Hi I have been struggling with the interface setup.
Mostly setting the latency. I keep getting during the set up "latency not detected" I am using a Behringer UMC22. I downloaded the driver from Behringer and tried a multitude of settings.
I can get play back on the head phones from videos etc. through the umc22. but the inputs are not being recognized. Locally on the UMC they are. any help would be appreciated
I am using a Scarlett 2i2 into a 2012 Macbook Air running OS 10.12 with a Thunderbolt adapter/ethernet connection. My interface latency is 10.3ms and I cannot figure out any way to improve this. My interface sample rate is synched at 48000 and my router download speed is 190.5mbps (this test was run on my Xfinity account page - I have, for some reason been unable to test my network speed on JKZ). Any ideas would be appreciated!
HI I am a drummer. I have plugged my electronic drums and mic into a Behringer analog mixer, and from there via RCA cables to mic input to Mac Laptop. The Mac/Jamkazam seems to pick up the input from the mixer, as when i speak the green/red lights for the level move left to right. Same with hitting the drums. However when I record, no audio.
Any tips? Should my setup work or is the analog to mac not compatible with Jamkazam?
My test results of audio equipment were good.
I must confess when that test runs at end of the audio setup the ping noise is very faint though my Mixer does not have a button to turn off direct whatever it was called.
Catman
List of audio devices shows the Focusrite Scarlett Solo (3rd Generation) as not having any issues. I just installed mine and ran it into a USB 3.0 port but the WDM indicator is orange and indicating the error notice that "I am using a problematic driver type". Everything else looks good. Is there a different driver I need to use? I have a win 10 64 bit pc.
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.
Another new users here, and trying to set up the audio. I have a Presonus AudioBox iOne, and the latest Presonus ASIO driver.
I can configure it for JK and can hear the guitar signal in a test session, and latency etc looks good. However when trying with a friend in a session there is a lot of breakup on my audio.
It appears that JK is using to smallest buffer size that the interface supports, which is 16. This is too small to be stable, which I know from using it with other applications, and I want to make this bigger, say 64, but I can not change what JK uses. If I open the Presonus control from JK and change the buffer size, after a few seconds JK switches it back to 16.
Does anyone have and idea on how I can change the ASIO buffer size that JK is setting? I even tried looking in the Audio.ini file but could not see where the 16 was coming from.
When using JamKazam and adjusting the ASIO buffers of my Focusrite Solo V3, I get a BAD POOL CALLER BSOD. The value selected is 96 with the ASIO control panel when setting up the AUDIO. I read in another thread to back the buffer down to a small value. 66 was fine. 96 was an immediate BSOD. This makes that experience dead in the water. Are there any fixes?
I am new to Jamkazam and the forum. I am using Audio Box iTwo as a microphone input and speaker output. Recording with video captures audio separately. When I play the video, there is no audio. I can play the audio separately.
HI I am a drummer. I have plugged my electronic drums and mic into a Behringer analog mixer, and from there via RCA cables to mic input to Mac Laptop. The Mac/Jamkazam seems to pick up the input from the mixer, as when i speak the green/red lights for the level move left to right. Same with hitting the drums. However when I record, no audio.