I'm planning on building a new machine and would like some advice. Right
now I think the main application of it's "power" will be in
recording/mixing in linux. People have been saying that a dual processor
is something to consider, but there's currently a problem: the amd
athlon's that would be at my price point (the 2600 - 2800's) have no
dual processor motherboard support, and it looks like there won't be any
b/c companies are just going to go straight to supporting the
new 64-bit opterons (which aren't in my price range). So if anyone can
comment on any of the following things, it would really help me out.
Front Side Bus speed: how important is this for recording? i could just
get two bargain athlon's with a slower FSB, would that work?
Would any of the sound apps out there, or even linux in general, make
any use of a 64 bit opteron anytime soon? (no i won't have more than 4GB
Hyperthreading - the new fancy P4's have it. Does it do anything on
linux? I saw some benchmarks where it really sped up video encoding (on
windows), how similar to sound processing is this?
In general, intel chips seem to do better in benchmarks on floating
point stuff (games and video) while amd's do better on integer heavy
apps (office software). I would think that sound stuff would therefore
run better on intel's but lots of sound people say they prefer amd's.
Any reason for this?
How fast does a system really need to be before it can handle recording
with practically no limits? (let's say fewer than 10 tracks at a time
such as with a delta1010)
thanks for any help, i'd be happy if i got responses to only a few of
I know that the RME HDSP card is probably what Oeyvind is looking for,
but I thought that perhaps some users might care to share their
experiences with it. Please cc any replies to him as well the list,
I have a Delta 1010LT audio card.
I have ALSA version 0.9.4 installed on a RedHat Linux 9.
I want to record multiple, unrelated audio streams that overlap in time
but do not necessarily stop and start at the same time. Imagine radios,
each tuned to a different station, plugged in to each audio input on the
I have an existing application, which uses OSS interfaces, which will do
what I want if I can just configure ALSA to create a different emulated
OSS sound device for each input channel on the sound card. (Or, perhaps
for stereo pairs of input channels). So, is there a way to configure
ALSA to do this, and if so, what is it?
I'm wondering if its possible to manipulate my sequencer clients using .asoundrc ?
I have my laptop with its cs46xx card which always gets installed as sequencer
client 64:0. When I plug in my midiman 2x2 it gets registered as clients 72:0
basically, I want to either place the midiman 2x2 as client 64:0 and 64:1 on
start up, or somehow use parameters in the modules.conf file to not load the
cs46xx's sequencer portion (since my laptop has no midi out).
I see there are some sequencer related things in the .asoundrc file, but I
really can't find what they do. Any suggestions ?
Sent through e-mol. E-mail, Anywhere, Anytime. http://www.e-mol.com
>Hyperthreading - the new fancy P4's have it. Does it do anything on
>linux? I saw some benchmarks where it really sped up video encoding (on
>windows), how similar to sound processing is this?
Alan Cox says HT provides 0-30% speedup.
I'm troubleshooting some problems with the Hydrogen rhythm composer,
and I wondered if anyone else on the list has been using the program.
For some reason yet unclear my saved drum kits are completely garbled
and crash Hydrogen when I try to load them. Has anyone else built and
saved their own drum kits in Hydrogen ?
== Dave Phillips
The Book Of Linux Music & Sound at http://www.nostarch.com/lms.htm
The Linux Soundapps Site at http://linux-sound.org
I just installed a Delta 1010LT multi-channel audio card into my PC. I'm
new to multi channel cards, and have some questions. I'm sorry if these
questions are dull or simple.
I just installed alsa 0.9.4, as described on the page
all modules seem to load fine.
I have the following questions:
I can also run alsamixer and env24control, but I can't run aumix or
other mixers that would use the OSS mixer interface. maybe the alsa ->
OSS mixer interface is not configured properly, as:
# cat /proc/asound/card0/oss_mixer
VOLUME "" 0
BASS "" 0
TREBLE "" 0
SYNTH "" 0
PCM "" 0
SPEAKER "" 0
LINE "" 0
MIC "" 0
CD "" 0
IMIX "" 0
ALTPCM "" 0
RECLEV "" 0
IGAIN "" 0
OGAIN "" 0
LINE1 "" 0
LINE2 "" 0
LINE3 "" 0
DIGITAL1 "" 0
DIGITAL2 "" 0
DIGITAL3 "" 0
PHONEIN "" 0
PHONEOUT "" 0
VIDEO "" 0
RADIO "" 0
MONITOR "" 0
- accessing the multiple input channels as separate OSS devices
actually the main goal of having this card is to be able to record
parallelly from the separate input channels it has. recording would be
done through opening and reading OSS-style /dev/dsp devices.
is this possible using this card and alsa drivers? if so, how? is this
related to /etc/asound.conf ?
all help would be appreciated,
This may be off your subject a bit, but I can confirm that multichannel
recording works with the Delta 1010LT card with Ardour.
If you run Jack, by the way, the jack patch bay will show all 10 inputs
and outputs on the 1010.
Again, this is probably not exactly what you want, but I thought I'd
Joe Dell'Orfano <fullgo(a)dellorfano.net>
Hello. I would like to have a look at full set of PDF manuals
for the following software. They comes for my personal use only
as I'm collecting manuals of audio and graphics software and
as I'm helping to make free, open source software better.
Cakewalk / Sonar 1.0
Emagic / Logic Audio Platinum 6.0 + effect plugins docs
Steinberg / Nuendo 2.0
I have already asked Emagic for their manual but the manuals
were available for the product owners, which is sad if somebody
like me wants document the audio and graphics software history.
In my quest to record from the different channels of the Delta 1010LT
have come so far that I can address the different hardware inputs using
ALSA device names, while using an appropriate /etc/asound.conf file.
e.g. I can
arecord -f cd -d 5 -D channel2 test.wav
and this would record from hardware input channel #2. I have four stereo
input channels, e.g channel1 ... channel4, plus the spdif channel. So
far, so good.
Now I would need a way to map these channels to OSS /dev/dsp interfaces.
/dev/dsp1 -> channel1
/dev/dsp2 -> channel2
/dev/dsp3 -> channel3
/dev/dsp4 -> channel4
I understand that I would need to use the kernel module snd-pcm-oss to
achieve this. how can I tell this module to map to the appropriate ALSA
BTW, the contents of this asound.conf file is:
# adcdac 1&2
# adcdac 3&4
# adcdac 7&8
#SPDIF channels only