[LAU] RME HDSP Multiface and PulseAudio

Niels Mayer nielsmayer at gmail.com
Fri Jun 25 21:08:44 UTC 2010

To solve your pulseaudio/ALSA integration issue, you will need to
create an ALSA mapping that pulseaudio expects to find in place, and
never does with multi-channel studio-oriented sound-cards (
https://bugzilla.redhat.com/show_bug.cgi?id=499435 ).

The following details what needs to be done on an ICE1712 soundchip
present in many "studio" soundcards:
http://www.jrigg.co.uk/linuxaudio/ice1712multi.html ... even if you're
not using multiple soundcards, you'll undoubtedly need the ALSA
"multi" bridging trick (
) to combine the digital and analog parts of many soundcards for use
by either Jack or ALSA....I'm sure there's equivalent setups like this
for the RME ( http://old.nabble.com/Is-it-possible-to-use-two-RME-cards-in-sync--to28513035.html
) but since I don't have an RME device I can only speak to my own

Speaking of which...

I recently switched to KDE (
) which uses 'phonon' for audio support, a more evolved and workable
alternative to gstreamer. Switching to the  'amarok' or 'juk' players,
I now can use jack as my workaday audio server, and also retain the
same setup should some musical inspiration hit, which is often killed
by having to switch setups and context.

By setting KDE's System Settings->Multimedia->{Music, Video, Games,
Communications, etc}, you can specify a priority listing of audio
devices to try for each kind of activity.  I set 'Music' using 'jack'
as top priority; if jack is not running, it skips that facility and
goes on to talking to the device using ALSA... Independent of whether
jack is running, whenever any phonon music-playing happens, it
automatically connects to the first two channels of a 10 channel's PCM
output presented by snd_ice1712 module (driver for Terratec DMX6Fire,
M-Audio Delta 66, etc.). Phonon auto-connects to PCM1+2 which is
dedicated solely to headphones or headphone monitoring. My mains are
connected via SPDIF/TOSLINK (PCM9+10), and I use the ice1712's
built-in digital mixer (on capture11+12)  to mix outputs PCM1-10,
analog inputs capture1-6, and spdif inputs capture9+10: the digital
mixer output is routed to digital output driving the mains via
external DAC/etc.

With this setup, playing music via 'juk' 'amarok' or auditioning via
'dolphin', I can hear the music on the headphone monitors right away,
using soundcard's physical volume knobs for quick volume adjustment.
If I want to turn up the music on the mains, I just make sure
envy24control->Monitor PCMs->{PCM Out 1,PCM Out 2} are unmuted and/or
adjust the levels. If I want to jam along with something, "playing on
the radio", or capture and record something, it's just a matter of
launching the right programs, outputting to PCM2-8 if needed,
headphone monitoring on PCM1-2, and using the soundcard digital mixer
alongside qjackctl for routing. (Pretty good for a $20.00 ebay
Terratec DMX6fire).

I think such a setup, where jack essentially replaces pulseaudio, is
superior as long as you can assure your "bus mix" sampling rate is the
same as everything driving it -- so if you're listening to CD's, your
jack better be running at a multiple of 44.1K... With video , usually
at 48K or  higher rates, this causes issues (that can be worked around
but the end result isn't as "point and click" as the above "music"

Alternately, if you want to work at a 96K, or higher, for music, you
may not want to deal with all the resampling that might be needed to
match 44.1K for CD's, 48K for digital video/tv, and 96K or 192k for
HD... for that's why I leave a TOSLINK output/device "free" for ALSA
video use (like my $0.99 dynex sc-5.1 with an ICE1724 and an optical
out). Setting that as the preferred "video" device, phonon will do the
right thing and drive your device at the sampling rate of the source
material. And then I have to go across the room and use a mechanical
TOSLINK switch to switch inputs to the DAC to my "non-jack" optical


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