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

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebband.c:
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c:
* gst/audiofx/audiocheblimit.h:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebband.c:
* tests/check/elements/audiocheblimit.c:
* tests/check/elements/audiochebyshevfreqband.c:
* tests/check/elements/audiochebyshevfreqlimit.c:
Rename audiochebyshevfreqband -> audiochebband and
audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS
surgery.
Closes: #491811
This commit is contained in:
Jan Schmidt 2008-02-06 23:44:43 +00:00
parent 8921eb2cd9
commit 22bea9fec3
23 changed files with 687 additions and 5149 deletions

View file

@ -1,3 +1,33 @@
2008-02-06 Jan Schmidt <jan.schmidt@sun.com>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebband.c:
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c:
* gst/audiofx/audiocheblimit.h:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebband.c:
* tests/check/elements/audiocheblimit.c:
* tests/check/elements/audiochebyshevfreqband.c:
* tests/check/elements/audiochebyshevfreqlimit.c:
Rename audiochebyshevfreqband -> audiochebband and
audiochebyshevfreqlimit -> audiocheblimit and do the requisite CVS
surgery.
Closes: #491811
2008-02-05 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: orjan <orjanf at axis dot com>

2
common

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

View file

@ -100,8 +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/audiofx/audiocheblimit.h \
$(top_srcdir)/gst/audiofx/audiochebband.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
$(top_srcdir)/gst/autodetect/gstautovideosink.h \
$(top_srcdir)/gst/avi/gstavidemux.h \

View file

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

View file

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

View file

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

View file

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

View file

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

View file

@ -33,7 +33,7 @@
*/
/**
* SECTION:element-audiochebyshevfreqband
* SECTION:element-audiochebband
* @short_description: Chebyshev band pass and band reject filter
*
* <refsect2>
@ -65,9 +65,9 @@
* <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
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
@ -85,13 +85,13 @@
#include <math.h>
#include "audiochebyshevfreqband.h"
#include "audiochebband.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
#define GST_CAT_DEFAULT gst_audio_cheb_band_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
GST_ELEMENT_DETAILS ("AudioChebBand",
"Filter/Effect/Audio",
"Chebyshev band pass and band reject filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
@ -122,26 +122,25 @@ enum
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0, "audiochebband element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
GST_BOILERPLATE_FULL (GstAudioChebBand, gst_audio_cheb_band,
GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
static void gst_audio_cheb_band_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
static void gst_audio_cheb_band_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
static gboolean gst_audio_cheb_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);
gst_audio_cheb_band_transform_ip (GstBaseTransform * base, GstBuffer * buf);
static gboolean gst_audio_cheb_band_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqBand * filter,
static void process_64 (GstAudioChebBand * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqBand * filter,
static void process_32 (GstAudioChebBand * filter,
gfloat * data, guint num_samples);
enum
@ -150,9 +149,9 @@ enum
MODE_BAND_REJECT
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_cheb_band_mode_get_type ())
static GType
gst_audio_chebyshev_freq_band_mode_get_type (void)
gst_audio_cheb_band_mode_get_type (void)
{
static GType gtype = 0;
@ -165,7 +164,7 @@ gst_audio_chebyshev_freq_band_mode_get_type (void)
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values);
gtype = g_enum_register_static ("GstAudioChebBandMode", values);
}
return gtype;
}
@ -173,7 +172,7 @@ gst_audio_chebyshev_freq_band_mode_get_type (void)
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_band_base_init (gpointer klass)
gst_audio_cheb_band_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
@ -187,9 +186,9 @@ gst_audio_chebyshev_freq_band_base_init (gpointer klass)
}
static void
gst_audio_chebyshev_freq_band_dispose (GObject * object)
gst_audio_cheb_band_dispose (GObject * object)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object);
if (filter->a) {
g_free (filter->a);
@ -202,7 +201,7 @@ gst_audio_chebyshev_freq_band_dispose (GObject * object)
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
GstAudioChebBandChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
@ -219,8 +218,7 @@ gst_audio_chebyshev_freq_band_dispose (GObject * object)
}
static void
gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
klass)
gst_audio_cheb_band_class_init (GstAudioChebBandClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
@ -230,9 +228,9 @@ gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
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;
gobject_class->set_property = gst_audio_cheb_band_set_property;
gobject_class->get_property = gst_audio_cheb_band_get_property;
gobject_class->dispose = gst_audio_cheb_band_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
@ -265,15 +263,15 @@ gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
"Number of poles to use, will be rounded up to the next multiply of four",
4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
GST_DEBUG_FUNCPTR (gst_audio_cheb_band_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_start);
}
static void
gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandClass * klass)
gst_audio_cheb_band_init (GstAudioChebBand * filter,
GstAudioChebBandClass * klass)
{
filter->lower_frequency = filter->upper_frequency = 0.0;
filter->mode = MODE_BAND_PASS;
@ -289,7 +287,7 @@ gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter,
generate_biquad_coefficients (GstAudioChebBand * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
{
@ -520,7 +518,7 @@ calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
}
static void
generate_coefficients (GstAudioChebyshevFreqBand * filter)
generate_coefficients (GstAudioChebBand * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
@ -535,7 +533,7 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
GstAudioChebBandChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
@ -553,7 +551,7 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
filter->channels = g_new0 (GstAudioChebBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
@ -565,7 +563,7 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
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);
filter->channels = g_new0 (GstAudioChebBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
return;
}
@ -591,9 +589,9 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 5);
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
filter->channels = g_new0 (GstAudioChebBandChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
GstAudioChebBandChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
@ -714,10 +712,10 @@ generate_coefficients (GstAudioChebyshevFreqBand * filter)
}
static void
gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
gst_audio_cheb_band_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object);
switch (prop_id) {
case PROP_MODE:
@ -763,10 +761,10 @@ gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
}
static void
gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
gst_audio_cheb_band_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object);
switch (prop_id) {
case PROP_MODE:
@ -796,17 +794,16 @@ gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
gst_audio_cheb_band_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
filter->process = (GstAudioChebBandProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
filter->process = (GstAudioChebBandProcessFunc)
process_64;
else
ret = FALSE;
@ -817,8 +814,8 @@ gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
}
static inline gdouble
process (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
process (GstAudioChebBand * filter,
GstAudioChebBandChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
@ -857,7 +854,7 @@ process (GstAudioChebyshevFreqBand * filter,
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstAudioChebyshevFreqBand * filter, \
process_##width (GstAudioChebBand * filter, \
g##ctype * data, guint num_samples) \
{ \
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
@ -878,10 +875,9 @@ DEFINE_PROCESS_FUNC (64, double);
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
gst_audio_cheb_band_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
@ -900,11 +896,11 @@ gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
}
static gboolean
gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
gst_audio_cheb_band_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqBandChannelCtx *ctx;
GstAudioChebBandChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if

View file

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

View file

@ -29,7 +29,7 @@
*/
/**
* SECTION:element-audiochebyshevfreqlimit
* SECTION:element-audiocheblimit
* @short_description: Chebyshev low pass and high pass filter
*
* <refsect2>
@ -61,9 +61,9 @@
* <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
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
@ -81,13 +81,13 @@
#include <math.h>
#include "audiochebyshevfreqlimit.h"
#include "audiocheblimit.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
#define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
GST_ELEMENT_DETAILS ("AudioChebLimit",
"Filter/Effect/Audio",
"Chebyshev low pass and high pass filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
@ -117,27 +117,25 @@ enum
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
DEBUG_INIT);
GST_BOILERPLATE_FULL (GstAudioChebLimit,
gst_audio_cheb_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
static void gst_audio_cheb_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
static void gst_audio_cheb_limit_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
static gboolean gst_audio_cheb_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);
gst_audio_cheb_limit_transform_ip (GstBaseTransform * base, GstBuffer * buf);
static gboolean gst_audio_cheb_limit_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqLimit * filter,
static void process_64 (GstAudioChebLimit * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqLimit * filter,
static void process_32 (GstAudioChebLimit * filter,
gfloat * data, guint num_samples);
enum
@ -146,9 +144,9 @@ enum
MODE_HIGH_PASS
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_cheb_limit_mode_get_type ())
static GType
gst_audio_chebyshev_freq_limit_mode_get_type (void)
gst_audio_cheb_limit_mode_get_type (void)
{
static GType gtype = 0;
@ -161,7 +159,7 @@ gst_audio_chebyshev_freq_limit_mode_get_type (void)
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
gtype = g_enum_register_static ("GstAudioChebLimitMode", values);
}
return gtype;
}
@ -169,7 +167,7 @@ gst_audio_chebyshev_freq_limit_mode_get_type (void)
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
gst_audio_cheb_limit_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
@ -183,9 +181,9 @@ gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
}
static void
gst_audio_chebyshev_freq_limit_dispose (GObject * object)
gst_audio_cheb_limit_dispose (GObject * object)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
if (filter->a) {
g_free (filter->a);
@ -198,7 +196,7 @@ gst_audio_chebyshev_freq_limit_dispose (GObject * object)
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
GstAudioChebLimitChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
@ -215,8 +213,7 @@ gst_audio_chebyshev_freq_limit_dispose (GObject * object)
}
static void
gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
klass)
gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
@ -226,9 +223,9 @@ gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
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;
gobject_class->set_property = gst_audio_cheb_limit_set_property;
gobject_class->get_property = gst_audio_cheb_limit_get_property;
gobject_class->dispose = gst_audio_cheb_limit_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
@ -256,16 +253,15 @@ gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
"Number of poles to use, will be rounded up to the next even number",
2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_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);
GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_start);
}
static void
gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitClass * klass)
gst_audio_cheb_limit_init (GstAudioChebLimit * filter,
GstAudioChebLimitClass * klass)
{
filter->cutoff = 0.0;
filter->mode = MODE_LOW_PASS;
@ -281,7 +277,7 @@ gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
generate_biquad_coefficients (GstAudioChebLimit * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2,
gdouble * b1, gdouble * b2)
{
@ -471,7 +467,7 @@ calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
}
static void
generate_coefficients (GstAudioChebyshevFreqLimit * filter)
generate_coefficients (GstAudioChebLimit * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
@ -486,7 +482,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
GstAudioChebLimitChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
@ -504,7 +500,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
@ -516,7 +512,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
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);
filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return;
} else if (filter->cutoff <= 0.0) {
@ -524,7 +520,7 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
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);
filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return;
}
@ -540,9 +536,9 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 3);
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
filter->channels = g_new0 (GstAudioChebLimitChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
GstAudioChebLimitChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
@ -623,10 +619,10 @@ generate_coefficients (GstAudioChebyshevFreqLimit * filter)
}
static void
gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
gst_audio_cheb_limit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
@ -666,10 +662,10 @@ gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
}
static void
gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
gst_audio_cheb_limit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
@ -696,17 +692,16 @@ gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
gst_audio_cheb_limit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
filter->process = (GstAudioChebLimitProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
filter->process = (GstAudioChebLimitProcessFunc)
process_64;
else
ret = FALSE;
@ -717,8 +712,8 @@ gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
}
static inline gdouble
process (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
process (GstAudioChebLimit * filter,
GstAudioChebLimitChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
@ -757,7 +752,7 @@ process (GstAudioChebyshevFreqLimit * filter,
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstAudioChebyshevFreqLimit * filter, \
process_##width (GstAudioChebLimit * filter, \
g##ctype * data, guint num_samples) \
{ \
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
@ -778,10 +773,9 @@ DEFINE_PROCESS_FUNC (64, double);
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
gst_audio_cheb_limit_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
@ -801,11 +795,11 @@ gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
static gboolean
gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
gst_audio_cheb_limit_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqLimitChannelCtx *ctx;
GstAudioChebLimitChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if

View file

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

View file

@ -1,922 +0,0 @@
/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Chebyshev type 1 filter design based on
* "The Scientist and Engineer's Guide to DSP", Chapter 20.
* http://www.dspguide.com/
*
* For type 2 and Chebyshev filters in general read
* http://en.wikipedia.org/wiki/Chebyshev_filter
*
* Transformation from lowpass to bandpass/bandreject:
* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm
* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm
*
*/
/**
* SECTION:element-audiochebyshevfreqband
* @short_description: Chebyshev band pass and band reject filter
*
* <refsect2>
* <para>
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The number of poles and the ripple parameter control the rolloff.
* </para>
* <para>
* This element has the advantage over the windowed sinc bandpass and bandreject filter that it is
* much faster and produces almost as good results. It's only disadvantages are the highly
* non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
* </para>
* <para>
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
* </para>
* <para>
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
* </para>
* <para>
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
* <para><note>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
* </note></para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include <math.h>
#include "audiochebyshevfreqband.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
"Filter/Effect/Audio",
"Chebyshev band pass and band reject filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MODE,
PROP_TYPE,
PROP_LOWER_FREQUENCY,
PROP_UPPER_FREQUENCY,
PROP_RIPPLE,
PROP_POLES
};
#define ALLOWED_CAPS \
"audio/x-raw-float," \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER," \
" rate = (int) [ 1, MAX ]," \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqBand * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqBand * filter,
gfloat * data, guint num_samples);
enum
{
MODE_BAND_PASS = 0,
MODE_BAND_REJECT
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
static GType
gst_audio_chebyshev_freq_band_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_BAND_PASS, "Band pass (default)",
"band-pass"},
{MODE_BAND_REJECT, "Band reject",
"band-reject"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_band_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &element_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
}
static void
gst_audio_chebyshev_freq_band_dispose (GObject * object)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE,
MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type",
"Type of the chebychev filter", 1, 2,
1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider without */
g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
g_param_spec_float ("lower-frequency", "Lower frequency",
"Start frequency of the band (Hz)", 0.0, 100000.0,
0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
g_param_spec_float ("upper-frequency", "Upper frequency",
"Stop frequency of the band (Hz)", 0.0, 100000.0,
0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple",
"Amount of ripple (dB)", 0.0, 200.0,
0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: What to do about this upper boundary? With a frequencies near
* rate/4 32 poles are completely possible, with frequencies very low
* or very high 16 poles already produces only noise */
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next multiply of four",
4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
}
static void
gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandClass * klass)
{
filter->lower_frequency = filter->upper_frequency = 0.0;
filter->mode = MODE_BAND_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
filter->have_coeffs = FALSE;
filter->num_a = 0;
filter->num_b = 0;
filter->channels = NULL;
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
{
gint np = filter->poles / 2;
gdouble ripple = filter->ripple;
/* pole location in s-plane */
gdouble rp, ip;
/* zero location in s-plane */
gdouble rz = 0.0, iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
gint type = filter->type;
/* Calculate pole location for lowpass at frequency 1 */
{
gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
}
/* If we allow ripple, move the pole from the unit
* circle to an ellipse and keep cutoff at frequency 1 */
if (ripple > 0 && type == 1) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (1.0 / es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
} else if (type == 2) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
}
/* Calculate inverse of the pole location to move from
* type I to type II */
if (type == 2) {
gdouble mag2 = rp * rp + ip * ip;
rp /= mag2;
ip /= mag2;
}
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
gdouble mag2;
rz = 0.0;
iz = cos (angle);
mag2 = rz * rz + iz * iz;
rz /= mag2;
iz /= mag2;
}
/* Convert from s-domain to z-domain by
* using the bilinear Z-transform, i.e.
* substitute s by (2/t)*((z-1)/(z+1))
* with t = 2 * tan(0.5).
*/
if (type == 1) {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t) / d;
x1 = 2.0 * x0;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
} else {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t * iz * iz + 4.0) / d;
x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
}
/* Convert from lowpass at frequency 1 to either bandpass
* or band reject.
*
* For bandpass substitute z^(-1) with:
*
* -2 -1
* -z + alpha * z - beta
* ----------------------------
* -2 -1
* beta * z - alpha * z + 1
*
* alpha = (2*a*b)/(1+b)
* beta = (b-1)/(b+1)
* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
* b = tan(1/2) * cot((w1 - w0)/2)
*
* For bandreject substitute z^(-1) with:
*
* -2 -1
* z - alpha * z + beta
* ----------------------------
* -2 -1
* beta * z - alpha * z + 1
*
* alpha = (2*a)/(1+b)
* beta = (1-b)/(1+b)
* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
* b = tan(1/2) * tan((w1 - w0)/2)
*
*/
{
gdouble a, b, d;
gdouble alpha, beta;
gdouble w0 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w1 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
if (filter->mode == MODE_BAND_PASS) {
a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0);
alpha = (2.0 * a * b) / (1.0 + b);
beta = (b - 1.0) / (b + 1.0);
d = 1.0 + beta * (y1 - beta * y2);
*a0 = (x0 + beta * (-x1 + beta * x2)) / d;
*a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d;
*a2 =
(-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) +
alpha * alpha * (x0 - x1 + x2)) / d;
*a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d;
*a4 = (beta * (beta * x0 - x1) + x2) / d;
*b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d;
*b2 =
(-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) +
2.0 * beta * (-1.0 + y2)) / d;
*b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d;
*b4 = (-beta * beta - beta * y1 + y2) / d;
} else {
a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0);
alpha = (2.0 * a) / (1.0 + b);
beta = (1.0 - b) / (1.0 + b);
d = -1.0 + beta * (beta * y2 + y1);
*a0 = (-x0 - beta * x1 - beta * beta * x2) / d;
*a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d;
*a2 =
(-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) -
alpha * alpha * (x0 + x1 + x2)) / d;
*a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d;
*a4 = (-beta * beta * x0 - beta * x1 - x2) / d;
*b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d;
*b2 =
-(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) +
alpha * alpha * (-1.0 + y1 + y2)) / d;
*b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d;
*b4 = -(-beta * beta + beta * y1 + y2) / d;
}
}
}
/* Evaluate the transfer function that corresponds to the IIR
* coefficients at zr + zi*I and return the magnitude */
static gdouble
calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
gdouble zi)
{
gdouble sum_ar, sum_ai;
gdouble sum_br, sum_bi;
gdouble gain_r, gain_i;
gdouble sum_r_old;
gdouble sum_i_old;
gint i;
sum_ar = 0.0;
sum_ai = 0.0;
for (i = num_a; i >= 0; i--) {
sum_r_old = sum_ar;
sum_i_old = sum_ai;
sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
}
sum_br = 0.0;
sum_bi = 0.0;
for (i = num_b; i >= 0; i--) {
sum_r_old = sum_br;
sum_i_old = sum_bi;
sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
}
sum_br += 1.0;
sum_bi += 0.0;
gain_r =
(sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
gain_i =
(sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
return (sqrt (gain_r * gain_r + gain_i * gain_i));
}
static void
generate_coefficients (GstAudioChebyshevFreqBand * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
filter->have_coeffs = TRUE;
if (filter->upper_frequency <= filter->lower_frequency) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
return;
}
if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) {
filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2;
GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency");
}
if (filter->lower_frequency < 0.0) {
filter->lower_frequency = 0.0;
GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0");
}
/* Calculate coefficients for the chebyshev filter */
{
gint np = filter->poles;
gdouble *a, *b;
gint i, p;
filter->num_a = np + 1;
filter->a = a = g_new0 (gdouble, np + 5);
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 5);
filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
}
/* Calculate transfer function coefficients */
a[4] = 1.0;
b[4] = 1.0;
for (p = 1; p <= np / 4; p++) {
gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4;
gdouble *ta = g_new0 (gdouble, np + 5);
gdouble *tb = g_new0 (gdouble, np + 5);
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1,
&b2, &b3, &b4);
memcpy (ta, a, sizeof (gdouble) * (np + 5));
memcpy (tb, b, sizeof (gdouble) * (np + 5));
/* add the new coefficients for the new two poles
* to the cascade by multiplication of the transfer
* functions */
for (i = 4; i < np + 5; i++) {
a[i] =
a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] +
a4 * ta[i - 4];
b[i] =
tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] -
b4 * tb[i - 4];
}
g_free (ta);
g_free (tb);
}
/* Move coefficients to the beginning of the array
* and multiply the b coefficients with -1 to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
b[4] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 4];
b[i] = -b[i + 4];
}
/* Normalize to unity gain at frequency 0 and frequency
* 0.5 for bandreject and unity gain at band center frequency
* for bandpass */
if (filter->mode == MODE_BAND_REJECT) {
/* gain is sqrt(H(0)*H(0.5)) */
gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0);
gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0);
gain1 = sqrt (gain1 * gain2);
for (i = 0; i <= np; i++) {
a[i] /= gain1;
}
} else {
/* gain is H(wc), wc = center frequency */
gdouble w1 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w2 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w0 = (w2 + w1) / 2.0;
gdouble zr = cos (w0), zi = sin (w0);
gdouble gain = calculate_gain (a, b, np, np, zr, zi);
for (i = 0; i <= np; i++) {
a[i] /= gain;
}
}
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
"mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB",
(filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject",
filter->type, filter->poles, filter->lower_frequency,
filter->upper_frequency, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz",
20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
{
gdouble w1 =
2.0 * M_PI * (filter->lower_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w2 =
2.0 * M_PI * (filter->upper_frequency /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble w0 = (w2 + w1) / 2.0;
gdouble zr, zi;
zr = cos (w1);
zi = sin (w1);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->lower_frequency);
zr = cos (w0);
zi = sin (w0);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) ((filter->lower_frequency + filter->upper_frequency) / 2.0));
zr = cos (w2);
zi = sin (w2);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->upper_frequency);
}
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
GST_AUDIO_FILTER (filter)->format.rate / 2);
}
}
static void
gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
switch (prop_id) {
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->mode = g_value_get_enum (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_TYPE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->type = g_value_get_int (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_LOWER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (filter);
filter->lower_frequency = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_UPPER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (filter);
filter->upper_frequency = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_RIPPLE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->ripple = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_POLES:
GST_BASE_TRANSFORM_LOCK (filter);
filter->poles = GST_ROUND_UP_4 (g_value_get_int (value));
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_TYPE:
g_value_set_int (value, filter->type);
break;
case PROP_LOWER_FREQUENCY:
g_value_set_float (value, filter->lower_frequency);
break;
case PROP_UPPER_FREQUENCY:
g_value_set_float (value, filter->upper_frequency);
break;
case PROP_RIPPLE:
g_value_set_float (value, filter->ripple);
break;
case PROP_POLES:
g_value_set_int (value, filter->poles);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqBandProcessFunc)
process_64;
else
ret = FALSE;
filter->have_coeffs = FALSE;
return ret;
}
static inline gdouble
process (GstAudioChebyshevFreqBand * filter,
GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
val += filter->a[i] * ctx->x[j];
j--;
if (j < 0)
j = filter->num_a - 1;
}
for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
val += filter->b[i] * ctx->y[j];
j--;
if (j < 0)
j = filter->num_b - 1;
}
if (ctx->x) {
ctx->x_pos++;
if (ctx->x_pos > filter->num_a - 1)
ctx->x_pos = 0;
ctx->x[ctx->x_pos] = x0;
}
if (ctx->y) {
ctx->y_pos++;
if (ctx->y_pos > filter->num_b - 1)
ctx->y_pos = 0;
ctx->y[ctx->y_pos] = val;
}
return val;
}
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstAudioChebyshevFreqBand * filter, \
g##ctype * data, guint num_samples) \
{ \
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
gdouble val; \
\
for (i = 0; i < num_samples / channels; i++) { \
for (j = 0; j < channels; j++) { \
val = process (filter, &filter->channels[j], *data); \
*data++ = val; \
} \
} \
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
#undef DEFINE_PROCESS_FUNC
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
if (!filter->have_coeffs)
generate_coefficients (filter);
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static gboolean
gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqBandChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if
* already existing */
if (channels && filter->channels) {
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
if (ctx->x)
memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
if (ctx->y)
memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
}
}
return TRUE;
}

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@ -1,79 +0,0 @@
/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type())
#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand;
typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass;
typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint);
typedef struct
{
gdouble *x;
gint x_pos;
gdouble *y;
gint y_pos;
} GstAudioChebyshevFreqBandChannelCtx;
struct _GstAudioChebyshevFreqBand
{
GstAudioFilter audiofilter;
gint mode;
gint type;
gint poles;
gfloat lower_frequency;
gfloat upper_frequency;
gfloat ripple;
/* < private > */
GstAudioChebyshevFreqBandProcessFunc process;
gboolean have_coeffs;
gdouble *a;
gint num_a;
gdouble *b;
gint num_b;
GstAudioChebyshevFreqBandChannelCtx *channels;
};
struct _GstAudioChebyshevFreqBandClass
{
GstAudioFilterClass parent;
};
GType gst_audio_chebyshev_freq_band_get_type (void);
G_END_DECLS
#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */

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@ -1,823 +0,0 @@
/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Chebyshev type 1 filter design based on
* "The Scientist and Engineer's Guide to DSP", Chapter 20.
* http://www.dspguide.com/
*
* For type 2 and Chebyshev filters in general read
* http://en.wikipedia.org/wiki/Chebyshev_filter
*
*/
/**
* SECTION:element-audiochebyshevfreqlimit
* @short_description: Chebyshev low pass and high pass filter
*
* <refsect2>
* <para>
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
* </para>
* <para>
* This element has the advantage over the windowed sinc lowpass and highpass filter that it is
* much faster and produces almost as good results. It's only disadvantages are the highly
* non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
* </para>
* <para>
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
* </para>
* <para>
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
* </para>
* <para>
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
* <para><note>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
* </note></para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include <math.h>
#include "audiochebyshevfreqlimit.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
"Filter/Effect/Audio",
"Chebyshev low pass and high pass filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MODE,
PROP_TYPE,
PROP_CUTOFF,
PROP_RIPPLE,
PROP_POLES
};
#define ALLOWED_CAPS \
"audio/x-raw-float," \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER," \
" rate = (int) [ 1, MAX ]," \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
DEBUG_INIT);
static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqLimit * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqLimit * filter,
gfloat * data, guint num_samples);
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
static GType
gst_audio_chebyshev_freq_limit_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_LOW_PASS, "Low pass (default)",
"low-pass"},
{MODE_HIGH_PASS, "High pass",
"high-pass"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &element_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
}
static void
gst_audio_chebyshev_freq_limit_dispose (GObject * object)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode",
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider without */
g_object_class_install_property (gobject_class, PROP_CUTOFF,
g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
100000.0, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
200.0, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: What to do about this upper boundary? With a cutoff frequency of
* rate/4 32 poles are completely possible, with a cutoff frequency very low
* or very high 16 poles already produces only noise */
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next even number",
2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
}
static void
gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitClass * klass)
{
filter->cutoff = 0.0;
filter->mode = MODE_LOW_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
filter->have_coeffs = FALSE;
filter->num_a = 0;
filter->num_b = 0;
filter->channels = NULL;
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2,
gdouble * b1, gdouble * b2)
{
gint np = filter->poles;
gdouble ripple = filter->ripple;
/* pole location in s-plane */
gdouble rp, ip;
/* zero location in s-plane */
gdouble rz = 0.0, iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
gint type = filter->type;
/* Calculate pole location for lowpass at frequency 1 */
{
gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
}
/* If we allow ripple, move the pole from the unit
* circle to an ellipse and keep cutoff at frequency 1 */
if (ripple > 0 && type == 1) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (1.0 / es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
} else if (type == 2) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
}
/* Calculate inverse of the pole location to convert from
* type I to type II */
if (type == 2) {
gdouble mag2 = rp * rp + ip * ip;
rp /= mag2;
ip /= mag2;
}
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
gdouble mag2;
rz = 0.0;
iz = cos (angle);
mag2 = rz * rz + iz * iz;
rz /= mag2;
iz /= mag2;
}
/* Convert from s-domain to z-domain by
* using the bilinear Z-transform, i.e.
* substitute s by (2/t)*((z-1)/(z+1))
* with t = 2 * tan(0.5).
*/
if (type == 1) {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t) / d;
x1 = 2.0 * x0;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
} else {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t * iz * iz + 4.0) / d;
x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
}
/* Convert from lowpass at frequency 1 to either lowpass
* or highpass.
*
* For lowpass substitute z^(-1) with:
* -1
* z - k
* ------------
* -1
* 1 - k * z
*
* k = sin((1-w)/2) / sin((1+w)/2)
*
* For highpass substitute z^(-1) with:
*
* -1
* -z - k
* ------------
* -1
* 1 + k * z
*
* k = -cos((1+w)/2) / cos((1-w)/2)
*
*/
{
gdouble k, d;
gdouble omega =
2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
if (filter->mode == MODE_LOW_PASS)
k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
else
k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
d = 1.0 + y1 * k - y2 * k * k;
*a0 = (x0 + k * (-x1 + k * x2)) / d;
*a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
*a2 = (x0 * k * k - x1 * k + x2) / d;
*b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
*b2 = (-k * k - y1 * k + y2) / d;
if (filter->mode == MODE_HIGH_PASS) {
*a1 = -*a1;
*b1 = -*b1;
}
}
}
/* Evaluate the transfer function that corresponds to the IIR
* coefficients at zr + zi*I and return the magnitude */
static gdouble
calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
gdouble zi)
{
gdouble sum_ar, sum_ai;
gdouble sum_br, sum_bi;
gdouble gain_r, gain_i;
gdouble sum_r_old;
gdouble sum_i_old;
gint i;
sum_ar = 0.0;
sum_ai = 0.0;
for (i = num_a; i >= 0; i--) {
sum_r_old = sum_ar;
sum_i_old = sum_ai;
sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
}
sum_br = 0.0;
sum_bi = 0.0;
for (i = num_b; i >= 0; i--) {
sum_r_old = sum_br;
sum_i_old = sum_bi;
sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
}
sum_br += 1.0;
sum_bi += 0.0;
gain_r =
(sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
gain_i =
(sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
return (sqrt (gain_r * gain_r + gain_i * gain_i));
}
static void
generate_coefficients (GstAudioChebyshevFreqLimit * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
filter->have_coeffs = TRUE;
if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return;
} else if (filter->cutoff <= 0.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return;
}
/* Calculate coefficients for the chebyshev filter */
{
gint np = filter->poles;
gdouble *a, *b;
gint i, p;
filter->num_a = np + 1;
filter->a = a = g_new0 (gdouble, np + 3);
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 3);
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
}
/* Calculate transfer function coefficients */
a[2] = 1.0;
b[2] = 1.0;
for (p = 1; p <= np / 2; p++) {
gdouble a0, a1, a2, b1, b2;
gdouble *ta = g_new0 (gdouble, np + 3);
gdouble *tb = g_new0 (gdouble, np + 3);
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
memcpy (ta, a, sizeof (gdouble) * (np + 3));
memcpy (tb, b, sizeof (gdouble) * (np + 3));
/* add the new coefficients for the new two poles
* to the cascade by multiplication of the transfer
* functions */
for (i = 2; i < np + 3; i++) {
a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
}
g_free (ta);
g_free (tb);
}
/* Move coefficients to the beginning of the array
* and multiply the b coefficients with -1 to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
b[2] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 2];
b[i] = -b[i + 2];
}
/* Normalize to unity gain at frequency 0 for lowpass
* and frequency 0.5 for highpass */
{
gdouble gain;
if (filter->mode == MODE_LOW_PASS)
gain = calculate_gain (a, b, np, np, 1.0, 0.0);
else
gain = calculate_gain (a, b, np, np, -1.0, 0.0);
for (i = 0; i <= np; i++) {
a[i] /= gain;
}
}
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
"mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB",
(filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
filter->type, filter->poles, filter->cutoff, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
{
gdouble wc =
2.0 * M_PI * (filter->cutoff /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble zr = cos (wc), zi = sin (wc);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->cutoff);
}
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
GST_AUDIO_FILTER (filter)->format.rate / 2);
}
}
static void
gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->mode = g_value_get_enum (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_TYPE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->type = g_value_get_int (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_CUTOFF:
GST_BASE_TRANSFORM_LOCK (filter);
filter->cutoff = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_RIPPLE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->ripple = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_POLES:
GST_BASE_TRANSFORM_LOCK (filter);
filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_TYPE:
g_value_set_int (value, filter->type);
break;
case PROP_CUTOFF:
g_value_set_float (value, filter->cutoff);
break;
case PROP_RIPPLE:
g_value_set_float (value, filter->ripple);
break;
case PROP_POLES:
g_value_set_int (value, filter->poles);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_64;
else
ret = FALSE;
filter->have_coeffs = FALSE;
return ret;
}
static inline gdouble
process (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
val += filter->a[i] * ctx->x[j];
j--;
if (j < 0)
j = filter->num_a - 1;
}
for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
val += filter->b[i] * ctx->y[j];
j--;
if (j < 0)
j = filter->num_b - 1;
}
if (ctx->x) {
ctx->x_pos++;
if (ctx->x_pos > filter->num_a - 1)
ctx->x_pos = 0;
ctx->x[ctx->x_pos] = x0;
}
if (ctx->y) {
ctx->y_pos++;
if (ctx->y_pos > filter->num_b - 1)
ctx->y_pos = 0;
ctx->y[ctx->y_pos] = val;
}
return val;
}
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstAudioChebyshevFreqLimit * filter, \
g##ctype * data, guint num_samples) \
{ \
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
gdouble val; \
\
for (i = 0; i < num_samples / channels; i++) { \
for (j = 0; j < channels; j++) { \
val = process (filter, &filter->channels[j], *data); \
*data++ = val; \
} \
} \
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
#undef DEFINE_PROCESS_FUNC
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
if (!filter->have_coeffs)
generate_coefficients (filter);
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static gboolean
gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if
* already existing */
if (channels && filter->channels) {
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
if (ctx->x)
memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
if (ctx->y)
memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
}
}
return TRUE;
}

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@ -1,78 +0,0 @@
/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type())
#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit;
typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass;
typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint);
typedef struct
{
gdouble *x;
gint x_pos;
gdouble *y;
gint y_pos;
} GstAudioChebyshevFreqLimitChannelCtx;
struct _GstAudioChebyshevFreqLimit
{
GstAudioFilter audiofilter;
gint mode;
gint type;
gint poles;
gfloat cutoff;
gfloat ripple;
/* < private > */
GstAudioChebyshevFreqLimitProcessFunc process;
gboolean have_coeffs;
gdouble *a;
gint num_a;
gdouble *b;
gint num_b;
GstAudioChebyshevFreqLimitChannelCtx *channels;
};
struct _GstAudioChebyshevFreqLimitClass
{
GstAudioFilterClass parent;
};
GType gst_audio_chebyshev_freq_limit_get_type (void);
G_END_DECLS
#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */

View file

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

View file

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

View file

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

File diff suppressed because it is too large Load diff

View file

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

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