Lab 4: Properties of DTFT

EE 341 Winter 2013

Emily Allstot (Section AA)

Assignment 1
In Assignment 1, animal sounds in .wav format are read and their time-domain
waveforms are plotted directly from the data, along with their frequency
spectrum. The discrete-time Fourier transform (DTFT) is used to find the
frequency spectrum of each animal sound, which is possible because the sounds
are discrete in the time-domain while being continuous in frequency using the
built-in MATLAB function freqz(), as detailed in the provided Lab 4 specifications
(Lab4.pdf).
Table 1. Animal sounds table

Name

Fs

cat.wav
bird.wav
toad.wav
tiger.wav
whale.wav

22050
22050
22255
22255
16000

Length
(sec)
1.00
1.14
0.67
0.753
3.45

Length
(Samples)
22052
25197
14843
16763
55190

Comments on what you see

from the DTFT magnitude

Using the max() built-in function, the fundamental frequency (first peak of the
magnitude response) of the cat sound occurs at occurs at 733 Hz (0.209 rad).
The precise amplitude value of the fundamental frequency is 379.79. These
values are confirmed by the bottom two graphs of Figure 1.

Figure 1. Cat sound: time sequence, DTFT (rad), and DTFT (Hz)

Figure 2. Bird sound: time sequence, DTFT (rad), and DTFT (Hz)

Figure 3. Toad sound: time sequence, DTFT (rad), and DTFT (Hz)

Figure 4. Tiger sound: time sequence, DTFT (rad), and DTFT (Hz)

Figure 5. Whale sound: time sequence, DTFT (rad), and DTFT (Hz)

Assignment 2
Using a given file with Microsoft Stock prices over time, convolution is performed
using the MATLAB built-in filter() function. The following given moving-average
filter is used to smooth the stock:

The original stock and the resultant smooth filtered stock are both shown in the
top plot in Figure 6. The frequency spectrum of h[n] is shown by the DTFT
magnitude in the bottom plot in Figure 6, which is.

Figure 6. Original and smoothed stock prices, frequency response

The moving average filter makes the stocks look smoother because it is a LPF
see how in the DTFT, the majority of the signal power is preserved near
baseband frequency, but higher frequencies are filtered out. So that is why using
this LPF has smoothed out the high-frequency micro-fluctuations in stock
prices.

Assignment 3
Assignment 3 implements a digital graphic equalizer on a given music.wav file,
and the role of dB magnitude to characterize gain and using freqz() with sampling
rate as input is observed.
If stereo (two channel) sound lasts for 10 seconds, the number of samples one
would expect to see is 882,000, as calculated using the following formula:

Additionally, in order to transmit at this level of CD audio quality without

compression, it would take 1,411,200 bits, as calculated using the following
formula:

Table 2. Filter coefficients and type of filter

Filter coefficients
Filter 1
Filter 2
Filter 3

B1 = 0.0495 0.1486
A1 = 1.0000 -1.1619
B2 = 0.1311
0
A2 = 1.0000 -0.4824
B3 = 0.0985 -0.2956
A3 = 1.0000 0.5722

0.1486 0.0495
0.6959 -0.1378
-0.2622
0
0.8101 -0.2269
0.2956 -0.0985
0.4218 0.0563

LP, HP,
or BP?
LP
0.1311
0.2722

BP
HP

Figure 7. Simple 3-bands equalizer filter frequency responses

While performing subband filtering, the three filters gains are changed to G1 =
0.1, G2 = 0.1, and G3 = 10. This means that the 1st and 2nd filters are attenuated
as much as possible (-20 dB) for a typical equalizer while the 3rd filter is boosted
to the max (+20 dB) for a typical equalizer. As expected, these new weights for
each filter make the song sound much more tinny and lacking bass tones a lot
like a cheap and/or small speaker sounds.
The gain of each filter is defined by the following formula:

Thus, the gain of each filter is as follows:

Filter 1: -20.1 dB
Filter 2: -36.6 dB
Filter 3: -7.73 dB
The reason that the output of each filter isnt heard as a distinct source is
because they are implemented in parallel, attenuating or boosting their own
range frequencies. The sound that is heard at the output of the equalizer is the
superposition of the effect of all three filters on the same one sound.