With the latest version of JK I get a message that the audio setup is not configured for best latency.  The report gave me some recommendations for my Focusrite Scarlett 2i2 Studio (3rd Gen.) which I have been able to implement, but one is stumping me: "Average Audio Latency: Audio input/out latency is 7.45833 ms. Tuning or changing your audio device could reduce latency to less than 5 ms."  I have the latest driver (Focusrite Control 3.6.0), the framesize is 2 ms and the maximum network packet rate is 400 Pkt/s. 

I see that this latency is considered "Average" as opposed to Excellent or Poor.  Is this just JK's way of advising me that I have a low end audio adapter (which is otherwise working adequately for my needs) or is there a problem here that should be addressed?

I work for a school district and we are interested in purchasing several paid subscriptions for some of our students. Does anyone know who I could contact to purchase a block of licenses? I don't want students to have to subscribe to one of their own for a school-supported application.

thanks

B

JamKazam was working well for 2 of us in Dec. Then, in the middle of a session I was unable to hear my friend. He could hear me and himself and I could hear me. We both rebooted without success. There is mild latency: 12-16. We are both hard wired although on my computer it always says he is on Wifi.
We are both using Mac desktop with Artesia a22xt.
Any advice for next step? Until we can get this to work we will hold off on paying monthly, but will begin when problem resolved.

Hello! Each time I want to start a session, I get "disconnected from server". I can't change it. What I'm doing wrong?
Thanks for your answers. Gerry

When trying to submit a crash, this happens:     

I also can reproduce a crash when changing the "Audio Devices" in System Preferences, e.g., the Format to 48,000. Using Big Sur 11.0.1. I uploaded the crash log here.

2nd issue, my singer has intermittent problems and gets the "not receiving audio packets message" and I see this alert:
   


What's the deal here? Why would one pay for this unreliability?

I and a friend setup a session - we each had 2 independent inputs to send to each other, e.g. vocals and guitar. Each local musician shows both tracks but only shows one track for the remote musician. This track is only mono so the two channels cannot be separated. I cannot find any options to do this.

I am currently on the free plan and would consider upgrading to get this, but found this when in the trial period (which is supposed to be gold plan).

Does anyone know how to do this?

I was in a two person session, where we were getting intermittent loud bursts of static/noise that were bad enough to make us stop.

My Audio configuration report says "1. BAD Driver.  Unpredictable latency".   I'm not sure why.   I have the latest Solo driver 4.63.24 with firmware 1052 (not sure if that's the latest or not.).   Edit-- Found there's a new driver, 4.65.5.   Will try that  EdIt 2--- this new driver shows up as 21.0ms latency in a solo session ,where the 4.63.24 driver showed up as 0.  I don't understand this at all.


My friend's report had "bad sample rate 44.1" on his Presonus 96.  My sample rate is 48000, buffer size of *120* according to .    We are only 5-8 miles apart geographically.

All the other reported settings were 'excellent'

He will try to adjust his sample rate to 48k, though his "Ultimate Control" panel was missing the controls for this.

Where should we focus our attention to address this?
Thanks.
Jim

Understanding Frame Size:
the calculations to get theoretical interface latency from Buffer size/No of samples/second  and Sample Rate khz( audio quality) khz isn't difficult:
Buffersize / Sample Rate X 1000=Latency ms
Also  the cpu load as expressed in buffer fills is
Buffer fills per second (cpu load) =Sample Rate / Buffer size

Anybody turning on the compressor in the Pro audio mixer window?  Adds to latency?

   

For your own backing tracks used in a session ( not jamtracks) , 3 of the main file types you can use are WAV, OGG and FLAC.  These usually vary quite a bit in size and quality.  I've been using FLAC but I'm wondering how quality or file size impacts the performance in JK.  Specifically in terms of glitches or being able to lower Audio FrameSize below 2.5ms.

In my experience 2.5 is the minimum for glitch-free playback of backing files.
WAV files are too large, would I be better off using OGG at around 224 kb/second, which is better than most mp3s and only a fourth the size of my FLAC files.