原创 Designing an Equalizer using a multiple feedback bandpass filter

Let’s take the example of the CD. How do you think the information was stored in the CD itself? It would have been saved as a .wav file on the CD or even on your mp3 player. The data would be in a series of digital bits. The song was originally sung by a singer in a recording studio. Then an equalizer was used to modify the frequency components of the signal. The signal was sampled, quantized and finally modulated using a technique called pulse code modulation.

Pretty heavy, huh? I guess so. So let’s take it one part at a time. Let’s see what an equalizer is. Simply put, an equalizer is a device which varies the amplitude of a select group of frequencies thus altering the quality of the sound. We’ll see a simple example. You may have seen the graphic equalizer on any media player in your PC. It looks like a set of sliders which can be moved up and down. Each of those sliders corresponds to a frequency and moving the slider will set how much of that frequency you want in the sound that is playing. Suppose you want to hear a better bass effect, you may have to vary the amplitude of the 63 Hz signal. Similarly for many other frequencies. Like this the equalizer in the recording studio will change the quality of the signal (the amplitude of some of the frequency components). The major component of our equalizer is the multiple feedback bandpass filter (MFB filter) which was discussed in my previous blog.

It’s not that complicated as it looks. As you can see I have divided it into three parts – A, B and C.

Part A: Let’s say we are recording a person singing. The microphone converts the sound wave into a voltage. This voltage signal will be a sum of different signals of different frequencies like the voice, and the various musical instruments. This is simulated by this adder circuit where five sine waves of different frequencies are added. In real life the signals need not be pure sine waves. But let them be sine waves in this simulation.

Part B: This part is made of the filters we discussed earlier. Each filter will extract one of the signals so their gain can be controlled. In our example let’s consider only two frequencies – 250 Hz and 1 kHz. Thus by adjusting the resistors R4 and R6 we can adjust the gain of each of the filters, in turn controlling the amplitude of each frequency in the signal.

Part C: This part is a summer circuit again. The frequency components that were varied must be added to the main signal with the other components. This is accomplished by this adder. So in a nutshell, this is what happens: The required frequency components are separated from the input signal, altered and added back to the audio signal. Thus the signal at the output will be a modified version of the input.

To see how the filtering takes place, see the outputs at the voltmeters Output_250 and Output_1k and calculate their frequencies. In addition you can also run a frequency analysis for the circuit and observe how the amplitude varies for the various frequencies present in the signal. Adjust the gain of any of the two frequencies and see the changes in the frequency response. It’s possible to get the response equal throughout all frequencies (increase the resistances R3 and R11 to get higher gain). And hence the name “equalizer”.

To design it yourself, please visit:

Graphic Equalizer:

http://www.docircuits.com/circuit-editor/116