Leonardo Gabrielli | 22 Oct 13:06 2014
Picon

Re: [LAU] Small instrument hardware module


On 22/10/2014 11:45,
linux-audio-user-request@... wrote:
> In my experience there's a greater risk of overheating without a fan and
> the ARM (allwinner) chipsets are prone to that. My bet is a low power x86
> processor/unit with a (quiet) fan will out perform and outlast an ARM
> chipset without.
I did some simple benchmarks on a Allwinner A20 board (cubieboard) 
recently. The benchmark consist of computing a bunch of sine oscillators 
(second order resonator filter), generally used for modal synthesis and 
other types of sound synthesis. The results I got from the A20 when 
clocked at 1GHz are suprisingly good: 1000 theoretical oscillator can be 
computed in a 128 samples period, while on my quad core-i5 I get 1500. 
On a 7-years onld Centrino Duo I get about 850. While this don't stand 
as a real-world benchmark (buffer transfers are not taken into account) 
and I haven't optimized for the architectures (but just let g++ go with 
-O2) you get the idea.

I didn't experience overheating on the A20 but the tests are not 
continuous as you would during a performance, so I won't bet it will 
last long. :)

I have a sensation that generally the kernel is also quite unstable on 
most platforms unless a silicon manufacturer is there to help (as it 
happens with some TI chips) and in general I would prefer Intel for 
reliable live performance. However as a researcher I am trying to 
squeeze ARMs to perform as musical instruments and I think they can work 
well if the industry supports kernel development. But I'm wondering if 
this will continue to happen, since the eastern mobile market is 
crushing the sales of the reliable manufacturers.
(Continue reading)

Patrick Shirkey | 22 Oct 09:56 2014

[LAU] The Passion of Raul

Hi,

Something for Halloween.

Raul is a Vampire who is driven by his unquenchable passion in the dead of
night.

https://www.youtube.com/watch?v=2KJTGA4S0zQ

This is a re-edit of an earlier private work titled "Rauls Redeption" by
Volker Allert, http://volkerallert.eu, a renown physical effects creator
for the Hollywood film industry.

All audio recorded with Linux tools and edited in Blender on Linux of course.

--
Patrick Shirkey
Boost Hardware Ltd
Will Godfrey | 21 Oct 19:12 2014
Picon

[LAU] A survey if you don't mind

I'm posting this in a number of places to try and get some idea of ALSA/JACK
usage, so I'd be grateful for responses.

https://www.surveymonkey.com/s/JZVV7K9

--

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Ivica Bukvic | 21 Oct 14:40 2014
Picon

[LAU] ANN: SEAMUS 2015 -- Call for submissions

Apologies for x-posting:

The SEAMUS 2015 conference will be held at Virginia Tech during March 26-28, 2015. The conference theme is "Emotion and Electroacoustic Music." The submission deadline is October 31, 2014. Please see http://seamus.music.vt.edu/main/ for further details on the conference, and http://www.seamusonline.org/ for further information about SEAMUS.

Any questions regarding the conference may be directed to seamus <at> music.vt.edu.

Best,

Ico

_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@...
http://lists.linuxaudio.org/listinfo/linux-audio-user
Vaclav Mach | 21 Oct 13:20 2014

[LAU] ALSA and RME Raydat troubleshooting

Hi all,

I'm the team leader of Voice Communication Systems development in ARTISYS company. We try to utilize RME
RayDat sound device on Linux system with ALSA sound interface. After a lot of experiments we are not able to
set up proper sound output from this device, so we kindly ask you for a help to solve our problem. Here is a
brief description:

General task of our troubleshooting is that our audio signal output is not continuous. The output signal is
rather interrupted. Here is our configuration:

PC with Linux (kernel v. 3.17.0) with ALSA sound libraries compiled in kernel. We have also additional
packages installed: alsa-tools, alsa-utils and alsa-lib (all version 1.0.28).

Inside the PC there is PCI Express card RME Raydat, whose drivers are also compiled in the Linux kernel. This
interface is connected by optical fibres (ADAT) to the Ferrofish A16 MK-II (2 pairs of TOSLINK cables).

Playing test files using commands

aplay -D pcm.out_test -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Left.wav -vv 
and
aplay -D pcm.out_test2 -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Right.wav -vv

causes audible interruptions. The configuration of the ALSA device is following:

pcm.out_dmix {
	type dmix
	ipc_key 56874
	ipc_key_add_uid false
	ipc_perm 0666
	slave {
		pcm "hw:2,0"
		period_size 2048
		channels 36
		rate 48000
	}
	bindings {
		0 0 # from 0 => to 0
		1 1 # from 1 => to 1
	}
}

pcm.out_test {
	type plug
	slave.pcm "out_dmix"
	ttable.0.0 1
}

pcm.out_test2 {
	type route
	slave.pcm "out_dmix"
	ttable.0.1 1
}

Let me describe other two example experiments:

1. If I try to record sound using the same parameters as out_dmix device but of type "dsnoop" (equivalent of
dmix for capturing), there are no interruptions and the recorded sound is perfect.

2. If I play single sound file to the output of the RME card while the ALSA device is of type "route", the sound
output is perfect. Playing second file on different output channel this way I'm not able to open
other output channel simultaneously by another program because the device is busy. There is this
experiment's ALSA device configuration:

pcm.out_test {
	type route
	slave.pcm "hw:2,0"
	slave.format "S32_LE"
	slave.channels 36
	ttable.0.0 1
}
pcm.out_test2 {
	type route
	slave.pcm "hw:2,0"
	slave.format "S32_LE"
	slave.channels 36
	ttable.0.1 1
}
This should be solved by "dmix" type ALSA device, which currently causes interruptions in our configuration.

The question is: how can we play multiple audio streams to multiple output channels using our equipment?

Thank you. 
Dear Mr. Knoth,

I'm the team leader of Voice Communication Systems development in ARTISYS company. We try to utilize RME
RayDat sound device on Linux system with ALSA sound interface. After a lot of experiments we are not able to
set up proper sound output from this device, so we kindly ask you for a help to solve our problem. Here is a
brief description:

General task of our troubleshooting is that our audio signal output is not continuous. The output signal is
rather interrupted. Here is our configuration:

PC with Linux (kernel v. 3.17.0) with ALSA sound libraries compiled in kernel. We have also additional
packages installed: alsa-tools, alsa-utils and alsa-lib (all version 1.0.28).

Inside the PC there is PCI Express card RME Raydat, whose drivers are also compiled in the Linux kernel. This
interface is connected by optical fibres (ADAT) to the Ferrofish A16 MK-II (2 pairs of TOSLINK cables).

Playing test files using commands

aplay -D pcm.out_test -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Left.wav -vv 
and
aplay -D pcm.out_test2 -r 48000 -f S32_LE /usr/share/sounds/alsa/Front_Right.wav -vv

causes audible interruptions. The configuration of the ALSA device is following:

pcm.out_dmix {
	type dmix
	ipc_key 56874
	ipc_key_add_uid false
	ipc_perm 0666
	slave {
		pcm "hw:2,0"
		period_size 2048
		channels 36
		rate 48000
	}
	bindings {
		0 0 # from 0 => to 0
		1 1 # from 1 => to 1
	}
}

pcm.out_test {
	type plug
	slave.pcm "out_dmix"
	ttable.0.0 1
}

pcm.out_test2 {
	type route
	slave.pcm "out_dmix"
	ttable.0.1 1
}

Let me describe other two example experiments:

1. If I try to record sound using the same parameters as out_dmix device but of type "dsnoop" (equivalent of
dmix for capturing), there are no interruptions and the recorded sound is perfect.

2. If I play single sound file to the output of the RME card while the ALSA device is of type "route", the sound
output is perfect. Playing second file on different output channel this way I'm not able to open
other output channel simultaneously by another program because the device is busy. There is this
experiment's ALSA device configuration:

pcm.out_test {
	type route
	slave.pcm "hw:2,0"
	slave.format "S32_LE"
	slave.channels 36
	ttable.0.0 1
}
pcm.out_test2 {
	type route
	slave.pcm "hw:2,0"
	slave.format "S32_LE"
	slave.channels 36
	ttable.0.1 1
}
This should be solved by "dmix" type ALSA device, which currently causes interruptions in our configuration.

The question is: how can we play multiple audio streams to multiple output channels using our equipment?

Thank you. 
Yours sincerely

Ing. Vaclav Mach
Voice Communications System team leader
ARTISYS
www: http://www.artisys.aero    
Benoît Rouits | 20 Oct 23:32 2014
Picon

[LAU] [OT] loudspeaker enclosure design software, testers welcomed.

Hello,

A long time i didn't post here, and now for something a bit OT:
I just finished to write a tiny Qt based software for DIYers on
loudspeaker enclosure design, under Linux. It is inspired by the
discontinued gspeakers Gtk+ based software. For the moment it has
basic only features, but it is usable, though there is a long TODO.
I you are interested, (and if some debian packager is around here, 
[whisling] ;-) you can checkout the svn tree and try it yourself:

$ svn co https://dbx.gtmp.org/svn/qspeakers/ [*]
$ cd qspeakers; qmake -config release && make && sudo make install
(it will install in /usr/local/..)

There is a 4 or 5 speakers DB for a start, but it won't be visible
if qspeaker is not _installed_. After installation, first run will
copy the DB into  ~/.local/share/data/Herewe/QSpeakers so that you
can edit it with your own speaker's Thiele-Small parameters.

Thank you for trying !
- Ben

[*] thanks to Mid', a friend of mine, who gave me some space on his server.
rosea grammostola | 20 Oct 11:18 2014
Picon

[LAU] Small instrument hardware module

Hi,

Is it possible to install JACK etc. and for instance Pianoteq on a device like this, to create your own instrument module?

http://www.rikomagic.com/en/product/showpro_id_58_pid_22.html


Regards,
Dirk
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@...
http://lists.linuxaudio.org/listinfo/linux-audio-user
Rui Nuno Capela | 19 Oct 22:49 2014

[LAU] [ANN] QjackCtl 0.3.12 released, feat. JACK Pretty-names aliasing

Again, a classic needs no introduction...

   QjackCtl 0.3.12 is now released!

Change-log:
- JACK client/port pretty-name (metadata) support is being introduced 
and seamlessly integrated with old Connections client/port aliases 
editing (rename) (refactored from an original patch by Paul Davis, 
thanks). (EXPERIMENTAL)
- Application close confirm warning is now raising the main window as 
visible and active for due top level display, especially applicable when 
minimized to the system tray.
- Messages standard output capture has been slightly improved as for 
non-blocking i/o, whenever available.
- Translations install directory change.
- Allow the build system to include an user specified LDFLAGS.
- Missing input/output-latency parameter settings now settled for the 
D-BUS controlled JACK server and firewire back-end driver.

Website:
   http://qjackctl.sourceforge.net

Project page:
   http://sourceforge.net/projects/qjackctl

Downloads:
- source tarball:
   http://downloads.sourceforge.net/qjackctl/qjackctl-0.3.12.tar.gz
- source package (openSUSE 13.1):

http://downloads.sourceforge.net/qjackctl/qjackctl-0.3.12-21.rncbc.suse131.src.rpm
- binary packages (openSUSE 13.1):

http://downloads.sourceforge.net/qjackctl/qjackctl-0.3.12-21.rncbc.suse131.i586.rpm

http://downloads.sourceforge.net/qjackctl/qjackctl-0.3.12-21.rncbc.suse131.x86_64.rpm

Weblog (upstream support):
   http://www.rncbc.org

License:
   QjackCtl is free, open-source software, distributed under the terms 
of the GNU General Public License (GPL) version 2 or later.

See also:
   http://www.rncbc.org/drupal/node/826

Enjoy && Have fun!
--
rncbc aka Rui Nuno Capela
rncbc@...
Philipp Überbacher | 19 Oct 22:10 2014

[LAU] [SEMI-OT] Looking for simple audio project

Hi there,

I'm attending a basic multimedia on mobile devices course at
university. During that course I'll have to do a small project,
possibly together with one or two others. I want to do something audio
related of course. The university staff is pretty much focused on
video, so they won't be of much help.

I'm looking for a reasonably low skill/work project idea.

The problem is that I have to do this for iOS (hence the SEMI-OT). I
wouldn't want to develop for or use that horrible platform but if I have
to do it anyway I might as well get some audio coding experience.

Maybe there is something that would be useful to port? Maybe there's
some need for a iPad as remote control thingy? Or some little thing
from scratch?

Suggestions are welcome.

Regards,
Philipp
Ken Restivo | 19 Oct 02:55 2014

[LAU] I made a thing (for ogg vorbis concatenation)

I made this one-off hack based on the vorbis lib sample code, for concatenating ogg files. The README has
more details:

http://github.com/kenrestivo/thrashcat

That itch is now scratched; but others might find it useful too, so here it is.

-ken

[LAU] Ardour: Taking out bits of a song

Hello all,

  I have a project in Ardour which has much more iterations of verses
than the fnal project wil have, for the sake of explorations,
improvisations, etc.  Is it practical to work on the structure itself
or should I redo all the tracks from scratch with a new count of
verses ?  By 'practical' I mean are there tools and tricks that would
make it easier to take out one verse made out of say, 7 tracks ?  The
project is in sync with the metronome, if it helps.  I can see that it
is possible to select one verse across all the tracks and then delete
it.  Since I do not know that much all that Ardour can do, I'm asking
here if there are better ways to do that using specific Ardour commands
that might be better to the task than selecting with a mouse on a
finer-grained display of the tracks.  Is it also possible to store a
cut-out verse somewhere persistently to use it later ?  - thanks for
any comments and suggestions !

Cheers.

Gmane