Hi everybody. My first mail here.
I have a question about send mtc between two different cities over network.
Something like the osx rtp-midi for linux.
I've heard that you can do it with jack, but I´m not sure if it´s only for a lan network.
Just want to know if possible before starting.
Thanks!
Hello, dear folks.
I'm still (constantly) tweaking my raspberry here and there when I
can. I have some configurations/sessions that run ok, Looper
(jack+sooperlooper), Looper+Fx
(jack+sooperlooper+rakarrack/guitarrix). But I'm still having some
annoying problems that wanted to share with you:
1) Simple session script, Looper, runs quite well when launched (via
ssh -X or from console) logged as PI user. When I run it as a
boot/init script (simple launcher script placed in /etc/init.d and
then installed with "update-rc.d launcherscript defaults 99"), almost
in every first start there's crackles and noise and my loops sound
distorted, even with clean guitar; if I restart then it goes well (2nd
or 3d time).
2) Next step forward I added Rakarrack in charge of guitar FX routed
to the Looper. First, it happens almost same thing as mentioned about
noise. Well, I guess It should have something to do with the signal
path and things connected to the same power supplier, so I'll have to
reserve some time to debug and switch off things until I get a cleaner
signal and get rid of the hum and crackles.
But the main issue is that there's problems with Rakarrack not been
able to start because several problems with not enough Jack ports and
Jack and RT. I didn't have any of these problems before.
I got to solve the problem with ports changing -p16 to -p32 when
launching jack+sooperlooper (-p24 is enough for just Rakarrack).
Then I made a launcher script LSB compliant (service script placed in
/etc/init.d and then installed with "update-rc.d servicescript
defaults", with LSB flags and functions for "start" and "stop", as a
proper newer Debian service, and configured to launch last in boot
phase) which then call the Jack-config+Rakarrack+Sooperlooper session
launcher script. I can see on screen in the init messages this error:
"JACK is running in realtime mode, but you are not allowed to use
realtime schedule..."
But I know RT is already configured, so I guessed it was something to
do with the boot/init stage configuration and environment.
I tried using "su" with several different parameters to run the
launcher script as "pi" user, as It runs quite well when logged as
this user (calling just the script and even with "sudo
/etc/init.d/script start").
So, maybe you can't get RT in init phase on RaspberryPI and you can
only when you get to the login prompt and everything is loaded and
ready. But I think there's a "Puredata on Raspberry PI" project that
runs some scripts in the init stage; and it seems that in order to get
a headless FX+looper station, running some configuration+apps-launcher
at boot time is an obvious choice.
I'm sure something escapes me after so much try-error-code cicles.
Maybe you have some ideas.
Thanks as always.
--
Carlos sanchiavedraz
* Musix GNU+Linux
http://www.musix.es
On Tue, July 16, 2013 4:54 am, Julien Claassen wrote:
> Hello Paul1
> Did you change it again or was theLAN thing only in either of the
> twoJACK
> versions? I definitely did want to try and we had to set up a VPN tunnel
> to
> use the JACK net driver. Maybe I'm just antiquated in my knowledge.
At least one end has to have a "visible" IP. Physical distance of the two
ends affects the latency, as do the switches/routers in the path. The
lowest bandwidth link in the path determines the maximum bandwidth of
information a netjack link deal with. Obviously MTC is low bandwidth, but
MTC without some other information (either a midi or audio stream) is of
limited value :)
--
Len Ovens
www.OvenWerks.net
Hello everyone!
I'm very pleased to announce that my second album is finished and online! :D
It's called "Ordinary Day Montage", and it consists of 8 tracks, and is a
bit more electronic than last one.
I prepared a page for this at my new website,
http://zthmusic.se/Ordinary_Day_Montage , but my host is a little shaky, so
it might not always work. So, if that link doesn't work, you can find the
album at:
Bandcamp
http://zthmusic.bandcamp.com/album/ordinary-day-montage
Soundcloud
https://soundcloud.com/zthmusic/sets/ordinary-day-montage
FLAC/OGG/MP3-formats for download at Piratebay:
http://thepiratebay.sx/user/zthmusic/
I'm very excited to be finished and to have completed this. Everything was,
as always, 100% made with Linux and Linux software. It's also licensed
CC-BY-SA. I wrote a bit about the album and what I've used technically too
at http://zthmusic.se/Ordinary_Day_Montage , if anyone wants to check that
out!
Anyway, thank you for taking your time to listen! I greatly appreciate it!
>Hello!
> I need to get a vocal microphone. I've had this project several times
>before, but now I'm decided. There are a couple of alternatives. there
is the
>Shure SM5x and there is the SE-220, if my memory doesn't betray me.
Any more
>suggestions on the topic? I want to pay around 100-200 EUR maximum. This
>microphone will be meant solely for singing. No instruments to mic.
> Warm regards and sorry for the OT again
> Julien
Hi Julien,
I think it's the Audix OM-6 that I have here. It is nicely built and has
a lot of clarity,
maybe too much presence in some cases.
Also, check out the Heil PR-20 which I think is a little more neutral
sounding and less expensive.
And the AKG C700 is a very good value. As with most mics, with a good
voice it can sound great.
Grekim
Hello everyone!
this is perghaps not straight on topic, but I couldn't find a satisfactory
answer anywhere else.
I'm having trouble with my JV-1080. Only every second note gets played. I
checked different input devices and different cables, direct and indirect
connections.
It seems to be a problem of the Roland itself. Internal sounds, demos and
the preview button, work fine. Any idea, what this might be? It looks so
systematic, as if it might be by design, but I don't know any feature, which
should allow for that.
Sorry to spam the list and thanks for following me so far.
Warm regards
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
Hi list,
I know this is a contravercial subject, but I'd like to hear some viewpoints.
I use a delta 1010lt card. I'm wondering if I should carefully set my
input levels for each input source, or, if the digital convenience of
normalizing and dc ofset makes up for level defficiencies, without
adding noise.
I know the concern here is signal to noise ratio.
Would my delta add noise if I boosted input levels initially or, in
normalizing, am I just boosting the audio with the snr which I would
have recordedif I had boosted my input levels in the first place?
Does that make sense?
Thanks!
Rusty
Hello!
I need to get a vocal microphone. I've had this project several times
before, but now I'm decided. There are a couple of alternatives. there is the
Shure SM5x and there is the SE-220, if my memory doesn't betray me. Any more
suggestions on the topic? I want to pay around 100-200 EUR maximum. This
microphone will be meant solely for singing. No instruments to mic.
Warm regards and sorry for the OT again
Julien
----------------------------------------
http://juliencoder.de/nama/music.html
I guess it was off-list by accident?!
-------- Forwarded Message --------
From: Ralf Mardorf <ralf.mardorf(a)alice-dsl.net>
To: Rusty Perez <rustys.lists(a)gmail.com>
Subject: Re: [LAU] maximum input level, or normalization and dc offset
correction?
Date: Fri, 12 Jul 2013 20:40:23 +0200
On Fri, 2013-07-12 at 11:20 -0700, Rusty Perez wrote:
> My question was really more of a question to understand what unwanted
> noise I may be adding to my recording, which I'm not hearing, either
> because of deficient equipment or deficient hearing. :-)
A valid question, but don't care about the dynamic, the real issue with
mastering music that should sound as good as possible on as much
equipment as possible is the whole mix regarding to frequencies, phases
vs mono and stereo.
It's unlikely that noise will become an issue, it's more likely that the
frequencies are biased or that there will be phasing between the
channels.
What faders do you use? The analog inputs have an optimized working
point, perhaps the digital side then is too high or lower than
"optimal". I don't know what positive or negative effects are caused by
digital remachining. Keep the level below 0 dBFS, resp. add headroom.
Too high in all cases is bad, but even a real too low level might be
inaudible.
"I never normalize the tracks in the DAW when mixing. I can't do it with
my analog machine, so I don't do it with my DAW." - Ricardus Vincente
and we can use the volume control of our amps to make the music louder
and quieter. However, optimal leveling at recording time is better, than
postprocessing. Analogy: If a recording does miss frequencies, you can't
raise the missing frequencies, you only can rase frequencies that are
there, but to silent, but this will come with side effects.