Graphic EQ: A Joke?

I thought digital EQ's (as in plugins) do not use phase reversal for filtering like analog EQ's do. Don't they simply lower amplitude values at given frequencies?

Besides, how can you consider phase to be a problem if it accomplishes the result you seek?

RawDepth
 
i've never heard of a linear phase eq in the analog.... and IIRC the phase issues in the digital are actually simulated and dont occur naturally... ya may want to google IIR and FIR filters ... cant remeber at the moment which is which but one does have the phasing... infinate impulse response and finite impulse response... for some reason i'm thinking its the fir that is more like analog...
 
I love my old TC Native EQ plugin. It's a "paragraphic". It's laid out like a graphic and works like one with the default settings, but you can change the freq., Q, and gain/boost on all the bands. Real easy to use.
 
I love my old TC Native EQ plugin. It's a "paragraphic". It's laid out like a graphic and works like one with the default settings, but you can change the freq., Q, and gain/boost on all the bands. Real easy to use.

Those are usually called "paragraphic eq's".
 
I use a Klark Teknik 1/3 octave eq that gives phenominal results. It is a good ole stand alone eq that I have had for over 15 years. Unbelievable shaping of sound.

I also use plugins and all my UAD stuff. All togeather the hardware and software based items can sound real good togeather.
 
most digital filters are based on the same principals as analogue ones, but there is a kind of filtering you can only do in digital (FIR) that has really good quality in some respects, I can't remember what tho...

It could be what you are hearing is that the graphic eq used a loeer res fft approach (another digital only process I'm sure) that didn't yield quality results.
 
Here's an EE-student's perspective, having taken classes both in continuous-time and discrete-time signal processing:


Any time you use any kind of frequency-response (more specifically amplitude response)-changing filter (analog or digital), you're going to alter the phase response (versus frequency) as well. It's not a function of imperfections in analog gear, or the lack thereof in digital signal processing, it's a function of the relationships between time-domain and frequency domain (both amplitude and phase versus frequency)

Don't confuse phase shifts with digital processing latency (even phase shifts they can and do cause, and are actually the very directly related to, time delays). They are very different things (though, as above, any time delay is invariably related to a phase shift, and vice versa)

If anyone wants to learn more about the relationship between time-domain, and frequency-domain representations of signals, do some reading up on Fourier Series/Transform, which relate time-domain to frequency-domain signal representations. Actually, knowing a little bit about fourier transforms is very useful in understanding a lot of things about audio - for example all the arguments about sample rate and Nyquist frequencies (although technically you'd want to look at the Z-transform for this discrete time discussion, but Fourier is pretty analogous)

Any time you're changing something in one domain, you're changing it in the other - changing the way a signal is in the time-domain (amplitude versus time) changes it in terms of amplitude versus frequency, and phase versus frequency

(of course some changes could presumably result in some of these things having a change of zero - i.e. change the frequency responses amplitude but not phase, but that's really not the issue here).


When we talk about linear phase filters, that doesn't mean that the phase shift if THE SAME over all frequencies, it means that the phase shift versus frequency is LINEAR - if you double the frequency, you double the phase shift, etc. If we shifted all frequencies by the same PHASE ANGLE, they'd still get all whack in their time relationships. The benefit of linear phase is that if you look at the appropriate transformation, this linear phase shift versus frequency corresponds to all frequencies being delayed in time by the same amount - the whole signal is just time-delayed. This can be shown by examining the Z-transform of a digital signal (The transform in discrete time - aka "sampled" - that corresponds to the Fourier transform in continuous time). We find that z^-1, inverse of the complex discrete-time frequency variable z, which is used in (one way of) describing a digital filter, is actually equal to a delay by one sample.

The ability to create a linear phase filter is one of the important benefits of digital filters. Another being the ability to create a filter with any number of poles or zeros by simply changing DSP software functions, rather than adding costly components.
 
At least it's english compared to what I had to learn about that from.... That's the downside of having teachers that write their own material... every thing sounds like a textbook, which never make sense. A few months ago I almost knew how to do that stuff... now, not so much.
 
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Graphics have their place. For instance the 512a or b is great for guitars. But... I like api's 560 used with drums is really cool. Just a different flavor of eq.
 
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