I have a Zoom H1n recorder with a built-in USB audio interface. I'm
trying to hear the input from its microphone in real time on my
computer, while hearing output from an audio player (audacious) at the
same time.
My OS is Fedora ThirtyOne, running on a ThinkPad.
I've managed to do this with JACK and Audacity using the software pass-
through setting, but there is a noticeable audio delay from the Zoom.
In Ardour, I can't seem to connect an audio track to the Zoom.
Any suggestions to accomplish this?
Hi.
For years, I used an RME HammerFall under Linux. I love this card, but
I finally goota say goodbye to PCI. What USB/ThunderBolt multichannel
card with mixing/routing capabilities would you recommend? I have been
out of the loop regarding semipro soundcard for many years now, so I
would be happy about some tips that get me started on choosing a device.
I mostly need analog channels. Ideally with a lot of headroom, since I
might be feeding it with quite hot signals (eurorack).
I have heard there are also mixers these days that work as a
multichannel soundcards. Maybe that might be interesting.
However, I will only be using it with linux, so I am really looking for
something that is known to work nicely.
Will be using it with JACK and SuperCollider.
--
CYa,
⡍⠁⠗⠊⠕
Hey hey,
after a long time I have just gone back to the LS SFZ documentation and found
it to be quite powerful and feature rich. But there are two questions I
haven't been able to solve yet:
1. Does the osccilator keyword really work and if so in which section (global,
group, region,...)? I get asn error message using it with LinuxSampler 2.1.1
(the ArchLinux package).
2. The waveguide keyword: how are other audio sources sent to it? By
lokey-hikey definitions or does a group have to have two regions with a sample
as source and a sample as waveguide?
As a question of secondary importance: in conjunction with the wavetable
oscillator is the oscillator_table_size honoured? I tried it with the
Adventure Kid WaveForms (AKWF www.adventurekid.se) and set it to 600 (600
samples per wave cycle), but still had to tune the resulting sample
accordingly by two semitones and a few cents.
Best wishes and many thanks,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* SoundCloud: https://soundcloud.com/jeanette_c
* Twitter: https://twitter.com/jeanette_c_s
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
And I love the way with just one whisper
You tell me everything <3
(Britney Spears)
Dear List,
I will be giving a talk and demo on
"Dynamic 3D Audio Scenes with ASDF + SSR"
at the Audio Communication Group at TU Berlin:
November 19th, 16:15
Einsteinufer 17c (Raum EN 324)
https://www.ak.tu-berlin.de/menue/fachgebiet_audiokommunikation/adresse_und…
See you there!
Matthias
---
Abstract:
Presentation of an open-source software for real-time spatial audio
reproduction called SoundScape Renderer (SSR). Review of available
reproduction methods, focussing on binaural rendering and Wave Field
Synthesis (WFS). Discussion of ongoing work, e.g. extension to 3D (only
binaural renderer for now) and a browser-based 3D GUI.
Presentation of the Audio Scene Description Format (ASDF), a format for
dynamic audio scenes which is currently under development. It allows for
moving sources along pre-defined 3D trajectories. Open questions and
future work will be discussed.
Time permitting, there will be a demonstration of the SSR running on a
loudspeaker array in one of the studios @ FG Audiokommunikation.
https://github.com/mgeier/fk2019
Hi all,
I've originally posted this issue on the Linux Musicians forum, and despite
having received several suggestions, we were not able to solve the issue.
(for those interested in the previous thread:
https://linuxmusicians.com/viewtopic.php?f=27&t=20629)
I'm an experienced Linux Audio user and have utilised a fair amount of
soundcards on Linux systems.
I've read most of the Scarlett 2nd Gen. post I could find here and
elsewhere, and noticed there's little info about the 3rd Gen.
I'm taking this chance to share my experience with a 18i8 3rd Gen. and see
if my problem can be solved.
SPECS:
I'm on a DELL M4800 Precision, Linux Mint 19 with Cinnamon, kernel
4.15.0-66-lowlatency.
Soundcard: Focusrite Scarlett 18i8 3rd Generation. I'm in the audio group
and the audio group has access to RT.
After the necessary tricks to force the soundcard out of MSD Storage mode,
now the device is recognised by and fully functioning with both Pulseaudio
and Jackd. However.....
PROBLEM:
When using the soundcard (either via Pulseaudio or via Qjackctl), I can hear
click noises at almost regular intervals. The clicks vary in loudness (some
are very loud, other almost unnoticeable). Independently from the CPU or
Disk usage, the clicks are always there, making the soundcard effectively
unusable.
Interestingly, when using the card via Qjackctl I see no XRuns, but I do
clearly hear the clicks.
ACTIONS TAKEN:
I excluded a defected soundcard, broken USB cable or USB port issues
(sometimes USB3 can cause troubles) because when using the soundcard on a
Windows partition on the same computer, the sound is clean, no clicks.
Although, perhaps I'm missing something here.
Running realTimeConfigQuickScan gives me a mostly positive check, the only
"not good" items are the following ones:
- Checking swappiness... 60
- Checking checking sysctl inotify max_user_watches... < 524288
- Checking access to the high precision event timer... not readable
- Checking access to the real-time clock... not readable
- Kernel with Real-Time Preemption... not found
these "not good" items have always been the same on my laptop, and they
never gave me troubles with two other Scarlett soundcards. I've also read on
the ConfigQuickScan page that - except Swappiness - the other items are not
so important these days any more. So I'm assuming the clicks with the
Scarlett 18i8 3rd Gen. are likely not related to any of those items. But I
may be wrong...
I've also tested with 48000 and 44100, no difference.
Does anyone have any hint on the issue? Or other things I could check?
thanks in advance,
--
Sent from: http://linux-audio.4202.n7.nabble.com/linux-audio-user-f5.html
Dear linux-multichannel-audio-users,
we are running a setup with two RME MADI cards to drive a system with
approx. 100 playback channels. We are using jack2 and zita-j2a to
achieve a sufficient number of playback channels. The sample clocks are
synchronized via word clock, therefor the resampling is deactivated in
zita-j2a.
On a similar setup (with two different card types) I noticed that the
latency between the cards is fixed as long as jack/zita-j2a is running,
however, the latency differs between starts of jack. This seems also be
the case in the setup with two MADI cards.
Is there a way on the software side to solve this problem? Or is there a
way to measure the latency without access to the analog I/O (knowing the
latency every time would be sufficient)? How are other people solving
synchronized playback of >64 channels?
Best,
Giso
Hi, all. I'm considering replacing my M-Audio Delta 1010 an Mackie Big Knob
monitor controller with an RME Babyface Pro so that I have preamps available
at my desktop without breaking out my big mixer, and hardware volume control
for my monitors.
I know a few folks here have the Babyface Pro (Hi, Fons and Brent!) and
I'm curious as to whether you're all still happy with it. I've been reading
that in the class-compliant mode, which is required for Linux use), that
various features are not enabled.
My biggest concern is about phantom power, as I'd seen a mention that
PP is only settable from the TotalMix software, which is a no-op in CC mode.
I'm also interested in any "gotchas" or anecdotes about living with this
unit under Linux.
Thanks!
--
======================================================================
Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com
Without deviation from the norm, "progress" is not possible. - FZappa
Hello all,
First post so obviously a newbie question. And this, I know, is a question
without answer, as it's a bit subjective. So what is the most convinient
distro to get started on.
I've previosuly been using an Ubuntu distro which worked up to a point, that
point being 'rosegarden'. I really wanted to play with rosegarden so I could
enter musical notation and list to the results. That was a step too far for
me. I could get qsynth installed and get it making a noise by connecting the
qjack keyboard to it in qjackctl, but rosegarden would never play a note. Just
could not get it connected up and happy. QJackCtl is fairly straight forward
if you have things to interconnect. If there's nothing to connect to when you
start up rosegarden you're kinda not going anywhere.
So that was Ubuntu, but this laptop, which I've just returned to is running
OpenSUSE Leap 15.1, which don't have, or doesn't appear to have, a low latency
kernel option. That might mean a return to OpenSUSE Tumbleweed, or other. In
most cases Ubuntu would be the easy way to start but I've already been down
that road and rosegarden just never showed up in QJackCtl.
I should say that the specific multi media distros seem to have fallen a bit
behind so they don't look like an option. So perhaps later in the week
Tumbleweed with GNOME and try to get a note, just a single note.
I like to announce the initial release for Xmonk.lv2 version 0.1
Xmonk is a sound generator based on the Faust `SFFormantModelBP` from
physmodels.lib by Mike Olsen.
It's a monophonic formant/vocal synthesizer.
The source is a sawtooth wave and the "filter" is a bank of resonant
bandpass filters.
Xmonk provide a interface to drive the filter/synthesizer via mouse
movement.
Additional Xmonk provide a midi in port to drive it with any midi
controller.
On top Xmonk provide a virtual keyboard to drive it with your PC
keyboard/mouse combination.
Surely you could use your DAW automation to drive it the way you want.
Xmonk could play in two different modes, on is the usual note on/off,
the other is sustain, means the last note play forever, while you play
with the vowel filter settings.
Xmonk could be used with a free scaled temperament, or as selected with
Equal Tempered scales from 12 to 53.
Here you get it:
https://github.com/brummer10/Xmonk.lv2
So, have some fun with it,
hemrann
