Setting up new audio gear, I was getting the error message:

   "Your input device samplerate (44100) and output device sample rate (48000) are assymmetric"

While the "New Gear" setup page in Jamkazam has a dropdown menu where you can adjust the output sample rate, or choose to "match" input and output sample rates, the functionality does not seem to work.

MAC
On a Mac (I'm on MacOSx 10.13), I found that the input and/or output sample rates can be adjusted using the:

   "Audio Midi Setup" app

... which was already present on my Mac.  I found the app using the "Spotlight" search function (upper right corner of screen).  Since my input device supports a sample rate of 48000, I set it as such to match the output device -- and voila, no more sample rate errors!

WINDOWS (using ASIO)
On Windows 7/8,8.1/9/10; open the dedicated ASIO Control Panel by going 'Start' > search for 'ASIO Control Panel' > open this. The sample rate can be changed here.

I found these solutions on a Focusrite support page ( https://support.focusrite.com/hc/en-gb/a...-interface ).

Hello

is there a benefit to have the person with lowest or highest latency being the host of the session ?

Our drummer has the worst latency of the band.  (Total latency around 40ms, when I hover on his icon in the session)
The host is generally the keyboard guy, with about 30ms Total.
Bass/vocals and Guitar(me), we have in the 20ms Total Latency.

Whenever I start a song and the drum needs to join after a few bars, he gets in a bit late... after a few more bars, we usually "rejoin each other"... 
But wondering if a simple host switch would help.

JP

My wife and I are testing Jamkazam.
Were both using Mac pros with Focusrite audio interfaces.
Were able to play together fine but the audio recording has a delay.
Does anyone know how to alleviate this issue?
Thanks for your help,
Christopher

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.