Digital Signal Processing

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

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Finite Impulse Response (FIR) Filter

• 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)

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Downsampling

• 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

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Simple DSP Functionality

• 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
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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