Hello all!
Just wanted to share with all of you the great news I received today.
Apparently my 2-quarter class on Linux & Multimedia has been approved by the
Composition, Theory, and Musicology Department at the College-Conservatory
of Music, University of Cincinnati, so I will be returning as an Adjunct
faculty to teach the course there. Yay! :-)
The reason why I find this to be so important (and I hope that you will
agree with me) is because the students are the ones who will be making
important choices of tomorrow, including the choice of OS for the music
and/or academic studios. This way, they will have an opportunity to be
exposed to Linux and therefore more likely to embrace it down the road (as
opposed to not knowing about it at all and thinking that there is nothing
beyond Pro Tools), especially considering that Linux in many ways portrays
the true freedom of the free society. Furthermore, their compositions that
will be made with the help of Linux during the course of this class will
likely further enhance the importance and strength of Linux as the
multimedia platform, suggesting that Linux not only offers competitive
software versatility and stellar performance, but also the much needed
flexibility that does not encumber the flow of artistic thought.
At any rate, just wanted to share with you the good news I got today ;-)
Best wishes,
Ivica Ico Bukvic, composer & multimedia sculptor
http://meowing.ccm.uc.edu/~ico/
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>This is something that has bugged me ever since I built my Linux DAW.
> When I boot up, when I start jack, when I open a web page with
>Flash, when I get a licq message -- anything that uses my audio
>driver, I hear a fairly loud low-end pop.
>
>Is this common? Is it a known issue I can fix? I figure it's a
>problem somewhere in the following (backwards) chain:
I get this as well (DA converter -> RME DIGI9632-PAD SPDIF out -> ALSA -> JACK)
But it only happens the first time I start Jack, after that it doesn't do it anymore _unless_ I change which device on the card is being used by JACK.
My guess is that devices on sound cards have a default sample rate, bit rate or some other type of setting and when something accesses the device and changes one of those settings, it results in that pop sound.
-Reuben
Normally, I don't announce minor releases, but this is an exception.
The latest Specimen has a completely redone GUI, which I hope will make
using it suck a whole lot less.
Downloads: www.gazuga.net/downloads.php
Screenshots: www.gazuga.net/screenshots.php
This version of Specimen also relies on PHAT, so be sure you have the
latest version (0.2.2).
PHAT: www.gazuga.net/phat.php
This is a relatively ambitious endeavor, so please inundate me with bug
reports. And yes, MIDI control is coming really, really soon.
Seriously.
--Pete
I just tonight switched from the Planet CCRMA RH9 to FC1. The install
was purely from the CDROMs dated 4/25/2004.
I've seen all the latest posts about interrupts and did the required
reading on the internet. I really managed to get RH9 cleaned up but in
FC1 I'm seeing something a little different. I have a Delta 1010 and I'm
running an AMD Barton 2.6 with 5 PCI slots. The question is what the
heck are IRQ 16 and 22? I moved the sound and ethernet cards around to
get them to 16 and 22 as they used to be eth0 on 21 and ICE1712 on 22. I
have ACPI turned off as a service but don't have a "disable" option in
the BIOS. I did turn off USB support in the BIOS.
Is 16 like the equivalent of IRQ 3 since it's following 15?
[brad@mars brad]$ cat /proc/interrupts
CPU0
0: 81690 IO-APIC-edge timer
1: 75 IO-APIC-edge keyboard
2: 0 XT-PIC cascade
8: 1 IO-APIC-edge rtc
9: 0 IO-APIC-level acpi
12: 836 IO-APIC-edge PS/2 Mouse
14: 10789 IO-APIC-edge ide0
15: 735 IO-APIC-edge ide1
16: 0 IO-APIC-level ICE1712
22: 21 IO-APIC-level eth0
NMI: 0
LOC: 81633
ERR: 0
MIS: 0
I'm getting 5.8 msec latency in JACK with 128 frames/period at 44100 and
2 periods/buffer. A huge improvement over the 46.1 msec using RH9 with
capabilities.
Thanks, Brad.
Hi all,
I tried a couple of months ago to get USB sound working( headset + mic).
And during the system bootup, the headset would work as headphones(
unitl i unplugged the device). But not the mic.
Does usb sound even including handling mic info?
And if so, does alsa need any extra configuration to get this working?
thanks in advance,
bryce
It seems to me you could use the "save state" options, always
working from the "edit history" (i.e. the program that describes
the changes you made at various points). You'd still have various
in-between takes, but they'd be handled automatically by Snd;
your originals would never be touched, and you could backup to
any point in the sequence. I didn't realize this was a stumbling
block -- will look at it more closely -- perhaps all it needs
is better documentation.
I'm looking for a way to run multiple instances of Jamin. The purpose behind this is that I want to use Jamin as an insert with different settings for two separate busses in ardour, but Jamin segfaults when there is already an instance running. I already tried to make a copy of Jamin under a different name but with the same results.
Any suggestions would be helpful.
Thanks,
Reuben
>Well, you're *supposed* to use the -n option to set a different client
>name for the second instance. But, it looks like the new OSC control
>logic is crashing trying to open the same control port.
>
>When I try it I get...
>
> $ jamin -n metoo
> jamin 0.9.03
> (c) 2003 J. Depner, S. Harris, J. O'Quin, R. Parker and P. Shirkey
> This is free software, and you are welcome to redistribute it
> under certain conditions; see the file COPYING for details.
> liblo server error 9904 in path (null): cannot find free port
> Segmentation fault
>
>Is that what you're seeing?
Yes. I've tried with and without the -n option to set a different client name and that is exactly what I am seeing. (and I am also using 0.9.03 as well)
>Building jamin with `./configure --disable-osc' might work. I can't
>say for sure. My system can't comfortably run multiple instances, so
>I haven't tried it in ages.
>
>Meanwhile we need to debug that segfault.
I will try that and see if it helps.
Thanks.
-Reuben
Was: Converting sample rate: failed...
Erik,
Regarding your recent post on this subject:
> Funnily enough the issue you see as a "serious misconception" I see
> as a "significant advantage of libsamplerate over your converter". [0]
> [0] You claim that the highly localized behavior of the truncated
> windowed sinc is a bad thing. I claim that this localization is
> a good thing for converting a general digital audio signal because
> these signals are already highly localized (ie snare drum hit at
> 10.03 seconds from the start). From your description, your
> converter spreads these highly localized events over the whole
> of the output signal which I think is a bad thing.
-----------------------------------------
This is absolute nonsense. Your misconception is even more deeply
rooted than I had thought. You are obviously missing some theoretical
background I had assumed you had. If you had had this background, you
probably wouldn't have written the rest of your post, in addition to
not having posted this nonsense.
You mentioned an interest in certain characteristics of the resampling
method I'm using. Either do the analysis or do the literature research.
I've already told you everything you need to know: FFT with and without
overlap; overlap accomplished with raised cosine windows. Only someone
who is utterly clueless would insist on obtaining my code for measurements.
(And only someone who is very immature would characterize this standard
advice as "hot air" before they even saw it.)
Good luck,
Dave.