Glenn Stanton

excellent, i didn't notice it and i was looking all over, and boom, there it is!

i could just put the MIDI on the instrument track. but i found that i can reuse the same midi track on other instruments to see how it works without having to move it around. and once i go into mixing mode, i just hide all muted tracks so it gets simpler pretty quickly. and i'm in the "buss" side of the equation so it's about 12 channels of instruments (if using all) and 4-5 ambience busses, and 3-4 stem busses. the latter i don't really change until i think the mix is overall balanced, then the stems are the final check and adjust - vocals up/down (in my case mostly down 🙂 ) lows up/down, etc. lastly a noise injection test maybe some slight tweaks on the sub-mix busses, a quick "mastering simulation check" by engaging a -6db limiter threshold and a tilt EQ - see if things are still ok at the increased volume - levels, harshness, etc. the template with all 15 soft-synths, and another ~50 effects (mainly the PC EQ, a few reverbs and compressors) across 48 tracks, and 20 or busses, i think a new project from the template takes maybe 25-30 seconds to load up. most of the soft-synth and effects are turned off and/or muted so loading is fairly low esp during recording (i'll set the ASIO down to 5ms - about 240 samples) and then during mix up to 2048 samples to reduce load when engaging/adding the effects.

is this some configuration setting i'm missing?

we may need some additional eyes on this one: @msmcleod @scook - any thoughts on this one?

unfortunately, Ozone 9 has issues if the GPU/UI resolution/OpenGL settings aren't perfectly aligned - i struggled trying different settings until it finally worked on my desktop... then had to retry a bunch to get it to run on my laptop... in general, if my standalone Ozone app is working correctly, i don't get crashes or hanging - but - i do find the plugins tend to be very heavy on CPU so i only use the Ozone plugins in the standalone app rather than in the DAW. it's extra steps in the workflow but saves a lot of anguish in DAW overloading...

not seeing ALL in the channel strip. missing things like sends, etc in the channel strip in the inspector, but visible in the console view (scrolling up) and in the track view. note: in the track control manager they're all selected. console view - when scrolled up it's there

there is a "slope" in the response, so no setting is generally a perfect "cut" or "boost" at a given frequency (hence "cross over") and determining where on the slope to transition from one driver to another has plagued acoustic transducer design since time immemorial 🙂 (well, since at least 2000BC)...

maybe check the stereo panning rule - i'm not sure if that is used when bouncing from a stereo or dual mono clip, or multiple mono clips into a single mono clip, but it might make sense if it was.

each track has a default PC (High Pass EQ only) and each sub-mix buss (one or more instrument tracks per each) as same PC (High Pass EQ only). then depending on what i want to do, of course i'm not limited to what effects i can put on the tracks or busses, and if i need more busses or tracks i can add them. add more effects like delays, reverb etc for 3D, and so on. it's just a starting point to the journey 🙂 i don't use dedicated instrument busses (unless we mean them to be the same as the "sub-mix" busses) since having a buss per individual instrument seems redundant. of course if your sub-mixes (similar to perhaps the "mains busses" in a HW console) need more, simply add more - if you do a lot of orchestral work, you might break it out as separate string, horn, and percussion section busses. the sub-mix controls the general levels per instrument group. The monitoring buss has the option to enable Waves Nx (in my templates) which is the virtual room for headphones. it goes directly to the HW IO. the master buss is where i tap off for the export since it has little or no effects, EQ, etc on it and i can choose to listen to the mix down via the virtual room while exporting. if you're not using a virtual room, or do not want to simulate the mastering levels, then you can remove it and simply route the master to the HW IO. the ambience busses are just (again) gross levels for a stem group - more reverb on the vocals or drums, more room across everything, less vibe during quiet parts, etc. also possible to automate. obviously these don't address instrument specific effects and those generally go on the instrument tracks and sometimes the sub-mix busses (i like to add Vocal Doubler on the BVOX sub-mix). for soft-synths - i print those to audio before mixing once i'm happy with the general sounds. i use midi tracks so i can disconnect them from the audio tracks later in the process, or re-use them for other instruments if i change my mind later. once i have printed things and disconnected and muted the soft-synths, i'll hide the muted stuff to get it out of the way. the noise uses a simple noise generating plugin (or you could import a clip) to check levels. i usually set it so it's about the level of the snare or a bit less so i can check that there is instrumentation and vocals above that (unless it's a snare exercise video 😉 ) i seldom print the "stem" busses for my own work. and again you could add more or use less "stems" as desired. in my case, i use them to simply make broad adjustments like bring up the vocals 1db, or bring down the solo by 2db, add 1db more bass and kick (low) etc. and they can be automated if needed 🙂 if i'm doing work for someone who specs they want stems, then i'll use those busses as the export source to the stem files.

same results here. even switching to use separate tracks. maybe the only way to edit outside of the comping tools is to use the full WAV file.

never heard of the 10Hz rule... in general, a LF spec of (for example) 50Hz on a speaker may be the -6db roll-off point or a -12db roll-off point where the manufacturer felt comfortable making the claim about 50Hz... so in general, see if they have a publish frequency response chart you can validate it and also keep in mind that speakers are generally tested in an "infinite baffle" type of configuration to reduce or eliminate baffle step and refraction impacts on the response. if they're honest they'll also publish a polar distribution chart or two to show how it acts in a free space. and some even go so far as to show how the speaker reacts in wall and corner scenarios. (reference: https://www.trueaudio.com/st_spcs1.htm)

yeah, the UNICA font can be tricky that way 😉 i do remember the groove 3 stuff from long ago... in this case, i focused mainly on the flow from recording -> mix -> export and less on specific effects or track processing since that is really material dependent. the addition of the dedicate "noise output" and a virtual room (monitoring) (or turning it off for speaker monitoring) are a couple of interesting ideas i've picked up - using noise to partially mask the overall mix to see what jumps out or isn't audible, and having a dedicated monitoring output so i can export from the master buss without changing it (like forgetting i have a virtual room enabled or the noise going...) in my use case, for example, i use a bunch of Slate and Waves plugins, as well are various instrument synths like the Kontakt-based Session Guitarist where i want to augment (or even replace) my own performance since it changes up the feel which can make things sound different than perhaps i was expecting.

a general purpose mixing and recording template - includes the use of the Pro Channel EQ and in some tracks & busses, compressors. Ambience busses include the Sonitus reverb, and/or Sonitus delay. I deliberately left out the soft synths because there are many many choices, and in my own templates, i use licensed products which may not be generally available. besides, would you ever learn something if it was simply handed to you? 🙂 in essence: tracks flow into sub-mix busses grouped by type of instrument tracks, sub-mixes feed into "stem" busses for overall gross-level setting, and those feed into the master buss. sub-mixes may have send to the "ambience" busses for drums, vocals, instruments, solos. stems may have send into a "room" buss for overall gluing. q&a yes - there are many opportunities to change, improve, or even wreck this template 🙂 hopefully it's a just start for folks. no - there is no support for this other than kind sharing between people and positive commentary. attached are some diagrams which provide insight into how the flow and console is set up. 000_mix_record_template.cwt

it may but i'd start with a more clinical approach to get the balance then tweak if needed - based on your experience with a given piece and how it should sound - i find usually this is done with some final monitor positioning and perhaps some very small monitoring EQ movements.

i have the SP-5B and a Sony sub (120W) and i set the crossover to 90hz and because it has a frequency bandwidth setting as well i allow up to 150hz ( a slight bump in EQ) to compensate for a room mode. but the main crossover point on my x-over unit is 90 and the setting on the sub is 90. that seems to produce a fairly even response. even with the Avantone Cube, 5.1 JBL satellites, and my Bose 901 as well. best bet - make sure the sub is well-placed and run a series of noise and frequency sweep tests to iron out the best crossover for your room and system. i would say it took be probably 2-3 hours of testing and tweaking to get it just right for me.

hmmm, in the open project plugins browser, sometimes i see both the 32-bit and 64-bit versions get installed and i usually find the 64-bit ones listed in the appropriate category (not all the times) and the 32-bit versions listed in "uncategorized".

maybe have a "MASTER" buss to be used for the export , and output that (or a pre-fader send from MASTER) into a "MONITOR" buss which has the Sonarworks or Slate (in my case Waves NX) on it. then export the MASTER buss to the file. output pre-fader send export MASTER

i've not used OMF much before, but freezing the track so the panning is embedded in a stereo track audio clip - it creates a stereo clip out of a mono track and the panning levels are embedded in the clip. i tried it with splitting other tracks and it seemed ok but maybe the export processing bounced them? i would recommend (like any project sharing effort) to bounce tracks to single clips which are then stretched to align on time 0, save the project as a new version of itself to a new folder so only the actual in-use tracks are copied, then export to OMF from there. this would also provide opportunity to change clip sample rates, bit depth etc for your sharing partner.

just stumbled across this and thought it might be of interest since there is a bunch of information here - sometimes quire old - that i hadn't seen elsewhere. https://www.sweetwater.com/sweetcare/manufacturer/cakewalk/

so first off, the foam is virtually doing nothing. except maybe off-gassing some chemical fumes. to improve the symmetry - close the closet door closest to the corner. then make a stack in each corner if semi-rigid insulation. you do this be running a 2x2 up each wall or on the closet door, and string across to hold it in place. cover with flame retardant treated cloth. curtains are such a fabric. do this in the back corners as well. on each side of the desk - build a 2'x4'x4" deep (600mm x 1200mm x 100mm) set of absorbers plus two for the ceiling. you could put the one in the closet behind the guitars if needed. make sure you have an equilateral triangle of your listening position and the acoustic centers of the speakers. the acoustic centers align with the tops of your ears and the point on the listening position is 12' (300mm) behind your head. this may mean the speakers need to come closer together and angled a bit more straight - try to get a 30° angle (or 60° depending on how you view things...) for each leg relatively. this positioning of the speakers and listening will help, as well as actual trapping in the room. you can glue the foam on the new absorbers if the styling of the foam is important to you 😛

yeah, it takes a few projects to fully appreciate the wonders of RX...

there just was an AAS SS2 update ( downloaded by not installed yet) a few days ago - sometimes version updates scramble previous version setting files 😞 although given the files are 1-2 years old, there may have been multiple updates... maybe the directories are off a bit, or a master INI (or equiv) file needs to be reloaded... worst case setting file formats/structure has changed...

not to mention that standard electrical levels are -10db consumer and +4db professional 🙂

you can treat the room with the absorption but also plan on adding some panels on the faces so you're not lost in the anechoic space 🙂 also, scattered absorption equals diffusive so some exposure of walls and ceiling can improve the listening space. angling clouds can help with adjusting response as well. badmac studio.skp

the ARC products use a variety of EQ and timing functions to try to flatten the response for a very specific location in the room. some include options to switch positions, reduce flattening for larger listening, etc. at the end of the day as @bitflipper pointed out the modal response will influence your listening and depending on the modes, may not be achieved solely with an ARC setting. presumably your room has an 8' ceiling and is 10'x10'. so as @msmcleod noted a square (or worse a cube) is hard to treat. that said, low volume listening, speakers with poor LF responses, etc somewhat enjoy the LF boost from such a room. so treating your corners (these are the vertical corners as well as wall-ceiling corners and sometimes wall-floor). foam treatments are generally not going to handle significant mode shifting as well as deep traps, and sometimes, tuned traps as well.