CS 312: SIGNAL PROCESSING LAB

CS 312 TA

1. Objectives
• To become familiar with the convolution operator and how it applies
to signal processing.
• To apply the fast fourier transform to a real world problem.
• To understand how a divide and conquer algorithm can increase
efficiency.

2. Background
Signal processing deals with the analysis and manipulation of signals. A
signal is any time varying quantity, and can be thought of as a function
over time. An example of a signal is a codified message being sent accross a
communication channel where at each time t, a certain symbol is received.
A sound recording is an example of a signal which has a specific amplitude
(or volumn) at each time t. The pitch heard is determined by the frequency
of the changing signal. Signals can be analyzed and manipulated just like
functions. For example two signals a and b can be added together to produce
a + b = (a0 + b0 , a1 + b1 , . . . , an−1 + bn−1 ).
One way of combining two signals is the convolution. Suppose you have
two vectors or signals of length m and n respecively: a = (a0 , a1 , . . . , am−1 ),
b = (b0 , b1 , . . . , bn−1 ). The convolution of a and b, written a ∗ b, will be a
vector of length m + n − 1, where coordinate k is equal to the sum of the
products a and b who’s indecies sum to k.
X
ai bj .
(i,j):i+j=k

For example, if c = a ∗ b, then c4 = a0 b4 + a1 b3 + a2 b2 + a3 b1 + a4 b0 because

each index of a and b sum to 4.
So,

a∗b = (a0 b0 , a0 b1 +a1 b0 , a0 b2 +a1 b1 +a2 b0 , . . . , am−2 bn−1 +am−1 bn−2 , am−1 bn−1 ).
1
2 CS 312 TA

One way to get the convolution is to imagine the matrix

a0 b0 a0 b1 ... a0 bn−2 a0 bn−1
a1 b0 a1 b1 ... a1 bn−2 a1 bn−1
a2 b0 a2 b1 ... a2 bn−2 a2 bn−1
.. .. .. .. ..
. . . . .
am−1 b0 am−1 b1 . . . am−1 bn−2 am−1 bn−1
then each coordinate of the convolution vector is found by summing along
the diagonals, starting on the left entry of column one and summing di-
agonnaly up and to the right.
The convolution operator is used widely in signal processing for many
different applications. When two signals are convolved they produce a new
signal that is a combination of the other two. With sound signals this can
produce many different effects like echo, noise cancelation, and various types
of noise filters.
In order to compute the convolution of two vectors a and b the following
sum needs to be calculated
X
ai bj
(i,j):i+j=k

which is just the definition of convolution. Here we will describe the “reverse-
and-slide” method for computing this sum. Let c = a ∗ b. Start by listing
the vector a in reverse order and then list the vector b underneath it so that
the first entry of b is right under the first entry of a as shown below.
am−1 am−2 . . . a2 a1 a0
b0 b1 b2 . . . bn−2 bn−1
Now the first index of the convolution, c0 , is found by multiplying columns
together and adding the results together, treating the empty spots as 0. So
c0 = a0 b0 . Now to get the next index shift the bottom row to the left by
one and repeat the sum of the column multiplication.
am−1 am−2 . . . a2 a1 a0
b0 b1 b2 . . . bn−2 bn−1
This results in c1 = a1 b0 + a0 b1 . After the next iteration
am−1 am−2 . . . a2 a1 a0
b0 b1 b2 . . . bn−2 bn−1
You get c2 = a2 b0 + a1 b1 + a0 b2 . Continue in this manner until the last two
iterations which look like this
am−1 am−2 . . . a2 a1 a0
b0 b1 b2 . . . bn−2 bn−1
am−1 am−2 . . . a2 a1 a0
b0 b1 b2 . . . bn−2 bn−1
 CS 312: SIGNAL PROCESSING LAB 3

Giving you cm+n−2 = am−1 bn−2 + am−2 bn−1 and cm+n−1 = am−1 bn−1 for
the last two terms.
Analyzing this algorithm yields a run time of O(m ∗ n) or O(n2 ) if the
signals are the same length. Go ahead and prove it to yourself.
There is another way to compute the convolution of two vectors or sig-
nals that will run in O(n log n). This requires using the Fast Fourier Trans-
form. We will treat our vectors/signals as polynomials. The vector a =
(a0 , a1 , . . . , am−1 ) becomes the polynomial A(x) = a0 + a1 x + a2 x2 + . . . +
am−1 xm−1 and vector b becomes the polynomial B(x) = b0 + b1 x + b2 x2 +
. . .+bn−1 xn−1 . Now C(x) = A(x)B(x) is a polynomial where the coefficients
form a vector c which is equal to the convolution of a and b. The algorithm
proceeds as follows:
(1) First we choose m + n values x1 , x2 , . . . , xm+n and evaluate A(xj )
and B(xj ) for each of j = 1, 2, . . . , m + n.
(2) We can now compute C(xj ) for each j very easily: C(xj ) = A(xj )B(xj ).
(3) Finally, we recover C from its values on x1 , x2 , . . . , xm+n . By using
polynomial interpolation any polynomial of degree d can be recon-
structed from its values on any set of d + 1 or more points. Since A
and B each have degree at most m − 1 and n − 1 respectively, their
product C has degree at most m + n − 2 so it can be reconstructed
from m + n − 1 or more points.

3. Signal Input
The files that you will be given as signals to convolve are .raw files which
are essentially .wav files that had the headers removed. They are all stan-
dardized to the following standard, Sample rate: 22050, 16 bit word sizes
per sample, with linear 2’s compliment encoding, and 1 channel (mono as
opposed to stereo). Which means that each sample of the sound file is a
16 bit 2’s compliment number, so you can just read in 16 bits and interpret
them as an integer (or short). Since it is a binary file there are no spaces or
parsing, just a stream of 16 bit words. A helpful class you may want to use
for reading in binary data is a BinaryReader.
To output the convolved signal the program should just output 16 bit
integers (or shorts) to a stream one after another. Make sure that you don’t
just output standard 32 bit integers or there will be extra data in the sound
file and it will sound garbled. Again, a helpful class you may want to use is
the BinaryWriter.
Once you output the raw sound file you can listen to it if you have a
player that can play raw sound files. One that I know of is GoldWave.
(Trial available at http://www.goldwave.com/release.php) When opening a
raw file it asks for the file format (sample rate, word size, encoding, and
channel) and when you enter the format (as specified above: 22050 sample
rate, 16 bit word size, linear 2’s comp. encoding, 1 channel) it will then play
the raw file. Alternatively you can convert the file to a wav file by using
4 CS 312 TA

Sox (downloadable at http://sox.sourceforge.net/). To convert a file from

.raw to .wav using our format you just enter this command in the command
prompt.
sox -r 22050 -s -w rawfilename.raw -r 22050 -s -w newfilename.wav
You can also convert any .wav file to our .raw file format by entering the
command
sox wavfilename.wav -r 22050 -s -w newfilename.raw
In these commands -r is the option to set the sample rate, -s is for linear 2’s
complement encoding, and -w is for 16 bit word size. These options must
always be specified for .raw files, whereas in .wav files this information is
contained in the header.

4. To Do
Your assignment for this project is to implement the Fast Fourier Trans-
form algorithm in C#. Your program will read in two different signals from
two different files. It will then produce a new signal that is the convolution
of the two input signals and output this new signal to a new file. The file-
names for both input signals and output signal will need to be entered on
the command line as parameters. Running your program from the command
line must look like this
C:\>convolution.cs inputfile1.raw inputfile2.raw outputfile.raw
Here are some guidelines/helps for implementing Fast Fourier Transform
(1) Follow the algorithm as it shows in the slides.
(2) Use the complex roots of unity to decide which points to evaluate
A(xj ) and B(xj ) at.
(3) Your program should use divide and conquer to evaluate the func-
tions A and B in O(n log n) time.
(4) For polynomial interpolation define a new function D(x) as shown
in class and use your divide and conquer algorithm to evaluate D(x)
at the 2nth roots of unity.
(5) Once you have D(x) find all the coefficients of C(x) using cs =
1 2πji/k .
2n D(ω2n−s,2n ) where ωj,k is the complex number e
(6) One easy way to do the inverse FFT is as follows:
(a) take the conjugate of each point of the signal.
(b) take the forward FFT.
(c) take the pointwise conjugate again.
(d) divide each point by n.

5. paper
Once you have your program completed, you need to convolve the five
pairs of signals found in table 1 which show five different applications of
convolution. Complete a lab write-up containing the following.
 CS 312: SIGNAL PROCESSING LAB 5

Table 1. Signal and Impulse Response pairs that should be

convolved together to pass off the project.

Signals Impulse Responses Desired Result

BlakesPiano.raw StNicolaesChurch.raw Blake’s piano in St. Nicolaes Church.
Classical.raw lowpass.raw Filter out the high pitched sounds.
Classical.raw allpass.raw Filter out high and low sounds. (telephone)
Classical.raw highpass.raw Filter out the low sounds.
NoisySong.raw notch60hz.raw Should remove the buzzing sound.

• An estimate of the time it took to complete the lab along with a

brief description of anything cool/extra that you implemented in
the project.
• We described the “reverse-and-slide” method of computing the con-
volution in the background section. Compare and contrast this al-
gorithm with the FFT algorithm implemented in this project.
Email your report, your code, an executable of your program that can
be run using the command line description above, and the 5 files containing
the convolution of the 5 pairs of convolved signals in .wav format to the TA
Paul using blackboard.