+2007-08-16 Sebastian Dröge <slomo@circular-chaos.org>
+
+ reviewed by: Stefan Kost <ensonic@users.sf.net>
+
+ * gst/audiofx/Makefile.am:
+ * gst/audiofx/audiochebyshevfreqband.c:
+ (gst_audio_chebyshev_freq_band_mode_get_type),
+ (gst_audio_chebyshev_freq_band_base_init),
+ (gst_audio_chebyshev_freq_band_dispose),
+ (gst_audio_chebyshev_freq_band_class_init),
+ (gst_audio_chebyshev_freq_band_init),
+ (generate_biquad_coefficients), (calculate_gain),
+ (generate_coefficients),
+ (gst_audio_chebyshev_freq_band_set_property),
+ (gst_audio_chebyshev_freq_band_get_property),
+ (gst_audio_chebyshev_freq_band_setup), (process), (process_64),
+ (process_32), (gst_audio_chebyshev_freq_band_transform_ip),
+ (gst_audio_chebyshev_freq_band_start):
+ * gst/audiofx/audiochebyshevfreqband.h:
+ * gst/audiofx/audiochebyshevfreqlimit.c:
+ (gst_audio_chebyshev_freq_limit_mode_get_type),
+ (gst_audio_chebyshev_freq_limit_base_init),
+ (gst_audio_chebyshev_freq_limit_dispose),
+ (gst_audio_chebyshev_freq_limit_class_init),
+ (gst_audio_chebyshev_freq_limit_init),
+ (generate_biquad_coefficients), (calculate_gain),
+ (generate_coefficients),
+ (gst_audio_chebyshev_freq_limit_set_property),
+ (gst_audio_chebyshev_freq_limit_get_property),
+ (gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
+ (process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
+ (gst_audio_chebyshev_freq_limit_start):
+ * gst/audiofx/audiochebyshevfreqlimit.h:
+ * gst/audiofx/audiofx.c: (plugin_init):
+ Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
+ Fixes #464800.
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.cvsignore:
+ * tests/check/elements/audiochebyshevfreqband.c:
+ (setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
+ (GST_START_TEST), (audiochebyshevfreqband_suite), (main):
+ * tests/check/elements/audiochebyshevfreqlimit.c:
+ (setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
+ (GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
+ Add unit tests for the chebyshev filters.
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ And add docs for the chebyshev filters. While doing
+ that also run make update in docs/plugins.
+
2007-08-16 Stefan Kost <ensonic@users.sf.net>
* ext/annodex/gstcmmltag.c:
$(top_srcdir)/gst/audiofx/audiodynamic.h \
$(top_srcdir)/gst/audiofx/audioinvert.h \
$(top_srcdir)/gst/audiofx/audiopanorama.h \
+ $(top_srcdir)/gst/audiofx/audiochebyshevfreqlimit.h \
+ $(top_srcdir)/gst/audiofx/audiochebyshevfreqband.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
$(top_srcdir)/gst/autodetect/gstautovideosink.h \
$(top_srcdir)/gst/avi/gstavidemux.h \
<xi:include href="xml/element-audioinvert.xml" />
<xi:include href="xml/element-audioamplify.xml" />
<xi:include href="xml/element-audiodynamic.xml" />
+ <xi:include href="xml/element-audiochebyshevfreqlimit.xml" />
+ <xi:include href="xml/element-audiochebyshevfreqband.xml" />
<xi:include href="xml/element-autoaudiosink.xml" />
<xi:include href="xml/element-autovideosink.xml" />
<xi:include href="xml/element-avidemux.xml" />
</SECTION>
<SECTION>
+<FILE>element-audiochebyshevfreqlimit</FILE>
+<TITLE>audiochebyshevfreqlimit</TITLE>
+GstAudioChebyshevFreqLimit
+<SUBSECTION Standard>
+GstAudioChebyshevFreqLimitClass
+GST_AUDIO_CHEBYSHEV_FREQ_LIMIT
+GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>element-audiochebyshevfreqband</FILE>
+<TITLE>audiochebyshevfreqband</TITLE>
+GstAudioChebyshevFreqBand
+<SUBSECTION Standard>
+GstAudioChebyshevFreqBandClass
+GST_AUDIO_CHEBYSHEV_FREQ_BAND
+GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS
+</SECTION>
+
+<SECTION>
<FILE>element-autoaudiosink</FILE>
<TITLE>autoaudiosink</TITLE>
GstAutoAudioSink
<ARG>
<NAME>GstVertigoTV::speed</NAME>
<TYPE>gfloat</TYPE>
-<RANGE>[0,01,100]</RANGE>
+<RANGE>[0.01,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Speed</NICK>
<BLURB>Control the speed of movement.</BLURB>
-<DEFAULT>0,02</DEFAULT>
+<DEFAULT>0.02</DEFAULT>
</ARG>
<ARG>
<NAME>GstVertigoTV::zoom-speed</NAME>
<TYPE>gfloat</TYPE>
-<RANGE>[1,01,1,1]</RANGE>
+<RANGE>[1.01,1.1]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Zoom Speed</NICK>
<BLURB>Control the rate of zooming.</BLURB>
-<DEFAULT>1,01</DEFAULT>
+<DEFAULT>1.01</DEFAULT>
</ARG>
<ARG>
<ARG>
<NAME>GstDV1394Src::port</NAME>
<TYPE>gint</TYPE>
-<RANGE>[G_MAXULONG,16]</RANGE>
+<RANGE>[-1,16]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>Port number (-1 automatic).</BLURB>
<ARG>
<NAME>GstGamma::gamma</NAME>
<TYPE>gdouble</TYPE>
-<RANGE>[0,01,10]</RANGE>
+<RANGE>[0.01,10]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Gamma</NICK>
<BLURB>gamma.</BLURB>
<DEFAULT>2</DEFAULT>
</ARG>
+<ARG>
+<NAME>GstAudioChebyshevFreqBand::lower-frequency</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Lower frequency</NICK>
+<BLURB>Start frequency of the band (Hz).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqBand::mode</NAME>
+<TYPE>GstAudioChebyshevFreqBandMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Mode</NICK>
+<BLURB>Low pass or high pass mode.</BLURB>
+<DEFAULT>Band pass (default)</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqBand::poles</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[4,32]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Poles</NICK>
+<BLURB>Number of poles to use, will be rounded up to the next multiply of four.</BLURB>
+<DEFAULT>4</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqBand::ripple</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Ripple</NICK>
+<BLURB>Amount of ripple (dB).</BLURB>
+<DEFAULT>0.25</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqBand::type</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[1,2]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Type</NICK>
+<BLURB>Type of the chebychev filter.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqBand::upper-frequency</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Upper frequency</NICK>
+<BLURB>Stop frequency of the band (Hz).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqLimit::cutoff</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Cutoff</NICK>
+<BLURB>Cut off frequency (Hz).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqLimit::mode</NAME>
+<TYPE>GstAudioChebyshevFreqLimitMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Mode</NICK>
+<BLURB>Low pass or high pass mode.</BLURB>
+<DEFAULT>Low pass (default)</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqLimit::poles</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[2,32]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Poles</NICK>
+<BLURB>Number of poles to use, will be rounded up to the next even number.</BLURB>
+<DEFAULT>4</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqLimit::ripple</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Ripple</NICK>
+<BLURB>Amount of ripple (dB).</BLURB>
+<DEFAULT>0.25</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioChebyshevFreqLimit::type</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[1,2]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Type</NICK>
+<BLURB>Type of the chebychev filter.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
<description>Source for DV data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
- <version>0.10.6</version>
+ <version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins source release</package>
+ <package>GStreamer Good Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
Daniel Fischer <dan@f3c.com>
Wim Taymans <wim@fluendo.com>
Zaheer Abbas Merali <zaheerabbas at merali dot org></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/x-dv, format=(string){ NTSC, PAL }, systemstream=(boolean)true</details>
+ </caps>
+ </pads>
</element>
</elements>
</plugin>
\ No newline at end of file
</pads>
</element>
<element>
+ <name>audiochebyshevfreqband</name>
+ <longname>AudioChebyshevFreqBand</longname>
+ <class>Filter/Effect/Audio</class>
+ <description>Chebyshev band pass and band reject filter</description>
+ <author>Sebastian Dröge <slomo@circular-chaos.org></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>audiochebyshevfreqlimit</name>
+ <longname>AudioChebyshevFreqLimit</longname>
+ <class>Filter/Effect/Audio</class>
+ <description>Chebyshev low pass and high pass filter</description>
+ <author>Sebastian Dröge <slomo@circular-chaos.org></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
<name>audiodynamic</name>
<longname>AudioDynamic</longname>
<class>Filter/Effect/Audio</class>
<description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
<filename>../../ext/dv/.libs/libgstdv.so</filename>
<basename>libgstdv.so</basename>
- <version>0.10.6</version>
+ <version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins source release</package>
+ <package>GStreamer Good Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<class>Codec/Decoder/Video</class>
<description>Uses libdv to decode DV video (smpte314) (libdv.sourceforge.net)</description>
<author>Erik Walthinsen <omega@cse.ogi.edu>,Wim Taymans <wim@fluendo.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/x-dv, systemstream=(boolean)false</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/x-raw-yuv, format=(fourcc)YUY2, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)24, endianness=(int)4321, red_mask=(int)65280, green_mask=(int)16711680, blue_mask=(int)-16777216, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]</details>
+ </caps>
+ </pads>
</element>
<element>
<name>dvdemux</name>
<class>Codec/Demuxer</class>
<description>Uses libdv to separate DV audio from DV video (libdv.sourceforge.net)</description>
<author>Erik Walthinsen <omega@cse.ogi.edu>, Wim Taymans <wim@fluendo.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/x-dv, systemstream=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>video</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>video/x-dv, systemstream=(boolean)false</details>
+ </caps>
+ <caps>
+ <name>audio</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>audio/x-raw-int, depth=(int)16, width=(int)16, signed=(boolean)true, channels=(int){ 2, 4 }, endianness=(int)1234, rate=(int){ 32000, 44100, 48000 }</details>
+ </caps>
+ </pads>
</element>
</elements>
</plugin>
\ No newline at end of file
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
- <details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ]</details>
+ <details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]</details>
</caps>
</pads>
</element>
<author>Wim Taymans <wim@fluendo.com></author>
<pads>
<caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 8, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
- </caps>
- <caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 8, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
</pads>
</element>
<element>
<author>Wim Taymans <wim.taymans@tvd.be></author>
<pads>
<caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
- </caps>
- <caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
</pads>
</element>
<element>
<author>Wim Taymans <wim@fluendo.com></author>
<pads>
<caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
- </caps>
- <caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
</pads>
</element>
<element>
<author>Wim Taymans <wim@fluendo.com></author>
<pads>
<caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
- </caps>
- <caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
</pads>
</element>
</elements>
<author>Wim Taymans <wim@fluendo.com></author>
<pads>
<caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>image/png</details>
- </caps>
- <caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-rgb, bpp=(int)32, depth=(int)32, endianness=(int)4321, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, alpha_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>image/png</details>
+ </caps>
</pads>
</element>
<element>
<author>Jeremy SIMON <jsimon13@yahoo.fr></author>
<pads>
<caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>image/png, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
- </caps>
- <caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-raw-rgb, bpp=(int)32, depth=(int)32, endianness=(int)4321, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, alpha_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>image/png, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ </caps>
</pads>
</element>
</elements>
</element>
<element>
<name>rtpdepay</name>
- <longname>RTP payloader</longname>
+ <longname>RTP depayloader</longname>
<class>Codec/Depayloader/Network</class>
<description>Accepts raw RTP and RTCP packets and sends them forward</description>
<author>Wim Taymans <wim@fluendo.com></author>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
- <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(int){ 20, 30 }</details>
+ <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(string){ 20, 30 }</details>
</caps>
</pads>
</element>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
- <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(int){ 20, 30 }</details>
+ <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(string){ 20, 30 }</details>
</caps>
</pads>
</element>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
- <details>video/mpeg, systemstream=(boolean)false</details>
+ <details>video/mpeg, mpegversion=(int)2, systemstream=(boolean)false</details>
</caps>
<caps>
<name>sink</name>
<description>Sends data to an icecast server using libshout2</description>
<filename>../../ext/shout2/.libs/libgstshout2.so</filename>
<basename>libgstshout2.so</basename>
- <version>0.10.6</version>
+ <version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>libshout2</package>
<author>Wim Taymans <wim.taymans@chello.be>
Pedro Corte-Real <typo@netcabo.pt>
Zaheer Abbas Merali <zaheerabbas at merali dot org></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>application/ogg; audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ]</details>
+ </caps>
+ </pads>
</element>
</elements>
</plugin>
\ No newline at end of file
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
- <version>0.10.6</version>
+ <version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins source release</package>
+ <package>GStreamer Good Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<class>Codec/Decoder/Audio</class>
<description>Decodes Wavpack audio data</description>
<author>Arwed v. Merkatz <v.merkatz@gmx.net>, Sebastian Dröge <slomo@circular-chaos.org></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-int, width=(int)32, depth=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], endianness=(int)1234, signed=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true</details>
+ </caps>
+ </pads>
</element>
<element>
<name>wavpackenc</name>
<class>Codec/Encoder/Audio</class>
<description>Encodes audio with the Wavpack lossless/lossy audio codec</description>
<author>Sebastian Dröge <slomo@circular-chaos.org></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-raw-int, width=(int)32, depth=(int)[ 1, 32 ], endianness=(int)1234, channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], signed=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>wvcsrc</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>audio/x-wavpack-correction, framed=(boolean)true</details>
+ </caps>
+ </pads>
</element>
<element>
<name>wavpackparse</name>
<class>Codec/Demuxer/Audio</class>
<description>Parses Wavpack files</description>
<author>Arwed v. Merkatz <v.merkatz@gmx.net>, Sebastian Dröge <slomo@circular-chaos.org></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>wvcsrc</name>
+ <direction>source</direction>
+ <presence>sometimes</presence>
+ <details>audio/x-wavpack-correction, framed=(boolean)true</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-wavpack, framed=(boolean)false; audio/x-wavpack-correction, framed=(boolean)false</details>
+ </caps>
+ </pads>
</element>
</elements>
</plugin>
\ No newline at end of file
audiopanorama.c \
audioinvert.c \
audioamplify.c \
- audiodynamic.c
+ audiodynamic.c \
+ audiochebyshevfreqlimit.c \
+ audiochebyshevfreqband.c
# flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
$(GST_BASE_LIBS) \
$(GST_CONTROLLER_LIBS) \
$(GST_PLUGINS_BASE_LIBS) \
- -lgstaudio-$(GST_MAJORMINOR)
+ -lgstaudio-$(GST_MAJORMINOR) \
+ $(LIBM)
libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
# headers we need but don't want installed
noinst_HEADERS = audiopanorama.h \
audioinvert.h \
audioamplify.h \
- audiodynamic.h
+ audiodynamic.h \
+ audiochebyshevfreqlimit.h \
+ audiochebyshevfreqband.c
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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.
+ */
+
+/*
+ * Chebyshev type 1 filter design based on
+ * "The Scientist and Engineer's Guide to DSP", Chapter 20.
+ * http://www.dspguide.com/
+ *
+ * For type 2 and Chebyshev filters in general read
+ * http://en.wikipedia.org/wiki/Chebyshev_filter
+ *
+ * Transformation from lowpass to bandpass/bandreject:
+ * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm
+ * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm
+ *
+ */
+
+/**
+ * SECTION:element-audiochebyshevfreqband
+ * @short_description: Chebyshev band pass and band reject filter
+ *
+ * <refsect2>
+ * <para>
+ * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
+ * band. The number of poles and the ripple parameter control the rolloff.
+ * </para>
+ * <para>
+ * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
+ * some frequencies in the passband will be amplified by that value. A higher ripple value will allow
+ * a faster rolloff.
+ * </para>
+ * <para>
+ * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
+ * be at most this value. A lower ripple value will allow a faster rolloff.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/controller/gstcontroller.h>
+
+#include <math.h>
+
+#include "audiochebyshevfreqband.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
+ "Filter/Effect/Audio",
+ "Chebyshev band pass and band reject filter",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_MODE,
+ PROP_TYPE,
+ PROP_LOWER_FREQUENCY,
+ PROP_UPPER_FREQUENCY,
+ PROP_RIPPLE,
+ PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-float," \
+ " width = (int) { 32, 64 }, " \
+ " endianness = (int) BYTE_ORDER," \
+ " rate = (int) [ 1, MAX ]," \
+ " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
+ GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
+ GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqBand * filter,
+ gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqBand * filter,
+ gfloat * data, guint num_samples);
+
+enum
+{
+ MODE_BAND_PASS = 0,
+ MODE_BAND_REJECT
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_band_mode_get_type (void)
+{
+ static GType gtype = 0;
+
+ if (gtype == 0) {
+ static const GEnumValue values[] = {
+ {MODE_BAND_PASS, "Band pass (default)",
+ "band-pass"},
+ {MODE_BAND_REJECT, "Band reject",
+ "band-reject"},
+ {0, NULL, NULL}
+ };
+
+ gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values);
+ }
+ return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_band_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &element_details);
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+ caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_band_dispose (GObject * object)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqBandChannelCtx *ctx;
+ gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
+ klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseTransformClass *trans_class;
+ GstAudioFilterClass *filter_class;
+
+ gobject_class = (GObjectClass *) klass;
+ trans_class = (GstBaseTransformClass *) klass;
+ filter_class = (GstAudioFilterClass *) klass;
+
+ gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property;
+ gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property;
+ gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose;
+
+ g_object_class_install_property (gobject_class, PROP_MODE,
+ g_param_spec_enum ("mode", "Mode",
+ "Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE,
+ MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_TYPE,
+ g_param_spec_int ("type", "Type",
+ "Type of the chebychev filter", 1, 2,
+ 1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
+ g_param_spec_float ("lower-frequency", "Lower frequency",
+ "Start frequency of the band (Hz)", 0.0, G_MAXFLOAT,
+ 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
+ g_param_spec_float ("upper-frequency", "Upper frequency",
+ "Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT,
+ 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_RIPPLE,
+ g_param_spec_float ("ripple", "Ripple",
+ "Amount of ripple (dB)", 0.0, G_MAXFLOAT,
+ 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_POLES,
+ g_param_spec_int ("poles", "Poles",
+ "Number of poles to use, will be rounded up to the next multiply of four",
+ 4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
+ trans_class->transform_ip =
+ GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
+ trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
+}
+
+static void
+gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
+ GstAudioChebyshevFreqBandClass * klass)
+{
+ filter->lower_frequency = filter->upper_frequency = 0.0;
+ filter->mode = MODE_BAND_PASS;
+ filter->type = 1;
+ filter->poles = 4;
+ filter->ripple = 0.25;
+ gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+ filter->have_coeffs = FALSE;
+ filter->num_a = 0;
+ filter->num_b = 0;
+ filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter,
+ gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
+ gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
+{
+ gint np = filter->poles / 2;
+ gdouble ripple = filter->ripple;
+
+ /* pole location in s-plane */
+ gdouble rp, ip;
+
+ /* zero location in s-plane */
+ gdouble rz = 0.0, iz = 0.0;
+
+ /* transfer function coefficients for the z-plane */
+ gdouble x0, x1, x2, y1, y2;
+ gint type = filter->type;
+
+ /* Calculate pole location for lowpass at frequency 1 */
+ {
+ gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
+
+ rp = -sin (angle);
+ ip = cos (angle);
+ }
+
+ /* If we allow ripple, move the pole from the unit
+ * circle to an ellipse and keep cutoff at frequency 1 */
+ if (ripple > 0 && type == 1) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+
+ vx = (1.0 / np) * asinh (1.0 / es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ } else if (type == 2) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+ vx = (1.0 / np) * asinh (es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ }
+
+ /* Calculate inverse of the pole location to move from
+ * type I to type II */
+ if (type == 2) {
+ gdouble mag2 = rp * rp + ip * ip;
+
+ rp /= mag2;
+ ip /= mag2;
+ }
+
+ /* Calculate zero location for frequency 1 on the
+ * unit circle for type 2 */
+ if (type == 2) {
+ gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
+ gdouble mag2;
+
+ rz = 0.0;
+ iz = cos (angle);
+ mag2 = rz * rz + iz * iz;
+ rz /= mag2;
+ iz /= mag2;
+ }
+
+ /* Convert from s-domain to z-domain by
+ * using the bilinear Z-transform, i.e.
+ * substitute s by (2/t)*((z-1)/(z+1))
+ * with t = 2 * tan(0.5).
+ */
+ if (type == 1) {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t) / d;
+ x1 = 2.0 * x0;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ } else {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t * iz * iz + 4.0) / d;
+ x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ }
+
+ /* Convert from lowpass at frequency 1 to either bandpass
+ * or band reject.
+ *
+ * For bandpass substitute z^(-1) with:
+ *
+ * -2 -1
+ * -z + alpha * z - beta
+ * ----------------------------
+ * -2 -1
+ * beta * z - alpha * z + 1
+ *
+ * alpha = (2*a*b)/(1+b)
+ * beta = (b-1)/(b+1)
+ * a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
+ * b = tan(1/2) * cot((w1 - w0)/2)
+ *
+ * For bandreject substitute z^(-1) with:
+ *
+ * -2 -1
+ * z - alpha * z + beta
+ * ----------------------------
+ * -2 -1
+ * beta * z - alpha * z + 1
+ *
+ * alpha = (2*a)/(1+b)
+ * beta = (1-b)/(1+b)
+ * a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
+ * b = tan(1/2) * tan((w1 - w0)/2)
+ *
+ */
+ {
+ gdouble a, b, d;
+ gdouble alpha, beta;
+ gdouble w0 =
+ 2.0 * M_PI * (filter->lower_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w1 =
+ 2.0 * M_PI * (filter->upper_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+
+ if (filter->mode == MODE_BAND_PASS) {
+ a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
+ b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0);
+
+ alpha = (2.0 * a * b) / (1.0 + b);
+ beta = (b - 1.0) / (b + 1.0);
+
+ d = 1.0 + beta * (y1 - beta * y2);
+
+ *a0 = (x0 + beta * (-x1 + beta * x2)) / d;
+ *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d;
+ *a2 =
+ (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) +
+ alpha * alpha * (x0 - x1 + x2)) / d;
+ *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d;
+ *a4 = (beta * (beta * x0 - x1) + x2) / d;
+ *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d;
+ *b2 =
+ (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) +
+ 2.0 * beta * (-1.0 + y2)) / d;
+ *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d;
+ *b4 = (-beta * beta - beta * y1 + y2) / d;
+ } else {
+ a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
+ b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0);
+
+ alpha = (2.0 * a) / (1.0 + b);
+ beta = (1.0 - b) / (1.0 + b);
+
+ d = -1.0 + beta * (beta * y2 + y1);
+
+ *a0 = (-x0 - beta * x1 - beta * beta * x2) / d;
+ *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d;
+ *a2 =
+ (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) -
+ alpha * alpha * (x0 + x1 + x2)) / d;
+ *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d;
+ *a4 = (-beta * beta * x0 - beta * x1 - x2) / d;
+ *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d;
+ *b2 =
+ -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) +
+ alpha * alpha * (-1.0 + y1 + y2)) / d;
+ *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d;
+ *b4 = -(-beta * beta + beta * y1 + y2) / d;
+ }
+ }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+ gdouble zi)
+{
+ gdouble sum_ar, sum_ai;
+ gdouble sum_br, sum_bi;
+ gdouble gain_r, gain_i;
+
+ gdouble sum_r_old;
+ gdouble sum_i_old;
+
+ gint i;
+
+ sum_ar = 0.0;
+ sum_ai = 0.0;
+ for (i = num_a; i >= 0; i--) {
+ sum_r_old = sum_ar;
+ sum_i_old = sum_ai;
+
+ sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+ sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+ }
+
+ sum_br = 0.0;
+ sum_bi = 0.0;
+ for (i = num_b; i >= 0; i--) {
+ sum_r_old = sum_br;
+ sum_i_old = sum_bi;
+
+ sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+ sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+ }
+ sum_br += 1.0;
+ sum_bi += 0.0;
+
+ gain_r =
+ (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+ gain_i =
+ (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+ return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqBand * filter)
+{
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqBandChannelCtx *ctx;
+ gint i;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "rate was not set yet");
+ return;
+ }
+
+ filter->have_coeffs = TRUE;
+
+ if (filter->upper_frequency <= filter->lower_frequency) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
+ return;
+ }
+
+ if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) {
+ filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2;
+ GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency");
+ }
+
+ if (filter->lower_frequency < 0.0) {
+ filter->lower_frequency = 0.0;
+ GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0");
+ }
+
+ /* Calculate coefficients for the chebyshev filter */
+ {
+ gint np = filter->poles;
+ gdouble *a, *b;
+ gint i, p;
+
+ filter->num_a = np + 1;
+ filter->a = a = g_new0 (gdouble, np + 5);
+ filter->num_b = np + 1;
+ filter->b = b = g_new0 (gdouble, np + 5);
+
+ filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+ for (i = 0; i < channels; i++) {
+ GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
+
+ ctx->x = g_new0 (gdouble, np + 1);
+ ctx->y = g_new0 (gdouble, np + 1);
+ }
+
+ /* Calculate transfer function coefficients */
+ a[4] = 1.0;
+ b[4] = 1.0;
+
+ for (p = 1; p <= np / 4; p++) {
+ gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4;
+ gdouble *ta = g_new0 (gdouble, np + 5);
+ gdouble *tb = g_new0 (gdouble, np + 5);
+
+ generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1,
+ &b2, &b3, &b4);
+
+ memcpy (ta, a, sizeof (gdouble) * (np + 5));
+ memcpy (tb, b, sizeof (gdouble) * (np + 5));
+
+ /* add the new coefficients for the new two poles
+ * to the cascade by multiplication of the transfer
+ * functions */
+ for (i = 4; i < np + 5; i++) {
+ a[i] =
+ a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] +
+ a4 * ta[i - 4];
+ b[i] =
+ tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] -
+ b4 * tb[i - 4];
+ }
+ g_free (ta);
+ g_free (tb);
+ }
+
+ /* Move coefficients to the beginning of the array
+ * and multiply the b coefficients with -1 to move from
+ * the transfer function's coefficients to the difference
+ * equation's coefficients */
+ b[4] = 0.0;
+ for (i = 0; i <= np; i++) {
+ a[i] = a[i + 4];
+ b[i] = -b[i + 4];
+ }
+
+ /* Normalize to unity gain at frequency 0 and frequency
+ * 0.5 for bandreject and unity gain at band center frequency
+ * for bandpass */
+ if (filter->mode == MODE_BAND_REJECT) {
+ /* gain is sqrt(H(0)*H(0.5)) */
+
+ gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0);
+ gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0);
+
+ gain1 = sqrt (gain1 * gain2);
+
+ for (i = 0; i <= np; i++) {
+ a[i] /= gain1;
+ }
+ } else {
+ /* gain is H(wc), wc = center frequency */
+
+ gdouble w1 =
+ 2.0 * M_PI * (filter->lower_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w2 =
+ 2.0 * M_PI * (filter->upper_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w0 = (w2 + w1) / 2.0;
+ gdouble zr = cos (w0), zi = sin (w0);
+ gdouble gain = calculate_gain (a, b, np, np, zr, zi);
+
+ for (i = 0; i <= np; i++) {
+ a[i] /= gain;
+ }
+ }
+
+ GST_LOG_OBJECT (filter,
+ "Generated IIR coefficients for the Chebyshev filter");
+ GST_LOG_OBJECT (filter,
+ "mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB",
+ (filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject",
+ filter->type, filter->poles, filter->lower_frequency,
+ filter->upper_frequency, filter->ripple);
+
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+ {
+ gdouble w1 =
+ 2.0 * M_PI * (filter->lower_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w2 =
+ 2.0 * M_PI * (filter->upper_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w0 = (w2 + w1) / 2.0;
+ gdouble zr, zi;
+
+ zr = cos (w1);
+ zi = sin (w1);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) filter->lower_frequency);
+ zr = cos (w0);
+ zi = sin (w0);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) ((filter->lower_frequency + filter->upper_frequency) / 2.0));
+ zr = cos (w2);
+ zi = sin (w2);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) filter->upper_frequency);
+ }
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+ GST_AUDIO_FILTER (filter)->format.rate / 2);
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->mode = g_value_get_enum (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_TYPE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->type = g_value_get_int (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_LOWER_FREQUENCY:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->lower_frequency = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_UPPER_FREQUENCY:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->upper_frequency = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_RIPPLE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->ripple = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_POLES:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->poles = GST_ROUND_UP_4 (g_value_get_int (value));
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ g_value_set_enum (value, filter->mode);
+ break;
+ case PROP_TYPE:
+ g_value_set_int (value, filter->type);
+ break;
+ case PROP_LOWER_FREQUENCY:
+ g_value_set_float (value, filter->lower_frequency);
+ break;
+ case PROP_UPPER_FREQUENCY:
+ g_value_set_float (value, filter->upper_frequency);
+ break;
+ case PROP_RIPPLE:
+ g_value_set_float (value, filter->ripple);
+ break;
+ case PROP_POLES:
+ g_value_set_int (value, filter->poles);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
+ GstRingBufferSpec * format)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+ gboolean ret = TRUE;
+
+ if (format->width == 32)
+ filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+ process_32;
+ else if (format->width == 64)
+ filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+ process_64;
+ else
+ ret = FALSE;
+
+ filter->have_coeffs = FALSE;
+
+ return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqBand * filter,
+ GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
+{
+ gdouble val = filter->a[0] * x0;
+ gint i, j;
+
+ for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+ val += filter->a[i] * ctx->x[j];
+ j--;
+ if (j < 0)
+ j = filter->num_a - 1;
+ }
+
+ for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+ val += filter->b[i] * ctx->y[j];
+ j--;
+ if (j < 0)
+ j = filter->num_b - 1;
+ }
+
+ if (ctx->x) {
+ ctx->x_pos++;
+ if (ctx->x_pos > filter->num_a - 1)
+ ctx->x_pos = 0;
+ ctx->x[ctx->x_pos] = x0;
+ }
+
+ if (ctx->y) {
+ ctx->y_pos++;
+ if (ctx->y_pos > filter->num_b - 1)
+ ctx->y_pos = 0;
+
+ ctx->y[ctx->y_pos] = val;
+ }
+
+ return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqBand * filter,
+ gdouble * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqBand * filter,
+ gfloat * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+ guint num_samples =
+ GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+ if (!gst_buffer_is_writable (buf))
+ return GST_FLOW_OK;
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+ gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+ if (!filter->have_coeffs)
+ generate_coefficients (filter);
+
+ filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}
+
+static gboolean
+gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+ GstAudioChebyshevFreqBandChannelCtx *ctx;
+ gint i;
+
+ /* Reset the history of input and output values if
+ * already existing */
+ if (channels && filter->channels) {
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ if (ctx->x)
+ memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+ if (ctx->y)
+ memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+ }
+ }
+ return TRUE;
+}
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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 __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand;
+typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass;
+
+typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint);
+
+typedef struct
+{
+ gdouble *x;
+ gint x_pos;
+ gdouble *y;
+ gint y_pos;
+} GstAudioChebyshevFreqBandChannelCtx;
+
+struct _GstAudioChebyshevFreqBand
+{
+ GstAudioFilter audiofilter;
+
+ gint mode;
+ gint type;
+ gint poles;
+ gfloat lower_frequency;
+ gfloat upper_frequency;
+ gfloat ripple;
+
+ /* < private > */
+ GstAudioChebyshevFreqBandProcessFunc process;
+
+ gboolean have_coeffs;
+ gdouble *a;
+ gint num_a;
+ gdouble *b;
+ gint num_b;
+ GstAudioChebyshevFreqBandChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqBandClass
+{
+ GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_band_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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.
+ */
+
+/*
+ * Chebyshev type 1 filter design based on
+ * "The Scientist and Engineer's Guide to DSP", Chapter 20.
+ * http://www.dspguide.com/
+ *
+ * For type 2 and Chebyshev filters in general read
+ * http://en.wikipedia.org/wiki/Chebyshev_filter
+ *
+ */
+
+/**
+ * SECTION:element-audiochebyshevfreqlimit
+ * @short_description: Chebyshev low pass and high pass filter
+ *
+ * <refsect2>
+ * <para>
+ * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
+ * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
+ * </para>
+ * <para>
+ * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
+ * some frequencies in the passband will be amplified by that value. A higher ripple value will allow
+ * a faster rolloff.
+ * </para>
+ * <para>
+ * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
+ * be at most this value. A lower ripple value will allow a faster rolloff.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/controller/gstcontroller.h>
+
+#include <math.h>
+
+#include "audiochebyshevfreqlimit.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
+ "Filter/Effect/Audio",
+ "Chebyshev low pass and high pass filter",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_MODE,
+ PROP_TYPE,
+ PROP_CUTOFF,
+ PROP_RIPPLE,
+ PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-float," \
+ " width = (int) { 32, 64 }, " \
+ " endianness = (int) BYTE_ORDER," \
+ " rate = (int) [ 1, MAX ]," \
+ " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
+ gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
+ DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
+ GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqLimit * filter,
+ gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqLimit * filter,
+ gfloat * data, guint num_samples);
+
+enum
+{
+ MODE_LOW_PASS = 0,
+ MODE_HIGH_PASS
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_limit_mode_get_type (void)
+{
+ static GType gtype = 0;
+
+ if (gtype == 0) {
+ static const GEnumValue values[] = {
+ {MODE_LOW_PASS, "Low pass (default)",
+ "low-pass"},
+ {MODE_HIGH_PASS, "High pass",
+ "high-pass"},
+ {0, NULL, NULL}
+ };
+
+ gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
+ }
+ return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &element_details);
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+ caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_dispose (GObject * object)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqLimitChannelCtx *ctx;
+ gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
+ klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseTransformClass *trans_class;
+ GstAudioFilterClass *filter_class;
+
+ gobject_class = (GObjectClass *) klass;
+ trans_class = (GstBaseTransformClass *) klass;
+ filter_class = (GstAudioFilterClass *) klass;
+
+ gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
+ gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
+ gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
+
+ g_object_class_install_property (gobject_class, PROP_MODE,
+ g_param_spec_enum ("mode", "Mode",
+ "Low pass or high pass mode",
+ GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_TYPE,
+ g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_CUTOFF,
+ g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
+ G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_RIPPLE,
+ g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
+ G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_POLES,
+ g_param_spec_int ("poles", "Poles",
+ "Number of poles to use, will be rounded up to the next even number",
+ 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ filter_class->setup =
+ GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
+ trans_class->transform_ip =
+ GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
+ trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
+ GstAudioChebyshevFreqLimitClass * klass)
+{
+ filter->cutoff = 0.0;
+ filter->mode = MODE_LOW_PASS;
+ filter->type = 1;
+ filter->poles = 4;
+ filter->ripple = 0.25;
+ gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+ filter->have_coeffs = FALSE;
+ filter->num_a = 0;
+ filter->num_b = 0;
+ filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
+ gint p, gdouble * a0, gdouble * a1, gdouble * a2,
+ gdouble * b1, gdouble * b2)
+{
+ gint np = filter->poles;
+ gdouble ripple = filter->ripple;
+
+ /* pole location in s-plane */
+ gdouble rp, ip;
+
+ /* zero location in s-plane */
+ gdouble rz = 0.0, iz = 0.0;
+
+ /* transfer function coefficients for the z-plane */
+ gdouble x0, x1, x2, y1, y2;
+ gint type = filter->type;
+
+ /* Calculate pole location for lowpass at frequency 1 */
+ {
+ gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
+
+ rp = -sin (angle);
+ ip = cos (angle);
+ }
+
+ /* If we allow ripple, move the pole from the unit
+ * circle to an ellipse and keep cutoff at frequency 1 */
+ if (ripple > 0 && type == 1) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+
+ vx = (1.0 / np) * asinh (1.0 / es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ } else if (type == 2) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+ vx = (1.0 / np) * asinh (es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ }
+
+ /* Calculate inverse of the pole location to convert from
+ * type I to type II */
+ if (type == 2) {
+ gdouble mag2 = rp * rp + ip * ip;
+
+ rp /= mag2;
+ ip /= mag2;
+ }
+
+ /* Calculate zero location for frequency 1 on the
+ * unit circle for type 2 */
+ if (type == 2) {
+ gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
+ gdouble mag2;
+
+ rz = 0.0;
+ iz = cos (angle);
+ mag2 = rz * rz + iz * iz;
+ rz /= mag2;
+ iz /= mag2;
+ }
+
+ /* Convert from s-domain to z-domain by
+ * using the bilinear Z-transform, i.e.
+ * substitute s by (2/t)*((z-1)/(z+1))
+ * with t = 2 * tan(0.5).
+ */
+ if (type == 1) {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t) / d;
+ x1 = 2.0 * x0;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ } else {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t * iz * iz + 4.0) / d;
+ x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ }
+
+ /* Convert from lowpass at frequency 1 to either lowpass
+ * or highpass.
+ *
+ * For lowpass substitute z^(-1) with:
+ * -1
+ * z - k
+ * ------------
+ * -1
+ * 1 - k * z
+ *
+ * k = sin((1-w)/2) / sin((1+w)/2)
+ *
+ * For highpass substitute z^(-1) with:
+ *
+ * -1
+ * -z - k
+ * ------------
+ * -1
+ * 1 + k * z
+ *
+ * k = -cos((1+w)/2) / cos((1-w)/2)
+ *
+ */
+ {
+ gdouble k, d;
+ gdouble omega =
+ 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
+
+ if (filter->mode == MODE_LOW_PASS)
+ k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
+ else
+ k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
+
+ d = 1.0 + y1 * k - y2 * k * k;
+ *a0 = (x0 + k * (-x1 + k * x2)) / d;
+ *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
+ *a2 = (x0 * k * k - x1 * k + x2) / d;
+ *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
+ *b2 = (-k * k - y1 * k + y2) / d;
+
+ if (filter->mode == MODE_HIGH_PASS) {
+ *a1 = -*a1;
+ *b1 = -*b1;
+ }
+ }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+ gdouble zi)
+{
+ gdouble sum_ar, sum_ai;
+ gdouble sum_br, sum_bi;
+ gdouble gain_r, gain_i;
+
+ gdouble sum_r_old;
+ gdouble sum_i_old;
+
+ gint i;
+
+ sum_ar = 0.0;
+ sum_ai = 0.0;
+ for (i = num_a; i >= 0; i--) {
+ sum_r_old = sum_ar;
+ sum_i_old = sum_ai;
+
+ sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+ sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+ }
+
+ sum_br = 0.0;
+ sum_bi = 0.0;
+ for (i = num_b; i >= 0; i--) {
+ sum_r_old = sum_br;
+ sum_i_old = sum_bi;
+
+ sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+ sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+ }
+ sum_br += 1.0;
+ sum_bi += 0.0;
+
+ gain_r =
+ (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+ gain_i =
+ (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+ return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqLimit * filter)
+{
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqLimitChannelCtx *ctx;
+ gint i;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "rate was not set yet");
+ return;
+ }
+
+ filter->have_coeffs = TRUE;
+
+ if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
+ return;
+ } else if (filter->cutoff <= 0.0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "cutoff is lower than zero");
+ return;
+ }
+
+ /* Calculate coefficients for the chebyshev filter */
+ {
+ gint np = filter->poles;
+ gdouble *a, *b;
+ gint i, p;
+
+ filter->num_a = np + 1;
+ filter->a = a = g_new0 (gdouble, np + 3);
+ filter->num_b = np + 1;
+ filter->b = b = g_new0 (gdouble, np + 3);
+
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ for (i = 0; i < channels; i++) {
+ GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
+
+ ctx->x = g_new0 (gdouble, np + 1);
+ ctx->y = g_new0 (gdouble, np + 1);
+ }
+
+ /* Calculate transfer function coefficients */
+ a[2] = 1.0;
+ b[2] = 1.0;
+
+ for (p = 1; p <= np / 2; p++) {
+ gdouble a0, a1, a2, b1, b2;
+ gdouble *ta = g_new0 (gdouble, np + 3);
+ gdouble *tb = g_new0 (gdouble, np + 3);
+
+ generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
+
+ memcpy (ta, a, sizeof (gdouble) * (np + 3));
+ memcpy (tb, b, sizeof (gdouble) * (np + 3));
+
+ /* add the new coefficients for the new two poles
+ * to the cascade by multiplication of the transfer
+ * functions */
+ for (i = 2; i < np + 3; i++) {
+ a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
+ b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
+ }
+ g_free (ta);
+ g_free (tb);
+ }
+
+ /* Move coefficients to the beginning of the array
+ * and multiply the b coefficients with -1 to move from
+ * the transfer function's coefficients to the difference
+ * equation's coefficients */
+ b[2] = 0.0;
+ for (i = 0; i <= np; i++) {
+ a[i] = a[i + 2];
+ b[i] = -b[i + 2];
+ }
+
+ /* Normalize to unity gain at frequency 0 for lowpass
+ * and frequency 0.5 for highpass */
+ {
+ gdouble gain;
+
+ if (filter->mode == MODE_LOW_PASS)
+ gain = calculate_gain (a, b, np, np, 1.0, 0.0);
+ else
+ gain = calculate_gain (a, b, np, np, -1.0, 0.0);
+
+ for (i = 0; i <= np; i++) {
+ a[i] /= gain;
+ }
+ }
+
+ GST_LOG_OBJECT (filter,
+ "Generated IIR coefficients for the Chebyshev filter");
+ GST_LOG_OBJECT (filter,
+ "mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB",
+ (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
+ filter->type, filter->poles, filter->cutoff, filter->ripple);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+ {
+ gdouble wc =
+ 2.0 * M_PI * (filter->cutoff /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble zr = cos (wc), zi = sin (wc);
+
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) filter->cutoff);
+ }
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+ GST_AUDIO_FILTER (filter)->format.rate / 2);
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->mode = g_value_get_enum (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_TYPE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->type = g_value_get_int (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_CUTOFF:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->cutoff = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_RIPPLE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->ripple = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_POLES:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ g_value_set_enum (value, filter->mode);
+ break;
+ case PROP_TYPE:
+ g_value_set_int (value, filter->type);
+ break;
+ case PROP_CUTOFF:
+ g_value_set_float (value, filter->cutoff);
+ break;
+ case PROP_RIPPLE:
+ g_value_set_float (value, filter->ripple);
+ break;
+ case PROP_POLES:
+ g_value_set_int (value, filter->poles);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
+ GstRingBufferSpec * format)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+ gboolean ret = TRUE;
+
+ if (format->width == 32)
+ filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+ process_32;
+ else if (format->width == 64)
+ filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+ process_64;
+ else
+ ret = FALSE;
+
+ filter->have_coeffs = FALSE;
+
+ return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqLimit * filter,
+ GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
+{
+ gdouble val = filter->a[0] * x0;
+ gint i, j;
+
+ for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+ val += filter->a[i] * ctx->x[j];
+ j--;
+ if (j < 0)
+ j = filter->num_a - 1;
+ }
+
+ for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+ val += filter->b[i] * ctx->y[j];
+ j--;
+ if (j < 0)
+ j = filter->num_b - 1;
+ }
+
+ if (ctx->x) {
+ ctx->x_pos++;
+ if (ctx->x_pos > filter->num_a - 1)
+ ctx->x_pos = 0;
+ ctx->x[ctx->x_pos] = x0;
+ }
+
+ if (ctx->y) {
+ ctx->y_pos++;
+ if (ctx->y_pos > filter->num_b - 1)
+ ctx->y_pos = 0;
+
+ ctx->y[ctx->y_pos] = val;
+ }
+
+ return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqLimit * filter,
+ gdouble * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqLimit * filter,
+ gfloat * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+ guint num_samples =
+ GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+ if (!gst_buffer_is_writable (buf))
+ return GST_FLOW_OK;
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+ gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+ if (!filter->have_coeffs)
+ generate_coefficients (filter);
+
+ filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}
+
+
+static gboolean
+gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+ GstAudioChebyshevFreqLimitChannelCtx *ctx;
+ gint i;
+
+ /* Reset the history of input and output values if
+ * already existing */
+ if (channels && filter->channels) {
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ if (ctx->x)
+ memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+ if (ctx->y)
+ memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+ }
+ }
+ return TRUE;
+}
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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 __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit;
+typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass;
+
+typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint);
+
+typedef struct
+{
+ gdouble *x;
+ gint x_pos;
+ gdouble *y;
+ gint y_pos;
+} GstAudioChebyshevFreqLimitChannelCtx;
+
+struct _GstAudioChebyshevFreqLimit
+{
+ GstAudioFilter audiofilter;
+
+ gint mode;
+ gint type;
+ gint poles;
+ gfloat cutoff;
+ gfloat ripple;
+
+ /* < private > */
+ GstAudioChebyshevFreqLimitProcessFunc process;
+
+ gboolean have_coeffs;
+ gdouble *a;
+ gint num_a;
+ gdouble *b;
+ gint num_b;
+ GstAudioChebyshevFreqLimitChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqLimitClass
+{
+ GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_limit_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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.
+ */
+
+/*
+ * Chebyshev type 1 filter design based on
+ * "The Scientist and Engineer's Guide to DSP", Chapter 20.
+ * http://www.dspguide.com/
+ *
+ * For type 2 and Chebyshev filters in general read
+ * http://en.wikipedia.org/wiki/Chebyshev_filter
+ *
+ * Transformation from lowpass to bandpass/bandreject:
+ * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm
+ * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm
+ *
+ */
+
+/**
+ * SECTION:element-audiochebyshevfreqband
+ * @short_description: Chebyshev band pass and band reject filter
+ *
+ * <refsect2>
+ * <para>
+ * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
+ * band. The number of poles and the ripple parameter control the rolloff.
+ * </para>
+ * <para>
+ * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
+ * some frequencies in the passband will be amplified by that value. A higher ripple value will allow
+ * a faster rolloff.
+ * </para>
+ * <para>
+ * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
+ * be at most this value. A lower ripple value will allow a faster rolloff.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/controller/gstcontroller.h>
+
+#include <math.h>
+
+#include "audiochebyshevfreqband.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
+ "Filter/Effect/Audio",
+ "Chebyshev band pass and band reject filter",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_MODE,
+ PROP_TYPE,
+ PROP_LOWER_FREQUENCY,
+ PROP_UPPER_FREQUENCY,
+ PROP_RIPPLE,
+ PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-float," \
+ " width = (int) { 32, 64 }, " \
+ " endianness = (int) BYTE_ORDER," \
+ " rate = (int) [ 1, MAX ]," \
+ " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
+ GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
+ GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqBand * filter,
+ gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqBand * filter,
+ gfloat * data, guint num_samples);
+
+enum
+{
+ MODE_BAND_PASS = 0,
+ MODE_BAND_REJECT
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_band_mode_get_type (void)
+{
+ static GType gtype = 0;
+
+ if (gtype == 0) {
+ static const GEnumValue values[] = {
+ {MODE_BAND_PASS, "Band pass (default)",
+ "band-pass"},
+ {MODE_BAND_REJECT, "Band reject",
+ "band-reject"},
+ {0, NULL, NULL}
+ };
+
+ gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values);
+ }
+ return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_band_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &element_details);
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+ caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_band_dispose (GObject * object)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqBandChannelCtx *ctx;
+ gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
+ klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseTransformClass *trans_class;
+ GstAudioFilterClass *filter_class;
+
+ gobject_class = (GObjectClass *) klass;
+ trans_class = (GstBaseTransformClass *) klass;
+ filter_class = (GstAudioFilterClass *) klass;
+
+ gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property;
+ gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property;
+ gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose;
+
+ g_object_class_install_property (gobject_class, PROP_MODE,
+ g_param_spec_enum ("mode", "Mode",
+ "Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE,
+ MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_TYPE,
+ g_param_spec_int ("type", "Type",
+ "Type of the chebychev filter", 1, 2,
+ 1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
+ g_param_spec_float ("lower-frequency", "Lower frequency",
+ "Start frequency of the band (Hz)", 0.0, G_MAXFLOAT,
+ 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
+ g_param_spec_float ("upper-frequency", "Upper frequency",
+ "Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT,
+ 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_RIPPLE,
+ g_param_spec_float ("ripple", "Ripple",
+ "Amount of ripple (dB)", 0.0, G_MAXFLOAT,
+ 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_POLES,
+ g_param_spec_int ("poles", "Poles",
+ "Number of poles to use, will be rounded up to the next multiply of four",
+ 4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
+ trans_class->transform_ip =
+ GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
+ trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
+}
+
+static void
+gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
+ GstAudioChebyshevFreqBandClass * klass)
+{
+ filter->lower_frequency = filter->upper_frequency = 0.0;
+ filter->mode = MODE_BAND_PASS;
+ filter->type = 1;
+ filter->poles = 4;
+ filter->ripple = 0.25;
+ gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+ filter->have_coeffs = FALSE;
+ filter->num_a = 0;
+ filter->num_b = 0;
+ filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter,
+ gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
+ gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
+{
+ gint np = filter->poles / 2;
+ gdouble ripple = filter->ripple;
+
+ /* pole location in s-plane */
+ gdouble rp, ip;
+
+ /* zero location in s-plane */
+ gdouble rz = 0.0, iz = 0.0;
+
+ /* transfer function coefficients for the z-plane */
+ gdouble x0, x1, x2, y1, y2;
+ gint type = filter->type;
+
+ /* Calculate pole location for lowpass at frequency 1 */
+ {
+ gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
+
+ rp = -sin (angle);
+ ip = cos (angle);
+ }
+
+ /* If we allow ripple, move the pole from the unit
+ * circle to an ellipse and keep cutoff at frequency 1 */
+ if (ripple > 0 && type == 1) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+
+ vx = (1.0 / np) * asinh (1.0 / es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ } else if (type == 2) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+ vx = (1.0 / np) * asinh (es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ }
+
+ /* Calculate inverse of the pole location to move from
+ * type I to type II */
+ if (type == 2) {
+ gdouble mag2 = rp * rp + ip * ip;
+
+ rp /= mag2;
+ ip /= mag2;
+ }
+
+ /* Calculate zero location for frequency 1 on the
+ * unit circle for type 2 */
+ if (type == 2) {
+ gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
+ gdouble mag2;
+
+ rz = 0.0;
+ iz = cos (angle);
+ mag2 = rz * rz + iz * iz;
+ rz /= mag2;
+ iz /= mag2;
+ }
+
+ /* Convert from s-domain to z-domain by
+ * using the bilinear Z-transform, i.e.
+ * substitute s by (2/t)*((z-1)/(z+1))
+ * with t = 2 * tan(0.5).
+ */
+ if (type == 1) {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t) / d;
+ x1 = 2.0 * x0;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ } else {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t * iz * iz + 4.0) / d;
+ x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ }
+
+ /* Convert from lowpass at frequency 1 to either bandpass
+ * or band reject.
+ *
+ * For bandpass substitute z^(-1) with:
+ *
+ * -2 -1
+ * -z + alpha * z - beta
+ * ----------------------------
+ * -2 -1
+ * beta * z - alpha * z + 1
+ *
+ * alpha = (2*a*b)/(1+b)
+ * beta = (b-1)/(b+1)
+ * a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
+ * b = tan(1/2) * cot((w1 - w0)/2)
+ *
+ * For bandreject substitute z^(-1) with:
+ *
+ * -2 -1
+ * z - alpha * z + beta
+ * ----------------------------
+ * -2 -1
+ * beta * z - alpha * z + 1
+ *
+ * alpha = (2*a)/(1+b)
+ * beta = (1-b)/(1+b)
+ * a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
+ * b = tan(1/2) * tan((w1 - w0)/2)
+ *
+ */
+ {
+ gdouble a, b, d;
+ gdouble alpha, beta;
+ gdouble w0 =
+ 2.0 * M_PI * (filter->lower_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w1 =
+ 2.0 * M_PI * (filter->upper_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+
+ if (filter->mode == MODE_BAND_PASS) {
+ a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
+ b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0);
+
+ alpha = (2.0 * a * b) / (1.0 + b);
+ beta = (b - 1.0) / (b + 1.0);
+
+ d = 1.0 + beta * (y1 - beta * y2);
+
+ *a0 = (x0 + beta * (-x1 + beta * x2)) / d;
+ *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d;
+ *a2 =
+ (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) +
+ alpha * alpha * (x0 - x1 + x2)) / d;
+ *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d;
+ *a4 = (beta * (beta * x0 - x1) + x2) / d;
+ *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d;
+ *b2 =
+ (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) +
+ 2.0 * beta * (-1.0 + y2)) / d;
+ *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d;
+ *b4 = (-beta * beta - beta * y1 + y2) / d;
+ } else {
+ a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
+ b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0);
+
+ alpha = (2.0 * a) / (1.0 + b);
+ beta = (1.0 - b) / (1.0 + b);
+
+ d = -1.0 + beta * (beta * y2 + y1);
+
+ *a0 = (-x0 - beta * x1 - beta * beta * x2) / d;
+ *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d;
+ *a2 =
+ (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) -
+ alpha * alpha * (x0 + x1 + x2)) / d;
+ *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d;
+ *a4 = (-beta * beta * x0 - beta * x1 - x2) / d;
+ *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d;
+ *b2 =
+ -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) +
+ alpha * alpha * (-1.0 + y1 + y2)) / d;
+ *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d;
+ *b4 = -(-beta * beta + beta * y1 + y2) / d;
+ }
+ }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+ gdouble zi)
+{
+ gdouble sum_ar, sum_ai;
+ gdouble sum_br, sum_bi;
+ gdouble gain_r, gain_i;
+
+ gdouble sum_r_old;
+ gdouble sum_i_old;
+
+ gint i;
+
+ sum_ar = 0.0;
+ sum_ai = 0.0;
+ for (i = num_a; i >= 0; i--) {
+ sum_r_old = sum_ar;
+ sum_i_old = sum_ai;
+
+ sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+ sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+ }
+
+ sum_br = 0.0;
+ sum_bi = 0.0;
+ for (i = num_b; i >= 0; i--) {
+ sum_r_old = sum_br;
+ sum_i_old = sum_bi;
+
+ sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+ sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+ }
+ sum_br += 1.0;
+ sum_bi += 0.0;
+
+ gain_r =
+ (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+ gain_i =
+ (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+ return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqBand * filter)
+{
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqBandChannelCtx *ctx;
+ gint i;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "rate was not set yet");
+ return;
+ }
+
+ filter->have_coeffs = TRUE;
+
+ if (filter->upper_frequency <= filter->lower_frequency) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
+ return;
+ }
+
+ if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) {
+ filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2;
+ GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency");
+ }
+
+ if (filter->lower_frequency < 0.0) {
+ filter->lower_frequency = 0.0;
+ GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0");
+ }
+
+ /* Calculate coefficients for the chebyshev filter */
+ {
+ gint np = filter->poles;
+ gdouble *a, *b;
+ gint i, p;
+
+ filter->num_a = np + 1;
+ filter->a = a = g_new0 (gdouble, np + 5);
+ filter->num_b = np + 1;
+ filter->b = b = g_new0 (gdouble, np + 5);
+
+ filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+ for (i = 0; i < channels; i++) {
+ GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
+
+ ctx->x = g_new0 (gdouble, np + 1);
+ ctx->y = g_new0 (gdouble, np + 1);
+ }
+
+ /* Calculate transfer function coefficients */
+ a[4] = 1.0;
+ b[4] = 1.0;
+
+ for (p = 1; p <= np / 4; p++) {
+ gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4;
+ gdouble *ta = g_new0 (gdouble, np + 5);
+ gdouble *tb = g_new0 (gdouble, np + 5);
+
+ generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1,
+ &b2, &b3, &b4);
+
+ memcpy (ta, a, sizeof (gdouble) * (np + 5));
+ memcpy (tb, b, sizeof (gdouble) * (np + 5));
+
+ /* add the new coefficients for the new two poles
+ * to the cascade by multiplication of the transfer
+ * functions */
+ for (i = 4; i < np + 5; i++) {
+ a[i] =
+ a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] +
+ a4 * ta[i - 4];
+ b[i] =
+ tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] -
+ b4 * tb[i - 4];
+ }
+ g_free (ta);
+ g_free (tb);
+ }
+
+ /* Move coefficients to the beginning of the array
+ * and multiply the b coefficients with -1 to move from
+ * the transfer function's coefficients to the difference
+ * equation's coefficients */
+ b[4] = 0.0;
+ for (i = 0; i <= np; i++) {
+ a[i] = a[i + 4];
+ b[i] = -b[i + 4];
+ }
+
+ /* Normalize to unity gain at frequency 0 and frequency
+ * 0.5 for bandreject and unity gain at band center frequency
+ * for bandpass */
+ if (filter->mode == MODE_BAND_REJECT) {
+ /* gain is sqrt(H(0)*H(0.5)) */
+
+ gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0);
+ gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0);
+
+ gain1 = sqrt (gain1 * gain2);
+
+ for (i = 0; i <= np; i++) {
+ a[i] /= gain1;
+ }
+ } else {
+ /* gain is H(wc), wc = center frequency */
+
+ gdouble w1 =
+ 2.0 * M_PI * (filter->lower_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w2 =
+ 2.0 * M_PI * (filter->upper_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w0 = (w2 + w1) / 2.0;
+ gdouble zr = cos (w0), zi = sin (w0);
+ gdouble gain = calculate_gain (a, b, np, np, zr, zi);
+
+ for (i = 0; i <= np; i++) {
+ a[i] /= gain;
+ }
+ }
+
+ GST_LOG_OBJECT (filter,
+ "Generated IIR coefficients for the Chebyshev filter");
+ GST_LOG_OBJECT (filter,
+ "mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB",
+ (filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject",
+ filter->type, filter->poles, filter->lower_frequency,
+ filter->upper_frequency, filter->ripple);
+
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+ {
+ gdouble w1 =
+ 2.0 * M_PI * (filter->lower_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w2 =
+ 2.0 * M_PI * (filter->upper_frequency /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble w0 = (w2 + w1) / 2.0;
+ gdouble zr, zi;
+
+ zr = cos (w1);
+ zi = sin (w1);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) filter->lower_frequency);
+ zr = cos (w0);
+ zi = sin (w0);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) ((filter->lower_frequency + filter->upper_frequency) / 2.0));
+ zr = cos (w2);
+ zi = sin (w2);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) filter->upper_frequency);
+ }
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+ GST_AUDIO_FILTER (filter)->format.rate / 2);
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->mode = g_value_get_enum (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_TYPE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->type = g_value_get_int (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_LOWER_FREQUENCY:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->lower_frequency = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_UPPER_FREQUENCY:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->upper_frequency = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_RIPPLE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->ripple = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_POLES:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->poles = GST_ROUND_UP_4 (g_value_get_int (value));
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ g_value_set_enum (value, filter->mode);
+ break;
+ case PROP_TYPE:
+ g_value_set_int (value, filter->type);
+ break;
+ case PROP_LOWER_FREQUENCY:
+ g_value_set_float (value, filter->lower_frequency);
+ break;
+ case PROP_UPPER_FREQUENCY:
+ g_value_set_float (value, filter->upper_frequency);
+ break;
+ case PROP_RIPPLE:
+ g_value_set_float (value, filter->ripple);
+ break;
+ case PROP_POLES:
+ g_value_set_int (value, filter->poles);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
+ GstRingBufferSpec * format)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+ gboolean ret = TRUE;
+
+ if (format->width == 32)
+ filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+ process_32;
+ else if (format->width == 64)
+ filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+ process_64;
+ else
+ ret = FALSE;
+
+ filter->have_coeffs = FALSE;
+
+ return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqBand * filter,
+ GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
+{
+ gdouble val = filter->a[0] * x0;
+ gint i, j;
+
+ for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+ val += filter->a[i] * ctx->x[j];
+ j--;
+ if (j < 0)
+ j = filter->num_a - 1;
+ }
+
+ for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+ val += filter->b[i] * ctx->y[j];
+ j--;
+ if (j < 0)
+ j = filter->num_b - 1;
+ }
+
+ if (ctx->x) {
+ ctx->x_pos++;
+ if (ctx->x_pos > filter->num_a - 1)
+ ctx->x_pos = 0;
+ ctx->x[ctx->x_pos] = x0;
+ }
+
+ if (ctx->y) {
+ ctx->y_pos++;
+ if (ctx->y_pos > filter->num_b - 1)
+ ctx->y_pos = 0;
+
+ ctx->y[ctx->y_pos] = val;
+ }
+
+ return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqBand * filter,
+ gdouble * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqBand * filter,
+ gfloat * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+ guint num_samples =
+ GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+ if (!gst_buffer_is_writable (buf))
+ return GST_FLOW_OK;
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+ gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+ if (!filter->have_coeffs)
+ generate_coefficients (filter);
+
+ filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}
+
+static gboolean
+gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
+{
+ GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+ GstAudioChebyshevFreqBandChannelCtx *ctx;
+ gint i;
+
+ /* Reset the history of input and output values if
+ * already existing */
+ if (channels && filter->channels) {
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ if (ctx->x)
+ memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+ if (ctx->y)
+ memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+ }
+ }
+ return TRUE;
+}
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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 __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand;
+typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass;
+
+typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint);
+
+typedef struct
+{
+ gdouble *x;
+ gint x_pos;
+ gdouble *y;
+ gint y_pos;
+} GstAudioChebyshevFreqBandChannelCtx;
+
+struct _GstAudioChebyshevFreqBand
+{
+ GstAudioFilter audiofilter;
+
+ gint mode;
+ gint type;
+ gint poles;
+ gfloat lower_frequency;
+ gfloat upper_frequency;
+ gfloat ripple;
+
+ /* < private > */
+ GstAudioChebyshevFreqBandProcessFunc process;
+
+ gboolean have_coeffs;
+ gdouble *a;
+ gint num_a;
+ gdouble *b;
+ gint num_b;
+ GstAudioChebyshevFreqBandChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqBandClass
+{
+ GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_band_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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.
+ */
+
+/*
+ * Chebyshev type 1 filter design based on
+ * "The Scientist and Engineer's Guide to DSP", Chapter 20.
+ * http://www.dspguide.com/
+ *
+ * For type 2 and Chebyshev filters in general read
+ * http://en.wikipedia.org/wiki/Chebyshev_filter
+ *
+ */
+
+/**
+ * SECTION:element-audiochebyshevfreqlimit
+ * @short_description: Chebyshev low pass and high pass filter
+ *
+ * <refsect2>
+ * <para>
+ * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
+ * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
+ * </para>
+ * <para>
+ * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
+ * some frequencies in the passband will be amplified by that value. A higher ripple value will allow
+ * a faster rolloff.
+ * </para>
+ * <para>
+ * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
+ * be at most this value. A lower ripple value will allow a faster rolloff.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/controller/gstcontroller.h>
+
+#include <math.h>
+
+#include "audiochebyshevfreqlimit.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
+ "Filter/Effect/Audio",
+ "Chebyshev low pass and high pass filter",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_MODE,
+ PROP_TYPE,
+ PROP_CUTOFF,
+ PROP_RIPPLE,
+ PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+ "audio/x-raw-float," \
+ " width = (int) { 32, 64 }, " \
+ " endianness = (int) BYTE_ORDER," \
+ " rate = (int) [ 1, MAX ]," \
+ " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
+ gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
+ DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
+ GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqLimit * filter,
+ gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqLimit * filter,
+ gfloat * data, guint num_samples);
+
+enum
+{
+ MODE_LOW_PASS = 0,
+ MODE_HIGH_PASS
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_limit_mode_get_type (void)
+{
+ static GType gtype = 0;
+
+ if (gtype == 0) {
+ static const GEnumValue values[] = {
+ {MODE_LOW_PASS, "Low pass (default)",
+ "low-pass"},
+ {MODE_HIGH_PASS, "High pass",
+ "high-pass"},
+ {0, NULL, NULL}
+ };
+
+ gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
+ }
+ return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &element_details);
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+ caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_dispose (GObject * object)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqLimitChannelCtx *ctx;
+ gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
+ klass)
+{
+ GObjectClass *gobject_class;
+ GstBaseTransformClass *trans_class;
+ GstAudioFilterClass *filter_class;
+
+ gobject_class = (GObjectClass *) klass;
+ trans_class = (GstBaseTransformClass *) klass;
+ filter_class = (GstAudioFilterClass *) klass;
+
+ gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
+ gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
+ gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
+
+ g_object_class_install_property (gobject_class, PROP_MODE,
+ g_param_spec_enum ("mode", "Mode",
+ "Low pass or high pass mode",
+ GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_TYPE,
+ g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_CUTOFF,
+ g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
+ G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_RIPPLE,
+ g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
+ G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ g_object_class_install_property (gobject_class, PROP_POLES,
+ g_param_spec_int ("poles", "Poles",
+ "Number of poles to use, will be rounded up to the next even number",
+ 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ filter_class->setup =
+ GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
+ trans_class->transform_ip =
+ GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
+ trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
+ GstAudioChebyshevFreqLimitClass * klass)
+{
+ filter->cutoff = 0.0;
+ filter->mode = MODE_LOW_PASS;
+ filter->type = 1;
+ filter->poles = 4;
+ filter->ripple = 0.25;
+ gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+ filter->have_coeffs = FALSE;
+ filter->num_a = 0;
+ filter->num_b = 0;
+ filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
+ gint p, gdouble * a0, gdouble * a1, gdouble * a2,
+ gdouble * b1, gdouble * b2)
+{
+ gint np = filter->poles;
+ gdouble ripple = filter->ripple;
+
+ /* pole location in s-plane */
+ gdouble rp, ip;
+
+ /* zero location in s-plane */
+ gdouble rz = 0.0, iz = 0.0;
+
+ /* transfer function coefficients for the z-plane */
+ gdouble x0, x1, x2, y1, y2;
+ gint type = filter->type;
+
+ /* Calculate pole location for lowpass at frequency 1 */
+ {
+ gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
+
+ rp = -sin (angle);
+ ip = cos (angle);
+ }
+
+ /* If we allow ripple, move the pole from the unit
+ * circle to an ellipse and keep cutoff at frequency 1 */
+ if (ripple > 0 && type == 1) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+
+ vx = (1.0 / np) * asinh (1.0 / es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ } else if (type == 2) {
+ gdouble es, vx;
+
+ es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
+ vx = (1.0 / np) * asinh (es);
+ rp = rp * sinh (vx);
+ ip = ip * cosh (vx);
+ }
+
+ /* Calculate inverse of the pole location to convert from
+ * type I to type II */
+ if (type == 2) {
+ gdouble mag2 = rp * rp + ip * ip;
+
+ rp /= mag2;
+ ip /= mag2;
+ }
+
+ /* Calculate zero location for frequency 1 on the
+ * unit circle for type 2 */
+ if (type == 2) {
+ gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
+ gdouble mag2;
+
+ rz = 0.0;
+ iz = cos (angle);
+ mag2 = rz * rz + iz * iz;
+ rz /= mag2;
+ iz /= mag2;
+ }
+
+ /* Convert from s-domain to z-domain by
+ * using the bilinear Z-transform, i.e.
+ * substitute s by (2/t)*((z-1)/(z+1))
+ * with t = 2 * tan(0.5).
+ */
+ if (type == 1) {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t) / d;
+ x1 = 2.0 * x0;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ } else {
+ gdouble t, m, d;
+
+ t = 2.0 * tan (0.5);
+ m = rp * rp + ip * ip;
+ d = 4.0 - 4.0 * rp * t + m * t * t;
+
+ x0 = (t * t * iz * iz + 4.0) / d;
+ x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
+ x2 = x0;
+ y1 = (8.0 - 2.0 * m * t * t) / d;
+ y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
+ }
+
+ /* Convert from lowpass at frequency 1 to either lowpass
+ * or highpass.
+ *
+ * For lowpass substitute z^(-1) with:
+ * -1
+ * z - k
+ * ------------
+ * -1
+ * 1 - k * z
+ *
+ * k = sin((1-w)/2) / sin((1+w)/2)
+ *
+ * For highpass substitute z^(-1) with:
+ *
+ * -1
+ * -z - k
+ * ------------
+ * -1
+ * 1 + k * z
+ *
+ * k = -cos((1+w)/2) / cos((1-w)/2)
+ *
+ */
+ {
+ gdouble k, d;
+ gdouble omega =
+ 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
+
+ if (filter->mode == MODE_LOW_PASS)
+ k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
+ else
+ k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
+
+ d = 1.0 + y1 * k - y2 * k * k;
+ *a0 = (x0 + k * (-x1 + k * x2)) / d;
+ *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
+ *a2 = (x0 * k * k - x1 * k + x2) / d;
+ *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
+ *b2 = (-k * k - y1 * k + y2) / d;
+
+ if (filter->mode == MODE_HIGH_PASS) {
+ *a1 = -*a1;
+ *b1 = -*b1;
+ }
+ }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+ gdouble zi)
+{
+ gdouble sum_ar, sum_ai;
+ gdouble sum_br, sum_bi;
+ gdouble gain_r, gain_i;
+
+ gdouble sum_r_old;
+ gdouble sum_i_old;
+
+ gint i;
+
+ sum_ar = 0.0;
+ sum_ai = 0.0;
+ for (i = num_a; i >= 0; i--) {
+ sum_r_old = sum_ar;
+ sum_i_old = sum_ai;
+
+ sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+ sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+ }
+
+ sum_br = 0.0;
+ sum_bi = 0.0;
+ for (i = num_b; i >= 0; i--) {
+ sum_r_old = sum_br;
+ sum_i_old = sum_bi;
+
+ sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+ sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+ }
+ sum_br += 1.0;
+ sum_bi += 0.0;
+
+ gain_r =
+ (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+ gain_i =
+ (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+ return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqLimit * filter)
+{
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+ if (filter->a) {
+ g_free (filter->a);
+ filter->a = NULL;
+ }
+
+ if (filter->b) {
+ g_free (filter->b);
+ filter->b = NULL;
+ }
+
+ if (filter->channels) {
+ GstAudioChebyshevFreqLimitChannelCtx *ctx;
+ gint i;
+
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ g_free (ctx->x);
+ g_free (ctx->y);
+ }
+
+ g_free (filter->channels);
+ filter->channels = NULL;
+ }
+
+ if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "rate was not set yet");
+ return;
+ }
+
+ filter->have_coeffs = TRUE;
+
+ if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
+ return;
+ } else if (filter->cutoff <= 0.0) {
+ filter->num_a = 1;
+ filter->a = g_new0 (gdouble, 1);
+ filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
+ filter->num_b = 0;
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ GST_LOG_OBJECT (filter, "cutoff is lower than zero");
+ return;
+ }
+
+ /* Calculate coefficients for the chebyshev filter */
+ {
+ gint np = filter->poles;
+ gdouble *a, *b;
+ gint i, p;
+
+ filter->num_a = np + 1;
+ filter->a = a = g_new0 (gdouble, np + 3);
+ filter->num_b = np + 1;
+ filter->b = b = g_new0 (gdouble, np + 3);
+
+ filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ for (i = 0; i < channels; i++) {
+ GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
+
+ ctx->x = g_new0 (gdouble, np + 1);
+ ctx->y = g_new0 (gdouble, np + 1);
+ }
+
+ /* Calculate transfer function coefficients */
+ a[2] = 1.0;
+ b[2] = 1.0;
+
+ for (p = 1; p <= np / 2; p++) {
+ gdouble a0, a1, a2, b1, b2;
+ gdouble *ta = g_new0 (gdouble, np + 3);
+ gdouble *tb = g_new0 (gdouble, np + 3);
+
+ generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
+
+ memcpy (ta, a, sizeof (gdouble) * (np + 3));
+ memcpy (tb, b, sizeof (gdouble) * (np + 3));
+
+ /* add the new coefficients for the new two poles
+ * to the cascade by multiplication of the transfer
+ * functions */
+ for (i = 2; i < np + 3; i++) {
+ a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
+ b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
+ }
+ g_free (ta);
+ g_free (tb);
+ }
+
+ /* Move coefficients to the beginning of the array
+ * and multiply the b coefficients with -1 to move from
+ * the transfer function's coefficients to the difference
+ * equation's coefficients */
+ b[2] = 0.0;
+ for (i = 0; i <= np; i++) {
+ a[i] = a[i + 2];
+ b[i] = -b[i + 2];
+ }
+
+ /* Normalize to unity gain at frequency 0 for lowpass
+ * and frequency 0.5 for highpass */
+ {
+ gdouble gain;
+
+ if (filter->mode == MODE_LOW_PASS)
+ gain = calculate_gain (a, b, np, np, 1.0, 0.0);
+ else
+ gain = calculate_gain (a, b, np, np, -1.0, 0.0);
+
+ for (i = 0; i <= np; i++) {
+ a[i] /= gain;
+ }
+ }
+
+ GST_LOG_OBJECT (filter,
+ "Generated IIR coefficients for the Chebyshev filter");
+ GST_LOG_OBJECT (filter,
+ "mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB",
+ (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
+ filter->type, filter->poles, filter->cutoff, filter->ripple);
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+ {
+ gdouble wc =
+ 2.0 * M_PI * (filter->cutoff /
+ GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble zr = cos (wc), zi = sin (wc);
+
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+ (int) filter->cutoff);
+ }
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+ 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+ GST_AUDIO_FILTER (filter)->format.rate / 2);
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->mode = g_value_get_enum (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_TYPE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->type = g_value_get_int (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_CUTOFF:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->cutoff = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_RIPPLE:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->ripple = g_value_get_float (value);
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ case PROP_POLES:
+ GST_BASE_TRANSFORM_LOCK (filter);
+ filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
+ generate_coefficients (filter);
+ GST_BASE_TRANSFORM_UNLOCK (filter);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+ switch (prop_id) {
+ case PROP_MODE:
+ g_value_set_enum (value, filter->mode);
+ break;
+ case PROP_TYPE:
+ g_value_set_int (value, filter->type);
+ break;
+ case PROP_CUTOFF:
+ g_value_set_float (value, filter->cutoff);
+ break;
+ case PROP_RIPPLE:
+ g_value_set_float (value, filter->ripple);
+ break;
+ case PROP_POLES:
+ g_value_set_int (value, filter->poles);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
+ GstRingBufferSpec * format)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+ gboolean ret = TRUE;
+
+ if (format->width == 32)
+ filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+ process_32;
+ else if (format->width == 64)
+ filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+ process_64;
+ else
+ ret = FALSE;
+
+ filter->have_coeffs = FALSE;
+
+ return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqLimit * filter,
+ GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
+{
+ gdouble val = filter->a[0] * x0;
+ gint i, j;
+
+ for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+ val += filter->a[i] * ctx->x[j];
+ j--;
+ if (j < 0)
+ j = filter->num_a - 1;
+ }
+
+ for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+ val += filter->b[i] * ctx->y[j];
+ j--;
+ if (j < 0)
+ j = filter->num_b - 1;
+ }
+
+ if (ctx->x) {
+ ctx->x_pos++;
+ if (ctx->x_pos > filter->num_a - 1)
+ ctx->x_pos = 0;
+ ctx->x[ctx->x_pos] = x0;
+ }
+
+ if (ctx->y) {
+ ctx->y_pos++;
+ if (ctx->y_pos > filter->num_b - 1)
+ ctx->y_pos = 0;
+
+ ctx->y[ctx->y_pos] = val;
+ }
+
+ return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqLimit * filter,
+ gdouble * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqLimit * filter,
+ gfloat * data, guint num_samples)
+{
+ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+ gdouble val;
+
+ for (i = 0; i < num_samples / channels; i++) {
+ for (j = 0; j < channels; j++) {
+ val = process (filter, &filter->channels[j], *data);
+ *data++ = val;
+ }
+ }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+ guint num_samples =
+ GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+ if (!gst_buffer_is_writable (buf))
+ return GST_FLOW_OK;
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+ gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+ if (!filter->have_coeffs)
+ generate_coefficients (filter);
+
+ filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}
+
+
+static gboolean
+gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
+{
+ GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+ gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+ GstAudioChebyshevFreqLimitChannelCtx *ctx;
+ gint i;
+
+ /* Reset the history of input and output values if
+ * already existing */
+ if (channels && filter->channels) {
+ for (i = 0; i < channels; i++) {
+ ctx = &filter->channels[i];
+ if (ctx->x)
+ memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+ if (ctx->y)
+ memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+ }
+ }
+ return TRUE;
+}
--- /dev/null
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.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 (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 __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit;
+typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass;
+
+typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint);
+
+typedef struct
+{
+ gdouble *x;
+ gint x_pos;
+ gdouble *y;
+ gint y_pos;
+} GstAudioChebyshevFreqLimitChannelCtx;
+
+struct _GstAudioChebyshevFreqLimit
+{
+ GstAudioFilter audiofilter;
+
+ gint mode;
+ gint type;
+ gint poles;
+ gfloat cutoff;
+ gfloat ripple;
+
+ /* < private > */
+ GstAudioChebyshevFreqLimitProcessFunc process;
+
+ gboolean have_coeffs;
+ gdouble *a;
+ gint num_a;
+ gdouble *b;
+ gint num_b;
+ GstAudioChebyshevFreqLimitChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqLimitClass
+{
+ GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_limit_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */
#include "audioinvert.h"
#include "audioamplify.h"
#include "audiodynamic.h"
+#include "audiochebyshevfreqlimit.h"
+#include "audiochebyshevfreqband.h"
/* entry point to initialize the plug-in
* initialize the plug-in itself
gst_element_register (plugin, "audioamplify", GST_RANK_NONE,
GST_TYPE_AUDIO_AMPLIFY) &&
gst_element_register (plugin, "audiodynamic", GST_RANK_NONE,
- GST_TYPE_AUDIO_DYNAMIC));
+ GST_TYPE_AUDIO_DYNAMIC) &&
+ gst_element_register (plugin, "audiochebyshevfreqlimit", GST_RANK_NONE,
+ GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT) &&
+ gst_element_register (plugin, "audiochebyshevfreqband", GST_RANK_NONE,
+ GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND));
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
elements/alphacolor \
elements/audiopanorama \
elements/audioinvert \
+ elements/audiochebyshevfreqband \
+ elements/audiochebyshevfreqlimit \
elements/audioamplify \
elements/audiodynamic \
elements/avimux \
audioamplify
audiodynamic
audioinvert
+audiochebyshevfreqband
+audiochebyshevfreqlimit
level
matroskamux
cmmldec
--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * audiochebyshevfreqband.c: Unit test for the audiochebyshevfreqband element
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/check/gstcheck.h>
+
+#include <math.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 */
+GstPad *mysrcpad, *mysinkpad;
+
+#define CAPS_STRING \
+ "audio/x-raw-float, " \
+ "channels = (int) 1, " \
+ "rate = (int) 44100, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 64" \
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+
+GstElement *
+setup_audiochebyshevfreqband ()
+{
+ GstElement *audiochebyshevfreqband;
+
+ GST_DEBUG ("setup_audiochebyshevfreqband");
+ audiochebyshevfreqband = gst_check_setup_element ("audiochebyshevfreqband");
+ mysrcpad =
+ gst_check_setup_src_pad (audiochebyshevfreqband, &srctemplate, NULL);
+ mysinkpad =
+ gst_check_setup_sink_pad (audiochebyshevfreqband, &sinktemplate, NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return audiochebyshevfreqband;
+}
+
+void
+cleanup_audiochebyshevfreqband (GstElement * audiochebyshevfreqband)
+{
+ GST_DEBUG ("cleanup_audiochebyshevfreqband");
+
+ g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (audiochebyshevfreqband);
+ gst_check_teardown_sink_pad (audiochebyshevfreqband);
+ gst_check_teardown_element (audiochebyshevfreqband);
+}
+
+/* Test if data containing only one frequency component
+ * at 0 is erased with bandpass mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_bp_0hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at band center is preserved with bandpass mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_bp_11025hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 4) {
+ in[i] = 0.0;
+ in[i + 1] = 1.0;
+ in[i + 2] = 0.0;
+ in[i + 3] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms >= 0.6);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is erased with bandpass mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_bp_22050hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at 0 is preserved with bandreject mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_br_0hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandreject */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at band center is erased with bandreject mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_br_11025hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandreject */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 4) {
+ in[i] = 0.0;
+ in[i + 1] = 1.0;
+ in[i + 2] = 0.0;
+ in[i + 3] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is preserved with bandreject mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_br_22050hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandreject */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+Suite *
+audiochebyshevfreqband_suite (void)
+{
+ Suite *s = suite_create ("audiochebyshevfreqband");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_bp_0hz);
+ tcase_add_test (tc_chain, test_bp_11025hz);
+ tcase_add_test (tc_chain, test_bp_22050hz);
+ tcase_add_test (tc_chain, test_br_0hz);
+ tcase_add_test (tc_chain, test_br_11025hz);
+ tcase_add_test (tc_chain, test_br_22050hz);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audiochebyshevfreqband_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * audiochebyshevfreqlimit.c: Unit test for the audiochebyshevfreqlimit element
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/check/gstcheck.h>
+
+#include <math.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 */
+GstPad *mysrcpad, *mysinkpad;
+
+#define CAPS_STRING \
+ "audio/x-raw-float, " \
+ "channels = (int) 1, " \
+ "rate = (int) 44100, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 64" \
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+
+GstElement *
+setup_audiochebyshevfreqlimit ()
+{
+ GstElement *audiochebyshevfreqlimit;
+
+ GST_DEBUG ("setup_audiochebyshevfreqlimit");
+ audiochebyshevfreqlimit = gst_check_setup_element ("audiochebyshevfreqlimit");
+ mysrcpad =
+ gst_check_setup_src_pad (audiochebyshevfreqlimit, &srctemplate, NULL);
+ mysinkpad =
+ gst_check_setup_sink_pad (audiochebyshevfreqlimit, &sinktemplate, NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return audiochebyshevfreqlimit;
+}
+
+void
+cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit)
+{
+ GST_DEBUG ("cleanup_audiochebyshevfreqlimit");
+
+ g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (audiochebyshevfreqlimit);
+ gst_check_teardown_sink_pad (audiochebyshevfreqlimit);
+ gst_check_teardown_element (audiochebyshevfreqlimit);
+}
+
+/* Test if data containing only one frequency component
+ * at 0 is preserved with lowpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_lp_0hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to lowpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is erased with lowpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_lp_22050hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to lowpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at 0 is erased with highpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_hp_0hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to highpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is preserved with highpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_hp_22050hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to highpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+Suite *
+audiochebyshevfreqlimit_suite (void)
+{
+ Suite *s = suite_create ("audiochebyshevfreqlimit");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_lp_0hz);
+ tcase_add_test (tc_chain, test_lp_22050hz);
+ tcase_add_test (tc_chain, test_hp_0hz);
+ tcase_add_test (tc_chain, test_hp_22050hz);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audiochebyshevfreqlimit_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * audiochebyshevfreqband.c: Unit test for the audiochebyshevfreqband element
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/check/gstcheck.h>
+
+#include <math.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 */
+GstPad *mysrcpad, *mysinkpad;
+
+#define CAPS_STRING \
+ "audio/x-raw-float, " \
+ "channels = (int) 1, " \
+ "rate = (int) 44100, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 64" \
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+
+GstElement *
+setup_audiochebyshevfreqband ()
+{
+ GstElement *audiochebyshevfreqband;
+
+ GST_DEBUG ("setup_audiochebyshevfreqband");
+ audiochebyshevfreqband = gst_check_setup_element ("audiochebyshevfreqband");
+ mysrcpad =
+ gst_check_setup_src_pad (audiochebyshevfreqband, &srctemplate, NULL);
+ mysinkpad =
+ gst_check_setup_sink_pad (audiochebyshevfreqband, &sinktemplate, NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return audiochebyshevfreqband;
+}
+
+void
+cleanup_audiochebyshevfreqband (GstElement * audiochebyshevfreqband)
+{
+ GST_DEBUG ("cleanup_audiochebyshevfreqband");
+
+ g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (audiochebyshevfreqband);
+ gst_check_teardown_sink_pad (audiochebyshevfreqband);
+ gst_check_teardown_element (audiochebyshevfreqband);
+}
+
+/* Test if data containing only one frequency component
+ * at 0 is erased with bandpass mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_bp_0hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at band center is preserved with bandpass mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_bp_11025hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 4) {
+ in[i] = 0.0;
+ in[i + 1] = 1.0;
+ in[i + 2] = 0.0;
+ in[i + 3] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms >= 0.6);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is erased with bandpass mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_bp_22050hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at 0 is preserved with bandreject mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_br_0hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandreject */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at band center is erased with bandreject mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_br_11025hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandreject */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 4) {
+ in[i] = 0.0;
+ in[i + 1] = 1.0;
+ in[i + 2] = 0.0;
+ in[i + 3] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is preserved with bandreject mode and a
+ * 2000Hz frequency band around rate/4 */
+GST_START_TEST (test_br_22050hz)
+{
+ GstElement *audiochebyshevfreqband;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqband = setup_audiochebyshevfreqband ();
+ /* Set to bandreject */
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqband,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
+ 44100 / 4.0 - 1000, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
+ 44100 / 4.0 + 1000, NULL);
+ inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 1024; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 1024; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 1024.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
+}
+
+GST_END_TEST;
+
+Suite *
+audiochebyshevfreqband_suite (void)
+{
+ Suite *s = suite_create ("audiochebyshevfreqband");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_bp_0hz);
+ tcase_add_test (tc_chain, test_bp_11025hz);
+ tcase_add_test (tc_chain, test_bp_22050hz);
+ tcase_add_test (tc_chain, test_br_0hz);
+ tcase_add_test (tc_chain, test_br_11025hz);
+ tcase_add_test (tc_chain, test_br_22050hz);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audiochebyshevfreqband_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * audiochebyshevfreqlimit.c: Unit test for the audiochebyshevfreqlimit element
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/check/gstcheck.h>
+
+#include <math.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 */
+GstPad *mysrcpad, *mysinkpad;
+
+#define CAPS_STRING \
+ "audio/x-raw-float, " \
+ "channels = (int) 1, " \
+ "rate = (int) 44100, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 64" \
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-float, "
+ "channels = (int) 1, "
+ "rate = (int) 44100, "
+ "endianness = (int) BYTE_ORDER, " "width = (int) 64")
+ );
+
+GstElement *
+setup_audiochebyshevfreqlimit ()
+{
+ GstElement *audiochebyshevfreqlimit;
+
+ GST_DEBUG ("setup_audiochebyshevfreqlimit");
+ audiochebyshevfreqlimit = gst_check_setup_element ("audiochebyshevfreqlimit");
+ mysrcpad =
+ gst_check_setup_src_pad (audiochebyshevfreqlimit, &srctemplate, NULL);
+ mysinkpad =
+ gst_check_setup_sink_pad (audiochebyshevfreqlimit, &sinktemplate, NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return audiochebyshevfreqlimit;
+}
+
+void
+cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit)
+{
+ GST_DEBUG ("cleanup_audiochebyshevfreqlimit");
+
+ g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (audiochebyshevfreqlimit);
+ gst_check_teardown_sink_pad (audiochebyshevfreqlimit);
+ gst_check_teardown_element (audiochebyshevfreqlimit);
+}
+
+/* Test if data containing only one frequency component
+ * at 0 is preserved with lowpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_lp_0hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to lowpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is erased with lowpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_lp_22050hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to lowpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at 0 is erased with highpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_hp_0hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to highpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i++)
+ in[i] = 1.0;
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms <= 0.1);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+/* Test if data containing only one frequency component
+ * at rate/2 is preserved with highpass mode and a cutoff
+ * at rate/4 */
+GST_START_TEST (test_hp_22050hz)
+{
+ GstElement *audiochebyshevfreqlimit;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gdouble *in, *res, rms;
+ gint i;
+
+ audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
+ /* Set to highpass */
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
+
+ fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
+ NULL);
+ inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+ in = (gdouble *) GST_BUFFER_DATA (inbuffer);
+ for (i = 0; i < 128; i += 2) {
+ in[i] = 1.0;
+ in[i + 1] = -1.0;
+ }
+
+ caps = gst_caps_from_string (CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gdouble *) GST_BUFFER_DATA (outbuffer);
+
+ rms = 0.0;
+ for (i = 0; i < 128; i++)
+ rms += res[i] * res[i];
+ rms = sqrt (rms / 128.0);
+ fail_unless (rms >= 0.9);
+
+ /* cleanup */
+ cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
+}
+
+GST_END_TEST;
+
+Suite *
+audiochebyshevfreqlimit_suite (void)
+{
+ Suite *s = suite_create ("audiochebyshevfreqlimit");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_lp_0hz);
+ tcase_add_test (tc_chain, test_lp_22050hz);
+ tcase_add_test (tc_chain, test_hp_0hz);
+ tcase_add_test (tc_chain, test_hp_22050hz);
+
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = audiochebyshevfreqlimit_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
projects / platform / upstream / gstreamer.git / commitdiff