bitflipper

Two posts, both recommending the same loop library. I think User 905133 is right, this is someone making money through referral links, and is probably not even a Cakewalk user. "Her" email address is suspiciously nothing close to "Helen Aoki". That doesn't mean the library isn't worth $10. It actually doesn't sound bad, if building EDM songs out of canned loops is what you call "composing". If this user keeps posting the same links he/she/it (could be a bot) will be banned.

Last night's gig was on an island and the ferries had stopped running before it was over. As is our custom, we all crashed at the sax player's house on the island. My head hit the pillow around 2:00 AM. Exhausted, I put on headphones and was drifting off to the sounds of Hans Zimmer, when the drummer came in - furious and threatening to quit the band if we didn't do something about the bass player. The issue is that the bassist drinks too much. He's a great player and a good singer, scores points with me by helping to set up the PA as well as hauling gear in and out. But when he drinks his timing goes all to hell and he makes stupid mistakes like starting a song in the wrong key. The rest of us have all agreed to not drink during a gig, which has been my own policy since forever, in every band I've ever been in. He doesn't seem to be able to go along with that, making me think he's an alcoholic. Plus he's "sick" a lot. Plus he gets very argumentative whenever the subject comes up. I've known enough alcoholics to recognize the symptoms. Not sure how to proceed. Do I confront him and demand that he stop drinking at gigs, knowing it will probably be fruitless and just engender resentment? Or do I start advertising for a bassist and hope I find one as good as him but without the psychological baggage? (Insert bass player joke here. Ed?) Singing bassists are surprisingly hard to come by.

Electrical engineers actually have to deal with propagation delay, what we'd call latency, based on the lengths of conductors. Actually, it's about the inductive reactance that every conductor has. However, this only becomes a significant factor at very high frequencies, not at audio frequencies. You'd have to have a VERY long cable to get any noticeable delay. Inductance is also a factor in high-power transmission lines, but that's about synchronizing the 50/60Hz waveforms across multiple sources. There are, however, practical considerations when dealing with long audio cables. If they are coiled, their inductance will increase. If in a bundle, they will have inductive crosstalk. Then there is the distributed capacitance between the two internal conductors, which increases linearly with length. These don't affect timing, but they do affect frequency response. Standing waves affect the overall impedance and can cause a nonlinear attenuation and impedance mismatches. Then there are the electrical properties of inductive sources (guitar pickups) that also become part of the circuit. These are the reasons for recommending the shortest cables you can practically use. It has virtually nothing to do with latency. That said, a nice long 14-gauge power cable can be used as an improvised towing rope when the band van has slipped off a snowy road enroute to a gig. True fact.

Sheesh, they've sent me at least six emails about this. They are sure determined to give me Elements! Ironic that their thank-you for already buying Ozone 3, 4, 5, 6, 7 and 8 is a free copy of Ozone Elements. Can I exchange that for something useful, maybe a free upgrade to RX9?

Note how in Cactus' screenshot above soloing the MIDI track ("Kick") also solos the instrument it's driving ("SI-Drums"). Sometimes I'll use this as a quick way to check routing in a complex project where the audio and MIDI tracks aren't right next to each other. You should see the same thing when you solo your MIDI track: a companion audio track should also be automatically highlighted. If you don't, then that MIDI track may be mis-routed, e.g. to an instrument that's been deleted.

Dither is noise. Every time you dither, you add noise. So it's best to avoid it when it's not necessary. In your situation, it's really only necessary for that final 16-bit export. There is no advantage to working in 16 bits throughout, beyond saving some disk space. There are good reasons why we do most of our work on 32-bit files.

Good point. I had to check its specs because I couldn't believe such a relatively expensive (for a 2-input interface) device didn't include a jack for an external power source. It doesn't. You can also run into issues with inadequate current if you use a USB breakout box. For example, my keyboard has two very handy USB ports built into it. But I cannot plug in a second device if either my external backup drive or a DVD are already plugged in. Not enough current. There are also some mics that require quite a lot of current to operate correctly (e.g. Earthworks) that simply wouldn't be compatible with a USB-powered interface. I'm a fan of Focusrite, but I would not want this one.

Allows for multiple users/profiles. Makes it harder for malware to mess with it. Keeps things tidy with hierarchical organization. One database to back up everything. Common programmatical interface with built-in O/S support. There are a few good reasons for the registry. But yeh, there are times when I wish for a good old ini file.

Could be a corrupt registry. Make sure you're clicking Reset and not just rescanning.

This is a long-forgotten blues guy named PJ Proby. What makes the recording historical isn't PJP, but rather the backing band, who went on to form a pretty successful combo. They are John Paul Jones, John Bonham, Jimmy Page and Robert Plant (harmonica). I was unable to find out who played piano, could have been JPJ. It was the first time all four had been together in the studio. At first I wondered if this might contradict the popular story wherein Plant was found via an ad in Melody Maker, and that he'd had no prior professional experience. But the 1968 session date suggests that the band was already in formation, given that the first LZ record came out in the summer of 1969, IIRC.

Gotta agree with Carlos. I'm certain it's high quality; duh, it's from Spectrasonics. But $149 apiece does seem a bit steep, considering you already have to own Omnisphere to use them. For comparison, Indiginus' Renaxxance may not have 328 patches, but it's $59 and a damn fine nylon guitar. Mark, aren't you supposed to be on vacation? Get out of the hotel, man, and see the sights.

Here's a good introduction:

I have an NT1-A, too. It's actually a fairly bright mic. But back when I first got it (it was my first condenser) I did in fact accidentally record into the back side of it more than once because looking at it, it isn't obvious which side's which. So I got into the habit of singing into one side and then the other as a test, to make sure I had it the right way around. Another thing about the NT1-A is that it has a strong proximity effect and it's easy to overload it with plosives. It's a mic that likes to keep its distance, so you'll probably get better results by simply backing away from the mic a bit. Assuming, of course, that your room sounds OK and you have a little acoustical absorption in place. If the room sounds bad, standing further from the mic won't be an option. I, too, use a Focusrite interface. Not the same one you have (it's the Firewire version of Byron's 18i20) but with the same type preamps. They're pretty transparent and work well with the NT1-A. Their only limitation is they lack sufficient gain to do well with ribbons, but condensers like the Rode put out plenty of signal.

When I read "low end boominess in vocals" I assumed your issue was with the proximity effect. Standard mitigation tactics are a) use a low-pass filter, preferably one that's built-in to the microphone, and/or b) sing further away from the mic. Or use a different kind of microphone, e.g. a ribbon ($). But then you said that you could identify specific notes that boom, which points to possibly being an entirely different problem. One that's caused by room acoustics, and for which mitigation is not a simple process. First step is to experiment with different mic positions, but you've tried that already without success, suggesting that the room may just not be a good place to record vocals. Even applying acoustically absorptive materials may not help because you probably don't have enough space to apply the kind of treatment that absorbs low frequencies. Assuming, then, that the vocals won't ever get a whole lot better, we're down to fixing them in the mix. It sucks, because filtering resonances is a tricky business and rarely works well. But it may be where you need to start. The magic plugin you'll use for that is a dynamic equalizer. The best one I've found to date for fixing vocal resonances (yes, even better than FabFilter ProQ3 for this application) is MDynamicEQ from Meldaproduction. It's $84 but often goes on sale for half price. Get on the Melda mailing list so you'll get notified when it does, and snatch it up. (Skip the fancier MAutoDynamicEQ, you don't need it.)

Not following what this means...are you running SONAR as administrator and then loading Rapture into a project, or running Rapture standalone? Either way should allow the instrument to write to the registry as long as the parent process is Administrator. Make sure they're really running as administrator. If launching from a command prompt, you can verify that by looking for "administrator" in the DOS window's title bar. Anything you start from within that DOS prompt will inherit admin perms, including a Windows Explorer window.

As Promidi points out, the controller just spits out a standard range of values and it's up to the virtual instrument to decide what to do with them. Most sample-based guitar libraries limit the pitch bend range to a few semitones because beyond that it starts to sound weird. That's just a limitation of how pitch mod works on sampled material. Synthesizers, OTOH, have no such restriction and usually let you define the pitch bend range to almost anything you want. Two possible solutions for you. Get a sample library where slides are baked into the samples. Indiginus, for example, has some guitars that can do a one-octave slide. The other solution is to use a synth that can sound more or less like a guitar. Sometimes, substituting a synth works surprisingly well, not because anybody's going to buy it as a guitar but because it puts a personal spin on your interpretation of the song.

Because it's not what I think it is, it follows that it must be something I don't think it is. Making a list of things I don't think it is shouldn't be that hard. Or...if I think it's something musical, then haven't I already eliminated every possibility? Just every musical possibility, so it can only be non-music product. Something to compete against Far Cry 6. And because I definitely don't think that's what it is, then logically it must be that - you're welcome.

Someone in the YT comments said that Rush got their first radio airplay on an Ohio radio station. I wondered why Ohio in particular had a connection to Rush, and why it wasn't a marching band from, say, Ontario.

Ohio State Marching Band halftime show on 9 Oct. Great job by the arranger - drum line must have been especially pleased with the project. And some of the best marching choreography ever.

Of course, somebody has to come along and suggest the most expensive option...but it needs to be said: Spectrasonics' Trilian is just about the last word on basses. Not only does it have a great-sounding standup, it has a couple other acoustic basses too, e.g. a Martin that I like a lot. It's really a one-stop-shop for all things bass, including acoustic, electric and synthetic basses. Plus if you have Omnisphere you can layer Trilian + Omnisphere patches. At $299 it's pricey, but really a bargain if you consider how many instruments you're getting. If you have Kontakt, Orange Tree Samples' CoreBass Pear is excellent and it's only $59.

Oh yeh, I've had them both going at once.

By "problem", do you mean PT exports the same project in less time?

Learn something new every day! Thanks, Steve. You've saved me at least five seconds twice a year with that one tip alone.

First, it's not up to the user to tell an application how to distribute the workload across cores. That's automatic, and it has to be that way. An explanation as to why that is would fill a book, and has indeed filled several. Parallel processing is a tricky business. Second, seeing 25% CPU usage at the top of the CW window is not a bad thing. It just means the CPU has had enough time to process the data before having to spit out a buffer-full of data to the interface or to the disk drive. That's what you want to see. The CPU isn't slacking off; it's not a spigot you can open wider to get more work done. (That would be analogous to replacing the CPU with one that runs at a faster clock speed.) Now, if CW isn't using all the cores (actually, it can't use them all because Windows needs some), that's another matter. Although it is possible for the user to limit the number of cores that a program can use, by default Win10 assumes you want to use every core available, so you have to go out of your way to limit an application to N cores. Consequently, any time you see a core's usage drop, it's because the DAW is doing something at that moment that can't be further divided into separate tasks until that something finishes. That, or the O/S itself has some high-priority task that must take precedence, such as servicing an IRQ. There was a time when we'd tell Windows to give the DAW priority in thread scheduling, so that more CPU cycles could be dedicated to our primary application. We don't do that anymore because it's as likely to screw up performance as to improve it. You can, however, take a look at background tasks that might be competing for resources and see if any of them can be disabled during your DAW session.

Reminds me yet again how much I dig Solid State Symphony. Still my most fun Kontakt instrument for live jamming.