Starship Krupa

And of course my favorite soundpacks are the Chromophone ones. Y'know, the thing where Player can (presumably accurately) play back patches from any of their instruments suggests that they have one engine, and the various instruments just emphasize or give access to different aspects of it. Player is just the version of the core engine with nothing but a patch browser. Could be. It can certainly chew up audio engine resources. I switched on Cakewalk's 2X oversampling and whoa. Sounded great, but I'd probably want to keep it down to a single instance, or only oversample on rendering. BTW, for those who (like me) got free or bundled Strum soundpacks, or want to use the Strum patches that come with Player, I'm working on a how-to for that. It's similar to how it's done in Strum, but it's not done with switched articulations. Great fun once I figured it out. It's kind of odd that A|A|S sell Strum soundpacks without any instructions on how to use them with Player alone, but whatever.

I like to think that the polite but firm email that I wrote them detailing the mess that their installer made had some effect. I went into how after every single install of even just a soundpack I had to go around and delete all these megs of unwanted .dll's and 32-bit VST3's. Two releases later, and there's not a single spurious file left after an A|A|S install. Bravo to those brilliant Canadians. Having said that, though, the first time I open an A|A|S UI takes longer to display the UI (with all other visual elements in the DAW static until it finishes) than it takes to start Cakewalk. Same in Mixcraft. I don't know what they're doing, but it's annoying. Subsequent openings of the UI are fine, it's just the first that sits there. And if you're seeing performance hits with Player or probably the full versions, cutting the number of voices down from 12 to 8 helps immensely. Some of their pad sounds have really long tails that can stack up.

Yes, Vegas Pro Edit. I started years ago with Sony Vegas Pro, then upgraded via Humble Bundles. Feature-wise, I can't tell the difference between Edit and the full deal. I did a bit of hacking and copied the Vegas Pro FX to the same folder under the Movie Studio Platinum program directory and they all showed up and work perfectly. I realize that there could be an issue with my older nVidia card and using video editing programs: it's the older Fermi architecture, and supposedly Kepler was a huge improvement for video and photo editing. I used to have a point of frustration when video card specs and reviews were all about gaming performance, when I want something that will be good at running a preview window on a second monitor while applying effects. Also rendering, which can take hours on my current system. If I had known, I would have spent a little more and found a Kepler card. I just checked, and there are now sites that talk about which cards are good for video and photo editing and encoding, which is great. Even nVidia's site has banners for "Studio" solutions. I'm about to hit eBay up for a used card that has Kepler and DDR5 RAM. Should be an improvement.

I suppose I'm interested in producing sounds that other critical listeners can enjoy. And the others can get the gist, or maybe even experience it subliminally. Being able to pick out detail and soundstage in a recording is an exhilarating experience, even an emotional one. I've read people on hi-fi listening forums use the term "tearjerker" to describe the experience of listening to a favorite recording for the first time on a really good system with bit perfect playback. I've experienced it myself. It has an emotional impact, realizing how much is in there that I had missed. Just because most casual listeners won't notice it doesn't mean that nobody will hear it. And even for them, I do believe that aliasing and compression artifacts can lead to ear fatigue. Radiohead's "Everything In Its Right Place" is the song I use to test sample rates and compression, and it has so many little hidden things in it that I can only hear with a good setup. Tiny little reverse reverb on the lead vocal. Little sound snippets hard panned. When I first got a bit perfect setup going, I called my housemate down to listen to the song, and he did literally get tears in his eyes. It's one of his favorite songs. I'd like to create music that can help people have that experience, that rewards critical listening. It creates joy in me, and I'd like to do that for other listeners. My ambitions are humble, I guess. I'd just like to get my songs up on Bandcamp. Bandcamp has a really good sounding CODEC for their website, and they featured FLAC downloads. I spend money there because I can get my music in the format I prefer. So, to me, not moot. I'm not producing music aimed at teenyboppers (although I love it when my friends' kids listen to my stuff and like it). People don't stay teenagers forever (in numbers, anyway), and perhaps when they get older and we don't need to compress music to fit a thousand songs on a phone, they'll learn to listen closely. Whatever I put out there, I don't want it to sound crappy in 20 years when (I hope) playback systems will have advanced. My stuff may be forever buried by the sands of time, but who knows.

This song is great. Traditional Country seems to be such a rarity today. When I check out "Country," it just sounds like rock with a Southern accent. The world could use some more of what you do! Also, from a mix/mastering viewpoint, it's excellent. And condolences about Maggie. I used to have to close the door to my studio space to try to keep my dog Kittery out while I played drums, as I didn't want to injure her ears. She would then go around the other side of the house, through my workspace, through the guest bedroom, through the kitchen just to be with me while I was playing. She outsmarted me. Some say that dogs have the intelligence of a 4 or 5 year old human, but I swear, with my baby girl, it was more like an 8-year-old. My girlfriend moved out, and we shared custody for a while, then she moved to Seattle. I went up to visit them a few months before Kittery passed and we had some fun, took a trip to Don Bennet's Drum Shop in Ballard. She followed me into the vintage drum parts room, we found some parts I needed for the Slingerland kit, and then I sat down to have a tap on a Gretsch kit they had. Same as ever, she wanted to be by my side while I was playing, so I had to tie her off to a shelf in another part of the shop. BTW, for any drummers in that area, the shop is not only amazing, they love doggies. Amen to this. When it's going well, musicians (and other artists) seem to channel the spirit (by whatever name). I don't know if it's possible to be a musician and not be in touch with the infinite. We are given the gift of speech, but there are things that can't be communicated by words. When we're with a loved one, it's touch. A hug or caress says "I love you." I'm with you and Shane on the inability to recreate guitar solos.?I've done more than one "scratch" take on guitar that I couldn't top after several tries. And then I had to go back and learn it the same way I'd have to do with another player's part, note by note. And I think "where did that come from?" In my mind, John Coltrane, especially in the years with the quartet that included Elvin Jones, is one of the best I've heard at channeling it. There's even a church in San Francisco (where else, eh?) that considers him a saint. I didn't know what they were on about until I started playing drums 9 years ago and got into Elvin Jones. The gentleness and yet sometimes ferocity with which they play touches me in spiritual way.

What operating system are you using? Windows 10? Put some more information about your system specs in the post (see my sig for an example) and it will make it much easier and more likely for someone to be able to give you good advice. I Googled the error message (just put quotes around it before you search) and got lots of hits, many of them saying that it happens when some app or other (maybe a plug-in) tries to talk to the Microsoft Access database connector. Somehow the database connector gets corrupted. Then they outline how to repair the database connector.

That must be it. Find by trial and error which ones make a difference and only upsample those. The thing is, in my testing, the ones it made the most difference with (A|A|S Player, Phoenix Stereo Reverb) are ones that really challenge the audio engine. My Plugin Alliance fx seemed happy either way, and didn't really impact the engine either.

I use buses exclusively. The only time I use aux tracks is when I want to bounce some audio to one so that I can apply special effects like reverse reverb. One of the things that I like about buses in Cakewalk is that they are separated from the other tracks. Opposite from what you like. As far as templates go, yeah, I guess it's a tradeoff. I have tried to create templates whenever I notice myself repeating actions during a project. Such as putting my favorite compressor on the snare and kick, or my favorite reverb on a send bus, or setting up a bus for monitoring, my typical master bus chain, etc. The repetitive drudge work at the start of most projects. The more different a project is, the less valuable templates are to me. I have yet to get going with track templates, but there's some promise in that concept.

@LittleStudios, The odd thing about it, which @Teegarden points out, is that the Cakewalk documentation specifically says that plug-in oversampling on playback is supposed to be less of a processor hit than running at 88.2 or 96. The results of my testing (albeit on older systems) indicate otherwise. I'm not so concerned about the size of the audio files; 500G SSD's go for about $50. I have a 3TB backup drive. Of course, reading them is potentially another performance hit, but I have an SSD, so I'll just have to see. My current strategy is that I'm going to enable 2X on most of my plug-ins (except the Meldaproduction and other ones that have internal sampling) during render only. At this point, with the trailing edge hardware I have, it's more important to me have stutter-free playback while mixing than it is to have the quality bump. At render time, it doesn't matter, I can crank my buffers up to 200 and let it churn. Then I can be pleasantly surprised after rendering at hearing more detail in the finished mix. ?

@Teegarden, you were curious as to whether the performance hit results I saw were specific to my vintage laptop. I repeated my playback experiments with running the project at 88.2 vs. upsampling all the plug-ins, and same thing, switching the rate to 88.2 incurred less of a performance hit than 2X'ing all the plug-ins. And, when I played the project at 88.2 and all the plug-ins 2X'd, I got the same "bum note" in my bass arp track as I did when rendering that way. Did you read my post where I tried various forms of higher sample rates (plug-in and project)? I must be in the top.1%, because I can hear a difference, and to me it's not even that subtle. Yes, I agree that after going through <256 lossy conversion, the differences probably wouldn't be audible, but I buy FLAC's on Bandcamp and purchase uncompressed or FLAC albums and songs from sites like HDTracks (OMG, Radiohead's Moon Shaped Pool at 48/24). This is becoming more of a trend. If you look at the Cakewalk documentation, one of the benefits of higher sampling rates is said to be "Phase shift is drastically reduced." p. 976 of the Ref Guide. Now I've speculated about how it can be that two power amps can sound different, how it can be that there's so much of a difference in detail and image. In my studio, I have two power amps for powering passive monitors. One is a Crown D60 amp from the '70's, designed for radio station studio monitoring and other pro audio uses. It's "only" 30 watts per channel. The other is an Alesis RA-100, originally designed and sold as a mate to the Alesis Monitor Ones that make up one of my monitoring systems. It's rated at 70 watts per channel. I tried an experiment where I switched my Monitor Ones and my Boston A70's back and forth between the two, and the difference was stunning. I had another musician friend in the studio at the time and he was also blown away by the difference. He's by no means any kind of audio freak, but he described the speakers being driven by the RA-100 as "squeezed" or "choked." or just "smaller." Even in mono, the Crown had a more vivid image. I suspect that the Crown was designed with a lot of attention to phase distortion and maybe group delay. I think something similar goes on with lossy compression and plug-ins that may sound optimal at higher rates. So why would lower rate lossy compression sound "2D" rather than "3D." Theory says that since humans can't hear anything above the frequency range, it's not possible to hear a difference. My (fully admit self-educated) theory is that there are other factors like phase and group delay that get messed up. It's not just about frequency response. @bitflipper suggested that I read Fletcher and Munson to get a better understanding of it, but I haven't yet. Another factor is what genre we're talking about. I'm in awe of artists like Tipper and Telefon Tel Aviv, who create detailed sonic immersive spaces. There's a higher percentage of their fans who know that their stuff will sound better in lossless and with bit perfect playback, on a system with less phase distortion, etc. I'd like to impress those people too. Even for people who don't listen "critically," I think the extra detail can come through subliminally.

@LittleStudios, can you try your experiment but turn off upsampling at render and instead set your export rate to 88.2 or 96? I tried my experiment with playback in my MIDI-only project, and exporting at 88.2 gave pretty much the same psychoacoustic results as upsampling all the plug-ins at 2X. I know that you're trying to avoid the larger rendered file sizes, so just as an experiment. A possible benefit to rendering at a higher sample rate is that any ProChannel plug-ins will benefit. And those include an 1176 clone, a console emulator and a saturation effect, all possibly able to generate frequencies beyond the Nyquist. And Chris, thanks again for opening this topic up. As you know, I was skeptical, but I can hear a difference.

Neither Silverlight nor sliverlight (sic) has anything to do with SONAR or Cakewalk and its user interface. Silverlight was an audio/video streaming technology developed by Microsoft. Netflix and Amazon Prime both used it. Microsoft announced 8 years ago that they were going to end-of-life it in favor of HTML5. Skylight, on the other hand, is the name that Cakewalk, Inc. gave the concept of the dockable rearrangeable views. How there's the main one being the track view, and in addition to that the other ones which may be opened or closed or floated as needs and screen real estate dictate. With the Multidock, which is the usual home for the larger views like PRV, Staff, Console. The overall concept is called The Skylight Interface, the name has nothing to do with 3rd-party technology. It's one of Cakewalk's strong suits, IMO. Once I got good at it, I could really fly through the various views as I needed to focus on them, much more efficiently than something like Ableton Live! that tries to have everything up on the screen at once. I guess the idea is to have as much as possible instantly available (not surprising given the program's origins). The reality is that there are people like me still using a14" screen notebook. A DAW user is usually going to be focused on one task or another and doesn't need to see the Piano Roll and the Console at the same time. I have Vegas Movie Studio open on my other monitor while it's taking over an hour to render out a 4 minute video. I don't mind the layout, although it's kind of homely, but it forces me to keep open views and panels that I don't care about, and none of them can be floated outside the main window. It's poop for dual monitor work. The big brother, Vegas Pro, is a little more dual monitor friendly because I can float my preview window over to another monitor.

You can already drag and drop from the Browser directly to Sitala. Also drag a clip from an audio track to Sitala. The latter is nice for if I decide to switch over to triggering Sitala for one or all of my samples instead of having them in audio tracks. This also works with Speedrum Lite.

If I have a plug-in set to upsample on render, does turning the 2X button off affect rendering? Or does it only affect things when you have playback upsampling enabled?

When I moved the directory, I didn't know ahead of time that Cakewalk lacks front-facing configuration for some of the ancillary file locations, and wouldn't respond to just moving everything back. Turns out that if it doesn't find the files where it thinks they are, it just goes ahead and resets things to Cakewalk Core in your AppData folder. The last time I tried messing with mklink, I had a hard time getting my head around the syntax, but I'll give it another shot.

A classic zebra in the annals of punkadelia.

Websters says "marked by the ability or power to create : given to creating." In my older age, I have a tighter definition than I used to. In order to be "creative," a person must actually create things. Otherwise, all we're doing is creating thoughts inside our heads, and anyone can do that. The fact that my thoughts are about music or visual art instead of, say, car stereos, doesn't make me creative any more than thinking about boats would make me a sailor. Part of this attitude comes from dating "art girls" who came equipped with a portfolio of projects they did in art school, and who looked and acted the part, but after I got to know them, I noticed that they never actually created anything. Well, except for confusion and bitterness in the minds of people they were involved with. ? It's just a label, anyway. Or a conceit. Bottom line, create something. The process of actually doing so tends to stimulate more ideas and more creations. Even if they're for nobody other than me, I'm fine with that. I would ideally like to share my creations with others, as I have enjoyed the creations of others. But that's not mandatory. Making music makes me happy in and of itself and that's enough. Anything else is gravy. YMM certainly V!

I just added a second SSD to my laptop and moved C:\Cakewalk Content to it. I also changed the location of my Documents folder to the new drive. I lost most of my folder locations once I did this, but I sorted it out in Preferences. The only thing that's still missing, even after I copied Cakewalk Content back to C:\ are the stock arpeggiator patterns. How can I fix this? I looked in the registry, in the Cakewalk INI files, and could find nowhere that the location is specified.

Oh dear, I was afraid of this. Curses on you @LittleStudios! ? I'll write more later, but preliminary results indicated an audible benefit to either enabling 2X oversampling of the plug-ins or rendering the project at 88.2. It's similar to the difference I can perceive between <256 MP3's and FLAC's. Individual elements sound more discrete and there's more apparent depth. Switching the 64bit engine on while rendering made no change that I could notice. I did renders of the same piece with and without upsampling at 44.1, then with and without 64bit. All renders were to 24-bit WAV's. Repeated the renders again at 88.2. So the potential for highest quality would have been 88.2 with 2X upsampling using the 64-bit engine. More about how that turned out later.* I enabled upsampling on both the synths and the fx, I imagine that I could narrow down which plug-ins make the biggest difference via trial and error, and I would start with A|A|S Player and Phoenix Stereo Reverb. The audible difference seems to be the same for both states, the version with the 2X upsampled plugs sounds pretty much the same as the one rendered at 88.2, at least in the wee hours of the morning through my speakers. I'll do more listening tests tomorrow. *Weirdly, I did have one synth plug-in severely and repeatably misbehave when I enabled both 2X upsampling and rendered at 88.2. The bass arp track, which is A|A|S Player, starts emitting bad notes. The notes still sound, but one of them is off key when it cycles through the arp. Another synth also possibly sounded off rhythm with those settings, but it might have been that the bass arp was the culprit and was throwing me off. They sounded fine with all the other permutations. Conclusions so far: try rendering the same project at 88.2 or 96 and compare it to itself rendered at 44.1 or 48, because I hear a difference, and I was carrying a healthy skepticism going in. I'm actually biased against hearing a difference, because I would rather not have to concern myself with more settings. I have the suspicion that the improvement comes at a certain threshold and that 4X upsampling or rendering at 192 wouldn't make a further difference. It might, though, I haven't tried it yet. I can't draw any conclusions from the 2X/88.2 render because it's out of tune. Except it does suggest that my concerns about possible negative effects were well-founded. Now I'm really interested to know what @Noel Borthwick makes of my impressions.

Sure it's possible. It's a decade-old hand-me-down Dell Latitude E6410. About a year ago, I upgraded it by replacing its original i5 with an i7. I'm using its onboard (not Realtek) audio with WASAPI. I figured that even if the clock speed were slower, twice the cores and more cache would help with DAW and NLE use. The A|A|S Player engine sounds amazing, probably because its algorithms are constantly crunching a ton of analog modeling. Likely similar with Phoenix Stereo Reverb. This makes sense, the more modeling of "real" objects and spaces, the greater the load. I usually kick resource-hungry plug-ins to the curb, but both of these sound better to me than anything else of their kind. I'll repeat the test on my main system and report back. I sort of recall that one of Noel's systems is (or was) an i7 3770 like mine, so it may yield results closer to the documentation. For me, this stuff is interesting to mess with, but I think the true test is to throw on my best set of cans, upsample every plug-in for playback, and A-B test playback with the 2X button. Or upsample them all for rendering and make two exports, one with and one without. If they sound different, even if it's a placebo effect, then the can of worms is open.?

I just tried all of those operations and they worked. Given the oddities, are you sure the clips you were testing with were linked? It was hard for me to know until I figured out the options, as shown above. The documentation is misleading. The dotted outlines are hard for me to see. To get what we normally think of as a string of linked clips that link to each other and the original, you have to tick both boxes.

Indeed, for which I thank you. ?

Maybe my posts were too verbose, but if you scroll up a couple, I quoted a couple of companies' take on what fx may benefit. It seems that the more a plug-in might generate harmonics above 22KHz, the greater the chance that this information, while inaudible to humans, will get aliased back down into the audible range. If Omni Channel generates saturation or other distortion, then it might potentially benefit. Vojtech Meluzin (aka Meldaproduction) states the opinion that running the whole engine at 88.2 or higher is the best practice. But he also says that listening to the results is the way to know. @Noel Borthwick mentions that oversampling (and presumably also running at a higher rate) may help with phase shift, which very much interests me. He also mentions reverbs as being a type that may benefit. If you take a look at the results of the tests I ran, I came to the conclusion that if you want to experiment with the possible benefits of oversampling, running at 88.2 incurred a significantly lower performance hit than enabling all the plug-ins. But the plug-in that caused the biggest hit (no pun intended) was a synth, not a processor. My Plugin Alliance elysia mpressor, alpha compressor, and Millenia NSEQ caused much less performance hits. AIR Hybrid, much less of a hit than A|A|S Player. A thing to remember is that all of this is only important during mixing and rendering, so if you crank up the latency in your driver settings, any performance hit will be less of an issue. Latency is usually only an issue during overdubbing, and recording soft synths, so during that process, you can hit the 2X button or just bypass your performance-hungry FX. I deliberately dropped my usual mixing latency on the laptop to make the results more obvious. One of the things that led to confusion when I first started using Cakewalk was its use of the term "global." I've always understood that to mean "in the entire program," but that's often not what it means in Cakewalk. If you turn your ProChannel off on one track, the button to do that is labeled "Global," but it only means that all ProChannel modules on that track will be bypassed, not in the entire project. With plug-in oversampling, "global" means that if you enable it in one specific plug-in, let's say elysia mpressor, all instances of elysia mpressor will be oversampled. The "2X" button better fits my usual idea of "global."

I'm working on my 10-year-old laptop, and, inspired by this thread, decided to use it to see what performance hit I might see from enabling upsampling for the plug-ins in this project. It uses 4 synths and 6 FX, and the laptop can play it back fine with no upsampling engaged. Onboard audio CODEC, 736 samples. 8G RAM, i7 760 processor. Kinda old by current standards, but I keep it optimized and so far it has refused to fail to perform whatever tasks I've thrown at it, including DAW and video NLE work. It has a discrete nVidia GPU. Results: With no upsampling, engine load hovers around 45%, with a spike up to 62%. With all 10 plug-ins set to upsample 2X, it hovers around 80%, with multiple late buffers, stutters and usage spikes up to 146%. Clearly, this suggests there is some expense involved in upsampling. One of the synths, however, is A|A|S Player running a String Studio patch that arpeggiates. A|A|S's engines, while they sound amazing, are also the most demanding I run on my system. I limit them to 8 simultaneous voices so that they won't bring the show down. Turning it off for just that synth restored gapless playback with spikes up to 82%. Also turning it off for iZotope Exponential Phoenix Stereo Reverb brought the spike down to 72%. I was just guessing at what would be the most expensive plug-ins. Interestingly, switching the sampling rate in Preferences to 88.2KHz resulted in smooth playback, with usage spiking up to the high 90's. Barely viable, but less expensive than per-plug-in upsampling. This suggests that the tradeoff will be as Chris says, the amount of room that recorded and rendered audio will take up on the hard drive. If space on your SSD is dear, but you have a faster, modern processor with many cores, per-plug-in may be the answer. Since for me it's the other way (as it is for me, I just swapped my DVD+R drive for a second SSD on the portable and don't need to archive projects on it anyway), if I wanted the potential benefits, I'd run at 88.2. Rendering at that rate would mean an extra rate change before distribution, though. I'll try the same project on the main DAW and see what it says. It may just be that the upsample button on the laptop will have to stay off. I always do final renders on the main system anyway. Jury's still out on whether I hear any differences, but I will do further tests, including with headphones. Thank you, @LittleStudios for bringing this up. I've long wondered what the performance hit might be.

No, not at all, Chris. And I certainly apologize if I gave the impression that I thought you were "wrong" about anything. You've definitely experimented more with plug-in oversampling than I have. I come here for discussions, and I usually assume that "why are you seeking that feature?" when I don't understand, is a fair matter for discussion. I've learned some things from people that way. I understand that it can come off as "why the heck would anyone want to do that??" You said in your first post that "Upsampling plugins instead of a high sample rate on the project keeps audio file sizes wayyyyyyy down, keep the audio quality as good as digital can get it," which sounded simplistic to me. There are many things that influence keeping "audio quality as good as digital can get it," and from what I've read elsewhere, plug-in oversampling is a pretty minor one. YMMV. I don't really know much about this topic except what I've read in articles that all take pains to warn against seeing plug-in oversampling as a guaranteed positive thing and what I've heard switching it off and on myself. Which was nothing, but then I don't use a lot of guitar amp sims. Supposedly one of the best tests of whether a distortion plug-in is aliasing is to play a double stop and bend one of the notes. Never tried it. That's why I quoted articles and asked questions. I wasn't coming from a position of knowing it all, rather the opposite. Also, my ears turned 60 this year, and I played in rock bands for about 1/3 of that time, so if the aliasing is up above 15KHz, I may be missing it entirely. However, I'm one of those people who must set his computer's music playback up to have as close to bit-perfect reproduction as possible. I can hear the difference between CD audio played back via ASIO or WASAPI and via Direct Sound, and it's a big one. The former sounds "3-D" and has "depth" to me, while the latter sounds "blurred" and "flat" by comparison. The night that I first set my system up that way I stayed up until dawn listening to my favorite albums because I was hearing so much stuff in them I never had before. This effect is as apparent to me on the onboard sound in my laptop as it is on the interfaces in my studio, so I'm always watching out for this "blurring" being introduced by software I use and want to know about any tips for preventing it. I only wish recording and mixing at 88 or 96 made a difference I could hear, I'd do it every time. As it is, CD quality is plenty as long as it's reproduced correctly, without further conversion. I have witnessed the effects of aliasing via a spectrum analyzer while QA testing DAW software. I ran some tests of sine sweeps through a DAW's (not Cakewalk) sample rate conversions and found that a couple of the permutations resulted in visible (and in the audible range, 10K area) artifacts. I was inspired to do so by this page: https://src.infinitewave.ca/ If you're inclined, you may find it as fascinating as I did. And notice that SONAR, as it was when they tested it, gave some of the most crystal clear results. I have had one effect, ADHD Leveling Tool, a very nice LA/3Aish compressor, mess up badly when Cakewalk's 2X and 64-bit engine were both engaged. Its output level dropped way down. This suggests to me that there may be other, possibly negative, side effects to oversampling certain plug-ins, and I don't know which plug-ins or side effects they would be, so I keep cautious. I turn on the plug-ins' internal 2X oversampling for my Meldaproduction fx during rendering only, because what the heck, it's free, and I trust Vojtech to code the function correctly. I listen for unwanted effects and don't hear any, so it's all good. I have to confess that I don't hear any positive effect either, though. If anyone else wants to let it rip with 16X oversampling all their plug-ins, it ain't up to me to tell them not to, but I myself approach it with some caution and wanted to say so. The possibility exists, at least in my mind, that feeding audio into these things at the rate they're expecting might be best practice, at least for some of them. I'm not sure though. I doubt the Cakewalk engineers would create such a feature and make it that easily accessible if there were too many possible drawbacks. By all means, carry on, I'm glad that you had the experience, as I have many times, of finding out that the feature you were looking for already exists in Cakewalk. And if you can share more specific experiences with plug-in oversampling, I'd love to hear about them.