Hello,
I've been having this issue with zita-a2j launching into 100% CPU usage
fairly frequently when I am running Carla, one VST (Audeze Reveal Plugin),
and using Ubuntu Studio 19.04.
Has anyone else seen this kind of behavior before and knows what might be
happening?
--
Scott
http://www.scottcazan.com
Twitter: @scazan <https://twitter.com/scazan>
Instagram: @scazan <https://www.instagram.com/scazan/>
Hello People,
I have a strange problem with high latency for both JConvolver or BruteFIR.
My Jack settings are low(3ms) and stable and the latency doesn't seem to
tally with the partition size?
My Hardware:
i5-8259U - Intel NUC
8GB DDR4
PCIe SSD
M-Audio - MTrack8 USB Sound
Software:
Ubuntu Studio 19.04 - Have tried updated and fresh install also a blank
Debian 10.1
Jconvolver 0.9.3-2
BruteFIR 1.0o-1
Jackd2 1.9.12
Settings:
Jack Settings - 48KHz 64Buffer 3Period
BruteFIR - 7 Filters each 8192 Taps, These are each split into 512sample
partitions
Jconvolver - 7 Filters each 8192 Taps, These are split into 64sample
partitions
Results:
JConvolver Latency = 85mS, CPU Load = <5%
Brutefir Latency = 104mS, CPU Load = 55-60%
These latencies are measured inside Jack(Not using sound card IO) using a
jack signal generator and a jack oscilloscope. But also verified externally.
I dont understand why I am getting such long latencies, I would be very
grateful if anyone could shed some light on why the latencies are so long
and even more grateful if you could share how to fix it!
Thank you in advance,
Alex
--
Sent from: http://linux-audio.4202.n7.nabble.com/linux-audio-user-f5.html
Hi all.
Tomorrow it's the scheduled meeting at c-base again. I don't know if
I'll make it (I've stayed home from work today not feeling great), but
maybe Louigi or Robin can gather people?
Cheers
/Daniel
Hello All,
i have a special little audio problem with kdenlive which is annoying me
a lot, perhaps you can reproduce it or have an solutiuon:
I like to do video cuts exactly synchronized with music song beats. So i
would like to "tap in" the beats with markers before cutting to have the
cut positions to snap to. That's not possible on my system, i always
have about 500ms of delay when i listen to the music track and tap in
the beats. So the markers are always a little to late and out of sync...
Can you reproduce this? Do you have an idea?
I am on Arch latest kernel, KDE, Ryzen System, tried it on Pulseaudion
and Firewire external Audio Interface, Same Problem.
Thanks for ideas, Daniel.
--
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Daniel Wilhelm
Hello!
I am familiar with this error you described. What is the SSD's file system
formatted to? I get the same error when writing to my NTFS partition. I
can't remember where I read this but it is best to record to ext4 or
another more Linux friendly format.
Hope this helps!
Best,
n!
Hey hey,
I have had issues with Aeolus for a while now, but always let it go due to
other priorities.
Having attacked it again, I find that this problem persists both in the
installed package from Arch Linux, version 0.9.7 and a freshly downloaded and
compiled tarball from kokkinizita.
It seems that Aeolus segfaults the moment that it calls rl_initialize, gdb
giving a point in libncursesw as the last stack frame before shutdown.
I don't understand the problem since all other programs running with ncursesw
and/or readline alone work perfectly.
I'd be glad for any pointers to further investigation or even straight
solutions.
Thanks and best wishes,
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
You should take me as I am
'Cause I can promise you
Baby, what you see is what ytou get <3
(Britney Spears)
Hi LAU,
I have anecdotally, but recurrently observed on laptops - including a
rather recent one - that there is always a 'best' USB port for external
sound cards.
For instance, on my latest machine with a decent 'realtime audio'
configuration/set-up (real-time kernel, /etc/limits stuff, 'performance'
CPU governor, (wireless) network switched off), I'm able to have a
pleasant xrun-free session recording in Ardour including a bunch of
tracks with effects playing at 64 frames and period of 3 with a
relatively cheap card (UMC202) on one of the USB ports.
On the other hand, in the exact same conditions I get incidental xruns
at even 128 frames and xrun instability at 64 frames on the other USB ports.
I wonder:
1. Is there a more scientific (well, precise at least) method to assess
this USB port performance? What to test or look into?
2. Is there a way to change (e.g. improve the not-so-good USB port
performance) OS/software wise, or is this usually hard-wired in notebooks?
2a. Are IRQs relevant on laptops and if so can a whole USB port (or
the device attached to it) be optimised from the OS?
Of course I _can_ live with one 'good sound-card port' on a laptop but
I'm quite curious about people's experiences and the gurus' wisdom -
albeit on my former machine this was the left-side port which was closer
to where the sound-card usually sits, now it's on the right, too bad! :)
Hopefully other LAU have mused about such USB-related mysteries in the
past...
Lorenzo.
Hi,
I have no problems with xruns and that stuff during recording with
Harrison Mixbus, but rarely (but very annoying when it happens), Mixbus
complains that my system can't catch up.
I'm using three SSD disks, one for the system, one for audio projects
and one for my sound libraries. I can usually record on very short
latencies when also playing or trigger software players or synths
without getting into trouble, but occasionally, Mixbus32C (and even
Ardour) says that the system can't catch up and the recording is
interrupted. This is not very fun when I'm in a creative mode and need
everything to flow, I normally make sure that every track I hear is a
wave track as much as possible in order to avoid this problem and my
SSDs are Samsung EVOs.
So here is what I'm thinking and wondering: Is it possible (or any idea)
to record into a RAM file system while recording and let the OS shuffle
everything into the SSD during less busy periods? I have 32 GB RAM so
it's no worries when it comes to RAM size. If it might be an idea,
what's the best way to do it?
I might have to look further into IRQ's and so on but can't see anything
else that the system is very optimized. I have a i7-4790K CPU@4GHz, the
soundcard has exclusive access to IRC 19 and I have done all the
ordinary audio settings an so on. So for now, I'm really curious to find
out if a RAM-solution is anything good or not. What do you think?
Jostein
