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authorStefan Kost <ensonic@users.sf.net>2009-01-28 12:29:42 +0200
committerStefan Kost <ensonic@users.sf.net>2009-01-28 12:32:59 +0200
commita99d3f8769ed3fd1266d5216ecefebfd1bdcf663 (patch)
tree4a5cf5e0f2f44b1f9ccea5344c38ef98f0a92990 /gst/audiofx
parent00fdca0c14eb9a5fe6b8b9f2d5ce2313e3b32f23 (diff)
Update and add documentation for plugins with no deps (gst).
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
Diffstat (limited to 'gst/audiofx')
-rw-r--r--gst/audiofx/audioamplify.c9
-rw-r--r--gst/audiofx/audiochebband.c29
-rw-r--r--gst/audiofx/audiocheblimit.c27
-rw-r--r--gst/audiofx/audiodynamic.c10
-rw-r--r--gst/audiofx/audioecho.c16
-rw-r--r--gst/audiofx/audiofirfilter.c18
-rw-r--r--gst/audiofx/audioiirfilter.c18
-rw-r--r--gst/audiofx/audioinvert.c10
-rw-r--r--gst/audiofx/audiokaraoke.c10
-rw-r--r--gst/audiofx/audiopanorama.c10
-rw-r--r--gst/audiofx/audiowsincband.c15
-rw-r--r--gst/audiofx/audiowsinclimit.c15
12 files changed, 71 insertions, 116 deletions
diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c
index ce4a0042..da5412a4 100644
--- a/gst/audiofx/audioamplify.c
+++ b/gst/audiofx/audioamplify.c
@@ -21,19 +21,16 @@
/**
* SECTION:element-audioamplify
- * @short_description: Amplifies an audio stream with selectable clipping mode
*
- * <refsect2>
* Amplifies an audio stream by a given factor and allows the selection of different clipping modes.
* The difference between the clipping modes is best evaluated by testing.
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * <refsect2>
+ * |[
* gst-launch audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 method=wrap-positive ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c
index bf9c205e..7ecb7956 100644
--- a/gst/audiofx/audiochebband.c
+++ b/gst/audiofx/audiochebband.c
@@ -34,42 +34,35 @@
/**
* SECTION:element-audiochebband
- * @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>
+ *
+ * <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>
+ * </note>
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* 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>
*/
diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c
index b4efbb3a..5d91909e 100644
--- a/gst/audiofx/audiocheblimit.c
+++ b/gst/audiofx/audiocheblimit.c
@@ -30,42 +30,35 @@
/**
* SECTION:element-audiocheblimit
- * @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>
+ * <note><para>
* 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></note>
* <para>
- * <programlisting>
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
* 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>
*/
diff --git a/gst/audiofx/audiodynamic.c b/gst/audiofx/audiodynamic.c
index 7a84fa81..240c270e 100644
--- a/gst/audiofx/audiodynamic.c
+++ b/gst/audiofx/audiodynamic.c
@@ -20,21 +20,19 @@
/**
* SECTION:element-audiodynamic
- * @short_description: Compressor and Expander
*
- * <refsect2>
* This element can act as a compressor or expander. A compressor changes the
* amplitude of all samples above a specific threshold with a specific ratio,
* a expander does the same for all samples below a specific threshold. If
* soft-knee mode is selected the ratio is applied smoothly.
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audioecho.c b/gst/audiofx/audioecho.c
index dd209a23..6676d791 100644
--- a/gst/audiofx/audioecho.c
+++ b/gst/audiofx/audioecho.c
@@ -20,24 +20,22 @@
/**
* SECTION:element-audioecho
+ * @Since: 0.10.12
*
- * <refsect2>
* audioecho adds an echo or (simple) reverb effect to an audio stream. The echo
* delay, intensity and the percentage of feedback can be configured.
- * <para>
+ *
* For getting an echo effect you have to set the delay to a larger value,
* for example 200ms and more. Everything below will result in a simple
* reverb effect, which results in a slightly metallic sounding.
- * </para>
- * <para>
- * <programlisting>
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
* gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
* gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
- *
- * Since: 0.10.12
*/
#ifdef HAVE_CONFIG_H
diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c
index 3ee4d832..df4e2dcb 100644
--- a/gst/audiofx/audiofirfilter.c
+++ b/gst/audiofx/audiofirfilter.c
@@ -21,31 +21,27 @@
/**
* SECTION:element-audiofirfilter
- * @short_description: Generic audio FIR filter
*
- * <refsect2>
- * <para>
* audiofirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Finite_impulse_response">FIR filter</ulink>. Before usage the
* "kernel" property has to be set to the filter kernel that should be
* used and the "latency" property has to be set to the latency (in samples)
* that is introduced by the filter kernel. Setting a latency of n samples
* will lead to the first n samples being dropped from the output and
* n samples added to the end.
- * </para>
- * <para>
+ *
* The filter kernel describes the impulse response of the filter. To
* calculate the frequency response of the filter you have to calculate
* the Fourier Transform of the impulse response.
- * </para>
- * <para>
+ *
* To change the filter kernel whenever the sampling rate changes the
* "rate-changed" signal can be used. This should be done for most
* FIR filters as they're depending on the sampling rate.
- * </para>
+ *
+ * <refsect2>
* <title>Example application</title>
- * <para>
- * <include xmlns="http://www.w3.org/2003/XInclude" href="element-firfilter-example.xml" />
- * </para>
+ * |[
+ * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/audiofx/firfilter-example.c" />
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c
index 76112c6f..1f063122 100644
--- a/gst/audiofx/audioiirfilter.c
+++ b/gst/audiofx/audioiirfilter.c
@@ -21,27 +21,23 @@
/**
* SECTION:element-audioiirfilter
- * @short_description: Generic audio IIR filter
*
- * <refsect2>
- * <para>
* audioiirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Infinite_impulse_response">IIR filter</ulink>. Before usage the
* "a" and "b" properties have to be set to the filter coefficients that
* should be used.
- * </para>
- * <para>
+ *
* The filter coefficients describe the numerator and denominator of the
* transfer function.
- * </para>
- * <para>
+ *
* To change the filter coefficients whenever the sampling rate changes the
* "rate-changed" signal can be used. This should be done for most
* IIR filters as they're depending on the sampling rate.
- * </para>
+ *
+ * <refsect2>
* <title>Example application</title>
- * <para>
- * <include xmlns="http://www.w3.org/2003/XInclude" href="element-iirfilter-example.xml" />
- * </para>
+ * |[
+ * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/audiofx/iirfilter-example.c" />
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audioinvert.c b/gst/audiofx/audioinvert.c
index 188793f5..a6911697 100644
--- a/gst/audiofx/audioinvert.c
+++ b/gst/audiofx/audioinvert.c
@@ -21,20 +21,18 @@
/**
* SECTION:element-audioinvert
- * @short_description: Swaps upper and lower half of audio samples
*
- * <refsect2>
* Swaps upper and lower half of audio samples. Mixing an inverted sample on top of
* the original with a slight delay can produce effects that sound like resonance.
* Creating a stereo sample from a mono source, with one channel inverted produces wide-stereo sounds.
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch audiotestsrc wave=saw ! audioinvert invert=0.4 ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert invert=0.4 ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audioinvert invert=0.4 ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audiokaraoke.c b/gst/audiofx/audiokaraoke.c
index ec505681..fe34971e 100644
--- a/gst/audiofx/audiokaraoke.c
+++ b/gst/audiofx/audiokaraoke.c
@@ -20,17 +20,15 @@
/**
* SECTION:element-audiokaraoke
- * @short_description: Voice removal element
*
- * <refsect2>
* Remove the voice from audio by filtering the center channel.
* This plugin is useful for karaoke applications.
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audiopanorama.c b/gst/audiofx/audiopanorama.c
index e38f10d2..3f57648d 100644
--- a/gst/audiofx/audiopanorama.c
+++ b/gst/audiofx/audiopanorama.c
@@ -21,20 +21,18 @@
/**
* SECTION:element-audiopanorama
- * @short_description: audio stereo pan effect
*
- * <refsect2>
* Stereo panorama effect with controllable pan position. One can choose between the default psychoacoustic panning method,
* which keeps the same perceived loudness, and a simple panning method that just controls the volume on one channel.
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch audiotestsrc wave=saw ! audiopanorama panorama=-1.00 ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiopanorama panorama=-1.00 ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audiopanorama panorama=-1.00 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audiopanorama method=simple panorama=-0.50 ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c
index 69bf5c19..70c85b48 100644
--- a/gst/audiofx/audiowsincband.c
+++ b/gst/audiofx/audiowsincband.c
@@ -30,28 +30,23 @@
/**
* SECTION:element-audiowsincband
- * @short_description: Windowed Sinc band pass and band reject filter
*
- * <refsect2>
- * <para>
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The length parameter controls the rolloff, the window parameter
* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
* worse stopband attenuation, the other way around for the Blackman window.
- * </para>
- * <para>
+ *
* This element has the advantage over the Chebyshev bandpass and bandreject filter that it has
* a much better rolloff when using a larger kernel size and almost linear phase. The only
* disadvantage is the much slower execution time with larger kernels.
- * </para>
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiosincband mode=band-pass lower-frequency=3000 upper-frequency=10000 length=501 window=blackman ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsincband mode=band-reject lower-frequency=59 upper-frequency=61 length=10001 window=hamming ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiowsincband mode=band-pass lower-frequency=1000 upper-frequency=2000 length=31 ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/
diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c
index 1f33ad2d..73bdbe50 100644
--- a/gst/audiofx/audiowsinclimit.c
+++ b/gst/audiofx/audiowsinclimit.c
@@ -30,28 +30,23 @@
/**
* SECTION:element-audiowsinclimit
- * @short_description: Windowed Sinc 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 length parameter controls the rolloff, the window parameter
* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
* worse stopband attenuation, the other way around for the Blackman window.
- * </para>
- * <para>
+ *
* This element has the advantage over the Chebyshev lowpass and highpass filter that it has
* a much better rolloff when using a larger kernel size and almost linear phase. The only
* disadvantage is the much slower execution time with larger kernels.
- * </para>
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsinclimit mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * ]|
* </refsect2>
*/