Introduction Many signals in modern communication systems are digital. Additionally, analog signals are transmitted digitally. Digitizing a signal results in reduced distortion and improvement in signal-to noise ratios. A digital signal is superior to an analog signal because it is more robust to noise, and can easily be recovered, corrected and amplified. For this reason, the tendency today is to change an analog signal to digital data. The sampling theorem Sampling is the process of taking periodic sample of the waveform to be transmitted. The process of transmitting signals in the form of pulses (discontinuous signals) by using special techniques. The signal is sampled at regular intervals such that each sample is proportional to the amplitude of signal at that instant.This technique is called “sampling”. Sampling is common in all pulse modulation techniques. Analog signal is sampled every Ts secs. Ts is referred to as the sampling interval. fs = 1/Ts is called the sampling rate or sampling frequency. “The more samples that are taken, the more final outcome looks like the original wave; However if fewer samples are taken, then other kinds information could be transmitted”. There are 3 sampling methods: Ideal sampling-an impulse at each sampling instant (at Ts sec instant). Natural sampling-a pulse of short width with varying amplitude. Flat top sampling-sample and hold, like natural but with single amplitude value. Sampling theorem (Nyquist’s theorem) Is used to determine minimum sampling rate for any signal so that the signal will be correctly restored at the receiver. Nyquist’s theorem states that, “The original information signal can be reconstructed at the receiver with minimal distortion if the sampling rate in the pulse modulation system is equal to or greater than twice the maximum information signal frequency. According to the sampling/Nyquist theorem, the sampling rate must be at least 2 times the highest frequency contained in the signal. fs =sampling frequency and fm = maximum frequency of the modulating signal. Basic condition of sampling process 1. Sampling at fs =2fm(max) In practice it is difficult to design a low pass filter, in order to restore the original modulating signal. 2. Sampling at fs > 2fm(max) This sampling rate creates a guard band between fm(max) and the lowest frequency component (fs-fm(max)) of the sampling harmonics. Therefore, a more practical LPF can be used to restore the modulating signal. 3. Sampling at fs< 2fm(max) When the sampling rate is less than the minimum value, distortion will occurs. This distortion is called aliasing. Aliasing effect can be eliminated by using an anti-aliasing filter prior to sampling and using a sampling rate slightly higher than Nyquist rate (fs=2fm).
Sampling alone is not a digital technique.
The immediate result of sampling is a pulse- amplitude modulation (PAM) signal. Nyquist sampling rate for low-pass and bandpass signals Quantization and Encoding Quantization: Is the process of dividing the maximum value of the analog signal into a fixed number of levels in order to convert the PAM into a Binary Code. The levels obtained are called “quantization levels”. The process of converting an analog signal into PCM is called coding, the inverse operation is called decoding. We will discuss these points later in the PCM section. Fig: Carrier for Continuous Wave and Pulse Modulation Need for Pulse Modulation Comparing to continuous wave modulation (like AM, FM), the performance of all pulse modulation schemes except PAM in presence of noise is very good. Due to better noise performance, it requires less power to cover large area of communication. Due to better noise performance and requirement of less signal power, the pulse modulation is most preferred for the communication between space ships and earth. ANALOG PULSE MODULATION (APM) In APM, the carrier signal is in the form of pulse waveform, and the modulated signal is where one of the characteristic (either amplitude, width or position) is changed according to the modulating/audio signal. The three common techniques of APM are: Pulse Amplitude Modulation (PAM), Pulse Width Modulation (PWM) and Pulse Position Modulation (PPM). Pulse Amplitude Modulation /PAM The simplest form of pulse modulation. In PAM, amplitude of pulses is varied in accordance with instantaneous value of modulating signal.
Fig: Generation of PAM
Fig: Waveforms of PAM The amplitude of a constant width, constant position pulsed carrier signal is varied according to the instantaneous amplitude of the modulating signal. Basically the modulating signal is sampled by the digital train of pulses and the process is based upon the sampling theorem. The PAM signal can be detected by passing it through a low pass filter. Advantages of PAM • It is easy to generate and demodulate PAM. Disadvantages of PAM • Since PAM does not utilize constant amplitude pulses, output is distorted due to additive noise so that it is infrequently used. • Transmission bandwidth required is too large. • Transmitted power is not constant. Application of PAM • Used in radio telemetry for remote monitoring and sensing. Pulse Width Modulation (PWM) When the width of pulsed carrier varies in accordance with the instantaneous amplitude of modulating signal, is called PWM where amplitude and position remains constant. Also known as Pulse Duration Modulation (PDM). PWM gives better signal to noise ratio performance than PAM. Advantages of PWM 1. More immune to noise. 2. Synchronization between transmitter and receiver is not required. PWM still works if synchronization between transmitter and receiver fails, whereas PPM does not. 3. Possible to separate out signal from noise. 4. PWM has advantage, when compared with PPM, that is its pulses are of varying width and therefore of varying power content. Applications of PWM • PWM is used in special purpose communication systems mainly for military but is seldom used for commercial digital transmission system. Pulse Position Modulation (PPM) PPM is when the position of a constant width and constant amplitude pulse within prescribed time slot is varied according to the amplitude of the modulating signal. PPM has the advantage of requiring constant transmitter power output. But, it has the disadvantage of depending on transmitter-receiver synchronization. PPM has less noise due to amplitude changes, because the received pulses may be clipped at the receiver, thus removing amplitude changes caused by noise. Advantages of PPM 1. Good noise immunity. 2. Requires constant transmitter power output. Disadvantages of PPM 1. Requires synchronization between transmitter and receiver. 2. Large Bandwidth requirement. Applications of PPM 1. It is used for optical communication system where there is no multipath interference. 2. PPM is useful for narrowband FM channel allocation, with these channel characteristics in the radio control and model aircraft, boats and cars. 3. PPM is also used for military applications. Comparison of PAM, PWM and PPM Digital Pulse Modulation It is mainly of two types: Pulse Code Modulation(PCM) Delta Modulation(DM) Pulse Code Modulation(PCM) Pulse-Code Modulation (PCM) is the most commonly used digital modulation scheme. PCM is a form of digital modulation where group of coded pulses are used to represent the analog signal. The analog signal is sampled and converted to a fixed length, serial binary number for transmission. In PCM, the available range of signal voltages is divided into levels and each is assigned a binary number. Each sample is represented by a binary number and transmitted serially. The number of levels available depends upon the number of bits used to express the sample value. PCM consists of three steps to digitize an analog signal: 1. Sampling 2. Quantization 3. Binary encoding Before we sample, we have to filter the signal to limit the maximum frequency of the signal . Filtering should ensure that we do not distort the signal, i.e. remove high frequency components that affect the signal shape.
Fig. A block diagram of PCM system (single channel)
Principles of PCM Three main process in PCM transmission are sampling, quantization and coding. 1. Sampling– is a process of taking samples of information signal at a rate of Nyquist’s sampling frequency. 2. Quantization– is a process of assigning the analog signal samples to a predetermined discrete levels. The number of quantization levels ,L, depends on the number of bits per sample, n, used to code the signal. Where, The magnitude of the minimum step size of the quantization levels is called resolution, ∆V. It is equal in magnitude to the voltage of the least significant bit of the magnitude step size of the digital to analog converter (DAC). The resolution depends on the maximum voltage, Vmax, and the minimum voltage Vmin of the information signal, where, Quantization error or quantization noise is the distortion introduced during the quantization process when the modulating signal is not an exact value of the quantized level. It is the difference between original signal and the quantized signal magnitude that is :
Quantization error can be reduced by increasing the
number of quantization level BUT this will increase the ENCODING: This is a process where each quantized sample is digitally encoded into n-bits code word.
Where, n = number of bits/sample and
L = number of quantization levels Transmission bit rate (R):is the rate of information transmission (bits/sec). It depends on the sampling frequency and the number of bit per sample used to encode the signal and is given by Fig: Components of PCM encoder To recover an analog signal from a digitized signal we follow the following steps: We use a hold circuit that holds the amplitude value of a pulse till the next pulse arrives. We pass this signal through a low pass filter with a cutoff frequency that is equal to the highest frequency in the pre-sampled signal. The higher the value of L, the less distorted a signal is recovered. Fig: Components of PCM decoder Delta Modulation In Delta modulation, only one bit is transmitted per sample. That bit is a one if the current sample is more positive than the previous sample, and a zero if it is more negative. Since so little information is transmitted, delta modulation requires higher sampling rates than PCM for equal quality of reproduction. This scheme works well for small changes in signal values between samples. If changes in amplitude are large, this will result in large errors. Components of Delta Modulation Delta demodulation components Examples: 1. A complex low-pass signal has a bandwidth of 200 kHz. What is the minimum sampling rate for this signal? Solution The bandwidth of a low-pass signal is between 0 and f, where f is the maximum frequency in the signal. Therefore, we can sample this signal at 2 times the highest frequency (200 kHz). The sampling rate is therefore 400,000 samples per second. 2. We want to digitize the human voice. What is the bit rate, assuming 8 bits per sample? Solution: The human voice normally contains frequencies from 0 to 4000 Hz. So the sampling rate and bit rate are calculated as follows: 3. A sinusoidal input wave of 3kHz is to be sampled at the lowest rate for transmission as pulses. Calculate the minimum sampling frequency required, so that all components of the wave can be reconstructed at the receiver. 4. The PCM sampled are encoded into 4-bits system. If the minimum sampling rate used is 8kHz, calculate a) The frequency of the information signal b) The quantization level. c) The transmission rate d) The transmission bandwidth THANK U!!!!!