Hello everyone,
I'm running Ubuntu Jaunty 9.04 (on a different laptop, not the one
running AVLinux). Till recently, my linuxsampler setup on this
particular system was working without (major, noticable) glitches, i.e
very few xruns and not many dropped notes. Since the past few days,
however, I'm seeing a host of xruns and buzzes and drop notes. I tried
switching to the RT kernel (kernel 2.6.28-3-rt, via synaptic). The
result - a *huge*, unmanageable latency. I also noticed this warning
which I'd not seen before:
Thread: WARNING, can't assign realtime scheduling to thread!
I'm using jackdmp 1.9.4, compiled from source:
/usr/local/bin/jackd -S -R -dalsa -dhw:0 -r44100 -p256 -n2
I'm sure there are better realtime kernels out there, but before I
switch the kernel, I'd like to understand why exactly linuxsampler is
not being allocated realtime priority. I don't know much about this.
Incidentally, here's what 'top' reveals: (Note: these are with the
vanilla kernel).
PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+
COMMAND
29516 guru 20 0 35872 13m 10m R 10.3 1.4 0:28.17
qsampler
29503 guru 20 0 437m 436m 76m S 2.7 46.5 0:11.57
linuxsampler
29493 guru 20 0 88632 74m 72m S 2.3 8.0 0:11.20
jackd
No user-owned entries have RT priority, but down the list some
root-owned processes do, for e.g:
3 root RT -5 0 0 0 S 0.0 0.0 0:00.00 migration/0
All user-oned processes have the same priority: 20
My /etc/security/limits.conf file ends with this:
@audio - rtprio 19
@audio - memlock unlimited
@audio - nice -19
And I'm a member of the group 'audio'.
Can someone help me out with this?
Thanks in advance!
Cheers,
Guru
--
Guru Prasad B.R.
Centre for Ecological Sciences
Indian Institute of Science
Bangalore - 560 012
India
Ph:+91-80-22933103
I've been running Fons's great Auto-Wah on my 32-bit Netbook for many months now, using it live constantly, and I love it.
I'm going into the studio Friday, so I've dusted off my 64-bit Core 2 Duo system and putting it through its paces.
Turns out that on my Core2Duo, JACK RACK running Fons's auto-wah plugin sucks up 6% CPU even if there is no sound going through it.
I asked Fons about it; he can't duplicate the problem since he doesn't have a 64-bit system. Hell, I can't even duplicate it on my 32-bit system; it's 64-bit dependent.
I looked at the output with Bitmeter at Fons' suggestion, and it seems there are no denormals, inf's, or nans-- just a very small -230dB DC offset in the code (probably to avoid denormals).
The CPU usage ramps up before any signal is fed into the plugin. It runs normally when there is signal, but then it keeps sucking up CPU afterwards. I'm feeding it from fluidsynth, and bitmeter says that it isn't sending anthing at all during silent moments.
It doesn't run away to 99% CPU (or 50% on a Core2Duo), like it would if there were a denormal. But still, why is it using up CPU when there's no signal? That seems odd. And why only on my 64-bit machine? FWIW, the 64-bit and 32-bit machines I have are even running the same kernel and RT-patch versions (2.6.26.8-rt12).
Can someone with a 64-bit system help me by trying to duplicate the problem? I'm looking to see if JACK-RACK running just Fons' plugin uses as much CPU with silence as when there's signal running through it.
Oh, and in case it's setting-dependent, here is the rack with the settings I'm using:
http://restivo.org/misc/clavy-phaser.rack
-ken
Hello,
I released an updated version of lashstudio module.
It works with puppy 431 retro variants (k2.6.25 and k2.6.21).
It includes: jack-0.118.0, muse sequencer 1.0 with dssi support,
zynaddsubfx-2.4.0, fluidsynth-1.1.1, jack-mixer-8 and much more....
downlod it here http://sourceforge.net/projects/lashstudio/
marius
Hi,
My 5 year old AMD64 machine smoked yesterday so I'm looking to
build something new. On interesting new processor is the Intel I5-661.
I'm wondering whether very many folks are using Intel motherboard in
Linux DAWs? I've not used one before for any reason so it's a new
vendor to me. The model I read about in a test document was the
DH55TC, but I'm partial to the DH55HC as I need more than 1 PCI slot:
http://www.newegg.com/Product/Product.aspx?Item=N82E16813121396
Comments and ideas welcome.
Thanks in advance,
Mark
On 13 January 2010 at 13:35, <nettings(a)stackingdwarves.net> wrote:
> i just stumbled upon this really slick "made with ardour" logo:
> http://thorwil.wordpress.com/logo-design/
Those logos Thorwil created are really superb!
> in fact this is something i've been looking for, cause it's way
> sexier than just a note saying "created with ardour". the problem
> with this particular logo is that it doesn't work very well when
> space is limited (such as on a cd jewel case inlay), because the
> fontwork that expresses the actual message is rather small compared
> to the total size, and that it imposes too many design constraints
> through its color.
>
> what i'm looking for is something that
>
> a) will advertise the fact that this product here has been created
> with ardour and other free software
> b) will look really cool and
> professional
> c) will scale down to ridiculously small size while
> still transporting the message
> d) will work in black and transparent, or alternatively a set of
> two or three (at most) custom-chosen colors matching
> whatever sleeve design i have to deal with (because usually
> clients will need to be talked into accepting it, so it has
> to blend in)
In addition to "cover art", I would think a "made with <your open
source program(s)>" logo could also be used on websites, e.g.
MySpace, and in downloadable music files, e.g. MP3s. For open
source use, I would really like to see logos licensed via the
author's choice of some combination of Creative Commons options.
Nice topic...
--
Kevin
I've been a big fan of the Dr. Who theme song for a long time, but for
whatever reason, I had no idea who Delia Derbyshire was until
recently.
http://www.delia-derbyshire.org/http://en.wikipedia.org/wiki/Delia_Derbyshire
I recently watched on YouTube the making of the Dr. Who theme song,
and learned that it was done prior to the synthesizer, using amazing
techniques (early electronic music-type stuff):
http://www.youtube.com/watch?v=pzF9PyM-FCA
Wow. Just wow.
So I thought it would be fun to see if you all would be interested in
having a little contest in honor of Derbyshire. Let's each remake the
Doctor Who theme using any app/styles you see fit (using Linux/FLOSS
tools, of course) and submit the entries. I would suspect that this
would be a lot of fun, and we would get a lot of interesting results.
So in summary, here are the rules:
Doctor Who theme, any way you see fit (techno, country, abstract, etc.)
Use Linux/FLOSS tools
Due by Feb 1st???
The only downside is I don't see that there would be a way to pick a
clear winner, and honestly there doesn't need to be one. Just make
sure you describe your process so we can all learn from it.
What say ye all? Would this be interesting enough for all of you to
get involved in?
--
Josh Lawrence
http://www.hardbop200.com
Hi guys, I just wanted to give an update on the integration of
librubberband into TerminatorX.
To understand rubberband, i wrote a small console Jack app that can play
wav files at diferent speeds, keeping pitch. It works well. The code is
attached.
I have a question to for the devs out there though: It seems to me that
i have to run process() a few times with a fixed blocksize before,
getRequiredSamples() returns something >0 in Realtime-mode. All other
options are default options. Is this true? I need some help on this
issue.
thanx Gerald
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Hi folks
Dumb question ? Yeah maybe but I don't know how to, so I'll ask anyway :)
I've used 64Studio for ages now, but the need of some decent video work
have forced me to tryout Ubuntu studio, with all packages installed, and
it runs surprisingly well without any xruns even at low latency, but I
have a problem !
How do I handle automatic start of my sound device "RME Multiface" isn't
runlevel scripts outdated ? I recall heard/seen somewhere that usb and
firewire devices startup could/can be managed in a file, but which ?
I'm running Ubuntu Studio 9.10 with kernel 2.6.31-9-rt on a Lenovo T61,
and the RME interface card is PCMCIA.
Thanks in advantage
Sv-e
All,
I'm looking for a way to balance the sound between regular playing
and pop/push type of playing on a accoustic bass guitar. The guitar is
both miked (M-Audio pulsar) and has a balanced line-in for the embedded
pick-up (of not very good quality as it lacks picking up the high
notes). That's two stereo tracks in Ardour as I feed both mono signal
in the right and left of each track. Maybe I shouldn't do that.
In any case, the push/pop notes are blasting when compared to
regular playing. A bit of blasting is OK, but there's too much
contrast now.
Is there a receipe in recording techniques that addresses that kind
of configuration and enables control over the loudness difference
from a sinle intrument in a single take ?
Any ideas/suggestions appreciated,
Cheers.