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5 <title>PulseAudio: FAQ
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12 <h1>Frequently Asked Questions
</h1>
15 <li><p><b>How does PulseAudio compare with ESOUND/aRts/NAS?
</b></p>
17 <p>PulseAudio is sound daemon similar to ESOUND and NAS, but much more
18 powerful. aRts is a realtime-synthesizer-cum-sound-server, i.e. it
19 does much more than PulseAudio. However, I believe that PulseAudio
20 does what it does much better than any other free sound server.
</p>
23 <li><p><b>What about ESOUND compatibility?
</b></p>
24 <p>PulseAudio is a drop in replacement for ESOUND. That means: you can
25 load a esound compatibility module which implements an ESOUND
26 compatible protocol which allows you to use most of the classic ESOUND
27 compatible programs (including the command line programs like
31 <li><p><b>Is PulseAudio a GNOME program?
</b></p>
32 <p>No, PulseAudio has no dependency on GNOME/GTK/GLIB. All it requires
33 is a UNIX-like operating system and very few dependency
34 libraries. However, the accompanying GUI tools are written with
35 gtkmm, i.e. require both GLIB and GTK.
</p></li>
37 <li><p><b>Can I integrate PulseAudio in my GLIB/GTK/GNOME application?
</b></p>
38 <p>Yes! PulseAudio comes with a GLIB main loop adapter. You can embed
39 both the client library and the daemon (!) into your GLIB based
42 <li><p><b>Can I integrate PulseAudio in my Qt/KDE application?
</b></p>
43 <p>Yes! PulseAudio uses a main loop abstraction layer that allows you
44 to integrate PulseAudio in any program that supports main
45 loops. Unfortunately there is no adapter for Qt publicly available yet.
</p></li>
47 <li><p><b>I want to write a new driver for PulseAudio, are there any docs?
</b></p>
48 <p>Currently, only the client API is documented with doxygen. Read
49 the source and base your work on a simple module like
50 <tt>module-pipe-sink
</tt>.
</p></li>
52 <li><p><b>What about compatibility with NAS?
</b></p>
53 <p>Is not available (yet?). It is doable, but noone has implemented it yet.
</p></li>
55 <li><p><b>What about compatibility with aRts?
</b></p>
56 <p>Is not available. Since aRts is as synthesizer application you'd have to
57 reimplement very much code for PulseAudio. It should be easy to
58 implement limited support for
<tt>libartsc
</tt> based
59 applications. Noone has done this yet. It is probably a better idea to
60 run
<tt>arts
</tt> on top of PulseAudio (through a PulseAudio driver
61 for aRts, which nobody has written yet). Another solution would be to
62 embed PulseAudio in the aRts process.
</p></li>
64 <li><p><b>I often hear noises when playing back with PulseAudio, what can I do?
</b></p>
65 <p>There are to possible solutions: run PulseAudio with argument
66 <tt>--high-priority=
1</tt> and make yourself member of the group
67 <tt>realtime
</tt>, or increase the fragment sizes of the audio
68 drivers. The former will allow PulseAudio to activate
69 <tt>SCHED_FIFO
</tt> high priority scheduling (root rights are dropped
70 immediately after this). Keep in mind that this is a potential security hole!
</p></li>
72 <li><p><b>The
<tt>pulseaudio
</tt> executable is installed SUID root by default. Why this? Isn't this a potential security hole?
</b></p>
74 <p>PulseAudio activates
<tt>SCHED_FIFO
</tt> scheduling if the user
75 passes
<tt>--high-priority=
1</tt>. This will only succeed when
76 executed as root, therefore the binary is marked SUID root by
77 default. Yes, this is a potential security hole. However, PulseAudio
78 tries its best to minimize the security threat: immediately after
79 startup PulseAudio drops all capabilities except
80 <tt>CAP_SYS_NICE
</tt> (At least on systems that support it, like Linux; see
<tt>man
7
81 capabilities
</tt> for more information). If the calling user is not a
82 member of the group
<tt>realtime
</tt> (which is required to have a GID
83 <
1000), root rights are dropped immediately. This means, you can
84 install
<tt>pulseaudio
</tt> SUID root, but only a subset of your users (the
85 members of the group
<tt>realtime
</tt>) may make use of realtime
86 scheduling. Keep in mind that these users might load their own binary
87 modules into the PulseAudio daemon which may freeze the machine. The
88 daemon has a minimal protection against CPU hogging (the daemon is
89 killed after hogging more than
70% CPU for
5 seconds), but this may
90 be circumvented easily by evildoers.
</p></li>
92 <li><p><b>I want to run PulseAudio only when it is needed, how do I do this?
</b></p>
94 <p>Set
<tt>autospawn = yes
</tt> in
<tt>client.conf
</tt>. That
95 configuration file may be found either in
<tt>/etc/pulse/
</tt> or
96 in
<tt>~/.pulse/
</tt>.
</p></li>
98 <li><p><b>How do I list all PulseAudio modules installed?
</b></p>
100 <p><tt>pulseaudio --dump-modules
</tt></p>
102 <p>Add
<tt>-v
</tt> for terse usage instructions.
</p>
104 <li><p><b>How do I use PulseAudio over the network?
</b></p>
106 <p>Just set
<tt>$PULSE_SERVER
</tt> to the host name of the PulseAudio
107 server. For authentication you need the same auth cookies on all sides. For
108 that copy
<tt>~./pulse-cookie
</tt> to all clients that shall
109 be allowed to connect.
</p>
111 <p>Alternatively the authorization cookies can be stored in the X11 server.
</p></li>
113 <li><p><b>Is PulseAudio capable of providing synchronized audio playback over the network for movie players like
<tt>mplayer
</tt>?
</b></p>
115 <p>Yes! Unless your network is congested in some way (i.e. transfer latencies vary strongly) it works perfectly. Drop me an email for experimental patches for MPlayer.
</p>
117 <li><p><b>What environment variables does PulseAudio care about?
</b></p>
119 <p>The client honors:
<tt>PULSE_SINK
</tt> (default sink to connect to),
<tt>PULSE_SOURCE
</tt> (default source to connect to),
<tt>PULSE_SERVER
</tt> (default server to connect to, like
<tt>ESPEAKER
</tt>),
<tt>PULSE_BINARY
</tt> (the binary to start when autospawning a daemon),
<tt>PULSE_CLIENTCONFIG
</tt> (path to the client configuration file).
</p>
121 <p>The daemon honors:
<tt>PULSE_SCRIPT
</tt> (default CLI script file run after startup),
<tt>PULSE_CONFIG
</tt> (default daemon configuration file),
<tt>PULSE_DLPATH
</tt> (colon separated list of paths where to look for modules)
</p></li>
124 <li><p><b>I saw that SIGUSR2 provokes loading of the module
<tt>module-cli-protocol-unix
</tt>. But how do I make use of that?
</b></p>
126 <p>A brilliant guy named Lennart Poettering once wrote a nifty tool
128 href=
"http://0pointer.de/lennart/projects/bidilink/">bidilink
</a>. To
129 connect to a running PulseAudio daemon try using the following commands:
</p>
131 <pre>killall -USR2 pulseaudio
132 bidilink unix-client:/tmp/pulse-$USER/cli
</pre>
134 <p><i>BTW: Someone should package this great tool for Debian!
</i></p>
136 <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>
139 <li><p><b>How do the PulseAudio libraries decide where to connect to?
</b></p>
140 <p>The following rule applies:
</p>
142 <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>
143 <li>If the environment variable
<tt>PULSE_SERVER
</tt> is defined the library connects to that server. If the connection fails, the library fails too.
</li>
144 <li>If
<tt>$DISPLAY
</tt> is set, the library tries to connect to that server and looks for the root window property
<tt>POYLP_SERVER
</tt> for the host to connect to. If
<tt>PULSE_COOKIE
</tt> is set it is used as authentication cookie.
</li>
145 <li>If the client configuration file (
<tt>~/.pulse/client.conf
</tt> or
<tt>/etc/pulse/client.conf
</tt>) sets the server address, the library connects to that server. If the connection fails, the library fails too.
</li>
146 <li>The library tries to connect to the default local UNIX socket for PulseAudio servers. If the connection fails, it proceeds with the next item.
</li>
147 <li>The library tries to connect to the default local TCP socket for PulseAudio servers. If the connection fails, it proceeds with the next item.
</li>
148 <li>If
<tt>$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>
149 <li>The connection fails.
</li>
153 <li><p><b>Why the heck does libpulse link against libX11?
</b></p>
154 <p>The PulseAudio client libraries look for some X11 root window
155 properties for the credentials of the PulseAudio server to access. You
156 may compile PulseAudio without X11 for disabling this feature.
</p></li>
158 <li><p><b>How can I use PulseAudio as an RTP based N:N multicast
159 conferencing solution for the LAN?
</b></p> <p>After loading all the
160 necessary audio drivers for recording and playback, just load the RTP
161 reciever and sender modules with default parameters:
</p>
164 load-module module-rtp-send
165 load-module module-rtp-recv
168 <p>As long as the PulseAudio daemon runs, the microphone data will be
169 streamed to the network and the data from other hosts is played back
170 locally. Please note that this may cause quite a lot of traffic. Hence
171 consider passing
<tt>rate=
8000 format=ulaw channels=
1</tt> to the
172 sender module to save bandwith while still maintaining good quality
173 for speech transmission.
</p></li>
175 <li><p><b>What is this RTP/SDP/SAP thing all about?
</b></p>
177 <p>RTP is the
<i>Realtime Transfer Protocol
</i>. It is a well-known
178 protocol for transferring audio and video data over IP. SDP is the
<i>Session
179 Description Protocol
</i> and can be used to describe RTP sessions. SAP
180 is the
<i>Session Announcement Protocol
</i> and can be used to
181 announce RTP sessions that are described with SDP. (Modern SIP based VoIP phones use RTP/SDP for their sessions, too)
</p>
183 <p>All three protocols are defined in IETF RFCs (RFC3550, RFC3551,
184 RFC2327, RFC2327). They can be used in both multicast and unicast
185 fashions. PulseAudio exclusively uses multicast RTP/SDP/SAP containing audio data.
</p>
187 <p>For more information about using these technologies with PulseAudio have a look on the
<a href=
"modules.html#rtp">respective module's documentation
</a>.
189 <li><p><b>How can I use PulseAudio to stream music from my main PC to my LAN with multiple PCs with speakers?
</b></p>
191 <p>On the sender side create an RTP sink:
</p>
194 load-module module-null-sink sink_name=rtp
195 load-module module-rtp-send source=rtp_monitor
199 <p>This will make
<tt>rtp
</tt> the default sink, i.e. all applications will write to this virtual RTP device by default.
</p>
201 <p>On the client sides just load the reciever module:
</p>
203 load-module module-rtp-recv
206 <p>Now you can play your favourite music on the sender side and all clients will output it simultaneously.
</p>
209 <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>
211 <li><p><b>How can I use PulseAudio to share a single LINE-IN/MIC jack on the entire LAN?
</b></p>
213 <p>On the sender side simply load the RTP sender module:
</p>
216 load-module module-rtp-send
219 <p>On the reciever sides, create an RTP source:
</p>
222 load-module module-null-sink sink_name=rtp
223 load-module module-rtp-recv sink=rtp
224 set-default-source rtp_monitor
227 <p>Now the audio data will be available from the default source
<tt>rtp_monitor
</tt>.
</p></li>
229 <li><p><b>When sending multicast RTP traffic it is recieved on the entire LAN but not by the sender machine itself!
</b></p>
231 <p>Pass
<tt>loop=
1</tt> to the sender module!
</p></li>
233 <li><p><b>Can I have more than one multicast RTP group?
</b></p>
235 <p>Yes! Simply use a new multicast group address. Use
236 the
<tt>destination
</tt>/
<tt>sap_address
</tt> arguments of the RTP
237 modules to select them. Choose your group addresses from the range
238 <tt>225.0.0.x
</tt> to make sure the audio data never leaves the LAN.
</p></li>
241 <li><p><b>Can I use PulseAudio to playback music on two sound cards simultaneously?
</b></p>
243 <p>Yes! Use
<a href=
"modules.html#module-combine"><tt>module-combine
</tt></a> for that.
</p>
246 load-module module-oss-mmap
device=
"/dev/dsp" sink_name=output0
247 load-module module-oss-mmap
device=
"/dev/dsp1" sink_name=output1
248 load-module module-combine sink_name=combined master=output0 slaves=output1
249 set-sink-default combined
252 <p>This will combine the two sinks
<tt>output0
</tt> and
253 <tt>output1
</tt> into a new sink
<tt>combined
</tt>. Every sample
254 written to the latter will be forwarded to the former two. PulseAudio
255 will make sure to adjust the sample rate of the slave device in case
256 it deviates from the master device. You can have more than one slave
257 sink attached to the combined sink, and hence combine even three and
258 more sound cards.
</p> </li>
260 <li><p><b>Can I use PulseAudio to combine two stereo soundcards into a virtual surround sound card?
</b></p>
262 <p>Yes! You can use use
<a href=
"modules.html#module-combine"><tt>module-combine
</tt></a> for that.
</p>
265 load-module module-oss-mmap
device=
"/dev/dsp" sink_name=output0 channel_map=left,right channels=
2
266 load-module module-oss-mmap
device=
"/dev/dsp1" sink_name=output1 channel_map=rear-left,rear-right channels=
2
267 load-module module-combine sink_name=combined master=output0 slaves=output1 channel_map=left,right,rear-left,rear-right channels=
4
270 <p>This is mostly identical to the previous example. However, this
271 time we manually specify the channel mappings for the sinks to make
272 sure everything is routed correctly.
</p>
274 <p>Please keep in mind that PulseAudio will constantly adjust the
275 sample rate to compensate for the deviating quartzes of the sound
276 devices. This is not perfect, however. Deviations in a range of
277 1/
44100s (or
1/
48000s depending on the sampling frequency) can not be
278 compensated. The human ear will decode these deviations as minor
279 movements (less than
1cm) of the positions of the sound sources
284 <li><p><b>Why did you rename Polypaudio to PulseAudio?
</b></p>
286 <p>Please read this
<a href=
"http://0pointer.de/blog/projects/pulse.html">blog story
</a> for an explanation.
</p>
293 <address class=
"grey">Lennart Poettering
<@PACKAGE_BUGREPORT@
>, April
2006</address>
294 <div class=
"grey"><i>$Id$
</i></div>