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245 lines
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245 lines
12 KiB
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<?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: FAQ</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>Frequently Asked Questions</h1>
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<ol>
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<li><p><b>How does Polypaudio compare with ESOUND/aRts/NAS?</b></p>
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<p>Polypaudio is sound daemon similar to ESOUND and NAS, but much more
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powerful. aRts is a realtime-synthesizer-cum-sound-server, i.e. it
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does much more than Polypaudio. However, I believe that Polypaudio
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does what it does much better than any other free sound server.</p>
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</li>
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<li><p><b>What about ESOUND compatibility?</b></p>
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<p>Polypaudio is a drop in replacement for ESOUND. That means: you can
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load a esound compatibility module which implements an ESOUND
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compatible protocol which allows you to use most of the classic ESOUND
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compatible programs (including the command line programs like
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<tt>esdcat</tt>).</p>
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</li>
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<li><p><b>Is Polypaudio a GNOME program?</b></p>
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<p>No, Polypaudio has no dependency on GNOME/GTK/GLIB. All it requires
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is a UNIX-like operating system and very few dependency
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libraries. However, the accompanying GUI tools are writen with
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gtkmm, i.e. require both GLIB and GTK.</p></li>
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<li><p><b>Can I integrate Polypaudio in my GLIB/GTK/GNOME application?</b></p>
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<p>Yes! Polypaudio comes with a GLIB main loop adapter. You can embed
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both the client library and the daemon (!) into your GLIB based
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application.</p></li>
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<li><p><b>Can I integrate Polypaudio in my Qt/KDE application?</b></p>
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<p>Yes! Polypaudio uses a main loop abstraction layer that allows you
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to integrate Polypaudio in any program that supports main
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loops. Unfortunately there is no adapter for Qt publicly available yet.</p></li>
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<li><p><b>I want to write a new driver for Polypaudio, are there any docs?</b></p>
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<p>Currently, only the client API is documented with doxygen. Read
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the source and base your work on a simple module like
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<tt>module-pipe-sink</tt>.</p></li>
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<li><p><b>What about compatibility with NAS?</b></p>
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<p>Is not available (yet?). It is doable, but noone has implemented it yet.</p></li>
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<li><p><b>What about compatibility with aRts?</b></p>
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<p>Is not available. Since aRts is as synthesizer application you'd have to
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reimplement very much code for Polypaudio. It should be easy to
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implement limited support for <tt>libartsc</tt> based
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applications. Noone has done this yet. It is probably a better idea to
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run <tt>arts</tt> on top of Polypaudio (through a polypaudio driver
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for aRts, which nobody has written yet). Another solution would be to
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embed Polypaudio in the aRts process.</p></li>
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<li><p><b>I often hear noises when playing back with Polypaudio, what can I do?</b></p>
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<p>There are to possible solutions: run polypaudio with argument
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<tt>--high-priority=1</tt> and make yourself member of the group
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<tt>realtime</tt>, or increase the fragment sizes of the audio
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drivers. The former will allow Polypaudio to activate
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<tt>SCHED_FIFO</tt> high priority scheduling (root rights are dropped
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immediately after this). Keep in mind that this is a potential security hole!</p></li>
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<li><p><b>The <tt>polypaudio</tt> executable is installed SUID root by default. Why this? Isn't this a potential security hole?</b></p>
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<p>Polypaudio activates <tt>SCHED_FIFO</tt> scheduling if the user
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passes <tt>--high-priority=1</tt>. This will only succeed when
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executed as root, therefore the binary is marked SUID root by
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default. Yes, this is a potential security hole. However, polypaudio
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tries its best to minimize the security threat: immediately after
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startup polypaudio drops all capabilities except
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<tt>CAP_SYS_NICE</tt> (At least on systems that support it, like Linux; see <tt>man 7
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capabilities</tt> for more information). If the calling user is not a
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member of the group <tt>realtime</tt> (which is required to have a GID
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< 1000), root rights are dropped immediately. This means, you can
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install polypaudio SUID root, but only a subset of your users (the
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members of the group <tt>realtime</tt>) may make use of realtime
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scheduling. Keep in mind that these users might load their own binary
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modules into the polypaudio daemon which may freeze the machine. The
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daemon has a minimal protection against CPU hogging (the daemon is
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killed after hogging more than 70% CPU for 5 seconds), but this may
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be circumvented easily by evildoers.</p></li>
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<li><p><b>I want to run polypaudio only when it is needed, how do I do this?</b></p>
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<p>Set <tt>autospawn = yes</tt> in <tt>client.conf</tt>. That
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configuration file may be found either in <tt>/etc/polypaudio/</tt> or
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in <tt>~/.polypaudio/</tt>.</p></li>
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<li><p><b>How do I list all polypaudio modules installed?</b></p>
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<p><tt>polypaudio --dump-modules</tt></p>
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<p>Add <tt>-v</tt> for terse usage instructions.</p>
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<li><p><b>How do I use polypaudio over the network?</b></p>
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<p>Just set <tt>$POLYP_SERVER</tt> to the host name of the polypaudio
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server. For authentication you need the same auth cookies on all sides. For
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that copy <tt>~./polypaudio-cookie</tt> to all clients that shall
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be allowed to connect.</p>
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<p>Alternatively the authorization cookies can be stored in the X11 server.</p></li>
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<li><p><b>Is polypaudio capable of providing synchronized audio playback over the network for movie players like <tt>mplayer</tt>?</b></p>
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<p>Yes! Unless your network is congested in some way (i.e. transfer latencies vary strongly) it works perfectly. Drop me an email for experimental patches for MPlayer.</p>
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<li><p><b>What environment variables does polypaudio care about?</b></p>
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<p>The client honors: <tt>POLYP_SINK</tt> (default sink to connect to), <tt>POLYP_SOURCE</tt> (default source to connect to), <tt>POLYP_SERVER</tt> (default server to connect to, like <tt>ESPEAKER</tt>), <tt>POLYP_BINARY</tt> (the binary to start when autospawning a daemon), <tt>POLYP_CLIENTCONFIG</tt> (path to the client configuration file).</p>
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<p>The daemon honors: <tt>POLYP_SCRIPT</tt> (default CLI script file run after startup), <tt>POLYP_CONFIG</tt> (default daemon configuration file), <tt>POLYP_DLPATH</tt> (colon separated list of paths where to look for modules)</p></li>
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<li><p><b>I saw that SIGUSR2 provokes loading of the module <tt>module-cli-protocol-unix</tt>. But how do I make use of that?</b></p>
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<p>A brilliant guy named Lennart Poettering once wrote a nifty tool
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for that purpose: <a
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href="http://0pointer.de/lennart/projects/bidilink/">bidilink</a>. To
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connect to a running polypaudio daemon try using the following commands:</p>
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<pre>killall -USR2 polypaudio
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bidilink unix-client:/tmp/polypaudio/cli</pre>
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<p><i>BTW: Someone should package this great tool for Debian!</i></p>
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<p><b>New:</b> There's now a tool <tt>pacmd</tt> that automates sending SIGUSR2 to the daemon and running a bidilink like tool for you.</p>
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</li>
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<li><p><b>How do the polypaudio libraries decide where to connect to?</b></p>
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<p>The following rule applies:</p>
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<ol>
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<li>If the the application using the library specifies a server to connect to it is used. If the connection fails, the library fails too.</li>
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<li>If the environment variable <tt>POLYP_SERVER</tt> is defined the library connects to that server. If the connection fails, the library fails too.</li>
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<li>If <tt>$DISPLAY</tt> is set, the library tries to connect to that server and looks for the root window property <tt>POYLP_SERVER</tt> for the host to connect to. If <tt>POLYP_COOKIE</tt> is set it is used as authentication cookie.</li>
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<li>If the client configuration file (<tt>~/.polypaudio/client.conf</tt> or <tt>/etc/polypaudio/client.conf</tt>) sets the server address, the library connects to that server. If the connection fails, the library fails too.</li>
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<li>The library tries to connect to the default local UNIX socket for polypaudio servers. If the connection fails, it proceeds with the next item.</li>
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<li>The library tries to connect to the default local TCP socket for polypaudio servers. If the connection fails, it proceeds with the next item.</li>
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<li>If <tt>$DISPLAY</tt> is set, the library tries to connect to the default TCP port of that host. If the connection fails, it proceeds with the next item.</li>
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<li>The connection fails.</li>
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</ol>
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</li>
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<li><p><b>Why the heck does libpolyp link against libX11?</b></p>
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<p>The Polypaudio client libraries look for some X11 root window
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properties for the credentials of the Polypaudio server to access. You
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may compile Polypaudio without X11 for disabling this feature.</p></li>
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<li><p><b>How can I use Polypaudio as an RTP based N:N multicast
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conferencing solution for the LAN?</b></p> <p>After loading all the
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necessary audio drivers for recording and playback, just load the RTP
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reciever and sender modules with default parameters:</p>
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<pre>
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load-module module-rtp-send
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load-module module-rtp-recv
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</pre>
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<p>As long as the Polypaudio daemon runs, the microphone data will be
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streamed to the network and the data from other hosts is played back
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locally. Please note that this may cause quite a lot of traffic. Hence
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consider passing <tt>rate=8000 format=ulaw channels=1</tt> to the
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sender module to save bandwith while still maintaining good quality
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for speech transmission.</p></li>
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<li><p><b>What is this RTP/SDP/SAP thing all about?</b></p>
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<p>RTP is the <i>Realtime Transfer Protocol</i>. It is a well-known
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protocol for transferring audio and video data over IP. SDP is the <i>Session
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Description Protocol</i> and can be used to describe RTP sessions. SAP
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is the <i>Session Announcement Protocol</i> and can be used to
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announce RTP sessions that are described with SDP. (Modern SIP based VoIP phones use RTP/SDP for their sessions, too)</p>
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<p>All three protocols are defined in IETF RFCs (RFC3550, RFC3551,
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RFC2327, RFC2327). They can be used in both multicast and unicast
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fashions. Polypaudio exclusively uses multicast RTP/SDP/SAP containing audio data.</p>
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<p>For more information about using these technologies with Polypaudio have a look on the <a href="modules.html#rtp">respective module's documentation</a>.
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<li><p><b>How can I use Polypaudio to stream music from my main PC to my LAN with multiple PCs with speakers?</b></p>
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<p>On the sender side create an RTP sink:</p>
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<pre>
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load-module module-null-sink sink_name=rtp
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load-module module-rtp-send source=rtp_monitor
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set-default-sink rtp
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</pre>
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<p>This will make <tt>rtp</tt> the default sink, i.e. all applications will write to this virtual RTP device by default.</p>
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<p>On the client sides just load the reciever module:</p>
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<pre>
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load-module module-rtp-recv
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</pre>
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<p>Now you can play your favourite music on the sender side and all clients will output it simultaneously.</p>
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<p>BTW: You can have more than one sender machine set up like this. The audio data will be mixed on the client side.</p></li>
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<li><p><b>How can I use Polypaudio to share a single LINE-IN/MIC jack on the entire LAN?</b></p>
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<p>On the sender side simply load the RTP sender module:</p>
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<pre>
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load-module module-rtp-send
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</pre>
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<p>On the reciever sides, create an RTP source:</p>
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<pre>
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load-module module-null-sink sink_name=rtp
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load-module module-rtp-recv sink=rtp
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set-default-source rtp_monitor
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</pre>
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<p>Now the audio data will be available from the default source <tt>rtp_monitor</tt>.</p>
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<li><p><b>When sending multicast RTP traffic it is recieved on the entire LAN but not by the sender machine itself!</b></p>
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<p>Pass <tt>loop=1</tt> to the sender module!</p></li>
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<li><p><b>Can I have more than one multicast RTP group?</b></p>
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<p>Yes! Simply use a new multicast group address. Use
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the <tt>destination</tt>/<tt>sap_address</tt> arguments of the RTP
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modules to select them. Choose your group addresses from the range
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<tt>225.0.0.x</tt> to make sure the audio data never leaves the LAN.</p></li>
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</ol>
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<hr/>
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<address class="grey">Lennart Poettering <@PACKAGE_BUGREPORT@>, April 2006</address>
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<div class="grey"><i>$Id$</i></div>
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</body> </html>
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