Paul Winkler <pw_lists(a)slinkp.com> wrote:
>On Tue, Feb 25, 2003 at 01:34:57PM -0500, linuxmedia4(a)netscape.net wrote:
>> ...they taught us in the recording classes... it's important to put
>> the mics in a "cross pattern" (like an X).
>well, sometimes...
>> The heads of the mics need
>> to be 45 degrees apart from each other. If you don't then you can get
>> "phase cancellations".
>I never went to recording classes, but ...
>Phase cancellations have at least as much to do with mic position
>as with mic angle. angling the mics is not a magic solution.
>Most cardioid mics have plenty of pickup at 90 degrees.
>It sounds like what you're describing is the classic "X/Y" stereo
>mic technique, with a pair of cardioids placed so that
>* The mics are angled 90 degrees relative to each other, and
> 45 degrees relative to "center"
>* The mic heads are placed as close together as possible.
After reading several of the responces to my original email, I'm realizing that I may be confusing mic angling with preventing phase cancalations when the (mic angling) part of the class may have just been about micing techniques. And you make a good point. Different mics have different pick-up patterns. So even (if) angling does (in some cases) prevent phase cancellations, it's not going to if the mic has a wide pick-up pattern. At this point, I don't even remember if angling has _anything_ to do with preventing phase cancellations.
>check the rec.audio.pro FAQ for more good info on stereo micing.
Since it's been 15 years since I took the recording class, It may be time to brush up on recording tecniques again.
>> ...If you don't do the cross pattern you (could) hear unwanted sounds
>> (and/or dropouts).
>dropouts?? what on earth does mic technique have to do with dropouts?
I (thought) the teacher was saying that phase cancellations can also cause dropouts. But hey, 15-year-old information can get blurred with other info in the mind. I've been busy for the last 10 years Busking on the streets of the various Cities that I've traveled to. I haven't used any of the tecniques that I learned. So I'm sure I've either forgot certain details, or have blurred different discussions in my mind.
Rocco
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Hello,
I just released polarbear. I had the code lying around, and just merged
it with the jack/alsa i/o code of tapiir. Note that this is the first
public release. I did not test it thoroughly, and I am not sure if the
GUI is obvious enough (it should be if you are familiar with complex
filters), so any input is welcome.
polarbear is a tool for designing filters in the complex domain. Filters
can be designed by placing any number of poles and zeros on the z plane.
>From this the filter coefficients are calculated, and the filter can be
applied in real time on an audio stream.
polarbear can be found at
http://www.iua.upf.es/~mdeboer/projects/polarbear/
For the (far) future, the idea is that polarbear and tapiir can work
together, in the sense that the filter coefs calculated by polarbear can
be used to control the gains of tapiir. maybe polarbear and tapiir might
even merge. that would be some animal :-)
Maarten
Hi everyone...
This may sound like a stupid question to a bunch of pros like you... but...
I am trying something simple. Soon, I will be building a system for home
recording... but for now...something simple.
Can anyone tell me how I might get my soundblaster live (emlu10k1) to duplex???
I have audacity installed, and krecord, neither will record anything! I
hear my voice amplified throught he speakers, but I get nothing.
I am using OSS drivers, both the ones from the 2.4.19 kernel source and the
ones from sourceforge (most current). No dice. I tried the ALSA
drivers... but that was severl "tries" ago....
I know this isn't hard.. but I can't find any help on this anywhere! Last
time I tried this salvation lay in the "full duplex" setting on kde... but
no dice this time...
Thanks in advance....
Chris
Hello,
I'm a bit of a newbie (okay, REALLY newbie) when it comes to working with
audio, but I am a strong linux user.
Basically, I have the following:
2 - instrument microphone (XLR)
1 - vocal microphone (XLR)
1 - keyboard w/ midi interface
I want to do some multitrack recording, using anywhere fmro 2-4 of the
above items at once. I am getting ready to start building a dedicated box
for this. So, what kind of hardware should I be looking at? I'm looking
for something somewhat inexpensive (RME Hammerfall was about $600, Midiman
delta 1010 LT was closer to $700, both too much).
Also, should I be looking at any additional items like a mixing board,
amplifiers and such? I would greatly appreciate any ideas, thoughts,
hints, pointers, links!
Thank you in advance,
Bryan
P.S. I know it seems like a lot for a "newbie", but I just like doing
things right the first time.
Just FYI, American Musical Supply has a (1) dent and scratch ST-Audio
C-Port DSP 2000 for $380 US. Usually $450 US. I wish that had been
there when I bought mine. I'm really happy with it, works well with
Ardour.
Jan
"Darren Landrum" <consul(a)studioconsul.net> wrote:
> I would like to reiterate how much I like the sound you got form the
> drums. It sounds so natural, and quite perfect for the type of music
> you're doing.
>
>-- Darren
>> Jan \Evil Twin\ Depner
>> Anthony,
>>
>> I used two cheap (Marshall) overhead condenser mics for the
>> drums (about 2 meters up and slighty back of the
>> drummer) and a Shure SM57 on the kick.
One thing I wanted to point out (that they taught us in the recording
classes I took) was that in the (above) setup, it's important to put
the mics in a "cross pattern" (like an X). The heads of the mics need
to be 45 degrees apart from each other. If you don't then you can get
"phase cancellations". It has to do with the the amount of time it
takes sound to hit one mic and then the other. If the mics are _not_ at
a 45 degree position, then they can be close enough to cause one mic
to pick up the sound source and the other one to pick up the _same_
sound source (just milliseconds later). And when you have the same exact
sound being offsetted by only milliseconds, you get "phasing". In fact,
that exactly how you get the (guitar) effect they call "phasing".... by taking a sound and offsetting the same sound by just a few milliseconds.
I think (if my memory serves me well) that the 45 degree angle isolates each mic to the point were it can't pick up the same sound in each mic.
Actually, Jan may have some kind of knack for this thing, because we were taught that (the tecnique jan used) was the way to go if you only have 3 mics - (two overhead and one for the kick drum). In fact, I think Jan's choice to use condencers for overhead and sm57 on the kick is a good choice. But please trust me. If you don't do the cross pattern you (could) hear unwanted sounds (and/or dropouts).
While I'm at it... that whole recording sounds great.
Rocco
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Hi John,
Thanks, but I covered that.
Any other suggestions?
Chris
At 07:22 PM 2/24/2003 -0500, you wrote:
>On Mon, 24 Feb 2003, Chris wrote:
> > Subject: [linux-audio-user] Full Duplex woes
> >
> > Hi everyone...
> >
> > This may sound like a stupid question to a bunch of pros like you... but...
> >
> > I am trying something simple. Soon, I will be building a system for home
> > recording... but for now...something simple.
> >
> > Can anyone tell me how I might get my soundblaster live (emlu10k1) to
> duplex???
> >
> > I have audacity installed, and krecord, neither will record anything! I
> > hear my voice amplified throught he speakers, but I get nothing.
> >
> > I am using OSS drivers, both the ones from the 2.4.19 kernel source and
> the
> > ones from sourceforge (most current). No dice. I tried the ALSA
> > drivers... but that was severl "tries" ago....
> >
> > I know this isn't hard.. but I can't find any help on this anywhere! Last
> > time I tried this salvation lay in the "full duplex" setting on kde... but
> > no dice this time...
> >
> >
> > Thanks in advance....
> >
> > Chris
> >
>
>If you're using kmix (sounds like you're using KDE) make sure you have the
>correct input marked as the recording device - the red LEDs along the
>bottomr determine which input to record from - make sure you have it set
>to 'mic' or 'line' whichever you're using.
>
>HTH - JB
>
># John Bleichert
># http://vonbek.dhs.org/latest.jpg
Hello All
If/when I get my Delta 44, how are all the inputs and outputs mapped
into/out of the system? /dev/dsp0 - /dev/dsp7? How exactly does this work?
Does the driver for these cards provide any routing capabilities between
the inputs and outputs?
Any info appreciated.
Thanks - JB
# John Bleichert
# http://vonbek.dhs.org/latest.jpg
Kai,
Sorry. I forgot about ecasound. I haven't used it myself but from what I've heard it's the most stable multitracker around.
Jan
-----Original Message-----
From: "linux-audio-user-admin(a)music.columbia.edu" <linux-audio-user-admin(a)music.columbia.edu> on behalf of "Kai Vehmanen" <kai.vehmanen(a)wakkanet.fi>
Sent: Tue, 25 Feb 2003 12:53:07 +0200 (EET)
To: "Audio-User" <linux-audio-user(a)music.columbia.edu>
Subject: Re: [linux-audio-user] Full Duplex woes
On 24 Feb 2003, Jan "Evil Twin" Depner wrote:
> I am assuming that this means Linux. I have only done multitrack
> recording with SLab and Ardour. If you're going to just record two
> tracks of 16 bit 44.1KHz at once (while playing other tracks) SLab is
> pretty good (it actually has some very nice effects built in). If you
> want to handle more tracks or more resolution you'll have to use Ardour
> or something else. Ardour is *not* trivial to set up (check my web page
> http://myweb.cableone.net/eviltwin69/ALSA_JACK_ARDOUR.html).
Don't forget ecasound (w/ ALSA):
ecasound -a:mon -i drums.wav -o alsa,default \
-a:rec1,rec2 -f:16,2,44100 -i alsa,default \
-a:rec1 -f:16,1,44100 -o first_rec_channel.wav \
-a:rec2 -erc:2,1 -f:16,1,44100 -o second_rec_channel.wav
... with OSS you just replace 'alsa,default' with '/dev/dep' and it'll
work (assuming your OSS drivers support full-duplex).
... with JACK it's even easier:
ecasound -a:mon -f:16,1,44100 -i drums.wav -o jack \
-a:rec1 -i jack -o first_rec_channel.wav \
-a:rec2 -i jack -o second_rec_channel.wav
... you'll have to use qjackconnect or jack_connect to connect the port to
the soundcard ports.
.. more examples at:
http://www.wakkanet.fi/~kaiv/ecasound/Documentation/examples.html
And ecasound *is* trivial to set up. :)
--
http://www.eca.cx
Audio software for Linux!