Not strictly a Linux problem, but I hope for advice.
We're using a Behringer X32-Rack mixer for our church PA system. One of
the inputs that we need is
for people to plug in laptops - usually via the headphone socket into a
Stagg dual DI box. We seem
to get a lot of hum pickup on that, which I think is down to the
proximity to the laptops. We have
another identical DI box at the other end of the room which takes
keyboard and bass and I've had no
problems with that. Is there an alternative method we can try? We do
raise the ground lift switch on
the DI box, which reduces the hum somewhat but still get enough hum to
be noticeable when nothing
else is playing. I wondered if the matching transformers in the DI box
are acting as pickups for the
RF noise generated by the laptops. The mixer on EQ display with RTA
shows noise across the audible
spectrum, but most of the audible sound is mains hum.
Bill
--
+----------------------------------------+
| Bill Purvis |
| email: bill(a)billp.org |
+----------------------------------------+
Hey hey,
I've done this remix for Lewitt audio's
https://www.lewitt-audio.com
mix/remix competition. They offered nicely recorded and unprocessed tracks of
Spitting Ibex's Seeds of your sorrows. I only used the vocals and part of one
guitar:
https://youtu.be/gPqIs5Q9Ypk
It's a progressive rock remix, going for the big symphonic sound. It uses
LinuxSampler with the Salamander drumkit, free Gigatron (mellotron) library,
Sampletekk/PMI's White grand piano and dulcimer from orchestral instruments,
Yoshimi (for the pad), Fluidsynth with ndbass (naturally decaying bass guitar)
soundfont, setBfree and Arturia MiniBrute 2s and Behringer Neutron for leads
and arpeggios.
Recorded and mixed in Midish and Nama.
The competition is still open until June 16 (next Wednesday).
I hope you enjoy it, feedback is welcome! :)
Best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
I saw your smile
Stay with me a while <3
(Britney Spears)
Hello list!
I am on (K)Ubuntu Studio 20.04) and needed a couple of extra in- and
outputs, so I connected 2 USB audio interfaces to my computer (a
FocusRite Scarlett 2i4 and an Edirol UA-25ex - they're old, but reliable
devices).
Through Studio Controls I set the Scarlett as master device and started
Jack, then I experienced recording and playback with Ardour in (as far
as I can hear) perfect sync.
I was amazed at the ease of this setup and the problem-free routing of
audio signals, so just for testing I split a stereo source recording one
channel on each device to one stereo track in Ardour. To my ears, the
sync is still perfect.
However, from this article
https://jackaudio.org/faq/multiple_devices.html it seems I have to
expect sync problems over time unless I do some manual configuring using
the alsa_in and alsa_out clients.
Is this still valid?
Not sure if you meant to send it privately, but I suppose putting it
publicly won't hurt.
-------- Forwarded Message --------
Subject: Re: [LAU] Hum pickup in DI boxes
Date: Tue, 8 Jun 2021 17:39:26 +0100
From: Bill Purvis <bill(a)billp.org>
Reply-To: bill(a)billp.org
To: Brandon Hale <bthaleproductions(a)gmail.com>
Hi Brandon,
On 08/06/2021 17:29, Brandon Hale wrote:
> We have the Behringer x32 at the place of my work. Why not just use a
> 3.5mm male to RCA L+R male out and plug into the aux input on the
> Behringer? That's what those inputs are for, right? That's what we do
> at work and have had success with it running small conferences with
> little to no audible noise.
>
> I hope I'm not muddying the waters more here,
>
> Brandon Hale
That would be fine, except the rack is in a separate, locked, room, and
the operating desk only has access to
XLR sockets in the floor.
Failure on my part to anticipate the need! ;-)
Bill
--
+----------------------------------------+
| Bill Purvis |
| email: bill(a)billp.org |
+----------------------------------------+
I just saw that this is probably going in the 5.14 kernel:
https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?h=f…
This is the description:
"USB-audio driver behaves a bit strangely for the playback stream --
namely, it starts sending silent packets at PCM prepare state while
the actual data is submitted at first when the trigger START is kicked
off. This is a workaround for the behavior where URBs are processed
too quickly at the beginning. That is, if we start submitting URBs at
trigger START, the first few URBs will be immediately completed, and
this would result in the immediate period-elapsed calls right after
the start, which may confuse applications.
"OTOH, submitting the data after silent URBs would, of course, result
in a certain delay of the actual data processing, and this is rather
more serious problem on modern systems, in practice.
"This patch tries to revert the workaround and lets the URB submission
starting at PCM trigger for the playback again. As far as I've tested
with various backends (native ALSA, PA, JACK, PW), I haven't seen any
problems (famous last words :)
"Note that the capture stream handling needs no such workaround, since
the capture is driven per received URB."
Curious if anyone has a setup that can backport this and see if it changes
the behavior of different latency every time you start.
If I get time to work on that this weekend, what would be the best setup?
Just patch output to input and run jack_iodelay over and over? Have
Ardour run latency test multiple times? Both?
--
Chris C
Ladies and Gentlemen,
this is some form of release.
Example first: https://laborejo.org/multichannelplayer/
Have you ever tried to convince your friends and family to install a Linux system just so they can replicate your modular setup to listen to a pre-production where they are supposed to play the saxophone solo? Me neither. But I don't think it would work. Instead:
This web software with the catchy name Multichannel_Web_Video_Audio_Player_With_Volume_Mixer is a tool for the boss (you) of ensembles and bands to distribute pre-productions for practice. But nothing keeps you from using it for real music with, as some kind of novelty player.
The user can adjust the volume levels of individual instruments and follow along a visual presentation (notation, conductor video etc.). This includes optional tracks, such as metronome clicks, spoken instructions or alternative versions of tracks.
Because it relies on simple video files there is no limit to quality and flexibility.
Works in every browser. No downloads. No obscure technology is involved. This can be used by the 60+ generation in your choir.
The provided Git/README (see link below) gives more ideas and examples why this tool could be beneficial to you.
The README also helps to setup your own player and offers a workflow to produce videos, which is quite complicated to be honest.
This is not a "proper" release because there is nothing to package. You just copy files to your web server.
License is AGPL 3 or later.
https://git.laborejo.org/lss/Multichannel_Web_Video_Audio_Player_With_Volum…
Yours,
Nils
Laborejo Software Suite
https://www.laborejo.org
Does anyone know if jackd (or the Ardour ALSA backend) can be forced to
use interleaved mode? I am trying to work around a problem where a driver
only supports interleaved access, but apparently jackd and Ardour try to
open in non-interleaved mode. In parallel I am helping someone else check
if there is some function missing that should inform applications that
only interleaved mode is supported, and someone else is going to look at
adding non-interleaved mode, but in the mean time I would like to find a
work around if possible.
--
Chris Caudle
Dear list!
I just spent a lot of time finding a good setup for telephone
interviews. In case someone else is interested for making podcasts or
live shows of some sorts, I thought I'd share the setup:
# Hardware
Laptop, smartphone, USB interface, iRig 2, microphone, headphones.
XLR cable for the mic, 2 Jack - Jack 1/4" TS cables, USB cable.
I'm switching between an Edirol UA-25ex and a Focusrite Scarlett 2i4.
Proper connection of the phone to the setup was a headache and I spent
quite some time testing a range of adapters. Most of them require an
electret mic to function, which does not go well with my setup. Thanks
to https://m.youtube.com/watch?v=wl6YuVu_v_g for pointing me to the iRig.
# Connections
Mic > USB 1
USB out 1 > iRig instrument input
iRig amp out > USB 2
iRig < > telephone
iRig set to FX
To cancel echo from my own voice the Input / Monitor selector on the USB
needs to be set almost fully counter-clockwise (almost input only).
# Software
Kubuntu Studio 20.04, Ardour 6.6, Studio Controls 2.1.64, Carla 2.3.0,
Ubuntu studio backports PPA.
# Software connections
1. Set Ardour to handle monitoring
2. Set Ardour Master to send L/R to System playback 2
3. Set up a mono foldback bus in Ardour for mix minus (everything except
the caller). Foldback sends to System Playback 1
4. Use Carla > Patchbay to disconnect System playback from pulse_out
Using a cheap Shure PG58 mic this setup is close to noiseless, even in
my bedroom home office :-)
Regards,
Alf
Hello LAUs,
Here is a list I am working on at the moment that I wanted to share with you.
I feel like I've written this already in 2012, but some stuff is worth repeating :)
https://hilbricht.net/foss-sampled-instruments.html
"This is a hand selected and curated list of sampled music instrument libraries for software samplers that musicians can use without worrying, but which are also free and open source. Through strict requirements and thorough research we hope that this list can be trusted."
(Guest appearances by Aeolus and setBFree :))
It is work in progress, but already useful, I hope.
Yours,
Nils