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Introduction to Digital
Signal Processing
Allen Downey
Olin College
Start here
http://allendowney.github.io/ThinkDSP/tutorial
http://tinyurl.com/DSP17Tut
Whirlwind?
We’re going to cover a semester in 3 hours.
We will not achieve deep understanding.
Goals:
1. Overview of the big ideas.
2. Stimulate appetite.
Why?
Signal processing is the coolest thing I know.
● It’s useful for many things.
● The algorithm is clever.
● The math is elegant.
● It explains so much.
But...
Non-engineers usually don’t know about it.
And engineers learn it the hard way.
http://www.amazon.com/dp/0966017633
Opportunity
Two ways to make this more fun:
● Sound
● Python
Sound
Because our ears do something similar to FFT.
Vi Hart, “What is up with Noises?
(The Science and Mathematics of Sound, Frequency, and Pitch)”
Python
Because you can explore interactively.
Downey, “Cacophony for the whole family”
Think DSP
Published by O’Reilly Media,
available under a free license
from Green Tea Press.
Starting at the end
1. Clone the repository.
2. Launch Jupyter.
3. Open chap10preview.ipynb
Put in your headphones!
Black magic?
Not if you understand
1. Discrete Fourier transform.
2. Convolution theorem.
3. Linear time-invariant theory.
And that’s the plan.
thinkdsp.py
Signal represents a continuous mathematical function.
Wave contains an array of discrete samples.
Spectrum contains the discrete Fourier transform of a Wave.
Signal, Wave, Spectrum
The notebooks
1. Open chap01.ipynb
2. Read the text.
3. Run the code.
Stop when you get to “Interaction”.
Exercise
Starting with this:
cos_sig = thinkdsp.CosSignal(freq=440, amp=1.0, offset=0)
sin_sig = thinkdsp.SinSignal(freq=880, amp=0.5, offset=0)
wave = mix.make_wave(duration=0.5, start=0, framerate=11025)
Make and plot the spectrum of wave.
Spectral analysis
Any signal can be represented as a sum of
single-frequency components.
The spectrum of a signal encodes the amplitude
of each component.
DFT
Discrete Fourier Transform of a wave is its spectrum.
Fast Fourier Transform is an efficient algorithm.
(But people use FFT and DFT interchangeably)
Opening the hood
Wave.make_spectrum is a wrapper for numpy.fft.rfft
rfft stands for “real FFT”.
The values in the Wave are real (not complex).
class Wave:
def make_spectrum(self):
hs = np.fft.rfft(self.ys)
n = len(self.ys) # number of samples
d = 1 / self.framerate # time between samples
fs = np.fft.rfftfreq(n, d)
return Spectrum(hs, fs, self.framerate)
How does DFT work?
Computes correlation
between the signal and
sinusoids with different
frequencies.
http://paulbourke.net/miscellaneous/dft/
Filtering
A filter modifies a signal
by amplifying/attenuating
some components more
than others.
Low-pass filter
Attenuates high frequencies, lets low frequencies pass.
Sounds “muffled”, like under water, through a pillow or wall…
Because everything that transmits sound is a low pass filter.
Waveforms and harmonics
1. Open chap02.ipynb.
2. Read and run the first section,
“Waveforms and Harmonics”.
Odd harmonics only Even and odd harmonics
Slow
dropoff
Fast
dropoff
?
See chap02soln.ipynb, last exercise.
Aliasing
Signal is continuous in time.
Wave is a sequence of discrete samples.
We lose information when we sample.
Aliasing
If the sample rate is 10000 Hz,
the highest we can measure is 5000 Hz.
What happens to frequencies above the
“folding frequency”?
http://denethor.wlu.ca/pc364/images/fft_alias.png
Aliasing
1. Back to chap02.ipynb
2. Jump to “Aliasing interaction”.
3. Explore the spectrum of a sawtooth signal
sampled at different rates.
Complex amplitudes
class Wave:
def make_spectrum(self):
hs = np.fft.rfft(self.ys)
n = len(self.ys) # number of samples
d = 1 / self.framerate # time between samples
fs = np.fft.rfftfreq(n, d)
return Spectrum(hs, fs, self.framerate)
Complex amplitudes
Spectrum contains frequencies, fs, and
complex amplitudes, hs.
Each element of hs encodes a magnitude
and a phase offset.
Complex numbers
Real part, x, and
imaginary part, y.
OR
Magnitude, r, and
angl e, φ (phi) .
https://en.wikipedia.org/wiki/Complex_number
Phase offset
Amplitude, r, indicates
loudness.
Phase offset, φ, indicates
where the signal “starts”.
https://upload.wikimedia.org/wikipedia/commons/5/55/Phase_shift.svg
Phase
1. Back to chap02.ipynb
2. Read “Amplitude and phase”.
3. Read and listen to
“What does phase sound like?”
What does phase sound like?
For more, see phase.ipynb.
Chirps
Signals so far don’t change over time.
But most sounds do.
Chirp: Signal with linearly-increasing frequency.
Chirps
1. On to chap03.ipynb
2. Read the first section “Chirp”.
3. Skip “Leakage”.
4. Read the third section “Spectrogram”.
5. Stop when you get to “Spectrum of a chirp”.
Short Time Fourier Transform
1. Divide a wave into segments.
2. Compute the DFT of each segment.
Gabor limit
Bigger segments:
● Better frequency resolution,
● Worse time resolution.
Gabor limit
Smaller segments:
● Better time resolution,
● Less frequency resolution.
Sawtooth Chirp
Chirp is based on a sinusoid.
How about a Sawtooth?
Sawtooth Chirp
1. Open chap03soln.ipynb
2. Read the second exercise.
3. Listen to the solution.
Familiar?
Bird
Song
Hero
https://academy.allaboutbirds.org/features/bird-song-hero/bird-song-hero-tutorial
Smoothing and convolution
Moving average is used to smooth
“noisy” time series data.
Convolution
Moving average is the convolution of a signal and a window.
signal ∗ window
Convolution
1. Multiply the signal by the window elementwise.
2. Add up the product.
3. Record the result.
4. Shift the window one element to the right.
5. Repeat.
In Python
smoothed = np.zeros(N) # space to store result
rolled = window.copy() # copy the window
for i in range(N):
smoothed[i] = sum(rolled * segment.ys)
rolled = np.roll(rolled, 1)
Smoothing and convolution
1. Open chap08.ipynb.
2. Read and run “Smoothing”.
3. Read and run “Smoothing sound signals”.
Smooth sawtooth?
Smoothing signals
1. Looking at the Signal, it gets smoother.
2. Looking at the Spectrum, it gets ???
Domain
Time domain: looking at Waveform
(function of time).
Frequency domain: looking at
Spectrum (function of frequency).
Conjecture
Smoothing the signal in the time domain
corresponds to a low pass filter in the
frequency domain.
But what kind of filter is it?
wave ∗ window =
spectrum × filter =
Estimating the filter
Maybe we can characterize the filter by
comparing the spectrum before and after.
In Python
amps = spectrum.amps
amps2 = spectrum2.amps
ratio = amps2 / amps
ratio[amps<280] = 0
Smoothing and convolution
1. Back to chap08.ipynb.
2. Read and run “Frequency domain”.
wave ∗ window =
spectrum × filter =
Convolution theorem
Convolution with a window in the time domain
corresponds to
multiplication by a filter in the frequency domain.
And...
Convolution theorem
Convolution with a window in the time domain
corresponds to
multiplication by a filter in the frequency domain.
The filter is the DFT of the window.
Convolution theorem
DFT(wave ∗ window) = DFT(wave) × DFT(window)
Convolution theorem
DFT(wave ∗ window) = DFT(wave) × DFT(window)
Convolution theorem
DFT(wave ∗ window) = DFT(wave) × DFT(window)
Why?
Quadratic is worse than N log N
Boxcar window
The boxcar window corresponds to a low-pass filter.
But not a very good one.
Because it looks like a square wave.
Which has slowly-decaying harmonics.
Gaussian window
A smoother window (in the time domain) has a
sharper cutoff (in the frequency domain).
Convolution theorem
1. Back to chap08.ipynb.
2. Read and run “Gaussian window”.
3. And try out the interactive widget.
Gaussian window
Wider window means more smoothing.
Which means a lower cutoff frequency.
Gaussian window
If std is too big, or M is too small:
1. The Gaussian gets cut off,
2. It looks more like a boxcar, and
3. The high frequency bounces come back.
LTI Theory
We’ve got DFT and the Convolution Theorem.
All we need now is Linear Time-Invariant Theory!
System
A system is anything that takes a signal as input
and produces a signal as output.
An LTI system is...
Time invariant
If input1 produces output1 now,
it always has,
and always will.
Linear
If input1 produces output1
and input2 produces output2
then
input1 + input2 produces output1 + output2
Acoustics are LTI
An acoustic system can:
1) Delay a signal and echo.
2) Attenuate some frequencies more than others.
But it can’t:
1) Output more energy
than input.
2) Move energy from one
frequency to another.
And that means...
System characterization
If we know what a system does to
each frequency component,
we know what it will do to a
signal.
How?
You could input one frequency at a time and
record the result.
Or sweep through a range of frequencies.
Or input all the frequencies at the same time!
All frequencies?
What signal has equal
power at all frequencies?
All frequencies!
Impulse
The input is an impulse.
The output is the “impulse response”.
DFT of the impulse response is the “transfer function”.
System characterization
1. Open chap10.ipynb.
2. Skim the first two sections.
3. Read and run “Acoustic Impulse Response”.
wave ∗ window =
spectrum × filter =
wave ∗ impulse response =
spectrum × transfer function =
Black magic?
Not if you understand
1. Discrete Fourier transform ✔
2. Convolution theorem ✔
3. Linear time-invariant theory ✔
And now you do.
Want more?
Think DSP, published by O’Reilly
Media, available under a free
license from Green Tea Press.
surveymonkey.com/r/pycon163

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Digital Signal Processing Tutorial Using Python