Hey y'all,
I've been cleaning house after a move and getting ruthless
about recycling old technology accumulated in various boxes
that I need to have a lot less of. And in an old parted-out
Macintosh Quadra 650 I ran across an Audiomedia II board.
I remember this board, and I'm pretty sure it's still fully
functional. (The Mac is definitely not.) So I thought I'd
toss it out here. I know it's pretty old, but if it's still
usable, maybe somebody on this list could use such a thing?
I'm just gonna throw it in the recycle bin if nobody wants
it. I'll be happy to send it anywhere, if you pay shipping.
I only have the board, no audio cables, but the outputs are
all rca jacks.
Write me off-list if you want it. If there's more than one
taker, well, I'll just have myself a little drawing. ;-)
Cheers,
~Jos~
This seemed to start up, but it hung. The command used was:
/usr/bin/jackd -dalsa -dhw:0 -r44100 -p1024 -n2 -s -m -S -H
Also, I get the following message on start:
playback and capture sample rates do not match (44100 vs. 44099)
cannot tie an input port
cannot tie an input port
The first one is kind of wierd--anyone have a 44099 system :-) ??
What does the sedond one mean?
Hi,
I have been discussing the design of a drum sample
library for Hydrogen with Comix (Hydrogen author),
Bill Bailey (my studio partner and a drummer) and
Bobby Josland (drummer). I don't think any of us has
experience with designing a high quality sample
library. We've got some questions and hope to get
feedback in the form of "this is my experience" and "I
don't know but imagine...."
The objective is to create a high quality sample
library that has the potential to "sound" like a live
drummer. Hydrogen has features that will help achieve
sound and feel. From the perspective of the library
I'm only concerned with "sound."
Snare Drum
For the sake of argument, assume there are three
primary sounds; 1, 2 and 3 and that each sound has
three volume levels; low, mid and high. This means
there needs to be nine snare drum samples--the reality
is that we'll have many, many more than that. The
objective is to sequence a part that has the dynamics
of a live player--snare roll or whatever.
Are three volume/dynamic samples for every single
sound enough for a sequence to achieve or get close
enough to the dynamics of a live player? Does anyone
know how the commercial drum libraries are produced?
Maybe we need five volume/dynamic samples for every
sound.
Stereo Image
I'm interested in knowing what people think is the
best solution for achieving live sounding stereo
imaging. I've spent several days recording test sounds
and conlcuded that some instruments are probably best
treated as stereo samples. Dry (no reverb) snare drums
can potentially be mono tracks but I suspect cymbals
must be stereo.
The cymbal recording strategy will be to form a
somewhat natural semicircle and possibly use an X/Y
mic pattern. When sequencing ride cymbals they need to
bleed from the listeners Hard left side across the
stereo field and into the Hard right side. If we don't
treat cymbals as stereo images they could fail to
sound like a live player when sequenced.
Snare drums might achieve live sounding stereo imaging
from 1 in 2 out reverbs. In addition to dry mono snare
drums, we intend to produce a number of reverb
processed (wet) stereo samples.
Anyway, I'd appreciate any thoughts on these issues. I
don't expect production to begin for another two or
three weeks and that will be somewhat dictated by the
usability of Ardour. With the push for Ardour 1.0 I'm
preoccupied with testing and not using it.
Incidentally, the production will be done in our
studio which has accoustically tuned rooms and with
some high quality signal chains; Neumann -> Avalon,
and some average chains. The overheads are SM81 which
run about $600.00 U.S. a pair. They aint great but
they are good enough. Those signal paths will run
through a Soundtracs Solo consol and insert into a
Tascam Dm-24. The drums will sound great because Bill,
Bobby and I know what we're doing. If we don't know
what we're doing someone should shoot us--we've got
about 90 years of combined experienced with playing
and recording drums.
ron
__________________________________
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The latest release of Specimen, a midi controllable audio sampler for
Linux, adds 4 LFOs which can modulate volume, panning, cutoff and
resonance into mix. Additionally, the LFOs can be tempo synced to
either midi or the jack-transport mechanism.
Available for immediate leeching from www.gazuga.net, or
you can download the tarball directly here:
http://www.gazuga.net/specimen-0.3.0.tar.gz
I'm working on a roadmap for Specimen, and I'm interested in hearing
what features people are most interested in having. If there's
something you want Specimen to do, let me know; the most popular
features will get incorporated first.
[pb]
Hi Fernando,
first of all thenk you for your help!
I'm quite new to linux world, so I did not know until now that Envy24control
accepted parameters:-P.
I just runned Envy24control with -c parameter and the system correctly
recognize the three cards installed (IRQ3, IRQ5, IRQ7).
Now my problem is I can't find in Envy24control the option to set cards 1
and 2 (or hw:1 and hw:2) as slave: concerning hardware settings I can find
only a table which let me choose the clock (from 22050 to 96000, plus s/pdif
option) and "locked" and "reset" option can be activated after choosing the
sample rate.
I can't fund any other option in Envy24control which seems to help to set
other boards as slaves.
I also noticed that running BruteFir is not possible if I set S/pdif in
sample rate options, BruteFir comes out with a "broken pipe error" at its
start and exits.
Thank you very much for your help, sorry for bother you!
Michele
P.S. You are doing an incredible job at PlanetCCarma! My best wishes!
> It is not completely clear to me if you have run envy24control on the
> other two cards to set them to slave to wordclock. Envy24control can be
> directed to use a particular card with the "-c x" parameter, where x is
> 0 for the first card, 1 for the second and so on and so forth.
>
> -- Fernando
Hi Robert,
I'm here again to bother you guys on this nice mailing list: I configured
BruteFir that, as Robert supposed, can handle more than one soundcard.
I can run BruteFir, but I have a problem: when I activate more than one
soundcard by BruteFir I can hear some "POPS" coming out from my speaker.
Everything is fine activating only one soundcard.
The soundcards are properly installed and connected by internal wordclock
cable: so two cards are slaved to the master one clock (this is done by HW
settings, I cannot find any information about how to set one card as master
and others as slaves, which is required, for example, under Windows).
A CD player is connected to S/Pdif input of master soundcard, and this card
is setted up by Envy24 Control utility to generate internally a 44100Hz
clock, while others inputs are empty.
I tried to set the card to get the clock signal form s/pdif, but BruteFir
returns a "broken pipe error" with this setup.
This setting can be made only on one card by Envy24 Control utility: I do
not know if it is normal dueing to internal clockword connection between
cards.
I was wondering if "glueing" my three cards into one card as Robert
suggested could solve the "POPS" problem.
Unluckily I have no idea about how to do it, I'm quite new in Linux world
and any help would be really appreciated!
Thank you very much in advance to everyone who can help!
Michele
> > Hi guys,
> >
> > I installed Suse 9.1 with 3 Terratec EWS88D soundcars: Alsa properly
> > recognize the 3 cards installed (3 Ice1712 divers are loaded), now I'm
> > using BruteFir which require to set the Alsa's parameters for managing
with
> > multiple soundcards.
>
> Oh, sounds like a nice setup.
>
> >
> > The parameter is:
> >
> > hw:0,0
> >
> > if you have a 1 card system, now does someone know how this parameters
work
> > with 3 soundcards?
> >
> I think it is specified as: <devicetype>:<device>,<subdevice>
>
> So in your case the devicetype you wish to use is hw, the device is 0->2
and
> subdevice depends entirely if the card HAS subdevices and if you wish to
use
> them, I think it probably best to leave it at 0.
>
> so:
> hw:0,0 (first card)
> hw:1,0 (second card)
> hw:2,0 (third card)
>
> As I understand you brutefir supports natively to use several cards. If
this
> is not the case then it IS possible to glue the cards together so they
form
> one "card". This requires .asoundrc trickery that is beyond most peoples
> abilities though ;)
>
> /Robert
Anyone using this device with jack?
and Ardour?
Thanks
--
Tim Gorman
Information Systems Specialist
Petr-All Petroleum Corp.
office (315) 446-0125 x126
cell (315) 415-8108
Hi
I normally use grip for ripping my CDs to mp3, which works well. But
sometimes I'd like to be able to rip a CD without an internet
connection, which leaves me without the freedb.org server which means I
have to type all the track names myself. Is there a way to rip a CD to
mp3 and later name the files from a freedb.org lookup?
--
peace, love & harmony
Atte
http://www.atte.dk
>From: Malcolm Baldridge <linux-audio(a)paypc.com>
>Subject: Re: [linux-audio-user] Ardour, Jack, and 2.6 kernels + XRuns
> with the Audiophile 24/96 M-Audio
>
>1) You can't use more than one drive per IDE channel.
[ ... ]
>It's VERY bad on performance to mix very slow devices (such as CD-ROMs)
>with fast ones: they hold a lock on the bus during all operations,
I recently removed CD-ROM device completely. I ended up doing that
because I got disk errors. The disks are perfectly ok, however.
Some example symptoms were:
-When the CD-ROM device slided in the disk, Alsa audio play was
interrupted for 7 seconds
-File copy sometimes made errors -- such as the highest bit of one
byte of 300 MB file turned from 0 to 1
-File was temporarily corrupted -- error stayed at the same place,
no matter how many times the file was verified; after the boot,
the file was again ok; I made sure the caches were emptied, and so,
it looks like a certain bit pattern or certain address caused the
change in data
I'm still testing and waiting for errors to happen.
This problem has driven me to verify that files are copied ok,
that the created tar.bz2 packages are ok (both the tar.bz2 file
and its content). I also have set Emacs to make the numbered backup
copies and make them by moving (not by copying). Backups, flac, mp3
and ogg codings need to be verified too.
The point is this: how you would know about the rare bit changes
if you never verify the files?
In audio work such a bit changes can go unnoticed. How many of you
verify, e.g., that a simple edit to a file did not made a bit error?
I'm sure that no one verifies. Use of a dither makes the detection
quite hard.
>3) Good/correct IDE drivers for the chipset. With Linux at least, you have
>pretty good control over this: just make sure you build your kernel with the
>IDE chipset driver compiled into the kernel statically [it needs to activate
>prior to the mounting of /].
How to know what drivers there already are? How to know if the driver
in use is not the correct one? Can the default IDE drivers work for
a IDE controller with possible limitations? That is, the problem
with IDE driver could go unnoticed without an expert help.
How to drop the speed of the IDE? Reliability vs. speed.
>If not, you can set them up with:
>
>hdparm -c1 -u1 -d1 -m16 /dev/hda
I made the change, but the speed stayed the same:
% hdparm -t
/dev/hda:
Timing buffered disk reads: 64 MB in 2.23 seconds = 28.70 MB/sec
Regards,
Juhana
To play an Ogg Vorbis stream through a firewall requiring authentication
try :
wget --proxy-user="myusername" --proxy-passwd=mypassword --ignore-length
-O - http://ogg.smgradio.com/vc32.ogg | ogg123 -d arts -
Replace "-d arts" with the appropriate system on your box - see man ogg123
Replace "http://ogg.smgradio.com/vc32.ogg" (Virgin Classic) with your
favourite.
Enjoy!
Pete.
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