bitflipper

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.

That does indeed look like it could be a bug. Unfortunately, I was unable to duplicate it. I followed your (excellent!) video as closely as possible, first overlaying two clips and then muting one of them via the clip context menu. Then I bounced to a new track. The muted clip was not included in the bounce, having been ignored completely, as expected. Does it act the same way if you bounce to clips rather than to a new track? By that I mean select all clips, right-click on a clip and choose "Bounce to Clip(s)". Technically, it's the same as bouncing to another track except the new clip stays in the original track.

It's probably not a plugin, either. Easy way to check: create a new (empty) project and see if the noise is still there.

You can also CTL-drag plugins from one track's fx bin to another track to duplicate them.

^^^ This. Not an EW Play user here, either, but I assume it has a visual representation of the mod wheel in the UI, and that it is animated to reflect the value of either CC1 or CC11. If it's not moving, that would be a clue that the instrument isn't seeing mod wheel events. There may also be an expression slider that determines how the instrument responds to CC11. When it's all the way down, CC11 would be ignored. I don't know if this particular instrument offers that feature, but it is a common one with orchestral libraries. I'd guess it's probably not a MIDI channel problem, since the track does play, which indicates the notes are on the right channel. If you're putting CCs in manually via the PRV, they should automatically go on the same MIDI channel.

That pop is unavoidable when the audio's numerical value at the split point is anything other than zero, because otherwise you're telling Cakewalk to go from X to nothing in an instant. Speakers can't go from X to zero instantly, but they'll try to comply and move as fast as they can. That results in the pop you hear. It's the sound of your speakers being kneecapped. Ouch. The term "zero crossing" just means a point in the waveform where the sample value happens to be zero. It's called a "crossing" because unless the clip is silent that zero value occurs as the waveform is crossing from a positive to a negative value, or vice versa. Open Preferences (press "P") and scroll down to Customization -> Snap to Grid. There is a checkbox labeled "Snap to Nearest Audio Zero Crossings". Even if you don't have grid snap enabled generally, with this option selected you'll always select a zero crossing when splitting a clip. No more pops.

Could be a lot of things, unfortunately. Electromagnetic interference (EMI) from a nearby appliance, for example. We'll need more clues in order to help. One thing it's definitely not is a Cakewalk bug.

^^^ Well, there ya go. gustabo saves the day! Sounds like Play's version of Kontakt's DSD preload buffer adjustment. It determines how hard your disk drive has to work to stream samples.

I have experienced similar problems in the past with various instruments. Not with EW Play, as that hasn't been part of my toolkit for a very long time. But it's not necessarily specific to any one instrument. It's a speed issue with streaming buffers and/or disk I/O (e.g. if you're short on RAM). Unfortunately it's probably not something a DAW can address. If Play has an option similar to Kontakt's preload buffer size setting, that might help (depends on whether the issue is slow disk I/O or too-little RAM). Easiest solution: turn off 64-bit processing. Trust me, you will not be able to hear any difference. But you'll be using half the memory. There are some instruments that just don't like fast bounce, but having to wait on a real-time bounce is annoying. Fortunately, in your case 32-bit processing apparently gives your system enough of a breather to mitigate the symptoms, so that's a better solution.

"I'll read the manual. Maybe she can make me a sandwich." Glad to see the return of humor to the genre, which has always suffered from taking itself too seriously.