Hi there,
I am looking for a way to improve the sound from an old VHS tape.
The tape has been played an played and played.... the osund has now some
fluctuations that I'd like to get rid of.
I know how to use SOX, GWC or Audacity but do not have the knowledge about
sound processing. Bcast 2000 seems to have useful tools as well, but really
it's above what I know (apart from gain and LPF).
So if anybody has a link, an idea, a trick, anything will do at this stage :-)
Thanks a lot
Edouard
Hi all,
someone know this error:
# modprobe snd-es1938
/lib/modules/2.4.20-20.9/kernel/sound/pci/snd-es1938.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module parameters,
including invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.20-20.9/kernel/sound/pci/snd-es1938.o: insmod
/lib/modules/2.4.20-20.9/kernel/sound/pci/snd-es1938.o
failed/lib/modules/2.4.20-20.9/kernel/sound/pci/snd-es1938.o: insmod
snd-es1938 failed
Please can you have some suggestions ?
My system have RedHat 9.0 with kernel 2.4.20-20.9
and the soundcard is ESS1969
Many thanks,
--
Davide Morano
Q-Lab Team
Wind Telecomunicazioni
Davide.Morano(a)mail.wind.it
http://wwww.libero.it
Does anyone on this list have an Echo card installed in their machine?
(Gina, Darla, Mia, etc...)
If so, could you email me the output of lspci -v please?
It would be greatly appreciated.
Austin
--
Austin Acton
Synthetic Organic Chemist, Teaching Assistant, Ph.D. Candidate
Department of Chemistry, York University, Toronto
MandrakeLinux Volunteer Developer, homepage: www.groundstate.ca
Yes, I would say that is a latency issue. On the other hand, since Ardour is multi-threaded, the GUI degrades gracefully. So it might have started recording even though the record button hasn't popped back.
Taybin
-----Original Message-----
From: Aaron Trumm <aaron(a)nquit.com>
Sent: Oct 6, 2003 9:16 PM
To: linux-audio-user(a)music.columbia.edu
Subject: [linux-audio-user] recording delay in Ardour...
hello - I think this is a simple question - right now when I record into
ardour, there's dang near a half second delay before it actually spits it
back out - that delay is reflected in the recording, too - I didn't THINK
this was happening yesterday...
is this a latency thing? I know I haven't maximized my latency, I'm running
the redhat kernel from planet ccrma, not the actual planet ccrma kernel and
when I check to see if low latency is turned on, I find that I don't even
have a low latency patch :)
the weird thing is I SWEAR is wasn't happening yesterday :)
--
--------------
Aaron Trumm
NQuit
www.nquit.com
--------------
Hi folks,
I'm starting to think that the M-Audio Ozone is just the thing for me
since I want to play with both instrument recording and soft synths,
and I don't have much room to spare. Does anybody know how well this
gizmo works with Linux? Any details are appreciated!
Thanks,
-Nathan
--
>>>-- Nathaniel Gray -- Caltech Computer Science ------>
>>>-- Mojave Project -- http://mojave.cs.caltech.edu -->
So I got a bunch of samples lying around from my FruityLoops days, and I
want to make use of 'em. The catch is, they're not standard PCM encoded
WAV files. While they are waves, they use format '674f', and I can't
figure out exactly what it is. XMMS can play it, sox doesn't know what
to do with it, and I've grepped the crap out of libsndfile and
libaudiofile and haven't found reference to it. Anybody know anything
about this? I'm about to go through the XMMS sources, so my post here
might be premature, but any and all leads will be appreciated.
Peace,
=Pete
--
You can only run configure at the top level of the Ardour source tree.
You don't want to know why this is true. Don't try to work around it.
Greetings all:
I am trying to learn about the LADSPA plugins. I have downloaded and
installed the ladspa_sdk, read the ladspa.h header file and read Dave
Phillips article on the O'Reilly network. I've also added the CMT plugin
library.
Okay, so I've got my plugins but I need some helping making them go. I don't
really understand how they work. I am trying to learn about them in a
console environment, that is to say just by themselves (if that is possible,
maybe that is part of the difficulty). I understand the basics of the C
language and the Linux operating system but most of the information I have
found seems to cover developing plugins and consequently goes right over my
head.
Lets say I wanted to use the sine.so plugin from the sample library to play
a continuos pitch from my sound card, how would I do that? I'm confused by
the need of an input file for this as I see the sine.so plugin as the
source. Can the output file be a dev file? How about using the ALSA hw
plugin hw:0,0 as the output, how would I do that? I've read the ALSA library
documentation about plugins and LADSPA but again it goes right over my head.
Anyway, I though I'd ask these questions to this list and see what came of
it. I'm using recent ALSA drivers, with a 2.2.20 kernel on a very old
computer (hence the desire to learn about the plugins from the console).
Thank you in advance for any consideration and advice.
Sincerely,
Paul
>
>On Friday 10 October 2003 23:12, Jan Depner wrote:
> You work on code for supercollider? Interesting. A close friend of
> > mine was working on that before it got shut down. PhD in particle
>> physics. Worked at Fermi Lab prior to that. He uses Linux now as do
> > I. What, exactly, was your point?
oh no! I mean SuperCollider the music language app. look at
http://www.audiosynth.com
and help make the linux port of SC server!
http://sourceforge.net/projects/supercollider/
as to my original point, it's mostly being summed up in the other
thread of mp3 vs. ogg etc...
>Matthias writes:
>Well spoken! I think this is not only related to ogg, it's related to most of
>the technology in the linux audio scene. We need to show people that it can
>actually be used to create great stuff.
>
>IMPO Linux audio isn't ready for the average Windows/Mac-user, but there is
originally, I wrote to respond to questions regarding a lack of
prebuilt tools, a small user base, etc. I was giving my perspective
as an educated, experienced studio and computer music user, but
coming from the professional and academic worlds wherein we all used
Macintosh. My point was that I was excited by the prospect of
learning morel, being on the cutting edge, being in the community of
Linux users, but that, even as an experienced user, I was finding it
very very difficult to make the transition, but I thought it might be
intersting to understand why I was learning linux. The philosophizing
came in the form of questions like: "is it a good thing or a bad
thing to be an elite group of users?" "do linux audio users want to
exist as part of a specifically educated group or to make it useable
by joe-reason user?" basically pointing out that when you become so
far inside a specific knowledge base it is sometimes hard to see how
opaque it may appear from outside --cf. improv or avant-garde music
cliques...
I was just poking the hornet's nest, as usual. please don't take this
sort of response as complaining! I am really happy with the available
sounds and software and you can expect a note somewhere on any
releases i make from now on that linux was used.
--
_________________________________________________________________
Jonathan Segel -- MAGNETIC -- PO Box 460816 S.F. CA. 94146-0816
4014 Brookdale Ave. Oakland, CA 94619
jsegel(a)magneticmotorworks.com <-----> magsatellite(a)yahoo.com
http://www.MagneticMotorworks.com
tel (510) 534 7825 cell (510) 484 7415 fax (425) 955 4495
Hi all,
To be honest I'm posting this before investigating all possibilites
myself, but i'm fast approaching the 'hugely frustrated stage'.
How do I enable the spdif input on my audiophile 2496, please? A bit of
googling suggested I should change the line in
/usr/share/alsa/cards/ICE1712.conf from
slave.channels 10
to
slave.channels 20
which should enable input (and disable output since you can't have both
at the same time on this card.) Correct?
I'm looking at the envy24control mixer settings, and I've unmuted the
spdif input channels and turned up the sliders, but no input is showing
on the meters. Is there anything else I should be doing? I haven't got
an .asoundrc, should I?
thanks, Jordan.
hi fellow linuxicians !
Is there any sequencer out there, that is able to control abstract musical
paramenters (simply numbers, so to speak :) for synth-systems like pd, beast
etc... there may be some fine midi sequencer, but sometimes note-on /offs on
a
32-grid just ain't enough to control something like grain-loop machine, or
whatever....
I'm having something in mind, what could be something like a OSC(open sound
control) sequencer...with programmable event/signal templates like /start or
/stop ... and nifty curve-based drawing tools, like those found in sequencer
with decent automation-handling (eg. logic,protools,...)
I'm just wondering if there's anything like that out there
or maybe a fitting (sequencer) develompent-library..
cheers !
jan.
--
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