96kHz or 192kHz interface?

[hollowsun]

If we had larger hard disks at the beginning, would mp3s exist today?

MP3s came about more to do with internet 'sharing' and getting data down to the bare minimum because, in days of 56k dial-up, a WAV would have been impossible.

But most people found the format acceptable (and it IS for the most part) and it stuck.

Which begs the question....

Why do people sweat and worry over trivia like sample rates, etc., when (as dmills alluded to) there are many more important factors to consider and especially when the end result is more than likely to end up as an over compressed, squashed and 'square wave' master recording then encoded as a 128 kbps MP3 and uploaded to the likes of YouTube where it will be degraded further?!

I am not suggesting for one moment that we shouldn't strive for recording excellence but let's be realistic - recording in your untreated front room with budget mics and monitoring, etc., doesn't warrant much more than the CD standard of 44/16, 44/24 at a push I suppose (even though a fair few bits of those 24 are going to be lost in ambient room noise and D-A conversion) especially as that 44/16 spec is better than most classic albums of yesteryear were recorded with!

It's just arse gravy and there are more important things to be worrying about ... like musical content, for example! I wish more people would attend to that rather than sweating over which sample rate to use because content is all that counts! Ain't got that and, however impeccably recorded, you ain't got nothing!!!

[A Non O Miss]

It's just arse gravy and there are more important things to be worrying about ... like musical content, for example! I wish more people would attend to that rather than sweating over which sample rate to use because content is all that counts! Ain't got that and, however impeccably recorded, you ain't got nothing!!!

word!

[grab]

so should i record everything at 22khz to save disk space?? or doesnt it work that way??? why is there the option to be able to record at 96khz if its pointless??

You only need a high enough sample rate to capture the frequencies you want to record. Claude Shannon proved (60ish years ago) that you'd capture frequencies of up to half the sampling frequency. So for human-audible sound up to 20kHz, you want a sample rate above 40kHz. But much above that, you're recording inaudible frequencies - or perhaps you're not recording anything at all, if your mics don't get there.

The following SoS articles are now available for free, and are superb.

http://www.soundonsound.com/sos/feb08/a ... laudio.htm
http://www.soundonsound.com/sos/sep07/a ... lmyths.htm

[ken long]

No, the entire statement.

DAWs run 32-Bit in general.

Oh dear you are being very pedantic.

It isn't pedantry. It's clarity. You've also taken those quotes out of context.

Why is my CPU statement wrong? If Increase bit depth and samplerate then my CPU works harder. Things are standardised relative to the technology of the time.

OK - One more time then: there was NO CPU to speak of. There was no computer recording, hard disk recording etc at the time the 44.1 standard was chosen. The technology did not exist. That figure came about because of the limitations of storage media and the idea to use video tape.

You continue to suggest that overstretched CPU is the reason that the SR and bit depth values were chosen. That is incorrect. Both technically and historically.

As for 44.1k being a standard because of video, it isn't, it's down to Nyquist's sample rate which comes from telephony.

No it doesn't. I refer you back to my previous postings. (wtf? really?)

I think you are confusing my statements, which are based chronological.

It is you who is confusing both chronological events and technical aspects of the argument. I've offered the only explanation but you don't want to hear it so I think we're done here.

[ef37a]

MM?
I think you have to bow to Ken.
Twas all video tech'.

To give an example of how incredibly technology has changed...
I well remember reading a letter in Wireless World saying how great it would be if high quality audio could be stored without mechanical intervention thus eliminating wow, flutter and noise.

The response of course was that the memory (then) for one hour of 20kHz bandwidth stereo (16 bit no doubt)would be the size of a decent car and you would need the ouput of the local substation to power it.

Never going to happen they said and nobody argued!

Dave.

[Guest]

so should i record everything at 22khz to save disk space?? or doesnt it work that way??? why is there the option to be able to record at 96khz if its pointless??

You only need a high enough sample rate to capture the frequencies you want to record. Claude Shannon proved (60ish years ago) that you'd capture frequencies of up to half the sampling frequency. So for human-audible sound up to 20kHz, you want a sample rate above 40kHz. But much above that, you're recording inaudible frequencies - or perhaps you're not recording anything at all, if your mics don't get there.

The following SoS articles are now available for free, and are superb.

<a href="/sos/feb08/articles/digitalaudio.htm" target="_blank">http://www.soundonsound.com/sos/feb08/a ... dio.htm</a>
<a href="/sos/sep07/articles/digitalmyths.htm" target="_blank">http://www.soundonsound.com/sos/sep07/a ... ths.htm</a>

Yes Grab but the discussion is why does 96k give more clarity than 44.1k? when the problems should be sorted by 44.1k even though the problems of LSB,Jitter and filtering, of the old days, have said to be sorted.

Ok Quantization, samplerate and filtering are different processes but why does the grey area of 44.1k-96k stir up this discussion?

[narcoman]

Yes Grab but the discussion is why does 96k give more clarity than 44.1k?

It doesn't. Not really - moves artefact creating filters further away from the hearable range AND plugins work better - so it seems. But raw capturing clarity? I wouldn't completely go along with that.

[Guest]

OK - One more time then: there was NO CPU to speak of. There was no computer recording, hard disk recording etc at the time the 44.1 standard was chosen. The technology did not exist. That figure came about because of the limitations of storage media and the idea to use video tape.

You continue to suggest that overstretched CPU is the reason that the SR and bit depth values were chosen. That is incorrect. Both technically and historically.

Ok I won't go on, you win!

[Guest]

MM?
I think you have to bow to Ken.
Twas all video tech'.

To give an example of how incredibly technology has changed...
I well remember reading a letter in Wireless World saying how great it would be if high quality audio could be stored without mechanical intervention thus eliminating wow, flutter and noise.

The response of course was that the memory (then) for one hour of 20kHz bandwidth stereo (16 bit no doubt)would be the size of a decent car and you would need the ouput of the local substation to power it.

Never going to happen they said and nobody argued!

Dave.

I have done!

What you say is partly what I'm on about. PCM was just an idea but there was no technology to support it. Now with faster CPUs and more storage space etc we aren't restricted so much.

Remember the first samplers?

[Guest]

It's just arse gravy and there are more important things to be worrying about ... like musical content, for example! I wish more people would attend to that rather than sweating over which sample rate to use because content is all that counts! Ain't got that and, however impeccably recorded, you ain't got nothing!!!

Temper temper!

So which samplerate do you use?

[dmills]

Well, having a higher sample rate using effects and synths in the box makes them have more depth and clarity.

That comes squarely down to piss poor engineering of the software kind.

There are a LOT of complete muppets out there writing (Particularly audio plugin) DSP code without really understanding this stuff. There ARE processes in DSP that can produce harmonics (Anything non linear springs to mind), the sample rate AT THAT POINT must be sufficient to satisfy the sampling theorem, but that has little to nothing to do with the input or output rates.

It is called a multi rate system and is entirely standard in DSP software.

Further there are places where using an excessively high samplerate can cause serious problems, the coefficient precision needed for a low frequency IIR filter increases as the filter cutoff becomes a smaller fraction of the samplerate to give just one example.

Regards, Dan.

[ken long]

PCM was just an idea but there was no technology to support it.

The technology was there. Even at prosumer level.

http://www.palsite.com/pcmf1ovi.html

Now with faster CPUs and more storage space etc we aren't restricted so much.

That's right. And that's why I record mostly at 88.2kHz. It is neatly divisible by 44.1. Storage is cheap. I can hear the difference. I've decided these reasons are enough for me.

[Guest]

Thanks Dan and Ken.

Learn something new all the time here, even though I get slaughtered! Then again I've never been afraid to ask questions.

[hollowsun]

Temper temper!

Not at all

So which samplerate do you use?

I find 44.1kHz perfectly sufficient but if I am feeling a bit gay and carefree, I might use 48kHz.

But that fact of the matter is that aside from (most) mics and amps and monitors being unable to capture/reproduce much above 20kHz (even on a good day with a prevailing wind), most peoples' ears rarely meet the 20Hz-20kHz theoretical spec either and I'd wager most people here can't hear much (if anything) above 18kHz (and considerably less in many cases). There was a time when age was the determining factor but modern music lovers have loud concerts which can seriously damage hearing in extremis whilst young people (in particular) have the tyranny of earbuds relentlessly pounding away on the cilia strapped, as youngsters are, to their iPods listening to full on music with no dynamic range.

And we're discussing recording at 96kHz to give a (theoretical) bandwidth of 48kHz to play on amps and speakers that can't really reproduce anything over 20kHz to a pair of ears that can't hear much above 18kHz best case? Errmmmm!

[Stan]

Apart from the frequency bandwidth advantages, there must be more to these higher sample rates -
How many times you sample a sound per second must affect the detail of the captured sound - more samples more detail. More detail within the frequencies we CAN hear.
I kept my coat on ..

[ef37a]

Apart from the frequency bandwidth advantages, there must be more to these higher sample rates -
How many times you sample a sound per second must affect the detail of the captured sound - more samples more detail. More detail within the frequencies we CAN hear.
I kept my coat on ..

There isn't anymore "detail" TO hear! 18kHz is 18kHz 19.156kHz IS 19.1di da...So long as they are ALL there (and they are) EVERYTHING is there!

Running tape at 30ips gets you up to 30kHzish (all things else being right!) but that does NOT improve any *^%$g DETAIL at 15kHz!(which BTW we were all pretty happy with when the BBC did it right!).

Dave.

[. . . Delete This User . . .]

there is one thing i always find odd....

everyone focuses on upper limit frequency response when talking about higher sample rates....

without considering what it might mean to transient impulse response...

[ken long]

everyone focuses on upper limit frequency response when talking about higher sample rates....

without considering what it might mean to transient impulse response...

This.

[dmills]

Because if a sampler has an upper frequency limit of say 20K, it can accurately capture and reproduce ANY transient you can produce using only frequencies within that bandwidth and maximum level limits!
obviously this assumes competent design and no slew rate or power bandwidth limits (neither are a given).
Given a flat frequency response and constant group delay (granted sometimes not as much of a given as it should be), frequency response and transient response are different ways of seeing the same thing!

It is for this reason that all those brain dead "thought" experiments involving square waves being sampled are so bloody stupid, they assume something that does not exist (a square wave).....

And no, adding 'more dots' does not improve the accuracy, more word length lowers the noise floor (and that is all) and more samples increases the upper frequency response limit, but 3 points on a cycle is enough to completely specify a sine wave, and the limit of bandwidth as being LESS then 1/2 the sample rate ensures there are always at least 3 points....

This stuff is counter intuitive, but then much signals theory is.... Negative frequencies (yes really), all sorts of weirdness.

Regards, Dan.

[Guest]

And no, adding 'more dots' does not improve the accuracy, more word length lowers the noise floor (and that is all) and more samples increases the upper frequency response limit, but 3 points on a cycle is enough to completely specify a sine wave, and the limit of bandwidth as being LESS then 1/2 the sample rate ensures there are always at least 3 points....

Regards, Dan.

Yes but all the diagrams do is show us people that that's all that's going on and that's why we need to come here to clarify things Dan. 3 points needs the underlying mathematical equations to make it mean anything, otherwise it is just three dots. That's what isn't being told properly and that's why people, like me, get confused and misunderstand things.

Thanks

[Guest]

I find 44.1kHz perfectly sufficient but if I am feeling a bit gay and carefree, I might use 48kHz.

So George Michael and Elton John never use 44.1k. Only when they want to sell CDs!

[dmills]

3 points needs the underlying mathematical equations to make it mean anything, otherwise it is just three dots. That's what isn't being told properly and that's why people, like me, get confused and misunderstand things.
Thanks

Unless this forum supports LaTeX (A math typesetting language) in some sane form, I am not getting into trying to write a proof (You really do need to have the correct symbols available!), besides that proof is easy to look up (A level pure maths book, most of them should discuss this), and can probably be done geometrically (but I am NOT going to try it as ascii art!)....

Regards, Dan.

[narcoman]

everyone focuses on upper limit frequency response when talking about higher sample rates....

without considering what it might mean to transient impulse response...

This.

The only transient information one can "hear" is those within our limits anyway. We can feel shock waves below are hearing limit - and I'm sure above. But speakers can't properly reproduce a shock wave - too fast and too loud.

And Dans right - there isn't a such thing as a square wave. Also we only experience impulse responses bound by our own hearing bandwidths. The values we use in digital sampling go along with those natural characteristics.

Dots and samples ?Look up sinc functions and reconstruction filters.

[A Non O Miss]

an expert on this stuff i am not, however, my take on it all is this...

if you are doing classical 96k

if you are doing video 48k

everything else 44.1k

any other cogitating upon it is a little OTT and frankly seems to be nothing more then brain/knowledge flexing... not that having an intrinsic and thorough understanding of it all won't subconsciously help in the end product, but, that alone won't do anything to the end product as far as making it better without the creative content part first being there...

[Folderol]

As Dan said it's all confusingly counter-intuitive, even to those of us that think we've got a bit of a grasp of it!

Another factor that muddies things further, is that when people say they can hear the difference between 48k & 96k sampling rate, what they are most likely hearing is the performance of crap analogue filters. At the higher sample rate you can have a much more gentle slope which means fewer phase and stability compromises.