Greetings,
Searched the list archives but found nothing appropriate, so I thought I would
just ask.
I am a metal guitar player. I like a heavily saturated guitar tone. Could
someone point me in the right direction for processing a clean guitar signal
in real-time such that it can (even partially - i like weird sounding stuff)
emulate tube circuitry. I am thinking a chain of LADSPA plug-ins would do
the job. The sound im looking for doesn't have to emulate anything to a T,
it just has to be raunchy (think Nine Inch Nails - Broken, or any Nine Inch
Nails album for that matter).
Thanks!
--
Levi Burton
http://www.puresimplicity.net/~ldb/
>> cygnus:~/mars> vsound --timing -f output.wav realplay
>> _1790918_blur_beagle_vi.ram
>> About to start the application. The output will not be available until
the
>> application exits.
>> /usr/bin/vsound: line 163: 1366 Aborted
>> LD_PRELOAD="$pkglibdir/libvsound.so" "$@"
>I find I have to use vsound -t
hi,
vsound -t
and
vsound --timing
are meant to be synonyms, and indeed both give me the same problem. has
anyone else had this problem? i don't even know what it means. why does it abort
on this line?
cheers
jane
Thanks for all the responses, especially the curses based vu meters... off
to try them, that's exactly what i meant :)
> >>Personally I use arecord and then adjust the volume level with a sound
> >>editor later.
> >
> > Thats' all very well but when recording you don't want to (a) go over
> > full scale or (b) have the incoming level so low that you lose resolution.
> >
> > Adjusting the level digitally afterwards won't help with either of these
> > problems.
>
> You are completely correct but if you know your system well then this
> should not be a problem. If it is then do a retake or ditch the parts
> that don't work.
right, well, exactly. that's what i've been doing and i've been getting
very very bored with it. consequently now delighted with the meter
suggestions :)
thanks!
jane
Hi
thanks for your suggestions on simple apps for simple people...
>From Julien:
> For just recording and checking mixer-settings, I think ecasound or
> qtecasound, if you like GUIs, might be ok. Otherwise there is still
ardour,
> muse and the other big ones, but they contain a lot of functionality.
um, sorry to be silly, but how does ecasound tell me whether my mixer
levels are suitable for recording the input? don't i need something
like a VU meter, ..or?
> Recording mp3s: Why not just record audio-data to a raw (.raw or .cdr)
file
> or .wav-file and convert them afterwards?
? gosh i really wasn't clear. i'm not sure what you are suggesting, but,
well, i think i'm doing that. but i need to do a bit of editing along the
way.
Matthijs:
> Sweep is quite a nice audio editor, if not that great at working with
> large sound files:
thanks! it has the functionality of x-fades which is really nice,
and audacity doesn't. but... hmmm... whats this about large sound files...
what happens?? <worries about spending hours to have it swept and mangled>
Alexandre:
>
> But, oh god, I installed ReZound 0.7 yesterday... It has VU meters
> for both recording and playing and an EQ analyser as well.
> Give it a try!
i tried. it sounds perfect, but... hmmm can't yet get it to compile. it
depends on a lot a lot of libs that are >versions than mine. i am still
struggling (debian woody/unstable)
so i'm still open for suggestions for a VU meter, or even some smart way
that makes my recording levels easy to assess...
Now this is a vsound (0.5) problem I've had for a while. It occurs on
realmedia files. Somewhen during the stream realplayer just closes. This
is what it says:
cygnus:~/mars> vsound --timing -f output.wav realplay
_1790918_blur_beagle_vi.ram
About to start the application. The output will not be available until the
application exits.
/usr/bin/vsound: line 163: 1366 Aborted
LD_PRELOAD="$pkglibdir/libvsound.so" "$@"
Any suggestions?
thanks!
jane
Hello,
After installing Gentoo Linux over RedHat, I find that my Delta44
playes sounds a about 1.5 semitones too fast!. Boot into windows,
and it goes back to normal. Not sure if I suspect ALSA drivers,
or the kernel, or what.
It is not user error, like sample rate probs or anything. XMMS,
aplay etc... all play content two fast, no matter what the soundfile,
or mp3.
Thanks for any help!
Tobiah
> > > Yes, thats the main point:
> > > http://www.ecs.soton.ac.uk/~swh/jrss.png
> >
> > This link is broken :-(
>
> Works for me.
>
> - Steve
Fine from here, too.
Matt
http://plugin.org.uk/releases/0.3.7/
I've done a major code audit (with the help of valgrind :), and things are
a lot less crufty now.
There are still some outstanding known sound quality/noise/aliasing bugs,
I'l tackle them in the next release, but I'd appreciate more reports.
ChangeLog:
2003-02-23 Steve Harris <steve(a)plugin.org.uk>
* Fixed memory leak in gate
* Fixed filter implementation in gate
* Fixed key defaults in gate
* Made passes=0 work in GSM
* Added bandlimiting filter to GSM (less cruchy sounds)
2003-02-24 Steve Harris <steve(a)plugin.org.uk>
* Removed stale code from surround encoder
* Fixed memory leak in surround encoder
2003-02-24 Steve Harris <steve(a)plugin.org.uk>
* Fixed maths error in multiplexer
* Fixed buffer overrun in sifter
* Efficiency improvements to FAD delay
* Fixed infinite loop in FAD delay.
* Fixed (another) buffer overrun in FM oscillator
* Performance improvement for FM oscillator
* Fixed buffer overrun in multiband EQ
* Fixed aliasing in Hermes
* Fixed memory leaks in:
AM pitchshift
Analogue osc
Bode sifters
Comb
Comb splitter
Delayorama
Dyson compressor
FM oscilator
Giant flange
Gong
GVerb
Hermes filter
L/C/R delay
Multiband EQ
Plate reverb
Rate shifter
Retro flanger
Satan maximiser
SC*
Sifter
Single band parametric
Multiplexer
Tape delay
There are still known leaks in imp and the multiband EQ
I need some help understanding the basics of audio editing/filters using
the linux tools.
I've successfully used Broadcast2000 and SND to capture music input from
cassette tape and LP, and have written some to CD. Next I want to do
some work on the files, filter out what nasty stuff I can, but don't
know what filters have what effect. I'm also not sure of the steps to
think the process through; that is, if you see/hear symptom A, then X
process, or X filter. Actually, I know how to apply the "plugins" in
Broadcast 2000, but don't know what to pick or why. In SND I haven't
gotten that far, but know it's possible.
Any hints, howto suggestion or pointers would be GREATLY appreciated. My
audio equipment is pretty modest; a SB16 ISA...and I know I might have
to change that.
Thanks in advance
Reid
Greetings list,
Thanks again for your responses to my questions about iiwusynth, zynaddsubfx
and jack; you were all most helpfull. And now for something completely
different...
When creating music, there are various issues I run into. First, the way I
would like to work is to build tracks within something such as Rosegarden. I
use a hardware synth (alesis qs6), and other software tools. Since I have a
relatively slow computer, I would like to build the tracks, then record each
one separately, then composite them with something like ecasound.
My first problem is the recording of the various tracks. Right now, I use snd
because it has this record feature called "trigger". This feature is
extremely usefull. My question is do any other software packages support
this feature? For those who are not familiar with it (or cannot infer its
meaning), it essentially just waits until the audio level reaches a specified
threshold, then starts recording. I would love to know if ecasound can do
this.
My next question is one concerning synchronicity of my composited tracks, and
also of sample loops. I am not mathematically inclined, and i do in fact
suck at math. What I would like to know is if there is (im sure there is) a
formula for calculating audio segment lengths according to a specified BPM.
For example, say I have a loop of some recorded drums at 120 BPM. I would
like to know exactly how long the audio sample must be to match 120BPM,
rather then trial and error.
I'm sure I had more questions but I seem to have forgotten them at the moment,
but im sure this is enough for this particular post!
Thanks!
--
Levi Burton
http://www.puresimplicity.net/~ldb/
