = (gint32) CLAMP (tmp * G_MAXINT32 + 0.5, G_MININT32, G_MAXINT32);
- o++;
- i++;
- len--;
- }
- } else {
- g_assert_not_reached ();
- }
- } else {
- if (gst_speex_resample_use_int && resample->width == 8 && !resample->fp) {
- gint8 *i = (gint8 *) in;
- gint16 *o = (gint16 *) out;
- gint32 tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp << 8;
- o++;
- i++;
- len--;
- }
- } else if (!gst_speex_resample_use_int && resample->width == 8
- && !resample->fp) {
- gint8 *i = (gint8 *) in;
- gfloat *o = (gfloat *) out;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp / G_MAXINT8;
- o++;
- i++;
- len--;
- }
- } else if (!gst_speex_resample_use_int && resample->width == 16
- && !resample->fp) {
- gint16 *i = (gint16 *) in;
- gfloat *o = (gfloat *) out;
- gfloat tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp / G_MAXINT16;
- o++;
- i++;
- len--;
- }
- } else if (resample->width == 24 && !resample->fp) {
- guint8 *i = (guint8 *) in;
- gdouble *o = (gdouble *) out;
- gdouble tmp;
- guint32 tmp2;
-
- while (len) {
- tmp2 = GST_READ_UINT24 (i);
- if (tmp2 & 0x00800000)
- tmp2 |= 0xff000000;
- tmp = (gint32) tmp2;
- *o = tmp / GST_MAXINT24;
- o++;
- i += 3;
- len--;
- }
- } else if (resample->width == 32 && !resample->fp) {
- gint32 *i = (gint32 *) in;
- gdouble *o = (gdouble *) out;
- gdouble tmp;
-
- while (len) {
- tmp = *i;
- *o = tmp / G_MAXINT32;
- o++;
- i++;
- len--;
- }
- } else {
- g_assert_not_reached ();
- }
- }
-}
-
-static void
-gst_speex_resample_push_drain (GstSpeexResample * resample)
-{
- GstBuffer *buf;
- GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
- GstFlowReturn res;
- gint outsize;
- guint out_len, out_processed;
- gint err;
- guint num, den, len;
- guint8 *outtmp = NULL;
- gboolean need_convert = FALSE;
-
- if (!resample->state)
- return;
-
- need_convert = (resample->funcs->width != resample->width);
-
- resample->funcs->get_ratio (resample->state, &num, &den);
-
- out_len = resample->funcs->get_input_latency (resample->state);
- out_len = out_processed = (out_len * den + num - 1) / num;
- outsize = (resample->width / 8) * out_len * resample->channels;
-
- if (need_convert) {
- guint outsize_tmp =
- (resample->funcs->width / 8) * out_len * resample->channels;
- if (outsize_tmp <= resample->tmp_out_size) {
- outtmp = resample->tmp_out;
- } else {
- resample->tmp_out_size = outsize_tmp;
- resample->tmp_out = outtmp = g_realloc (resample->tmp_out, outsize_tmp);
- }
- }
-
- res =
- gst_pad_alloc_buffer_and_set_caps (trans->srcpad, GST_BUFFER_OFFSET_NONE,
- outsize, GST_PAD_CAPS (trans->srcpad), &buf);
-
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (resample, "failed allocating buffer of %d bytes",
- outsize);
- return;
- }
-
- len = resample->funcs->get_input_latency (resample->state);
-
- err =
- resample->funcs->process (resample->state,
- NULL, &len, (need_convert) ? outtmp : GST_BUFFER_DATA (buf),
- &out_processed);
-
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
- GST_WARNING_OBJECT (resample, "Failed to process drain: %s",
- resample->funcs->strerror (err));
- gst_buffer_unref (buf);
- return;
- }
-
- if (G_UNLIKELY (out_processed == 0)) {
- GST_WARNING_OBJECT (resample, "Failed to get drain, dropping buffer");
- gst_buffer_unref (buf);
- return;
- }
-
- /* If we wrote more than allocated something is really wrong now
- * and we should better abort immediately */
- g_assert (out_len >= out_processed);
-
- if (need_convert)
- gst_speex_resample_convert_buffer (resample, outtmp, GST_BUFFER_DATA (buf),
- out_processed, TRUE);
-
- GST_BUFFER_DURATION (buf) =
- GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
- GST_BUFFER_SIZE (buf) =
- out_processed * resample->channels * (resample->width / 8);
-
- if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
- GST_BUFFER_OFFSET (buf) = resample->next_offset;
- GST_BUFFER_OFFSET_END (buf) = resample->next_offset + out_processed;
- GST_BUFFER_TIMESTAMP (buf) = resample->next_ts;
-
- resample->next_ts += GST_BUFFER_DURATION (buf);
- resample->next_offset += out_processed;
- }
-
- GST_LOG_OBJECT (resample,
- "Pushing drain buffer of %u bytes with timestamp %" GST_TIME_FORMAT
- " duration %" GST_TIME_FORMAT " offset %" G_GUINT64_FORMAT " offset_end %"
- G_GUINT64_FORMAT, GST_BUFFER_SIZE (buf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_BUFFER_OFFSET (buf),
- GST_BUFFER_OFFSET_END (buf));
-
- res = gst_pad_push (trans->srcpad, buf);
-
- if (G_UNLIKELY (res != GST_FLOW_OK))
- GST_WARNING_OBJECT (resample, "Failed to push drain: %s",
- gst_flow_get_name (res));
-
- return;
-}
-
-static gboolean
-gst_speex_resample_event (GstBaseTransform * base, GstEvent * event)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_FLUSH_START:
- break;
- case GST_EVENT_FLUSH_STOP:
- gst_speex_resample_reset_state (resample);
- resample->next_offset = -1;
- resample->next_ts = -1;
- resample->next_upstream_ts = -1;
- case GST_EVENT_NEWSEGMENT:
- gst_speex_resample_push_drain (resample);
- gst_speex_resample_reset_state (resample);
- resample->next_offset = -1;
- resample->next_ts = -1;
- resample->next_upstream_ts = -1;
- break;
- case GST_EVENT_EOS:{
- gst_speex_resample_push_drain (resample);
- gst_speex_resample_reset_state (resample);
- break;
- }
- default:
- break;
- }
-
- return parent_class->event (base, event);
-}
-
-static gboolean
-gst_speex_resample_check_discont (GstSpeexResample * resample,
- GstClockTime timestamp)
-{
- if (timestamp != GST_CLOCK_TIME_NONE &&
- resample->next_upstream_ts != GST_CLOCK_TIME_NONE &&
- timestamp != resample->next_upstream_ts) {
- /* Potentially a discontinuous buffer. However, it turns out that many
- * elements generate imperfect streams due to rounding errors, so we permit
- * a small error (up to one sample) without triggering a filter
- * flush/restart (if triggered incorrectly, this will be audible) */
- GstClockTimeDiff diff = timestamp - resample->next_upstream_ts;
-
- if (ABS (diff) > (GST_SECOND + resample->inrate - 1) / resample->inrate) {
- GST_WARNING_OBJECT (resample,
- "encountered timestamp discontinuity of %s%" GST_TIME_FORMAT,
- (diff < 0) ? "-" : "", GST_TIME_ARGS ((GstClockTime) ABS (diff)));
- return TRUE;
- }
- }
-
- return FALSE;
-}
-
-static GstFlowReturn
-gst_speex_resample_process (GstSpeexResample * resample, GstBuffer * inbuf,
- GstBuffer * outbuf)
-{
- guint32 in_len, in_processed;
- guint32 out_len, out_processed;
- gint err = RESAMPLER_ERR_SUCCESS;
- guint8 *in_tmp = NULL, *out_tmp = NULL;
- gboolean need_convert = (resample->funcs->width != resample->width);
-
- in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
- out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
-
- in_len /= (resample->width / 8);
- out_len /= (resample->width / 8);
-
- in_processed = in_len;
- out_processed = out_len;
-
- if (need_convert) {
- guint in_size_tmp =
- in_len * resample->channels * (resample->funcs->width / 8);
- guint out_size_tmp =
- out_len * resample->channels * (resample->funcs->width / 8);
-
- if (in_size_tmp <= resample->tmp_in_size) {
- in_tmp = resample->tmp_in;
- } else {
- resample->tmp_in = in_tmp = g_realloc (resample->tmp_in, in_size_tmp);
- resample->tmp_in_size = in_size_tmp;
- }
-
- gst_speex_resample_convert_buffer (resample, GST_BUFFER_DATA (inbuf),
- in_tmp, in_len, FALSE);
-
- if (out_size_tmp <= resample->tmp_out_size) {
- out_tmp = resample->tmp_out;
- } else {
- resample->tmp_out = out_tmp = g_realloc (resample->tmp_out, out_size_tmp);
- resample->tmp_out_size = out_size_tmp;
- }
- }
-
- if (need_convert) {
- err = resample->funcs->process (resample->state,
- in_tmp, &in_processed, out_tmp, &out_processed);
- } else {
- err = resample->funcs->process (resample->state,
- (const guint8 *) GST_BUFFER_DATA (inbuf), &in_processed,
- (guint8 *) GST_BUFFER_DATA (outbuf), &out_processed);
- }
-
- if (G_UNLIKELY (in_len != in_processed))
- GST_WARNING_OBJECT (resample, "Converted %d of %d input samples",
- in_processed, in_len);
-
- if (out_len != out_processed) {
- if (out_processed == 0) {
- GST_DEBUG_OBJECT (resample, "Converted to 0 samples, buffer dropped");
-
- return GST_BASE_TRANSFORM_FLOW_DROPPED;
- }
-
- /* If we wrote more than allocated something is really wrong now
- * and we should better abort immediately */
- g_assert (out_len >= out_processed);
- }
-
- if (G_UNLIKELY (err != RESAMPLER_ERR_SUCCESS)) {
- GST_ERROR_OBJECT (resample, "Failed to convert data: %s",
- resample->funcs->strerror (err));
- return GST_FLOW_ERROR;
- } else {
-
- if (need_convert)
- gst_speex_resample_convert_buffer (resample, out_tmp,
- GST_BUFFER_DATA (outbuf), out_processed, TRUE);
-
- GST_BUFFER_DURATION (outbuf) =
- GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
- GST_BUFFER_SIZE (outbuf) =
- out_processed * resample->channels * (resample->width / 8);
-
- if (GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
- GST_BUFFER_TIMESTAMP (outbuf) = resample->next_ts;
- GST_BUFFER_OFFSET (outbuf) = resample->next_offset;
- GST_BUFFER_OFFSET_END (outbuf) = resample->next_offset + out_processed;
-
- resample->next_ts += GST_BUFFER_DURATION (outbuf);
- resample->next_offset += out_processed;
- }
-
- GST_LOG_OBJECT (resample,
- "Converted to buffer of %u bytes with timestamp %" GST_TIME_FORMAT
- ", duration %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT
- ", offset_end %" G_GUINT64_FORMAT, GST_BUFFER_SIZE (outbuf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
- GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
-
- return GST_FLOW_OK;
- }
-}
-
-static GstFlowReturn
-gst_speex_resample_transform (GstBaseTransform * base, GstBuffer * inbuf,
- GstBuffer * outbuf)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (base);
- guint8 *data;
- gulong size;
- GstClockTime timestamp;
- guint outsamples, insamples;
- GstFlowReturn ret;
-
- if (resample->state == NULL) {
- if (G_UNLIKELY (!(resample->state =
- gst_speex_resample_init_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- resample->quality, resample->fp))))
- return GST_FLOW_ERROR;
-
- resample->funcs =
- gst_speex_resample_get_funcs (resample->width, resample->fp);
- }
-
- data = GST_BUFFER_DATA (inbuf);
- size = GST_BUFFER_SIZE (inbuf);
- timestamp = GST_BUFFER_TIMESTAMP (inbuf);
-
- GST_LOG_OBJECT (resample, "transforming buffer of %ld bytes, ts %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
- G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
- size, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
- GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
-
- /* check for timestamp discontinuities and flush/reset if needed */
- if (G_UNLIKELY (gst_speex_resample_check_discont (resample, timestamp)
- || GST_BUFFER_IS_DISCONT (inbuf))) {
- /* Flush internal samples */
- gst_speex_resample_reset_state (resample);
- /* Inform downstream element about discontinuity */
- resample->need_discont = TRUE;
- /* We want to recalculate the timestamps */
- resample->next_ts = -1;
- resample->next_upstream_ts = -1;
- resample->next_offset = -1;
- }
-
- insamples = GST_BUFFER_SIZE (inbuf) / resample->channels;
- insamples /= (resample->width / 8);
-
- outsamples = GST_BUFFER_SIZE (outbuf) / resample->channels;
- outsamples /= (resample->width / 8);
-
- if (GST_CLOCK_TIME_IS_VALID (timestamp)
- && !GST_CLOCK_TIME_IS_VALID (resample->next_ts)) {
- resample->next_ts = timestamp;
- resample->next_offset =
- GST_CLOCK_TIME_TO_FRAMES (timestamp, resample->outrate);
- }
-
- if (G_UNLIKELY (resample->need_discont)) {
- GST_DEBUG_OBJECT (resample, "marking this buffer with the DISCONT flag");
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- resample->need_discont = FALSE;
- }
-
- ret = gst_speex_resample_process (resample, inbuf, outbuf);
- if (G_UNLIKELY (ret != GST_FLOW_OK))
- return ret;
-
- if (GST_CLOCK_TIME_IS_VALID (timestamp)
- && !GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
- resample->next_upstream_ts = timestamp;
-
- if (GST_CLOCK_TIME_IS_VALID (resample->next_upstream_ts))
- resample->next_upstream_ts +=
- GST_FRAMES_TO_CLOCK_TIME (insamples, resample->inrate);
-
- return GST_FLOW_OK;
-}
-
-static gboolean
-gst_speex_resample_query (GstPad * pad, GstQuery * query)
-{
- GstSpeexResample *resample = GST_SPEEX_RESAMPLE (gst_pad_get_parent (pad));
- GstBaseTransform *trans = GST_BASE_TRANSFORM (resample);
- gboolean res = TRUE;
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_LATENCY:
- {
- GstClockTime min, max;
- gboolean live;
- guint64 latency;
- GstPad *peer;
- gint rate = resample->inrate;
- gint resampler_latency;
-
- if (resample->state)
- resampler_latency =
- resample->funcs->get_input_latency (resample->state);
- else
- resampler_latency = 0;
-
- if (gst_base_transform_is_passthrough (trans))
- resampler_latency = 0;
-
- if ((peer = gst_pad_get_peer (trans->sinkpad))) {
- if ((res = gst_pad_query (peer, query))) {
- gst_query_parse_latency (query, &live, &min, &max);
-
- GST_DEBUG_OBJECT (resample, "Peer latency: min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
-
- /* add our own latency */
- if (rate != 0 && resampler_latency != 0)
- latency =
- gst_util_uint64_scale (resampler_latency, GST_SECOND, rate);
- else
- latency = 0;
-
- GST_DEBUG_OBJECT (resample, "Our latency: %" GST_TIME_FORMAT,
- GST_TIME_ARGS (latency));
-
- min += latency;
- if (max != GST_CLOCK_TIME_NONE)
- max += latency;
-
- GST_DEBUG_OBJECT (resample, "Calculated total latency : min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
-
- gst_query_set_latency (query, live, min, max);
- }
- gst_object_unref (peer);
- }
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
- }
- gst_object_unref (resample);
- return res;
-}
-
-static const GstQueryType *
-gst_speex_resample_query_type (GstPad * pad)
-{
- static const GstQueryType types[] = {
- GST_QUERY_LATENCY,
- 0
- };
-
- return types;
-}
-
-static void
-gst_speex_resample_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstSpeexResample *resample;
-
- resample = GST_SPEEX_RESAMPLE (object);
-
- switch (prop_id) {
- case PROP_QUALITY:
- resample->quality = g_value_get_int (value);
- GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
-
- gst_speex_resample_update_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- resample->quality, resample->fp);
- break;
- case PROP_FILTER_LENGTH:{
- gint filter_length = g_value_get_int (value);
-
- if (filter_length <= 8)
- resample->quality = 0;
- else if (filter_length <= 16)
- resample->quality = 1;
- else if (filter_length <= 32)
- resample->quality = 2;
- else if (filter_length <= 48)
- resample->quality = 3;
- else if (filter_length <= 64)
- resample->quality = 4;
- else if (filter_length <= 80)
- resample->quality = 5;
- else if (filter_length <= 96)
- resample->quality = 6;
- else if (filter_length <= 128)
- resample->quality = 7;
- else if (filter_length <= 160)
- resample->quality = 8;
- else if (filter_length <= 192)
- resample->quality = 9;
- else
- resample->quality = 10;
-
- GST_DEBUG_OBJECT (resample, "new quality %d", resample->quality);
-
- gst_speex_resample_update_state (resample, resample->width,
- resample->channels, resample->inrate, resample->outrate,
- resample->quality, resample->fp);
- break;
- }
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_speex_resample_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstSpeexResample *resample;
-
- resample = GST_SPEEX_RESAMPLE (object);
-
- switch (prop_id) {
- case PROP_QUALITY:
- g_value_set_int (value, resample->quality);
- break;
- case PROP_FILTER_LENGTH:
- switch (resample->quality) {
- case 0:
- g_value_set_int (value, 8);
- break;
- case 1:
- g_value_set_int (value, 16);
- break;
- case 2:
- g_value_set_int (value, 32);
- break;
- case 3:
- g_value_set_int (value, 48);
- break;
- case 4:
- g_value_set_int (value, 64);
- break;
- case 5:
- g_value_set_int (value, 80);
- break;
- case 6:
- g_value_set_int (value, 96);
- break;
- case 7:
- g_value_set_int (value, 128);
- break;
- case 8:
- g_value_set_int (value, 160);
- break;
- case 9:
- g_value_set_int (value, 192);
- break;
- case 10:
- g_value_set_int (value, 256);
- break;
- }
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-#define BENCHMARK_SIZE 512
-
-static gboolean
-_benchmark_int_float (SpeexResamplerState * st)
-{
- gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
- gfloat in_tmp[BENCHMARK_SIZE], out_tmp[BENCHMARK_SIZE / 2];
- gint i;
- guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
-
- for (i = 0; i < BENCHMARK_SIZE; i++) {
- gfloat tmp = in[i];
- in_tmp[i] = tmp / G_MAXINT16;
- }
-
- resample_float_resampler_process_interleaved_float (st,
- (const guint8 *) in_tmp, &inlen, (guint8 *) out_tmp, &outlen);
-
- if (outlen == 0) {
- GST_ERROR ("Failed to use float resampler");
- return FALSE;
- }
-
- for (i = 0; i < outlen; i++) {
- gfloat tmp = out_tmp[i];
- out[i] = CLAMP (tmp * G_MAXINT16 + 0.5, G_MININT16, G_MAXINT16);
- }
-
- return TRUE;
-}
-
-static gboolean
-_benchmark_int_int (SpeexResamplerState * st)
-{
- gint16 in[BENCHMARK_SIZE] = { 0, }, out[BENCHMARK_SIZE / 2];
- guint32 inlen = BENCHMARK_SIZE, outlen = BENCHMARK_SIZE / 2;
-
- resample_int_resampler_process_interleaved_int (st, (const guint8 *) in,
- &inlen, (guint8 *) out, &outlen);
-
- if (outlen == 0) {
- GST_ERROR ("Failed to use int resampler");
- return FALSE;
- }
-
- return TRUE;
-}
-
-static gboolean
-_benchmark_integer_resampling (void)
-{
- OilProfile a, b;
- gdouble av, bv;
- SpeexResamplerState *sta, *stb;
-
- oil_profile_init (&a);
- oil_profile_init (&b);
-
- sta = resample_float_resampler_init (1, 48000, 24000, 4, NULL);
- if (sta == NULL) {
- GST_ERROR ("Failed to create float resampler state");
- return FALSE;
- }
-
- stb = resample_int_resampler_init (1, 48000, 24000, 4, NULL);
- if (stb == NULL) {
- resample_float_resampler_destroy (sta);
- GST_ERROR ("Failed to create int resampler state");
- return FALSE;
- }
-
- /* Warm up cache */
- if (!_benchmark_int_float (sta))
- goto error;
- if (!_benchmark_int_float (sta))
- goto error;
-
- /* Benchmark */
- oil_profile_start (&a);
- if (!_benchmark_int_float (sta))
- goto error;
- oil_profile_stop (&a);
-
- /* Warm up cache */
- if (!_benchmark_int_int (stb))
- goto error;
- if (!_benchmark_int_int (stb))
- goto error;
-
- /* Benchmark */
- oil_profile_start (&b);
- if (!_benchmark_int_int (stb))
- goto error;
- oil_profile_stop (&b);
-
- /* Handle results */
- oil_profile_get_ave_std (&a, &av, NULL);
- oil_profile_get_ave_std (&b, &bv, NULL);
-
- gst_speex_resample_use_int = (av > bv);
- resample_float_resampler_destroy (sta);
- resample_float_resampler_destroy (stb);
-
- if (av > bv)
- GST_DEBUG ("Using integer resampler if appropiate: %lf < %lf", bv, av);
- else
- GST_DEBUG ("Using float resampler for everything: %lf <= %lf", av, bv);
-
- return TRUE;
-
-error:
- resample_float_resampler_destroy (sta);
- resample_float_resampler_destroy (stb);
-
- return FALSE;
-}
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
- GST_DEBUG_CATEGORY_INIT (speex_resample_debug, "audioresample", 0,
- "audio resampling element");
-
- oil_init ();
-
- if (!_benchmark_integer_resampling ())
- return FALSE;
-
- if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
- GST_TYPE_SPEEX_RESAMPLE)) {
- return FALSE;
- }
-
- return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
- GST_VERSION_MINOR,
- "audioresample",
- "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
- GST_PACKAGE_ORIGIN);
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* Copyright (C) 2007-2008 Jean-Marc Valin
- Copyright (C) 2008 Thorvald Natvig
-
- File: resample.c
- Arbitrary resampling code
-
- Redistribution and use in source and binary forms, with or without
- modification, are permitted provided that the following conditions are
- met:
-
- 1. Redistributions of source code must retain the above copyright notice,
- this list of conditions and the following disclaimer.
-
- 2. Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in the
- documentation and/or other materials provided with the distribution.
-
- 3. The name of the author may not be used to endorse or promote products
- derived from this software without specific prior written permission.
-
- THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
- IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
- OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
- INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
- (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
- SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
- STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
- ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
- POSSIBILITY OF SUCH DAMAGE.
-*/
-
-/*
- The design goals of this code are:
- - Very fast algorithm
- - SIMD-friendly algorithm
- - Low memory requirement
- - Good *perceptual* quality (and not best SNR)
-
- Warning: This resampler is relatively new. Although I think I got rid of
- all the major bugs and I don't expect the API to change anymore, there
- may be something I've missed. So use with caution.
-
- This algorithm is based on this original resampling algorithm:
- Smith, Julius O. Digital Audio Resampling Home Page
- Center for Computer Research in Music and Acoustics (CCRMA),
- Stanford University, 2007.
- Web published at http://www-ccrma.stanford.edu/~jos/resample/.
-
- There is one main difference, though. This resampler uses cubic
- interpolation instead of linear interpolation in the above paper. This
- makes the table much smaller and makes it possible to compute that table
- on a per-stream basis. In turn, being able to tweak the table for each
- stream makes it possible to both reduce complexity on simple ratios
- (e.g. 2/3), and get rid of the rounding operations in the inner loop.
- The latter both reduces CPU time and makes the algorithm more SIMD-friendly.
-*/
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#ifdef OUTSIDE_SPEEX
-#include <stdlib.h>
-
-#include <glib.h>
-
-#define EXPORT G_GNUC_INTERNAL
-
-static inline void *
-speex_alloc (int size)
-{
- return g_malloc0 (size);
-}
-
-static inline void *
-speex_realloc (void *ptr, int size)
-{
- return g_realloc (ptr, size);
-}
-
-static inline void
-speex_free (void *ptr)
-{
- g_free (ptr);
-}
-
-#include "speex_resampler.h"
-#include "arch.h"
-#else /* OUTSIDE_SPEEX */
-
-#include "../include/speex/speex_resampler.h"
-#include "arch.h"
-#include "os_support.h"
-#endif /* OUTSIDE_SPEEX */
-
-#include <math.h>
-
-#ifndef M_PI
-#define M_PI 3.14159263
-#endif
-
-#ifdef FIXED_POINT
-#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
-#else
-#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
-#endif
-
-#define IMAX(a,b) ((a) > (b) ? (a) : (b))
-#define IMIN(a,b) ((a) < (b) ? (a) : (b))
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-#ifdef _USE_SSE
-#include "resample_sse.h"
-#endif
-
-/* Numer of elements to allocate on the stack */
-#ifdef VAR_ARRAYS
-#define FIXED_STACK_ALLOC 8192
-#else
-#define FIXED_STACK_ALLOC 1024
-#endif
-
-typedef int (*resampler_basic_func) (SpeexResamplerState *, spx_uint32_t,
- const spx_word16_t *, spx_uint32_t *, spx_word16_t *, spx_uint32_t *);
-
-struct SpeexResamplerState_
-{
- spx_uint32_t in_rate;
- spx_uint32_t out_rate;
- spx_uint32_t num_rate;
- spx_uint32_t den_rate;
-
- int quality;
- spx_uint32_t nb_channels;
- spx_uint32_t filt_len;
- spx_uint32_t mem_alloc_size;
- spx_uint32_t buffer_size;
- int int_advance;
- int frac_advance;
- float cutoff;
- spx_uint32_t oversample;
- int initialised;
- int started;
-
- /* These are per-channel */
- spx_int32_t *last_sample;
- spx_uint32_t *samp_frac_num;
- spx_uint32_t *magic_samples;
-
- spx_word16_t *mem;
- spx_word16_t *sinc_table;
- spx_uint32_t sinc_table_length;
- resampler_basic_func resampler_ptr;
-
- int in_stride;
- int out_stride;
-};
-
-static double kaiser12_table[68] = {
- 0.99859849, 1.00000000, 0.99859849, 0.99440475, 0.98745105, 0.97779076,
- 0.96549770, 0.95066529, 0.93340547, 0.91384741, 0.89213598, 0.86843014,
- 0.84290116, 0.81573067, 0.78710866, 0.75723148, 0.72629970, 0.69451601,
- 0.66208321, 0.62920216, 0.59606986, 0.56287762, 0.52980938, 0.49704014,
- 0.46473455, 0.43304576, 0.40211431, 0.37206735, 0.34301800, 0.31506490,
- 0.28829195, 0.26276832, 0.23854851, 0.21567274, 0.19416736, 0.17404546,
- 0.15530766, 0.13794294, 0.12192957, 0.10723616, 0.09382272, 0.08164178,
- 0.07063950, 0.06075685, 0.05193064, 0.04409466, 0.03718069, 0.03111947,
- 0.02584161, 0.02127838, 0.01736250, 0.01402878, 0.01121463, 0.00886058,
- 0.00691064, 0.00531256, 0.00401805, 0.00298291, 0.00216702, 0.00153438,
- 0.00105297, 0.00069463, 0.00043489, 0.00025272, 0.00013031, 0.0000527734,
- 0.00001000, 0.00000000
-};
-
-/*
-static double kaiser12_table[36] = {
- 0.99440475, 1.00000000, 0.99440475, 0.97779076, 0.95066529, 0.91384741,
- 0.86843014, 0.81573067, 0.75723148, 0.69451601, 0.62920216, 0.56287762,
- 0.49704014, 0.43304576, 0.37206735, 0.31506490, 0.26276832, 0.21567274,
- 0.17404546, 0.13794294, 0.10723616, 0.08164178, 0.06075685, 0.04409466,
- 0.03111947, 0.02127838, 0.01402878, 0.00886058, 0.00531256, 0.00298291,
- 0.00153438, 0.00069463, 0.00025272, 0.0000527734, 0.00000500, 0.00000000};
-*/
-static double kaiser10_table[36] = {
- 0.99537781, 1.00000000, 0.99537781, 0.98162644, 0.95908712, 0.92831446,
- 0.89005583, 0.84522401, 0.79486424, 0.74011713, 0.68217934, 0.62226347,
- 0.56155915, 0.50119680, 0.44221549, 0.38553619, 0.33194107, 0.28205962,
- 0.23636152, 0.19515633, 0.15859932, 0.12670280, 0.09935205, 0.07632451,
- 0.05731132, 0.04193980, 0.02979584, 0.02044510, 0.01345224, 0.00839739,
- 0.00488951, 0.00257636, 0.00115101, 0.00035515, 0.00000000, 0.00000000
-};
-
-static double kaiser8_table[36] = {
- 0.99635258, 1.00000000, 0.99635258, 0.98548012, 0.96759014, 0.94302200,
- 0.91223751, 0.87580811, 0.83439927, 0.78875245, 0.73966538, 0.68797126,
- 0.63451750, 0.58014482, 0.52566725, 0.47185369, 0.41941150, 0.36897272,
- 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758,
- 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490,
- 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000
-};
-
-static double kaiser6_table[36] = {
- 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003,
- 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565,
- 0.71712752, 0.67172623, 0.62508937, 0.57774224, 0.53019925, 0.48295561,
- 0.43647969, 0.39120616, 0.34752997, 0.30580127, 0.26632152, 0.22934058,
- 0.19505503, 0.16360756, 0.13508755, 0.10953262, 0.08693120, 0.06722600,
- 0.05031820, 0.03607231, 0.02432151, 0.01487334, 0.00752000, 0.00000000
-};
-
-struct FuncDef
-{
- double *table;
- int oversample;
-};
-
-static struct FuncDef _KAISER12 = { kaiser12_table, 64 };
-
-#define KAISER12 (&_KAISER12)
-/*static struct FuncDef _KAISER12 = {kaiser12_table, 32};
-#define KAISER12 (&_KAISER12)*/
-static struct FuncDef _KAISER10 = { kaiser10_table, 32 };
-
-#define KAISER10 (&_KAISER10)
-static struct FuncDef _KAISER8 = { kaiser8_table, 32 };
-
-#define KAISER8 (&_KAISER8)
-static struct FuncDef _KAISER6 = { kaiser6_table, 32 };
-
-#define KAISER6 (&_KAISER6)
-
-struct QualityMapping
-{
- int base_length;
- int oversample;
- float downsample_bandwidth;
- float upsample_bandwidth;
- struct FuncDef *window_func;
-};
-
-
-/* This table maps conversion quality to internal parameters. There are two
- reasons that explain why the up-sampling bandwidth is larger than the
- down-sampling bandwidth:
- 1) When up-sampling, we can assume that the spectrum is already attenuated
- close to the Nyquist rate (from an A/D or a previous resampling filter)
- 2) Any aliasing that occurs very close to the Nyquist rate will be masked
- by the sinusoids/noise just below the Nyquist rate (guaranteed only for
- up-sampling).
-*/
-static const struct QualityMapping quality_map[11] = {
- {8, 4, 0.830f, 0.860f, KAISER6}, /* Q0 */
- {16, 4, 0.850f, 0.880f, KAISER6}, /* Q1 */
- {32, 4, 0.882f, 0.910f, KAISER6}, /* Q2 *//* 82.3% cutoff ( ~60 dB stop) 6 */
- {48, 8, 0.895f, 0.917f, KAISER8}, /* Q3 *//* 84.9% cutoff ( ~80 dB stop) 8 */
- {64, 8, 0.921f, 0.940f, KAISER8}, /* Q4 *//* 88.7% cutoff ( ~80 dB stop) 8 */
- {80, 16, 0.922f, 0.940f, KAISER10}, /* Q5 *//* 89.1% cutoff (~100 dB stop) 10 */
- {96, 16, 0.940f, 0.945f, KAISER10}, /* Q6 *//* 91.5% cutoff (~100 dB stop) 10 */
- {128, 16, 0.950f, 0.950f, KAISER10}, /* Q7 *//* 93.1% cutoff (~100 dB stop) 10 */
- {160, 16, 0.960f, 0.960f, KAISER10}, /* Q8 *//* 94.5% cutoff (~100 dB stop) 10 */
- {192, 32, 0.968f, 0.968f, KAISER12}, /* Q9 *//* 95.5% cutoff (~100 dB stop) 10 */
- {256, 32, 0.975f, 0.975f, KAISER12}, /* Q10 *//* 96.6% cutoff (~100 dB stop) 10 */
-};
-
-/*8,24,40,56,80,104,128,160,200,256,320*/
-#ifdef DOUBLE_PRECISION
-static double
-compute_func (double x, struct FuncDef *func)
-{
- double y, frac;
-#else
-static double
-compute_func (float x, struct FuncDef *func)
-{
- float y, frac;
-#endif
- double interp[4];
- int ind;
- y = x * func->oversample;
- ind = (int) floor (y);
- frac = (y - ind);
- /* CSE with handle the repeated powers */
- interp[3] = -0.1666666667 * frac + 0.1666666667 * (frac * frac * frac);
- interp[2] = frac + 0.5 * (frac * frac) - 0.5 * (frac * frac * frac);
- /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
- interp[0] =
- -0.3333333333 * frac + 0.5 * (frac * frac) -
- 0.1666666667 * (frac * frac * frac);
- /* Just to make sure we don't have rounding problems */
- interp[1] = 1.f - interp[3] - interp[2] - interp[0];
-
- /*sum = frac*accum[1] + (1-frac)*accum[2]; */
- return interp[0] * func->table[ind] + interp[1] * func->table[ind + 1] +
- interp[2] * func->table[ind + 2] + interp[3] * func->table[ind + 3];
-}
-
-#if 0
-#include <stdio.h>
-int
-main (int argc, char **argv)
-{
- int i;
- for (i = 0; i < 256; i++) {
- printf ("%f\n", compute_func (i / 256., KAISER12));
- }
- return 0;
-}
-#endif
-
-#ifdef FIXED_POINT
-/* The slow way of computing a sinc for the table. Should improve that some day */
-static spx_word16_t
-sinc (float cutoff, float x, int N, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- float xx = x * cutoff;
- if (fabs (x) < 1e-6f)
- return WORD2INT (32768. * cutoff);
- else if (fabs (x) > .5f * N)
- return 0;
- /*FIXME: Can it really be any slower than this? */
- return WORD2INT (32768. * cutoff * sin (M_PI * xx) / (M_PI * xx) *
- compute_func (fabs (2. * x / N), window_func));
-}
-#else
-/* The slow way of computing a sinc for the table. Should improve that some day */
-#ifdef DOUBLE_PRECISION
-static spx_word16_t
-sinc (double cutoff, double x, int N, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- double xx = x * cutoff;
-#else
-static spx_word16_t
-sinc (float cutoff, float x, int N, struct FuncDef *window_func)
-{
- /*fprintf (stderr, "%f ", x); */
- float xx = x * cutoff;
-#endif
- if (fabs (x) < 1e-6)
- return cutoff;
- else if (fabs (x) > .5 * N)
- return 0;
- /*FIXME: Can it really be any slower than this? */
- return cutoff * sin (M_PI * xx) / (M_PI * xx) * compute_func (fabs (2. * x /
- N), window_func);
-}
-#endif
-
-#ifdef FIXED_POINT
-static void
-cubic_coef (spx_word16_t x, spx_word16_t interp[4])
-{
- /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
- but I know it's MMSE-optimal on a sinc */
- spx_word16_t x2, x3;
- x2 = MULT16_16_P15 (x, x);
- x3 = MULT16_16_P15 (x, x2);
- interp[0] =
- PSHR32 (MULT16_16 (QCONST16 (-0.16667f, 15),
- x) + MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
- interp[1] =
- EXTRACT16 (EXTEND32 (x) + SHR32 (SUB32 (EXTEND32 (x2), EXTEND32 (x3)),
- 1));
- interp[3] =
- PSHR32 (MULT16_16 (QCONST16 (-0.33333f, 15),
- x) + MULT16_16 (QCONST16 (.5f, 15),
- x2) - MULT16_16 (QCONST16 (0.16667f, 15), x3), 15);
- /* Just to make sure we don't have rounding problems */
- interp[2] = Q15_ONE - interp[0] - interp[1] - interp[3];
- if (interp[2] < 32767)
- interp[2] += 1;
-}
-#else
-static void
-cubic_coef (spx_word16_t frac, spx_word16_t interp[4])
-{
- /* Compute interpolation coefficients. I'm not sure whether this corresponds to cubic interpolation
- but I know it's MMSE-optimal on a sinc */
- interp[0] = -0.16667f * frac + 0.16667f * frac * frac * frac;
- interp[1] = frac + 0.5f * frac * frac - 0.5f * frac * frac * frac;
- /*interp[2] = 1.f - 0.5f*frac - frac*frac + 0.5f*frac*frac*frac; */
- interp[3] =
- -0.33333f * frac + 0.5f * frac * frac - 0.16667f * frac * frac * frac;
- /* Just to make sure we don't have rounding problems */
- interp[2] = 1. - interp[0] - interp[1] - interp[3];
-}
-#endif
-
-#ifndef DOUBLE_PRECISION
-static int
-resampler_basic_direct_single (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const spx_word16_t *sinc_table = st->sinc_table;
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- spx_word32_t sum;
- int j;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *sinc = &sinc_table[samp_frac_num * N];
- const spx_word16_t *iptr = &in[last_sample];
-
-#ifndef OVERRIDE_INNER_PRODUCT_SINGLE
- float accum[4] = { 0, 0, 0, 0 };
-
- for (j = 0; j < N; j += 4) {
- accum[0] += sinc[j] * iptr[j];
- accum[1] += sinc[j + 1] * iptr[j + 1];
- accum[2] += sinc[j + 2] * iptr[j + 2];
- accum[3] += sinc[j + 3] * iptr[j + 3];
- }
- sum = accum[0] + accum[1] + accum[2] + accum[3];
-#else
- sum = inner_product_single (sinc, iptr, N);
-#endif
-
- out[out_stride * out_sample++] = PSHR32 (sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-#ifdef FIXED_POINT
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int
-resampler_basic_direct_double (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const spx_word16_t *sinc_table = st->sinc_table;
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- double sum;
- int j;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *sinc = &sinc_table[samp_frac_num * N];
- const spx_word16_t *iptr = &in[last_sample];
-
-#ifndef OVERRIDE_INNER_PRODUCT_DOUBLE
- double accum[4] = { 0, 0, 0, 0 };
-
- for (j = 0; j < N; j += 4) {
- accum[0] += sinc[j] * iptr[j];
- accum[1] += sinc[j + 1] * iptr[j + 1];
- accum[2] += sinc[j + 2] * iptr[j + 2];
- accum[3] += sinc[j + 3] * iptr[j + 3];
- }
- sum = accum[0] + accum[1] + accum[2] + accum[3];
-#else
- sum = inner_product_double (sinc, iptr, N);
-#endif
-
- out[out_stride * out_sample++] = PSHR32 (sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-#ifndef DOUBLE_PRECISION
-static int
-resampler_basic_interpolate_single (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- int j;
- spx_word32_t sum;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *iptr = &in[last_sample];
-
- const int offset = samp_frac_num * st->oversample / st->den_rate;
-#ifdef FIXED_POINT
- const spx_word16_t frac =
- PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
- st->den_rate);
-#else
- const spx_word16_t frac =
- ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
- st->den_rate;
-#endif
- spx_word16_t interp[4];
-
-
-#ifndef OVERRIDE_INTERPOLATE_PRODUCT_SINGLE
- spx_word32_t accum[4] = { 0, 0, 0, 0 };
-
- for (j = 0; j < N; j++) {
- const spx_word16_t curr_in = iptr[j];
- accum[0] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
- accum[1] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
- accum[2] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset]);
- accum[3] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
- }
-
- cubic_coef (frac, interp);
- sum =
- MULT16_32_Q15 (interp[0], accum[0]) + MULT16_32_Q15 (interp[1],
- accum[1]) + MULT16_32_Q15 (interp[2],
- accum[2]) + MULT16_32_Q15 (interp[3], accum[3]);
-#else
- cubic_coef (frac, interp);
- sum =
- interpolate_product_single (iptr,
- st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample,
- interp);
-#endif
-
- out[out_stride * out_sample++] = PSHR32 (sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-#ifdef FIXED_POINT
-#else
-/* This is the same as the previous function, except with a double-precision accumulator */
-static int
-resampler_basic_interpolate_double (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_word16_t * in, spx_uint32_t * in_len,
- spx_word16_t * out, spx_uint32_t * out_len)
-{
- const int N = st->filt_len;
- int out_sample = 0;
- int last_sample = st->last_sample[channel_index];
- spx_uint32_t samp_frac_num = st->samp_frac_num[channel_index];
- const int out_stride = st->out_stride;
- const int int_advance = st->int_advance;
- const int frac_advance = st->frac_advance;
- const spx_uint32_t den_rate = st->den_rate;
- int j;
- spx_word32_t sum;
-
- while (!(last_sample >= (spx_int32_t) * in_len
- || out_sample >= (spx_int32_t) * out_len)) {
- const spx_word16_t *iptr = &in[last_sample];
-
- const int offset = samp_frac_num * st->oversample / st->den_rate;
-#ifdef FIXED_POINT
- const spx_word16_t frac =
- PDIV32 (SHL32 ((samp_frac_num * st->oversample) % st->den_rate, 15),
- st->den_rate);
-#else
-#ifdef DOUBLE_PRECISION
- const spx_word16_t frac =
- ((double) ((samp_frac_num * st->oversample) % st->den_rate)) /
- st->den_rate;
-#else
- const spx_word16_t frac =
- ((float) ((samp_frac_num * st->oversample) % st->den_rate)) /
- st->den_rate;
-#endif
-#endif
- spx_word16_t interp[4];
-
-
-#ifndef OVERRIDE_INTERPOLATE_PRODUCT_DOUBLE
- double accum[4] = { 0, 0, 0, 0 };
-
- for (j = 0; j < N; j++) {
- const double curr_in = iptr[j];
- accum[0] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 2]);
- accum[1] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset - 1]);
- accum[2] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset]);
- accum[3] +=
- MULT16_16 (curr_in,
- st->sinc_table[4 + (j + 1) * st->oversample - offset + 1]);
- }
-
- cubic_coef (frac, interp);
- sum =
- MULT16_32_Q15 (interp[0], accum[0]) + MULT16_32_Q15 (interp[1],
- accum[1]) + MULT16_32_Q15 (interp[2],
- accum[2]) + MULT16_32_Q15 (interp[3], accum[3]);
-#else
- cubic_coef (frac, interp);
- sum =
- interpolate_product_double (iptr,
- st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample,
- interp);
-#endif
-
- out[out_stride * out_sample++] = PSHR32 (sum, 15);
- last_sample += int_advance;
- samp_frac_num += frac_advance;
- if (samp_frac_num >= den_rate) {
- samp_frac_num -= den_rate;
- last_sample++;
- }
- }
-
- st->last_sample[channel_index] = last_sample;
- st->samp_frac_num[channel_index] = samp_frac_num;
- return out_sample;
-}
-#endif
-
-static void
-update_filter (SpeexResamplerState * st)
-{
- spx_uint32_t old_length;
-
- old_length = st->filt_len;
- st->oversample = quality_map[st->quality].oversample;
- st->filt_len = quality_map[st->quality].base_length;
-
- if (st->num_rate > st->den_rate) {
- /* down-sampling */
- st->cutoff =
- quality_map[st->quality].downsample_bandwidth * st->den_rate /
- st->num_rate;
- /* FIXME: divide the numerator and denominator by a certain amount if they're too large */
- st->filt_len = st->filt_len * st->num_rate / st->den_rate;
- /* Round down to make sure we have a multiple of 4 */
- st->filt_len &= (~0x3);
- if (2 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (4 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (8 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (16 * st->den_rate < st->num_rate)
- st->oversample >>= 1;
- if (st->oversample < 1)
- st->oversample = 1;
- } else {
- /* up-sampling */
- st->cutoff = quality_map[st->quality].upsample_bandwidth;
- }
-
- /* Choose the resampling type that requires the least amount of memory */
- if (st->den_rate <= st->oversample) {
- spx_uint32_t i;
- if (!st->sinc_table)
- st->sinc_table =
- (spx_word16_t *) speex_alloc (st->filt_len * st->den_rate *
- sizeof (spx_word16_t));
- else if (st->sinc_table_length < st->filt_len * st->den_rate) {
- st->sinc_table =
- (spx_word16_t *) speex_realloc (st->sinc_table,
- st->filt_len * st->den_rate * sizeof (spx_word16_t));
- st->sinc_table_length = st->filt_len * st->den_rate;
- }
- for (i = 0; i < st->den_rate; i++) {
- spx_int32_t j;
- for (j = 0; j < st->filt_len; j++) {
- st->sinc_table[i * st->filt_len + j] =
- sinc (st->cutoff, ((j - (spx_int32_t) st->filt_len / 2 + 1) -
-#ifdef DOUBLE_PRECISION
- ((double) i) / st->den_rate), st->filt_len,
-#else
- ((float) i) / st->den_rate), st->filt_len,
-#endif
- quality_map[st->quality].window_func);
- }
- }
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_direct_single;
-#else
-#ifdef DOUBLE_PRECISION
- st->resampler_ptr = resampler_basic_direct_double;
-#else
- if (st->quality > 8)
- st->resampler_ptr = resampler_basic_direct_double;
- else
- st->resampler_ptr = resampler_basic_direct_single;
-#endif
-#endif
- /*fprintf (stderr, "resampler uses direct sinc table and normalised cutoff %f\n", cutoff); */
- } else {
- spx_int32_t i;
- if (!st->sinc_table)
- st->sinc_table =
- (spx_word16_t *) speex_alloc ((st->filt_len * st->oversample +
- 8) * sizeof (spx_word16_t));
- else if (st->sinc_table_length < st->filt_len * st->oversample + 8) {
- st->sinc_table =
- (spx_word16_t *) speex_realloc (st->sinc_table,
- (st->filt_len * st->oversample + 8) * sizeof (spx_word16_t));
- st->sinc_table_length = st->filt_len * st->oversample + 8;
- }
- for (i = -4; i < (spx_int32_t) (st->oversample * st->filt_len + 4); i++)
- st->sinc_table[i + 4] =
-#ifdef DOUBLE_PRECISION
- sinc (st->cutoff, (i / (double) st->oversample - st->filt_len / 2),
-#else
- sinc (st->cutoff, (i / (float) st->oversample - st->filt_len / 2),
-#endif
- st->filt_len, quality_map[st->quality].window_func);
-#ifdef FIXED_POINT
- st->resampler_ptr = resampler_basic_interpolate_single;
-#else
-#ifdef DOUBLE_PRECISION
- st->resampler_ptr = resampler_basic_interpolate_double;
-#else
- if (st->quality > 8)
- st->resampler_ptr = resampler_basic_interpolate_double;
- else
- st->resampler_ptr = resampler_basic_interpolate_single;
-#endif
-#endif
- /*fprintf (stderr, "resampler uses interpolated sinc table and normalised cutoff %f\n", cutoff); */
- }
- st->int_advance = st->num_rate / st->den_rate;
- st->frac_advance = st->num_rate % st->den_rate;
-
-
- /* Here's the place where we update the filter memory to take into account
- the change in filter length. It's probably the messiest part of the code
- due to handling of lots of corner cases. */
- if (!st->mem) {
- spx_uint32_t i;
- st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
- st->mem =
- (spx_word16_t *) speex_alloc (st->nb_channels * st->mem_alloc_size *
- sizeof (spx_word16_t));
- for (i = 0; i < st->nb_channels * st->mem_alloc_size; i++)
- st->mem[i] = 0;
- /*speex_warning("init filter"); */
- } else if (!st->started) {
- spx_uint32_t i;
- st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
- st->mem =
- (spx_word16_t *) speex_realloc (st->mem,
- st->nb_channels * st->mem_alloc_size * sizeof (spx_word16_t));
- for (i = 0; i < st->nb_channels * st->mem_alloc_size; i++)
- st->mem[i] = 0;
- /*speex_warning("reinit filter"); */
- } else if (st->filt_len > old_length) {
- spx_int32_t i;
- /* Increase the filter length */
- /*speex_warning("increase filter size"); */
- int old_alloc_size = st->mem_alloc_size;
- if ((st->filt_len - 1 + st->buffer_size) > st->mem_alloc_size) {
- st->mem_alloc_size = st->filt_len - 1 + st->buffer_size;
- st->mem =
- (spx_word16_t *) speex_realloc (st->mem,
- st->nb_channels * st->mem_alloc_size * sizeof (spx_word16_t));
- }
- for (i = st->nb_channels - 1; i >= 0; i--) {
- spx_int32_t j;
- spx_uint32_t olen = old_length;
- /*if (st->magic_samples[i]) */
- {
- /* Try and remove the magic samples as if nothing had happened */
-
- /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */
- olen = old_length + 2 * st->magic_samples[i];
- for (j = old_length - 2 + st->magic_samples[i]; j >= 0; j--)
- st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]] =
- st->mem[i * old_alloc_size + j];
- for (j = 0; j < st->magic_samples[i]; j++)
- st->mem[i * st->mem_alloc_size + j] = 0;
- st->magic_samples[i] = 0;
- }
- if (st->filt_len > olen) {
- /* If the new filter length is still bigger than the "augmented" length */
- /* Copy data going backward */
- for (j = 0; j < olen - 1; j++)
- st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] =
- st->mem[i * st->mem_alloc_size + (olen - 2 - j)];
- /* Then put zeros for lack of anything better */
- for (; j < st->filt_len - 1; j++)
- st->mem[i * st->mem_alloc_size + (st->filt_len - 2 - j)] = 0;
- /* Adjust last_sample */
- st->last_sample[i] += (st->filt_len - olen) / 2;
- } else {
- /* Put back some of the magic! */
- st->magic_samples[i] = (olen - st->filt_len) / 2;
- for (j = 0; j < st->filt_len - 1 + st->magic_samples[i]; j++)
- st->mem[i * st->mem_alloc_size + j] =
- st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]];
- }
- }
- } else if (st->filt_len < old_length) {
- spx_uint32_t i;
- /* Reduce filter length, this a bit tricky. We need to store some of the memory as "magic"
- samples so they can be used directly as input the next time(s) */
- for (i = 0; i < st->nb_channels; i++) {
- spx_uint32_t j;
- spx_uint32_t old_magic = st->magic_samples[i];
- st->magic_samples[i] = (old_length - st->filt_len) / 2;
- /* We must copy some of the memory that's no longer used */
- /* Copy data going backward */
- for (j = 0; j < st->filt_len - 1 + st->magic_samples[i] + old_magic; j++)
- st->mem[i * st->mem_alloc_size + j] =
- st->mem[i * st->mem_alloc_size + j + st->magic_samples[i]];
- st->magic_samples[i] += old_magic;
- }
- }
-
-}
-
-EXPORT SpeexResamplerState *
-speex_resampler_init (spx_uint32_t nb_channels, spx_uint32_t in_rate,
- spx_uint32_t out_rate, int quality, int *err)
-{
- return speex_resampler_init_frac (nb_channels, in_rate, out_rate, in_rate,
- out_rate, quality, err);
-}
-
-EXPORT SpeexResamplerState *
-speex_resampler_init_frac (spx_uint32_t nb_channels, spx_uint32_t ratio_num,
- spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate,
- int quality, int *err)
-{
- spx_uint32_t i;
- SpeexResamplerState *st;
- if (quality > 10 || quality < 0) {
- if (err)
- *err = RESAMPLER_ERR_INVALID_ARG;
- return NULL;
- }
- st = (SpeexResamplerState *) speex_alloc (sizeof (SpeexResamplerState));
- st->initialised = 0;
- st->started = 0;
- st->in_rate = 0;
- st->out_rate = 0;
- st->num_rate = 0;
- st->den_rate = 0;
- st->quality = -1;
- st->sinc_table_length = 0;
- st->mem_alloc_size = 0;
- st->filt_len = 0;
- st->mem = 0;
- st->resampler_ptr = 0;
-
- st->cutoff = 1.f;
- st->nb_channels = nb_channels;
- st->in_stride = 1;
- st->out_stride = 1;
-
-#ifdef FIXED_POINT
- st->buffer_size = 160;
-#else
- st->buffer_size = 160;
-#endif
-
- /* Per channel data */
- st->last_sample = (spx_int32_t *) speex_alloc (nb_channels * sizeof (int));
- st->magic_samples = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
- st->samp_frac_num = (spx_uint32_t *) speex_alloc (nb_channels * sizeof (int));
- for (i = 0; i < nb_channels; i++) {
- st->last_sample[i] = 0;
- st->magic_samples[i] = 0;
- st->samp_frac_num[i] = 0;
- }
-
- speex_resampler_set_quality (st, quality);
- speex_resampler_set_rate_frac (st, ratio_num, ratio_den, in_rate, out_rate);
-
-
- update_filter (st);
-
- st->initialised = 1;
- if (err)
- *err = RESAMPLER_ERR_SUCCESS;
-
- return st;
-}
-
-EXPORT void
-speex_resampler_destroy (SpeexResamplerState * st)
-{
- speex_free (st->mem);
- speex_free (st->sinc_table);
- speex_free (st->last_sample);
- speex_free (st->magic_samples);
- speex_free (st->samp_frac_num);
- speex_free (st);
-}
-
-static int
-speex_resampler_process_native (SpeexResamplerState * st,
- spx_uint32_t channel_index, spx_uint32_t * in_len, spx_word16_t * out,
- spx_uint32_t * out_len)
-{
- int j = 0;
- const int N = st->filt_len;
- int out_sample = 0;
- spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
- spx_uint32_t ilen;
-
- st->started = 1;
-
- /* Call the right resampler through the function ptr */
- out_sample = st->resampler_ptr (st, channel_index, mem, in_len, out, out_len);
-
- if (st->last_sample[channel_index] < (spx_int32_t) * in_len)
- *in_len = st->last_sample[channel_index];
- *out_len = out_sample;
- st->last_sample[channel_index] -= *in_len;
-
- ilen = *in_len;
-
- for (j = 0; j < N - 1; ++j)
- mem[j] = mem[j + ilen];
-
- return RESAMPLER_ERR_SUCCESS;
-}
-
-static int
-speex_resampler_magic (SpeexResamplerState * st, spx_uint32_t channel_index,
- spx_word16_t ** out, spx_uint32_t out_len)
-{
- spx_uint32_t tmp_in_len = st->magic_samples[channel_index];
- spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size;
- const int N = st->filt_len;
-
- speex_resampler_process_native (st, channel_index, &tmp_in_len, *out,
- &out_len);
-
- st->magic_samples[channel_index] -= tmp_in_len;
-
- /* If we couldn't process all "magic" input samples, save the rest for next time */
- if (st->magic_samples[channel_index]) {
- spx_uint32_t i;
- for (i = 0; i < st->magic_samples[channel_index]; i++)
- mem[N - 1 + i] = mem[N - 1 + i + tmp_in_len];
- }
- *out += out_len * st->out_stride;
- return out_len;
-}
-
-#ifdef FIXED_POINT
-EXPORT int
-speex_resampler_process_int (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len,
- spx_int16_t * out, spx_uint32_t * out_len)
-#else
-#ifdef DOUBLE_PRECISION
-EXPORT int
-speex_resampler_process_float (SpeexResamplerState * st,
- spx_uint32_t channel_index, const double *in, spx_uint32_t * in_len,
- double *out, spx_uint32_t * out_len)
-#else
-EXPORT int
-speex_resampler_process_float (SpeexResamplerState * st,
- spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len,
- float *out, spx_uint32_t * out_len)
-#endif
-#endif
-{
- int j;
- spx_uint32_t ilen = *in_len;
- spx_uint32_t olen = *out_len;
- spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
- const int filt_offs = st->filt_len - 1;
- const spx_uint32_t xlen = st->mem_alloc_size - filt_offs;
- const int istride = st->in_stride;
-
- if (st->magic_samples[channel_index])
- olen -= speex_resampler_magic (st, channel_index, &out, olen);
- if (!st->magic_samples[channel_index]) {
- while (ilen && olen) {
- spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
- spx_uint32_t ochunk = olen;
-
- if (in) {
- for (j = 0; j < ichunk; ++j)
- x[j + filt_offs] = in[j * istride];
- } else {
- for (j = 0; j < ichunk; ++j)
- x[j + filt_offs] = 0;
- }
- speex_resampler_process_native (st, channel_index, &ichunk, out, &ochunk);
- ilen -= ichunk;
- olen -= ochunk;
- out += ochunk * st->out_stride;
- if (in)
- in += ichunk * istride;
- }
- }
- *in_len -= ilen;
- *out_len -= olen;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-#ifdef FIXED_POINT
-EXPORT int
-speex_resampler_process_float (SpeexResamplerState * st,
- spx_uint32_t channel_index, const float *in, spx_uint32_t * in_len,
- float *out, spx_uint32_t * out_len)
-#else
-EXPORT int
-speex_resampler_process_int (SpeexResamplerState * st,
- spx_uint32_t channel_index, const spx_int16_t * in, spx_uint32_t * in_len,
- spx_int16_t * out, spx_uint32_t * out_len)
-#endif
-{
- int j;
- const int istride_save = st->in_stride;
- const int ostride_save = st->out_stride;
- spx_uint32_t ilen = *in_len;
- spx_uint32_t olen = *out_len;
- spx_word16_t *x = st->mem + channel_index * st->mem_alloc_size;
- const spx_uint32_t xlen = st->mem_alloc_size - (st->filt_len - 1);
-#ifdef VAR_ARRAYS
- const unsigned int ylen =
- (olen < FIXED_STACK_ALLOC) ? olen : FIXED_STACK_ALLOC;
- VARDECL (spx_word16_t * ystack);
- ALLOC (ystack, ylen, spx_word16_t);
-#else
- const unsigned int ylen = FIXED_STACK_ALLOC;
- spx_word16_t ystack[FIXED_STACK_ALLOC];
-#endif
-
- st->out_stride = 1;
-
- while (ilen && olen) {
- spx_word16_t *y = ystack;
- spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen;
- spx_uint32_t ochunk = (olen > ylen) ? ylen : olen;
- spx_uint32_t omagic = 0;
-
- if (st->magic_samples[channel_index]) {
- omagic = speex_resampler_magic (st, channel_index, &y, ochunk);
- ochunk -= omagic;
- olen -= omagic;
- }
- if (!st->magic_samples[channel_index]) {
- if (in) {
- for (j = 0; j < ichunk; ++j)
-#ifdef FIXED_POINT
- x[j + st->filt_len - 1] = WORD2INT (in[j * istride_save]);
-#else
- x[j + st->filt_len - 1] = in[j * istride_save];
-#endif
- } else {
- for (j = 0; j < ichunk; ++j)
- x[j + st->filt_len - 1] = 0;
- }
-
- speex_resampler_process_native (st, channel_index, &ichunk, y, &ochunk);
- } else {
- ichunk = 0;
- ochunk = 0;
- }
-
- for (j = 0; j < ochunk + omagic; ++j)
-#ifdef FIXED_POINT
- out[j * ostride_save] = ystack[j];
-#else
- out[j * ostride_save] = WORD2INT (ystack[j]);
-#endif
-
- ilen -= ichunk;
- olen -= ochunk;
- out += (ochunk + omagic) * ostride_save;
- if (in)
- in += ichunk * istride_save;
- }
- st->out_stride = ostride_save;
- *in_len -= ilen;
- *out_len -= olen;
-
- return RESAMPLER_ERR_SUCCESS;
-}
-
-#ifdef DOUBLE_PRECISION
-EXPORT int
-speex_resampler_process_interleaved_float (SpeexResamplerState * st,
- const double *in, spx_uint32_t * in_len, double *out,
- spx_uint32_t * out_len)
-#else
-EXPORT int
-speex_resampler_process_interleaved_float (SpeexResamplerState * st,
- const float *in, spx_uint32_t * in_len, float *out, spx_uint32_t * out_len)
-#endif
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i = 0; i < st->nb_channels; i++) {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_float (st, i, in + i, in_len, out + i, out_len);
- else
- speex_resampler_process_float (st, i, NULL, in_len, out + i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int
-speex_resampler_process_interleaved_int (SpeexResamplerState * st,
- const spx_int16_t * in, spx_uint32_t * in_len, spx_int16_t * out,
- spx_uint32_t * out_len)
-{
- spx_uint32_t i;
- int istride_save, ostride_save;
- spx_uint32_t bak_len = *out_len;
- istride_save = st->in_stride;
- ostride_save = st->out_stride;
- st->in_stride = st->out_stride = st->nb_channels;
- for (i = 0; i < st->nb_channels; i++) {
- *out_len = bak_len;
- if (in != NULL)
- speex_resampler_process_int (st, i, in + i, in_len, out + i, out_len);
- else
- speex_resampler_process_int (st, i, NULL, in_len, out + i, out_len);
- }
- st->in_stride = istride_save;
- st->out_stride = ostride_save;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int
-speex_resampler_set_rate (SpeexResamplerState * st, spx_uint32_t in_rate,
- spx_uint32_t out_rate)
-{
- return speex_resampler_set_rate_frac (st, in_rate, out_rate, in_rate,
- out_rate);
-}
-
-EXPORT void
-speex_resampler_get_rate (SpeexResamplerState * st, spx_uint32_t * in_rate,
- spx_uint32_t * out_rate)
-{
- *in_rate = st->in_rate;
- *out_rate = st->out_rate;
-}
-
-EXPORT int
-speex_resampler_set_rate_frac (SpeexResamplerState * st, spx_uint32_t ratio_num,
- spx_uint32_t ratio_den, spx_uint32_t in_rate, spx_uint32_t out_rate)
-{
- spx_uint32_t fact;
- spx_uint32_t old_den;
- spx_uint32_t i;
- if (st->in_rate == in_rate && st->out_rate == out_rate
- && st->num_rate == ratio_num && st->den_rate == ratio_den)
- return RESAMPLER_ERR_SUCCESS;
-
- old_den = st->den_rate;
- st->in_rate = in_rate;
- st->out_rate = out_rate;
- st->num_rate = ratio_num;
- st->den_rate = ratio_den;
- /* FIXME: This is terribly inefficient, but who cares (at least for now)? */
- for (fact = 2; fact <= IMIN (st->num_rate, st->den_rate); fact++) {
- while ((st->num_rate % fact == 0) && (st->den_rate % fact == 0)) {
- st->num_rate /= fact;
- st->den_rate /= fact;
- }
- }
-
- if (old_den > 0) {
- for (i = 0; i < st->nb_channels; i++) {
- st->samp_frac_num[i] = st->samp_frac_num[i] * st->den_rate / old_den;
- /* Safety net */
- if (st->samp_frac_num[i] >= st->den_rate)
- st->samp_frac_num[i] = st->den_rate - 1;
- }
- }
-
- if (st->initialised)
- update_filter (st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void
-speex_resampler_get_ratio (SpeexResamplerState * st, spx_uint32_t * ratio_num,
- spx_uint32_t * ratio_den)
-{
- *ratio_num = st->num_rate;
- *ratio_den = st->den_rate;
-}
-
-EXPORT int
-speex_resampler_set_quality (SpeexResamplerState * st, int quality)
-{
- if (quality > 10 || quality < 0)
- return RESAMPLER_ERR_INVALID_ARG;
- if (st->quality == quality)
- return RESAMPLER_ERR_SUCCESS;
- st->quality = quality;
- if (st->initialised)
- update_filter (st);
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT void
-speex_resampler_get_quality (SpeexResamplerState * st, int *quality)
-{
- *quality = st->quality;
-}
-
-EXPORT void
-speex_resampler_set_input_stride (SpeexResamplerState * st, spx_uint32_t stride)
-{
- st->in_stride = stride;
-}
-
-EXPORT void
-speex_resampler_get_input_stride (SpeexResamplerState * st,
- spx_uint32_t * stride)
-{
- *stride = st->in_stride;
-}
-
-EXPORT void
-speex_resampler_set_output_stride (SpeexResamplerState * st,
- spx_uint32_t stride)
-{
- st->out_stride = stride;
-}
-
-EXPORT void
-speex_resampler_get_output_stride (SpeexResamplerState * st,
- spx_uint32_t * stride)
-{
- *stride = st->out_stride;
-}
-
-EXPORT int
-speex_resampler_get_input_latency (SpeexResamplerState * st)
-{
- return st->filt_len / 2;
-}
-
-EXPORT int
-speex_resampler_get_output_latency (SpeexResamplerState * st)
-{
- return ((st->filt_len / 2) * st->den_rate +
- (st->num_rate >> 1)) / st->num_rate;
-}
-
-EXPORT int
-speex_resampler_skip_zeros (SpeexResamplerState * st)
-{
- spx_uint32_t i;
- for (i = 0; i < st->nb_channels; i++)
- st->last_sample[i] = st->filt_len / 2;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT int
-speex_resampler_reset_mem (SpeexResamplerState * st)
-{
- spx_uint32_t i;
- for (i = 0; i < st->nb_channels * (st->filt_len - 1); i++)
- st->mem[i] = 0;
- return RESAMPLER_ERR_SUCCESS;
-}
-
-EXPORT const char *
-speex_resampler_strerror (int err)
-{
- switch (err) {
- case RESAMPLER_ERR_SUCCESS:
- return "Success.";
- case RESAMPLER_ERR_ALLOC_FAILED:
- return "Memory allocation failed.";
- case RESAMPLER_ERR_BAD_STATE:
- return "Bad resampler state.";
- case RESAMPLER_ERR_INVALID_ARG:
- return "Invalid argument.";
- case RESAMPLER_ERR_PTR_OVERLAP:
- return "Input and output buffers overlap.";
- default:
- return "Unknown error. Bad error code or strange version mismatch.";
- }
-}
&& data[3] == 'd')
gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MID_CAPS);
}
+
/*** audio/mobile-xmf ***/
static GstStaticCaps mxmf_caps = GST_STATIC_CAPS ("audio/mobile-xmf");
{
guint8 *data = NULL;
- /* Search FileId "XMF_" 4 bytes */
+ /* Search FileId "XMF_" 4 bytes */
data = gst_type_find_peek (tf, 0, 4);
if (data && data[0] == 'X' && data[1] == 'M' && data[2] == 'F'
- && data[3] == '_') {
- /* Search Format version "2.00" 4 bytes */
- data = gst_type_find_peek (tf, 4, 4);
- if (data && data[0] == '2' && data[1] == '.' && data[2] == '0'
- && data[3] == '0') {
- /* Search TypeId 2 1 byte */
- data = gst_type_find_peek (tf, 11, 1);
- if (data && data[0] == 2 ) {
- gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MXMF_CAPS);
- }
- }
- }
+ && data[3] == '_') {
+ /* Search Format version "2.00" 4 bytes */
+ data = gst_type_find_peek (tf, 4, 4);
+ if (data && data[0] == '2' && data[1] == '.' && data[2] == '0'
+ && data[3] == '0') {
+ /* Search TypeId 2 1 byte */
+ data = gst_type_find_peek (tf, 11, 1);
+ if (data && data[0] == 2) {
+ gst_type_find_suggest (tf, GST_TYPE_FIND_MAXIMUM, MXMF_CAPS);
+ }
+ }
+ }
}
#endif
TYPE_FIND_REGISTER (plugin, "audio/midi", GST_RANK_PRIMARY, mid_type_find,
mid_exts, MID_CAPS, NULL, NULL);
- TYPE_FIND_REGISTER (plugin, "audio/mobile-xmf", GST_RANK_PRIMARY, mxmf_type_find,
- mxmf_exts, MXMF_CAPS, NULL, NULL);
+ TYPE_FIND_REGISTER (plugin, "audio/mobile-xmf", GST_RANK_PRIMARY,
+ mxmf_type_find, mxmf_exts, MXMF_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "video/x-fli", GST_RANK_MARGINAL, flx_type_find,
flx_exts, FLX_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "application/x-id3v2", GST_RANK_PRIMARY + 103,
mpeg4_video_type_find, m4v_exts, MPEG_VIDEO_CAPS, NULL, NULL);
TYPE_FIND_REGISTER (plugin, "video/x-h264", GST_RANK_PRIMARY,
h264_video_type_find, h264_exts, MPEG_VIDEO_CAPS, NULL, NULL);
- TYPE_FIND_REGISTER (plugin, "video/x-nuv", GST_RANK_SECONDARY,
- nuv_type_find, nuv_exts, NUV_CAPS, NULL, NULL);
+ TYPE_FIND_REGISTER (plugin, "video/x-nuv", GST_RANK_SECONDARY, nuv_type_find,
+ nuv_exts, NUV_CAPS, NULL, NULL);
/* ISO formats */
TYPE_FIND_REGISTER (plugin, "audio/x-m4a", GST_RANK_PRIMARY, m4a_type_find,
elements/adder \
elements/audioconvert \
elements/audiorate \
+ elements/audioresample \
elements/audiotestsrc \
elements/decodebin \
elements/ffmpegcolorspace \
elements/multifdsink \
elements/playbin \
$(check_subparse) \
- elements/speexresample \
elements/videorate \
elements/videotestsrc \
elements/volume \
elements_subparse_LDADD = $(LDADD)
elements_subparse_CFLAGS = $(CFLAGS) $(AM_CFLAGS)
-elements_speexresample_CFLAGS = \
+elements_audioresample_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) \
$(AM_CFLAGS)
-elements_speexresample_LDADD = \
+elements_audioresample_LDADD = \
$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
$(top_builddir)/gst-libs/gst/interfaces/libgstinterfaces-@GST_MAJORMINOR@.la \
$(GST_BASE_LIBS) \
/* 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) <2006> Tim-Philipp Müller <tim at centricular net>
#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_TEMPLATE_STRING \
+#define RESAMPLE_CAPS_INT \
"audio/x-raw-int, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ], " \
"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,
);
static GstElement *
-setup_audioresample (int channels, int inrate, int outrate)
+setup_audioresample (int channels, int inrate, int outrate, int width,
+ gboolean fp)
{
GstElement *audioresample;
GstCaps *caps;
GST_DEBUG ("setup_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);
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_element_set_state (audioresample,
gst_pad_set_caps (mysrcpad, 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);
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));
mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
gst_pad_set_active (mysinkpad, TRUE);
gst_pad_set_active (mysrcpad, TRUE);
+ gst_caps_unref (caps);
+
return audioresample;
}
int i, j;
gint16 *p;
- audioresample = setup_audioresample (2, inrate, outrate);
+ audioresample = setup_audioresample (2, inrate, outrate, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
for (j = 1; j <= numbuffers; ++j) {
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_OFFSET (inbuffer) = offset;
offset += samples;
GST_DEBUG ("inrate:%d outrate:%d samples:%d numbuffers:%d",
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);
fail_unless (gst_caps_is_fixed (caps));
GstBuffer *inbuffer;
GstCaps *caps;
- audioresample = setup_audioresample (1, 9343, 48000);
+ audioresample = setup_audioresample (1, 9343, 48000, 16, FALSE);
caps = gst_pad_get_negotiated_caps (mysrcpad);
fail_unless (gst_caps_is_fixed (caps));
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);
* 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));
GstEvent *newseg;
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
* rate 48000- and can only allocate buffers for that rate, but if someone
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));
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)
{
Suite *s = suite_create ("audioresample");
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;
}
+++ /dev/null
-/* 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);
projects / platform / upstream / gstreamer.git / commitdiff