I recently had some problems with static while playing back some wav files with Winamp3 on my W98 partition, and after reading this thread, tried Winamp2.91 (I think, 2.9x anyway). It seems to be working alright so far, thoough I haven't tested it with the problem files yet.
What's wrong with Winamp3? Is it the same kind of static problems that I encountered, or something else?
Oh, and BTW, Winamp2.x for the last couple of releases supports Ogg Vorbis without needing a plug-in.
Thanks,
Barton
Need a new email address that people can remember
Check out the new EudoraMail at
http://www.eudoramail.com
>i don't think anybody in *my* immediate
>working environment (band, studio, college) would be able to do it**.
>that combined with learning the new structure of the file system
>(where is this app? where are its libraries? where are the
>files/patches for it kept? how do i make it work?) make it a daunting
>task for any lay user of the more popular windows/mac
>softsynths/sequencers etc. to navigate.
To be quite honest, unless you have a technical background you probably won't ever get very involved in Linux as an OS of choice for media production. I was a computer science major before I changed to a digital media major, so I have somewhat of a background to fall back on when using Linux. (Not enough of a background to contribute as a developer, but enough to at least understand what is going on... at least _most of the time_)
Even though Linux has made great strides in becoming an end-user OS for server and office use lately, it is still a hobby OS as far as audio, video, graphics etc are concerned. You're not going to have much success convincing people to use it for media purposes unless they are eager to get their hands dirty. Many of the DM students argue constantly over Win vs. Mac and when I tell them I use Linux I just get this really weird look. (At least from the ones who even know what Linux is.) I would never convince any of them to try Linux and I'm not going to try. It has to be something you're already interested in.
BTW: I would personally recommend giving Gentoo a try. It takes some work, but the end result is a system that is _much_ more responsive when doing audio work. Even when compared with something like planet-ccrma. Plus the Gentoo community is amazing.
Reguards,
-Reuben
well, as my subject line says, I'm wondering about sysex dumping and receiving
- I always used to do it in Cakewalk to save synth setups or dump patches to
the synths, etc.
now I'm not seeing such a function in rosegarden (which is the only program
I've got to run so far - oh I haven't tried kmidi I'll do that)
is sysex implemented in any sequencers, or is there a stand alone?
thanks! :)
--
--------------
Aaron Trumm
NQuit
www.nquit.com
--------------
> isn't this a linux list? ;-)
True, but it's handy to be able to give .ogg files to Windows using friends,
and they do enquire how they're supposed to play 'em.
Winamp (2.x is better than 3.x), FooBar 2000, there's quite a few.
You can get Windows Media Player to play ogg files too(as long as they're actual files, not ogg streams) if you install a Direct Show filter for them. http://tobias.everwicked.com/oggds.htm
Reguards,
-Reuben
-----Original Message-----
From: Mark Knecht [mailto:mknecht@controlnet.com]
Sent: Thursday, October 09, 2003 9:07 AM
To: Linux-Audio-User
Subject: [linux-audio-user] ogg player for Windows?
Hi,
Windows Media Player doesn't handle ogg files. Is there a Windows app
that would allow my to listen to ogg files?
(That other hijacked thread has an ogg link, but I couldn't bear to
hijack the thread in yet another direction!) ;-)
Thanks,
Mark
An interesting discussion on a filesystem "shootout":
http://slashdot.org/articles/03/10/08/2348224.shtml?tid=106&tid=185
The benchmarks themselves:
http://fsbench.netnation.com/
Looks like I may be switching filesystems on the studio machine soon!
--
======================================================================
Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com
129 Petta. Lake Rd, Saunderstown, RI 02874 - vox 401.338.9214
Without deviation from the norm, "progress" is not possible. - FZappa
Hi,
I can record from the analogue inputs on my audiophile 24/96 using:
ecasound -i alsawh,0,0 -o whatever.wav
Is it possible to record from the spdif input? I've been googling quite
a bit, and I don't understand if or how I can relate device or subdevice
numbers to the spdif input.
(btw, using the latest alsa release).
Thanks, Jordan.
The amsynth guy accepts contributions. I've sent him some settings which he distributes along with amsynth now.
Taybin
-----Original Message-----
From: Mark Knecht <mknecht(a)controlnet.com>
Sent: Oct 8, 2003 9:40 AM
To: linux-audio-user(a)music.columbia.edu
Subject: RE: [linux-audio-user] recording delay in Ardour...
> >> I think we Linux soft synth users should try to make that happen.
> >
> Reaktor users == thousands
>
> Linux softsynth users == dozens
>
> Umm, maybe it's a numbers thing ??
>
> == dp
>
I think you're probably right, although of the thousands I wouldn't venture
a guess as to how many of those are warez types. More than NI might want to
admit.
That said, Reaktor Session comes with 35-40 prebuilt instruments, and each
instrument has many sounds ready to go. It comes with a large library of
wave files built into the instruments, but they can be extracted and reused
elsewhere. I think the method to that madness is that NI creates enough
usability out of the box to get users really turned on to create more. Then
the library comes, which benefits all the users again.
I'd love to see that happen with one or more of the Linux tools. I simple
don't have the time or interest to become a programmer. When I turn on my PC
and try to write music I need tools that get me there faster. I don't want
to spend an hour grabbing blocks and wiring them together. By the time I get
that done I've lost my inspiration for making music.
I may be the odd man out around here. I'm not sure...
- Mark
This is just to let those who are interested know that I just commited
some fixes which greatly improve the sound quality in jackEQ and allow
the crossfaders to be fully functional including mute and all fader options.
Apart from being able to internally assign jack i/os I feel this version
qualifies for professional mixing use. To prove that I am going to be
using it live everyday for the next week while being paid to do so :)
Still only available from cvs you may need to wait a day or two to get
the newest version depending on the sf lag at the moment.
http://jackeq.sf.net
TODO:
internally assigning jack i/os
multiple interfaces - Long interface
- Tall interface (current)
extra button functionality - mostly for ease of use
unlimited channel support (mostly a gui design issue)
variety of xfader gain slopes for better mixing.
MIDI fader console support.
--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
Being on stage with the band in front of crowds shouting, "Get off! No!
We want normal music!", I think that was more like acting than anything
I've ever done.
Goldie, 8 Nov, 2002
The Scotsman
Hi. If I want to resample from 96khz to 48khz, what would give the best
result: use libsamplerate, or just drop off odd (or even) samples?
I guess the libsamplerate and any of its interpolations would cause some
digital garbage anyway, or are they intellectual enough to detect that the
rate is halved and just perform a drop off?
Since I don't know any programs that would just halve the rate by
dropping, I'd have to code it myself, but it shouldn't be too hard.
Any suggestions?
Tommi Uimonen
