John Curl's Blowtorch preamplifier part II

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I quoted from a book. Of course, 6dB/octave is very slow, AND at ultrasonic frequencies, perhaps the negative effect described by Slot, goes away, BUT then I did not select 44.1KHz either, neither did Ampex, Philips, or anybody else, BUT Sony. Most everyone else started with 50KHz, and hoped to go higher. This made the filters easier to make, and gave the ears a potential break.
 
I quoted from a book. Of course, 6dB/octave is very slow, AND at ultrasonic frequencies, perhaps the negative effect described by Slot, goes away, BUT then I did not select 44.1KHz either, neither did Ampex, Philips, or anybody else, BUT Sony. Most everyone else started with 50KHz, and hoped to go higher. This made the filters easier to make, and gave the ears a potential break.
Yes, and 60 kHz would have provided a nice response comparable to professional tape. We really should leave CD behind in the history books - in selecting that standard, so many compromises were made based on short-term cost limitations.
 
I would again like to comment about the 'essence' of quality audio. We appear to be making progress in digital 'compromises' thanks to T and PMA, which might be the making of a technical paper in future by somebody. It is important to be able to tell the difference between specs. and performance. Sometimes, specs. can be misleading.
Let me give a parallel example:
Almost 60 years ago, I got a Tasco (Japanese) 60mm refractor telescope for Christmas. I was overjoyed, because at the time, I had hoped to be an astronomer (I still sometimes wish I had gone in that direction). Of course, this small telescope had limitations, BUT not as many as you would think. I saw the normal stuff, but sometimes, with a little effort, I could resolve what I was told that a scope this size could not resolve. It is no secret, in retrospect, why. This little scope was well made, perhaps by people who made military optics during WW2. In any case, it really worked to the extent of its specifications and it gave me years of enjoyment. Had it not been stolen, perhaps 5 years later, I would have continued to use it, until something bigger and better became affordable.
Well, years later, I decided to buy another Tasco telescope, much like the one I lost. I found one, seemed to work, BUT it just was not the same quality, YET it had the same optical specifications. I tried several Tasco telescopes, over the years, and settled with a slightly bigger one, 90mm, and it's a 'dog' too! Why? I should have had an optical advantage with the somewhat larger scope. Well, specs. are specs. and user quality is above and beyond the specs. It is the same with audio. A well maintained Dyna Stereo 70, coupled to the right speakers can operate beyond its 'specifications' in sound quality, I have heard it, myself, with another audio professional's systems. Of course, you have to work within its limitations, but with a pair of LS3-5A's it was awesome! Yet, I make amps with up to 20 times more power, almost effortlessly, but I would not have chosen a JC-1 pair of amps in this set-up. Of course, the guy's wife worked full time for Reference Recordings, etc., etc. so they had confidence in their trade-offs, and pulled it off! Enough for now.

Now we shall think over why we cannot produce products in the same quality like 60 years ago, but we have a so much better technical capabilities (CAD / CAM etc.) ...

Sensibility of human craftmanship cannot be implemented into machines.
 
rsradio said:
I'm not saying that a 1st-order filter wouldn't work, I'm merely stating that it's corner frequency would have to be sufficiently below 22 kHz in order to suppress a reasonable amount of the aliasing.
Are we talking about an anti-aliasing filter before the ADC or a reconstruction filter (to suppress images) after the DAC? A first-order anti-aliasing filter would be about as useful as a chocolate teaspoon. The NOS crowd will claim that a first-order reconstruction filter is overkill as their amp, speakers and ears do the job OK unassisted.
 
rs, thanks for the explanation. I looked at the spectra of a few random CDs I had ripped and can't find any significant energy over the Nyquist frequency. So if the phenomenon exists, I'd guess it's transient rather than systematic?
You talkin' to me? :)
You can't really see aliasing on the CD itself. Your description doesn't say whether you're analyzing the data on the disc or the analog output of your DAC.
What I described was my 48 kHz recordings of 44.1 kHz DACs being operated by musicians on a budget. It's impossible for 44.1 kHz data (CD) to contain information above Nyquist, but once the data is converted to analog by a particular DAC circuit you might easily find images. What I saw was completely different from one act to the next, even on the same sound system, because each artist brings different equipment. Most DAC gear, even some of the cheaper stuff, does fine when it comes to anti-aliasing the output, but a few times there are problems. This is probably most prevalent with those MP3 DJ softwares where a vinyl time code disk is used on a real turntable to alter the playback rate of the MP3, and the resulting digital resampling creates all sorts of mayhem, both in the data and in the converter hardware.

By the way, when looking at HD discs, i.e., media beyond CD, it's actually more common to see aliasing within the data itself. I've seen quite a few examples of 96 kHz "HD" audio where it was really nothing more than 44.1 kHz data plus aliased frequencies above 22.05 kHz. Disgusting and unforgivable!
 
Are we talking about an anti-aliasing filter before the ADC or a reconstruction filter (to suppress images) after the DAC? A first-order anti-aliasing filter would be about as useful as a chocolate teaspoon. The NOS crowd will claim that a first-order reconstruction filter is overkill as their amp, speakers and ears do the job OK unassisted.
The same filter is required both before ADC and after DAC. It doesn't really matter whether it's called an anti-alias filter or a reconstruction filter. At least for NOS they have to both be as sharp, because a DAC has aliased frequencies starting right at 22.05 kHz.

An oversampling DAC can certainly employ a low-order filter, but it would probably require many times oversampling to enable first-order to work (not to mention a steep digital filter before the DAC).
 
rsradio said:
The same filter is required both before ADC and after DAC. It doesn't really matter whether it's called an anti-alias filter or a reconstruction filter. At least for NOS they have to both be as sharp, because a DAC has aliased frequencies starting right at 22.05 kHz.
The name does matter, because the name identifies the function. One filter removes aliases, which are spurious components arising within the normal audio band from any signal components above 22.05kHz. These, if present, are likely to be audible and unpleasant. The other filter removes images, which are spurious components arising above the normal audio band which may or may not be audible and may or may not be unpleasant.

Both filters have to be sharp if the aim is to faithfully reproduce the band-limited version of the analogue signal. People differ on whether this should be the aim. It is clear that in some quarters, but hopefully not here, people are confused and blame the reconstruction filter for issues which are actually (and possibly unavoidably) caused by the anti-alasing filter. Maintaining the distinction between these two filters is thus a vital aid to clarity of thought and expression.
 
biological variations don't necessarily (or even often) result in smooth, symmetric, "infinite tail" distributions

assuming Gaussian distribution may be the "default" for some "canned" statistics packages but that doesn't make it predictive of real world data distributions

Although in psychoacoustic it quite often seems to be a normal distribution and in the case of the graph from Zwicker/Fastl the authors explicitely mentioned the 5% - 95% range which is normally a good hint for the assumption of a underlying normal distribution.

But if you want to use data from a small sample to draw conclusions wrt to the whole population, you _have_ to assume an underlying distribution otherwise a conclusion isn´t possible.
That is the difference between descriptive and interference statistics.

My objection is the that some want to "assume their conclusions" as a method of argument, "proof"

Let us concentrate on the arguments. If these are correct there is little room left.

to me the data in peer reviewed papers is not yet conclusive, 44.1 kHz is not "obviously" inadequate for Music reproduction
many accepted measurements, detailed models for human auditory perception, even typical recording, playback equipement limits make it plausible that content above 20 kHz isn't of 1st order importance....

I totally agree, but of course other people might percept it different (due to interindividual differences).
And that it is not "unlistenable" should not prevent us from doing better if possible .

.....small "time difference" resolution numbers, IATD, keep getting misinterpreted as implying channel bandwidth of ~ 1/dt - this has been pointed out as logically wrong many times now
see Kunchur's slide: (my Paint underlines, arrows)

If am not totally mistaken, Kunchur does not work with IATDs .
Your underlined sentence is exactly the essence of his studies, that with diotic signals the temporal resolution is higher than the upper frequency bound would predict.
 
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I meant IATD as an additional example of "unexpectedly" small time resolution numbers that people naively want to use to imply 100-200 kHz "hearing"

Kunchur's models don't require modifying "conventional" psychoacoustic understanding of high frequency sensitivity - he is ascribing the increased time resolution to the result of a convolution like process from phase locking of frequency content totally within conventional < 20 kHz human hearing limits

His explanation of filtered 7 kHz square wave results do appeal to possible 21 k
Hz nonlinear mixing, 14 kHz in band IMD – but not to direct sensitivity of hearing the 21 kHz fundamental


On the Slot ref – “sharpness” in psychoacoustics is correlated with increased high frequency content – rising sensitivity with frequency above ~ 3 kHz
So sharpness sensitivity to filter order for corner frequencies below our upper frequency hearing limit is “conventional” psychoacoustics – care to post what filter corner frequencies he was citing?
 
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