On 20 June 2010 18:10, Louigi Verona <louigi.verona(a)gmail.com> wrote: Yeah, that's as close as you can get. In live, keeping the panner in the centre on the mixer does it by duplicating the signal, but you can't replicate that with jack automatically if that's what you're asking. You do have the patchbay at your disposal, though. -- GPG/PGP ID: B42DDCAD

I have a question about a different way of doing this (soundcard mixer dependent), and whether it is advantageous (or not) to use a soundcard's built-in mixer, e.g. for direct monitoring at low-latency. If you have a soundcard like the Terratec DMX6Fire or the M-Audio Delta66 you certainly have the capability for this. If you have a supported soundcard, from "alsa-tools" you use , e.g. :/usr/bin/cspctl , /usr/bin/echomixer , /usr/bin/envy24control , /usr/bin/hdspconf , /usr/bin/hdspmixer/ , /usr/bin/rmedigicontrol , /usr/bin/sscape_ctl , /usr/bin/us428control ... The alternative is to route via jack using qjackctl and/or take care of mixing/monitoring in the DAW, e.g. ardour/rosegarden/qtractor. What are the advantages of using the soundcard's digital mixer, esp. envy24control & snd_ice1712 to route a single sound-source to an output for auditioning; or, if mixing is needed, use the built-in digital mixer to directly mix the different sound-sources? Note you have the option of digital-sync to a source that can be mixed back through the digital mixer's SPDIF inputs as well (e.g. reverb/room effects, madonna's-vocal-backing-track). And then route the digital mix as the "headphone mix" back to the performer. The main advantage I would predict (and can see) is slightly lower CPU usage, which might matter on a laptop or older computer, but just seems like a trivial amount of extra processing on today's fast 4 core machines. The other advantage would be latency -- I can see no way that routing in and out of soundcard(s) through jack would not add some latency -- the hardware mixing option is marketed as "0 latency monitoring" for a reason. However, in-practice, this practice is bedevilled by the mains or headphone monitor outputs automatically being routed back from the digital mixer L/R outputs, to their PCM0-3 outputs: at least when used via Jack/qjackctl . So eventually, you end up hearing silence (since I usually would then digital mix PCM4-7 leaving PCM0-3 unused or silent for this scenario). Any suggestions, help or workarounds on the latter would be appreciated as well. As I have yet to characterise what causes the automatic-routing-switchover, (sometimes it can happen after a few hours, other times after a few minutes) Niels http://nielsmayer.com