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gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements.

Browse source 
Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_mode_get_type),
(gst_audio_chebyshev_freq_band_base_init),
(gst_audio_chebyshev_freq_band_dispose),
(gst_audio_chebyshev_freq_band_class_init),
(gst_audio_chebyshev_freq_band_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_band_set_property),
(gst_audio_chebyshev_freq_band_get_property),
(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
(gst_audio_chebyshev_freq_band_start):
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_mode_get_type),
(gst_audio_chebyshev_freq_limit_base_init),
(gst_audio_chebyshev_freq_limit_dispose),
(gst_audio_chebyshev_freq_limit_class_init),
(gst_audio_chebyshev_freq_limit_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_limit_set_property),
(gst_audio_chebyshev_freq_limit_get_property),
(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
(gst_audio_chebyshev_freq_limit_start):
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Fixes #464800.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebyshevfreqband.c:
(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
* tests/check/elements/audiochebyshevfreqlimit.c:
(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
Add unit tests for the chebyshev filters.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
And add docs for the chebyshev filters. While doing
that also run make update in docs/plugins.
This commit is contained in:
Sebastian Dröge 2007-08-16 17:02:07 +00:00
parent 22bcaa904c
commit 842451a720
30 changed files with 5818 additions and 58 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

63
ChangeLog
View file

@ -1,3 +1,66 @@
2007-08-16 Sebastian Dröge <slomo@circular-chaos.org>
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_mode_get_type),
(gst_audio_chebyshev_freq_band_base_init),
(gst_audio_chebyshev_freq_band_dispose),
(gst_audio_chebyshev_freq_band_class_init),
(gst_audio_chebyshev_freq_band_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_band_set_property),
(gst_audio_chebyshev_freq_band_get_property),
(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
(gst_audio_chebyshev_freq_band_start):
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_mode_get_type),
(gst_audio_chebyshev_freq_limit_base_init),
(gst_audio_chebyshev_freq_limit_dispose),
(gst_audio_chebyshev_freq_limit_class_init),
(gst_audio_chebyshev_freq_limit_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_limit_set_property),
(gst_audio_chebyshev_freq_limit_get_property),
(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
(gst_audio_chebyshev_freq_limit_start):
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Fixes #464800.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebyshevfreqband.c:
(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
* tests/check/elements/audiochebyshevfreqlimit.c:
(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
Add unit tests for the chebyshev filters.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
And add docs for the chebyshev filters. While doing
that also run make update in docs/plugins.
2007-08-16 Stefan Kost <ensonic@users.sf.net>
* ext/annodex/gstcmmltag.c:

2
docs/plugins/Makefile.am
View file

@ -100,6 +100,8 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/audiofx/audiodynamic.h \
$(top_srcdir)/gst/audiofx/audioinvert.h \
$(top_srcdir)/gst/audiofx/audiopanorama.h \
$(top_srcdir)/gst/audiofx/audiochebyshevfreqlimit.h \
$(top_srcdir)/gst/audiofx/audiochebyshevfreqband.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
$(top_srcdir)/gst/autodetect/gstautovideosink.h \
$(top_srcdir)/gst/avi/gstavidemux.h \

2
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -19,6 +19,8 @@
<xi:include href="xml/element-audioinvert.xml" />
<xi:include href="xml/element-audioamplify.xml" />
<xi:include href="xml/element-audiodynamic.xml" />
<xi:include href="xml/element-audiochebyshevfreqlimit.xml" />
<xi:include href="xml/element-audiochebyshevfreqband.xml" />
<xi:include href="xml/element-autoaudiosink.xml" />
<xi:include href="xml/element-autovideosink.xml" />
<xi:include href="xml/element-avidemux.xml" />

20
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -81,6 +81,26 @@ GST_AUDIO_DYNAMIC
GST_AUDIO_DYNAMIC_CLASS
</SECTION>
<SECTION>
<FILE>element-audiochebyshevfreqlimit</FILE>
<TITLE>audiochebyshevfreqlimit</TITLE>
GstAudioChebyshevFreqLimit
<SUBSECTION Standard>
GstAudioChebyshevFreqLimitClass
GST_AUDIO_CHEBYSHEV_FREQ_LIMIT
GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS
</SECTION>
<SECTION>
<FILE>element-audiochebyshevfreqband</FILE>
<TITLE>audiochebyshevfreqband</TITLE>
GstAudioChebyshevFreqBand
<SUBSECTION Standard>
GstAudioChebyshevFreqBandClass
GST_AUDIO_CHEBYSHEV_FREQ_BAND
GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS
</SECTION>
<SECTION>
<FILE>element-autoaudiosink</FILE>
<TITLE>autoaudiosink</TITLE>

122
docs/plugins/gst-plugins-good-plugins.args
View file

@ -401,21 +401,21 @@
<ARG>
<NAME>GstVertigoTV::speed</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,01,100]</RANGE>
<RANGE>[0.01,100]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Speed</NICK>
<BLURB>Control the speed of movement.</BLURB>
<DEFAULT>0,02</DEFAULT>
<DEFAULT>0.02</DEFAULT>
</ARG>
<ARG>
<NAME>GstVertigoTV::zoom-speed</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[1,01,1,1]</RANGE>
<RANGE>[1.01,1.1]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Zoom Speed</NICK>
<BLURB>Control the rate of zooming.</BLURB>
<DEFAULT>1,01</DEFAULT>
<DEFAULT>1.01</DEFAULT>
</ARG>
<ARG>
@ -1141,7 +1141,7 @@
<ARG>
<NAME>GstDV1394Src::port</NAME>
<TYPE>gint</TYPE>
<RANGE>[G_MAXULONG,16]</RANGE>
<RANGE>[-1,16]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>Port number (-1 automatic).</BLURB>
@ -17241,7 +17241,7 @@
<ARG>
<NAME>GstGamma::gamma</NAME>
<TYPE>gdouble</TYPE>
<RANGE>[0,01,10]</RANGE>
<RANGE>[0.01,10]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Gamma</NICK>
<BLURB>gamma.</BLURB>
@ -17328,3 +17328,113 @@
<DEFAULT>2</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqBand::lower-frequency</NAME>
<TYPE>gfloat</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Lower frequency</NICK>
<BLURB>Start frequency of the band (Hz).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqBand::mode</NAME>
<TYPE>GstAudioChebyshevFreqBandMode</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Mode</NICK>
<BLURB>Low pass or high pass mode.</BLURB>
<DEFAULT>Band pass (default)</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqBand::poles</NAME>
<TYPE>gint</TYPE>
<RANGE>[4,32]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Poles</NICK>
<BLURB>Number of poles to use, will be rounded up to the next multiply of four.</BLURB>
<DEFAULT>4</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqBand::ripple</NAME>
<TYPE>gfloat</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Ripple</NICK>
<BLURB>Amount of ripple (dB).</BLURB>
<DEFAULT>0.25</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqBand::type</NAME>
<TYPE>gint</TYPE>
<RANGE>[1,2]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Type</NICK>
<BLURB>Type of the chebychev filter.</BLURB>
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqBand::upper-frequency</NAME>
<TYPE>gfloat</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Upper frequency</NICK>
<BLURB>Stop frequency of the band (Hz).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqLimit::cutoff</NAME>
<TYPE>gfloat</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Cutoff</NICK>
<BLURB>Cut off frequency (Hz).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqLimit::mode</NAME>
<TYPE>GstAudioChebyshevFreqLimitMode</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Mode</NICK>
<BLURB>Low pass or high pass mode.</BLURB>
<DEFAULT>Low pass (default)</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqLimit::poles</NAME>
<TYPE>gint</TYPE>
<RANGE>[2,32]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Poles</NICK>
<BLURB>Number of poles to use, will be rounded up to the next even number.</BLURB>
<DEFAULT>4</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqLimit::ripple</NAME>
<TYPE>gfloat</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Ripple</NICK>
<BLURB>Amount of ripple (dB).</BLURB>
<DEFAULT>0.25</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioChebyshevFreqLimit::type</NAME>
<TYPE>gint</TYPE>
<RANGE>[1,2]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Type</NICK>
<BLURB>Type of the chebychev filter.</BLURB>
<DEFAULT>1</DEFAULT>
</ARG>

12
docs/plugins/inspect/plugin-1394.xml
View file

@ -3,10 +3,10 @@
<description>Source for DV data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
<version>0.10.6</version>
<version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<package>GStreamer Good Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
@ -18,6 +18,14 @@
Daniel Fischer &lt;dan@f3c.com&gt;
Wim Taymans &lt;wim@fluendo.com&gt;
Zaheer Abbas Merali &lt;zaheerabbas at merali dot org&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-dv, format=(string){ NTSC, PAL }, systemstream=(boolean)true</details>
</caps>
</pads>
</element>
</elements>
</plugin>

42
docs/plugins/inspect/plugin-audiofx.xml
View file

@ -30,6 +30,48 @@
</caps>
</pads>
</element>
<element>
<name>audiochebyshevfreqband</name>
<longname>AudioChebyshevFreqBand</longname>
<class>Filter/Effect/Audio</class>
<description>Chebyshev band pass and band reject filter</description>
<author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
<element>
<name>audiochebyshevfreqlimit</name>
<longname>AudioChebyshevFreqLimit</longname>
<class>Filter/Effect/Audio</class>
<description>Chebyshev low pass and high pass filter</description>
<author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
<element>
<name>audiodynamic</name>
<longname>AudioDynamic</longname>

38
docs/plugins/inspect/plugin-dv.xml
View file

@ -3,10 +3,10 @@
<description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
<filename>../../ext/dv/.libs/libgstdv.so</filename>
<basename>libgstdv.so</basename>
<version>0.10.6</version>
<version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<package>GStreamer Good Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
@ -15,6 +15,20 @@
<class>Codec/Decoder/Video</class>
<description>Uses libdv to decode DV video (smpte314) (libdv.sourceforge.net)</description>
<author>Erik Walthinsen &lt;omega@cse.ogi.edu&gt;,Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-dv, systemstream=(boolean)false</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)YUY2, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)24, endianness=(int)4321, red_mask=(int)65280, green_mask=(int)16711680, blue_mask=(int)-16777216, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)720, framerate=(fraction)[ 1/1, 60/1 ]</details>
</caps>
</pads>
</element>
<element>
<name>dvdemux</name>
@ -22,6 +36,26 @@
<class>Codec/Demuxer</class>
<description>Uses libdv to separate DV audio from DV video (libdv.sourceforge.net)</description>
<author>Erik Walthinsen &lt;omega@cse.ogi.edu&gt;, Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-dv, systemstream=(boolean)true</details>
</caps>
<caps>
<name>video</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>video/x-dv, systemstream=(boolean)false</details>
</caps>
<caps>
<name>audio</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>audio/x-raw-int, depth=(int)16, width=(int)16, signed=(boolean)true, channels=(int){ 2, 4 }, endianness=(int)1234, rate=(int){ 32000, 44100, 48000 }</details>
</caps>
</pads>
</element>
</elements>
</plugin>

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

@ -47,7 +47,7 @@
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ]</details>
<details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]</details>
</caps>
</pads>
</element>

48
docs/plugins/inspect/plugin-jpeg.xml
View file

@ -16,18 +16,18 @@
<description>Decode images from JPEG format</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 8, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 8, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
</pads>
</element>
<element>
@ -37,18 +37,18 @@
<description>Encode images in JPEG format</description>
<author>Wim Taymans &lt;wim.taymans@tvd.be&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>image/jpeg, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
</pads>
</element>
<element>
@ -58,18 +58,18 @@
<description>Decode video from Smoke format</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
</pads>
</element>
<element>
@ -79,18 +79,18 @@
<description>Encode images into the Smoke format</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-smoke, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
</pads>
</element>
</elements>

24
docs/plugins/inspect/plugin-png.xml
View file

@ -16,18 +16,18 @@
<description>Decode a png video frame to a raw image</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>image/png</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-rgb, bpp=(int)32, depth=(int)32, endianness=(int)4321, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, alpha_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>image/png</details>
</caps>
</pads>
</element>
<element>
@ -37,18 +37,18 @@
<description>Encode a video frame to a .png image</description>
<author>Jeremy SIMON &lt;jsimon13@yahoo.fr&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>image/png, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/x-raw-rgb, bpp=(int)32, depth=(int)32, endianness=(int)4321, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, alpha_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>image/png, width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
</pads>
</element>
</elements>

8
docs/plugins/inspect/plugin-rtp.xml
View file

@ -137,7 +137,7 @@
</element>
<element>
<name>rtpdepay</name>
<longname>RTP payloader</longname>
<longname>RTP depayloader</longname>
<class>Codec/Depayloader/Network</class>
<description>Accepts raw RTP and RTCP packets and sends them forward</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
@ -332,7 +332,7 @@
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(int){ 20, 30 }</details>
<details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(string){ 20, 30 }</details>
</caps>
</pads>
</element>
@ -353,7 +353,7 @@
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(int){ 20, 30 }</details>
<details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)ILBC, mode=(string){ 20, 30 }</details>
</caps>
</pads>
</element>
@ -558,7 +558,7 @@
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/mpeg, systemstream=(boolean)false</details>
<details>video/mpeg, mpegversion=(int)2, systemstream=(boolean)false</details>
</caps>
<caps>
<name>sink</name>

10
docs/plugins/inspect/plugin-shout2send.xml
View file

@ -3,7 +3,7 @@
<description>Sends data to an icecast server using libshout2</description>
<filename>../../ext/shout2/.libs/libgstshout2.so</filename>
<basename>libgstshout2.so</basename>
<version>0.10.6</version>
<version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>libshout2</package>
@ -17,6 +17,14 @@
<author>Wim Taymans &lt;wim.taymans@chello.be&gt;
Pedro Corte-Real &lt;typo@netcabo.pt&gt;
Zaheer Abbas Merali &lt;zaheerabbas at merali dot org&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/ogg; audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ]</details>
</caps>
</pads>
</element>
</elements>
</plugin>

58
docs/plugins/inspect/plugin-wavpack.xml
View file

@ -3,10 +3,10 @@
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
<version>0.10.6</version>
<version>0.10.6.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<package>GStreamer Good Plug-ins CVS/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
@ -15,6 +15,20 @@
<class>Codec/Decoder/Audio</class>
<description>Decodes Wavpack audio data</description>
<author>Arwed v. Merkatz &lt;v.merkatz@gmx.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-int, width=(int)32, depth=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], endianness=(int)1234, signed=(boolean)true</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true</details>
</caps>
</pads>
</element>
<element>
<name>wavpackenc</name>
@ -22,6 +36,26 @@
<class>Codec/Encoder/Audio</class>
<description>Encodes audio with the Wavpack lossless/lossy audio codec</description>
<author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-int, width=(int)32, depth=(int)[ 1, 32 ], endianness=(int)1234, channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], signed=(boolean)true</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true</details>
</caps>
<caps>
<name>wvcsrc</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>audio/x-wavpack-correction, framed=(boolean)true</details>
</caps>
</pads>
</element>
<element>
<name>wavpackparse</name>
@ -29,6 +63,26 @@
<class>Codec/Demuxer/Audio</class>
<description>Parses Wavpack files</description>
<author>Arwed v. Merkatz &lt;v.merkatz@gmx.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>audio/x-wavpack, width=(int)[ 1, 32 ], channels=(int)[ 1, 2 ], rate=(int)[ 6000, 192000 ], framed=(boolean)true</details>
</caps>
<caps>
<name>wvcsrc</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>audio/x-wavpack-correction, framed=(boolean)true</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-wavpack, framed=(boolean)false; audio/x-wavpack-correction, framed=(boolean)false</details>
</caps>
</pads>
</element>
</elements>
</plugin>

11
gst/audiofx/Makefile.am
View file

@ -7,7 +7,9 @@ libgstaudiofx_la_SOURCES = audiofx.c\
audiopanorama.c \
audioinvert.c \
audioamplify.c \
audiodynamic.c
audiodynamic.c \
audiochebyshevfreqlimit.c \
audiochebyshevfreqband.c
# flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
@ -18,12 +20,15 @@ libgstaudiofx_la_LIBADD = $(GST_LIBS) \
$(GST_BASE_LIBS) \
$(GST_CONTROLLER_LIBS) \
$(GST_PLUGINS_BASE_LIBS) \
-lgstaudio-$(GST_MAJORMINOR)
-lgstaudio-$(GST_MAJORMINOR) \
$(LIBM)
libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
# headers we need but don't want installed
noinst_HEADERS = audiopanorama.h \
audioinvert.h \
audioamplify.h \
audiodynamic.h
audiodynamic.h \
audiochebyshevfreqlimit.h \
audiochebyshevfreqband.c

916
gst/audiofx/audiochebband.c Normal file
View file

@ -0,0 +1,916 @@
/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Chebyshev type 1 filter design based on
* "The Scientist and Engineer's Guide to DSP", Chapter 20.
* http://www.dspguide.com/
*
* For type 2 and Chebyshev filters in general read
* http://en.wikipedia.org/wiki/Chebyshev_filter
*
* Transformation from lowpass to bandpass/bandreject:
* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm
* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm
*
*/
/**
* SECTION:element-audiochebyshevfreqband
* @short_description: Chebyshev band pass and band reject filter
*
* <refsect2>
* <para>
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The number of poles and the ripple parameter control the rolloff.
* </para>
* <para>
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
* </para>
* <para>
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
* </para>
* <para>
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include <math.h>
#include "audiochebyshevfreqband.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
"Filter/Effect/Audio",
"Chebyshev band pass and band reject filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MODE,
PROP_TYPE,
PROP_LOWER_FREQUENCY,
PROP_UPPER_FREQUENCY,
PROP_RIPPLE,
PROP_POLES
};
#define ALLOWED_CAPS \
"audio/x-raw-float," \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER," \
" rate = (int) [ 1, MAX ]," \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqBand * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqBand * filter,
gfloat * data, guint num_samples);
enum
{
MODE_BAND_PASS = 0,
MODE_BAND_REJECT
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
static GType
gst_audio_chebyshev_freq_band_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_BAND_PASS, "Band pass (default)",
"band-pass"},
{MODE_BAND_REJECT, "Band reject",
"band-reject"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_band_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &element_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
}
static void
gst_audio_chebyshev_freq_band_dispose (GObject * object)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE,
MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type",
"Type of the chebychev filter", 1, 2,
1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
g_param_spec_float ("lower-frequency", "Lower frequency",
"Start frequency of the band (Hz)", 0.0, G_MAXFLOAT,
0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
g_param_spec_float ("upper-frequency", "Upper frequency",
"Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT,
0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple",
"Amount of ripple (dB)", 0.0, G_MAXFLOAT,
0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next multiply of four",
4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
}
static void
gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandClass * klass)
{
filter->lower_frequency = filter->upper_frequency = 0.0;
filter->mode = MODE_BAND_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
filter->have_coeffs = FALSE;
filter->num_a = 0;
filter->num_b = 0;
filter->channels = NULL;
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
{
gint np = filter->poles / 2;
gdouble ripple = filter->ripple;
/* pole location in s-plane */
gdouble rp, ip;
/* zero location in s-plane */
gdouble rz = 0.0, iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
gint type = filter->type;
/* Calculate pole location for lowpass at frequency 1 */
{
gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
}
/* If we allow ripple, move the pole from the unit
* circle to an ellipse and keep cutoff at frequency 1 */
if (ripple > 0 && type == 1) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (1.0 / es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
} else if (type == 2) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
}
/* Calculate inverse of the pole location to move from
* type I to type II */
if (type == 2) {
gdouble mag2 = rp * rp + ip * ip;
rp /= mag2;
ip /= mag2;
}
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
gdouble mag2;
rz = 0.0;
iz = cos (angle);
mag2 = rz * rz + iz * iz;
rz /= mag2;
iz /= mag2;
}
/* Convert from s-domain to z-domain by
* using the bilinear Z-transform, i.e.
* substitute s by (2/t)*((z-1)/(z+1))
* with t = 2 * tan(0.5).
*/
if (type == 1) {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t) / d;
x1 = 2.0 * x0;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
} else {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t * iz * iz + 4.0) / d;
x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
}
/* Convert from lowpass at frequency 1 to either bandpass
* or band reject.
*
* For bandpass substitute z^(-1) with:
*
* -2 -1
* -z + alpha * z - beta
* ----------------------------
* -2 -1
* beta * z - alpha * z + 1
*
* alpha = (2*a*b)/(1+b)
* beta = (b-1)/(b+1)
* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
* b = tan(1/2) * cot((w1 - w0)/2)
*
* For bandreject substitute z^(-1) with:
*
* -2 -1
* z - alpha * z + beta
* ----------------------------
* -2 -1
* beta * z - alpha * z + 1
*
* alpha = (2*a)/(1+b)
* beta = (1-b)/(1+b)
* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
* b = tan(1/2) * tan((w1 - w0)/2)
*
*/
{
gdouble a, b, d;
gdouble alpha, beta;
gdouble w0 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w1 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
if (filter->mode == MODE_BAND_PASS) {
a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0);
alpha = (2.0 * a * b) / (1.0 + b);
beta = (b - 1.0) / (b + 1.0);
d = 1.0 + beta * (y1 - beta * y2);
*a0 = (x0 + beta * (-x1 + beta * x2)) / d;
*a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d;
*a2 =
(-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) +
alpha * alpha * (x0 - x1 + x2)) / d;
*a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d;
*a4 = (beta * (beta * x0 - x1) + x2) / d;
*b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d;
*b2 =
(-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) +
2.0 * beta * (-1.0 + y2)) / d;
*b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d;
*b4 = (-beta * beta - beta * y1 + y2) / d;
} else {
a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0);
alpha = (2.0 * a) / (1.0 + b);
beta = (1.0 - b) / (1.0 + b);
d = -1.0 + beta * (beta * y2 + y1);
*a0 = (-x0 - beta * x1 - beta * beta * x2) / d;
*a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d;
*a2 =
(-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) -
alpha * alpha * (x0 + x1 + x2)) / d;
*a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d;
*a4 = (-beta * beta * x0 - beta * x1 - x2) / d;
*b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d;
*b2 =
-(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) +
alpha * alpha * (-1.0 + y1 + y2)) / d;
*b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d;
*b4 = -(-beta * beta + beta * y1 + y2) / d;
}
}
}
/* Evaluate the transfer function that corresponds to the IIR
* coefficients at zr + zi*I and return the magnitude */
static gdouble
calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
gdouble zi)
{
gdouble sum_ar, sum_ai;
gdouble sum_br, sum_bi;
gdouble gain_r, gain_i;
gdouble sum_r_old;
gdouble sum_i_old;
gint i;
sum_ar = 0.0;
sum_ai = 0.0;
for (i = num_a; i >= 0; i--) {
sum_r_old = sum_ar;
sum_i_old = sum_ai;
sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
}
sum_br = 0.0;
sum_bi = 0.0;
for (i = num_b; i >= 0; i--) {
sum_r_old = sum_br;
sum_i_old = sum_bi;
sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
}
sum_br += 1.0;
sum_bi += 0.0;
gain_r =
(sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
gain_i =
(sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
return (sqrt (gain_r * gain_r + gain_i * gain_i));
}
static void
generate_coefficients (GstAudioChebyshevFreqBand * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
filter->have_coeffs = TRUE;
if (filter->upper_frequency <= filter->lower_frequency) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
return;
}
if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) {
filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2;
GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency");
}
if (filter->lower_frequency < 0.0) {
filter->lower_frequency = 0.0;
GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0");
}
/* Calculate coefficients for the chebyshev filter */
{
gint np = filter->poles;
gdouble *a, *b;
gint i, p;
filter->num_a = np + 1;
filter->a = a = g_new0 (gdouble, np + 5);
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 5);
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
}
/* Calculate transfer function coefficients */
a[4] = 1.0;
b[4] = 1.0;
for (p = 1; p <= np / 4; p++) {
gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4;
gdouble *ta = g_new0 (gdouble, np + 5);
gdouble *tb = g_new0 (gdouble, np + 5);
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1,
&b2, &b3, &b4);
memcpy (ta, a, sizeof (gdouble) * (np + 5));
memcpy (tb, b, sizeof (gdouble) * (np + 5));
/* add the new coefficients for the new two poles
* to the cascade by multiplication of the transfer
* functions */
for (i = 4; i < np + 5; i++) {
a[i] =
a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] +
a4 * ta[i - 4];
b[i] =
tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] -
b4 * tb[i - 4];
}
g_free (ta);
g_free (tb);
}
/* Move coefficients to the beginning of the array
* and multiply the b coefficients with -1 to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
b[4] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 4];
b[i] = -b[i + 4];
}
/* Normalize to unity gain at frequency 0 and frequency
* 0.5 for bandreject and unity gain at band center frequency
* for bandpass */
if (filter->mode == MODE_BAND_REJECT) {
/* gain is sqrt(H(0)*H(0.5)) */
gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0);
gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0);
gain1 = sqrt (gain1 * gain2);
for (i = 0; i <= np; i++) {
a[i] /= gain1;
}
} else {
/* gain is H(wc), wc = center frequency */
gdouble w1 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w2 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w0 = (w2 + w1) / 2.0;
gdouble zr = cos (w0), zi = sin (w0);
gdouble gain = calculate_gain (a, b, np, np, zr, zi);
for (i = 0; i <= np; i++) {
a[i] /= gain;
}
}
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
"mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB",
(filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject",
filter->type, filter->poles, filter->lower_frequency,
filter->upper_frequency, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz",
20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
{
gdouble w1 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w2 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w0 = (w2 + w1) / 2.0;
gdouble zr, zi;
zr = cos (w1);
zi = sin (w1);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->lower_frequency);
zr = cos (w0);
zi = sin (w0);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) ((filter->lower_frequency + filter->upper_frequency) / 2.0));
zr = cos (w2);
zi = sin (w2);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->upper_frequency);
}
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
GST_AUDIO_FILTER (filter)->format.rate / 2);
}
}
static void
gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
switch (prop_id) {
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->mode = g_value_get_enum (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_TYPE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->type = g_value_get_int (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_LOWER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (filter);
filter->lower_frequency = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_UPPER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (filter);
filter->upper_frequency = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_RIPPLE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->ripple = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_POLES:
GST_BASE_TRANSFORM_LOCK (filter);
filter->poles = GST_ROUND_UP_4 (g_value_get_int (value));
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_TYPE:
g_value_set_int (value, filter->type);
break;
case PROP_LOWER_FREQUENCY:
g_value_set_float (value, filter->lower_frequency);
break;
case PROP_UPPER_FREQUENCY:
g_value_set_float (value, filter->upper_frequency);
break;
case PROP_RIPPLE:
g_value_set_float (value, filter->ripple);
break;
case PROP_POLES:
g_value_set_int (value, filter->poles);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
process_64;
else
ret = FALSE;
filter->have_coeffs = FALSE;
return ret;
}
static inline gdouble
process (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
val += filter->a[i] * ctx->x[j];
j--;
if (j < 0)
j = filter->num_a - 1;
}
for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
val += filter->b[i] * ctx->y[j];
j--;
if (j < 0)
j = filter->num_b - 1;
}
if (ctx->x) {
ctx->x_pos++;
if (ctx->x_pos > filter->num_a - 1)
ctx->x_pos = 0;
ctx->x[ctx->x_pos] = x0;
}
if (ctx->y) {
ctx->y_pos++;
if (ctx->y_pos > filter->num_b - 1)
ctx->y_pos = 0;
ctx->y[ctx->y_pos] = val;
}
return val;
}
static void
process_64 (GstAudioChebyshevFreqBand * filter,
gdouble * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
static void
process_32 (GstAudioChebyshevFreqBand * filter,
gfloat * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (!gst_buffer_is_writable (buf))
return GST_FLOW_OK;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (!filter->have_coeffs)
generate_coefficients (filter);
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static gboolean
gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if
* already existing */
if (channels && filter->channels) {
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
if (ctx->x)
memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
if (ctx->y)
memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
}
}
return TRUE;
}

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/*
* 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__ */

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/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Chebyshev type 1 filter design based on
* "The Scientist and Engineer's Guide to DSP", Chapter 20.
* http://www.dspguide.com/
*
* For type 2 and Chebyshev filters in general read
* http://en.wikipedia.org/wiki/Chebyshev_filter
*
*/
/**
* SECTION:element-audiochebyshevfreqlimit
* @short_description: Chebyshev low pass and high pass filter
*
* <refsect2>
* <para>
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
* </para>
* <para>
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
* </para>
* <para>
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
* </para>
* <para>
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include <math.h>
#include "audiochebyshevfreqlimit.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
"Filter/Effect/Audio",
"Chebyshev low pass and high pass filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MODE,
PROP_TYPE,
PROP_CUTOFF,
PROP_RIPPLE,
PROP_POLES
};
#define ALLOWED_CAPS \
"audio/x-raw-float," \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER," \
" rate = (int) [ 1, MAX ]," \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
DEBUG_INIT);
static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqLimit * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqLimit * filter,
gfloat * data, guint num_samples);
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
static GType
gst_audio_chebyshev_freq_limit_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_LOW_PASS, "Low pass (default)",
"low-pass"},
{MODE_HIGH_PASS, "High pass",
"high-pass"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &element_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
}
static void
gst_audio_chebyshev_freq_limit_dispose (GObject * object)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode",
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_CUTOFF,
g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next even number",
2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
}
static void
gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitClass * klass)
{
filter->cutoff = 0.0;
filter->mode = MODE_LOW_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
filter->have_coeffs = FALSE;
filter->num_a = 0;
filter->num_b = 0;
filter->channels = NULL;
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2,
gdouble * b1, gdouble * b2)
{
gint np = filter->poles;
gdouble ripple = filter->ripple;
/* pole location in s-plane */
gdouble rp, ip;
/* zero location in s-plane */
gdouble rz = 0.0, iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
gint type = filter->type;
/* Calculate pole location for lowpass at frequency 1 */
{
gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
}
/* If we allow ripple, move the pole from the unit
* circle to an ellipse and keep cutoff at frequency 1 */
if (ripple > 0 && type == 1) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (1.0 / es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
} else if (type == 2) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
}
/* Calculate inverse of the pole location to convert from
* type I to type II */
if (type == 2) {
gdouble mag2 = rp * rp + ip * ip;
rp /= mag2;
ip /= mag2;
}
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
gdouble mag2;
rz = 0.0;
iz = cos (angle);
mag2 = rz * rz + iz * iz;
rz /= mag2;
iz /= mag2;
}
/* Convert from s-domain to z-domain by
* using the bilinear Z-transform, i.e.
* substitute s by (2/t)*((z-1)/(z+1))
* with t = 2 * tan(0.5).
*/
if (type == 1) {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t) / d;
x1 = 2.0 * x0;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
} else {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t * iz * iz + 4.0) / d;
x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
}
/* Convert from lowpass at frequency 1 to either lowpass
* or highpass.
*
* For lowpass substitute z^(-1) with:
* -1
* z - k
* ------------
* -1
* 1 - k * z
*
* k = sin((1-w)/2) / sin((1+w)/2)
*
* For highpass substitute z^(-1) with:
*
* -1
* -z - k
* ------------
* -1
* 1 + k * z
*
* k = -cos((1+w)/2) / cos((1-w)/2)
*
*/
{
gdouble k, d;
gdouble omega =
2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
if (filter->mode == MODE_LOW_PASS)
k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
else
k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
d = 1.0 + y1 * k - y2 * k * k;
*a0 = (x0 + k * (-x1 + k * x2)) / d;
*a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
*a2 = (x0 * k * k - x1 * k + x2) / d;
*b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
*b2 = (-k * k - y1 * k + y2) / d;
if (filter->mode == MODE_HIGH_PASS) {
*a1 = -*a1;
*b1 = -*b1;
}
}
}
/* Evaluate the transfer function that corresponds to the IIR
* coefficients at zr + zi*I and return the magnitude */
static gdouble
calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
gdouble zi)
{
gdouble sum_ar, sum_ai;
gdouble sum_br, sum_bi;
gdouble gain_r, gain_i;
gdouble sum_r_old;
gdouble sum_i_old;
gint i;
sum_ar = 0.0;
sum_ai = 0.0;
for (i = num_a; i >= 0; i--) {
sum_r_old = sum_ar;
sum_i_old = sum_ai;
sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
}
sum_br = 0.0;
sum_bi = 0.0;
for (i = num_b; i >= 0; i--) {
sum_r_old = sum_br;
sum_i_old = sum_bi;
sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
}
sum_br += 1.0;
sum_bi += 0.0;
gain_r =
(sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
gain_i =
(sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
return (sqrt (gain_r * gain_r + gain_i * gain_i));
}
static void
generate_coefficients (GstAudioChebyshevFreqLimit * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
filter->have_coeffs = TRUE;
if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return;
} else if (filter->cutoff <= 0.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return;
}
/* Calculate coefficients for the chebyshev filter */
{
gint np = filter->poles;
gdouble *a, *b;
gint i, p;
filter->num_a = np + 1;
filter->a = a = g_new0 (gdouble, np + 3);
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 3);
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
}
/* Calculate transfer function coefficients */
a[2] = 1.0;
b[2] = 1.0;
for (p = 1; p <= np / 2; p++) {
gdouble a0, a1, a2, b1, b2;
gdouble *ta = g_new0 (gdouble, np + 3);
gdouble *tb = g_new0 (gdouble, np + 3);
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
memcpy (ta, a, sizeof (gdouble) * (np + 3));
memcpy (tb, b, sizeof (gdouble) * (np + 3));
/* add the new coefficients for the new two poles
* to the cascade by multiplication of the transfer
* functions */
for (i = 2; i < np + 3; i++) {
a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
}
g_free (ta);
g_free (tb);
}
/* Move coefficients to the beginning of the array
* and multiply the b coefficients with -1 to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
b[2] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 2];
b[i] = -b[i + 2];
}
/* Normalize to unity gain at frequency 0 for lowpass
* and frequency 0.5 for highpass */
{
gdouble gain;
if (filter->mode == MODE_LOW_PASS)
gain = calculate_gain (a, b, np, np, 1.0, 0.0);
else
gain = calculate_gain (a, b, np, np, -1.0, 0.0);
for (i = 0; i <= np; i++) {
a[i] /= gain;
}
}
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
"mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB",
(filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
filter->type, filter->poles, filter->cutoff, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
{
gdouble wc =
2.0 * M_PI * (filter->cutoff /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble zr = cos (wc), zi = sin (wc);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->cutoff);
}
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
GST_AUDIO_FILTER (filter)->format.rate / 2);
}
}
static void
gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->mode = g_value_get_enum (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_TYPE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->type = g_value_get_int (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_CUTOFF:
GST_BASE_TRANSFORM_LOCK (filter);
filter->cutoff = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_RIPPLE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->ripple = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_POLES:
GST_BASE_TRANSFORM_LOCK (filter);
filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_TYPE:
g_value_set_int (value, filter->type);
break;
case PROP_CUTOFF:
g_value_set_float (value, filter->cutoff);
break;
case PROP_RIPPLE:
g_value_set_float (value, filter->ripple);
break;
case PROP_POLES:
g_value_set_int (value, filter->poles);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_64;
else
ret = FALSE;
filter->have_coeffs = FALSE;
return ret;
}
static inline gdouble
process (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
val += filter->a[i] * ctx->x[j];
j--;
if (j < 0)
j = filter->num_a - 1;
}
for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
val += filter->b[i] * ctx->y[j];
j--;
if (j < 0)
j = filter->num_b - 1;
}
if (ctx->x) {
ctx->x_pos++;
if (ctx->x_pos > filter->num_a - 1)
ctx->x_pos = 0;
ctx->x[ctx->x_pos] = x0;
}
if (ctx->y) {
ctx->y_pos++;
if (ctx->y_pos > filter->num_b - 1)
ctx->y_pos = 0;
ctx->y[ctx->y_pos] = val;
}
return val;
}
static void
process_64 (GstAudioChebyshevFreqLimit * filter,
gdouble * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
static void
process_32 (GstAudioChebyshevFreqLimit * filter,
gfloat * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (!gst_buffer_is_writable (buf))
return GST_FLOW_OK;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (!filter->have_coeffs)
generate_coefficients (filter);
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static gboolean
gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if
* already existing */
if (channels && filter->channels) {
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
if (ctx->x)
memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
if (ctx->y)
memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
}
}
return TRUE;
}

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/*
* 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__ */

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/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Chebyshev type 1 filter design based on
* "The Scientist and Engineer's Guide to DSP", Chapter 20.
* http://www.dspguide.com/
*
* For type 2 and Chebyshev filters in general read
* http://en.wikipedia.org/wiki/Chebyshev_filter
*
* Transformation from lowpass to bandpass/bandreject:
* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm
* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm
*
*/
/**
* SECTION:element-audiochebyshevfreqband
* @short_description: Chebyshev band pass and band reject filter
*
* <refsect2>
* <para>
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The number of poles and the ripple parameter control the rolloff.
* </para>
* <para>
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
* </para>
* <para>
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
* </para>
* <para>
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include <math.h>
#include "audiochebyshevfreqband.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
"Filter/Effect/Audio",
"Chebyshev band pass and band reject filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MODE,
PROP_TYPE,
PROP_LOWER_FREQUENCY,
PROP_UPPER_FREQUENCY,
PROP_RIPPLE,
PROP_POLES
};
#define ALLOWED_CAPS \
"audio/x-raw-float," \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER," \
" rate = (int) [ 1, MAX ]," \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqBand * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqBand * filter,
gfloat * data, guint num_samples);
enum
{
MODE_BAND_PASS = 0,
MODE_BAND_REJECT
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
static GType
gst_audio_chebyshev_freq_band_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_BAND_PASS, "Band pass (default)",
"band-pass"},
{MODE_BAND_REJECT, "Band reject",
"band-reject"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_band_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &element_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
}
static void
gst_audio_chebyshev_freq_band_dispose (GObject * object)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE,
MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type",
"Type of the chebychev filter", 1, 2,
1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
g_param_spec_float ("lower-frequency", "Lower frequency",
"Start frequency of the band (Hz)", 0.0, G_MAXFLOAT,
0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
g_param_spec_float ("upper-frequency", "Upper frequency",
"Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT,
0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple",
"Amount of ripple (dB)", 0.0, G_MAXFLOAT,
0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next multiply of four",
4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
}
static void
gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandClass * klass)
{
filter->lower_frequency = filter->upper_frequency = 0.0;
filter->mode = MODE_BAND_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
filter->have_coeffs = FALSE;
filter->num_a = 0;
filter->num_b = 0;
filter->channels = NULL;
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
{
gint np = filter->poles / 2;
gdouble ripple = filter->ripple;
/* pole location in s-plane */
gdouble rp, ip;
/* zero location in s-plane */
gdouble rz = 0.0, iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
gint type = filter->type;
/* Calculate pole location for lowpass at frequency 1 */
{
gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
}
/* If we allow ripple, move the pole from the unit
* circle to an ellipse and keep cutoff at frequency 1 */
if (ripple > 0 && type == 1) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (1.0 / es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
} else if (type == 2) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
}
/* Calculate inverse of the pole location to move from
* type I to type II */
if (type == 2) {
gdouble mag2 = rp * rp + ip * ip;
rp /= mag2;
ip /= mag2;
}
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
gdouble mag2;
rz = 0.0;
iz = cos (angle);
mag2 = rz * rz + iz * iz;
rz /= mag2;
iz /= mag2;
}
/* Convert from s-domain to z-domain by
* using the bilinear Z-transform, i.e.
* substitute s by (2/t)*((z-1)/(z+1))
* with t = 2 * tan(0.5).
*/
if (type == 1) {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t) / d;
x1 = 2.0 * x0;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
} else {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t * iz * iz + 4.0) / d;
x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
}
/* Convert from lowpass at frequency 1 to either bandpass
* or band reject.
*
* For bandpass substitute z^(-1) with:
*
* -2 -1
* -z + alpha * z - beta
* ----------------------------
* -2 -1
* beta * z - alpha * z + 1
*
* alpha = (2*a*b)/(1+b)
* beta = (b-1)/(b+1)
* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
* b = tan(1/2) * cot((w1 - w0)/2)
*
* For bandreject substitute z^(-1) with:
*
* -2 -1
* z - alpha * z + beta
* ----------------------------
* -2 -1
* beta * z - alpha * z + 1
*
* alpha = (2*a)/(1+b)
* beta = (1-b)/(1+b)
* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
* b = tan(1/2) * tan((w1 - w0)/2)
*
*/
{
gdouble a, b, d;
gdouble alpha, beta;
gdouble w0 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w1 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
if (filter->mode == MODE_BAND_PASS) {
a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0);
alpha = (2.0 * a * b) / (1.0 + b);
beta = (b - 1.0) / (b + 1.0);
d = 1.0 + beta * (y1 - beta * y2);
*a0 = (x0 + beta * (-x1 + beta * x2)) / d;
*a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d;
*a2 =
(-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) +
alpha * alpha * (x0 - x1 + x2)) / d;
*a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d;
*a4 = (beta * (beta * x0 - x1) + x2) / d;
*b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d;
*b2 =
(-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) +
2.0 * beta * (-1.0 + y2)) / d;
*b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d;
*b4 = (-beta * beta - beta * y1 + y2) / d;
} else {
a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0);
alpha = (2.0 * a) / (1.0 + b);
beta = (1.0 - b) / (1.0 + b);
d = -1.0 + beta * (beta * y2 + y1);
*a0 = (-x0 - beta * x1 - beta * beta * x2) / d;
*a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d;
*a2 =
(-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) -
alpha * alpha * (x0 + x1 + x2)) / d;
*a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d;
*a4 = (-beta * beta * x0 - beta * x1 - x2) / d;
*b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d;
*b2 =
-(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) +
alpha * alpha * (-1.0 + y1 + y2)) / d;
*b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d;
*b4 = -(-beta * beta + beta * y1 + y2) / d;
}
}
}
/* Evaluate the transfer function that corresponds to the IIR
* coefficients at zr + zi*I and return the magnitude */
static gdouble
calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
gdouble zi)
{
gdouble sum_ar, sum_ai;
gdouble sum_br, sum_bi;
gdouble gain_r, gain_i;
gdouble sum_r_old;
gdouble sum_i_old;
gint i;
sum_ar = 0.0;
sum_ai = 0.0;
for (i = num_a; i >= 0; i--) {
sum_r_old = sum_ar;
sum_i_old = sum_ai;
sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
}
sum_br = 0.0;
sum_bi = 0.0;
for (i = num_b; i >= 0; i--) {
sum_r_old = sum_br;
sum_i_old = sum_bi;
sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
}
sum_br += 1.0;
sum_bi += 0.0;
gain_r =
(sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
gain_i =
(sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
return (sqrt (gain_r * gain_r + gain_i * gain_i));
}
static void
generate_coefficients (GstAudioChebyshevFreqBand * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
filter->have_coeffs = TRUE;
if (filter->upper_frequency <= filter->lower_frequency) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
return;
}
if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) {
filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2;
GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency");
}
if (filter->lower_frequency < 0.0) {
filter->lower_frequency = 0.0;
GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0");
}
/* Calculate coefficients for the chebyshev filter */
{
gint np = filter->poles;
gdouble *a, *b;
gint i, p;
filter->num_a = np + 1;
filter->a = a = g_new0 (gdouble, np + 5);
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 5);
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
}
/* Calculate transfer function coefficients */
a[4] = 1.0;
b[4] = 1.0;
for (p = 1; p <= np / 4; p++) {
gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4;
gdouble *ta = g_new0 (gdouble, np + 5);
gdouble *tb = g_new0 (gdouble, np + 5);
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1,
&b2, &b3, &b4);
memcpy (ta, a, sizeof (gdouble) * (np + 5));
memcpy (tb, b, sizeof (gdouble) * (np + 5));
/* add the new coefficients for the new two poles
* to the cascade by multiplication of the transfer
* functions */
for (i = 4; i < np + 5; i++) {
a[i] =
a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] +
a4 * ta[i - 4];
b[i] =
tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] -
b4 * tb[i - 4];
}
g_free (ta);
g_free (tb);
}
/* Move coefficients to the beginning of the array
* and multiply the b coefficients with -1 to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
b[4] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 4];
b[i] = -b[i + 4];
}
/* Normalize to unity gain at frequency 0 and frequency
* 0.5 for bandreject and unity gain at band center frequency
* for bandpass */
if (filter->mode == MODE_BAND_REJECT) {
/* gain is sqrt(H(0)*H(0.5)) */
gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0);
gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0);
gain1 = sqrt (gain1 * gain2);
for (i = 0; i <= np; i++) {
a[i] /= gain1;
}
} else {
/* gain is H(wc), wc = center frequency */
gdouble w1 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w2 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w0 = (w2 + w1) / 2.0;
gdouble zr = cos (w0), zi = sin (w0);
gdouble gain = calculate_gain (a, b, np, np, zr, zi);
for (i = 0; i <= np; i++) {
a[i] /= gain;
}
}
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
"mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB",
(filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject",
filter->type, filter->poles, filter->lower_frequency,
filter->upper_frequency, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz",
20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
{
gdouble w1 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w2 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w0 = (w2 + w1) / 2.0;
gdouble zr, zi;
zr = cos (w1);
zi = sin (w1);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->lower_frequency);
zr = cos (w0);
zi = sin (w0);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) ((filter->lower_frequency + filter->upper_frequency) / 2.0));
zr = cos (w2);
zi = sin (w2);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->upper_frequency);
}
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
GST_AUDIO_FILTER (filter)->format.rate / 2);
}
}
static void
gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
switch (prop_id) {
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->mode = g_value_get_enum (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_TYPE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->type = g_value_get_int (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_LOWER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (filter);
filter->lower_frequency = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_UPPER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (filter);
filter->upper_frequency = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_RIPPLE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->ripple = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_POLES:
GST_BASE_TRANSFORM_LOCK (filter);
filter->poles = GST_ROUND_UP_4 (g_value_get_int (value));
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_TYPE:
g_value_set_int (value, filter->type);
break;
case PROP_LOWER_FREQUENCY:
g_value_set_float (value, filter->lower_frequency);
break;
case PROP_UPPER_FREQUENCY:
g_value_set_float (value, filter->upper_frequency);
break;
case PROP_RIPPLE:
g_value_set_float (value, filter->ripple);
break;
case PROP_POLES:
g_value_set_int (value, filter->poles);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
process_64;
else
ret = FALSE;
filter->have_coeffs = FALSE;
return ret;
}
static inline gdouble
process (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
val += filter->a[i] * ctx->x[j];
j--;
if (j < 0)
j = filter->num_a - 1;
}
for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
val += filter->b[i] * ctx->y[j];
j--;
if (j < 0)
j = filter->num_b - 1;
}
if (ctx->x) {
ctx->x_pos++;
if (ctx->x_pos > filter->num_a - 1)
ctx->x_pos = 0;
ctx->x[ctx->x_pos] = x0;
}
if (ctx->y) {
ctx->y_pos++;
if (ctx->y_pos > filter->num_b - 1)
ctx->y_pos = 0;
ctx->y[ctx->y_pos] = val;
}
return val;
}
static void
process_64 (GstAudioChebyshevFreqBand * filter,
gdouble * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
static void
process_32 (GstAudioChebyshevFreqBand * filter,
gfloat * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (!gst_buffer_is_writable (buf))
return GST_FLOW_OK;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (!filter->have_coeffs)
generate_coefficients (filter);
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static gboolean
gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if
* already existing */
if (channels && filter->channels) {
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
if (ctx->x)
memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
if (ctx->y)
memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
}
}
return TRUE;
}

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/*
* 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__ */

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/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Chebyshev type 1 filter design based on
* "The Scientist and Engineer's Guide to DSP", Chapter 20.
* http://www.dspguide.com/
*
* For type 2 and Chebyshev filters in general read
* http://en.wikipedia.org/wiki/Chebyshev_filter
*
*/
/**
* SECTION:element-audiochebyshevfreqlimit
* @short_description: Chebyshev low pass and high pass filter
*
* <refsect2>
* <para>
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
* </para>
* <para>
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
* </para>
* <para>
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
* </para>
* <para>
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include <math.h>
#include "audiochebyshevfreqlimit.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
"Filter/Effect/Audio",
"Chebyshev low pass and high pass filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MODE,
PROP_TYPE,
PROP_CUTOFF,
PROP_RIPPLE,
PROP_POLES
};
#define ALLOWED_CAPS \
"audio/x-raw-float," \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER," \
" rate = (int) [ 1, MAX ]," \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
DEBUG_INIT);
static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqLimit * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqLimit * filter,
gfloat * data, guint num_samples);
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
static GType
gst_audio_chebyshev_freq_limit_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_LOW_PASS, "Low pass (default)",
"low-pass"},
{MODE_HIGH_PASS, "High pass",
"high-pass"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &element_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
}
static void
gst_audio_chebyshev_freq_limit_dispose (GObject * object)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode",
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_CUTOFF,
g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next even number",
2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
}
static void
gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitClass * klass)
{
filter->cutoff = 0.0;
filter->mode = MODE_LOW_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
filter->have_coeffs = FALSE;
filter->num_a = 0;
filter->num_b = 0;
filter->channels = NULL;
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2,
gdouble * b1, gdouble * b2)
{
gint np = filter->poles;
gdouble ripple = filter->ripple;
/* pole location in s-plane */
gdouble rp, ip;
/* zero location in s-plane */
gdouble rz = 0.0, iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
gint type = filter->type;
/* Calculate pole location for lowpass at frequency 1 */
{
gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
}
/* If we allow ripple, move the pole from the unit
* circle to an ellipse and keep cutoff at frequency 1 */
if (ripple > 0 && type == 1) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (1.0 / es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
} else if (type == 2) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
}
/* Calculate inverse of the pole location to convert from
* type I to type II */
if (type == 2) {
gdouble mag2 = rp * rp + ip * ip;
rp /= mag2;
ip /= mag2;
}
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
gdouble mag2;
rz = 0.0;
iz = cos (angle);
mag2 = rz * rz + iz * iz;
rz /= mag2;
iz /= mag2;
}
/* Convert from s-domain to z-domain by
* using the bilinear Z-transform, i.e.
* substitute s by (2/t)*((z-1)/(z+1))
* with t = 2 * tan(0.5).
*/
if (type == 1) {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t) / d;
x1 = 2.0 * x0;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
} else {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t * iz * iz + 4.0) / d;
x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
}
/* Convert from lowpass at frequency 1 to either lowpass
* or highpass.
*
* For lowpass substitute z^(-1) with:
* -1
* z - k
* ------------
* -1
* 1 - k * z
*
* k = sin((1-w)/2) / sin((1+w)/2)
*
* For highpass substitute z^(-1) with:
*
* -1
* -z - k
* ------------
* -1
* 1 + k * z
*
* k = -cos((1+w)/2) / cos((1-w)/2)
*
*/
{
gdouble k, d;
gdouble omega =
2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
if (filter->mode == MODE_LOW_PASS)
k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
else
k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
d = 1.0 + y1 * k - y2 * k * k;
*a0 = (x0 + k * (-x1 + k * x2)) / d;
*a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
*a2 = (x0 * k * k - x1 * k + x2) / d;
*b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
*b2 = (-k * k - y1 * k + y2) / d;
if (filter->mode == MODE_HIGH_PASS) {
*a1 = -*a1;
*b1 = -*b1;
}
}
}
/* Evaluate the transfer function that corresponds to the IIR
* coefficients at zr + zi*I and return the magnitude */
static gdouble
calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
gdouble zi)
{
gdouble sum_ar, sum_ai;
gdouble sum_br, sum_bi;
gdouble gain_r, gain_i;
gdouble sum_r_old;
gdouble sum_i_old;
gint i;
sum_ar = 0.0;
sum_ai = 0.0;
for (i = num_a; i >= 0; i--) {
sum_r_old = sum_ar;
sum_i_old = sum_ai;
sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
}
sum_br = 0.0;
sum_bi = 0.0;
for (i = num_b; i >= 0; i--) {
sum_r_old = sum_br;
sum_i_old = sum_bi;
sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
}
sum_br += 1.0;
sum_bi += 0.0;
gain_r =
(sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
gain_i =
(sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
return (sqrt (gain_r * gain_r + gain_i * gain_i));
}
static void
generate_coefficients (GstAudioChebyshevFreqLimit * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
filter->have_coeffs = TRUE;
if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return;
} else if (filter->cutoff <= 0.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return;
}
/* Calculate coefficients for the chebyshev filter */
{
gint np = filter->poles;
gdouble *a, *b;
gint i, p;
filter->num_a = np + 1;
filter->a = a = g_new0 (gdouble, np + 3);
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 3);
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
}
/* Calculate transfer function coefficients */
a[2] = 1.0;
b[2] = 1.0;
for (p = 1; p <= np / 2; p++) {
gdouble a0, a1, a2, b1, b2;
gdouble *ta = g_new0 (gdouble, np + 3);
gdouble *tb = g_new0 (gdouble, np + 3);
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
memcpy (ta, a, sizeof (gdouble) * (np + 3));
memcpy (tb, b, sizeof (gdouble) * (np + 3));
/* add the new coefficients for the new two poles
* to the cascade by multiplication of the transfer
* functions */
for (i = 2; i < np + 3; i++) {
a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
}
g_free (ta);
g_free (tb);
}
/* Move coefficients to the beginning of the array
* and multiply the b coefficients with -1 to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
b[2] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 2];
b[i] = -b[i + 2];
}
/* Normalize to unity gain at frequency 0 for lowpass
* and frequency 0.5 for highpass */
{
gdouble gain;
if (filter->mode == MODE_LOW_PASS)
gain = calculate_gain (a, b, np, np, 1.0, 0.0);
else
gain = calculate_gain (a, b, np, np, -1.0, 0.0);
for (i = 0; i <= np; i++) {
a[i] /= gain;
}
}
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
"mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB",
(filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
filter->type, filter->poles, filter->cutoff, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
{
gdouble wc =
2.0 * M_PI * (filter->cutoff /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble zr = cos (wc), zi = sin (wc);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->cutoff);
}
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
GST_AUDIO_FILTER (filter)->format.rate / 2);
}
}
static void
gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->mode = g_value_get_enum (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_TYPE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->type = g_value_get_int (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_CUTOFF:
GST_BASE_TRANSFORM_LOCK (filter);
filter->cutoff = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_RIPPLE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->ripple = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_POLES:
GST_BASE_TRANSFORM_LOCK (filter);
filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_TYPE:
g_value_set_int (value, filter->type);
break;
case PROP_CUTOFF:
g_value_set_float (value, filter->cutoff);
break;
case PROP_RIPPLE:
g_value_set_float (value, filter->ripple);
break;
case PROP_POLES:
g_value_set_int (value, filter->poles);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_64;
else
ret = FALSE;
filter->have_coeffs = FALSE;
return ret;
}
static inline gdouble
process (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
val += filter->a[i] * ctx->x[j];
j--;
if (j < 0)
j = filter->num_a - 1;
}
for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
val += filter->b[i] * ctx->y[j];
j--;
if (j < 0)
j = filter->num_b - 1;
}
if (ctx->x) {
ctx->x_pos++;
if (ctx->x_pos > filter->num_a - 1)
ctx->x_pos = 0;
ctx->x[ctx->x_pos] = x0;
}
if (ctx->y) {
ctx->y_pos++;
if (ctx->y_pos > filter->num_b - 1)
ctx->y_pos = 0;
ctx->y[ctx->y_pos] = val;
}
return val;
}
static void
process_64 (GstAudioChebyshevFreqLimit * filter,
gdouble * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
static void
process_32 (GstAudioChebyshevFreqLimit * filter,
gfloat * data, guint num_samples)
{
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
gdouble val;
for (i = 0; i < num_samples / channels; i++) {
for (j = 0; j < channels; j++) {
val = process (filter, &filter->channels[j], *data);
*data++ = val;
}
}
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (!gst_buffer_is_writable (buf))
return GST_FLOW_OK;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (!filter->have_coeffs)
generate_coefficients (filter);
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static gboolean
gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if
* already existing */
if (channels && filter->channels) {
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
if (ctx->x)
memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
if (ctx->y)
memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
}
}
return TRUE;
}

78
gst/audiofx/audiochebyshevfreqlimit.h Normal file
View file

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

8
gst/audiofx/audiofx.c
View file

@ -29,6 +29,8 @@
#include "audioinvert.h"
#include "audioamplify.h"
#include "audiodynamic.h"
#include "audiochebyshevfreqlimit.h"
#include "audiochebyshevfreqband.h"
/* entry point to initialize the plug-in
* initialize the plug-in itself
@ -48,7 +50,11 @@ plugin_init (GstPlugin * plugin)
gst_element_register (plugin, "audioamplify", GST_RANK_NONE,
GST_TYPE_AUDIO_AMPLIFY) &&
gst_element_register (plugin, "audiodynamic", GST_RANK_NONE,
GST_TYPE_AUDIO_DYNAMIC));
GST_TYPE_AUDIO_DYNAMIC) &&
gst_element_register (plugin, "audiochebyshevfreqlimit", GST_RANK_NONE,
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT) &&
gst_element_register (plugin, "audiochebyshevfreqband", GST_RANK_NONE,
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND));
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,

2
tests/check/Makefile.am
View file

@ -55,6 +55,8 @@ check_PROGRAMS = \
elements/alphacolor \
elements/audiopanorama \
elements/audioinvert \
elements/audiochebyshevfreqband \
elements/audiochebyshevfreqlimit \
elements/audioamplify \
elements/audiodynamic \
elements/avimux \

2
tests/check/elements/.gitignore vendored
View file

@ -3,6 +3,8 @@ alphacolor
audioamplify
audiodynamic
audioinvert
audiochebyshevfreqband
audiochebyshevfreqlimit
level
matroskamux
cmmldec

471
tests/check/elements/audiochebband.c Normal file
View file

@ -0,0 +1,471 @@
/* GStreamer
*
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* audiochebyshevfreqband.c: Unit test for the audiochebyshevfreqband element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/check/gstcheck.h>
#include <math.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
GstPad *mysrcpad, *mysinkpad;
#define CAPS_STRING \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64" \
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
GstElement *
setup_audiochebyshevfreqband ()
{
GstElement *audiochebyshevfreqband;
GST_DEBUG ("setup_audiochebyshevfreqband");
audiochebyshevfreqband = gst_check_setup_element ("audiochebyshevfreqband");
mysrcpad =
gst_check_setup_src_pad (audiochebyshevfreqband, &srctemplate, NULL);
mysinkpad =
gst_check_setup_sink_pad (audiochebyshevfreqband, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return audiochebyshevfreqband;
}
void
cleanup_audiochebyshevfreqband (GstElement * audiochebyshevfreqband)
{
GST_DEBUG ("cleanup_audiochebyshevfreqband");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audiochebyshevfreqband);
gst_check_teardown_sink_pad (audiochebyshevfreqband);
gst_check_teardown_element (audiochebyshevfreqband);
}
/* Test if data containing only one frequency component
* at 0 is erased with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_bp_0hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandpass */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at band center is preserved with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_bp_11025hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandpass */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms >= 0.6);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_bp_22050hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandpass */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_br_0hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandreject */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at band center is erased with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_br_11025hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandreject */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_br_22050hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandreject */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
Suite *
audiochebyshevfreqband_suite (void)
{
Suite *s = suite_create ("audiochebyshevfreqband");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_bp_0hz);
tcase_add_test (tc_chain, test_bp_11025hz);
tcase_add_test (tc_chain, test_bp_22050hz);
tcase_add_test (tc_chain, test_br_0hz);
tcase_add_test (tc_chain, test_br_11025hz);
tcase_add_test (tc_chain, test_br_22050hz);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audiochebyshevfreqband_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

341
tests/check/elements/audiocheblimit.c Normal file
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@ -0,0 +1,341 @@
/* GStreamer
*
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* audiochebyshevfreqlimit.c: Unit test for the audiochebyshevfreqlimit element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/check/gstcheck.h>
#include <math.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
GstPad *mysrcpad, *mysinkpad;
#define CAPS_STRING \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64" \
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
GstElement *
setup_audiochebyshevfreqlimit ()
{
GstElement *audiochebyshevfreqlimit;
GST_DEBUG ("setup_audiochebyshevfreqlimit");
audiochebyshevfreqlimit = gst_check_setup_element ("audiochebyshevfreqlimit");
mysrcpad =
gst_check_setup_src_pad (audiochebyshevfreqlimit, &srctemplate, NULL);
mysinkpad =
gst_check_setup_sink_pad (audiochebyshevfreqlimit, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return audiochebyshevfreqlimit;
}
void
cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit)
{
GST_DEBUG ("cleanup_audiochebyshevfreqlimit");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audiochebyshevfreqlimit);
gst_check_teardown_sink_pad (audiochebyshevfreqlimit);
gst_check_teardown_element (audiochebyshevfreqlimit);
}
/* Test if data containing only one frequency component
* at 0 is preserved with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_lp_0hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_lp_22050hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is erased with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_hp_0hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_hp_22050hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
Suite *
audiochebyshevfreqlimit_suite (void)
{
Suite *s = suite_create ("audiochebyshevfreqlimit");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_lp_0hz);
tcase_add_test (tc_chain, test_lp_22050hz);
tcase_add_test (tc_chain, test_hp_0hz);
tcase_add_test (tc_chain, test_hp_22050hz);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audiochebyshevfreqlimit_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

471
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/* GStreamer
*
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* audiochebyshevfreqband.c: Unit test for the audiochebyshevfreqband element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/check/gstcheck.h>
#include <math.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
GstPad *mysrcpad, *mysinkpad;
#define CAPS_STRING \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64" \
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
GstElement *
setup_audiochebyshevfreqband ()
{
GstElement *audiochebyshevfreqband;
GST_DEBUG ("setup_audiochebyshevfreqband");
audiochebyshevfreqband = gst_check_setup_element ("audiochebyshevfreqband");
mysrcpad =
gst_check_setup_src_pad (audiochebyshevfreqband, &srctemplate, NULL);
mysinkpad =
gst_check_setup_sink_pad (audiochebyshevfreqband, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return audiochebyshevfreqband;
}
void
cleanup_audiochebyshevfreqband (GstElement * audiochebyshevfreqband)
{
GST_DEBUG ("cleanup_audiochebyshevfreqband");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audiochebyshevfreqband);
gst_check_teardown_sink_pad (audiochebyshevfreqband);
gst_check_teardown_element (audiochebyshevfreqband);
}
/* Test if data containing only one frequency component
* at 0 is erased with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_bp_0hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandpass */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at band center is preserved with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_bp_11025hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandpass */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms >= 0.6);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_bp_22050hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandpass */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_br_0hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandreject */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at band center is erased with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_br_11025hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandreject */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_br_22050hz)
{
GstElement *audiochebyshevfreqband;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqband = setup_audiochebyshevfreqband ();
/* Set to bandreject */
g_object_set (G_OBJECT (audiochebyshevfreqband), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqband,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqband), "lower-frequency",
44100 / 4.0 - 1000, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqband), "upper-frequency",
44100 / 4.0 + 1000, NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 1024; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 1024.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqband (audiochebyshevfreqband);
}
GST_END_TEST;
Suite *
audiochebyshevfreqband_suite (void)
{
Suite *s = suite_create ("audiochebyshevfreqband");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_bp_0hz);
tcase_add_test (tc_chain, test_bp_11025hz);
tcase_add_test (tc_chain, test_bp_22050hz);
tcase_add_test (tc_chain, test_br_0hz);
tcase_add_test (tc_chain, test_br_11025hz);
tcase_add_test (tc_chain, test_br_22050hz);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audiochebyshevfreqband_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

341
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@ -0,0 +1,341 @@
/* GStreamer
*
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* audiochebyshevfreqlimit.c: Unit test for the audiochebyshevfreqlimit element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/check/gstcheck.h>
#include <math.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
GstPad *mysrcpad, *mysinkpad;
#define CAPS_STRING \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64" \
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) 1, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) 64")
);
GstElement *
setup_audiochebyshevfreqlimit ()
{
GstElement *audiochebyshevfreqlimit;
GST_DEBUG ("setup_audiochebyshevfreqlimit");
audiochebyshevfreqlimit = gst_check_setup_element ("audiochebyshevfreqlimit");
mysrcpad =
gst_check_setup_src_pad (audiochebyshevfreqlimit, &srctemplate, NULL);
mysinkpad =
gst_check_setup_sink_pad (audiochebyshevfreqlimit, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return audiochebyshevfreqlimit;
}
void
cleanup_audiochebyshevfreqlimit (GstElement * audiochebyshevfreqlimit)
{
GST_DEBUG ("cleanup_audiochebyshevfreqlimit");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (audiochebyshevfreqlimit);
gst_check_teardown_sink_pad (audiochebyshevfreqlimit);
gst_check_teardown_element (audiochebyshevfreqlimit);
}
/* Test if data containing only one frequency component
* at 0 is preserved with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_lp_0hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_lp_22050hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to lowpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 0, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is erased with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_hp_0hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms <= 0.1);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_hp_22050hz)
{
GstElement *audiochebyshevfreqlimit;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
audiochebyshevfreqlimit = setup_audiochebyshevfreqlimit ();
/* Set to highpass */
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "mode", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "poles", 8, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "type", 1, NULL);
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "ripple", 0.25, NULL);
fail_unless (gst_element_set_state (audiochebyshevfreqlimit,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (audiochebyshevfreqlimit), "cutoff", 44100 / 4.0,
NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
rms = 0.0;
for (i = 0; i < 128; i++)
rms += res[i] * res[i];
rms = sqrt (rms / 128.0);
fail_unless (rms >= 0.9);
/* cleanup */
cleanup_audiochebyshevfreqlimit (audiochebyshevfreqlimit);
}
GST_END_TEST;
Suite *
audiochebyshevfreqlimit_suite (void)
{
Suite *s = suite_create ("audiochebyshevfreqlimit");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_lp_0hz);
tcase_add_test (tc_chain, test_lp_22050hz);
tcase_add_test (tc_chain, test_hp_0hz);
tcase_add_test (tc_chain, test_hp_22050hz);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audiochebyshevfreqlimit_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}