Glenn Stanton

this is where a side=chained compressor on the other instruments could help by "ducking" some of the parts while the vocal is happening. might only need a couple of db to make it cut, sometimes using things like RBass/MaxxBass (or other LF harmonic generators) can help to cut low end on instruments by creating some additional higher frequency parts of the signal so you can reduce the instrument level slightly... which may/or not/ interfere with the vocal but it's something to try...

ok - so ignore previous suggestions. so the plugin is communicating outside of the audio IO path once it's configured to SEND. i'd think that the RECIEVE function would (if you needed it) be on a separate track or buss. one consideration: 0) MASTER is set to output to the Scarlett IO 1) install plugin on a new stereo buss which routes the SEND on Restream and also outputs the buss to MASTER. (so you can hear what you're sending to Reastream without colliding with the MASTER IO) 2) install plugin on another new stereo buss which has the RECEIVE (if needed) and also outputs to MASTER (so you can hear what is being sent to you on your audio system) 3) route output of your tracks to the REASTREAM SEND buss this approach should then let you independently manage your in-room/headphone volume separately from the Reastream send volume, and also mix in any receive levels.

it's a computer. everything is binary. one way or another. 🙂

definitely some nice specs: https://www.amd.com/en/products/cpu/amd-ryzen-9-5900x#product-specs makes me want to upgrade my whole PC...

i think he's referring to the clipboard history which is a collection of 25 or less last copied/cut items. note that there is a per-clip limit of 4mb and also, not all apps can use the clipboard history nor are all objects in an app storeable there...

which i also set to 24-bit 🙂

true - a 16-bit input being recorded at 24-bit will simply be padded, however once recorded, any "destructive" changes to the audio clip would benefit from the 24-bit setting as those LSB are populated from processing effects etc.

some thoughts to consider: "Clipboard history first appeared in Windows 10’s October 2018 Update (Version 1809). Currently, Clipboard history supports text, HTML, and images less than 4 MB in size. Larger items won’t be stored in the history. Clipboard history stores a maximum of 25 entries, with the oldest items disappearing as new ones appear. Also, unless an item is pinned to the Clipboard, the Clipboard history list will reset every time you restart your computer or device." https://thetechhacker.com/2020/11/04/how-to-view-clipboard-history-windows-10/

it might be helpful to post some links to short samples of audio where you hear the difference. it's possible that the same material through each is being processed differently - for example - dithering enabled? auto-level boosting? rounding in the processing? CbB using 64-bit internal then rounding down to 32-bit for file. floating point in and of itself is a "lossy" format (esp if you use it to calculate money 🙂 )

presuming your sending the output of your master channel (either a buss or your hardware interface)? it may be you need to route that output via your audio mix controller - this could be a mixer or audio interface then into the Reastream app. should check that you have exclusive control set off in your IO in the event Reastream and Cakewalk are vying for the same outputs/inputs. software-wise, it might require something like https://vb-audio.com/Cable/index.htm or https://vb-audio.com/Voicemeeter/index.htm note: i have not used these products but some folks here have mentioned them before.

i use busses for the sub mixes, but those feed into 3 busses (think of them as "stems"): low, vocal, instrument. the low only has kick and bass, instrument has everything else (drums, parallel drums, guitars, keys, etc) high passed like @marled noted. and the vocal buss contains the lead and backing vocals. separate busses for reverbs and delays - room, vibe, vocals, instruments. this is then all passed to the master buss which has a -2db threshold on the ProChannel limiter (or what you might have), auto-release (most times) and a slight -1db cut @ 250hz on the PC EQ. this provides a very light limit/gain on the master with the limit cutoff @ -1db. note: this approach leaves about 20 faders to manage the mix and general ambient/vibe effects even if there are 40-50 tracks behind it. so with the stem-like approach, moving the vocals +- 1-2db is easy, same for low and instruments, with the option to add side-chain compression to instruments so the vocals can turn down the instruments 1 or 2db, and same for ambience effects - those busses can be side-chained to bring the vocals a bit forward as well. this seems to provide a reasonable level of final balance (again think "stems") using the 3 main busses and a decent output to the mastering stage (me - Ozone 9 advanced separately on the WAV output).

excellent intro to vocals recording! i like how you pretty much nailed the verse in one take 🙂 i was struggling a little with my own breath control during that. 😉 one note for newbies you might want to include - save your project file once you get things set up like the import of the beat and the vocal tracks, this way all the audio etc is put into a project folder and of course you can use that trusty crtl+s to keep your work safe. cheers!

you should never mix on Bose 🙂 that said, i have a pair (or two) of 901's which are brutal for testing a mix - if they're right on the 901's, they're generally right every where. mixing on them? never. testing for translation? definitely. no idea about T1 but suspect the same as @John Vere

i use it mainly for monitoring the affinity of apps and the additional power performance settings but if i were to get into issues and needed to really fine tune it, i'd dig much deeper into the affinity and balancing settings.

this is a product i use: Bitsum. Real-time CPU Optimization and Automation https://bitsum.com/ they have a free version with some of the functionality.

and for those of us impaired with speaking a given language, there are translation services online 🙂 while not perfect by any means, they can be useful.

Avantone has a couple of NS-10-like units - a passive and newer active - both got decent reviews. http://www.avantonepro.com/cla-10a.php http://www.avantonepro.com/cla-10.php https://www.soundonsound.com/reviews/avantone-cla-10a i have Avantone MixCube (active version) which is very useful for true mono mixing

yes, hence the problem - which process gets priority on the execution side? OS and critical services (like file system, network, etc) - likely the OS - it's almost inevitable that one of the physical cores will get affinity set for OS which means one of the logical cores will operate impaired at best. and now with multiple cores on a chip plus issues with race-condition/timing issue security hacks, we'll see more physical cores in use than SMT/HT - certainly for the OS and related services. the cost is more power consumption and heat... but i'd like to have a CPU with 24 physical cores and 1TB RAM to make life a bit easier 😉

thanks @Wookiee - i deliberately let the french and english clash (historical maybe 😉 ) yeah, using Melodyne - you can convert a lot of stuff - guitar, bass, voice, piano, etc to MIDI. i often use the voice tracks to add some background pads, leads, etc. guitar i often convert chords to piano and organ, leads into sax, synth, etc. I have Melodyne 5 Editor so polyphonic works quite well. the only thing i don't find converts reliable are drums, but that's what drum replacer is for 🙂 my belief is: however you get the notes in, you can reuse them for all kinds of interesting instrumentation.

thanks! the woman is the French voice from google translate that i ran through with Melodyne to increase the melodic inflections a bit. Sax is my voice converted to MIDI then run through DimPro Tenor. i kinda deliberately left the drum parts simpler since there was a lot of other stuff going on.

it is simply the underlying physical architecture, memory channels, IO channels, inter-core communications, pre-execution queuing etc. so as much as Windows virtualizes the hardware, there is definitely affinity to the physical cores because at the end of the day the OS and its critical services need it in order to ensure it can do it's job. personally, i have to admit i have not looked too deeply into Windows 10, but i did set up some park control and affinity settings so my quad cores are as optimal as i can get them. perhaps it's not needed although as Noel pointed out they support this in the software...

the wire might not be actually faulty, just needed an upgrade 🙂 e.g. CAT3 -> CAT5-> CAT6 etc e.g. USB3 - can use same as USB2 but with the different protocol specs (faster speed, more power, etc) an older cable might not cut it.

something left over from the last, er, election-thingie. just in case.

the physical cores are generally responsible for IO activities, so it makes sense that the setting needs to be the physical cores. the "virtual cores" are simply some variant of a hyperthreading model to timeshare the physical core (since most software steps, RAM access, and IO activities take much longer than a thread swap, the manufacturers figure they can get away with that approach). Intel definitely seems to have a better handle on this than AMD but at about 2x the price for the equiv CPU... couple of other notes - most times a single core affinity is assigned for the OS as well network control, so if you have the option to set the core 0 to not be used by the software (sometimes this is the park control, or elsewhere...) this can help avoiding issues with unresponsive UI etc. (disclaimer, 8-9 years ago i spent 6 months working closely with Microsoft, HP, Intel, and SAP, and the MS SQL product team on optimization of SQL Server and physical IO for solving the record locking and concurrent disk IO issues for a 25K concurrent user SAP test POC) (and it's ok if none of that makes sense 🙂 )

the nff files are create by Melodyne - in a separations folder (either the global one, or per-project depending on using the UI or the region FX). these are the individual blobs you see in the UI. once rendered they're (supposed to be?) cleared, but sometimes i find them left over when close a project and forgot to render the track. finding the newest one and/or largest (sort of like organizing by alphabetical-height order) you can play it (assuming youo've set your media player to recognize nff files for audio playback) and it might help. all that said - it sounds like you got things sorted...