Hi folks
Does anyone know how to manage one of these cards, I've bought a used
one and put it in my computer without having the ADAT interface, I have
ordered one and hope to get it tomorrow, but I wasn't able to start Jack
and got the error message that jack couldn't find a propper recording
channel. I removed the card again so I can't give you the exact error
text.
By the way I'm running UbuntuStudio.
Take care
/Sv-e
Hi all,
Recently I've been working on 4 new tracks.
I've been trying new experiments.
http://www.archive.org/details/Coloured
More electro an crazyness than usual in:
- My super hero
- Telemorning bells
More feelings than usual, mainly using my voice:
- Put it away (this one is fresh from today :-) )
- Lights on
These are very early drafts, but I would appreciate your opinion, comments
in the very beginning so I can have an idea on how to continue them..
Thanks!
--
julien
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A few videos of some fretless guitar playing:
http://www.youtube.com/watch?v=VZuE2LWLXMchttp://www.youtube.com/watch?v=8OQm7vdt7UE
Heh, I thought this was pretty unique, then I looked in the little right-hand oolumn on YouTube and found plenty of other fretless guitars being played. Oh well.
The audio was recorded with one of my favourite little utilities: the ever-powerful jack_capture.
The video was recorded onto SD with a generic Kodak digital camera, then copied over.
I then processed the audio (using the great CAPS Amp plugins), and sync'ed it up with the video using Ardour, exported it, and used Avidemux to replace the camera's audio track with the processed one.
- -ken
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Quoting Victor Roetman <victory747(a)gmail.com>:
> The question I have is whether I should be recording 24 bit audio at
> 44.1 kHz or at 48 kHz if the end result will be Compact Disk audio. It
> seems that some pro-audio equipment tends to prefer 48 kHz, and many
> people do their recording at 48 kHz. The heart of the question is this:
> Will the slight slight increase in quality of recording and mixing at 48
> kHz outweigh the slight reduction in quality from re-sampling 48 kHz to
> 44.1 kHz?
All resampling is bad for the signal. The slight benefit achieved from the
higher frequency anti-alias filter at 48k compared to 44.1k is nothing
compared to the issues caused by resampling.
And to be honest, i'm not that sure that most sound cards even have
different anti-alias filters for 44.1k and 48k.
> How about if the project was also using loops recorded at 44.1 kHz? If
> the Ardour project is at 48 kHz, the loops would need to be up-sampled,
> and then the whole thing down-sampled again to 44.1 on export. Would
> that up-down sampling offset any benefit of using 48 kHz in the first
> place?
Even more reason to record at 44.1k. I always use 44.1k because CDs are the
target media. Less fuss, less processing on the signal.
Sampo
Hm, my reply to munging got damaged there!
Kurzweil developed a box to do this, I think as a project after selling his company to Young Chang. It was just such a PLL that would sync to the drummer so that the massively overproduced garange bands could play their music live, something that was impossible without miming the whole act. Basically the drummer would lead the tempo (some are notoriously bad at following a set tempo anyway), from that the software would derive a MIDI time code and away it went to drive the sequencers.
Not sure if it caught on or even worked. Firstly garage bands became more popular than overproduced, resampled and sync'ed music, but it never worked very well with extended endings as the sequencers would become confused - they understood if the drummer wanted to change the speed for mood, or for lack of drumming abilities, however if they wanted to play an extra chorus it kind of fell apart.
Have not heard of any open source examples, but, as Fons says, it is possible and would be an interesting project.
If it was not Kurzweil I am certain somebody can give a pointer in the right direction.
Nick.> Date: Wed, 22 Aug 2007 00:04:08 +0200> From: fons(a)kokkinizita.net> To: linux-audio-user(a)lists.linuxaudio.org> Subject: Re: [LAU] Progressive Quantisation> > On Tue, Aug 21, 2007 at 10:01:16PM +0100, Folderol wrote:> > > What I would like to see is quantisation algorythm the detects trends> > rather than absolute values, then progressively applies small> > corrections to keep overall timing correct. (it would of course have to> > operate over all tracks simultaneously).> > A PLL (phase locked loop) or DLL (delay locked loop) will do exactly that:> follow the general trend and ignore local variations. Neither of them is> difficult to program, but you'll need a good grasp of the theory to make> them work,> > There are mixed open/closed loop algorithms to do this as well. They can> work even better sometimes, but there's very little 'open' information> available on those.> > -- > FA> > Follie! Follie! Delirio vano è questo !> > > _______________________________________________> Linux-audio-user mailing list> Linux-audio-user(a)lists.linuxaudio.org> http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-user
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> How about if the project was also using loops recorded at 44.1 kHz?
> If
> the Ardour project is at 48 kHz, the loops would need to be
> up-sampled,
> and then the whole thing down-sampled again to 44.1 on export. Would
> that up-down sampling offset any benefit of using 48 kHz in the first
> place?
Hey, how about resampling them up, do everything you wanted to do with
the rest of the song, stripping the loops out, downsample the song,
adding the *44.1*-samples, and finish off the mix.
Go!
Gasten
Hi,
I believe it is possible to control the volume, panning, reverb, etc in
Ardour and Rosegarden using the knobs and sliders of a MIDI controler,
but I'm not getting it right. The controllers come with some default
assignments, but when I use the knobs and sliders nothing happens.
Probably the software has different assigned channels, where can I
change them to match the controllers settings? Little Help?
Thanks,
Lobo
Hi1
Is there any chance to play swf-videos on the commandline? I tried mplayer,
but it told me, that the format was not exactly supported. Is there probably
some kind of windows-dll or linux .so I can get somewhere and copy in my
mplayer's codes-directory?
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
Simon Williams wrote:
> Interestingly, amidi --dump gives absolutely nothing unless I press some
> keys. So where are these active sensing messages aseqdump keeps getting?
Filtered out unless you add the -a option. :)
> Here is what I got from the MK's own USB...
> 90 30 50
> 90 30 00
> 90 32 4C
> 90 32 00
> ...
>
> And here is what I got from the USB midi adapter...
> 90 30 50
> 30 00
The USB MIDI protocol doesn't allow runing status ...
> 00 32
> 4C 00
... and here is one zero byte too much. Apparently, this device doesn't
conform to the USB MIDI specification.
Please uncomment the "#define DUMP_PACKETS" in line 58 or so of
sound/usb/usbmidi.c, recompile the snd-usb-audio driver, and show what
the driver writes to the system log when such note-on messages are
received. (Warning: the driver will dump all packets, including those
containing active sensing messages.)
> Slightly OT: I would like to test this in windows, to prove that the
> device really is broken, but I can't get any sounds from it or my MK.
> Does anyone know of a free software synthesizer for windows which will
> play sounds from a midi instrument?
You should be able to use any sequencer to connect a MIDI input port to
Windows' built-in software synth.
Regards,
Clemens