Glenn Stanton

general thoughts: for simplicity - keep things one-to-one whenever possible yes, in theory the pedal could be "patched in" - in practice, you would test this 🙂 bearing in mind most times tehre are TRS (tip-ring-sleeve) cables so your patchbay would need to support TRS connections. one (1) output can feed one (1) or more inputs EXCEPT when those inputs are already connected to another output (meaning you have 2 ouput signals going into a single input -- this can cause all kinds of issues, or none. but generally unless you have some summing apparatus (like a resistor bridge) then you run the risk of overlouding the input, possibly damaging the other output when impedances are way off, etc). line level is preferred over higher voltages (phantom power, 70v audio, power amp outputs) transitioning (plug in, pull out) patch cables when this high voltage is live can result in sparks (which erode your patch contacts), damage to other circuits including the voltage driver, etc. here's a handy guide to understanding line level voltages: https://www.sweetwater.com/insync/understanding-signal-levels-audio-gear/

one way to do this: 1. have all your synth outputs and IO inputs connected in your first 2 patch bays (assuming 16 each) - this gives you what you need for directly recording the synths without changes (i would consider doing "normalled" as i could use a simply Y split cable if i need to "tap" off for some other reason like a "clean" vs" fx'd synth recording) 2. have your effects in another set of patch bays in/out wholly separated ("de-normalled") 3. you can now patch any synth to any effect w the caveat the effct patched in "breaks" the clean path and the effect is fully inserted in the path (i think this is typically the desired approach as many effects have "level" or "mix" controls to set how much of the effect is heard) of course all of this depends on your workflow and if you need multiple inputs for a single synth + fx (as noted in #1). generally my patch bays simply expose every in and out, and things like mic / instrument inputs are normalled to the IO. effects are all de-normalled for flexibility, and monitoring are also via the patch panel (for line levels) and all mics have XLR patch to avoid TRS insert while 48v phantom volt is on and forgot to turn it off 🙂 i used to use a spreadsheet to create the labels but it got painful because of 4 different brands of patchbays, so i bought "patchcad" which made life much much easier https://www.patchcad.com/ (just a happy customer).

long story but short version - Bob Katz (whose mastering audio book is still great reading) was one of the early advocates of standardizing volume levels and he developed a system which provided for setting the output level vs a dynamic range choice. this model was later refined and adopted by many in the broadcast standards we see today. this is a nice reference: https://www.meterplugs.com/blog/2016/10/14/k-system-metering-101.html

yes it would see Kontakt as a single synth to freeze. one option is to have multiple Kontakt synths - my understanding is there is little penalty for this approach with Kontakt as it has some form of memory sharing for the application part, and @msmcleod did some analysis on that. another option is disconnect the tracks you DO NOT want to freeze, then freeze the synth. then disconnect those tracks, mute those related MIDI tracks (or clips), and unfreeze the synth. then reconnect the other tracks you want to hear the MIDI audio from. a bunch of steps but not too long with a limited number of tracks.

thanks Lynn. the sax is indeed sampled (Real Tracks from BIAB soloist) and at first i was thinking i'd replace it (actually a number of times thinking i'd replace it) but then i realized - it's somewhat "angry" and in a way, the person experiencing the loss is somewhat angry at the unfairness of the universe. so i decided (for the moment) to keep it. i'll probably remove the delay effect as i tend to overuse it in my songs. cheers!

one of the nice features of AD - is it's built in mapping tool. depending on where i'm importing the MIDI from, i can usually find one to plug in directly using just a default AD drum map (and without one as well -- i use the drum map simply to have the names, not that i'm mapping anything specific).

i get a new OS when i get a new device 🙂 although proper backups are important, if my OS craps outs, since i've got the installation files and the content backed up, i can format my OS drive and reinstall the OS fresh, remove all the crud they bundle, then reinstall. takes me about 2 days to fully recover (mainly loading takes times) but then my system is as clean as it's ever going to be. my configurations, contents, licenses etc are all saved separately and fully re-installable. i may have to go to some sites to deactivate things so my licenses are still reusable from a count perspective, but i do that before the update.

how far will you go when everything that matters is gone? comments welcome.

in W11 the default is "C:\ProgramData" which is different than W10 which has "C:\Program Data" - either someone in MS oops or they found having a space in the directory name was always a bad choice and caused more issues than not...

the audio part of the secret are the anti-aliasing filters used. for the digital side: power, clocks, etc all contribute to how good (or not) a AD/DA unit "sounds". poor clock stability, under-powered, cheap opamps and filtering components all make a difference. and much of it is measurable. check out a DIY equipment forum where some folks get really in depth on the components, actual o-scope views, log traces, etc etc.

sure - share screenshots of your laptop and desktop display settings. also any video display information would help as well.

a number of years back, a person had done a whole exposé on The Beatles - key changes and tempo changes per song. there is this one but not the tempo map... https://www.beatlesbible.com/forum/recording-and-musicology/keys-that-the-beatles-used-now-that-youve-found-another-key/

one of my favorites as a kid (14?) was to take an old single tube photo preamp (from my old non-turning portable record player) and just plug the guitar straight in and then output to the amp (whatever one tube or solid state never really mattered at the time) - using the volume control to get just ther right distortion - from just light distort to that "modern" metal screaming distortion! the key was to add a volume pot in front of the input to limit the guitar level and a 10K resistor in series for the input load.

no idea. so i just make it a hard selection and ignore it. presumably it's because of whatever matching criteria is used is different - names, CRC, uuid, etc etc which don't align because the vendor gets creative and wants to make a splash with their VST3 version...

i just go into my plugins manager and exclude VST2 for which i have a VST3. takes a few minutes. then if i need the VST2 version for some reason (seldom) then it's right there, re-enable it and ready to go. no messing with installations, paths, and updates fro mthe vendor work as expected and very few times does it ever affect the excluded setting.