bitflipper

Ah, that makes more sense. I'll wait for a sale on the full package. This one's a bit limited, basically a set of curated arpeggiations with pretty good but fixed-sounding ensembles. The lack of mic positions is going to make it harder to blend in with other orchestral libraries such as solo instruments. I also don't see a way to route different sections to separate outputs, so all mixing has to be done within the instrument. I guess Spitfire has just spoiled me.

Lovely! The guitars sound great. Good move backing off the reverb; makes the performance sound more close-up and personal. My only suggestion would be to flesh out the drum track a bit. It sounds very nice but repetitive, drum-machiney (if that's a word). Some accents, ghost notes, ride bell hits, maybe some hand percussion (e.g. shakers, bongos, congas) could dress it up without too much effort. (Check out Skaka for quick and dirty shakers.)

It was $399 when initially released, so $99 is still a deal. This product isn't just for beginners. It's also perfect for lazy people, regardless of experience [raises hand].

They aren't particularly "accurate", but they sound very smooth and pleasant. I wouldn't trust them for mixing or mastering, but they're great for pleasure listening in the dark. They are my final QA for every mix. I wasn't super impressed with them on first listen, but after adding a headphone amplifier to my nightstand sound system they really came to life. They sound even better in my studio, where I have a fancier headphone amp. Just don't expect your phone to be able to push them adequately, or even necessarily your audio interface's headphone outs.

I paid $200 for mine back when they first came out. I pounced on it because it was such a bargain compared to $500 for the HD650. Even if it did come in a plain black box. So $195 is a very good deal indeed. There aren't a whole lot of comparable products out there in that price range. Once in awhile you do see DT-880s for less, and those are fine headphones, but I prefer the relative linearity of a true open-back design. Be forewarned, however: these are high-impedance headphones.

Ah, yes. CA-2A. I knew I was forgetting an important one. I'm just the opposite - a contrarian who never uses ProChannel plugins. They're great, of course; I just like the tidiness of seeing all effects neatly listed there in the fx bin. CA-2A - as a VST - is my second most-used compressor. Nothing's better on vocals. I'd also like to see Dim Pro make a comeback. I always liked it but was unable to re-authorize it after replacing my stolen DAW. If it was reintroduced as a BandLab product, they'd likely change its goofy licensing scheme and I could use it again. Sorry, OP. None of this has anything to do with console emulators. What's the tie in here, besides Craig's book...um, yes, I've got it: CW has some excellent goodies in the vault that would help your mixes more than any console emulator. You just can't have them. OK, that's better, now we're back on-topic.

While it's true there's no lower level, at least you're not alone down here in the basement, Tim. Some, believe it or not, are here by choice.

Unique and good stuff there, ramscapri. Chilled synthetic and organic New Age with traditional Indian undertones.

Me, too. I sadly broke my favorite Cakewalk mug. My extensive collection of vintage 12-Tone t-shirts all have holes in them. I know I'll never have another V-Studio shirt, and it was a sad day when I added it to the rag box. Plus it's a shame new users can't enjoy the Adaptive Limiter or Z3ta+. Not sure what else they're missing out on - the percussion strip? VX-64? What about Rapture and Dimension Pro? Can those be purchased today? At least Craig's book is still available.

I'd caution against generalizing based on your observations. All you've determined is that your computer can handle the miniscule overhead of upsampling a specific compressor, in your specific project with your specific plugins. This is far more about the plugin's efficiency than that of Cakewalk, since nearly all the extra processing that upsampling incurs happens within the plugin. It'll be running exactly the same instructions under Cakewalk as it would under Reaper or Studio One. Yes, being able to upsample individual plugins is indeed a great feature worthy of bragging about in the marketplace, but I doubt any knowledgeable DAW user is going to jump ship based on that feature alone. This discussion is meaningless without talking about aliasing. Aliasing is not only relevant, in the context of compression it's literally the only reason oversampling is ever prescribed. If you have no audible aliasing, you don't need oversampling to begin with. If you don't need 2x you likewise don't need 4x or 6x.

As I thought about this, a few dormant brain cells must have woken up because I suddenly recalled a vague memory of having the same problem long ago. Google brought up this thread (by me) from 2010: http://forum.cakewalk.com/Somewhat-Solved-Superior-Drummer-3db-louder-after-freezing-m2028203.aspx Although we were talking about SD2 at the time, the symptoms are identical to what you're experiencing. TL;DR the workaround I came up with back then: freeze the track normally but then set the track interleave to stereo.

You should be able to select one side of the stereo output. Do you not see the same selections shown in my screenshot above?

Well, I suppose there could be a problem within SD3 that I've just never noticed. Even if that's the case, you can still force mono within SD3 using the pan sliders. Note that SD3 gives you separate left and right sliders for every output channel. Put them both into the center and you've got mono out regardless of whether you use 1, 2 or 1 + 2.

Makes sense that you'd instinctively protect your dominant hand with the other. Assuming a right-hander...when protecting your face, is it not your left arm you throw up? When taking a fighter's stance, is it not your left side you present to your opponent? Do you not flip off rude drivers with your left hand? Instinct designates it as the sacrificial limb. Of course, on very cold nights I have a very different strategy for hand placement.

Make sure the audio track you're sending it to is set for mono interleave, then look at the frozen track and make sure it's still mono. If not, it may be because you've got a stereo plugin in its fx bin. Once you have it all set up and working properly, save it as a track template so next time you won't have to mess with it at all.

Don't know if this will help, but here's how I normally set up SD3. Output channels are designated Kick, Snare, Toms, etc. Each of these goes to its own audio track. In the screenshot below, the kick is taken from the Left channel and routed to a mono audio track. Overheads and Room remain stereo.

It won't matter which one you install first. If Essential is already installed along with Cakewalk, the Celemony installer will simply overwrite it with the Studio version. I prefer installing the DAW first and testing before any third-party add-ons are installed, so I've got a confident baseline to start with. (btw, I'd encourage you to consider investing in the Melodyne 5 upgrade. I was initially reluctant to do so, since version 4 already did everything I needed (so did versions 2 and 3, for that matter). But I am loving the new global leveling feature. It's a real timesaver if you have a lot of vocal tracks and/or multi-part harmonies.)

Nothing's broken, it's just arithmetic. Combining any two tracks, which is what happens when you convert stereo to mono, requires them to be summed. IOW, the left and right channels get added together, sample-by-sample. Whenever they are in phase, even briefly, you'll get a sum that's larger than either channel. 2 + 2 = 4, and there's nothing we can do about that. Here's the easy solution. Most samplers, include SD3, let you choose between stereo and mono outputs. If you use a mono out for the kick, and route it to a mono track in Cakewalk, you've now got mono from start to finish and there will be no increase in peak levels when you freeze. Anything you want to have in stereo, such as overheads, goes to a stereo output to a stereo audio track. There will still be no rise in peaks as long as you're freezing a stereo track as stereo.

I'm in a six-piece band. Pay's about the same as for a duo. In 1975. Thinking of a revised distribution scheme... 16.7% base pay less 1% for every song you screw up less 5% if you don't help set up the PA less 5% if you only pack out your own gear and then leave less 5% if we can't find you after break's over because you're chatting up the barmaid

Well, that makes sense. Where else are you going to spend all those BJZ/FM/CHB royalties? btw, my favorite setup for final critical listening is a pair of HD650's, but they did not reveal their true potential until I bought a headphone amp for them. Hi-Z, ya know. They're a nice complement to my main speakers, so any time I'm happy with how both sound it raises confidence in the mix.

EQ generally last because it's the one you're more likely to continue tweaking as the mix progresses. EQ changes levels, which assures you'll have to go back and readjust the comp threshold. Putting EQ last means you're less likely to screw up something else down the chain. This has long been the standard for hardware consoles that feature integrated channel compression, including my current Yamaha stage mixer. Of course, such compressors are primitive and don't offer anywhere near the flexibility of software compressors - such as sidechain filters. The exception to the EQ-last rule is if the compressor doesn't have a HPF on the sidechain input. In that case, EQ should precede compression, especially if you're cleaning up mud by rolling off lows on guitars and vocals.

You have to explicitly tell Windows to not put USB ports to sleep. Open Device Manager and scroll down to the USB devices. There may be a bunch of them, but if you don't know which one connects your audio interface it's OK to do this to all of them. Un-check the "allow computer to turn off" option. (It's not unchecked in the screenshot below because I don't use a USB interface and therefore don't need to worry about it.)

Try this: select a track, then choose Process -> Apply Effect -> Remove DC Offset, see if that changes the waveform. Or...apply a steep high-pass filter with a low cutoff frequency (< 50Hz) and see if that changes the waveform.

Aren't we all? Don't sweat console emulation. It's an attempt to replicate the flaws of old gear, and its primary function is to introduce a small amount of harmonic distortion. That's all. As noted by Erik above, the flaws in classic gear were unintentional. Those expensive boards were designed to deliver the highest fidelity available with the electronics of the day. Nowadays, we can easily achieve more linear frequency response, greater dynamic range, lower noise, finer control and greater consistency using inexpensive gear. As an old timer who battled analog signal chains and magnetic tape for decades, I fully embrace the crystal clarity of digital audio and feel no nostalgia for days gone by. So why does anybody bother with console emulation and tape sims? Because the flaws inherent in those things made mixing easier. A little harmonic distortion adds texture. Even white noise helps glue a mix together. Tape saturation does a lot of the blending for you. Personally, I do not use any of that stuff. Embrace the purity of digital, I say. If your mix sounds thin and disjointed, keep working on the mix.

Thanks, guys. I appreciate you giving it a listen. Looks like it's gonna be another weekend for musical doodling in the garage, as our guitarist has come down with COVID, forcing us to cancel tomorrow's gig. Unfortunately, I got the news AFTER packing up all my gear and loading it into the van. I think it's just going to stay in there for the week. Kills my gas mileage dragging all that stuff around, but moving it kills my back. And next week's gig might not happen, either.