Hello guys,
I'm having some trouble setting up my vst's..
When I load Bias FX2 Jamkazam crashes

When I load Bias Amp 2, it doesn't work... I mean, I can hear my direct signal input but it is not being processed by bias amp2..

I don't know if it may have something to do with my audio routing, but actually in earler versions of jamkazam I was able to use at least bias amp

here's my audio routing
guitar->mic preamp (Neve 1073) -> RME ad/da converter ->optical out->focusrite saffire pro 24 optical IN -> Macbook pro through Firewire port


Now, as I said I can hear my direct signal, so I suppose that the audio device is set correctly...

any advice is really appreciated

Is there a way to get more info on upcoming sessions?  In particular who posted it and their profile?

When I'm in a session I can hear myself, but I can't  hear the other person. I can see the audio levels, though.The other person can see and hear me. My gear hasn't changed at all. The other person has a new interface and a new mic, and has updated all the info in JK. We both have the latest JK version. Any ideas?

When I set up my chat mic (using my laptops builtin "mic array") in JK, it sounds horribly distorted.  It's not clipping, and it happens at any level.  It sounds more like lots and lots of small frames are being dropped so there's a constant background of chopping clicks.  (Anyone who's done a bad slice knows what this sounds like, except that's one click and this is happening constantly, but not regularly enough to have a pitch.)

My mic is working fine, by itself.  If I use Windows control panel to listen to the mic, it sounds fine.  The mic has a number of processing options, but the noise in JK happens with or without them.

I get this regardless of whether exclusive access is enabled (with priority to exclusive access.)

The device name in JK is "Microphone Array (Realtek High Definition Audio) - (Windows WASAPI)".  The other options are also WASAPI, but are my audio interface or Stereo Mix.

I'm not yet sure how well the automatic muting/unmuting of chat works, but it seems to me that it would be very handy to be able to mute/unmute the chat mic (or any selected channels) with the space bar.

Hello everyone,

I've plugged in my Edirol UA-101 interface in Jamkazam for the first time today.

I've tweaked the ASIO settings for buffer size and I'm on ultimate power settings, but I just get terrible latency numbers:

Latency: 73.86ms
I/O rate 397
Var 5.81

Latency was a whopping 350ms+ before tweaking everything.

Where it gets extremely odd is if I switch to WDM, which should be slower, I then get:

Latency: 20ms
I/O rate 100
Var 0.32

ASIO works fine with Cakewalk, I get no latency. I'm expecting latency in the low teens or even below 10ms.

I've tried it on my other (older and slower) windows 10 PC and I get the same problem.

Does anyone have the same issue? Or any pointers at to what I'm doing wrong?

Thanks for looking

Fred

I play different instruments, and my audio setup is often different.  Sometimes it's guitar on one channel and vocals on another.  Other times it's stereo keyboards with built-in mic for voice chat.  Or I may have my guitar but no mic set up, so I need voice chat.  No doubt I'll have other options in the future.  I'd rather not have to edit my channels each time I join a session.  I'd like to have my favorite setups at hand, with a name I provide, in the list of audio profiles.

Currently it seems we only get one profile per audio device, and the name is the name of the audio device.

The simplest solution might be to add "Save settings as ..." next to the "Save Settings" button in the "Configure Tracks" panel.

My band is interested in using JamKazam, but before we make the attempt to set it all up - has anyone had success jamming across the country? I'm in Boston and trying to play with my bandmate in Texas. Or will there be too much latency regardless of how strong our network connections are?

The Youtube videos show people practicing with low latency - but it notes that they are all located in the same state..

Thanks in Advance!

b34

• Probably not related but I wonder what auto throttle on the Audio Booster page is doing to the flow of data out into the internet.  Could it affect jitter ( also known as buffer bloat)?It's interesting that setting auto throttle on the network packet size grays out the individual choices but auto throttle on the bit rate doesn't gray out the individual bit rate choices.... I'm getting good latency now on the 1 millisecond experimental frame rate choice but when I used that a few weeks ago it Interfered with playing back jam tracks or my own audio files.... Other musicians in the session we're hearing Interference but they didn't hear that at 2.5

I'm using a hardwired ethernet connected Mac, OS 10.14, with a Steinberg UR22, and when joining a session with another musician who invited me, we both receive an error message that streaming audio may not be reaching others and to try to reboot the router or optimize my network. Is there a legitimate reason for this and what is the solution? Do I really need to reboot the router? How do I optimize my network?  

When joined into a session, my signal is greyed out.

Any help is appreciated.