1. Digital Signal Processing
CS 153, Spring 2007
Ian G. Harris
Department of Computer Science
University of California Irvine

2. Representing Sound Digitally
ADC 30 ,
120
160 .. .
microphone
Microphone - converts pressure waves in the air (sound) to an analog voltage waveform
Analog to Digital Converter (ADC) - converts analog voltage waveform to a series of digital voltage values

3. Digital Sampling
ADC samples the analog voltage waveform at regular intervals
Samples can be used to create analog waveform again later
Samples can be processed to alter the sound

4. Finite Impulse Response (FIR) Filter
2, 6, 7, 5, 3
FIR
5, 6, 5 5 6
Window size = 3
Q(t) = [p(t-3) + p(t-2) + p(t-1)] / 3
FIR filter performs a function on the most recent w samples
Window size w is the number of recent samples considered
Smoothing filter computes average of all samples in the window
Output rate is the same as the input rate (once per clock cycle)

5. Downsampling
2, 6, 7, 5, 3 DS
2, ,
Downsampling factor M = 2
Downsampling reduces sample frequency by dropping samples
Downsampling factor M is the ratio between old and new frequencies
Assume that sample frequency is clock frequency

6. Simple DSP Functionality
sample_in
F/DS
ctrl
reset
sample_out
Simple DSP
clk
Can perform either smooth filtering or downsampling
sample_in - 8 bits wide, sample inputs,1 sample each clock
F/DS - selects operation, 0 = smooth filtering, 1 = downsampling
ctrl - 3 bits wide, selects window size (0-7) or downsample factor (1, 2, 4)
reset - starts function from the beginning erases history

7. Process to Complete the Project
Write “pseudocode” for the behavior
- Like the GCD algorithm in the last project
Design a controller and datapath
Implement them like in the last project