Buffer size of UMC202HD automatically switched to 8 samples

[behri.stein]

Currently, I am using a Behringer UMC202HD under Windows 10 as audio gear. In the audio gear setup, I choose the UMC ASIO Driver, go to the ASIO settings, choose a buffer size of 64 samples but whenever I push the Resync-button, the buffer size switches back to 8 samples (resulting in a pretty good latency but a very poor sound quality). I am not able to switch to anything else than 8 samples since Jamkazam always switches back. Can anyone help?

[jamjam]

Currently, I am using a Behringer UMC202HD under Windows 10 as audio gear. In the audio gear setup, I choose the UMC ASIO Driver, go to the ASIO settings, choose a buffer size of 64 samples but whenever I push the Resync-button, the buffer size switches back to 8 samples (resulting in a pretty good latency but a very poor sound quality). I am not able to switch to anything else than 8 samples since Jamkazam always switches back. Can anyone help?

The JamKazam app captures frames samples in the same about needed to encode a audio frame. This is about 110 samples at 44.1 KHZ (or multiple thereof) or 120 samples at 48Kz (or multiple thereof). When it requests that number of samples from the audio driver, the driver switches to the closest frame size will allow that capture size. If you set the input sample rate to 48Khz based it usually retain the value as opposed to 44.1Hz.

[behri.stein]

Currently, I am using a Behringer UMC202HD under Windows 10 as audio gear. In the audio gear setup, I choose the UMC ASIO Driver, go to the ASIO settings, choose a buffer size of 64 samples but whenever I push the Resync-button, the buffer size switches back to 8 samples (resulting in a pretty good latency but a very poor sound quality). I am not able to switch to anything else than 8 samples since Jamkazam always switches back. Can anyone help?

The JamKazam app captures frames samples in the same about needed to encode a audio frame. This is about 110 samples at 44.1 KHZ (or multiple thereof) or 120 samples at 48Kz (or multiple thereof). When it requests that number of samples from the audio driver, the driver switches to the closest frame size will allow that capture size. If you set the input sample rate to 48Khz based it usually retain the value as opposed to 44.1Hz.

Thanks a lot for your reply. In fact - I already used a sample rate of 48 kHz and the buffer size gets switched to 8 samples whenever I press the resync button. So the sample rate does not seem to be the solution.

[Pete Savigny]

I have the same problem.
I was told to set the buffer size to 256 or so, in order to solve my problem of not hearing my co-musicians' sound, whereas they hear mine perfectly.

Has that been your problem as well?

Have you found out how to change the buffer size without JamKazam reverting to 8 samples upon resync?

[Michael Dolak]

I also have the same problem and it drives me crazy (actually with a UMC204HD):

When I connect with another musician just via audio everything works very smooth - but as soon as I open the video stream I get sound distortion. I guess this is a latency-issue and I tried to change the ASIO buffer size but it always changes back to 8 samples upon resync.

I really hope somebody can figure out a solution because rehearsing and jamming isn't possible for me without a proper video feed (if in fact my problem is latency-related)...

[Pete Savigny]

I also have the same problem and it drives me crazy (actually with a UMC204HD):
===
Hi Michael, 
that is good for me to hear, because I was wondering whether exchanging my UMC202HD and getting the UMC204HD could also solve that problem.
===
You write:
When I connect with another musician just via audio everything works very smooth -
===
It is also good to hear your audio works smoothly - in spite of the buffer being reset to 8 samples upon resync. That seems to mean that the fact that I do not get to hear my co-musicians, whereas they hear me perfectly, should not be caused by this reset to 8 samples. 

Maybe you can help me find out whether my UMC202HD must be at fault, and I will have to exchange it, or whether at least the audio can be made to run smoothly for me as well?

So I actually wish I just had the problem with the video stream. 

You write:
===
but as soon as I open the video stream I get sound distortion. I guess this is a latency-issue and I tried to change the ASIO buffer size but it always changes back to 8 samples upon resync.

I really hope somebody can figure out a solution because rehearsing and jamming isn't possible for me without a proper video feed (if in fact my problem is latency-related)...
===

I'm sure a proper video feed will also be something I will want - once I have solved the more basic audio problem.
Are you ready to help me with it?
Pete

[Pete Savigny]

More than 100 views, and nobody has a suggestion, as to how to make buffer settings prevail?

Please help if you can.

[Pete Savigny]

Someone on the Behringer Musictribe forum answered:
====
8 samples is what Jamkazam defaults most interfaces to for some reason. It still works.  

I recommend performing a complete audio gear setup on Jamkazam. To find audio gear go to top right of page and select audio gear. Even if you have done this I would try again. Make sure and select your device for input and output and check  boxes 1 and 2  for both directions. 

During the gear setup you should hear their test tone when you reach that page to test. When in a session direct monitoring should be off and you control your own volume by the level control under my live tracks when in a session. Do you see the levels light up on your friends under other live tracks?  

Also, the mute function will activate very easily if you are not careful when you are adjusting your or someone elses levels. 

If their avatar is grayed out this would be a network issue and not an audio issue. It could be on their end or your end. 

Many use the various Behringer interfaces and mixers on Jamkazam and have great results. 
You can also check your interface on a DAW to see if your audio is working properly on input and output. 
====

I went through the audio gear setup again. I cannot see that I did anything differently, apart from also going through my Windows sound settings again. And I made sure that no JamKazam process was still running on my task manager, before restarting JamKazam. 

And now I'm hearing my co-musicians. I cannot quite yet believe that this will keep working. 
I will try out with my tango band members.

[Pete Savigny]

Eric Hayes writes to me via personal message on JamKazam:
====
use the UMC driver 4.38 and you should be okay.
I had the same problem when I first hooked up the Behringer.
This is what solve it from going back to 8.
Let me know if this helps
====

I deinstalled UMC driver 4.59, installed 4.38. 
It still reverts to 8 samples upon resync. 
But I can now change the buffer size, and it will stay, until I resync.

Meanwhile it still works with different buffer sizes.  
Though my latency stays at 4.17ms which corresponds to my latency at 8 samples. 

I would have expected it to go up according to buffer size.
So, I'm not sure if the higher buffer size is actually effective.
Latency seems to increase, when I'm playing and listening to myself through the system and directly at the same time.

It seems that 8 samples does not bring up any quality issues with the sound getting through. 
So I suppose I can let JamKazam run on 8 samples.
What do you think?


I wrote earlier:

Someone on the Behringer Musictribe forum answered:
====
8 samples is what Jamkazam defaults most interfaces to for some reason. It still works.  

I recommend performing a complete audio gear setup on Jamkazam. To find audio gear go to top right of page and select audio gear. Even if you have done this I would try again. Make sure and select your device for input and output and check  boxes 1 and 2  for both directions. 

During the gear setup you should hear their test tone when you reach that page to test. When in a session direct monitoring should be off and you control your own volume by the level control under my live tracks when in a session. Do you see the levels light up on your friends under other live tracks?  

Also, the mute function will activate very easily if you are not careful when you are adjusting your or someone elses levels. 

If their avatar is grayed out this would be a network issue and not an audio issue. It could be on their end or your end. 

Many use the various Behringer interfaces and mixers on Jamkazam and have great results. 
You can also check your interface on a DAW to see if your audio is working properly on input and output. 
====

I went through the audio gear setup again. I cannot see that I did anything differently, apart from also going through my Windows sound settings again. And I made sure that no JamKazam process was still running on my task manager, before restarting JamKazam. 

And now I'm hearing my co-musicians. I cannot quite yet believe that this will keep working. 
I will try out with my tango band members.

[Michael Dolak]

I still have the same problem...

Does anyone know if it is possible to edit some sort of ini-file directly to change the preferred buffer size?