There are tools on Windows to automate the transfer of tapes to MP3: the
software can sense when the record is actually being played and it
starts recording, can sense when the record has ended and it stops
recording, it can even merge two sides of the same tape into the same
file, etc. It does that, I guess, by monitoring the signal on the inputs.
Of course, all this stuff can be done even without a specialized tool,
but when you have a huge number of tapes to transfer, it's nice to get a
little help from a specialized app like that.
So, is there anything like that available for Linux?
--
Florin Andrei
http://florin.myip.org/
it would be great to have a wiki with all the LA apps, there dev statuts,
description, updated often, kind of a database
--
Um abraço, Jorge Salgueiro
.................................................................
Use GNU/Linux: free culture for a free society
Sebastian:
>
> A QUESTION: I read something about a vst->ladspa wrapper (ladspavst.so
> or something). Is there such a wrapper, because I couldn't find anything
> by googling => In other words: Is there a way to use VST plugins in a
> LADSPA environment.
>
Yes, its called ladspavst, and I get 1320 hits when searching for
"ladspavst" in google...
You probably won't get it to work though. It just worked with the
old vstserver which won't compile anymore. However, making the
ladspavst ladspa plugin work with fst or dssi-vst shouldn't be much
work, maybe an hour or something. I'm surprise no one has ported it
the last 6 years.
Hi all,
I converted the bass track off a midi file to ogg, muting out all the
other tracks via timidity. The purpose being I'm about to buy some
monitor speakers and I'm deciding what low frequency range is
acceptable. I tried audacity and analyze-->plot spectrum, but I can't
get that to show me what the low end is doing (I'm sure I'm missing
something simple).
Anyways, here's a link. What frequency is that bass around? How can I
measure that?
http://braziloutsource.com/random/carnival_bass.ogg
If curious, I'm looking at the m-audio bx8a. I live in a remote place
where I can't just walk into a store and demo them, so I have to buy
off the internet.
Thanks!
Robert
[sent directly to me by Christopher, replying to the list so others can
answer]
Christopher Stamper wrote:
> Why not first try a good pair of studio-quality heaphones with a good
> sound interface... Seriously, it makes a huge difference.
It can make a huge difference, depends on how bad your existing speaker
and sound hardware are.
> SoundFonts are available for free on lots of sites, just google for free
> soundfonts...
Not all soundfonts are created equal.
> Anyone know if sf2midi.com <http://sf2midi.com> has pirated sf's?
Don't know - never heard of it!
> On Jan 10, 2008 4:36 AM, david wrote:
>
> Ken Restivo wrote:
> > On Wed, Jan 09, 2008 at 12:00:42AM -1000, david wrote:
> >> Simon Williams wrote:
>
> Actually, david wrote this part:
>
> >> I've been using and loving the Yamaha audio patches for years.
> >> Someone else on the list mentioned that a lot of the Yamaha audio
> >> sound is due to the licensing of some very sophisticated wave guide
> >> audio patents from MIT or some college like that. I suppose you
> >> could read the patents and figure it out, but ...
> >
> > Wasn't it CCRMA at Stanford?
>
> I think that was it. I don't really recall, I just enjoy the Yamaha
> sound.
>
> Someone on the list provided a link to the site with the information
> about the patents (which belonged to the university, just were licensed
> to Yamaha).
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
On Thursday 10 January 2008 09:00:03
linux-audio-user-request(a)lists.linuxaudio.org wrote:
> Message: 8
> Date: Wed, 09 Jan 2008 22:50:07 +0100
> From: Dragan Noveski <perodog(a)gmx.net>
> Subject: Re: [LAU] [ANN] new version of ssg (Simple Sine Generator)
> To: A list for linux audio users
> <linux-audio-user(a)lists.linuxaudio.org>
> Message-ID: <4785418F.9070502(a)gmx.net>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Dragan Noveski wrote:
> > Nedko Arnaudov wrote:
> >
> >
> >> New version of Simple Sine Generator is available.
> >>
> >> It now requires lv2core.
> >>
> >>
> >>
> >
> > hallo nedko, could you please provide us the URL to this library
> > (lv2core)?
> >
> > cheers,
> > doc
> >
> >
>
> the answer on my question just came with dave ANN - sorry for the noise!
>
> cheers,
> doc
I'm sorry, could someone explain the lv2core call? Download? I'm confused :)
Hey y'all
I'm looking for something like an array of bandpass filters with
feedback, to simulate resonance in different sized spaces. Doesn't have
to be particularly real-world-accurate.
I've looked at a bunch of jack-friendly and LADSPA effects, and things
like that. But none of them seem to be quite what I'm after.
Seems like the best option is JMax, or pd, or something like that. But
they seem like long steep learning curves.
bye
John
Hi,
> I'm new to ardour and would like to know how you created this "channel
> strip". I am accustomed to certain features already being part of each
> channels/tracks strip, like eq, under Windows recording apps, but have
> found this isn't the case with ardour.
> Could you please explain how you did this, thanks.
Well, I don't know if we mean the same, but I just put the desired
effects into the ardour-track in the mixer window. Nothing special.
Check out this mixer strip:
http://ardour.org/files/manual/images/mixerstrip.png
The plugins "Barry's Sa....", "Allpass de..." and "AM pitcha..." are the
plugins. And that is where you can load, unload, bypass, etc. the plugins.
> Try inserting the "Artificial latency" plugin from the swh-plugins
> mhmmm... that doesn't work. However, the latency of the swh plugin gets
> compensated like it should be.
> Those two statements contradict each other. What exactly is not
working on it? How exactly are you using it?
Hi.
I did what Lars suggested, I tried inserting the "Artificial latency"
plugin from the swh-plugins package after the VSTs. I understood "After
the VSTs..." as loading it into the same channel strip right after the
VST plugins. The channel routing is now:
Audio -> Stereo Bus -> VST plugins -> swh artificial intelligence ->
Ardour Fader/Panning ... -> Master Out.
Then I tried to change the delay value of the swh plugin. The Swh plugin
itself doesn't produce any effect. The VSTs are still delaying the
channel without getting compensated.
Am I doing something wrong?
Thanks in advance.
Best,
Sebastian.
Hi , i've got this message from ardour 2.1 when open a session with
two tracks :
[WARNING]: you cannot use colons to name objects with I/O connections
I bring this session from another machine , where I did the recording ,
to this one where from I'm writing now , in an SD card with Fat file
system.
Also , Jack produce tons of xruns untill it die.
The session has two streo tracks with diferent regions for each take .
The souncards are the same in both machines (Terratec DMX6 Fire -
ice1712 ) .
Here is a screenshot of the issue.
http://www.imaxenes.com/imagen/ardour_error1lk83py.png.html
It only happens with this session , I have some others with more tracks
and effects and everything works great .
Linux ubuntuStudio 2.6.22-14-rt #1 SMP PREEMPT RT Tue Dec 18 10:01:34
UTC 2007 i686 GNU/Linux
Jack 0.103 , Ardour-2.1 , in a Pentium IV 2.6 Ghz with 1gb of RAM .
Any idea?
Thanks a lot .
http://createdigitalmusic.com/2008/01/08/ces-free-transmission-audio-distro…
It is nice to see this coming along.
Meanwhile, does anyone know anything about the Burn program on this?
I'm curious to know more about if it has anything special to it,
particularly in the way of integration with the various DAW programs.