> I can't see where there's a difference.
>
> When searching for references to the srticle I remembered, I was struck by
> how many top end broadcast audio processors use split band clipping and
> limiting.
>
> 73, Steve
This is certainly true, some of the more up to date ones split the frequency
into five bands and then perform adaptive compression with different time
constants for each frequency band. They are able to give 12dB of increased
average level without audible artefacts on full bandwidth music signals.
They are implemented in DSP and have a slight 'look ahead' delay
(<4ms)incorporated to prevent peak overshoots. As a square wave can be
described as a Fourier series of sine waves then it seems logical that
removal of harmonics by a filter can indeed restore the original waveform,
if the odd harmonics are removed then it's no longer a square wave. Because
of the limited bandwidth of an SSB transmission a two band approach is more
than adequate and should provide excellent results if implemented properly.
I may just make one using a DSP sometime soon, it's easy enough these days
using Matlab. No programming required, just select the appropriate DSP
modules, set the parameters, see how it behaves and if you like it tell
simulink to write it to a TI DSP. You can tailor the response to exactly
what you want. There is no disadvantage to doing it at audio or baseband
provided it's implemented properly.
For those interested in the BC products look at TC Electronics' website.
Regards
Conrad G0RUZ
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