---

---

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

---

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:

---

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:

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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."

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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.

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