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Old 19th August 2009 | Show parent
  #15
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Quote:
Originally Posted by Paul Frindle View Post
Yes indeed - this effectively solves the issue :-)



Ok this is where it gets tricky. Soon as you send your stuff to the mastering stage current fashion dictates that they will once again push the levels to the absolute maximum. That's apparently all anyone wants these days - even though in many cases it's creates illegal program that will sound awful on the end user's equipment - and all bets are off as to what will happen. But listening to almost any popular music CD these days reveals the results - you can hear them on any cheap ghetto blaster :-( One could be forgiven for concluding that all this 'mastering' level war is killing our whole industry; but that's another discussion :-(

You can use something like the Oxford limiter that will detect and effectively fix up the peaks by reducing levels momentarily only during the exact places they happen. This is better than simply turning everything down as the loudness in not affected when the IS peaks don't happen.

But even this will fail if at a later stage the mastering process does anything at all with the program - quite simply the IS peaks can (and will) come back again with EQ, further compression, limiting or almost anything else. This is because the current trend in mastering leaves almost no headroom or dynamic range - a reduction of -0.5dB taken grudgingly and daringly by the mastering engineer in the hope he won't lose his job is just not enough :-(

Ok - so then this gets us back to the subject of the PSW thread. Given that mastering (and mixing) these days is all about actually creating and then handling what are effectively self-generated errors in the signal itself, how these respond and sound can vary considerably depending on how the gear handles these errors.

So for instance one way of dealing with it is to chose a suitable DAC (probably expensive) that does not not crack up too badly with the errors - feed the result to a load of analogue gear (that has headroom - because they can't be made any other way) - then re-coding the whole thing back to digital using an ADC (probably expensive) which will skim off the IS peaks (as it cannot do otherwise) and sound appropriately 'not too bad' as it does so.

However all of the above messing with signal paths to handle self-generated errors of course completely denies us all (musicians, engineers and consumers) the amazing potential advantages of an accurate and repeatable digital signal chain, which is of course that it can (or at least could if implemented correctly) transmit signals without any audible errors or degradation - right into the end users reproduction equipment.. That was supposed to be it's big advantage :-)
Question:

Would a LP filter set to near or below (20k) the nyquist frequency possibly reveal any clipping which may occur from the reconstruction filter?