Digital and Analog Transmission

Digital Transmission

Digital Transmission A computer network is designed to send information from one point to another. This information needs to be converted to either a digital signal (digital transmission) or an analog signal (analog transmission). In Digital transmission: (1) methods which convert digital data to digital signals (digital-to-digital conversion) and (2) methods which convert analog signals to digital signals (analog-to-digital conversion)

Digital-to-Digital conversion Line coding is the process of converting binary data, to a digital signal (Unipolar, Polar and Bipolar).

Analog-to Digital Conversion The techniques described earlier convert digital data to digital signals. Sometimes, however, we have an analog signal such as one created by a microphone or camera. Since digital signal is superior to analog signal in processing, the tendency today is to change analog signal to digital signal. The most common technique to change an analog signal to digital signal is called pulse code modulation (PCM).

PCM A PCM encoder has three processes: 1. Sampling: The analog signal is sampled 2. Quantization: The sampled signal is quantized 3. Encoding: The quantized values are encoded as stream of bits.

From analog signal to PCM digital code

Sampling Line coding and block coding can be used to convert binary data to a digital signal. If we want to store voice recording in the computer or send it digitally, we need to change it through a process called sampling. The sampling process is sometimes referred to as pulse amplitude modulation (PAM) After sampling, encoding can be used to convert it to a digital signal ready for transmission. Digital signals are less prone (rentan) to noise and distortion. A small change in an analog signal can change the received voice substantially, but it takes a considerable change to convert a 0 to 1 or a 1 to 0.

PAM

Pulse Code Modulation (PCM) Pulse Code Modulation (PCM) modifies the pulses created by PAM to create a completely digital signal. To do so, PCM first quantizes the PAM pulses. Quantization is a method of assigning integral values in a specific range to sampled instances. Each value is translated into its 7-bit binary equivalent, while the eighth bit indicates the sign. The binary digits are then transformed to a digital signal by using one of the line coding techniques. PCM is the sampling method used to digitize voice in T-line transmission in the North American telecommunication system

Quantized PAM signal

Quantizing by using sign and magnitude

PCM

Note: According to the Nyquist theorem, the sampling rate must be at least 2 times the highest frequency.

Nyquist theorem

Example 1 What sampling rate is needed for a signal with a bandwidth of 10,000 Hz (1000 to 11,000 Hz)? Solution The sampling rate must be twice the highest frequency in the signal: Sampling rate = 2 x (11,000) = 22,000 samples/s

Example 2 A signal is sampled. Each sample requires at least 12 levels of precision (+0 to +5 and -0 to -5). How many bits should be sent for each sample? Solution We need 4 bits; 1 bit for the sign and 3 bits for the value. A 3-bit value can represent 23 = 8 levels (000 to 111), which is more than what we need. A 2-bit value is not enough since 22 = 4. A 4-bit value is too much because 24 = 16.

Example 3 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. Sampling rate = 4000 x 2 = 8000 samples/s Bit rate = sampling rate x number of bits per sample = 8000 x 8 = 64,000 bps = 64 Kbps

Transmission Mode

Parallel Transmission In parallel mode, multiple bits are sent with each clock tick. In serial mode one bit is sent with each clock tick. In parallel transmission n wires are used to transmit n bits. This is limited to short distances as wiring is expensive

Parallel transmission

Serial Transmission In serial transmission, one bit follows another; so we need only one communication channel rather than n-channels to transmit data between two communicating devices. Serial communication reduces the cost by roughly a factor of n. Serial transmission occurs in of two ways: asynchronous or synchronous

Serial transmission

Asynchronous transmission In asynchronous transmission the timing or a signal is unimportant. To alert the receiver of the arrival of a new group of bits (usually a byte), an extra bit (0) (start bit) is added at the beginning of each byte. Another bit (1) (stop bit) needs to be transmitted at the end of the group to indicated that the transmission of a byte is finished. In addition, the transmission of each byte may then be followed by a gap of varying duration. This can be either represented by an idle channel or by a stream of additional stop bits. Asynchronous communication is slow but it is cheap and effective. It is used for communication between the keyboard and a computer

Note: In asynchronous transmission, we send 1 start bit (0) at the beginning and 1 or more stop bits (1s) at the end of each byte. There may be a gap between each byte.

Asynchronous transmission

Synchronous transmission In synchronous transmission, the bit stream is combined into ‘longer’ frames which may contain multiple bytes. Each byte is introduced on to the transmission link without a gap between two bytes. It is left to the receiver to separate the bit stream into bytes for decoding purposes. The advantage of synchronous transmission is speed, that’s why it is used for data transmission between computers.

Note: In synchronous transmission, we send bits one after another without start/stop bits or gaps. It is the responsibility of the receiver to group the bits.

Synchronous transmission

Analog Transmission

Analog Transmission Digital transmission is very desirable but a low-pass channel with a very large bandwidth is needed. Analog transmission is the only choice if we have a band-pass channel. We discuss methods for converting: Digital data to a bandpass analog signal (Digital-to-Analog conversion) A lowpass analog signal to a bandpass analog signal (Analog-to-Analog conversion)

Digital-to-Analog Conversion Digital-to-analog conversion is the process of changing one of the characteristics (amplitude, frequency, or phase) of an analog signal based on the information in a digital signal (0s and 1s). Transmitting data from one computer (digital) to another across a public access phone line (carries analog signals) requires that the digital data must modulated on an analog signal that has been manipulated to look like two distinct values corresponding to binary 0 and 1.

Digital-to-analog conversion

Types of digital-to-analog modulation

Note: Bit rate is the number of bits per second. Baud rate is the number of signal units per second. Baud rate is less than or equal to the bit rate.

Bit rate and Baud rate A signal unit is composed of one or more bits. The fewer signal units required, the more efficient the system and less bandwidth required to transmit more bits. The baud rate determines the bandwidth required to send the signal NOT the number of bits. In transportation, a baud is analogous to a car, and a bit is analogous to a passenger. A car can carry one or more passengers. The number of cars not the number of passengers determines the traffic and therefore the need for wider highways

Example 4 An analog signal carries 4 bits in each signal unit. If 1000 signal units are sent per second, find the baud rate and the bit rate Solution Baud rate = 1000 bauds per second (baud/s) Bit rate = 1000 x 4 = 4000 bps

Example 5 The bit rate of a signal is 3000. If each signal unit carries 6 bits, what is the baud rate? Solution Baud rate = 3000 / 6 = 500 baud/s

Carrier Signal In analog transmission, the sending device produces a high-frequency signal that acts as a basis for the information signal. This basic signal is called the carrier signal or carrier frequency. The receiving device is tuned (menyesuaikan) to the frequency of the carrier signal that it expects from the sender. Digital information then modulates the carrier signal by modifying one or more of its characteristics (amplitude, frequency, or phase). This is called modulation or shift keying and the information signal is called the modulating signal.

Amplitude Shift Keying (ASK) In ASK, the strength of the carrier signal is varied to represent binary 1 or 0. Both frequency and phase remain constant while the amplitude changes. ASK is highly susceptible (ren to noise interference because noise mainly affects amplitude.

ASK

Frequency Shift Keying (FSK) The frequency of the signal during each bit duration is constant, and its value depends on the bit (0 or 1). Both peak amplitude and phase remain constant. In FSK, the frequency of the carrier is varied to represent binary 1 or 0.

FSK

Phase Shift Keying (PSK) Both peak amplitude and frequency remain constant as the phase changes. The phase of the signal during each bit duration is constant, and its value depends on the bit (0 or 1). For example a phase shift of 0° can represent binary 0 and a phase shift of 180 ° to represent binary 1. This is then called 2-PSK or binary PSK, because only two phases are used. In PSK, the phase of the signal is varied to represent binary 1 or 0.

PSK

Phase Shift Keying (PSK) A constellation or phase-state diagram shows the relationship between phase and bit value. PSK is not susceptible to the noise degradation that affects ASK or to the bandwidth limitations of FSK. Therefore, we can use four variations of a signal, each representing 2 bits. This technique is called 4-PSK or Q-PSK. The pair of bits represented by each phase is called a dibit.

Phase Shift Keying (PSK) The 4-PSK idea can be extended to 8-PSK. Instead of 90° , we can vary the signal by shifts of 45°. With eight different phases, each shift can represent 3 bits (a tribit). The minimum bandwidth required for PSK transmission is the same as that required for ASK transmission (for the same reasons). While the maximum baud rates of ASK and PSK are the same for a given bandwidth, PSK bit rates using the same bandwidth can be 2 or more times greater.

PSK constellation

The 4-PSK method

The 4-PSK characteristics

The 8-PSK characteristics

Quadrature Amplitude Modulation (QAM) PSK is limited by the ability of the equipment to distinguish small differences in phase. So far we have been altering only one of the three characteristics of a sine wave at a time. What if we alter two? Bandwidth limitations make combinations of FSK with other changes practically useless. As such, we will combine ASK with PSK.

Quadrature Amplitude Modulation (QAM) If we have x variations in phase and y variation in amplitude, we will have x*y possible variations. This is called Quadrature Amplitude Modulation (QAM). Possible variations of QAM are numerous. To avoid noise interference, QAM always uses more phase shifts than amplitude shifts.

Note: Quadrature amplitude modulation is a combination of ASK and PSK so that a maximum contrast between each signal unit (bit, dibit, tribit, and so on) is achieved.

The 4-QAM and 8-QAM constellations

Time domain for an 8-QAM signal

Quadrature Amplitude Modulation (QAM) The first example (3 amplitudes and 12 phases) handles noise best because of a greater ratio of phase shift to amplitude. It is the ITU-T recommendation. The second example (4 amplitudes, 8 phases) is the ISO recommendation. Not all intersections in the graph are utilized out of 32 possible variations. QAM has a lower susceptibility to noise compared to ASK. The minimum bandwidth required for QAM is the same as that required for ASK and PSK.

16-QAM constellations

Table 5.1 Bit and baud rate comparison ASK, FSK, 2-PSK Bit 1 N Modulation Units Bits/Baud Baud rate Bit Rate ASK, FSK, 2-PSK Bit 1 N 4-PSK, 4-QAM Dibit 2 2N 8-PSK, 8-QAM Tribit 3 3N 16-QAM Quadbit 4 4N 32-QAM Pentabit 5 5N 64-QAM Hexabit 6 6N 128-QAM Septabit 7 7N 256-QAM Octabit 8 8N

Example 6 A constellation diagram consists of eight equally spaced points on a circle. If the bit rate is 4800 bps, what is the baud rate? Solution The constellation indicates 8-PSK with the points 45 degrees apart. Since 23 = 8, 3 bits are transmitted with each signal unit. Therefore, the baud rate is 4800 / 3 = 1600 baud

Telephone Modems Traditional phone lines can carry frequencies between 300 and 3300 Hz, giving them a bandwidth of 3000 Hz. All this range is used for transmitting voice, where a great deal of interference and distortion can be accepted without loss of intelligibility. Data signals, however, require a higher degree of accuracy to ensure integrity. To be on the safe side, the edges of the bandwidth range are not used for data communication. The effective bandwidth of a telephone line being used for data transmission is 2400 Hz (600-3000 Hz).

Note: A telephone line has a bandwidth of almost 2400 Hz for data transmission.

Telephone line bandwidth

Modem stands for modulator/demodulator. Note: Modem stands for modulator/demodulator.

Modulation/demodulation

Telephone Modems A modulator creates a band-pass analog signal from binary data. A demodulator recovers the binary data from the modulated signal. Today, the most popular modems available are based on the V-series standards published by ITU-T

Traditional Modems In traditional modems data exchange is between two computers, A and B through the digital telephone network. After modulation by the modem, an analog signal reaches the telephone company switching station, where it is sampled and digitized to be passed through the digital network. The quantization noise introduced into the signal at the sampling point limits the data rate according to Shannon capacity. This limit is 33.6Kbps. Because the sampling point exists in both directions, the maximum data rate is 33.6Kbps.

Traditional modems

56K Modem Communication today is via the internet. We still use modems to upload data to the Internet and down load data from the Internet In uploading, the analog signal must still be quantized at the switching station, which means that the data rate in uploading is limited to 33.6Kbps.However, there is no sampling in the downloading. The signal is not affected by quantization and hence the maximum data rate in the down loading direction is 56Kbps

56K modems

Analog-to-Analog conversion Analog-to-analog conversion or analog modulation is the representation of analog information by an analog signal. Why we need to modulate an analog signal, it is already analog? Modulation is needed if the medium is bandpass in nature or if only a bandpass channel is available. An example is radio. The government assigns a narrow bandwidth to each radio station. The analog signal produced by each station is a low-pass signal, all in the same range. To be able to listen to different stations, the low-pass signals need to be shifted, each to a different range. Analog-to-analog conversion can be accomplished in three ways: AM, FM and PM.

Analog-to-analog modulation

Types of analog-to-analog modulation

Note: The total bandwidth required for AM can be determined from the bandwidth of the audio signal: BWt = 2 x BWm.

Amplitude modulation

AM bandwidth

AM band allocation

Frequency Modulation (FM) In FM transmission, the frequency of the carrier signal is modulated to follow the changing voltage level (amplitude) of the modulating signal. The peak amplitude and phase of the carrier signal remains constant. The bandwidth of an FM signal is 10 times the bandwidth of the modulating signal. The bandwidth of an audio signal (speech and music) broadcast in stereo is almost 15KHz. Each FM station needs therefore a bandwidth of 150KHz. The FCC allows 200 KHz for each station to provide some room for guard bands. .

Note: The total bandwidth required for FM can be determined from the bandwidth of the audio signal: BWt = 10 x BWm.

Frequency modulation

Figure 5.30 FM bandwidth

Note: The bandwidth of a stereo audio signal is usually 15 KHz. Therefore, an FM station needs at least a bandwidth of 150 KHz. The FCC requires the minimum bandwidth to be at least 200 KHz (0.2 MHz).

FM band allocation

Phase Modulation (PM) Due to simpler hardware, PM is used in some systems as an alternative to FM. In PM transmission, the phase of the carrier signal is modulated to follow the changing voltage level (amplitude) of the modulating signal. The peak amplitude and frequency of the carrier signal remain constant. The analysis and final result (modulated signal) are similar to those of frequency modulation