There are times when I'm starting up a session when my latency is too high, so I have a red dot.  And the Scarlett 2i2 (3rd Gen) Device Settings pop up window will default to Buffer Size of *880*.  I don't know how this is happening, as I change it back down to 64, hit Resync, and it will go back to *880*.

It seems to be better if I have my Ableton DAW open.  Should I open my DAW before I start JamKazam?

How do I get the Buffer size to stick?  Can I set a default of 64 somehow, as it seems to do what it wants before it calms down?

Thanks.

I'm trying to set up a stereo input in JamKazam and the solution is eluding me. It seems like I've set everything up right and I see both channels in my session as well as in the power mixer. I can pan left and right and it works through my headphones. However, everyone else in the session only hears it coming in mono. They can pan my feed left and right but they don't hear the stereo mix I'm sending. What am I missing? Any advice would be greatly appreciated. Thanks!

Ray

I'm having troubles when configuring the audio input/output in jamkazam. It worked fine in my two (very old) Macs, but it just don't work in my bandmates' Windows PCs. I tried it in four different computers, all of them with JamKazam's system requeriments. And I also tried all of them with different interfaces (iMic, Line6 KB37, MX-4BT), as well as the computers mic inputs. I keep receiving an Invalid Audio Device message and a warning about a difference in the sample rate of my input and the one selected in Jamkazam. I tried to change this in Jamkazam's sample rate box, I tried to change it in the computers' audio control panel and it just don't work. Can you help me?

Hi all,

I am trying to help a JK friend shave some ms off their audio gear latency. 

Here are some details about their current setup:

   * on Windows
   * default WDM driver
   * they have no external audio interface
   * Framesize setting 10ms (I think this is the default for WDM)
   * Audio gear latency 20.0ms.

Does anyone have experience using the the ASIO4ALL driver without an external audio interface?  It would be especially helpful to hear from you.
Does it help at all to reduce latency?

Any tips suggestions for improving latency on Windows other than purchasing an audio interface?

Thank you

So far, with just two users, our JK sessions are fine.   I do see in speed tests that bufferbloat is bad on uploads, but I'm not sure my Centurylink DSL modem/router has QOS ( quality of service ) that's all that configurable.

DSLReports/speedtest results


http://www.dslreports.com/speedtest/64914973

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Hi everyone, quick question. I have 1st port to my guitar. 2nd port to my drum machine. I am using ID14. I can hear myself and the drum machine. I can't hear anyone else in any session. 

It was working fine until I added the drum machine to port 2. It's not the mute or volume level. I have the UPnP enabled on the router.

Guess this will not work for me until I get someone to help. Frustrating the lack of support, I'd rather pay for a program that works and has support than have a "free program" that I cannot use. ugh

Does anyone know the answer?

It would be good if there was an option to set a limit to the number of people that can be in your session.

I'm finding to often I will set up a session only to have it fill up to an extent that I can no longer participate or it gets hijacked by a group of people who have low latency to each other. Based in Australia I am a long way from the rest of the world and can have a good session with one or two others say from the USA or Europe but any more I'm basically blown out of my own session.

If I could set say a limit of 4 and others wanting to join simply get a session full message try back later would be really useful.

Cheers

Mike

Anybody delved into SQM QOS on their router?  I'm just trying to solve bufferbloat.
I'm only getting it on the uploads side ---
CenturyLink DSL


https://www.bufferbloat.net/projects/blo...fferbloat/

What is the difference between "been mastered and mixed" and "a stream mix available"?
Or: Why send out two messages that say almost the same thing?

Note that this is one and the same recording.
Mysterious!

Here is the full text of the two notifications:

"One of your recordings has been mastered and mixed and is ready to share."
"One of your recordings has a stream mix available and is ready to share."

See attached image

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All,

So - after getting the most current client Windows client installed - and installing ASIO drivers for my PreSonus 2|6 USB
I was able to get a working configuration.

However - if I try to add a second channel ( like for vocal or guitar) - the audio gets a nasty scratchy sound in it.

I am wonder if perhaps this is an issue with the performance capabilities of the PreSonus 2|6. It's a USB 2.0 interface.

Not sure. Any thoughts/experience with this particular USB 2.0 device being able to support 2 audio channels on JamKazam?


I don't mind updating to a USB-C device, if that's going to solve the problem. But don't want to buy one if it's not the issue.

Any recommendations for a specific Audio interface to support two audio channels? (one for vocal and one for guitar)

Thanks,
-Paul