> From: hayesjaj(a)notes.udayton.edu [mailto:hayesjaj@notes.udayton.edu]
>
> Have you managed to find a solution to this problem? I am
> having the same
> issues.
>
> Thanks,
> James
Most of my problems were fixed with the addition of three new features:
1) Jack knows to use controller 0 when accessing device 1
2) Jack makes a 16 bit connection when the connected interface doesn't support 32 bit
3) The ability to tell Jack how many ports to make available.
What kind of problems are you having?
regards
-Reuben
Hi,
I posted on this list a while ago, but still couldn't resolve one of my
problems. I have a Dela 1010LT card, which works nice with the envy24
ALSA driver. What I want to achieve is to record from this soundcard
through the OSS interface. I've been experimenting with /etc/asound.conf
and the aoss utility, but to no avail.
Basically what I want to achieve is to have a two channel /dev/dsp
device, which corresponds to adc 3&4 on the Delta card. in
/etc/asound.conf I have:
# adcdac 3&4
pcm.channel2 {
type plug
ttable.0.2 1
ttable.0.3 1
slave.pcm ice1712
}
which works nice when I use arecord -D channel2 ...
now, what I want to work is something following:
aoss sox -t ossdsp -r 44100 -c 2 -w /dev/dsp0 -t wav tmp.wav
but no matter how I try to generate a pcm.dsp0 section in
/etc/asound.conf, what I record is total silence :(
as far as I understand, I need a pcm.dsp0 section, which would describe
the /dev/dsp0 features. what I tried, among others is:
pcm.dsp0 {
type plug
slave.pcm "channel2"
}
but this doesn't work. (it may be a problem that there is actually an
OSS-driven /dev/dsp0 in the system. but setting pcm.dsp1 didn't work, it
gave me non-existent device errors) can someone give me hints on setting
up a proper pcm.dsp0 section?
Hi all,
the following announcement gets sent to all of linux-audio-dev,
linux-audio-user and linux-audio-announce mailing lists in order to
reach as many possible interested parties as possible; I'm sorry if you
receive this twice or even more often.
I would be glad if we get a lot of participation from your side!
Frank
-----------------------------------------------------------------------
>From April 29th to May 2nd, 2004, the Institute for Music and Acoustics
of ZKM Karlsruhe, Germany, will host the 2nd conference of the
Linux Audio Developers (LAD). As a new feature there will be
presentations of music in addition to technical talks. For this, we
are looking for music that has been produced completely or mostly
under Linux.
We are looking for:
* Interesting demos of sound synthesis, sound processing, etc.
* "Classical" computer music compositions, to be played in a concert
setting
* Pieces from areas such as Electronica, Chill-Out, Ambient etc.
If you would like to participate, please send your composition(s)
to this address:
Linux Sound Night
ZKM, Institut fuer Musik und Akustik
Lorenzstr. 19
D-76135 Karlsruhe
Germany
Please make use of one of the following media formats:
- Audio-CD, DVD or CD-ROM
Possible audio file formats: aiff or wav; mono, stereo or multi-channel;
44.1 or 48 kHz; 16 or 24 bit resolution.
Please include the following items with your submission (in English):
* A short commentary on the compositions
* A short Curriculum Vitae
* A completed and signed printout of the form available here:
http://www.zkm.de/lad
Deadline for submissions is February 29th, 2004.
A jury will select the compositions that will be performed/played.
The jury will award 3 grants to participants to contribute to their travel
expenses.
Terms and conditions for participation can be found in the form above.
Up-to-date information about the conference is available here:
http://www.zkm.de/lad
lau(a)hippie-online.de wrote:
>> I would like to record directly from my DAT recorder (48 kHz) via SPDIF.
>> As a greenhorn in harddisk recording I expect that there should be a way
>> to get an exact copy of the data on the tape with no input level / mixer
>> and no DA-AD conversion in between?!
>>
>> Unfortunately so far I did not succeed in recording from the SPDIF-input
>> at all. I am just able to route the DAT signal to the analog line out by
>> setting its volume using alsamixer. But the SPDIF-in does not appear in
>> other mixers like kmix and the signal is not available in recording
>> programs like audacity or krecord. :-(
>>
>> I am using:
>>
>> - SuSE 8.2 / kernel 2.4.20 / i86
>> - alsa 0.9.0
>> - emu10k1
>> - Soundblaster live! rev. 4
Joern Nettingsmeier wrote:
> i don't have an spdif source, so i can't test, but it used to work on my
> sblive when i set the capture flag in alsamixer on "IEC958 Coaxial"
> *and* the "capture" channel.
Yes, you are right, this works (and is not possible using alsamixergui)
- thank you! *But* it definitely results in a DA-AD conversion because
- I can record in arbitrary sampling rates without problems
- When I stop the tape the input level remains about -70 dB
Is this a particular weakness of the SB live? What card supports exact
recording from SPDIF-in?
Ciao,
HippiE
Hi all,
Should have mentioned this a while ago, but I have a new release availible here:
http://www.retinascan.de/ (nebogeo - 0 CD-R - including artwork by cristiana
yambo)
All done in linux of course, mostly with spiralsynth modular, a bit of spiral
loops and mastered with audacity, ecasound, sweep - and not to forget a ton of
LADSPA plugins :)
cheers!
dave
................................. www.pawfal.org/nebogeo
Hi everyone,
Curious about the use of this PDAudio-CF card under laptops. Dave
Phillips is the website's "poster boy" for this, and I thought maybe he
or somebody else that has checked it out could post an evaluation for it
here. I am considering it as the Hi-res soundcard for a Pismo Powerbook
running Gentoo, primarily because it seems cheaper than getting another
cardbus HDSP...
Thanks,
D.
--
derek holzer ::: http://www.umatic.nl
---Oblique Strategy # 57:
"Do the words need changing?"
Hi all!
I'm still trying to install a good working festival system. But still I'm
not sure about the components needed and in which order to install the items.
Can anyone help me with a bit of practical advise? Like: which tts-app to
choose, which languages are good (english and/or german)...
Does anyone know, if mbrola voices work together with festival? I read
something about it, but again I'm not sure.
Thanks for any help and advise!
Kindest regards
Julien
Julien Patrick Claassen
jclaassen(a)gmx.de
julien(a)c-lab.de
http://www.geocities.com/jjs_home
SBS C-LAB
Fuerstenallee 11
33102 Paderborn
Phone: (+49) 5251 60 6060
Fax: (+49) 5251 60 6065
www.c-lab.de
Hi
Horgand ... is a organ, jack capable who generates sound with a FM based
synthesizer, also provides DSP effects and a small programable accompaniment
in wave table.
Requires:
FLTK
ALSA
JACK
LIBSNDFILE
NEWS on 1.03
----------------------
- Solved small bugs.
- Solved bug now horgand read the config rhthm file from the installed dir.
- Rewrited GNU-Autotools scripts, now better.
- Rewrited alsa detection for compatibility with alsa 1.0 pre1.
- Rewrited jack support for compatibility with jack 0.80.
horgand is available in:
http://personal.telefonica.terra.es/web/soudfontcombi/http://www.telefonica.net/web/soudfontcombi/