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What can I do about very poor latency? - threewanderers - 04-20-2020
Hello all. I am a new user to JamKazam, and my technical skills are limited. I would love to use JamKazam to jam with friends, on guitar primarily, but my latency appears to be very poor, it is consistently 79.8, or red. Jitter in and out are green, as is gear driver. However, frame size is also red, at 40 ms. When I play with my friend, or play along with a jam track, there is a noticeable lag, which is very frustrating.
I am using a Windows laptop with Windows 10 (v 1903), processor is i7 6560U CPU, @ 2.2 Ghz, 16 gigs of RAM. I am using, for now, the built in mic, and an ethernet cable, download speed 45.59 mbps, upload 5.0 mbps.
My question is this: what are the things I can do to improve (reduce) my latency? Are there settings on my computer I can adjust, or is there equipment that I need to get? Would an interface device improve my latency? If I use an interface for my guitar, can I still use my computer's built in mic to talk to my jamming partner, or will I also have to buy an external mic?
If someone can make any suggestions to help me improve my JamKazam experience, I would be extremely grateful. With the current coronavirus situation, this is the best way I can think of to keep playing music with other people, which is helping me to keep sane during this difficult time in all our lives!
Joe
Oceanside, CA
RE: What can I do about very poor latency? - Hans Peter Augustesen - 04-20-2020
Look here
LATENCY, mostly TOTAL LATENCY
> https://forum.jamkazam.com/showthread.php?tid=171
And by the way:
Dont confuse GEAR latency with INTERNET latency and/or TOTAL latency
"Upload 5.0 mbps" is low - according to latest "moderator" news about that.
You allready have a audio interface. That is the internal type.
But a external one would usually giv much better result.
Read about it all elsewhere in forum - search
RE: What can I do about very poor latency? - threewanderers - 04-20-2020
Thanks for the info. I did watch a tutorial video on You Tube, not sure if it was official from the JamKazam team, but it says that an upload rate of 3-5 mbps should work fine. Anyway, without changing internet providers, I am stuck with the speed I have, so I need to find other ways to improve the latency/lag time. I will look into an external interface.
RE: What can I do about very poor latency? - Hans Peter Augustesen - 04-20-2020
You are very right about that 3-5 mbps. I think
I think that 5 should be more than enough - to 8 persons in a session.
But I am unsure about that (there has of course been some development in the internet since 2015-2016, so many conditions has changed)
No wonder that I am unsure and others too - the old (and lost) information was 200-300 kbps upload per person in session - without video, will say.
Actually in a session one can see the actual amount of audio data transmission for each person.
And the usual amount is about 400 kbps per person.
That will give 400 x 7 = 2.8 mbps. Only audio
And now it is declared that 25 mbps is the case. With both audio AND video, I believe.
Look for your self about that, as I said.
RE: What can I do about very poor latency? - Hans Peter Augustesen - 04-21-2020
And of course search for "built in mic" -"external mic" or/and similar
You can use built in mic along with a external audio interface - as you like. But it is usually low quality and the latency is greater.
List of Audio Interfaces
https://forum.jamkazam.com/showthread.ph...+interface
RE: What can I do about very poor latency? - Dimitri Muskens - 04-21-2020
>>>
Please bear in mind that in most households there is a lot more going on than just JamKazam.
Any use of your LAN & WAN connection will influence your jkz experience. Both in bandwidth and 'noise' (jitter)
So, any 'overhead' on whatever calculation one might do on hypothetical use of mbps' is welcome when trying to work online with live audio.
RE: What can I do about very poor latency? - blandis - 04-22-2020
Hi Joe,
I'm not sure the above posts directly answered your questions, but did point you to sources for answers. Given your comment of limited tech skills, I will provide my 2 cents on the most important item I read.
Audio interface: yes, a new audio interface will dramatically improve your experience, mostly through its reduction in experienced time delays, aka, latency. The main reason is that separate dedicated audio interfaces come with drivers that let you control settings to specifically control the latency as well as other parameters. Internal sound cards in PCs do not typically provide configurable control panels to adjust their settings (such as the most important one, buffer size, sometime referred to as buffer samples). Their electronics are optimized for low latency recording and playback, and they almost always are of better sound quality too. You don't need to spend a ton of money here; take a look at the list of audio interfaces here in the forum, and consider buying one that meets your needs online (Amazon, etc), in terms of number of audio channels. There are other buying guides and ranking info readily available on the Internet, but I do recommend picking one that the forum list confirms as verified working with JamKazam.
In the meantime, there is a somewhat universal free "audio driver wrapper" known as ASIO4ALL. This driver provides a layer between the Window's regular WDM driver (which is typically a driver provided by the PC manufacturer or the sound card/module maker, such as Realtek) and your audio application, which in this case is the JamKazam app. What ASIO4ALL does is essentially override or bypass standard Windows audio performance settings and connects directly to the low-level drivers of most commercial sound chipsets, providing user-configurable settings like buffer size and sample rate, and performance closer to what can be expected from a dedicated audio interface and its driver. In fact, even some lower-priced dedicated interfaces (example: Behringer UM2) only come with ASIO4ALL and no custom driver (although there are older dedicated ASIO drivers for the UM2 that work a little better than ASIO4ALL). You can search the forum for ASIO4ALL help and opinions.
Hope this helps a little
RE: What can I do about very poor latency? - djs - 04-23-2020
Here are a few things from a list I posted on another thread (most of these actually refer to your AUDIO INTERFACE LATENCY, but a couple will address INTERNET LATENCY - together they make up you TOTAL LATENCY):
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!
* Note that I see a help article is in the works about the need for sample rates to match, so we should soon have information about how I am actually wrong about using mixed sample rates - although it does work, it must not be ideal, and causes issues. I will do my best to update when that gets posted. But it should still be true that the higher the sample rate, the lower your latency will be. Curious to see what the say about the need to have everyone on the same Audio Frame Size.
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:
https://support.focusrite.com/hc/en-gb/a...ces-on-Mac
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."
* People are saying that a lot depends on which ASIO driver you are using. I also think I saw that this may be adjustable in the Audio Gear (from the dropdown menu next to your profile name)
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 8.8.8.8" 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 (8.8.8.8 is one of Googles). Check this out or search for yourself:
https://etherealmind.com/what-is-the-best-ip-address-to-ping-to-test-my-internet-connection/
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).
RE: What can I do about very poor latency? - Hans Peter Augustesen - 04-23-2020
Fiber is NOT better than "old" copper - for JamKazam
I firmly believe.
From search on the WEB:
"Is fiber better than copper?
The Difference Between Fiber and Copper
Underlying technologies cause the bandwidth gap between fiber and copper. ...
Copper uses electrons for data transmission, while fiber uses photons.
Light is faster than electrical pulses, so fiber can transmit more bits of data per second and offer higher bandwidth. 23. jan. 2019""
"... more bits of data per second and offer higher bandwidth ..."!?
Nothing about PING - PING is very crucial.
PING for fiber is NOT better than for copper. In short
Of course - one need to find better documentation for that claim.
Maybe I can do that
Now there are at least two threads about excactly the same topic.
It is confusing - not good.
Now one had to post the same message different places.
In fact - it is a total waste and crazy
RE: What can I do about very poor latency? - djs - 04-23-2020
(04-23-2020, 03:06 PM)Hans Peter Augustesen Wrote: Fiber is NOT better than "old" copper - for JamKazamHans,
I firmly believe.
From search on the WEB:
"Is fiber better than copper?
The Difference Between Fiber and Copper
Underlying technologies cause the bandwidth gap between fiber and copper. ...
Copper uses electrons for data transmission, while fiber uses photons.
Light is faster than electrical pulses, so fiber can transmit more bits of data per second and offer higher bandwidth. 23. jan. 2019""
"... more bits of data per second and offer higher bandwidth ..."!?
Nothing about PING - PING is very crucial.
PING for fiber is NOT better than for copper. In short
Of course - one need to find better documentation for that claim.
Maybe I can do that
Now there are at least two threads about exactly the same topic.
It is confusing - not good.
Now one had to post the same message different places.
In fact - it is a total waste and crazy
I'm not a cable / network specialist, so maybe you know better than me. I'm sure many people do and I welcome informed feedback and corrections. I'm trying to figure this out just like everyone else.
As far as your comment about me posting messages in different places, I didn't start those threads. I'm just trying to help the people that did. It's starting to feel like you have a problem with me.
A few days ago I posted a suggestion (in the appropriate place) that would be beneficial to me if it were to be implemented. And I thought it would help some others as well. You (and someone else) took issue with me doing so. Instead of just ignoring it you felt the need to basically shoot my suggestion down and insinuate that I am either lazy or don't know how to use the software.
Now I made this last post as a response to a couple people asking how to fix latency issues (yes in more than one thread) and you say it is a total waste and crazy. I don't get that. I'm just attempting to help some people as best I can. I didn't realize you were going to police the forum, and that I had to conform to your high standards and processes. I'm not looking to be on here to be picked apart or get into arguments, just to help and get help. I can just stop posting and leave the forum if you want. I'll be sure to only post this response to this thread (not both of your identical posts that are in 2 different threads).