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/
Hello all (especially UK and Europe based list members),
There is a trade show called Sounds Expo 2004 taking place at Wembley,
near London, on the 10th - 12th February next year. The exhibitors
include most of the major studio technology companies, and the media
sponsor is Sound on Sound.
Would members of this list be interested in doing a Linux
Expo/LinuxTag sort of stand at this event, showing off JACK,
Rosegarden, Ardour, JAMin, Audacity etc etc? In some ways it makes
sense to target musicians, engineers and producers rather than just
general Linux users. I think it could also be useful in raising
awareness among the hardware companies that will be there.
There is no community area at this event that I can see, but there are
some free seminars being hosted by SoS, a Steinberg/Native
Instruments theatre and 5.1 workshops. It's just possible that the
organisers may allow a free stand on the basis that it's a not for
profit (dis)organisation offering free advice to event visitors.
What do you lot reckon? If there are enough people who would be
willing to be on a stand to cover the three days, then I will
approach the event organisers.
Daniel
Hi,
I'm using the following:
IBM T40 laptop
Digigram VXPocket v2
RedHat 9
Kernel 2.4.22-ac4
ALSA drivers 0.9.8
vxloader 0.9.7-i686
Measurement gear
Audio Precision System Two Cascade
PrismSound DSA-1
Some of my other gear is having difficulties locking to the S/PDIF output
of the card at 44.1 kHz and I have a suspicion as to why this is
happening...
I have control over the sampling rate of the S/PDIF output of the card as I
would expect, however...
In the S/PDIF Channel Status Block information there is a portion of Byte 3
that is used to send the intended sampling rate (selectable between three
values, 32 kHz, 44.1 kHz and 48 kHz, depending on the arrangement of bits
0-3 in Byte 3. For specific information on this, please see Application
Note AN22REV2 from www.crystal.com). It is also possible, using bit 0 of
Byte 0, to indicate whether the signal is Professional or Consumer format.
Using my present configuration, I do not appear to have control over either
of these two variables. For example, even when I am sending a 44.1 kHz
signal at a rate of 44.1 kHz (or close enough... it's actually 44.0993, -17
ppm) Byte 3 of the Channel Status Block is saying that the signal is a 48
kHz transmission.
Are the VXPocket drivers in ALSA capable of changing these values in the
card? The official drivers for Mac OSX allow me to change the Pro/Consumer
flag using an application, and the sampling rate indicator changes
automatically with the actual sampling rate of the card's output.
Cheers
-geoff
____________________________________________________
Geoff Martin Ph.D.
Tonmeister, Bang & Olufsen a/s
email: ggm(a)bang-olufsen.dk
web: www.bang-olufsen.com
web: www.tonmeister.ca
phone: +45 96 84 49 54
Hi all!
I've been trying to stretch a soundfile. It is a piano track with mostly two
voices only. I did it with an fft plain phase vocoder. Now there are a lot of
parameters and I got some undesireable delay. In general the stretched sound
was good, but the delay is disturbing. So here's the question, how should I
set the following parameters to get the best result?
fft-length (as high as possible?), windows per second, window length,
windowtype (hamming, triangular, kasier with several coeffecients).
Any ideas or practical experience with that? - I'm NOT doing it in realtime,
so CPU-time is no problem. I've got time! :-)
Thanbks for any help or good ideas!
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
[ someone ]
>As someone who runs a business, why would I want to pay someone $600 to
>fix 10 documents when I can buy Microsoft's tools for $300 and have
>guaranteed compatibility? That's a tough sell...
Hello. We should have a law which says the file formats should be
open formats. People who write and make documents should have a vendor
independent access to the documents.
At meanwhile, why one should be able to read Word and Excel documents
in Linux? One can always ask clients to print to the good old paper
or to an image file.
Regards,
Juhana