bitflipper

Great composition, as always. Love the dynamics and panning choices that make your compositions a joy to listen to on headphones. Sometimes, though, it sounds surprisingly bright, at times almost harsh, here on both my speakers and (to a lesser extent) on my headphones. Especially noticeable on loud passages and percussion, e.g. some crash cymbal hits and tambourine. But other elements, even those that can sometimes be overly bright such as celeste and brass, sound great. I've never noticed this on your other stuff. Maybe this one's just mastered a little hotter than your other pieces? Or maybe it's just the libraries. I know some Spitfire libraries favor treble and often call for a little EQ, but I don't know about VSL - that pricey stuff's out of my league!

If nothing has changed other than the executable's replacement, then it's perfectly reasonable to assume the problem lies there. However, it's very common for users to assume nothing else has changed and be wrong about that. If you're recording a new project, then lots of things have changed. Do previous projects play without dropouts? Or does the problem only appear while recording new audio? To answer the original question, no, I have not experienced any dropouts over the past 20 or so (including beta releases) updates. However, I keep my ASIO buffer size at 2048 and almost never change it, so I'm pretty much immune to buffer underruns under normal circumstances.

FabFilter has never been known for unique plugins. Rather, they make unimaginative standard types of plugins, but do them with such uncompromising quality and friendly UIs that you'll want their stuff even though you already have plenty of other compressors, EQs, reverbs, limiters, etc. in your arsenal. So what generic types of plugins don't they have yet? Chorus/Phaser/Flanger. Analysis/metering tools. Multi-channel effects. A level-rider. I can't imagine the world needs more of any of these, but then the success of iZotope's Insight took me by surprise. The other thing FabFilter's famous for is selling new versions of existing products. Updated versions of their gate, de-esser or dynamic EQ seem unnecessary. A multi-channel spectrum analyzer would be welcome but other products already do this very well. Saturn would easily lend itself to expansion, perhaps offering some proper amp sims, so I'd be inclined to agree with Zo. I would also be inclined to buy a Saturn 2. I just hope it's not Pro-C 3 or Pro-L 3.

The main change to this version: Boogex can be used as a plain stereo convolver when the "Amp" stage is disabled, and the "Pre EQ Mix" set to 100. It already uses IR files for its cabinet emulations, so I'm assuming you can use other IRs now without using the cabinet sim feature.

My wife and I once toured a new housing development. Some of the homes were still under construction, which the salesman promoted as a plus, since we'd be able to customize the house. My wife asked if that included the color of the house. No, that was strictly forbidden, we were told. These colors - all a variations of gray - had been chosen by urban planning experts for maximum homogeneity and inoffensiveness. Likewise the limits on external decorations, including trees, shrubs and fences. You were not allowed to work on your car in your own driveway. Grass had to be trimmed weekly. Violating these rules would result in a fine. Looking around the development, we noticed that everybody living there also looked the same, as did their kids. Real Stepford Wives stuff. They all drove the same type of car. My guess is they all held the same social and political views, too. We couldn't get out of there fast enough. Driving home, we noticed that every new housing development was a clone of that one. We gained a new appreciation for our older neighborhood with no official color scheme, both messy and neatly manicured lawns, bicycles lying in the yard and cars on jack stands. And people who represented a cross section of humanity, not cookie-cutter clones of one another. If you think that kind of rigorously-enforced blandness couldn't possible apply to creative endeavors such as music and art, look around. Ever fall asleep in a movie because you knew exactly what was going to happen next? We're being guided into a world of eggshell-colored sameness. I don't want that, even if I could personally determine what the universal mastering standards would be. Call it AI if it makes you feel more progressive, but this is really just art by committee. And don't say there's no difference between AI and a human export who similarly follows rules and norms. An ME will always come back to the client and ask "what do ya think?", or even tell the mix engineer to give it another try because it's not ready for mastering yet. An ME evaluates whether the song lyrics are intelligible, something AI will never be able to do (what would it think of "goo goo goo joob?"). Brian, you are correct that software assistants will get better. Heck, I use them myself every day in the form of spectrum analyzers, correlation meters, goniometers and such. I argue against the absurdity of the "only use your ears" manifesto, which conveniently ignores the limitations of human ears, psychoacoustic perception, speakers and room acoustics. Software aids are good. But automated mastering attempts to completely remove the human brain from the process, and no matter how good the AI gets it will never yield anything other than average results. That's just how it works.

Apogee makes good stuff, no doubt. At least half the pro studios I've been in use Apogee. But their marketing department has long played fast and loose with facts. I don't trust them. Focusrite, OTOH, has always been helpful and quick with bug fixes and support answers. Their preamps are pretty good, albeit low-gain. The two manufacturers use different technologies for ADC/DAC, with Focusrite having greater internal latency but otherwise more or less comparable audio quality. I'd suggest choosing between the two interfaces based on features - whichever one suits your needs and is easier to use. I doubt you'd be able to differentiate between the A/D conversions of the two in a blind A/B test. Maybe if you're recording dolphins at 192 KHz, but not for rock 'n roll. If you're hearing an obvious difference, make sure it's actually the interface that's the deciding factor and not one of the dozens of other factors (e.g. input impedance). Also bear in mind that ADCs are optimized for a target sampling frequency because no converter can be equally precise across all sample rates. Pro units tend to be optimized for 96 KHz while prosumer products are likely to be optimized for 44100 Hz. For this reason, a less-expensive unit might actually perform better at 44,1 than a higher-end pro unit.

Unfortunately, among all the vast stores of wisdom on the internet, you stepped into a big pile of ain't-so, that 5% of the crowdsourced groupthink that's just wrong. Re-installing your DAW, or your audio driver, or re-tiling your bathroom will almost never fix a VST problem. I do not know what QuickFontCache.dll is, but it's not a dependency of Cakewalk. More likely, it is a dependency of a plugin, perhaps (just guessing) something intended to run only in FL Studio. You can identify which plugin it is via the debug log option on your Preferences screen. Once isolated, you can move that plugin out of your scan path.

Everybody's dancing around it, but I'll just say it out loud: automated mastering is snake oil. Many accomplished mix engineers also happen to also be accomplished mastering engineers (e.g. Phil Ramone). But they don't master their own records. The value of an ME isn't in his expensive gear, granite speakers or perfectly-tuned rooms. The true value is having a separate set of ears that can critically and objectively evaluate your mix, in an environment other than your own studio. To that end, you'd do just as well by offering to swap mastering duties with somebody from the Songs forum.

Good to see you back, Jerry. We can always use the perspective of a "real" composer around here. Unfortunately, I was unable to listen to your new movement ("this site can't be reached"). Might be because I'm on a dial-up connection - actually a gizmo that's a wifi router coupled to a cell phone and perched on a balcony above the garage. We improvise and make do out here on the island.

Insta-buy for me. Syntorus has long been a staple here. My first-call will probably always be Ubermod, but if it doesn't deliver then I know Syntorus will. I like that you can tempo-sync it - from the graphic is appears you can sync each channel independently, that would be cool.

I loved ST3. You could always dig up something inspirational in there. Need a didgeridoo or a hurdy-gurdy? They were in there. Totally agree about the toys-to-talent ratio, which I reckon to be about .01%. When I reinstalled all my sample libraries I was shocked at how many of them I'd forgotten I even had . That's when I realized I probably had enough of them. Running out of space on a 1TB drive should've been the first clue, I suppose.

Seems everyone at GC is working there solely for the employee discount. I'm always friendly toward them and treat them as equals. Even when they assume I am ignorant because I'm an old fart, perhaps there to get a clarinet for my grandkid. However, I cannot stand being BS'd. That's comes down from corporate, which commands them to sell, sell, sell the high-margin items. So they lie. Monster cables are always "the best". Amplifiers have 1000W outputs even though their UL nameplate says power consumption is 200W Brand names are the most important factor when buying guitar strings. Rokits are the best studio monitors money can buy and all the big studios use them. $100 is a bargain for a 10' mic cable because it's oxygen-free. And of course, everybody there is a guitar player. Ask about keyboards and they can only point to the keyboard room. But I never intentionally bait them with questions I know they can't answer. I save that for the computer department at the big-box appliance stores. Even though I have no animosity toward GC staff, I've not shopped there in 5 years and never will again. GC bought my stolen gear and refused to assist me in any way. Wouldn't let me see the security footage so I could identify the thieves ("it's a privacy issue"). Corporate told me to get lost. Fortunately, we still have a viable local chain as an alternative. Sales staff is just as clueless, but they're OK with "I don't know, I'll try to find out". That earns my respect.

I seem to recall that there was a usable P-Bass in there. That's about it. I was so disappointed with ST4 that when I replaced my computer I never even bothered to re-install it. Plus they had the audacity to want to charge me ten bucks to re-download it. Said no thanks.

Released recently with little fanfare, version 3 of this longstanding classic freebie. Yes, it's free. If you haven't ever tried this one, it's a quite good "character" compressor. I like it on bass, acoustic and electric guitars and occasionally as a remedy for overly-percussive vocals. A lot of people use it on individual drums, me not so much. But it does interesting things to your room mics if you like Bonham-style roominess. Version 3 adds a sidechain input, although the way I use this plugin (aggressive fast-attack scruncher) I don't know that I'll actually use the sidechain. Also adds a scrolling visual display that helps dial in the "sensitivity" (threshold) control. https://www.audiodamage.com/pages/free-downloads

First thing I'd do in such a situation is open the Event View and see if there is anything weird about those missing notes, e.g. very short durations.

I was introduced to it at my local music store where I hung out. Prior to that I'd only read about it in magazines, although I'd been exposed to Roland's proprietary interconnect scheme so I got the concept. I had one keyboard with the Roland interface but nothing to connect it to, and was shopping for a second one when Roland announced they'd be dropping their design in favor of MIDI. I had no MIDI-enabled hardware of my own, so I convinced my buddy at the music store to let me hook up every MIDI-equipped synthesizer in the shop. We had six synths going off one keyboard with me in the middle of it all happily jamming away - until the owner came out and told us to STFU. Apparently, not all the customers there that day appreciated the momentousness of the occasion. Some were just there for clarinet reeds. To be honest, I did not see the full potential of MIDI at first. There were no MIDI interfaces for computers yet. For me it was just a way to play piano and strings at the same time. That all changed with Cakewalk 1.0, the reason I bought my first MIDI interface.

It hasn't ever even occurred to me to try inserting a plugin during playback. Try to imagine all that goes on under the hood when you add a plugin. The plugin and DAW have to enter into a conversation to establish their relationship, e.g. the plugin's I/O needs and what its internal latency is. The plugin has to initialize itself, which, as Steve points out above, may involve significant buffering latency as well as file system overhead from checking licenses and loading default presets. The track may have to be reconfigured if the new plugin happened to force a change to the internal track interleave. All this has to happen within the brief window dictated by your buffer settings or you'll starve the output buffer and get a dropout. Finally, the DAW has to recalculate plugin delay compensation, and if the overall compensation has increased pause playback to get all the tracks back in sync. Like Kalle, I too would be shocked if inserting DSP during playback didn't cause a glitch. If not a full stop. Rod, if this didn't happen to you before, my guess is you were just lucky. Does this happen with any plugin, or certain ones? If the latter, what type of plugin were you inserting when you noticed the glitch? A reverb or delay, perhaps? Also, what are your buffer sizes?

I finish one song for every 50 I start. Such a low completion rate can lead to a paralysis of indecision and procrastination. I've gone months without any creative output because I wonder what's the point when it's probably just going to suck. What keeps me going is that wonderful feeling you get when a song suddenly clicks and you start to actually enjoy listening to it. It's always a surprise when that happens. Sometimes I spend weeks on a piece that goes nowhere. Other times I'm sure it's going to suck and then one day it doesn't. Rarely do I know up front when I've begun a project that it's going to come out OK. It's like putting one more dollar into the slot machine: sure, it just took my money the first 50 times, but you never know. Here's a tip. Every failure is going to have some little kernel of goodness in it. Might be a riff, a beat, a clever lyric or a cool synth patch. 5 seconds of brilliance buried in 5 minutes of crap. Those gems are like pocket change; toss one into a jar every day and over time they'll add up to something of value.

It's possible what the OP is hearing are dropouts, which when numerous can sound like very bad distortion. It could be that the overhead incurred by real-time DC offset detection (basically a HPF set to a very low cutoff frequency) is simply pushing the CPU over the edge. If that's the case, then simply increasing the size of the record buffers may fix it. Of course, that'd also increase your latency, if that's a problem.

You learn something new every day - I didn't even know that was an option during recording. It's not something I'd have even thought of doing, as I avoid any ITB processing at all while recording. Still, it's surprising that removing DC offset would actually corrupt files. Corrupt in what way? Dropouts, perhaps?

IIRC, the ART preamps can be set for +4 or -10 dBu. Make sure yours is set to the former.

Outstanding advice from Tim. A guy who walks the walk.

I like your logical thinking. However, there are some fundamental truths of physics that, if taken into account, might nudge your thinking in a slightly different direction. First of all, let's dismiss the often-heard argument that sine waves don't exist in the real world. They absolutely do, and in fact even the most complex sound can be shown to be constructed entirely of many sine waves. Refer to the groundbreaking book On the Sensations of Tone as a Physiological Basis for the Theory of Music by Hermann von Helmholtz, which you can read for free here. What makes it such a great introduction is that it was written in 1863 when nobody would have had any idea what he was talking about, so the explanations are given without any presumptions about what the reader already knows. When you see a transient, or any abrupt change in level, think of it as containing high-frequency content. When I was in electronics school, my instructor had us add sine waves by hand on graph paper. It was a tedious exercise but very enlightening. As I kept adding harmonics and plotting the algebraic sum of them, the resulting waveform took on new but familiar shapes. Depending on the harmonic relationships, I got a square wave or a triangle or a sawtooth. I then experienced an epiphany about how subtractive synthesis works: the complex waveforms that we use as raw material for sculpting tones are comprised of many frequencies (sine waves). And that the steepness of the leading edge of a square wave increases as you add more and more high frequency harmonics to it. Later, I went to work as an instructor at that same school teaching oscillators, amplifiers and filters. Many of the experiments I devised for my students revolved around my personal favorite topic, audio synthesis. We'd run a square wave oscillator into a low-pass filter to show how the leading edge got more rounded as you lowered the cutoff frequency. As well as proving that a truly square shape as often drawn in diagrams can't really exist in nature because it would require an infinite number of harmonics. And the most important lesson: showing that until you rolled off at a point below the upper limit of hearing, there was no audible difference in how the "square" wave sounded. Removing, say, 30 KHz from the signal made no difference in how the sound was perceived. However, the effect was clearly visible on an oscilloscope. Removing frequencies above the hearing range obviously changed the waveform, but did not change how it sounded. This is why we can safely ignore frequencies above 20 KHz in digital audio. That's fortuitous because the sampling theorem only applies to a band-limited system. If it was necessary to preserve ultrasonic content, you'd need a much, much higher sample rate. Digitize a 20 KHz square wave and you get a 20 KHz sine wave. But both sound exactly the same (assuming you can hear them at all). Digitize a 12 KHz sawtooth and you get a 12 KHz sine wave - and they both sound the same because the sinusoidal components that distinguish a sawtooth from a sine are above the range of human hearing. This is all a long-winded explanation for why a hearing test using only sine waves is valid.

That's surprising. The sound card doesn't have to change for an export, since the sound card isn't involved in that process. I guess it does that in case you want to actually hear your mix after exporting it. Maybe you're doing an audible export, meaning listening to it as it saves the file? Either way, it shouldn't change your project's SR. It'll still be 48 KHz and when you hit the spacebar to play it back your interface should revert to 48K.

True, Cakewalk's resampling algorithm is quite good. However, bear in mind that a "good" resampling algorithm isn't good because it improves the sound quality, but rather because it doesn't degrade it. Best-case scenario is that the upsampled version sounds exactly like the original. Even if those premium "high resolution" records were mastered at 192 or 96 KHz, it still wouldn't make any difference because what matters is the sample rate they were recorded and mixed at. Once recorded, it's not possible to improve them; all you can do is avoid making them sound worse. Remember, the only thing higher sample rates do for you is extend the frequency range. If there wasn't any >20KHz content in the original recording, upsampling isn't going to magically put some in.