I've see a few threads talking about distortion, and it seems the replies are implying a problem with audio levels. So I thought a thread for ideas on this problem and how to correct it was appropriate.
Some background; I've been involved with audio engineering and recording (analog and hardware digital recorders) for some time and have a few digital consoles but I have not really entered the full on ProTools world of complete computer based recording. Hopefully that gives a basis for the accuracy of my descriptions of this problem, especially in reference to typical 'digital distortion' due to improperly set gain structure or not understanding How 0 dbu differs with analog vs digital audio signals.
We run band JamKazam rehearsal sessions between 5 people sometimes for few hours, each using an audio interface for vocals and their instrument, and have been doing this for quite a few months. Occasionally, one person will complaint of 'static' or 'distortion', including me. By 'occasionally' I mean the signal between us is good, clean, and low latency but then the 'static' will happen for anywhere from a few seconds to someone needing to leave the session, reboot their setup and rejoin the session.
What we noticed was people using Behringer interfaces had no problems. We tried an older Focusrite Scarlet Solo, Focusrite 2i2 Gen3, and both and Allen & Heath QU16 and SQ5 digital mixers' interface functions. All except the Behringer hardware interfaces give that occasional distorted 'static'. Further details of out setup is hardwired Ethernet connections (of course), Windows computers all running 10, but some desktops and some laptops, typical total latency between members is 30ms or less, though we see 'red' indications for internet and I/O jitter occasionally. We are all using The manufacturers' ASIO drivers, and the sample buffer is forced by JamKazam to the lowest possible value available, which is 6ms I think. When we try to change it to, say, 32ms, JamKazam changes it back to the lowest setting within seconds.
We all now use Behringer interfaces, from the cheapest UM2 to the UMC1820 and we have not had any 'static' issues since. But I would rather use the QU16 as my interface if not for the 'static' problem.I've investigated USB signal jitter reducers and re-clockers, but I am not sure that is the problem.
Has anyone has similar experiences and, if so, how did you solve it? Would LOVE if a JamKazam programmer could chime in with the technical perspective of the cause.
My friend and I just started using JamKazam. We both have UMC 202 devices, successfully set up. I have a Windows laptop with downloaded UMC Driver, she has a MAC using built-in MAC drivers. We have tried several sessions together, and while can connect and hear (and chat, etc) - the music (even just one line input guitar) is unusable. It has significant latency and also some noise. We can each record separately locally through JamKazam on our own gear and that works fine.
While our Audio latency is generally green (occasionally a bit yellow) --- It Seems our network latency is "red" ( around 36-40 ms).
We have both run speedtest and have upwards of 80Mbps download, and about 10 Mbps Upload, from ewach of our Cable modem based providers respectively.
We have tried setting different output bit rates, and frame sizes - none of which has helped.
Even just one DI guitar is noticeably latent & distorted when heard at the other end of the network. Recordings clearly show this
The attached picture shows the diagnostics for someone I was playing with. The audio interface latency is 6.0 ms (mine was the same) and the internet latency is 14.5 ms. These add to 20.5 ms and yet the total latency is showing as 51 ms. The numbers were fairly steady. Even if you add in the jitter (should you?) the sum only comes to 27 ms. Would you expect the sum of individual latencies to add up to the total? In other sessions with different people I find they usually do.
Subjectively we could detect no effect of any latency on our ability to keep together. It felt just like playing in the same room. In a session with someone in Rhode Island (I am near London) we had a total latency of 80 ms of which 60 was internet. It was impossible to keep together with that amount of latency. One of us had to lead and the other to try to keep with that. Not very satisfactory. If 80 ms is almost unplayable it seems unlikely that 51 ms should be perfect. I am therefore wondering if the reported 51 ms is wrong and that it was actually nearer the 20 ms sum of interface and internet latencies.
On a separate point I notice that Frame Size is 10 ms and red. What causes that and can I change anything? We each have the Frame parameter set to 5 ms. Is that the same thing?
I'm just getting started with Jamkazam and am playing electric guitar into my apogee ensemble interface.
As you know the signal straight out of an electric guitar is a very poor sound and needs processing. In my computer I'm running Ableton 10 for my DAW and using a native instruments plugin called Guitar Rig 5. This is the processing I need to hear.
But when I turn off direct monitoring from the interface I get dry signal only, and when I record into Jamkazam I get dry signal only. I assume that's what my jamming partner 50 or 60 miles away will also hear when we connect.
How can I make the processed signal the one that Jamkazam uses? Thank you so much.
This is just for info since I have solved the problem. I had my internet connection routed through a wireless access point (WAP). I was using a wired connection but still got "no audio". My set-up was: main router to Netgear 8 port un-managed switch, to Netgear R6250 router set to WAP mode (i.e. all router functions disabled), to laptop running JK. All via Cat 5e cable. On the R6250 user interface all the security options were greyed out so you would assume that it was not doing any policing. I was just using the R6250 as a switch and also as a WiFi extender. The "No-audio" problem disappeared as soon as I removed the R6250 from the chain so it was obviously doing more than acting as a simple switch.
Just posting this in case anyone else has a similar problem.
Last night I tried using my Kemper Profiler with Jamkazam, via S/PDIF, and all I could get was maybe 5 seconds of audio from the Kemper after entering a solo session. After that, the S/PDIF audio was muted, and I could not figure out how to get it back. Are there any special settings that need to be made, to keep my S/PDIF connection active in Jamkazam, during a session?
A little additional info on my setup. I'm using a Roland Quad Capture USB interface in Windows 10. I connect my vocal microphone to input 1, and I route S/PDif from my Kemper to channels 3&4 of the Quad Capture . In Jamkazam, I have one audio input for the vocal mic (input ch1), and another audio input for guitar (ch3&4 selected).
Upon starting Jamkazam, and a new session, I can hear the Kemper via S/PDIF for about 5 seconds while the VST scan happens. Once the scan ends, the S/PDIF appears to get muted, and I can no longer hear the Kemper. Even if I leave the session and start a new one, it remains muted. If I shutdown the Jamkazam app, and reopen, It will work again for 5 seconds then the same thing happens. I'm not using any VST's on either channel of this profile BTW.
As a side note, If I open Cakewalk Bandlab, add a channel, and put that channel in record arm mode, I find that the S/PDIF from the Kemper is working perfectly. It just seems to be an issue with the Jamkazam software or configuration.
Is anyone else having tremendous stability and cra$ng issues? Since the last update on 6.23.2020 jamkazam is now useless with my MAC. Mac with OS X 10.11.6
I have in new Behringer UMC404 HD and I have it apparently correctly installed in JamKazam with the UMC ASIO driver 4.59 When I plug an instrument into channel 1 of the UMC, the signal appears on my live tracks (on both channels, if only input port 1 is activated in the audio setting of jamkazam) When I plug instruments into channels 2-4, no signal appears in my life tracks (but I can hear the signal in my phones on UMC) When I activate Input port 1 and 2, signals from channels 1 and 2 of the UMC appear in my life tracks (channel 1 left, channel 2 right, but visible only in the volume-controller), no signal comes from instruments in channels 3 or 4. Do you have an idea for a solution of the problem ?
