Hi,
One thing that I've found recently is the Antec Aria case
(http://www.antec-inc.com/aria.html) which fit regular mATX
motherboards, and since it's mATX most motherboards will handle 1 AGP
and 3 PCI cards. It's definitely a little bigger than a Shuttle, but
gives you a lot more choice in chipsets and processors.
later,
Steve
derek holzer wrote:
> Hi all,
>
> resurrecting an old thread for my own evil purposes ;-)
>
> I am looking to buy a Shuttle SN41G2 for use in an audio installation at
> European Media Arts Festival this month. Everything looks cool, except
> that VGA Shared Memory. As a devout follower of RME, I read everything
> on their site about hardware recommendations, and one of the big
> warnings is this Shared Graphics Memory. This means that the graphics
> card uses normal RAM for its graphics-handling. The Shuttle says it
> comes with a Geforce4 MX, and that it "features" 128 Mb of shared
> memory. This sounds to me more like 128 Mb that gets taken away from
> audio processing!!
>
> Can anybody tell me if they have run into audio dropout problems with
> this machine, and most importantly while using an add-on soundcard. I
> will be using a Delta 66 in this machine. I know that it gets glitches
> with PD + ALSA with default settings on the built-in Realtek card, but I
> won't be using that anyway....
>
> Other problem I know of is that new kernel modules are needed for the
> built-in NIC ethernet card, but I can handle that.
>
> robcanning wrote:
>
>> Does anybody know if i can access more than 2 the 5.1 outputs on the
>> onboard audio chip with PD or should i go ahead and get an SBLive for 4
>> outs?
>
>
> To answer Rob's question, look for the thread on pd list called "nForce
> chipset". It was generally concluded that using the 5.1 outputs calls on
> *extremely* proprietary Dolby [en/de]coding.
>
> best,
> Derek
>
hi,
I just bought one of these little boxes (Shuttle Spacewalker XPC SN41G2
nForce2 Aluminium Barebones System - AMD Athlon XP to use for an
installation and have installed redhat9 on it and soon will be
planetccrmaing it
- it is a very quiet machine as it comes and i believe you can get an
optional super quiet power supply for it - very easy to build/configure
- i'll let you know if i run into any audio worries.
Does anybody know if i can access more than 2 the 5.1 outputs on the
onboard audio chip with PD or should i go ahead and get an SBLive for 4
outs?
nVida Corporation nForce Multimedia audio controller [VIA VT82C686B]
(rev a2)
nVida Corporation nforce AC97 audio controller (MCP) (rev 1a)
rob
--
robcanning <rscanning(a)eircom.net>
www.robcanning.utvinternet.com
OK, here is what I want to do:
1) Take my linux laptop (now with Fedora core 1) and connect a
signal to the line-in jack and record audio to the laptop hard
drive. Ideally I would like to do something like generate an
mp3 file to save hard drive space. The goal here is to record
conference speakers and generate a CD containing MP3 files of
a days worth of people speaking. (Using something like MP3 is
not essential, but would save space on the hard drive). Since
all I am after is decent vocal audibility, I don't need exquisite
audio. I will be getting a feed from a mixer.
My laptop is a Toshiba Tecra 8100, which has a Yamaha YMF-744B
(supposedly this was supported under ALSA 0.5.8)
2) What I have done so far ...
(apart from just get bewildered by all the audio jargon under
linux -- there is OSS, ALSA, JACK, ... maybe that is most of what
I have tripped over so far)
Find fedora RPM's of the alsa stuff (vintage 1.0.2) and load them
onto the laptop.
Read stuff on the ALSA site and linux-sound.org (which I am
still doing).
I was playing yesterday on my desktop system -- I installed a
Sound Blaster Live card and had it playing a CD using the OSS
drivers, now I will see if it still works with the ALSA drivers.
I couldn't get gnome-sound-recorder to do anything, but maybe
it will work with the ALSA drivers. I am looking at an application
called qarecord (I like it since it has level meters -- I want to
see something move when I squak into a microphone and the gnome
recorder seems to have nothing like this), I have got the source,
but will have to build it.
So what am I asking? Any help and pointers -- I don't want to make
this unduly complicated. I wish I could just hook up a signal source
bring up a GUI, click on record and have a .wav file coming out of
stdout that I could pipe to some compression tool. Surely one
of you out there has done this -- my goal was to test fly this
Wednesday night, but so far I am a day and 1/2 into this and am
just getting deeper and deeper without seeing progress towards my
goal.
Thanks for anyone who has the time to offer some help.
Tom
--
Tom Trebisky
MMT Observatory
University of Arizona -- Tucson
tom(a)mmto.org
Hi
I released sfc-0.016 (SoundFontCombi) a MIDI router who emulates a
synthesizer, storing routes as sounds using MIDI soundfonts or other MIDI
capable devices.
News on v0.016
-----------------------
MIDI channel for own programs
Route MIDI Program Change
Refresh MIDI data when activate section.
MIDI Change Bank improved.
Fixed bugs in command line, MIDI BankLSB message, control messages and
auto-connect ALSA sequencer port.
Reduced the amount of memory needed.
sfc is available in:
http://personal.telefonica.terra.es/web/soudfontcombi/sfc-0.016.tar.gzhttp://www.telefonica.net/web/soudfontcombi/sfc-0.016.tar.gz
Hi,
I have some problems with 6 channel sis 7012 device - I'm using it as 3
stereo channels. I trigger 3 different instances of aplay at once and I get
underrun messages from aplay.
What does this mean ?
How to solve this (bigger buffer? ) ?
It sounds like it is interleaved in short chunks (all channels are heard on
all channels) - so it's not really bearable.
Regards,
Robert.
Hi!
> So, can it be done ? I tried using holborn's SoundFontCombi but
> couldn't get it to keep the settings I made. Every time I started
> playing a sequence it would flip back to its original state. Sigh...
probably that's because sfc recognizes MIDI program change for their own
programs and obviously changes .. good point ... i will put some
enable/disable button for that .. :-)
Thank you ..
Josep
Hi, just back from Musikmesse in Frankfurt.
FYI:
Videos of Mediastation X-76 and Lionstracs - Thomas Organ Musicstation
VKX-76
(basically the mediastation with 2 manuals, pedals, speakers in a wooden
case)
on the right side of the page, scroll down to VIDEO OF MUSIKMESSE,
you will find 4 videos
(under Linux you can play them with xine or mplayer if you have the
win32 codecs installed)
http://www.lionstracs.com/index.php?module=Static_Docs&func=view&f=/demos.h…
read these two links too:
http://www.synthzone.com/ubbs/Forum37/HTML/008798.htmlhttp://www.synthzone.com/ubbs/Forum37/HTML/008794.html
(in the organ videos Bernd Wurzenrainer plays the NI B4 under VST server
with a jazz base (.wav) :-) )
Some LADers that were at Musikmesse: Marek Peteraj, Frank Neumann,
Matthias Nagoni, Fons A. (aeolus).
Companies using Linux in musical gear besides Lionstracs: Plugzilla (a
rack that can play VSTs), Muse Receptor (similar concept), Hartman
Neuron (a synth). Unfortunately the others are based on pretty
closed design and most don't even tell you that's based on Linux.
Perhaps their attitude will change
in future.
As always thanks to everyone that contributes to Linux and Linux audio,
without these people these
musical instruments would not be a reality today.
cheers,
Benno
Greetings:
Is there some way to tell Qsynth that I want it to receive only on
certain channels and ignore the others ? I could really use that feature...
Best,
dp
The [1]Linux Audio Development web site talks about this mailing list,
unfortunately [2]the linked homepage seem to be under construction.
As I am learning both the french language, and Linux audio
development, being subscribed to that mailing list, if exists, would
be very useful for me.
Any feedback is welcome, thanks in advance.
Cordially, Ismael
1. http://www.linuxdj.com/audio/lad/resourceslists.php3
2. http://membres.lycos.fr/linuxsound/
Hi,
I am having a bit of trouble running Jack with the Emagic Emi 6|2m USB
audio box. The device works well with 2 inputs and 2 outputs at 96/24 or
less. It also works well with all 6 inputs and 2 outputs at 44.1/16.
Starting Jack makes the Emi correctly set the sample rate and the bit depth.
The problem is that when 6 channels are recorded with 24-bit depth the
Emi is able to provide 2 16-bit output channels at 44.1 KHz or no output
at 48 KHz.
Running jack with no -C or -P options tries to record 6 channels and
play back 2 channels at 24-bith which is impossible by the hardware. As
a result I get a lot of noise and mix-ups in the audio when trying to
record or play back something.
So - I would need to be able to tell jack to open the capture ports at
24-bit and playback at 16-bit.
Any ideas on how to do this?
Could I possibly create aliases for the capture and playback with the
desired options in my .aousndrc and use them with the Jack -C and -P
switches? If yes - how?
I heard that ecasound does use the card correctly so there should not be
anything wrong with the driver.
Below is what /proc/asound/card1/stream0 says.
Thanks,
-Jorma
Emagic GmbH Emagic EMI 6|2 m at usb-00:03.1-3.1 : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S16_LE
Channels: 2
Endpoint: 10 OUT (ASYNC)
Rates: 44100, 48000, 96000
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 10 OUT (ASYNC)
Rates: 44100, 48000, 96000
Capture:
Status: Stop
Interface 2
Altset 1
Format: S16_LE
Channels: 6
Endpoint: 12 IN (ASYNC)
Rates: 44100, 48000
Interface 2
Altset 2
Format: S24_3LE
Channels: 6
Endpoint: 12 IN (ASYNC)
Rates: 44100, 48000
Interface 2
Altset 3
Format: S16_LE
Channels: 2
Endpoint: 12 IN (ASYNC)
Rates: 44100, 48000, 96000
Interface 2
Altset 4
Format: S24_3LE
Channels: 2
Endpoint: 12 IN (ASYNC)
Rates: 44100, 48000, 96000
__
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