bitflipper

Yes and yes. They do it because it sounds good through earbuds, and they have a low opinion of their customers' technical acumen. Go ahead and name names, RR. We always like to hear about products to avoid.

Gotta wonder if there's really any money to be made creating presets for a Hammond. I suppose if you can sell beats, then you can surely sell drawbar settings as well. Trying to think what else hasn't been done yet...presets for light switches? (btw, not knocking the product; IMO it's the best Hammond emulation you can buy atm.)

So, to everybody who's already in 2021: how's it looking so far? Is it safe to come out yet?

Here's hoping that in 2021 people find the minds they've lost. Gotta have hope, eh?

Witch! Witch! She turned me into a newt! A newt? I got better.

You should have begun this conversation with the song, as it would clarify what you're going for. It's a good tune! I'm sure you'd agree that ambient music sounds simpler than it really is. The point is to make interesting and immersive textures, and your tune does that very well indeed. What you have to consider while mixing - and this goes for all genres - is constantly asking yourself "what do I want the listener to be listening to right here?". At any point in the song there should be a clear voice that says "Listen to me". Too many competing voices makes it difficult for the listener to follow along. There are places in your mix where it's not clear which element I should be focusing on. I can't tell you how to fix that, since it's a creative choice only you can make. But for every measure/phrase/section, ask yourself "what single element do I want the listener to be focused on, right here?". A more technical observation: the big climax is too hot. That's not a subjective call, but a measurable one. Overall, the macro-dynamics make a pleasant progression from quiet to loud, with a good peak-to-average ratio through most of it. Unfortunately, it takes too big a jump at around the three-minute mark and becomes distorted. Not the pleasant kind of distortion, but the fingernails-on-chalkboard kind that results from trying to exceed 0 dBFS and consequently incurring overs. I suspect you may have broken LANDR's algorithm on this one, as it doesn't seem to do well with big dynamic changes like this. Fortunately, you can fix this pretty easily. The principle here is that everything can't be loud. Start by just pulling all the faders down a few decibels to give yourself some headroom to work with. Then decide specifically which tracks are going to supply the beef, and pull the others down even more. I'd suggest a high-pass filter on the kick drum so it's not driving your (by which I mean LANDR's) mastering limiter so hard. Insert a spectrum analyzer (such as the free SPAN from Voxengo) and compare the guitars' spectra to the drums and see where they overlap so you can add some EQ to the guitars to help get them out of the way of the drums. The idea is to carve out a space for the drums and not have the guitars own the entire audible spectrum. You should post this tune to the Songs forum, where you'll get plenty of opinions, some of which will actually be helpful. I'd suggest posting an unmastered version, since that will reveal more measures that you can take to clean it up before LANDR mangles it. P.S. I, too am mostly "self-learned". Even after half a century of such self-learning, new lessons continue to remind me that there's no end to it.

A single reverb bus is usually adequate, but only if all the effected instruments need the same type of reverb. In your scenario, that's probably the case. The reverb plugin on that bus is usually set to 100% wet, and you'll use each guitar track's individual bus send slider to tweak how much each guitar gets. This send can also be automated, so if there are places in the song where one guitar needs a lot more reverb and other places where it needs to be drier, you can do that. The biggest mistake I hear in mixes is setting the reverb tail too long. That can sound great when you solo one track, but might get muddy in the full mix. How long is long enough depends on the song style and guitar tones, as well as the song's tempo. Think about what happens when a reverb tail carries on into the next note, and how that might make the part sound indistinct and distant. Decide whether the reverb should be an Effect that listeners are intended to hear prominently, or if it should be subtle, e.g. the reverb is there to simulate the guitars being in a reverberant physical space. If it's a David Gilmour style slow melody with pitch bends, slather on the reverb with 4-6 seconds or more. If it's a fast rocker, keep the times under 2 seconds. Bear in mind that at 120 bpm, each quarter note is a half-second and each measure is 2 seconds long. A two-second reverb tail will bleed into each subsequent measure, potentially blurring the melody. That can either be a good thing, or a recipe for mud.

You'll get as many recommendations as there are users, because everybody has their favorite go-to piano. Also, the piano is far and away the most-sampled instrument out there. That's good, because it means there are many options, ranging in price from free to $400+. There are a surprising number of good ones in the FREE category, but whether they're good enough depends on what you're after. Richard, you didn't say in your post whether or not you're a piano player. For some of us, the piano is our main instrument. For others, it's just one more element to throw into a mix. If you record solo piano pieces, or piano with orchestral backing, then you'll want a piano VI that sounds great on its own. I'm assuming that's where you're coming from. Which then begs the question: what does a piano sound like? A concert grand sounds different from an upright, a Yamaha sounds different from a Steinway. A serious "piano player" will likely have multiple virtual instruments to cover all the bases. Personally, I have at least 20 piano solutions here, and that's not unusual. My default go-to is a rather expensive one called Keyscape from Spectrasonics, but I'll often switch it up depending on the song and style. Some of my favorites were very inexpensive, although they typically require full Kontakt to use them. I'd start with YouTube. There are a number of videos there that compare virtual pianos. I know the thread asked for specific recommendations, but do the search as it will probably uncover something that fits your needs perfectly. Maybe even something we don't know about yet.

Something may have gone awry during the scan. I've had this happen when, for example, a needed file was missing. The scanner has a debug feature that can be helpful when troubleshooting a failed plugin. Rather than running the scanner from the top-level Utilities menu, open the Preferences dialog (press P) and scroll down to Files -> VST Settings. Check the "Generate Scan Log" option and click the Reset button. The reset forces a from-scratch scan, which is sometimes necessary after Cakewalk has marked a plugin as unusable because it failed the previous scan. Now click the Scan button to re-scan your plugins. The scan log will be found in %appdata%\cakewalk\logs\vstscan.log. This text file gives a blow-by-blow accounting of the scan process. The log can seem a bit intimidating at first, but feel free to post it here if you like. At the top of the file you'll see the pathnames of each VST folder where the scanner looked for plugins. Make sure the folder your amp sim was copied into is included in this list. If not, add the path and re-scan. Below that you'll see a list of every DLL that was found. Verify that your plugin is included. If not, you may have installed the DLL in a different folder than you thought you did. Next, search the log for the name of the amp sim's DLL. There will be a bunch of gobbledygook that might not mean anything to you, but again you can post the text here so others can have a look. Usually, if the scan failed it will say something like "plugin failed to load" with an error code. Cakewalk support can interpret the error code for you.

I remember you - one of the people who'd post serious replies to even the goofiest of queries. Good to have you back. The moniker change was a good idea. "losguy" just sounds like bad Spanish grammar. Should be either "losguys" or "elguy". Tone Ranger, much better.

I have actually used a PA for a similar purpose before, back in the 80's. I had a neighbor who insisted on playing his boombox in the backyard, every f*kin' day through the summer. Until late into the night and accompanied by the maddening boing-boing-boing of his kids' trampoline. The only sensible response was cannon fire, right? So I set up my PA - even dragged out the 18" subs - in my own backyard and drowned out his lame hair metal with the 1812 Overture.

My best - and most-expensive - purchase this year turned out to also be the biggest waste of money. A new PA for the band, which I bought in January when bands were still a thing and I still had a paying job. Sounds great, though.

Here's the background. And a link to the sick f&cks who are bringing down civilization this way. Turns out, the method works quite well with death metal. Who'd have guessed?

Do you love bass solos? Me neither. But in case you do, and just don't have the time to actually master the instrument yourself, AI has come to the rescue. This automated bass solo has been streaming nonstop since December 13th.

Low-pass filtering is usually a good place to start.

Oh, yeh. Voxengo SPAN was definitely a trojan horse, a back door that led to me blithely installing many useful plugins DIRECTLY FROM RUSSIA! (btw, this is a joke. AFAIK the Russians have no interest in tricking the NSA into compromising their sample rate conversions or master bus limiting.)

Start with an industry-standard microphone such as the EV RE-20, a solid boom and spider (that's the elastic web that isolates the mic from vibrations), then apply all the acoustical absorption you can afford. The BBC has long been the leading authority on setting up voice studios, having been the first to do so (going back to the 1930's) as they set up remote facilities all over the world for their world service. Lots of authoritative information is freely available online from the BBC on acoustic treatments. Many of their techniques involved a creative but science-based DIY approach, as they figured out how to use locally-sourced materials (e.g. bamboo diffusers) where things such as rigid fiberglass weren't available. If you're in the U.S. or U.K. then rigid fiberglass isn't hard to obtain, even if you have to special-order it from your local hardware store. You'll need lots of it. NPR voice tracks are heavily edited to minimize noise (e.g. breath noise, lip smacks, air-conditioner wind, traffic) and then heavily compressed to assure consistent loudness. There is a whole category of audio editing tools aimed at post-production treatments. You'd do well to invest in iZotope RX8 Advanced. Yes, it's expensive, about a grand IIRC. But it contains many time-saving features geared specifically for VO work.

I agree with Gswitz: there isn't really any such thing as high-fidelity earplugs. I have a pair of Etymotics, that I chose after careful research and finding that they are popular with classical orchestras. Yes, even all-acoustic music can damage your ears. A full orchestra going full-tilt can hit 130 dBSPL, waaay beyond the threshold of irreversible damage. I bought them because I'd just joined a rock band. It had been years since I'd been subjected to that level of aural assault, and I found the high volume annoying. I wasn't just concerned with protecting my already-compromised hearing , but also just trying to be able to hear what the heck was going on in the band. Long story short, I found them uncomfortable. Not the fit, that was fine. It was uncomfortable because everything sounded unnatural. Muffled. They're 1000x better than the foam inserts you get at the hardware store, but still definitely nonlinear. I stopped using them and shifted my strategy to talking the band into playing at lower volume. That worked.

Last week I noticed a news report that there'd been a widespread hack into government networks. It didn't set off any alarms in my head, since stealing data has been an international hobby for years. Then I watched the SANS Emergency Webcast from a couple days ago. And holy sh*t, this is big. When you think "hack" you picture some script-kiddie in his mom's basement trying to alter his high school grades. This ain't that. This is a highly sophisticated act of cyberwarfare. Caveat: the above-linked webcast will be very obtuse to most folks, as the intended audience is computer-security propellerheads. But I know there are a few here that will at least get the gist of it, even if you have to look up a few acronyms along the way.

Well, it is now.

Ironic. Behringer trying to protect their intellectual property.

Error 1 (buffer underrun) is the most common dropout code. It means that the computer couldn't fill the output buffer fast enough, so when the interface went to grab the next piece of data there wasn't anything in there. In short, the computer just couldn't keep up. Since increasing your buffer size didn't help, you're going to have to do some detective work. Something is preventing your CPU from having enough time to fill those buffers, and it probably has nothing to do with your project or your Focusrite. I've run over 80 tracks from a conventional drive (I also have a Focusrite, similar model but with Firewire) without dropouts. And my computer's not nearly as heavy-duty as yours. Your problem is that the CPU is dividing its attention between the DAW and something else. What could that something-else be? Wi-fi adapters are a frequent culprit. If your computer has one of those, try turning the wi-fi off and see if that makes a difference. But any hardware device can potentially cause similar problems. There are also dozens of background processes that can eat CPU cycles. You're going to have to do some sleuthing. I know, you just want to make music, not be a computer technician. Sadly, sometimes you have no choice. One tool that's often helpful is called LatencyMon from Resplendence. It's free. If your issue is with hardware, LatencyMon will tell you. Interpreting the information it provides can be a little confusing, but they do have some good tips on their site, and you can post the results here so somebody can have a look at them. Also check Task Manager, or better, Process Explorer. This free tool will show how much CPU each process is using, and thus identify any background process that's being too greedy with your precious CPU cycles. Given that it's a brand-new computer, I'd first look for bloatware that often comes preinstalled on new computers. Vendors will throw all kinds of useless crap in there (because software companies pay them to), and some of it is network-intensive (not good; network traffic trumps everything else, including audio). Getting rid of that garbage will make your computer happier in general, not just the DAW. BTW, what's your project sample rate?

Don't let anybody tell you it's not a real musical instrument!

I intend to try this myself as soon as we're legally allowed to have four people in close proximity. Jump forward to 14:30 for the demonstration part, in case you're averse to silly humor. Er, that'd more properly be "humour", I guess.