On Wed, Jan 13, 2010 at 12:55 PM, Rob Wentz wrote:
> Ok, just saying here.. It might be more helpful to post the contents or at
> least more of a clue to what your .zynaddsubfxXML.cfg contains, instead
> of how clueless you are about it's origin ... or how tasty miso soup is for
> that matter.. capish?
>
yes, I could have attached it in the first place,
done now.
The issue came up before:
http://linuxaudio.org/mailarchive/lau/2009/11/12/161948
I am still interested in where the file comes from, that's
why I mentioned not having a clue. I could have written
an explicit question as well.
Usually I stay on topic, IMHO the little OT line, more meant
to be ignored, is acceptable.
all the best,
d
Hi,
If you want to work on Linux the 'modular way', you need good Jack
Transport, for syncing apps and looping. Currently there are some
limitations, which you can find here:
http://trac.jackaudio.org/wiki/TransportLimitations
So looping is not really possible.
Maybe you as an Linux Audio User can describe a concrete 'user case
example'. Something you like to be able to do with some apps, but which
is not possible at the moment. Think about syncing but also looping with
Jack Transport.
This will help us getting a clear picture about the state of Jack
Transport and how it could/should be improved.
Thanks in advance,
\r
hi everybody!
my friends and I have started a mix tape club. basically, we dump
everything we are listening to on a CD and hand it out to all of our
friends. I'm 4 days late on the december deadline, and need to get
something in quick.
I've got all my tracks, but I'd really like to do something...errm,
strange with them. I read a post on reddit.com that feh can be used
to create collages of images:
http://www.reddit.com/r/linux/comments/akudo/feh_create_random_art_from_the…
is there an audio application that can make audio collages from audio
files? I understand that the end result would be somewhat strange,
and you might only get 1 out of 10 results that would be considered
listenable, but it might be fun to try anyway.
any ideas?
--
Josh Lawrence
http://www.hardbop200.com
I know it's an old can of worms but...
Anyone have a guitar tuner working with jack? From what I can tell:
Lingot should be jack capable with the "next release". Anyone built
this from cvs?
Qjacktuner is broken (not compatible with modern qt versions?)
Fmit is broken (not compatible with modern jack versions?)
Ben
Hi all,
Happy new decade!
Back on that WiFi jamming issue I wrote about a while ago. Thank you
all for the various suggestions. I have spent most of my time trying
to get somewhere with netjack as I found it the most promising, not
only in terms of efficiency but also scripting and control. However,
I have hit some brick walls which are certainly due to the nature of
WiFi networks (packet collisions and such) which make my setup very
unstable.
First of all, in my setup, there will be one "server" computer
collecting 5 signals from the 5 musicians and playing them for the
audience. Moreover, the operator (or software) will send back 5 mono
signals to performers (one signal per performer). The idea is that
each performer hears only one instrument at a time.
Currently the show stopper lies in alsa_in and alsa_out components.
Whenever I run both on my netbook (Atom 1.6G with Atheros
Communications Inc. AR5001 Wireless Network Adapter (802.11g)) they
consume around 70-80%CPU and they peak at around 50% each whenever
they choke on net over/underruns. They eventually segfault (and
sometimes bring the jack server down with them. This particularly
true whenever the WiFi signal signal strength is weak (around 50-60%).
I was especially surprised at the high CPU consumption of those apps,
as I figured that if all they do is schlep audio to and from the
soundcard they would keep a low profile... My built-in audio
interface is:
lspci -s 00:1b.0 -vv
00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High
Definition Audio Controller (rev 02)
Subsystem: Acer Incorporated [ALI] Device 019c
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr-
Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort-
<TAbort- <MAbort- >SERR- <PERR- INTx-
Latency: 0
Interrupt: pin A routed to IRQ 16
Region 0: Memory at 56340000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: HDA Intel
Kernel modules: snd-hda-intel
It is an Acer Aspire One netbook. Running Karmic with jack compiled
by hand, version 0.118.0
Any insight will be greatly appreciated.
./MiS
Hi All,
I want to rip a bunch of vinyls to FLAC. I first tried Audacity but when I
change the interface to my Audio 4 DJ in preferences Audacity gets killed.
Can anybody help or suggest another recording device? Also when recording
with the built-in Sound Recorder (UbuntuStudio Karmic) and then trying to
open (import) the files with Audacity just does nothing (although the Movie
Player plays the files correctly.
Thanks for any ideas!
Martin
Hi guys, I just wanted to give an update on the integration of
librubberband into TerminatorX.
To understand rubberband, i wrote a small console Jack app that can play
wav files at diferent speeds, keeping pitch. It works well. The code is
attached.
I have a question to for the devs out there though: It seems to me that
i have to run process() a few times with a fixed blocksize before,
getRequiredSamples() returns something >0 in Realtime-mode. All other
options are default options. Is this true? I need some help on this
issue.
thanx Gerald
Hi guys, I just wanted to give an update on the integration of
librubberband into TerminatorX.
To understand rubberband, i wrote a small console Jack app that can play
wav files at diferent speeds, keeping pitch. It works well. The code is
attached.
I have a question to for the devs out there though: It seems to me that
i have to run process() a few times with a fixed blocksize before,
getRequiredSamples() returns something >0 in Realtime-mode. All other
options are default options. Is this true? I need some help on this
issue.
thanx Gerald
Hi folks
I had my demo recently -- used nted with ubuntu karmic. [Had to
upgrade from hardy to karmic for the demo because earlier X versions
would not project -- but thats another story]. It was a huge success
as far as I can see to the extent that other schedules were cancelled
so that I could teach more singing to the group. So thanks to people
on this list and of course to Joerg for nted.
So now the question: Some of the folks there asked me: Did you write
this software?
I laughed: Use nted -- for that you need to use linux.
Could I give a more helpful answer? Whats the nearest windows
equivalent of nted?
I told them the nearest I know is Cubase...
Thanks
Rustom
OK well I just applied the patch I needed directly to the ALSA that was in the source tree, bumped my --append-to-version, and recompiled the whole kernel. Not too tedious on a 64-bit machine using -j2. Took slightly less time than compiling Ardour ;-)
It is this patch that I am trying to get working:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3249
And the version that applied cleanly to my kernel version (2.6.26.8-rt12) was the one called "fasttrackpro.patch", which was apparently intended for 2.6.27. Good enough.
48Khz now works for playback and capture, yay. But not in 24-bit mode; only in 16bit mode.
This is using 0x9 or 0xd as device_setup paramters; either way, I get only 48/16, not 48/24. I don't get the digital inputs even if I try, but that's OK, I don't need them anyway.
Why am I even bothering to do any of this? Because Lahar is going into the studio. A friend will record our basic tracks using his ProTools setup (alas, Digidesign Firewire interfaces won't ever work with Linux, so we're stuck with ProTools on a Mac). He tracks at 48Khz, so I needed to bump my FastTrack up to 48Khz so that I can use Ardour to mix it. So far so good. Yes, this means a mixed-with-Linux CD is coming soon.
Lack of 24-bit is neither tragic nor fatal-- it's all 32-bit floats inside JACK, and it'll get mastered either with JAMIN at 32 bits, or via a professional shop at 24 bits--, but it'd sure be nice to have those extra bits from the input of the FastTrack for recording overdubs if needed.
-ken