LAB5_03_09_2007

LAB5_03_09_2007 - ECE 3551 MICROCOMPUTER SYSTEMS 1 Lab...

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ECE 3551 MICROCOMPUTER SYSTEMS 1 Lab 5—Learn to process audio data OBJECTIVE: Learn how to process audio data in ADSP-BF533 EZ-Kit Lite. Learn how to design Finite Impulse Response ( fir ) filters with MATLAB fdatool. Learn how to implement digital fir filter in DSP using: 32-bit floating point emulation 16-bit integer arithmetic 16-bit fractional arithmetic Learn how to control the audio input/output. Create the Project 1. Before power on the board , make sure that switch SW9 pin1,pin2, pin3, pin4, pin5 and pin6 are turned on., which means all pins in SW9 must be on. 2. Copy the project of the previous completed Lab #3 to your u-drive “u:\ece3551\labs\lab5”. Modify the Project In this lab, you need to learn how to filter audio data and compare the original audio and the filtered audio. In this Lab exercise only stereo input 1 and stereo output 1 will be used. Read the data from the input and save it’s copy: one is the original data and another will 1 2 3 4 5 6
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be the filtered data. In the beginning, LED4 & LED5 are off, and the original data is send to output. So what is heard is the original audio. When PF8 is pressed: The LED4 should turn on, and The low-pass filtered data is send to output. What is heard therefore is filtered audio. By pushing PF8 again, the LED4 should turn off, and the original data is send to output; what is heard is the original audio again. When PF9 is pressed: The LED5 should turn on, and The high-pass filtered data is send to output. What is heard therefore is high-pass filtered audio. By pushing PF9 again, the LED5 should turn off, and the original data is send to output; what is heard is the original audio again. Designing FIR Filters: MATLAB exports FIR filters as a number of coefficients representing that are used to implement moving average (MA) of the input weighted by those coefficients. The general relationship of output and input given with a difference equation of the form: Note that this difference equation is equivalent to z transfer function of filter of the form: ( 29 ( 29 ( 29 M M z z z z X z Y z H - - - + + + + = = α 2 2 1 1 1 1 Again, note that z -1 denotes unit delay D operator and z -2 , denotes 2 unit delays, i.e, D 2 , etc. Y(z) is z-transform of the output, X(z) is z-transform of the input, M is the order of the filter. Contrasting FIR filter with IIR filter covered in the previous LAB one has to observe that the output is dependent only on the input for FIR filter (Moving Average part). Thus FIR filter as know in the literature as MA filters. For IIR filter the output is dependent on the input (Moving Average part of the filter) as well as the current and past values of the output (Auto Regressive part of the filter). IIR filters are known in literature as ARMA filters. For more information see PPT lecture notes Discrete-Time Signal Processing Framework2.ppt under:
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LAB5_03_09_2007 - ECE 3551 MICROCOMPUTER SYSTEMS 1 Lab...

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