Sampling The starting point we need to understand is sampling and its effect on the signal of interest.

To take an analog signal and convert it to a digital signal, we need to sample the signal using a device called an analogto-digital converter (ADC). The ADC measures the signal at rapid intervals, and these measurements are called samples. It outputs a digital signal proportional to the amplitude of the analog signal at that instant. The graphs in Figures 3.1 and 3.2 show two different sinusoidal signals being sampled. The slower moving signal (lower frequency) in Figure 3.1 can be represented accurately with the indicated sample rate, but the faster moving signal (higher frequency) in Figure 3.2 is not accurately represented by our sample rate. In fact, it actually appears to be a slow-moving (low frequency) signal, as indicated by the dashed line. This shows the importance of sampling fast enough for a given frequency signal.

The dashed line in Figure 3.2 shows how the sampled signal will appear if we connect the sample dots and smooth out the signal. Notice that since the actual (solid line) signal is changing so rapidly between sampling instants, this movement is not apparent in the sampled version of the signal. The sampled version actually appears to be a lower frequency signal than the actual signal. This effect is known as aliasing. We need a way to quantify how fast we must sample to accurately represent a given signal. We also need to better understand exactly what is happening when aliasing occurs. The idea may seem strange, but in some instances aliasing can be useful.

To understand how much to sample, refer a fantastic example in DSPGoodOne.pdf page 29-31. To summarize, whenever you have a sampled signal, you cannot really be sure of its frequency. But if you assume that the rule was followedthat the signal was sampled at more than twice the frequency of the signal itselfthen the sampled signal will really represent the same frequency as the actual signal prior to sampling. The critical frequency that the signal must not ever exceed, which is one half of the sampling frequency, is called the Nyquist frequency. If we follow this rule, then we can avoid the aliasing phenomenon (demonstrated with the moving wheel example in DSPGoodOne.pdf page 29-31). Normally, the ADC converter frequency is selected to be high enough to sample the signal on which we want to perform digital signal processing. To make sure that unwanted signals above the Nyquist frequency do not enter the ADC and cause aliasing, the desired signal in usually passed through an analog low-pass filter, which attenuates any unwanted high-frequency content signals, just prior to the ADC. A common example is the telephone system. Our voices are assumed to have a maximum frequency of about 3600 Hz. At the microphone, our voice is filtered by an analog filter to eliminate, or at least substantially reduce, any frequencies above 3600 Hz. Then the microphone signal is sampled at 8000 Hz, or 8 kHz. All subsequent digital signal processing occurs on this 8 kSPS (kilo-samples per second) signal. That is why if you hear music in the background while on the telephone, the music will sound flat or distorted. Our ears can detect up to about 15 kHz frequencies, and music generally has frequency content exceeding 3600 Hz. But little of this higher frequency content will be passed through the telephone system. Quantization So far, we have covered the most important effects of sampling, but there remains one last issue related to sampling: quantization. We saw earlier how a digital signal is represented numerically and how we must be sure that the numbers representing the sampled signal do not exceed the range of our binary or hexadecimal number range What happens if the signals are very small? Remember in our discussion of signed fractional 8-bit fixedpoint numbers, the range of values we could represent was from 1 to +1 (well, almost +1). The step size was 1/128, which works out to 0.0078125. Any signals having values in between of these 1/128 steps, will have errors during quantization process. This error will be seen as form of noise or unwanted signal by DSP. It is called quantization noise. The quantization noise is always present. But as the input signal decreases in level, the quantization noise becomes more significant in a relative sense. Eventually, for very small input signal levels, the quantization noise can become so significant that it degrades the quality of whatever signal processing is to be performed. Think of it as like static on a car radio. As you get farther from the radio station, the radio signal gets weaker, and eventually the static noise makes it difficult or unpleasant to listen to, even if you increase the volume. Suppose we exchange our 8-bit ADC for a 16-bit ADC? Then our maximum range is still from 1 to +1, but our step size is now 1/32,768, which works out to 0.000030518. What a difference! The actual error is always less than our step size, 1/32,768. But the error as a percent of signal level is dramatically improved. This is what we usually care about in signal processing. Because of the

much smaller step size of the 16-bit ADC, the quantization noise is much less, allowing even small signals to be represented with very good precision (<1%). Another way of describing this scenario is to introduce the concept of signal-to-noise power ratio, or SNR. This ratio describes the power of the largest signal compared to the background noise. This can be very easily seen on a frequency domain or spectral plot of a signal. There can be many sources of noise, but for now, we are considering only the quantization noise introduced by the ADC sampling. SNR is usually expressed in decibels (denoted dB), using a logarithmic scale. The SNR of an ideal ADC can be determined by the following equation: SNRquantization(dB) = 6.02 * (number of bits) + 1:76 Basically, for each additional bit of the ADC, 6 dB of SNR is gained. An 8-bit ADC is capable of representing a signal with an SNR of about 48 dB, a 12-bit ADC can do better at 72 dB, and a 16-bit ADC will give up to 96 dB. This accounts only for the effect of quantization noise; in practice there are other effects that also will degrade SNR in a system. There is one last important point regarding decibels. They are very commonly used in many areas of digital signal processing subsystems. A decibel is simply a signal power ratio, similar to a percentage. But because of the extremely high ratios commonly used (a billion is not uncommon), it is convenient to express this amount logarithmically. The logarithmic expression also allow chains of circuits or signal processing operations, each with its own ratio (e.g., of output power to input power), to simply be added up to find the final ratio.

1. The Low Pass Filter the low pass filter only allows low frequency signals from 0Hz to its cut-off frequency, c point to pass while blocking those any higher.

2. The High Pass Filter the high pass filter only allows high frequency signals from its cut-off frequency, c point and higher to infinity to pass through while blocking those any lower.

3. The Band Pass Filter the band pass filter allows signals falling within a certain frequency band setup between two points to pass through while blocking both the lower and higher frequencies either side of this frequency band.