bitflipper

That was my first thought, too - that a defective plugin might be causing the phase shift. But if the polarity reversal actually shows up as such in the phase button's icon, well, a plugin can't do that. I'd start with a global fx bypass and play the song "5 or 6 times" to verify that it's an effect that's causing the shift. If the problem goes away, start eliminating the types of plugins that affect phase - a tape echo, phaser, chorus, flanger or delay would be logical candidates.

Hey, I celebrated my birthday on Sunday, too!

Burgers, brats and a jam session in the garage. My birthday gift was tickets to Trans-Siberian Orchestra next month. My gift to myself was a ticket to Todd Rundgren. And I know it's extravagant, but I also sprang for an EP from an obscure Australian band that I bought off Bandcamp for $3.37.

That didn't just put a smile on my face, I literally laughed out loud.

Just because it's mind-blowingly technical doesn't mean it can't also be fun. Somebody should convey that idea to all those grim-faced metal guys who think you won't take them seriously unless they look like they're suffering.

Building commercial Kontakt instruments is a lot more complicated than you think, even if you've created some instruments yourself. It takes great attention to detail and multiple skill sets.

The last skill a library developer learns is scripting, and because most of them have never done any programming before it's a steep learning curve. Even though I'm a proponent of giving newcomers a chance, I'm more comfortable with more experienced developers who know where the pitfalls are (e.g. hard-coded keyswitches) and how to avoid them.

I get to try out a lot of sample libraries in the course of researching topics for review, but I select only a fraction of them to write about because so many are just not ready for market. Some of those developers subsequently withdraw from the market after realizing that being a good recording engineer is only the start for a fleshed-out commercial product.

Gregg Schlaepfer of Orange Tree Samples and Tracy Collins of Indiginus, for example, are two guys who've been doing it long enough to have got the scripting stuff down. They make everything configurable, including keyswitch assignments, and rarely screw anything up. That's why I tend to steer people toward companies that have been around awhile, and who do their own scripting in-house as opposed to farming it out.

YouTube's algorithm suggested this...pretty obscure and you can only buy it on their Bandcamp page (5 bucks Australian, that's $3.37 US, a real bargain). Track #2 is in the same metal-ish vein but on acoustic guitars. 

There must be a very specific set of circumstances for this to happen. It could even be a corrupt cwp file. The CW folks could solve the mystery easily if you sent them the project (assuming the symptoms are reproducible).

Like others, I freeze and un-freeze synths all the time, including Omnisphere (that synth in particular gets the freeze/thaw treatment often, because it can be such a resource hog). Over the years I've witnessed a number of weird things happening after an un-freeze, but attaching the wrong synth has not been among them.

Could it be that you moved Omnisphere since the track was frozen? I did that once, after deciding to get Omnisphere off my SSD to recover disk space. The problem with this hypothesis is that you'd normally just get the missing-synth dialog.

I have seen one strange phenomenon wherein the missing-plugin dialog did not come up and the wrong plugin was chosen instead. But that was a highly unusual situation, in which two plugins had the same CLSID. That doesn't seem likely in your case, but I'm just trying to remember any comparable symptoms I've seen in the past.

Not a new band, just new to me. I got tipped off to this guy by Rick Beato in a video about odd time signatures. It's become my new favorite falling-to-sleep album.

Thank you, Jose. Seriously, I never knew what that icon meant. I just assumed it was my computer's coat of arms.

No. Homonyms don't count - read the T.O.S.

I've seen that phenomenon a couple of times in older SONAR versions. The workaround was to simply copy the track to a new one without including automation. I've seen nothing like that in CbB, so you might consider just migrating. All your X2 projects will open in CbB just fine and as long as you don't uninstall X2 you'll still have all the SONAR goodies such as PerfectSpace and PX-64.

22 minutes ago, Notes_Norton said:

I called the FCC, and the radio station and the FCC used the same engineer.

In a previous life, I used to be the service manager in a shop that sold marine, ham and cb radios. The CB'ers were always getting complaints about interference because they wanted to talk to South America and the legal 5 watts wouldn't cut it. Their neighbors would then call the FCC, who would send me out to do a field-strength measurement. Me, the same guy whose boss had sold them the illegal gear in the first place. It was an easy gig - all I had to do was call the offender ahead of time and tell him I was coming over, so he could stash his 1KW amp in the closet. I'd fill out a report for the FCC and that was that.

But then it happened to a friend of mine. His neighbor's illegal CB was interfering with all the TVs in the neighborhood. We had to come up with a more permanent solution. I suggested that he insert a pushpin through the fellow's coax going up to his antenna. Next day, he shows up at my shop with blown power transistors. I repaired it and took time to demonstrate to him that it was working - so that when he returned the next day with more blown transistors he couldn't blame it on my repair. He must have gone to another shop after that, so I don't know how many more times he blew up his amp before figuring out his antenna was shorted.

Yes, this post is on-topic. It's about cables.

It isn't necessarily a problem, but worth some investigation because it could become a problem for future, larger projects.

Defragging doesn't hurt anything, but probably won't address the high disk I/O you're seeing. And of course it won't do anything at all for the high CPU.

What you're seeing may actually have little to do with the project, but rather might be indicative of some other high-CPU processes that are running in the background.

A DAW's "cpu meter" is an indirect measurement of CPU usage. It's really based on how much time is needed to service audio buffers. Your meter is not showing that the DAW is using up 60% of the available CPU cycles; it is showing that the CPU is taking 60% of the time available to assure uninterrupted audio. It's a significant distinction. Most important, it does not indicate exactly what those CPU cycles are being expended upon and therefore cannot distinguish between what's going on in your project versus what's going on with, say, your network adapter.

Start by getting a baseline. Open the Windows performance monitor after rebooting. Let the system settle down for 5-10 minutes and make note of what the resting CPU, I/O and page faults are. If it remains high, figure out which processes are consuming the most resources. You might solve the mystery right there.

Next, start up the DAW. Ignore the metrics while the DAW is first initializing. After a minute or so, compare the perfmon numbers to what they were before starting the DAW. Other than RAM usage, they should be essentially the same. Open a blank project - again, the numbers should not change significantly. Load up your favorite synths (give any sample-based synths time to load) and then check again. If any of them add a significant amount of resource usage, you've found your culprit.

Synths (and many effects) typically use CPU while idle, but it should not be excessive. Some older synths and fx go crazy with silence, being unable to distinguish it from random noise. If you have that problem, there are plugins available that will fix it so that you won't have to abandon a favorite plugin.

Make sure you haven't inadvertently changed your ASIO buffer size. The number shown in the DAW's cpu meter reflects how much of the available time is being used to fill output buffers. Larger buffers means fewer of them per second, so more time between them. That's why the meter readings go down when you bump up the buffer size, regardless of whether excessive system resources are being consumed within or external to your DAW.

I've got lots of reverbs, some of which have built-in ducking, but none of them do this metallic culvert / vacuum cleaner hose sound. 

$159 for ATH-M50s (that I've seen for as low as $99) and they don't even throw in the 1/4" adapter. Sheesh.

Plus even if you really like the ATH-M50s (and I do) one thing you cannot say about them is that they are any kind of "reference". 

Periodic silence usually indicates a plugin that's running in demo mode. Bypass all your effects and see if you still have those gaps.

Don't know about the distortion issue, except to suggest that maybe your ASIO buffers are set too low. Or maybe there is some volume/expression automation or other CC data you're unaware of. Does the distortion always occur at the same point during playback, or is it unpredictable?

Jeez, if I created a template with ALL my libraries I'd be waiting an hour for them to load, only to ultimately crash the DAW when I ran out of memory.

And yes, it definitely is a slippery slope. But then, so is just about everything else. Kontakt just happens to have an especially low viscosity.

However, it can be quite helpful to make templates, especially if you're into orchestral stuff or use super-configurable instruments such as Superior Drummer. And while I fully appreciate Bill's reluctance to be constrained by pre-selected instruments, templates needn't be restrictive. You can create as many as you like, and you're not obligated to always use them. 

On 9/16/2019 at 10:35 AM, Hidden Symmetry said:

...Think it was called the over under technique.

I've always heard it called "the roadie wrap". But I'm always shocked at how few roadies know it or use it.

Like Brian, this technique has become so second-nature to me that I have to make a conscious effort to not roll a cable this way.  Which is kinda silly when it's a 3-foot patch cord.

Not a waste of time - because you took the time to come back and share the solution and thus potentially help out somebody else down the road.

If it makes you feel any better, I had a similar issue just a few days ago. I have a 6-year-old great-grandson who cannot resist turning the knobs on my mixer. Hosted a jam session here on Sunday and when I turned on the PA it erupted into ear-splitting feedback. At least that problem was easy to diagnose.

I thought this was hilarious. Unfortunately it's also telling about how social media can be abused.

Or maybe it's actually good news, since the fraud was found out. Maybe we're not all as gullible as was feared. Then again, I'm sure you can all cite exceptions to that hypothesis.

The good news is that no, panning is not broken. It works just fine.

If your master bus interleave is correctly set to Stereo, your mono tracks are all set to Mono, and you use a proper stereo panner on your stereo tracks, then there should be no problem.

Otherwise, it is almost certainly a plugin that's forcing everything to mono. Try using the Global Bypass to bypass all effects and see if you now have stereo. If you do, go back and disable the fx bins on each track and bus until you identify the source of the problem.

On 9/12/2019 at 2:18 PM, Karmamechanic said:

...They all box in the sound and make it less open and more fatiguing, even when the threshold is so high that there isn't even any limiting going on. Merely turning them on boxes in the sound. I can close my eyes, randomly click on bypass to set it to a state unknown to me, and then switch back and forth between bypass and enabled with my eyes closed and can tell 100% percent of the time when it's enabled even when it is not actually limiting anything due to the less open sound that it creates.

Something else is wrong besides the limiters you're testing. A properly-designed limiter should literally do nothing at all until the input exceeds (or comes close to) its configured threshold. It could be a routing issue, as David suggested above, or it could be you're not doing proper gain compensation when A/B-ing bypass vs. enabled.

All of the limiters recommended in the previous posts are ones that I'd call "properly-designed". Some are easier to use than others, and some are better at preserving microdynamics when pushed hard, but none of them should have any audible effect just by being in-circuit. Yes, there are indeed some that do just that - it's on purpose because they're trying to closely model some real-world electronic device (e.g. units with vacuum tubes and input and/or output transformers). However, such products will proudly identify themselves as emulations, and most digital limiters don't go to those lengths to mimic hardware.

BTW, any quality limiter will give you the option to enable "true peak" detection. Even if there isn't a button for it, all you have to do is enable Cakewalk's built-in upsampling feature, because that's what a "true peak" limiter is actually doing internally: upsampling. It's not some magic algorithm, just a higher internal sample rate.

When all my gear was stolen, including maybe a hundred cables that had conveniently (for the thief) all been gathered into two gym bags for easy looting, I had to replace a LOT of cables in one go. $60-100 per item was going to be cost-prohibitive. So I crossed my fingers and took a chance on Sweetwater's own Pro Co line solely because they were relatively inexpensive. (Not because I blindly trust Sweetwater; they also sell Hosa cables.)

Three years later and haven't had one fail yet, despite frequent abuse from coiling/uncoiling, being stepped on, having helpful bandmates quickly yank them out so we can pack out faster from a venue. No RFI or EMI, either, even in venues with questionable wiring and neon beer signs. And if they ever do fail, they all have metal plugs that can be easily repaired.

I'd also second Geoff's observation about keeping cables short. That's just a best-practice prophylactic measure in any situation. I, too, keep an assortment of lengths on hand - which is why I had so many in those gym bags. 

It is essential to apply compression before the signal hits the ADC.

Audio only gets converted to floating-point data after the interface, which works exclusively with signed integers. That means that instead of clipping you get inversion, e.g. an overflow that turns a positive number into a negative number. This doesn't sound like analog clipping at all. It is particularly nasty-sounding and noticeable even at low levels.

There are only two options for preventing this: analog limiting prior to the interface, or recording at a low-enough level that clipping can't happen. The latter is indeed practicable -- if you have a quiet microphone, a quiet room, and acoustical treatments to mitigate boundary reflections and room resonances. 

Just one of many unanswered questions surrounding that tale. 

We once had an amazing saxophonist sit in at a gig. As he was stepping off the stage, an audience member asked him if what he did was hard. His reply: "I guess it would be if I was a guitarist."