How does PulseAudio compare with ESOUND/aRts/NAS?
- -PulseAudio is sound daemon similar to ESOUND and NAS, but much more - powerful. aRts is a realtime-synthesizer-cum-sound-server, i.e. it - does much more than PulseAudio. However, I believe that PulseAudio - does what it does much better than any other free sound server.
-What about ESOUND compatibility?
-PulseAudio is a drop in replacement for ESOUND. That means: you can - load a esound compatibility module which implements an ESOUND - compatible protocol which allows you to use most of the classic ESOUND - compatible programs (including the command line programs like - esdcat).
-Is PulseAudio a GNOME program?
-No, PulseAudio has no dependency on GNOME/GTK/GLIB. All it requires - is a UNIX-like operating system and very few dependency - libraries. However, the accompanying GUI tools are written with - gtkmm, i.e. require both GLIB and GTK.
Can I integrate PulseAudio in my GLIB/GTK/GNOME application?
-Yes! PulseAudio comes with a GLIB main loop adapter. You can embed - both the client library and the daemon (!) into your GLIB based - application.
Can I integrate PulseAudio in my Qt/KDE application?
-Yes! PulseAudio uses a main loop abstraction layer that allows you - to integrate PulseAudio in any program that supports main - loops. Unfortunately there is no adapter for Qt publicly available yet.
I want to write a new driver for PulseAudio, are there any docs?
-Currently, only the client API is documented with doxygen. Read - the source and base your work on a simple module like - module-pipe-sink.
What about compatibility with NAS?
-Is not available (yet?). It is doable, but noone has implemented it yet.
What about compatibility with aRts?
-Is not available. Since aRts is as synthesizer application you'd have to - reimplement very much code for PulseAudio. It should be easy to - implement limited support for libartsc based - applications. Noone has done this yet. It is probably a better idea to - run arts on top of PulseAudio (through a PulseAudio driver - for aRts, which nobody has written yet). Another solution would be to - embed PulseAudio in the aRts process.
I often hear noises when playing back with PulseAudio, what can I do?
-There are to possible solutions: run PulseAudio with argument ---high-priority=1 and make yourself member of the group -realtime, or increase the fragment sizes of the audio - drivers. The former will allow PulseAudio to activate - SCHED_FIFO high priority scheduling (root rights are dropped - immediately after this). Keep in mind that this is a potential security hole!
The pulseaudio executable is installed SUID root by default. Why this? Isn't this a potential security hole?
- -PulseAudio activates SCHED_FIFO scheduling if the user -passes --high-priority=1. This will only succeed when -executed as root, therefore the binary is marked SUID root by -default. Yes, this is a potential security hole. However, PulseAudio -tries its best to minimize the security threat: immediately after -startup PulseAudio drops all capabilities except -CAP_SYS_NICE (At least on systems that support it, like Linux; see man 7 -capabilities for more information). If the calling user is not a -member of the group realtime (which is required to have a GID -< 1000), root rights are dropped immediately. This means, you can -install pulseaudio SUID root, but only a subset of your users (the -members of the group realtime) may make use of realtime -scheduling. Keep in mind that these users might load their own binary -modules into the PulseAudio daemon which may freeze the machine. The -daemon has a minimal protection against CPU hogging (the daemon is -killed after hogging more than 70% CPU for 5 seconds), but this may -be circumvented easily by evildoers.
I want to run PulseAudio only when it is needed, how do I do this?
- -Set autospawn = yes in client.conf. That -configuration file may be found either in /etc/pulse/ or -in ~/.pulse/.
How do I list all PulseAudio modules installed?
- -pulseaudio --dump-modules
- -Add -v for terse usage instructions.
- -How do I use PulseAudio over the network?
- -Just set $PULSE_SERVER to the host name of the PulseAudio -server. For authentication you need the same auth cookies on all sides. For -that copy ~./pulse-cookie to all clients that shall -be allowed to connect.
- -Alternatively the authorization cookies can be stored in the X11 server.
Is PulseAudio capable of providing synchronized audio playback over the network for movie players like mplayer?
- -Yes! Unless your network is congested in some way (i.e. transfer latencies vary strongly) it works perfectly. Drop me an email for experimental patches for MPlayer.
- -What environment variables does PulseAudio care about?
- -The client honors: PULSE_SINK (default sink to connect to), PULSE_SOURCE (default source to connect to), PULSE_SERVER (default server to connect to, like ESPEAKER), PULSE_BINARY (the binary to start when autospawning a daemon), PULSE_CLIENTCONFIG (path to the client configuration file).
- -The daemon honors: PULSE_SCRIPT (default CLI script file run after startup), PULSE_CONFIG (default daemon configuration file), PULSE_DLPATH (colon separated list of paths where to look for modules)
I saw that SIGUSR2 provokes loading of the module module-cli-protocol-unix. But how do I make use of that?
- -A brilliant guy named Lennart Poettering once wrote a nifty tool -for that purpose: bidilink. To -connect to a running PulseAudio daemon try using the following commands:
- -killall -USR2 pulseaudio -bidilink unix-client:/tmp/pulse-$USER/cli- -
BTW: Someone should package this great tool for Debian!
- -New: There's now a tool pacmd that automates sending SIGUSR2 to the daemon and running a bidilink like tool for you.
-How do the PulseAudio libraries decide where to connect to?
-The following rule applies:
-Why the heck does libpulse link against libX11?
-The PulseAudio client libraries look for some X11 root window -properties for the credentials of the PulseAudio server to access. You -may compile PulseAudio without X11 for disabling this feature.
How can I use PulseAudio as an RTP based N:N multicast -conferencing solution for the LAN?
After loading all the -necessary audio drivers for recording and playback, just load the RTP -reciever and sender modules with default parameters:
- --load-module module-rtp-send -load-module module-rtp-recv -- -
As long as the PulseAudio daemon runs, the microphone data will be -streamed to the network and the data from other hosts is played back -locally. Please note that this may cause quite a lot of traffic. Hence -consider passing rate=8000 format=ulaw channels=1 to the -sender module to save bandwith while still maintaining good quality -for speech transmission.
What is this RTP/SDP/SAP thing all about?
- -RTP is the Realtime Transfer Protocol. It is a well-known -protocol for transferring audio and video data over IP. SDP is the Session -Description Protocol and can be used to describe RTP sessions. SAP -is the Session Announcement Protocol and can be used to -announce RTP sessions that are described with SDP. (Modern SIP based VoIP phones use RTP/SDP for their sessions, too)
- -All three protocols are defined in IETF RFCs (RFC3550, RFC3551, -RFC2327, RFC2327). They can be used in both multicast and unicast -fashions. PulseAudio exclusively uses multicast RTP/SDP/SAP containing audio data.
- -For more information about using these technologies with PulseAudio have a look on the respective module's documentation. - -
How can I use PulseAudio to stream music from my main PC to my LAN with multiple PCs with speakers?
- -On the sender side create an RTP sink:
- --load-module module-null-sink sink_name=rtp -load-module module-rtp-send source=rtp_monitor -set-default-sink rtp -- -
This will make rtp the default sink, i.e. all applications will write to this virtual RTP device by default.
- -On the client sides just load the reciever module:
--load-module module-rtp-recv -- -
Now you can play your favourite music on the sender side and all clients will output it simultaneously.
- - -BTW: You can have more than one sender machine set up like this. The audio data will be mixed on the client side.
How can I use PulseAudio to share a single LINE-IN/MIC jack on the entire LAN?
- -On the sender side simply load the RTP sender module:
- --load-module module-rtp-send -- -
On the reciever sides, create an RTP source:
- --load-module module-null-sink sink_name=rtp -load-module module-rtp-recv sink=rtp -set-default-source rtp_monitor -- -
Now the audio data will be available from the default source rtp_monitor.
When sending multicast RTP traffic it is recieved on the entire LAN but not by the sender machine itself!
- -Pass loop=1 to the sender module!
Can I have more than one multicast RTP group?
- -Yes! Simply use a new multicast group address. Use -the destination/sap_address arguments of the RTP -modules to select them. Choose your group addresses from the range -225.0.0.x to make sure the audio data never leaves the LAN.
Can I use PulseAudio to playback music on two sound cards simultaneously?
- -Yes! Use module-combine for that.
- --load-module module-oss-mmap device="/dev/dsp" sink_name=output0 -load-module module-oss-mmap device="/dev/dsp1" sink_name=output1 -load-module module-combine sink_name=combined master=output0 slaves=output1 -set-sink-default combined -- -
This will combine the two sinks output0 and -output1 into a new sink combined. Every sample -written to the latter will be forwarded to the former two. PulseAudio -will make sure to adjust the sample rate of the slave device in case -it deviates from the master device. You can have more than one slave -sink attached to the combined sink, and hence combine even three and -more sound cards.
Can I use PulseAudio to combine two stereo soundcards into a virtual surround sound card?
- -Yes! You can use use module-combine for that.
- --load-module module-oss-mmap device="/dev/dsp" sink_name=output0 channel_map=left,right channels=2 -load-module module-oss-mmap device="/dev/dsp1" sink_name=output1 channel_map=rear-left,rear-right channels=2 -load-module module-combine sink_name=combined master=output0 slaves=output1 channel_map=left,right,rear-left,rear-right channels=4 -- -
This is mostly identical to the previous example. However, this -time we manually specify the channel mappings for the sinks to make -sure everything is routed correctly.
- -Please keep in mind that PulseAudio will constantly adjust the -sample rate to compensate for the deviating quartzes of the sound -devices. This is not perfect, however. Deviations in a range of -1/44100s (or 1/48000s depending on the sampling frequency) can not be -compensated. The human ear will decode these deviations as minor -movements (less than 1cm) of the positions of the sound sources -you hear.
- -Why did you rename Polypaudio to PulseAudio?
- -Please read this blog story for an explanation.
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