--- En date de : Sam 27.3.10, rosea grammostola <rosea.grammostola(a)gmail.com> a écrit :
> Objet: Re: [LAU] [ANN] LADI Session Handler - Preview 2
> Debian, Mandriva, ..., on their way... (!?)
http://ladish.org/wiki/installing_on_mandriva
Mandriva has jack2 dbus&classic and ladish 0.2 packages in Cooker since february. They will be part of Mandriva 2010.1
Regards
I love Fons' jconvolver convolution engine, and have used it for years, but I'm mixing a record now and I'm wondering if there might be a convolution engine out there which is compatible with JACK freewheeling mode.
I discovered a hack some years ago to change the maxsise to a huge number to get freewheeling mode to work for mixes, but it caused glitches at the end of mixes, and also made Ardour lock up often while finishing an export.
I like the CALF reverb, but I'd like to possibly use convolution for some of the reverbs I'll need.
-ken
Throughout the decades, I've read various engineers and producers waxing poetically and rapturously about the Neve console filters and how great they sound.
Filters are filters, and there must be some way to measure their their frequency characteristics and emulate it in software. Has anyone done that? In LADSPA on Linux?
-ken
Dear folks,
I have box with Debian Lenny on an ATA disk and 64 studio sat on a second SATA drive. It is an AMD 64 3200 machine with 512 MB RAM. I could get a much more powerful computer if necessary at some point. I also have an M-Audio Delta 1010 that I have not used yet and a Seck 12 8 2 mixing desk. I also have a Roland GR-1 guitar synthesiser that I guess could drive Rosegarden in some way in theory but I would need a MIDI to USB cable to do it most likely.
I am now reading the 64 studio audio manual and learning to use the LinuxSampler program. I need to buy a music keyboard probably with a USB output on it from somewhere to drive the sampler and record something in Rosegarden.
I am trying to understand what some of the programs do here and what the relationship between them is. Ardour seems to be a program that is equivalent to a multitrack tape recorder in the old days. Jack seems to be a program that connects to audio from the outside world like the M Audio Delta 1010 and files produced by synthesisers etc. So I assume that if I would record myself playing my electric guitar through a microphone into the Seck mixing desk I have and then into the Delta 1010, that the digitally sampled sound would go into the PC and through JACK before it got recorded e.g. in Ardour......
Or something like that. I could then record a bunch of tracks with the guitar synthesiser driving Rosegarden (?) and maybe create a drum track with Hydrogen. I could then mix them all down to make a master CD or whatever with Jamin.
Maybe I haven't quite got it right here.... Please correct me a bit.
What does Audacity do? Would it be helpful in my above recording activity?
What does Qsynth do that ZynAddSubFX doesn't?
the old GR-1 guitar synthesiser I have has a synthesiser and a sequencer in it. The sequencer I think just programmed the synthesiser to play sounds within a given time frame as I recall it when I used it many years ago. Based on that experience, I assume that Rosegarden which is supposed to be a MIDI sequencer is software that can drive synthesisers etc according to a program that you enter into it.
What is the difference between Jamin and GCDMaster and also between Timemachine and Ardour?
I used to use a Portastudio when I was a kid. The kit I have now could make infinitely superior recordings at least in principle. You could record some enormous number of overdubs and parallel tracks in a computer I guess. I could even record a live drummer with a dozen microphones and mix them down to 8 channels in the Seck Desk and record those simultaneously on the PC. I could then record guitar and vocal tracks on top of that as overdubs and then a bunch of synthesiser stuff. Presumably the Jamin program can master some huge number of simultaneous tracks. The Delta 1010 seems to have tremendous signal to noise ratio and frequency response capability so it seems that now the quality of the recording is based the around the microphones and preamps you have etc...
It seems to be like having a 64 or 128 track tape recorder connected to a highly flexible set of sequencers and synthesisers plus a whole load of effects processing like equalisation and digital delay that improve drum and vocal sounds with ace signal to noise, frequency response and the opportunity to avoid distortion etc with a bit of effort, but actually less difficulty than the old Portastudio because the new tools are much better ones.
Regards
Michael Fothergill
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Due to my motherboard choice, it turns out I need to use a bleeding edge
linux, in my case Ubuntu 10.04 beta.
Anyone else using this? In the sound preferences control panel, on the
hardware tab, when I select ICE1712 (what the M-Audio 2496 shows up as),
in the settings dropdown, I only see digital outputs and inputs. The
HDA device lets me select something line "Analog Stereo Duplux", but
nothing analog shows up for the M-Audio card. Does anyone know how to
make more options appear hear?
In another machine, still running ubuntu 9.10, I have another M-Audio
card, and it gives me a full range of options including both Analog
Stereo Duplex and Digital Stereo Duplex.l That card seems to be CM8738
based though.
I may be getting side tracked here from what I really want though. What
I really want is to use the M-Audio card with Jack, while still having
pulseaudio running on the onboard sound so that I can have both nice
audio applications and Flash outputting at the same time (albeit to
seperate channels on the mixing board). I see in QJackCtl that I can
select the alsa device to use, but it still shuts down pulseaudio
globally when I select the second card. And still for some reason the
second card doesn't want to work correctly here. When I had this card
in a prior 8.04 system, I disabled the onboard sound and didn't try this
dual output trick, but otherwise everything worked auto-magically.
I finally got around to building Yoshimi and I like it rather a lot. It seems a lot more stable than Zyn.
The pad I was going to use on this record, is a Will J. Godfrey pad included with the Yoshimi distribution, called 0030-Slow Strings.xiz . There is one small problem. Loading this pad-- and actually many other of the Will Godfrey patches-- causes Yoshimi to lock up completely, and take part of my window manager with it!
Basically, Yoshimi just stops at the file open dialog box, doesn't seem to be using any CPU, but won't let go of the window manager either, freezing the screen. Yoshimi doesn't respond to kill signals either, but I can kill -9 it to get my screen back. I'm using ion3 so it'll only take out one section of a split screen, not my whole screen.
The only error message that I get-- which I can see once kill -9'ing it-- is "X_TranslateCoords: BadWindow (invalid Window parameter) 0x800099".
Bummer, because I'd rather use Yoshimi than take my chances with Zyn, but the main thing I'd use it for is Will's pads anyway.
-ken
Hello everyone!
I want to split an audio signal in three frequency bands, as used in
multiband compression/other mastering purposes. I'm using LADSPA effects for
that. But I have the feeling, that I'm not doing it completely right, as the
simple three-band filter already changes the sound of the audio. Here are my
settings:
low_pass: LADSPA unique ID: 1891, lowpass_cutoff: 120, stages: 1
mid_pass: LADSPA unique ID: 1892, band_centre_frequency: 420, BAND WIDTH: 800,
stages: 1
high_pass: LADSPA unique ID: 1890, highpass_cutoff: 920, stages: 1
Mathematically seen, the three bands just touch. But then there are the
filter stages. Should I set them higher, so the bands don't overlap? Or should
grow/shrink the limits of the bands/ What would be your best bet?
Thanks and kindest regards
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
Hi guys,
I've recently started recording some screencast tutorials on the basics
of analogue-style sound synthesis under Linux. I've been using
Xsynth-DSSI as my example synth for the most part so far, just because
it's so accessible, though my most recent tutorial covers the Specimen
sampler.
If you've wanted to know more about how to create your own synth sounds,
this will hopefully be a good starting point! I'd also appreciate any
suggestions/corrections/etc. from more experienced users. The
screencasts are on Youtube here:
http://www.youtube.com/pneumanlsd
or on my blog (where they're also available in Ogg Vorbis/Theora format)
here:
http://blag.linuxgamers.net/?cat=117
Thanks
Leigh
