Hi,
QMidiArp 0.5.2 has just seen the light of the day. It brings mainly
two improvements. One is a comeback, that of tempo changes on the fly,
and that now includes also tempo changes of a potential Jack Transport
master. Also the Jack Transport starting position is finally taken into
account, so that QMidiArp should be in sync also when starting the
transport master not at zero.
The second one is Non Session Manager support, mainly thanks to the work done by Roy Vegard Ovesen!
Note that for compiling in NSM support you will now need liblo as dependency.
Enjoy, and enjoy LAC in Graz this year
Frank
________________________________
QMidiArp is an advanced MIDI arpeggiator, programmable step sequencer and LFO.
Everything is on
http://qmidiarp.sourceforge.net
qmidiarp-0.5.2 (2013-05-09)
New Features
o Tempo changes are again possible while running, both manually or by
a Jack Transport Master
o Jack Transport position is now taken into account when starting,
QMidiArp used to start always at zero
o Muting and sequencer parameter changes can be deferred to pattern
end using a new toolbutton
o Modules in the Global Storage window have mute/defer buttons
o Global Storage location switches can be set to affect only the pattern
o Non Session Manager support with "switch" capability (thanks to
Roy Vegard Ovesen)
General Changes
o NSM support requires liblo development headers (liblo-dev package)
Hi Daniel,
I notice that I cannot help you without enough logs with which I can
realize what happens.
At lease, 'verbose' options for aplay/arecord, '#' nodes in
/proc/asound/cardX and dmesg.
Additionally, would you please add 'LANG=C' when you get output from
software.
I cannot read Spanish language...
> hi, it does not work.
Try in following steps:
1. $ arecord /tmp/test.wav -D hw:UFX1604 -c 16
(I expected you receive 'Available formats' output)
2. $ arecord /tmp/test.wav -D hw:UFX1604 -c 16 -f (here the format,
maybe S32_LE)
(I expected you receive 'rate is not accurate' output)
3. $ arecord /tmp/test.wav -D hw:UFX1604 -c 16 -f (the same above) -r
(got rate)
(I expected it will run correctly.)
I hope you to do these test when killing pulseaudio. The way is here:
https://wiki.ubuntu.com/PulseAudio/Log
Regards
Takashi Sakamoto
o-takashi(a)sakamocchi.jp
Hey Everybody,
I'm happy to announce OpenAV productions: http://openavproductions.com
OpenAV productions is a label under which I intend to release my
linux-audio software projects. The focus of the software is on the workflow
of creating live-electronic music and video.
The release system for OpenAV productions is one based on donations and
time, details are available on http://openavproductions.com/support
Sorcer is a wavetable synth, and is ready for release. Check out the
interface and demo reel on http://openavproductions.com/sorcer
Greetings from the LAC, -Harry
Hello everyone,
lately I had to fight big XRUN troubles, and thanks to this forum I
finally solved that. This excellent thread saved me:
http://linuxaudio.org/mailarchive/lau/2012/9/5/192706
On my long quest, I tried to see a little bit more what happened with
the IRQs on my system. I searched for a kind of 'top' utility to monitor
the interrupts, but the only apps I found were either deprecated, or
missed some cool features.
So, I ended up writing my own tool to monitor the file /proc/interrupts.
It's available a this address:
https://gitorious.org/elboulangero/itop
As its name indicates, it behaves pretty much like top, but for interrupts.
It's quite a simple thing, that I tried to enhance a bit with some cool
features:
+ refresh period can be specified.
+ two display modes: display interrupts for every CPU, or only a sum
of all CPU.
+ display every interrupt (sorted like /proc/interrupts), or only
active interrupts (sorted by activity).
+ in case the number of interrupts changes during the execution of
itop (due to a rmmod/modprobe), it's handled without any fuss.
+ command-line options are also available as hotkeys for convenience.
+ at last, the program display a summary on exit. The idea is that
this summary could be copied/pasted in emails to help debugging.
If anyone is interested, feel free to try and comment !
Cheers
On Mon, Feb 24, 2014 at 12:48 PM, David J Myers <
david.myers(a)amg-panogenics.com> wrote:
> 0.121.2 is the version I got from ubuntu with apt-get install.
>
> I removed that version and downloaded and built 0.124.1, however when I
> now try to run
>
> jackd -d alsa
>
> I get back
>
> jackd: unknown driver 'alsa'
>
>
>
> What's going on?
>
Your system is missing the development version of the ALSA library. So JACK
was built without ALSA support.
What version of Ubuntu are you using? I am unable to believe that 0.121 is
the latest version available for any reasonably recent version of that
distro.
--p
Hi there everyone, specially developers.
I think we should stop assuming releasing source code is enough.
[GNU/] Linux is getting more user friendly, and most users are not able
to compile software,
plus some distributions make it specially hard (debian, ubuntu, fedora,
opensuse) by having the libs installed but not the headers.
Releasing software on windows or mac, even open-source, *always* comes
in a binary,
and most users come from there.
Now, I have a "toolchain" repository for ubuntu 10.04 with gcc4.8,
python3+qt4 and a bunch of other useful stuff.
I use this to get generic linux binaries that (from what I know) work
everywhere.
I can make a developer-oriented tutorial on how to use that, so that
developers can provide linux binaries to its users.
Would that be something useful to Linux Audio?
Hi LADs
We are working on an ARM based MOD device and while fiddling with JACK some
questions appeared.
I didn't know whether to post on the ALSA os JACK lists, so I decided to
post here :-)
When I start Jack in my PC I get the well know message:
/usr/bin/jackd -P80 -dalsa -r48000 -p128 -n2 -D -Chw:0 -Phw:0
creating alsa driver ... hw:0|hw:0|128|2|48000|0|0|nomon|swmeter|-|32bit
configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods
ALSA: final selected sample format for capture: 32bit integer little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit integer little-endian
ALSA: use 2 periods for playback
When starting it on the Beaglebone Black I get:
/usr/bin/jackd --realtime -P80 -dalsa -r48000 -p128 -n2 -Xraw
creating alsa driver ... hw:0|hw:0|128|2|48000|0|0|nomon|swmeter|-|32bit
configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods
ALSA: final selected sample format for capture: 32bit integer little-endian
ALSA: use 16 periods for capture
ALSA: final selected sample format for playback: 32bit integer little-endian
ALSA: use 16 periods for playback
- what are those "periods" in the ALSA lines? (2 for the PC and 16 for the
BBB) ? I read a post by Jeremy Jongepier about the Cubieboard2 stating that
the hardcoded values for the Cubie (Minimum number of periods was 4 and
minimum buffer size was 1024) where unfit for realtime operation. Why is
that so?
- on the ArchWiki page on Jack there is a D-bus call
jack_control eps realtime true
stating that it "Sets JACK to realtime mode in its own internal setup."
What is this internal setup? How do I address this when not using D-Bus?
King regards all
Gianfranco Ceccolini
The MOD Team
PS: We will be at LAC againd this year. I'm very eager to meet most of this
distinguished community members there. We have lots of good new for the MOD
in 2014.
Hi *!
I'm trying to get my Gigaport HD+ to run at 48000kHz. The specs say it
is capable of 8ch @ 44k1/16, 6ch at 44k1/24 and 48k/24.
When I try to start it at 48k, it comes up ok but ends up running at
44k1. It shows 8 channels, so that's expected.
Now, how do I tell it to use only 6, so that I can get to 48k?
I've tried setting -o6, which gives the usual "cannot set playback
channel count".
Next I looked at amixer -D hw:3 controls, and there is a playback switch
map that is set to "on,on,on,on,on,on,on,on". I set it to
"on,on,on,on,on,on,off,off" and retry. No luck. Still fails to set
playback channel count, JACK still comes up with 8 outs and falls back
to 44k1.
What am I missing? Is there some .asoundrc magic that might come to the
rescue? Maybe define a device that has only 6 channels to begin with?
But how do I do that?
Any hints much appreciated.
Best,
Jörn
--
Jörn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487
Meister für Veranstaltungstechnik (Bühne/Studio)
Tonmeister VDT
http://stackingdwarves.net
Hi,
I saw a message with this title in LAA:
lm3jo en deb version 1.1.1-5
It has a link to a deb file, which looks very suspicios. Is it malware for
Debian? It would be the proof that this is the year of the Linux desktop.
Cheers,
Andrés