Hi listmembers,
Looking over the info here:
http://www.m-audio.com/products/en_us/
It looks as if the Delta 1010LT has 2 XLR inputs, whereas the Delta 1010
only has 1/4" jacks.
http://www.m-audio.com/images/en/callouts/big/delta_1010.jpg
I have 1" condenser mics (AKG 3000B) which require phantom power. I
assume neither card provides it.
Can owners of these cards confirm my assumptions? What recommendations
would you give for me to purchase.
We have a 5 member band, two instruments have built in pickups, the harp
has a 1/2" condenser mic (I'm not sure if it is battery powered or
requires phantom power), all of the other instruments need miking
(Uilleann, Great Highland and shuttle bagpipes, anglo concertina,
fiddle, whistles, flutes, bodhran, bones and egg rattle).
I am building a box with Gentoo Linux intended for recording and
editing.
TIA
--
Phil
Our 2nd CD: http://www.cdbaby.com/naomisfancy
Naomi's Fancy performances: http://naomisfancy.virtualave.net/schedule.html
Hello -
Can anyone recommend the best way to set this combination up? My system
is crashing during playback with the Korg connected to the Delta1010LT.
Do I need something special in a .asoundrc ? What are the steps to set
up the Korg through the Delta? Will it always be hidden behind the Delta?
Thanks!
Steve
Hi Arnold,
> So maybe someone
> could combine these two and produce a stereoset of IR's of the following
> scenario:
> - A normal-to-good studio room with reasonable acoustics.
> - A pair of studio-speakers (which most of you seem to have) and a
> binaural-recording-mic in the optimal stereo-listening position.
> The resulting sound (after feeding this IR into jack_convolve, etc.) should
> (from my experience and understanding) be as if you are listening the music
> in the studio...
> Thats what I wish, perhaps someone has the resources to create this.
If I understood your request correctly, I have already done what you have
asked for. Sorry if you missed my previous postings or misunderstood the
purpose:
http://home.earthlink.net/~davidrclark/linux_audio_users
The program to produce IR's for the situation you described is Boxster.
The program to do stereo convolution is contained in Mixster. You could
also use Flo's recent program for this, provided it accepted WAV IR's.
Steve Harris has worked on LADSPA plugins for doing convolution, also,
but I'm not aware of the latest. Mine is non-realtime whereas Flo's is
realtime. Making mine realtime would be very straightforward. I've also
offered a whole set of IR's for a large concert hall. These IR's produce
binaural images complete with reverb, echo, equalization, "panning," etc.
There are other IR's available on the web, but I've not heard any that
were binaural, nor have I heard any that I considered to be very good,
possibly because they were recorded rather than calculated.
I'm willing to help serious potential users, but not students with a semester
project nor those who like to try everything out to see how it works.
Been there, done that (times 20 or more including other projects over the
years). I don't mind at all helping professors in constructing class
materials, but I have not done that yet with this work.
Dave.
Andrew/Dave:
By way of odd, somewhat related observations, I recently got a new receiver
which has DTS Neo:6 surround decoding. There is slight phasiness about the
Neo:6 sound when used on a pure Blumlein X/Y recording that made me think it
might work to externalize the stereo image when using headphones. (The
surround sound seems shoved away from the listener toward the side walls of
the hall.) After reading your posts, it tried listening to the front channels
only on headphones. It definitely works. Of course, that leaves open the
question of what to do with the rear channels.
I read that some think DTS Neo:6 is good for decoding Ambisonic UHJ
recordings, but it is clearly doing more than the old UHJ decoders, for which
the formulas are readily available on the Internet and which did not seem to
have the same kind of phasiness.
Does anyone know anything about the theory behind DTS Neo:6?
John
Hi,
i'm writing this email, because i'm interested in what plans the
different linux audio developers have for the year 2005 Any new
revolutionary applications planned? Major changes to some of the
existing apps? So let us know. What are your roadmaps for 2005? What are
you guys up to?
Where is help needed?
Anyways, i start off with my own stuff:
amidimon - a terminal midi monitoring app which is very incomplete, but
works good for me. I will finally add the autoconnection feature
somewhere in the spring of this year. I have seen some alternative midi
monitoring apps on the mailing lists in the last year, so maybe i just
dump the project completely instead
rtc_mtc_gen - a small MTC generator app for alsa_seq. I will teach it
drop modes and FPS other than 30
Session - a small gtk2 app to organize collections of programs which
make up a "session". I plan to add LASH support sometime this year. I
think LASH needs a major revisiting though as imho the adoption of it is
rather slow. Maybe it's already too intrusive to applications (i will
start another thread on LASH specifically i think)? Session is still
alpha (planning stage) and if anyone wants to get involved i will be all
ears to suggestions..
The 2.6.x linux audio wiki - i need to definitely update it with newer
information. Sadly spammers have discovered my wiki and i had to disable
public editing access. This is the first thing i'll do after i have
finished some university stuff.
Find my stuff here:
http://www.affenbande.org/~tapas/wiki/index.php?Ware
and here:
http://www.affenbande.org/~tapas/linux-2.6.x-ll.html
Regards,
Florian Schmidt
--
Palimm Palimm!
http://affenbande.org/~tapas/
I'm sure the 96 kHz converters have not been designed for
this kind of application. Only choise is to record with
the normal speed.
It could be more realistic to buy multiple cassette players.
That would reduce the digitization time easily. You would
not need quality players if you have only speech on the
tapes. The neat thing is that multiple players would reduce
your idle time as you would need to change the cassettes
all the time.
If there would be simple robots available, one could
leave the cassette changing to the robot.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
I have a few papers on mirror image methods if we want implement
something. Mirror image method is perfectly ok for a musical
and a virtual reality reverberator.
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
Hi,
I'm trying to build fmit, and it can't find glut, where can I get it for
Fedora Core2 Planet CCRMA?
GLFormants.cpp:174:21: GL/glut.h: No such file or directory
GLFormants.cpp: In member function `virtual void GLFormants::paintGL()':
GLFormants.cpp:242: error: `GLUT_BITMAP_HELVETICA_18' undeclared (first use
this function)
GLFormants.cpp:242: error: (Each undeclared identifier is reported only once
for each function it appears in.)
GLFormants.cpp:242: error: `glutBitmapCharacter' undeclared (first use this
function)
GLFormants.cpp:278: error: `GLUT_BITMAP_HELVETICA_10' undeclared (first use
this function)
Thanks,
Andrés
Hi Arnold,
That is very close to Andrew's original question about "out of the
head" sound from CD's.
There are headphone amplifiers that *attempt* to produce binaural sound
from normal stereo recordings. There may be software, but I haven't
seen any. I ran some tests with my own software, but was very discouraged
with the resulting sound. It's almost impossible to recreate a true
binaural image from two-channel stereo because a lot of signal is
duplicated in both channels. Binaural listening requires a very clean
separation between the two channels and very precise phase differences.
It seems far better to do it right from the beginning.
I have the same thought as you: That *some* improvement might popularize
the method and turn the tide. However, there are also a lot of people with a
vested interest in keeping things as they are. As long as the general
populace is kept in the dark, nothing will change.
Dave.
I have a soundblaster live 24-bit card and am following the directions
at http://www.alsa-project.org/alsa-doc/doc-php/template.php?
company=Creative+Labs&card=Sound+Blaster+Live
+Value.&chip=emu10k1&module=emu10k1
I get to the last final command: Now insert the modules into the kernel.
modprobe snd-emu10k1;modprobe snd-pcm-oss;modprobe snd-mixer-oss;modprobe snd-seq-oss
and I get this error below. I do not have a /etc/modules.conf
or /etc/conf.modules. I do have a /etc/modprobe.conf file. I am brand
new to linux and am at a serious loss. I am not even sure if this is
the module I am really supposed to be installing for sb live 24-bit.
You help would be greatly appreciated.
Sincerely and without sound,
Angela
modprobe snd-card-emu10k1
FATAL: Module snd_card_emu10k1 not found.
[root@localhost services]# modprobe snd-emu10k1
WARNING: Error inserting snd_ac97_codec
(/lib/modules/2.6.10-1.760_FC3/kernel/sound/pci/ac97/snd-ac97-codec.ko):
Unknown symbol in module, or unknown parameter (see dmesg)
WARNING: Error inserting snd_ac97_codec
(/lib/modules/2.6.10-1.760_FC3/kernel/sound/pci/ac97/snd-ac97-codec.ko):
Unknown symbol in module, or unknown parameter (see dmesg)
FATAL: Error inserting snd_emu10k1
(/lib/modules/2.6.10-1.760_FC3/kernel/sound/pci/emu10k1/snd-emu10k1.ko):
Unknown symbol in module, or unknown parameter (see dmesg)
FATAL: Error running install command for snd_emu10k1