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Alécio Costa 2 January 2003

Hi friends!
I had previosly posted this issue here and at PSWbut up to now, no one had come with a very good explanation. You can talk about polos, zeroes, transfer function, delay group, phase distortion and so. I will do all that I can to understand properly this... here itgoes..
Whenever I apply HPF's to digital equalizer like Q10 ( Waves Q series), I sometimes face a nasty distorion no matter if I am applying up to -6d uts at some bands an that the input signal is properly under 0 db.
some talk about distortion phase that HPF's add to the transfer function and so..
So, let us see if now we get a better in deep explanation
Thanks again
Nice 2003 to all
:)

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audiowkstation Thu, 01/02/2003 - 19:06

This is what is happening. When you EQ, you are changing the shapes of the interactions of the waves. To change the shapes properly, the upper harmonics that will be generated will be upwards of 30KhZ and 44.1 is a brick wall at 22,050HZ.

This brick wall filtering is bad enough for one channel.

As the wave shapes are formed with upper harmonics in mind during the render, the inability of them becoming formed causes a DC shift in the audible waves. IT will actually throw them off center and shape the zero point slightly forward of where it should be. The folowing zero point actually trys to move back and this is what causes the DC shift.

Like I say, it is bad enough for 1 channel but get 32 of them running and this spells trouble. One way to avoid this is to use very small increases or decreases per render, usually a 1/2 octave below or above the fundamental. Doing this render several times will bunch them up again so the best thing to do is track more properly and stay away from the C-10 if possable.

Next thing you can do is use a higher sampling freqency that will allow for more upper harmonics. You will still have the problem of adding tracks so that is one reason I do my critical equalization with the room, microphone or the source.

I hope this helps.
Their are other factors too as the math cannot allow even a -10 at times.

audiowkstation Thu, 01/02/2003 - 19:41

That is right Bob!

This "ringing", is the "overshoot, trying to correct itself"

That is one reason I loath 12,18 and 24dB/octave slopes on loudspeaker crossovers, passive or active.

I even designed a 3dB per octave slope for a 2 way once because of this ringing and overshoot.. (simple 6dB audibility)and it always seems to crawl up the odd order harmonics as well. Very audible. It is odd order intermodulation distortion PLUS and VS harmonic distortion.

I like to call it nastyness.

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