Glenn Stanton

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.

testing it out, i don't see any extraneous locations so it seems to save it within the project itself. you can add multiple photos per track but only a single comment section. the trick is getting it off your phone and into the PC/Mac filesystem so you can load it via "Add Images". their site hinted sync via dropbox, but presumably if you have Google photo, Adobe Central, etc photo sync and can access via file system, you're good to go.

i noted in another posting Session Drummer 3 has multiple assignable outputs and works pretty well. I generally use AD 2 though.

the biggest challenge is the Mackie Control Protocol is undocumented, as is HUI. it would be nice if those specs could be published somewhere. this is about as close as one gets: http://www.midibox.org/dokuwiki/doku.php?id=mc_protocol_mappings

looks like my '61 Les Paul Deluxe is going to worth a lot, oh wait. never mind.... ?

this is almost worse than when Computer Associates acquires something... "where great ideas go to die"...

maybe just keep them in a PowerPoint in your project folder, or a Google Slides named as your project and keep one slide per revision (or some other ordering thing). this way you can easily reference them in another window on your PC or even your mobile device, as well as readily revisit settings for projects without having to actually open the project. it's a bit more work but my be a more effective solution once you get into a habit of using it. i keep photos in my Google photos under an album "0000 HW settings" and this way i can readily access them - but only if i'm connected... my option (if needed) would be to Create a Slides presentation and export it as PPT to my local disk so i can go offline.

not sure it can do that now, but it would be nice. a shortcut would be to copy those notes and flatten them into the respective key switches. then once you've done all the artics you want that way, convert to articulation map (if that is desirable)

@Harley Dear as a quick note - you might investigate the Session Drummer 3. when reviewing some of my older projects, i'd forgotten i used it for a lot of percussion as well as drums, and when i checked the PRV, it was all nicely laid out as if using a drum map, but it wasn't. there are 12 assignable stereo output. audio is based on WAV samples. tuning each instrument is an option as well as "width". you'd have to check out the latency between MIDI and sound, but it seemed ok to me. SS_OldZep.cwt

one of my oldest songs from the mid-80s - parents keep an eye on your kids and don't do drugs! or something like that ?

there's a limit to the testing that can be done without some loopback on the USB port(s). there is one from Microsoft https://docs.microsoft.com/en-us/windows-hardware/drivers/usbcon/microsoft-usb-test-tool--mutt--devices but it requires loopbacks. the same for PassMark etc. otherwise it seems like most tests are skewed to disk performance. not sure if someone clever wrote a USB test using a audio interface as a loopback, although the REW app does use for defining the IO hardware baseline calibration. in general, LatencyMon is the utility most likely to identify an issue and perhaps, unchecking hardware acceleration on the IO could help (if it has HW acceleration...).

latency - it's a combination of the drum synth taking take to output the audio based on the MIDI input, and the audio roundtrip between the software outputting it via your sound device and you hearing it, reacting via your guitar, and the audio path + any effects you might use (an amp sim for example). so several options: 1) render the audio on the drums and turn off the synth. 2) turn off as many effects and other tracks as possible while recording, 3) turn down the sample buffer on your sound device. if you're round trip is under 5ms you'll probably be ok, however, you may still need to tweak the recorded audio tracks. i use the "nudge" setting @ 2ms per nudge to quickly align the start of the recorded audio.

Addictive Drums 2 Session Percussion, a number of old soundfonts via sforzando, and recently i (re)found Session Drummer 3 packages include some decent hand percussion readily accessed via the PRV. for synth percussion sounds - Z3TA+2 simply rocks.

so, the quick answer is simple: calibrate your monitoring across your recording and playing equipment. 1) load up a band-pass pink noise WAV (not MP3) running -20db (you'll find a bunch online) 2) load that into a track in Cakewalk and loop it @ 0db on the sound card output 2a) adjust Windows and speaker settings as needed to get the levels (read up on gain staging if you need to) 3) adjust your loudest monitoring setting on your speakers so you're at 83db at your listening position (i recommend a proper sound level meter but many phone apps seem to be ok for general usage) repeat steps 2 & 3 with each audio player with all FX turned off and adjust players to get same level as CbB. if you're using separate audio interfaces, make sure you adjust the volume level at each interface on your Windows sound control panel and not the record/mix software ? for example, in my Windows sound card settings i have my UMC1820 set at 100%, my IO2 @ 98%, my RealTek at 80%, and fiber to 5.1 @ 70%. in my speaker controller output settings - SP-5 and Avantone @ 75% (3 o'clock on the pots) and 50% on the speakers themselves (12 o'clock) , and Bose 901 eq/amp @ 60%. 901 amp at 50% (12 o'clock ~250W/ch) ***if you have a speaker controller - check frequency and set sub crossover correctly with the speaker sets. example: my 2310 x-over set for 90hz crossover, parametric EQ set for Bose 901 series 1 simulation, and on the sub crossover dial is set @ 90hz with a slight 2-3db EQ bump @ 150hz (due to a room mode at my listening position so this flattens it for me) -- this yields an even (+/- 1db) 83db (at my listening position as measured by a digital sound level meter) at max volume (0db) for all my recording, editing, playing software. generally i listen at the 1/2 volume setting on the controller or around 75db. ----the exception are a pair of old beatup Fisher speakers run off the 5.1 amp (150W/ch) set along the instrument (guitars, bass, etc) amp wall for simply raw, don't care, translation testing, and general party level music @ 105-110db @ 2m.