Craig Anderton

Well, it nulled, so what can I say? Maybe PDC is being applied properly to the dry track, which I believe is the way it should work. It's not hard to reproduce the test if you want to see if I'm doing something wrong in the test itself.

I've never had a problem with upsampling when used as intended according to the documentation, and wondered if maybe a recent update broke it or something. So I did a test. 1. Duplicated a drum track so there were two parallel tracks of the same events. 2. Inserted the TS-64 Transient Shaper on one track, and verified that the two tracks nulled when one was thrown out of phase (I chose the TS-64 because it's possible to have settings that don't alter the sound, which allowed for an accurate null test). 3. I enabled Upsample on Playback for the TS-64, and the two tracks still nulled. Then I did Upsample on Render, rendered, and the two tracks still nulled. So it worked as expected, although I didn't try more exotic use cases like sidechaining, using plug-ins with different internal oversampling rates, using plug-ins whose internal oversampling doesn't allow upsampling, plug-ins with widely varying latencies, etc.

At one point, oversampling was important. I've been an advocate of it, but selectively. It only makes a difference with sounds or effects generated in the box. In other words, if you're just recording a vocal, oversampling won't make a difference. It can make a difference with some virtual instruments (and to a lesser extent, amp sims and limiters), but no difference with others. Bear in mind that more and more, effects and virtual instruments have internal oversampling. Also, you can "oversample" by changing to a higher sample rate, rendering, then returning to the original sample rate. This saves a lot of CPU compared to running everything at a higher sample rate. Finally, real-time oversampling will not necessarily be as good as the algorithms used for offline oversampling. In a way, I think calling it a "game-changer" is being a little late to the party. These days, when rendering virtual instruments, I'll likely just render as audio at 192 kHz. If I can hear a difference, I'll keep the audio. Of course, this likely won't work if the plug-in is already oversampling internally, so expecting to do, for example, 4X internal oversampling at 192 kHz is not going to end well. In that case, I'll render at a lower sample rate. A lot of plug-ins simply cannot oversample above 96 kHz. But here's another consideration. I've done workshops as far back as 2015 where I compared the purer sound of an oversampled synth vs. a sound that had aliasing. It was not always a given that people would prefer the oversampled version. Although I never conducted a formal test over multiple workshops, I would say that at least half the audience preferred the sound of the version with aliasing.

The cost of doing print is out of control. I've stopped doing paper books completely, everything is a downloadable PDF. It just makes so much sense on so many levels, but in my case, the main one is being able to update technology-oriented books by just switching out a file, and having virtually no negative environmental impact.

Disclaimer: Despite my prior involvement with Cakewalk, I have no insider knowledge about what's going on. Although I still use Cakewalk, I haven't had a relationship with the company since it was purchased. So the following might be 100% wrong...but that won't keep me from speculating In my experience, Asian companies in general take a longer-term, more strategic view of business. It's not just about what's happening in the next quarter. Several "hit" products from a variety of companies have grown out of obscure, money-losing "proof of concept"-type research. Also, bear in mind that Meng is on a mission about music. My sense is that he wants to make money in the process, but he's motivated by more than just money. DAWs like Cakewalk will never be mass-market items. Even GarageBand, basically a dumbed-down version of Logic (not an insult), has relatively high adoption but it's still not a mass-market product. It doesn't cost BandLab all that much to keep Cakewalk alive. In return, they have a technology platform that could be of tremendous value someday, either because parts of it get ported over to BandLab, or it provides a path for the BandLab users who become serious about music. If BandLab saw Cakewalk as a vehicle for making money, they would have opened a Cakewalk online store with plug-ins, presets, etc. Clearly, they have much bigger fish to fry. But by keeping Cakewalk on the rear burner, they'll be able to get it up to temperature quickly, should the need arise. That would be far better than having to start something from scratch. Bottom line is I don't see Cakewalk going away any time soon, nor do I expect to see BandLab start charging Cakewalk users. If they want to make money from Cakewalk, it would be much easier to sell add-ons to existing users instead of trying to do customer acquisition. Making it free, and still being around, seems to be all that's needed for customer acquisition.

Would it be possible/doable/worth it to develop GPU-friendly plug-ins to supplement the native versions, and recommend some specific graphics cards? At that point, it would be like having a Universal Audio accelerator version and a UA Spark version of a plug-in, but you wouldn't be locked into one software company. That would also eliminate the issue of trying to make it work across multiple graphics cards.

Just to follow up...I checked out a bunch of virtual instruments for aliasing, and as rule of thumb, the older it is, the more likely you'll hear issues. There were several instruments I couldn't upsample, because they were already oversampled internally and attempting to upsample them put their sample rate into the stratosphere. So this is to confirm that yes, if you don't hear anything, don't worry about it. But it's definitely worth checking whether older plug-ins oversample internally or not. There is one other wrinkle: some people think that a synth's oversampling algorithm won't give as good a performance as offline rendering, because the synth has to do it in real time. I'm not sure if it makes an audible difference, but I mention it just in case it matters.

Except for when it's incredibly obvious But, that depends entirely on what's generating the audio. I've never heard aliasing with sounds coming from the outside world into a computer, only sounds generated within the computer. I really should do a deep dive on this on my youtube channel someday.

Assuming you have access to the z3ta, it has the option to change the internal sample rate. With some presets, the difference is quite dramatic. Check out this video to hear the difference between running it at 44.1, 96, and 192 kHz with the factory Clav patch. This demo is actually more about reproducing sound accurately than the sound of aliasing, but if you have an older sampled string library, play some of the high notes. The artifacts you hear are what aliasing sounds like. I think I need to do a long-form video about this for my YouTube channel, it's a question that comes up quite often.

Yes, that's what works for me. But as a caveat, I have an interface that doesn't get weird when I change sample rates.

What you hear are artifacts in the audio range. For example, the aliasing could be 4 or 5 kHz. Unless you've been mixing front of house for a heavy metal band the past few years, you'll be able to hear that However...to be fair, at seminars when I've demoed the difference between "pure" sounds and sounds with aliasing, some people prefer the grit of the aliasing. I've been mixing a song for a Ukraine rally in Washington, DC that has people in it from Public Enemy and the B-52s (among others), and the producer insisted on keeping the aliasing in a string part - even though I offered to replace it. Ultimately it's about art, not science!

Not off-topic! But by "192 kHz if possible," I was referring to the system as a whole. Some plug-ins that are oversampled won't respond to higher sample rates because they're already at their limit.

I agree with the bolded part - if you can't hear it, don't bother. However, I've found a major difference with quite a few synthesizers that generate harmonically rich sounds. To avoid the CPU hit, I change Cakewalk's sample rate as high as it will go with the plug-in (even 192 kHz if possible), render the track as audio, then ratchet the sample rate back down to whatever I was using. When you compare a synth rendered at the higher sample rate to the same synth sound rendered at a lower sample rate, the difference can be quite noticeable.

Exactly. The reality is that this is one of the most helpful forums ever. I don't understand how it has avoided TFS (toxic forum syndrome), but hey, I'll take it! The point of a response to this kind of thread is NOT primarily for the OP, it's just gravy if they respond. It's for all the people who are considering the same thing, but then they read a post, and they think "hmmm, I remember someone saying Intel has a free utility that will check your computer and make sure you have the latest drivers...maybe I should do that." And then they live happily ever after. It's all good.

In my experience, when the majority of people don't have problems with a piece of software, and a few people do, it's something about the setup. Sometimes it can be updates of multiple elements being out of sync with each other, e.g., an operating system update using a graphics card that hasn't been updated. I don't care what anyone uses, just because Cakewalk is a seriously good program doesn't mean there might not be others that are more appropriate to someone's needs or workflow. However, I think it's a mistake to encounter outlier problems and assume the solution would be to use a different program, because whatever caused the problems could cause new or different problems with the new program. I've seen people re-install Windows thinking that was necessary to solve some system issue, yet all they really needed to do was update a driver. Sure, maybe re-installing Windows installed the more recent driver and solved the problem, but why go through all that hassle? It seems a shame to throw away decades years of familiarity with a program before first isolating why there's a problem. If the issue is "I don't want to hassle with computers, I just want to make music," I understand that mindset. For those people, I recommend a Mac, with the caveat that it too has problems, but the odds are better that an out-of-the-box setup will work well for audio.

Mark is right, the only way to aggregate devices is with Windows native drivers. They keep getting better, and rumor has it the Windows roadmap expects to catch up in a year or two with what Apple was doing almost 20 years ago (!). However, even today the aggregation process is easier on Windows, and the WASAPI drivers have (at least IMO) acceptably low latency. Although not everything supports WDM or WASAPI (Cakewalk has kind of spoiled its users by adopting new protocols quickly), the situation continues to improve. Although I usually use ASIO, sometimes I need to aggregate stuff, and the Windows native drivers do the job.

I always update now. I used to hold off, but haven't had any problems.

Everything said here about bloatware is true, but I'll add one more wrinkle: some of it is difficult to clean out. I bought my mom a computer and it ran super-slow. I uninstalled everything but it didn't make much improvement. Long story short: "uninstalling" Norton still left bits and pieces that slowed down the computer. Several pages deep in the Norton website, there were instructions on how to remove it completely. As soon as I did that, the computer zipped along. Also +1 to an external audio interface with ASIO (not ASIO4ALL) drivers.

Apologies if I'm not understanding the issue correctly, but perhaps the simplest option is to change the balance of the two sends to affect the threshold. For example if you want the keyboards compressed more when the voice is singing but less when the guitar is playing by itself, send in a higher vocal level than guitar.

Maybe because the nice people asking for help get help from nice people, and go back to making music

My 2000 VW Beetle (the second of only two cars I've owned in my life) would probably explode at 12K RPM

Normally I couldn't really add anything to this kind of thread, but it does provide an excuse to say something that might be helpful in general. My experience with software is the better the computer, the less likely the chances of any software crashing. I use several DAWs in my work, and they're about the same in terms of reliability. Overall, I would consider them all very reliable. There are the occasional hiccups, but they're almost always from pushing a program too hard. I can't complain about that any more than I could complain about my car engine overheating because I always run it at 4,500 RPM. Overall I've been very happy with Cakewalk's stability since it became BandLabified. However, I do keep the program, the OS, and all drivers updated religiously, and image my drive periodically just in case an update introduces a problem. Computer maintenance is as much of a fact of life these days as cleaning and demagnetizing heads, and then adjusting bias, was with tape recording.

When I was writing an article about convolution and wanted to be able to reference a free plug-in, I chose Convology. It worked fine. Here's a link to that as well as other free convolution reverbs. One of them is bound to work for you The impulse responses for PerfectSpace should be able to load into whatever you choose, but if not, circle back and we'll see what we can do.

Everything is mixed in the box, no console. I don't think clippers would help, but it's hard to answer because we have quite different philosophies. I try to avoid effects as much as possible, and do a lot of my processing on tracks using destructive DSP, especially for the vocals. I think the focus you hear in my music is due to track EQ. I always start mixes in mono, and use EQ so that each instrument owns a specific part of the frequency spectrum. Once all the parts are distinct in mono, then I start thinking about stereo. I also do a lot of multiband processing, which helps with imaging and focus (especially distorted guitar). I also never use processing on the master bus. I see mixing and mastering as very different processes: mixing is about obtaining a balance, mastering is about polishing that balance to perfection. After finishing a mix I let it sit for a few weeks, and then I master it. Perhaps some of the effects that you add to every track would be more effective if used while mastering? Also, don't overlook transient shapers. For masters, the best I've found is Waves Trans-X, because it's multiband. I haven't used a real amp for recording in decades - even when I was doing concerts back in the 60s, I used clean FRFR keyboard amps, and got my sound before it hit the amp's input jack. And yes, that is me playing all the parts and doing all the singing. A comparison to Bryan Ferry is a first, though...usually people say I sound like David Byrne or Roger McGuinn. I'll take any of those The album The Singles was from 2019, when my wife was dying, so it was influenced by the turbulence of that year. You might want to check out some other projects. Take Me Back to Tomorrow is from 2020, and one listener called it "tropical pop" which I think is really accurate. It has an upbeat, island vibe. But for 2021, I got into a 60s retro mode with Anything Goes - lots of guitars, Chicago blues harmonica, etc. I never plan these "themes"...I don't have to worry about being "commercial," so I just do whatever feels good at the time. You might also like Neo-, which is short for "Neo-Psychedelic Music for the 21st Century." It was my take on what psychedelic music would sound like if it was invented today. It's the most layered, with the thickest mix and lots of guitar. There are other projects on my YouTube channel, but I think those would interest you the most, and maybe give you some ideas Let me know if you have any other questions, I'd be happy to help.

You're probably not interested in a workaround, but as someone who thinks tempo changes are important, I've found the best way to introduce them is in the two-track mix. That way you can record with tempo-synched effects, quantize parts if needed, but once mixed they will follow any tempo changes. The quality seems to be better as well, compared to trying to change project tempo when a bunch of clips all need to change at once. I use several DAWs and wrote up how to do this in Studio One if you need more details; the same technique works with Cakewalk, and works well. Bring the two-track mix into Cakewalk. Select the two-track mix, open AudioSnap, and select Clip Follows Project. The tempo track doesn't need to match the two-track master, because any tempo changes are relative, not absolute. Just move tempo markers around on the tempo track to change tempo. This technique also makes it easy to add accelerandos, ritardandos, etc. If you want to hear what this sounds like, every song in my album The Singles (the link goes to YouTube) has tempo changes that I think breathe life into the songs.