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bitflipper last won the day on May 13 2019

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About bitflipper

  • Birthday 10/02/1951

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  1. 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?
  2. 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.
  3. 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.
  4. 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?
  5. IIRC, the ART preamps can be set for +4 or -10 dBu. Make sure yours is set to the former.
  6. 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.
  7. 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.
  8. 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.
  9. Yes, there are music sites that offer 96 KHz and even 192 KHz wave files - for a premium price. It's a scam that unethically takes advantage of consumer ignorance. Given that the material was in all likelihood simply upsampled from the original source, whatever perceived benefit from the higher sample rate couldn't possibly be in there. You're paying for the placebo effect.
  10. I haven't seen this before, where all settings on all synths are lost after unfreezing. I have, however, seen it many times with specific synthesizers. And after moving a project to another machine. Since then, I make a habit of saving a preset with the same name as the project. That's a surefire safeguard, and also makes it easier to find previously-created favorites for re-use in new projects. I just tried unfreezing a soft synth to make sure it wasn't a new bug. Good news is that it isn't. Sorry, that's all I can offer for now.
  11. Have you made any hardware changes to your system? Amplitube's license is tied to your hardware configuration and has to be re-licensed if you change the motherboard or network card. Run the scan with the debug log enabled. It may yield a clue as to why it failed.
  12. You would have to convert every file in your project to 96kHz, import it into a new project and re-mix. Most likely to no benefit. The only good reason to convert on export is if you'll be sending the files to a third party who has specifically requested 96k.
  13. Start with Matt's advice, as it's always the first step and often solves the issue. However, there are many potential reasons for dropouts, clicks and pops. Wifi adapters, for instance, are notorious for eating CPU cycles and should be disabled during audio sessions. Kontakt alone has its own extensive set of issues and solutions, but mostly it's just very resource-hungry (CPU, RAM and disk I/O). For example, 8GB should be plenty but some Kontakt libraries gobble up more than 2GB per instance and it's not impossible to run out of memory and start paging. So try increasing the buffers first, then report back if that didn't resolve the problem.
  14. The BSOD message should include the name of the driver that raised the error. That could be an important clue. It could be any device driver, not necessarily audio-related.
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