hello all - got a question - I've only recently been stopping and taking a
look at my studio computer's performance and in the almost year since I
change from Red Hat 9 to gentoo, it's been more solid on some things, but I
notice a huge latency difference - ie: I have to run Jack at -p 8192 to get
anything done in Ardour
Anybody have any tips on what to look at to tweak it? Seems like it should
do better than that... I didn't see it as a problem until in the last few
days I started playing with playing softsynths live directly into Ardour -
you've gotta be running at -p 1024 or there's a latency that screws up your
playing - at 8192 it's a downright 8th note delay...
Here's some vitals that I can think of:
OS: gentoo 2.6.6-rc1 kernel (alsa built in)
jack: 0.99.0
ardour: beta28
jack command line:
jackd -R -d alsa -d hw:0 -r 48000 -p 8192 <------- (or whatever)
harddrive:
multicount on
io support: 32 bit
unmaskirq on
use dma on
keepsettings off
readonly off
readahead on
chip: 2ghz amd (I THINK - not at computer now)
ram: 512MB
thanks for any ideas! :)
---------------------
Aaron Trumm
www.nquit.com
-----------------------
Hi Folks,
So, I have spouted off about a distro before (I know some of you checked
it out) and it was not as good as I hoped it would be! :(
Trust me...THIS is not the case this time!!!
Announcing PClinuxOS (PCLOS for short)
http://www.pclinuxonline.com/pclos/
This is based on Mandrake but minus the bloat! One ISO built as a
LiveCD! You can boot it and not install...no commitment unless you want
to. The install is simple as can be and fast. KDE 3.4, Fluxbox, Gnome,
etc. The distro was started by Texstar who used to build packages for
Mandrake a few years ago.
So, whats that got to do with Music?? Just this; Thac has decided to
start packaging for PCLOS just as he has with Mandrake. For those not
familiar with his work...
http://rpm.nyvalls.se/sound10.1.html
I have relied on Thac's RPMS for audio since I started Linux audio work.
He keeps his packages updated and has about everything worth having in
his repositories. They have been invaluable to me.
As recently as yesterday, Thac has started his own 3rd party Apt
repository for PCLOS and as I write I am downloading the first of his
packages and new mm kernel, etc. We are chatting on the Pclinuxos IRC
chat room and ironing out a few minor bugs. It will be a few days before
he has all of his packages done.
Texstar suggested that we might make a liveCD with the key audio apps
and a R/T kernel for easy usage and evaluation. Obviously some time is
needed to work the bugs out but this is very likely in the not too
distant future! I should think those of you working with MDK right now
would see this as an easy step....others...well, the proof will be in
the pudding.
More soon....
R~
I don't think I understand this jackplug concept.
I'm trying to get an alsa client, such as aplay to play through jackd.
I've setup jackplug in /etc/asound.conf
pcm.jackplug
{
type plug
slave
{
pcm "jack"
}
}
pcm.jack
{
type jack
playback_ports
{
0 alsa_pcm:playback_1
1 alsa_pcm:playback_2
}
capture_ports
{
0 alsa_pcm:capture_1
1 alsa_pcm:capture_2
}
}
I then tell aplay to use:
#aplay -d jackplug foobar.wav
But aplay plays the file even if jackd is not started; why?. Isn't
jackplug supposed to show up in my jack connections when aplay is
playing?
I seem to be missing something vital.
Is there some way to make a "fake audio device", like hw:3,0 that alsa
applications can connect to so that I can get the audio out and then
send it through jack, process it and then send it to the real audio
device?
--
Esben Stien is b0ef(a)esben-stien.name
http://www.esben-stien.name
irc://irc.esben-stien.name/%23contact
[sip|iax]:b0ef@esben-stien.name
Someone brought this device to my attention:
http://www.emu.com/products/product.asp?product=2209&category=754&maincateg…
Creative may donate a device to an ALSA developer to develop the
necessary support, if they think Linux users would be interested. There
is already a volunteer to develop the driver support. If you would
consider buying this device were it supported (I certainly would) please
contact Creative and politely ask them.
I think if just a few people write them along the lines of "I really
like looks of the 1212m, and I'd like to buy one, if I could use it with
Linux. Can you provide one to the ALSA developers?", they will do it.
Every person who takes the time to contact them, they will assume
represents N users.
Lee
On Wednesday 19 January 2005 19:02,
linux-audio-user-request(a)music.columbia.edu wrote:
> > So how would Raton-Conductor work? Again, not as simple as it sounds.
> > Minimally, one would move the mouse in a ecliptic (or circular) motion
> > (as suggested by MagicBaton's instructions for beginners). The size of
> > the vertical diameter (or average diameter) would be the
> > volume/expression and tempo or time-codes set by the change in vertical
> > direction from down to up. Real conducting patterns are more complex but
> > these two principals would more or less remain.
>
> I started tinkering after I read your message, and I created a little
> gadget that traces mouse motions and recognizes fairly general conducting
> patterns. One can extract timing and intensity information for the purpose
> of generating MIDI clock events as well as MIDI controller values. I'm
> tentatively calling it Boa Conductor.
Cool!
>
> I've got a few questions:
> 1. Is anyone interested in a tool like Boa Conductor? What I've done so
> far was just for kicks; I now have to decide how much time and effort
> to put into polishing it.
I, of course, would be interested.
> 2. Are there any MIDI sequencers/players for Linux that can be driven
> by external clock messages? My understanding is that Rosegarden does
> not currently work as a slave but that may change in the future.
> MusE can work as a slave, but I never used it before (has anyone
> tried driving MusE with clock messages?). I don't think timidity
> expects to be driven by a MIDI clock. How about other MIDI players?
> 3. Would it make sense to have a feature that uses JACK Transport
> rather than MIDI clock?
There are a few alternatives. Not much that I have on Windows or Linux support
clock messages. MagicBaton was a MIDI-player that added events based on the
mouse-conducting.
Alternatives:
1. Run the Boa in Parallel with whatever sequencer or player is going through
jack or such. Boa would then put out simply omni/overall level and tempo
events.
2. Run Boa as a plug-in Rosegarten, Muse or other such program. In this case,
it would work on one track/channel and insert expression (and tempos) or in
an omni mode as above. For rehearsing (MagitBaton's parlance) one track, one
would probably want to disable tempo changes and conduct expression.
Omin/overall would do level and tempo.
3. Controlling another software via midi-clock and some volume control device.
This assume, naturally, that this software is available :-)
Hi,
Since I kept coming across the linux audio user discussions in my
continuing quest to make my soundcard work under linux, I thought I'd
join and start bugging you with questions.
I have the soundcard mentioned in subj.
I have a set of cambridge soundworks hooked up to the coax digital out
and a set of headphones to the headphones output.
It took me a while, but I got sound playing out of the coax digital
output. I did this by fiddling with a .asoundrc and alsamixer.
Finally I found a combo that worked. I had to lock my internal mixer
rate to 32000 and use that as the rate in .asoundrc.
I never tried setting the rate that low, since the rate I normally see
in conjunction with audio equipment and digital connections is 44100 or
48000.
I then tried getting sound of the analog outputs, and that took me ages.
I have no idea what I did, since it was nothing I hadn't done several
times previously, but suddenly I got sound out of the analog outputs.
I could use any matched set of alsamixer internal clock rate and
.asoundrc rate up to 192000.
I did a little victory dance.
Then I hit the volume button on my keyboard, which changes the volume
with kmix. The sound disappeared.
I could get sound out of the digital coax again (changing the default
device in .asoundrc), but no matter what I did, I couldn't get analog
sound back.
I tried rebooting, turning off the machine, etc. Nothing helped.
Then suddenly after a couple of days, I tried switching back to analog
output and it worked for no apparent reason. I thought I knew what I did
right this time (messing with the multi track rate reset button in
alsamixer - I still have no clue what this is for) and bravely hit the
volume button to see if I could break the sound again.
I could. And I couldn't get it back until a few days later when it
mysteriously reappeared. After the next reboot it was gone again.
Now I have my analogue sound and I am not keen to launch kmix no time
soon. Or to reboot.
I still have no idea what I am doing wrong, but I am very keen to see
the .asoundrc and alsamixer settings of someone with a Aureon 7.1
Universe card.
Another thing I'd like to do is to make alsa play on the digital and
analog outputs at the same time, instead of having to choose between two
different pcm devices in .asoundrc.
I have RTFMed as well as I could, exerimented and googled a fair amount.
I even tried modifying envy24control to work with ICE1724 cards, but
quickly got stuck.
Any clues, pointer or help is greatly appreciated.
Brian Meidell Andersen
Hi.
I see that Behringer has released a new gadget:
http://www.behringer.com/BCA2000/index.cfm?lang=ENG
Has anyone tried this thing ? Does it work with the
generic USB audio driver or do we have to wait for
ALSA drivers to become available ? I am _almost_ off
to buy one :-).
Cheers
-- Jan Holst Jensen, Denmark
__________________________________
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VMWare is 1.--expensive, 2.--not designed for real-time work but more for OS
testing, et. al. 3--is not opensourced so you could not play with it even if
you knew how.
There are some alternatives, none of which I have ventured to try:
qemo - open source
zen - open source
Lin4win - less expensive than vmware but restricted to i386.
I'd love to see some of this stuff work with WINE but fewer and fewer sound
things seem to work with it as newer versions come out. Hint. If you try
them, select one of the 32-bit windows models.
Hey everyone,
I'm looking at an audio interface/sound card for pro recording. Anyone had
any success with the Delta 1010 under Linux (Fedora Core 3 - CCRMA)? I found
a great deal on a used on so thought I'd check.
Thanks,
Kevin