96k is better than 44.1k sample rate. why?

Hi_Flyer said:
neat stuff. I might have to try my next project at 48kHz. I don't think I can hear much of those frequencies though. My ears aren't what they used to be. I've played with too many loud drummers.
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My ears are not good either. Even so, give it a test and decide for yourself. In the end, you'll need a good SRC to go to 44.1 without hurting your mix, but it might be worth it.
 
TheDewd said:
2) A myth? Sample rate conversion is complicated whenever there isn't a common integer by which you can divide the samples. In that case, you have to actually choose which sample you keep, while having twice the sample frequency allows you to keep one sampe out of two.
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SRCs at the simplest level work by multipling to a number than has both rates as a common denominator.

Also, consider the work your A/D converter is doing: it actuallly has a sample rate in excess of 2mHz. 44.1 or 48 kHz is simply the data rate; it takes a few bits at a very high rate and reduces that to lots of bits at a number of different lower rates. Whether you select 44.1, 48, 88.2, 96, or 192, the true sample rate is unchanged.
 
to add to mshillarious

remember unless you like driving a car without brakes, drink out of a toilet, and jump out of perfectly good airplanes, your going to be running an anti aliasing filter as you downsample

so theres some new numbers youll be multiplying in there anyway.
 
Going back to the original question...Nyquist answered it best. Now, can you hear the difference?

I'm too old to tell
 
Lets be sure of what nyquist said

IF you have a bandlimited waveform, you can *completely* describe that signal by taking pictures of it at twice the highest frequency of the signal. Even further, any BANDLIMITED signal can be perfectly described with only two points, with those points being twice the frequency of that waveform

Hopefully Im paraphrasing correctly

It says nothing of 44k vs 96k, except that you could accurately describe a BANDLIMITED signal up to 20khz with a sampling frequncy of 40Khz

We gotta agree where the frequency is that doesnt matter anymore

Unfortunately, gear that is pretty flat from DC to light, doesnt usually have a mode or variation that is bandlimited to something much lower, so when we say those pieces of gear that go WAY up sound better than gear that doesnt, its not a fair comparison as so much else could be involved.

We also, and heres the critical deal, must be able to bandlimit the signal

You're going to have a WAY easier and nicer time bandlimiting a signal under 22-24khz sampling at 96khz than at 44.1khz. Dan Lavry seems to say something around 60khz would be great

44.1khz, thats cutting it WAY close, but in theory, IF WE AGREE that 20khz is the upper limit, it should be doable
 
You're absolutely right...won't argue a thing you've posted. The "critical" frequency is much to high to be concerned with in this application. I was simply beating around the bush with the concept of more samples per cycle renders a closer more true representation of the original signal (in broad and general terms of course). I also mention that the differences between the two sampling rates is very difficult to detect with the human ear.

Good stuff!
 
punkin said:
I also mention that the differences between the two sampling rates is very difficult to detect with the human ear.

Good stuff!
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W00t the actual IMPORTANT part of the question, and yet the one that always seems to get the LEAST attention in these type of threads LOL
 
pipelineaudio said:
W00t the actual IMPORTANT part of the question, and yet the one that always seems to get the LEAST attention in these type of threads LOL
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mshilarious - Great stuff!

Another thing that seems to get lost in these types of threads is the affects from processing. While the final format is 44/16 how many would process their data at 16 bits over 24 bits? The same sort of logic applies to sampling rates.

The biggest benefit of higher bit depths and sampling rates is to help reduce as many artifacts as possible during the processing stage. By having higher resolution data during processing you are going to be reducing the affects of anti-alias filtering multiple times, quantization distortion etc.

Can you hear the difference between audio that was recorded at 44/16 and audio recorded at 96/24 then brought down to 44/16? Obviously debatable.

The other question is can you hear the difference between the two after both have been processed digitally multiple times and then converted?
 
OK more fun graphs. Today I test some newer converters than the AI3s I used to own, now I can test 96kHz too :)

Again I used my old Yamaha digital piano, since it has some canned sequences that are 100% repeatable, direct into my ART Digital MPA, which has a very good converter chip. I forget which one it is, but search my old posts, you'll find it :o

I won't torture you with the audio samples, but if you are desperate to hear them, maybe I'll post a link later. I have not done a formal A/B test, but I think 96 sounds a little crisper than 44.1, I haven't A/Bed 48 at all yet. These graphs take time :(

Here is 44.1 vs. 96:
 
So then I started to wonder how UA does SRCs with no attenuation. I'm guessing maybe some sort of EQ adjustment, like a preemphasis or something.

Here was my very rough attempt at the same, an 8dB boost at 24kHz (while at 96kHz rate) with the Q set at 20dB/octave:
 
Am I reading the graphs correctly? It looks like trhe difference doesn't even start until 19705hz - I can't hardly hear anything at that frequency anyway.

There must be more to it than that???
 
NL5 said:
Am I reading the graphs correctly? It looks like trhe difference doesn't even start until 19705hz - I can't hardly hear anything at that frequency anyway.

There must be more to it than that???
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At 44.1kHz, it starts at about 18.5kHz. I can hear a sine wave at that frequency at 70dBSPL or so, but I think there is more to it than that, the difference is audible, but hard to describe. It doesn't sound "dark" or anything, maybe a little fuzzy.

Anyway, it's pretty much what the theory describes, so the remarks about audible differences at 7kHz and 10kHz are, from my point of view, without theoretical basis or experimental evidence.
 
Audible difference or no?

So does a studio that has a sample rate of 96khz better than the 44.1khz?
What is the conclusion with all this research and graphs?
 
Is English your first language? Your sentence made no sense.

Dumby said:
So does a studio that has a sample rate of 96khz better than the 44.1khz?
What is the conclusion with all this research and graphs?
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All very interesting, and I have a lot of respect for people who really know their stuff. But...I can't help but wonder how much difference (between, say, standard CD quality and 192/24-bit/super-pro audiophile quality) most of us who have spent years playing in loud rock bands can really hear. Most of my friends think that MP3s are the ultimate in sound quality, and anything better than a $200 home stereo would be wasted on them.
 
mshilarious said:
So then I started to wonder how UA does SRCs with no attenuation. I'm guessing maybe some sort of EQ adjustment, like a preemphasis or something.

Here was my very rough attempt at the same, an 8dB boost at 24kHz (while at 96kHz rate) with the Q set at 20dB/octave:
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So it looks like you hit the nail on the head, good guessing.
 
Filter/converter issues aside (definitely a legit point): is it the frequency response and sampling rate of your equipment that really matters, or is it the frequency response and sampling rate of an actual set of human ears??? can digital encoders actually hear sound more continuously than we can? mathematically, it seems like a distinct possibility...if a tree falls where no one can hear it, does it make a sound? and if it doesn't make a sound, can a computer record it? :eek:
 
drossfile said:
is it the frequency response and sampling rate of your equipment that really matters, or is it the frequency response and sampling rate of an actual set of human ears???
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the question why? why go 192Khz? can we hear the improvement?
after reading this very interesting thread, it comes down to using the best technology available, whether we humans can hear it or not. imo.

from a manufacturing viewpoint, eventually the manufacturers make the decison for us by shutting down the old line, for example the 8bit dacs become almost extinct like the dinosaur as the shelf fills with 24/192's.

Or are the audiophiles leading the pack again? and its a real deal!
"The consumers don't have 24/192 playback cardecks.yet!
if people can really hear an improvement they'll buy it. Like HDTV, its a real noticeable improvement and people are gobbling the TV's and converters up as the price comes down.

But when do the converters become like this typing-keyboard I'm using?
it can type faster than me already, so why upgrade?
 
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