Simple Telecom : Technologies

Telecommunication transmission technology is often lumped into two categories: analog transmission or digital transmission. Figure 4.1 depicts a basic telecommunications transmission system that transfers digital information from one source to an information receiver. The information source is supplied to an end-node that converts the information to a form that can be transmitted through the transmission medium (air, copper, or fiber). The receiving end-node converts the transmission signal into a form that is compatible for the receiver of the information.


Basic Transmission System

Analog
Analog transmission is a process of transferring signals between end-nodes than can have many different signal levels and frequencies. Because analog signals can continuously change to many different levels (voltages) at changing rates (frequencies), the transfer of analog signals (such as an audio signal) requires the transmission medium to have similar transfer characteristics to all parts (levels and frequencies) of the transmission signal. Analog transmission systems must be robust to transfer the signal unaltered for specific voltage levels and frequency components (e.g., high and low frequency).

Figure below shows an analog transmission system. This diagram shows that an audio acoustic (sound) signal is converted by a microphone to an audio electrical signal prior to transmission on a copper line. This audio electrical signal is amplified by an end-node to increase the signal level for transmission on a copper wire. This amplification is necessary to overcome the transmission loss of the copper wire. As the signal progresses down the copper wire, some of the signal energy is converted to heat reducing the signal level. Another amplifier (the receiving end-node) is located at the receiving end to increase the signal to a level suitable for the information receiver (audio speaker).


Analog Transmission System

Digital
Digital transmission is the process of transferring information from node to node in a form that can only have specific levels (usually logic 1 and logic 0). Digital signals have a limited number of different levels (voltages) that represent digital information. Transferring digital signals (such as a computer’s data signal) only requires the transmission medium to transfer two levels without precisely (linearly) transferring levels in between the two levels.

Figure below shows a digital transmission system. This diagram shows a computer that is sending digital data (one equal to +5 volts and zero equal to 0 volts) to an end-node. The end-node is a channel service unit (CSU) and digital service unit (DSU) that converts the levels from the computer to levels suitable for the copper wire transmission medium (logic 1 = +5V and logic 0 = -5V). As the digital signal transfers down the copper wire, some of the energy is converted to heat and some of the frequency components are attenuated resulting in a slightly distorted (rounded) digital pulse arriving at the receiving end-node. Because the receiving CSU/DSU only needs to sense two levels, it is able to re-create the original undistorted digital signal (also known as digital signal regeneration).


Digital Transmission System

Digital Signal Processing

Digital signals processing refers to a category of electronic devices that represent and process information that are in discrete signal level (digital) formats. Digital signal processing refers to the manipulation of digital signals to change their content and to add error detection and correction capability.

Digital signals typically vary in two levels; on (logic 1) and off (logic 0). A bit is the smallest part of a digital signal, typically called a data bit. A bit typically can assume two levels: either a zero (0) or a one (1). A byte is an agreed-upon group of bits, typically eight. A byte typically represents one alphabetic or special character, two decimal digits, or eight binary bits of information.

When analog signals are converted to digital format, the digital signals represent the original analog waveform. Just like analog signals that may be processed by filters, shaping circuits, combiners and amplifiers, digital signals can be processed to produce similar functions. However, because the signal is in digital form, these functions are performed by software programs that manipulate the data.

Unlike analog signals, digital signals can be recreated to their original form. This process is called signal regeneration. To increase the efficiency of a transmission signal (allow more users per channel), digital signals can be analyzed for redundancy and the digital signal data can be compressed. Digital signals can also be processed in a way that helps overcome the effects signal distortion that can result in the incorrect determination of a digital signal (whether a zero or one had been sent). This is called error detection and error correction processing. When digital signals represent the original analog signal, advanced echo canceling software programs can be used to reduce the effects of echoes that are caused by feedback in the audio and transmission system. Some systems use dedicated digital signal processors (DSPs) to manipulate the incoming digital information via a program (stored instructions) that produce a new digital output. This allows software programs to perform many functions (such as signal filtering) that previously required complex dedicated electronic circuits.

Digitization of an Analog Signal
Analog signals must be converted to digital form for use in a digital wireless system. To convert analog signals to digital form, the analog signal is digitized by using an analog-to-digital (pronounced A to D) converter. The A/D converter periodically senses (samples) the level of the analog signal and creates a binary number or series of digital pulses that represent the level of the signal.

The common conversion process is Pulse Code Modulation (PCM). For most PCM systems, the typical analog sampling rate occurs at 8000 times a second. Each sample produces 8 bits digital that results in a digital data rate (bit stream) of 64 thousand bits per second (kbps).

Figure below shows how an analog signal is converted to a digital signal. This diagram shows that an acoustic (sound) signal is converted to an audio electrical signal (continuously varying signal) by a microphone. This signal is sent through an audio band-pass filter that only allows frequency ranges within the desired audio band (removes unwanted noise and other non-audio frequency components). The audio signal is then sampled every 125 microseconds (8,000 times per second) and converted into 8 digital bits. The digital bits represent the amplitude of the input analog signal.


Signal Digitization

Digital bytes of information are converted to specific voltage levels based on the value (weighting) of the binary bit position. In the binary system, the value of the next sequential bit is 2 times larger. For PCM systems that are used for telephone audio signals, the weighting of bits within a byte of information (8 bits) is different than the binary system. The companding process increases the dynamic range of a digital signal that represents an analog signal; smaller bits are given larger values that than their binary equivalent. This skewing of weighing value give better dynamic range. This companding process increases the dynamic range of a binary signal by assigning different weighted values to each bit of information than is defined by the binary system.

Two common encoding laws are Mu-Law and A-Law encoding. Mu-Law encoding is primarily used in the Americas and A-Law encoding is used in the rest of the world. When different types of encoding systems are used, a converter is used to translate the different coding levels.

Digital Signal Regeneration
To overcome the effects of noise on transmitted signals, digital transmission systems use digital signal regeneration to restore the quality of the signal as it moves through a network. Digital signal regeneration is the process of reception and restoration of a digital pulse or lightwave signal to its original form after its amplitude, waveform, or timing have been degraded by normal factors during transmission. The resultant signal is virtually free of noise or distortion.

Figure below shows the process of digital signal regeneration. This example shows an original digital signal (a) and added noise (b) to produce a combined digital signal with noise (c). The regeneration process detects maximum and minimum expected values (threshold points) and recreates the original digital signal (d).


Digital Signal Regeneration

Data Compression
To increase the amount of information that a transmission system can transfer, digital systems may use data compression. Data compression is a processing technique for encoding information so that fewer data bits of information are required to represent a given amount of data. Compression allows the transmission of more data over a given amount of time and circuit capacity. It also reduces the amount of memory required for data storage.

Digital compression analyzes a digital signal for either redundant information (repeated 1’s or 0’s) or may analyze the information content of the digital signal into component parts (such as speech patterns or video frames). All of this processing allows the data transmission rate to be reduced by sending only the characteristics of the signal rather than the complete digital signal. Some data compression systems can only reduce data rates by a factor of 2:1 (e.g., ADPCM audio compression) while advanced digital audio compression can only reduce data rates by a factor of approximately 200:1 (e.g., MPEG video compression). When used in combination of data compression and decompression, the device is called a COder/DECoder (CoDec).

When a digital signal is compressed for voice communications, it is called a voice coder (Vo-coder) or speech coder. The Vo-coder is a digital signal processing device that analyzes speech signals so that it can produce a lower data rate compressed digital signal. The difference between standard data compression and voice data compression is the analysis of the information source (speech) and elimination of compression process for non-voice signals. Speech coding usually involves the use of data tables (called code books) that represent information parts that can be associated with human sound. Because non-human sounds can be eliminated from the code book, this allows the number of bits can be used to create a compressed digital voice signal to be reduced.

Figure below shows the digital voice compression process. In this example, a digital signal is continuously applied to a digital signal analysis device. The analysis portion of the speech coder extracts the amplitude, pitch, and other key parameters of the signal and then looks up related values in the code book for the portion of sound it has analyzed. Only key parameters and code book values are transmitted. This results in data compression ratios of 4:1 to over 16:1.


Digital Voice Compression

Error Detection and Error Correction
To help reduce the effects of errors on data transmission, error detection, and error protection systems are used in most communication systems. Error detection systems use a process of adding some data bits to the transmitted data signal that are used to help determine if bits were received in error due to distorted transmission. Error correction is made possible by sending bits that have a relationship to the data that is contained in the desired data block or message. These related bits permit a receiver of information to use these extra information bits to detect and/or correct for errors that may have occurred during data transmission.

A common measurement of the performance of a communication system is the amount of bits received in error, called the bit error rate (BER). The BER is the ratio of bits received in error compared to the total number of bits received.

Error detection processing involves the creation of additional bits that are sent with the original data. The additional check bits are created by using a formula calculation on the digital signal prior to sending the data. After the digital signal is received, the formula can be used again to create check bits from the received digital signal. If the check bits match, the original digital signal was received correctly. If the check bits do not match, some (or all) of the digital signal was received in error. This process is called error detection.

Some digital systems use sophisticated mathematical formulas to create the check bits so that the check bits can be used to make corrections (or predictions of the correct bits) to the received digital signal. This process is called error correction.

Figure below shows the basic error detection and correction process. This diagram shows that a sequence of digital bits is supplied to a computing device that produces a check bit sequence. The check bit sequence is sent in addition to the original digital bits. When the check bits are received, the same formula is used to check to see if any of the bits received were in error.


Error Detection and Correction

Echo Cancellation
Echo cancellation is a process of extracting a delayed version of an original transmitted signal (audio echo) from the received signal. Echoes may be created through acoustic feedback where some of the audio signal transferring from a speaker into a microphone.

Echoed signals cause distortion and may be removed by performing via advanced signal analysis and filtering. Figure 3.18 shows an example of the echo cancellation process. This diagram shows how the combining of two signals, the original plus a delayed version of the original produces a complex signal. The echo canceling system analyzes the complex signal and uses the signal analysis to create variations of the likely echo signal. This prediction of echoed signal is subtracted from the complex signal to reproduce the original signal without the echo.


Echo Cancellation

Echoed signals can also occur in signals other than audio signals. When echoes occur on radio channels (the broadband signal), it is usually the result of the same signal that travels on different paths to reach its destination. This is called multipath propagation. Echo canceling can be used to reduce the effects of radio multipath propagation.

Digital Signal Processor (DSP)
A digital signal processor (DSP) is an electronics device or assembly (typically an integrated circuit) that is designed to process signals through the use of embedded microprocessor instructions. The use of DSPs in communication circuits allows manufacturers to quickly and reliably develop advanced communications systems through the use of software programs. The software programs (often called modules) perform advanced signal processing functions that previously complex dedicated electronics circuits. Although manufacturers may develop their own software modules, DSP software modules are often developed by other companies that specialize in specific types of communication technologies. For example, a manufacturer may purchase a software module for echo canceling from one DSP software module developer and a modulator software module from a different DSP software module developer. Because DSPs use these software modules, if new technologies such as speech compression, channel coding, or modulation techniques are developed, the manufacturer only has to change the software programs in the DSP to utilize the new technology.

Figure below shows typical digital signal processor that is used in a digital communication system. This diagram shows that a DSP contains a signal input and output lines, a microprocessor assembly, interrupt lines from assemblies that may require processing, and software program instructions. This diagram shows that this DSP has 3 software programs, digital signal compression, channel coding, and modulation coding. The digital signal compression software analyzes the digital audio signal and compresses the information to a lower data transmission rate. The channel coding adds control signals and error protection bits. The modulation coding formats (shapes) the output signal so it can be directly applied to an RF modulator assembly. This diagram also shows that an optional interface is included to allow updating of the software programs that are stored in the DSP.


Digital Signal Processor (DSP)

Basic Communication Technologies : Signal Types

Telecommunication technology involves the transfer of information signals through wires, fiber, or through the air by the by means of electrical or optical signals. Communication signals are usually characterized by their intensity (voltage and current) and frequency (cycles per second). To allow information to be transferred using communication signals, an information source (audio, data or video) is either represented by the signal itself (called the baseband signal) or the information slightly changes the wave shape of the communication signal (called the broadband signal). The information is imposed on the carrying signal (called the carrier) by varying the signal level or time changes (frequency shift).

Signal Types
There are two basic types of signals: analog and digital. Many communication systems receive analog signals (e.g., audio signals), convert them to a digital format, transport the digital signals through a network, and reconvert the digital signals back to their analog form when they reach their destination.

Analog
An analog signal can vary continuously between a maximum and minimum value and it can assume an infinite number of values between the two extremes.

Figure below shows a sample analog signal created by sound. In this example, as the sound pressure from a person’s voice is detected by a microphone, it is converted to its equivalent electrical signal. Also, the analog audio signal continuously varies in amplitude (height, loudness, or energy) as time progresses.



Digital
Digital signals have a limited number of discrete states, usually two, in contrast to analog signals that vary continuously and have an infinite number of states. Digital signals transfer discrete signal levels at predetermined time intervals. Digital signals typically have two levels: on (logic 1) and off (logic 0). The information contained in a single time period is called a bit. The number of bits that are transferred in one second is called the data transfer rate or bits per second (bps). Because many bits are typically transferred in 1 second, the data rate is typically preceded by a multiplier k (thousand) or M (million). For example, if the data transfer rate is 3 million bits per second, 3 Mbps would indicate this. Bits are typically combined into groups of 8 bits to form a byte. When the reference is made to bytes instead of bits, the b is capitalized. For example, 10 thousand bytes is represented by kB. Figure below shows a sample digital signal. In this example, the bits 01011010 are transferred in 1 second. This results in a bit rate of 8 bps.



The earliest form of digital radio communication was Morse Code. To send Morse Code, the radio transmitter was simply turned on and off to form dots and dashes. The receiver would sense (detect) the radio carrier to reproduce the dots and dashes. A code book of dots and dashes was used to decode the message into symbols or letters. The on and off pulses or bits that comprise a modern digital signal is sent in a similar way.

The trend in communication systems, just as in other types of electronics products such as compact discs, is to change from analog systems to digital systems. Digital systems have a number of important advantages including the fact that digital signals are more immune to noise. Unlike analog systems, even when noise has been introduced, any resulting errors in the digital bit stream can be detected and corrected. Also, digital signals can be easily manipulated or processed in useful ways using modern computer techniques.

Video Communications

Video communication is the transmission and reception of video (multiple images) using electrical or optical transmission signals. Telecommunications systems can transfer video signals in analog or digital form.

Analog Video
Analog video is the representation of a series of multiple images (video) through the use of rapidly changing signals (analog). This analog signal indicates the position, luminance, and color information within the video signal.

Sending a video picture involves the creation and transfer of a sequence of individual still pictures called frames. Each frame is divided into horizontal and vertical lines. To create a single frame picture on a television set, the frame is drawn line by line. The process of drawing these lines on the screen is called scanning. The frames are drawn to the screen in two separate scans. The first scan draws half of the picture and the second scan draws between the lines of the first scan. This scanning method is called interlacing. Each line is divided into pixels that are the smallest possible parts of the picture. The number of pixels that can be displayed determines the resolution (quality) of the video signal. The video signal television picture into three parts: the picture brightness (luminance), the color (chrominance), and the audio.

Digital Video
Digital video is a sequence of picture signals (frames) that are represented by binary data (bits) that describe a finite set of color and luminance levels. Sending a digital video picture involves the conversion of a scanned image to digital information that is transferred to a digital video receiver. The digital information contains characteristics of the video signal and the position of the image (bit location) that will be displayed.