Hi all!
I'm running reaper 6.05 on a raspberry pi 4 running raspbian/debian buster. I can't get rid of the last 5ms of latency between the midi note send and the trigger of my midi-to-cv interface (controlling my modular).
I played around with buffer settings in jack and the delay gets smaller as I decrease the buffer. However, no matter how low I set the buffer size, the "offset output to this device" in the preferences will not make the delay less than 5ms.
My setup is like this: reaper -> midi mate ii (usb midi interface) -> midi splitter -> cv.ocd (midi to cv interface) gate out -> soundcraft mtk 22 (mixer with usb audio interface). I also tested with the midi splitter out of the picture, same results.
I realize there will always be some delay, what surprises me, it that I can't adjust the delay so lo that the midi note and the recorded trigger align. With offset at 0 I get 15 ms delay with "reasonable audio buffer", adjusting the offset even to -100ms will still give me the trigger 5ms after the mid note. I'd expect -100ms to over compensate to the point of the trigger being recorded 85ms early...
Any input would be highly appreciated!
https://www.dropbox.com/s/hlvmls7c850i2ia/midi_delay_reaper.png?dl=0
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Atte
http://latestyoutube.a773.dk | http://a773.dk
Hi list,
I guess it's a known issue that zita-a2j will eat all memory if the
device it was bridging was disconnected. Is there a way to make it
handle disconnects more gracefully without taking down the system after
some time?
m.
Hello,
using zita-njbridge between two x86_64 systems works great. However,
when I try to connect an arm-based system (Banana Pi M1) with a 64 bit
system things get less stable (dropouts even with large buffers, error
messages, segmentation fault on the 64bit side). I tried the Ubuntu
18.04 provided version (0.1.1) as well as a freshly compiled version
(0.4.4). Did anyone successfully try zita-njbridge on arm-based systems?
Giso
(when the segfault happens it is always in netrx.cc, line 264 in the
function write_zeros)
Hi list,
I am running a telephony daemon on a laptop that gives no audio
interface configuration options and I wish to use it with Jack. I'm
confused as to whether or not it is using ALSA. If Jack is already
running (with pulseaudio-module-jack uninstalled) and configured to use
an external USB sound card, starting the daemon results in its audio i/o
being connected with the laptop's internal mic and speaker. If Jack is
not running, the daemon connects with card 1 listed by "aplay -l", which
happens to be the USB card. However, if Jack is running with
pulseaudio-module-jack installed and all other non-jack audio apps are
routed via the sink to the USB card, the telephony daemon again connects
and runs correctly with laptop's internal hardware.
I don't think ALSA-Jack loopback scripts will work as I think
pulseaudio-module-jack does this job now. Is there a solution with
kernel modules? Can someone please put me in the right direction?
Thanks!
Iain
