On Fri, Jun 25, 2010 at 04:55:39PM +0100, Steve Fosdick wrote: Could be that PA just doesn't know what to do with a card having so many channels. But you can run PA as a Jack client (don't ask me how, I don't use PA at all). Ciao, -- FA O tu, che porte, correndo si ? E guerra e morte !

I have a system with an RME HDSP card (Multiface II system) and I faced the exact same issue. So I hacked an ALSA 2 Jack bridge: http://alsa.opensrc.org/index.php/Jack_and_Loopback_device_as_Alsa-to-Jack_… Works really good for what I am using it for. No pulseaudio, no gstreamer, pure ALSA only. Good luck :) J.

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 ( http://old.nabble.com/Re:-QXGEdit-(was-Re:-Suzuki-QC1-Qchord-Digital-Songca… ) 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--to285130… ) but since I don't have an RME device I can only speak to my own experiences. Speaking of which... I recently switched to KDE ( http://old.nabble.com/KDE-4.4's-nice-integration-of-Pulseaudio,-ALSA,-a… ) 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" scenario). 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 output.... Niels http://nielsmayer.com

On Fri, Jun 25, 2010 at 5:55 PM, Steve Fosdick <sjflists(a)btinternet.com>wrote;wrote: In case it helps, there is a similar problem with the ICE1712 cards. See http://ubuntuforums.org/showthread.php?t=1360685 The hack explained in Comment #31 has solved the problem for me. I use pavucontrol to select the playback device and configuration (before, analog stereo was missing in the pavucontrol configuration tab). Maybe editing your alsa module conf file helps? Cheers! Pablo

On Fri, Jun 25, 2010 at 2:14 PM, Paul Davis <paul(a)linuxaudiosystems.com> wrote: to something not imply "alternative" ? sees gstreamer; in that gnome applications use settings from ~/.gconf/system/gstreamer/0.10/ to determine which sound device or services to use, whereas KDE applications have a more useful mechanism achieved via phonon priority list. Gstreamer applications just lockup or fail when a device is busy -- phonon does the right thing, and notifies you via GUI of what it is doing in case of a failure, and then automatically selects the next device in the priority list. Thus my statement "a more evolved and workable alternative to gstreamer." The only place gstreamer is explicitly seen to the KDE user is under System Settings->Multimedia->Backend where Xine is the preferred backend and Gstreamer is the secondary/fallback alternative. -- Niels http://nielsmayer.com