Hi all of you,

I would like to add a practical point of view. 

My violin dimension library take more than 60 Gig on my hard disk (more than 700 Go for my still incomplete VSL's collection; I can already figure that the whole DImension String will take about 200 Gig). That stuff is using a lot of Ram too. So I planned buying a second SSD (drive SSD 1 is allocated to OS and aplications) to replace my 2 Tb internal HD. I'll need a 1Tb SSD and a lot of new memory ("just" 16 Gig at the moment). Moving to 96kHz would cost at least 200% $.

But space is not all. There's also the CPU… My iMac24 2011 (i7) is OK for now. I'm not sure at all it will be powerful enough for my future projects. The new generation of Intel's CPU does a gain of 20-30%. Interesting, but enough? Luckly, I have a second computer, my new MacBook Retina (i7 and SSD). I can use it to relieve my first computer. I am using MIR. So I will have to buy a second license (MIR 24?). I could save money by sending the audio signal from each of the instruments that are on my MacBook to MIR on my iMac. This can work, but the number of instruments should not be too large, otherwise the Ethernet connection between the two devices could be saturated. All this is in the real world with 24bit/44kHz samples. I dare not imagine what would happen with 96kHz samples. A world at 192kHz seems simply unrealistic for still many years.

As many have pointed out, the idea that a sound recording is better by sampling with resolution higher than 48khz is a myth. One can speculate that such a resolution could be useful when complex calculations (EQ, stretching, etc.). But scientific studies with blind tests show that there is no real interest to keep this "quality" for the final product.

Anyway, what I wanted to put forward in my comment is that only important quality gain can justify the additional investment resulting from the transition to the 96kHz standard. In practice, I join those who think it would be a waste of resources (financial as well as technical).

Have a nice day.

@gregb said:

I can hear it with any decent flat set of monitors and any decent mic - more air and less harsh top. Fact is 16/44 was sold as being perfect and it clearly is not....24/44.1 is also not perfect. Why dont you try your own scientific 24/44 24/192 test? :o) I'm out.

You really haven't been reading what's been stated here, from a true 100% accurate scientific point of view - you cannot hear anything above 22KHz - so scientifically recording at 96KHz will make no audable difference, because 44.1KHz will capture everything and beyond what you can hear. If you really can hear a difference, then you best buy a  better quality interface that doesn't add distortion to your signal.

Still,  the whole market relies on gullible people that think something is better when it actually isn't - it's called marketing.... 

Dear Gregb,

Of course I tried 24/92 kHz (on acoustic guitar particularly). Very good mic and monitors and top quality Headphones, more than decent preamp and DA converter, etc. I made blind comparaison. Tested it with some friends too. 24/48 kHz recordings were choosen about 60% (seems to sound a little bit less «Digital»). After two years of hard working, I finished a CD few weeks ago (mix of samples and real audio takes). At my surprize, it seems to sound a little bit better on my audio system after I went from 24/48Khz to 16/44kHz! Dithering added some good «noise»… 😉. So I tried to understand why and I read some interesting papers like this one (see particularly last pages and conclusion).

The 16/24 bit choice is a totally different question. 24 bit adds obvious and tangible advantage for studio working (much much more dynamics). Not so sure for final listener since most of modern music (on CD or other media) is very compressed («maximising» could be a more appropriate word).

Our sound (and music) perception is as subjective than objective. And perhaps it's a good thing after all.

Sorry if you're going out.

 No, I don't assume 44.1KHz captures perfectly upto 20KHz,  I know it DOES.

Here is an intresting web pages that demonstrates this:-

http://www2.egr.uh.edu/~glover/applets/Sampling/Sampling.html

It is showing that there is absolutely no point in sampling more than twice the required frequency, that was discovered in 1928 and proved in 1959.

You must never overlook the fact that human beings cannot hear anything above 22KHz (most people struggle above 18KHz), so any harmonics above that will never be heard and thus there is no point in capturing them.

Actually, have you realised that any fundamental waveform above 10KHz will sound like a sine wave to us?  Because the second harmonic is beyond what we can hear.

@andyjh said:

 No, I don't assume 44.1KHz captures perfectly upto 20KHz,  I know it DOES.

Here is an intresting web pages that demonstrates this:-

http://www2.egr.uh.edu/~glover/applets/Sampling/Sampling.html

It is showing that there is absolutely no point in sampling more than twice the required frequency, that was discovered in 1928 and proved in 1959.

You must never overlook the fact that human beings cannot hear anything above 22KHz (most people struggle above 18KHz), so any harmonics above that will never be heard and thus there is no point in capturing them.

Actually, have you realised that any fundamental waveform above 10KHz will sound like a sine wave to us?  Because the second harmonic is beyond what we can hear.

 The half frequency theory (Nyquist) does state that it assumes a perfect bandwidth limit (a theoritcal brick wall filter - which does not seem possible),  and there is a slight "grey" area when it comes to exactly half the frequency (that would be 22.05 KHz sampling at 44.1KHz).  So this is why the frequency response of a sampling frequency is always less than half, to allow for a real world filter.

You must not allow anything over half the sampling frequency to be sampled, because it will alias, and you cannot sort out aliasing on playback, because the frequencies have already been "folded back" are then part of the sub 20KHz signal.

It is how good this filter is, that determines the quality of an A/D conversion, if the filter is weak it may allow aliasing to happen, if it is over zealous, it will start to affect audible frequencies that it should not be affecting.  A good quality filter design will perfectly deal with the critical 22Khz frequency, yet be perfectly out at 20KHz.  But - some converters will be struggling.  If you shift the sampling to 96KHz, you can put a very lazy filter in there that easily kills any aliasing, and does not affect sub 20KHz frequencies.  So it's a lot easier to make a perfect 96KHz sampling frequency than it is a 44.1KHz one. Which is why some audio interfaces "seem" to sound better at 96KHz.

It is however better to buy a top quality audio interface and run at 44.1KHz (or 48KHz if your output medium requires it), than invest in a super fast computer that has to deal with 96KHz sampling.  The end result wil be the same. Especially as the final product is unlikely to be at 96KHz anyway.

@andyjh said:

 The half frequency theory (Nyquist) does state that it assumes a perfect bandwidth limit (a theoritcal brick wall filter - which does not seem possible),  and there is a slight "grey" area when it comes to exactly half the frequency (that would be 22.05 KHz sampling at 44.1KHz).  So this is why the frequency response of a sampling frequency is always less than half, to allow for a real world filter.

You must not allow anything over half the sampling frequency to be sampled, because it will alias, and you cannot sort out aliasing on playback, because the frequencies have already been "folded back" are then part of the sub 20KHz signal.

It is how good this filter is, that determines the quality of an A/D conversion, if the filter is weak it may allow aliasing to happen, if it is over zealous, it will start to affect audible frequencies that it should not be affecting.  A good quality filter design will perfectly deal with the critical 22Khz frequency, yet be perfectly out at 20KHz.  But - some converters will be struggling.  If you shift the sampling to 96KHz, you can put a very lazy filter in there that easily kills any aliasing, and does not affect sub 20KHz frequencies.  So it's a lot easier to make a perfect 96KHz sampling frequency than it is a 44.1KHz one. Which is why some audio interfaces "seem" to sound better at 96KHz.

It is however better to buy a top quality audio interface and run at 44.1KHz (or 48KHz if your output medium requires it), than invest in a super fast computer that has to deal with 96KHz sampling.  The end result wil be the same. Especially as the final product is unlikely to be at 96KHz anyway.

@gregb said:

---- You have not explained fig 8. i.e. more than 2 signals both sharing the same sampling points but both below nyquist/2..... according to you this error should not exist.

Fig 8 is showing 3 sines at the same frequency, but phase shifted by +/- 90 degrees and at the critical frequency (which is half the sampling frequency), which is why I suggested "half" the frequency is a grey area, because it needs to be just under half (-1Hz). An audio waveshape would not look like this, it would be one composite wave shape, but we do need to describe things in terms of sine waves as that is how the maths works.

I guess the required sampling frequency should be described as double plus 1Hz,  though at 44.1KHz we are already double plus 2050Hz - which well clears the critical frequency.  Aliasing filters is the enemy not the critical frequency.