I run the emu 1212m successfully on 64 bit linux. Some of the subcards (
hw0.1 etc) have problems with 44.1, I use 48 so it's not an issue for me.
Make sure your sync is set correctly in alsamixer (adat,spdif,int44) and
play with using different settings for hardware.. hw0,1, 0,0 0,3 etc.. hw:0
will *not* work for recording.
the card support is problematic and I don't recommend it. I had to patch
1.15 to get it working correctly and for the life of me I cannot find where
I got the patches. The original author of the patch hasn't done any major
work on it for over a year. Check the alsa dev wiki page on emu10k/emu on
the discussion page, thats as much documentation as anyone has collected on
the card.
taken from:
http://www.alsa-project.org/main/index.php/Matrix_Talk:Module-emu10k1
MajorBytes
majorbytes.com -- >> music site
http://myspace.com/allenwarrenmusic
--
host-69-146-29-10.static.bresnan.net (2007-05-29 18:36:16)
You use jack, right? Then you have to set which alsa device will be
used for capture and playback (the simplest way is to use qjackctl for
that) .
There are several possibilities on e-mu. Let's say e-mu is the only
card in your system, so it has index 0.
Then hw:0,0 is stereo, 16bit only. Do not use it for capture.
However, it's probably the best device for playback for now,
because multichannel playback is currently broken. hw:0,2 is
multichannel capture capable of 8 channles at 24 bit resolution.
Always use that for capture. hw:0,3 is multichannel playback of 16
channels at 16bit. It does not work reliably a need to be reworked (24
bit, proper sync at 44kHz, etc..).
So most reliable setup for e-mu cards is hw:0,2 for capture and
hw:0,0 for playback (you can use them simultaneously).
Now about the routing. There are a lot of ports called "DSP <number>"
it he mixer. At first look it's rather confusing.
Actually, there are two groups od "DSP" ports. One group is for input
(DSP 0 to DSP 15, hexadecimal numbering! ) and it's the first bunch
of "DSP" from left in alsamixer (bottom line). The other group is for
playback (DSP 0 to DSP 31, decimal numbering, OMG!!!).
First 8 inputs (DSP 0 - DSP 7) are used for capture (you will see and
hear them in the jack). Other inputs are mapped directly
onto their output brothers with same number (don't forget, that inputs
are hexadecimal, so DSP A is mapped to DSP 10, DSP 12 to DSP 18,
etc..).
Otput works same, so if you use hw:0,0 (2 channels) for playback, then
you will hear them on outputs DSP 0 and DSP 1.
this isn't a card for a beginner, but with coaxing it does work.
best
bradley newton haug
On Jan 28, 2008 4:29 PM, Darren Landrum <darren.landrum(a)sbcglobal.net>
wrote:
Darren Landrum wrote:
Norval Watson wrote:
> I got the same processor running 64-bit debian sid running jack at
44.1khz, 64
sample buffer on Audiophile 2496 soundcard with no xruns.
> Are you running realtime kernel with realtime
box checked in
qjackctl/settings
> (or with -R option on command line)?
> (I got 4GB dual-channel RAM and 2 SATA DRIVES...)
> My kernel: 2.6.23.11-rt14 #1 SMP PREEMPT RT Wed Dec 26 08:47:49 EST
2007
x86_64 GNU/Linux
Norv
Checking realtime makes no difference. Still 50-60 xruns a second at 128
samples, and if I go down to 64 samples, it crashes.
You know, I just noticed something. The xruns as they are reported are
exact multiples of 53. The first is 53, then 106, then 159, etc. The
qjackctl app looks like it updates the display once a second, so it's
not 50-60 like my previously lazy reporting said, it's exactly 53. So...
What's happening 53 times a second?
-- Darren
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