If you really mean that you can generate new frequencies, or shift frequencies, with a linear circuit, better get your hat and tails ready to go to Sweden.
Are you saying sidebands cannot be created without a nonlinearlty? I've looked for, but can't find, evidence to the contrary.
Did you find any evidence of envelope modulation in musical instruments creating sidebands?I am surprised that I could not find any published literature on the possibility of modulation sidebands hitting the brick wall. Lord knows, I did an awful lot of googling to find it..
Derfy,
High frequency loss depends greatly on the humidity in the air. I suspect it is less than 2 dB in most home reproduction systems.
What I have mentioned a few times before is that the original CD A/D converters were 9 bit linear with a 7 bit tail. The idea was to avoid zero crossing issues.
BTY I was at the AES convention when the CD standard was passed. I didn't run into anyone who thought the high frequency limit was wonderful. The attitude was we got a standard passed that folks could actually build.
Virtually all the original converters used multistage LC filters. I tried looking up the data sheet for the single can multipole filters that were mass produced for the early CD players. No luck, maybe the amazing George can do it.
Also to confuse the issue there are small children who respond to dog whistles!
Why waste time on folks with fixed ideas?
My thrust was more in terms the live-to-reproduced simulation. Namely, let's assume an extremely intimate venue/absolutely front row seats where you're sitting 10 m from the performers: propagation loss @ 20 C / 45% RH at 20 kHz is 5.5 dB; 30 kHz is 9.5, which I think is safe to call a substantial attenuation. Close mic'ing, unless patched through an appropriate set of filters which (drumroll) roll off the higher frequencies, like, say, an antialiasing filter.
The greater point I'm making is that for any of a hundred thousand reasons stereo playback is its own highly processed medium, not remotely an accurate simulation of anyone's prior experience. So there's a practical question of whether this entire dragged out back and forth has the slightest grounding in even one person's experience. That is not to say that one could prefer a certain type of synthesized music that pushes the limits of what can be recreated in 44.1/16.
None of this argues against the value of higher sampling rates in the transfer medium, as it gives more room between fs/2 and where music/humanly perceptible audio peters out, but it'd be really really nice if people could step away from their perfect mathematical transforms about fairies dancing on a pin to say whether this has a smidgen of practical value. Not even going to lengths of an audibility test -- close mic'ed with wideband microphones and high sampling rate, bang a cymbal or triangle, what's lost in the downsample to 44.1/16?
The science and understanding of modulation predates the birth of everybody on this thread.
But you're getting modulation with a linear circuit. That's new, essentially self-contradictory by classic thought.
All the best,
Chris
Perhaps one way to envision it is through consideration of why "windowing" is used in an FFT. The various windowings used appear to me to be a form of amplitude modulation, giving the analyzed signal a "soft start" - and a soft end. Without windowing, the FFT will show frequencies that arent there in a continuous, steady state signal being analyzed.
The theory wants nothing above Fs. Since Fs is so low (in CDs), the filter is needed to attempt to eliminate the stuff above Fs since much music has stuff up there.
The filter can never be perfect and will introduce effects below Fs - phase perturbations, ripple, more depending on the filter used.
dave
What an odd ripple in the output. Looks to be about 3Khz.
I can't tell if it's something in your output, or a pixellation artifact. The other two do not exhibit anything like that.
The envelope is pretty slow as well.
Also, it's impossible to determine if the envelope modification is a consequence of the abrupt start, or it and modulation envelope slope.
That is why I specifically mentioned using a time mirrored exponential rise into a flat top, then exponential decay.
While you casually dismiss the alteration of the envelope by the filtering process, an audio researcher would probably not.
jn
I can't tell if it's something in your output, or a pixellation artifact. The other two do not exhibit anything like that.
The envelope is pretty slow as well.
Also, it's impossible to determine if the envelope modification is a consequence of the abrupt start, or it and modulation envelope slope.
That is why I specifically mentioned using a time mirrored exponential rise into a flat top, then exponential decay.
While you casually dismiss the alteration of the envelope by the filtering process, an audio researcher would probably not.
jn
The filter can never be perfect and will introduce effects below Fs - phase perturbations, ripple, more depending on the filter used.
This is simply not true but only a statement of limited available compute power.
But you're getting modulation with a linear circuit. That's new, essentially self-contradictory by classic thought.
All the best,
Chris
What linear circuit does a bell use when I strike it and it exhibits exponential decay?
Did you just use a strawman?
jn
What linear circuit does a bell use when I strike it and it exhibits exponential decay?
Why does a bell have to be linear? Not following you.
All the best,
Chris
I went one step further in my attempt to find a replacement circuit for the ESL63, this time with both transformers in place.
Again, any comment is welcome.
Hans
View attachment 821761
Here is some facinating info about the delay line inductors Quad 63 (and later) Delay Line Inductors let are around 2-3H air core inductors and quite lossy with a shorted coil inside.
I'll attempt a model for the delay line to see what it does. The model will be flawed for various reasons but still a start.
While you casually dismiss the alteration of the envelope by the filtering process, an audio researcher would probably not.
I tried some more tests with possibly more realistic signals. More importantly I actually listened to the 20k, 16k sum which had a 4k envelope and could hear nothing so I don't see how the presence or absence of the 24k could make a difference. I know at lower frequencies the signal envelope can be perceived.
If you really mean that you can generate new frequencies, or shift frequencies, with a linear circuit, better get your hat and tails ready to go to Sweden.
All the best fortune,
Chris
You are welcome JN :-D
I promise to invite you to hear my system 😉
//
Why does a bell have to be linear? Not following you.
All the best,
Chris
It's the struck part he's alluding to; in steady state your ideal bell will have its fundamental and harmonics, but being struck we have a semi-step and decay modulation pattern multiplied (NOT added, where we'd see no new frequencies) on top of it, hence the additional harmonics.
Convolution in time = multiplication in frequency (vice versa)
Here is some facinating info about the delay line inductors Quad 63 (and later) Delay Line Inductors let are around 2-3H air core inductors and quite lossy with a shorted coil inside.
I'll attempt a model for the delay line to see what it does. The model will be flawed for various reasons but still a start.
Thank you for the link.
Hans
I tried some more tests with possibly more realistic signals. More importantly I actually listened to the 20k, 16k sum which had a 4k envelope and could hear nothing so I don't see how the presence or absence of the 24k could make a difference. I know at lower frequencies the signal envelope can be perceived.
I don't think I could hear that high either.
I also don't know how I would distinguish between a real cymbal in a room, one with CD brick filtering, and a hi res one. Apparently some believe they can.
jn
Some instruments are certainly nonlinear, I'd be fascinated to see the sidebands generated but the envelopes of different instruments. I can't find anything at all about it searching on line. The challenge, of course, with bells and cymbals, for example, would be separating the sidebands from the complex spectra.Why does a bell have to be linear? Not following you.
All the best,
Chris
I'm confident the closest I will get to the King of Sweden is a picture of him.You are welcome JN :-D
I promise to invite you to hear my system 😉
//
One day, I walked up to a women staring at a Nobel Prize picture, asked her if I could be of any assistance, answer any question...
She said, no, thank you. I was just looking at the prince, he grew up so much since I was there..
me:===>😱😱😱
jn
This confirms that the 4k beat envelope (clearly visible in time domain but does not show up in FFT) is inaudible.... I actually listened to the 20k, 16k sum which had a 4k envelope and could hear nothing ...
Hi everybody! From what I gather in this discussion, CD (might) be compromised by difficult audio signals interacting with the anti-aliasing filter that is used, somewhere in the chain. How we fix this is to use 24-96K or even higher resolution recordings. Many quality recordings are made like this today, and that is what we seriously use for audio component evaluation or alternatively, a really classy vinyl playback system with at least a $2500 MC phono pickup. Everything else is too compromised to evaluate hi end audio systems.
- Home
- Member Areas
- The Lounge
- The Black Hole......