"Jim Lesurf" wrote in message
...
In article , Peter Scott
wrote:

I'm afraid it didn't make much sense to me given the parts I snipped this
time, nor did I agree with what Wally wrote. :-)

Clipping after the amplifying stage does not require the amplifier to
produce the frequencies.

Alas, the output stage of an amplifier *is* part of the amplifier, not
'after' it. Nor is it invariable that clipping only occurs at the output
stage. Depends on the design, etc. Also, the components created by
clipping *won't* always be at higher frequencies than those in the input.
Clipping 'musical' waveforms can lead to unwanted components at all sorts
of frequencies, many of which are lower than, or comparable with, those in
the input waveform.

Yes the new waveform might well have new low frequency components
needed to synthesise it. But I come back the point about the hf. Surely
if the clipping is caused by effects in the output transistors, that drive
them
into non-linearity, it is in effect after the amp stages? Otherwise how
could
components be generated that lie outside the range of the amp. Or is it
that fourier doesn't come into it. The fact that you need this range of
frequencies to synthesise the wave doesn't mean that the amp has to produce
them. All that is needed is a slew rate that is at least equal to the
steepest
slope on the wave. As you can tell I'm struggling a bit with this but I do
want to understand it. I had exactly the same problem with discrete
cosine transformation when learning about jpeg and mp3.

Hence you would need an 'adaptive' filter that always knew what components
to pass and what to reject, right across the audio band. The usual
arrangement for this is - an amplifier that is not clipping in the first
place. :-)

Definitely. Hence I am now buying power amps with oodles of power. Max
Headroom lives!

Peter Scott