Glenn Stanton

you can use an external drive for OS (the boot settings will allow it), but it's not really interchangeable with another machine because of physical component differences that Windows will detect and require activation on. however, the external drive (assuming USB) will be slower than an internal drive and cause more interrupts (like interfering with the audio IO on the USB). one thing i do for my laptops - i swap the DVD drive socket with my OS (the DVD drive is the slower speed connection in most cases) and put my content / project disk into the primary slot (faster, sometimes 2x faster). (also -- all SSD drive). then i have a couple of "low usage" external drives for my cloud content (i only sync to the cloud drives not operate with them directly to avoid USB interrupts and file contention) and archive, downloads, etc type content.

things to double check: - A/V exceptions for Cakewalk.exe and the sound reference ID apps (make sure MS defender isn't "protecting" you from your apps. - sound reference ID set to virtual driver vs APO (i've found the virtual setting let's me switch it out in my CW preferences whereas the APO setting seems to take over the IO interface and no option to bypass it) - re-run the redistributables for CW and the sound reference installer (i've found that MS likes to install all kinds of fun stuff in MS Edge and hot fixes for critical security and things get weird afterwards)

"as far as i know" AFAIK

it's not capacity - it's lifespan. the memory used in USB keys degrade over time and also poor connector quality can cause headaches with the Waves licenses (not exactly sure why). so spend the $40 on a high quality brand instead of a generic $5 or $10. i replaced my cheaper Sandisk one with a high end Sandisk model 2 years ago now and 0% issues.

thanks Max! i seldom use envelopes for dropouts - if i do need it, i usually just split the tracks i need to drop and carve the opening, esp since i seldom drop all tracks - and most times the arrangement is addressing most of the dynamics in the performance anyways. i wan't sure if the quick group could affect multiple volume (etc) envelopes or not. i'll have to try that out as a new song i'm working on needs a 1 beat drop across all 31 tracks and i'm reluctant to adjust the performances...

depending on how tightly these need to happen, one track at a time with automation write enabled to drag down the fader and bring it up. one other option is do a split on the clips start and end, then edit the volume envelope - the cursor sets to the size of the selected clip, then just drag that down. repeat per track. not great but reasonably fast.

one thought - the send levels look like you have them set very low. usually what you want is to hit the effects harder and then use the buss level to balance the overall level into the mix. another tip - have the delay buss do a send to the reverb buss to add some reverberation to the delayed content.

one challenge is when sample libraries (or some synths) only have 44.1 sample rates and the implicit conversion to 48 has (in my experience) caused more problems than not (which is why 2 years ago a reverted to 44.1 after using 48 for a couple of years and dealing with stoopid sample libraries and synths (of which i have too many to re-purchase or convert). but now, once i have everything all settled at 44.1 - windows, synths, samples, etc i seldom run into issues. when i do need to run 48 or 96, it's because i have inbound client files or i'm deliberating recording at 96. in which case, i simply reset to 48/96 on the os, and set IO to 48/96 and i'm good. if i need to add something from a synth or sampler - i use Kontakt as that seems to have the least number of issues with 44.1 -> 48/96 implicit conversions.

you can copy and paste envelopes.

it sounds like one of the analytical plugin type functions to give you stats on the project. i use Expose on rendered files but plugins like YouLean, Insight, etc you can get LUF type info. as far as L-R + phase (stereo/mono compatability) balance, EQ balance per genre, etc you can use some products / plugins like Izotope RX, Ozone, Acoustica, SoundForge etc. to measure those things.

so if you start a blank project and set this up, it still doesn't work?

try something like this to check everything is working as expected, then adjust levels:

same here. i only have the Slate subscription because there are some plugins and synths i use a lot. otherwise i'm avoiding subscriptions and trying to wean off the Slate.

if you use a high quality usb key (don't skimp on this) to host your Waves licenses, you can readily switch between machines, just not be able to run both at the same time. my second secret trick is to have a bunch of free plugins (or paid ones without restriction) which i can substitute - i have the Cakewalk ones from Sonar 7 onwards, and a bunch of free ones which provide a lot of capabilities (and this how i have several friends setup their systems). so, probably 80/20 - 80% of the time i'm using Waves, Slate, etc and other single machine at a time plugins, and 20% of the time using freebies or builtin.

presumably you are trying: track -> master buss (or equiv) track -> send to reverb buss or aux track -> reverb 100% wet / 0% dry -> output to master buss ? a few things to check: 1) is the reverb getting a signal from the track? 2) is the reverb output set to the same output as the tracks? 3) is the mix (wet/dry) settings on the reverb set to 100% wet and 0% dry? (otherwise you're just getting more dry signal) 4) is the reverb powered on / enabled / not controlled by an envelope which could be turning off the sends or the buss itself?

i keep the licenses on my ilok and waves thumb drives, then use the installers to install the plugins (as expected) but use freefile sync to keep the plugin folders, settings / presets, and content sync'd to the cloud and then sync from cloud to devices (laptop and desktop machines). whichever machine is the "current one" where i may have installed updates or new presets, overwrites the cloud set and then the switchover is simply move the USB keys and hit the sync, and after a few minutes (most times) it's all done. application updates occur via the vendor plugin managers as normal. the nice thing about the keys and cloud - if i'm doing something with my friends setup, i can quickly load the plugins, load the presets and content, and activate and get to work. although most times i'll just bring my laptop and drives. 🙂

i'm finding i do a lot of composition in Hookpad and Musescore. or both. a rough draft in Hookpad, export MIDI and bring into Musescore to then fine tune, merge, expand, create the melodies etc. export to MIDI for CbB and any final tweaks and edits. maybe re-export from the DAW in needed to ensure my sheet music is aligned. it's a weird combination of "blocks" and music notation. i'm not really a fan of tabs (for me it's harder than plain old sheet music, no idea why). 🙂

not true, for all spambots to truly get it all, they have to buy the list from the DMV or voter rolls 🙂 lol

thanks, but in reality, the ink system is embedded in the typing system - so the "ink workspace" can be turned off, but the internals of the ink system are embedded in the typing system - so it's never really turned off from a collection of keystrokes and mouse movements. my guess -- people were using drawing to communicate on chats and meeting product whiteboards to avoid typing and therefore a means of collecting the mouse (or other drawing means) was needed to collect it. not sure if they track cursor key movements though. so maybe the old etch-a-sketch means of signaling could be used? (like square wave binary code or even morse code lol) 🙂

another thought (and what i do) is you can use the arranger as in #1 - then export a 2-track (or mono) WAV file, then import that into a new project (e.g. projectname VOX.cwp) and do all your vocals in that project: lead, comps, backing, ad libs, etc etc. then export those finished vocals and import into the main project. the arranger removes the solo (and any other sections you don't need) and the 2-track WAV file let's you do all the vocal work without the weight of all the tracks in the main project (which lightens the CPU load and let's you work with lower latency).

make sure in the device manager they're listed and enabled in "software devices" and "sound, video, and game controllers".

"artificial" or "automatic" "double tracking" ADR was invented by an Abbey Road engineer (Ken Townsend) (https://en.wikipedia.org/wiki/Automatic_double_tracking) because John was always complaining about having to do double takes on vocals. lol. Waves has a plugin (which i use often on background vocals, not so much lead vocals) which simulates that effect. https://assets.wavescdn.com/pdf/plugins/reel-adt.pdf the double tracking technique is more than just shifting the timing but doing small (or large) variations whilst it is running to create differences which make it appear more lifelike. so in addition to shifting the time, some modulation on one or both tracks can add more realistic double tracking (or cool effects). before i got the Waves plugin i would do as mettelus suggested as well as use the Sonitus modulation plugin on each track. or on a separate use the Sonitus (or other split delay effect) with the modulation to create a stereo split using the delay timing + modulations and then center the lead vocal. the delay + modulation was mainly for background vocals.

not really - i like to use no latency approach to recording live instruments and vocals - so the monitoring is not echoed from the DAW, but from the mixer and monitoring gear. so that also means i can specify the latency on the recording. say 100ms one-way. then after i record, i know the tracks are shifted 100ms and simply move them to the left 100ms. done. no possibilities of crackles, pops, or problems with performances impacted by latency from the DAW and IO echo. some folks are ok with 3, 5, 10, or even 20ms latency (roundtrip) with the DAW recording-echo approach. i think it's just simpler to accept it and move it to align the tracks. takes mere seconds to set and ready for next round of recording. actually, if you watch the Paul McCartney @ Abbey Roads special (2011? https://youtu.be/9elQeVfrLOo?si=qm8jLR_xLK-e2T_Z) where he is demonstrating (all solo performances except audience sing & clap along) the old recording processes he (and the Beatles) used with tape, and then his tech using pro tools, you can see the tech doing the manual latency shifts in near real time after each take and before Paul begins a new track (granted much older technology but same principle). much less complex than say trying to time align or phase align drums, etc. so no issues for me. note that my UMC1820 can get roundtrip down to 2.3ms on my desktop, and 5.2ms on my laptop, but really, why bother with the potential for combing effects from even minor latency when direct monitoring basically solves that. and my mixers (Behringer EURORACK UB1202FX ) built-in effects like reverb and delay provide singers with some of that without using up any of my CPU processing as well. so my path = mic / DI -> preamp -> IO -> IO USB to/from PC (record in + play out) + direct out -> mixer into headphone monitoring w/ any effects desired. so all clean mic and DI signals in via IO, and the direct monitoring and recorded play out via the mixer/monitor.

thanks! maybe another song for more cowbell 🙂 the xylophone with the Bb (which i think if louder would be too dragging) is used to make the swing between G & C slightly discordant and resolving alternately... cheers!

thanks! these are Synth V voices - Kevin is the lead male on the verses and singalong, and Natalie is the female voice on the bridge. both are also the backup vocals as well as a two more harmony voices - Anri and Solaria. https://dreamtonics.com/synthesizerv/ https://synthv.fandom.com/wiki/SynthV_Wiki i did use a xylophone to provide a constant Bb chord throughout the verse (when i play it live on guitar, it's the Bb note on the C and G chords) which somewhat acts like a cowbell 🙂 i did tweak the mix a bit yesterday to bring in the instruments a tiny bit (1db or so) more and reduce some of the bass level (as i hadn't really done a proper check across listening devices like i should have...)