This is a set of ZynAddSubFX tutorials produced by the man himself.
I can recommend them all, and be prepared to be amazed :O
http://www.youtube.com/user/zynaddsubfx
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Dear Linux Audio developer, user, composer, musician, philosopher
and anyone else interested, you are invited to the...
Linux Audio Conference 2010
The conference about Open Source Software for music and audio
May 1-4 2010
Hogeschool voor de Kunsten Utrecht (HKU)
Utrecht, The Netherlands
Registration is open, and so is the call for abstracts and papers.
More information can be found on the website:
http://lac.linuxaudio.org/2010
For previous editions, look here:
http://lac.linuxaudio.org
For concerts, music and workshops a submission system and protocol will
be available soon. In the meantime, ideas and announcements can be sent by
e-mail ("lac -at- linuxaudio -dot- org ")
or written on the wiki:
http://wiki.linuxaudio.org/lac2010
We hope to see you all in Utrecht !
Kind regards on behalf of the LAC team,
Marc Groenewegen, lecturer music software design @ HKU
Hi,
I found this info:
"USB and jack
The USB interrupt period is 1 msec. To be able to get lower latency with
jack when using it with an USB device, you have to use a setting as
48kHz and 3 period. It will makes the buffer time a multiple of 1 msec
and you will get a much lower latency as with the default 2 period.
Additionaly, loading the snd-usb-audio with the parameter "nrpacks=1"
will give you a much lower latency (for this to work take care that
CONFIG_USB_BANDWIDTH is not set and CONFIG_USB_DYNAMIC_MINORS is not set
in your running kernel)."
http://proaudio.tuxfamily.org/wiki/index.php?title=Howto_RT_Kernel#USB_and_…
1) is this info still up-to-date?
2) how do I exactly take care of this:
"Additionaly, loading the snd-usb-audio with the parameter "nrpacks=1"
will give you a much lower latency (for this to work take care that
CONFIG_USB_BANDWIDTH is not set and CONFIG_USB_DYNAMIC_MINORS is not set
in your running kernel)"
(Debian (based) systems)
\r
Hello all!
This should have been ready before, but alas I couldn't get up to it. :-)
The song is titled "Carlos' Night" and can be downloaded here;
OGG version:
http://juliencoder.de/nama/carlos_night.ogg
MP3 version:
http://juliencoder.de/nama/carlos_night.mp3
Instruments and software used: Of course Nama for recording and mastering,
LinuxSampler for the piano, hydrogen for the drums, beatrix for the organ and
my roland XP30 hardware synth for the mellotron string and choir and the bass.
Enjoy and have fun
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 all, two quick questions (new to this list, let me know if I'm doing
this wrong),
1) What's support like on Linux on the USB Audio Interfaces? Specific
list below. I guess this is pretty much a 'does alsa support these
devices', at the current moment? Obviously this is the primary
requirement compared to 2).
2) I've got a Shure SM57 dynamic mic which needs amp-ing to get a
useable signal for my laptop. Instead of just getting a pre-amp I was
figuring I may as well get a USB audio interface, enabling me to finally
use the MIDI output of my Korg as well as record a presumably good
signal direct from my SM57. I've seen various complaints on the Fast
Track Pro, for example, that the SM57 volume is just too low for quiet
recording (ex. recording accoustic guitar), and its been confirmed as an
issue by one of their engineers
(http://forums.m-audio.com/showthread.php?t=14610). So the second
question is, is this a general issue with the SM57 and budget USB audio
interfaces, or would one of the list below be better?
I'm looking at the following interfaces:-
a) M-Audio Fast Track Pro
b) Alesis IO|2 Audio/Midi Interface
c) Tascam US-144 USB 2.0 Audio and Midi Interface
d) Edirol UA-4FX USB Audio/MIDI Interface
Please feel free to recommend alternatives with similar feature set (XLR
input, 1/4" input, some sort of MIDI is a preferred bonus).
Thank you all.
Yesterday i asked for assistance figuring out how to get mpd + pulse
audio upsampling to 24/192. The results are *fantastic* Bass is deeper
and tighter, decay is longer and vocals are clearer. I don't know why.
It kinda makes sense and it doesn't. It might all be in my head.
Anyhow, now that MPD and pulse are set to 192 they sound great. However,
gstreamer apps drop out after a few seconds, and sometimes pause for a
few then come back in for 5 seconds, then pause, come back in for 5
seconds (playing at the right speed however). Flash (adobe not gnash or
swf) stutters or doesn't come in at all.
Is this a buffering issue or because *they* are playing at the (now)
wrong bit rate? Setting pulse back to default fixes that but i'd really
like to have gstreamer and flash resample to 192 to synch with
pulseaudio.
Looking around i cant figure out a way to tell my system "every thing
you play with gstreamer, resample it to 192 first", and there's
*nothing* on doing it with adobe flash. Any ideas or pointers? Do i
need to have them resample or so i need to increase buffers (somehow)
If i can't do it that preferred way, isn't there a way i can set up
pulse so i can switch the bitrate back and forth with a toggle?
If this is too much hand-holding or not appropriate for this mailing
list just tap my muzzle and point it in the right direction.
Thanks
--
Bearcat M. Şandor
Jabber/xmpp/gtalk/email: bearcat(a)feline-soul.net
MSN: bearcatsandor(a)hotmail.com
Yahoo: bearcatsandor
AIM: bearcatmsandor
Nature of xruns
I never paid attention to mentions of xruns since I rarely got
them using a stock Fedora 8 kernel (and F6 before that) on a
x86_64 architecture, 4GB RAM. Yesterday though, I got plenty of
them and my observation is that in this case any real-time
feature associated to the kernel does not matter. In this case
the xruns are made by the way applications handles
stuff (whatever that is). So that makes me think that there
ought to be a guideline, a best practices in developing Linux
audio/jack applications.
Here's the observation setup.
First, the song. The song does not matter, It always plays
fine. The song is made by the use of Seq24 which drives several
soundfonts loaded by Qsynth in different engine instances, as
well as two sounds made by Zyn. Qsynth has several instances of
FluidR3_GM (120MB) and one SGM (250MB) loaded.
Second, the xruns. I take the outputs and drive them through
jackmix, add electric guitar line-in from a Vox amp, and have the
outputs of jackmix feed into Qarecord. Start recording. The
.wav result is polluted with xruns.
Third, same song w/o xruns. I terminate jackmix and Qarecord
and launch Ardour. I run all outputs into Ardour, and run the
electric guitar as well. Add some Freeverb and MultiEQ. Record.
No xruns at all in the resulting .wav file.
From this it is obvious that the application layer helps a lot
in promoting xruns. It is safe to assume that Qarecord is a
quick hack. But, there's seemingly a big difference in
application design regarding the handling of audio/jack stuff
between Qarecord and Ardour since one produces a festival of
xruns and the other none.
My question is, is there a guide/HOWTO around that addresses
the proper design of audio/jack applications or, is - in 2009 -
the only to know about that is by actually going through the source
code of successful Linux audio/jack applications ?
Cheers.
Hi
I missed the rpm challenge 2009, and I'm eager to join this year.
However the website (http://www.rpmchallenge.com/) still doesn't mention
2010 and have 2009 all over the place. Anyone knows what's up? Will
there be a rpm10 or? How about last year, when was it possible to join
the challenge?
--
Atte
http://atte.dkhttp://modlys.dk
I've been getting by so far with an amd64 box running debian, jack, ardour
etc through Terratec DMX-6fire (Envy24 based) card, but now find that I need
to record more channels simultaneously.
I would like (and can just about afford) the M-Audio FastTrack Ultra, but it
doesn't look as though it is going to be easy to get it to work under Linux.
It's a USB 2.0 device - it needs those 480Mb/s to record 8 channels
simultaneously - so it isn't class-compliant.
Has anyone succeeded in getting this interface to work properly in Linux?
If not, can anyone suggest alternatives in the same sort of price range?
Thanks,
Edward
Just wondering. Without an RT kernel here, my 2 laptops seem to run my
simple audio needs pretty well at 64msec latency. At least, it's never
bothered my playing along with computer-generated audio.
I don't do any heavy-duty audio work here. Once I tried Jackrack, put
one effect in it (that worked) or one amplifer (that worked) but trying
to use both didn't. But I don't know if that had so much to do with
latency or lack of RT kernel as with a smallish amount of memory and an
underpowered processor driving the whole thing. Now that I''ve upgraded
the memory on both laptops, perhaps it would work? On musicbox, with
512MB, using a single good quality (larger) soundfont was enough to
cause problems. With 768MB in it, it works without problems.
I see people on the list running much lower latencies than 64msec, and
seemingly trying to get even lower ...
So, just wondering.
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community