CHAPTER 4

DIGITAL TRANSMISSION
 ANALOG-TO-DIGITAL CONVERSION
 Pulse Code Modulation (PCM)
 Technique to change an analog signal to digital data (digitization) is called pulse code modulation
(PCM).
 A PCM encoder has three processes

 The analog signal is sampled.

 The sampled signal is quantized.
 The quantized values are encoded as streams of bits.
 Sampling
 The first step in PCM is sampling.
 The analog signal is sampled every Ts s, where Ts is the sample interval or period.
 The inverse of the sampling interval is called the sampling rate or sampling frequency and
denoted by fs, where fs = 1/Ts.
 There are three sampling methods—ideal, natural, and flat-top

 In ideal sampling, pulses from the analog signal are sampled.

 In natural sampling, a high-speed switch is turned on for only the small period of time when the
sampling occurs. The result is a sequence of samples that retains the shape of the analog signal.
 The most common sampling method, called sample and hold
 The sampling process is sometimes referred to as pulse amplitude modulation (PAM).
 Sampling Rate
 According to the Nyquist theorem, the sampling rate must be at least 2 times the highest
frequency contained in the signal.
 First, we can sample a signal only if the signal is band-limited. In other words, a signal with an
infinite bandwidth cannot be sampled.
 Second, the sampling rate must be at least 2 times the highest frequency, not the bandwidth.
 If the analog signal is low-pass, the bandwidth and the highest frequency are the same value.
 If the analog signal is bandpass, the bandwidth value is lower than the value of the maximum
frequency.
 A complex low-pass signal has a bandwidth of 200 kHz. What is the minimum sampling rate for
this signal?
 Solution
 The bandwidth of a low-pass signal is between 0 and f, where f is the maximum frequency in the
 signal. Therefore, we can sample this signal at 2 times the highest frequency (200 kHz). The
sampling rate is therefore 400,000 samples per second.

 A complex bandpass signal has a bandwidth of 200 kHz. What is the minimum sampling rate for
this signal?
 Solution
 We cannot find the minimum sampling rate in this case because we do not know where the
bandwidth starts or ends. We do not know the maximum frequency in the signal.
 Quantization
 The result of sampling is a series of pulses with amplitude values between the maximum and
minimum amplitudes of the signal.
 The set of amplitudes can be infinite with nonintegral values between the two limits. These
values cannot be used in the encoding process.
 The following are the steps in quantization:
 1. We assume that the original analog signal has instantaneous amplitudes between Vmin and
Vmax.
 2. We divide the range into L zones, each of height Δ (delta).

 3. We assign quantized values of 0 to L − 1 to the midpoint of each zone.

 4. We approximate the value of the sample amplitude to the quantized values.
 assume that we have a sampled signal and the sample amplitudes are between −20 and +20 V.
 We decide to have eight levels (L = 8). This means that Δ = 5 V.
 The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to 0, 0 to +5, +5 to +10, +10 to +15, +15 to
+20
 The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5, 17.5
 The value at the top of each sample in
the graph shows the actual amplitude.
 In the chart, the first row is the
normalized value for each sample
(actual amplitude/Δ).
 The quantization process selects the
quantization value from the middle of
each zone. This means that the
normalized quantized values (second
row) are different from the normalized
amplitudes.
 The difference is called the normalized
error (third row).
 The fourth row is the quantization code
for each sample based on the
quantization levels at the left of the
graph.
 The encoded words (fifth row) are the
final products of the conversion.
 Quantization Error
 One important issue is the error created in the quantization process.
 Quantization is an approximation process.
 The input values to the quantizer are the real values; the output values are the approximated
values. The output values are chosen to be the middle value in the zone.
 If the input value is also at the middle of the zone, there is no quantization error; otherwise, there
is an error.
 The value of the error for any sample is less than Δ/2.
 In other words, we have −Δ/2 ≤ error ≤ Δ/2.
 The quantization error changes the signal-to-noise ratio of the signal, which in turn reduces the
upper limit capacity according to Shannon.
 It can be proven that the contribution of the quantization error to the SNRdB of the signal
depends on the number of quantization levels L, or the bits per sample nb, as shown in the
following formula:
 What is the SNRdB in the example of Figure 4.26?
 Solution
 We can use the formula to find the quantization. We have eight levels and 3 bits per sample, so
SNRdB = 6.02(3) + 1.76 = 19.82 dB. Increasing the number of levels increases the SNR.

 A telephone subscriber line must have an SNRdB above 40. What is the minimum number of bits
per sample?
 Solution
 We can calculate the number of bits as
 Telephone companies usually assign 7 or 8 bits per sample.
 Encoding
 The last step in PCM is encoding.
 After each sample is quantized and the number of bits per sample is decided, each sample can be
changed to an nb-bit code word.
 A quantization code of 2 is encoded as 010; 5 is encoded as 101; and so on.
 Note that the number of bits for each sample is determined from the number of quantization
levels. If the number of quantization levels is L, the number of bits is nb = log2 L.
 In our example L is 8 and nb is therefore 3. The bit rate can be found from the formula

 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. So the sampling rate and
bitrate are calculated as follows:
 Original Signal Recovery
 The recovery of the original signal requires the PCM decoder.
 The decoder first uses circuitry to convert the code words into a pulse that holds the amplitude
until the next pulse.
 After the staircase signal is completed, it is passed through a low-pass filter to smooth the
staircase signal into an analog signal.
 The filter has the same cutoff frequency as the original signal at the sender.
 If the signal has been sampled at (or greater than) the Nyquist sampling rate and if there are
enough quantization levels, the original signal will be recreated.
 Note that the maximum and minimum values of the original signal can be achieved by using
amplification.
  PCM Bandwidth
 Suppose we are given the bandwidth of a low-pass analog signal. If we then digitize the signal,
what is the new minimum bandwidth of the channel that can pass this digitized signal?
 the minimum bandwidth of a line-encoded signal is Bmin = c ×N × (1/r). We substitute the value
of N in this formula:

 When 1/r = 1 (for a NRZ or bipolar signal) and c = (1/2) (the average situation), the minimum
bandwidth is

 This means the minimum bandwidth of the digital signal is nb times greater than the bandwidth
of the analog signal.
 Delta Modulation (DM)
 PCM is a very complex technique. The simplest is delta modulation.
 PCM finds the value of the signal amplitude for each sample; DM finds the change from the
previous sample.

 Note that there are no code words here; bits are sent one after another.
 scheme sends only the difference between pulses, if the pulse at time tn+1 is higher in amplitude
value than the pulse at time tn, then a single bit, say a “1”, is used to indicate the positive value.
 If the pulse is lower in value, resulting in a negative value, a “0” is used.
 This scheme works well for small changes in signal values between samples.
 If changes in amplitude are large, this will result in large errors.
 Modulator
 The modulator is used at the sender site to create a stream of bits from an analog signal.
 The process records the small positive or negative changes, called delta δ.
 If the delta is positive, the process records a 1; if it is negative, the process records a 0.
 The modulator builds a second signal that resembles a staircase. Finding the change is then
reduced to comparing the input signal with the gradually made staircase signal.

 The modulator, at each sampling interval, compares the value of the analog signal with the last
value of the staircase signal.
 If the amplitude of the analog signal is larger, the next bit in the digital data is 1; otherwise, it is
0.
 If the next bit is 1, the staircase maker moves the last point of the staircase signal δ up; if the next
bit is 0, it moves it δ down.
 Demodulator
 The demodulator takes the digital data and, using the staircase maker and the delay unit, creates
the analog signal. The created analog signal, however, needs to pass through a low-pass filter for
smoothing.

 Quantization Error
 It is obvious that DM is not perfect. Quantization error is always introduced in the process.
 The quantization error of DM, however, is much less than that for PCM.
 TRANSMISSION MODES
 The transmission of binary data across a link can be accomplished in either parallel or serial
mode.
 In parallel mode, multiple bits are sent with each clock tick.
 In serial mode, 1 bit is sent with each clock tick.

 Parallel Transmission
 Binary data, consisting of 1s and 0s, may be organized into groups of n bits each.
 By grouping, we can send data n bits at a time instead of 1. This is called parallel transmission.
 The mechanism for parallel transmission is a conceptually simple one: Use n wires to send n bits
at one time.
 The advantage of parallel transmission is speed. All else being equal, parallel transmission can
increase the transfer speed by a factor of n over serial transmission.
 But there is a significant disadvantage: cost.
 Parallel transmission requires n communication lines (wires in the example) just to transmit the
data stream.
 Serial Transmission
 In serial transmission one bit follows another, so we need only one communication channel
rather than n to transmit data between two communicating devices.

 The advantage of serial over parallel transmission is that with only one communication channel,
serial transmission reduces the cost of transmission over parallel by roughly a factor of n.
 Since communication within devices is parallel, conversion devices are required at the interface
between the sender and the line (parallel-to-serial) and between the line and the receiver (serial-
to-parallel).
 Serial transmission occurs in one of three ways: asynchronous, synchronous, and isochronous
  Asynchronous Transmission
 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.
 Without synchronization, the receiver cannot use timing to predict when the next group will
arrive.
 To alert the receiver to the arrival of a new group, therefore, an extra bit is added to the beginning
of each byte. This bit, usually a 0, is called the start bit.
 To let the receiver know that the byte is finished, 1 or more additional bits are appended to the
end of the byte. These bits, usually 1s, are called stop bits.
 By this method, each byte is increased in size to at least 10 bits, of which 8 bits is information
and 2 bits or more are signals to the receiver.
 In synchronous transmission,
 the bit stream is combined into longer “frames,” which may contain multiple bytes.
 Each byte, however, is introduced onto the transmission link without a gap between it and the
next one.
 data are transmitted as an unbroken string of 1s and 0s, and the receiver separates that string
into the bytes, or characters, it needs to reconstruct the information.

 The advantage of synchronous transmission is speed. With no extra bits or gaps to introduce at
the sending end and remove at the receiving end,
 synchronous transmission is faster than asynchronous transmission.
 For this reason, it is more useful for high-speed applications such as the transmission of data
from one computer to another.
  Isochronous
 In real-time audio and video, in which uneven delays between frames are not acceptable,
synchronous transmission fails.
 For example, TV images are broadcast at the rate of 30 images per second; they must be
viewed at the same rate.
 If each image is sent by using one or more frames, there should be no delays between
frames.
 For this type of application, synchronization between characters is not enough; the entire
stream of bits must be synchronized.
 The isochronous transmission guarantees that the data arrive at a fixed rate.