bitflipper

HD280 Pro, then. Good for editing, also good for vocal tracking, and they're cheap. I would never use them to kick back and listen for pleasure, as they have no low end. But that's what makes them good for editing and tracking. That and their very good isolation. ATM-H50 is also a good choice for editing, just don't be fooled by their hyped bass. Also inexpensive. Either of these can be had for under $100. FWIW, I've had both of the above for many years, during which the Audio Technicas have broken twice, while the Sennheisers are still like new. Nowadays both have been retired for editing duties, replaced by Sennheiser HD-558s. These are open-back, though, so may not be suitable if you need to be quiet or if others in your household refuse to be. But I am fortunate to have a quiet space here and appreciate the 558s' comfortable velour pads.

Depends what for. Tracking? Editing? Monitoring a live performance? Working out? Or just lying in the dark enjoying some herbal therapy? I have separate cans for each of these. Except for the working out use case. Don't do that.

Ever notice how well trombones go with distorted guitar? Me neither.

I'd suggest getting an SSD for your samples before replacing the boot disk. When I replaced my C drive with an SSD, the speedup was disappointing. Noticeably faster, but not life-changing. That's because most of what happens on your C drive is of a transient one-time nature, e.g. loading programs. Cakewalk and other applications start up quickly, but projects take just as long to load. Where the SSD pays for itself is when reading audio data.

Hibernation puts your computer back into the exact state it was in prior to sleeping, restoring the entire contents of memory. Including any wasted memory and unneeded background processes. Only a hard reboot gives you a clean slate. Yes, it takes longer to get going initially. I just go make a pot of coffee. I'd try to discover which virtual instruments are taking the most time to load/save. Unfortunately, there is no easy way to do that other than the painstaking process of saving and restoring different versions of your project with different instruments deleted from it. However, logic dictates that the largest memory consumers will likely be the prime culprits, and you can find out that information from Kontakt. I have identified instruments in past projects that gobbled insane amounts of RAM even though they played a small role in the composition, and those are the first things to get frozen. A frozen instrument has zero load time. Granted, I often have to un-freeze some of them later to tweak the composition, but in the meantime I can enjoy very fast loading times while working on other stuff. One thing I avoid doing is loading a lot of instruments into a single instance of Kontakt. Yes, doing so does save some CPU, but it makes freezing and un-freezing more time-consuming. Separate Kontakt instances for smaller groups of instruments gives you finer granularity when deciding which tracks to freeze. So while it may make perfect logistical sense to combine a whole orchestra into one instance, chances are there's a piccolo flourish or a tambourine hit that you won't need to edit and can therefore be moved into its own instance and frozen. Same idea goes for Omnisphere, an even worse memory hog than Kontakt. I've had slow-loading projects suddenly speed up after freezing a single Omnisphere instance. Non-sample-based synths are never a problem, so I'll often try Zebra for percussive hits and bells over Omnisphere. It might not have the same tonal complexity, but if it works in context that'll be one track that won't slow me down. Are you using an SSD for audio/samples? The newer NVMe drives are so frickin' fast it's almost like reading/writing to memory. I don't have one yet, but I'm just waiting for prices to fall some more. Unfortunately, I have over a Terabyte in just Kontakt libraries alone, so moving them to an NVMe SSD will be expensive.

Rebooting is the only way to fully reclaim allocated memory. What you've probably been seeing is the result of gradual memory loss (not yours, your computer's) that inevitably occurs over time. If you have a lot of RAM it may take weeks before the problem becomes noticeable - unless you're running extremely memory-intensive applications. This happens with all operating systems, but Windows has historically has been particularly guilty of memory leakage issues. I remember it being discussed a lot when Windows 3.0 came out. Back then a typical system might have only 4 MB of RAM, so memory management was something you had to be cognizant of. Nowadays we have so much RAM we can be pretty sloppy, leaving the computer running for months at a time and rarely if ever checking memory usage. So if you currently have 16 GB, doubling that to 32 GB should let you go longer between restarts.

The reason I asked is because my band's singer had asked me if it was possible. She's jonesin' for some quality band time but of course like everybody else around these parts we are all self-quarantined. No gigs, no rehearsals. Personally, I'm content jamming alone, but that's not an option for the singer nor her drummer husband. I'd told her I didn't think it was possible over the internet, but researched it anyway. I ran across this thread on Gearslutz where it was being discussed. I checked out the free open source software but didn't download it because I'm still convinced it isn't practical. Maybe on a local area network, but not over the internet. I stopped short of recommending a conventional online collaboration, given that of this six-piece band only one other member is conversant in DAW-speak.

How are they doing this? I had assumed that online jam software over large physical distances was inherently impractical due to the unavoidable network latency.

It'll work. However, the reason I use a smaller monitor (34") is that anything wider is going to impinge on line of sight to the monitors and skew their frequency response. If that's not enough screen real estate, get two and mount them one above the other. My older 24" screens are mounted on an arm and sit above the 34" display. One is used for plugins and browser, the other for the PRV or Melodyne.

Create a separate cue mix bus and put the reverb there. The singer will hear it in the headphones but it won't be recorded. Caveat: any software reverb is going to introduce latency that can make the vocal timing sound sloppy. I'd recommend buying a cheap hardware reverb unit just for vocal monitoring.

So the reason you can't use the configuration that works (ASIO) is that there are problems with 96 KHz and above? Is there a reason why you need to run the interface at 96 KHz? Does Pianoteq sound better at the higher rate, or are you just doing it for the lower latency?

Try to give legit acoustical advice and this is the treatment I get.

I work at home anyway, but I have little to do work-wise since most of my customers are shut down. And because I can't bring myself to bill anyone under these circumstances, I probably won't see any income at all for months to come. So no musical purchases this year. Unfortunately, I'd recently spent a couple grand on keyboard amplification that I now won't get to use, given that there aren't any gigs. I should have spent the money on toilet paper instead. I imagine 5,000 rolls stacked in the corners would make great bass traps.

Clean Audio Folder removes files that are no longer referenced. User 905133's problem is the opposite: there is a reference but no matching file.

In that case, there is still a (bogus) link to some nonexistent clip in one of your tracks. Unfortunately, there is no convenient way to identify which track it is. Here's how I handled that scenario when it happened to me... First, make a copy of the entire project folder. This will be a temporary workspace for diagnostics and you won't need it afterward. Now open that copy and start deleting tracks and saving the project. Repeat until it comes up without the missing audio warnings, and then you'll know which track was having the problem. In my case, I had some 60 tracks in the project so I used a binary approach to save time, deleting half the tracks, then half the remaining tracks and so on. Once you've identified the problem track, clone it but don't clone events (just plugins, automation and routing). Select all the clips in the original track and CTL-SHIFT-drag them down into the new track, which will be clean of orphaned links. Finally, delete the original problem track and away you go.

Your audio interface plays no role in exports, unless you're using external inserts. First thing I'd do is import one of those files back into the project and verify that its level has really changed, as opposed to being altered by SoundForge. Just to be sure, you're exporting these tracks as 24- or 32-bit wave files, right?

If you're looking specifically for a versatile envelope-follower/modulatable filter, I know of nothing else that's in Volcano's league. Certainly not any freebies. Main downside to Volcano is it's rather complicated if you want to dive deep. Meldaproduction has a simple but pretty powerful one (MFilter), but I'd wait for it to go on sale for $27. You can demo it for free.

You can't draw conclusions from statistics you don't have. In the US, we have tested less than 0.2% of the population for the virus. Based on sample sizes like that, one could conclude that not only are sharks not dangerous, they probably don't even exist. Ask anyone in Nebraska if they've ever seen one.

Are these mono tracks being exported as mono waves?

I sure hope 5-pin DINs and UARTs don't ever go away. They are superior to USB in almost every regard. As to whether hardware always beats software, that's another can o' worms. Digital hardware synths will usually have design restrictions that a computer needn't be subject to, such as reduced storage space and slower CPUs. The supposed superiority of analog synths is a moot point - those pristine waveforms are going to end up as digital data eventually. I have found balance by using both hardware and software. I play a high-end digital synthesizer on stage, but in the studio it's solely a MIDI controller. I can get any kind of sound I want from it while playing/recording, thus avoiding the need to ever play a software synth in real time. Consequently, latency is never a concern despite keeping my buffers at their highest setting all the time.

Almost no USB MIDI here. I have a MIDI multiplexer, a sort of 5-pin DIN patch bay with presets. Also some standalone splitters and a combiner so I can drive a common MIDI device from two keyboards onstage. My only USB connection is to a tiny portable keyboard that fits into my laptop bag. Its only purpose is to give me something to do in airport waiting lounges. No need to chain it to anything.

Maybe this is the true value of 432 Hz tuning. I've settled on 433.5 Hz just to be safe. Empirical testing has shown that to let in just enough cosmic vibrations and still fall outside the parameters of automated melody detection algorithms.

^^^ This is my logic, exactly. I'm self-quarantining. But I've been soaking my microphones in alcohol just to be safe. The surgical mask over the pop filter didn't work out so well.

Step one: verify that the DLL is actually where you think it is. Yeh, that falls under the "duh" category but it's always gotta be step 1. I've been bitten by not making it step #1, e.g. the plugin's installer put it somewhere I didn't expect. Once you've verified the DLL is present and where you think it should be, there are only two reasons for it to not show up: 1. The VST's internal registration failed, or 2. Cakewalk never initiated the registration because it didn't know about the VST's existence. Open a scan log and re-scan all your VSTs. Do a text search within the resulting log file on the DLL's filename. If it's not there at all, then it's scenario #2 (you haven't told the scanner where to find the DLL). If it's in the log but raised an error during initialization, the error may provide a clue as to what went wrong, e.g. a missing dependency or insufficient Windows permissions to create some file.