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In Reply to: Re: Fuck you, Hansen nt posted by Middleground on August 22, 2003 at 06:33:30:
CD was *not* an advancement! ...Except insofar as noted in the article liked below. It was a marketing ploy, to get people to buy a whole new record collection, sound be damned.And if one of those Sony/Phillips guys could explain why tweaking a CD produces superior results, I for one would be *very surprised*. I don't think they really understand digital audio.
clark
Follow Ups:
For those of us who are too young to have ever bought more than a few silly records, the CD freed us from the cassette tape. We need to keep in mind that not too many people ever listened to vinyl in their cars or on easily portable systems. In that context, I think the CD is a pretty damn good solution. Sure, it could have been much better for the audiophile, but us loons are a small part of the market and difficult customers to boot.Regardless of the countless techincal sonic issues with CDs, it is a much more convenient medium than anything else available in the mid-80's, and we all know the mass market loves convenience.
Of course, you're probably multiple decades older than I, so I wouldn't expect us to have the same context on an issue like this :-) If I had any sort of meaningful vinyl collection when CDs emerged, I would probably think/feel differently, too.
As for your wish, if I remember correctly, I believe a Sony rep did chime in on one of the threads regarding CD track sequencing versus jumping. The gentleman stated that he had heard the effects of this phenomenon, and that his rig was immune to it due to the numerous pieces of expensive gear in his digital front end. I don't recall his name and can't find the post for the life of me though, so I may be nuts.
-Pete
CDs are certainly an improvement over cassette.The only problem from an audiophile perspective is that it lowered the absolute maximum attainable sound quality.
It acted sort of like a compressor; it brought the lowest fidelity systems up to a pretty high level (witness what's now possible in a car), but it also lowered the top end slightly.
It's taken until very recently to get CD playback up to the level of vinyl.
I abandoned vinyl (mostly) in favor of CD, and my 750 records say unplayed for the last 10 years. A month ago, I invested in a reasonably high end table, and I am completely blown away by (or maybe just reminded of) the musicality of vinyl.
Just a reminder Clark; this is 2003 not 1983. The CD has exposed more flaws in recording techniques than most people care to imagine. However, people like you are essentially dysfunctional because you are so accustomed to the high noise and distortion of 78's , tape, and poorly designed tube gear, that you really have lost any grip on what live unamplified acoustic music is really all about.
Many of the recent recordings I've heard are compressed to death; the only goal seems to be to get the level as high as possible. This (unfortunately) also seems to be the norm for concert PA management -- crank everything up until it's simply a wall of distortion with zero dynamic range.With a few exceptions, it seems like we're going backwards, not forwards...
I wonder if the convenience/portability/durability of CD has contributed to this? For example, lots of people (including me) play CDs in their cars, where you have to use compression or you lose music down in the noise.
BTW, in the CD-is-good-or-evil debate, there's one thing that is rarely talked about:
The CD was pushed to market with whatever sample rate was possible given the silicon available at the time, not the sample rate that was determined to be appropriate for good sound. Given the Moore's Law rate of chip improvement, a 12-24 month delay in the release of CD might have given us a far better medium with less sonic compromise.
Anyone who looks you in the eye and claims that a brickwall filter at 20kHz and a 16-bit sample rate is "perfect sound" is deluding themselves. Anyone looked at the distortion spectrum of a CD player at -40dB (where a lot of the music is) instead of at 0dB?
Notice that Sony is now touting the audibe benefits of SACDs extended bandwidth? But wait, we can't hear anything above 20kHz.........
Peter
Stop listening to pop music which tends to be deliberately done in this fashion. Listen to some TELARC classical discs and then tell us what you hear. You may want to check out this article I wrote for Audioholics.com which may help you understand what's going in in the studio.
http://www.audioholics.com/techtips/specsformats/CurrentFormatTrends.php
I just wish it was a bigger percentage....We've got two different directions happening:
The people who care are making better and better recordings; those who don't are getting worse and worse.
Peter
a
""But wait, we can't hear anything above 20kHz.........""peterProve it....
Human hearing directionality response is capable of 20 microsecond ear to ear delay, with some actually able to discern to 10 uSec.
That is the realm of 1 foot side movement of a source ten feet away. So it is my expectation that 44Khz sampling rate is unable to duplicate soundstage with total accuracy.
Cheers, John
The timing resolution of a 44.1kHz system is greater that the sample rate ie 22us.steve
Agreed. But the transient information within the recording can only be in 22 uS time steps.On a sample by sample basis, that could blur the soundstage, a transient could be very close to the sampling time, and on occasion show up in the stream one sample later.
Cheers, John
HowdyNope, if you for a moment assume that there is no level quantization, an appropriately bandlimited transient can show up in any phase and be correctly reconstructed. E.g. the sampled point isn't necessarily the peak, the correct peak will be there after the reconstruction filter.
-Ted
I wasn't talking peak level.What I was getting at, is that an analog transient that crosses zero (arbitrary point of course) at any instant can show up in the sampled datastream up to 22 usec later. In fact, any analog transient superimposed on a slower signal can have the temporal relation shifted between the two waveforms by a max of 22 uS..
Cheers, John
HowdyFine: it's also true that with the same assumptions a zero crossing can have any phase after the appropriate reconstruction filter. The phase isn't quantized by the sampling rate. The sampling rate affect the highest representable frequency.
Think about it, with no quantization of level, any appropriately bandlimited signal can be reproduced accurately. Putting it more simply, any sine wave with a freq less than the Nyquist rate can be easily represented and reconstructed and only if it's freq is rationally related to the sampling rate will there not be all possible phases of zero crossings present.
The real fun starts when you start quantizing the levels and/or you try to time limit the signal.
-Ted
Ted:When you say
an appropriately bandlimited transient can show up in any phase and be correctly reconstructed. E.g. the sampled point isn't necessarily the peak, the correct peak will be there after the reconstruction filter
You're right, but doesn't "appropriately band limited" mean containing no signals with a component at more than half the sample rate?
So a 44.1kHz stream can't contain any anything above 22.05kHz, i.e nothing with a period of less than 45uS.
If there is information in the original signal that the ear/brain can discriminate down to the 20uS range, does bandlimiting (and therefore the subsequent A/D/A) have the effect of eliminating/smoothing/smearing that information?
Peter
HowdyMy whole point is that the sample rate isn't in and of it's self responsible for quantizing phase. As I point out in my response to John, with no quantization of level, all phases are possible. When you quantize the level things are more interesting and the sample does come into the picture, but it's been many years since I took DSP courses and my books are packed so I won't even try to come up with the relationship of quantization of level to resultant phase effects.
Perhaps Werner or some one else who does this stuff actively for a living will chime in.
-Ted
you're right that anything up to half the sample frequency will be perfectly reconstructed (assuming no jitter, of course....)Now how does that jive with the fact that we can apparently hear arrival time differences down to around 20uS, or half a sample period? Doesn't sound like a CD has that level of time resolution.
I know I'm mixing up a bunch of concepts here, which is why I'm having trouble connecting the dots....
Peter
HowdySince you can represent any given sine with an arbitrary phase, you just add enough up to give you the attack or transient you want with what ever phase you want. (Limited by Nyquist for highest freq and ultimately the level quantization introducing enough errors which smear the time resolution.)
CDs can represent left and right channel phase differences which are quite small, smaller than a sample period. This is probably how great CD players with low jitter get a precise soundstage, even for higher freq instruments like the triangle, etc. (I'm just hand waving here.)
I don't know the exact time resolution of Redbook, but the simplistic argument that it's one sample period is clearly hooey.
-Ted
""If there is information in the original signal that the ear/brain can discriminate down to the 20uS range, does bandlimiting (and therefore the subsequent A/D/A) have the effect of eliminating/smoothing/smearing that information?""peterMy take? absolutely.
In fact, take a look at a 44Khz sample rate of a 15 Khz sine wave..The reconstructed waveform looks like three 1 Khz sines superimposed in three phase..so the peaks are riding three crests..
The only way to get back to the real wave is to reconstruct with a wide window, whereas cd replay is point to point..Blowin Nyquist theory out the window..
Cheers, John
HowdyNope, if you use a proper reconstruction filter a 15k sine will be precisely a 15k sine after reconstruction.
CD is not point to point.
-Ted
Look at the waveform..You will see what happens..I wish I could post a pic here.John
HowdyI have. I used to write software for digital audio workstations.
Be careful what hardware and software you're using. Many software programs display samples connected by lines not what the waveform will look like when run thru the reconstruction filter.
-Ted
that was fast..Yes, by looking at the dac immediately before any filtering, you see the step waveform. And the peak values at that point are visibly riding three sines. But, post filtering cannot reconstruct the peak value of the 15Khz signal on a cycle by cycle basis unless the filter is resonating at 15Khz.
Nyquist theory assumes the ability to reconstruct using more than one sample, much as an FFT requires many points.
It would be nice to have someone chime in who can support the math to show how point by point reconstruction is not really pure nyquist theory. I can't.
How do I get pics here?
Cheers, John
HowdyHere, for example is a 14kHz tone (actually to keep your #'s correct I used a 14031.81818181Hz tone at 44.1kHz) displayed by a real program Cool Edit Pro.
-Ted
HowdyPoint by point reconstruction introduces higher freqs than the Nyquist rate.
Anyway, we are off track. The whole point is that a proper filter allows finer time resolution than a naive 1/sample rate argument would indicate. An improper filter is more sloppy on the time resolution.
-Ted
HowdyAs I said we are assuming band limited signals. If you are talking about only one cycle of a wave form that is decidedly not bandlimited.
Nyquist shows that as long as you have incrementally more than 2 samples per cycle of your wave you are OK unless you are extremely unlucky, e.g. you have a signal whose freq is exactly a rational relationship with the sample rate. Since the rational #'s are sparse in the real #'s this is rare :)
And if you are bandlimited you have many cycles of your signal so you see the correct peaks, etc.
If you have a picture host you just put the URL for your picture in the 'Optional Image URL' when you post. You can also put in the HTML IMG tag, if you want it somewhere other than at the top of your post. If you don't have an image host, the Asylum provides inmate picture galleries for contributors.
-Ted
Thanks for the pic info Ted..""And if you are bandlimited you have many cycles of your signal so you see the correct peaks, etc.""ted
Agreed..but look at your 14 khz waveform...you really have to wait around to see the sample coincide with the actual peak of the waveform. You are making the connection in your thinking that when the two coincide, that is the actual wave amplitude..
Reconstruction of that signal by a DAC...For the dac to really output the correct amplitude of that signal, it has to look over many samples to get the right amplitude..Hence, my point..
Now, if you were to "connect the dots", so to speak, with some filtering, you should see some envelope modulation..what is interesting is that the top of that envelope is not symmetric with the bottom one, there will be a frequency component there. And, nyquist math does not produce that artifact, unless you include the FIR response of real filters, representing the very few samples that are actually available to the filter. Also, don't forget, Nyquist math requires sample lookahead in order to work..
Thanks Ted
John
HowdyHave you read all of my responses on this thread?
You don't have to wait for the peak. Look at the waveform that Cool Edit Pro displayed. The program had no idea what I gave it. Here's what I gave it:
#include#define _USE_MATH_DEFINES
#includeint main () {
const double sr = 44100;
const double sp = 1/sr;
FILE * h = fopen("out.pcm", "w");
for (double t = 0; t < 1; t += sp) {
double a = sin(t * (M_PI * 2) * (sr / (44000.0l/14000)));
short s = int(a * 32767);
fwrite(&s, 2, 1, h); }
fclose(h); }
Cool Edit Pro (and your DAC) have no idea what waveform I gave, but the peak is reconstructed accurately from the samples present.
Anyway go listen to your 14kHz sampled and reconstructed at 32k, 44.1k, 48k, 96k, etc. They all sound the same to my ears and have the same amplitude. It also agrees with theory. Why would I believe yours eyes instead of mine and my ears and the theory?
-Ted
HowdyShould anyone want to play with this stuff to recreate my examples you might want this slightly corrected code. If you don't open the file in binary you'll get weird extra bytes now and then :)
#include#define _USE_MATH_DEFINES
#includeint main () {
const double sr = 44100;
const double sp = 1/sr;
FILE * h = fopen("out.pcm", "w+b");
for (double t = 0; t < 20; t += sp) {
double a = sin((2 * M_PI) * (t * 154350.0l/11));
short s = int(a * 32767);
fwrite(&s, 2, 1, h); }
fclose(h); }
-Ted
""Cool Edit Pro (and your DAC) have no idea what waveform I gave, but the peak is reconstructed accurately from the samples present.""TEDTed...try presenting it the samples from .19625 to .19655 only.
Bet it doesn't give you the correct peak..
Looked at the notation you provided..fraid I'm clueless as to what it says..my fault, not yours..
What is the math cool edit pro uses to reconstruct the underlying waveform? is it the same math that is applied to the output of a CD??
At first, I had thought the oversampling techniques did that, but apparently not.
""Anyway go listen to your 14kHz sampled and reconstructed at 32k, 44.1k, 48k, 96k, etc. They all sound the same to my ears and have the same amplitude.""ted
The only thing I have to play it back on here is cd type stuff..so I can't yet do that..My newer 24/192k card may support that, but it's still in the box..I think I'll just build a new computer around it instead of the infernal problem of "new hardware found".
""It also agrees with theory.""ted
In theory, yes..but the application is not up to the theory. My DSP prof's never used truncated signals to support nyquist. They did, however, show how truncated signals cannot provide optimal reconstruction. Some kind of convolution between the sampling window and the signal of interest, from what I recall..definitely fuzzy on that, though.
""Why would I believe your eyes instead of mine and my ears and the theory? :)""ted
Ummmm because I'm such a nice guy???. :-)
A really cool test of what we speak? Take the pure file, .wav it, play it into a typical CD player, and scope it..Then use a parametric set high Q, look for the artifacts I see with my eyes..by scanning the parametric across the 500 hz to 5 k range.
You're correct..It's my eyes here, not my ears..And I also think I wouldn't hear a diff over the rates..
I love this crap....
Thanks Ted,
John
BTW, does 25 dollars get me the ability to put pics on AA? Cause I really like AR for that capability.
HowdyIf you give a DAC or the program too few samples you are convolving a rectangular window which by definition has infinite bandwidth.
The code I gave is C++ and it generated the .wav file I used as input to Cool Edit Pro and that I played out my DACs.
Cool Edit Pro is simulating a DAC by applying the correct brick wall filter and then resampling at the rate dictated by the number of pixels on the screen (or thereabouts) to draw the waveform.
Sorry I don't have to do your parametric filter test, I've been there, done that, etc. If I had my Digital Audio Workstation alive I would do the experiment just to humor you, but then you'd need to understand it's algo's to believe that it's doing the experiment you want. On the other hand I don't happen to have this kind of analog equipment laying around at my house to do the experiment in the analog domain. But like I said I used to do this stuff for a living and remember people arguing about this till the cows came home. I also have bad memories of people not getting it in school when they were only doing the math and clearing things up for them in another lab where I had the equipment to demo things for people. In a previous life I've written similar display code to that of Cool Edit Pro, just so customers didn't get confused.
Just because I thought you might ask, tho, I did play this file thru three of my DACs and got the same answer, no 1k envelope modulations (boy I'm glad my daughter wasn't here when I played that, she'd be complaining that I'm deaf at that freq.)
Yep, make a donation to the Asylum and get hosting for anywhere. Rod doesn't limit the pictures to just be displayed at this site.
-Ted
Did you try the limited sample size on the program? When you do, please post the pic..I remain unconvinced...But I'm really enjoying the conversation..thanks Ted..
When I get the chance, I'll put together the analog stuff to prove it either right or wrong..but I fear my time budget makes for a large queue.
I don't know if it's 1k envelope mods, or some weird thing. But I'll certainly think about all you've said..
Donation? Cool, I'd love to post some pics..got my 100 foot linecord/mike/snake that doesn't hum..my 18 pole resistive high bandwidth coaxial load..and soon enough, my IA setup..along with the pc test setup.
Cheers, John
HowdyHere's a selection:
.JPG)
Here's the display after pasting that selection into a new waveform:
You can see the window effects caused by assuming that the other samples are zero.
-Ted
I know..that's what I've been talking about..The DAC does not have the information from the future, nor the past..The output filter will only have the past..
And note..how would a filter output a signal that is higher in amplitude than the sample? points 4.5 and 9 come to mind.
Ted...thanks for this cool discussion..going offline now, be back monday..gotta take the kidlets to see medallion..
Cheers, John
HowdyOf course a reconstruction filter produce outputs higher than the input, otherwise how could they reproduce the original waveform without waiting for a peak :) A reconstruction filter isn't just a smoothing filter. Anyway ever hear of filters ringing? If you hit a reconstruction filter with a single non-zero sample, you'll not only get an output of some amplitude at that point but also non-zero outputs at other points, when there are more than one sample those other "spurious" outputs can overlap constructively or destructively and cause the output at any given point to be higher or lower than you might otherwise expect.
The reconstruction filter is a part of the DAC.
To do a proper brickwall filter it must look into the future a little, (by delaying the output a little) but that has it's own problems...
There is no such thing as an ideal implementation of a reconstruction filter, but in any case the better the filter, the better it's time resolution... which is what brought the whole thing up in the first place.
-Ted
is the idea that you can completely create a sine wave at or near the Nyquist freq with only a couple of points.I think the key concept is that the data points allow only one waveform to be reconstructed that does not have components above the Nyquist freq. In other words, in order to create any sharper curvature in the waveform, you would need higher freq components that cannot exist.
This also means, of course, that you don't need to have the peaks, or any other particular part of the waveform, sampled in order to recreate the band-limted wave (as you said before).
Boy, this is easier to understand than it is to explain.....
Peter
Howdy"Boy, this is easier to understand than it is to explain....."
You said a mouth full there. There is a reason some (well motivated) people drop out of signal processing theory classes :) And here on the internet you don't even have the advantage of knowing the audience and the audience having the requisite background.
-Ted
Agree on mouthful and audience.To recover the peak signal, the reconstruction math has to work quite hard, and requires samples around the peak of interest.
There will be a tradeoff between the window size and the ability to reconstruct the waveform beyond the simple DAC hold between samples.
I make the assumption that most CD outs are simple filtering of the staircase DAC output signal after the hold circuit.
I do like the thought of a digital brickwall filter, as that could easily include a large sample delay for better implimentation and reconstruction. My thinking has been only on analog filtering.
Cheers, John
I don't know what is right, because I don't do digital,but thanks for the info. Ted. I can learn something from this exchange.
You may well be right, but the 44.1kHz sampling rate issue pales in comparison to loudspeaker/room issues. Also note that most microphones roll off very quiclky above 20 kHz. The few that have extended frequency response are rarely used because the noise level is too high.
But, with two identical speakers, with their identical foibles (identical being relatively so), soundstage can get mucked up if the system delays some of the info in the 20 usec range.How do typical cd players run?..My soundcard, at that rate, alternates channels for A/D conversion. So there is a built in right to left time shift. My vinyl does that when my cartridge is not tangential to the groove.
Cheers, John
You've got interleaved converters? If that's the case, my suggestion is to get something with uniform group delay. I think you'll appreciate it.
I don't know about my cd players or my denon 1800..Just my soundblaster platinum live..That's why the 24/192 card...Cheers, John
I have a The Phillips CD1 test Disc which has about 99 different tracks for CD test, including delay. Should I get you a copy? That and a scope will tell you what you need to know.
I'll be taking you up on that at a later date. thanks..Cheers, John
about the 20kHz limit. I think extended bandwidth is intuitively better, provided you don't start running into interference/EMI/ radio pickup issues. Less chance for causing unintended problems.I find it funny that the same company that claimed you didn't need to go past 20kHz (when they couldn't) now claims the benfits to going to 100kHz (because they can).
And yes, I also believe that steady state sine wave hearing tests don't tel you much about how we hear music.
Peter
nt
I really can't hear a direct tone much above 14Khz.
Which makes me think that how we "hear" above 20Khz is
by changes in image location and size and shape.Why else have two ears, if not to hear these subtle
timing differences?
I'm close to 17Khz.But, I'm really surprised by the ability to hear 20 microsecond timing delays, cause that implies that audio systems need to be phase accurate way over hearing BW to keep directional information intact, otherwise soundstage is compromised..but we actually don't hear the freq's up there..
Cheers, John
""But, I'm really surprised by the ability to hear 20 microsecond timing delays, cause
that implies that audio systems need to be phase accurate way over hearing BW to
keep directional information intact, otherwise soundstage is compromised..but we
actually don't hear the freq's up there..""
What has time delay to do with freq's up there?Is it not to totally different things?
Thanks. Jens.
A 20 Khz sound has a period of 50 microseconds, for reference.Many systems are described by either the bandwidth, expressed as the frequency where the response is 3 db down, or by the risetime of the system, how fast it can respond to a change of signal. The relationship between the two is inverse, as in Freq is proportional to 1/slew rate.
A system which is capable of responding to a 5, or 10 uSec transient usually is a system that has a very high bandwidth, one which is far beyond human hearing capability.(I've been saying 20 uSec, but the study I saw had several subjects clearly distinguishing down to 10uSec.
To say that a human can distinguish a 10 uSec difference in delay almost implies that we can hear in the 50 to 100 Khz range, which is clearly not possible.
But to get 10uSec response from an audio device requires the designer make the amplifier capable of very high bandwidth response.
As John Curl said, he claimed he heard the diff between a 35 Khz filter and a 100Khz filter..But, he is not saying he can hear up there. My take is that he can hear the transient response differences, although some would try to take him to task as saying that he is hearing 35Khz stuff, which he did not say.Cheers, John
Take 3 kHz as referance.
Take two signals, and delay one 15 deg.
Take stereo, and a very good radiating speaker, one would have no problem telling the shift in location of the signal.The brain can tell what ear is first. What has that to do with very high bandwidth? That is what I do not understand. I do find it to be two totally different things.
Thanks. Jens.
is that the fact that we can localize sounds using delays in the 10-20uS range, which is less than the period of a 20kHz sine wave. In your example, both signals are arriving at both ears, separated by very small delays. How does the brain distinguish one signal from the other?The conventional 20kHz upper limit of hearing is based on tests using (I believe) steady state tones.
Doesn't this imply that the notion that we don't need info > 20kHz because we can't hear sine waves over that frequency may be a gross oversimplification?
So, you are right that there is not a CLEAR connection between these two things, but one does seem to contradict the other.
Peter
It is a gross simplification that because we cannot hear over 20Khz, that 10 uSec information is useless..Calculate the difference in time between right and left ear for a source ten feet away, and one foot off axis..that seems like a reasonable thing for a species that used to depend on hunting to survive..Course, it also is a reasonable thing for the existance of a soundstage between two speakers 10 feet apart, and ten feet away from the listener.
Mtry on AR (or was it Radar) gave me some links to the studies that showed the human subject response to L-R delays...Some subjects ran down to 10 uSec capability, statistically valid..And yes, it certainly does seem to contradict..
So I certainly wonder how a 22 uSec sampling period can be capable of conveying 10 uSec information necessary for soundstage.
Cheers, John
Your example of 3K, 15 degrees..1/3k is .33 milliseconds per cycle, 330 microsec's.
15 degrees is 15/360, or .041 of 330, 13 uSec...good example...
For a stereo amp to reproduce that with R-L coherency..yes, it doesn't need 100 Khz capabilities..in fact, 3 Khz would do, as long as the channels match..
But, for an amp to faithfully follow a complex audio signal without introducing transient delays at that speed? Hmmm..each channel has to faithfully reproduce the audio signal, without affecting the time relationships between, regardless of the signal content..I do not know how to design an amp that is low bw. but still faithfully passes 10 uSec transients..
To answer your question...some believe it is necessary to have high bandwidth in order to maintain the time relationship between the lower frequency information and the higher freq stuff. I think I belong to that group. :-)
I am absolutely confident that I have not answered your question to your satisfaction...I'll think about it...Sorry...
Cheers, John
NT.
It removes the tension between the "tests have shown you can't hear > 20kHZ" crowd and the anecdotal reoprts that extended bandwidth matters.I think there is way too little attention paid to the time-domain behavior of audio systems, and the impact on sound quality.
As Dan B. would certainly point out ;-), the time domain errors of most speakers are much worse than those of the upstream electronics.
However, if you could look at the cumulative time delay errors of the entire chain from microphone to speaker terminals, I bet it would look pretty horrific. I suspect that 44.1kHz A/D and D/A makes that even worse.
Peter
Peter, I would like to inject in here, that in 1974, Jon Meyer and I had our really good research lab. We used B&K 4133 condenser mikes that had an extended response to 40 KHz or so, and a pretty good pulse response, better than 9us risetime. This is better than most microphones then or now, BUT Mark Levinson, Stellavox, The Grateful Dead, and others used these mikes as their working reference. We found that they sounded 'more extended' DUH ! ;-) They were quiet enough, thanks to me, because I lowered their noise by about 10dB in the electronics.
Better than this is difficult, but it sure beats CD's, even perfect ones. We found that our hearing is not completely limited to what people say: Many people, especially when young, can hear extended respose to about 25KHz, but what seems to be really important is 'transient response' . I have run 'informal' tests myself where we could easily tell the difference between a 35KHz single pole filter and an 100KHz single pole rolloff. Tough nuts to those who have never tried it, properly.
""but what seems to be really important is 'transient response' . I have run 'informal' tests myself where we could easily tell the difference between a 35KHz single pole filter and an 100KHz single pole rolloff""JCAnd that is of course, consistent with the oversampling crowd with their use of higher breakpoint filters.
The only writeups I've seen so far on the oversampling didn't actually say we could hear way up there, but they also didn't mention any transient response things that are bonafide proven, like the ability to hear 20 uSec L-R delay changes. Perhaps they should..
Cheers, John
20usec is 50Khz.In order to accurately preserve timing info,
does that mean 10x... so 500Khz?How many cables have good characteristics up
in this range?
I wonder what the entire audio chain looks like viewed from this perspective?pretty ugly...
Peter
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