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add documentation for the new RTP modules

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git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@731 fefdeb5f-60dc-0310-8127-8f9354f1896f
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Lennart Poettering 2006-04-16 13:34:09 +00:00
parent 2f3fa42ca6
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doc/modules.html.in
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@ -283,7 +283,7 @@ module and point your browser to <a
href="http://localhost:4714/">http://localhost:4714/</a>. This module takes the same arguments
as <tt>module-cli-protocol-tcp</tt>.</p>
<h2>Miscellaneous</h2>
<h2>X Window System</h2>
<h3>module-x11-bell</h3>
@ -315,6 +315,94 @@ and import credential data from/to the X11 display.</p>
cookie to store in the X11 display. If ommited the cookie of an
already loaded protocol module is used.</td></tr> </table>
<h2>Volume Control</h2>
<h3>module-mmkbd-evdev</h3>
<p>Adjust the volume of a sink when the special multimedia buttons of modern keyboards are pressed.</p>
<table>
<tr><td><tt>device=</tt></td><td>Linux input device ("<tt>evdev</tt>", defaults to <tt>/dev/input/event0</tt>)</td></tr>
<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
</table>
<h3>module-lirc</h3>
<p>Adjust the volume of a sink when the volume buttons of an infrared remote control are pressed (through LIRC).</p>
<table>
<tr><td><tt>config=</tt></td><td>The LIRC configuration file</td></tr>
<tr><td><tt>appname=</tt></td><td>The application name to pass to LIRC (defaults to <tt>polypaudio</tt>)</td></tr>
<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
</table>
<a name="rtp"/>
<h2>RTP/SDP/SAP Transport</h2>
<p>Polypaudio can stream audio data to an IP multicast group via the
standard protocols <a
href="http://en.wikipedia.org/wiki/Real-time_Transport_Protocol">RTP</a>,
<a
href="http://en.wikipedia.org/wiki/Session_Announcement_Protocol">SAP</a>
and <a
href="http://en.wikipedia.org/wiki/Session_Description_Protocol">SDP</a>
(RFC3550, RFC3551, RFC2327, RFC2327). This can be used for multiple
different purposes: for sharing a single microphone on multiple
computers on the local LAN, for streaming music from a single
controlling PC to multiple PCs with speakers or to implement a simple
"always-on" teleconferencing solution.</p>
<p>The current implementation is designed to be used exlusively in
local area networks, though Internet multicasting is theoretically
supported. Only uncompressed audio is supported, hence you won't be
able to multicast more than a few streams at the same time over a
standard LAN.</p>
<p>Polypaudio implements both a sender and a reciever for RTP
traffic. The sender announces itself via SAP/SDP on the same multicast
group as it sends the RTP data to. The reciever picks up the SAP/SDP
announcements and creates a playback stream for each
session. Alternatively you can use any RTP capable client to
recieve and play back the RTP data (such as <tt>mplayer</tt>).</p>
<h3>module-rtp-send</h3>
<p>This is the sender side of the RTP/SDP/SAP implementation. It reads
audio data from an existing source and forwards it to the network
encapsulated in RTP. In addition it sends SAP packets with an SDP
session description.</p>
<p>In combination with the monitor source of <tt>module-null-sink</tt>
you can use this module to create an RTP sink.</p>
<table>
<tr><td><tt>source=</tt></td><td>The source to read the audio data from. If ommited defaults to the default source.</td></tr>
<tr><td><tt>format=, rate=, channels=</tt></td><td>Sample format to use, defaults to the source's.</td></tr>
<tr><td><tt>destination=</tt></td><td>Destination multicast group for both RTP and SAP packets, defaults to <tt>224.0.0.56</tt></td></tr>
<tr><td><tt>port=</tt></td><td>Destination port number of the RTP
traffic. If ommited defaults to a randomly chosen even port
number. Please keep in mind that the RFC suggests to use only even
port numbers for RTP traffic.</td></tr>
<tr><td><tt>mtu=</tt></td><td>Maximum payload size for RTP packets. If ommited defaults to 1280</td></tr>
<tr><td><tt>loop=</tt></td><td>Takes a boolean value, specifying whether locally generated RTP traffic should be looped back to the local host. Disabled by default.</td></tr>
</table>
<h3>module-rtp-recv</h3>
<p>This is the reciever side of the RTP/SDP/SAP implementation. It
picks up SAP session announcements and creates an RTP playback stream
for each.</p>
<p>In combination with <tt>module-null-sink</tt> you can use this
module to create an RTP source.</p>
<table>
<tr><td><tt>sink=</tt></td><td>The sink to connect to. If ommited defaults to the default sink.</td></tr>
<tr><td><tt>sap_address=</tt></td><td>The multicast group to join for SAP announcements, defaults to <tt>224.0.0.56</tt>.</td></tr>
</table>
<h2>Miscellaneous</h2>
<h3>module-sine</h3>
<p>Creates a sink input and generates a sine waveform stream.</p>
@ -360,24 +448,6 @@ already loaded protocol module is used.</td></tr> </table>
<p>Publish all local sinks/sources using mDNS Zeroconf.</p>
<h3>module-mmkbd-evdev</h3>
<p>Adjust the volume of a sink when the special multimedia buttons of modern keyboards are pressed.</p>
<table>
<tr><td><tt>device=</tt></td><td>Linux input device ("<tt>evdev</tt>", defaults to <tt>/dev/input/event0</tt>)</td></tr>
<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
</table>
<h3>module-lirc</h3>
<p>Adjust the volume of a sink when the volume buttons of an infrared remote control are pressed (through LIRC).</p>
<table>
<tr><td><tt>config=</tt></td><td>The LIRC configuration file</td></tr>
<tr><td><tt>appname=</tt></td><td>The application name to pass to LIRC (defaults to <tt>polypaudio</tt>)</td></tr>
<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
</table>
<hr/>
<address class="grey">Lennart Poettering &lt;@PACKAGE_BUGREPORT@&gt;, April 2006</address>