Quote:
Originally Posted by
Pyeguy
β‘οΈ
I provided at an 'interesting read', which was an unbiased offering. Then, it was implied that it was nonsense because, they're in the business of 'selling analog'.
I said I thought it was FUD, and I do.
Not just because I know the guy is trying to sell analog, but because I read the whole thing and it read as FUD.
In the middle that Sharp11 skipped, there's a litany of scary things that can go wrong in writing a plugin.
To pick one example, there's failing to upsample enough for the HF content of intermediate waveforms, and then getting aliasing.
Yes, it hurts when you do that.
As a Doctor (of EE and CS) I have a simple prescription:
don't do that, then. Figure out your algorithms correctly and do enough upsampling.
He warns that sometimes you don't just need to upsample the obvious audio waveforms, you may need to upsample the control signals as well, because the control signals can modulate the audio and generate frequencies in the resulting audio that were not in the original audio.
Well,
of course they can. If you turn the volume up and down 1000 times a second, you have an
amplitude modulator. And if you vary the speed 1000 times a second, you have a frequency modulator. Either will throw off sideband signals above and below the original frequences.
Anybody who knows the first thing about audio signal processing should be well aware of that, and know about the frequency content of their control signals and how they're modulating the audio signal.
Normally, this isn't much of a problem because you can ensure that the control signals don't have any high frequency content, and the sidebands they generate are just not far from the original frequencies, and therefore not much higher. You may not need any additional oversampling, or not much. But if you do need it,
do that, then.
This stuff is generally not rocket surgery. There is an entire well-developed field of computer science known as "numerical analysis" that's all about figuring out how to approximate things as near as you need to, putting bounds on the possible errors (distortions), and keeping things computationally efficient.
There are college courses and textbooks and research journals on numerical analysis and on digital signal processing, and handbooks and standardized libraries and cookbooks for a lot of this stuff.
You need to be properly educated, by taking the relevant courses, or by reading the right books and papers, if you have a solid background in CS and a decent grasp of engineering math.
If you like and understand math and CS, it's just not as scary as that article makes it out to be. It's not lions, tigers, and bears, and it's definitely not Here Be Dragons.
He presents a scary-looking equation that's actually not very complicated by the standards of mathematicians and engineers, presumably because it looks intimidating to most
musicians.
And he talks about things like how capacitors aren't just capacitors and have resistance and inductance, too. Oh no! There's some thing called "equivalent series resistance" that sounds scary.
That's AC circuits 101 stuff, but you wouldn't know it the way he talks about it. It's called equivalent series resistance because it's
easy to analyze by modeling it as a
resistor in series with the capacitor. No big deal.
He's making basic stuff that real engineers and computer scientists do before breakfast sound like terrifying, effectively impossible stuff for mere humans to get right.
Yeah, it's FUD.
Yeah, it's biased.
I don't know if the guy believes his own bullshit, and I don't care. It's bullshit.