Well since people were commenting on so many emails being written by one
person, Ill toss out one by me, even if it is a bit of a plug:) I
apologize in advance if this is against the policy for the list, but I
couldn't find where that gets spelled out anywhere. Of course now that I
have said that someone is likely to point out the blatantly obvious link I
missed:)
Tube is an open movie project being made with open source tools primarily
(Not sure on the music, but the SFX definitely though I will likely be
using LinuxDSP etc. plugins). It has just a few more days left in it's
funding drive, and while we hit the bare minimum goal, we are trying to
reach $50k all together. The goal is to release the video with all sources
so that anyone can use them, and for my part this includes releasing the
SFX sessions and source files, I will be recording all of them for this
project for this purpose. If anyone is interested you can find out more
info at the Kickstarter page or help fund it possibly.
http://www.kickstarter.com/projects/1331941187/the-tube-open-movie
I also am attaching a copy of the message I sent to Ardour-User close to a
month back explaining my own involvement and the plan for using Linux Audio
programs to do the SFX design. Since then I have gotten the test version
of A3 compiling and running on my machine, and it is likely I will do the
final SFX design and possibly final mix in it as well, but Mixbus is being
used for the initial draft for animators to use for inspiration before I
re-record all the sounds.
Thanks for the noise...
-----------------------------------------------------------------------------
Whenever an open movie gets made by the Blender institute, there seems to
always pop up a discussion in various places that deal with Linux Audio the
same question. "Why wasn't Ardour/etc. used to make this instead of
Logic?". While I won't go into the answers here, other than to say I agree
with Jan's thought process on this, I will say that there is an open movie
being made that is incorporating Ardour and Mixbus into it.
Tube is an open movie being made by a crew of people, many of whom are also
part of the Blender community, and Bassam was the director for the very
first open movie released by the Blender Institute, "Elephant's Dream".
While this open movie does not have anything directly to do with the
Blender Institute, I can say that it is being made with Ardour and Mixbus
for SFX design, taking another step into the realm of openness.
I can say this because I am working on the SFX for Tube in those programs
myself. The goal is to replace all of the current sounds with sounds I
record myself so that we can release all the sounds openly, and the
sessions as well, but for the moment I am using a mix of my own recordings
and library sounds as I am very short for time to go out and do field
recordings. None the less, we are attempting to finish up this movie this
year and have started fundraising for it if you want to help out and
participate. You can find more info on the kickstarter page...
http://www.kickstarter.com/projects/1331941187/the-tube-open-movie
Seablade
aka Thomas Vecchione
PS. Yes the audio for the video was edited and mixed in Mixbus on Linux
with a variety of software plugins including restoration VSTs (WaveARTS)
via Festige, LinuxDSP plugins, and a few others. Again credit to Robin for
the excellent Jadeo software, even if it seems to want to turn everyone in
the video into Smurfs while I am viewing in it, but I suspect that is more
due to a very out of date version.
Replying to a digest, sorry if that screws up mail threading.
From: Robin Gareus <robin(a)gareus.org>
> Subject: Re: [LAU] USB audio interfaces >= 8 channels
> Here it mainly concerns the outputs: In my case there's
> a ground loop between my Screen (Asus VE278) and the
> active speakers when connected via computer and an
> USB UA-25 (not UA25-ex which features a ground-lift switch)
> -> 1/4inch TRS -> B2031A speakers.
You have found that there are ways to design balanced inputs improperly.
The Audio Engineering Society devoted an entire issue of the journal (June
2005) to such issues.
> A multimeter shows a constant 1mA current and ~3-4mV AC potential.
Should not be a problem for properly designed balanced inputs and outputs,
can be for improperly designed equipment.
> /me is pondering to cut the ground-wire from the screen..
> but I have so far refrained from doing that.
Could cause other problems. Probably your choices are either to modify
the equipment so it is not quite so improperly constructed, use a
transformer to isolate the equipment from the shield current, or possibly
to construct a cable which works around the offending equipment.
From: Fons Adriaensen <fons(a)linuxaudio.org>
> Very few multimeters are capable of measureing AC current with
> any level of accuracy. If you have 150 mV between two points, and
> zero current when you short-circuit them (as a current meter is supposed
> to do) then at least one of the two measurements is bogus.
Probably what happens is that when the screen is not connected, there is a
high impedance voltage difference between the devices, and when the
shields are connected together (through the ammeter) current actually
flows, but is so low that the ammeter does not measure it correctly.
When the VGA monitor is connected, the monitor has low enough impedance
leakage path that it can source a couple of milliamps through the shield,
and the USB interface or speaker (or both) has a common impedance path
for that current to flow on the reference potential node of a high gain
stage and amplifies the noise current.
From: Robin Gareus <robin(a)gareus.org>
> Any suggestions before I add a switch to the ground of the screen?
Verify the wiring, make sure that there are no broken conductors or broken
solder joints in the shield connections of the cables between the audio
interface and the speakers.
Are you using TRS-TRS cables, or TRS-XLR cables for the connection from
audio interface to speakers?
Which of the equipment (video monitor, computer, active speakers) have
three wire power connections with safety ground connected, and which (if
any) have only two wire connections. Equipment in the US is typically
double insulated and so does not require a safety earth connection, but I
don't know if that also applies on equipment shipped for use on European
240V power distribution.
I ask because you need to give the ground current someplace to flow to
complete the circuit which does not flow across the reference potential
node (sometimes called the "ground connection") of a high gain stage.
Could be in the output of the audio interface, or the input of the
speaker, or both, where that current is being converted to audible noise.
One way to do that is to make sure the shield of the cable connects well
to the shield of the connector. You generally have to connect the cable
that way in a TRS connector, but XLR connectors do not by default have a
connection between pin1 and the connector shell. The equipment should
connect pin 1 to a low impedance equipment shield connection internally,
but many designs do not. In that case connecting pin 1 to the connector
shell inside the connector can sometimes help.
If some of the equipment has an earthed chassis and some does not,
sometime making an external connection (using wire or copper braid)
between the different chassis can reduce the potential difference enough
that the current flowing on the audio cable becomes low enough to be
inaudible.
Something like one of these would probably help:
http://jensentransformers.com/dm2xx.htmlhttp://jensentransformers.com/pi2xx.html
But good transformers are not inexpensive, it might be the same cost to
get an audio interface which did not inject so much noise current into the
output. If the speakers are causing the problem and not the audio
interface, then you may be able to make some modifications to the input
connections of the speaker interface to solve the issue. Depends on how
the amplifier assembly is physically constructed.
--
Chris Caudle
Hi *,
I'm looking into buying a new audio-interface. It should have at least 8
analog in and 8 analog outputs - preferably both unbalanced XML and work
with various laptops for the foreseeable future (which likely rules out
firewire and cardbus).
It should be able to provide phantom-power and have decent preamps but
they don't need to be excellent (that's what real preamps are for).
It must be portable and robust (both physical as well as electrical ie.
ground-lift) and it goes without saying: be supported by Linux.
It should also not be more expensive than 1000 euros; and preferably
cheaper.
So far I'm eying the Presonus Audiobox 1818VSL:
http://www.presonus.com/products/Detail.aspx?ProductId=65
It has only balanced outputs but otherwise fits the bill.
What seems weird with with this box: it has a word-clock out but no
word-clock input. Also, while it does support up to 96K SPS digitally,
all analog I/O is apparently limited to 20Hz-20kHz (the latter might be
a good thing, though). Oh well. good enough. What is not clear to me is
if the balanced I/Os are [or can be] ground-lifted.
Does anyone have experience with this device? Pros/Cons?
What's the minimum round-trip latency achieved with jack?
Any alternative suggestions?
Bonus-point if the device comes without useless bundled software :)
TIA,
robin
hi all,
i'm wondering if anyone can help me with some software i am looking for.
i recall some years back, when i was using cakewalk/sonar in the late
90s/early 00s, a facility called studioware which allowed me to
control a hardware synthesiser from within cakewalk. this was driven
by files which one could load, which then represented each of the
functions of a midi-connected hardware synthesiser/sampler/fx unit by
an on-screen button, slider, dial, etc.
a large number of the various hardware synthesisers available at the
time had a studioware panel defined for them, and there was several
repositories of these files, where users shared these files - they
were fairly simple to create, some versions of cakewalk/sonar software
included a design environment.
is there a similar facility, maybe as a plugin, available for any
linux DAW software, e.g. rosegarden/ardour? i have a couple of fairly
complicated hardware synths i'd like to control, and the tiny screen
and two physical knobs they provide are not conducive to in-depth
editing.
cheers,
--
robin
http://fu.ac.nz - Auckland's Free University
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Hash: SHA1
gjacktransport is a gtk application that allows to control
JACK-transport. It includes a stand-alone /big-jack-clock/ application
called gjackclock.
Version 0.5.3 released last night is a maintenance update that allows
the apps to run on the GNU Hurd.
Thanks for the bug-report and patch by Cyril Roelandt.
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=671586
The sourcecode and more information can be found at
http://gjacktransport.sf.net/
Cheers!
robin
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On 05/16/2012 01:30 AM, Brian Quandt wrote:
> Yeah only complaint I hear from folks is the AD-DA circuitry, but they
> are from folks who don't give specifics. I think the only diffs are
> what people get used to there, as well as how much money they might
> spend on gear and therefore be influenced by their own choices.
> This 1818 seems pretty cool in my mind, and frankly for what it cost and
> can do, it's awesome!
+1
@Fons: I remember you were measuring preamps and AD/DA behaviour not too
long ago (some RME card?!).
Can one reproduce this using jnoisemeter or does one need special
hardware equipment?
Setting the output-volume to -0db, `jnoise` could be used to calibrate
the input levels, even though it's bad practice to rely on the device
itself, I suppose it'd work to measure the noise-floor and S/N, right?
...and `jaaa` will do the frequency plot... I love your tools!
best,
robin
Hi folks,
I'm on Lucid 10.04.
I installed the rt kernel 2.6.33-8 from here [1], because the new nvidia
driver doesn't like the rt 2.6.31.
Now, with the new rt kernel qjackctl starts fine, then the gui crashes
after about 1 minute, and qjackctl.bin freeze in the background.
Then I have to shut down my machine pressing the halt button, because
everything is frozen (cannot launch a terminal, nor type in it if it's
open).
It happened 7 times over 7 trials.
This makes it unusable. And I need the new kernel to do audiovideo stuff
with my nvidia card.
Any hint?
I read that this kernel has some problems with PulseAudio, but I don't know
if this could relate to qjack.
many thanks in advance!
[1] https://launchpad.net/~bojo42/+archive/rt
--
Marco Donnarumma
New Media + Sonic Arts Practitioner, Performer, Teacher, Director.
ACE, Sound Design MSc by Research (ongoing)
The University of Edinburgh, UK
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Portfolio: http://marcodonnarumma.com
Research: http://res.marcodonnarumma.com | http://www.thesaddj.com |
http://www.flxer.net
Director: http://www.liveperformersmeeting.net
So, since I am a long time lurker (and occasional poster) on this list, I'd
suggest that we give everyone who reads the list the freedom to write (post)
to the list as long as it is in the spirit of the Mission Statement.
What! There is no Mission Statement? (I could have sworn I had it in my
back pocket a few minutes ago). I guess I have to do another google search
and figure out what I did with the Mission Statement...
Well, Ardour's got one...
LinuxMuscian's got one...
Linux Audio DOT org makes me have to make a security exception because their
certificate can't be traced. They don't have a mission, so maybe if I read
the Policy Statement it would be good enough...
"The aim of the Linuxaudio.org consortium is to promote and enable the use
of Linux kernel based systems for professional audio use."
Well, their policies look pretty good and the Policy Statement aim could do
for a mission statement in a pinch.. A generic statement like this is ok,
but it doesn't produce a hard-on or make me want to give my life for it.
OK, so maybe we have to craft a mission statement:
How about a mission statement like this: "This mailing list consists of a
bunch of cheap bastard techno-dweebs that have no life of their own who are
mainly attracted to linux audio because it is so cheap to implement as long
as you don't count your own time as being worth anything. We have long ago
lost the ability to make music on our own because we have fallen into the
hole of continually tweaking and trying to make something work. We are
lost in our own underwear and can't find our way out. We haven't noticed
that the rest of the audio world uses Windows or Mac O/S and don't care
because those apps aren't on Pirate's Bay and we are too self absorbed to
give any attention to anything we'd have to pay for. We do not normally
work together and use this mailing list to espouse our individual opinions
or problems."
Naw, this one is a bit long.
Here's another potential mission statement: "Linux is good"
Nope. This one is too short.
OK, here's a last one:
"The LAU list is dedicated to providing freedom for the developer, musician,
and audience thru the application of music".
This one seems just right.
If people who post to this list are advancing the causes identified in the
mission statement, then we can easily live with the diversity created by
individuals within the community who have different approaches to solving
the problems and questions inherent in moving the mission forward. If you
read your post and it moves the mission forward, you can then feel free to
hit the "Send" button.
Make sure that "membership" in this "club" is something that you are proud
of and would be happy to recommend to your close friends. Remember that
your friends are weird too and the inclusion of some of them into this club
should move the achievement of the mission statement a bit closer.
Normally I don't get into Mission Statements because most of them are
Corporate GroupSpeak half truths with no meaning. However, this small
group is VERY talented as a whole and is in danger of drifting off into
non-relevance, especially if we spend a lot of energy attacking each other
instead of helping each other.
Please provide your best alternative "Mission Statement" if this one should
be improved on (and most things can be improved on).
-Mike Mazarick
In behave of the guitarix development team I'm happy to announce a new
bugfix release: guitarix 0.22.3.
This is the 3. bugfix release after we reach version 0.22.0 as you can
properly see on the version number.
Below is a list of the squashed bugs, thanks goes to all users witch
report them to us.
version 0.22.3
fix: denormals generated under special circumstances
fix: switch off auto_startup_notification for splash window (unity)
convolver bugfix: use correct channel count
Convolver bugfix: delay and maxsize must be based on system samplerate
fix: preset_button in config mode
version 0.22.2
fix: save scratch preset before switching to a newly created one
version 0.22.1
fix: changed "requires" tag for gtk+ to 2.20 in gx_distortion_ui.glade
ladspa_guitarix: fix preset loading
ladspa_guitarix: fix module loading
ladspa_guitarix: fix loading (undefined symbols)
bugfix: wrong variable in crybaby UI
get it here:
http://sourceforge.net/projects/guitarix/
greets
hermann
Hi
This has been bothering me for a while, and trying to catch up on the
rtirq thread, I thought I'd ask here:
I'm running arch, and I used to be quite happy with the performance with
the stock arch kernel. But something happened around 3.1 (or so). Now I
get xruns every once in a while, even with conservative buffer settings.
It used to work just fine with both alsa and through jack, and now the
problem is both places, which leads me to believe that's it's got
nothing to do with my jack settings (but I might be wrong).
Before I cat my whole box here, maybe someone had a similar experience
and knows a solution. Or could extract the conclusion from the rtirq
thread, the information in there seems overwhelming.
--
Atte
http://atte.dkhttp://modlys.dk
