How Science Got Sound Wrong

I found it interesting.

I find Analog LP's (lots of old analog record cutter produced ones) and Reel to Reel, (analog mic to analog tape) more INVOLVING than Digital.

I have simplified the difference as "Analog gets Overtones Right". 

I wish the preserved timing advantages of analog, and the resultant timing of overtones was discussed.
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I agree about precise positioning of speakers.

Happily I have an old wood floor, like grid paper on the floor, which allows me to move them into a few 'situational' positions without a ruler, including matched toe-in. I squished a speck of paper into the grid for the 2 front speaker corners.

They are very heavy, on 3 wheels, (3 will always settle with no wobble, and their weight prevents any vibration, no spikes needed for these). 

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I also believe in time alignment of various frequencies, so I tilted my speakers base a bit back (also changes reflections off both floor and ceiling, and resultant back wall).

And I use the floor's wood grid as an assist for listening chair(s) positioning, as those chairs are moved 'situationally' also, turning around to be part of home theater, back around for music system, centered for 1, off center for two. I also have a vase on the window sill perfectly centered, which helps center my head when sitting this way or that, and gives the brain a center before/during listening.

This setup is how I found that very slight balance adjustments can make a big difference on certain tracks. Remote balance from listening position allows refinement, track to track. I wish my integrated amp had remote balance. I use my Chase Remote Line Controller now.

Interesting read.....thank you for posting and sharing!

Thank you David. I will have to think about that.

As for the Nyquist-Shannon theorem, yes, I am familiar with it, and am not convinced that it says what some engineers think it does. For one thing, it involves a limit in terms of an infinite series (or integration over infinite time), and infinite time is available for relatively few signals.

My reference, A Handbook of Fourier Theorems by Champeney (Cambridge University Press), is a little too dense for casual reading, but I’ll persevere for a time.

Thanks again for the discussion, and also for re-igniting an interest in that branch of mathematics.

terry9,

Excellent catch on infinite series, but also easily addressed. As we are dealing with audio, there is effectively no information below 10Hz, and some would argue 20, but let’s say 10Hz. For that reason, any real single data set, i.e. a song file, can be modelled as an infinite series as there is a maximum rise time and minimum fall time at beginning and end, hence you can "set" all data outside to 0 (whatever your 0 is) for all points when applying the theorem. Any "errors" in bit level would be in the silence at the beginning and end of the track. In some ways, this is like a natural windowing function.

There are lots of papers, proofs, course books, material, etc. that goes into detail, including size of error when you don’t have an infinite series, which in a practical audio case, would be much smaller than other error sources.

If you want to play with "math", GNU Octave is a free-ware version of Mathcad (not as graphical) and would let you simulate any of these concepts.