bitflipper

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.

Here's the change log. In all the years I've been using FF plugins, I can't recall ever running into a showstopper bug. Most post-release fixes have been obscure and often involve VST3, AU and Ableton. I use the VST2 versions and don't have a Mac, so most of this list does not apply to me. Of course, I'll grab the update anyway. You never know, someday I might actually want to add a high shelf boost to a reverb tail. Added native Apple Silicon support to the AU, VST and VST3 plug-ins on Mac. Of course, Intel Macs are still supported via universal binaries. The minimum macOS requirement is now macOS 10.10. Fixed a bug that caused the right-click menu in VST3 plug-ins on Mac to appear in the wrong location on Retina displays. Fixed a bug in the VST3 plug-ins that could cause some plug-in parameters to reset to the default value after leaving the plug-in open without running audio for a long time (for example overnight). Pro-C 2: Fixed a bug that could occur in the VST3 plug-in in Ableton Live and possibly other hosts that caused attack and release to work incorrectly when multiple small audio clips are present on a track with silence in between. Pro-L 2, Pro-Q 3: Added support for Dolby Atmos 7.0.2 and 7.1.2 in the VST3 plug-ins. Pro-L 2, Timeless 2, Twin 2: Fixed glitches when moving the interface from a High DPI monitor to a regular monitor or back. Saturn 2, Timeless 2, Twin 2, Volcano 2: Fixed a crash that could occur in MIDI Learn mode or when making a drag-and-drop modulation connection if the number of active sources (e.g. XLFOs) is automated by the host at the same time. One, Twin 2: Fixed a crash when validating the Intel AU plug-in on an Apple Silicon Mac via Rosetta 2 in a native Apple Silicon host like Logic Pro. Pro-R: Fixed a bug that could cause Decay Rate EQ bands set to High Shelf with a factor above 100% to have no effect on the sound. Pro-Q 3: With Gain-Q Interaction enabled, Cmd/Ctrl-dragging a positive dynamic range in the display would flip the gain to negative. Pro-Q 3: Double-clicking on the instance name in the Analyzer panel and making it empty would not work correctly when the name was already set to the default. Saturn 2: Optimized CPU usage with various distortion styles, most notably British Rock and British Pop. Volcano 2: The mono AU plug-in now no longer uses the "FF" prefix in the plug-in name. Volcano 2: Fixed a bug that caused modulation connections to a filter on the right or side channel to display an incorrect modulation target.

For me, it's about 50-50. Like rsinger says, it depends. Reasons for using internal effects: Convenience, as they're often integrated into presets, and some are customized for a specific synth. Allows separate FX for each voice in a multi-timbral synth while still using a single stereo pair as its output and not needing an extra bus. Some effects are synced to or modulated by an internal synth parameter. Efficiency. Many built-in FX are simpler and more CPU-efficient. Saves having to own every effect, e.g. a separate flanger Simplicity. Most internal effects have limited controls; great if you don't need every parameter. Reasons for using plugins instead: Third-party plugins are often superior to a synth's built-in effects. You can freeze synths independent of their effects, leaving more options for the final mix. Reverb, in particular, is best applied to many instruments in a common bus if you're after a natural sound, as if those virtual instruments were actually in the same room. Plus it conserves CPU. Routing, e.g. sidechaining. Lots more possibilities for making your mix more dynamic through automation. More options, finer control and a larger UI. Ability to upgrade independent of the instrument. Fewer FX overall means you can learn them better, have a deeper understanding of how they work.

There is certainly plenty to talk about regarding the technical side of mastering. You probably don't want to get me started. However, the direction this thread has taken is appropriate, given that at the end of the day the goal of mastering is making sure your record sounds as good as possible wherever and however it's heard. You just know somebody's going to listen to it on Apple earbuds or laptop speakers, and it's going to sound like cr... er...sh...,er garbage. Somebody's going to listen to it on a train or in the car. The one thing you know for sure is that hardly anybody is going to ever hear it on your speakers in your studio. Mastering tries to make it sound the best it can, regardless of the circumstances. The single best way to assure that is to have somebody else do the mastering, preferably someone who's using a combination of technical standards, full-range neutral speakers in a neutral acoustical environment, trained ears and experience. Unfortunately, those people charge for their services. If paying for such a service is not an option, it's up to you to get as close as possible. It's actually do-able. A couple years ago, a fellow came onto the mastering forum at Gearslutz and made waves there by declaring that he could master his own stuff just as well as a professional ME. Needless to say, his comments were met with pushback ranging from skepticism to derision. It is, after all, a forum frequented by some of the best MEs in the business. Being a fair-minded person, I decided to listen to his material and see for myself. I did not know who he was, but a google search informed me that he'd had a successful band in the past, was now a solo artist and I could get his latest record from Amazon. So I did. And I was absolutely gobsmacked. Turned out, he was right. The record was brilliantly mixed and mastered, with an impressive dynamic range and clarity that's rare nowadays. And he does it all himself, from composing to tracking to mixing to mastering. Maybe he even has a shrinkwrap machine in his house, I don't know. But the final product isn't just as good as anything out there, it puts many contemporary releases to shame. So good that he's been hired to remix and remaster many classic albums such as In the Court of the Crimson King, Aqualung, and Tales from Topographic Oceans. He's done, or is working on the entire catalogs of Yes, King Crimson, Jethro Tull and Gentle Giant. If you want some technical details, Sound on Sound did a writeup that should satisfy your curiosity.

MAutoVolume from Meldaproduction is one I've used. Works well for lead instruments, too. Hornet has Autogain Pro, which is similar but IMO not as easy to use as the Melda one. But it's cheaper. There's also a blatant clone called VocRider. I haven't tried it, so you'll have to google it, but IIRC it's a freebie. What these have in common, including Vocal Rider, is that they'll adjust a track's volume up and down to keep the ratio constant against a reference. Sometimes it works well and sometimes this is definitely NOT what you want to accomplish. It's certainly not the only way to do it. Assuming your vocal track has been properly leveled already, it's usually going to work better if you leave the vocal levels alone and instead lower whatever is competing with it. This can be done with any compressor that has a filtered side chain input (meaning most of the better compressors out there). I sometimes use FabFilter Pro-C2 this way. Even better is to carve out a space in the spectrum for the vocal. Wavesfactory Trackspacer works like this. I use MSpectralDynamics from Meldaproduction this way if it's a super-busy mix, but usually that level of precision isn't necessary so I'll use a dynamic equalizer with a sidechain input. FabFilter Pro-MB can do this, as well as MDynamicEQ from Melda. Another technique uses volume automation instead of a volume-adjusting plugin. Something like Bluecat Audio's DP Meter Pro, for example, can create automation envelopes from any track that can then be used to automate another track or feed a compressor sidechain. There are a bunch more, I just can't remember them all. I recall one that predated Vocal Rider and was so similar when VR came out I remember thinking "Hey, Waves ripped those guys off!". Can't think of what it was called now. Maybe somebody else can remember what it was. Sadly, it was one of the things I lost when my computer got stolen.

Not really. In MS Word the clipboard contains the last thing you cut or copied. As soon as you cut or copy anything else, the original data is discarded and replaced by the new selection, whether it's text, formatting, or an image. It makes sense that Cakewalk would work the same way. The procedure under the hood would logically remain the same regardless of how you copied the data. The ability to copy a clip by holding down the CTL key while dragging is a great convenience, but it's just a shortcut to the standard copy-and-paste mechanic. Which is not to say Cakewalk couldn't offer separate clipboards for different types of data (which they may do, internally). However, the design goal is to maintain consistency so that all types of data (MIDI, audio, automation, markers, clip effects, ARA regions) are manipulated in the same way from the user perspective. So yeh, it's just something to get used to. But worth it in the end.

I should have said "I've no doubt scook will come along to answer your question".

Yeh, Stevie Wonder seems to do alright. That said, personally I prefer an analytical approach. Keeping your tracks below -8 dB is a good practice, keep doing that. But as noted above, relative levels are more important than absolute values. Back in tape days, we had to work hard to keep levels high in order to maintain a decent signal-to-noise ratio. But digital audio has (mostly) freed us from that concern. Tracks peaking at -20 or even -30 dB aren't necessarily a problem. Concentrate on getting the relative levels in balance first, then worry about bringing everything up in volume. Another salient point is that volume perception isn't about peaks, but about average levels. You can easily have a track that peaks right up at 0 dB and still sounds too quiet. Or you can have a track that peaks at -18 dB and is still overwhelming other tracks that have higher peaks. A compressor (or limiter, a type of compressor) is used to raise average levels without raising peak levels. Compression is almost always applied to vocals, often quite aggressively in most popular genres. So once you've gotten your tracks reasonably balanced, add compressors before proceeding to the final mix. Don't be surprised if you find that you end up turning those compressed vocals down!