Glenn Stanton

i use the channel tool to set widths on stereo instruments - pianos, drums, guitars, backing vocals, etc so i can pan them while still having some width to them. basically a proper side-to-side balance within the overall stereo field is enabled (like in PT with the two pan controls on stereo tracks). another option (and i use this one without the channel tool) is using the surround panner to set the width, depth and position on a 2.0 or 2.1 mix.

one could imagine that the routing to several synths could be notes or controls. or each synth might have a MIDI track for controlling while listening to the overall set of notes, or vice versa... the ability to send several synths articulations or keyswitches might be handy even if they're playing separate notes (e.g. an orchestral part with trilling/vibratos across several instruments). however, v1 should probably just share all for all instruments "subscribing" to MIDI source. v2 - filtering rules, v3 - combining MIDI (ala patch point?) to allow separate streams of MIDI to provide notes and controls...

and then after we get v1 of this, we'll want v2 to some ability to apply filtering rules on the routing table 🙂

check your workspace settings.

thanks azslow -- for me, the added 2ms on echo is a grace note, which if i'm playing more straight ahead, makes me check myself. then i discovered turning echo off 😛 and now for years, i'm happy.

true but for some reason, i really don't like track folders, i think it's because i can't apply fx and do other things enmasse.

i number tracks in the name (in my templates) so i can sort in a variety of ways and still return to my default list. even if i add tracks, i add the # to the name so it sorts back to default. e.g. I (instrument) ## = I10 KICK, I11 SNARE, etc I70 VOX1, I75 BVOX1 etc. leaving space between instrument types so i can insert new items easily. this also helps when exporting tracks as well...

you shouldn't play games on your DAW machine anyways 🙂 it's why God created gaming consoles...

same, i simply rest my wrist on the desk and use fast mouse pointer move so i'm only moving the mouse slightly. proper posture for the reading and keyboard usage. using a mouse since 1987... no wrist issues. just need to tune your seating position / desk levels properly and support the wrist.

for the most part - i created a bunch of difference workspaces to rapidly switch views, sizes etc. and then i use the browser or inspector sides to display full track info or instrument list. the built-in list of workspaces get overwritten whenever there is an update, so don't bother editing those and saving them. if you set your wide/narrow selection in a given workspace, they return to their sizes. the challenge becomes if you're adding them in a project - the workspace won't "remember" them. so i also advise building yourself some templates which make managing all this stuff (like narrow/wide, tracks, busses, routing, default effects and settings etc) to rapidly get to a place where you can focus on recording or mixing or mastering, vocal takes, solos etc.

i use WASAPI Shared which enables me to select both the Spark Mini input and my UMC202HD output (but not inputs - but then i'm only interested in the Spark Mini during that time)

with drums, even a 2ms latency will be audible to most people. so a 28+ms latency will be nearly impossible to ignore (basically the premise for this entire thread). and as an ahole, i'll repeat my suggestion of monitoring live the drum performance vs the other instrumentation, and if need be, include the metronome in the playback. this way you're playing to the material (without looping backing) and hearing your drumming live via the analog path in your IO. turn off DAW monitoring of your MIDI and drum audio from the DAW.

sadly, all he needed to do was live monitor the audio of the kit vs the recording, and then capture the MIDI (and for me i also capture the audio). did this all the time for many recordings. no latency on any DAW. just my generally sloppy drumming 🙂 easily fixed by edits and quantizing... LOL

like a guitar foot pedal collection, DC powered equipment (or low powered AC) - you might look into getting a high quality power supply (or a few) to reduce the noise into your system. a single source of DC (many of the higher end units have multiple voltages and current ratings) should help. most "wall wart" type power supplies are crap - a simple bridge rectifier and an inductor for some spike management, and even laptop power supplies don't have much filtering. running all devices (as noted by amberwolf) off a single grounded power strip, keep your power and audio lines aligned (avoiding ground loops) and prefer the lowest noise outlet you can find (no other equipment as this can have other systems feeding noise).

and make sure to calibrate your listening system - your DAW output should be unity to the monitoring system, and each monitoring system (SoundID, ARC, speakers, headphones, BT speaker, etc) should all provide a consistent level, if not relative frequency response (in my case i want the same approx levels but ok with varying freq response to identify things in the mix). i use a -12LUFS noise signal to get me to 75db level (which for me is "loud") so when mixing, if something seems loud, i know i'm approaching the -12LUFS. and generally i'm mixing at ~65db by turning down the volume to a known value on each system (e.g. my direct UMC = -10.2db, SoundID = -6.2db, Sparx Mini = -7.6db, etc etc). now i can rapidly switch across monitoring systems and not be (too) skewed by level shift, but more from freq content (which is what i want to ensure translation is tested as best as possible on other systems) (and in context of the material... not everything needs to be "full spectrum")