> BTW, one of the beautifullest examples of extreme stretching isn't
> Jane Fonda, but the "Beetstretch" project of Leif Inge:
It was done with Snd/CLM.
Hi,
here's an update to the notorious problem with some M-Audio USB
devices on kernel 2.6. I won't be able to test this in the next time,
but maybe some other owners of Quattro, Duo or Audiophile USBwant to
give it a try. If yes, please report to Alan Stern or linux-usb-devel.
ciao,
--
Frank
----- Forwarded message from Alan Stern <stern(a)rowland.harvard.edu> -----
Date: Tue, 16 Mar 2004 10:12:45 -0500 (EST)
From: Alan Stern <stern(a)rowland.harvard.edu>
Cc: USB development list <linux-usb-devel(a)lists.sourceforge.net>
Subject: Re: invalid alternate setting
Here's an announcement for people interested in using USB devices with
invalid configuration descriptors. I've completed a series of patches
that will make the system accept such devices, to the extent that it can.
You will have to apply all three patches in sequence; each is a
prerequisite for the next. They should go on top of a 2.6.4 system. The
patches are:
as221: http://marc.theaimsgroup.com/?l=linux-usb-devel&m=107937025411336&w=2
as222: http://marc.theaimsgroup.com/?l=linux-usb-devel&m=107937021027794&w=2
as223: http://marc.theaimsgroup.com/?l=linux-usb-devel&m=107937029404286&w=2
The first two patches just improve the error logging; you might like to
see what happens when you plug in your devices. The third is the one that
changes the kernel's behavior. Please let me know if the patches don't
work as expected.
Not all the USB drivers will work properly with that third patch
installed, so be warned. The USB audio class driver should be okay. I
have no idea what will happen with the ALSA USB audio/MIDI driver; it
hasn't been updated yet.
Alan Stern
----- End forwarded message -----
--
Frank Barknecht _ ______footils.org__
If specimen is too complicated, I would suggest using a modular synth like
galan.
Here's a quick howto for galan:
Per sample you have to create:
1/2x Sampled voice (mono/stereo), attach the samples to these
1/2x Randomaccess to realtime converter (mono/stereo)
1 x Control
To get anything out:
1 x oss/alsa/jack output
* Attach the sampled voice(s) to the randomaccess converter(s)
* Attach the control to the converter(s) (bottom round thingy to the
converters upmost round thingy)
* Attach the converter(s) output (right side thingy) to the sound output
* Create a button for the control (via the context menu new
control->button)
Then open the control panel (window->Show control panel)
Press the button in the control panel, and the sample will play.
You can add as many samples as you wish, just use different sources,
converters and controls for each sample.
(I used an old version of galan to do this, so there might be stereo
"sampled voice" plugins in the newer ones)
Sampo
daniel.vincent(a)blueyonder.co.uk:
> Me:
> > Just wondering if anyone has any recommendations for a piece of
> software
> > that I can play samples from, as simple interface-wise as possible.
>
> Sampo:
> > How about specimen?
> > http://www.gazuga.net/
>
> Thanks Sampo
>
> Even Specimen is a bit overly complicated, I literally want to be able to
> trigger samples whilst playing keyboards etc if that makes sense. I've
> created somthing similar in VB for Windows, which is basically a grid of
> buttons, each one linked to a sound file, configurable by an ini file I
> keep with the executable. Obviously I'd like to do something similar
> under Linux.
>
> www.southlondonlive.org
> www.onionjack.com
>
>
Hello all,
this comes a bit late, but in case you happen to be somewhere
in South Finland tomorrow...
I'll be giving a short presentation on Linux audio tomorrow at Otaniemi
(hut.fi) at 14.00 [1]. I'll talk a little about general issues of
developing apps for GPOSes, introduce ALSA, JACK and LADSPA, and finally
show a quick demo of well-known Linux audio apps (full PlanetCCRMA setup).
The presentation is part of the "Digital Signal Processors and Audio
Signal Processing" course [2] at HUT, but everyone is welcome
to come and listen (and ask tricky questions ;)). The presentation
is in Finnish, but I can translate on-the-fly if needed. :)
[1] Auditorium S1, Otakaari 5
http://www.hut.fi/Current/getting_to_otaniemi.htmlhttp://www.hut.fi/Current/otaniemi_map.html
[2] http://www.acoustics.hut.fi/teaching/S-89.510/
--
http://www.eca.cx
Audio software for Linux!
The tried and true method for setting speaker delay is to use a microphone
and a sharpie at the FOH position and adjust the second cluster by ear.
Matthew Polashek
Associate Editor, Silver Burdett Ginn - Music
Scott Foresman/Pearson Education
299 Jefferson Road
Parsippany, NJ 07054-0480
office: 973.739.8709
fax: 973.739.8098
Matthew.Polashek(a)scottforesman.com
> ----------
> From: Mark Knecht
> Reply To: A list for linux audio users
> Sent: Tuesday, March 16, 2004 2:10 PM
> To: A list for linux audio users
> Subject: Re: [linux-audio-user] measuring sound volume
>
> Patrick Shirkey wrote:
> >
> > Definitely
> >
> > So if a zone is 10 meters a minimum distance between systems would be
> > 100 meters?
> >
> > Could that be cut down to 50 to provide fusion at the edges of the
> zones?
> >
> >
>
> We used to set up some large outdoor concerts here in California. Don't
> forget that there are timing differences based on where people are
> standing WRT each set of speakers. If someone is just beyond the second
> set they will hear both the main set on stage as well as the set close
> to them.
>
> We used to set up all the speakers going in the same direction away from
> the stage and then digitally delay the later speakers by 1mS/foot that
> the two sets were separated. This worked really well and gave much
> better sound in the back of the area we were covering.
>
> I would think this would be very easy with Jack and some little digital
> delay app running under Linux.
>
> HTH,
> Mark
>
>
>
>
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I have been asked to work on a large outdoor party again this year.
This time we are going to have multiple zones with powerful equipment.
Can anyone give me some tips on appropriate spacing to enable a good mix
of cleaness and sound fusion?
For example if I have 4 x 20KW systems how far apart from each other
should they be so they don't overpower each other?
--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.comHttp://www.djcj.org/LAU/guide/ - The Linux Audio Users guide
Http://www.djcj.org/gigs/ - Gigs guide Korea
========================================
Apparently upon the beginning of the barrage, the donkey broke
discipline and panicked, toppling the cart. At that point, the rockets
disconnected from the timer, leaving them strewn around the street.
Tethered to the now toppled cart, the donkey was unable to escape before
the arrival of U.S. troops.
United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM
Frank Barknecht:
>
> BTW, one of the beautifullest examples of extreme stretching isn't
> Jane Fonda, but the "Beetstretch" project of Leif Inge:
> http://www.notam02.no/9/ who stretched Beethoven's 9th Symphony -
> which originally is about 1 hour - to be 24 hours long. The
> audio-files are several gigabytes large, an obvious use case for
> Bittorrent.
>
> Maybe Kjetil would elaborate a bit, how this was done, as I read that
> he was involved in the Beetstretch?
>
It was made with common lisp music.
The stretching-function was made by the
composer and programmer Anders Vinjar. The function
is basicly based on the make-granulate function with parameters
finetuned for the current stretch. Not very advanced,
most credits for the nice sound goes to Bill Schottstaedt
for the make-granulate function.
I did the dirty work, setting up scripts, high-level
lisp programming, doing conversions, fixing disk-space etc.
--
Hi,
I think the problems you described are fairly common with the Multiface. I have never gotten PD to work with a Multiface. XMMS does not work well, either, nor does mplayer (oss emulation is really no good, and it mplayer will not play through alsa with this card). Alsaplayer will work pretty well (better through Jack on my system), but I had to disable cddb-lookup because it would stall with CD's it didn't recognize. Xine will not work, and with the particular incarnation of kernel/drivers I'm using (ccrma), Audacity has been slowed to a complete crawl (it takes between 1 and 2 minutes to start up, and 1-2 minutes after you enter a command to execute it) - hopefully this will change when I upgrade. I suspect a lot of this is because the drivers are relatively new. I think Thomas Charbonnel (sp?) wrote the HDSP drivers (kudos), so he may welcome bug reports.
Matt
PS, which flavor of linux are you running?
->Question two: I get this error with pd and the multiface
using non-interleaved audio input
Sample width set to 4 bytes
snd_pcm_hw_params_set_access (output): Invalid argument
using non-interleaved audio
pd: pcm.c:1133: snd_pcm_readi: Assertion `pcm->access == SND_PCM_ACCESS_RW_INTERLEAVED\\\'
failed.
Pd: signal 6
and xmms sees the card (proabably from my .asoundrc file) but wont play through it. Ive run
hdsploader and the little red host error light is gone, so whats up?<-
hello list,
a colleague of mine seems to be extremely interested in performing
exaggerated time stretching of audio files. i wrote a csound instrument
based on the new streaming phase vocoding opcodes by richard dobson.
adjusting window size, it can perform passable time stretching up to 6 or 8
times the duration of the original.
but this colleague would like to have much more than that. he told me in in
a visit to cologne he used a POD system, where he could stretch sounds up to
400 or 500 times...
my questions are:
- which linux applications can stretch sounds with good quality up to let's
say at least 20 times?
- is there anything like POD for linux? anyone knows how POD does the
time-stretching? granular synthesis?
thanks as always.
lj
--
Hi,
I've been working on getting my linux system set up for recording with
jack/ardour. I came across the "Low Latency mini howto":
http://www.djcj.org/LAU/guide/Low_latency-Mini-HOWTO.php3
At the end, it talks about setting up IRQs such that the sound card is
at a high priority relative to other devices in the system. However,
all the information in the doc pertains to non-APIC machines. What
about an APIC-enabled machine?
The output of 'cat /proc/interrupts' shows my sound card (ICE1712) dead
last:
CPU0 CPU1
0: 6628091 0 IO-APIC-edge timer
1: 6 0 IO-APIC-edge keyboard
2: 0 0 XT-PIC cascade
9: 0 0 XT-PIC acpi
12: 803 0 IO-APIC-edge PS/2 Mouse
15: 5 0 IO-APIC-edge ide1
16: 0 0 IO-APIC-level usb-uhci, usb-uhci
18: 4178 0 IO-APIC-level libata, usb-uhci
19: 320 0 IO-APIC-level usb-uhci
20: 6406 0 IO-APIC-level eth0
21: 0 0 IO-APIC-level ICE1712
NMI: 0 0
LOC: 6628191 6628246
ERR: 0
MIS: 0
output of: lspci -v -s 02:01.0
02:01.0 Multimedia audio controller: IC Ensemble Inc ICE1712 [Envy24]
(rev 02)
Subsystem: IC Ensemble Inc: Unknown device d634
Flags: bus master, medium devsel, latency 32, IRQ 21
I/O ports at 8000 [size=32]
I/O ports at 8400 [size=16]
I/O ports at 8800 [size=16]
I/O ports at 8c00 [size=64]
Capabilities: [80] Power Management version 1
I've tried manually setting IRQs in the BIOS, moving the card around,
etc. No amount of tweaking can move it any higher in the list. (It's
set to IRQ 10 in the BIOS)
I guess my question is: Does the IRQ priority matter when IRQs are
abstracted by the APIC? If so, how does one change the priority in this
situation? I'm running 2.4.22, with the low latency and kernel
preemption patches installed.
Thanks,
Greg
--
Greg Jednaszewski
greg(a)attenuated.org