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Add audioiirfilter and audiofirfilter elements which allow generic IIR/FIR filters to be implemented by providing the...

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Original commit message from CVS:
* configure.ac:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofirfilter.c: (gst_audio_fir_filter_base_init),
(gst_audio_fir_filter_class_init),
(gst_audio_fir_filter_update_kernel), (gst_audio_fir_filter_init),
(gst_audio_fir_filter_setup), (gst_audio_fir_filter_finalize),
(gst_audio_fir_filter_set_property),
(gst_audio_fir_filter_get_property):
* gst/audiofx/audiofirfilter.h:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioiirfilter.c: (gst_audio_iir_filter_base_init),
(gst_audio_iir_filter_class_init),
(gst_audio_iir_filter_update_coefficients),
(gst_audio_iir_filter_init), (gst_audio_iir_filter_setup),
(gst_audio_iir_filter_finalize),
(gst_audio_iir_filter_set_property),
(gst_audio_iir_filter_get_property):
* gst/audiofx/audioiirfilter.h:
Add audioiirfilter and audiofirfilter elements which allow
generic IIR/FIR filters to be implemented by providing the
filter coefficients. Fixes bug #567577.
* 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/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.signals:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
Add documentation for the audioiirfilter and audiofirfilter
elements.
* tests/check/Makefile.am:
* tests/check/elements/audiofirfilter.c: (on_message),
(on_rate_changed), (on_handoff), (GST_START_TEST),
(audiofirfilter_suite):
* tests/check/elements/audioiirfilter.c: (on_message),
(on_rate_changed), (on_handoff), (GST_START_TEST),
(audioiirfilter_suite):
* tests/examples/Makefile.am:
* tests/examples/audiofx/Makefile.am:
* tests/examples/audiofx/firfilter-example.c: (on_message),
(on_rate_changed), (main):
* tests/examples/audiofx/iirfilter-example.c: (on_message),
(on_rate_changed), (main):
Add unit tests and example applications for the two filter
elements.
This commit is contained in:
Sebastian Dröge 2009-01-13 19:23:57 +00:00
parent 5f5ae768b8
commit 75c1c9f378
28 changed files with 1594 additions and 29 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

56
ChangeLog
View file

@ -1,3 +1,59 @@
2009-01-13 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofirfilter.c: (gst_audio_fir_filter_base_init),
(gst_audio_fir_filter_class_init),
(gst_audio_fir_filter_update_kernel), (gst_audio_fir_filter_init),
(gst_audio_fir_filter_setup), (gst_audio_fir_filter_finalize),
(gst_audio_fir_filter_set_property),
(gst_audio_fir_filter_get_property):
* gst/audiofx/audiofirfilter.h:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioiirfilter.c: (gst_audio_iir_filter_base_init),
(gst_audio_iir_filter_class_init),
(gst_audio_iir_filter_update_coefficients),
(gst_audio_iir_filter_init), (gst_audio_iir_filter_setup),
(gst_audio_iir_filter_finalize),
(gst_audio_iir_filter_set_property),
(gst_audio_iir_filter_get_property):
* gst/audiofx/audioiirfilter.h:
Add audioiirfilter and audiofirfilter elements which allow
generic IIR/FIR filters to be implemented by providing the
filter coefficients. Fixes bug #567577.
* 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/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.signals:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
Add documentation for the audioiirfilter and audiofirfilter
elements.
* tests/check/Makefile.am:
* tests/check/elements/audiofirfilter.c: (on_message),
(on_rate_changed), (on_handoff), (GST_START_TEST),
(audiofirfilter_suite):
* tests/check/elements/audioiirfilter.c: (on_message),
(on_rate_changed), (on_handoff), (GST_START_TEST),
(audioiirfilter_suite):
* tests/examples/Makefile.am:
* tests/examples/audiofx/Makefile.am:
* tests/examples/audiofx/firfilter-example.c: (on_message),
(on_rate_changed), (main):
* tests/examples/audiofx/iirfilter-example.c: (on_message),
(on_rate_changed), (main):
Add unit tests and example applications for the two filter
elements.
2009-01-13 Michael Smith <msmith@songbirdnest.com>
Patch by: Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br>

1
configure.ac
View file

@ -1105,6 +1105,7 @@ po/Makefile.in
tests/Makefile
tests/check/Makefile
tests/examples/Makefile
tests/examples/audiofx/Makefile
tests/examples/equalizer/Makefile
tests/examples/level/Makefile
tests/examples/rtp/Makefile

6
docs/plugins/Makefile.am
View file

@ -112,8 +112,10 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/audiofx/audiopanorama.h \
$(top_srcdir)/gst/audiofx/audiocheblimit.h \
$(top_srcdir)/gst/audiofx/audiochebband.h \
$(top_srcdir)/gst/audiofx/audioiirfilter.h \
$(top_srcdir)/gst/audiofx/audiowsincband.h \
$(top_srcdir)/gst/audiofx/audiowsinclimit.h \
$(top_srcdir)/gst/audiofx/audiofirfilter.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosrc.h \
$(top_srcdir)/gst/autodetect/gstautovideosink.h \
@ -177,7 +179,9 @@ EXTRA_HFILES = \
# example code that needs to be converted to xml and placed in xml/
EXAMPLE_CFILES = \
$(top_srcdir)/tests/examples/level/level-example.c \
$(top_srcdir)/tests/examples/spectrum/spectrum-example.c
$(top_srcdir)/tests/examples/spectrum/spectrum-example.c \
$(top_srcdir)/tests/examples/audiofx/firfilter-example.c \
$(top_srcdir)/tests/examples/audiofx/iirfilter-example.c
# Images to copy into HTML directory.
HTML_IMAGES =

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

@ -19,8 +19,10 @@
<xi:include href="xml/element-audioamplify.xml" />
<xi:include href="xml/element-audiochebband.xml" />
<xi:include href="xml/element-audiocheblimit.xml" />
<xi:include href="xml/element-audioiirfilter.xml" />
<xi:include href="xml/element-audiowsincband.xml" />
<xi:include href="xml/element-audiowsinclimit.xml" />
<xi:include href="xml/element-audiofirfilter.xml" />
<xi:include href="xml/element-audiodynamic.xml" />
<xi:include href="xml/element-audioinvert.xml" />
<xi:include href="xml/element-audiopanorama.xml" />

32
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -102,6 +102,20 @@ GST_TYPE_AUDIO_CHEB_LIMIT
gst_audio_cheb_limit_get_type
</SECTION>
<SECTION>
<FILE>element-audioiirfilter</FILE>
<TITLE>audioiirfilter</TITLE>
GstAudioIIRFilter
<SUBSECTION Standard>
GstAudioIIRFilterClass
GST_AUDIO_IIR_FILTER
GST_AUDIO_IIR_FILTER_CLASS
GST_IS_AUDIO_IIR_FILTER_BAND
GST_IS_AUDIO_IIR_FILTER_CLASS
GST_TYPE_AUDIO_IIR_FILTER
gst_audio_iir_filter_get_type
</SECTION>
<SECTION>
<FILE>element-audiodynamic</FILE>
<TITLE>audiodynamic</TITLE>
@ -178,7 +192,7 @@ GST_AUDIO_WSINC_BAND_CLASS
GST_IS_AUDIO_WSINC_BAND
GST_IS_AUDIO_WSINC_BAND_CLASS
GST_TYPE_AUDIO_WSINC_BAND
audio_wsincband_get_type
gst_audio_wsincband_get_type
</SECTION>
<SECTION>
@ -193,7 +207,21 @@ GST_AUDIO_WSINC_LIMIT_CLASS
GST_IS_AUDIO_WSINC_LIMIT
GST_IS_AUDIO_WSINC_LIMIT_CLASS
GST_TYPE_AUDIO_WSINC_LIMIT
audio_wsinclimit_get_type
gst_audio_wsinclimit_get_type
</SECTION>
<SECTION>
<FILE>element-audiofirfilter</FILE>
<TITLE>audiofirfilter</TITLE>
GstAudioFIRFilter
<SUBSECTION Standard>
GstAudioFIRFilterClass
GST_AUDIO_FIR_FILTER
GST_AUDIO_FIR_FILTER_CLASS
GST_IS_AUDIO_FIR_FILTER_BAND
GST_IS_AUDIO_FIR_FILTER_CLASS
GST_TYPE_AUDIO_FIR_FILTER
gst_audio_fir_filter_get_type
</SECTION>
<SECTION>

40
docs/plugins/gst-plugins-good-plugins.args
View file

@ -19668,3 +19668,43 @@
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioFIRFilter::kernel</NAME>
<TYPE>GValueArray*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Filter Kernel</NICK>
<BLURB>Filter kernel for the FIR filter.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioFIRFilter::latency</NAME>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Latecy</NICK>
<BLURB>Filter latency in samples.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioIIRFilter::a</NAME>
<TYPE>GValueArray*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>A</NICK>
<BLURB>Filter coefficients (numerator of transfer function).</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioIIRFilter::b</NAME>
<TYPE>GValueArray*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>B</NICK>
<BLURB>Filter coefficients (denominator of transfer function).</BLURB>
<DEFAULT></DEFAULT>
</ARG>

4
docs/plugins/gst-plugins-good-plugins.hierarchy
View file

@ -56,10 +56,14 @@ GObject
GstAudioKaraoke
GstAudioAmplify
GstAudioDynamic
GstAudioFXBaseIIRFilter
GstAudioChebLimit
GstAudioChebBand
GstAudioIIRFilter
GstAudioFXBaseFIRFilter
GstAudioWSincLimit
GstAudioWSincBand
GstAudioFIRFilter
GstIirEqualizer
GstIirEqualizerNBands
GstIirEqualizer3Bands

16
docs/plugins/gst-plugins-good-plugins.signals
View file

@ -332,3 +332,19 @@ guint arg1
guint arg2
</SIGNAL>
<SIGNAL>
<NAME>GstAudioFIRFilter::rate-changed</NAME>
<RETURNS>void</RETURNS>
<FLAGS>l</FLAGS>
GstAudioFIRFilter *gstaudiofirfilter
gint arg1
</SIGNAL>
<SIGNAL>
<NAME>GstAudioIIRFilter::rate-changed</NAME>
<RETURNS>void</RETURNS>
<FLAGS>l</FLAGS>
GstAudioIIRFilter *gstaudioiirfilter
gint arg1
</SIGNAL>

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

@ -26,7 +26,7 @@
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ]</details>
<details>audio/x-raw-int, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)16, signed=(boolean)true</details>
</caps>
</pads>
</element>
@ -41,7 +41,7 @@
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ]</details>
<details>audio/x-raw-int, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)16, signed=(boolean)true</details>
</caps>
<caps>
<name>src</name>

50
docs/plugins/inspect/plugin-audiofx.xml
View file

@ -35,7 +35,7 @@
<longname>Band pass &amp; band reject filter</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>
<author>Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>
<caps>
<name>sink</name>
@ -56,7 +56,7 @@
<longname>Low pass &amp; high pass filter</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>
<author>Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>
<caps>
<name>sink</name>
@ -93,6 +93,48 @@
</caps>
</pads>
</element>
<element>
<name>audiofirfilter</name>
<longname>Audio FIR filter</longname>
<class>Filter/Effect/Audio</class>
<description>Generic audio FIR filter with custom filter kernel</description>
<author>Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>
<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>
<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>
</pads>
</element>
<element>
<name>audioiirfilter</name>
<longname>Audio IIR filter</longname>
<class>Filter/Effect/Audio</class>
<description>Generic audio IIR filter with custom filter kernel</description>
<author>Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>
<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>
<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>
</pads>
</element>
<element>
<name>audioinvert</name>
<longname>Audio inversion</longname>
@ -161,7 +203,7 @@
<longname>Band pass &amp; band reject filter</longname>
<class>Filter/Effect/Audio</class>
<description>Band pass and band reject windowed sinc filter</description>
<author>Thomas Vander Stichele &lt;thomas at apestaart dot org&gt;, Steven W. Smith, Dreamlab Technologies Ltd. &lt;mathis.hofer@dreamlab.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<author>Thomas Vander Stichele &lt;thomas at apestaart dot org&gt;, Steven W. Smith, Dreamlab Technologies Ltd. &lt;mathis.hofer@dreamlab.net&gt;, Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>
<caps>
<name>sink</name>
@ -182,7 +224,7 @@
<longname>Low pass &amp; high pass filter</longname>
<class>Filter/Effect/Audio</class>
<description>Low pass and high pass windowed sinc filter</description>
<author>Thomas Vander Stichele &lt;thomas at apestaart dot org&gt;, Steven W. Smith, Dreamlab Technologies Ltd. &lt;mathis.hofer@dreamlab.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
<author>Thomas Vander Stichele &lt;thomas at apestaart dot org&gt;, Steven W. Smith, Dreamlab Technologies Ltd. &lt;mathis.hofer@dreamlab.net&gt;, Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>
<caps>
<name>sink</name>

4
docs/plugins/inspect/plugin-avi.xml
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File diff suppressed because one or more lines are too long

12
docs/plugins/inspect/plugin-flac.xml
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@ -16,18 +16,18 @@
<description>Decodes FLAC lossless audio streams</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int){ 8, 16, 32 }, depth=(int)[ 4, 32 ], rate=(int)[ 1, 655350 ], channels=(int)[ 1, 8 ]</details>
</caps>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-flac</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int){ 8, 16, 32 }, depth=(int)[ 4, 32 ], rate=(int)[ 1, 655350 ], channels=(int)[ 1, 8 ]</details>
</caps>
</pads>
</element>
<element>

4
docs/plugins/inspect/plugin-mulaw.xml
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@ -26,7 +26,7 @@
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ]</details>
<details>audio/x-raw-int, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)16, signed=(boolean)true</details>
</caps>
</pads>
</element>
@ -41,7 +41,7 @@
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ]</details>
<details>audio/x-raw-int, rate=(int)[ 8000, 192000 ], channels=(int)[ 1, 2 ], endianness=(int)1234, width=(int)16, signed=(boolean)true</details>
</caps>
<caps>
<name>src</name>

2
docs/plugins/inspect/plugin-video4linux2.xml
View file

@ -20,7 +20,7 @@
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>video/x-raw-rgb, bpp=(int)8, depth=(int)8, red_mask=(int)224, green_mask=(int)28, blue_mask=(int)3, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YVU9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)UYVY, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y42B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41P, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV21, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUV9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]</details>
<details>video/x-raw-rgb, bpp=(int)8, depth=(int)8, red_mask=(int)224, green_mask=(int)28, blue_mask=(int)3, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)1234, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)15, red_mask=(int)31744, green_mask=(int)992, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, red_mask=(int)16711680, green_mask=(int)65280, blue_mask=(int)255, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-rgb, bpp=(int)32, depth=(int)32, red_mask=(int)-16777216, green_mask=(int)16711680, blue_mask=(int)65280, endianness=(int)4321, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-gray, bpp=(int)8, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YVU9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUY2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)UYVY, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y42B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41B, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)Y41P, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV12, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)NV21, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)YUV9, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-yuv, format=(fourcc)I420, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-raw-bayer, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; image/jpeg, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-dv, systemstream=(boolean)true, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-sonix, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc1, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]; video/x-pwc2, width=(int)[ 1, 32768 ], height=(int)[ 1, 32768 ], framerate=(fraction)[ 0/1, 100/1 ]</details>
</caps>
</pads>
</element>

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

@ -20,7 +20,7 @@
<name>wavparse_src</name>
<direction>source</direction>
<presence>sometimes</presence>
<details>audio/ms-gsm, rate=(int)[ 1, 96000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)2, rate=(int)[ 16000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ]; audio/x-vorbis, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-ac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ]; audio/x-dts; audio/mpeg, mpegversion=(int)4, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ]; audio/x-alaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-mulaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-wms, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)microsoft, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)dvi, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-truespeech, rate=(int)8000, channels=(int)[ 1, 2 ]; audio/x-wma, wmaversion=(int)1, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ], block_align=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)2, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ], block_align=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)3, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ], block_align=(int)[ 1, 2147483647 ]; audio/x-vnd.sony.atrac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-raw-float, endianness=(int)1234, width=(int){ 32, 64 }, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ]; audio/x-voxware, voxwaretype=(int)117, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-adpcm, layout=(string)dk4, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-adpcm, layout=(string)dk3, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]; application/x-ogg-avi</details>
<details>audio/ms-gsm, rate=(int)[ 1, 96000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)2, rate=(int)[ 16000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], rate=(int)[ 1000, 192000 ], channels=(int)[ 1, 8 ]; audio/x-vorbis, rate=(int)[ 1000, 192000 ], channels=(int)[ 1, 2 ]; audio/x-ac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ]; audio/x-dts; audio/mpeg, mpegversion=(int)4, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 8 ]; audio/x-alaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-mulaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-wms, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)microsoft, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)dvi, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ], block_align=(int)[ 1, 2147483647 ]; audio/x-truespeech, rate=(int)8000, channels=(int)[ 1, 2 ]; audio/x-wma, wmaversion=(int)1, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ], block_align=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)2, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ], block_align=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)3, bitrate=(int)[ 0, 2147483647 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 6 ], block_align=(int)[ 1, 2147483647 ]; audio/x-vnd.sony.atrac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-raw-float, endianness=(int)1234, width=(int){ 32, 64 }, rate=(int)[ 1000, 192000 ], channels=(int)[ 1, 8 ]; audio/x-voxware, voxwaretype=(int)117, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-adpcm, layout=(string)dk4, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-adpcm, layout=(string)dk3, rate=(int)[ 8000, 96000 ], channels=(int)[ 1, 2 ]; application/x-ogg-avi</details>
</caps>
<caps>
<name>wavparse_sink</name>

6
gst/audiofx/Makefile.am
View file

@ -12,9 +12,11 @@ libgstaudiofx_la_SOURCES = audiofx.c\
audiofxbaseiirfilter.c \
audiocheblimit.c \
audiochebband.c \
audioiirfilter.c \
audiofxbasefirfilter.c \
audiowsincband.c \
audiowsinclimit.c
audiowsinclimit.c \
audiofirfilter.c
# flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
@ -39,8 +41,10 @@ noinst_HEADERS = audiopanorama.h \
audiofxbaseiirfilter.h \
audiocheblimit.h \
audiochebband.h \
audioiirfilter.h \
audiofxbasefirfilter.h \
audiowsincband.h \
audiowsinclimit.h \
audiofirfilter.h \
math_compat.h

273
gst/audiofx/audiofirfilter.c Normal file
View file

@ -0,0 +1,273 @@
/*
* GStreamer
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*/
/**
* SECTION:element-audiofirfilter
* @short_description: Generic audio FIR filter
*
* <refsect2>
* <para>
* audiofirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Finite_impulse_response">FIR filter</ulink>. Before usage the
* "kernel" property has to be set to the filter kernel that should be
* used and the "latency" property has to be set to the latency (in samples)
* that is introduced by the filter kernel. Setting a latency of n samples
* will lead to the first n samples being dropped from the output and
* n samples added to the end.
* </para>
* <para>
* The filter kernel describes the impulse response of the filter. To
* calculate the frequency response of the filter you have to calculate
* the Fourier Transform of the impulse response.
* </para>
* <para>
* To change the filter kernel whenever the sampling rate changes the
* "rate-changed" signal can be used. This should be done for most
* FIR filters as they're depending on the sampling rate.
* </para>
* <title>Example application</title>
* <para>
* <include xmlns="http://www.w3.org/2003/XInclude" href="element-firfilter-example.xml" />
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include "audiofirfilter.h"
#define GST_CAT_DEFAULT gst_audio_fir_filter_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
SIGNAL_RATE_CHANGED,
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_KERNEL,
PROP_LATENCY
};
static guint gst_audio_fir_filter_signals[LAST_SIGNAL] = { 0, };
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_fir_filter_debug, "audiofirfilter", 0, \
"Generic audio FIR filter plugin");
GST_BOILERPLATE_FULL (GstAudioFIRFilter, gst_audio_fir_filter, GstAudioFilter,
GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
static void gst_audio_fir_filter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_fir_filter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_audio_fir_filter_finalize (GObject * object);
static gboolean gst_audio_fir_filter_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
/* Element class */
static void
gst_audio_fir_filter_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class,
"Audio FIR filter", "Filter/Effect/Audio",
"Generic audio FIR filter with custom filter kernel",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
gst_audio_fir_filter_class_init (GstAudioFIRFilterClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_fir_filter_set_property;
gobject_class->get_property = gst_audio_fir_filter_get_property;
gobject_class->finalize = gst_audio_fir_filter_finalize;
g_object_class_install_property (gobject_class, PROP_KERNEL,
g_param_spec_value_array ("kernel", "Filter Kernel",
"Filter kernel for the FIR filter",
g_param_spec_double ("Element", "Filter Kernel Element",
"Element of the filter kernel", -G_MAXDOUBLE, G_MAXDOUBLE, 0.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LATENCY,
g_param_spec_uint64 ("latency", "Latecy",
"Filter latency in samples",
0, G_MAXUINT64, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fir_filter_setup);
/**
* GstAudioFIRFilter::rate-changed:
* @filter: the filter on which the signal is emitted
* @rate: the new sampling rate
*
* Will be emitted when the sampling rate changes. The callbacks
* will be called from the streaming thread and processing will
* stop until the event is handled.
*/
gst_audio_fir_filter_signals[SIGNAL_RATE_CHANGED] =
g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioFIRFilterClass, rate_changed),
NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
}
static void
gst_audio_fir_filter_update_kernel (GstAudioFIRFilter * self, GValueArray * va)
{
gdouble *kernel;
guint i;
gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
(self));
if (va) {
if (self->kernel)
g_value_array_free (self->kernel);
self->kernel = va;
}
kernel = g_new (gdouble, self->kernel->n_values);
for (i = 0; i < self->kernel->n_values; i++) {
GValue *v = g_value_array_get_nth (self->kernel, i);
kernel[i] = g_value_get_double (v);
}
gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
kernel, self->kernel->n_values, self->latency);
}
static void
gst_audio_fir_filter_init (GstAudioFIRFilter * self,
GstAudioFIRFilterClass * g_class)
{
GValue v = { 0, };
GValueArray *va;
self->latency = 0;
va = g_value_array_new (1);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_set_double (&v, 1.0);
g_value_array_append (va, &v);
g_value_unset (&v);
gst_audio_fir_filter_update_kernel (self, va);
self->lock = g_mutex_new ();
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fir_filter_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioFIRFilter *self = GST_AUDIO_FIR_FILTER (base);
if (self->rate != format->rate) {
g_signal_emit (G_OBJECT (self),
gst_audio_fir_filter_signals[SIGNAL_RATE_CHANGED], 0, format->rate);
self->rate = format->rate;
}
return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
}
static void
gst_audio_fir_filter_finalize (GObject * object)
{
GstAudioFIRFilter *self = GST_AUDIO_FIR_FILTER (object);
g_mutex_free (self->lock);
self->lock = NULL;
if (self->kernel)
g_value_array_free (self->kernel);
self->kernel = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_audio_fir_filter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioFIRFilter *self = GST_AUDIO_FIR_FILTER (object);
g_return_if_fail (GST_IS_AUDIO_FIR_FILTER (self));
switch (prop_id) {
case PROP_KERNEL:
g_mutex_lock (self->lock);
gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
(self));
gst_audio_fir_filter_update_kernel (self, g_value_dup_boxed (value));
g_mutex_unlock (self->lock);
break;
case PROP_LATENCY:
g_mutex_lock (self->lock);
self->latency = g_value_get_uint64 (value);
gst_audio_fir_filter_update_kernel (self, NULL);
g_mutex_unlock (self->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_fir_filter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioFIRFilter *self = GST_AUDIO_FIR_FILTER (object);
switch (prop_id) {
case PROP_KERNEL:
g_value_set_boxed (value, self->kernel);
break;
case PROP_LATENCY:
g_value_set_uint64 (value, self->latency);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}

72
gst/audiofx/audiofirfilter.h Normal file
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@ -0,0 +1,72 @@
/*
* GStreamer
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*/
#ifndef __GST_AUDIO_FIR_FILTER_H__
#define __GST_AUDIO_FIR_FILTER_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include "audiofxbasefirfilter.h"
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_FIR_FILTER \
(gst_audio_fir_filter_get_type())
#define GST_AUDIO_FIR_FILTER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_FIR_FILTER,GstAudioFIRFilter))
#define GST_AUDIO_FIR_FILTER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_FIR_FILTER,GstAudioFIRFilterClass))
#define GST_IS_AUDIO_FIR_FILTER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_FIR_FILTER))
#define GST_IS_AUDIO_FIR_FILTER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_FIR_FILTER))
typedef struct _GstAudioFIRFilter GstAudioFIRFilter;
typedef struct _GstAudioFIRFilterClass GstAudioFIRFilterClass;
/**
* GstAudioFIRFilter:
*
* Opaque data structure.
*/
struct _GstAudioFIRFilter {
GstAudioFXBaseFIRFilter parent;
GValueArray *kernel;
guint64 latency;
/* < private > */
GMutex *lock;
gint rate;
};
struct _GstAudioFIRFilterClass {
GstAudioFXBaseFIRFilterClass parent;
void (*rate_changed) (GstElement * element, gint rate);
};
GType gst_audio_fir_filter_get_type (void);
G_END_DECLS
#endif /* __GST_AUDIO_FIR_FILTER_H__ */

8
gst/audiofx/audiofx.c
View file

@ -32,8 +32,10 @@
#include "audiodynamic.h"
#include "audiocheblimit.h"
#include "audiochebband.h"
#include "audioiirfilter.h"
#include "audiowsincband.h"
#include "audiowsinclimit.h"
#include "audiofirfilter.h"
/* entry point to initialize the plug-in
* initialize the plug-in itself
@ -60,10 +62,14 @@ plugin_init (GstPlugin * plugin)
GST_TYPE_AUDIO_CHEB_LIMIT) &&
gst_element_register (plugin, "audiochebband", GST_RANK_NONE,
GST_TYPE_AUDIO_CHEB_BAND) &&
gst_element_register (plugin, "audioiirfilter", GST_RANK_NONE,
GST_TYPE_AUDIO_IIR_FILTER) &&
gst_element_register (plugin, "audiowsinclimit", GST_RANK_NONE,
GST_TYPE_AUDIO_WSINC_LIMIT) &&
gst_element_register (plugin, "audiowsincband", GST_RANK_NONE,
GST_TYPE_AUDIO_WSINC_BAND));
GST_TYPE_AUDIO_WSINC_BAND) &&
gst_element_register (plugin, "audiofirfilter", GST_RANK_NONE,
GST_TYPE_AUDIO_FIR_FILTER));
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,

291
gst/audiofx/audioiirfilter.c Normal file
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@ -0,0 +1,291 @@
/*
* GStreamer
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*/
/**
* SECTION:element-audioiirfilter
* @short_description: Generic audio IIR filter
*
* <refsect2>
* <para>
* audioiirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Infinite_impulse_response">IIR filter</ulink>. Before usage the
* "a" and "b" properties have to be set to the filter coefficients that
* should be used.
* </para>
* <para>
* The filter coefficients describe the numerator and denominator of the
* transfer function.
* </para>
* <para>
* To change the filter coefficients whenever the sampling rate changes the
* "rate-changed" signal can be used. This should be done for most
* IIR filters as they're depending on the sampling rate.
* </para>
* <title>Example application</title>
* <para>
* <include xmlns="http://www.w3.org/2003/XInclude" href="element-iirfilter-example.xml" />
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include "audioiirfilter.h"
#define GST_CAT_DEFAULT gst_audio_iir_filter_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
SIGNAL_RATE_CHANGED,
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_A,
PROP_B
};
static guint gst_audio_iir_filter_signals[LAST_SIGNAL] = { 0, };
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_iir_filter_debug, "audioiirfilter", 0, \
"Generic audio IIR filter plugin");
GST_BOILERPLATE_FULL (GstAudioIIRFilter, gst_audio_iir_filter, GstAudioFilter,
GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
static void gst_audio_iir_filter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_iir_filter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_audio_iir_filter_finalize (GObject * object);
static gboolean gst_audio_iir_filter_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
/* Element class */
static void
gst_audio_iir_filter_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class,
"Audio IIR filter", "Filter/Effect/Audio",
"Generic audio IIR filter with custom filter kernel",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
gst_audio_iir_filter_class_init (GstAudioIIRFilterClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_iir_filter_set_property;
gobject_class->get_property = gst_audio_iir_filter_get_property;
gobject_class->finalize = gst_audio_iir_filter_finalize;
g_object_class_install_property (gobject_class, PROP_A,
g_param_spec_value_array ("a", "A",
"Filter coefficients (numerator of transfer function)",
g_param_spec_double ("Coefficient", "Filter Coefficient",
"Filter coefficient", -G_MAXDOUBLE, G_MAXDOUBLE, 0.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_B,
g_param_spec_value_array ("b", "B",
"Filter coefficients (denominator of transfer function)",
g_param_spec_double ("Coefficient", "Filter Coefficient",
"Filter coefficient", -G_MAXDOUBLE, G_MAXDOUBLE, 0.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_iir_filter_setup);
/**
* GstAudioIIRFilter::rate-changed:
* @filter: the filter on which the signal is emitted
* @rate: the new sampling rate
*
* Will be emitted when the sampling rate changes. The callbacks
* will be called from the streaming thread and processing will
* stop until the event is handled.
*/
gst_audio_iir_filter_signals[SIGNAL_RATE_CHANGED] =
g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioIIRFilterClass, rate_changed),
NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
}
static void
gst_audio_iir_filter_update_coefficients (GstAudioIIRFilter * self,
GValueArray * va, GValueArray * vb)
{
gdouble *a = NULL, *b = NULL;
guint i;
if (va) {
if (self->a)
g_value_array_free (self->a);
self->a = va;
}
if (vb) {
if (self->b)
g_value_array_free (self->b);
self->b = vb;
}
if (self->a && self->a->n_values > 0)
a = g_new (gdouble, self->a->n_values);
if (self->b && self->b->n_values > 0)
b = g_new (gdouble, self->b->n_values);
if (self->a) {
for (i = 0; i < self->a->n_values; i++) {
GValue *v = g_value_array_get_nth (self->a, i);
a[i] = g_value_get_double (v);
}
}
if (self->b) {
for (i = 0; i < self->b->n_values; i++) {
GValue *v = g_value_array_get_nth (self->b, i);
b[i] = g_value_get_double (v);
}
}
gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
(self), a, (self->a) ? self->a->n_values : 0, b,
(self->b) ? self->b->n_values : 0);
}
static void
gst_audio_iir_filter_init (GstAudioIIRFilter * self,
GstAudioIIRFilterClass * g_class)
{
GValue v = { 0, };
GValueArray *a, *b;
a = g_value_array_new (1);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_set_double (&v, 1.0);
g_value_array_append (a, &v);
g_value_unset (&v);
b = NULL;
gst_audio_iir_filter_update_coefficients (self, a, b);
self->lock = g_mutex_new ();
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_iir_filter_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioIIRFilter *self = GST_AUDIO_IIR_FILTER (base);
if (self->rate != format->rate) {
g_signal_emit (G_OBJECT (self),
gst_audio_iir_filter_signals[SIGNAL_RATE_CHANGED], 0, format->rate);
self->rate = format->rate;
}
return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
}
static void
gst_audio_iir_filter_finalize (GObject * object)
{
GstAudioIIRFilter *self = GST_AUDIO_IIR_FILTER (object);
g_mutex_free (self->lock);
self->lock = NULL;
if (self->a)
g_value_array_free (self->a);
self->a = NULL;
if (self->b)
g_value_array_free (self->b);
self->b = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_audio_iir_filter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioIIRFilter *self = GST_AUDIO_IIR_FILTER (object);
g_return_if_fail (GST_IS_AUDIO_IIR_FILTER (self));
switch (prop_id) {
case PROP_A:
g_mutex_lock (self->lock);
gst_audio_iir_filter_update_coefficients (self, g_value_dup_boxed (value),
NULL);
g_mutex_unlock (self->lock);
break;
case PROP_B:
g_mutex_lock (self->lock);
gst_audio_iir_filter_update_coefficients (self, NULL,
g_value_dup_boxed (value));
g_mutex_unlock (self->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_iir_filter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioIIRFilter *self = GST_AUDIO_IIR_FILTER (object);
switch (prop_id) {
case PROP_A:
g_value_set_boxed (value, self->a);
break;
case PROP_B:
g_value_set_boxed (value, self->b);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}

71
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@ -0,0 +1,71 @@
/*
* GStreamer
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*/
#ifndef __GST_AUDIO_IIR_FILTER_H__
#define __GST_AUDIO_IIR_FILTER_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include "audiofxbaseiirfilter.h"
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_IIR_FILTER \
(gst_audio_iir_filter_get_type())
#define GST_AUDIO_IIR_FILTER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_IIR_FILTER,GstAudioIIRFilter))
#define GST_AUDIO_IIR_FILTER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_IIR_FILTER,GstAudioIIRFilterClass))
#define GST_IS_AUDIO_IIR_FILTER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_IIR_FILTER))
#define GST_IS_AUDIO_IIR_FILTER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_IIR_FILTER))
typedef struct _GstAudioIIRFilter GstAudioIIRFilter;
typedef struct _GstAudioIIRFilterClass GstAudioIIRFilterClass;
/**
* GstAudioIIRFilter:
*
* Opaque data structure.
*/
struct _GstAudioIIRFilter {
GstAudioFXBaseIIRFilter parent;
GValueArray *a, *b;
/* < private > */
GMutex *lock;
gint rate;
};
struct _GstAudioIIRFilterClass {
GstAudioFXBaseIIRFilterClass parent;
void (*rate_changed) (GstElement * element, gint rate);
};
GType gst_audio_iir_filter_get_type (void);
G_END_DECLS
#endif /* __GST_AUDIO_IIR_FILTER_H__ */

2
tests/check/Makefile.am
View file

@ -72,10 +72,12 @@ check_PROGRAMS = \
elements/audioinvert \
elements/audiochebband \
elements/audiocheblimit \
elements/audioiirfilter \
elements/audioamplify \
elements/audiodynamic \
elements/audiowsincband \
elements/audiowsinclimit \
elements/audiofirfilter \
elements/avimux \
elements/avisubtitle \
elements/deinterleave \

169
tests/check/elements/audiofirfilter.c Normal file
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@ -0,0 +1,169 @@
/* GStreamer
*
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU 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/check/gstcheck.h>
static gboolean have_eos = FALSE;
static gboolean
on_message (GstBus * bus, GstMessage * message, gpointer user_data)
{
GMainLoop *loop = (GMainLoop *) user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
case GST_MESSAGE_WARNING:
g_assert_not_reached ();
g_main_loop_quit (loop);
break;
case GST_MESSAGE_EOS:
have_eos = TRUE;
g_main_loop_quit (loop);
break;
default:
break;
}
return TRUE;
}
static void
on_rate_changed (GstElement * element, gint rate, gpointer user_data)
{
GValueArray *va;
GValue v = { 0, };
fail_unless (rate > 0);
va = g_value_array_new (6);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 1.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_object_set (G_OBJECT (element), "kernel", va, NULL);
g_value_array_free (va);
}
static gboolean have_data = FALSE;
static void
on_handoff (GstElement * object, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
if (!have_data) {
gdouble *data = (gdouble *) GST_BUFFER_DATA (buffer);
fail_unless (GST_BUFFER_SIZE (buffer) > 5 * sizeof (gdouble));
fail_unless (data[0] == 0.0);
fail_unless (data[1] == 0.0);
fail_unless (data[2] == 0.0);
fail_unless (data[3] == 0.0);
fail_unless (data[4] == 0.0);
fail_unless (data[5] != 0.0);
have_data = TRUE;
}
}
GST_START_TEST (test_pipeline)
{
GstElement *pipeline, *src, *filter, *sink;
GstBus *bus;
GMainLoop *loop;
have_data = FALSE;
have_eos = FALSE;
pipeline = gst_element_factory_make ("pipeline", NULL);
fail_unless (pipeline != NULL);
src = gst_element_factory_make ("audiotestsrc", NULL);
fail_unless (src != NULL);
g_object_set (G_OBJECT (src), "num-buffers", 1000, NULL);
filter = gst_element_factory_make ("audiofirfilter", NULL);
fail_unless (filter != NULL);
g_signal_connect (G_OBJECT (filter), "rate-changed",
G_CALLBACK (on_rate_changed), NULL);
sink = gst_element_factory_make ("fakesink", NULL);
fail_unless (sink != NULL);
g_object_set (G_OBJECT (sink), "signal-handoffs", TRUE, NULL);
g_signal_connect (G_OBJECT (sink), "handoff", G_CALLBACK (on_handoff), NULL);
gst_bin_add_many (GST_BIN (pipeline), src, filter, sink, NULL);
fail_unless (gst_element_link_many (src, filter, sink, NULL));
loop = g_main_loop_new (NULL, FALSE);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_signal_watch (bus);
g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop);
gst_object_unref (GST_OBJECT (bus));
fail_if (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
g_main_loop_run (loop);
fail_unless (have_data);
fail_unless (have_eos);
fail_unless (gst_element_set_state (pipeline,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
}
GST_END_TEST;
static Suite *
audiofirfilter_suite (void)
{
Suite *s = suite_create ("audiofirfilter");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_pipeline);
return s;
}
GST_CHECK_MAIN (audiofirfilter);

179
tests/check/elements/audioiirfilter.c Normal file
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@ -0,0 +1,179 @@
/* GStreamer
*
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU 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/check/gstcheck.h>
static gboolean have_eos = FALSE;
static gboolean
on_message (GstBus * bus, GstMessage * message, gpointer user_data)
{
GMainLoop *loop = (GMainLoop *) user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
case GST_MESSAGE_WARNING:
g_assert_not_reached ();
g_main_loop_quit (loop);
break;
case GST_MESSAGE_EOS:
have_eos = TRUE;
g_main_loop_quit (loop);
break;
default:
break;
}
return TRUE;
}
static void
on_rate_changed (GstElement * element, gint rate, gpointer user_data)
{
GValueArray *va;
GValue v = { 0, };
fail_unless (rate > 0);
va = g_value_array_new (6);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_value_set_double (&v, 1.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_object_set (G_OBJECT (element), "a", va, NULL);
g_value_array_free (va);
va = g_value_array_new (6);
g_value_set_double (&v, 0.0);
g_value_array_append (va, &v);
g_value_reset (&v);
g_object_set (G_OBJECT (element), "b", va, NULL);
g_value_array_free (va);
}
static gboolean have_data = FALSE;
static void
on_handoff (GstElement * object, GstBuffer * buffer, GstPad * pad,
gpointer user_data)
{
if (!have_data) {
gdouble *data = (gdouble *) GST_BUFFER_DATA (buffer);
fail_unless (GST_BUFFER_SIZE (buffer) > 5 * sizeof (gdouble));
fail_unless (data[0] == 0.0);
fail_unless (data[1] == 0.0);
fail_unless (data[2] == 0.0);
fail_unless (data[3] == 0.0);
fail_unless (data[4] == 0.0);
fail_unless (data[5] != 0.0);
have_data = TRUE;
}
}
GST_START_TEST (test_pipeline)
{
GstElement *pipeline, *src, *filter, *sink;
GstBus *bus;
GMainLoop *loop;
have_data = FALSE;
have_eos = FALSE;
pipeline = gst_element_factory_make ("pipeline", NULL);
fail_unless (pipeline != NULL);
src = gst_element_factory_make ("audiotestsrc", NULL);
fail_unless (src != NULL);
g_object_set (G_OBJECT (src), "num-buffers", 1000, NULL);
filter = gst_element_factory_make ("audioiirfilter", NULL);
fail_unless (filter != NULL);
g_signal_connect (G_OBJECT (filter), "rate-changed",
G_CALLBACK (on_rate_changed), NULL);
sink = gst_element_factory_make ("fakesink", NULL);
fail_unless (sink != NULL);
g_object_set (G_OBJECT (sink), "signal-handoffs", TRUE, NULL);
g_signal_connect (G_OBJECT (sink), "handoff", G_CALLBACK (on_handoff), NULL);
gst_bin_add_many (GST_BIN (pipeline), src, filter, sink, NULL);
fail_unless (gst_element_link_many (src, filter, sink, NULL));
loop = g_main_loop_new (NULL, FALSE);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_signal_watch (bus);
g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop);
gst_object_unref (GST_OBJECT (bus));
fail_if (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE);
g_main_loop_run (loop);
fail_unless (have_data);
fail_unless (have_eos);
fail_unless (gst_element_set_state (pipeline,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
}
GST_END_TEST;
static Suite *
audioiirfilter_suite (void)
{
Suite *s = suite_create ("audioiirfilter");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_pipeline);
return s;
}
GST_CHECK_MAIN (audioiirfilter);

4
tests/examples/Makefile.am
View file

@ -1,3 +1,3 @@
SUBDIRS = equalizer level rtp spectrum
SUBDIRS = audiofx equalizer level rtp spectrum
DIST_SUBDIRS = equalizer level rtp spectrum
DIST_SUBDIRS = audiofx equalizer level rtp spectrum

7
tests/examples/audiofx/Makefile.am Normal file
View file

@ -0,0 +1,7 @@
noinst_PROGRAMS = firfilter-example iirfilter-example
firfilter_example_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
firfilter_example_LDADD = $(GST_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstfft-@GST_MAJORMINOR@ $(LIBM)
iirfilter_example_CFLAGS = $(GST_CFLAGS)
iirfilter_example_LDADD = $(GST_LIBS) $(LIBM)

161
tests/examples/audiofx/firfilter-example.c Normal file
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@ -0,0 +1,161 @@
/* GStreamer
* Copyright (C) 2009 Sebastian Droege <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/* This small sample application creates a bandpass FIR filter
* by transforming the frequency response to the filter kernel.
*/
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/fft/gstfftf64.h>
static gboolean
on_message (GstBus * bus, GstMessage * message, gpointer user_data)
{
GMainLoop *loop = (GMainLoop *) user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
g_error ("Got ERROR");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_WARNING:
g_warning ("Got WARNING");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_EOS:
g_main_loop_quit (loop);
break;
default:
break;
}
return TRUE;
}
static void
on_rate_changed (GstElement * element, gint rate, gpointer user_data)
{
GValueArray *va;
GValue v = { 0, };
GstFFTF64 *fft;
GstFFTF64Complex frequency_response[17];
gdouble tmp[32];
gdouble filter_kernel[32];
guint i;
/* Create the frequency response: zero outside
* a small frequency band */
for (i = 0; i < 17; i++) {
if (i < 5 || i > 11)
frequency_response[i].r = 0.0;
else
frequency_response[i].r = 1.0;
frequency_response[i].i = 0.0;
}
/* Calculate the inverse FT of the frequency response */
fft = gst_fft_f64_new (32, TRUE);
gst_fft_f64_inverse_fft (fft, frequency_response, tmp);
gst_fft_f64_free (fft);
/* Shift the inverse FT of the frequency response by 16,
* i.e. the half of the kernel length to get the
* impulse response. See http://www.dspguide.com/ch17/1.htm
* for more information.
*/
for (i = 0; i < 32; i++)
filter_kernel[i] = tmp[(i + 16) % 32];
/* Apply the hamming window to the impulse response to get
* a better result than given from the rectangular window
*/
for (i = 0; i < 32; i++)
filter_kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / 32));
va = g_value_array_new (1);
g_value_init (&v, G_TYPE_DOUBLE);
for (i = 0; i < 32; i++) {
g_value_set_double (&v, filter_kernel[i]);
g_value_array_append (va, &v);
g_value_reset (&v);
}
g_object_set (G_OBJECT (element), "kernel", va, NULL);
/* Latency is 1/2 of the kernel length for this method of
* calculating a filter kernel from the frequency response
*/
g_object_set (G_OBJECT (element), "latency", 32 / 2, NULL);
g_value_array_free (va);
}
gint
main (gint argc, gchar * argv[])
{
GstElement *pipeline, *src, *filter, *conv, *sink;
GstBus *bus;
GMainLoop *loop;
gst_init (NULL, NULL);
pipeline = gst_element_factory_make ("pipeline", NULL);
src = gst_element_factory_make ("audiotestsrc", NULL);
g_object_set (G_OBJECT (src), "wave", 5, NULL);
filter = gst_element_factory_make ("audiofirfilter", NULL);
g_signal_connect (G_OBJECT (filter), "rate-changed",
G_CALLBACK (on_rate_changed), NULL);
conv = gst_element_factory_make ("audioconvert", NULL);
sink = gst_element_factory_make ("autoaudiosink", NULL);
g_return_val_if_fail (sink != NULL, -1);
gst_bin_add_many (GST_BIN (pipeline), src, filter, conv, sink, NULL);
if (!gst_element_link_many (src, filter, conv, sink, NULL)) {
g_error ("Failed to link elements");
return -2;
}
loop = g_main_loop_new (NULL, FALSE);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_signal_watch (bus);
g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop);
gst_object_unref (GST_OBJECT (bus));
if (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
g_error ("Failed to go into PLAYING state");
return -3;
}
g_main_loop_run (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
return 0;
}

137
tests/examples/audiofx/iirfilter-example.c Normal file
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@ -0,0 +1,137 @@
/* GStreamer
* Copyright (C) 2009 Sebastian Droege <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/* This small sample application creates a lowpass IIR filter
* and applies it to white noise.
* See http://www.dspguide.com/ch19/2.htm for a description
* of the IIR filter that is used.
*/
#include <string.h>
#include <math.h>
#include <gst/gst.h>
/* Cutoff of 4000 Hz */
#define CUTOFF (4000.0)
static gboolean
on_message (GstBus * bus, GstMessage * message, gpointer user_data)
{
GMainLoop *loop = (GMainLoop *) user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
g_error ("Got ERROR");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_WARNING:
g_warning ("Got WARNING");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_EOS:
g_main_loop_quit (loop);
break;
default:
break;
}
return TRUE;
}
static void
on_rate_changed (GstElement * element, gint rate, gpointer user_data)
{
GValueArray *va;
GValue v = { 0, };
gdouble x;
if (rate / 2.0 > CUTOFF)
x = exp (-2.0 * M_PI * (CUTOFF / rate));
else
x = 0.0;
va = g_value_array_new (1);
g_value_init (&v, G_TYPE_DOUBLE);
g_value_set_double (&v, 1.0 - x);
g_value_array_append (va, &v);
g_value_reset (&v);
g_object_set (G_OBJECT (element), "a", va, NULL);
g_value_array_free (va);
va = g_value_array_new (1);
g_value_set_double (&v, x);
g_value_array_append (va, &v);
g_value_reset (&v);
g_object_set (G_OBJECT (element), "b", va, NULL);
g_value_array_free (va);
}
gint
main (gint argc, gchar * argv[])
{
GstElement *pipeline, *src, *filter, *conv, *sink;
GstBus *bus;
GMainLoop *loop;
gst_init (NULL, NULL);
pipeline = gst_element_factory_make ("pipeline", NULL);
src = gst_element_factory_make ("audiotestsrc", NULL);
g_object_set (G_OBJECT (src), "wave", 5, NULL);
filter = gst_element_factory_make ("audioiirfilter", NULL);
g_signal_connect (G_OBJECT (filter), "rate-changed",
G_CALLBACK (on_rate_changed), NULL);
conv = gst_element_factory_make ("audioconvert", NULL);
sink = gst_element_factory_make ("autoaudiosink", NULL);
g_return_val_if_fail (sink != NULL, -1);
gst_bin_add_many (GST_BIN (pipeline), src, filter, conv, sink, NULL);
if (!gst_element_link_many (src, filter, conv, sink, NULL)) {
g_error ("Failed to link elements");
return -2;
}
loop = g_main_loop_new (NULL, FALSE);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_signal_watch (bus);
g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop);
gst_object_unref (GST_OBJECT (bus));
if (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
g_error ("Failed to go into PLAYING state");
return -3;
}
g_main_loop_run (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_main_loop_unref (loop);
gst_object_unref (pipeline);
return 0;
}