NATIONAL UNIVERSITY OF SINGAPORE Department of Electrical and Computer Engineering

EE3304: Digital Control Systems CA1 Report

Project Report Prepared by: Ow Min Qi U083984N

Introduction
The definition of correct sampling is simple. As highlighted in class, suppose a continuous signal is sampled in some manner. If the sampled signal is exactly reconstructed from the analog signal, proper sampling has been done. Even if the sampled data appears confusing or incomplete, the important information has been captured. However, when a signal is improperly sampled, aliasing may result and the information capture can be incorrect and misleading. For example, if an signal containing ultrasonic 45 KHz tone is sampled at 44.75kHz, the resulting signal contains a 0.25kHz, approximately middle C. The Nyquist-Shannon sampling theorem states that a signal can be properly sampled, only if the sampling rate is twice the highest frequency of the original signal.

First Case: A simple look at sine waves sampled

a)

This is a signal that is sampled at about 11 times the original analog frequency. This might represent a 90 cycle/second sine wave sampled at 1000 sampled/second, where 11.1 samples are taken at each cycle. This is proper sampling as no other sinusoid or combinations of any can produce this pattern of samples. Hence these sampled patterns correspond to one analog signal only, making it unique. The digital data can then be reconstructed. b)

This second signal is sampled at about 3.23 times the original analog frequency, where about 3.23 samples are taken every cycle. The resulting sample seems to be sparse and does not seem to capture the information very well. However, due to the fact that is the sampled signal still represents the original signal uniquely. In this case, the signal can once still be reconstructed. c)

In this case, the sampled signal rate is only about 1.05 times the frequency of the original signal. Hence the sampled signal only takes a mere 1.05 sample per cycle. The samples results in a different sine wave formed. Thus it has misrepresented the original signal. This is the case where aliasing occurs. Since the digital data is no longer uniquely related to a particular analog signal, an unambiguous reconstruction is impossible.

Second Case: Moire Patterns, a result of undersampling

Aliasing can cause problems in digital imaging. In images, the repetition is in space domain rather than time. If a repetitive pattern image of high spatial frequency is sampled at a low resolution (the sampling frequency actually becomes lower in relative), Moire patterns appear. Hence it is always troublesome to scan half-tone pictures or newspaper especially, where dots appear all the time if an anti-aliasing filter is not used. Consider a brick-like image. A successful resize of the image would result in the exact image but with a small resolution. However, without a anti-aliasing filter, dots may appear and the image would be distorted.

Original Image

Distorted Image The reason behind this strange aliasing is because when taking adequate samples, a few samples are taken per tone in the image. However, in the case of aliasing, the sample taking may land on the same tone at each interval, resulting in the reproducing of the same tone in wrong parts of the image. A more obvious effect of aliasing can be seen in the image below.

Original image

Image with aliasing This situation shows that when an image does not have enough pixel representation. The image can become distorted.

Third Case: Undersampling or Bandpass sampling

Undersampling or bandpass sampling is a technique where a bandpass filtered signal is sampled at a sample rate below the usual Nyquist sampling rate. When a bandpass signal is sampled, samples are equal to samples of a low-frequency alias of the high-frequency signal. This technique reduces the need for oversampling when the highest frequency of the bandpass signal is very high.

a) A simple case

Consider the above case where . When we choose resultant signal are shifted spectral of replicas that are somehow fits just nicely.

, the

(k is odd) When k is even, the spectral patterns are reversed. b) The general case

In this case highlighted, the kth replica should only touch the original signal barely in order for the sampled signal to be properly. Hence, and . Hence, the formula for the sampling frequency is: [ c) Example: FM Radio in Singapore For bandpass signals which are used in FM radio, its low and high band limits allow more flexibility to the sampling criterion. The condition, once again, is such that when sampled the bands in the positive and negative domains must not overlap each other. ]

In Singapore, FM radio operates on the frequency range of 88MHz to 108MHz. Letting and . The bandwidth is . Hence for the sampling criterion to be satisfied the above equation has to be used to calculate the available frequencies. The sampling conditions are met if: [ Therefore k can be 1, 2, 3, 4, or 5. If k = 5 is used, the lowest sampling frequency is used: . ]

The result is the presence of a baseband alias and replicas that barely touches the original signal. However an anti-alias filter would have to be precisely designed to fit the radio band. For k = 4, the sampling frequency can be The result spectrum can be seen below:

The ends of the replica would appear at and . This would allow more room for design of a bandpass filter to eliminate the aliased signals. Note the orientation of the replicas would be reversed as k is even. This means the resulting spectrum of the radio band is reversed. In undersampling of a real signal, in this case the FM radio, the sampling of the signal should be made in a short interval so that it can represent the instantaneous value of the signal at the highest

frequency. Hence the signal sampled must be able to capture a signal with a frequency of 108MHz, and not 43.2MHz. Thus, the sampling frequency may be only a little larger than 43.2MHz.

Conclusion
As seen from above the sampling theorem is very convenient for sampling band-limited signals. However, most source signal waveforms are not quite band-limited. There could be out-of-band signals or noise that could be incorporated into the sampled signal. This then leads to a need to first filter the source waveform to achieve a band-limited signal before doing bandpass sampling.