Glenn Stanton

in general, unless the source is stereo (e.g. keyboard direct in, drum overheads or room, guitar stompbox chorus effect, etc) record it in mono. on outboard effects applied to an instrument, it's always a good idea to record another track clean in case you want to "re-amp" later or just overlay a clean version etc. for MIDI capable instruments, record the MIDI as well. you don't want a good performance to not be captured with all options for using it in the mix. you can always delete unneeded stuff later if space or organization is a concern.

in general, if you have discreet mono tracks, LCR is just that. to widen a given instrument in stereo you can use mid-side EQ, delays (Haas effect) or have it recorded with a stereo mic setup (recommend M/S setup so it can collapse in mono). listening to the mix in mono (meaning tracks, busses, and a single center speaker all mono) allows you to find places for everything using panning and volume and EQ. then opening it all back up to stereo is often an amazing thing ? however rechecking in mono is important. why? most night/dance clubs use mono on their systems to avoid weirds holes in the music or other audio, people often share one side of their ear buds etc. so if the balance is good in mono or fairly even in ear side, your audience will often benefit from that effort. the Ozone Imager uses M/S EQ and has a stereoize Haas effect. which can result in phase issues. poorly done mic'ing can result in phase issues and time issues. skewing a double track (even if played fairly closely) can lead to what sounds like phase issues since it mimics a poor mic setup. and spanning an instrument via a delay can result in a "big mono" sound and may be what you want to not. so setting a stereo instrument within a stereo track (e.g. a leslie organ) means you need to have a means of controlling each side of that track with a panner or a plugin (like channel tool) to apportion it within the sound field (e.g. the leslie spans C+10% to L+60%)

if the tracks and busses are all set to mono and things are still "wide" then i'd look at: - windows sound settings - disable all enhancements; - speaker placement and make sure all DSP is turned off. having "room correction" running when the room is badly in need of fixing is going to result in poor stereo monitoring. as a general rule all mono input to the speakers should result in a stable center phantom image. if not, then something is wrong in the speaker path, placement, or room interference causing a tiny sweet spot (meaning even small head movements are catching phase shift effects) as a result of comb filtering etc. once that is working correctly, switching the DAW back to stereo and finding the plugins which are causing the "widening" effect should be much easier.

and then it's all crushed down into a 128K MP3 to be listened to on tiny ear buds ?

yeah, i have a bunch of old synth plugins which are 32-bit and i like to keep around. as more 64-bit synths come out that are better sounding i'll make the switch. so far JBridge is handling things perfectly for me.

JBridge, and then don't worry about 32-bit. worth every $. https://jstuff.wordpress.com/jbridge/

if the mixer has a USB IO connection, it can generally be used for your IO. if not, you could take the main output of the mixer into the 2i4 IO, and the outputs of that into an aux return on mixer or into a pair of line level channels (caring not to set up the routing to cause feedback loops). most times a mixer has only stereo in/out on the USB unless you buy a high end unit which supports multiple channels. as an example - i have an old Behringer MX2004A which has inserts on the 8 mic channels, so i normalize via patch panel the IO of the UMC1820 from the mixer which has better preamp levels for mics although when i want something really clean, or i am using a line out keyboard etc,, i patch in directly to the UMC inputs. since i'm only using the mixer for inputs, i route the outputs from the UMC into my cross-over and then monitor switcher which feed the powered monitors, the amp for the 901s, or the optical out to the 5.1 receiver.

if the USB mic is the source in CbB, then unless the mic has it's own builtin volume control, the raw input into the DAW will be 100%, so "turning it down" in your monitoring doesn't solve that problem. best bet - get yourself a 2 input IO unit with XLR and an SM57 or SM58. probably total under $200. as far as vocal-room interactions - you could build a simple set of frames - tied together - and hang heavy packing blankets of them. if the frames are tall enough, hang a packing blanket over the top. leave some gaps for air. some ideas: https://3dwarehouse.sketchup.com/model/d7c034208826744bfda112de017f7f93/Recording-Studio-Inexpensive-Vocal-Booth https://3dwarehouse.sketchup.com/model/1ed68da16d67650b6d2f7f4796305f20/Cheap-Vocal-Booth-Redux

trying using only a single Avantone with the master in mono. i also go back and forth between the single cube and the stereo monitors + sub. then a pass on the Bose 901 for the brutal truth test ? and then repeat ?

a while back i was watching a mix engineer show how he would calibrate the in and out levels so each effect/chain had net zero increase on levels - turn up the gain a channel, that gain is (mostly) the same on the output of the chain (and levels between each effect to avoid clipping - both on the external analog gear and plugins). so for the past few years i've been doing that - gain staging to avoid clipping and net zero level increase on the effects/chains. when trying things i ignore this temporarily, but once settled on something i'll balance them, save as a preset, and off to the next track or buss. and of course, sometimes i violate the rule to make it sound like i like it ? i had tried a number of the gain management plugins but they were generally was too resource intensive for my systems. reusing gain staged and level balanced effects chains is pretty simple so i just go with that and no additional plugins needed for a minute or two of work...

firstly - set low freq sources to mono. you could split the low frequency instruments a send to a low freq buss and another to the higher freq buss. low pass filter everything on the low freq buss so only 150hz and below are coming through and set it to mono. high pass the high freq buss at around the same. if you have the Waves StudioRack then you can use the built-in split on freq option. you could just send the mono kick and mono bass outputs to a low freq buss, mono, compressed (to taste), high pass everything else (varying as needed) (this is what i do) there are a number of "bass tightener" effects which work to mono-ize low freq into the center "sanford bass tightener", Waves has one, "bass lane" is another.

for tracks, i always just use the PC FX chain for non PC effects and leave the effects rack turned off. sometimes on busses i'll use the effects racks but i'm gradually moving away from that and just using the PC rack. it might have been a leftover from the days where people suggested not using a master buss because the recording comes out cleaner (something about the summing process not being perfect) when all tracks are exported "equally" to the output file or IO.

agreed, there are definitely times you want the aux track or buss approach.

i put the side chain compressor (or expander) directly on the target track (e.g. bass) and then use a pre-fader send from the source track (e.g. kick) to the target track compressor. no additional aux tracks or busses needed. doing this i can add several pre-fader sends to other tracks (e.g. rhythmically triggering compressor or other effects on pad tracks).

worldwide- how many are actually using a DAW? when you factor that in to the total number of people with computers and phones, it's tiny. it's probably why many people are using 2-3-4 DAW products in their personal and professional work, not including related stuff like audio/video editors, movie/video creation tools, and not mentioning effects even... i use CbB pretty much exclusively (and SPLAT before) but i have PT, Cubase, Audacity, and Live and a bunch of ancillary software to allow me to handle projects originated in other DAW where i cannot simply import and export the track audio/MIDI (which i can probably 80% of the time and only need the "real" software for the 20% when a project must remain in their preferred software).

agreed - cross fades work nicely, but many times i split and trim the sections and add fades (in and out) where needed to suppress pesky transients. this is because sometimes the cross fades are not enough to eliminate sharp transients. so - cross fades first, then split and trim second if there are still transients...

the Waves version gets some high marks as far as emulation goes given its harmonic characteristics and is often on sale... ? i tend to use the Waves version instead of the CW version because i like how it sounds... ymmv

not sure its a copyright violation as much as an equivalent to software theft or "unlicensed software". the CA-2A needs the license number as well as the install files. so someone would need to share their personal license number.

is it the same underlying disk or a partition on the same disk as the OS? i recommend a separate physical disk because (esp if its a rotating disk drive) the access contention will be higher.

this is a rabbit hole ? i prefer the dome tweeters myself - i find the ribbons are too harsh for me. but i know a bunch of clients who have the ribbons and absolutely swear by them.

sometimes drum replacer is your friend if it can't be used as @Bruno de Souza Lino noted about old school mixing ?

i have my OneDrive and Google Drive on my D content drive. With the Google Drive you can pick folders outside of the base folder, OneDrive unfortunately only allows choices within the base folder. when setting up the OneDrive you need to unlink it after the initial install, then move the folder, and relink it and then point to the folder you moved. https://support.microsoft.com/en-us/office/change-the-location-of-your-onedrive-folder-f386fb81-1461-40a7-be2c-712676b2c4ae

the problem with keep the OneDrive (or Google Drive) active while using it - they're aggressive in trying to synchronize the files and i've many times run into a collision which caused the DAW to crash or report permission issues. so before starting a session - i reboot my PC so its as clean as it will be, then i pause the OneDrive and Google Drive (and sometimes the network) and i'm good to go. Once done, i resume the OneDrive and Google Drive to sync and go take a break. since i've gotten into this habit - no crashes or reported conflicts with files, and only take a few seconds to resume.

i usually just record to a separate track while i operate the speed controls. or use the automation recording to do the same - depends on how much of a rush i'm in ?

1980 - HP 2116 / HP 2100 w/ 4K core RAM. when a bit got stuck you take it out and shake it and back in. programmed via front panel switches and mini bulbs. green tube dumb terminal and HP 7900 tapes for advanced stuff ?