Funniest snake oil theories

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Using woofer towers (my opinion) or dipole woofers (Linkwitz's opinion) is likely to reduce boominess due to room acoustic problems.

Woofer tower? If single woofer, it means the position is far from floor. If multiple driver, it is in one enclosure so no better than separate subs where you can position each of them more flexibly to cancel some room modes.
 
One other question someone could ask: did the generator have the usual 50 ohms source impedance? Did the spectrum analyzer have the customary 50 ohms input impedance?

Jan
Of course, yes. It can be selected for 600 Ohm output impedance or 50 Ohm impedance. The spectrum analyzer also have 50 ohms input impedance.
Edited: I am almost forget the spectrum analyzer have two input impedance, high impedance and 50 Ohm impedance.
 
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maybe not as well, your perception of bass differs depending on the level of high frequency sound present, so possibly the output of the tweeters is different causing the bass to be perceived differently. The only way would be some form of measurement to see how much if any modulation is occurring, or go active and split the frequency range up as soon as possible to limit intermodulation.
 
maybe not as well, your perception of bass differs depending on the level of high frequency sound present, so possibly the output of the tweeters is different causing the bass to be perceived differently. The only way would be some form of measurement to see how much if any modulation is occurring, or go active and split the frequency range up as soon as possible to limit intermodulation.

Actually, my system I biamped, with each driver separately amped and passively filtered at line level.
Eventually figured this is easier to swap out drivers without re doing the speaker level filters.
And I did crudely level match tweeter comparisons, with warble test tones and a radio shack sound pressure level meter.
Still believe the harmonics sum together; this is really only obvious on acoustic instruments, far less so on synthesized or electric ones.
 
ANYWAY a new theorem has been pointed out to me...
Some Audiophiles can determine how a DAC will sound by looking at the digital square waves fed in to the DAC, it seems the shape of the square wave has a direct correlation to the resulting analogue output and how it sounds!!!!!!!!!

:scratch1::scratch1::scratch1:
 
ANYWAY a new theorem has been pointed out to me...
Some Audiophiles can determine how a DAC will sound by looking at the digital square waves fed in to the DAC, it seems the shape of the square wave has a direct correlation to the resulting analogue output and how it sounds!!!!!!!!!

:scratch1::scratch1::scratch1:

What digital square waves? :scratch1::scratch1::scratch1:
Digital signal accuracy is a function of time not shape
 
But many people can hear the "intermodulation" of high frequency with low frequency.

!!! Okay , that's the first reasonable explanation I've come across as to why my bass seems to change when tweeters are swapped. Some of that tweeter output must combine with the woofer for pleasant or unpleasant harmonics.

What you are hearing may not be intermodulation but a change in the perceived attack of the bass notes.
The attack is always a treble sound and you can change that by, for example, varying the eq on a bass drum hit in the kHz region. The main body of the sound remains the same as there aren't all that many harmonics produced by a bass drum, it merely changes the initial rise time.
If you boost the treble it sounds as if the drummer is playing harder and if you cut it will sound as if he is playing softer.
You could conceivably get a similar result from changing tweeters. Also lesser tweeters may well distort that short impulse.

In electronic music the attack is predetermined in the synth via the envelope generator. This will usually not change depending on how hard (loud) it is played, very much unlike acoustic instruments.
With synths you may also get something called a slew generator which reduces the rise time of any signal fed through it. It is basically just a specialized low pass filter.
 
ANYWAY a new theorem has been pointed out to me...
Some Audiophiles can determine how a DAC will sound by looking at the digital square waves fed in to the DAC, it seems the shape of the square wave has a direct correlation to the resulting analogue output and how it sounds!!!!!!!!!

:scratch1::scratch1::scratch1:
I cant imagine that there are any holes in that theory, That is valid, not snake oil at all.
 
I can see depending on the how the DAC input is built, rise time could effect temperature. Overshoot, ringing and other such analog issues ( digital is a concept, not physical) could effect various stages within the dac. Power distribution for one that may get imprinted on the analog output. Now, is the claim snake oil just because there is viable physics behind it? You would have to convince me that the dac system was correctly executed first. Poor layout or addressing the power supply would likely make the situation far worse.

I can see it now, all the reviewers coming up with "objective" tests with which they will ascribe garbage to nirvana depending on who bought the back page add.
 
I can see depending on the how the DAC input is built, rise time could effect temperature. Overshoot, ringing and other such analog issues ( digital is a concept, not physical) could effect various stages within the dac. Power distribution for one that may get imprinted on the analog output. Now, is the claim snake oil just because there is viable physics behind it? You would have to convince me that the dac system was correctly executed first. Poor layout or addressing the power supply would likely make the situation far worse.

I can see it now, all the reviewers coming up with "objective" tests with which they will ascribe garbage to nirvana depending on who bought the back page add.

Er we are talking digital, we are talking about looking at the digital waveform on a scope and from that you can correlate what the sound is going to be like, how come in my little world none of this ever comes up regarding digital design and layout, Ok we don't do consumer audio layout, but do pro audio analogue/digital layout, aerospace analogue/digital, medical analogue/digital layout, military analogue/digital layout, but we do follow digital design rules and procedures such as those put forward by Howard Johnson and others.
Now if you do not follow the digital design guidelines and information put out by the chip manufacturers, Howard Johnson, Henry Ott, Eric Bogatin etc. etc. you are likely to end up with something that is incorrect and may not work, but following audiophile layout beliefs and the forever chanting "well digital is analogue" you are probably guaranteed a failure.
But looking at the square wave going into a DAC will not correlate to the sound that is output.
 

OK thanks. And here is my opinion:

1) digital square waves is used to decode the analog signal. Inaccuracy can happen if there is too much drift in transmission time domain. In other words, inaccuracy will happen when there is certain amount or minimum jitter.

2) is there relationship between the shape of the digital square wave with jitter? I guess so. Bad shape is a cause of system imperfection. The same imperfection can promote data drift due to weak control on rise and fall times.

3) is jitter audible? If too much, of course. If a little? Then we need to find out what is the minimum amount to be audible.

4) to find this minimum we need (a) good ears. One may be able to hear what bad ears can't (b) resolving sound system, especially speaker.

5) My experience with reclocking. Long time ago my system was not as good as today. AND I didn't know if minimum jitter can give negative impact to sound. What I knew was that reclocking will improve things (and I knew more about digital theory then than now). So when I heard the result after reclocking, I got the feeling that I like the sound better before reclocking, so I took the time to restore my system to original setup. Please note, I expected an improvement and I didn't know if degradation was possible. So read between the lines.

6) if you want to do a jitter blind test, IMO I'm a good listener. I have good ears, the best I know, and I have very resolving speaker. You can consult Mooly and PMA who i believe can create test files with this synthetic jittery.
 
The funniest thing I've read to date is people in the audio community have stopped using their ears and are buying their audio gear based on 3 measurements:

-frequency response
-harmonic distortion of a 1kHz sinewave
-noise floor

A few pages in, this poster is defensive that if the THD and noise are below .005%, then it all sounds the same. Objective vs subjective thoughts
 
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