Archived from groups: rec.audio.tech (
More info?)
> I'm curious as to the technical differences between these new breed of
> amplifier circuits and standard. From what I've read, they seem to use
> digital pulsing to reproduce AC signal instead of true harmonic
> (sinusoidal) AC.
Here's a link to a technical description of how these amplifiers work:
http/users.ece.gatech.edu/~mleach/ece4435/f01/ClassD2.pdf
The so-called "digital" or class "D" amplifiers use pulse-width modulation
of a square wave that is then filtered to analog. The speaker gets an analog
signal, just like any other amplifier output. Pulse width modulation is
actually an analog process. A square pulse can have any width: It can
smoothly go from "always off" to "always on". The output filter "integrates
the area under the curve" [1] -- also an analog process.
The advantages of class "D" are very high efficiency (lower power
consumption and less heat) compared to traditional class "A", "AB" or "B"
amplifiers. The have been around in experimental form since the '70s, but
they seem to be gaining in popularity due to the large number of power
amplifiers required for multichannel surround sound. The primary
disadvantage, as far as I can tell, is class "D" switching amplifiers don't
attain the performance of the highest quality traditional designs, but they
might eventually.
This is a digression, but... The class "D" modulated output is actually very
similar to the unfiltered output of a single-bit sigma-delta D/A converter.
I don't know if anyone uses this D/A output signal directly as a raw signal
source for class "D" power amps. As you can see, the distinction between
analog and digital can become blurred, but that's not surprising when you
are converting between digital data and the end product, which must still be
analog sound.
[1] A complete coverage of integration is probably beyond the scope of this
NG, but if you're interested, see "integration" in any calculus text.