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Hello folks,
I recently have taken the effort installing ffado from SVN on Ubuntu
Studio. My Presonus works quite OKayish with freeBoB but I cannot set
the sync to external via S/PDIF - unfortunately it looks, like ffado
does not offer any method to set this either. As I try to start the
ffadomixer it says, my device is not supported :-(
Any hints?
And: does any body know, what devices are supported by ffadomixer ?
best regards
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dgarcia(a)iua.upf.edu wrote:
> We are jubilous to announce CLAM 1.2 'GSoCket plugged-in release'.
> We had to wait for some months to make this release as we had to
> redeploy the multiplatform release infrastructure [1]. Thus, the
> feature buffer for this release is pretty full. It incorporates both,
> the results of the Summer of Code [2] students work and the
> involvement of David and Pau with Barcelona Media Foundation Audio
> Research Lab[3].
>
> We want to thank the involvement of GSoC students Hernan Hordiales[4],
> Bennet Kolasinsky[5], Greg Kellum[6], Andreas Calvo, Roman Goj[7] and
> Abe Kazemzadeh, Google Inc, and Barcelona Media audio lab members
> for their precious involvement in CLAM.
>
> [1] http://clam.iua.upf.edu/testfarm/
> [2] http://clam.iua.upf.edu/wikis/clam/index.php/GSoC_2007
> [3] http://www.barcelonamedia.org/index.php/linies/10/en
> [4] http://h.ordia.com.ar
> [5] http://bennettdoesclam.blogspot.com
> [6] http://gregkellum.com
> [7] http://ro-baczek.blogspot.com
>
> A summarized list of changes follows. See also the CHANGES files[8]
> for details, or the development screenshots[9] for a visual guided tour.
> As usual binary packages for Windows, MacOSX and several flavors of Linux
> are available to download.
>
> [8] http://iua-share.upf.edu/svn/clam/trunk/CLAM/CHANGES
> [9] http://clam.iua.upf.edu/wikis/clam/index.php/Development_screenshots
>
>
> Summary of changes:
>
> The most exciting feature is the new plugin system (acalvo)
> which enables third party algorithms to be distributed separately
> from the core binaries. LADSPA plugins support has been enhanced
> and a first iteration on FAUST[10] integration. The wiki[11] contains
> very nice how-to's that cover most of that.
>
> [10] http://faust.grame.fr/
> [11] http://clam.iua.upf.edu/wikis/clam
>
> Most of the GSoC work come as plugins: a SMS Synthesizer (gkellum),
> a Voice synthesis/analysis (akazem) and some some cool guitar effects
> (hordia). Also not included as plugins but in the main repository
> several enhancements have been done on the SMS transformations (hordia)
> and the tonal analysis (rgoj).
>
> Some interesting work has been done on the Barcelona Media Audio Lab
> on having a system to simulate 3D room acoustics which can be reproduced
> on several exhibition systems. Some precomputed room databases are
> available to try. Check the wiki NetworkEditor Tutorial for more
> information.
>
> Regarding the applications, Network Editor incorporates new usability
> enhancements, a new on-line Tutorial and a new Spectrogram like view.
> The Annotator received Bennet Kolasinsky attention improving its the
> flexibility of its interface, the practical effects are multiple
> segmentation and low-level descriptors panes and that we are pretty
> close to visualization and auralization plugins.
>
> Enjoy.
> The CLAM Team
>
>
>
yes, i am really enjoying the new release!
compiles good and runs much more stable on my debian-testing.
please, keep up the good work!
cheers,
doc
LAC 2008: bandwidth to burn and volunteers needed
The 6th annual Linux Audio Conference is taking place in Cologne, Germany, Feb
28th to March 2nd, 2008. As with each previous year this year's conference will
be streamed live over the internet in ogg theora via icecast. The stream server
is up at: http://lac2008.khm.de:8000/
There is nothing to see at the moment, but keep checking over the coming days
as we hope to have a test stream up soon.
This year we are in the unique situation of having a Gigabit link donated by
CITIZENMEDIA: http://www.ist-citizenmedia.org/ They have asked us to use up as
much of their bandwidth as we can so they can see how well the link performs.
This year the core team, Joern Nettingsmeier and myself, are recruiting
volunteers to spread the workload. To that end we have set up a mailing list
and irc channel to coordinate our efforts. We will also have a wiki shortly.
If you will be coming to Cologne for the conference please consider signing up
to help. If you are not coming, please enjoy the fruits of our labors by
watching the streams and participating via irc.
stream team mailing list: http://zhevny.com/mailman/listinfo/lac-streams
general conference chat: #lac2008 on irc.freenode.net
stream team tech talk: #lac2008-tech on irc.freenode.net
Thanks,
Eric Rz.
--
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dangerous content by MailScanner, and is
believed to be clean.
Dear List,
having strange Problems with the numbering of devices in /proc/asound/cards.
The computer has a built-in RME9652 and two usb-Midi-Interfaces.
I have specified the order in the /etc/modules file as follows:
snd_rme9652 index 0
snd_usb_audio index 1,2
although from time to time, when unplugging and plugging in the
usb-Midi-interfaces, they get inserted as devices 0 and 1, thus shifting
the RME card to position 2 which causes great confusion.
This setup is to be used for an important performance, so it is
absolutely crucial to get this sorted out, and i hope for great hints
and suggestions as usual from this wonderful list.
with regards,
Peter
Hey guys,
I just recently posted here stating I was looking to convert my laptop
for a live performance machine using linux and mostly Pd (I was
thinking pd, but I said Max MSP instead). After browsing around for
various distros, I chose JAD become it was the most recently updated
one, was advertised to have a realtime kernel and could run with ASIO
drivers.
The problem was that I couldn't install it from the disk since my DVD
drive is semi kaput and can really only read CDs. So instead, I opted
to install openSuse 10.3 instead because they offered to install it
via a net connection. Its up and running (aside from my flipping
wireless! BROADCOM!!!) and I want to start some wireless work by
converting it into a JAD type workstation. The problem is that JAD is
based on OpenSuse10.2 and I'm running 10.3, which has me concerned.
There is a guide
(http://wiki.jacklab.net/index.php/3_Steps_to_JAD_for_Beginners) on
how to convert Suse10.2 to jad, but I'm afraid of compatibility issues
since my version is 10.3 and I don't know if the techniques described
there transfer over. Furthermore, I snooped around the Jacklab
repository and found a OpenSuse 10.3 folder, but as soon as I added
it, I got a warning from YaST telling me the description file was
broken... or something.
Furthermore, is there a way to install all the audio packages at once
instead of downloading them individually.
Thanks
-Chris
Hi,
Are you sure the clics and/or pops are in the source material? Your previous
problem hints that your jack setup might not be optimal and the problems
could come from jackd xruns.
If the problem is with the source material, you could try gnome wave cleaner:
http://gwc.sourceforge.net/
Sampo
Quoting Nicolas Peltier <peltier.nicolas(a)gmail.com>:
> Hello,
>
> you may recognize my name : i'm one of the linuxaudio users who yelled
> about
> watchdog timeout (jack problem).
> it still doesn't work and i'm still interested in any advice...
>
> BUT my mail isn't about that problem. I couldn't record and playback
> simultaneously so i tried another solution : recording through my Micro
> BR
> and then use ardour to playback and mix the audio files.
>
> The only problem that remains is i have clics or pops (i dont know the
> difference) on my guitar record...i tried to suppress them with
> audacity,
> but it seems not to work (with me as a user).
>
> Could i have explanations about using that function?
> Is there a more efficient/ easier tool to do that?
> why do clics and pops exist (it's a wav file)?
>
Hello,
you may recognize my name : i'm one of the linuxaudio users who yelled about
watchdog timeout (jack problem).
it still doesn't work and i'm still interested in any advice...
BUT my mail isn't about that problem. I couldn't record and playback
simultaneously so i tried another solution : recording through my Micro BR
and then use ardour to playback and mix the audio files.
The only problem that remains is i have clics or pops (i dont know the
difference) on my guitar record...i tried to suppress them with audacity,
but it seems not to work (with me as a user).
Could i have explanations about using that function?
Is there a more efficient/ easier tool to do that?
why do clics and pops exist (it's a wav file)?
Just curious, but I've heard that 64-bit OSes are slower than 32-bit
ones. That seems odd to me. Any truth to it?
--
David
gnome(a)hawaii.rr.com
authenticity, honesty, community
The Aqualung development team is pleased to announce the latest
release of the Aqualung music player.
Aqualung is an advanced music player originally targeted at the
GNU/Linux operating system, today also running on FreeBSD, OpenBSD and
Microsoft Windows. It plays audio CDs, internet radio streams and
podcasts as well as soundfiles in just about any audio format and has
the feature of inserting no gaps between adjacent tracks.
The ChangeLog is included below.
Please note that our website has moved to http://aqualung.factorial.hu
Please update your links and bookmarks.
Enjoy,
Tom
2008-02-10 Tom Szilagyi <tszilagyi at users dot sourceforge dot net>
* Aqualung 0.9beta9.1
http://aqualung.factorial.hu
This is a bugfix, stability and performance oriented release also
containing a few updates to existing functionality. By using this
version, your Aqualung will be more stable, and in some cases
significantly faster. All users are encouraged to upgrade.
The project homepage has moved to http://aqualung.factorial.hu
Please upgrade your pointers and bookmarks.
Notable changes:
* Playlist code refactoring for improved performance. Please note
that incompatible changes have been made to the playlist format:
this means that your old playlists won't be parsed, you'll have to
re-create them.
(NOTE: Music Store contents are unaffected. If we ever change the
Music Store format in a backward-incompatible way, we will provide
tools to migrate your precious store data.)
* Fix threading problems that caused random crashes for some users.
* Fix lurking bug that sometimes resulted in getting stuck at the
beginning of a track when Sinc interpolator sample rate converters
were used.
* Modified the way of opening ALSA output to achieve non-exclusive
driver access.
* New, more versatile title string generating templates. Make sure
to check the documentation.
* Added option to periodically save playlist.
* Several fixes concerning command line file and playlist loading,
esp. with the -L flag.
* Better metadata handling for external files in playlist.
* Increment CDDB revision number on resubmitting an existing
disc. This is essential for correcting existing CDDB entries,
otherwise the CDDB server rejects the submission. The latest CVS
version from http://libcddb.sf.net is required for this to work
(Aqualung-Win32 is built with this version).
* OpenBSD-related portability fixes. Aqualung should now compile
cleanly and be fully functional OOTB on OpenBSD 4.2.
* Updated German, Hungarian and Italian translations. Added
Russian translation.
Hallo,
Roman Haefeli hat gesagt: // Roman Haefeli wrote:
> On Sun, 2008-02-10 at 00:40 +0100, Frank Barknecht wrote:
>
> >
> > > And why 48KHz? 44.1 is standrd CD.
> >
> > 48 ist standard DVD. And standard jackd. ;)
>
> since when does jackd have a standard?
48kHz is the default samplerate in the ALSA backend, if you don't
specify anything else.
Ciao
--
Frank Barknecht _ ______footils.org__