I have been reading about the -ck patchset
(http://ck.kolivas.org/faqs/audio_hints,
http://members.optusnet.com.au/ckolivas/kernel/) and was wondering if
it is or is not good for audio?
The reason I am asking is that Ubuntu's kernel is patched and a lot of
users rely on those patches, and although I have received a -rt patch
that applies to the Ubuntu kernel, it does give me many issues
(interestingly, only with RT apps like JACK and the like).
The Ubuntu kernel can be patched with -ck rather easily, and most
people that have tried it seem to say they have been results than
using the standard Ubuntu kernel (me excluded).
I saw that it uses CFQ as the default scheduler, but that can be
changed back to anticipatory easily enough.
I understand that Ingo's -rt patches are best, but is the -ck patchset
better than a vanilla kernel, for audio, as a medium? Why or why not?
Also, why does Studio To Go! use it if it is bad?
Thanks for any insight in advance.
Dana
----- Esben Stien <b0ef(a)esben-stien.name> wrote:
> "Mike Taht" <mike.taht(a)gmail.com> writes:
>
> > a call in radio station
>
> This is my idea as well. I want listeners to call in using VOIP.
>
> > If ladspa plugins could be run through asterisk or a jack compliant
> > sip phone you could give your outgoing voice calls a little bass
> boost
> > for that "voice of god" effect...
>
> This is fully possible and it's working without problems for me using
> oss2jack.
In my experience oss2jack is /very/ application dependent. What softphone did you successfully use it with?
I too have used a softphone for a radio station, but that was running on a sepperate computer and going through a hardware mixing board, using the classic aux send phone hybrid style. The addition of seperate input and output with the softphone makes a traditional hybrid obsolete, which made me very happy. All of this can work if a softphone which runs on Linux can connect to JACK, which I think is the point of this discussion.
-lee
>From: timg <timg(a)expressmart.com>
>
>But if these actions are found to be within legal limits what can be done?
>I must admint that a company that does not invest in it's own R&D is one
>that will fail... and lose (at least my) respect.
Monopoly game --- 500 million have played, inventor became millionare.
Except -- the inventor stole and patented the game from public domain.
Zero R&D; even a mispelling was copied to the patented game.
And the game was not sold cheap like the Behringers.
(Hasbro still repeats the wrong history of the game at their webpages.
Monopoly aged to 70 a few weeks ago and we all heard the wrong history.
I got interested in because I wondered why I have not seen any similar
games in the shops -- cheaper clone games, say.
But check Atlantik for open source computer game.
I'm still looking for print-your-own-Monopoly PDF files or such.)
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
Hello, all,
I'm a new subscriber to this list, but I've been a long time linux
user, even using audio for quite some time. I want to move up to "the
next level", and am having problems. I tried to gather information
from the list archives, but I didn't get too far, thought I'd just
ask.
I have 3 basic questions for now. I'll follow them with additional
background information that you might need to answer the questions.
1. M-Audio Quattro, can it be made to work? I now know nobody seems
to have a very high opinion of it.
I am having problems using it (surprise). I'm running Fedora
Core 3, with some stuff installed from CCRMA. Kernel is
2.6.11-0.3.rdt.rhfc3.ccrma, /proc/asound/version gives:
Advanced Linux Sound Architecture Driver Version 1.0.9rc1.
I used the sample Quattro asoundrc file from alsa-project.org (in a
message from Patrick Shirkey from 2002), and the example arecord
command lines given (like arecord -r 44100 -c 4 -f s16_le -D q4 -d 5
/home/xxx/q4.wav). I seem to still have endian problems -- playback
sounds like a bunch of loud white noise unless I reduce the input
level to something very low -- which I assume means I get most samples
to fit in the least significant byte with zeroes elsewhere. I've read
about some patches, but it seemed like they should have been already
incorporated?
Is there at least one person out there that has this interface work
(well?) for them? If so, I'd be very interested in exchanging email
with you.
2. Is a 700 MHz athlon a reasonable system for a few tracks of audio
with Jackd and Ardour?
I gave up on the Quattro for a while, and installed a SoundBlaster
Live Value. I got it to work quite well. But I'm getting xruns all
quite often from jackd (version 0.99.36) with ardour. Asside from
installing the planet CCRMA kernel, I haven't begun to really try and
optimize settings yet. But I'd like to know what I can reasonably
expect from this older system. (As I mention below, it has "only"
192M of RAM, maybe that's a factor.)
3. Can ALSA use a Sound Blaster Audigy 2 to get 24 bit audio? Can it
do four channels in and four out simultaneously?
I don't have the money to spend on a better supported Pro level audio
interface right now (assuming I have to give up on the Quattro). But
I think I want something a bit (few bits?) better than the SB Live,
so I won't be disappointed down the road at the audio work I do
today. But I saw an old message that said the 24 bit converters were
different than the 16 bit converters, and that they weren't supported
yet. Did they become supported at some time?
Thanks in advance for any replies!
Here's some DETAILS that may fill in the blanks:
I've been running linux for a long time; I have some Slackware 96
disks, that may have been my first linux load at home. Actually ran
Novel Unixware at home before that, and have run various versions of
Red Hat (4 through 9) and now Fedora.
My main linux machine is much faster, but does some server things for
the rest of the house (mail service, file service, DNS, etc.) that I
don't think I want on the same system I'm using for audio work. I've
now started to believe I should migrate all that low-priority stuff
off to a lesser machine, but don't have a whole lot of spare time on
my hands to make such a change...
The 700MHz athalon system has only 192M ram, which might contribute to
my xrun problems. It dual boots with Win98. I intend to make it my
dedicated audio system. (If this proves too slow, I've been thinking
of putting together a new machine to be a MythTV box, and I could
probably make that a dual purpose machine, sometimes running mythtv,
sometimes running audio.)
As far as what to record into it, besides my son who's starting to
play guitar: I have a couple of synths (Ensoniq VFX and Roland
JV-1080), a few mics (best ones are two shure sm58s), a Mackie MS-1202
mixer (original, not newer VLZ), and a Behringer DDX3216 digital mixer
with ADAT interface card, that I picked up when it went on blowout
pricing, just after I purchased the Quattro. An audio interface that
worked with linux and had an ADAT interface would be REALLY SWEET with
this setup. But I don't think I can afford the RME stuff. (If
there's any place I can help apply pressure to get Emu to release the
details of the 1212M card, sign me up!)
I also have my turntable and cassette deck hooked up so I can make CDs
and MP3s of my old albums. And I have a MiniDisc recorder that I've
done some "bootleg" style recordings of musicians at my church, that I
then run into the computer and make CDs from as well.
Up till now, for most of my recording to computer I've used a program
I wrote myself (by the dates on the files, goes back to 1997!) when I
couldn't find anything linux based with VU meters for recording. It
is a simple, console and OSS based application, strictly 16-bit 44.1k
stereo, and it is far surpassed by the stuff that's out there now.
But I am used to it... I have a small script that follows recording
the wav file with two passes of sox to calculate the scale value and
then normalize the file -- if no scaling needs to be done, I usually
assume I clipped somewhere and go back and re-record.
That worked on a 100MHz pentium, so I expect I should be able to do
similar things with a 700MHz athalon and current software, but I might
be wrong which is why I ask.
My needs right now really don't reach much past what one could do with
a cassette porta-studio, or the current digital equivalents. I just
have this stuff I've accumulated, and want to make it useful!
My wish to go to 24 bits is based on this: my target is 16 bit CDs,
and I want a few extra bits resolution so I can record low enough to
not worry about clipping, and still have (pretty much) full 16 bits
resolution after normalizing. So even 20 bits is probably enough, and
I figure an SB Audigy 2 might not have the greatest noise floor but
would probably still meet those needs. If anyone's read this far, let
me know if that sounds spot-on, or misguided.
I bought the Quattro after checking the Alsa soundcard matrix. But I
didn't do a google search until after I had problems getting it to
work. I'm thinking maybe the matrix should be edited and have that
entry marked as not having the greatest success rate!
Again, thanks to all of you for your time in reading this, and for any
help you provide!
--> Steve
--
Steve Wahl steve(a)pro-ns.net
Fools ignore complexity. Pragmatists suffer it.
Some can avoid it. Geniuses remove it.
-- Perlis's Programming Proverb #58, SIGPLAN Notices, Sept. 1982
Hello all,
I recently obliterated my website while trying to do a Wordpress
upgrade and lost all of my content. (Note to self - back up your
databases!) I am trying to rebuild it, and I need your help. I try
to evangelize Linux, and more specifically Ubuntu, in my little corner
of the world. I know that there are a few of you that have blogs or
websites specifically relating to this. (At one time, there was an
Ubuntu Studio website, and another individual named Oktobyr that had a
website with some Ubuntu information.) If you have any links that you
believe would be helpful, please send them to me at
hardbop200(a)gmail.com so I can rebuild my Linux audio section of the
site.
Thank you very much for your help.
--
Josh Lawrence
http://www.hardbop200.com
>Since the original posting here was about a commercial product, I
>thought it was inappropriate ('cause spam isn't allow, I can't see how
>this is any different)
> . . . . .
>Just wanted to explain my reason for asking him. I didn't want to tell
>him NOT to ask for money for his product (however I do question the
>appropriateness of posting it here).
Since Artemiy is obviously a member of the oss community - making many free
contributions himself, and offering a very inexpensive product as well, I
didn't really view this as spam - but yes, perhaps this list wasn't the
appropriate place to promote it.
What I took exception to was the censure he seemed to be receiving for
asking money for his product..
>Ah, I was wondering where the 'OT' dicussions were... LOL, not really OT
>but simply with a philosophical focus, and just as important.
:)
Until I am much more code-savvy and knowledgeable about electronics, I'm
afraid 'OT' is mostly all I can contribute.
But this community is centered around certain philosophies bearing on
copyright, free software, free music, etc., and these philosphies need to
be periodically revisited and subject to vigorous debate.
We're obviously not 'there' yet with it all - lot's of issues still need to
be worked out.
>I believe the reason why digital products with a cost are frowned upon
>so heavily is simply that there is no cost (or very little cost) for
>reproduction,
'Reproduction' is not the only cost factor.
There is a considerable cost in *production* that must also be taken into
account.
Computer hardware and high-quality audio hardware are not cheap, nor are
fine musical instruments.
Neither are the equipment and supplies needed to manufacture and ship your
own CDs, and neither is the cost of hosting a website, maintaining a
merchant account, etc.
Costs of production/reproduction aside, there remains the intrinsic worth
of the creative work itself.
Even were it to cost nothing to produce, a beautiful work of art still
holds value simply for what it is.
>I do not believe that proper compensation is not possible under these
>circumstances. The proper way to do it has simply not been implemented
>yet very widely.
Agreed.
>This guy
>
>http://tipping.selfpromotion.com
>
>might be a good place to start looking.
>If you start a tipping economy, AND you assume that everyone will tip
>his or her highest price possible (which they will, if they are grateful
>enough for your product and have been able to earn some money with it as
>well), and further assume they will if you succeed in making them very
>grateful for your service, you need no such compomise; your yield is
>represented by the area of the complete triangle.
>
>Not to mention free viral advertising.
Yup - no better advertising than word-of-mouth.
>For linux software (and other free software) what we need is a good
>method for tipping distribution, because there are so many people
>involved who all deserve their fair share, so people simply can tip to
>'linux audio' and the bulk will be covered... including kernel,
>libraries... And they then may proceed to tipping towards special
>sympathies :)
>
>You know, I just know this can work, and I'm looking forward to
I think so too, but as I said - we're not there yet.
As for tipping, it is a good approach, but doesn't seem to be working very
well yet.
Perhaps that is due to the fact that we don't have what you call a tipping
'economy'.
I wish very much that I could just tip for electricity, internet access,
web hosting, food, shelter, etc., etc.
Tipping is an intermediate stage between a full-scale currency-based
economy and a barter-based one.
Believe me, I despise currency probably more than anyone here.
I wrote a treatise against the concept of 'money' and in favor of barter
when I was 6 years old, and have not altered my views about it
significantly since then. :).
(However I don't really see how technological innovations such as we now
enjoy could have come about without it).
>To me the hypocrisy is questioning about these issues without taking the
>tipping option into consideration. Personally, I went in this direction.
Wherever did you get the idea that I wasn't taking tipping into
consideration?
I was taking issue with the concept that music or software must be *free*
as opposed to receiving reasonable remuneration for one's effort.
Tipping *is* a form of remuneration.
>Is it working?
>
>Not yet.
It obviously is not working well at this time.
If someone is making a modest living off of tips, I would really love to
hear about it and learn their techniques.
I think Ardour is a perfect case in point.
As I understand it, Paul has received little more than $2K in donations -
mostly from *one* individual - vs the $60K+ of his own he has put into
development.
He had to go to work part or full-time (I'm not sure which) to make up for
the lack of support in the way of donations.
I don't think anyone here questions the value of this great software and
its importance to the community, but 'tipping' doesn't seem to be working
here, does it?
This is not right.
>The reason - and this is the strongest hypocrisy - is also that people
>think that if you don't sell your work, but just ask for voluntary
>donations, you can afford it.
It's actually worse than that: many people think that if you don't sell
your work, it is because it is not *worth* anything.
I know of many cases in the real world, where people selling various forms
of arts and crafts actually experienced an *increase* in sales when they
raised their prices considerably.
There are some very obtuse mindsets at work here - brand-name snobbery, and
the like.
>Another reason, and this is why I was sarcastic with your first post,
>Maluvia, is that there are still people who believe that a printed cd
>sounds better than a cd-r or a flac file downloaded from the Net. Or
>just that they're two completely differen things.
Ah - another misunderstanding there, I think.
I was not equating what I perceive to be 'audiophile' quality of music with
its monetary worth.
There is the music itself - the content which has intrinsic worth, no
matter how poor the production quality may be.
And it is precisely this value which seems to be getting ignored in all
this discussion of 'cost of reproduction', etc.
The creative work has value in and of itself, regardless of the tangible
form it takes or means by which it is distributed.
>I contacted the same
>magazines that reviewed some of my printed records rating them as
>masterpieces of the genre. When they saw that I was giving my newer
>stuff as free downloads they din't even care to listen.
That is utterly absurd, but ubiquitous in this society.
It is precisely the type of shallow mindset I was referring to earlier.
>Dana, to me it is great that you put a donation button on
>ubuntustudio.org. And your ambition should be that one day this could
>give you even a greater income than just enough to pay the hosting fees.
Yes! - if there is not an actual 'profit', then it is still 'free' from the
producer's standpoint.
>It won't sound better, but there are other advantages to buying a real
>CD (even if it's a CDR that the band's produced themselves) -- having
>a nicely-printed case with liner notes, and having a physical artefact
>that represents the music you've paid for. . . .
>but I'm quite happy to buy CDs from musicians who've made their music
>available online (and have done, many times -- it's something like a
>third of my CD collection now).
> . . . .
>>music. I don't think it's a business model that's going to make anyone
>>fantastically wealthy, but it does seem to work pretty well for lots
>>of musicians at the moment...
>>
>Not judging from what I read on forums. And anyway it is only a matter
>of time.
I doubt anyone is making a living from this yet, but some bands are having
varying degrees of success selling their CDs.
We did best just selling them at live performances - I'd say 90% of our CD
sales have been at gigs.
>For instance, I bought 20+ cds (and paid lot for them, since they were
>not printed in Italy but imported from england) in the past from a band
>who put a lot of emphasis on the fetishism of their phisical artefacts.
>They also used to emphasize the fact that those were *limited editions*,
>but the truth were that they just couldn't afford to print more copies,
>or they knew they wouldn't have sold more copies anyway.
>
>Now they're selling their whole catalogue on iTunes.
>
>Have they lost credibility? To me, yes.
Not sure I'm following your reasoning here.
This would certainly undermine the idea that their 'limited artifacts' have
some enhanced value, but doesn't make them totally worthless.
That would be the same reasoning adopted by the magazines that wouldn't
listen to your music once they realized you were offering some of it for
free.
We sell our music both in CD form and through iTunes and other
paid-download sites.
I see nothing wrong with this, except that what we make from the downloads
is negligible.
But we'll take it - not any different from taking 'tips' for the music.
>there is a difference between a CD-R and a CD... not directly the sound
>quality, but how long the data is preserved... CD-R's decay much faster
>than printed CD's, even faster when not stored right. Of course, this
>also depends on the brand of CD-R you get: some are better than others.
>So in the end, it may be cheaper to get the real CD instead of a burned
>copy of it... as you have to renew the second one from time to time. Of
>course, if you don't like the music anymore after a few years, then
>there's no problem...
I think that burning the master audio files directly to the marketed CD-Rs
- if they are of high quality, like Mitsui - can yield a superior sounding
product vs. pressed CDs - but that is due to my perception of generational
degradation involved going from the original masters to the final pressed
product.
(Don't want to debate that one again here. :) )
The durability of this media is another matter altogether, but even pressed
CDs have a limited life-span.
I've even heard of peoples CDs deteriorating due to some kind of
plastic-eating fungus!
>But my argument was that people's perception of an artefact that stores
>a digitalized information (music in this case) is still tied with the
>physical value, when what matter are just the bits.
Not just the bits - what the bits represent - the artistic content matters.
Not all compositions are created equal.
That is why Pat Metheny can still sell his "New Chautauqua" CD for $18 even
though it's 27 years old.
>Once something has been digitalized it's archived for the eternity, or
>better as long as somebody owns a backup. This is also saving old
>records and films that would've been lost otherwise.
I don't think there yet exists a media which has virtually 'eternal'
archival qualities.
Perhaps when we can learn to encode data into crystal lattices, or as
Arthur C. Clarke envisioned, lattices of light?
In short, I think tipping *is* a good step on the way to reforming the way
in which artists, programmers, etc. - maybe everyone - are compensated for
their work, but it must all be in the context of a more general reformation
in how goods and services are exchanged and paid for.
Removing the middleman whenever possible is a very good way to reduce the
costs of delivering goods and hence reducing the costs needing to be
recouped by the producer.
That is one of the things the internet makes possible.
Those who are willing to adjust their lifestyle to a more modest,
sustainable, and self-sufficient one will certainly be in the best position
to take advantage of these new models of trade and exchange.
And I sincerely hope it will become possible someday for technological
innovation and specialization to exist without the need for currency.
- Maluvia
Peder Hedlund writes:
> Have you tried replacing "-t ub" with "-t uw" ?
That created an interesting effect. The original data are
unsigned linear 8-bit samples. Silence is represented by 0x7f or 0x80
with 0 being the low extreme and 0xff being the highest extreme.
Using uw or unsigned words told sox that this was 8,000 16-bit samples
per second. It did that just fine and produced audio that was twice
the correct pitch.
> There's also a "-u" flag for "unsigned linear".
> What does 'file' say about your file?
There is no header on that file so it just says "data."
The -u flag appeared to have no effect since sox already
understood the data were unsigned linear.
sox: resample opts: Kaiser window, cutoff 0.950000, beta 16.000000
sox: Input file cdda.ub: using sample rate 8000
size bytes, encoding unsigned, 1 channel
sox: Do not support unsigned with 16-bit data. Forcing to Signed.
sox: Writing Wave file: Microsoft PCM format, 2 channels, 44100 samp/sec
sox: 176400 byte/sec, 4 block align, 16 bits/samp
sox: Output file output.wav: using sample rate 44100
size shorts, encoding signed (2's complement), 2 channels
sox: Output file: comment "Processed by SoX"
sox: resample: rate ratio 80:441, coeff interpolation not needed
sox: Finished writing Wave file, 125728820 data bytes 62864410 samples
I began to wonder if the problem was in the resampling
algorithm. If you listen to the sound, the glitches are at regular
intervals at about 10 hits per second. When there is voice present,
it sounds as if the "sixty Minutes" stopwatch was ticking away at
around ten ticks per second and there were little segments missing
from words. As I previously sed, the pitch was correct.
I then tried the following line to change the algorithm:
sox -V -r8000 cdda.ub -t wav -c 2 -w -r44100 output.wav polyphase .95
You can't tell any audible difference between the upconverted
.wav file and the original unsigned linear data file.
Being at 8000 samples/sec, the audio is voice grade and is a
recording of two-way radio communications so it isn't the best anyway,
but it appears that using the polyphase algorithm fixed the problem.
The latest Debian port of sox is
sox: Version 12.17.7. I am not sure but what the version was
different a couple of years ago when I first wrote that script using
the resample algorithm instead of polyphase. Maybe the previous
version of sox had some different defaults and it forced polyphase,
somehow.
Thanks, Peder, for your suggestions on things to try as they
got me to look at everything more closely.
Martin McCormick WB5AGZ Stillwater, OK
Systems Engineer
OSU Information Technology Department Network Operations Group
Peder Hedlund writes:
> If the file sounds corrupt then perhaps it is corrupted during
> the capture.
If I cat the original PCM file which is unsigned 8-bit mono
>/dev/dsp, it sounds normal. I think the corruption occurs when that
unsigned byte audio is read as signed audio.
Thank you.
Martin McCormick WB5AGZ Stillwater, OK
Systems Engineer
OSU Information Technology Department Network Operations Group
hi crew, this issue is probably getting boring now but i've hit it.
i am trying to get my Multiface PCMCIA card to work under linux and
have tried many things but still a bit lost. i have it running on my
windows setup on the same computer(another partition) without a hitch.
for this reason i can't understnd why it would not function here in
linux.
i am running an Asus v6800v (v6v) laptop and would really appriciate
some advice on where to move next...
i have reached the point where alsa is set up
.....
localhost tom # cat /proc/asound/cards
0 [DSP ]: H-DSP - Hammerfall DSP
RME Hammerfall DSP + Multiface at 0x54000000, irq 17
1 [ICH6 ]: ICH4 - Intel ICH6
Intel ICH6 with ALC650F at 0xdffff800, irq 17
.....
lspci | grep CardBus; lspci | grep RME
03:01.0 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev b3)
04:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall
DSP (rev 32)
.....
uname -a
Linux localhost 2.6.14-gentoo-r5 #1 SMP PREEMPT Tue Jan 17 05:35:27
EST 2006 i686 Intel(R) Pentium(R) M processor 1.73GHz GenuineIntel
GNU/Linux
.....
i have tried using setpci latency scripts found on this thread http :
lists.infradead.org/pipermail/linux-pcmcia/2004-April/000750.html
However, i must admit i'm a really lost when it comes to this level of
computing. so i'm stabbing in the dark and currently trying to learn
about what all this means.
script written by Daniel Ritz
> #!/bin/sh
>
> # set lateny timers for the bridges
> CB=`lspci | grep CardBus | cut -d" " -f1`
> for i in $CB; do
> echo "setting latency timer for CB $i"
> setpci -s $i 0x0d.b=0xff > /dev/null
> setpci -s $i 0x1b.b=0xff > /dev/null
>
> # for EnE only, others should ignore it
> setpci -s $i 0xc9.b=0x06 > /dev/null
> done
this script did something but i'm not entirely sure what? pd seems to
detect audio input but can't output,(anyone know if i need to compile
pd with rme support to make it work.)
after doing this i can get sound but its all distortorted and
horrible. it feels like on the right track here but i don't understand
the jargon or where to go next with it?
......
no problems loading hdsploader firmware
......
localhost tom # hdsploader
hdsploader - firmware loader for RME Hammerfall DSP cards
Looking for HDSP + Multiface or Digiface cards :
Card 0 : RME Hammerfall DSP at 0x54000000, irq 17
Upload firmware for card hw:0
Firmware uploaded for card hw:0
Card 1 : Intel ICH6 with ALC650F at 0xdffff800, irq 17
......
when trying to access change settings on hdspconf
......
tom@localhost ~ $ hdspconf
HDSPConf 1.4 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org>
This program comes WITH ABSOLUTELY NO WARRANTY
HDSPConf is free software, see the file copying for details
Looking for HDSP cards :
Card 0 : RME Hammerfall DSP + Multiface at 0x54000000, irq 17
Multiface found !
Card 1 : Intel ICH6 with ALC650F at 0xdffff800, irq 17
1 Hammerfall DSP card found.
Error accessing ctl interface on card hw:0
.....
i have read that some people have had problems when modules are set on
the same IRQ.
i have no idea if this is a problem as my RME cardbus, wireless
(ipw2200), onboard sound (AC97) and Yenta are all on IRQ 17. this is
the same in windows and my bios is pretty limiting when it comes to
changing this stuff around.
/proc/interrupts
CPU0
0: 700336 IO-APIC-edge timer
1: 6148 IO-APIC-edge i8042
9: 2885 IO-APIC-level acpi
12: 2177 IO-APIC-edge i8042
14: 4815 IO-APIC-edge ide0
16: 2 IO-APIC-level uhci_hcd:usb5, ohci1394
17: 97673 IO-APIC-level yenta, hdsp, Intel ICH6, ipw2200
19: 0 IO-APIC-level uhci_hcd:usb4, skge
20: 92955 IO-APIC-level ehci_hcd:usb1, uhci_hcd:usb2
21: 0 IO-APIC-level uhci_hcd:usb3
NMI: 0
LOC: 50568
ERR: 0
MIS: 0
.....
Any suggetions links greatly appriciated!
thanks
tom.
also....
have already tried pci=noacpi and acpi=off and there was no difference
to the setups, irq and the like.
bit worried about upgrading my bios as the new asus bios for the v6 is
apparently noisy and has fan issues..... no good for my studio...