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Lecture 5
Equipment less complex and expensive than digital-to-analog modulation equipment Permits use of modern digital transmission and switching equipment
Some transmission media will only propagate analog signals E.g., optical fiber and unguided media Analog data in electrical form can be transmitted easily and cheaply Done with voice transmission over voice-grade lines
An increase in data rate increases bit error rate An increase in SNR decreases bit error rate An increase in bandwidth allows an increase in data rate
Signal spectrum
With lack of high-frequency components, less bandwidth required With no dc component, ac coupling via transformer possible Transfer function of a channel is worse near band edges Ease of determining beginning and end of each bit positionSignal interference and noise immunity Performance in the presence of noise The higher the signal rate to achieve a given data rate, the greater the cost
Clocking
When only analog transmission facilities are available, digital to analog conversion required A higher frequency may be needed for effective transmission Modulation permits frequency division multiplexing
Spectrum of AM signal
Amplitude Modulation
Transmitted power
na Pt ! Pc 1 2
Sends only one sideband Eliminates other sideband and carrier Only half the bandwidth is required Less power is required Suppressed carrier cant be used for synchronization purposes
Advantages
Disadvantages
Pulse Modulation Analogue modulated systems are quite widely used, because of their simplicity. An alternative to analogue modulated systems is Pulsed systems. This system is based on digital signals or pulses. The basis of such a system is the use of a digital carrier signal, which is modulated by an analogue signal. There are various ways in which this can be achieved, giving rise different systems.
Sampling of signals
An analogue signal is transmitted continuously in its entire form. This need not be done provided certain conditions are satisfied. Samples of the analogue signal may be transmitted at given intervals of time. The original signal may then be recovered at the receiving end from the transmitted samples. This technique is known as sampling and it underlies pulsed systems.
1 f ! T
If the pulse train were amplitude modulated by the analogue signal, the result will be pulses whose amplitudes are samples of the analogue signal at time intervals T.
If the amplitude modulated pulse train is then analysed, its spectrum will consist of Fourier components,
0, f , 2 f , 3 f ...
At each of this there will be a set of sum and difference frequencies (lower and upper sidebands) due to each frequency component of the analogue signal.
If W and f-W do not overlap then it is possible to separate the group of frequencies at the receiving end. This can be achieved by use of a low-pass filter with a cut-off frequency f = W. The separated frequencies are then those of the analogue signal transmitted. From this the following condition is derived
f W uW
f u 2W
This means the repetition frequency must be at least twice the highest frequency component in the analogue signal. The minimum sampling frequency is then
f min ! 2W
This is also called the Nyquist rate. If the sampling rate is less than 2W the lower sideband will overlap the baseband and it will not be possible to separate them. This effect is known as aliasing. It can be avoided by passing the signal through a filter before sampling.
Telephone siganls range from 300 Hz to 3.4 kHz, the internationally agreed sampling frequency is 8kHz. It means there is a guard band of 1.2kHz between the lower side band and the baseband. Comment: It means W has an upper limit The technique can only be used if the bandwidth can be restricted to W without destroying the essential information. To achieve this, bandlimited signals are used.
Sampling Theorem
t Any function of time F
whose highest frequency is W Hz can be completely determined by sampled amplitudes spaced at
time intervals
1 2W
apart.
If a signal f(t) is sampled at regular intervals of time and at a rate higher than twice the highest frequency, then the samples contain all of the information of the original signal. The function f(t) may be reconstructed from these samples by the use of a low pass filter.
Sampler
An analogue sampler commonly known as the sample-andhold is used in sampling the input analogue signal voltage and maintaining that voltage until the next sampling instant.
The FET (Field Effect Transistor) acts like a simple switch. When turned "on," it provides a low-impedance path to deposit the analogue sample voltage on capacitor. The time that the FET is "on" is called the aperture or acquisition time. Essentially, the capacitor is the hold circuit. When the switch FET is "Off," the capacitor does not have a complete path to discharge through and therefore stores the sampled voltage.
The storage time of the capacitor is also called the conversion time because it is during this time that the unit converts the sample voltage to a digital code. The acquisition time should he very short. This assures that a minimum change occurs in the analogue signal while it is being deposited across the capacitor. If the input to the sampler is changing while it is performing the conversion, distortion results. This distortion is called aperture distortion.
Thus, by having a short aperture time and keeping the input to the constant relatively constant, the sample-and-hold circuit reduces aperture distortion. If the analogue signal is sampled for a short period of time and the sample voltage is held at constant amplitude during the conversion time, this is called flat-top sampling. If the sample time is made longer and the analogue-to-digital conversion takes place with a changing analogue signal, this is called natural sampling. Natural sampling introduces more aperture distortion than flattop sampling and requires a faster A/D converter.
Based on the sampling theorem Each analog sample is assigned a binary code
The digital signal consists of block of n bits, where each n-bit number is the amplitude of a PCM pulse
Quantization
Is the process of converting the sampled signal to a binary value Each voltage level will correspond to a different binary number The magnitude of the minimum step size is called the resolution. The error resulting from quantizing is called the quantization noise. Its value is 1/2 the resolution
Dynamic Range This is the ratio of the largest to smallest analogue signal that can be transmitted.
Vmax DR ! Vmin
But Vmin is the resolution and can be written as Vmax q ! Vmin ! n 2 It follows that Vmax DR ! ! 2n Vmin
DR(dB) ! 6n
From
DR ! 2n
Maximum binary number for a system. With one code used for 0V which is not considered in calculating DR, it is observed that DR ! 2n 1
Example
Given a PCM system with the following parameters: Maximum analog input frequency 3kHz A maximum decoded voltage at the receiver of +/- 1.27V A minimum dynamic range of 35dB. Find the minimum sample rate The number of bits required The resolution The quantization error
Quantization Noise
1 I 1 2 I dI ! Root mean square error q 3 q q q 2
3
q 2
error (I )
q 2
q 2
q 2
q ! 12
! q ! 12 R
2
q 2 3
No additive noise, can be used over long distances No intermodulation noise For a given bandwidth the signal/noise ratio is superior
Conversion to digital signaling allows use of more efficient digital switching techniques
Problems
Large bandwidth required. Might consider using optical fibres Circuits for implementation are costly uneconomical over short distances less than 5km. Might consider using integrated circuits to deal with last two.