When joining a session, I typically get only 1 or 2 of the participants, the rest is "no audio".
My own audio setup looks ok and working (3 msec latency), in the router upnp is set, firewalls are open.
My internet connection is DS Lite type, meaning I only have a ipv6 adress. Virtual ipv4 adress is created through the DS Lite tunnel. Typical setup in Germany.....
Actual guess is that this comes from my missing public ipv4 adress. Does anybody experience the same trouble, and eventually have a fix???
Hello, new to jamkazam and struggling to get latency down. Using Tascam US MKII DI and in audio test get latency 14ms, I/O in the green. Looked at a new DI but been told more about tweaking computer than interface. Any help, or suggestions. Tascam is old 2012, I thought that was it but open to any ideas to avoid purchasing a new one if possible. Thanks
I have a rather large setup, as I use a lot of synths, and generally, it ends up coming to about 10 stereo channels or so. Is there a way to have it so that Jamkazam will just take the output of my sound interface instead of sending out the individual channels? I think that would make a huge difference, in terms of bandwidth. It's not feasible for me to use a mixer at this point, because I can't afford one that would handle everything, at this point. I have my levels mixed properly with my interface's software mixer. Another reason this would be useful is that I don't always use all of the channels, but want to have them available if I do, so that's more bandwidth being wasted.
If I can't do this, it would probably be best to just put an option in such as "use monitor output from interface".
I am only about 3 weeks into using JamKazam. A friend and I were doing sessions in my studio before the whole covid-19 thing started and we found JamKazam and it has been especially useful to keep us going. We'd been jamming to some Band In A Box backing tracks (bass and drum), so we have bounced those down to FLAC files to play in JK. This has worked well for us, although getting the audio levels correct has been a mystery to some extent. FWIW, I ran a recording studio for about 10 years using primarily Nuendo and ProTools, so I have some background with this.
We spent a lot of time running some tests last night to better understand how the audio path / gain staging works - not as straightforward as most DAW software which normally shows you levels and also 0 is 0 full scale and you go from there and stay modestly away from full scale so as not to clip the mix.
Background:
I created a -20 dBFS Test Tone Sine Wave Flac file to measure audio levels - this file, when played into an input audio track will show all green and no orange on the JK Tall Meters – the ones on the fader. So I'm guessing that -20 is the nominal input level for JK. (The American SMPTE standard defines -20 dBFS as the alignment level.) I mainly did this for my friends use since he does not have anything to readily create a test tone. I can inject a tone from my converters, and I did do this initially to verify that my test tone flac behaved identically to a directly generated tone.
My friend on the guitar is using a Behringer Interface.
I am using an RME Interface - this interface has the very handy TotalMix application which allows for loopback (an output can be fed into an input) and really accurate meters for setting levels.
Our process included:
- Opening the Test Tone as an Audio File in JamKazam - we both did this to see how that measured depending on who opened the file.
- Looping Back the Test Tone into an Input on the RME so that it would play as an instrument track in JamKazam to set and measure levels.
- For all of these tests, we zeroed out all the JamKazam faders (the default - mostly) and left the Audio Boost faders at their defaults.
Observations:
- The Master Mix is 5 dB lower than the Personal Mix. It's not a big deal but is confusing that the levels drop when the Master Mixer is opened for mixing. One solution is to raise the master output level to 5 and leave it there to make up for this.
- Instrument inputs lose about 10 dB to Personal Mix outputs (and 15 dB to Master Mix outputs) - so when I loop the -20 dBFS Test Tone into a track, the JamKazam Personal Mix output level is about -30 dBFS and the Master Mix Output level is about -35 dBFS.
- Audio Files open with the Fader set to -10 dB by default. I do not know why this is - maybe just an anomaly. I set it to zero for the tests. They lose about 5 dB through the system yet have the same output level in both the Personal Mix and the Master mix rather than the 5 dB difference of instrument tracks.
- Audio Files are about 15 dB hotter at receivers compared to the person loading and playing it. So, when my friend plays the -20 dB audio file in JK he sees -25 dBFS and I see it in my Mixer and on the RME meters at -10 dBFS. If I play the file, I see it as -25 and my friend will see it as -10. I cannot imagine why this would be by design and is probably the most confusing thing about getting levels if using an audio file to play along with and there are two or more people in a session.
- Audio File levels can only be set by the person who loads it and starts the playback and the person setting the Master Mix (session starter). This can be an issue because the file may be coming in 15 dB hot. What we did is I started the Session so I could set that audio file level in the Master, and my friend started the audio file so he could also set the level. And since I was on the receiving end of it, I had to turn it down 15 dB. If there were a 3rd person in the session, they would have no control over that audio file level and would have to compensate with their headphone level down and then turn everything else up roughly 15 dB. That will work but could compromise ideal gain staging for that person or cause their mix to clip or force the other players into lowering all their levels to bring the audio file playback down for the 3rd person.
- Panners for Stereo Tracks cannot be zeroed out - they reset to 4 Right. This seems like a new "feature" as I do not recall that happening a few weeks ago. I have also seen it where I added a second track and could zero it if I recall correctly, so the problem may not always happen. Not a huge deal; it just made checking levels a little confusing on the meters.
- If one person in the session has two instruments or an instrument and a mic, the other person cannot independently mix those tracks in their Personal Mix - they only get one mix fader for each additional player and I don't know if that can be changed by some setting in Audio Boost or not.
- Audio Boost. There are several faders there that have a range of 1 to 3 - and I don't know what the units are - they are labeled as multipliers. Mine are all set to 1.5 by default except for Peer Input Channels Music which is 2.5. When I play my looped back test tone as an instrument and raise the Output Channels/Mix fader to 3, the output level I measure increases from -30 to -24. If I lower it to 1, the output level falls to -33.3. So, the audio level range on that fader from 1 to 3 is roughly 10 dB which is why it is called a multiplier, I guess. Other than the units, these faders are fairly straightforward. I don't know what the "My Peer Input" faders do specifically because my friend and I didn't test those. I assume they affect the input levels into my mix for his audio. I'll test those next time we get into a session.
Net on all of this: Audio File playback makes setting the mix complicated, but most users may not even do this. Otherwise setting the audio so that the meters are hitting full green and no orange is a good place to start, sounds decent on playback, and probably leaves enough headroom in the mix for several people playing together. If anyone in the session is playing more than one instrument or mic, then they will have to balance the levels of those as the other players will see it as one audio stream.
There are a number of odd things in the audio path: the 5 dB lower level of the Master Mix, the input to output loss through the software of 10-15 dB, and the odd 15 dB difference in Audio File playback levels depending on who initiates the playback. Now that we know all this, it is easier to setup up levels on both ends and when things do not sound as expected, we know where to look to sort it out. I hope this information helps other new users get their mix set more easily.
We are four musicians. For three of us, jamkazam runs on windows. Our drummer uses a MacBookPro16.1 with Mac OS X 10.15.4 and audio device UMC204HD.
In the middle of the session, the audio connection to the others is broken on the Mac, our drummer no longer hears the other three. When he presses jamkazams "Settings" button on the Mac, the connection comes back. The other session members do not experience a loss of audio connection. In the Mac-Logfile we see (please find logfile attached):
"Audio IO callback interval is unstable. Expected 1.00. Actual median: 0.00 var: -36315952700202157443055670525952.00 - off by -100.00" JSB vol call back not defined JSB vol call back not defined ...... x 143 repetitions startMixerCleanupTimer Closed PortAudio stream 0x7f8ce844aba0 PAudioOutput: Shutdown.
Special phenomenon: if the bass guitar plays everything an octave higher, the connection remains constant. Or if only three musicians are logged in, the connection remains constant. Does anyone have a solution?
We are four musicians. For three of us, jamkazam runs on windows. Our drummer uses a MacBookPro16.1 with Mac OS X 10.15.4 and audio device UMC204HD.
In the middle of the session, the audio connection to the others is broken on the Mac, our drummer no longer hears the other three.
When he presses jamkazams "Settings" button on the Mac, the connection comes back. The other session members do not experience a loss of audio connection.
In the Mac-Logfile we see (please find logfile attached):
"Audio IO callback interval is unstable. Expected 1.00. Actual median: 0.00 var: -36315952700202157443055670525952.00 - off by -100.00" JSB vol call back not defined JSB vol call back not defined ...... x 143 repetitions startMixerCleanupTimer Closed PortAudio stream 0x7f8ce844aba0 PAudioOutput: Shutdown.
Special phenomenon:
if the bass guitar plays everything an octave higher, the connection remains constant.
Or if only three musicians are logged in, the connection remains constant.
In a duo session with a mate a few miles away, my end of the session has fine diagnostics at ~23ms total latency. On his end, he sees my Jitter Queue as being red in the 7.x ms range with the balance of parameters green or yellow and similar to what I see on my end. Audio Boost adjustments don't seem to have an impact. I'm on a 2019 MacBook Pro, he's on a desktop running windows. Both of us on cable router/modem.
Please put your instrument in your profile so it shows up in the sessions. There is nothing more frustrating than combing through the Open sessions looking for one that has an opening for your instrument, only to find out after you get in that someone is already covering that instrument!
