ECE 3512 Spring 2011 (Assignment #2)

Interpreting the Fourier Transform

Nabidur Rahman & Asish Mathew
AbstractFourier Transforms are used to transform a time domain signal into its frequency domain. In doing so, we gain a better understanding of the signals properties and its frequency components. If the duration of the signal in the time domain is increased, the resulting signals Fourier Transform will decrease. The application of a Fourier Transform can also allow us to analyze audio samples in their frequency domain, allowing a further analysis of the signal and discussion on the type of sound that is observed.

II. PROCEDURE AND RESULTS (PART 1) The first part of the assignment was to create a cosine signal x(t) with a frequency and duration of our choosing. We assigned our x(t) as x(t)=cos(2t) and our time duration as 5, 15, and 100 seconds. After inputting the selected values in MATLAB and using the myFFT function to compute the Fourier Transform, the results of this experiment are as follows:
Duration for 5 sec |X(w)| 0.5 0 -2

I. INTRODUCTION

HE objective of this assignment is to gain experience in interpreting the Fourier Transform of various signals. Using MATLAB, we can show how taking the Fourier Transform of several functions reveals its time and frequency domains and its alterations from the original function to the transformed function. The assignment is to be completed entirely using MATLAB using the given function myFFT. This function will compute the Fourier Transform of various signals inputted. The equation for computing a Fourier Transform is shown below: = 1

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The theoretical application of a Fourier Transform is that if the time domain of a signal is increased, the frequency component of the signal will decrease, resulting into an impulse response. An impulse response describes the reaction of the system as a function of time, or known as the delta function, . The reason behind this phenomenon is that we are actually multiplying our specified signal with a square wave, which results in the convolution in the frequency domain. An equation sharing this relationship is shown below: }{ }{ = } {2 * means convolution, with {} denoting Fourier Transform We can further validate the need for a Fourier Transform by testing out different audio signals and computing their Fourier Transforms using different time durations. By taking the Fourier Transform of different audio signals, we can gain knowledge as to how time affects the results and how different sounds can affect the transform of a signal. Signal processing helps us to understand how sound works and why some are heard differently than others. The first step in understanding sound waves is to understand the principle meaning of a Fourier Transform and its applications in signal processing.

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Figure 1- Resulting Fourier Transforms of x(t) with three different time durations We can observe that as the time duration of our cosine signal is increased, the resulting sinc function is becoming more and more compressed. To verify if our results are correct, we can observe the center point of the sinc function and see if matches with our x(t). Since our x(t) is defined as cos(2t), we can use the equation = 2 to find the value of f (frequency in Hz). Taking that into account, the resulting equation will become f= = 1Hz. The resulting Fourier Transform should be centered at those frequencies, both positive and negative since the Fourier series of a cosine wave is two impulse responses at both frequencies. The reason as to why the Fourier Transform of our cosine wave results in a sinc function is because our cosine signal is actually being multiplied by a square wave with a height of 1 and width of the duration we specified for our x(t). Using equation 2, we can see the properties of convolution taking place in the frequency domain, with the sinc function and delta functions centered at the base frequency of 1. We can observe this result at 100 seconds, where the sinc function is compressed almost into an impulse response. This concludes

ECE 3512 Spring 2011 (Assignment #2) our theory on the characteristics of a Fourier Transform at different time durations: the longer the time duration of the signal, the resulting Fourier Transform will decrease into an impulse response. III. PROCEDURE AND RESULTS (PART 2) The second part of this assignment instructs us to take the Fourier Transform of different sound clips. Using Audacity, we created three different sound clips: one with bass tones, one with vocal/midrange tones, and one with treble tones. As in Part 1, we used myFFT to compute the Fourier Transform of each sound clip. The results are as follows:
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2 IV. PROCEDURE AND RESULTS (PART 3) The final part of this assignment instructs us to take a previous sound clip used in Part 2 and modify it using one of the audio effects in the Audacity Effect Menu. We decided to use our vocal/midrange sound clip and implement a low pass filter to see how it would affect the resulting Fourier Transform. The results from this experiment are as follows:
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Figure 3 Original Fourier Transform compared with Low Pass Filtered Signals Fourier Transform In the Audacity Effects Menu, we used a low pass filter with a cutoff frequency of 1400Hz. Using the knowledge learned in lecture about low pass filters, we know that low pass filters are designed to reject frequencies above the cutoff and retain frequencies below it. From Figure 3, we can see that the filtered signal has eliminated all the high frequencies above 1400Hz, leaving very low amounts of energy at the remaining frequencies. We also observed that the filtered signal sounded different, as if it was dense with more bass and less clarity in the treble and mids. Since the filter rejected the high frequency components, the resulting sound wave should sound muffled, a resulting effect that low-pass filters have on all audio signals. V. DISCUSSION After completing the assignment, we learned how different effects can ultimately change the Fourier Transform of a given signal. Part 1 gave us an idea the mechanism at work: by increasing the time domain, we observed that the frequency content decreased into an impulse response. We conclude that this applies to all discrete signals based on multiple tries with different cosine inputs. Part 2 gave us an understanding as to how different audio tones output different Fourier Transforms. Part 3 of this assignment allowed us to apply an effect to a sound clip previously used in Part 2 and comment on its Fourier Transform. After the assignment was completed, we observed the purpose of Fourier Transforms and its useful insight in examining a signal and its properties.

Figure 2 Resulting Fourier Transforms of three different sound clips We can observe that bass tones carry high energies at low frequencies as opposed to treble tones which carry high energies at higher frequencies. For the bass tone, most of the energy is located from 50-150 Hz, with the remaining frequencies showing almost no energy at all. For the vocal/midrange tone, we can see that most of the energy is located at 0-250 Hz, with the remaining frequencies showing very low energy levels. The treble tone shows sporadic amounts of energy from 0-1800Hz, with 2000 Hz containing the highest amount of energy for that signal. To validate if our results are correct, we can observe the type of sound used and its characteristics. We know that audio signals that consist of bass tones are made out of low frequency components. The resulting Fourier Transform shows that high energies are located at low frequencies, which verifies that our results for the bass tone signal are correct. Likewise, treble tones are made out of high frequency components. Figure 2 shows that the resulting Fourier transform of the treble tone to have higher energies at higher frequencies, validating that our results are correct. By comparing the magnitude plot of all three Fourier transforms, we can observe the characteristics of each sound and what frequency components they consist of.