Robin Blandford [ ByteSurgery.com - Digital Media Engineering ]

26/11/05 Digital Signal Processing - Tough Week

This week could be summed up as the week that saw ridiculously low amounts of sleep. Last weekend ByteSurgery saw a huge job through over a Fri-Mon deadline, and will be profiled here on its launch day. Many thanks go to Jon Hanlon for his excellent flash work and short-notice contracting!

Following this, a DCU assignment in Digital Signal Processing was carried out to construct a Low Pass Finite Impulse Response Filter implementation in both C and assembly language and hardware version. After a week of hard work and sweat; the [print] button was pressed at 5:35am on Thursday night and delivered the next morning to the assignment box. Thanks to the genius of the web and Adobe Acrobat, I can give you a copy to read of the 44 page documentation and source-code for those interested. While this might require the lecturer to change it for next year’s class, I don’t think it’s against any rules to publish my work, but please correct me if anyone knows better :-)

Assignment Specification:

To design a ‘FIR Low Pass Filter’ in the C programming language. The program should ask for the number of taps, the sampling frequency, the cut-off frequency and whether windowing is to be used.

Use your programme to design a filter (i) with and (ii) without windowing, whose number of taps is equal to the last two digits of your ID number. The sampling frequency is to be 8000Hz and the cut-off frequency 200Hz.

Write a program in C or C++ to calculate the frequency response (Amplitude & Phase) of a digital filter and use it to plot the frequency response of the filters you designed in 2. Compare the responses (Amplitude & Phase)

Make a file of about 400 samples [at 8kHz] of you own speech waveform for the area between a fricative ’shh’ and a vowel ‘ay’ sound. Plot the original waveform, process it through your filter and plot the resultant output. Compare the input and the output signals.

Using Matlab or your FFT program, plot the spectrum of the input and output [of the vowel part] of the speech signals. Comment.

Construct a real-time implementation of your filter on the C31 DSK. Record few seconds of your speech.

Playback the original speech and hear the signal filtered by your filter. Listen carefully to both and comment on what you hear.

Tags: DCU, Digital Signal Processing, General

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