I have been attempting to design a 15 band audio equalizer using Simulink. My
design right now is as follows (please let me know if there are any significant
design flaws here):

-Audio signal goes into preamp.

-Preamp output is split into 15 branches, 1 for each filter.

-Each branch contains a BPF, followed by a gain block.

-The 15 branches are summed, and that is my output.

The output sounds decent, except for low frequencies seem quite distorted, so I
started to do some trouble shooting. Checked out the phase for each filter...
they all appear to be approximately linear within the passband, and what happens
to the phase after the passband is irrelevant, right?

Next, I noticed something a little more troubling with the low frequency filters
don't nearly meet the passband edge frequency characteristics I specify, since
the sampling frequency must be 44100-Hz. I guess my main question is how should
I design the filters such that their magnitude specifications are properly met,
using such a high sampling frequency? Is there a better design method to use for
audio equalizers?

Also, can someone explain Q values a little better for me, or point me to a good
reference? From what I've read, a constant Q value throughout the equalizer is
the best design, but trying to implement various constant Q values made my
current design sound worse.

# Audio Equalizer Filter design

Started by ●April 19, 2011

Reply by ●April 19, 20112011-04-19

Olsen-

Audio equalizers are a complex subject. I know people who have spent chunks of their careers developing AEs with

better performance, with very low power consumption (suitable for embedded systems), that equalize based on an

arbitrary room response, etc.

So what you're doing is not easy. Suggest that you do lots of online research and find other academic projects. In

particular, you might search on parametric filters.

With that said, what type of filters are you using? If you want high quality low frequency filters, you will need to

use IIR filters. Also lower frequency IIR filters will require high numerical precision to be stable.

-Jeff

> I have been attempting to design a 15 band audio equalizer using Simulink. My design right now is as follows (please

> let me know if there are any significant design flaws here):

>

> -Audio signal goes into preamp.

> -Preamp output is split into 15 branches, 1 for each filter.

> -Each branch contains a BPF, followed by a gain block.

> -The 15 branches are summed, and that is my output.

>

> The output sounds decent, except for low frequencies seem quite distorted, so I started to do some trouble shooting.

> Checked out the phase for each filter... they all appear to be approximately linear within the passband, and what

> happens to the phase after the passband is irrelevant, right?

>

> Next, I noticed something a little more troubling with the low frequency filters don't nearly meet the passband edge

> frequency characteristics I specify, since the sampling frequency must be 44100-Hz. I guess my main question is how

> should I design the filters such that their magnitude specifications are properly met, using such a high sampling

> frequency? Is there a better design method to use for audio equalizers?

>

> Also, can someone explain Q values a little better for me, or point me to a good reference? From what I've read, a

> constant Q value throughout the equalizer is the best design, but trying to implement various constant Q values made

> my current design sound worse.

Audio equalizers are a complex subject. I know people who have spent chunks of their careers developing AEs with

better performance, with very low power consumption (suitable for embedded systems), that equalize based on an

arbitrary room response, etc.

So what you're doing is not easy. Suggest that you do lots of online research and find other academic projects. In

particular, you might search on parametric filters.

With that said, what type of filters are you using? If you want high quality low frequency filters, you will need to

use IIR filters. Also lower frequency IIR filters will require high numerical precision to be stable.

-Jeff

> I have been attempting to design a 15 band audio equalizer using Simulink. My design right now is as follows (please

> let me know if there are any significant design flaws here):

>

> -Audio signal goes into preamp.

> -Preamp output is split into 15 branches, 1 for each filter.

> -Each branch contains a BPF, followed by a gain block.

> -The 15 branches are summed, and that is my output.

>

> The output sounds decent, except for low frequencies seem quite distorted, so I started to do some trouble shooting.

> Checked out the phase for each filter... they all appear to be approximately linear within the passband, and what

> happens to the phase after the passband is irrelevant, right?

>

> Next, I noticed something a little more troubling with the low frequency filters don't nearly meet the passband edge

> frequency characteristics I specify, since the sampling frequency must be 44100-Hz. I guess my main question is how

> should I design the filters such that their magnitude specifications are properly met, using such a high sampling

> frequency? Is there a better design method to use for audio equalizers?

>

> Also, can someone explain Q values a little better for me, or point me to a good reference? From what I've read, a

> constant Q value throughout the equalizer is the best design, but trying to implement various constant Q values made

> my current design sound worse.

Reply by ●April 26, 20112011-04-26

I found a few years ago a
technique for making high performance bell curves for the implementation of
parametric or graphic EQ. The technique uses an all pass filter whose output is
added or subtracted with the source signal.

I don't recall details,
but used the method on code for a Freescale 56311, Even at
low frequencies and high Q, it accurately matched Matlab
simulations.

For references, search
for Dana Massie and Sanjit Mitra (coverage n a text text book).

I believe the method is
also good from the standpoints of coefficient sensitivity, noise, and simplicity
of parameter variation.

To use for a graphic, you
will cascade the band sections, not parallel them.

Chris
Moore

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