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Tips to Improve Latency
Hey gang. Need your help. I think we need ONE specific thread with general recommendations for lowering latency for audio devices. If you scour all the threads there are tips, but we need them all listed together.

My audio device works, but I need to get my latency down. Would be great to have a list of "try these 10 things" all in one thread. Can y'all help me here?

And if this ends up being a useful thread, maybe an Admin can stick this so it stays at the top of this forum topic.
I agree, there are a number of different threads about latency, but I many of them are specific to a type of gear, etc. not useful to a general audience. Also, the ones I've seen are pretty technical, again it would be helpful to have some tips in plain, clear, explanatory language for those of us who are not tech experts. It would be very helpful to have an overall topic about latency, what are the best ways to improve the JamKazam experience, either through settings on our computers, or additional equipment.

Thanks all, and happy jamming!
"ONE specific thread with general recommendations for lowering latency for audio devices" would be good, yes.

Lowering audio interface latency:

"fast/faster" device + tweaking

And then of course you must provide information about the specific circumstances.

Otherwise it is impossible to give a precise and specific answer.

If your latency is as low as for example 4 milliseconds there is no need for any change
It would not be a good idea to stick this thread so it stays at the top of this forum topic

It is already too bulky compared to the scarce content.

But if one could sort and concentrate it and all the other material here - then
LeeBert, here are a few for you:

1) Audio Frame Rate (main menu at top: Manage --> Audio Settings --> Audio Booster  *also found in network settings):
The higher the audio frame size, the higher the latency you will have.  Because we are dealing with round trip audio, the added latency is essentially double the audio frame size (that's my theory anyway - and you can confirm the numbers if you look at your interface latency before and after a change).  So an audio frame size of 1 = 2ms added latency to your interface.  Audio frame size of 2 = 4ms of added latency.  2.5 = 5ms of additional latency, and so on.  If you have a newer powerful computer you should be able to run at frame size of 1, but test it out.  If you hear clicks, pops, etc., you will need to change to a higher frame size.  I am using a very powerful computer but it is 12 years old and I suspect the bus speed may be limiting this, but it could have something to do with the os (I'm stuck on 10.10.5 until I buy a new computer).  So the lowest setting I can use is 2.5.  If I upgrade and can get to an audio frame size of 1 I would shave off 3ms of latency.  FWIW, everyone I know that is using a newer Macbook Pro has no issue running a frame size of 1. 

2) Bit rate (main menu at top: Manage --> Audio Settings --> Audio Booster  *also found in network settings):
Grateful Dead Jams claims that changing this number to 128 made a noticeable improvement on his session.  I haven't seen a measurable change when switching to lower a bit rate from the default 320 (I was the only one in my session that changed it - while the others looked at my numbers), but it makes sense that if everyone did it things could be better.  I will have everyone try it on my next session and see what happens.  There is definitely a noticeable difference in quality as you change this setting.  Low and high end will suffer digital artifacts more noticeably (especially for bass players and other instruments heavy in those frequencies).  You will also notice a change in how forward the mids are.  I actually prefer the 320 setting to the 512, all else being equal (that is for vocals and electric guitar anyway).  Ultimately, if you can't play with someone else due to sync issues, it doesn't matter how good it sounds.  So if lower quality helps you be able to play, that is what you need to do.  I can deal with lower quality audio, but not big delay and time $fts. 

3) You interface's sample rate:
One person I play with was able to make a considerable savings in his interface latency by changing his interface's sample rate from 44.1 to 48.  And JK allows for each player to be at a different sample rate.  Not sure if this creates other issues though.  I could see how it may increase JK's buffer needs or possibly jitter.  I think that is because it is converting your audio to the bitrate you select (see above) and that is what is being sent.  Not sure how it deals with mixed bit rates on the other end though.  I have a feeling that the more settings everyone in a session has the same, the smother everything will work.  Although, I have no proof of this as I haven't even tested it and don't know how the software is handling many things.  But ultimately, if someone can shave 10 or more ms off their latency, that may be a better return then slightly reduced jitter or something.  After all, I think the system has to cater to the worst setup in the group of players.  That means that everyone's experience will significantly improve if one person can shave that much off of their latency.  Good luck out there!

4) System Buffer (not normally adjustable on Mac's Core Audio - I think ASIO on windows can be adjusted, maybe someone on windows can chime in?):
*BigStu posted this for anyone using a Focusrite interface on Mac:
"I just was searching the Focusrite website support area, and found this article:
In it, you will find a link to a driver you can install to lower latency. I just installed it, and my latency went from 10.4ms down to 7.8ms, which puts me in the green zone on JamKazam. Not sure if this will help enough if your latency is up around 20ms, but it's worth a try.
Also, I am using Ableton Live 10 Lite, and I have lowered my buffer size setting to 64 samples to help speed things up.
Hope this all helps."

5) Interface / drivers:
Your interface and it's drivers can make a huge difference.  Bad ones can be over 30 or 40ms.  Great ones will be under 9m or even lower.  Some exceptional ones with high bandwidth cable may be under 5 with the right settings but this would be very difficult considering the rest of what is going on with the software and computer (see other items here).  Many manufactures have specs for latency on their websites or in their manuals.  Some test results can be found on forums such as Gearslutz, or elsewhere on the web.  I can tell you that RME, UA, and Slate all have very low latency interfaces but higher costs as well.  Those numbers will vary depending on the unit, cable type, if you need a conversion cable to get to your computer and how good that is, the design of that particular interface (there is variation between units from a give manufacturer... usually), and the drivers for that unit.  So make sure you have the most recent drivers for your interface!  
* Keep in mind that your "interface latency" as shown in JK is your actual interface's latency (round trip) as well as some stuff happening in the computer / JK software.  So even if your interface specs a latency of 1ms in and 1ms out, your "interface latency" as displayed in JK will be higher than 2ms.  Adding audio frame size to that for starters puts you at at least 3ms (with an audio frame size of 1).  I think there are other things in the software contributing to the number, but I don't fully understand them yet.  Core Audio (Mac) is definitely buffering and that is adding on as well. Not sure what those numbers are.  Does anyone know? 

6) Ethernet / wifi:
If you don't already know this you MUST be on a wired ethernet connection and NOT wifi for this to work.  This really should be number 1 on the list, but I'll assume everyone already knows this if you made it to the forums.  

7) Modem:
Make sure you have a good modem with the most up to date software / firmware.  

8) Internet:
The better your internet service, the better your experience will be.  Get fiber if you can.  You need good download AND upload speeds.  You can run tests to check your speeds both ways, as well as ping and jitter which are both very important.  Also look at packet loss if you can see that.  There are many sites on the web to use for this.  A quick Google search will bring up a bunch of them.  I recommend trying a few different ones.  

On a Mac you can even go into Terminal and type "ping" and press enter.  Press "Control" and "c" together to stop.  It will then show you info on packet loss and ping stats.  These may differ from what you actually experience in a session depending on providers and where everyone is.  There are other number / sites to try as well ( is one of Googles).  Check this out or search for yourself:

I'm sure this can be done on PC as well, but I don't know how.  Any PC people want to chime in?

Those are the things I know of at the moment.  I was hoping to come up with 10, but oh well.  I hope this helps some people.  Good luck and let me know if you find anything else or have additional info on what I wrote (or corrections).
To comment on the Windows ASIO buffer size adjustability... it depends on the ASIO driver you're using. My only experience is with ASIO4ALL, and while there is a nice little slider to adjust the buffer size, it doesn't always actually change. I have a feeling that "real" ASIO drivers would allow easier adjusting of the buffer. Often the software itself is supposed to request a buffer size from the ASIO driver. It seems that is what JamKazam is doing, but unfortunately there is no way to adjust JK's buffer size request. This would be a seemingly simple feature to add (I'm no programmer, so that's to say from my seat), that would help a lot of people trim down their latency much easier. Hopefully the JK team will be able to implement something in the future.

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