bitflipper

That stuff makes my throat hurt. Right after the Waves debacle, there were a whole cluster of videos on the theme of "how to dump Waves". This one was pretty extensive.

Just yesterday I was thinking "sure these 6 trombones sound pretty epic, but how much better would it be if I added 60 more?"

That's exactly the kind of thing I wanted to know. Thanks Simeon. I'll be checking out your video. Slides and legato transitions are key to all string libraries' believability. Most of the time, I resort to fiddly scroll wheel tweaks to implement them. If a sampled guitar includes slides, they are usually pre-recorded samples with a fixed duration and span, which can sound quite authentic but are only useful at specific intervals and note durations. That's why my go-to solo violin is Audio Modeling's modeled SWAM Violin: being synthesized rather than sampled, it allows you to define your own transitions. For Kontakt libraries, this requires clever scripting and doesn't always sound natural. OTS' Slide Acoustic and Slide Lap Steel do a pretty good job of this, but at the cost of being fiddly and time-consuming to program. I'm waiting for a true modeled guitar to come along. In the meantime, my own secret trick is that many of the "guitars" in my recordings are actually Zebra2.

And you are correct. However, the master bus isn't the last place it goes. It's second-to-last. From there the audio is handed off to the audio interface via its driver, and is labeled in the dropdown list of routing destinations with the name that Windows gave it. On my system, for example, it's called "Speakers (Saffire Audio)" because that's what Windows calls my Focusrite Saffire Pro interface. Like Steve says above, it wouldn't matter whether you sent the reference track to the master or to the outside world directly -- if, and only if, there is no processing being done on the master bus. Usually, there is. And you want to bypass that so that the DAW is having zero affect on the sound of your reference.

This one might be pretty good, but it's hard to tell as there are no detailed walkthroughs. They've tried to make the interface simple by not using keyswitches and limiting the number of articulations, but I'd need to see a real demo to know if they've succeeded or just dumbed it down too much. Either way, it'd be hard to separate me from my longtime go-to nylon guitar, Renaxxance from Indiginus. It doesn't have as many articulations and options as the OTS Nylon, but for the price of the OTS product you could buy Renaxxance plus two more instruments.

Anyone know if the "New" Iron Pack 7 is an update to the old "Iron Pack 7"? I'll gladly give them another 3 bucks if they've added content.

Ah, the true Golden Era. Young folks complain that they'll never be able to own a home, jealous that their grandparents bought theirs for less than the price of a pickup truck today. Little do they know, that wasn't the best perc us boomers enjoyed.

It seems I was mistaken about UM2 not supporting direct monitoring. Sorry about that. Odd that the manufacturer didn't think to mention it, but made a big deal of it when advertising its more expensive siblings. AFAIK, Cakewalk has no built-in loopback test. It's something you'd only ever do once, after installing a new interface or switching drivers. However, the DAW does query the interface's driver so that it knows how much compensation to apply. If you're using ASIO, you can see the reported roundtrip latency in the Driver Settings panel of Preferences, under "ASIO Reported Latencies". btw, the only way to guarantee an accurate round-trip latency measurement is via a loopback test, where you send audio out from your computer, route it back through the interface and record it.

Sure. They just don't venture down into the basement often.

Now he may actually be a ghost from a wishing well.

Since we're strolling down memory lane, anybody remember Sickvision? At first I defended the fellow, assuming that he was not a native English speaker. I was wrong. He's from Boston. (btw, lest you think I made up that quote for the sake of humor, the original thread is here.)

You guys do realize that newer forumites will not get any of these references, right?

Good answer! The only place you'll ever see a square square wave is in the icon silkscreened onto your synthesizer next to the waveform selector.

There's always a mystery to tug at your brain, which is one of the reasons ours is the best hobby. I commend you for answering your own questions through experimentation. Most folks just ask a question on some forum, accept whatever explanation is offered and incorporate it as an eternal fact from then on. As for what the volume slider does, it sets the starting value for CC7 for the track. It does not have anything to do with velocities. You can observe the action of the slider by setting it to something other than "(101)". If your soft synth respects CC7, you can watch its volume control move as you move the volume slider in the track header. You can also see that if the volume slider is first set to, say, 112, and you then add a volume automation envelope to the track, its initial value will also be 112. "(101)" isn't a real value. It just indicates that the DAW will not be forcing an initial value for CC7. Do all synths respect CC7 for volume? No. It's entirely up to the developer how or whether they want to implement any continuous controller. Do all synths adjust volume based on velocity? No. Again, it's at the developer's discretion. Because volume often does go up with velocity, I suspect that's why people might conflate volume and velocity.

While floating-point approximations are a real thing, and might be an issue for astrophysicists, they do not affect audio quality.

I've never used that interface, but from what I've read it would seem that the UM2 has a built-in latency that you can't circumvent. Most audio interfaces support something called "zero-latency monitoring", aka "direct monitoring", where you're basically listening to the sound coming into the interface directly rather than running it through the computer first. The tracks you record should be perfectly in time with previously recorded tracks, because the DAW knows what the actual latency is and automatically adjusts the start time of your recorded clip accordingly. Unfortunately, I've found no information to suggest that the UM2 has that feature. You might want to think about upgrading your interface. The UM2's bigger brother, the UMC202HD has direct monitoring and is only $100. M-Audio has a similar product that supports direct monitoring that's even cheaper at $70.

Therein lies the crux of your error - not that you didn't have the right idea but that it was derailed by one assumption: that the sound source was immutable. Sample library developers and soft synth programmers go to a lot of trouble to introduce unpredictability, in order to make the instrument sound more natural, e.g. round robins, randomized modulations and effects. It isn't easy, given that samples are by nature static recordings and software oscillators are algorithmic. Unfortunately, such unpredictability torpedoes attempts to make objective, repeatable measurements. That's why we use test signals such as sine waves for testing, despite their being far removed from anything musical. It's about consistency, removing variables that might cause unpredictable or unreproducible results. I'd like to hear about your observations using unprocessed audio files instead of samples. You could even export the same loops you used initially and bring them back in as audio for the tests. You might well observe that the same DAW yielded different results this time! As old Joe F himself might have said: Stay Curious! Or the French equivalent, anyway.

Holy crap, Rain, this is getting scary. That black van is now parked in my frickin' garage! They now have access to not only my frozen zombie apocalypse stores but also to many potentially lethal gardening tools. Oh hell, now there's a light blinking on the router...I'm stubbing out this joint right now, turning out the lights and locking the doors.

I'm still on version 7, so mine is named "iZotope Ozone 7 Imager.dll". Starting with version 8, they stopped putting the version number in the file names, so I'd guess you're looking for the same name but without the "7". The vst3 version will be in \program files\common files\iZotope. The vst2 versions can be anywhere. Mine are under program files\cakewalk\vstplugins. Note that there are two similarly-named files for each effect in Ozone Advanced. Curiously, it's the larger of the two that is used by the main Ozone container, with much smaller files for the standalone plugins (my Imager dlls are 25MB and 3MB, respectively). The larger file (iZOzone7Imager.dll on my system) is not a VST. It gets scanned by the scanner and is included in the inventory in the register, but does not show up in the Apply Audio FX context menu because the scanner recognizes that it's not a VST. Anyway, that's about all I know concerning Ozone file names and locations. Maybe it will help turn on the proverbial light bulb.

Is this what the error message said? Unrecognized format? Were you using the standalone imager, or a full instance of Ozone? If you've updated Ozone since X3 days, you might just need to rescan plugins in X3. I don't remember if old Sonar had the Reset option for the scanner, but if it does I'd start with that.

And why I am still using Adobe Audition 3.

I was listening to some Tomita the other day and thinking how cool his cosmic string washes still sound 50 years later. So I decided to try to recreate them. All I knew was that he used many layers and lots of effects. What kinds of effects would have been available to him in 1973? Tape delays and reverb, basically. Maybe a chorus. So I started with various harmonically-rich sounds and began throwing anything I had on hand at them. The combination that came closest - so far - was a a low-res pipe organ (from the free Kawai N1 that someone on this forum reminded me existed) run through MVibratoMB (Rotary 4 model), Valhalla Delay (Quartz algorithm) and some reverb (FabFilter Pro-R, but others worked just as well). I layered this on top of an existing string part (Amadeus) and added volume automation to give the pipe organ a slow attack. I'm still fiddling with it, but so far it sounds very Tomita-like. Given the complex fx chain, I'm now thinking about eventually putting it into a Kontakt instrument for convenience. Still on my to-do list: try some of Omnisphere's choir patches underneath the pipe organ. All inspired by a $9 plugin I didn't need.

Do you get a crash dump when it happens? Or a message of any kind? What is the format of the audio you're trying to paste, e.g. wav vs. mp3? Are you bringing in the files from outside the project or copying clips within a project? Can you do the latter but not the former?

It was really just an installment payment option. Actual subscription plans, like the one Waves recently floated, only let you use the product as long as you keep paying.

Vintage Rotary excels at dirty Leslie sounds. For organ my usual go-to is IKM's Hammond, which has its own (very good) Leslie sim. But when I'm looking for raunchiness I'll turn off the IKM Leslie and use the Melda Vintage Rotary. I have also used it as a vocal effect. I picked up MVibratoMB, because for ten bucks, why not? It won't be replacing my other Leslie sims, but it'll find a use, I'm sure. The coolest thing I've done with it on so far was a bell patch with delay that serves the song as more of a percussive element than a melodic one. I made the upper mid frequencies swish from side to side, which was a pleasant effect. That led to trying it out on other higher-pitched percussion, such as the 8Dio bongos and SoundIron's bamboo ensemble. Makes them sound like the players are dancing around the room while playing. This might end up being an alternative to tired ping-pong delays.