Jim Roseberry

Version that was just released solves the issue some of us were having the past ~48 hours.

Heard back from Morten. They are aware of the issue... and they're working to resolve.

I've contacted Morten. I'll post when I have a response/solution.

For anyone else experiencing this issue: Just installed a new version of the Bandlab Assistant (available today). Same error message upon trying to install CbB

Today I went to load CbB for a client... and got this message (below). Uninstalled CbB from my own machine... and now receive the same error message. I wish there was a dedicated download/installer. Dead-in-the-water... Unhandled Promise Rejection Error: spawn EBUSY at t.<computed> (electron/js2c/asar_bundle.js:5:2208) at C:/Users/Jim Roseberry/AppData/Local/Programs/bandlab-assistant/resources/app.asar/index-win.js:403:7 at new Promise (<anonymous>) at install (C:/Users/Jim Roseberry/AppData/Local/Programs/bandlab-assistant/resources/app.asar/index-win.js:402:12) at queueInstallers (C:/Users/Jim Roseberry/AppData/Local/Programs/bandlab-assistant/resources/app.asar/index-win.js:720:7) at C:/Users/Jim Roseberry/AppData/Local/Programs/bandlab-assistant/resources/app.asar/index-win.js:794:13 at runMicrotasks (<anonymous>)

What audio interface are you using?

Some audio interfaces report their latency accurately... others don't. Take a short high-transient audio spike... and put it on a track in CbB. Re-record this high-transient audio spike on a second track (via an analog input on your audio interface). Zoom-in and measure the difference (in samples) between the two. This is the amount of "Record Offset" (in samples). Preferences>Audio>Sync And Caching>Record Latency Adjustment: Enter the "Record Offset" in the Manual Offset field. Repeat the high-transient spike test used above... and verify there's no offset between the two.

Stream Deck is a great solution for having lots of Hot-Key/Macro functions immediately available. You can custom design icons for all the buttons (each has a LCD). Particularly nice for transport controls, record-arm, etc. You can save different Stream Deck configurations for each of your applications (DAW, Video Editing, etc).

For working at ultra low latency settings, clock-speed is the single most important factor. ie: Some audio interfaces like the Antelope Orion Studio Synergy Core will allow you to run at 96k using a 32-sample ASIO buffer size... resulting in 1ms total round-trip latency. Running amp-sim plugins at these settings isn't something that lends itself to being heavily multi-threaded (spread across multiple cores). More cores is certainly beneficial (especially at higher buffer sizes), but not at the expense of significant clock-speed. In a perfect scenario, you want highest clock-speed... AND the most cores you can get. Tested the 11900k recently. It's a performance improvement (vs the 10900k) in most areas... but not all. There were some changes in the CPU architecture (for Rocket Lake) that are a bit more latent. Working at larger buffer sizes, you'd not notice. If you're trying to run Neural DSP plugins, Helix Native, etc... at 96k using a 32-samples ASIO buffer size (or smaller), you'll hear glitches. That's the one area Rocket Lake is a step backward (ultra low latency audio). For the performance/cost, it's still hard to beat the 10900k.

If you're into AMD, the 5950x is significantly better at low-latency audio (than Threadripper)... and still offers great multi-threaded performance. TDP is a much more manageable 105w.

With 280w TDP, there's no such thing as a truly quiet build (as in near dead silent). Then, you have active-cooled motherboard chipsets... ? Even the Floe Riing RGB 360 TT Premium Edition (one of the quietest closed-loop coolers available) is appreciably louder than something like a Noctua D15. Of course, if you try to use a D15 with Threadripper, it'll thermal-throttle (defeating the whole purpose).

And would you believe a $400 CPU bests it at ultra low latency performance?!?! ?

FWIW, You'll see *zero* performance difference using an original Quantum 26x32 (Thunderbolt-2) vs. the newer Quantum 2626 (Thunderbolt-3). The Apple Thunderbolt-3 to Thunderbolt-2 adapter ($50) works perfectly. If the original Quantum is a better feature match, I wouldn't give the Thunderbolt-2 connection a second thought. I still have an original Quantum... along with a RME UFX+.

Quantum is a great audio interface. When it comes to round-trip latency, Quantum is an exceptional performer (can achieve sub 1ms). Obviously the machine has to be able to keep up with the load... or you'll hear glitches. Part of the reason why Quantum can achieve such low round-trip latency is there's no onboard DSP. IOW, Quantum doesn't offer hardware based monitoring/mixing/routing/loop-back-recording. All routing/mixing has to be done via software (in your DAW application).

USB-2/USB-3 audio interfaces can't get much below ~4ms total round-trip latency. Thunderbolt audio interfaces can get down below 1ms total round-trip latency. Think of Thunderbolt as "external PCIe".

Yes, Quantum-2 was Thunderbolt-2... but works fine using a Thunderbolt-3 to Thunderbolt-2 adapter. For the same $600, you can now get the Quantum 2626.

A couple of thoughts How do the overheads sound? If the overheads are well-placed yielding a good overall balance of the full kit, you can easily get away without using the spot mic on the HiHat. If you feel the HiHat mic is absolutely necessary, first thing I'd do is run a high-pass filter on it. You don't need a lot from the HiHat mic... just a little attack/articulation. Run the high-pass filter frequency up where it's pulling out the body of the snare drum. You'll still hear the attack... but less of the "shell/resonance". Don't worry if you loose some of the "chunk" on the HiHat. I'm not a fan of close-mic'd cymbals. As a test, put your ear close to a cymbal... and listen. You'll hear nasty gong-like overtones. Step back a couple feet and listen again. Now you hear the shimmer/articulation... without harsh/brash overtones. Since you're dealing with already recorded tracks, you probably don't have the luxury of re-recording. In that case, you may also find that some "bleed" isn't necessarily a bad thing. In reasonable amounts, it can actually make the drums sound more 3D/real. With a high-pass filter... and for as little as you need that HiHat mic, I have no doubt you can make it work. First thing I'd do is check phase across all the drum mics. Once you're sure the drum tracks are all in-phase, then I'd start with the overheads. Get the overheads sounding balanced... giving a good representation of the full kit. Next, add Kick and Snare spot mics... to add some "beef" to those drums. If you have close mics on the Toms, add those. If the drums aren't well-tuned, you'll struggle more with close-mic Tom tracks (EQ can help). Now, listen to the balance of the overall drum-kit. You may find you don't need any close-mic'd cymbals. If you decide to use those close-mic'd cymbal tracks, you won't need much.

I use an Arturia Audio Fuse 8 Pre (connected via lightpipe) to provide more analog I/O for a Fireface UFX+. In my case, I have the Audio Fuse 8 Pre look to its lightpipe input (word-clock "Slave") The UFX+ (which is the word-clock "Master") is sending lightpipe (embedded word-clock) to the Audio Fuse 8 Pre. If I change sample-rates in the RME Fireface UFX+, the Audio Fuse 8 Pre automatically follows.

Hi Michael, Lightpipe carries embedded word-clock. When connecting two pieces of gear (digitally), they both need to be running from the same clock-source. If each are running on separate clocks, you'll hear small pops/ticks when the digital audio streams are merged. You'll have to choose either the Audio Interface... or the DP88 as the word-clock "Master". Have the other device (word-clock "Slave") look to its lightpipe input for word-clock. (The word-clock "Slave" must have lightpipe routed to its lightpipe input... and that's where it'll look for word-clock sync.)

I was just curious to hear your take on the OX. I'm certainly not a fan of UA Developers' Apple bias. UA hasn't really done much (firmware wise) to expand the OX. It got a few new Cabs (including v30 speakers)... and the ability to use with Solid-State amps.

John Suhr would explain it much better than I. The simple explanation is that the impedance curve of the Suhr Reactive-Load is nearly identical to a 4x12 with Greenback speakers. To my knowledge, there's no other reactive-load that's more accurate (in that regard). The OX impedance curve is (IIRC) based on a 2x12 and not as accurate (an approximate curve). Not sure what the Two Notes Captor-X reactive-load is based on (speaker wise), but it's also more of an approximate curve. Two Notes IR loading/capabilities are far more advanced than the Suhr Reactive Load IR. The Suhr is limited to running a single 1024-sample Cab IR. Two Notes allows running/mixing two Cab IRs... each up to 4096-samples. Cab IR loading is pretty spartan in the Boss Waza TAE. 1024-sample Cab IRs capture about 22-25ms (depending on the sample-rate). Longer Cab IRs capture a little more low-end.

Significant attenuation noticeably affects the sound. Of those mentioned above, the Suhr doesn't have attenuation abilities beyond ~3dB I was strictly interested in going cab-free (noise-free recording)... so attenuation wasn't a factor.

No worries! This is your area of expertise. ? The OX (or any of these units using digital processing) will have some amount of latency. Suhr Reactive-Load IR: ~1.2ms Captor-X: ~1.2-4.8ms (varies depending on length of IRs) Boss Waza TAE: ~2ms OX: ~2ms (similar technology to their "Unison" plugins for Apollo) Is it the sound/feel that you find inferior to the Palmer boxes? I'd like to see Suhr's reactive-load... along with Celestion's "SpeakerMix Pro" (dynamic Cab IRs)... in a hardware box.

Have also had the Boss TAE. You can tweak the TAE's Reactive-Load (bottom and top) for the specific amp. The only one that allows this. Again, you can achieve good/great sounds. I didn't care for the onboard SS power-amp. Can't go wrong too far wrong with any of the above.

I've owned all of them. Suhr has the best Reactive-Load... but the IRs are limited to 1024-Samples (short). Captor X allows you to run a pair of simultaneous Cab IRs. The Reactive-Load isn't as nice as the Suhr... but the IRs can be up to four times the length (plus you can run two simultaneously). OX Reactive-Load isn't as good as the Suhr. Cab models aren't IRs... they're slightly more dynamic models. UA Plate Reverb, Dynamics, and EQ are familiar to those who've used UAD/Apollo. You really can't make a bad decision from any of the three. IME, None is totally heads and shoulders above the others. You can get good/great sounds out of any of the three. I still have an OX and Captor X