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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.
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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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