Starship Krupa

Charlie Daniels Band. Used to like him better when he wanted the world to "leave this long-haired country boy alone." Less a fan of his later work. (CBD is cannabidiol, an ingredient in a recreational substance that the leader of the CDB claimed to ingest in the morning)

Hit that "FX" button up at the top in the Control Bar and see if the problem goes away. If it does, you set up one or more of your FX in a weird way. Go through them and figure out what you did. Sounds like an interesting, possibly useful effect. Make note of it for possible future use. ?

No, it is past time to look for closed ear headphones. ? In the spirit of good humor, it does puzzle me when people say on forums that they are trying to salvage already-recorded tracks and then get suggestions that they use a different mic, or put a mic on the kick drum, or have the vocalist use a stand instead of hand-holding the mic....and this is when the OP has not self-identified as Time Lord nor are they in possession of any time travel device or method. I see that it's also past the time when processing suggestions would be useful for the OP, but I love the challenge of salvaging tracks. Maybe some future reader (see what I did there) can benefit. I might throw a dynamic EQ or gate at this one. First take a parametric like the Quadcurve or my favorite, MEqualizer. Dial in one of the bands to boost about 10dB. Then sweep the frequency across until the sound of the metronome click is at its most obnoxious and you've found the frequency you need to cut. You can start with trying to just cut it with the para and see if it sounds too intrusive. To get fancier, set up a dynamic eq to slap the frequency down a bit when it comes up. Another cool trick is to use something like Boz' Gatey-Watey to gate out only the frequencies you want to gate out. What, buy a $20 plug just for this? Nope, turns out that the Sonitus fx gate has this feature built right in, once you figure out what freq you want to nuke, just move the sliders, tell it how much, and set the detector. The thing is, usually while I'm playing, my guitar sound is covering up the click anyway (or it should be, ha).

It just means that the user 1. has Recycle Plug-In Windows set in Preferences and 2. clicks on the icon in the upper right corner of the plug-in UI that looks like a push-pin. If you have Recycle Plug-In Windows turned off by default, all your plug-in GUI's are "pinned" to begin with and you have to close them to get rid of them. This should have the effect of keeping all such "pinned" plug-in UI's open, but the behavior I have observed is that it only does so when the plug-in UI is open over the main window. It doesn't work over floated multidock windows. @Rod L. Short, maybe our issues are the same, and we always drag our floated windows to the second monitor! I'll try it the other way around and see which it is. IIRC, it's that the GUI's will stay open over your Track View, so please try it and we'll see.

It's possible that the custom uninstaller program is responsible for the antique version of the Visual C++ libraries. Those uninstaller programs can be a mixed bag, with the occasional baby going out with the bathwater in the case of such things as shared libraries. A few things that I have learned about AAS's software, being a big fan: First, they recently updated most of their stuff, so make sure you have the latest. Second, their installer is compliant with the Windows 10 uninstall list, so if you want to uninstall their stuff, it's all there in the Apps list in Windows Settings. Third, in my experience AAS are right behind Waves as far as strewing copies of their plug-ins all over the place. Seriously, I have wondered whether if a folder has the string "VST" in its name they just go ahead and stick a copy of their .DLL's in there. I've found copies of their plug-ins in my Cakewalk folder, my Programs folder, all over the place. 32-bit, AAX, 32-bit AAX, whatever. I don't like there to be more than one copy of a given plug-in on my system because multiple copies can cause issues with version control, so after an AAS install I go around and manually root out the extras. @Noel Borthwick, I, too, seem to now be having AAS woes in a project that previously played fine. My issue is more nebulous at the moment, so I can't chime in except to say that I'm also seeing higher resource usage with corresponding stutter and crackle in the audio output. This is on my lesser-powered i5 notebook using the onboard chip with WASAPI, not the system shown in my sig. Switching ThreadScheduling back and forth between 2 and 3 seems to result in a bit smoother operation with 2, but results are as yet inconclusive.

Always in motion is the future. If you had asked in late October 2017 about the next update of SONAR, you would have gotten all sorts of answers, probably none of which would have been "there will be no 'next' update of SONAR." On your computer you say Windows 10 itself worked correctly for a while then stopped working correctly, so who knows? People stay on Windows 7 to try to save money (and upgrade hassle) by holding on to their old computer for as long as possible before upgrading it or buying a new one. As time passes, the hassle level will continue to rise until it reaches a point at which it is more hassle to stay with Windows 7 than it would be to upgrade their system to Windows 10. The hassle comes first in the form of looming uncertainty about compatibility, later it will be in the form of incompatibility itself. How long a person wishes to hold out probably depends on their comfort level with the uncertainty until of course the wall is hit and the hardware breaks down or the software just plain won't run any more, and holding on to the old system is a false economy. Until that time: nothing is guaranteed. "Supported" means that the tech support staff and developers (including QA) will do what they can (within reason) to see to it that Cakewalk will run on any Windows 10 system that meets the spec. Also in the case of the developers, that they stay on top of the latest improvements to the way the OS deals with multimedia and use that information to make the program work better. With a supported platform you can breathe more easily. I was a Windows 7 user until last year when a Cakewalk developer said in this forum that they were no longer actively developing for that platform. I couldn't switch over fast enough once I heard that. I'm not going to be standing on the ice floe drifting away wondering why my computer programs are starting to act weird and I can't download any new ones because they need the current OS. Windows 10 didn't work as well as my very optimized Windows 7 system at first, but I spent some time and figured out why, and now I have a very optimized Windows 10 system where I can shut off realtime malware scans permanently, delay system updates for weeks at a time if I want, etc. because I learned how to control it.

I think the days of Microsoft being concerned about making money by selling individuals upgrade licenses for Windows are probably in the past. For that income, they make money from selling licenses in bulk to people who sell the OS installed on brand new computers. They let people upgrade to Windows 10 at no cost for a good while, now they've left a little loophole open, which itself is evidence that they either don't really give much of a crap or that they even would still like us all to upgrade for free, but for marketing purposes, they had to put some deadline on the program to give people a sense of urgency.

I am pretty sure that the ones that come (hidden) with Cakewalk by BandLab are the same ones that shipped with the last versions of SONAR Professional and Platinum. Can't think of why the developers would update plug-ins that are disabled by default. Thanks all for reminding me to re-enable them! I need to come up with a script, maybe a .REG or .BAT or .CMD file that will do it.

When I suggested being able to turn off the helpy number bubble, I thought I would either turn it off and leave it off or leave it on depending on what kind of project I was doing, but after installing the EA build and messing with a bit of Alt-X, I'm finding that's not always the case. Having the ability to toggle the number bubble off and on is letting me use it in a precise fashion, as I would digital calipers or similar. It's there when I want its precision and out of the way when I want to focus on the ruler underneath.

Oh, I am really happy about this!

Thank you! The second answer was the one I was looking for, except now I need to figure out why I don't get the Note Names dialog when I right click on my piano keys....

I seem to recall that when the Piano Roll View first got note names on the MIDI notes, a couple of people mentioning that it was cool that one could also get drum names to show up on the MIDI notes. I didn't get around to asking them how to accomplish this at the time, but I do so now. Is it possible to get individual drum instrument names to show up on the MIDI notes in the PRV? If so, how do I do it?

I don't mean having it as a plug-in. It can still behave like an offline "process," just one that doesn't mess with my audio file. Other DAW's do it that way. The same way that volume automation, edits, phase invert, EQ and most other processing is handled: the audio file sits unaltered on the disk and the DAW applies the processing during playback/mixdown.

Non-destructive normalize is surely at the top of my feature request list. The fact that it's destructive in the first place seems kind of weird, like why normalization out of all the things that you can do to audio in Cakewalk? Do anything you want to do to your sound except....normalize its level?

SONAR Platinum came with more modules including compressors and a limiter, so more PC modules do exist and are in the possession of BandLab. They would need to test them with the current Cakewalk and decide whether to re-release them. There were also a few companies whose products were released in the format. The only one I can think of at the moment is Boz Digital's Bark of Dog I, but there were others.

I get you, and thank you for reminding me of Heimholz and On the Sensations of Tone and how any waveform can be created using sines. In my musings I had sort of forgotten about ol' Hermann's work. I would like to point out that I did not say that "sine waves do not exist in the real world," I said "almost entirely devoid of pure sine waves," which is different. I also did not mean to imply that the standard hearing test was somehow not valid nor suited to its purpose of measuring hearing loss, although I can see where it looked as if I was. What I was contending, and still do, is that when discussing audio reproduction and perception, there may be abilities or acuities we haven't studied on the human side, and things on the reproduction side that relate to that. We already know, for instance, that different people have different curves and different ranges. What if we also have differing degrees of sensitivity to IM distortion or group delay or phase coherence or transient sharpness or whatever? Maybe researchers are studying that, I don't know. The most prominent blind test I've seen for whether people could choose lossless vs. lossy music files, the NPR one, was completely flawed due to the delivery system being a web browser. Yet hundreds of thousands of people accept it as proof that even recording engineers can't spot the difference. Well, not through Firefox I can't. I am a skeptic. I accept 100% that recording at 96K should make no difference whatsoever in what a normal human can hear. My hearing rolls off around 12K anyway. I record at 44.1. I am also open to the possibility that recording at higher rates may have some side effect that makes the audio sound better to some people. Maybe not all people, maybe just people with the ears that are extra sensitive to whatever. 50 years ago guitar amplifier designers and musicians and audio engineers were told that vacuum tubes would soon be phased out in favor of solid state devices, that transistor amplification was in every way superior, if they heard any difference it was imaginary, or the solid state device would sound superior, etc. and so on. The only people who bucked this were the musicians, and since they were all on drugs anyway, nobody paid them any mind. I will not bore you with recapping the story of how that turned out, but I will say that I have made a good living recapping tube amplifiers from 50 years ago.

Although I have what I consider to be a decent quiver of mics for a home studio (Audio Technica, MXL, Sterling, Shure), every so often I like to set up something weird just to see what I can do with it. The other night I recorded some acoustic guitar stereo mic'd using a tie clip lav of unknown provenance into the mic in on my Dell laptop for one channel and the laptop's built-in mic for the other. Once I pulled the room resonance out of it with proximity eq and put on MAutoAlign it started to sound like it could be usable as a background atmosphere track. A BM800 is not a useless microphone for a beginner to start using and man I would have loved to have one 35 years ago. The best thing to do when asking for help in improving tracks is to post an example track so that we can hear how your track sounds now. The BandLab site is a great way to host tracks for free for those of us on the forum to listen to. I'm in agreement with the others who have suggested that from the photo, putting up some sound-deadening on your walls or going to a room with as few reflections as possible will probably make things easier. With a Chinese medium diaphragm condenser, which I believe the BM800 is, I'd start out with the "address" side of the mic facing the 12th fret about 12" away. Record and see how it sounds. If it's too boomy, move it away, too thin, get closer. Once recorded, use a para EQ to roll off the bass up to about 200Hz, and find the "honk" frequency and notch it down by about 5dB. Then add a bit of medium room reverb. Tezza's suggestions are excellent. Trial and error, there's going to have to be some of that. YouTube viewing, some of that, too. Post recording(s)!

As I described, "pristine," meaning clean, without adding or subtracting anything. Put in a 10Hz-20KHz sweep recorded at 44.1K, convert it to 96K, compare the two waveforms using the appropriate analysis tools and they should match. By "mastered," I meant whether they had had some kind of final mixdown or processing done at that rate, for whatever reason. On another subject, an aspect of audio perception that I just thought of that I have never seen discussed is possible differences in people's ability to detect transients. I've never been tested for anything like that by an audiologist. They put headphones on me and run a series of pure monophonic sine waves and that's the extent of my hearing test. It's 100% frequency response. Seeing how our hearing evolved in an environment that was almost entirely devoid of pure sine waves, yet filled with directional transients that were important to hear if one were to survive, maybe there's more to the story than can be learned by playing sine waves to people. Is it possible that frequency response has been deemed the thing that matters because that's the thing the researchers know how to test for? ? It wouldn't be the first time....

I've not looked into their descriptions, but yeah, it sure would be nice to know whether the record in question was actually mastered at that rate. ?Otherwise, as you say, even if the buyer does think that there's some benefit to be gained by playing a higher bit rate file back through their fancy DAC, they could get the same effect by running their 44.1K or 48K lossless file through a good resampling program. And when I say "good" resampling program, I only wish that every audio program could up and down sample audio at every rate without bungling it. The fact that not all of them do is an empirically proven fact. Our dear Cakewalk's resampling algorithms handle the job in pristine fashion. (There's a site that shows test results on this, and I've resampled sine waves in upward and downward directions in programs I use then compared the results using SPAN. Oh man.) I was surprised that the Radiohead album was offered lossless at 48K and not at 44.1K. That implies to me that they're working at 48 rather than the old CD-friendly 44.1. I still work at 44.1 because....habit? Tradition? My Firepod takes a moment to switch rates when I play back different files.

According to the Reference Guide, Quick Grouping should allow you to do this, but the feature doesn't work. It seems like it was designed in but never implemented. If you do a bit of digging you can find the page in the Guide where it says it should work just as you want it to. Feel free to start a thread in Feedback about how much you would like to see the bug fixed.

HDTracks and its fellow HQ download sites beg to differ. I bought Radiohead's A Moon-Shaped Pool from them because it was the only downloadable way to get it lossless and oh man does that record sound good in 48K. They have other albums available for purchase in 96K. I don't think I would buy in that format myself, as I find that my limit of quality perception goes to "lossless" and that's about it. 256K AAC is pretty good, but I like my music like it was made, lossless.

(this rantlet is not directed at you personally, G, I just sound off about higher rates every once in a while) This assumes of course that all there is to being able to measure what all humans are able to perceive with our ears is frequency response and that there is nothing more to learn about human hearing and sound perception and reproduction. Sure, 95% of people can't tell the difference between a heinous low bitrate MP3 and a pristine FLAC, but for 99% of people, "Las Meninas" is also a picture of a bunch of history-looking people standing around in a bedroom. Sayin' you might have to know what to look/listen for, and that hearing perception and acuity can possibly be learned. I say "possibly" because I am not 100% sure and I haven't performed rigorous double-blind tests. I will say that I hear the difference between 125 MP3's and FLAC's being played back in MusicBee but I failed the snot out of that NPR test thing. What it told me was that I can't tell the difference between those formats when they are delivered through a web browser. Gimme the same files and a bit-perfect music player and then see how I do. There is a wide variation in color perception and even visual acuity (look at how many people wear eyeglasses) in the human population. 4% of music students possess the innate ability to identify absolute pitch while the rest of us have to learn relative pitch. Maybe there is more to hearing perception than can be measured with a frequency counter. Why can I, at age 58 after playing in loud rock bands for years, still dig a tiny finger squeak in a guitar track out of a dense mix? I'm not saying it's so, I'm saying that I'm open to the possibility that there may be more to it than raw frequency response, and my own empirical observations suggest that there's a good chance. There may not be information up there that we can hear (and that our primitive paper-or-plastic drivers can even reproduce), but recording with the extra bandwidth may have an effect on things other than just frequency response, like phase or group delay of higher frequencies. I personally still track at 44.1, but if I were running a pro studio, and had the disk space, I might do more at 88.2 or 96 just because we can. It will give future generations more to work with if they ever dig up what we do and want to work with it. And, BTW, for the OP, the thing to do if you intend to mix down at 96 is to start the project at that rate. Otherwise, at least with Cakewalk, there is little point. Any advantage as far as plug-in sampling is already covered elsewhere in the program.

Nice clean interface. I'll add it to the Favorite Freeware Instruments Thread. I see that they are targeting it as a drum sampler, how does it perform as a general phrase sampler?

Nice work, Alan! Sounds like we might have been listening to some of the same stuff back then. I'm getting whiffs of Eno, Talking Heads, Bowie, Fripp, Belew, Nelson?

Good heavens! For decades, "disable the onboard sound" has been such a part of DAW tuning/troubleshooting folklore that whenever I've obtained a computer for DAW use, one of the first things I do is open the case, pry the Realtek chip off the MB, toss it in the outside trash, then go through any printed manuals that came with the system and cross out any reference to built-in audio.?