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Latency & Syncing & using the Jamkazam Distributed Metronome
Latency & Syncing & using the Jamkazam Distributed Metronome

As a new user, i am sharing the importance of watching and understanding the Distributed Metronome. Regardless of uncontrollable latency, this feature appears to be the best solution to sync users with their playing, since the same clock source is distributed to all users


The hardware and Internet Service Provider my music buddy I have is excellent. We have excellent audio interfaces, both on the same ISP, 2 miles away from each other, and very low latency. But still, the law of physics apply and as guitarists we easily fall out of sync. When we start playing we both notice we are slowing down asking each other, “why are you slowing down?”.

We experimented with the metronome and it greatly improved our playing experience  because it gave both of us something to sync to. Using a Met can be annoying yes, but it can also make you a better player over all.

The Met features are interesting. We chose the Knock sound. You can change the volume independently for the Met. Also, there is an option to include a visual metronome.

Cluster Beat? In the Met controls you can listen to a Cluster Beat. This is where things become interesting. There are two beats! It sounds like they are an 8th note beat away from each other. We kept it off so we can sync to the main beat and recorded a couple of songs and listened back to them. The Met is not included in the recording and overall we felt we achieved some success with playing in sync. The pre-mixed recording revealed a slight off set of the tracks at times, but not all the time. If you take your recordings seriously you can correct this by loading the individual tracks in your DAW and fix the off-set.

Not without some glitches, there will always be some unexpected anomalies. While playing on occasion the Met skipped, or dropped a beat causing the players to fall off the beat. This happened twice during a 15 minute session. I suspect a network hiccup can cause this.

If two guitarist are playing and one takes a lead while the other hold the chords, the lead player will notice the other player is not on the beat, but the after beat. In our case we just decided to go with what made musical sense and play the song, regardless of what the metronome was doing. The good news is, we did not feel like we were slowing down on each other. The point being, as long as one is keeping the tempo going based on the Metronome clock source, you can have a good experience.

Drums. I have a Digitech Trio which can play a drum beat in different styles. It is not MIDI, and has it’s own internal clock source. It’s a lot of fun playing solo with this device but I suspect for other players in the session, even though it provides a steady beat it is not the same as the distributed metronome. This makes me think, even for an actual drummer in the session, they too will need to depend on the distributed metronome so all players are in sync with the Met, not the drummer.

I have some more experimenting to do but wanted to at least post this in hopes it will put other players in a direction they may find helpful. Remember, latency is always going to be happening, it’s a matter of who we play with the latency that can give us a better playing experience.

Update April 13, 2020

Did some more experimenting with the metronome. It too has its moments and sometimes is not dependable. It is worth noting
the lower tempos under 100 or 90 were less likely to skip a beat. Anything a 100 and it was a struggle because the tempo would $ft.

One interesting test you can perform between each person in the session is, put your headphones to your microphone so the other person can hear your Met click, with theirs and vice versa. In many cases it was close enough.

One thing that seemed to work the best was, setting a playing rule between players. One person is responsible for playing the chords and keeping the tempo and should never change even if the other players begin to slow down. Once we established this rule, we played about 4 songs reasonably in sync.

Networking Packet Rate Configuration Testing: we lowered down the “maximum outing music bitrate” to 128 kbit/s. After doing so we are more in sync. Expect the sound quality to be lessened. In my opinion, the trade off is worth it. The default is 320, and we had sync issues also at 256. Screenshots attached.

Update April 26 2020

Time for a General Update. First, thanks djs, for your 4/22 posting comments. While I think there is a lot of value exploring the network settings and get familiar with the app, my latest observations about the Distributed Metronome are not as encouraging. Even with it, my band mate and I notice it too can have some bad moments keeping us in time and sync.

One thing we notice, we have exceptionally good jam sessions even without the Met until about an hour + into our session. Things go downhill form there. It’s impossible for us to tame the latency. The theory is, perhaps it’s because we share the same ISP and live 2 miles apart from each other, and there could be increased bandwidth traffic happening. We usually start at 4:30 Pacific time, and around 5:40 / 6P, the latency is uncontrollable.  This is an important reminder that closer, does not always mean better.

There is no question, the higher bit rates sound nicer but we still find lowering down to 128 K does seem to help and is worth the trade off. The 1ms Frame Rate still seems favorable.

Attached Files Thumbnail(s)
Very helpful info for this newbie.  Thanks.
Good stuff GDJ.  Thanks for posting.  For anyone reading your post and looking at changing the bit rate, also note his Audio Frame Size is 1.0. 

This setting will give you the lowest latency.  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.  

I haven't seen a measurable change when switching to lower bit rate (I was the only one 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 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.  

One other quick note.  These setting are found as GDJ's pic shows (network settings), but can also be found in Manage --> Audio Settings --> Audio Booster.  They are the same window, just two different ways to get there.  

And while we are at it, play with your 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 a different sample rates.  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!

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