Jimbroni

Well, no idea if logic can do this (cubase can) but if it can: Zoom WAY in on the pop and draw out the wavform so it connects better. That was probably a bad way of explaining it. Basically I'm telling you to re draw the wavform. yeah I dunno logic, but if you have the option to manually redraw the waveforms (the pencil tool) that's the way to go. I've tried a bunch of declickers and never found one that did the job without ruining the sound. Of course the pencil tool is more time consuming given alot of clicks and pops, but in the end it does the job without affecting the overall sound. I'm not sure if you know what a click looks like on a waveform, but in general its where the waveform will either spike straight line up or down (big volume change with no time component) or a wave curve will just suddenly drop off like a chunk has been taken out instead of continuing to slope back to zero. In both cases its the Instantaneous volume change(straight vertical line) that causes clicks.

Well regardless of the ongoing debate about sound quality. There are other factors that highend cards provide that are very important if not more important. When it comes to sound quality, your money is better spent on microphones, then onto preamps, but also room treatments. For interfaces, sound is important but.... more important are the connections. Does it have balanced or unbalanced line inputs or does it have pre's or does it have both and is switchable? Does it have toslink or aesbu or spdif? Does it have word clock or adat type or midi time code sync? How expandable is it? Can you eventually run 16 or 24 or 48 channels by adding on? Does it handle midi as well? Also its a sound card, so you have to remember the computer angle. How good are the drivers, or better yet how good is the driver support? My experience has left me very satisfied with RME. There driver support is top notch. But more importantly their customer service is great. I've dudes from their group call me on a saturday, regarding setup questions. I wasn't having product defects, I was experimenting with syncing two breakout boxes and an adat, and this guy called me back and spent an hour and a half trying to figure out how to make my software control it. Thats what you get with paying RME. Motu and RME have great units with a zillion ways to connect stuff and different things together, that is worth its weight in gold in a studio. However, with Motu I was hearing about far too many customer service issues. So I went RME. But my point don't just focus on the sound quality thing, because thats largely subjective. You have to remember this thing is the nerve center, so IMO you're better with a unit that has alot of connection types and expandability and comes from a reputable company that doesn't have alot of service issues. I've used Maudio stuff on other people's studios and quite frankly I thought it was decent stuff. But when I was looking to setup my thing, Maudio just didn't have right setup I was looking for, their stuff tended to be very stripped down and simple, and just didn't have everything I needed. Overall, I think they make some of best cheap {censored} out there. But its just not very expandable. I wanted a solid bedrock that I could keep building upon. Figure out what you want and need, Then Search the forums, for sound card info. Thats all you can do. Whatever DAW you use start at their forum, then look up how various sound cards work with it. See who's complainin' about what.

I'm still looking forward to seeing his oscilloscope results, because I just can't figure out what he's talking about when he says "parallel waveforms". Because there pretty much has to be only one waveform. How would you display more than one waveform on an oscilloscope anyway, unless you were using multiple test leads and what sense would that make? So I just don't know what he's saying. So I'm hoping to clarify that anyway. I don't mind people disagreeing with me. I just don't like it when we can't get our terminology on the same page, and I feel like that's happening here. Yeah I kinda agree. I think AN made some intriguing points but it was kinda like shatter shotgun, that confused alot of people. I would like to see these oscilloscope results as well. I have my opinions on what makes digital and analog sound different. Mainly the fact digital provides more of the spectrum, but more importantly the boundary conditions are different. Analog will never truncate, instead tapers or smears the boundaries, and digital will truncate. Providing more of the spectrum, IMO may not always be a good thing. In real life our ears will dull down that super crisp highend or harshness from a cymbal splash. With digital it just gets captured, and if you address in your eqing, whallah you got harsh top end. Either way if AN has some data that provide some clues into even more differences, I'm all for it.

yes and no. but it will never be simultenous like analog. Thats the quality thats really being debated. The occurance of simultenous data broadcast. but what your neglecting is that each of those transforms occurs in rapid succesion. Not all at once. No it is all at once. When the waveform is reconstructed, thats what you hear. The time it takes to decompose, do the series of summations, and reconstruction is why we have latency.

digital equipment does not do anything simultanoesuly ever. you have a clock and in that clock cycle you can execute instructions. However you can only execute one instruction or a series of instructions per clock cycle. Ie 8+8 = 16 then the next 4+4=8. then you can add 16+8 = 24 You can't do this 8+8=16 4+4=8 and have them both sum and translate to 24 in one cycle in parralellel. Actually no. This is exactly what fourier transform does. It decomposes the signal into a series of sinewave additions and then reconstructs them. Thats the definition of a summation series. It doesn't just lump it all together and try to guess, it breaks it down into series of summations, and then lumps it back together.

So far even though Gubu was providing alot of misinformation, he really has been the only one to really give a valid excuse for using high sample rates. Capturing harmonics for 20 hz fundamental. However, in the real world mixing. I don't think there any instruments that have 20 hz fundamental, so he would really be refering to 2nd, 3rd, 4th or higher order harmonics that are getting reduced to sinewaves. My point being that those harmonics, are so low dynamically that what real impact do they have to the mixing environment. My answer it depends alot on the instrument being recorded. In the world of pop, rock, metal, blah, blah, blah. I say there is basically no value up there. As the instrument fundamental are all below 10K and the harmonics past 20K aren't as valueable relatively speaking. However, once you get into the world of classical, with french horns, trumpets, violins, etc, the harmonic information is more valuable and very well may justify higher sample rates. So in my world, when recording rock n roll and pop, I've found no value in recording above 48K, which means for a 6K note I can capture a 3rd order harmonic. The highest note on a 24 fret guitar is 1175. So one could argue what about cymbals, well even then your getting your second and third order harmonics, as most cymbals are in the 2 to 6K range. Which to my ears is just fine. But then you get into the problem that even if you capture all the upper harmonics in the world, speaker systems do not reproduce those harmonics any way. Analog never equipment never reproduced those harmonics, Which is probably alot of the reason people go for that warm analog sound, it doesn't contain all that extra information. The difference of course being that instead of truncating the info, its just blurred out or not there at all.

Not if the higher harmonics that make it a square wave are being filtered out by the LPF that would be active when converting at 44.1 good point, but those higher harmonics aren't there to begin with. No microphone will produce them.

But if I input a 20kHz square or other oddly shaped waveform, it will always come out as a sine wave? nope it comes out as a square wave, because a square wave is a bunch of sine waves added together.

Math does not give you better time domain resolution. Only a higher sample rate can do that. Is it because there's electricity in them wires or because of my coffee beans?

No buddy, the voltage comes from the input. Actually it originates at the source (instrument/microphone). But that proves your point how? Because coffee beans give me energy, 96K sample rates are only way to accurately produce 20k signals.

So there's no voltage being sampled at all? It's all just math is it? Nah The voltage comes from the coffee grinder.

Put this into your calculator then:- How does a convertor running at 44.1kHz know whether a 20kHz waveform is sinusoidal, square, sawtooth or Mickey Mouse doing a tap dance when it's only sampling the voltage 2 times per cycle? I'm sure the complex math gives you back some shape of a waveform, just not necessarily the waveform you put in. Until someone who actually develops and designs convertors from scratch comes on here and tells me different, I am not accepting that 44.1kHz or 48kHz samples frequencies at the upper limit of human hearing as accurately as 96kHz or 192kHz. I've done A/B recording tests and there is an audible difference, so you guys can tell me what you think I heard and tell me that I'm full of {censored} from now til the end of time, the map is not the waveform, no matter how convoluted the math is in sampling it and reproducing it. More accuracy in the time domain, for that's what a higher sample rate gives you, produces a more accurate reproduction of the waveform. Good luck to ye All waves are sine waves. A square wave is just a few sinewaves added together and so on.

"sampling at 44.1k produces square waves. And that's a fact!" Sincerely that is funniest BS I've hear in long time. And I've heard some doosies. I honestly have no idea how you come to that conclusion, all I can imagine is somehow you think this is like connect the dots, with a straight edge. The math of sound is based on complex addition/ subtraction of sinewaves not straight lines. Because sinewaves are symmetrical, all you is need two points anywhere within its period and you can determine everything you need to redraw it.

The mics capture the waveform up to their threshold, that's fine but what you've said there is like saying that there is no difference between Mozart and my grandmother when you play them thru a transistor radio. I'm sorry sir, but there is... mozart, grandma, and math have nothing in common.

It can't possibly do so. And ok maybe it might, I'm not a boffin, but how about 10 samples per 9kHz waveform than 5, that's gotta be a better representation of what went in, no? Sure its better, but your ears can't hear it, your microphones can't capture it, and your speakers can't play it.