Hi all,
It is my pleasure to announce the latest release of Superkaramba theme
"Borealis" version 2.0 (not to be mistaken with the associated soundtheme
"Borealis") that offers system monitoring as well as user-friendly interface
for the most commonly used desktop features while utilizing minimal amounts
of CPU (2-8% on a mobile AMD64 3000+ running in powersave mode at 800MHz).
For more info please see:
http://www.kde-look.org/content/show.php?content=13876
as well as the online manual:
http://meowing.ccm.uc.edu/~ico/Linux/Borealis/Borealis_manual/index.html
Changelog:
*REQUIRES Superkaramba v.0.35 or higher
*Tons of optimizations and script
improvements.file:///home/ico/Borealis_screenshots/install.html
*Tons of bug-fixes.
*Utilizes latest additions to the superkaramba package (CVS).
*Inclusion of sound events.
*Completely revamped drawing engine.
*Systray and desks bars resize automatically to accomodate space for
externally added desktops and systray icons.
*Included script for background and font color customization.
*8 variations of background bar designs included.
*CPU footprint is now down to 2-8%.
*Expanded documentation.
*Added new wireless networking bar.
*Many additional features (consult the new Manual for more info).
Best wishes,
Ivica Ico Bukvic, composer & multimedia sculptor
http://meowing.ccm.uc.edu/~ico/
Hello! I'm new to the list, and only marginally familiar with Linux.
I've used Linux on and off for several years, but never for audio
before. I've compiled the kernal in the past, but never done anything
quite so complicated as setting up SMP and low-latency, which I plan
to. My target machine is a dual PII-350.
I'm looking at putting together a system that I can run outboard MIDI
gear from to start with, using either a Tracker or general-purpose MIDI
sequencer.
I've seen the compatibility chart, and think I'd like to pick up a
small M-Audio interface. The MidiSport 2x2 is probably too small, as
I'd outgrow it quickly. The 4x4 is just about the right size... but
the 2x4 is insanely cheap on some webstores. Unfortunately, it's not
mentioned on the list, and I haven't seen any mention of it in the
archives.
Does anyone know if the MidiSport is supported? And if not, how can I
tell what chipset it has, so that I can guess if its even a worthwhile
attempt? Thanks for any advice!
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Hello LAU,
I'm designing a modular synth engine, and would like to get some input
on desired features/abilities to make sure it's designed in a way that
will make it the most flexible.
So, what features are important to you in a synth? What annoys you
about the current crop of linux synths?
Already planned (* = working right now):
* Low latency realtime operation via Jack
* 100% LADSPA plugins, not Yet Another Internal Plugin Format
- (DSSI and perhaps VST are being considered)
* Seperate engine and frontend(s), ALL communication via OSC (Open Sound
Control)
The main GUI will be a modular a lot like AMS/SSM/etc. but there's no
reason it has to be, the engine can be used from the command line, in
programs/scripts (from any language that can talk OSC), hardcoded in
custom synths, etc. etc. This will allow, for example, algorithmic
composition in C without the limitations of MIDI and discrete notes.
The use of OSC means you can run the engine transparently over the
network (on 20 machines if you so please..)
- Polyphonic, runtime selectable (in the works right now)
- Multitimbral
* ALSA MIDI in will be through a seperate application, which will allow
binding of any MIDI controller to any param in the synth
- Sending OSC messages from module ports
* Sample accurate everything
- The main modular frontend will support LASH (formerly LADCCA), and
allow building of control panels ala galan
I'm very open to suggestions, that's why I started this thread.. let's
hear 'em
Cheers,
-DR-
Greetings all,
Can anyone tell me what the REAL process for getting jack_fst installed
is?
For fst-1.6:
fixheaders needs to be run from the $HOME/fst-x.x/vst directory as
"../fixheaders" not from the $HOME/fst-x.x directory as advised by Dave
Phillips (http://www.djcj.org/LAU/quicktoots/toots/vst-plugins/).
Small problem... Like Dave said, this stuff isn't for rank beginners.
mkinstalldirs, called from Makefile, was missing. I copied the one
supplied with Hydrogen. Annoying problem.
For jack_fst-1.2:
fixheaders, used above, does not actually fix the headers. gcc doesn't
understand the initializer. This is what I'm not sure about: How to emulate
C++ for struct VSTFileTypes initialization. Although I've written a bunch
of code for C++ classes, I've written virtually nothing for structs in C.
Lines near 918 in aeffectx.h need to be further modified. This appears
to be yet another "audioMaster" -related problem.
Thanks for your attention and any info.
Regards,
Dave.
Thanks to those who responded, and in particular to Christian Henz.
I had not seen this before (from Paul Davies):
"The implementation of DRI by certain video interface drivers means
that we end up trying to lock the video memory as well, and this tends
to fail for various reasons."
Uh-huh. Well, that certainly explains what I'm seeing, and it's more
than just " *trying* to lock the video memory;" it's quite successful at
doing that. The X xerver gets hosed.
Regards,
Dave.
Hey y'all - thought I'd ask what thoughts people had on this (very cliche'd and normal, really)
basically the deal is i'm getting a lot more x runs since i changed out my mobo and i'm trying to figure out what else i can do...
i'm running gentoo with the latest ebuilds of jack and ardour, which is what i was doing before, when my motherboard's memory slots and possibly other parts of the board went haywire - it was a soyo dragon plus - so i took the processor out, got new ram (same amount - half gig), and replaced the soyo with a chaintech mobo which I think has the same via chipset (i think)
but what's weird is, i have to run jack at -p 1024 to even HOPE to be able to work, and sometimes at 2048 - hell i even did one mix the other day at 8192 or whatever that step is because i kept getting like one xrun and screwing it up
but before the mobo was replaced (same hd, didn't reinstall OS or kernel but did rsync and update jack and ardour) i could run at -p 256 just fine
so i dicked around, did some hard drive tuning with hdparm, found out about "nice" and tried runnning jack and ardour with nice at the highest 2 priorities, which really seemed to help - but i'm not back to where i can run at 512 or 256 (which is better because you don't get that crazy latency in ardour when you're monitoring what you're recording) - i'm not able to reliably record without having at least one drop out at 1024 which is the highest i've been able to record at withut having ardour actually not place the audio in quite the right spot...with nice maybe i am but i'm still curious what's up that i can't get back to 256...
the only other thing i can think of is that there may be some kernel config options that could stand to be adjusted and the kernel rebuilt...but i so far have been lazy and unconvinced that that'll help...
wow that was a lot of typing...so what do people think?
-------
NQuit
www.nquit.com
-------
Greetings all,
A few of us have found MusE to lock up our systems. I've had this problem
since 0.6.3 running on 2.6.X and on a relatively new machine. I think
I've run MusE successfully on this machine, but if I did, it was long ago.
The symptoms are that the machine totally locks up and requires a hard
reboot (reset), followed by ()fsck'ing all mounted drives on reboot.
The CPU heats up as some sort of race condition occurs. It is not a
kernel crash as has been suggested elsewhere, at least not for me;
I wish it were.
I found that I can set ulimit -l to some fraction of memory so that
it isn't "unlimited." This does produce various warnings, but on this
machine, with what little I've done thus far, MusE *seems* to run OK.
It would seem to me, with what little I know about mlock() policy, that
there is an mlock() SNAFU in some of the Linux audio programs. If
that's not the case, I'd really like to hear the explanation. (Thanks
to anyone who contributes info on this.)
ulimit -l could very well be set to "unlimited" in many distributions
rather than the distributors setting it to some sort of guess.
Developers should count on it.
I have a similar problem with qjackctl, but temporarily disabled memory
locking entirely. I'll check it with ulimit set to some reasonable value.
I believe that qamix also had this problem. I think that was it --- so
far....
Thanks for your attention, and regards to everybody out there,
Dave.
Hi,
I recently completed production of an eight minute
film score. The final destination is DVD-Video. I need
to know what levels the audio should be mastered to
play at. The answer seems obvious but I don't want to
make any assumptions, can't make a mistake and have
become a little bit paranoid.
My job is simply the audio so I should hand off a
16bit, 44100 digital master with no peaks above
0.0dbfs and the video production house can adjust to
suit the final destination format; broadcast, analog,
digital, whatever.
The problem is, the video production house never asked
the client what the destination is but assumed it is
analog and for broadcast. The client contacted me and
stated that they had to turn their home stereo
playback volume up all the way. And even then it
wasn't loud enough. Incidentally, the client is happy
with my master. They stated that it sounds "great" on
their home stereo.
There is a great dynamic range between the peaks and
valleys which is exactly appropriate for this sound
track. Reguardless, my knee jerk response was to
suggest making up some of the volume by raising the
average level through remastering or remixing and
mastering. However, this is a mistake because the
existing product is fine.
My conclusion is, I need to provide the video house
with a product that they will not adjust in any way. I
don't know how this production house merges audio and
video so I can provide a test tone for them to
calibrate with. Of course I'd prefer they simply
dran-n-drop my file into an audio track but maybe it's
just not done that way.
I assume audio for DVD-Video should be mastered with
no peaks above 0.0dbfs, exactly like any other digital
audio. Correct?
Well, I decided to relay the long version of the story
because I'm certain that someone else here will
eventually share my joy. :) Until this is resolved, I
will continue; just living in a cave, cleaning the
guns for jah, watchin' clouds rollin' by and watchin'
fishes jumpin' high.
ron
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I upgraded my kernel from 2.4.18 to 2.4.26, and now I can't play
sounds. Any time I try, the program just hangs until I ^C out of it.
No messages appear in /var/log/messages, or on the console.
I have a Debian unstable system with an SBLive soundcard. (There's
also a sound system on the motherboard, but that never confused it
before.)
I have in the past managed to run Alsa, but getting it installed has
never been fun, so since I'm only doing fairly simple-minded things
like playing MIDI and having card sounds on my solitaire game, I've
been just running OSS.
I can't see anything wrong with the permissions on the /dev files, or
with the list of modules that's being loaded. I just uploaded the
latest version of that kernel-image, and the July 22 version has the
same problem as the May 1 version.
Does anyone have any ideas about what to try next?
--
Laura (mailto:lconrad@laymusic.org , http://www.laymusic.org/ )
(617) 661-8097 fax: (501) 641-5011
233 Broadway, Cambridge, MA 02139
Hello,
I am using Audacity 1.2.0.
I'd like to see where I am in the sound file, in seconds, samples or
something. But the timeline has no numbers in it.
Perhaps it dsoesn't find a font or something?
BTW, Audacity constantly outputs to the xterm where I started it things
like:
** (audacity:5262): CRITICAL **: file pango-layout.c: line 2002
(pango_layout_get_extents): assertion `layout != NULL' failed
Perhaps this has something to do with the problem?
Yours, Mikhail Ramendik