Craig Anderton

A few point...when you import into your target DAW, check that it has the same pan law settings as what you used in Sonar or CbB.

Also remember that everything has a failure mode. You can usually count on DVDs or Blu-Ray (which I prefer) for 5 years if stored properly, but they don't last forever. Flash drives and SSDs have redundancy built in because they're constantly deteriorating, and over time, they will fail. Hard drives can last for 10 or more years - if they don't fail in the first six months. Linear tape is still the preferred backup medium for most large companies, supplemented by the cloud. The to key to storage is the same as investing: a diversified portfolio If you have data on a hard drive, an SSD, the cloud, and a Blu-Ray in a bank's safe deposit box, hopefully you're covered.

Very cool (as usual), Azslow3. FWIW, for S-series keyboards, I wrote How to Use Cakewalk with Komplete Kontrol. However it doesn't cover the M or A-series keyboards.

And companies certainly didn't want you to! It's not like the Sony CD players had big stickers on them saying "New! Now not with 16-bit resolution!" But yes, the dynamic range of today's music is so limited that even 10 bits seems generous. That gives approximately the same noise floor as tape recorders without noise reduction, or AM radio, so at those levels, you probably won't notice distortion all that much.

A lot of early CD players used 12-bit DACs, which gave about 10 bits of "real" resolution. But I do think that might be why people didn't think CDs sounded as good as viny when CDs first came out. Even though people may not have consciously noticed the degradation, it may have been something they sensed nonetheless..."I dunno, it just doesn't sound right." As to the OP, although it's risky to say something will "never" happen, once your noise floor is the sound of electrons bumping into each other, you're not going to get better resolution than that. Unless, of course, Noel comes up with electrons 2.0

+1. However, there's one reason I would recommend aiming for a relatively consistent level - e.g., not have some tracks peaking at -3, and others at -18, etc. Compressors, amp sims, and other level-sensitive effects are affected by input level, because that determines how often the signal level goes over the threshold or starts saturating. For example, my amp sim presets assume an input level that peaks around -3 dB. That way if I normalize the guitar to -3 dB before mixing, then I know the amp sim preset will sound as intended. With compressors, I include the intended amount of gain reduction in the preset name, like "Warm vocal -6." Then I know to adjust the level coming in to provide maximum gain reduction of 6 dB. That doesn't mean I won't tweak the compression threshold, but it gives a consistent starting point.

There's also a free version of AmpliTube that has lots of gear in it.

No, they are not. I'll send you a PM as to why However, I will be doing a desktop mastering eBook in the future. Thanks for your interest!

Ha! Brings back memories FWIW I'm planning on releasing a MIDI 2.0 book in 2023 if MIDI 2.0 has gotten enough traction by then. It will be for Sweetwater Publishing, like all my other books (except for the Cakewalk one).

Thanks for the props!!

I always took the cautious approach, but in practice, found it wasn't really necessary. Cakewalk is remarkably tolerant when it comes to backward compatibility.

Here's some excerpts from the The Huge Book of Cakewalk by BandLab Tips eBook that you might find helpful, because it lists the actual character counts and text editing protocols. It's part of a longer tip called "The Virtual Back of the Tape Box," which goes into finding file attributes and such. The Various “Notepads" Cakewalk by BandLab makes it easy to take notes on individual tracks, takes, and track folders. However, as with Project Info, there are no text formatting options. To take notes on Tracks, click the Track tab in the Inspector. You can enter around 1,000 characters in the description field. To take notes on Takes, double-click in the rectangle toward the right of the Take header (e.g., to the right of the Take’s record button). This can also store about 1,000 characters. You can enter line breaks in the Take Lane notes section by holding Ctrl and pressing Enter, which is immensely useful when making time-based notes on takes like… 00:00-00:50 Good 00:51-00:55 Mistake 00:56-1:30 Final (C1) To take notes on Track Folders, double-click in the rectangle below the listing of the number and types of tracks contained in the folder. Track names can hold 128 characters. You can’t see them all at once, but if you double-click as if you were doing text entry, you can use the right and left arrow keys to scroll through what you entered. Or, hover the cursor over the track name, you’ll see a label that gives the full title. The Browser also contains a section for Notes. This text editor can hold over 29,000 characters—almost 5,000 words. To provide perspective, that’s like a two-page magazine article...which should be enough to hold you! You can do the usual cut/copy/paste commands, but there’s no formatting (font, size, bold, etc.). Also, there’s no scroll bar, so you need to use the up and down arrow keys to move through the text. One considerate feature is that when entering text, it becomes larger and bolder so it’s easier to read. You can also show/hide file stats with the File Stats button, as well as upload an image (like album art). Note that if you used Project Info in older versions of SONAR, that info will be imported into the Notes tab in Cakewalk by BandLab.

Also you can make really long track names that go into quite a bit of detail.

I've only seen a few of his videos, but I like them a lot. I think the biggest deal with Reaper in this case is being to do oversampling once for an entire effects chain. That said...my primary instrument is guitar, and I want the lowest possible latency for real-time playing. My experience with real-time oversampling is that it causes a major hit to your CPU, and you have to increase latency to compensate. So, for the moment, I think I'll be sticking to "render-listen-keep or toss."

It's definitely an edge case, no argument there! The main point I keep trying to make is listen, and if something sounds better, use what sounds better (which may be the version with aliasing). It's easy enough to raise the project sample rate, render, go back to the lower sample rate, and compare. Most of the time there will be no audible difference. But then out of seemingly nowhere, you hear a difference. The problem I encounter is that there are a lot of variables, like what waveform the synth is using, and whether the real-time oversampling algorithm is as detailed as the offline one. I would like to think that I could simply identify particular synths as needing or not needing oversampling, but that hasn't been the case. Same with amp sims. Sometimes it makes a difference on high-gain amps, most of the time it does not. But I still check anyway, just in case.