Communication in networks requires at least two parties; the sender and recipient, along with the medium for transmission and the message/ data to be transmitted. All communications start with the sender, who must figure out how to encode the data/ information to be sent, so as to convey meaning. Encoding refers to the process of putting a character sequence, such as numbers, symbols, and letters into a special format to enable efficient transmission and in some cases, storage. The encoded information is then decoded at the recipient end where the encoded information is encoded (converted) back into the original character sequence. The encoding aims at producing a signal from the encoded data stream to be transmitted over a specific medium; the encoding must take into consideration the medium for transmission. Encoding is the process of using a code to convert original information/ data into a form usable by an external process. The process of encoding defines the way that signals are represented on a given (physical) line of communication; hence, there is a need to have a format that is standard for seamless communication [1]. The encoded signal is manipulated in such a way that the sender and the recipient can recognize the changes (encoding) made. Information to be sent over a network can be analog (such as voice and video data) or digital.
Networks have become more complex and highly interconnected; for instance, the World Wide Web is among the biggest network of interconnected computers and servers. Copious volumes of data are exchanged at any given time within networks, hence the need for encoding. Further, new transmission models, such as fiber optic have been developed, requiring a standardized method of transmitting information over different mediums until they reach their destination in a desirable format [2]. To ensure transmission is optimized, a signal has to be encoded to facilitate transmission over a physical medium that can be a copper cable, a network cable, or a fiber cable; wireless mediums also necessitate encoding. Encoding is essential in keeping the lines balanced even if the data keeps changing its state often enough and to make it distinguishable from a line that is dead. For instance, if the data stream has uneven 1s and 0s, an unbalanced offset voltage will be developed by the receiver. Encoding also ensures that long 1s or 0s strings are eliminated and that over time, the total number of 1s and 0s remains balanced all the time. Further, signals cannot be sent just in the forms of 0s or 1s; the signals have to be encoded into signals having two states, such as;
There are two main ways in which encoding can be achieved; two level encoding and three level encoding. In two-level encoding, the signal is encoded in such a way that it can only take on a value that is either strictly positive (+x) or strictly negative (-x), where x is a value of the physical quantity that is being used to transport signals (the physical medium). In three level encoding, the signal can take on a value that is strictly positive (+x), strictly negative (-x), or null (0). There are various methods for signal encoding, as discussed in the next section [3].
The network uses standard protocols for the transmission and delivery of data over a distance, with the network designed in layers, each performing a specific function. The standard network has various layers including the physical layer (layer 1), the data link layer (layer 2), and the presentation layer (layer 3). The Transmission of data is based on the TCP (Transmission Control Protocol) and it is found in layer one (physical layer), where cannel coding takes place. Data transmission refers to the transfer of data over a point to another point or multipoint communication channels. The channels used for transmissions include optical fiber, copper wires, computer buses, wireless communication channels, and storage media. The data being transmitted is presented in the form of electromagnetic signals, for example radio waves, microwaves, infrared signals, or electrical voltages. There are various kinds of data transmission, including analog transmission where data, voice, image, video information or signals are transmitted using a continuous signal that varies in phase, amplitude, or other property in direct proportion to that of a variable [2]. The messages are represented by a set of limited continuous varying wave forms (a process termed pass band transmission) through digital modulation, or by a pulse sequence using a line code (a method termed baseband transmission). Modem equipment undertakes the process of passband modulation and its corresponding demodulation. The bit streams represented by both passband and baseband transmission are both considered digital transmission. The transmitted data can be an analog source, such as from video or phone call digitized into bit stream signals or digital such as from a keyboard. Before being transmitted over networks, information/ data must be encoded before being transported across a given media such as copper or fiber optic. This implies that the current or voltage waveform pattern used for representing 1s and 0s are encoded by adjusting them. There are various ways in which encoding of signals for transmission over networks can be achieved;
The chosen mechanism for encoding depends largely on the technology available at a given time and the application needs.
This is one of the oldest methods for encoding and entails converting analog data into digital signals and was (is) employed in telephony to transmit voice data over telephone (copper) lines. Analog data is represented in their baseband as analog frequencies. For instance, when using a telephone, there are two ways in which the data is sent; the analog voice signal is transmitted at the baseband signal. Alternatively, the data can be transmitted by combining the signals into another signal which acts as the carrier and the combined signals transmitted at a different frequency. When the pitch of a signal wave is modified, its termed frequency modulation; modifying a wave’s strength is termed amplitude modulation, while modifying its (wave) natural flow is termed phase modulation [4]
AM works by having the data/ message encoded in the amplitude of a signal pulse series; the amplitudes of carrier pulses train are varied based on the sample value of the data/ message signal. The bulk of carrier signals and and a waveform in the baseband with the lower sideband being slightly lower than the frequency of the carrier while the upper side band is slightly higher [5]. AM can is expressed as;
s(t) = [1+na x(t)] cos 2 πfc t
The resulting signal envelope is 1+na x (t)
As long as na <1, the envelope becomes the original signal’s exact reproduction. If na >= 1, the standard AM modulator will fail since as the wave envelop negative excursions cannot fall below zero; therefore, the received modulation becomes distorted. Because of its nature, the author feels it’s more suited for wide area applications but unsuitable for situations where quality is required as it has levels of noise. Further, it’s unsuitable for data or multimedia because of a limited bandwidth and has a weaker signal
Is an analog data encoding method in which data is encoded into an alternating current (AC) wave through changing the waves instantaneous frequency and can be used for encoding either digital or analog data. In digital FM, the data is in the form of 0s and 1s, and there are abrupt signal changes of the carrier frequency. The number of carrier frequencies is represented in powers of two (2), corresponding with the ON/OFF frequency states. Analog FM has a continuous/ smooth AC carrier wave that can be represented as a sine wave [6]. If the baseband signal is x m (t) and the sinusoidal carrier wave is
Xc (t) = Ac cos (2 π fc t)
Where fc is the base frequency of the carrier and Ac is its amplitude the modulator combines the baseband data signal with the carrier to obtain the transmitted signal;
The author feels FM is suitable for data transfers over short distances due to a larger bandwidth, low noise, is more efficient as it requires less amplification during transmission and compression (photo-acoustic) can be applied to it. However, it requires complicated demodulators due to the need for amplitude limiter.
This involves conveying digital signals by shifting phases; this technique is basically used for satellite communication and digital signaling. The phase numbers used in representing the information being transmitted can significantly impact the amount of transmitted information. When more than two phases are used, it is termed multi-level signaling [7]. This, on further evaluation, is an easier encoding method compared to FM and more information such as Doppler can be obtained with PSM. However, extending its modulation beyond 180 degrees results in phase ambiguity and it requires frequency multipliers, limiting its application for modern data transfer needs
This is a method of encoding that is at present, commonly used in transmitting data over digital facilities, such as computer data over networks. This model uses less complex equipment as well as being less expensive, compared to methods such as digital to analog. Digital signals are discrete sequences of discontinuous voltage pulses with every pulse having a signal element. It entails encoding binary data bit into signal elements in order to transmit data. The encoding scheme entails mapping signal elements from data bits; a mark is the binary digit 1 while the binary digit 0 signifies a space. The common techniques used in digital to digital encoding include NRZ (no return to zero) encoding, NRZI (no return to zero inverted) encoding, Bipolar AMI (alternate mark inversion), and Manchester Encoding.
This is among the simplest and earliest used encoding systems for digital to digital encoding; it entails transforming the 1s into -X and the 0s into +X resulting into a bipolar encoding where the signal can never be null. As a consequence, the recipient of the message/ data can determine easily whether there is a signal present or not. NRZ is often used in slow speed type of communications interfaces for asynchronous and synchronous data transmissions [13]. I think this technique makes synchronization difficult and causes higher power losses for transmitted DC power and adds costs to transmission as the lines must be DC coupled. Further, it’s difficult to achieve clock recovery from signals and errors may be introduced over long distances
This type of digital to digital encoding entails the signal changing state after ticking of the clock when the bit value is 1. When value of the bit is 0, there is no change in state of the signal value. This type of encoding is advantageous because it enables detecting whether there is a signal or not and the transmission current is low voltage, albeit with the problem of continuous current during sequences of 0s [8]. This method is suitable for data transmissions over long distances because many transitions can be introduced to enable clock recovery from the signal and so limit errors; a clear advantage over NZR.
This is a digital to digital encoding where transitions from one logical state to another state represents data bits and each data bit length is set by default. It entails an exclusive performance of the OR (XOR) of a signal with the clock signal that results into a raising edge when the value of the bit is 0 and a falling edge in the opposite case. The direction of the transition determines the state of a bit. Data encoded using Manchester Encoding contains frequent transition levels that allow the extraction of the clock signal by the receiver using the DPLL (digital phase locked loop) [8]. Manchester encoding works based on the rules shown below;
|
Original data |
Sent value |
|
Logic 1 |
1 to 0; downward transition at the bit center |
|
Logic 0 |
0 to 1; transition at the bit center |
There are various types of digital data to analog signal data encoding for data transmission. Digital to analog encoding works on the principle of shift keying using the following techniques;
This equates the digital to analog encoding of AM (amplitude modulation) in which digital data is represented as variations in the carrier wave amplitude. With ASK, the 1 (binary symbol) representation is attained by transmitting a carrier wave with a fixed amplitude and fixed frequency for T seconds of bit duration period. If the signal has a value of 1, the carrier signal is transmitted; however, if the value of the signal is 0, then a 0 value signal is transmitted. Digital data transmission over optical fiber is achieved using the ASK technique. In LED transmission, a short light pulse represents the binary 1 with the absence of light represented by the binary 0. in Laser transmission, there is a fixed bias current that results in the device emitting light at low levels; the low level light represents the binary 0 while binary 1 is represented by higher amplitude light.
Where
ht(f) = transmission carrier signal
hc(f)= channel impulse response
n(t) = channel introduced noise
hr(f) = the receiver filter
L = level numbers used in transmission
Ts = Time taken between two symbols generation
When data is coming out of a transmitter, the signal s(t) is expressed using the relation;
s(t) = v[n] . ht (t- nTs)
After filtering in the receiver, the hr (t), the signal is expressed as
z(t) = nr(t) + . gt (t- nTs) [11]
I think it is useful for digital data transmission over optical fibers due to high bandwidth efficiency and the modulation-demodulation processes are inexpensive along with simple receiver design. This means it can be efficiently applied to networks such as large data-centers and WANs. Frequency Shift Keying (FSK)
In FSK, only the frequency changes with the phase and amplitude remaining unchanged. In this modulation scheme, the digital data is transmitted via discrete changes in frequency of the carrier signal. FSK is used in applications such as remote metering and in caller ID in telephone lines. FSK helps solve the line noise challenges posed by ASK since the receiver is tuned to a specific frequency. FSK needs two carrier signals, the lower frequency for the binary 0s and the higher frequency one for the 1s binary signals. FSK is used for encoding in applications such as modems, to a maximum frequency of 1200 bps; it works by making guard bands whose role is to avoid overlap in signals [11]. Using low power micro controllers, the binary FSK signal can be demodulated fast and efficiently using the Goertzel algorithm which has two stages; the first to compute the intermediate sequence y[n];
s[n] = x[n] + 2 cos (w0)s [n-1] – s [n-2]
and the second stage that applies the filter s[n] to generate the output sequence y[n];
y[n] = s[n] – e -jwo s [n-1]
The image below depicts FSK operation ;
Digital to analog encoding works by converting data using modulation, by changing the reference signal phase (carrier wave). The cosine and sine inputs are varied at precise times to attain modulation. Binary bit 0 has a phase shift of 0, but the binary bit 1 has a phase shift of 180. The wave shift in signals is achieved through delaying the signal while retaining its frequency and/ or amplitude. The modified signal is assigned a given binary value, such as 1, while the carrier signal that remains unchanged has the binary value 0. A finite number of well-defined signals are used to represent data [11]. The two binary phases are represented by 0 and 180 degrees and the resulting one bit time transmitted signal is given by the equation. Its characteristics make it highly suitable in modern data transfer needs such as RFID, Bluetooth, and WLANs (wireless local area networks); the diagram shows the process;
This is a method of encoding that can be used to either decrease or increase the amounts of data sent without the need for increasing bandwidth. QAM operates by combining PSK and ASK to give maximum contrast between tri-bits, di-bits, and quad bits. The recommended QAM is the 16 QAM in which three amplitudes and twelve phases are combined; this is because it is the most efficient model for reducing noise because the ratio of phase shifts to amplitude is the highest. The bit rate then becomes thrice the baud rate to further enhance its efficiency [12]. Two signals transmitted through QAM modulation will be in the form;
S(t) = Re {[I(t) + iQ (t)] e12πF0t }
= I(t) cos (2πf0t) – Q(t) sin 2 πf0t
Where i2 is -1, I(t) and Q(t) are modulating signals, Re{} is the real section, and f0 is carrier frequency. Analog to Digital Encoding
This is a form of encoding popularly known as digitization and is achieved either by pulse code modulation or delta modulation. Quantization and sampling are the main important factors in analog to digital modulation
This is a form of encoding popularly known as digitization and is achieved either by pulse code modulation or delta modulation. Quantization and sampling are the main important factors in analog to digital modulation Pulse Code Modulation (PCM)
This is a technique of converting analog data into digital signals such as in computer digital audio, digital telephony, and CDs (compact discs). In the PCM stream, the analog signals’ amplitude is regularly sampled at uniform intervals, with every sample being quantized within a given range of digital steps, to the nearest value. PCM enables analog data to be converted into digital signals to enable the transmission of the analog signal through digital communication channels and networks. The signal is transported in digital format and then converted again into analog signals at the receiving end. PCM entails three main steps; sampling, followed by quantization, and then coding. Before sampling, the signal must be filtered first before sampling so the maximum signal frequency is limited as this has a direct effect on the rate of sampling. Sampling occurs when every Ts seconds (sampling interval), an analog signal is sampled [15].
Fs = 1/Ts becomes the sampling frequency. Sampling can be ideal, natural, or flat top. Sampling is a form of PAM (pulse amplitude modulation) that results in an analog signal. It is can be applied to fields such as video streaming and VOIP very well, however, it needs a large bandwidth.
The PCM signal bit rate can be computed from the relation
Bit rate = nb x fs
Where fs is sampling rate and nb is the numbers of bits in each sample
This is a technique for converting analog data into digital signals and is frequently used for transmitting voice data in applications where the quality is not of primal importance. Oversampling techniques are utilized to ensure the signal to noise ratio is high; meaning less noise in the transmitted data by sampling the signal at rates several times higher than Nyquist rate [14]. Delta modulation restricts the input signal amplitude since if the transmitted signal possesses a large derivative, the modulated signal will fail to follow the input signal, resulting in slope overload.
If input signal is m(t) = A cos (wt),
The input signal derivative becomes
{m (t)] max = wA
The required condition for avoiding slope overload is
[m (t)]max = wA < afs
As such, the the input signal maximum amplitude becomes
Amax = afs/w
Where;
fs = sampling frequency
w = input signal frequency
a = quantization step size
Amax = maximum possible amplitude that can be transmitted by DM without causing slope overload [15].
This is a recent technique for encoding used in wide band digital communication in applications such as audio and television broadcasting, wireless networks, DSL internet access, 4 G mobile communication, and power line networks. It is a frequency division multiplexing scheme employed as a multi carrier method of modulation. It work s by using a large number of orthogonal sub carrier signals that are closely spaced for carrying data on multiple parallel data channels/ streams. A conventional modulation scheme such as PSK or QAM is used for modulation each carrier at low symbol rates. This ensures total data rates are maintained in a way similar to schemes used in single carriers on the same bandwidth. OFDM has benefits over single carrier schemes such as FM is that it can cope well with channel conditions that are severe such as high frequency attenuation in very long copper wire mediums or narrow band interference [16]
This is a technique for combining multiple digital and analog signals into a single signal over shared mediums in which case the medium is a scarce resource. This entails dividing the channel into several logical channels, with each channel handling a single data stream or signal. At the r3ceiver end, de-multiplexing reverses this process to recover the signal. A common approach to multiplexing is the frequency division multiplexing which can be considered an analog method. Several distinct signal frequencies are sent over a single channel using electrical signals and is applied in cable television and tv channel broadcasting [17]
Conclusion
Encoding is an integral and highly essential component in networks and data transmission to enable transmission of data between two or more terminals. Encoding also ensures the lines are balanced even when there are changes in the state of the data as to be indistinguishable from a dead line or has uneven numbers. Encoding allows different media to be transmitted, from voice to videos and computer data in a standardized manner. Different techniques are used for data transmission, depending on the type of data, the medium, and distance as well as the application. There are four major ways of encoding, namely analog to analog (AM, FM, PSM), digital to digital (NRZ, NRZI, ME), analog to analog, and digital to digital (ASK FSK PSK QAM). These are standard/ regularly used methods; however, as applications become more resource intensive such as video streaming and cloud computing new approaches to handle such demands are necessary. New techniques include orthogonal frequency division multiplexing and advanced methods of multiplexing
References
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