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Final Project(1)

Final Project(1) - Final Project Digital Equalizer Class...

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Final Project: Digital Equalizer Class: ECE - 3551 Submitted on: December, 12, 2011 Submitted by: John Ernsberger GSA: Jacob Zurasky Purpose: The purpose of this project is to design a fixed and user controlled digital equalizer, where the user can increase the amplitude of individual frequency ranges or choose a predetermined setting. Theory: Digital Equalizer: A Digital equalizer works by running an audio input through several band pass filters. These filters are designed to only allow a certain frequency range to pass through them. Thus, when that range is multiplied by a scalar It augments the frequencies only within that filter. Because of the specific frequencies needed for the filtering second order IIR Filters are necessary. If you use FIR filters, although easier to implement, the coefficient level is far too high to be practical for the CPU to handle. Thus a much lower order is needed making IIR filters a prime selection. These filters have a low coefficient count and are extremely precise. Below is a visual representation of how the equalizer works. First the input is passed through several Band Pass filters. The output of those filters is multiplied by an individual gain and summed. The summed augmented outputs of the filters are then multiplied by a master gain, which is 1/( the number of filters ). To discover the individual gains for your filters you can use a db calculator. The maximum increase is around 12 db or multiplying your filter by four. This is because anymore
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amplification will result in too much energy in the system and the quality of the music will decrease. Second Order IIR Filters: Second Order IIR filters are very useful as they are precise and have a very low coefficient count. However, the implementation of the code is more complex than FIR filters. The equation used to calculate second order IIR filters is : This equation can be implemented as code using the following two equations: wn = ( input ) - ( a1 * wn1 ) - ( a2 * wn2 ) ; yn = ( b1 * wn ) + ( b2 * wn1 ) + ( b3 * wn2 ) ; where the wn is the current modified input value and wn1 and wn2 are the past modified inputs. To complete the filtering operations you must pass the wn1 to wn2 and wn to wn1 thus allowing the past two modified inputs to be recalled for later use within the filter. Method: 1. Generate second order IIR pass band filters using MATLAB for the following frequencies. a. 100 - 200 Hz b. 200 - 400 Hz c. 400 - 800 Hz d. 800 - 1,500 Hz e. 1,500 - 3,000 Hz f. 3,000 - 5,000 Hz g. 5,000 - 7,000 Hz h. 7,000 - 10,000 Hz i. 10,000 - 15,000 Hz 2. For the second fixed equalizer create second order IIR filters for the following frequency ranges. a.
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Final Project(1) - Final Project Digital Equalizer Class...

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