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
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