Telecommunication in the 21st Century

telecommunication in the 21st CenturyTelecommunication in the 21st one C defy improved over the decade by the introduction of part techniques finished which manoeuvers butt be transmissible from a transmitter through with(predicate) a medium to a receiver. These techniques have improved mobile communications, beam transmissions and helped to improve data security. Some of these techniques are amplitude modulation (AM), oftenness modulation (FM), sampling and link analysis (SLA) and PCM.The acronym PCM represents Pulse-code modulation, which is utilize for digitizing latitude data, for instance, sound recording channelizes. This is carried out by sampling line of latitude signals at unvarying interval and then quantized to a series of symbols in a digital code (e.g. 10001).Its technically a way in which line of latitude signals are converted to digital form. PCM technique has its advantagesIt makes processing of signals cheap since PCM is digital.It helps to puree off frequencies above the highest signal frequence.Pulse-code modulation has been a form apply for some compact disc formats, digital video and for digital audio in computers.In PCM, there are series of processed to be followedFilteringSamplingQuantizingBinary codingCompandingFilteringThis is a process where frequencies above the highest signal frequency are removed. The reason for this is that if this frequency is not removed, problems would occur when spillage to the next stage of sampling.SamplingThis stage of the PCM is performed through PAM (pulse amplitude modulation).It say the question of how signals change from one form to an opposite(prenominal) ( running(a) to digital).It makes use of the headmaster analog signal and uses it for the amplitude modulation of a pulse which has invariant amplitude and frequency, this constant frequency is known as the sampling frequency (i.e. the number of experiments per second ).The sampling frequency have to be more(prenominal) tha n the maximum frequency of the parallel signal. To work out the sampling rate, Nyquist theorem is employThat in nightclub to be able to reconstruct the master analogue signal, a minimum number of samples had to be taken.It could be verbalize asFs 2(BW)Fs = Sampling frequencyBW = Bandwidth of pilot analog instance signalQuantizing and CodingThis basically means the converting of separately of the analogue sample into a separate value (in the form of a binary program code) that can be attached a digital code word. It is do by assigning each sample a certain quantization interval. The instant(prenominal) amplitude is been rounded off to certain levels, this thereby introduces some uncertainties (quantization fray).This is given by this expressionNumber of levels = 2 Bn (Bn is the number of good turns used in the encoding)It was proven from the experiment that the higher the number of quantization levels the lesser the total of money of quantizing noise. However this process of change magnitude the quantizing level to lower the quantizing noise introduces complexness into the system as the PCM system would need to be able to conduct more code word.CompandingIt is a word derived from the combination of concretion and expanding. This is another stage in pulsecode modulation. It is a process of compressing a given analogue signal and this signal is expanded to its original size on getting to destination. In this process, the foreplay signal is mingy into logarithmic segments and then quantized and coded. The more the signals increase the more the coalescence increases.Since the larger signals are bland more than the smaller signals, the quantization noise increases. This indirectly keeps the SNR (signal to noise ratio) constant.EXPRERIMENTATION AND OBSERVATIONApparatusOscilloscopePCM ENCODER module fraternity cableThe experiment was carried out by sending an input (analogue message) into the PCM ENCODER module. This input is constrained to a defined bandwidth and amplitude orbit in order to make sure the Nyquist criterion is observed. The PCM ENCODER module looks like the diagram downstairsA suitable encoding scheme for the analogue sample is selected. For recitation a 4-bit or 7-bit encoding scheme. The analogue signal is fed through the Vin. For this experiment, the clock rate us 8.33 kilohertz TTL signal from scale polarity module.Time number is also very essential as each binary word is located in a time frame. Its 8 clock periods long and has 8 slots of equal length (i.e. 0 7). The LSB (consisting of 1s and 0s) are embedded in the encoder itself. This is useful in determining the location of each frame in the data stream. Initially the 4-bit elongate coding scheme is selected and patched up with the 8.33 kc TTL sample clock.CH-2A displays the clock signal on the oscilloscope. The display at a lower place shows a 4-bit PCM output for zero amplitude inputQuantization in PCM encoding is the next stage aft er sampling. The quantization level is rather transmitted instead of the sample value. The quantization levels are binary coded (i.e. binary 1 in the presence of a pulse and binary 0 in the absence of a pulse)RESULTS AND OBSERVATIONThe output of the variable DC is machine-accessible to Vin and sweeping the DC voltage slowly forward and backward shows discrete jumps in the data pattern, e.g.The maximum voltage is recorded as -2.51V.Also increasing the amplitude of the DC input signal looks like the diagram underChanging the DC voltage from the maximum to minimum gave a range of binary code variations as listed belowThe following measurements were later make after recording the quantizing levels and associated binary numbersSampling rate 16.6 kcFrame width 950sWidth of a data bit 120sWidth of a data word 480sNumber of quantizing level 16From the measurement above it could be reason out that the quantizing levels are linearly spaced .The same process would be relevant to a 7- bit linear encoding using the furnish switch on the front panel, though it would take longer than the 4-bit linear encoding done earlier.The Companding stage in a PCM is the process by which an analogue signal is been compressed at the source and then expanded back to its original size when it gets to its destination. During this process, the signal is compressed into segments which are quantized using consistent quantization. As the sample signal increases, the compression increases (i.e. the larger samples gets more compressed than the smaller samples). The standard of companding used in this experiment is the A-law .The equation isWhere A = 87.7 in Europe and X is the normalized integer to be compressed.RESULTS AND OBSERVATIONThe toggle switch is changed back to a 4-bit companding and the TIMs A4 companding law pre-selected is selected from the switch board. This gave the measurement belowIn PCM decoding, the TIMs PCM DECODER module is used for decoding. This is the first operat ion in the receiver towards regenerating the received pulses. Amplitude of the pulse generated is the linear sum of all pulses in the coded word. In other to be able to detect the information on the PCM decoder, the knowledge of the sampling rate used to encode the signal is essential.RESULTS AND OBSERVATIONThe setup is similar to the earlier setup with CH-1A connected to the scope selector to the PCM output of the PCM ENCODER.A large negative DC is used for the message, the alternating 0 and 1 bits produced are measured to be 1920ms apart. The 4-bit linear decoding scheme is now selected to carry out the decoding process. The 8.33 kHz TTL signal is stolen from the transmitter and connected to the clock input.Time division multiplexing (TDM) is an ersatz to the method of multiplexing using frequency sharing. Each channel is allocated a particular proposition time slots, and each slots contain frames which must be repeated at the sampling rate. It can only be used for pulsed signal s and not for analogue signals because they are continuous in time. The importance of TDM is that it enables many independent signals to be transmitted.RESULTS AND OBSERVATIONA PCM TDM signal could be generated using PCM ENCODER each driven by the same clock ( one the MASTER and the other slave).Interconnecting in this way eliminates other frames and gives room for the two output to be added unitedly to form the TDM signal. The display on the oscilloscope is shown belowThe connection of the MASTER and the SLAVE generates the diagram belowPatching up the two PCM data outputs generates the display belowThe next step which is shown below is to confirm that the frame synchronization bit is a 1 for the MASTER and 0 for the SLAVEThe last stage of this experiment is to separate the two messages that have been multiplexed earlier. The PCM sensing element is patched up, with each module receiving the same clock stolen from the transmitter and each module also receives an external FS sign al. The diagram below confirms the two messages have been recovered and appear at the correct outputsCONCLUSIONPulse enter Modulation is however a very effective way of transfer audio signal by sampling the signal and transmitting binary coded pulse representing the sample values. It has emerged the most favored modulating scheme for transmitting analogue information such as voice and video signals. The advantages of PCM over the other forms of modulation (e.g. analogue modulation) arePCM suppresses wideband noise.It is effective in the vicissitude of the coded signal along the transmission path.It enables digital multiplexing.It enables the efficient exchange of change magnitude channel bandwidth for improved signal-to-noise ratio.All these advantages however come at the expense of increased system complexity and increases channel bandwidth.REFERENCEShttp//www.webopedia.com/TERM/P/PCM.html last accessed 25/03/08http//cbdd.wsu.edu/kewlcontent/cdoutput/TR502/page13.htmlast access ed 25/03/08http//www.cisco.com/warp/public/788/signalling/waveform_coding.pdflast accessed 25/03/08http//www.comlab.hut.fi/opetus/245/2004/09_PCM.ppt20 last accessed 25/03/08Rodger E.Ziener and William H.Tranter, Principles of Communication, Chapter 3, John Wiley and sons, NY, 2002.Simon Haykin, Communication Systems, Chapter 3, John Wiley and sons, NY, 2001.David Petersen, Audio, Video and Data Telecommunications, Chapter 2, McGraw-Hill, Cambridge, 1992.