No I've not.. . I still have never used beyond 44.1.. I really should!

I just got an audio interface a little while back capable of 96khz recording but still have never bothered... I'll do some tests soon.

I suspect that it's not possible to tell the difference, but there might be some kind of strange unexpected audible artifacts that appear in music when its converted from 44.1 to an analogue signal, that you dont get if you convert 96 to analogue. . or something like that..

I'm not sure what exactly, but its possible that a 44.1khz signal put through D/As creates a lower frequency interference somehow... some kind of resonance in electronics components or something, who knows!

This reminds me also of Robert Henke (one of the guys who works on ableton live) talking about how all the major DAWs recently shifted to using internal procesing at 64bit instead of 32bit to give more headroom and lower possible signal-noise ratios etc..


He basically described that going above 32 bit is pointless (the amount of headroom you have is like 192db already with 32bit - in other words, enough to send you deaf in a very short time, with perfect clarity of sound.), its just a gimmick that people buy into - one DAW changed to 64 bit so tehy could advertise as being 64bit compared to all the other's which were 32 bit.

Technically, it is of absolutely no use whatsoever - and it wastes processing power. But once one DAW went 64 bit, all the others were forced to follow or else they would be perceived as being backwards or not as good.

wow... industry trends i guess. and youre right izotope ozone 3 was advertised to have 64bit blah blah blah. but in the same breath mentioned that "you wont really hear it but, just in case..."


well maybe not those exact words but in a round about way thats what it was...

if you do get a chance, do like a 1 min or less test that u can post. if thats not asking too much. my mbox only goes to 48k so im sol.

Credit: Dr. Nyquist discovered the sampling theorem, one of technology’s fundamental
building blocks. Dr. Nyquist received a PhD in Physics from Yale University. He discovered his
sampling theory while working for Bell Labs, and was highly respected by Claude Shannon,
the father of information theory.

Nyquist Sampling Theory: A sampled waveforms contains ALL the information without any
distortions, when the sampling rate exceeds twice the highest frequency contained by the
sampled waveform.

Introduction

While this article offers a general explanation of sampling, the author's motivation is to help
dispel the wide spread misconceptions regarding sampling of audio at a rate of 192KHz. This
misconception, propagated by industry salesmen, is built on false premises, contrary to the
fundamental theories that made digital communication and processing possible.

The notion that more is better may appeal to one's common sense. Presented with analogies
such as more pixels for better video, or faster clock to speed computers, one may be misled to
believe that faster sampling will yield better resolution and detail. The analogies are wrong.
The great value offered by Nyquist's theorem is the realization that we have ALL the
information with 100% of the detail, and no distortions, without the burden of "extra fast"
sampling.

Nyquist pointed out that the sampling rate needs only to exceed twice the signal bandwidth.
What is the audio bandwidth? Research shows that musical instruments may produce energy
above 20 KHz, but there is little sound energy at above 40KHz. Most microphones do not pick
up sound at much over 20KHz. Human hearing rarely exceeds 20KHz, and certainly does not
reach 40KHz. The above suggests that 88.2 or 96KHz would be overkill. In fact all the
objections regarding audio sampling at 44.1KHz, (including the arguments relating to pre
ringing of an FIR filter) are long gone by increasing sampling to about 60KHz.

Sampling at 192KHz produces larger files requiring more storage space and slowing down the
transmission. Sampling at 192KHz produces a huge burden on the computational processing
speed requirements. There is also a tradeoff between speed and accuracy. Conversion at
100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach
50-60Hz we get near 24 bits. Speed related inaccuracies are due to real circuit considerations,
such as charging capacitors, amplifier settling and more. Slowing down improves accuracy.

So if going as fast as say 88.2 or 96KHz is already faster than the optimal rate, how can we
explain the need for 192KHz sampling? Some tried to present it as a benefit due to narrower
impulse response: implying either "better ability to locate a sonic impulse in space" or "a more
analog like behavior". Such claims show a complete lack of understanding of signal theory
fundamentals. We talk about bandwidth when addressing frequency content. We talk about
impulse response when dealing with the time domain. Yet they are one of the same. An
argument in favor of microsecond impulse is an argument for a Mega Hertz audio system.
There is no need for such a system. The most exceptional human ear is far from being able to
respond to frequencies above 40K. That is the reason musical instruments, microphones and
Copyright Dan Lavry, Lavry Engineering, Inc, 2004
speakers are design to accommodate realistic audio bandwidth, not Mega Hertz
bandwidth.

Audio sample rate is the rate of the audio data. Such data may be generated by an AD
converter, received and played by a DA converter, or even altered by a Sample Rate
converter.
Much confusion regarding sample rates stems from the fact that some localized processes
happen at much faster rates than the data rate. For example, most front ends of modern AD
(the modulator section) work at rates between 64 and 512 faster than a basic 44.1 or 48KHz
system. This is 16 to 128 times faster than 192KHz. Such speedy operation yields only a few
bits. Following such high speed low bits intermediary outcome is a process called decimation,
slowing down the speed for more bits. There is a tradeoff between speed and accuracy. The
localized converter circuit (few bits at MHz speeds) is followed by a decimation circuit, yielding
the required bits at the final sample rate.

Both the overall system data rate and the increased processing rate at specific locations (an
intermediary step towards the final rate) are often referred to as “sample rate”. The reader is
encouraged to make a distinction between the audio sample rate (which is the rate of audio
data) and other sample rates (such as the sample rate of an AD converter input stage or an
over sampling DA’s output stage).


^^^the above taken from Dan Lavry's web page..Lavry makes kick ass converters

Deta!l... I mean Platypus Beird, droppin science... like woah. Good stuff man.

^^ yes that was the shyt! clears up alot!

I've always just looked at it as sample rate is all about audible frequency and bitrate is all about noise floor. As mentioned, 44.1khz is right at the human threshold so unless you've got dog ears, I never saw the point in wasting the space nor putting extra load on the machine for what's likely to be imperceivable. However a little bump in bitrate (ie, 24bit) can be good because the noise floor drops significantly... which meanss that you can technically record at a lower volume, have more headroom for mixing, and still introduce less noise.

That being said I keep all projects on my computer at 44.1-16bit, tho I use all sorts of lower sample rates in the creative process. I just throw the idea of noise floor out the window since it's likely I'm recording from a 4-track. The s/n ratio on the 4-track introduces tons of lovely noise.