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494 lines
22 KiB
XML
494 lines
22 KiB
XML
<?xml version="1.0" encoding="iso-8859-1"?> <!-- -*-html-helper-*- -->
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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd">
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<html xmlns="http://www.w3.org/1999/xhtml">
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<head>
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<title>polypaudio: Loadable Modules</title>
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<link rel="stylesheet" type="text/css" href="style.css" />
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</head>
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<body>
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<h1>Loadable Modules</h1>
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<p>The following loadable modules are provided with the <tt>polypaudio</tt> distribution:</p>
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<h2>Device Drivers</h2>
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<p>All device driver modules support the following parameters:</p>
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<table>
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<tr><td><tt>format=</tt></td><td>The sample format (one of <tt>u8</tt>, <tt>s16</tt>, <tt>s16le</tt>, <tt>s16le</tt>, <tt>float32</tt>, <tt>float32be</tt>, <tt>float32le</tt>, <tt>alaw</tt>, <tt>ulaw</tt>) (defaults to <tt>s16</tt>)</td></tr>
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<tr><td><tt>rate=</tt></td><td>The sample rate (defaults to 44100)</td></tr>
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<tr><td><tt>channels=</tt></td><td>Audio channels (defaults to 2)</td></tr>
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<tr><td><tt>sink_name=</tt>, <tt>source_name=</tt></td><td>Name for the sink (resp. source)</td></tr>
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<tr><td><tt>channel_map=</tt></td><td>Channel map. A list of
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comma-seperated channel names. The currently defined channel names
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are: <tt>left</tt>, <tt>right</tt>, <tt>mono</tt>, <tt>center</tt>,
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<tt>front-left</tt>, <tt>front-right</tt>, <tt>front-center</tt>,
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<tt>rear-center</tt>, <tt>rear-left</tt>, <tt>rear-right</tt>,
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<tt>lfe</tt>, <tt>subwoofer</tt>, <tt>front-left-of-center</tt>,
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<tt>front-right-of-center</tt>, <tt>side-left</tt>,
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<tt>side-right</tt>, <tt>aux0</tt>, <tt>aux1</tt> to <tt>aux15</tt>,
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<tt>top-center</tt>, <tt>top-front-left</tt>,
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<tt>top-front-right</tt>, <tt>top-front-center</tt>,
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<tt>top-rear-left</tt>, <tt>top-rear-right</tt>,
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<tt>top-rear-center</tt>, (Default depends on the number of channels
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and the driver)</td></tr> </table>
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<h3>module-pipe-sink</h3>
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<p>Provides a simple test sink that writes the audio data to a FIFO
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special file in the file system. The sink name defaults to <tt>pipe_output</tt>.</p>
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<p>The following option is supported:</p>
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<table>
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<tr><td><tt>file=</tt></td><td>The name of the FIFO special file to use. (defaults to: <tt>/tmp/music.output</tt>)</td></tr>
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</table>
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<h3>module-pipe-source</h3>
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<p>Provides a simple test source that reads the audio data from a FIFO
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special file in the file system. The source name defaults to <tt>pipe_input</tt>.</p>
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<p>The following option is supported:</p>
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<table>
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<tr><td><tt>file=</tt></td><td>The name of the FIFO special file to use. (defaults to: <tt>/tmp/music.input</tt>)</td></tr>
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</table>
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<h3>module-null-sink</h3>
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<p>Provides a simple null sink. All data written to this sink is silently dropped. This sink is clocked using the system time.</p>
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<p>This module doesn't support any special parameters</p>
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<a name="module-alsa-sink"/>
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<h3>module-alsa-sink</h3>
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<p>Provides a playback sink for devices supported by the <a href="http://www.alsa-project.org/">Advanced Linux
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Sound Architecture</a> (ALSA). The sink name defaults to <tt>alsa_output</tt>.</p>
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<p>In addition to the general device driver options described above this module supports:</p>
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<table>
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<tr><td><tt>device=</tt></td><td>The ALSA device to use. (defaults to "plughw:0,0")</td></tr>
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<tr><td><tt>fragments=</tt></td><td>The desired fragments when opening the device. (defaults to 12)</td></tr>
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<tr><td><tt>fragment_size=</tt></td><td>The desired fragment size in bytes when opening the device (defaults to 1024)</td></tr>
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</table>
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<h3>module-alsa-source</h3>
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<p>Provides a recording source for devices supported by the Advanced
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Linux Sound Architecture (ALSA). The source name defaults to <tt>alsa_input</tt>.</p>
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<p>This module supports <tt>device=</tt>, <tt>fragments=</tt> and <tt>fragment_size=</tt> arguments the same way as <a href="#module-alsa-sink"><tt>module-alsa-sink</tt></a>.</p>
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<a name="module-oss"/>
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<h3>module-oss</h3>
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<p>Provides both a sink and a source for playback, resp. recording on
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<a href="http://www.opensound.com">Open Sound System</a> (OSS) compatible devices.</p>
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<p>This module supports <tt>device=</tt> (which defaults to <tt>/dev/dsp</tt>), <tt>fragments=</tt> and <tt>fragment_size=</tt> arguments the same way as <a href="#module-alsa-sink"><tt>module-alsa-sink</tt></a>.</p>
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<p>In addition this module supports the following options:</p>
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<table>
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<tr><td><tt>record=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the recording on this device. (defaults: to 1)</td></tr>
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<tr><td><tt>playback=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the playback on this device. (defaults: to 1)</td></tr>
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</table>
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<p>The sink name (resp. source name) defaults to <tt>oss_output</tt> (resp. <tt>oss_input</tt>).</p>
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<h3>module-oss-mmap</h3>
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<p>Similar to <tt>module-oss</tt> but uses memory mapped
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(<tt>mmap()</tt>) access to the input/output buffers of the audio
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device. This provides better latency behaviour but is not as
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compatible as <tt>module-oss</tt>.</p>
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<p>This module accepts exactly the same arguments as <a href="#module-oss"><tt>module-oss</tt></a>.</p>
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<h3>module-solaris</h3>
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<P>Provides a sink and source for the Solaris audio device.</p>
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<p>In addition to the general device driver options described above this module supports:</p>
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<table>
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<tr><td><tt>record=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the recording on this device. (defaults: to 1)</td></tr>
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<tr><td><tt>playback=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the playback on this device. (defaults: to 1)</td></tr>
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<tr><td><tt>buffer_size=</tt></td><td>Record buffer size</td></tr>
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</table>
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<h3>module-waveout</h3>
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<P>Provides a sink and source for the Win32 audio device.</p>
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<p>This module supports all arguments thet <tt>module-oss</tt> supports except <tt>device=</tt>.</p>
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<a name="module-combine"/>
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<h3>module-combine</h3>
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<p>This combines two or more sinks into one. A new virtual sink is
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allocated. All data written to it is forwarded to all connected
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sinks. In aequidistant intervals the sample rates of the output sinks
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is recalculated: i.e. even when the sinks' crystals deviate (which is
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normally the case) output appears synchronously to the human ear. The
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resampling required for this may be very CPU intensive.</p>
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<table>
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<tr><td><tt>sink_name=</tt></td><td>The name for the combined sink. (defaults to <tt>combined</tt>)</td></tr>
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<tr><td><tt>master=</tt></td><td>The name of the first sink to link into the combined think. The sample rate/type is taken from this sink.</td></tr>
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<tr><td><tt>slaves=</tt></td><td>Name of additional sinks to link into the combined think, seperated by commas.</td></tr>
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<tr><td><tt>adjust_time=</tt></td><td>Time in seconds when to readjust the sample rate of all sinks. (defaults to 20)</td></tr>
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<tr><td><tt>resample_method=</tt></td><td>Resampling algorithm to
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use. See <tt>libsamplerate</tt>'s documentation for more
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information. Use one of <tt>sinc-best-quality</tt>,
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<tt>sinc-medium-quality</tt>, <tt>sinc-fastest</tt>,
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<tt>zero-order-hold</tt>, <tt>linear</tt>. If the default happens to
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be to slow on your machine try using <tt>zero-order-hold</tt>. This
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will decrease output quality however. (defaults to
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<tt>sinc-fastest</tt>)</td></tr> </table>
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<h3>module-tunnel-{sink,source}</h3>
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<p>Tunnel a remote sink/source to a local "ghost"
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sink/source. Requires a running polypaudio daemon on the remote server
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with <tt>module-native-protocol-tcp</tt> loaded. It's probably a
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better idea to connect to the remote sink/source directly since some
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buffer control is lost through this tunneling.</p>
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<table>
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<tr><td><tt>server=</tt></td><td>The server to connect to</td></tr>
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<tr><td><tt>source=</tt></td><td>The source on the remote server. Only available for <tt>module-tunnel-source</tt>.</td></tr>
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<tr><td><tt>sink=</tt></td><td>The sink on the remote server. Only available for <tt>module-tunnel-sink</tt>.</td></tr>
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<tr><td><tt>cookie=</tt></td><td>The authentication cookie file to use.</td></tr>
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</table>
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<h3>module-esound-sink</h3>
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<p>Create a playback sink using an <a href="http://www.tux.org/~ricdude/apps.html">ESOUND</a> server as backend. Whenever you can, try to omit this
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module since it has many disadvantages including bad latency
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and even worse latency measurement. </p>
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<table>
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<tr><td><tt>server=</tt></td><td>The server to connect to</td></tr>
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<tr><td><tt>cookie=</tt></td><td>The authentication cookie file to use.</td></tr>
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</table>
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<h2>Protocols</h2>
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<a name="module-cli"/>
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<h3>module-cli</h3>
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<p>Provides the user with a simple command line interface on the
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controlling TTY of the daemon. This module may not be loaded more than
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once.</p>
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<p>For an explanation of the simple command line language used by this
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module see <a href="cli.html"><tt>cli.html</tt></a>.
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<p>This module doesn't accept any arguments.</p>
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<a name="module-cli-protocol-unix"/>
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<a name="module-cli-protocol-tcp"/>
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<a name="module-cli-protocol"/>
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<h3>module-cli-protocol-{unix,tcp}</h3>
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<p>An implemenation of a simple command line based protocol for
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controlling the <tt>polypaudio</tt> daemon. If loaded, the user may
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connect with tools like <tt>netcat</tt>, <tt>telnet</tt> or
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<a href="http://0pointer.de/lennart/projects/bidilink/"><tt>bidilink</tt></a> to the listening sockets and execute commands the
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same way as with <tt>module-cli</tt>.</p>
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<p><b>Beware!</b> Users are not authenticated when connecting to this
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service.</p>
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<p>This module exists in two versions: with the suffix <tt>-unix</tt>
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the service will listen on an UNIX domain socket in the local file
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system. With the suffix <tt>-tcp</tt> it will listen on a network
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transparent TCP/IP socket. (Both IPv6 and IPv4 - if available)</p>
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<p>This module supports the following options:</p>
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<table>
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<tr><td><tt>port=</tt></td><td>(only for <tt>-tcp</tt>) The port number to listen on (defaults to 4712)</td></tr>
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<tr><td><tt>loopback=</tt></td><td>(only for <tt>-tcp</tt>) Accepts
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a numerical binary value. If 1 the socket is bound to the loopback
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device, i.e. not publicly accessible. (defaults to 1)</td></tr>
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<tr><td><tt>listen=</tt></td><td>(only for <tt>-tcp</tt>) The IP address to listen on. If specified, supersedes the value specified in <tt>loopback=</tt></td></tr>
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<tr><td><tt>socket=</tt></td><td>(only for <tt>-unix</tt>) The UNIX socket name (defaults to <tt>/tmp/polypaudio/cli</tt>)</td></tr>
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</table>
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<h3>module-simple-protocol-{unix,tcp}</h3>
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<p>An implementation of a simple protocol which allows playback by using
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simple tools like <tt>netcat</tt>. Just connect to the listening
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socket of this module and write the audio data to it, or read it from
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it for playback, resp. recording.</p>
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<p><b>Beware!</b> Users are not authenticated when connecting to this
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service.</p>
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<p>See <tt>module-cli-protocol-{unix,tcp}</tt> for more information
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about the two possible suffixes of this module.</p>
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<p>In addition to the options supported by <a href="module-cli-protocol"><tt>module-cli-protocol-*</tt></a>, this module supports:</p>
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<table>
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<tr><td><tt>rate=</tt>, <tt>format=</tt>, <tt>channels=</tt></td><td>Sample format for streams connecting to this service.</td></tr>
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<tr><td><tt>playback=</tt>, <tt>record=</tt></td><td>Enable/disable playback/recording</td></tr>
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<tr><td><tt>sink=</tt>, <tt>source=</tt></td><td>Specify the sink/source this service connects to</td></tr>
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</table>
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<h3>module-esound-protocol-{unix,tcp}</h3>
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<p>An implemenation of a protocol compatible with the <a
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href="http://www.tux.org/~ricdude/EsounD.html">Enlightened Sound
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Daemon</a> (ESOUND, <tt>esd</tt>). When you load this module you may
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access the <tt>polypaudio</tt> daemon with tools like <tt>esdcat</tt>,
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<tt>esdrec</tt> or even <tt>esdctl</tt>. Many applications, such as
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XMMS, include support for this protocol.</p>
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<p>See <tt>module-cli-protocol-{unix,tcp}</tt> for more information
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about the two possible suffixes of this module.</p>
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<p>In addition to the options supported by <a href="module-cli-protocol"><tt>module-cli-protocol-*</tt></a>, this module supports:</p>
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<table>
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<tr><td><tt>sink=</tt>, <tt>source=</tt></td><td>Specify the sink/source this service connects to</td></tr>
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<tr><td><tt>auth-anonymous=</tt></td><td>If set to 1 no authentication is required to connect to the service</td></tr>
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<tr><td><tt>cookie=</tt></td><td>Name of the cookie file for authentication purposes</td></tr>
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</table>
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<p>This implementation misses some features the original ESOUND has: e.g. there is no sample cache yet. However: XMMS works fine.</p>
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<h3>module-native-protocol-{unix,tcp}</h3>
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<p>The native protocol of <tt>polypaudio</tt>.</p>
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<p>See <tt>module-cli-protocol-{unix,tcp}</tt> for more information
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about the two possible suffixes of this module.</p>
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<p>In addition to the options supported by <a href="module-cli-protocol"><tt>module-cli-protocol-*</tt></a>, this module supports:</p>
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<table>
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<tr><td><tt>auth-anonymous=</tt></td><td>If set to 1 no authentication is required to connect to the service</td></tr>
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<tr><td><tt>auth-group=</tt></td><td>(only for <tt>-unix</tt>): members of the specified unix group may access the server without further auhentication.</td></tr>
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<tr><td><tt>cookie=</tt></td><td>Name of the cookie file for authentication purposes</td></tr>
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</table>
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<h3>module-native-protocol-fd</h3>
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<p>This is used internally when auto spawning a new daemon. Don't use it directly.</p>
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<h3>module-http-protocol-tcp</h3>
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<p>A proof-of-concept HTTP module, which can be used to introspect
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the current status of the polypaudio daemon using HTTP. Just load this
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module and point your browser to <a
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href="http://localhost:4714/">http://localhost:4714/</a>. This module takes the same arguments
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as <tt>module-cli-protocol-tcp</tt>.</p>
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<h2>X Window System</h2>
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<h3>module-x11-bell</h3>
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<p>Intercepts X11 bell events and plays a sample from the sample cache on each occurence.</p>
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<table>
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<tr><td><tt>display=</tt></td><td>X11 display to connect to. If ommited defaults to the value of <tt>$DISPLAY</tt></td></tr>
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<tr><td><tt>sample=</tt></td><td>The sample to play. If ommited defaults to <tt>x11-bell</tt>.</td></tr>
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<tr><td><tt>sink=</tt></td><td>Name of the sink to play the sample on. If ommited defaults to the default sink.</td></tr>
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</table>
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<h3>module-x11-publish</h3>
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<p>Publishes the access credentials to the Polypaudio server in the
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X11 root window. The following properties are used:
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<tt>POLYP_SERVER</tt>, <tt>POYLP_SINK</tt>, <tt>POLYP_SOURCE</tt>,
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<tt>POLYP_COOKIE</tt>. This is very useful when using SSH or any other
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remote login tool for logging into other machines and getting audio
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playback to your local speakers. The Polypaudio client libraries make
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use of this data automatically. Instead of using this module you may
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use the tool <tt>pax11publish</tt> which may be used to access, modify
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and import credential data from/to the X11 display.</p>
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<table>
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<tr><td><tt>display=</tt></td><td>X11 display to connect to. If ommited defaults to the value of <tt>$DISPLAY</tt></td></tr>
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<tr><td><tt>sink=</tt></td><td>Name of the default sink. If ommited this property isn't stored in the X11 display.</td></tr>
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<tr><td><tt>source=</tt></td><td>Name of the default source. If ommited this property isn't stored in the X11 display.</td></tr>
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<tr><td><tt>cookie=</tt></td><td>Name of the cookie file of the
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cookie to store in the X11 display. If ommited the cookie of an
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already loaded protocol module is used.</td></tr> </table>
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<h2>Volume Control</h2>
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<h3>module-mmkbd-evdev</h3>
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<p>Adjust the volume of a sink when the special multimedia buttons of modern keyboards are pressed.</p>
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<table>
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<tr><td><tt>device=</tt></td><td>Linux input device ("<tt>evdev</tt>", defaults to <tt>/dev/input/event0</tt>)</td></tr>
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<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
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</table>
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<h3>module-lirc</h3>
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<p>Adjust the volume of a sink when the volume buttons of an infrared remote control are pressed (through LIRC).</p>
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<table>
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<tr><td><tt>config=</tt></td><td>The LIRC configuration file</td></tr>
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<tr><td><tt>appname=</tt></td><td>The application name to pass to LIRC (defaults to <tt>polypaudio</tt>)</td></tr>
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<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
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</table>
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<a name="rtp"/>
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<h2>RTP/SDP/SAP Transport</h2>
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<p>Polypaudio can stream audio data to an IP multicast group via the
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standard protocols <a
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href="http://en.wikipedia.org/wiki/Real-time_Transport_Protocol">RTP</a>,
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<a
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href="http://en.wikipedia.org/wiki/Session_Announcement_Protocol">SAP</a>
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and <a
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href="http://en.wikipedia.org/wiki/Session_Description_Protocol">SDP</a>
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(RFC3550, RFC3551, RFC2327, RFC2327). This can be used for multiple
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|
different purposes: for sharing a single microphone on multiple
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|
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>
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|
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<p>The current implementation is designed to be used exlusively in
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|
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>
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|
|
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<h3>module-rtp-send</h3>
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|
|
|
<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>JACK Connectivity</h2>
|
|
|
|
<p>Polypaudio can be hooked up to a <a
|
|
href="http://jackit.sourceforge.net/">JACK Audio Connection Kit</a> server which is a specialized sound server used for professional audio production on Unix/Linux. Both a
|
|
Polypaudio sink and a source are available. For each channel a port is
|
|
created in the JACK server.</p>
|
|
|
|
<h3>module-jack-sink</h3>
|
|
|
|
<p>This module implements a Polypaudio sink that connects to JACK and registers as many output ports as requested.</p>
|
|
|
|
<table>
|
|
<tr><td><tt>sink_name=</tt></td><td>The name for the Polypaudio sink. If ommited defaults to <tt>jack_out</tt>.</td></tr>
|
|
<tr><td><tt>server_name=</tt></td><td>The JACK server to connect to. If ommited defaults to the default server.</td></tr>
|
|
<tr><td><tt>client_name=</tt></td><td>The client name to tell the JACK server. If ommited defaults to <tt>polypaudio</tt>.</td></tr>
|
|
<tr><td><tt>channels=</tt></td><td>Number of channels to register. If ommited defaults to the number of physical playback ports of the JACK server.</td></tr>
|
|
<tr><td><tt>connect=</tt></td><td>Takes a boolean value. If enabled (the default) Polypaudio will try to connect its ports to the physicial playback ports of the JACK server</td></tr>
|
|
</table>
|
|
|
|
<h3>module-jack-source</h3>
|
|
|
|
<p>This module implements a Polypaudio source that connects to JACK
|
|
and registers as many input ports as requested. Takes the same
|
|
arguments as <tt>module-jack-sink</tt>, except for <tt>sink_name</tt>
|
|
which is replaced by <tt>source_name</tt> (with a default of <tt>jack_in</tt>) for obvious reasons.</p>
|
|
|
|
<h2>Miscellaneous</h2>
|
|
|
|
<h3>module-sine</h3>
|
|
|
|
<p>Creates a sink input and generates a sine waveform stream.</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>frequency=</tt></td><td>The frequency to generate in Hertz. Defaults to 440.</td></tr>
|
|
</table>
|
|
|
|
<h3>module-esound-compat-spawnfd</h3>
|
|
|
|
<p>This is a compatibility module for <tt>libesd</tt> based autospawning of polypaudio. Don't use it directly.</p>
|
|
|
|
<h3>module-esound-compat-spawnpid</h3>
|
|
|
|
<p>This is a compatibility module for <tt>libesd</tt> based autospawning of polypaudio. Don't use it directly.</p>
|
|
|
|
<h3>module-match</h3>
|
|
|
|
<p>Adjust the volume of a playback stream automatically based on its name.</p>
|
|
|
|
<table>
|
|
<tr><td><tt>table=</tt></td><td>The regular expression matching table file to use</td></tr>
|
|
</table>
|
|
|
|
<p>The table file should contain a regexp and volume on each line, seperated by spaces. An example:</p>
|
|
|
|
<pre>
|
|
^sample: 25
|
|
</pre>
|
|
|
|
<p>The volumes of all streams with titles starting with <tt>sample:</tt> are automatically set to 25. (FYI: All sample cache streams start with <tt>sample:</tt>)</p>
|
|
|
|
<h3>module-detect</h3>
|
|
|
|
<p>Automatically detect the available sound hardware and load modules for it. Supports OSS, ALSA, Solaris and Win32 output drivers.
|
|
|
|
<table>
|
|
<tr><td><tt>just-one=</tt></td><td>If set to <tt>1</tt> the module will only try to load a single sink/source and than stop.</td></tr>
|
|
</table>
|
|
|
|
<h3>module-zeroconf-publish</h3>
|
|
|
|
<p>Publish all local sinks/sources using mDNS Zeroconf.</p>
|
|
|
|
|
|
<hr/>
|
|
<address class="grey">Lennart Poettering <@PACKAGE_BUGREPORT@>, April 2006</address>
|
|
<div class="grey"><i>$Id$</i></div>
|
|
</body> </html>
|