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Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs alrea...
Original commit message from CVS: * docs/plugins/gst-plugins-ugly-plugins-docs.sgml: * docs/plugins/gst-plugins-ugly-plugins-sections.txt: * ext/a52dec/gsta52dec.c: * ext/amrnb/amrnbdec.c: * ext/amrnb/amrnbenc.c: * ext/amrnb/amrnbparse.c: * ext/lame/gstlame.c: * ext/mad/gstmad.c: * ext/sidplay/gstsiddec.cc: * gst/asfdemux/gstrtspwms.c: * gst/mpegaudioparse/gstxingmux.c: * gst/realmedia/rademux.c: * gst/realmedia/rdtmanager.c: * gst/realmedia/rtspreal.c: * gst/synaesthesia/gstsynaesthesia.c: Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs already). Fix doc-markup: use convinience syntax for examples (produces valid docbook), add several refsec2 when we have several titles. Fix some types.
This commit is contained in:
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16 changed files with 124 additions and 149 deletions
23
ChangeLog
23
ChangeLog
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@ -1,3 +1,26 @@
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2008-06-13 Stefan Kost <ensonic@users.sf.net>
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* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
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* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
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* ext/a52dec/gsta52dec.c:
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* ext/amrnb/amrnbdec.c:
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* ext/amrnb/amrnbenc.c:
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* ext/amrnb/amrnbparse.c:
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* ext/lame/gstlame.c:
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* ext/mad/gstmad.c:
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* ext/sidplay/gstsiddec.cc:
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* gst/asfdemux/gstrtspwms.c:
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* gst/mpegaudioparse/gstxingmux.c:
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* gst/realmedia/rademux.c:
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* gst/realmedia/rdtmanager.c:
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* gst/realmedia/rtspreal.c:
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* gst/synaesthesia/gstsynaesthesia.c:
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Add missing elements to docs. Restore alphabetical order in section
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file. Document mad (it was included in docs already).
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Fix doc-markup: use convinience syntax for examples
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(produces valid docbook), add several refsec2 when we have several
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titles. Fix some types.
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2008-06-13 Stefan Kost <ensonic@users.sf.net>
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* ext/lame/gstlame.c:
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@ -18,6 +18,10 @@
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<xi:include href="xml/element-amrnbparse.xml" />
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<xi:include href="xml/element-lame.xml" />
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<xi:include href="xml/element-mad.xml" />
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<xi:include href="xml/element-rademux.xml" />
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<xi:include href="xml/element-rdtmanager.xml" />
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<xi:include href="xml/element-rtspreal.xml" />
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<xi:include href="xml/element-rtspwms.xml" />
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<xi:include href="xml/element-siddec.xml" />
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<xi:include href="xml/element-synaesthesia.xml" />
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<xi:include href="xml/element-xingmux.xml" />
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@ -54,6 +54,20 @@ GST_TYPE_AMRNBPARSE
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gst_amrnbparse_get_type
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</SECTION>
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<SECTION>
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<FILE>element-lame</FILE>
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<TITLE>lame</TITLE>
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GstLame
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<SUBSECTION Standard>
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GstLameClass
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GST_LAME
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GST_LAME_CLASS
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GST_IS_LAME
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GST_IS_LAME_CLASS
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GST_TYPE_LAME
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gst_lame_get_type
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</SECTION>
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<SECTION>
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<FILE>element-mad</FILE>
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<TITLE>mad</TITLE>
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@ -72,17 +86,18 @@ gst_mad_tag_list_to_id3_tag
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</SECTION>
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<SECTION>
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<FILE>element-lame</FILE>
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<TITLE>lame</TITLE>
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GstLame
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<FILE>element-rademux</FILE>
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<TITLE>rademux</TITLE>
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GstRealAudioDemux
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<SUBSECTION Standard>
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GstLameClass
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GST_LAME
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GST_LAME_CLASS
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GST_IS_LAME
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GST_IS_LAME_CLASS
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GST_TYPE_LAME
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gst_lame_get_type
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GstRealAudioDemuxClass
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GstRealAudioDemuxState
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GST_REAL_AUDIO_DEMUX
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GST_REAL_AUDIO_DEMUX_CLASS
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GST_IS_REAL_AUDIO_DEMUX
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GST_IS_REAL_AUDIO_DEMUX_CLASS
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GST_TYPE_REAL_AUDIO_DEMUX
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gst_real_audio_demux_get_type
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</SECTION>
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<SECTION>
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@ -101,21 +116,6 @@ gst_rdt_manager_get_type
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gst_rdt_manager_plugin_init
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</SECTION>
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<SECTION>
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<FILE>element-rademux</FILE>
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<TITLE>rademux</TITLE>
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GstRealAudioDemux
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<SUBSECTION Standard>
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GstRealAudioDemuxClass
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GstRealAudioDemuxState
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GST_REAL_AUDIO_DEMUX
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GST_REAL_AUDIO_DEMUX_CLASS
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GST_IS_REAL_AUDIO_DEMUX
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GST_IS_REAL_AUDIO_DEMUX_CLASS
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GST_TYPE_REAL_AUDIO_DEMUX
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gst_real_audio_demux_get_type
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</SECTION>
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<SECTION>
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<FILE>element-rtspreal</FILE>
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<TITLE>rtspreal</TITLE>
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@ -20,25 +20,16 @@
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/**
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* SECTION:element-a52dec
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*
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* <refsect2>
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* <para>
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* Dolby Digital (AC-3) audio decoder.
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* </para>
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* </refsect2>
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*
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* <refsect2>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* |[
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* gst-launch dvdreadsrc title=1 ! dvddemux ! a52dec ! audioresample ! audioconvert ! alsasink
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* </programlisting>
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* Play audio track from a dvd.
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* </para>
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* <para>
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* <programlisting>
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* ]| Play audio track from a dvd.
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* |[
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* gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink
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* </programlisting>
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* Decode a stand alone file and play it.
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* </para>
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* ]| Decode a stand alone file and play it.
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* </refsect2>
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*/
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@ -21,17 +21,14 @@
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* SECTION:element-amrnbdec
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* @see_also: #GstAmrnbEnc, #GstAmrnbParse
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*
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* <refsect2>
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* <para>
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* This is an AMR narrowband decoder based on the
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* AMR narrowband decoder based on the
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* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
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* </para>
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*
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* <refsect2>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* |[
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* gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink
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* </programlisting>
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* </para>
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* ]|
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* </refsect2>
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*/
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* SECTION:element-amrnbenc
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* @see_also: #GstAmrnbDec, #GstAmrnbParse
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*
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* <refsect2>
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* <para>
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* This is an AMR narrowband encoder based on the
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* AMR narrowband encoder based on the
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* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
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* </para>
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*
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* <refsect2>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* |[
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* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr
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* </programlisting>
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* </para>
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* Please not that the above stream misses the header, that is needed to play
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* ]|
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* Please note that the above stream misses the header, that is needed to play
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* the stream.
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* </refsect2>
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*/
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@ -21,16 +21,13 @@
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* SECTION:element-amrnbparse
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* @see_also: #GstAmrnbDec, #GstAmrnbEnc
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*
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* AMR narrowband bitstream parser.
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*
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* <refsect2>
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* <para>
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* This is an AMR narrowband parser.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* |[
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* gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink
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* </programlisting>
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* </para>
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* ]|
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* </refsect2>
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*/
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@ -23,61 +23,45 @@
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* SECTION:element-lame
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* @see_also: mad, vorbisenc
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*
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* <refsect2>
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* <para>
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* This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream.
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* Note that <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not
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* a free format, there are licensing and patent issues to take into
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* consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
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* for a royalty free (and often higher quality) alternative.
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* </para>
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*
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* <refsect2>
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* <title>Output sample rate</title>
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* <para>
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* If no fixed output sample rate is negotiated on the element's src pad,
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* the element will choose an optimal sample rate to resample to internally.
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* For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
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* get resampled to 32 KHz. Use filter caps on the src pad to force a
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* particular sample rate.
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* </para>
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* </refsect2>
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* <refsect2>
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* <title>Writing metadata (tags)</title>
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* <para>
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* Whilst the lame encoder element does claim to implement the GstTagSetter
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* interface, it does so only for backwards compatibility reasons. Tag writing
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* has been removed from lame. Use external elements like id3v2mux or apev2mux
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* to add tags to your MP3 streams. The same goes for XING headers: use the
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* xingmux element to add XING headers to your VBR mp3 file.
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* </para>
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* </refsect2>
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* <refsect2>
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* <title>Example pipelines</title>
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* <para>
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* Encode a test sine signal to MP3.
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* </para>
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* <programlisting>
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* |[
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* gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lame ! filesink location=sine.mp3
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* </programlisting>
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* <para>
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* Record from a sound card using ALSA and encode to MP3
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* </para>
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* <programlisting>
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* ]| Encode a test sine signal to MP3.
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* |[
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* gst-launch -v alsasrc ! audioconvert ! lame bitrate=192 ! filesink location=alsasrc.mp3
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* </programlisting>
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* <para>
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* Transcode from a .wav file to MP3 (the id3v2mux element is optional):
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* </para>
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* <programlisting>
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* ]| Record from a sound card using ALSA and encode to MP3
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* |[
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* gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lame bitrate=192 ! id3v2mux ! filesink location=music.mp3
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* </programlisting>
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* <para>
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* Encode Audio CD track 5 to MP3:
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* </para>
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* <programlisting>
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* ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional)
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* |[
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* gst-launch -v cdda://5 ! audioconvert ! lame bitrate=192 ! filesink location=track5.mp3
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* </programlisting>
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* <para>
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* Encode to a fixed sample rate:
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* </para>
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* <programlisting>
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* ]| Encode Audio CD track 5 to MP3
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* |[
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* gst-launch -v audiotestsrc num-buffers=10 ! audio/x-raw-int,rate=44100,channels=1 ! lame bitrate=48 mode=3 ! filesink location=test.mp3
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* </programlisting>
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* ]| Encode to a fixed sample rate
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* </refsect2>
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*
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* Last reviewed on 2007-07-24 (0.10.7)
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-mad
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* @see_also: lame
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*
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* MP3 audio decoder.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch filesrc location=music.mp3 ! mad ! audioconvert ! audioresample ! autoaudiosink
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* ]| Decode the mp3 file and play
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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@ -21,26 +21,20 @@
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/**
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* SECTION:element-siddec
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*
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* <refsect2>
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* <para>
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* This element decodes .sid files to raw audio. .sid files are in fact
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* small Commodore 64 programs that are executed on an emulated 6502 CPU and a
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* MOS 6581 sound chip.
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* </para>
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* <para>
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*
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* This plugin will first load the complete program into memory before starting
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* the emulator and producing output.
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* </para>
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* <para>
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*
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* Seeking is not (and cannot be) implemented.
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* </para>
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* |[
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* gst-launch -v filesrc location=Hawkeye.sid ! siddec ! audioconvert ! alsasink
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* </programlisting>
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* Decode a sid file and play back the audio using alsasink.
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* </para>
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* ]| Decode a sid file and play back the audio using alsasink.
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* </refsect2>
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*
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* Last reviewed on 2006-12-30 (0.10.5)
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/**
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* SECTION:element-rtspwms
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*
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* <refsect2>
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* <para>
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* A WMS RTSP extension
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-07-25 (0.10.14)
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*/
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@ -24,23 +24,19 @@
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/**
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* SECTION:element-xingmux
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*
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* <refsect2>
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* <para>
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* xingmux adds a Xing header to MP3 files. This contains information about the duration and size
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* of the file and a seek table and is very useful for getting an almost correct duration and better
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* seeking on VBR MP3 files.
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* </para>
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* <para>
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*
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* This element will remove any existing Xing, LAME or VBRI headers from the beginning of the file.
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* </para>
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*
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* <refsect2>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* |[
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* gst-launch audiotestsrc num-buffers=1000 ! audioconvert ! lame ! xingmux ! filesink location=test.mp3
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* gst-launch filesrc location=test.mp3 ! xingmux ! filesink location=test2.mp3
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* gst-launch filesrc location=test.mp3 ! mp3parse ! xingmux ! filesink location=test2.mp3
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* </programlisting>
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* </para>
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* ]|
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* </refsect2>
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*/
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@ -20,25 +20,18 @@
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/**
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* SECTION:element-rademux
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*
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* <refsect2>
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* <para>
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* Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
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* </para>
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*
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* <refsect2>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* |[
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* gst-launch filesrc location=interview.ra ! rademux ! ffdec_real_288 ! audioconvert ! audioresample ! alsasink
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* </programlisting>
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* Read a RealAudio file and decode it and output it to the soundcard using
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* ]| Read a RealAudio file and decode it and output it to the soundcard using
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* the ALSA element. The .ra file is assumed to contain RealAudio version 2.
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* </para>
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* <para>
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* <programlisting>
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* |[
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* gst-launch gnomevfssrc location=http://www.example.org/interview.ra ! rademux ! a52dec ! audioconvert ! audioresample ! alsasink
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* </programlisting>
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* Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
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* ]| Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
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* and output it to the soundcard using the ALSA element.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-10-24 (0.10.5)
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|
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@ -43,12 +43,9 @@
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/**
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* SECTION:element-rdtmanager
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* @see_also: GstRtspSrc
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*
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* <refsect2>
|
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* <para>
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* A simple RTP session manager used internally by rtspsrc.
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||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* Last reviewed on 2006-06-20 (0.10.4)
|
||||
*/
|
||||
|
|
|
@ -21,11 +21,7 @@
|
|||
/**
|
||||
* SECTION:element-rtspreal
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
* A RealMedia RTSP extension
|
||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* Last reviewed on 2007-07-25 (0.10.14)
|
||||
*/
|
||||
|
|
|
@ -21,18 +21,14 @@
|
|||
* SECTION:element-synaesthesia
|
||||
* @see_also: goom
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
* Synaesthesia is an audio visualisation element. It creates glitter and
|
||||
* pulsating fog based on the incomming audio signal.
|
||||
* </para>
|
||||
* <refsect2>
|
||||
* <title>Example launch line</title>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* |[
|
||||
* gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ximagesink
|
||||
* gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ffmpegcolorspace ! xvimagesink
|
||||
* </programlisting>
|
||||
* </para>
|
||||
* ]|
|
||||
* </refsect2>
|
||||
*/
|
||||
|
||||
|
|
Loading…
Reference in a new issue