If anyone cares to comment about this I would appreciate it
and it could be useful for other DAW novices. First, the
bass boost for 2 seconds here just rocks my socks... for my
tastes it is perfect.
Could anyone literally describe the technical details of
what it would take to get the kind of bottom end specifically
from 32 to 34 seconds, the first brief sound part ?
http://homepage.mac.com/hschot/PodCasts/HijackEffectsTest-050203.mp3
(the last 1/3 is over the top rubbish)
Secondly (shudder), the reverb applied from 40 seconds to
1:10 is also excellent. I don't care if it's real or the sound
of a forest, it just sounds commercially 'ken excellent to
my ears. Could anyone advise me how to get pretty well
exactly that kind of reverb on our favourite tux powered
OS with that same clarity and transparency ?
All the reverbs I've heard so far (but with minimal exposure
to all LADSPA plugins) sound unnatural and somewhat like "down
a drainpipe".
--markc
Hi,
since a couple of days I have not been able to get anonymous CVS access
to Jackit and Jamin. The login just times out. Can anybody confirm
this issue or comment on this?
Regards
Christian Schumann
--
------------------------------------------------------------------------
Dipl.-Phys. Christian Schumann |Technische Universitaet Kaiserslautern
Mail: schumann(a)physik.uni-kl.de | Fachbereich Physik
Tel.: 0631/205-4842 | http://www.physik.uni-kl.de
Fax.: 0631/205-3902 |
Post: Erwin-Schroedinger-Strasse, D-67663 Kaiserslautern
-------------------------------------------------------------------------
Hi Bob,
As I mentioned to Tim, I also have uses for over-the-top reverb, just not
applied to room modelling.
> Can't say I've had a lot to do with Synth reverbs, I come from a ' live'
> sound engineering background so tended to use reverbs very sparingly.
> As a recording engineer and moving to digital this is a different ball
> game. I think it really depends on the type of music you are aiming for.
Yes it does depend on the type of music, but again, most any kind of music
sounds better in a good room where "good room" is up to you to design.
As a live sound engineer, you probably chose specific venues for recording.
> I must admit to struggling with reverb settings as they sound ok one
> minute and then not the next as the track moves on.
I'm just guessing, but with your experience in live recording, you're probably
very sensitive to errors in the physics. As you know, there are simplistic
approaches used for reverb, echo, and stereo separation. Although you
can set all this optimally for one instrument at a particular time, the
settings don't extrapolate to other cases well at all because the modelling
is physically incorrect (it's signal-based). If you have a good math
background, then you probably know that tuned polynomial models of things
usually fail miserably outside the tuned space --- sometimes within the tuned
space. Same thing here. If you turn the effects down, as you're used to
doing anyway, then problems probably are not as obvious. If you turn the
effects way up, then you have a new instrument, and it may sound OK. You're
no longer attempting to simulate a room with a weak model.
This discussion probably doesn't help you too much as a working engineer,
but perhaps in the not-too-distant future, other methods will be developed
which change the methodology or at least offer marketable alternatives.
Meanwhile, you'll probably develop a useful bag of digital tricks that work
for your customers.
Regards,
Dave.
Hello,
I've installed jackit-0.99.0-2mdk.i586.rpm and
qjackctl-0.2.14-1mdk.i586.rpm and try to start jackd and get the
following error message :
Patch bay deactivated
Statistics reset
MIDI connection graph change
MIDI connection change
Startup script
artsshell -q terminate
Startup script terminate with exit status=256
Jack is starting ...
/us/bin/jackd -v R -dalsa -ddefault -r48000 -p1024 -n2
JACK was started with PID=9259 (0x242b)
getting driver descriptor from /usr/lib/jack/jack_alsa.so
getting driver descriptor from /usr/lib/jack/jack_dummy.so
getting driver descriptor from /usr/lib/jack/jack_oss.so
could not open driver '/usr/lib/jack/jack_oss.so':
/usr/lib/jack/jack_oss.so: undefined symbol: jack_create_thread
jackd 0.99.0
Copyright 2001-2003 Paul Davis and others
Jackd comes with ABSOLUTELY NO WARRANTY
This is a free softaware, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
registered builtin port type 32 bit float mono audio
cannot lock down memory for jackd (operation not permitted)
/usr/bin/jack: symbol lookup error: /usr/bin/jackd: undefined symbol:
jack_create_thread
JACK was stopped with exit status=127
what's the matter, can you help me?
speak slowly I'm new with Linux
See you,
Greetings.
I have just set up my first partition table.
When you specify the mount point for each partition you can use your own title or
one of those in the menu. I can't remember all of them, but they had names like /
var, /tmp etc etc.
Where can I find information about which of these I need to create partitions for and
what they're supposed to be used for.
Currently I have (ext3):
/
swap
/home
/audio
I guess when I'm installing software an arbitrarily structured partition table is likely to
result in chaos something I'm naturally quite keen to avoid.
Many thanks in advance,
Tom
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i keep getting this error when i start qjackctl
its not an important thing but its bugging me
tried reinstalling qjackctl
but it didnt help:(
any thoughts?
if i could find a copy of the file id just stick it in there ;)
I'm using a configure for v1.0.8 drivers like this...
./configure --without-oss --with-sequencer=yes \
--with-cards=usb-usx2y
but I keep getting...
checking for which soundcards to compile driver for...
configure: error: Unknown soundcard usb-usx2y
and yet this URL clearly states it should exist...
http://alsa-project.org/alsa-doc/doc-php/template.php?company=Tascam&card=U…
Any suggestions ?
--markc
Hi Marije,
Sorry I did not repost it previously:
http://home.earthlink.net/~davidrclark/linux_audio_users
The IR generator method I'm using is constuction of Green's functions
using eigenfunctions. So far I've stuck to rectangular geometries, but
do have a circular one (not posted but available). This method,
in case you're not familiar with it, would be like ray tracing every
possible ray throughout the entire geometry for a particular pair of
source and solution points. However, it's usually limited to simpler
geometries and boundary conditions, as you may know.
Ray-tracing is good for more irregular geometries, but it's a lot like
Monte Carlo techniques: You launch something and hope that it produces
a useful result each time --- but usually produces a result that is
similar to previous results. I'm sure there are optimizations that
are done, just as with MC techniques, but I'm not familiar with
what those might be. The useful ones probably somehow ensure that
the ray goes very, very close to both source and listener fairly soon.
But it seems that this would be difficult to ensure in advance for
any rays other than the obvious ones. I would guess that this is
a serious limitation on resolution for large rooms and high frequencies.
It's also bound to be comparatively slow without rather extensive
optimizations.
Regards,
Dave.
Hi Steve,
Thanks for your comments on impulses, etc.
My main concern was that, and I'm sure you know this perfectly well, that
mathematical physicists are fond of saying about impulses, in the context
of integration, that the shape of the impulse doesn't matter too much.
It's a handy way to sweep away a difficult derivation!
This is true for many physical phenomena. But in audio --- and again
I'm sure you know this but others may not --- the ear is an extraordinarily
sensitive detector, and audio includes many different physical situations,
sometimes on very short time scales. So I think it's important to ensure
that everyone understands that "doesn't matter too much" doesn't necessarily
mean "isn't audible."
---------------------------------
(The following is more specific, hopefully of interest to at least some
LAUer's.)
On my web page is a song "latest" that uses calculated IR's. Some software
is there to produce them, also. But I know you're a busy guy! The song
"latest" is for headphones, as usual. Long ago you and some others
encouraged me to post the IR generators. They were so user unfriendly
that I took some time to "put some lipstick" on them. Hopefully they're
not too bad now.
Regarding synthetic space --- ironically a problem with the Green's function
technique that I'm using, the calculated IR's are a little too accurate.
They have extremely sharp spikes, and the timing is very accurate. You can
hear echos where you might actually hear more of a reverb (with actual
diffuse reflections). Amplitudes can also become very large, yet the
sound have very little power. Here again, I think that diffusion and diffuse
reflections need to be incorporated, or something mimicing them. However,
this accuracy produces crystal clear sound, and I find more and more that
I like it compared to the mushy sound that a lot of audio software produces.
Some reverb models sound hilarious to me because they're so physically
erroneous. I can't hear the music over the sound of my own laughter.
I'm not a big fan of ray-tracing at this time. It just seems to me that
it's equivalent to a sparse grid without some sort of optimization or
pruning method. But I have no experience other than reading technical
articles.
I've worked in the semiconductor industry with finite difference codes, so
naturally it occurs to me to try and build a IR generator using that
technique. Do you have any experience along those lines? Please feel
free to contact me via email if you prefer, because that's probably not
of interest to LAUer's. My crude estimates lead to a monstrous mesh and
no chance of reducing the time step. This may be OK for IR generation...
Thanks for any further comments.
Best regards,
Dave.
Hi,
I need to do some voiceover work, and I was wondering if there was an
application or a set of apps to do this with Linux. The video I was
given is an MPEG file, and I need to redo the voiceovers of the
presentation. The only voiceover work I have done was in Windoze with
Sonar 3, which I don't have access to anymore. I can think of a few
ways of doing this (coming up with an EDL, recording the voice, and
editing with Audacity or something). What I really want is an
application that will let me easily replace the audio track of a video
with another (preferably with a GUI). I know that I can use tools
like mplayer to dump out just the video and mencoder to re-encode, but
if there is an easier tool (more automated), I'd like to check it out.
Thanks.
Jon