Glenn Stanton

Windows does not have a native ASIO driver. what IO are you using? did you install the product ASIO drivers? if you recently did an update, did you reinstall the drivers? as a note, i find myself periodically reinstalling some of my drivers after a large Windows update and that has generally taken care of the problems related to ASIO driver behaving oddly (or not at all). i'm guessing sometimes the stack of SDK related bits installed gets confused and some of the winapi get used as the wrong version for some software etc... backwards compatibility not withstanding...

very nice work! couple of thoughts: 1) ASIO4ALL - maybe tweak this video or add an appendix video to showing how ASIO4ALL isn't really ASIO so it's fixed in peoples minds... 2) WASAPI is "was-A.P.I." or "W.A.S.A.P.I" not "was-sappy" ? like ASIO is not "ah-see-oh" ?

not all plugins install where the app manager can perform an uninstall, and many don't have an easy uninstall option either (esp if the plugin is installed as part of another software package). using the plugin manager you can see the location of the plugin file and the file name, then simply (using the File Explorer) click on the file, click again (now it's going to let you edit the name), use CRTL+-> or CTRL+END and type ! (so it's now .dll! or vst3!, etc), hit enter to close the edit, and say Yes to window warning you the file may not function... if you have a bulk renamer (i use StEx) you can select all the files you want to drop from the scans, and use the replace .dll with .dll! (or .vst3 with .vst3!). if the plugins have an associated folder, you can do the same thing, just add a ! at the end. then rescan. you should find all the files you appended with the ! are no longer showing up, and yet, if you decide to "restore" the plugin, just remove the ! from the file (and associated directory).

yeah, like HP PC of yore, a number of interfaces need their own ASIO drivers to function. Windows will recognize them in the Device Manager as Audio Interface, etc but only the IO device drivers will work. usually they make a WASAPI driver for general Windows usage, and an ASIO "equiv" for "power users"... not sure how much ASIO is there when they do this. to wit: newer UMC series by Behringer, M-Audio, Focusrite, Avid, etc. many seem to adopt the old "we don't need to use no stinkin' class compliant device drivers" that HP used to do to us, because, ya know? then again, if it's well made device with great features, ok then...

yeah, it would be nice to have the full latency total for a given track and by plugin in a small display or metric popup... hmmmm

in general, you need to know the "maximum" or "largest" latency which is used by the DAW to then set it across the board so everything plays in sync. so if i have only a single track with 5 plugins and none of the other tracks have anything on them, all the tracks will use the latency reported from that one track to set the overall latency so everything plays as expected. e.g. if that one track plugins add 10ms of latency, then all other tracks will be delayed by 10ms so all tracks are playing at the same time. (this is a simplification since internally i'm sure exactly what is happening but the effective results is the same). note that # of buffers used, as well as the sample rate will dictate the overall latency of the chain. CbB has a button "PDC" (plugin delay compensation) which if you turn it off will (should?) remove the compensation added based on what CbB determined based on the latency reported by the plugins etc. this (to me) should identify which track has the most because it will be out of sync. so you'd want to measure the latency of the tracks (or busses) to see which one has the most, and then how much latency that track (or buss) has. one way (manually) would be to use something like https://www.voxengo.com/product/latencydelay/ and pick a track with nothing else on it and adjust the time/samples until it aligns with the longest track. or try out https://www.kvraudio.com/product/vst-plugin-analyser-by-christian-budde some references on internal latency in a DAW: https://www.macprovideo.com/article/recording-and-production/understanding-plug-in-delay-compensation https://help.ableton.com/hc/en-us/articles/360010545559-How-Latency-Works https://www.16sounds.com/blog/zero-latency-plugins/

my son and his friends did a few months of recording using a 1920's washboard (apparently it was too much effort to setup and use my drum kit...) and apparently learned a lot on the way... if you're ok with someone with limited experience in playing a washboard and if he thinks he can fit it into his schedule, i'll have him reach out to you via this board.

as a note about the latency - when using direct monitoring - you'll be playing to the output from the DAW and hearing your drums directly - should all sound normal, however, once in playback, you'll likely find the latency has put the drums out of sync with the other tracks. having an audible click track (use it during recording or not) will make shifting the drum tracks to the proper position easier (rather than simply trying to line up based on the bass or other instrument peaks).

it looks like a muted waveform is there in the circle. read up on how to use the comping tools and you should be able to unmute/unarchive/etc whatever mode that take is in and use it.

in my experience - yes they're saved - you could verify this by setting up your FX rack, saving the file, closing, and re-open it. if the effects have their settings intact, you're all set. as a note, for complex set ups (say TH-3 with multiple amps, cabinets, and effects) i'd save them as a preset in the effect itself to ensure you are in fact saving things, and also you might reuse them as a starting point somewhere else. i'm not sure if saving a track as a "track template" captures all the settings or if they're reset upon importing that template, but again, i tend to err on the side of creating a preset within the effect itself. if the settings are something specific to a project or even a song, i'll save the preset with a name making it clear what it is, and with the project files as well as in the general presets folder.

i don't know of a shared template repository but i suppose if someone asked the moderators if it were possible to set up a persistent thread which people only posted their templates and related info only (a separate thread of people to make comments etc on to keep it down). and i suppose it would probably be nice to have some naming convention agreed to searching for a template could be made easier ?

pretty sure meter changes would be on the measure line, so if you have 3x 4/4 measures, then want to add 2x 3/4 measures, the 3/4 measures would start on the 4th measure: 4:01:000 and return to 4/4 on 6:01:000 not sure you can do it less than full measures...

yeah, that's the challenge. i think a lot of people who don't want to go down the road of mastering the tech in great detail usually learn enough to get a good recording and then take it to someone for mixing and mastering - including vocal corrections etc. one thing always impressed me is watching the John Lennon and Paul McCartney in a studio setting - how they were fascinated by the tech and learned to use it and expand on it in ways people are still trying to emulate to this day. it's somewhat amazing when Paul is doing some demo of his recording and he's just operating all the audio inputs and tape machines or laptop to get his recording... he then has his expert assistance doing the looping, rough mix etc in near real-time as well ?

i only have a single avantone cube for mono listening, and i like it. i use it extensively during mixing. the cool thing is when you final think your mix is sounding good, then you flip to the bigger stereo monitors and - wow! i am thinking of the new NS-10 active clones which have decent reviews.

it's not a bug because it is a manual step, and probably should remain so. making it easier though would be nice to have. i have projects at 44.1, 48, and 96K done at different times. right now, mostly using 48 (mainly because i'm working remotely and only have my remote rig which is limited to 48K and 16-bit) . i would not want my 96/24 files down sampled to 48/24 (files are always 24 bit, even with the 16 bit IO) when i just open it. i'd rather set my IO to 96/24 (assuming it can support, which it does on my home rig). same for a 44.1/24 project - i just set my interface to 44.1/16 (it's limited there but the IO truncation on the output stream is something i live with, the files are still 24 bit). in your case, you bought an interface which only supports 48K. so in that case you'll likely have to change the project sample rates on a project by project basis. presumably you're not editing or planning on editing all 200 projects...

are you rendering the region effect or just trying to "bounce to clip"? you need to render the region fx in order to incorporate the changes. one way (i do this) is to copy (clone) the track, do the region fx and render, then mute/archive the original. this way you can always go back to the original.

hmmm. it is strange. if i use a surround bus in 5.1 (with a stereo track feeding it), and route each channel to a separate physical IO channel (FLCR, Sub, RLR = channel 1-6 on my UMC1820) and it works as expected (i can pan it around a set of 6 speakers, for me to use my dedicated 5.1 cinema system, i have to encode it since it needs a Dolby (or equiv) stream). presumably you could use a stereo track and sends to go directly - you set it up for main out to ch 1-2, c and sub to 3-4, and rear to ch 5-6 - i do this in my UMC1820 and it is different than a surround buss because there is no surround panner - i have to balance it via the main and send levels to position it. i set up a stereo bus like a track - however - sending a track to this buss, there is no surround panner available (same as the track, no surround panner).

and once you go J-Bridge, there's no going back ?

a number of software companies offer bounties for finding bugs... not a lot of money, but whatever gets people looking and reporting them. of course whether or not the bug is fixed remains to be seen ? and then there are "spellcheckers and grammar checkers" (like auto-wreck, er, i mean auto-correct) which don't ever seem to care about getting things right... i generally agree with what Craig was saying "Me: Doctor, every time i touch my rib, it hurts. Doctor: then don't do that" ?

are the speakers using the 192/6? or something else? i think either way you should use the M-Audio drivers - i have a pair of M-Audio Delta (1010LT, 66) PCI cards in my desktop rig and they need to use the M-Audio driver (as limited as they are - at least they're clock sync'd to my UMC1820) https://www.m-audio.com/support/drivers-search AIR 192|61.0.310-09-2019AIR 192|6 Window Driver v1.0.3

Rupert was so badass that he had 2 Lava Lamps on his home rig!

one challenge for a new CPU (likely the best path for this MOBO) is the LGA1155 socket was superseded in 2013... so you're going to be limited to older Intel CPU (which from a price perspective may be good). an i7-3770K seems like it would be a small performance increase but 4 more cores and threads may be bigger than simple comparison shows: https://www.cpu-world.com/Compare/583/Intel_Core_i5_i5-3570K_vs_Intel_Core_i7_i7-3770K.html i'd update the CPU first then add RAM if you're loading large libraries/samples.

i use 12's on my '64 fender mustang, it's single coil pickups give it the proper "surfer guitar" sound. even when i don't want it to ? of course a decent fender telecaster will do as well. and as Bruno points out - spring reverb on a fender 12w amp will complete it. in theory, you could use some amp sims to create that rig - TH-U, Amplitube, Revalver, etc etc. however the source of the twangy-ness is that fender guitar single pickup sound...

i've found on some projects, my automation envelope was the issue - it was changed to a solid instead of the dashed (skip) line, so it went on forever. changing it to skip (dashed line) made it stop since the envelope was just a bit longer than the audio (by design - lets reverb and delays settle out).

sorry, the project (includes audio files) and the audio content (Cakewalk Content, libraries, etc) folders are in the Google Drive configuration - so all files are kept on the (separate) drive but not under the Google Drive folder. the OneDrive has all my other work (recording studio design, and other architecture) files, and yes, those are in the OneDrive folder and configured to always be kept on the device.