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 slaves=zone2,zone3,zone4 #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 slaves=zone2,zone3,zone4 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 slaves=zone2,zone3,zone4 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 slaves=zone2,zone3,zone4 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? Matt 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 >>>> >>>> >>>> >>>> _______________________________________________ >>>> pulseaudio-discuss mailing list >>>> pulseaudio-discuss at mail.0pointer.de >>>> https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss >>>> >>>> >> >> _______________________________________________ >> pulseaudio-discuss mailing list >> pulseaudio-discuss at mail.0pointer.de >> https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss >> > > ------------------------------------------------------------------------ > _______________________________________________ > pulseaudio-discuss mailing list > pulseaudio-discuss at mail.0pointer.de > https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss > > ------------------------------------------------------------------------ > > _______________________________________________ > pulseaudio-discuss mailing list > pulseaudio-discuss at mail.0pointer.de > https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss > -------------- next part -------------- An HTML attachment was scrubbed... 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