It is also the standard form for digital audio in computers and the compact disc "red book" format. Why PCM? Even in the presence of noise, the presence or absence of a pulse can be easily determined. Since PCM is digital, a more general reason would be that digital signals are easy to process by cheap standard techniques. This makes it easier to implement complicated communication systems such as telephone networks.
In the following sections we will discuss the stages of PCM in an extended way with full details. Simply we can say that filtering stage removes frequencies above the highest signal frequency.
These frequencies if not removed, may cause problems when the signal is going through the stage of sampling. Next we will discuss extensively the other important three processes which together called the modulation process.
PCM mostly based on the sampling theorem which states that If a signal f t is sampled at regular intervals of time and at a rate higher than twice the highest significant signal frequency, then the samples contain all the information of the original signal. The function f t may be reconstructed from these samples by the use of a low-pass filter. Figure 2 briefly describe the functionality of sampler process.
If voice data are limited to frequencies below Hz, a conservative procedure for intelligibility, samples per second would be sufficient to completely characterize the voice signal. Note, however, that these are analog samples. When the voice waveform is sampled, a train of short pulses is produced, each representing the amplitude of the waveform at the specific instant of sampling. The envelope of the PAM samples replicates the original waveform. In PAM the successive sample values of the analog signal s t are used to effect the amplitudes of a corresponding sequence of pulses of constant duration occurring at the sampling rate.
No quantization of the samples normally occurs Figure 3a, b. In principle the pulses may occupy the entire time between samples, but in most practical systems the pulse duration, known as the duty cycle, is limited to a fraction of the sampling interval.
Such a restriction creates the possibility of interleaving during one sample interval one or more pulses derived from other PAM systems in a process known as time-division multiplexing TDM. Figure 3 - a Analog signal, s t. For digital transmission, further processing is required. Pulse Code Modulation is a technique used to convert the PAM samples to a binary weighted code for digital transmission.
The first step is quantization, where each sample is assigned a specific quantizing interval. Each is discussed in the text that follows. Converting PAM samples to a digital signal involves assigning the amplitude of a PAM sample one of a whole range of possible amplitude values, which are divided into quantizing intervals.
There are possible quantizing intervals, positive and negative. The boundaries between adjacent quantizing intervals are called decision values. Below Figure 4 show the simple representation of quantization process. The sample value is rounded to the closest quanta level. A guide gap tg is kept between two pulses. An example to the PCM steps explained up to here is given in Figures 5 and 6 respectively.
For example a quantization code of 2 is encoded as ; 5 is encoded as ; and so on. Note that the number of bits for each sample is determined from the number of quantization levels. For a NRZ system to be synchronized using in-band information there must not be long sequences of identical symbols, such as ones or zeroes.
For binary PCM systems, the density of 1-symbols is called ones- density. Ones-density is often controlled using pre-coding techniques such as Run Length Limited encoding, where the PCM code is expanded into a slightly longer code with a guaranteed bound on ones-density before modulation into the channel. In other cases, extra framing bits are added into the stream which guarantees at least occasional symbol transitions. Another technique used to control ones-density is the use of a scrambler polynomial on the raw data which will tend to turn the raw data stream into a stream that looks pseudo-random, but where the raw stream can be recovered exactly by reversing the effect of the polynomial.
In this case, long runs of zeroes or ones are still possible on the output, but are considered unlikely enough to be within normal engineering tolerance. In other cases, the long term DC value of the modulated signal is important, as building up a DC offset will tend to bias detector circuits out of their operating range. In this case special measures are taken to keep a count of the cumulative DC offset, and to modify the codes if necessary to make the DC offset always tend back to zero.
Many of these codes are bipolar codes, where the pulses can be positive, negative or absent. In the typical alternate mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to generate special symbols used for framing or other special purposes.
After each sampling period has passed, the next value is read and a signal is shifted to the new value. As a result of these transitions, the signal will have a significant amount of high-frequency energy. Some systems use digital filtering to remove some of the aliasing, converting the signal from digital to analog at a higher sample rate such that the analog filter required for anti-aliasing is much simpler.
The sampling theorem suggests that practical PCM devices, provided a sampling frequency that is sufficiently greater than that of the input signal, can operate without introducing significant distortions within their designed frequency bands.
The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signal. They produce on their output a voltage or current depending on type that represents the value presented on their inputs. This output would then generally be filtered and amplified for use.
The sampling clock needs to be either sent to, or regenerated at, the receiving side to determine when each full sequence of sampling begins and ends. The data clock is also needed to determine exactly when to read each bit of information. If there is one channel or 30 TDM channels the sampling period is fixed at Ms and this period is known as a frame. Therefore the frame clock must have a period of Ms.
The falling edge of the data clock informs the receiver that it must read the data bit. When the bit stream is transmitted along a line the pulses become distorted and the rise and fall times become significant. Telephony is full duplex so that there is a coder and a decoder at each port, but each would use the same clock.
To minimize the number of circuits it is possible to use a line-coding scheme which allows the receiver to extract the clocks from the PCM signal. Marian Marciniak. A short summary of this paper. Jaworski itl. Marciniak itl. Two methods of increasing the SBS threshold are compared: well known dithering method and new one called Initial Pulse Modulation Method, in which optical signal spectrum is widening and suppressed due to FWM in initial fiber with high normal dispersion and Kerr nonlinearity, e.
DCF fiber. Requirement of high transmissions quality causes that phenomena normally omitted in conventional telecommunication systems must be taken into account.
Such undesirable phenomena are distortion — caused by nonlinearity of fiber appeared for high transmission power level, and multi-path interference MPI of signals directly coming to optical receiver and after double reflection.
Module for increasing SBS threshold power Fig. The acoustic wave creates patterns of periodically changing refractive index fiber Bragg gratings. These gratings reflect the optical wave. For transmitted power higher than the SBS threshold power approximately 7 dBm the reflected power rapidly increases and consequently interferometric noise MPI — multi path interference increases.
Scattering the optical spectrum can decrease spectral density. Unfortunately, when the transmitted spectrum is in order of tens of GHz modulation instability MI occurs in the case of fiber with anomalous positive dispersion.
Two methods of increasing the SBS threshold are presented below: well known dithering method and new one called Initial Pulse Modulation Method. Two methods of increasing SBS threshold power: a optical phase modulation dithering , b initial pulse modulation and spectrum scattering by propagation in DCF fiber A. Dithering Method Wideband optical phase modulation dithering is well known method of increasing the SBS threshold power.
External phase modulator or direct laser current modulation produces the dithering. Making choice of the modulation frequency is very important. Additional increasing of the SBS threshold can be achieved by multi-frequency modulation [2]. The main drawback of the frequency modulation either single-tone or multi-tone in long CATV link of total span higher then km is spectrum fluctuations of the propagating power Fig.
Proper shape of optical spectrum can be realised by transmitting short periodical pulses through highly nonlinear fiber with high normal negative dispersion Fig. Pulses are quickly widening during propagation due to fiber dispersion, which means that time dependence of optical power diminishes. As a result, optical power at the fiber output is near constant, with only small component of pulse repetition frequency. The components interfering each other due to Kerr nonlinearity it is well known Four Photons Mixing process and dispersion, Fig.
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