Craig Anderton

[deleted due to being slightly different topic]

No, it shows the Windows logo, no dots chasing each other, and then throws up the error message. Not being able to use the Windows Recovery disc I did for my last image is disturbing. In the BIOS, all the drives are present and accounted for. This was a Windows 10 over Windows 7 install back in the day. I suppose I need to start over, but would it be possible to buy a new copy of Windows 10 and try to do an in-place install from a USB drive to avoid nuking all my programs and plug-ins?

Everything was working fine with 1809, but then today, Windows decided it was time for another update. Can't get out of the BSOD. Automatic repair doesn't work. I can't take the time to re-install everything from scratch. So I went to uninstall the lastest feature update. Got the "restoring previous version" or whatever it says, but then when it was done, said that "A required device isn't connected or can't be accessed." In the BIOS, all the drives show up. I can't boot from a recovery disc to re-image my 😄 drive, even though it's set as the priority in the boot order. I am totally hosed...all because of a Windows 10 update. I have no idea how I can fix this.

That's a great tip, thanks.

I wonder if there's any significance to the fact that the only plug-ins referenced as being problematic were amp sims - Guitar Rig, Helix Native, and TH3.

My comments were in the locked thread, but after way too many hours, I read that people were having a hard time with the latest W10 update if Avast or a couple other programs were installed. Although Malwarebytes wasn't on the list, I uninstalled it and tried again to update. Again, after updating, I got a BSOD. After a restart, the problem sorted itself out, and the system now seems to be stable. I've noticed in the past that when Windows gives BSODs, just keep turning off and rebooting. Eventually either Windows figures out what's wrong, does an sfc /scannow disc repair, or lets you go into advanced troubleshooting where you can at least try to fix things. Don't think Macs are exempt, either. Last night I was informed I needed a Safari and security update. After doing so, my MacBook Pro couldn't boot into the OS. Instead, it said the installation could not be completed, and displayed an error log. The local Mac store walked me through four different procedures, each more invasive than the last one. The fifth one was opening in Restore mode and downloading the OS from scratch, which let me use the Mac again. At least it doesn't make you re-install your programs, but still, IMHO today's computers are a house of cards. I have great sympathy for any company trying to develop for today's computers, let along remain compatible with yesterday's.

In my experience, the newer the laptop, the more likely the emphasis on battery life. I use an old laptop for my workshops. It's big and generates heat, but it doesn't screw up. OTOH I can't get more than 60 minutes from the battery, even with throttling down the CPU max % when not plugged in, and a dim screen.

Let me elaborate about a "modular" DAW structure. I would like to see it not just for Cakewalk, but as an industry standard. ReWire has shown that it's possible on a basic level. There are a lot of possibilities. For example, only CbB and Acid let you create and edit Acidized files. If there was a modular architecture, Cakewalk could sell their Loop Construction view as a module to plug into, for example, Cubase or Pro Tools if they too were modular. Also, people here often talk about how they prefer CbB's Console View compared to other programs. It would be great if you could "rewire" Cakewalk's mixing console into other programs that don't implement it as well. And a modular approach would solve the Staff View issue Standards have really helped drive music software. Imagine where we'd be if ASIO and VST didn't exist. We'd be waiting for Microsoft to finally develop their audio drivers, and have probably nothing but plug-ins locked to specific DAWs. I know the idea of modular DAW is kind of blue sky stuff, but hey...sometimes the sky is blue.

Which reminds me...I wonder whatever happened to the content I created for Sonar (like the 30th anniversary material and the Anderton Collection FX Chains), and the Rapture Expansion Packs that were never released. As to the video, I'm about ready to do one like that on the latest Windows 10 update, which bricked my computer (even with multiple attempts), until I did a pre-update system restore. I think Cakewalk and other companies might be having a hard time trying to keep up with the moving goal posts. I realize some of these bugs have been around for a long time, but as Mark said: ...and it would be even worse if the problem is due to something that keeps changing. Also, Brian's comment is telling: So if Mark's theory is correct and CbB loads the plug-in before it initializes itself - i.e., Cakewalk is faster than another program - then the other, less efficient program would appear to be working correctly whereas Cakewalk (and similarly efficient programs) would not be. For any program, I try to stay within the program's ecosystem as much as possible. Thankfully I still have all my old Sonar plug-ins so I can still use mostly Cakewalk plug-ins in my CbB projects. The other stalwarts I use in all my other DAWs - Waves, IK, and Native Instruments - don't give me any problems. I haven't personally experienced the issues the video references.

Sound quality also depends on whether you're monitoring the loop at the host tempo, compared to the acidized sound that conforms to tempo.

Also make sure you have the latest versions of plug-ins. When doing an S1 project with two instances of the Line 6 Helix (which admittedly, I max out by doing multiband processing with several amps), I couldn't understand why I was getting terrible crackling. The performance monitor showed around 47% CPU for each instance. I went to the Line 6 website, and saw that I was several revisions behind. After updating, it went down to 32% for each instance - apparently someone has been doing optimizations

Your historical trivia for today: "moonies" were devotees of Sun Myung Moon, and often very vocal in their support for same.

You're preaching to the choir! I've been pushing for this with music software in general - levels like video games. Lenses create a more beginner-friendly environment, but they're still lenses to a complicated program. With 8.5, where you had all those customization options, I did an interface that was basically an ADAT with faders. No MIDI, no VIs, no waveform editing, no plug-ins except for a reverb send on each channel and a limiter on the output. The idea was that you could give something like that away for free, and hook people. I don't know if it's possible from a software standpoint to create a modular DAW, but that would be ideal. For example after mastering the ADAT interface, you could then add the waveform editing module. Or the virtual instrument module. But since no DAW does this, it's probably not possible. Even better would be a standardized module format, so you could use, for example, Cakewalk's virtual mixer module in Pro Tools, or Cubase's staff editor in Cakewalk.

When there's a locked plug-in I really want to use, I use the closest plug-in in the program I'm using. Before mixing, I export a premix and the file I want to process in the other DAW, render it, and bring it back into the main DAW. Sure, that's less convenient than having cross-program plug-ins, but it's better than not using plug-ins at all.

At the very least, I'm a reasonable facsimile thereof... Can you believe I'm still writing a Guitar Player column every month?!?

I think you mean time stretch, not tempo change? This was changed quite a while ago to be in conformance with the Melodyne keyboard shortcut. Hold down Ctrl+Shift while you click and drag. Or choose F8 for the timing tool. Note that this doesn't work on acidized or REX files, only conventional WAV files.

I think for professionals, the days of "I use X DAW" are over. Now it's "I use X, Y, Z" DAWs, and this is the world into which newbies will be introduced in the years ahead. CbB will become part of their arsenal of tools. As to current market penetration, the CbB thing happened only a little over a year ago. It's not surprising that there will be more mentions of programs that are one or even two decades old. They're established, and also, for a long time the Mac was king of the hill. That's been changing to where people are more accepting of running Windows - although like many people, I'm a bi-platform kinda guy (even though overall, I greatly prefer Windows). For example...nothing does the "parse any audio and create a Tempo track" like CbB does. Nothing does the polyphonic Harmonic Editing or Scratchpad things that Studio One does. Nothing has as stable an audio engine, or handles loop-based music, as well as Ableton Live. Nothing has a suite of plug-ins as good as Reason. And, if you collaborate with major studios, nothing is as ubiquitous as Pro Tools...which is why I use ALL these programs in my day-to-day work. I also use Cubase from time to time for its "special sauces," and DP's excellent amp sims. The reason I switched to Sonar back In 2000 was because at the time, I was bouncing back and forth between Cubase (good MIDI and hard disk recording) and Acid (only thing that could handle loops). When Sonar came along, I could do everything in one program, and I used it for commercial classical music projects, narration, video soundtracks, songwriting, loop library development, and rock/EDM projects. As long as DAWs went through a period of trying to be more like each other and adopt each other's features, then all I really needed was one program (although I used Studio One for mastering and Live for live performance). But now, DAWs have particular personalities and user bases, and are going off in more specialized directions. This is great - it's like if the only guitar that you had was a solid-body electric, but then you were introduced to a semi-hollow body, an acoustic, a Dobro, and a 12-string. All of a sudden, your options were multiplied...sure, you had to learn some different playing techniques, but they had more similarities than differences. I gave a seminar about mixing at a university last week, and showed the tempo track extraction feature in CbB and some of the mixing features, as well as the Harmonic Editing in Studio One and the Reason instruments. With CbB being free, I'd bet that a lot of the attendees went home and downloaded it so they could use some of the CbB-exclusive features I mentioned. Probably some of them looked seriously at adding Studio One to their Logic or Pro Tools-based world (or even switching), while others started thinking about rewiring Live or Reason into their DAW of choice. If that's indeed the way things are going, then CbB is in a good position because Bandlab bought an essentially fully-formed program for what I assume was a very reasonable price. It will become part of what people do, and for those who have really good chemistry with it, it will become their primary DAW. Different people have different needs, and there will always be some people for whom a particular DAW will be the right fit. I think it's also important to remember that culturally, Asian companies often take a longer-term view. Consider that many of the CbB updates have been about stability. It's smart not to put a huge push on something until you know it's going to make a good impression on first-time users. The last person I turned on to the free CbB download (who uses both Windows and Mac) hit it off immediately, and is no longer using Logic. It's all good.

Another option, if you want to compare the waveforms in non-real time: Duplicate the track. Render the effect on the duplicate track, but not on the original one. If you want to see a spectrum analyzer view in real time: In the track with the audio and the compressor, open up the QuadCurve flyout (it defaults to pre-FX Rack; change if necessary). Turn on the QuadCurve, but don't turn up any gain controls. Solo this track. Now you'll see the spectrum analysis for the pre-compressor audio Open the QuadCurve flyout in the master output bus. Turn on the QuadCurve, but don't turn up any gain controls. This will show the spectrum analysis for the post-compressor audio. Compare the two QuadCurves flyouts.

Hey Mark - does that work every time, or is it hit-or-miss? I want to upgrade Melodyne for use with ARA2-friendly programs, but still want to be able to use CbB with minimal issues.

+1 backatcha, and to add to what you said, presets for dynamics processors are more or less useless unless the input level at which the presets were created is known. The presets I make for myself assume peaks of -3 dB, so being able to normalize to that level prior to dynamics processing saves time doing final tweaks on the presets.

True indeed, but trust me...it's still time-consuming

One advantage of the pitch-based approach I mentioned rather than speed-based one is that you can specify how much transposition you want, in semitones and cents. However I should have asked about how much transposition you wanted to do. The technique I mentioned is optimized for smaller amounts, like a semitone. Otherwise there will be a timbral shift when you pitch back up. However, in many cases, this is considered desirable. Perhaps an even simpler option, which may not alter timbre as much: 1. Create a premix. 2. Transpose it down x number of semitones with either the DSP function (semitones only) or Melodyne (any shift amount). 3. Sing along with the premix. 4. Transpose the vocal up by x number of semitones with either the DSP function or Melodyne.

Actually one reason why I referenced LUFS in my previous post about normalization is because it is about perceived volume, and that includes frequency-domain elements. There's much more to "loud" than normalization or compression. For example, the human ear is most sensitive in the 3 to 4 kHz range. If you boost frequencies in that range, the perceived level will be louder (although a little goes a long way - if you go too far, the music will sound harsh, and lead to listener fatigue). Also, I have a technique called "micro-mastering" for producing loud masters that retain dynamics by doing the following to an uncompressed/non-limited stereo mix. 1. Locate all peaks above a certain level, like -3 dB. Hopefully there will be only a couple dozen (that's why it's important the mix not be compressed). 2. Normalize the level of individual cycles, or even individual half-cycles, to -3 dB (yes, it's time-consuming ). 3. Now you can raise the level of the entire track by 3 dB. Because you're changing the level of individual half-cycles, they go by so quickly that nothing seems limited. Also, because you're doing this with DSP, there's zero pumping, breathing, artifacts, etc. If you then add compression or limiting, you won't need to add as much as if you hadn't used this technique.

There are several ways to do variable-speed techniques in CbB, including your particular application. Check out the section in the article titled "Fine-Tuning Varispeed." If you have "The Huge Book of Cakewalk by BandLab Tips," this technique is described starting on page 366.

A few things about normalization... Normalization doesn't affect the sound or the dynamics. It's no different from turning up the level. If you plan to add limiting or compression, turning up the processor's input level, or lowering its threshold, is functionally equivalent to normalizing the input signal to a higher value. If you normalize to 0, you run the risk of intersample distortion due to the D/A converter's smoothing process. A signal that registers as "0" on a meter that's reading sample values could actually be creating a signal that's 3 dB over the available headroom after conversion...so there's distortion even though the meter says there isn't. True Peak meter readings take this into account. If you're doing an album project, ultimately it's up to your ears to decide if the levels are balanced. One way to do that is to normalize all the cuts in the album to something under 0 True Peak initially. Decide which one is softest, and reduce the levels of the other one to match it. However, that's not a particularly scientific approach, and requires a lot of trial and error to get the right balance. Level is less of an issue these days than it was with the CD. I recommend checking out my article What Is LUFS, and Why Should I Care? It addresses level-matching, loudness, the R128 standard, and how to deal with optimum level-setting in the post-CD era.