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Rename audiochebyshevfreqband -> audiochebband and audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS...

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Original commit message from CVS:
* 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-audiofx.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebband.c:
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c:
* gst/audiofx/audiocheblimit.h:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebband.c:
* tests/check/elements/audiocheblimit.c:
* tests/check/elements/audiochebyshevfreqband.c:
* tests/check/elements/audiochebyshevfreqlimit.c:
Rename audiochebyshevfreqband -> audiochebband and
audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS
surgery.
Closes: #491811
This commit is contained in:
Jan Schmidt 2008-02-06 23:44:43 +00:00
parent 8921eb2cd9
commit 22bea9fec3
23 changed files with 687 additions and 5149 deletions
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30
ChangeLog
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@ -1,3 +1,33 @@
2008-02-06 Jan Schmidt <jan.schmidt@sun.com>
* 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-audiofx.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebband.c:
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c:
* gst/audiofx/audiocheblimit.h:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebband.c:
* tests/check/elements/audiocheblimit.c:
* tests/check/elements/audiochebyshevfreqband.c:
* tests/check/elements/audiochebyshevfreqlimit.c:
Rename audiochebyshevfreqband -> audiochebband and
audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS
surgery.
Closes: #491811
2008-02-05 Wim Taymans <wim.taymans@collabora.co.uk> 2008-02-05 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: orjan <orjanf at axis dot com> Patch by: orjan <orjanf at axis dot com>

2
common

@ -1 +1 @@
Subproject commit 3c5473161ce19a3530bad279b842d542895b1500 Subproject commit 8b37d7ee833fab1d25b484d8574df3dae231b5f2

4
docs/plugins/Makefile.am
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@ -100,8 +100,8 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/audiofx/audiodynamic.h \ $(top_srcdir)/gst/audiofx/audiodynamic.h \
$(top_srcdir)/gst/audiofx/audioinvert.h \ $(top_srcdir)/gst/audiofx/audioinvert.h \
$(top_srcdir)/gst/audiofx/audiopanorama.h \ $(top_srcdir)/gst/audiofx/audiopanorama.h \
$(top_srcdir)/gst/audiofx/audiochebyshevfreqlimit.h \ $(top_srcdir)/gst/audiofx/audiocheblimit.h \
$(top_srcdir)/gst/audiofx/audiochebyshevfreqband.h \ $(top_srcdir)/gst/audiofx/audiochebband.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \ $(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
$(top_srcdir)/gst/autodetect/gstautovideosink.h \ $(top_srcdir)/gst/autodetect/gstautovideosink.h \
$(top_srcdir)/gst/avi/gstavidemux.h \ $(top_srcdir)/gst/avi/gstavidemux.h \

4
docs/plugins/gst-plugins-good-plugins-docs.sgml
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@ -16,8 +16,8 @@
<xi:include href="xml/element-apedemux.xml" /> <xi:include href="xml/element-apedemux.xml" />
<xi:include href="xml/element-apev2mux.xml" /> <xi:include href="xml/element-apev2mux.xml" />
<xi:include href="xml/element-audioamplify.xml" /> <xi:include href="xml/element-audioamplify.xml" />
<xi:include href="xml/element-audiochebyshevfreqband.xml" /> <xi:include href="xml/element-audiochebband.xml" />
<xi:include href="xml/element-audiochebyshevfreqlimit.xml" /> <xi:include href="xml/element-audiocheblimit.xml" />
<xi:include href="xml/element-audiodynamic.xml" /> <xi:include href="xml/element-audiodynamic.xml" />
<xi:include href="xml/element-audioinvert.xml" /> <xi:include href="xml/element-audioinvert.xml" />
<xi:include href="xml/element-audiopanorama.xml" /> <xi:include href="xml/element-audiopanorama.xml" />

48
docs/plugins/gst-plugins-good-plugins-sections.txt
View File

@ -58,35 +58,35 @@ gst_audio_amplify_get_type
</SECTION> </SECTION>
<SECTION> <SECTION>
<FILE>element-audiochebyshevfreqband</FILE> <FILE>element-audiochebband</FILE>
<TITLE>audiochebyshevfreqband</TITLE> <TITLE>audiochebband</TITLE>
GstAudioChebyshevFreqBand GstAudioChebBand
<SUBSECTION Standard> <SUBSECTION Standard>
GstAudioChebyshevFreqBandClass GstAudioChebBandClass
GstAudioChebyshevFreqBandProcessFunc GstAudioChebBandProcessFunc
GST_AUDIO_CHEBYSHEV_FREQ_BAND GST_AUDIO_CHEB_BAND
GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS GST_AUDIO_CHEB_BAND_CLASS
GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS GST_AUDIO_CHEB_BAND_GET_CLASS
GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND GST_IS_AUDIO_CHEB_BAND
GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS GST_IS_AUDIO_CHEB_BAND_CLASS
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND GST_TYPE_AUDIO_CHEB_BAND
gst_audio_chebyshev_freq_band_get_type gst_audio_cheb_band_get_type
</SECTION> </SECTION>
<SECTION> <SECTION>
<FILE>element-audiochebyshevfreqlimit</FILE> <FILE>element-audiocheblimit</FILE>
<TITLE>audiochebyshevfreqlimit</TITLE> <TITLE>audiocheblimit</TITLE>
GstAudioChebyshevFreqLimit GstAudioChebLimit
<SUBSECTION Standard> <SUBSECTION Standard>
GstAudioChebyshevFreqLimitClass GstAudioChebLimitClass
GstAudioChebyshevFreqLimitProcessFunc GstAudioChebLimitProcessFunc
GST_AUDIO_CHEBYSHEV_FREQ_LIMIT GST_AUDIO_CHEB_LIMIT
GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS GST_AUDIO_CHEB_LIMIT_CLASS
GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS GST_AUDIO_CHEB_LIMIT_GET_CLASS
GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT GST_IS_AUDIO_CHEB_LIMIT
GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS GST_IS_AUDIO_CHEB_LIMIT_CLASS
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT GST_TYPE_AUDIO_CHEB_LIMIT
gst_audio_chebyshev_freq_limit_get_type gst_audio_cheb_limit_get_type
</SECTION> </SECTION>
<SECTION> <SECTION>

26
docs/plugins/gst-plugins-good-plugins.args
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@ -17399,7 +17399,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqBand::lower-frequency</NAME> <NAME>GstAudioChebBand::lower-frequency</NAME>
<TYPE>gfloat</TYPE> <TYPE>gfloat</TYPE>
<RANGE>[0,100000]</RANGE> <RANGE>[0,100000]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17409,8 +17409,8 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqBand::mode</NAME> <NAME>GstAudioChebBand::mode</NAME>
<TYPE>GstAudioChebyshevFreqBandMode</TYPE> <TYPE>GstAudioChebBandMode</TYPE>
<RANGE></RANGE> <RANGE></RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Mode</NICK> <NICK>Mode</NICK>
@ -17419,7 +17419,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqBand::poles</NAME> <NAME>GstAudioChebBand::poles</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[4,32]</RANGE> <RANGE>[4,32]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17429,7 +17429,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqBand::ripple</NAME> <NAME>GstAudioChebBand::ripple</NAME>
<TYPE>gfloat</TYPE> <TYPE>gfloat</TYPE>
<RANGE>[0,200]</RANGE> <RANGE>[0,200]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17439,7 +17439,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqBand::type</NAME> <NAME>GstAudioChebBand::type</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[1,2]</RANGE> <RANGE>[1,2]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17449,7 +17449,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqBand::upper-frequency</NAME> <NAME>GstAudioChebBand::upper-frequency</NAME>
<TYPE>gfloat</TYPE> <TYPE>gfloat</TYPE>
<RANGE>[0,100000]</RANGE> <RANGE>[0,100000]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17459,7 +17459,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqLimit::cutoff</NAME> <NAME>GstAudioChebLimit::cutoff</NAME>
<TYPE>gfloat</TYPE> <TYPE>gfloat</TYPE>
<RANGE>[0,100000]</RANGE> <RANGE>[0,100000]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17469,8 +17469,8 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqLimit::mode</NAME> <NAME>GstAudioChebLimit::mode</NAME>
<TYPE>GstAudioChebyshevFreqLimitMode</TYPE> <TYPE>GstAudioChebLimitMode</TYPE>
<RANGE></RANGE> <RANGE></RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Mode</NICK> <NICK>Mode</NICK>
@ -17479,7 +17479,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqLimit::poles</NAME> <NAME>GstAudioChebLimit::poles</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[2,32]</RANGE> <RANGE>[2,32]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17489,7 +17489,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqLimit::ripple</NAME> <NAME>GstAudioChebLimit::ripple</NAME>
<TYPE>gfloat</TYPE> <TYPE>gfloat</TYPE>
<RANGE>[0,200]</RANGE> <RANGE>[0,200]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
@ -17499,7 +17499,7 @@
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstAudioChebyshevFreqLimit::type</NAME> <NAME>GstAudioChebLimit::type</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
<RANGE>[1,2]</RANGE> <RANGE>[1,2]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>

74
docs/plugins/inspect/plugin-audiofx.xml
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@ -14,59 +14,59 @@
<longname>AudioAmplify</longname> <longname>AudioAmplify</longname>
<class>Filter/Effect/Audio</class> <class>Filter/Effect/Audio</class>
<description>Amplifies an audio stream by a given factor</description> <description>Amplifies an audio stream by a given factor</description>
<author>Sebastian Dr303266ge &lt;slomo@circular-chaos.org&gt;</author> <author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads> <pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
<caps> <caps>
<name>src</name> <name>src</name>
<direction>source</direction> <direction>source</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads> </pads>
</element> </element>
<element> <element>
<name>audiochebyshevfreqband</name> <name>audiochebband</name>
<longname>AudioChebyshevFreqBand</longname> <longname>AudioChebBand</longname>
<class>Filter/Effect/Audio</class> <class>Filter/Effect/Audio</class>
<description>Chebyshev band pass and band reject filter</description> <description>Chebyshev band pass and band reject filter</description>
<author>Sebastian Dr303266ge &lt;slomo@circular-chaos.org&gt;</author> <author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads> <pads>
<caps> <caps>
<name>sink</name> <name>src</name>
<direction>sink</direction> <direction>source</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
<caps> <caps>
<name>src</name> <name>sink</name>
<direction>source</direction> <direction>sink</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
</pads> </pads>
</element> </element>
<element> <element>
<name>audiochebyshevfreqlimit</name> <name>audiocheblimit</name>
<longname>AudioChebyshevFreqLimit</longname> <longname>AudioChebLimit</longname>
<class>Filter/Effect/Audio</class> <class>Filter/Effect/Audio</class>
<description>Chebyshev low pass and high pass filter</description> <description>Chebyshev low pass and high pass filter</description>
<author>Sebastian Dr303266ge &lt;slomo@circular-chaos.org&gt;</author> <author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads> <pads>
<caps> <caps>
<name>sink</name> <name>src</name>
<direction>sink</direction> <direction>source</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
<caps> <caps>
<name>src</name> <name>sink</name>
<direction>source</direction> <direction>sink</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
@ -77,17 +77,17 @@
<longname>AudioDynamic</longname> <longname>AudioDynamic</longname>
<class>Filter/Effect/Audio</class> <class>Filter/Effect/Audio</class>
<description>Compressor and Expander</description> <description>Compressor and Expander</description>
<author>Sebastian Dr303266ge &lt;slomo@circular-chaos.org&gt;</author> <author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads> <pads>
<caps> <caps>
<name>sink</name> <name>src</name>
<direction>sink</direction> <direction>source</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
<caps> <caps>
<name>src</name> <name>sink</name>
<direction>source</direction> <direction>sink</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
@ -98,17 +98,17 @@
<longname>AudioInvert</longname> <longname>AudioInvert</longname>
<class>Filter/Effect/Audio</class> <class>Filter/Effect/Audio</class>
<description>Swaps upper and lower half of audio samples</description> <description>Swaps upper and lower half of audio samples</description>
<author>Sebastian Dr303266ge &lt;slomo@circular-chaos.org&gt;</author> <author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads> <pads>
<caps> <caps>
<name>sink</name> <name>src</name>
<direction>sink</direction> <direction>source</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
<caps> <caps>
<name>src</name> <name>sink</name>
<direction>source</direction> <direction>sink</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details> <details>audio/x-raw-int, depth=(int)16, width=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)32, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps> </caps>
@ -121,18 +121,18 @@
<description>Positions audio streams in the stereo panorama</description> <description>Positions audio streams in the stereo panorama</description>
<author>Stefan Kost &lt;ensonic@users.sf.net&gt;</author> <author>Stefan Kost &lt;ensonic@users.sf.net&gt;</author>
<pads> <pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)32; audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
</caps>
<caps> <caps>
<name>src</name> <name>src</name>
<direction>source</direction> <direction>source</direction>
<presence>always</presence> <presence>always</presence>
<details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)2, endianness=(int)1234, width=(int)32; audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)2, endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details> <details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)2, endianness=(int)1234, width=(int)32; audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)2, endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
</caps> </caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)32; audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
</caps>
</pads> </pads>
</element> </element>
</elements> </elements>

8
gst/audiofx/Makefile.am
View File

@ -8,8 +8,8 @@ libgstaudiofx_la_SOURCES = audiofx.c\
audioinvert.c \ audioinvert.c \
audioamplify.c \ audioamplify.c \
audiodynamic.c \ audiodynamic.c \
audiochebyshevfreqlimit.c \ audiocheblimit.c \
audiochebyshevfreqband.c audiochebband.c
# flags used to compile this plugin # flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \ libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
@ -29,6 +29,6 @@ noinst_HEADERS = audiopanorama.h \
audioinvert.h \ audioinvert.h \
audioamplify.h \ audioamplify.h \
audiodynamic.h \ audiodynamic.h \
audiochebyshevfreqlimit.h \ audiocheblimit.h \
audiochebyshevfreqband.h audiochebband.h

114
gst/audiofx/audiochebband.c
View File

@ -33,7 +33,7 @@
*/ */
/** /**
* SECTION:element-audiochebyshevfreqband * SECTION:element-audiochebband
* @short_description: Chebyshev band pass and band reject filter * @short_description: Chebyshev band pass and band reject filter
* *
* <refsect2> * <refsect2>
@ -65,9 +65,9 @@
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * <para>
* <programlisting> * <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebband 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 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebband 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 * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
* </programlisting> * </programlisting>
* </para> * </para>
* </refsect2> * </refsect2>
@ -85,13 +85,13 @@
#include <math.h> #include <math.h>
#include "audiochebyshevfreqband.h" #include "audiochebband.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug #define GST_CAT_DEFAULT gst_audio_cheb_band_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details = static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand", GST_ELEMENT_DETAILS ("AudioChebBand",
"Filter/Effect/Audio", "Filter/Effect/Audio",
"Chebyshev band pass and band reject filter", "Chebyshev band pass and band reject filter",
"Sebastian Dröge <slomo@circular-chaos.org>"); "Sebastian Dröge <slomo@circular-chaos.org>");
@ -122,26 +122,25 @@ enum
" channels = (int) [ 1, MAX ]" " channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \ #define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element"); GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0, "audiochebband element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band, GST_BOILERPLATE_FULL (GstAudioChebBand, gst_audio_cheb_band,
GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_chebyshev_freq_band_set_property (GObject * object, static void gst_audio_cheb_band_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec); guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_band_get_property (GObject * object, static void gst_audio_cheb_band_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec); guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter, static gboolean gst_audio_cheb_band_setup (GstAudioFilter * filter,
GstRingBufferSpec * format); GstRingBufferSpec * format);
static GstFlowReturn static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, gst_audio_cheb_band_transform_ip (GstBaseTransform * base, GstBuffer * buf);
GstBuffer * buf); static gboolean gst_audio_cheb_band_start (GstBaseTransform * base);
static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqBand * filter, static void process_64 (GstAudioChebBand * filter,
gdouble * data, guint num_samples); gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqBand * filter, static void process_32 (GstAudioChebBand * filter,
gfloat * data, guint num_samples); gfloat * data, guint num_samples);
enum enum
@ -150,9 +149,9 @@ enum
MODE_BAND_REJECT MODE_BAND_REJECT
}; };
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ()) #define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_cheb_band_mode_get_type ())
static GType static GType
gst_audio_chebyshev_freq_band_mode_get_type (void) gst_audio_cheb_band_mode_get_type (void)
{ {
static GType gtype = 0; static GType gtype = 0;
@ -165,7 +164,7 @@ gst_audio_chebyshev_freq_band_mode_get_type (void)
{0, NULL, NULL} {0, NULL, NULL}
}; };
gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values); gtype = g_enum_register_static ("GstAudioChebBandMode", values);
} }
return gtype; return gtype;
} }
@ -173,7 +172,7 @@ gst_audio_chebyshev_freq_band_mode_get_type (void)
/* GObject vmethod implementations */ /* GObject vmethod implementations */
static void static void
gst_audio_chebyshev_freq_band_base_init (gpointer klass) gst_audio_cheb_band_base_init (gpointer klass)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps; GstCaps *caps;
@ -187,9 +186,9 @@ gst_audio_chebyshev_freq_band_base_init (gpointer klass)
} }
static void static void
gst_audio_chebyshev_freq_band_dispose (GObject * object) gst_audio_cheb_band_dispose (GObject * object)
{ {
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object);
if (filter->a) { if (filter->a) {
g_free (filter->a); g_free (filter->a);
@ -202,7 +201,7 @@ gst_audio_chebyshev_freq_band_dispose (GObject * object)
} }
if (filter->channels) { if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx; GstAudioChebBandChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) { for (i = 0; i < channels; i++) {
@ -219,8 +218,7 @@ gst_audio_chebyshev_freq_band_dispose (GObject * object)
} }
static void static void
gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass * gst_audio_cheb_band_class_init (GstAudioChebBandClass * klass)
klass)
{ {
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstBaseTransformClass *trans_class; GstBaseTransformClass *trans_class;
@ -230,9 +228,9 @@ gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
trans_class = (GstBaseTransformClass *) klass; trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass; filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property; gobject_class->set_property = gst_audio_cheb_band_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property; gobject_class->get_property = gst_audio_cheb_band_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose; gobject_class->dispose = gst_audio_cheb_band_dispose;
g_object_class_install_property (gobject_class, PROP_MODE, g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode", g_param_spec_enum ("mode", "Mode",
@ -265,15 +263,15 @@ gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
"Number of poles to use, will be rounded up to the next multiply of four", "Number of poles to use, will be rounded up to the next multiply of four",
4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); 4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup); filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_setup);
trans_class->transform_ip = trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip); GST_DEBUG_FUNCPTR (gst_audio_cheb_band_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start); trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_start);
} }
static void static void
gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter, gst_audio_cheb_band_init (GstAudioChebBand * filter,
GstAudioChebyshevFreqBandClass * klass) GstAudioChebBandClass * klass)
{ {
filter->lower_frequency = filter->upper_frequency = 0.0; filter->lower_frequency = filter->upper_frequency = 0.0;
filter->mode = MODE_BAND_PASS; filter->mode = MODE_BAND_PASS;
@ -289,7 +287,7 @@ gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
} }
static void static void
generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter, generate_biquad_coefficients (GstAudioChebBand * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3, gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4) gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
{ {
@ -520,7 +518,7 @@ calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
} }
static void static void
generate_coefficients (GstAudioChebyshevFreqBand * filter) generate_coefficients (GstAudioChebBand * filter)
{ {
gint channels = GST_AUDIO_FILTER (filter)->format.channels; gint channels = GST_AUDIO_FILTER (filter)->format.channels;
@ -535,7 +533,7 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
} }
if (filter->channels) { if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx; GstAudioChebBandChannelCtx *ctx;
gint i; gint i;
for (i = 0; i < channels; i++) { for (i = 0; i < channels; i++) {
@ -553,7 +551,7 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
filter->a = g_new0 (gdouble, 1); filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0; filter->a[0] = 1.0;
filter->num_b = 0; filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); filter->channels = g_new0 (GstAudioChebBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet"); GST_LOG_OBJECT (filter, "rate was not set yet");
return; return;
} }
@ -565,7 +563,7 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
filter->a = g_new0 (gdouble, 1); filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0; filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
filter->num_b = 0; filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); filter->channels = g_new0 (GstAudioChebBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "frequency band had no or negative dimension"); GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
return; return;
} }
@ -591,9 +589,9 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
filter->num_b = np + 1; filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 5); filter->b = b = g_new0 (gdouble, np + 5);
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); filter->channels = g_new0 (GstAudioChebBandChannelCtx, channels);
for (i = 0; i < channels; i++) { for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i]; GstAudioChebBandChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1); ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1); ctx->y = g_new0 (gdouble, np + 1);
@ -714,10 +712,10 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
} }
static void static void
gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id, gst_audio_cheb_band_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec) const GValue * value, GParamSpec * pspec)
{ {
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object);
switch (prop_id) { switch (prop_id) {
case PROP_MODE: case PROP_MODE:
@ -763,10 +761,10 @@ gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
} }
static void static void
gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id, gst_audio_cheb_band_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec) GValue * value, GParamSpec * pspec)
{ {
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object);
switch (prop_id) { switch (prop_id) {
case PROP_MODE: case PROP_MODE:
@ -796,17 +794,16 @@ gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */ /* GstAudioFilter vmethod implementations */
static gboolean static gboolean
gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base, gst_audio_cheb_band_setup (GstAudioFilter * base, GstRingBufferSpec * format)
GstRingBufferSpec * format)
{ {
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
gboolean ret = TRUE; gboolean ret = TRUE;
if (format->width == 32) if (format->width == 32)
filter->process = (GstAudioChebyshevFreqBandProcessFunc) filter->process = (GstAudioChebBandProcessFunc)
process_32; process_32;
else if (format->width == 64) else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqBandProcessFunc) filter->process = (GstAudioChebBandProcessFunc)
process_64; process_64;
else else
ret = FALSE; ret = FALSE;
@ -817,8 +814,8 @@ gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
} }
static inline gdouble static inline gdouble
process (GstAudioChebyshevFreqBand * filter, process (GstAudioChebBand * filter,
GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0) GstAudioChebBandChannelCtx * ctx, gdouble x0)
{ {
gdouble val = filter->a[0] * x0; gdouble val = filter->a[0] * x0;
gint i, j; gint i, j;
@ -857,7 +854,7 @@ process (GstAudioChebyshevFreqBand * filter,
#define DEFINE_PROCESS_FUNC(width,ctype) \ #define DEFINE_PROCESS_FUNC(width,ctype) \
static void \ static void \
process_##width (GstAudioChebyshevFreqBand * filter, \ process_##width (GstAudioChebBand * filter, \
g##ctype * data, guint num_samples) \ g##ctype * data, guint num_samples) \
{ \ { \
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
@ -878,10 +875,9 @@ DEFINE_PROCESS_FUNC (64, double);
/* GstBaseTransform vmethod implementations */ /* GstBaseTransform vmethod implementations */
static GstFlowReturn static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, gst_audio_cheb_band_transform_ip (GstBaseTransform * base, GstBuffer * buf)
GstBuffer * buf)
{ {
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
guint num_samples = guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
@ -900,11 +896,11 @@ gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
} }
static gboolean static gboolean
gst_audio_chebyshev_freq_band_start (GstBaseTransform * base) gst_audio_cheb_band_start (GstBaseTransform * base)
{ {
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels; gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqBandChannelCtx *ctx; GstAudioChebBandChannelCtx *ctx;
gint i; gint i;
/* Reset the history of input and output values if /* Reset the history of input and output values if

36
gst/audiofx/audiochebband.h
View File

@ -18,8 +18,8 @@
* Boston, MA 02111-1307, USA. * Boston, MA 02111-1307, USA.
*/ */
#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ #ifndef __GST_AUDIO_CHEB_BAND_H__
#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ #define __GST_AUDIO_CHEB_BAND_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/base/gstbasetransform.h> #include <gst/base/gstbasetransform.h>
@ -27,16 +27,16 @@
#include <gst/audio/gstaudiofilter.h> #include <gst/audio/gstaudiofilter.h>
G_BEGIN_DECLS G_BEGIN_DECLS
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type()) #define GST_TYPE_AUDIO_CHEB_BAND (gst_audio_cheb_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_AUDIO_CHEB_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEB_BAND,GstAudioChebBand))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) #define GST_IS_AUDIO_CHEB_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEB_BAND))
#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) #define GST_AUDIO_CHEB_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEB_BAND,GstAudioChebBandClass))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) #define GST_IS_AUDIO_CHEB_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEB_BAND))
#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) #define GST_AUDIO_CHEB_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEB_BAND,GstAudioChebBandClass))
typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand; typedef struct _GstAudioChebBand GstAudioChebBand;
typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass; typedef struct _GstAudioChebBandClass GstAudioChebBandClass;
typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint); typedef void (*GstAudioChebBandProcessFunc) (GstAudioChebBand *, guint8 *, guint);
typedef struct typedef struct
{ {
@ -44,9 +44,9 @@ typedef struct
gint x_pos; gint x_pos;
gdouble *y; gdouble *y;
gint y_pos; gint y_pos;
} GstAudioChebyshevFreqBandChannelCtx; } GstAudioChebBandChannelCtx;
struct _GstAudioChebyshevFreqBand struct _GstAudioChebBand
{ {
GstAudioFilter audiofilter; GstAudioFilter audiofilter;
@ -58,22 +58,22 @@ struct _GstAudioChebyshevFreqBand
gfloat ripple; gfloat ripple;
/* < private > */ /* < private > */
GstAudioChebyshevFreqBandProcessFunc process; GstAudioChebBandProcessFunc process;
gboolean have_coeffs; gboolean have_coeffs;
gdouble *a; gdouble *a;
gint num_a; gint num_a;
gdouble *b; gdouble *b;
gint num_b; gint num_b;
GstAudioChebyshevFreqBandChannelCtx *channels; GstAudioChebBandChannelCtx *channels;
}; };
struct _GstAudioChebyshevFreqBandClass struct _GstAudioChebBandClass
{ {
GstAudioFilterClass parent; GstAudioFilterClass parent;
}; };
GType gst_audio_chebyshev_freq_band_get_type (void); GType gst_audio_cheb_band_get_type (void);
G_END_DECLS G_END_DECLS
#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */ #endif /* __GST_AUDIO_CHEB_BAND_H__ */

120
gst/audiofx/audiocheblimit.c
View File

@ -29,7 +29,7 @@
*/ */
/** /**
* SECTION:element-audiochebyshevfreqlimit * SECTION:element-audiocheblimit
* @short_description: Chebyshev low pass and high pass filter * @short_description: Chebyshev low pass and high pass filter
* *
* <refsect2> * <refsect2>
@ -61,9 +61,9 @@
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * <para>
* <programlisting> * <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiocheblimit 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 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit 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 * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
* </programlisting> * </programlisting>
* </para> * </para>
* </refsect2> * </refsect2>
@ -81,13 +81,13 @@
#include <math.h> #include <math.h>
#include "audiochebyshevfreqlimit.h" #include "audiocheblimit.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug #define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details = static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit", GST_ELEMENT_DETAILS ("AudioChebLimit",
"Filter/Effect/Audio", "Filter/Effect/Audio",
"Chebyshev low pass and high pass filter", "Chebyshev low pass and high pass filter",
"Sebastian Dröge <slomo@circular-chaos.org>"); "Sebastian Dröge <slomo@circular-chaos.org>");
@ -117,27 +117,25 @@ enum
" channels = (int) [ 1, MAX ]" " channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \ #define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element"); GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit, GST_BOILERPLATE_FULL (GstAudioChebLimit,
gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, gst_audio_cheb_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
DEBUG_INIT);
static void gst_audio_chebyshev_freq_limit_set_property (GObject * object, static void gst_audio_cheb_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec); guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_limit_get_property (GObject * object, static void gst_audio_cheb_limit_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec); guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter, static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter,
GstRingBufferSpec * format); GstRingBufferSpec * format);
static GstFlowReturn static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, gst_audio_cheb_limit_transform_ip (GstBaseTransform * base, GstBuffer * buf);
GstBuffer * buf); static gboolean gst_audio_cheb_limit_start (GstBaseTransform * base);
static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqLimit * filter, static void process_64 (GstAudioChebLimit * filter,
gdouble * data, guint num_samples); gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqLimit * filter, static void process_32 (GstAudioChebLimit * filter,
gfloat * data, guint num_samples); gfloat * data, guint num_samples);
enum enum
@ -146,9 +144,9 @@ enum
MODE_HIGH_PASS MODE_HIGH_PASS
}; };
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ()) #define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_cheb_limit_mode_get_type ())
static GType static GType
gst_audio_chebyshev_freq_limit_mode_get_type (void) gst_audio_cheb_limit_mode_get_type (void)
{ {
static GType gtype = 0; static GType gtype = 0;
@ -161,7 +159,7 @@ gst_audio_chebyshev_freq_limit_mode_get_type (void)
{0, NULL, NULL} {0, NULL, NULL}
}; };
gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values); gtype = g_enum_register_static ("GstAudioChebLimitMode", values);
} }
return gtype; return gtype;
} }
@ -169,7 +167,7 @@ gst_audio_chebyshev_freq_limit_mode_get_type (void)
/* GObject vmethod implementations */ /* GObject vmethod implementations */
static void static void
gst_audio_chebyshev_freq_limit_base_init (gpointer klass) gst_audio_cheb_limit_base_init (gpointer klass)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps; GstCaps *caps;
@ -183,9 +181,9 @@ gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
} }
static void static void
gst_audio_chebyshev_freq_limit_dispose (GObject * object) gst_audio_cheb_limit_dispose (GObject * object)
{ {
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
if (filter->a) { if (filter->a) {
g_free (filter->a); g_free (filter->a);
@ -198,7 +196,7 @@ gst_audio_chebyshev_freq_limit_dispose (GObject * object)
} }
if (filter->channels) { if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx; GstAudioChebLimitChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) { for (i = 0; i < channels; i++) {
@ -215,8 +213,7 @@ gst_audio_chebyshev_freq_limit_dispose (GObject * object)
} }
static void static void
gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass * gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
klass)
{ {
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstBaseTransformClass *trans_class; GstBaseTransformClass *trans_class;
@ -226,9 +223,9 @@ gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
trans_class = (GstBaseTransformClass *) klass; trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass; filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property; gobject_class->set_property = gst_audio_cheb_limit_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property; gobject_class->get_property = gst_audio_cheb_limit_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose; gobject_class->dispose = gst_audio_cheb_limit_dispose;
g_object_class_install_property (gobject_class, PROP_MODE, g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode", g_param_spec_enum ("mode", "Mode",
@ -256,16 +253,15 @@ gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
"Number of poles to use, will be rounded up to the next even number", "Number of poles to use, will be rounded up to the next even number",
2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup = filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup);
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
trans_class->transform_ip = trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip); GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start); trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_start);
} }
static void static void
gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter, gst_audio_cheb_limit_init (GstAudioChebLimit * filter,
GstAudioChebyshevFreqLimitClass * klass) GstAudioChebLimitClass * klass)
{ {
filter->cutoff = 0.0; filter->cutoff = 0.0;
filter->mode = MODE_LOW_PASS; filter->mode = MODE_LOW_PASS;
@ -281,7 +277,7 @@ gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
} }
static void static void
generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter, generate_biquad_coefficients (GstAudioChebLimit * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2, gint p, gdouble * a0, gdouble * a1, gdouble * a2,
gdouble * b1, gdouble * b2) gdouble * b1, gdouble * b2)
{ {
@ -471,7 +467,7 @@ calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
} }
static void static void
generate_coefficients (GstAudioChebyshevFreqLimit * filter) generate_coefficients (GstAudioChebLimit * filter)
{ {
gint channels = GST_AUDIO_FILTER (filter)->format.channels; gint channels = GST_AUDIO_FILTER (filter)->format.channels;
@ -486,7 +482,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
} }
if (filter->channels) { if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx; GstAudioChebLimitChannelCtx *ctx;
gint i; gint i;
for (i = 0; i < channels; i++) { for (i = 0; i < channels; i++) {
@ -504,7 +500,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
filter->a = g_new0 (gdouble, 1); filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0; filter->a[0] = 1.0;
filter->num_b = 0; filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet"); GST_LOG_OBJECT (filter, "rate was not set yet");
return; return;
} }
@ -516,7 +512,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
filter->a = g_new0 (gdouble, 1); filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0; filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
filter->num_b = 0; filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency"); GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return; return;
} else if (filter->cutoff <= 0.0) { } else if (filter->cutoff <= 0.0) {
@ -524,7 +520,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
filter->a = g_new0 (gdouble, 1); filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0; filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
filter->num_b = 0; filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff is lower than zero"); GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return; return;
} }
@ -540,9 +536,9 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
filter->num_b = np + 1; filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 3); filter->b = b = g_new0 (gdouble, np + 3);
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
for (i = 0; i < channels; i++) { for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i]; GstAudioChebLimitChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1); ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1); ctx->y = g_new0 (gdouble, np + 1);
@ -623,10 +619,10 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
} }
static void static void
gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id, gst_audio_cheb_limit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec) const GValue * value, GParamSpec * pspec)
{ {
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
switch (prop_id) { switch (prop_id) {
case PROP_MODE: case PROP_MODE:
@ -666,10 +662,10 @@ gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
} }
static void static void
gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id, gst_audio_cheb_limit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec) GValue * value, GParamSpec * pspec)
{ {
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
switch (prop_id) { switch (prop_id) {
case PROP_MODE: case PROP_MODE:
@ -696,17 +692,16 @@ gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */ /* GstAudioFilter vmethod implementations */
static gboolean static gboolean
gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base, gst_audio_cheb_limit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
GstRingBufferSpec * format)
{ {
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
gboolean ret = TRUE; gboolean ret = TRUE;
if (format->width == 32) if (format->width == 32)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc) filter->process = (GstAudioChebLimitProcessFunc)
process_32; process_32;
else if (format->width == 64) else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc) filter->process = (GstAudioChebLimitProcessFunc)
process_64; process_64;
else else
ret = FALSE; ret = FALSE;
@ -717,8 +712,8 @@ gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
} }
static inline gdouble static inline gdouble
process (GstAudioChebyshevFreqLimit * filter, process (GstAudioChebLimit * filter,
GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0) GstAudioChebLimitChannelCtx * ctx, gdouble x0)
{ {
gdouble val = filter->a[0] * x0; gdouble val = filter->a[0] * x0;
gint i, j; gint i, j;
@ -757,7 +752,7 @@ process (GstAudioChebyshevFreqLimit * filter,
#define DEFINE_PROCESS_FUNC(width,ctype) \ #define DEFINE_PROCESS_FUNC(width,ctype) \
static void \ static void \
process_##width (GstAudioChebyshevFreqLimit * filter, \ process_##width (GstAudioChebLimit * filter, \
g##ctype * data, guint num_samples) \ g##ctype * data, guint num_samples) \
{ \ { \
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \ gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
@ -778,10 +773,9 @@ DEFINE_PROCESS_FUNC (64, double);
/* GstBaseTransform vmethod implementations */ /* GstBaseTransform vmethod implementations */
static GstFlowReturn static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, gst_audio_cheb_limit_transform_ip (GstBaseTransform * base, GstBuffer * buf)
GstBuffer * buf)
{ {
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
guint num_samples = guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
@ -801,11 +795,11 @@ gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
static gboolean static gboolean
gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base) gst_audio_cheb_limit_start (GstBaseTransform * base)
{ {
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels; gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqLimitChannelCtx *ctx; GstAudioChebLimitChannelCtx *ctx;
gint i; gint i;
/* Reset the history of input and output values if /* Reset the history of input and output values if

36
gst/audiofx/audiocheblimit.h
View File

@ -18,8 +18,8 @@
* Boston, MA 02111-1307, USA. * Boston, MA 02111-1307, USA.
*/ */
#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ #ifndef __GST_AUDIO_CHEB_LIMIT_H__
#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ #define __GST_AUDIO_CHEB_LIMIT_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/base/gstbasetransform.h> #include <gst/base/gstbasetransform.h>
@ -27,16 +27,16 @@
#include <gst/audio/gstaudiofilter.h> #include <gst/audio/gstaudiofilter.h>
G_BEGIN_DECLS G_BEGIN_DECLS
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type()) #define GST_TYPE_AUDIO_CHEB_LIMIT (gst_audio_cheb_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_AUDIO_CHEB_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEB_LIMIT,GstAudioChebLimit))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) #define GST_IS_AUDIO_CHEB_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEB_LIMIT))
#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) #define GST_AUDIO_CHEB_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEB_LIMIT,GstAudioChebLimitClass))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) #define GST_IS_AUDIO_CHEB_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEB_LIMIT))
#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) #define GST_AUDIO_CHEB_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEB_LIMIT,GstAudioChebLimitClass))
typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit; typedef struct _GstAudioChebLimit GstAudioChebLimit;
typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass; typedef struct _GstAudioChebLimitClass GstAudioChebLimitClass;
typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint); typedef void (*GstAudioChebLimitProcessFunc) (GstAudioChebLimit *, guint8 *, guint);
typedef struct typedef struct
{ {
@ -44,9 +44,9 @@ typedef struct
gint x_pos; gint x_pos;
gdouble *y; gdouble *y;
gint y_pos; gint y_pos;
} GstAudioChebyshevFreqLimitChannelCtx; } GstAudioChebLimitChannelCtx;
struct _GstAudioChebyshevFreqLimit struct _GstAudioChebLimit
{ {
GstAudioFilter audiofilter; GstAudioFilter audiofilter;
@ -57,22 +57,22 @@ struct _GstAudioChebyshevFreqLimit
gfloat ripple; gfloat ripple;
/* < private > */ /* < private > */
GstAudioChebyshevFreqLimitProcessFunc process; GstAudioChebLimitProcessFunc process;
gboolean have_coeffs; gboolean have_coeffs;
gdouble *a; gdouble *a;
gint num_a; gint num_a;
gdouble *b; gdouble *b;
gint num_b; gint num_b;
GstAudioChebyshevFreqLimitChannelCtx *channels; GstAudioChebLimitChannelCtx *channels;
}; };
struct _GstAudioChebyshevFreqLimitClass struct _GstAudioChebLimitClass
{ {
GstAudioFilterClass parent; GstAudioFilterClass parent;
}; };
GType gst_audio_chebyshev_freq_limit_get_type (void); GType gst_audio_cheb_limit_get_type (void);
G_END_DECLS G_END_DECLS
#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */ #endif /* __GST_AUDIO_CHEB_LIMIT_H__ */

922
gst/audiofx/audiochebyshevfreqband.c
View File

@ -1,922 +0,0 @@
/*
* 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>
* This element has the advantage over the windowed sinc bandpass and bandreject filter that it is
* much faster and produces almost as good results. It's only disadvantages are the highly
* non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
* </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>
* <para><note>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
* </note></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));
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider without */
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, 100000.0,
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, 100000.0,
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, 200.0,
0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: What to do about this upper boundary? With a frequencies near
* rate/4 32 poles are completely possible, with frequencies very low
* or very high 16 poles already produces only noise */
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;
}
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstAudioChebyshevFreqBand * filter, \
g##ctype * 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; \
} \
} \
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
#undef DEFINE_PROCESS_FUNC
/* 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_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
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;
}

79
gst/audiofx/audiochebyshevfreqband.h
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@ -1,79 +0,0 @@
/*
* 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__ */

823
gst/audiofx/audiochebyshevfreqlimit.c
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@ -1,823 +0,0 @@
/*
* 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>
* This element has the advantage over the windowed sinc lowpass and highpass filter that it is
* much faster and produces almost as good results. It's only disadvantages are the highly
* non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
* </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>
* <para><note>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
* </note></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));
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider without */
g_object_class_install_property (gobject_class, PROP_CUTOFF,
g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
100000.0, 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,
200.0, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: What to do about this upper boundary? With a cutoff frequency of
* rate/4 32 poles are completely possible, with a cutoff frequency very low
* or very high 16 poles already produces only noise */
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;
}
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstAudioChebyshevFreqLimit * filter, \
g##ctype * 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; \
} \
} \
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
#undef DEFINE_PROCESS_FUNC
/* 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_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
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;
}

78
gst/audiofx/audiochebyshevfreqlimit.h
View File

@ -1,78 +0,0 @@
/*
* 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__ */

12
gst/audiofx/audiofx.c
View File

@ -29,8 +29,8 @@
#include "audioinvert.h" #include "audioinvert.h"
#include "audioamplify.h" #include "audioamplify.h"
#include "audiodynamic.h" #include "audiodynamic.h"
#include "audiochebyshevfreqlimit.h" #include "audiocheblimit.h"
#include "audiochebyshevfreqband.h" #include "audiochebband.h"
/* entry point to initialize the plug-in /* entry point to initialize the plug-in
* initialize the plug-in itself * initialize the plug-in itself
@ -51,10 +51,10 @@ plugin_init (GstPlugin * plugin)
GST_TYPE_AUDIO_AMPLIFY) && GST_TYPE_AUDIO_AMPLIFY) &&
gst_element_register (plugin, "audiodynamic", GST_RANK_NONE, 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_element_register (plugin, "audiocheblimit", GST_RANK_NONE,
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT) && GST_TYPE_AUDIO_CHEB_LIMIT) &&
gst_element_register (plugin, "audiochebyshevfreqband", GST_RANK_NONE, gst_element_register (plugin, "audiochebband", GST_RANK_NONE,
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)); GST_TYPE_AUDIO_CHEB_BAND));
} }
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,

4
tests/check/Makefile.am
View File

@ -55,8 +55,8 @@ check_PROGRAMS = \
elements/alphacolor \ elements/alphacolor \
elements/audiopanorama \ elements/audiopanorama \
elements/audioinvert \ elements/audioinvert \
elements/audiochebyshevfreqband \ elements/audiochebband \
elements/audiochebyshevfreqlimit \ elements/audiocheblimit \
elements/audioamplify \ elements/audioamplify \
elements/audiodynamic \ elements/audiodynamic \
elements/avimux \ elements/avimux \

4
tests/check/elements/.gitignore vendored
View File

@ -2,8 +2,8 @@
alphacolor alphacolor
apev2mux apev2mux
audioamplify audioamplify
audiochebyshevfreqband audiochebband
audiochebyshevfreqlimit audiocheblimit
audiodynamic audiodynamic
audioinvert audioinvert
audiopanorama audiopanorama

514
tests/check/elements/audiochebband.c
View File

File diff suppressed because it is too large Load Diff

338
tests/check/elements/audiocheblimit.c
View File

@ -2,7 +2,7 @@
* *
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org> * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
* *
* audiochebyshevfreqlimit.c: Unit test for the audiochebyshevfreqlimit element * audiocheblimit.c: Unit test for the audiocheblimit element
* *
* This library is free software; you can redistribute it and/or * This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License * modify it under the terms of the GNU Lesser General Public License
@ -63,26 +63,24 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
); );
GstElement * GstElement *
setup_audiochebyshevfreqlimit () setup_audiocheblimit ()
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GST_DEBUG ("setup_audiochebyshevfreqlimit"); GST_DEBUG ("setup_audiocheblimit");
audiochebyshevfreqlimit = gst_check_setup_element ("audiochebyshevfreqlimit"); audiocheblimit = gst_check_setup_element ("audiocheblimit");
mysrcpad = mysrcpad = gst_check_setup_src_pad (audiocheblimit, &srctemplate, NULL);
gst_check_setup_src_pad (audiochebyshevfreqlimit, &srctemplate, NULL); mysinkpad = gst_check_setup_sink_pad (audiocheblimit, &sinktemplate, NULL);
mysinkpad =
gst_check_setup_sink_pad (audiochebyshevfreqlimit, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE); gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE); gst_pad_set_active (mysinkpad, TRUE);
return audiochebyshevfreqlimit; return audiocheblimit;
} }
void void
cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit) cleanup_audiocheblimit (GstElement * audiocheblimit)
{ {
GST_DEBUG ("cleanup_audiochebyshevfreqlimit"); GST_DEBUG ("cleanup_audiocheblimit");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL); g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers); g_list_free (buffers);
@ -90,9 +88,9 @@ cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit)
gst_pad_set_active (mysrcpad, FALSE); gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE); gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audiochebyshevfreqlimit); gst_check_teardown_src_pad (audiocheblimit);
gst_check_teardown_sink_pad (audiochebyshevfreqlimit); gst_check_teardown_sink_pad (audiocheblimit);
gst_check_teardown_element (audiochebyshevfreqlimit); gst_check_teardown_element (audiocheblimit);
} }
/* Test if data containing only one frequency component /* Test if data containing only one frequency component
@ -100,25 +98,24 @@ cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit)
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_32_lp_0hz) GST_START_TEST (test_type1_32_lp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -144,7 +141,7 @@ GST_START_TEST (test_type1_32_lp_0hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -154,25 +151,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_32_lp_22050hz) GST_START_TEST (test_type1_32_lp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -200,7 +196,7 @@ GST_START_TEST (test_type1_32_lp_22050hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -210,25 +206,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_32_hp_0hz) GST_START_TEST (test_type1_32_hp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -254,7 +249,7 @@ GST_START_TEST (test_type1_32_hp_0hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -264,25 +259,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_32_hp_22050hz) GST_START_TEST (test_type1_32_hp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -310,7 +304,7 @@ GST_START_TEST (test_type1_32_hp_22050hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -320,25 +314,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_64_lp_0hz) GST_START_TEST (test_type1_64_lp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -364,7 +357,7 @@ GST_START_TEST (test_type1_64_lp_0hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -374,25 +367,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_64_lp_22050hz) GST_START_TEST (test_type1_64_lp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -420,7 +412,7 @@ GST_START_TEST (test_type1_64_lp_22050hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -430,25 +422,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_64_hp_0hz) GST_START_TEST (test_type1_64_hp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -474,7 +465,7 @@ GST_START_TEST (test_type1_64_hp_0hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -484,25 +475,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type1_64_hp_22050hz) GST_START_TEST (test_type1_64_hp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -530,7 +520,7 @@ GST_START_TEST (test_type1_64_hp_22050hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -540,25 +530,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_32_lp_0hz) GST_START_TEST (test_type2_32_lp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -584,7 +573,7 @@ GST_START_TEST (test_type2_32_lp_0hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -594,25 +583,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_32_lp_22050hz) GST_START_TEST (test_type2_32_lp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -640,7 +628,7 @@ GST_START_TEST (test_type2_32_lp_22050hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -650,25 +638,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_32_hp_0hz) GST_START_TEST (test_type2_32_hp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -694,7 +681,7 @@ GST_START_TEST (test_type2_32_hp_0hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -704,25 +691,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_32_hp_22050hz) GST_START_TEST (test_type2_32_hp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gfloat *in, *res, rms; gfloat *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer); in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -750,7 +736,7 @@ GST_START_TEST (test_type2_32_hp_22050hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -760,25 +746,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_64_lp_0hz) GST_START_TEST (test_type2_64_lp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -804,7 +789,7 @@ GST_START_TEST (test_type2_64_lp_0hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -814,25 +799,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_64_lp_22050hz) GST_START_TEST (test_type2_64_lp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to lowpass */ /* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -860,7 +844,7 @@ GST_START_TEST (test_type2_64_lp_22050hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -870,25 +854,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_64_hp_0hz) GST_START_TEST (test_type2_64_hp_0hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++) for (i = 0; i < 128; i++)
@ -914,7 +897,7 @@ GST_START_TEST (test_type2_64_hp_0hz)
fail_unless (rms <= 0.1); fail_unless (rms <= 0.1);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
@ -924,25 +907,24 @@ GST_END_TEST;
* at rate/4 */ * at rate/4 */
GST_START_TEST (test_type2_64_hp_22050hz) GST_START_TEST (test_type2_64_hp_22050hz)
{ {
GstElement *audiochebyshevfreqlimit; GstElement *audiocheblimit;
GstBuffer *inbuffer, *outbuffer; GstBuffer *inbuffer, *outbuffer;
GstCaps *caps; GstCaps *caps;
gdouble *in, *res, rms; gdouble *in, *res, rms;
gint i; gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit (); audiocheblimit = setup_audiocheblimit ();
/* Set to highpass */ /* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL); g_object_set (G_OBJECT (audiocheblimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL); g_object_set (G_OBJECT (audiocheblimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 2, NULL); g_object_set (G_OBJECT (audiocheblimit), "type", 2, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 40.0, NULL); g_object_set (G_OBJECT (audiocheblimit), "ripple", 40.0, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit, fail_unless (gst_element_set_state (audiocheblimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing"); "could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0, g_object_set (G_OBJECT (audiocheblimit), "cutoff", 44100 / 4.0, NULL);
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble)); inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer); in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) { for (i = 0; i < 128; i += 2) {
@ -970,16 +952,16 @@ GST_START_TEST (test_type2_64_hp_22050hz)
fail_unless (rms >= 0.9); fail_unless (rms >= 0.9);
/* cleanup */ /* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit); cleanup_audiocheblimit (audiocheblimit);
} }
GST_END_TEST; GST_END_TEST;
Suite * Suite *
audiochebyshevfreqlimit_suite (void) audiocheblimit_suite (void)
{ {
Suite *s = suite_create ("audiochebyshevfreqlimit"); Suite *s = suite_create ("audiocheblimit");
TCase *tc_chain = tcase_create ("general"); TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain); suite_add_tcase (s, tc_chain);
@ -1007,7 +989,7 @@ main (int argc, char **argv)
{ {
int nf; int nf;
Suite *s = audiochebyshevfreqlimit_suite (); Suite *s = audiocheblimit_suite ();
SRunner *sr = srunner_create (s); SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv); gst_check_init (&argc, &argv);

1540
tests/check/elements/audiochebyshevfreqband.c
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1020
tests/check/elements/audiochebyshevfreqlimit.c
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