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
this is not an offical release
http://plugin.org.uk/meterbridge/meterbridge-0.9.2.tar.gz
The release cycle has slowed down a lot, so I wanted to make sure there
aren't any subtle regressions before announcing it.
I've fixed a couple of major bugs, and reduced the CPU cost slightly, but
there are no new features.
- Steve
Brian Redfern wrote:
> Out of curiosity (being somewhat hardware illiterate), what is a
> "DFU loader"?
The "Device Firmware Upgrade Specification" describes a vendor- and
device-independent way of downloading a firmware image to a USB
device. Older USB devices from Midiman (MidiSport interfaces,
KeyStation keyboards) required a device-specific firmware loader, but
when a device implements the DFU specification (such as the Sonica or
the Ozone), it is possible to use one loader for all these devices, so
user's don't have to wait until somebody manages to write a loader for
their device. (Of course, now you have to wait for somebody to write
the DFU loader, but that's only once. :)
Regards,
Clemens
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Hi,
I want to record audio from my USB port and seek a command line or
reliable and simple GUI app. Can anyone suggest what might suit me?
I have Edirol UA-1A USB interface which is seen as /dev/dsp1. I intend
to record 80 minutes of stereo music on my laptop. The laptop suffers
overheating so I need something which uses minimal CPU activity.
Otherwise I'd be using Audacity. I'm unable to even compile this
software due to the overheating.
Debian 3.0r0/Linux 2.4.18
TIA, Tim
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hi all,
i'd like to:
* record a single chorus of "girl from ipanema" on my accoustic guitar.
* play it endlessly in a loop while i practise jazz improvisation.
* it would be nice for my computer to provide a metronome while i play
(and record that metronome over my recording), but that's not necessary
since i can use a real metronome and have it play in the background.
i went to dave phillip's site, and found a staggering number of
recorders and players.
can someone suggest a nice easy to use set of tools to do this with?
as a first stab, i tried sound-recorder, but when i play the resulting
wav file with wavp, wavp exits immediately. when i try to play the
resulting wav file with "play", it complains about premature end of a
wav file.
i assume thousands of linux using musicians are doing exactly what i
want to do. what are some good tools that i should look at?
looking for ease of use rather than features, at least for now.
thanks!
pete
--
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