Glenn Stanton

if it's a single parameter, it seems like you just pop into the effect and change it 3x. once per buss. 10 seconds? esp if you're focus is on the room size only. that said, depending on what your routing through those busses, sometimes different room sizes may make it more interesting - drums, guitars, vocals, strings, etc can all benefit from different settings including room size. using an envelope to control the room size parameter is also an option - but essential the same as directly modifying each. the benefit of the envelope is you can adjust throughout the song - e.g. enlarge the room for the chorus or verse and then shrink back. use the same to bring instruments back and forward.

thanks! agreed, originally i had it simple but it felt too simple... so, i'm thinking of ways to keep it moving without being busy...

probably one of the first songs i wrote, erm, back in the early 1980's. storyline: some guy daydreaming about his girl as she's walking on her way home (in the city, so it's not too far) ? comments welcome. while creating this version, the half-time just kind of jumped out as fun. BPM 120 Key A Drums, Bass, Acoustic Guitar, Piano, E Piano, B3 Organ, and Sax Solo Lead Vocal + two Backup Vocals ---------------------------------------- i can tell by the way she's walking she's thinking of me i can tell by the way she smiles just what she sees and what she sees in me remains a mystery i know, we might seem a lot of fun yeah i know, we must be on the run lord, i know, we're in love i can tell by the way she takes my hand and gives me her smile that when she tells me she wants to spend the night i know she means a little while yet what she sees in me remains a mystery i know, we might seem a lot of fun yeah i know, we must be on the run lord, i know, we're in love i can tell by the way she touches me everything is all right i can tell by the way she kisses me and holds me tight still what she sees in me remains a mystery i know, we might seem a lot of fun yeah i know, we must be on the run lord, i know, we're in love [solo] i can tell by the way she's walking she's thinking of me i can tell by the way she smiles just what she sees so what she sees in me remains a mystery i know, we might seem a lot of fun yeah i know, we must be on the run lord, i know, we're in love yeah, i can tell

make a shortcut to "C:\Program Files\Youlean\Youlean Loudness Meter 2\Youlean Loudness Meter 2.exe" and put it on your desktop

and after the prehistory of writing on the tape box, text editors and sometimes screenshots into the project information folder ?

so, export the file to MP3 with the plugin on the master showing (after fully playing it) an integrated LUFS of -14db. close the project (after saving it of course). then open the standalone version of YouLean and drag the MP3 file onto it, and it reads? in my projects, if i use Expose 2 or YouLean, they're usually within about 1/10's LUFS for integrated level based on my project settings of -14LUFS. the short term, dynamic range, etc values can be somewhat different because the tools may be using variants of the "standards" in their application (for example, using AES in both results in slightly different values, but again usually are in 1/10's of differences).

there is a thread on the Soundtoys plugins on this forum and it seems several of them have issues, which as faik, have not yet been resolved - but a couple did get updates as a "beta". of course, i had just purchased them before finding this out... lol

i would think that since you're recording a performance, that after each recording, the band would organize for the next performance including tuning, effects they might use on their instruments, cheat sheets and lyrics etc. so basically just closing the recorded project (after saving it ? ) and then creating a new one from the template and saving as <name>+<date> etc and a quick check to make sure the tracks are all live and getting signal, could happen while the band is prepping? if not, then perhaps setting up several projects at a time to enable nearly continuous record would work. if you do need the macros - Auto HotKey could do it. AutoHotkey Downloads https://www.autohotkey.com/download/

ODIN 2, VK-1, TyrellIN6 are all nice here's some i found doing a simple google search (bolded are ones i regularly use): Very low CPU (~1%) u-he Podolski & TripleCheese Waldorf PPG Wave Dexed NI FM8 KORG Gadget Collection Full Bucket FB-7999 SQ8L (32-bit only) Synth1 Low CPU (~2%) u-he Zebra (2/CM/lette) NI Massive & Absynth

or use a better synth which is effective for low CPU and low RAM processing. like we used to do. ? then again, if surge isn't the issue (i uninstalled it in mine because it sucked the life out of my PC) then try another DAW.

i've seen some of the pro tools folks have videos with 12 or more PER TRACK and 40+ tracks + another 20+ busses sometimes with 5-10 plugins... so something like 400+ plugins on a project... of course if they had recorded it properly things would likely be easier... one had crazy layers like EQ, compressor, pitch, EQ, compressor, saturation, tape effect, de-esser, EQ, final compressor, slap echo, then reverb... on each of 8 vocal tracks... but i guess that's what it takes sometimes to get platinum records... lol

Butt memory isn't funny, many people have said i have my brains in my as* and so i take it seriously. ?

adjust the level for the project. basically turning the volume up or down. assuming all your audio is ending up on your master buss, one option is to create a "print" buss and feed it off the master using a send. set the print buss output to none (to avoid increasing the monitoring level). this should be quick to do (2-3 mins) and non-destructive. tracks + busses ---> master (as normal) -> (add) pre-fader send ---> (new) "print" buss (output = off) w/ YouLean meter then export using "print" instead of master. your listening level is now controlled by the master volume, and the print buss output will be the export source. (setting the print buss output to none means the sound on that buss won't add to the master level and cause you to hear things louder) and then you use your level meter on the print buss and adjust the send to create the desired level.

yeah the Apple "we don't do backwards compatibility" policy is not so great unless you like to keep buying new hardware to get compatible/reliable drivers...

part 2- yes - re-open the project and adjust levels and re-export. if you have exported as a WAV file, you could use other tools to adjust the volume (like SoundForge, Acoustica, etc etc) and export as MP3

use your search to find the desktop app. then create a shortcut on your desktop ? pg 13 of the manual. or use the File -> Analyze File

my experience with the Steinberg IO (2 ch and 4 ch units) is completely different - built like tank, solid drivers on Windows, and excellent audio performance. definitely NOT the best for any desired "preamp" magick, but then again IO units (imho) should be about perfect transparency and not some nuanced preamp magick. ? that's what your rack of 1073's are for...

even worse, referring to "real life experience" -- how quint in the 21st century... lol ?

not sure, but not an issue for me. when i open the PRV i see all the MIDI on the track regardless of number of clips or takes on that track. so it's later that if i know i'm going to copy the MIDI onto another track, i'll select the clips, bounce to clip (almost instant) and continue.

these ideas might be useful in your next project if you have a lot of automation and won't use the offset methods on the tracks with automation. calibration is straightforward - get an inexpensive sound level meter (i bought an older Radio Shack digital for $25 off ebay) and i set my main monitoring volume to 0db, and adjust my amplification and OS sound settings to get to 75db. then a prefader send to the -12db monitoring buss. simple to do a quick switch between the two. and of course the main one can be adjusted as needed. it's important to note: i have templates which i use to get all this set up before i start a project so it's already built-in for me. i recommend the same for you - create templates to set up your "standard" routing etc. i use Ozone 10 EQ and Maximizer on the master with the option to use or not on the Print buss (file export).

most times i start with volume controls set to -6db per track and -3db per sub-buss. then adjust my gain to get somewhere around-12db (using clip gain as needed) on tracks and busses, while keeping my master, with all tracks up, to around -6db. this is before i apply compression/expansion on any tracks or busses. this is a bit "hot" but good for mixing (i use a calibrated 75db level on main monitor buss and also 63db on -12db monitoring buss) and then easily adjusted to accommodate the target export levels later. another reasonable tactic - set a lower monitoring volume and add a noise track with -18db mono pink noise (use a plugin or an audio clip). then raise each track until you can just barely hear it over the noise. one track at a time (mute the already raised track). then unmute all. you should now have a pretty good set of levels to start with - then work out the clip gains to get those individual levels approximately equal and get to a decent static mix.

in Windows 11, if you want to reduce the hibernate file and keep fast start, or simply delete it, running your cmd window as administrator: reduce (e.g. 10% of memory) delete:

the challenge with the gain control to shift volume on an automated track (presuming overall level change) is the impact on effects or other gain staging impacts (over / under drive - undesirable clipping / loss of the saturation effect). so, for large overall change -> use offset mode, for small spot changes -> more automation envelope edits ? use the clip gain envelope as a "pre-compressor / expander / de-esser" function (etc) to get the consistency on the clip levels.

is this for live performances or recording? if recording, the advantage of 6 separate channels would be to create multiple instruments (like string section) to perform the "chord" or small runs integral to the piece, and keeping it all in sync without "overdub". i'd probably go with option 1 - simpler and cleaner. i've done some recording (and some live stuff) with people using expensive guitar MIDI gear - and ultimately, when recording, the challenge is getting a clean MIDI performance (even folks with great touch) to avoid errant notes. most times, we're simply resorted to using the guitar audio and Melodyne to clean up and convert, then split as needed across tracks. live performance - most times you're outputting audio for the audience so the synths (yes, a number of people use several chained and i guess the VSL could do that as well) to get the full set of sounds -- but it requires lots of prep to get all modules and program switching etc setup (and the DAW can help or using other programs to coordinate). will be curious to see what other folks have experienced.

W11 is fine for use. like any OS there will be things you want to tweak but un general, my W11 (home) is working just fine for my DAW and my design work with very few changes - mainly turning off unneeded services and setting my A/V exclusions. occasionally using airplane mode when some plugins seem more impacted (e.g. Waves Abbey Road Chambers seems to be in this category) and thus reduce any network interrupts.