hello everyone,
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
and 72:1.
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 ?
cheers
rob
_______________________________________________________
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.
Greetings:
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 ?
Best regards,
== Dave Phillips
The Book Of Linux Music & Sound at http://www.nostarch.com/lms.htm
The Linux Soundapps Site at http://linux-sound.org
Hi,
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
http://www.alsa-project.org/alsa-doc/doc-php/template.php3?company=Midiman&…
all modules seem to load fine.
I have the following questions:
- mixer
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,
Akos
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
chime in!
-Joe
--
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.
Best regards,
Juhana
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.
something like:
/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
devices?
BTW, the contents of this asound.conf file is:
pcm.ice1712 {
type hw
card 0
device 0
}
# adcdac 1&2
pcm.channel1 {
type plug
ttable.0.0 1
ttable.1.1 1
slave.pcm ice1712
}
# adcdac 3&4
pcm.channel2 {
type plug
ttable.0.2 1
ttable.1.3 1
slave.pcm ice1712
}
#adcdac 5&6
pcm.channel3 {
type plug
ttable.0.4 1
ttable.1.5 1
slave.pcm ice1712
}
# adcdac 7&8
pcm.channel4 {
type plug
ttable.0.6 1
ttable.1.7 1
slave.pcm ice1712
}
#SPDIF channels only
pcm.ice1712_spdif {
type plug
ttable.0.8 1
ttable.1.9 1
slave.pcm ice1712
}
Hi,
Can someone show me an example asoundrc for the delta 66 I can't get
mine working (everything works fine using oss emulation so I'm pretty
sure this is were the problem lies at the moment mine looks like this.
pcm.via {
type hw
card 0
}
ctl.via {
type hw
card 0
}
pcm.ice1712 {
type hw
card 1
}
ctl.ice1712 {
type hw
card 1
}
And if I try using it with aplay -D ice1712 it complains thusly:
aplay: set_params:805: Sample format non available
--
rob <mailingLists(a)pangolin.org.uk>
I'm still playing around and trying to get my Delta 1010LT to work, so I
can record from the different input channels.
I'm now simply trying to record using arecord, but I can't seem to
record anything but silence. the specific problems I get:
- I can't recrod from the hardware device as configured /etc/asound.conf
I try:
$ arecord -f cd -d 5 -D hw test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
arecord: set_params:805: Sample format non available
or
$ arecord -f cd -d 5 -D ice1712 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
arecord: set_params:805: Sample format non available
where
$ cat /etc/asound.conf
pcm.ice1712 {
type hw
card 0
}
ctl.ice1712 {
type hw
card 0
}
is the config that was hinted at the 1010LT ALSA driver installation page.
BTW,
$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: 1010LT [M Audio Delta 1010LT], device 0: ICE1712 multi [ICE1712
multi]
Subdevices: 1/1
Subdevice #0: subdevice #0
anyways, what does it mean by "Sample format non available", as the
sound card supports this format? also, how do I list the available formats?
- I can record from the 'default' device, but only silence
I can succesfully run:
$ arecord -f cd -d 5 -D default test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
but, it only records silence, even though all my channels are _not_
muted, and I have input running in to the sound card at hardware inputs
3 and 4. I can see this from envy24control. I also experimented with
default:0,1 ... 0,3 and default:1,0 ... 3,0, but the same occurs.
what am I doing wrong?
Akos