bitflipper

Just occurred to me that I'd also have to buy another disk drive if I want to retain the older Kontakt factory libraries. I currently have two 1TB drives dedicated to samples, both nearly full.

That's what it always comes down to: some must-have library comes along that demands an upgrade. Maybe history will repeat, I don't know. I already have well over a terabyte of Kontakt libraries. Every instrument category from basses to jug percussion is already covered, often many times over. My wishlist is currently empty. My favorite library developers sensibly (why intentionally limit your customer pool?) remain on K5 and probably will for some time. I go through this with every Kontakt release. As well as every iZotope release and every FabFilter update. Eventually, they'll get me with a price I can't refuse. Fifty bucks may just be that magic number, but I still have to justify it in my head. I do like vocal libraries, though.

I have Kontakt 5 (as well as 2, 3 and 4). It's stable, hosts all my current libraries and offers all the features I need. I don't need Kontakt 6. Please - anybody - convince me I'm wrong.

Welcome back, SiberianK. You weren't the only one who panicked when Gibson locked up the office. Many migrated to other DAWs, with the most popular destinations seemingly Studio One and Reaper. I was not among the refugees. However, I did try just about every Windows-based DAW out there, just so I'd have a backup plan (I liked Samplitude, only because it was most similar to SONAR in workflow, but not nearly as sophisticated.) But I kept on using SONAR, knowing that a) it wasn't going to suddenly stop working, and b) already had all the features I needed. The good news is that today's Cakewalk is in fact SONAR under a new banner. This would be, what, SONAR X6 by now? Same brilliant fellow is handling its development, except that now the focus is on usability and stability enhancements rather than marketing-driven bells 'n whistles. Although there have been some truly excellent bells 'n whistles added that even a set-in-his-ways curmudgeon like me can love. Articulation maps!

Automate the cutoff frequency for the filter(s) in the EQ. That's a little tricky if you're using the Sonitus "Telephone" preset, as it uses three filters, IIRC. But do-able. When I do that particular effect, three automations are involved: 1. Cutoff frequency on a high-pass filter is automated (start high, automate to gradually lower) 2. Volume is automated (quieter at start, then louder as the full-frequency range comes in) 3. Reverb send is automated (more reverb transitioning to less reverb)

My grid's always visible whether snap is enabled or not. Is there more than one way to toggle it? I always click on the snap unit button to turn it off and on. I don't really care whether the TV and PRV snap settings are linked, because I don't actually ever use grid snap in the track view, only in the PRV. That's just me, though.

Some plugins require an extra step to authorize them. I don't know about your specific instrument (I have a policy about never buying any product that uses the word "Beatz":) but instructions were either included with documentation you downloaded or perhaps in an email they sent to you after purchase.

Exactly. You want all the tracks to be in sync with one another, and Cakewalk goes to great pains to assure that's always the case. Consequently, total latency can never be shorter than the most time-consuming effect. There is really only one reliable solution: track dry and add effects later. Even with a very fast computer, any fx that require significant internal buffering (e.g. reverb) is going to determine latency regardless of CPU speed or buffer sizes. Everybody's approach is going to be a little different. Personally, I am never concerned with latency at all. That's because I treat fx as part of the mixing and/or mastering phase, separate from tracking. My buffers are always at 2048. Direct monitoring through my interface lets me do that for audio inputs, which are always either recorded dry or with external fx. My solution for MIDI tracking might not suit you, though: I use an external synthesizer, recording only MIDI, and then substitute a soft synth afterward. It's a simple process and I don't have to jump through any hoops to keep everything in sync.

This problem didn't occur during registration, but rather during installation. The error code suggests a corrupt or missing file. When you re-install it, do so as Administrator. That should take care of it. But if the problem persists, try renaming the Melodyne folders files and re-install again. Melodyne lives in c:\program files\Celemony, c:\program files\Common Files\Celemony, and c:\program files\Common Files\vst3\Celemony.

Yup, that's how it's supposed to work. Any time I plop notes into the PRV by hand, copy-and-paste bits or append an existing MIDI clip, I end up with many clips. It's really not a problem; the last thing I do on such a MIDI track is select the entire track, right-click on any clip and select "Bounce to Clip(s)". That combines all the little clips into one big one.

If you're after simple pulsating rhythms, as opposed to being sequenced, any tremolo plugin will do the trick. Create a bus, route all the tracks to it that you want to effect this way, and insert the effect on the bus. Meldaproduction has a free one that lets you morph the LFO waveform. Cakewalk's own Sonitus Modulator can do it, too.

That sounds like you may have a plugin running in demo mode. Try bypassing all effects (using the global bypass feature) and see if the volume drops cease.

It could just be a peculiarity of the plugin you're using. The way to test if the oversampling is sufficient is to look for aliasing. If there is none, or it's very low (< -70 dB below the signal) then oversampling is working. It's easiest to use a sine wave test tone for this, along with a spectrum analyzer such as Voxengo SPAN. As the amount of clipping distortion is raised, you'll start to see higher and higher levels of odd-order harmonics appear in SPAN's display (e.g. if your test tone is 100 Hz, you'll start to see 300, 500, 700, etc. Hz components appear). This is normal and is the primary reason you use a clipper. However, if some of those manufactured frequencies exceed the Nyquist frequency (~ half the sample rate), you'll get aliasing - unpleasant frequencies that are not harmonics. They're pretty easy to spot with SPAN, but not necessarily a problem if they're low enough to be inaudible. Oversampling raises the Nyquist frequency, shifting the aliased components upward where they can be filtered out with a low-pass filter without altering the tone. Each time you double the sample rate you double the Nyquist frequency, thus doubling the highest "legal" frequency the plugin can handle before aliasing starts. If the first aliased frequency is, say, 10 KHz but it's 96 dB below the main signal, you're good. If it's 1KHz and 12 dB below the main signal, then you need to increase the oversampling rate. This test will tell you how much oversampling you need for this specific plugin. It may end up being 2x or 4x, but probably not higher than that - unless the plugin is very badly designed. And you may get away with a lower rate if the plugin is followed by a low pass filter set to a high cutoff frequency.

You probably missed it because you were looking for something labeled "real time bounce", and the option is called "fast bounce". You un-check the box to enable real-time rendering.

Rendering is never real-time unless you expressly tell it to be by selecting the real-time bounce option. So normally, the CPU should have all the time in the world. I can't think of an explanation for your dropouts off the top of my head, but I can say that unless the clipper is extraordinarily poorly designed, 32x oversampling is overkill and unnecessary. Under most circumstances, 2x should be adequate. If it isn't, and you're hearing aliasing at 2x, try a different plugin.

My troubleshooting methodology for issues such as this: start stripping the project until the problem either goes away or you're left with a single track, no fx and the problem is still there. If the problem goes away during the process, you'll have identified - or at least greatly narrowed down - the cause. If it doesn't go away, you now have a minimal project that illustrates the problem. If you reach that point, send that minimal project to Noel & company. Or post it so the rest of use can have a look.

Recopying the project files should fix your problem. Hopefully, you still have the old computer handy and aren't relying on an external drive to transfer the files. If that doesn't do the trick, rename the project folders (don't delete them) and recopy the project folders.

I like your attitude. Don't lose it by doing things the hard way. Using the virtual keyboard is very clunky - timing can be tough and it won't register velocity like a proper MIDI controller. Take Byron's advice and get yourself a keyboard controller. A very basic one can be had for 50 bucks. If you're going to be programming drums and you're not a keyboard player, spend a little more for one that has pressure pads, which are more intuitive to most people. You might have fun with something like this: https://www.sweetwater.com/store/detail/LaunchK3mini--novation-launchkey-mini-mk3-keyboard-controller

I've found a way to keep my foot off the sustain pedal: play standing up. Can't play a four-hour gig standing on one foot. First determine if it's really the pedal that's causing the stuck notes. It's not the only reason for stuck notes. Open the PRV and the controller pane and select CC64. You'll easily see if there is a pedal-down event (CC value > 64) in there. If there are none, then it's probably not a problem with the pedal. (Gotta say "probably" because it's not impossible - but only if there is another MIDI track routed to the same instrument, which would be rare.)

Every DAW has to be told where to look for VST2 plugins. If one DAW finds it while another doesn't, it's just because the latter wasn't told where to look.

Plenty of bruises on my forehead from beating it against the proverbial wall. Not from anything music- or recording-related, though. Those hard-earned badges of honor came from trying to understand people, not technology.

I had a similar puzzle once. It turned out there was a duplicate clip buried behind the "real" one. That's why you need to examine the project - it could turn out to be something totally unexpected.

To be fair, you have to first be familiar with the concept of assigning different MIDI channels to key splits before you can even begin to look it up in the manual. In any endeavor, knowing what can be done is 90% of the process; figuring out how to do it is the easy part.

Try inserting the synth with just "MIDI Source" and "First Synth Audio Output" checked. See if that works. Dim Pro isn't actually a multi-timbral synth, but does allow layering. Are you trying to direct each layer to a separate track? I don't know if that's even possible, but it would be the only reason I can think of for using multiple mono audio outputs with this instrument.

What you'll need to do is go into the Launchkey's settings and find where it lets you assign different MIDI channels to the lower and upper portions. Any MIDI controller that supports splits should have that capability. You know what they say: when all else fails, read the manual.