EE330 Digital Signal Processing
Introduction
Arbab Latif
Spring 2024
Resources:
Discrete Time Signal Processing, A. V. Oppenheim and R. W. Schaffer, 3rd Edition, 2010 (Chapter 2.1-2.7)
2 Major topic areas
• Signal processing in the time domain:
convolution
• Frequency-domain processing:
– The DTFT and the Z-transform
– Complementary signal representations
• Sampling and change of sampling rate
• The DFT and the FFT
• Digital filter implementation
• Digital filter design
• Selected applications
3
Some application areas
(we will not get to all of these)
• Linear prediction and lattice filters
• Adaptive filtering
• Optimal Wiener filtering
• Two-dimensional DSP (image processing)
• Short-time Fourier analysis
• Speech processing
4
Signal representation: why perform
signal processing?
• A speech waveform in time:
“Welcome to DSP I”
5
A time-frequency representation
of “welcome” is much more informative
6 Downsampling the waveform
Downsampling the waveform by factor of 2:
7 Consequences of downsampling by 2
Original:
Downsampled:
8 Upsampling the waveform
Upsampling by a factor of 2:
9 Consequences of upsampling by 2
Original:
Upsampled:
10 Linear filtering the waveform
x[n] y[n]
Filter 1:
y[n] = 3.6y[n–1]+5.0y[n–2]–3.2y[n–3]+.82y[n–4]
+.013x[n]–.032x[n–1]+.044x[n–2]–.033x[n–3]+.013x[n–4]
Filter 2:
y[n] = 2.7y[n–1]–3.3y[n–2]+2.0y[n–3]–.57y[n–4]
+.35x[n]–1.3x[n–1]+2.0x[n–2]–1.3x[n–3]+.35x[n–4]
11 Filter 1 in the time domain
12
Output of Filter 1 in the frequency
domain
Original:
Lowpass:
13 Filter 2 in the time domain
14
Output of Filter 2 in the frequency
domain
Original:
Highpass:
15
Let’s look at the lowpass filter from
different points of view …
x[n] y[n]
Difference equation for Lowpass Filter 1:
y[n] = 3.6y[n–1]+5.0y[n–2]–3.2y[n–3]+.82y[n–4]
+.013x[n]–.032x[n–1]+.044x[n–2]–.033x[n–3]+.013x[n–4]
Lowpass filtering in the time domain:
16
the unit sample response
17
Lowpass filtering in the frequency domain:
magnitude and phase of the DTFT
18 The z-transform representation…
x[n] y[n]
Difference equation for Lowpass Filter 1:
The corresponding z-transform of the
system:
19
The poles and zeros of the lowpass
filter
20
Lowpass filtering in the frequency domain:
magnitude and phase of the DTFT
21
Another type of modeling:
the source-filter model of speech
• A useful model for representing the
generation of speech sounds:
Pitch Amplitude
Pulse train
source p[n]
Vocal tract model
Noise source
22
Signal modeling: let’s consider the
“uh” in “welcome:”
23 The raw spectrum
24 All-pole modeling: the LPC spectrum
25
An application of LPC modeling: separating
the vocal tract excitation and and filter
• Original speech:
• Speech with 75-Hz excitation:
• Speech with 150 Hz excitation:
• Speech with noise excitation:
• Comment: this is a major techniques used
in speech coding
26
Classical signal enhancement: compensation
of speech for noise and filtering
• Approach of Acero, Liu, Moreno, et al.
(1990-1997)…
“Clean” speech Degraded speech
x[m]
h[m] z[m]
Linear filtering n[m]
Additive noise
• Compensation achieved by estimating
parameters of noise and filter and applying
inverse operations
27
“Classical” combined compensation improves
accuracy in stationary environments
Complete
retraining
–7 dB 13 dB Clean
VTS (1997)
Original
CDCN (1990)
“Recovered”
CMN (baseline)
• Threshold shifts by ~7 dB
• Accuracy still poor for low SNRs
Another type of signal enhancement:
28
adaptive noise cancellation
• Speech + noise enters primary channel, correlated noise
enters reference channel
• Adaptive filter attempts to convert noise in secondary
channel to best resemble noise in primary channel and
subtracts
• Performance degrades when speech leaks into reference
29
Simulation of noise cancellation for a PDA
using two mics in “endfire” configuration
• Speech in cafeteria noise, no noise
cancellation
• Speech with noise cancellation
•
30
Signal separation: speech is quite intelligible,
even when presented only in fragments
• Procedure:
– Determine which time-frequency time-
frequency components appear to be
dominated by the desired signal
– Reconstruct signal based on “good”
components
• A Monaural example:
– Mixed signals -
– Separated signals -
31
Practical signal separation: Audio samples
using selective reconstruction based on ITD
RT60 (ms) 0 300
No Proc
Delay-sum
ZCAE-bin
ZCAE-cont
32
Phase vocoding: changing time
scale and pitch
• Changing the time scale:
– Original speech
– Faster by 4:3
– Slower by 1:2
• Transposing pitch:
– Original music
– After phase vocoding
– Transposing up by a major third
– Transposing down by a major third
• Comment: this is one of several
33 Summary
• Lots of interesting topics that teach us how
to understand signals and design filters
• An emphasis on developing a solid
understanding of fundamentals
• Will introduce selected applications to
demonstrate utility of techniques
• I hope that you have as much fun in signal
processing as I have had!
