Glenn Stanton

what happens when you create a new blank project?

reinstall CW and/or the redistributable package.

it looks like you have stereo tracks and master buss and the waveforms look to be stereo - so maybe somehow your conversion to "mono" didn't make the tracks mono? but just one side of stereo or split mono?

it would be interesting to get a list of the words that are offensive in Washington state...

the making of his So album - where the producer/engineer got him to use cymbals resulting an another massive album... one thing i found cool - even in his new studio - using an SM57 on a flex neck for recording over a boombox for monitoring his vocals... scary good skills there.

ASIO4ALL is a wrapper for the WDM drivers - so it might work as you can adjust the ASIO4ALL buffer size. EXOVERB, per their user guide is restricted to: Supported sample rates by installation: 44.1 kHz, 48 kHz and 96 kHz Download reverb files for additional support: 88.2 kHz, 176.4 kHz and 192 kHz Supported buffer sizes: 32, 64, 128, 256, 512 and 1024 i use ASIO4ALL when an older softsynth needs even boundary buffers (which EXOVERB requires) the SHARED WASAPI is itself limiting to "best" buffer size in a shared usage. ASIO, EXCLUSIVE WASAPI, etc all being full control have the option to give the active app the control to adjust the buffer. so not a "bug" in either product, just an unfortunate intersection of restrictions due to your IO HW.

one method is to setup a volume leveler (like concrete limiter etc) to set max levels and even the dynamic range (reasonably) and then a measuring tool to set the overall levels. for example: using a limiter (or maximizer) (i use Ozone) and a metering solution which can report on the overall loudness as well as peaks and dynamic range (i use Insight but there are YouLean and other metering products) to set the overall level (in this case - the "integrated loudness" which is the sum of all), short term (longer than peaks but maybe several seconds worth), momentary loudness (peaks generally), and the dynamic range (loudest vs softer sounds). so pick a level - for backing tracks in live situations - say -11LUFS (loudness units full scale = ~db) and a dynamic range of say 3-5db would (imho) give you enough loudness for those tracks and not have them disappear during lower levels (your volume control). your material will dictate the dynamic range but for live popular forms of music, you probably want it flatter than wider (whereas in a recording you might want it more dramatic) one other aspect - if the source material is WAV, maybe export as WAV and play as WAV. MP3 (and other lossy formats can change the mix somewhat because of the masking algorithms used to compress the file). and so an MP3 version could have unexpected changes due to that effect whereas the WAV file tends to preserve the mix "more correctly".

maybe it's something that the WASAPI doesn't like about the virtualized IO using Jack? https://jackaudio.org/ i've seen a number of products which require an "even" boundary on the samples in order to operate (e.g. 441 does not work for a specific VST, but 512 does).

it's important to always ask ourselves: what kind of a world are we going to leave for Keith Richards?

dear pay-for-software-company admin, please reset my password and send to my new email: joe.hakker@darkweb.net as i'm desperate to order new software - here's my licence # to prove i'm authentic. this way i can get to my credit card and other personal information like shipping address in order to buy lots of your great products for my, erm, friends. best regards and love, joe

divide and conquer. always try a new blank project when troubleshooting problems to define if it is system or project related....

just thinking out loud that the region fx state of takes in sequential takes embedded are being handled differently than the composite level. so one short fix would be to apply the region fx to the composite level vs at the take level. understood separate takes = separate physical file, and comp takes are embedded as appended.

unless i'm mistaken, like a database, new takes for a given clip are appended physically to a given track's wav files, and the "clips"/"takes" are the logical representations - so pointers are needed to create the composite view.

is the volume control on the IO unit up? sometimes when i switch i turn my master volume on the IO i'm not using. down. and then of course sometimes forget to turn it back up. so i suggest getting into "pilot" mode of checking settings: IO volume up - check, sound id profile correct for headphone type - check, master fader volume up - check, H/W fader (usually hidden) up - check. and now several years of this habit i seldom ever get any points during which no audio happens due to system settings. still happens when i forget to unmute a track here and there though... lol.

since it doesn't happen when you place the region fx on the main track, but does happen on the unmuted and muted take lanes (which is why you see two (or more) unmuted sets of blobs, i doubt this is an ARA issue as much as it's something that wasn't seen in a mono version of Melodyne and rears its head in polyphonic versions. i.e. when the region fx is applied to a take (which physically is a snippet of audio in a long WAV file which the UI displays as a "clip") -- if the information passed to the region fx is not aware of the clip mute/not mute status in that level of operation - so it seems that either Melodyne is choosing to ignore the state, or CW is passing all takes aligned to that clip.

yeah, that definitely seems like a bug -- the take position data seems ok but it's muted status is ignored so passing both takes... does this also happen if the Melodyne is applied to the main track or only when its in the take segment?

under $200 - Steinberg UR22MkII and not under $200 (usually) the Steinberg UR24C -- in my experience these are rock solid, well built and very clean. that said, i also use the UMC202HD and UMC1820 as my main IO units now.

try to keep your plugins into "standard" locations and then make sure those locations are in your CW plugin scanner directory list. almost all plugin installers (and install instructions) will want to place plugins here: C:\Program Files\Cakewalk\VSTPlugins C:\Program Files\Common Files\VST3 C:\Program Files (x86)\Common Files\VST3 C:\Program Files\Steinberg\VstPlugins C:\Program Files (x86)\Steinberg\VstPlugins any number of plugins will default to the Steinberg folders... so acceptance is ok... ? then C:\Program Files\Cakewalk\Shared Utilities\Internal (Channel tools and some older CW plugins) plus maybe the %localappdata% directory one (not sure what was put there, not by me ? )

i use the AD2 Jazz Brushes. or just play them myself (i play with brushes a lot, but not so much for recordings). if not for the EZ3 or AD2 brushes, i'd do like John suggested and make your own samples. it's really really hard to get authentic brush performances (really they're effectively "noise" by design - which in and of itself is cool since you can use brushes in anything from 20's Jazz though EDM)

popularity has its price - i'm guessing a large number of people want a "live" Melodyne version... i'd like a version where i can use my keyboard to assign notes, but i'm ok with it working as-is for now as well. plus, i have several "live" pitch products i use for "fixing" things on the fly to test if a take is worth keeping or not, or the singer needs some confidence ? (and if the side effects of pitch correction aren't too bad, then i know later in the editing process, things will likely work out).

yes. while they experimented on us to produce the best DAW in the world! mwah hah <fades to cackling in darkness> ?‍?

maybe troubleshoot the Melodyne issue? i have the studio version and multiple tracks with polyphonic instruments converted to MIDI - no issues over the last 4-5 years. so maybe there is some settings (like A/V exceptions? separation folder permissions?) which are causing it? one note - i'm running 44.1/24 and just noticed you have 48/24 - i've done 48/24 and used Melodyne without issue but it's just one thing to check - alignment of all devices and software to use 48/24. shouldn't impact it but also worth checking.

when exporting MP3 - all FX & etc + best rate possible (usually 320K) and Pow-r3 (if i'm posting to something with a maximum size (like 8Mb) then i may resort to 256K) when exporting tracks or stems to be reused in another mix - no FX & etc and whatever bit rate (for me typically 44.1 or 48K) and 24-bit or 32-bit float and no dithering. when exporting 2-track mixes, all FX & etc for the raw mix - with the Pow-r3 dithering; and for the mastering mix - no dithering applied. one interesting feature of Ozone is to listen to the effects of different codecs & bit rates and hear what is being removed. it's a good way to check mixes that are going into MP3 / AAC formats because it does affect the mix (like masking frequencies do).

Wifi definitely has more interrupts going on than Ethernet simply because of the way it works in terms of using multiple sets of connections to establish speed. one downside to using "airplane mode" to turn off Wifi is any apps that use the internet (A/V, OneDrive, Drive, Box, spy-ware, etc) almost all of them keep trying to communicate or at the very least "test" the connection -- so if you're going to turn off your Wifi, also pause or quit those apps as well (or boot into a "clean" profile).

and esp if you don't have A/V exceptions for the CW (and any related apps and folder) as the latest MS defender is not only trying to "cloud protect" you but also then uploading to the Internet while scanning all kinds of possible "signatures" which eat the disk IO alive. by adding exceptions to CW, Melodyne, and other apps and tools, even with the network on, my load times are very fast - 2 second for CW to load and about 5-10 seconds for most projects (unless i have a lot of Kontakt instruments to load, then 20-30 seconds for 5-8 instruments and about 6gb of their content).