[pulseaudio-discuss] Network Audio with Pulse

Matthew Patterson matt at v8zman.com
Tue Apr 8 07:53:26 PDT 2008

I'm sorry, I think I mispoke in the last email, I meant combine sink. 
Here's some sample config:

When simulating my matrix switch idea, before I got the multiple sound 
cards I used remap to make it seem like I had 4 stereo sound cards. You 
can use this same idea to make one 6 channel card appear as 3 stereo cards:

# remap things so it seems like we have 4 stereo zones
load-module module-remap-sink sink_name=zone1 
master=alsa_output.pci_8086_2668_alsa_playback_0 channels=2 
master_channel_map=front-left,front-right channel_map=front-left,front-right
load-module module-remap-sink sink_name=zone2 
master=alsa_output.pci_8086_2668_alsa_playback_0 channels=2 
master_channel_map=front-left,front-right channel_map=front-left,front-right
load-module module-remap-sink sink_name=zone3 
master=alsa_output.pci_8086_2668_alsa_playback_0 channels=2 
master_channel_map=front-left,front-right channel_map=front-left,front-right
load-module module-remap-sink sink_name=zone4 
master=alsa_output.pci_8086_2668_alsa_playback_0 channels=2 
master_channel_map=front-left,front-right channel_map=front-left,front-right

Then I used the combine module to join all the zones together 4 times so 
there would be four inputs per zone, which can be muted to control what 
is heard:

# we leave only one of the outputs unmuted at startup, that is our 
player selection
load-module module-combine sink_name=p1 master=zone1 
#set-sink-input-mute 4 1
set-sink-input-mute 5 1
set-sink-input-mute 6 1
set-sink-input-mute 7 1
load-module module-combine sink_name=p2 master=zone1 
set-sink-input-mute 8 1
#set-sink-input-mute 9 1
set-sink-input-mute 10 1
set-sink-input-mute 11 1
load-module module-combine sink_name=p3 master=zone1 
set-sink-input-mute 12 1
set-sink-input-mute 13 1
#set-sink-input-mute 14 1
set-sink-input-mute 15 1
load-module module-combine sink_name=p4 master=zone1 
set-sink-input-mute 16 1
set-sink-input-mute 17 1
set-sink-input-mute 18 1
#set-sink-input-mute 19 1

Does that help?


Jim Duda wrote:
> Matt,
> I don't understand how the remap_sink module helps (or works for that 
> matter).  I'm having trouble getting my head around the inputs and 
> outputs of this module.
> Would you mind posting an example of how you use remap_sink?
> Thanks,
> Jim
>     "Matt Patterson" <matt at v8zman.com <mailto:matt at v8zman.com>> wrote
>     in message news:47FAB22D.7000408 at v8zman.com...
>     Yeah, sounds like you have the rtp thing. I assume you realize you
>     can have multiple multicast addresses so there can be simultaneous
>     streams that don't collide/get mixed.
>     I don't think there is a way you can avoid the mesh (multicast is
>     basically a mesh, just for free) unless you designate a server
>     machine. In which case you could set up a single tunnel sink to
>     each client machine and then have all the switching happen on that
>     machine. I use the remap module to split each sink into 4 inputs
>     (could be a tunnel sink), then connect each input to a different
>     mpd instance, and control what is heard out each device by muting
>     3 of the 4 inputs. I end up with 16 remapped sinks in this case (4
>     output devices * 4 remapped sinks each). I will be adding a 5th
>     zone to my whole home audio soon, so that will make it 25. The 16
>     sinks/streams seems to cause no undue load on the system (Core 2
>     Duo 2180), we'll see how 25 does :)
>     I haven't played with the tunnel sink module.
>     Matt
>     Jim Duda wrote:
>>     Matt,
>>     Thanks for the feedback.
>>     I understand your point about using the command line interface module. 
>>     I actually would end up using the socket approach from perl once I had 
>>     it all working properly.
>>     I believe I understand how the rtp approach would work. I think what you 
>>     are doing is as follows.  All stream senders would send on the rtp_send 
>>     side, connecting the rtp_send.monitor to the default alsa sink (for 
>>     local sound).  All other machines would have an rtp_recv, and send the 
>>     output of rtp_recv to the default alsa sink.  Each of these rtp_recv 
>>     would be muted by default.  If machine B wants to join in, machine B 
>>     would unmute it's rtp_recv and thereby get the stream.  Do I have a 
>>     basic understanding of how this approach would work?  I have played with 
>>     this to some degree, so I think I understand.
>>     I assume using the tunnels would work in a similar fashion.  However, 
>>     you need to build a mesh of tunnel connections.  In my case, with 4 
>>     nodes, the mesh is 3 tunnels for each mode, 4*3=12 tunnels in total. 
>>     Each receiving node would then mute each tunnel by default, turning on 
>>     the one it wants.  The annoying part of this approach is that you have 
>>     to decide which source you want to connect to, whereas, with the rtp 
>>     approach, you simply join the "collective".
>>     I have played with the combined_sinks somewhat too.  However, since 
>>     upgrading to FC8, the pulseaudio server keeps crashing when I attempt to 
>>     use a combined sink.  I've been trying to get a core dump to Lennart, 
>>     but I haven't been able to get gdb to help me out, I keep getting some 
>>     problems with some threading library (or something of that nature).
>>     I'm now trying to understand how the paprefs gui mechanism works.  I 
>>     haven't been able to get any of the options to be enabled for operation, 
>>     all the controls are grayed out, trying to understand why.
>>     Jim
>>     Matt Patterson wrote:
>>>     I played with something similar but my goal was an audio multiplex 
>>>     switch all on the same machine to the rtp lag issue was less apparent. 
>>>     As for controlling it, I just wrote a simple python app that connects to 
>>>     the unix socket (same thing pacmd does) and I issue commands to load 
>>>     modules, mute inputs, etc so things can be controlled. I then wrote a 
>>>     php wrapper around the python app so my web based audio control could 
>>>     come about.
>>>     To go this route you have to make sure the command line interface is 
>>>     available either via TCP or Unix socket (I chose unix socket). If you 
>>>     like I would be happy to send my hacktastic python code to help get 
>>>     things moving.
>>>     I believe that using the tunnels allows you to have the sync feature 
>>>     where rtp doesn't, so maybe play around with getting them working???
>>>     Matt
>>>     Jim Duda wrote:
>>>>     There was a similar thread, back around New Year's regarding Network 
>>>>     Audio.  I've read the entire thread a few times.  I'm having similar 
>>>>     problems, yet different.
>>>>     I'm looking for some advice as to how best to use network audio with pulse.
>>>>     I have multiple linux computers in my house, four to be specific.  One 
>>>>     operates as a file server, one as a desktop, and the other two as 
>>>>     diskless think clients which basically operate as media players.
>>>>     I use these computers in a home automation network in my house using the 
>>>>     misterhouse home automation software (misterhouse.net).
>>>>     All machines are running stock fedora 8.  The two thin clients, are not 
>>>>     running the full suite of services which a desktop would.  For example, 
>>>>     they are not currently running avahi or hal (but could if necessary).  I 
>>>>     can certainly turn on what needs to be running.
>>>>     I'm hoping to perform the following using pulseaudio.
>>>>     Let's call my machines A, B, C, D.
>>>>     Let's assume that some stream is started on machine A, playing in the 
>>>>     living room.  I would like to be able to have that same stream play on 
>>>>     machines A and B simultaneously.  I don't care if I have to go to stream 
>>>>     A and say send to machine B now, or, go to machine B and ask B to fetch 
>>>>     a stream from machine A.  I can make both work.  I want to be able to 
>>>>     drop the stream to B at anytime.  I realize that if the source stream 
>>>>     stops, then all streams would in essence stop too.
>>>>     I need to be able to access the controls to switch streams using a 
>>>>     command line application which I can call from perl using the system 
>>>>     call.  I've seen the stream switch in pavucontrol.  I've seen the 
>>>>     move-sink-input in pactl (but failed to get it to work, I guess I don't 
>>>>     understand how the params work as I always get some error message).
>>>>     At some other time, I may want to have machine C join in the stream with 
>>>>     machines B, C.
>>>>     How is this best to accomplish?
>>>>     1) Should I use combine_sink on the source machine?
>>>>     2) Should I use rtp?
>>>>     3) Should I use tunnel_sink?
>>>>     I've played with rtp.  Although it works, the audio isn't synchronized. 
>>>>         Maybe it should be synchronized, but I haven't found that to be 
>>>>     true.  I can hear latency delay between multiple machines.
>>>>     I know how to play across the network, using the pulseaudio alsa plugin.
>>>>     I'm now trying to play with the network options in the paprefs 
>>>>     application.  On my main server and desktop, all the network audio 
>>>>     options in paprefs, configure local sound server, are all grayed out.
>>>>     Each machine has these modules installed from FC8.
>>>>     sudo yum list '*pulse*'
>>>>     Installed Packages
>>>>     akode-pulseaudio.i386             2.0.2-4.fc8            installed
>>>>     alsa-plugins-pulseaudio.i386      1.0.15-3.fc8.1         installed
>>>>     pulseaudio.i386                   0.9.8-5.fc8            installed
>>>>     pulseaudio-core-libs.i386         0.9.8-5.fc8            installed
>>>>     pulseaudio-esound-compat.i3       0.9.8-5.fc8            installed
>>>>     pulseaudio-libs.i386              0.9.8-5.fc8            installed
>>>>     pulseaudio-libs-devel.i386        0.9.8-5.fc8            installed
>>>>     pulseaudio-libs-glib2.i386        0.9.8-5.fc8            installed
>>>>     pulseaudio-libs-zeroconf.i386     0.9.8-5.fc8            installed
>>>>     pulseaudio-module-gconf.i386      0.9.8-5.fc8            installed
>>>>     pulseaudio-module-jack.i386       0.9.8-5.fc8            installed
>>>>     pulseaudio-module-x11.i386        0.9.8-5.fc8            installed
>>>>     pulseaudio-module-zeroconf.i386   0.9.8-5.fc8            installed
>>>>     pulseaudio-utils.i386            0.9.8-5.fc8             installed
>>>>     Available Packages
>>>>     audacious-plugins-pulseaudio.i386 1.3.5-3.fc8            fedora
>>>>     fluxbox-pulseaudio.i386           1.0.0-2.fc8            updates
>>>>     gstreamer-plugins-pulse.i386      0.9.5-0.4.svn20070924. fedora
>>>>     kde-settings-pulseaudio.noarch    3.5-38.fc8             updates
>>>>     pulseaudio-module-bluetooth.i386  0.9.8-5.fc8            updates
>>>>     pulseaudio-module-lirc.i386       0.9.8-5.fc8            updates
>>>>     Both the avahi and gconf modules are loaded as displayed in the Modules 
>>>>     section of the Paprefs Manager display.  What else is necessary?
>>>>     I have auth-anonymouns=1 loaded for both native-protocol-unix and native 
>>>>     -protocol-tcp.
>>>>     I've read all the documentation on the pulse wiki many times.  I've 
>>>>     browsed through all the postings on the mailing list over the past 6 months.
>>>>     I'm just playing now with the server and desktop which have full blown 
>>>>     stock fc8 installs, just to figure out how all this works, then I'll 
>>>>     incorporate the thin clients later.
>>>>     The whole package is rather complicated and I haven't had much success 
>>>>     in putting it all together.
>>>>     I've done my homework.  I just cannot get it working ...
>>>>     Thanks,
>>>>     Jim
>>>>     _______________________________________________
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