25 Jun
2010
25 Jun
'10
5:55 p.m.
I have, for the moment, my RME HDSP Multiface at home along with a nice
pair of Sennheiser headphones. The headphones show up the shortcomings
of the integrated audio I/F on the motherboard so I'd like to be able to
route "normal" audio to the RME, e.g. use it for things like listening
to music with rhtythmbox, which seems to mean persuading PulseAudio to
output to it.
So far the PulseAudio sound preferences does not see the HDSP even after
restarting PulseAudio. The kernel module is loaded, the firmware loaded
and the card works fine with jack and ardour.
What do I have to do to persuade PulseAudio to see it?
The distribution is Ubuntu 10.04. There is no /etc/asound.conf
or .asoundrc file. I did try creating a .soundrc to enable ALSA
applications to output to Jack in the hope this would cause PulseAudio
to offer jack as an output option but this didn't seem to make any
difference.
Interestingly aplay -l shows the card and aplay -L does not:
$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: DSP [Hammerfall DSP], device 0: RME Hammerfall DSP + Multiface
[RME Hammerfall DSP + Multiface]
Subdevices: 0/1
Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog]
Subdevices: 1/2
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
card 1: NVidia [HDA NVidia], device 1: VT1708B Digital [VT1708B Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
$ aplay -L
pulse
Playback/recording through the PulseAudio sound server
front:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
Front speakers
surround40:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
4.0 Surround output to Front and Rear speakers
surround41:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
4.1 Surround output to Front, Rear and Subwoofer speakers
surround50:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
5.0 Surround output to Front, Center and Rear speakers
surround51:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
5.1 Surround output to Front, Center, Rear and Subwoofer speakers
surround71:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Analog
7.1 Surround output to Front, Center, Side, Rear and Woofer speakers
iec958:CARD=NVidia,DEV=0
HDA NVidia, VT1708B Digital
IEC958 (S/PDIF) Digital Audio Output
Any ideas, please?
25 Jun
25 Jun
6:11 p.m.
On Fri, Jun 25, 2010 at 04:55:39PM +0100, Steve Fosdick wrote:
I have, for the moment, my RME HDSP Multiface at home
along with a nice
pair of Sennheiser headphones. The headphones show up the shortcomings
of the integrated audio I/F on the motherboard so I'd like to be able to
route "normal" audio to the RME, e.g. use it for things like listening
to music with rhtythmbox, which seems to mean persuading PulseAudio to
output to it.
So far the PulseAudio sound preferences does not see the HDSP even after
restarting PulseAudio. The kernel module is loaded, the firmware loaded
and the card works fine with jack and ardour.
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 !
6:52 p.m.
On Fri, Jun 25, 2010 at 11:55 AM, Steve Fosdick <sjflists(a)btinternet.com> wrote:
I have, for the moment, my RME HDSP Multiface at home
along with a nice
pair of Sennheiser headphones. The headphones show up the shortcomings
of the integrated audio I/F on the motherboard so I'd like to be able to
route "normal" audio to the RME, e.g. use it for things like listening
to music with rhtythmbox, which seems to mean persuading PulseAudio to
output to it.
no, it doesn't. gstreamer has a jack plugin, and you can just have RB
send directly to JACK with it. just set either your gstreamer
properties or system sound properties to use jackaudiosink the
gstreamer plugin has its own issues, the most annoying of which being
that it disconnects and reconnects for every song, but i use it to
listen to my music collection and i-radio all day while doing ardour
development.
as fons note, pulseaudio really doesn't have a clue what to do with a
card like the HDSP. it could be coerced into doing something
reasonable, but its hardly worth it.
7:21 p.m.
From: Paul Davis <paul(a)linuxaudiosystems.com>
Subject: Re: [LAU] RME HDSP Multiface and PulseAudio
To: "Steve Fosdick" <sjflists(a)btinternet.com>
Cc: linux-audio-user(a)lists.linuxaudio.org
Date: Friday, June 25, 2010, 12:52 PM
On Fri, Jun 25, 2010 at 11:55 AM,
Steve Fosdick <sjflists(a)btinternet.com>
wrote:
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.
I have, for the moment, my RME HDSP Multiface at
home
along with a nice
pair of Sennheiser headphones. The headphones
show
up the shortcomings
of the integrated audio I/F on the motherboard so
I'd
like to be able to
route "normal" audio to the RME, e.g.
use it for
things like listening
to music with rhtythmbox, which seems to mean
persuading PulseAudio to
output to it.
no, it doesn't. gstreamer has a jack plugin, and you can
just have RB
send directly to JACK with it. just set either your
gstreamer
properties or system sound properties to use jackaudiosink
the
gstreamer plugin has its own issues, the most annoying of
which being
that it disconnects and reconnects for every song, but i
use it to
listen to my music collection and i-radio all day while
doing ardour
development.
as fons note, pulseaudio really doesn't have a clue what to
do with a
card like the HDSP. it could be coerced into doing
something
reasonable, but its hardly worth it.
7:58 p.m.
On Fri, Jun 25, 2010 at 1:21 PM, James Warden <warjamy(a)yahoo.com> wrote:
Works really good for what I am using it for. No
pulseaudio, no gstreamer, pure ALSA only.
many people seem confused by what gstreamer is. its not anything like
JACK or PulseAudio or ALSA. if you use Rythmbox, for example, it will
be using gstreamer no matter how the audio gets out of your machine.
gstreamer is an intra-application framework for handling streaming
data. it has a variety of "plugins" that can be connected together to
form a pipeline *WITHIN* an application. one of those plugins is
generally going to be an output plugin, and it might be the ALSA
output plugin, the Pulse output plugin, the JACK output plugin etc.
etc. etc.
put differently, you can dump JACK, Pulseaudio, ALSA from your system
and it won't remove the use of gstreamer by applications, just like it
won't remove the use of libsndfile.
--p
11:08 p.m.
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
11:14 p.m.
On Fri, Jun 25, 2010 at 5:08 PM, Niels Mayer <nielsmayer(a)gmail.com> wrote:
) which uses 'phonon' for audio support, a
more evolved and workable
alternative to gstreamer. Switching to the 'amarok' or 'juk' players,
i have no vested interest in either phonon or gstreamer, but just to
keep the record straight, i don't believe that phonon is an
alternative to gstreamer.
in fact, if you look on this page:
http://phonon.kde.org/cms/1034
you'll see that its actually a wrapper around existing libraries like gstreamer.
26 Jun
26 Jun
2:12 a.m.
On Fri, Jun 25, 2010 at 2:14 PM, Paul Davis <paul(a)linuxaudiosystems.com> wrote:
...i don't believe that phonon is an alternative
to gstreamer.
in fact, if you look on this page:
http://phonon.kde.org/cms/1034
you'll see that its actually a wrapper around existing libraries like gstreamer
From a user-perspective, how does "wrapper"
or any meta-level approach
to something not imply "alternative" ?
From a user's perspective the kde user sees
phonon&xine, and gnome
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
4:13 a.m.
On Fri, Jun 25, 2010 at 8:12 PM, Niels Mayer <nielsmayer(a)gmail.com> wrote:
On Fri, Jun 25, 2010 at 2:14 PM, Paul Davis
<paul(a)linuxaudiosystems.com> wrote:
well, partly because it means that any failings in whatever is wrapped
are likely to exist in the wrapper too. if the backend can't do X,
then neither can the wrapper.
...i don't believe that phonon is an
alternative to gstreamer.
in fact, if you look on this page:
http://phonon.kde.org/cms/1034
you'll see that its actually a wrapper around existing libraries like gstreamer
From a user-perspective, how does "wrapper" or any meta-level approach
to something not imply "alternative" ? Gstreamer applications just lockup
or fail when a device is busy
this depends entirely on the output plugin used, and what parameters
it is used with (as demonstrated by the way that Phonon manages to use
it).
25 Jun
25 Jun
11:25 p.m.
On Fri, Jun 25, 2010 at 5:55 PM, Steve Fosdick <sjflists(a)btinternet.com>wrote;wrote:
So far the PulseAudio sound preferences does not see the HDSP even after
restarting PulseAudio. The kernel module is loaded, the firmware loaded
and the card works fine with jack and ardour.
What do I have to do to persuade PulseAudio to see it?
The distribution is Ubuntu 10.04.
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
26 Jun
26 Jun
1:25 a.m.
On Fri, Jun 25, 2010 at 2:25 PM, Pablo Fernandez <pablo.fbus(a)gmail.com> wrote:
In case it helps, there is a similar problem with the
ICE1712 cards.
See http://ubuntuforums.org/showthread.php?t=1360685
Indeed, as Lennart Poettering points out on 2009-08-17 in
https://bugzilla.redhat.com/show_bug.cgi?id=499435 (link from my prev
post):
http://pulseaudio.org/ticket/624
See also https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/178442
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10 comments
6 participants
participants (6)
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fons＠kokkinizita.net -
James Warden -
Niels Mayer -
Pablo Fernandez -
Paul Davis -
Steve Fosdick