How does Polypaudio compare with ESOUND/aRts/NAS?
Polypaudio 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 Polypaudio. However, I believe that Polypaudio does what it does much better than any other free sound server.
What about ESOUND compatibility?
Polypaudio 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 Polypaudio a GNOME program?
No, Polypaudio 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 writen with gtkmm, i.e. require both GLIB and GTK.
Can I integrate Polypaudio in my GLIB/GTK/GNOME application?
Yes! Polypaudio 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 Polypaudio in my Qt/KDE application?
Yes! Polypaudio uses a main loop abstraction layer that allows you to integrate Polypaudio 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 Polypaudio, 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 Polypaudio. 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 Polypaudio (through a pulseaudio driver for aRts, which nobody has written yet). Another solution would be to embed Polypaudio in the aRts process.
I often hear noises when playing back with Polypaudio, 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 Polypaudio 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?
Polypaudio 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/pulseaudio/ or in ~/.pulseaudio/.
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 $POLYP_SERVER to the host name of the pulseaudio server. For authentication you need the same auth cookies on all sides. For that copy ~./pulseaudio-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: POLYP_SINK (default sink to connect to), POLYP_SOURCE (default source to connect to), POLYP_SERVER (default server to connect to, like ESPEAKER), POLYP_BINARY (the binary to start when autospawning a daemon), POLYP_CLIENTCONFIG (path to the client configuration file).
The daemon honors: POLYP_SCRIPT (default CLI script file run after startup), POLYP_CONFIG (default daemon configuration file), POLYP_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/pulseaudio/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 Polypaudio client libraries look for some X11 root window properties for the credentials of the Polypaudio server to access. You may compile Polypaudio without X11 for disabling this feature.
How can I use Polypaudio 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 Polypaudio 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. Polypaudio exclusively uses multicast RTP/SDP/SAP containing audio data.
For more information about using these technologies with Polypaudio have a look on the respective module's documentation.
How can I use Polypaudio 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 Polypaudio 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 Polypaudio 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. Polypaudio 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 Polypaudio 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 Polypaudio 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.