Thanks for the info Nikki-k. I already had an idea about some of what you were talking about in your post. From what I've kind of been led to understand from my time spent trying to learn that a lot of the higher sample rate stuff was kind of along the lines of the rage of over-sampling back in the early 90's. I know that you remember. Everyone had 2X and then 4X and eventually it got up to about 64X. I mean that's great and everything, but from what I can figure out the only thing that really did for anyone was allow them to use cheaper parts to create a steeper cutoff for the low pass filters. And to be honest I can see where the advantage would be in the recording side; I don't get it for playback or D>A conversion, but that's besides the point here. I'm just saying that I do get the point that the converters are what make or break a device and that by increasing the sample rate improves quality, but its usually in the name of saving money on the discrete components that add cost first and foremost.
So that does explain (at least for me) the phenomenon with the 11R. Sure I could use other sample rates, but they all sounded like crap (and I'm not just bashing the thing here because it's not that bad) unless you used 96kHz. I mean the options are nice to have, but if it sounds bad then why even have it in there? I suppose that kind of explains why Cliff chose one sample rate to rule them all so to speak.
And yeah, I've done a lot of testing of my gear and my current audio interface does a really good job at various sample rates and like I said earlier, I can't hear the difference between 96kHz and 48kHz. Its the same noise floor, high end, texture, levels, frequency response, whatever. Its the same. But taking a digital signal straight from the 11R at a down-sampled rate sounded horrendously muddy and muffled.
Im a noob, but 96khz what?
And just to back up what clarky was saying about bit-depth and stuff for nellings; I do feel really comfortable with this area. I look at it like someone giving me a ruler to take measurement (which is all you're doing when you digitize something). The sample is like a snapshot of the signal's voltage at a specific point in time and you have to assign a value to it by measuring it. Now if you have something that you have to measure that is between 0-12 inches and use a 12" ruler marked with only inches (pretend that's 8-bit) you can get close. Write down that measurement and then take the next sample, measure it, write it down, etc. Now because you know how many times per second you wrote those measurements down (sample rate) you can recreate them by 'playing' them back at that same rate or speed at which you wrote them down.
Now any measurement your took that was exactly in one inch increment is fine, but once you have stuff in fractions its been averaged and you lose accuracy. Most of the time that's not a huge deal to be honest. Where it really rears its ugly head is when you are taking consecutive measurements where there's no dynamic range (almost no variations in the signal's voltage). Once you start getting similar measurements in a row it becomes this big old nasty flat line which sounds like clipping, but it's usually at very lower volumes. A higher bit depth like 16-bit will give you the equivalent to a ruler with 1/64th increments. Now all those measurements below 1 inch can actually be measured and given a real value other than just 0 or 1 inch so you don't get that straight line.
This is why you may see it said in places that 16-bit gives you -96dB of SNR. Basically that's saying that it can register a value -96dB from 0dBFS (the highest value you can record without clipping). That's like trying to measure 13 inches on our ruler; we can only go up to 12.
In case you haven't noticed this is getting a little long winded. I'll try to get to the point.
8-bit is 256 values because you've got 8 registers in your byte. It can range from 00000000-111111111. The highest value is all ones and anything higher than that is still going to be all ones. That's your classic clipping from overdriving your A>D converter. The other disadvantage with 8-bit is that it can only get down to -48dB for SNR. That sounds pretty decent on paper and if you record something that is constantly loud its not going to be a problem. But if you have quiet passages it will distort. You'll get a bunch of zeros in multiple consecutive bytes (your actual measurements) and it sounds like really quiet digital clipping. I believe that it's called granulation noise. You also have 256 values so there's a lot of averaging going on for every sample. 8-bit sounds like crap, but because it's a smaller byte it does have its applications. Just not in music.
16 bit gives you 65,536 levels, -96dB and that's actually pretty good for most things. Its what CD's use.
But 24-bit gives you over a million levels and -144dB SNR. That's like taking a micrometer to make those measurements. Granulation noise is damn near non-existent. Honestly I've never really needed anything higher than 16-bit until I started getting into re-amping. The signal level is like -40dB at its peaks. That extra resolution at those lower voltage levels makes a world of difference in that case.
Hopefully nothing in there is wrong; like I said I'm more of a concept guy.
Just to try to summarize my opinions; sample rates are cool; look up Nyquist theorem if you're really interested. The jist of it is that your sample rate has to be at least twice the frequency of the highest sound you want to record. Even if someone claims they can hear above 20kHz no musical information is in that range so they can kiss my butt.
Sample rate will determine the highest frequency you can replicate digitally and as long as you've got good quality A>D converters you can get the full range of human hearing (and more importantly any type of music) at 44.1kHz with no problem at all. And not to get too far off track here, but how many people actually
like to hear stuff above about 15kHz anyway? That's like the scan rate of an old TV set. If you can't hear an old TV whining with no volume then you can't hear anything above what a 32kHz sample rate would give you. If you like to listen to an old TV whine then you need professional help.
But to get back to my point; too much emphasis has been placed on sample rate; its all about the bit depth in my book.
Sorry, I was all over the place.