I've just read SoS's excellent article on recording The Darkness' new album. I noticed, with interest, that once the tracks were recorded using analogue tape, it was edited in Pro Tools at 96kHz.

Now, correct me if I'm wrong, but, there seemed to be no hint of compromise in the production of the album, and yet they didn't use 192kHz...

So, what is the point in 192k?

Is it just case of "Hi-spec for Hi-Spec's sake", to fool naive punters into upgrading their soundcards?

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www.thediplomatz.com

No, this is not the case, but I'm sure someone with more technical know how than me will elaborate...

P

Well, recording using 96kHz will reproduce frequencies of over 40kHz - more than twice the human theoretical hearing range, and well above any frequency which could potentially modulate sounds within our hearing range, so I'd like to hear what the justification of 192kHz is.

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www.thediplomatz.com

This could be a long thread. Personally I've yet to hear any justification for 192k that convinces me, but maybe I've got cloth ears as a (very) few people claim to be able to hear a difference.

Since decent analogue kit will comfortably go up to 192k and way beyond, actually dropping it into PT or any digital domain for editing could even be argued (by some) to be a retrograde step!!

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Dynamite with a laser beam...

True indeed.

My point is, if the Darkness wanted 192kHz, they would have used it. They did not.

If they choose not to use it, then why should the lower-budgeted, hard-disk-space-conscious producers entertain the idea of 192k?

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www.thediplomatz.com

Dan Lavry (who makes very pukka Lavry ADCs & DACs) argues quite strongly that current materials/manufacturing technology/whatever are limiting what can feasibly be achieved with increasing sample rates octave upon octave (as we are).

I think that the current crop of options available (I'm still using 24bit 44.1kHz!!) are 'good enough' for the moment - I feel there's other areas of our game which could do with development & improvement before further increases of sample rate yield significant benefits, even ignoring the possibility they may not be capturing all that they claim...

Mike

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www.ehsound.co.uk - Live Sound Hire & Services

Hugh RJ has written extensively, both on this and the older forums ands in the magazine, on this subject. This is a subect that he has explained with clarity and insight and I suggest to one and all that they read his articles.

Without going into details which are easily looked up in the back issues and on the first and the second forums, good low-res is far better than poor hi-def.

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www.the-byre.com No longer Forum Member

Some claim that phsycho-acoustics come into play at very high frequencies and that even if we don't hear past 20k, humans perceive frequencies well into the 40k and on past 60k.

When I think about this from a more practical point of view, going from say 20k up to 80k is just 2 octaves. Now 2 octaves is now a far stretch for the ears. We can easily perceive beating from tones that are 2 octaves apart but lower in the frequency spectrum. Why wouldn't the same thing apply higher up the scale.

This brings me to another point, get two signals, one at 65k the other at 75k...mix them...you get sum and difference signals (sometimes known as "beating" or "aliasing")...the difference is 10k, well within perceptable range... This is how AM radio works folks, but with a much higher frequency "carrier" signal.

As for why they used 96k instead of 192k...maybe they only had 96k converters available with them at the time.

Let's face it, even 96kHz was completely wasted on that project! They can't've been that bothered about quality if they stuck everything through Protools - though it's about all the "music" deserved.

On a slightly more serious note, the high sample rate issue is about so much more than mere frequency response or just the digital side of things. As The Byre said, it's been discussed at length on these forums; try a search of this forum, the old Infopo (V2) forum and the main SOS site.

aaaah yes, grim, but sum and difference tones ARE all captured within current technologies.

And to counter an earlier point - there may be SOME individuals who (dubiously in my books) claim to hear detail in larger rate samples - however they are the occasional and most think digital radio is good quality....

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Battenburg to the power of 20 - said by Richie Royale in a moment of genius. 4pm. Wed 16th Nov 2011. Remember where you were....

Quote 0VU:

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Let's face it, even 96kHz was completely wasted on that project! They can't've been that bothered about quality if they stuck everything through Protools - though it's about all the "music" deserved.

---

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www.ehsound.co.uk - Live Sound Hire & Services

No-one has convincingly proven that humans can hear much beyond 20kHz (the one Japanese study that claimed to show this has since been discredited).

However, many people claim to hear differences with gear that works at 192kHz. I wouldn't necessarily put this down to the sample rate - there are other factors which may be affecting the signal in the normally audible frequency range.

Cheers

James.

PS - I know that one of 0VU's mastering friends thinks that 384kHz is only just about comparable to analogue!

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JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net

Edited by James Perrett (20/01/06 10:53 AM)

Quote:

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PS - I know that one of 0VU's mastering friends thinks that 384kHz is only just about comparable to analogue!

---

Lol Who've you been talking to?

Several of them do Some feel that it still isn't good enough

We don't necessarily agree about everything

Good analogue still sounds nicer than even good digital though. Not necessarily better, just nicer. Which is fine if you're looking for a nice sound. Sometimes accurate is better - and 384 is pretty accurate - though perhaps not as accurate as 176.4. Or even 88.2. Too many variables.

I'm going back to listening to the tape I'm supposed to be working on next week. Full track mono 1/4" recorded in 1959

a guy came into where i used to work wanting to transfer all his vinyl at 192khz. When i enquired why he wanted to do this he said that "I had to hear the quality to believe it"

I explained to hime that when vinyl is mastered there is a pretty steep roll off above 16khz as anything above that would cause the needle to jump out of the groove on the record. I said that sampling so high would merely highlight the limitations of the source.

This guy was about 65 and probably therefore couldnt hear much above 13khz anyway...

However he still insisted that he could hear the difference. when i showed him something done at 48khz. Where i could not.

Obviosly he had read somwhere that this was the thing to do.

The moral of this story is that if somebody reads something in "what HI-FI" etc they tend to believe it.

In my opinion things like 192khz are used to sell expensive gear to people who are too ignorant of the actual physics of audio to know any better.

just my 2c

Quote:

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The moral of this story is that if somebody reads something in "what HI-FI" etc they tend to believe it.

---

Perhaps. Or perhaps the moral is, you do it and charge him for a high resolution transfer instead of an ordinary one, he goes away happy in his belief and you go away happy with a few extra quid in your pocket and your belief in his stupidity reinforced.

If you get a lot of that kind of work you might want to upgrade your cables too. And perhaps take on an agency for one of the more esoteric cable brands.

Cynical exploitation? Me?

It does seem a shame as let's face it,
some of the Darkness's best stuff is doubtless happening up above 80kHz
I for one would rather hear them up there than their 20 - 20kHz output

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~~~ A weasel hath not such a deal of spleen as you are tossed with! www.feline1.co.uk ~~~

Quote narpin99:

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In my opinion things like 192khz are used to sell expensive gear to people who are too ignorant of the actual physics of audio to know any better.

---

It's certainly true that there's a lot of BS abroad in the hifi world to hook in the gullible.

On the other hand, there are a number of extremely non-gullible professionals who are convinced by it (like Elliot Mazer) simply because they think it sounds better.

I think there's a moral here - whatever the technical arguments for and against, if you test 192KHz and you genuinely think it improves your sound, then go for it. You shouldnt need technical arguments to confirm or contradict what your ears are telling you.

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www.myspace.com/benjunctionbox

Well, my two cents. You simply have to try with your own equipment.

In a perfect world 44.1kHz is perfect up to about 20kHz. And 96 is, well, even more perfect? Not to talk about 192 that should be, well, even more perfect again?

Reality seems to be different. Somehow the limitations in AD and DA converters and processing plugins and such make the sounds different. And most probably, there is an optimum frequency for every combination. Not necessarily the fastest the boxes can go.

What I would expect from real world converters is that with a higher sample rate I would get slightly increased distortion. And as every tape fan knows, distortion can make things sound nicer.

Personally, I go at 44.1 using Lavry Blues. To put it short, the sample rate is not the limiting factor in my recordings. Maybe in 20 years time when my mic positioning is perfect... (I do mostly on location classical music thoung, guess that is less demanding than distorted electrical guitars).

Gunnar

Well, there you have it - SoS forums in a nutshell... We start off with a technical discussion, and end up slagging-off "The Darkness".

On that particular tangent, I quite like 'em. Although, their current album is mastered with the loudness knob set to number 11. Taking their Spinal Tap comparision a step too far, they've definately gone "One Louder".

If you want to discuss the necessity of 192kHz sampling, I guess you need to read Dan's white paper on Sampling Theory first. It's quite sobering:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

B.

Quote feline1:

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It does seem a shame as let's face it,
some of the Darkness's best stuff is doubtless happening up above 80kHz
I for one would rather hear them up there than their 20 - 20kHz output

---

LOL!

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madtheory creations
Synths and pianos for Kontakt

Most people tend to forget what it takes to make 196k ADC's. To make a premium quality 196k ADC, one must have extremely fast sample and hold circuitry. This circuitry makes sampling of square waves of much higher "fidelity". Square waves are the "Primo" source of harmonics.

This is why top quality analog is so much better than ordinary digital. But getting into "extreme" sampling rates enhances the ability to capture that gnarly square wave. Which in turn gets you higher up the fidelity scale.

I for one would love to see 1Mhz sampling rates, but then again you could buy top quality analog and get that.

Quote Steve Hill:

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Since decent analogue kit will comfortably go up to 192k and way beyond...

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Some analogue electronics are wideband and will pass frequencies of 100kHz or more. However, I'm yet to find an analogue tape recorder that can do much better than 50kHz

hugh

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Technical Editor, Sound On Sound

Quote Grimm Reaper Sound:

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Square waves are the "Primo" source of harmonics.

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A square wave is created from a continuous series of odd harmonics at specific relative amplitudes. Great if you like odd harmonics. Bugger all use if you want something a little kinder on the ears

Quote:

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But getting into "extreme" sampling rates enhances the ability to capture that gnarly square wave. Which in turn gets you higher up the fidelity scale.

I for one would love to see 1Mhz sampling rates, but then again you could buy top quality analog and get that.

---

Er... I fear this shows the all-too-common misunderstanding of sampling theory and the inherent bandwidth limitations of analogue equipment.

hugh

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Technical Editor, Sound On Sound

Quote Grimm Reaper Sound:

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Most people tend to forget what it takes to make 196k ADC's. To make a premium quality 196k ADC, one must have extremely fast sample and hold circuitry. This circuitry makes sampling of square waves of much higher "fidelity". Square waves are the "Primo" source of harmonics.

This is why top quality analog is so much better than ordinary digital. But getting into "extreme" sampling rates enhances the ability to capture that gnarly square wave. Which in turn gets you higher up the fidelity scale.

I for one would love to see 1Mhz sampling rates, but then again you could buy top quality analog and get that.

---

You obviously have next to zero knowledge about the basics of sampling. Why don't you scroll a few posts up from yours and read my post and the document linked there first? That would help people avoid reading posts like yours that are full of nonsense rubbish.

I'll quote that message of mine here for your convenience:

Quote Barish:

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If you want to discuss the necessity of 192kHz sampling, I guess you need to read Dan's white paper on Sampling Theory first. It's quite sobering:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

B.

---

IMO unless you have very poor hearing, you will clearly hear the difference between recordings made at 44.1 and 192kHz.
That said, I still very much appreciate the 16bit 44.1kHz CD format and would hate to loose it now that we 'own' it, so to speak.

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.. is this thing on?

Quote Hugh Robjohns:

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A square wave is created from a continuous series of odd harmonics at specific relative amplitudes. Great if you like odd harmonics. Bugger all use if you want something a little kinder on the ears

Quote:

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But getting into "extreme" sampling rates enhances the ability to capture that gnarly square wave. Which in turn gets you higher up the fidelity scale.

---

hugh

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And perfect square waves, with an infinite bandwidth of odd harmonmics don't even exist .. therefore the square wave argument is not valid.

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"Nothing in the world can take the place of persistence. Talent will not; nothing is more common than unsuccessful people with talent."

Quote Barish:

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If you want to discuss the necessity of 192kHz sampling, I guess you need to read Dan's white paper on Sampling Theory first. It's quite sobering:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

B.

---

Thanks for the link Barish, very interesting stuff.

--------------------
Looking for musicians?
www.partysounds.co.uk

There are plenty of ADC's that will do 1MHz - and the editing software that I use will work up to 10MHz but the trick is getting them to work together

Cheers

James.

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JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net

Hello,

This is a question, not an argument

Whenever I read arguments for high sampling rates (> 48k, say) that are based on the high-frequency-content's production of artifacts in the audible range I become confused. Wouldn't the artifactual information be present in the signal if it were recorded at a lower samplr rate, even if the higher frequency components that produced it weren't? Isn't this also why we can hear the artifactual information in the first place?

I have a feeling that there is a simple answer to my confusion that I am just not getting. Enliightment would be much appreciated!

Regards

Quote Stan:

---

IMO unless you have very poor hearing, you will clearly hear the difference between recordings made at 44.1 and 192kHz.
That said, I still very much appreciate the 16bit 44.1kHz CD format and would hate to loose it now that we 'own' it, so to speak.

---

I'm not sure you have read Dan's white paper before making a comment on this.

I'd like to remind you that he is one of the people who design and produce the AD/DA converters so that we can buy and use them in order to make comments as such, not to mention that he also happens to be a musician himself and the converters he designs and manufactures are right up there with the best available on this planet.

Please read the following and then make a comment, which is going to have to be as scientific as his in order to stand. Otherwise, it will have to remain subjective/your personal preference and not something scientifically accurate:

Quote Dan Lavry:

---

There are reports of better sound with higher sampling rates. No doubt, the folks that like the "sound of a 192KHz" converter hear something. Clearly it has nothing to do with more bandwidth: the instruments make next to no 96KHz sound, the microphones don't respond to it, the speakers don't produce it, and the ear can not hear it.


Moreover, we hear some reports about "some of that special quality captured by that 192KHz is retained when down sampling to 44.1KHz. Such reports neglect the fact that a 44.1KHz sampled material can not contain above 22.05KHz of audio.


Some claim that that 192K is closer to the audio tape. That same tape that typically contains "only" 20KHz of audio gets converted to digital by a 192K AD, than stripped out of all possible content above 22KHz (down sample to CD).


“If you hear it, there is something there” is an artistic statement. If you like it and want to use it, go ahead. But whatever you hear is not due to energy above audio. All is contained within the "lower band". It could be certain type of distortions that sound good to you. Can it be that someone made a real good 192KHz device, and even after down sampling it has fewer distortions? Not likely. The same converter architecture can be optimized for slower rates and with more time to process it should be more accurate (less distortions).


The danger here is that people who hear something they like may associate better sound with faster sampling, wider bandwidth, and higher accuracy. This indirectly implies that lower rates are inferior. Whatever one hears on a 192KHz system can be introduced into a 96KHz system, and much of it into lower sampling rates. That includes any distortions associated with 192KHz gear, much of which is due to insufficient time to achieve the level of accuracy of slower sampling.


Conclusion:


There is an inescapable tradeoff between faster sampling on one hand and a loss of accuracy, increased data size and much additional processing requirement on the other hand.
AD converter designers can not generate 20 bits at MHz speeds, yet they often utilize a circuit yielding a few bits at MHz speeds as a step towards making many bits at lower speeds.


The compromise between speed and accuracy is a permanent engineering and scientific reality.


Sampling audio signals at 192KHz is about 3 times faster than the optimal rate. It compromises the accuracy which ends up as audio distortions.


While there is no up side to operation at excessive speeds, there are further disadvantages:


1. The increased speed causes larger amount of data (impacting data storage and data transmission speed requirements).


2. Operating at 192KHz causes a very significant increase in the required processing power, resulting in very costly gear and/or further compromise in audio quality.


The optimal sample rate should be largely based on the required signal bandwidth. Audio industry salesman have been promoting faster than optimal rates. The promotion of such ideas is based on the fallacy that faster rates yield more accuracy and/or more detail. Whether motivated by profit or ignorance, the promoters, leading the industry in the wrong direction, are stating the opposite of what is true.


Sampling Theory Page 26
Copyright Dan Lavry, Lavry Engineering, Inc, 2004

---

barish, thanks for the reference. i started a thread with similar thoughts last summer, and was appalled at the level of misinformation people were spouting like gospel. ("ignorance is not impediment to utterance" a wise man once said). by the time most rockers hit their late 20s, their hearing limit is creeping downn to 10 kHz. read pete townsend's recent warnings to people using ipods for the danger of headphones...

this reminds me of the vinyl purists who disdain cds. i've often wanted to add some surface noise to a cd's output and a/b w/ vinyl to these folks. (before i'm flamed on this, let me add: *if* you have b&w 801s, mark levinson amps, bryston pre-amps, a perfect listening space - oh hell, flame away, most of us *still* won't hear the difference!)

anyway, i suppose we should hope that there are enough technology-philes to buy the latest and greatest, hence keeping the manufacturers going. not least because then the luddites among us (call me ned ludd, to quote the late, great robert calvert) can buy killer product as it's discontinued and sold for pennies...

cheers,

d

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"Patsy had the drug tolerance of Keith Richards and the moral rectitude of Brian Jones." - Dr. Walter Bishop, "Fringe"

Hi Barish, I trust you read Dan's white paper more carefully than my post.
I agree with Dan ''If you like it and want to use it, go ahead.''

Have you tried it yourself yet?

--------------------
.. is this thing on?

Yes. It is different. But different does not necessarily mean "better". Differences are subject to many other conditions in design topology and you'll find that a 192I/O operated at 192kHz will sound different than a 96I/O operated at 192kHz, so will a Rosetta 800 or AD-16X/DA-16X or an RME or what have you. Even different products from same manufacturer sound different. That does not prove a point.

From a scientific point of view, a 192kHz sampling is not better than 96kHz for the audible frequency range in terms of fidelity to the original, and in fact, it is worse, as explained in that paper.

If it sounds better to "you", that's an artistic statement and electronic devices are not designed with artistic decisions, but rather mathematical calculations. We're only making an art using something that was designed with non-artistic values.

B.

This might be one reason why 192kHz sounds so good.

The effect of oversampling and noise shaping on quantisation noise in A/D conversion.
If you expand the range of frequencies that can pass through the system by sampling more quickly, more and more of the noise shaping (dither) can be filtered into higher frequency bands that can not be heard.

Paul Frindle of Sony Professional stated in an interview in 1998 that it was easier to make agood converter at 96kHz than at 48kHz.
I take it from this a 192kHz converter has some advantages for the manufacturer as well.

Edited by Stan (20/01/06 08:39 PM)

Quote Barish:

---

Quote Grimm Reaper Sound:

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Most people tend to forget what it takes to make 196k ADC's. To make a premium quality 196k ADC, one must have extremely fast sample and hold circuitry. This circuitry makes sampling of square waves of much higher "fidelity". Square waves are the "Primo" source of harmonics.

This is why top quality analog is so much better than ordinary digital. But getting into "extreme" sampling rates enhances the ability to capture that gnarly square wave. Which in turn gets you higher up the fidelity scale.

I for one would love to see 1Mhz sampling rates, but then again you could buy top quality analog and get that.

---

You obviously have next to zero knowledge about the basics of sampling. Why don't you scroll a few posts up from yours and read my post and the document linked there first? That would help people avoid reading posts like yours that are full of nonsense rubbish.

I'll quote that message of mine here for your convenience:

Quote Barish:

---

If you want to discuss the necessity of 192kHz sampling, I guess you need to read Dan's white paper on Sampling Theory first. It's quite sobering:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

B.

---

---

Well according to that Dan Lavry,he claims that the human ear can only pick up signals up to 40k and quotes that there is no need for a MHz audio system.PREPOSTEROUS.
Just listen to the very noticable difference in the quality of AM radio(in KHz) and FM(in Mhz).Surely he cant be saying we cant hear radio 1.I have to put up with the bloody thing every morning when my bloody sister wakes up.
Richard Steed
www.soundclick.com/steedie


--------------------
RSteed

Quote Grimm Reaper Sound:

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When I think about this from a more practical point of view, going from say 20k up to 80k is just 2 octaves. Now 2 octaves is now a far stretch for the ears. We can easily perceive beating from tones that are 2 octaves apart but lower in the frequency spectrum. Why wouldn't the same thing apply higher up the scale?

---

I'm not sure I quite understand the technical nature of the "beating" you refer to, but maybe because the notes of the harmonic series get closer together as they get higher?

ie, two octaves above a fundamental note is the distance to its fourth harmonic. Two octaves above that brings you to its sixteenth harmonic, and so on. Thus the perception of difference between high notes is not the same as the perception of difference between low notes.

You can hear this if you play a chord at the top of the piano and then the same chord at the bottom of the piano. Any dissonances (or even thirds, which can be perceived as dissonance down there) or out of tune notes jump out and beat like all hell in the low version, whereas the ear is much more forgiving in the high version, since it picks up sypathetic resonances with the lower strings and appears as a group of overtones higher up the harmonic series (where such dissonances are more natural).

Again I'm not sure this is connected with what you meant. Just pointing out that the way we hear is not a straightforward linear scale up the frequency spectrum. An octave is a much smaller interval, in actual aural terms, down low than it is up high, since it is the distance between one or two harmonics rather than the distance between many.

Quote Richard Steed:

---

yada yada

---

Classic.

Barish, I would check that document you so proudly keep bringing up and look at page 25...The reproduction of the square wave shown is BANDWIDTH LIMITED to 20kHz before hitting the converters...Obviously this works!
Yes his document is technically correct for the most part but some people can perceive a 40kHz filter being turned on in the signal path.

What I meant to say was that without bandwidth limiting and without bringing up data transfer rates for high frequency sampling and the inherent FIR limitations. You cannot reproduce a square wave in the digital domain like you have in analog.

As for the use of this in music, push two oscilator into high frequency (over 20kHz) mix them back together and start playing with that combined signal, you then get an idea of what is available as a difference signal. And yes some analog synths can do this.

So yes using 196kHz can get you better fidelity.

Check before roasting next time

There isnt really much of a significant and apparent difference though between 44.1 khz and 192khz.Obviously,there would be when audio is heard in Mhz.
Stick to 44.1 Khz i say.......'cause I do now-every morning,while I'm having my cornflakes.

--------------------
RSteed

Edited by Richard Steed (20/01/06 09:59 PM)

---

---

Well according to that Dan Lavry,he claims that the human ear can only pick up signals up to 40k and quotes that there is no need for a MHz audio system.PREPOSTEROUS.
Just listen to the very noticable difference in the quality of AM radio(in KHz) and FM(in Mhz).Surely he cant be saying we cant hear radio 1.I have to put up with the bloody thing every morning when my bloody sister wakes up.
Richard Steed

is this a joke?

Considering companies like Sony have released mobile phones with gigs of storage so people can play MP3'S I cant really see any need for more than 16 bit.
How many albumns have many of us got that have been mastered on tape?
I think there is a whole industry making money off the backs off home studio's.
What ever next?

--------------------
Child birth was painfull so dont waste your life being a slave chase those dreams and take a chance on happiness.

Quote Richard Steed:

---

Well according to that Dan Lavry,he claims that the human ear can only pick up signals up to 40k and quotes that there is no need for a MHz audio system.PREPOSTEROUS.
Just listen to the very noticable difference in the quality of AM radio(in KHz) and FM(in Mhz).Surely he cant be saying we cant hear radio 1.I have to put up with the bloody thing every morning when my bloody sister wakes up.
Richard Steed
www.soundclick.com/steedie

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You are kidding right?

What's this? A kindergarten or something? How old are you? What are your electronic qualifications? Could you please tell me the difference between an AM radio signal and FM radio signal in terms of audio quality, apart from the frequency band content? Like for instance, why can AM only carry mono audio information as opposed to FM's stereo/mono compatibility etc?

Let me tell you this: The audio quality difference between AM and FM is not about the amount of carrier frequency, but rather about how the audio content is modulated onto the carrier frequency. Go read about it a bit more and then come back. I've spent 3 years in a technical school just to read that concept from ground up.

Quote Grimm Reaper Sound:

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Barish, I would check that document you so proudly keep bringing up and look at page 25...The reproduction of the square wave shown is BANDWIDTH LIMITED to 20kHz before hitting the converters...Obviously this works!
Yes his document is technically correct for the most part but some people can perceive a 40kHz filter being turned on in the signal path.

What I meant to say was that without bandwidth limiting and without bringing up data transfer rates for high frequency sampling and the inherent FIR limitations. You cannot reproduce a square wave in the digital domain like you have in analog.

As for the use of this in music, push two oscilator into high frequency (over 20kHz) mix them back together and start playing with that combined signal, you then get an idea of what is available as a difference signal. And yes some analog synths can do this.

So yes using 196kHz can get you better fidelity.

Check before roasting next time

---

Some people? Who for example? Under what tests?

Well, I am Jesus Christ then. Worship me. Proof? Well, I know I am.

We're talking something scientific here, dude. Not hear-say. "Some people can perceive the changes in 40kHz." Yeah sure.

Even if we accepted that "some" people do perceive as high as 40kHz of audio, rather "super-audio" to us mortals, 80kHz sampling rate would sort that out with no problems.

I'm still not sure you're getting the point. The guy is saying "96kHz is more than enough. No need for 192".


Yours is like saying "let's capture ultraviolet and infrared light when filming as well, cos some people said that they didn't see them but perceived them."


That's nonsense.


This subject has been caned to death in its all entirety at Dan's forum at R/E/P for months, with contributions from many other manufacturer/mastering engineer people. I'm not going to waste any more time trying to explain it all over again here. Sorry. If you are so convinced then go buy the next 384kHz converters when they are out. Or if you are interested to know more about it, you can check it in Dan's own forum. He also has a forum at his website http://www.lavryengineering.com but that's totally Lavry product range related. R/E/P forum is general technical discussion.

B.

In fact, below thread could be a good starter for you:

http://recforums.prosoundweb.com/index.php/t/2997/0

and this "high frequency transients fallacy":

http://recforums.prosoundweb.com/index.php/t/4097/0


and this "EQ for 192kHz sampling":

http://recforums.prosoundweb.com/index.php/t/2666/0




B.

I see and agree - most of the advantages of a higher sampler rate would indeed appear to be covered by sampling at 96kHz.
I'm also gathering that these sonic advantages are more to do with the converters and their 'ways' than the increased frequency range.

If I ever get the opportunity to record a great artists performance, I would like to be able to play safe and record it at 192kHz.
No harm in that is there?

--------------------
.. is this thing on?

Quote Stan:

---

If I ever get the opportunity to record a great artists performance, I would like to be able to play safe and record it at 192kHz.
No harm in that is there?

---

Only that you need to have invested in 192kHz converters to do it.

Then, whatever medium you're recording to, you need over four times as much storage space as you would to record at 44.1khz.

Then you need massively more CPU power for the same number of simultaneous tracks, and to consider whether you're pushing the stability of your system to the limit and thereby compromising the security of the session. Or if you're using a non-computer hard disk recorder, you need one with the spec and power to do that many tracks at 192khz.

Weighing all this up, you may find that it costs many, many times more money to be set up to record the session, and do so effectively, at 192 khz as it does at 44.1 khz. If the only person who can hear the difference is your dog, then that's the harm: you've spent a load of money and got no discernable improvement for it.

I'm not saying there's no difference - I don't have sufficient experience of higher sample rates to have settled on an opinion. All I'm saying is that IF there's no difference, then there's no point spending the money.

Good point Wurlitzer. I dont think I'd try it without a test flight.
My CPU is an AMD Athlon 2.8 so 192kHz mulititracking is not on the cards for me, yet.
Processors will get faster.
I can get impressive results stereo sampling at 192kHz with my E-mu 1212m.

--------------------
.. is this thing on?

barish, take heart and know you fought the good fight. there are some people who will not be convined no matter what technical details you present.

when i last had this argument a few months ago with individuals who i'll leave unnamed, i got so disgusted with their ignorance that i went on the web and found a whole set of college lecture notes available online concerning electrical and communications engineering, from M.I.T. (Massechusetts Institute of Technology, #1 US university in science and engineering.) these notes covered all the stuff in the white paper you cited, but in much more detail, startingn with the nyquist sampling theorem, etc. i was going to post the urls on the particular forum.

then i remembered what a friend had taught me many years ago: you can lead a whore to culture but you can't make her think. so i dropped the matter. i only chimed in on your thread cause i wanted to show solidarity, not in any hope of educating the yahoos (read gulliver's travels for the original reference). that and do what i do: buy up end-of-life kit that *only* does 44.1/16. thanks to the punters who buy the latest gear, i can equip a pentium 3 pc with a pro audio sound for pennies. in my home studio i've got 7 machines linked by adat optical, running reason and assorted vst instruments, all connected by fast ethernet switching fabric. total cost, pcs, soundcards,etc? under 500 pounds.

btw, i dig kebabs!

cheers,

d

--------------------
"Patsy had the drug tolerance of Keith Richards and the moral rectitude of Brian Jones." - Dr. Walter Bishop, "Fringe"

Here is a direct copy from the SOS July 1998 issue.
CD: The Next Generation: Super Audio Compact Disc

"
The Sony/Philips proposed hybrid disc format does not store high-resolution audio data in the conventional multi-bit Pulse Code Modulation form (PCM) used in current digital recording systems. Instead it uses a process known as Direct Stream Digital (DSD), upon which Sony have been working for some time, as a new recording, mastering and archiving format.

DSD is claimed to provide an audio bandwidth between DC and 100kHz and a realisable dynamic range well in excess of 120dB (the signal-to-noise ratio is specified as better than -120dBFS at 20kHz and the equivalent audio resolution better than 24 bits). Sony and Philips have been fine-tuning the system over recent months through extensive listening and comparative sessions held around the world, with numerous artists, producers, recording and mastering engineers, and even audiophile consumers, taking part.
.
.
Current DSD systems sample audio at 2.8224 MHz (that is, 64 x 44.1kHz) and the resulting 1-bit data stream is obviously pretty big. However, it is actually 'only' four times bigger than that of a conventional 16-bit/44.1kHz PCM signal, and so is well within the capabilities of many current tape and disk recording systems -- Sony are using PCM800 (DTRS format) machines with a custom interface for experimental DSD work in the UK, and in America commercial stereo DSD recordings are being made on hard disk-based recorders (Sonic Solutions manufacture a compatible system, for example).
"

Quote Grimm Reaper Sound:

---

.

So yes using 196kHz can get you better fidelity.

---

What kind of speakers would I need to get the full
benefit from this increased fidelity?

--------------------
Phatcat Studios--"I love the smell of Genelecs in the morning"

Quote Grimm Reaper Sound:

---

Here is a direct copy from the SOS July 1998 issue.
CD: The Next Generation: Super Audio Compact Disc

"
The Sony/Philips proposed hybrid disc format does not store high-resolution audio data in the conventional multi-bit Pulse Code Modulation form (PCM) used in current digital recording systems. Instead it uses a process known as Direct Stream Digital (DSD), upon which Sony have been working for some time, as a new recording, mastering and archiving format.

DSD is claimed to provide an audio bandwidth between DC and 100kHz and a realisable dynamic range well in excess of 120dB (the signal-to-noise ratio is specified as better than -120dBFS at 20kHz and the equivalent audio resolution better than 24 bits). Sony and Philips have been fine-tuning the system over recent months through extensive listening and comparative sessions held around the world, with numerous artists, producers, recording and mastering engineers, and even audiophile consumers, taking part.
.
.
Current DSD systems sample audio at 2.8224 MHz (that is, 64 x 44.1kHz) and the resulting 1-bit data stream is obviously pretty big. However, it is actually 'only' four times bigger than that of a conventional 16-bit/44.1kHz PCM signal, and so is well within the capabilities of many current tape and disk recording systems -- Sony are using PCM800 (DTRS format) machines with a custom interface for experimental DSD work in the UK, and in America commercial stereo DSD recordings are being made on hard disk-based recorders (Sonic Solutions manufacture a compatible system, for example).
"

---

That text talks about a technology when it was a baby and everything mentioned there were yet to be proven, and it is based on a press release by the developers 8 years ago. EIGHT YEARS AGO. You can not find many people on these boards who still use a professional digital audio product that they bought eight years ago. As a non-native speaker of English language, I can even see that when I read it and I am deeply disappointed that you could not see what I see in it, yet you still try to come up to the surface with that. And for your information, everybody knows that SACD technology has been thrown out in the bath water a while ago. It was destined to be defunct, only a few companies tried to support it and it failed to impress the masses anyway. If you were not so ignorantly and reactionistly stubborn against the scientific facts, you would have read those thread links that I had given up there and seen the part where Mr Lavry says:

http://recforums.prosoundweb.com/index.php/mv/msg/2997/37350/0

We all based our opinions on what was available at that time and as the time moves on we all develop, experience and progress in what we know. That's how the science improves.

Let it go.

B.

Since when SACD has died a death, and Sony have pledged $millions to compensate users who they arbitrarily chose to infect with embedded viruses!

192k is bound to have its defenders - lots of manufacturers have a massive vested interest in taking it up to the limit. The reality is I can't hear a difference.

I don't even bother to tell clients what I use (I can switch to quite a few things), I just ask if they like the sound. If they are happy at 44.1k, so am I. Far less strain on CPU and very respectable results for most (not all...) genres.

--------------------
Dynamite with a laser beam...

Dubbmann thank you for understanding.

Regards,

B.

Mate, Ive read a lot of your posts. How do you sleep at night being so f*cking perfect?

Nobody's mentioned that we still have to live with 24 Hz when we go watch a movie at the cinema. 24!!! This is outrageous !

Never mind kilo hetz - we are talking hertz here!

That's such a low resolution that it should have died before the stoneage ! It is not fair !

What are we complaining about? This needs to be rectified!

Immediately!



(PS - I really hope that was funny because it took me ages to get all those italics and bolds right - phew!)
ML

Quote ow:

---

Mate, Ive read a lot of your posts. How do you sleep at night being so f*cking perfect?

---

I am actually not. I am a perfectionist right enough, which is not a bad thing IMO, but I am not perfect. It's just that I only talk about things that I am sure I know right. Otherwise I shut up and read on (or listen up, or keep working). That's why you may think I know everything. There's always room for improvement for every one of us.


Oh, look, it's half twelve in the morning and I already feel sleepy. Nite nite





B.

Nite Barish

Sweet dreams Barish, I'd just like to add , it does not take a bat or a dog to hear the difference sampling at 192kHz.
Dont knock it 'till you try it.
Long live science and long live art.

--------------------
.. is this thing on?

I thought the reason why an analog system that could go to into 100k+ frequency response, sounds better is that they have the ability to critically track the wave form. This comes from there “high slew rates", not to be confused with the 20-20k audible range. So it also would to work with high sampling rates.
“Less digital fatigue”

I think it might be best just to retire to our respective corners on this one. No one is likely to be convinced of the counter arguments. Minds are closed and beliefs are two deeply held.

For the record, I'm of the opinion that 44.1 or 48kHz should be sufficient, but the practicalities of implementing a suitable design make it difficult to get right at budget prices. Pay big money for something at the real cutting edge, like a Prism converter, and you'll soon realise that there is nothing much wrong with 44.1 or 48kHz sampling.

For the rest of us, moving up to 96kHz is a very cost effective and practical workaround. Moving the cut-off point up another octave relaxes the constraints sufficiently that it becomes possible to get an excellent sound for the kind of budgets that are reasonable for hobbyists.

Moving it up again to 192kHz, in my opinion, can start to become counter-productive. Not just because of the data storage and processor overheads, but also because it starts to put a lot more strain on the design of some specific aspects of the converter. The result is that you have to go back to the top flight designers and pay a lot more more money to get something that works as intended.

But this is a very complex subject indeed, and there are a great many compromises and trade-offs involved -- most of which aren't obvious to the non-specialist.

If you happen to like the sound of manufacturer A's converter at 192kHz over manufacturer B's converter at 44.1, then great. It's a subjective choice and you are free to choose whatever works for your ears.

However, you cannot infer from that selection anything truly objective about the merits of 192 over 44.1, because there are far too many variables involved.

We all know how different analogue circuitry or circuit topologies can sound subtly (or even blatently) different. A lot of the sonic differences between otherwise similar converters is purely down to the design and layout of the analogue circuitry and the nature of the power supply system(s).

In the case above, Manufacturer A's converter would probably still sound different to manufacturer B's product at the same sample rate, simply becaue of the different analogue circuitry involved. We could then argue about which one was better -- the more technically accurate, or the one that sounded more 'analogue'... but we still wouldn't reach a concensus.

Objectivity doesn't get much easier even if you compare two converters from the same manufacturer, operating at different sample rates. Let's say unit A operating at 44.1 and unit B at 192. Same analogue electronics this time, but completely different decimation filters involved with different slopes, ripples and phase responses. The sonic differences here could easily be down to the choice of different decimation filters as to the sample rate itself.

In fact, some high end converter manufacturers actually incorporate four or five user-selectable filter characteristics, and switching between these without changing the sample rate at all produces distinct sonic differences easily as great as comparing two identical converters operating at different sample rates!

So come on, let's stop the pointless arguing. The theory is plain and incontrovertible. 44.1 should be enough, but practical construction constraints tends to let the theory down. 96kHz provides a reasonably convenient workaround. If done properly, 192 shouldn't be any worse, but doesn't really offer any advantages -- and if not done properly it can actually be less accurate than 96!

Of course, some people might like the less accurate version because it sounds 'more analogue'... and round we go again...

Oh, and film is shot with a sample rate of at 24 frames a second but the film projector oversamples the output by a factor of two or three times by using a rotating shutter.

Funny old world, isn't it!

hugh

--------------------
Technical Editor, Sound On Sound

Digitising audio at 44100 samples a second means that, according to Nyquist's theorem, effectively record frequencies up to 22050 Hz. However, the higher the frequency, the less the detail - a 200 Hz sine wave will be far more accurately represented than a 13,000 Hz sine wave, and thus will sound much better.

For instance:

1Hz: 44100 / 1 = 44100 samples
30Hz: 44100 / 30 = 1470 samples
300Hz: 44100 / 300 = 147 samples
3000Hz: 44100 / 3000 = 14.7 samples
13000Hz: 44100 / 13000 = 3.39 samples

A 13000Hz sine recorded at 16-bit, 44100 Hz looks like this:



So, as you can see, high frequencies are not represented very well at 44100 Hz. Raising the bar to 96000 Hz, on the other hand, allows us, in the case of 13000 Hz, to have 7.38 samples per second, which is over double the accuracy of 44100:



We can see that the waveform is starting to approximate to a sine wave a lot clearly now. Not perfect, by any means, but this does show that even though a sampling rate of 44.1 Khz is adequate for representing frequencies over the human hearing threshold, high frequencies are represented horribly.

No no no no no no no no no no no no I definitely won't get involved.

--------------------
Yes, good, very good, but everything LOUDER!

Magic Window, it isn't that simple. Any sinewave accurately may be reconstucted using only two samples.

Search the forum and main SOS site for some of the many posts and explanatory articles about how digital audio works. Or try getting hold of a copy of "The Art of Digital Audio" by John Watkinson (or pretty much any of his other books on the subject - they're all much the same) and find out about how the system works to produce an audio waveform from the stored data.

Quote Magic Window:

---

1Hz: 44100 / 1 = 44100 samples
30Hz: 44100 / 30 = 1470 samples
300Hz: 44100 / 300 = 147 samples
3000Hz: 44100 / 3000 = 14.7 samples
13000Hz: 44100 / 13000 = 3.39 samples

---

Rubbish. You only need 2 sample points for a sine wave.

Barish, there is an interesting discussion going on on the Pro Audio mailinglist about inter-sample peaks above 0 db FS. One comment has been made that increasing sample rates might help for avoiding inter-sample peaks that clip in the DCA.

Stan, you converters might sound better at 192Khz but that could just mean that your converters are broken at 44.1Khz... (without getting into how good humans are at convincing themselves of things that really don't exist). Things really are not as simple as they may seem.

UnderTow

MagicWindow,

Do you know what "upsampling" and "downsampling" are and the concepts behind them?

Look, I have a life here. I am 35 and the time I have left for making music is getting less and less in my life. I have shown you the way to a specialist "designer/manufacturer" forum. Not a Pro audio power user forum. Go post your simple wireframe hangman diagrams there and see the technical roasting you are getting.

I can't get into an argument with people who approach to sampling concept with a four-operation calculator arithmetic. It won't get me anywhere.

That's as far as my contribution to this thread goes.


Ah, one last thing, for the chap who talks about generating square wave in synthesizer as part of the music:

In music, or any analog audio related concept for that matter, square wave means clipping. It is the menace that blows your tweeter, bursts your woofer, burns your amplifier's output stage and hurts your ears like a chinese torture. Why would you want to sample a squarewave in the first place? To convert it into a.... err... square wave?


Why don't you just connect the SPDIF output to your power amplifier and leave the converter alone?


That's all from me fellas.

B.

Nice to see the 'join the dots' misunderstanding rearing its ugly head again. How I have missed it.

--------------------
Tesco Value Tonmeister

To be fair to Magic Window, they weren't his dots, they were screen grabs from Audacity (Magic Window correct me if I'm wrong). Now that I've read Dan Lavry's article, I might invest in a more expensive audio editor

Thank you to the knowledgeable people who've shared their expertise in this thread; it's very educative for non-mathematicians like me. My one reservation about Dan Lavry's article was that I hope his maths is more accurate than his English. Confusing "whether" and "weather" might not make any difference to his argument, but if his summing graphs contain the same sort of elementary errors, then the overall picture might not be as he says it is.

Quote Ivories:

---

Confusing "whether" and "weather" might not make any difference to his argument, but if his summing graphs contain the same sort of elementary errors, then the overall picture might not be as he says it is.

---

Well, with that spelling the man designed, made and sold some of the most appraised converters in the world today so he must know a bit or two about these things, don't you think?

Plus why would one want to piss in the reverse direction when everyone else has already jumped on the bandwagon in this 192kHz hype?

Let me quote Matthias Carstens of RME from an email (not directly to me, but to a distributor friend of mine who forwarded it to me), on Lavry's paper, before I go:

Quote Matthias Carstens:

---

"No surprise.

Nobody wants to hear the real truth. And if you tell someone the real truth, he won't believe you. I know this document, it's not new. The doc has been discussed all over the web. I think the basic information is correct.

Of course there is always a 'but if then'."

---

A food for thought.


Nite nite.

B.

Barish, I owe you an apology for my rediculous 'perfect' outburst the other night. Dont know what came over me. I dont even know you man... i'm so sorry... sincerely... T

Hugh Robjohns, not for the first time, I really enjoyed your reply to this thread. It may be old hat to some but for me digital audio technology is fascinating and totally baffling at times. I love it.
Over the years I have noticed an SOS stand on 192kHz. I would call it diplomatic. It's why I trust you guys.
Salutations from a non-specialist.

--------------------
.. is this thing on?

Quote 0VU:

---

Magic Window, it isn't that simple. Any sinewave accurately may be reconstucted using only two samples.

---

What about a non sine wave?

Still makes no sense.

If you had two samples to represent a sine wave, surely the best you could do is a discontinous leap from one value to another? A square wave?

Quote The Bob Campbell:

---

Quote 0VU:

---

Magic Window, it isn't that simple. Any sinewave accurately may be reconstucted using only two samples.

---

What about a non sine wave?

---

Every other wave can be described as a combination of sine waves. In other words, there are only sine waves.

UnderTow

Quote Magic Window:

---

Still makes no sense.

If you had two samples to represent a sine wave, surely the best you could do is a discontinous leap from one value to another? A square wave?

---

This is where people usually make the conceptual mistake: Digital audio isn't sound. It is an encoded signal _representing_ sound waves. To retrieve the encoded signal you need a decoder. In this case a Digital to Analogue Converter (DAC). Part of the decoder is a (lowpass) reconstruction filter that removes all the extra harmonics.

What happens when you filter out all the extra harmonics of a (theoretical) square wave? Yes, thats right: A perfect sine wave.

UnderTow

Quote Barish:

---

Well, with that spelling the man designed, made and sold some of the most appraised converters in the world today so he must know a bit or two about these things, don't you think?
B.

---

I'm quite prepared to accept your opinion of his convertors (since they're way out of my price range, I can't try them for myself). However, since you asked for a scientific debate, I don't see how one person loving the sound of his 96 kHz convertor is any more admissible as evidence than another person loving another manufacturer's 192 kHz product. If Lavry's paper had been published in a peer-reviewed scientific journal, then non-scientists like me would probably be safe to assume that the technical content was beyond reproach; since he's simply published it on his own website, I think we're entitled to look at its obvious flaws and wonder whether they extend to the maths as well. That's why I like reading the arguments between you experts on the SOS forum, since you're more competent to judge that than I am .

Magic window, you seem to be in the same position mathematically as me, so I'll have a go at explaining the sine wave reconstruction bit: a DA convertor doesn't use straight lines join together the points on the graph made by the samples in the digitized waveform. The point of all the maths in the article Barish referred to was to prove what shape the lines ought to be. If you get the maths right, then the waveform will be reconstructed accurately, provided the sample rate is at least double the highest frequency present in the original waveform.

Quote Magic Window:

---

Still makes no sense.

If you had two samples to represent a sine wave, surely the best you could do is a discontinous leap from one value to another? A square wave?

---

Hint: Open the book called "Calculus and Analytic Geometry" and study the sections about Derivatives and Integrals.

That'll give you the basic idea how mathematical functions are derived into simpler forms via Derivation and then reconstructed back to their original states by Integration.

That's why I'm saying I can't waste time arguing with a logic that approaches to sampling concept with four basic mathematical operation (addition, subtraction, multiplication and division) mentality. You won't get anywhere with that logic. You can't design or explain an AD/DA converter with that level of mathematics either.

B.

Quote UnderTow:

---

Quote The Bob Campbell:

---

Quote 0VU:

---

Magic Window, it isn't that simple. Any sinewave accurately may be reconstucted using only two samples.

---

What about a non sine wave?

---

Every other wave can be described as a combination of sine waves. In other words, there are only sine waves.

UnderTow

---

What happens when the wave you are describing with a combination of sine waves happens to be right at the upper limit of your sample rate? e.g. The theory shows that a 22.050khz sine wave can be correctly sampled and reproduced as long as the sampling rate is 44.1khz. If you take a theoretically perfect sine wave at either 1hz or 22.050khz normal sampling theory proves that a 44.1khz sampling system can reproduce it.

The difficulty I have with this is that waves are not perfect.

For example a soft bass guitar note may have a strong sine wave component (if we can effectively call it a sine wave) but it will contain many higher frequency harmonics and imperfections. A real world sine wave at 22.050 can also contain those higher harmonics and imperfections, which in reality would be components of frequencies higher that the 22.050. A bandpass filter in the converter will remove these minutae and therefore they are not passed to the sampling process.

For music's sake, this is not an issue, and the A-D is doing it's job correctly.

I would simply point out that it is not wholly accurate, you are employing a level of averaging to contain the information being fed to the digital process below a limit. If you wanted to be truly accurate, you would have to increase the sampling rate several times, even though you are effectively recording information that no-one can hear.

The question that should then be asked - is that extra information useless, or in a complex piece of music does it somehow add up to something that should be there?

Bob

p.s. if you think that information above the frequency range of human hearing does not need to be captured, think about this, if I take a 1khz sample, then combine it with another of equal volume but 1.01khz, I can create very audible "beats" that manifest themselves as a lower frequency. What if the two adjacent frequencies creating this lower frequency beat effect are not within the permitted frequency range, but lie outside it? Is a "human audible" sub-22050khz beat effect created by higher frequencies being lost when we filter out everything above 22050?

Quote Ivories:

---

If Lavry's paper had been published in a peer-reviewed scientific journal, then non-scientists like me would probably be safe to assume that the technical content was beyond reproach; since he's simply published it on his own website, I think we're entitled to look at its obvious flaws and wonder whether they extend to the maths as well.

---

Only if you knew that in the forum the I directed you to, all his peers, including Herr Zeiss from Switzerland are debating on his paper for months.

I also remind you to read my quote from Herr Cartens of RME Audio about the paper. Of course, if you think he is a peer of Lavry's.


Too much argument with too little knowledge, that's what I see in this thread. That's why you are all running round in circles with "what if it's not sine wave and this wave, that wave."

You need to get basic sampling and mathematical concepts right first before getting into arguing the differences between waveforms. There are many sources where you can get information about these concepts and I don't want to look like Dan Lavry's champion here, but they are also available in Lavry's website in concise papers. Just read:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

http://www.lavryengineering.com/white_papers/sample.pdf

http://www.lavryengineering.com/white_papers/dnf.pdf

http://www.lavryengineering.com/white_papers/fir.pdf

http://www.lavryengineering.com/white_papers/iir.pdf

http://www.lavryengineering.com/white_papers/jitter.pdf


Cheers.

B.

Thanks for not being a hostile, bitter old bastard like our friend Barish over there. The calculus is irrelevant for one's understanding of how digital audio works on a basic level.

I understand it now, I think - samples are 'snapshots' of amplitude levels, but because speaker cones cannot move from one point to another instantaneously, you get a non-linear movement that "fills in" the sine wave even though there are only two sample points, right?

Quote Barish:

---

Too much argument with too little knowledge, that's what I see in this thread. That's why you are all running round in circles with "what if it's not sine wave and this wave, that wave."

---

That's why it is always refreshing to receive messages from the pro-audio mailing list. At least you know that the discussions there are about things that will improve our understanding of audio rather than endless discussion of basic sampling theory.

Dan Lavry spent a fair bit of time discussing his ideas on the pro-audio list before giving them a wider public airing. Unfortunately it looks like the discussions are no longer in the list archives but, if you are really interested in the finer points of audio then take a look at http://www.pgm.com/mailman/listinfo/proaudio

Cheers

James.

--------------------
JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net

Quote Ivories:

---

To be fair to Magic Window, they weren't his dots, they were screen grabs from Audacity (Magic Window correct me if I'm wrong). Now that I've read Dan Lavry's article, I might invest in a more expensive audio editor

---

That's what I mean about the misunderstanding - these screen grabs seem to give a very misleading message, since they neither show a stream of samples (as quantised steps) or a reconstructed waveform. I wasn't implying that MagicWindow had drawn himself some dots and joined them together!

Quote Magic Window:

---

I understand it now, I think - samples are 'snapshots' of amplitude levels, but because speaker cones cannot move from one point to another instantaneously, you get a non-linear movement that "fills in" the sine wave even though there are only two sample points, right?

---

Not quite - you're right about the samples being snapshots, but the reconstruction of the waveform from the samples happens within the DAC, way before it gets to any speakers. In a reconstruction filter, in fact. It really does require some nasty maths to describe it properly, I'm afraid.

Quote UnderTow:

---

Every other wave can be described as a combination of sine waves. In other words, there are only sine waves.

---

AHA! I've been doing some reading since this post and apparantly the above statement I made is incorrect! (or incomplete).

Here is a quote from Dan Lavry on the Prosoundweb forum:

Quote:

---

...music is not all sine waves, and music can not be "taken apart" to sine waves.

Fourier found that any PERIODIC wave (endless repetitive identical cycles) can be "broken down" into "basic elements" - sine and cosine wave, all at frequencies that relate to the basic pitch by somle integers.

But much of music is not made of PERIODIC waves. The attack, the decay of a note (and much more) are NON PERIODIC. The "near periodic" part of a note is that duration after trhe attack and before the decay, where some instruments (like a pipe organ) "stays steady for a while. That "steady portion" can be looked at as "almost Fourier like.

But as a rule, music is not the sum of sine waves.

Yet Sine waves are great tools for testing part of the system behaviour (not all).

And while at it, if one wishes to "break" a signal up into "basic elements", it can be done as follows:

A. Define a proper impulse response wave:
Take the maximum signal bandwidth. That bandwidth corresponds to some impulse response (the more bandwidth the narrower the impulse). The shape of the impulse is the sinc function (sine X/X).

B. sampling:
Now sample the signal at a rate slightly greater then twice the bandwidth.

C. At each sample time, insert the impulse wave (see A. above), and make the each inserted impulse amplitude equal to the corresponding sample value.

D. Add all the impulses from C. That is it! the "elements" (properly scaled impulse waves placed at all sample times) will add up to your original wave (which can be an hour of Beethoven symphony, or the sound of the wind or whatever... within the agreed original bandwidth.

---

Sorry for confusing things.

PS: Anyway, Barish didn't comment on this so either he isn't paying attention or his knowledge isn't as complete as he is claiming.

PPS: It is a bad teacher that blames his students for not understanding the lesson ... I'm sure you can figure what this comment is directed at.

UnderTow

Edited by UnderTow (25/01/06 06:10 PM)

I'd forgotten why I quit posting, now it's come back to me, people have a habit of not reading (or responding to) what is typed....

Bob

Yes, that's my mistake. I should have let this thread go after my first post.

Other folks, call me any name you want. It's not my money you'll be spending after all. Why should I care.

Get what makes you happy.


B.

Quote Barish:

---

Yes, that's my mistake. I should have let this thread go after my first post.

Other folks, call me any name you want. It's not my money you'll be spending after all. Why should I care.

Get what makes you happy.


B.

---

I have been tempted to post on this thread about half a dozen times, but I have resisted. I don't want to get into arguements or putting people down.

Lets face it, the highest quality medium (widely) available to the consumer is 44.1khz 16bit CD. With the loudness war, the quality of most commercial material is awful, and most people are now listening to low res MP3. I always advise people to work with 44.1, using the highest quality equipment they can afford. There is no substitute for good Mikes, Preamps, and ADC's. Also a good set of monitors, and avoiding the over use of cheap digital plugins.

The quality of your equipment, and especially your ability to use it are going to have a far bigger impact than the sample rate.

Dave

--------------------
Lighthouse Mastering

Quote Magic Window:

---

Thanks for not being a hostile, bitter old bastard like our friend Barish over there.

---

I think Barish has more knowledge in his little finger than you, my friend, are likely to acquire in several lifetimes based on the slender evidence of your very few posts on this forum to date.

However, if you wish to continue to make a complete prat of yourself, go ahead. I'm sure you will be indulged. For a while.

--------------------
Dynamite with a laser beam...

Quote:

---

What is the point in 192kHz?

---

it's the decimal . tells you it's 192000.00 Hz. Not 192.00000 Hz or 1920.0000 or 19200.000 Hz

.




I don't consider there to be any real world advantage in actually working at 192KHz.

HOWEVER

I DO consider there to be an advantage in the 192KHz capable converters and so on....



Bluntly, the Digidesign 192 interface is the best sounding AD/DA thing they have EVER made....

EVEN WHEN WORKING AT 44.1KHz

and the same applies to the MOTU 192HD

( and their 896HD)


The same applies to the Apogee Rosettas, Lavry, DCS, Benchmark and a whole heap of other stuff ..

all these PROPER High Rate capable high end converters also sound better than their predecessors did.... even when working at more mundane and sensible sample rates.,

IMHO there is a worthwhile improvement to be had out of 96KHz sample rate for serious work, but 192 as a working rate is not only pointless , but self defeating... But in general 44.1/48 KHz is largely sufficient.

My (educated) guess is that the knock on benefit of the manufacturers tooling up to work at these elevated rates, is that we get better clocks and converters to continue working at the rates we feel most comfortable with in the balance between ultimate fidelity and practical usefulness.

--------------------
if you don't know who i am, i aint gonna tell you.

I'm intrigued by all this. So ive tuned to 192Khz long wave of course. And ive got a very interesting French phone in show. Dont know what theyre saying but if I hang around theres bound to be a woman phone in and they always sound like they are talking about shagging to me.

From a slightly sideways non technical view... I could never get on with the film *Full metal jacket* and it's for a seemingly strange reason, and maybe even a totally superfulous one.

The clothing is accurate, the hardware is accurate, the language is spot on. But the light is all wrong. The sky over the London Docklands can never look like Vietnam becasue its way too North too. The colour is different even when it is cloudy.


Now i didn't know that that was where it was shot when i saw it first ,but said after, the light was wrong, sorry it ruined it for me. Now i accept that the vast majority of people probably 1... didn't even notice and 2.. couldnt give a flying duck anyway. However, i did it just made the whole experience totally phoney for me.

To go to all that effort to be convincing and then throw the baby out with the bathwater with the sky seemed to me to be way more stupid than just playing totally fast and loose with the whole continuity. He could have made just as good a movie with the same message using a wholly fictitious setting and war.

Now the relevance to 192 is this... if you haven't fallen asleep already.... 192 offers all the proper continuity and the right location, where location is the recorded acoustic in full, whereas lower sampling rates are the equivalent of a studio recreaton of the location.

Yes, to most people it will not worry or bother them , even if they notice at all, but there will always be that small section of the public who demand more.

It is up to the artist and the project manager to decide whether they want to indulge in total continuity and do so in the full knowledge that it will mostly go over peoples heads.

Be that as it may, maybe, just maybe, musical recordings that have true longevity will more often come from the school of total continuity because even if it is only subconciousm, people have an intrinsic sense of when they are listening or watching the *real thing* as opposed to a studio recreation.

--------------------
nibbled to death by an Okapi http://www.soundclick.com/tubilahdog

Quote Hugh Robjohns:

---

I think it might be best just to retire to our respective corners on this one. No one is likely to be convinced of the counter arguments. Minds are closed and beliefs are two deeply held.

For the record, I'm of the opinion that 44.1 or 48kHz should be sufficient, but the practicalities of implementing a suitable design make it difficult to get right at budget prices. Pay big money for something at the real cutting edge, like a Prism converter, and you'll soon realise that there is nothing much wrong with 44.1 or 48kHz sampling.

For the rest of us, moving up to 96kHz is a very cost effective and practical workaround. Moving the cut-off point up another octave relaxes the constraints sufficiently that it becomes possible to get an excellent sound for the kind of budgets that are reasonable for hobbyists.

Moving it up again to 192kHz, in my opinion, can start to become counter-productive. Not just because of the data storage and processor overheads, but also because it starts to put a lot more strain on the design of some specific aspects of the converter. The result is that you have to go back to the top flight designers and pay a lot more more money to get something that works as intended.

But this is a very complex subject indeed, and there are a great many compromises and trade-offs involved -- most of which aren't obvious to the non-specialist.

If you happen to like the sound of manufacturer A's converter at 192kHz over manufacturer B's converter at 44.1, then great. It's a subjective choice and you are free to choose whatever works for your ears.

However, you cannot infer from that selection anything truly objective about the merits of 192 over 44.1, because there are far too many variables involved.

We all know how different analogue circuitry or circuit topologies can sound subtly (or even blatently) different. A lot of the sonic differences between otherwise similar converters is purely down to the design and layout of the analogue circuitry and the nature of the power supply system(s).

In the case above, Manufacturer A's converter would probably still sound different to manufacturer B's product at the same sample rate, simply becaue of the different analogue circuitry involved. We could then argue about which one was better -- the more technically accurate, or the one that sounded more 'analogue'... but we still wouldn't reach a concensus.

Objectivity doesn't get much easier even if you compare two converters from the same manufacturer, operating at different sample rates. Let's say unit A operating at 44.1 and unit B at 192. Same analogue electronics this time, but completely different decimation filters involved with different slopes, ripples and phase responses. The sonic differences here could easily be down to the choice of different decimation filters as to the sample rate itself.

In fact, some high end converter manufacturers actually incorporate four or five user-selectable filter characteristics, and switching between these without changing the sample rate at all produces distinct sonic differences easily as great as comparing two identical converters operating at different sample rates!

So come on, let's stop the pointless arguing. The theory is plain and incontrovertible. 44.1 should be enough, but practical construction constraints tends to let the theory down. 96kHz provides a reasonably convenient workaround. If done properly, 192 shouldn't be any worse, but doesn't really offer any advantages -- and if not done properly it can actually be less accurate than 96!

Of course, some people might like the less accurate version because it sounds 'more analogue'... and round we go again...

Oh, and film is shot with a sample rate of at 24 frames a second but the film projector oversamples the output by a factor of two or three times by using a rotating shutter.

Funny old world, isn't it!

hugh

---

So id say that basically different sample rates dont really play a significant role in the quality of the production when the quality of it is really determined by the use of different use of decimation filters.
I really think we shouldnt worry about different sample rates at all.We should all be thinking about the different use of filters and equalizers which both determine the main quality of production.
Richie Steed.
www.soundclick.com/steedie

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RSteed

Of all the gin joints complex threads, in all the towns forums, in all the world, she had to walk into mine this one...

Well I'm sure "we" will all sleep better now that Richie Steed has cut the chaff out of the argument and revealed its beating heart for all lesser mortals to see clearly. Unless of course that was the 'Royal "we"', in which case I think "we" should leave well alone and get back to the colouring books...

Barish, Mr Robjohns et al, you have my sympathies.....

J/

Quote Magic Window:

---

Digitising audio at 44100 samples a second means that, according to Nyquist's theorem, effectively record frequencies up to 22050 Hz. However, the higher the frequency, the less the detail - a 200 Hz sine wave will be far more accurately represented than a 13,000 Hz sine wave, and thus will sound much better.

---

This is a common argumant, but is technically incorrect I'm afraid. If you re-assess the Nyquist theorem you quote, you'll see that the signal can be fully reconstructed -- without any waveshape distortion and with perfect accuracy -- provided there are at least two samples per cycle. The maths proves this without any doubt whatever.

Quote:

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We can see that the waveform is starting to approximate to a sine wave a lot clearly now.

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Again, a common mis-interpretation of the graphics. Your waveform has not been processed correctly. This is a very crude join-the-dots represenation of the D-A process. The essential 'recontruction filtering' element of the D-A process has not been performed here. The reason for the apparently poor waveshape reconstruction is because the signal as shown is still full of image frequencies. Pas the same data through an appropriate reconstrcution filter to remove those images, and the pure sinewave signal will be recovered in all its pristine glory.

Quote:

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Not perfect, by any means, but this does show that even though a sampling rate of 44.1 Khz is adequate for representing frequencies over the human hearing threshold, high frequencies are represented horribly.

---

As explained, it shows no such thing!

hugh

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Technical Editor, Sound On Sound

Quote The Bob Campbell:

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What if the two adjacent frequencies creating this lower frequency beat effect are not within the permitted frequency range, but lie outside it? Is a "human audible" sub-22050khz beat effect created by higher frequencies being lost when we filter out everything above 22050?

---

This could indeed be an issue, but only in some very specific circumstances. Consider the case of an orchestra -- strings and brass are known to generate a fairly significant ultrasonic spectrum. These ultrasonic elements will produce beats as the sound from the various instrumetns mixes in the air before reaching the (stereo) mic. Hence the mics will capture the audible beats and they will be recorded int a digital system in exactly the same way as in an analogue recording system.

The same will be true of multiple live sources mixed through an analogue console before recording digitally. Good analogue circuitry has a very wide bandwidth, and ultrasonic signals will be able to interact and produce audible beats within the analogue circuits.

The potential fly in the ointment comes when recording tracks individually, and when overdubbing. In these cases, ultrasonic signals are removed as the source is recorded, and hence they can not interact with those of other signals during the mixing stage.

Perhaps these thoughts go some way to explaining why multitracked digital projects seem to lose some of the 'warmth' we associated with analogue, whereas straight-to-stereo projects seem to benefit from digital.

hugh

--------------------
Technical Editor, Sound On Sound

Quote Magic Window:

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Thanks for not being a hostile, bitter old bastard like our friend Barish over there.

---

In fact, of course, he is none of those things. He is knowledgeable, accurate, right and patient beyond all reason!

Quote:

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The calculus is irrelevant for one's understanding of how digital audio works on a basic level.

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Maybe, but so far there appears to be a lack of even a rudimentary understanding of the basics from your corner, and a refusal to accept the generous help offered by those who clearly do understand the subject well.

Quote:

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I understand it now, I think - samples are 'snapshots' of amplitude levels

---

Yes, they are, but they cannot be contemplated as individual events. They are utterly meaningless as individual samples -- the audio signal is not conveyed by individual samples but by the ongoing stream. One sample by itself is just a click. You cannot determine the fundamental frequency of a signal by looking at a single sample -- or even a pair.

So clearly, your 'understanding' that a sample is a snapshot of signal amplitude is very limited in its ability to help explain the principles of digital audio.

Quote:

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... but because speaker cones cannot move from one point to another instantaneously, you get a non-linear movement that "fills in" the sine wave even though there are only two sample points, right?

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Wrong. The 'filling in' as you call it is done by the reconstruction filtering stage of the D-A conversion.

hugh

--------------------
Technical Editor, Sound On Sound

Quote Hugh Robjohns:

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Quote Magic Window:

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Thanks for not being a hostile, bitter old bastard like our friend Barish over there.

---

In fact, of course, he is none of those things. He is knowledgeable, accurate, right and patient beyond all reason!

MagicWindow Quote:

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The calculus is irrelevant for one's understanding of how digital audio works on a basic level.

---

Maybe, but so far there appears to be a lack of even a rudimentary understanding of the basics from your corner, and a refusal to accept the generous help offered by those who clearly do understand the subject well.

---

Hugh,

I am humbled.

Among all that commotion I didn't have a chance to make a point about Dan Lavry's negative comments about your integrity/sincerity in your articles in his post there. I personally don't agree with his views on that, which is why I am here participating. We all base our views on what is available to us in any given time and as the time passes we all find out that sometimes the things we thought to be right were actually wrong and the ones we thought to be wrong were in fact, right. I hope Dan Lavry has a chance to see that in your articles, AND in his comments as well.

Quote MagicWindow:

---

The calculus is irrelevant for one's understanding of how digital audio works on a basic level.

---

That's correct. But your problem is, the area that you are trying to understand/explain is not the basic level of how digital audio works. It is quite an advanced part of it.

That's why no matter how many times people tell you about reconstruction filters it keeps going over your head.

If you don't know the concepts of derivation and integration, there is no way you can fully picture the processes of AD and DA respectively in your mind.


Open up the Calculus and analytic geometry book and look at the graphical representations of the mathematical functions and their similarity to the graphic representation of sound waves.


Sound is mathematic.


It can be mathematically represented in a function.


Functions can be derived (which is what Analog-to-Digital conversion is all about).

Derivatives can be intregrated (which is what Digital-to-Analog conversion is all about).


If four basic operations is all you know about mathematics, then you have no other choice but to connect two dots with a straight line. But as soon as it turns into a curve, you need a higher mathematics. If it turns into a spline, it requires even higher mathematics.

That's why people keep missing the point when reconstruction filter is mentioned.

It is a very complex issue and there are some facts that straight logic falls short to explain. You certainly need a certain amount of higher mathematic knowledge in order to debate on the fidelity issues in AD/DA process.

B.

Quote Stevedog:

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From a slightly sideways non technical view... I could never get on with the film *Full metal jacket* and it's for a seemingly strange reason, and maybe even a totally superfulous one.

The clothing is accurate, the hardware is accurate, the language is spot on. But the light is all wrong. The sky over the London Docklands can never look like Vietnam becasue its way too North too. The colour is different even when it is cloudy.

---

I had a bigger problem with the Beckton Alp dry ski slope which loomed up in the background from time to time! Charlie don't ski!

Barish,

My apologies, sir. My rude comments earlier were a kneejerk reaction to some pretty bad news. You know how humans are very bad with dealing with bombshells. I am grateful for you and Hugh and everyone to explain to me what is really going on.

No hard feelings, please?

If I may continue, I would like to know how more complex signals are treated. Am I right in maintaining that the reconstruction filter does the same for say, white noise, as it does for a sine wave? A friend of mine suggested that the characteristics of the filter may 'colour' it a certain way, and this is why you've been talking about using one's ears and personal taste to determine if an audio system is to your taste?

I love debates like this (except people usually ignore me, nevermind..)

I fully appreciate that for some sound sources a higher frequency sampling rate is preferable.

Surely though, if the music is going to end up on CD at 44,100Hz then recording at whole number multiplications of 44.1KHz is preferable thereby leaving headroom in the system but more importantly eliminating (?) any aliasing and other artefacts of sample rate conversion in the mastering stage.

Someone please acknowledge this and tell me if I'm right or wrong; it's like talking at a brick wall (limiter) in here sometimes....

Quote smiling stranger:

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Surely though, if the music is going to end up on CD at 44,100Hz then recording at whole number multiplications of 44.1KHz is preferable thereby leaving headroom in the system but more importantly eliminating (?) any aliasing and other artefacts of sample rate conversion in the mastering stage.

---

The sampling rate has no effect on the headroom. That is determined entirely by how much headroom you choose to leave when recording/post-producing. Furthermore, the sampling rate has no effect on aliasing artefacts when sample-rate converting: that is down to the accuracy of the maths involved in creating the appropriate reconstruction filtering.

With early sample rate converters, the limited (by modern standards) accuracy of the process meant that there were often sonic advantages to using simple multiples of sample rates -- 88.2 for conversion to 44.1 rather than 96, say. However, these days, modern SRCs are stunningly accurate -- far more so than most converters in fact, regardless of the input/output rate ratios.

As proof of this, the superb Benchmark DAC 1 -- one of the best sounding D-As available -- passes everything through a sample rate converter which outputs at roughly 110kHz regardless of the input rate. Everyone complements the DAC1 on its sound quality and no one is the least worried about the non-interger sample rate conversion!

hugh

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Technical Editor, Sound On Sound

I think we need to remember a simple fact about digitising analogue waveforms:

It is widely accepted that to capture a waveform with some accuracy, the sample rate must be at least twice that of the highest frequency to be recorded. 44.1kHz would appear to be adequate for audio signals up to 22.5kHz for example.

However, a significant problem at 22.5kHz is that a pure sine wave would appear slightly 'triangulated' if digitised literally. This is because there are a maximum of two samples at that frequency, and the line that joins them is not a curve but a straight line. This is, perhaps, why high frequencies sound harsher on digital recordings at 44.1kHz.

Although there are software algorithms to correct this flaw, few can claim to know what exactly occurs between the sample 'points' and so can only give an approximation as it might be more complex than a sine wave.

Clearly, a higher sampling rate improves the digitisation at high frequencies (including those within the hearing range). This assumes that the AD conversion is of decent quality and not compromised by the extra speed.

Some argue that it will be mixed down to 44.1kHz anyway (for CD) and that you would require, say, a 96kHz player to appreciate the difference. However, if the mixing of various waveforms is done at a higher sampling resolution than 44.1, the resultant waveform will reproduce the interplay of harmonics better, and the final master will include most of these subtleties.

Quote Stevedog:

---

Now the relevance to 192 is this... if you haven't fallen asleep already.... 192 offers all the proper continuity and the right location, where location is the recorded acoustic in full, whereas lower sampling rates are the equivalent of a studio recreaton of the location.

---

I had a chance to talk to a very successful sound engineer @ NAMM about recording at higher sampling rates (96kHz / 192kHz), and he felt that even though we're outside the range of frequencies that are audible to human ears, there IS a slight improvement in terms of the sense of space, imaging, and localization of the instruments in the recording. Recordings with higher sampling rates, when listening under ideal conditions, seem better for capturing the reverberation characteristics, or the "air", of the room the recording was made.

As mentioned, there's greater hassles in dealing with larger amounts of data, and the approach is not beneficial for all types of recording (more applicable to minimalistic classical music recordings or 5.1 surround, but not to closed-miced multi-track rock studio stuff).

As stated, this is a situation of diminishing returns: while some listeners MAY able to appreciate a difference, for the clear majority of listeners (using consumer-grade audio equipment) the difference is imperceptible.

Don't flame me: that's what this egg-headed recording engineer told me....

(And after someone guffaws, I'll share how many Grammys he has won for best-engineered classical music recording...)

Chris

--------------------
1.5GHz Al 17" Powerbook G4 (2.0GB RAM, Hitachi 60GB 7,200 rpm drive), running Logic Pro 7 under OSX 10.4.5

Quote:

---

Eduardo:
I think we need to remember a simple fact about digitising analogue waveforms:

It is widely accepted that to capture a waveform with some accuracy, the sample rate must be at least twice that of the highest frequency to be recorded. 44.1kHz would appear to be adequate for audio signals up to 22.5kHz for example.

However, a significant problem at 22.5kHz is that a pure sine wave would appear slightly 'triangulated' if digitised literally. This is because there are a maximum of two samples at that frequency, and the line that joins them is not a curve but a straight line. This is, perhaps, why high frequencies sound harsher on digital recordings at 44.1kHz........ *snip*

.....done at a higher sampling resolution than 44.1, the resultant waveform will reproduce the interplay of harmonics better, and the final master will include most of these subtleties.

---

Here we go again!

You clearly either didn't read or didn't understand the rest of this thread! Before you start reminding people of simple facts can I suggest you do read it and try to understand it because you seem to be missing a few simple facts that we've been over..and over...and over....

Gegen die Dummheit kaempfen die Goetter selbst vergebens.

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www.the-byre.com No longer Forum Member

Quote Eduardo:

---

I think we need to remember a simple fact about digitising analogue waveforms:

It is widely accepted that to capture a waveform with some accuracy, the sample rate must be at least twice that of the highest frequency to be recorded. 44.1kHz would appear to be adequate for audio signals up to 22.5kHz for example.

However, a significant problem at 22.5kHz is that a pure sine wave would appear slightly 'triangulated' if digitised literally. This is because there are a maximum of two samples at that frequency, and the line that joins them is not a curve but a straight line. This is, perhaps, why high frequencies sound harsher on digital recordings at 44.1kHz.

Although there are software algorithms to correct this flaw, few can claim to know what exactly occurs between the sample 'points' and so can only give an approximation as it might be more complex than a sine wave.

Clearly, a higher sampling rate improves the digitisation at high frequencies (including those within the hearing range). This assumes that the AD conversion is of decent quality and not compromised by the extra speed.

Some argue that it will be mixed down to 44.1kHz anyway (for CD) and that you would require, say, a 96kHz player to appreciate the difference. However, if the mixing of various waveforms is done at a higher sampling resolution than 44.1, the resultant waveform will reproduce the interplay of harmonics better, and the final master will include most of these subtleties.

---

Lord Almighty please give me patience...


Please read, dude. PLEASE.


Do everyone here a favour. Please READ the thread all throughout before posting.


I have spent days here trying to explain why the opinions like yours are wrong, with the simplest technical and mathematical language possible so that the simple fellas with four-operation maths like you can understand.


What's more dangerous than being ignorant about something is to be unaware of how ignorant oneself actually is.


Please don't be that.


Look what I'd written a few posts above yours:

Quote Barish:

---

If you don't know the concepts of derivation and integration, there is no way you can fully picture the processes of AD and DA respectively in your mind.


Open up the Calculus and analytic geometry book and look at the graphical representations of the mathematical functions and their similarity to the graphic representation of sound waves.


Sound is mathematic.


It can be mathematically represented in a function.


Functions can be derived (which is what Analog-to-Digital conversion is all about).

Derivatives can be intregrated (which is what Digital-to-Analog conversion is all about).


If four basic operations is all you know about mathematics, then you have no other choice but to connect two dots with a straight line. But as soon as it turns into a curve, you need a higher mathematics. If it turns into a spline, it requires even higher mathematics.

That's why people keep missing the point when reconstruction filter is mentioned.

It is a very complex issue and there are some facts that straight logic falls short to explain. You certainly need a certain amount of higher mathematic knowledge in order to debate on the fidelity issues in AD/DA process.

---

B.

I must admit, I am rather astonished by how little people know and yet they claim to know what they're talking about.

Now, I certainly don't claim to know what I'm talking about with any authority, but the things I think I have picked up over the last few years are these:

1. You only need 2 points I.e. 2 samples of amplitudes, to create a sine wave at the reconstruction filter I.e. it joins the dots with a smooth curve - not a straight line.

2. The maximum frequency you can hear (or just over) is ALL the information you need (ignoring "beat" effects etc). Therefore any extra information that would be between the two points sampled is excess info and can be lost. I.e. the highest frequency sine wave you recreate is literally the highest frequency you need and hence all other *detail* can be discarded.. This gives you the sample rate for 22.5kHz:
22.5kHzx2(samples)=44.1kHz

3. A visual description might help: The wave form you would see in a wave editor can be thought of as a combination of many frequencies of sine waves occurring at different times and amplitudes, but adding together to form one continuous line or one wave. If you magnified an actual soundwave at a point and looked closely, you would see detail that is smaller in wavelength than a 22.5kHz wave BUT as you can't hear this detail you don't need it when digitising.

I've tried to make some sense of this in my own head and I'm hoping I've pretty much got there - and therefore my simplified explanation should be able to help others...... However, If I'm wrong, please feel free to shoot me down.

D

http://jn.physiology.org/cgi/content/full/83/6/3548

The industry's focus on the 192kHz issue is pretty ironic. It seems more concerned over trying to catch supersonic sounds (which tape could capture anyway) than it does in dealing with the real issues clouding sound fidelity these days ... namely a) the dominance of sub-standard audio codecs like MP3 and WMA and b) the so-called "loudness wars" which have resulted in most popular music today sounding harsh and brittle at any level of volume, and being mostly void of any dynamics.

--------------------
"Nothing in the world can take the place of persistence. Talent will not; nothing is more common than unsuccessful people with talent."

Quote The Bob Campbell:

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http://jn.physiology.org/cgi/content/full/83/6/3548

---

That's the Oohashi paper that I mentioned in an earlier post. No-one has been able to repeat the findings as far as I know. Unfortunately I can't find any of the postings that refute his claims or question the methodology but I believe that people have tried to repeat the experiments and haven't seen the same results.

Cheers

James.

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JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net