Muziekschuur at home


Hi Eratu,
 
Freddie,

In the nineties the same was happening with consoles. It was about how many khz their preamps and mixingconsoles let through.
In the end SSL Neve and D&R (among others) created great consoles wich let through up to 100khz and sometimes up to 200khz.

If you have a lowpass filter to cut down noise in a cd-player it damages some high frequencies because of the lack of steepness of analog filtering. Thus take away some the "realistic feel" of the recording.
44.1 was a wrongly chosen sampleformat created by Philips. But at the time the lack of wow and flutter a noise a big advantage over analog records. So Philips did not wait untill digital systems were more powerfull. They simply released technology they could create. The rest is history.
 
When you record music with high frequency content (every bang on cymbals will give frequencies up to 50khz (and above but much more weakened). When recording at 44.1. khz. (every sample is half a wavefile so 44.1 can hold up a frequency response of up to 22,5 khz. We loose much of of the high frequency content when loosing those 50khz to 22.1 khz content. We allso loose the lower harmonics of that part of the signal.

And by the way. We are sensitive to phase up to 200khz.

So, much of the localisation content of the signal is lost when recording at 44.1 and 48khz.

This is what the managing director of D&R (Duco de Rijke) had to say about this subject (about analog consoles).
Taken from the D&R website.... I think it is pretty interesting:
--------------------------------------------------------------------------------------------------------------
Did YOU know why analog sounds better than digital?

D&R started long ago (1972) making mixing consoles equipped with tubes.

One of the nicest parts of these hot valves were that they sounded great.

The sound was pleasantly warm and overload was not an issue with over 300 volts on the power rails and a bandwidth of up to 100kHz.

Then we experimented with transistors and could not get the same sound as we had with tubes.

Up to today we are using in most of our consoles integrated circuits mostly known as IC’s.

There is an enormous offer of IC’s on the market and they all have their specific pro’s and con’s.

They are different from tubes and transistors and sound different. Some sound hard and others more smoothly. But connected to a high end measuring system such as System-1, specs are pretty much the same. So what causes the differences we are experiencing?



We human beings are capable of perceiving sound in extremely small details.

We can even notice extremely small amounts of distortion in the order of 0.001% of the nominal signal level. We are extremely unpleasantly sensitive to odd harmonics such as the 3rd, 5th, 7th, 9th 11th and so on of the fundamental frequency.

Another problem in IC’s is the crossover distortion in the output stage producing not only dissonant harmonics but also switching transients when using a not so good op-amp.

We have selected our op-amps to behave in class A/B towards class A resulting in a minimum of distortion that we are so extremely sensitive to.

From a well known designer in our industry, the well respected Mr. Rupert Neve, I was pleasantly informed about the results of the following test, that anyone can repeat when interested.

This test mostly confirmed what we already knew by experience but we could not prove that designing our consoles this way improved the sound dramatically as it did.





WIDEBAND WIDTH DESIGN.



Imagine listening to a sine wave of 1kHz, this should be a nice pure round sound like a pure tone of an old Hammond organ. Then switch this 1kHz frequency to a square wave, now you are listening to a sine wave plus on top of it lots of odd harmonics such as 3rd, 5th, 7th and so on .

Actually you are listening to 1 kHz plus on top of this on a very low level frequency 3khz, 5kHz , 7kHz 11khz and so on. So the difference in perceived sound is quite clear to any listener, nothing mystical here.

Now switch the fundamental 1kHz sine wave to let’s say 12kHz. Again a pure very high frequency tone could be noticed.

But now the funny part of it. If this 12kHz sine wave is switched to a square wave we all still can hear the difference between sine and square wave.





THIS IS AMAZING.



If we all agree on this, it means that we can hear or at least notice signals beyond 20kHz.

Remember the first harmonic content above 12kHz is 24kHz , and then 36kHz and so on.

These generated frequencies above 20kHz (even in small amounts) prove that it is important to have an audio system that has a bandwidth well over 20kHz.

If you repeat this test for a fundamental of let’s say 16kHz, the first order harmonic is around 32khz and higher. Even on this frequency most of the experienced professional listeners could easily determine when they were listening to a sine wave or a square wave!



Amazingly is not it? At least I was amazed and immediately knew why we should continue making our console having a bandwidth of 100kHz and higher.

We trim our op-amps on the board for as long as I can remember to be stable at 40kHz square waves without overshoots and frequency roll off. We are convinced that if we tune our op-amps this way they are capable of amplifying any sum or difference of the fundamental frequency within the audio band without any coloration. Our reputation for perfect sounding consoles proved we were right.



The simple test described above proves that we are capable of perceiving sound information well above 20kHz and that this information absolutely leads to a warmer and richer sound . How we humans are doing this is still a miracle to me and many others I heard.

It is absolutely a fact that a small lift in frequency response of audio equipment well above 20kHz could easily give you the impression of giving you more air or transparency. Based on this principle we have designed our equalizers in a way that the upper band is lifting the fundamental frequency well above 20kHz if needed., resulting in a smooth breeze of air in your control room sound wise.





DIGITAL SHORTCOMINGS



Can you imagine why most digital consoles today sound harsh and can not even touch the smoothness of well designed analog consoles. This is easy understandable now with frequency responses that sharply fall off at 20kHz. No information above 20kHz means no warm open sound.

Imagine what a digital upper shelving high frequency control is doing when lifting those frequencies by 16dB just before they sharply fall off at 20kHz.

I think that digital technology has along way to go before it can even touch analog audio sound wise.

Think about this when you are ready for a new console!
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Here is the thread about this subject I started a year ago;
http://www.gearslutz.com/board/high-end/128405-hearing-above-20khz-analog-consoles.html

Dan Lavry was nice enough to comment there aswell. 6 core cpu's will be available in a few months. When the technology advances It might well be possible sampling at 96khz and above will get much more accurate.

Sony Oxford created Hyper Mac for that. And supports over Gb-lan up to 384khz samplerates.

So we will see what the future brings.

More importantly. Bruell & Kjaer did create microphones in the nineties who could capture that high frequency content. Up to that point high frequencies could not be captured in that accuracy. (allthough tape machines record up to 50khz.

Bruell & Kjaer now has a brand called DPA microphones. When you hear those microphones (like the 4006 or the 4090) you soon realise that high frequencies add alot to a mix....

Muziekschuur

 
 
HEAR HEAR!

bitflipper

If this 12kHz sine wave is switched to a square wave we all still can hear the difference between sine and square wave...THIS IS AMAZING.

That would be amazing, indeed. Unfortunately, it is also untrue.

How are you generating the 12KHz test tone? You'll need a laboratory signal generator to do that; no synthesizer I know of can even go half that high.

How are you playing the 12Khz test tone? A few speakers (Tannoys come to mind) advertise extended high-frequency response up to about 24KHz, but you'd need to be able to reproduce 36KHz just to get the lowest of the significant harmonic components.

 
 
?
Bit!
 
Take a---> EQ-->add 12 kHz "done"--->  now I have generator that generator that!
 
Regards
Freddie

UnderTow

Muziekschuur at home

In the nineties the same was happening with consoles. It was about how many khz their preamps and mixingconsoles let through.
In the end SSL Neve and D&R (among others) created great consoles wich let through up to 100khz and sometimes up to 200khz.

This makes sense because having your -3 dB (or whatever) point at 100Khz or more means a very flat linear response in the audible band.

If you have a lowpass filter to cut down noise in a cd-player it damages some high frequencies because of the lack of steepness of analog filtering.

Non-sense. Only someone that doesn't understand modern converters would say such a thing. The ADC is oversampled and DAC is upsampled so the analogue filter does not have to be so steep. The oversampling can be anything up to 1024 the base rate.

44.1 was a wrongly chosen sampleformat created by Philips.

You can not say that without knowing why that sample rate was chosen. It was most probably an engineering decision and as most engineering decisions, a compromise between many factors. (Btw, the Beethoven 9th or 5th or whatever story is an urban myth).

When you record music with high frequency content (every bang on cymbals will give frequencies up to 50khz (and above but much more weakened). When recording at 44.1. khz. (every sample is half a wavefile so 44.1 can hold up a frequency response of up to 22,5 khz. We loose much of of the high frequency content when loosing those 50khz to 22.1 khz content. We allso loose the lower harmonics of that part of the signal.

In a properly designed system we only lose inaudible frequencies.

And by the way. We are sensitive to phase up to 200khz.

So, much of the localisation content of the signal is lost when recording at 44.1 and 48khz.

This is absolutely NOT true. Read one of my previous posts about timing/phase accuracy in digital audio. It goes way beyond what the human ear can detect even at 44.1 Khz.

This is what the managing director of D&R (Duco de Rijke) had to say about this subject (about analog consoles).
Taken from the D&R website.... I think it is pretty interesting:
--------------------------------------------------------------------------------------------------------------
Did YOU know why analog sounds better than digital?

D&R started long ago (1972) making mixing consoles equipped with tubes.

One of the nicest parts of these hot valves were that they sounded great.

The sound was pleasantly warm and overload was not an issue with over 300 volts on the power rails and a bandwidth of up to 100kHz.

I think he means pleasantly distorted. We moved on from tubes because they were not clean enough.

Then we experimented with transistors and could not get the same sound as we had with tubes.

No indeed, good transistor designs are much cleaner.

Imagine listening to a sine wave of 1kHz, this should be a nice pure round sound like a pure tone of an old Hammond organ. Then switch this 1kHz frequency to a square wave, now you are listening to a sine wave plus on top of it lots of odd harmonics such as 3rd, 5th, 7th and so on .

Actually you are listening to 1 kHz plus on top of this on a very low level frequency 3khz, 5kHz , 7kHz 11khz and so on. So the difference in perceived sound is quite clear to any listener, nothing mystical here.

Now switch the fundamental 1kHz sine wave to let’s say 12kHz. Again a pure very high frequency tone could be noticed.

But now the funny part of it. If this 12kHz sine wave is switched to a square wave we all still can hear the difference between sine and square wave.

This points to either a bad wave generator or the signal is going into a non-linear device causing inter-modulation distortion. EDIT: Or the signals were not level matched for the in-band frequencies. We can NOT hear the harmonics. If we can hear anything, we are hearing artefact within the audible bandwidth.  Mr Neve has designed some great analogue equipment but his comments on digital audio make me wonder if he really understands how it all works. Even some of his remarks and conclusions about analogue equipment are very strange to say the least.

THIS IS AMAZING.

No. This is a flawed test.

If we all agree on this, it means that we can hear or at least notice signals beyond 20kHz.

Remember the first harmonic content above 12kHz is 24kHz , and then 36kHz and so on.

These generated frequencies above 20kHz (even in small amounts) prove that it is important to have an audio system that has a bandwidth well over 20kHz.

Again, no. This is a flawed test. Double blind tests with good wave generators have demonstrated that people can not hear the difference when you filter out stuff above 20 Khz.

It is absolutely a fact that a small lift in frequency response of audio equipment well above 20kHz could easily give you the impression of giving you more air or transparency.

Again if this happens, this is due to changes in the audible band. Probably a phase shift but possibly a saturating (thus non-linear) stage somewhere in the EQ or elsewhere in the signal path.

Can you imagine why most digital consoles today sound harsh and can not even touch the smoothness of well designed analog consoles. This is easy understandable now with frequency responses that sharply fall off at 20kHz. No information above 20kHz means no warm open sound.

This is complete non-sense. If digital consoles sound harsh (which is not true for the good ones) it has nothing to do with the limited bandwidth. On the contrary. It goes even further than that, I heard a story once by someone that was one of the first listeners of a new Neve console. He and the other listeners found it harsh and aggressive sounding. Mr Neve said he could fix that. At the next listening session everyone agreed that it sounded much better, much warmer. When asked what he had changed Mr Neve answered that he had added low-pass filters to every channel.

  Imagine what a digital upper shelving high frequency control is doing when lifting those frequencies by 16dB just before they sharply fall off at 20kHz.

It depends of the design.

I think that digital technology has along way to go before it can even touch analog audio sound wise.

Only if you don't understand it and anyway, what are the alternatives as a storage medium? Tape? Tape is MUCH more coloured than even the cheapest Soundblaster sound card.

Here is the thread about this subject I started a year ago;
http://www.gearslutz.com/board/high-end/128405-hearing-above-20khz-analog-consoles.html

Cool. Will read. :)

More importantly. Bruell & Kjaer did create microphones in the nineties who could capture that high frequency content. Up to that point high frequencies could not be captured in that accuracy. (allthough tape machines record up to 50khz.

Bruell & Kjaer now has a brand called DPA microphones. When you hear those microphones (like the 4006 or the 4090) you soon realise that high frequencies add alot to a mix....

It could be that those microphones just sound different. Or maybe they are more linear in the audible range and what you hear is LESS artefacts from ultrasonic frequencies.

UnderTow

Hear Hear! Also true!

When we use the 64bit mode in either Sonar64 or Sonar32bit we get a 64bit masterbus. When we use digital (convolution) reverb and add this to our recorded audio the extra bits up to 64 may be needed so the calculations of added reverb to the recorded 24bit audio is calculated with the best accuracy. And not rounded off when each piece of audio is hitting severall busses.
 
The TC Electronics system 6000 & Lexicon 960 are fixed point 40bit sharks at 96khz. Not sure why those reputable companies went that high when we all claim we can't hear above a 44.1 khz 24bit samplerate. When there are loads of threads stating those reverbs are tha bomb.
 
At least with the current Cakewalk technique we can match 40 bit fixed point with 64bit float. And match their samplerate....
 
Muziekschuur  

Muziekschuur at home


The TC Electronics system 6000 & Lexicon 960 are fixed point 40bit sharks at 96khz. Not sure why those reputable companies went that high when we all claim we can't hear above a 44.1 khz 24bit samplerate. When there are loads of threads stating those reverbs are tha bomb.

Being able to hear above 20kHz is not the only possible reason for using higher sample rates. As was discussed here many, many pages ago, filter artifacts below 20kHz, aliasing (from non-linear processing), or, in this case, digital interfacing with other gear running at 96kHz are other possible reasons you might do this.

And if those reverbs are still "tha bomb" when used for producing 16bit/44kHz CD's, it isn't because of anything they're doing above 22kHz.

drewfx

batsbrew


g0d d*mn, this thread has a sh!t load of yapping!!!!


LOL

it LOOKED like the discussion was ended - but - damn - its starting again - now with arguments lie "everything is like this AND like that" "everyone is right" and whatever....
 
hahahahah ...

One other thing .....

Having done most of my work at 88.2/24 and a few (usually by accident) at 44.1/24, and FWIW, I can tell you that SONAR is MUCH better behaved for me at 44.1/24 - very few dropouts, glitches, corruption, etc. etc. etc. Whereas, 88.2 projects have been much more touchy. This is even though I'm running a Quad Core with 4G and even though my CPU hit rarely gets above half on any of the processors even at 88.2

Undertow,

I agree with your points. They all are valid. It's just.... I allways wanted a TC6000 and could not afford. Now with my next computer probably being a 4 or 6 core machine with more memory then I would ever need I can process that quality in my pc. So that large setup with outbox this and outbox that is getting less and less nessesary.

I love surround mixing. And for that purpose I like having mono or stereo signals with a clear signal so it will be easier to create a believeable soundstage. The higher samplerate helps. I do work at 48khz. And it is a big difference to 16bit44.1. I have done some projects at 96khz. It will be fun to stay there and just..... don't have to worry about this anymore ...... and have better... (for my purpose) recordings.

I'm happy with less at the compression front. As I have never touched the compression section I do not have an opinion on that. I have heard that the TC Card for pc's slowly is getting the goodies. I allready have a UAD1. So maybe I should get a TC card in my next pc..... with Sonar9 that is....


Cheers Undertow. 

Does it sound better in 96kHz?
Yes & No, all depends, its many factor to consider!


The picture has nothing to do with me or the subject! 



It sound "different" compare to 48kHz due to software oversampling...but is it distortion I hear compare to the sound I hear in the 48kHz? It sound "different"!
As many have already comment, its a possibility you get worse sound in 96kHz compare to 48kHz too. The main thing is is it worth the 3x more CPU usage in 96 kHz? 48kHz sounds perfect too. I can not complain over the 48 kHz sampling quality at all.

All depends what the software has in internal sampling frequency too? Is the software in 44.1 kHz, 48kHz, 96kHz 32bit floating? If its in 48kHz default---> then it will benefit run in 48kHz. Is it in 48--> and then oversampling---> to 96kHz it can well loose its overall high quality it had instead...


Friends have and share the same thoughts & comments. We all do in this thread. 
Some say that you can benefit internal working with software's & plugins “oversampling” in higher frequency like 96kHz. But you will hardly notice  any big difference compare to 48Khz.
Recording vocals and guitars and so on, hardly any benefits compare to 48 kHz 32bit floating... Reverb and FX can fit better in the mix in 96kHz VS 48kHz but if this to due to oversampling I don't know?

 
 

Conclusions!

I was wrong; for now its not worth it---> 3x overall CPU usage in 96kHz!
48 kHz sounds great so I'm back with 48kHz for now. Perhaps in the future when we get even better CPU:s then the i7... then perhaps it worth it.

I always work with huge projects +100 channels so even if my computer can manage 96kHz I want to have the overall “CPU-headroom” to work with. Today it barley reach more then 50 % in 48kHz projects in SONAR 8.5 x64bit. X64bit makes it all run even smoother and I have no limits of use of RAM-memory so that's great too.
Also 48kHz sounds fantastic good and with data-bits of 32bit floating or higher; you have the guarantee that you will not loose any quality working with your software synths and your audio files “internal” in SONAR.

The only thing I will not recommend anyone to use is 44.1 kHz, 16bit, but is another story and I think we all can agree on that...



Best Regards
Freddie

Freddie H


FLac! I think that will soon change too. They will be start streaming in higher format, spotify!

Flac is a great format.

And not encumbered by any royalty payments etc etc 

I have an iRiver media player that plays flac files natively.

Please, I have sincer question! 44.1 kHz VS 48 kHZ?


Can we hear he different in the softwares? I have problem with my Kontakt in 48kHz and up so it seems that I get unfortunately forced to work and use 44.1 kHz again instead...
At least until this get fixed! I feel it doesn't sound as good as 48 kHz though, but I want to hear any good thoughts & comments about this?
http://forum.cakewalk.com/tm.aspx?m=1854900
 

 
I do listen on what you all have to say, you all know that!

Best Regards
Freddie

Well ... it's used in miniDV, digital TV, DVD, Digital Audio Tape.

I believe the original reason was to get enough headroom for simple filters to operate without introducing audible artifacts (read: aliasing). Of course good filters have now come down in price and is not as much of an issue as it was originaly in digital video products.

Dude

Well ... it's used in miniDV, digital TV, DVD, Digital Audio Tape. I believe the original reason was to get enough headroom for simple filters to operate without introducing audible artifacts (read: aliasing). Of course good filters have now come down in price and is not as much of an issue as it was originaly in digital video products. Dude

I know where its used what I don't get is why. CD were out way before DVD or any other video format for disks. I just suspect it is another "we decided to do it this way for incompatibility reasons". The CD format was not developed in Japan thus when DVD was being speced out here was a way to make it different for difference sake. The 24 bit part was good the 48 kHz sample rate seems arbitrary.

Freddie,

I can understand you wanting to record at 192KHz more than I can compehend why you would record in 32 bit.  At least the former gives you more data (even if you can't hear it), but the latter doesn't.  It just makes the files bigger for no reason.  Converters can only go up to 24 bits.  Setting them to record at 32 bit fills the file with silence by padding 0's at the end.  There's no audio quality gain and you're only giving your computer a hard time.  It's a lost case my friend :-)


HTH