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Analog-to-Digital Conversion Techniques

Chapter 4 discusses analog-to-digital conversion, focusing on pulse code modulation (PCM) as the primary method for digitizing analog signals. It covers the processes of sampling, quantization, and encoding, as well as transmission modes, including parallel and serial transmission. The chapter also highlights the importance of sampling rates, quantization levels, and the implications for signal recovery and data transmission efficiency.
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0% found this document useful (0 votes)
9 views47 pages

Analog-to-Digital Conversion Techniques

Chapter 4 discusses analog-to-digital conversion, focusing on pulse code modulation (PCM) as the primary method for digitizing analog signals. It covers the processes of sampling, quantization, and encoding, as well as transmission modes, including parallel and serial transmission. The chapter also highlights the importance of sampling rates, quantization levels, and the implications for signal recovery and data transmission efficiency.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PPT, PDF, TXT or read online on Scribd

Chapter 4

4.1 ANALOG-TO-DIGITAL CONVERSION

4.2 TRANSMISSION MODES


Objective
 Discussion on analog-to-digital conversion.

 Pulse code modulation is described as the main method


used to sample an analog signal..

 Transmission modes.

 Data can be digitally transmitted


4-2 ANALOG-TO-DIGITAL CONVERSION

•Discussed how to convert digital data to digital signals.

•When we have an analog signal such as one created by a


microphone or camera, the tendency today is to change an
analog signal to digital data.

•Two techniques are

1)Pulse code modulation(PCM)


2)Delta modulation. (DM)
Pulse Code Modulation (PCM)
•The most common technique to change an analog
signal to digital data (digitization) is called pulse code
modulation (PCM).

A PCM encoder has three processes: Sampling,
Quantization and Encoding.

1. The analog signal is sampled.


2. The sampled signal is quantized.
3. The quantized values are encoded as stream of
bits.
Components of PCM encoder
Sampling

First step in PCM is sampling(Process of measuring
instantaneous values of continuous signal in discrete form)
 Analog signal is sampled every TS secs.
T is the sample interval or period.

The inverse of sampling interval is called Sampling rate
or sampling frequency denoted by fs, where
fs=1/Ts.

The three sampling methods are
1. Ideal
2. Natural
3. Flat-top.
Continued...

In Ideal sampling, pulses from the analog signal are
sampled.

Cannot be easily implemented.

In Natural sampling, a high speed switch is turned on
only for the small period of time when sampling occurs.

Result is a sequence of samples that retains the shape of
the analog signal.

In Sample and hold sampling, creates flat-top samples
by using a circuit.
Three different sampling methods for PCM
Continued..

The sampling process is sometimes refereed to as Pulse
amplitude modulation(PAM).

Result is still an analog signal with non integral values.
Sample rate:
What are the restrictions on TS ?????

Answered by Nyquist theorem. “To reproduce the
original analog signal, necessary condition is that
sampling rate be at least twice the highest frequency in
the original signal”
i.e Sample rate= 2 * Fmax
Nyquist sampling rate for low-pass and
band pass signals
Continued..
1) We can sample signal only if signal is band-limited.
i.e signal with an infinite bandwidth cannot be
sampled.
2)Sampling rate must be at least two times the highest
frequency, not bandwidth.
If analog signal is low pass Bandwidth and
highest frequency are same values.
If band pass Bandwidth value is lower than
the value of maximum frequency.
Example
The frequency domain is more compact and useful when we
are dealing with more than one sine wave.
 Figure below shows three sine waves at three sampling
rates
 fs=4f(2 times Nyquist rate),
fs=2f(Nyquist rate) and
fs=f(One half the Nyquist rate)

If the signal is sampled at Nyquist rate, original sine wave


can be created.
Recovery of a sine wave with different sampling
rates.
Example
Telephone companies digitize voice by assuming a
maximum frequency of 4000 Hz. The sampling rate
therefore is 8000 samples per second.
Example
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.
Example
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 = 400,000 samples/ second.
Quantization
 Quantization is an approximation process.
 The result of sampling is
1. A series of pulses with amplitude values between
maximum and minimum of the signal.
2. The set of amplitudes can be infinite with non
integral values between two limits.
3. These values can not be used in the encoding
process.
Steps in quantization
1. Assume that the original analog signal has
instantaneous amplitudes between Vmin and Vmax.
2. Divide the range into L zones, each of height ∆.
∆= (Vmax-Vmin)/L
3. Assign quantized values of 0 to L-1 to the midpoint of
each zone.
4. 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 to +20V.
Let number of levels L= 8, Therefore ∆=5v.
Quantization and encoding of a sampled signal
Quantization levels
• Choice of L(Number of levels) depends on the range
of amplitudes of the analog signal and how accurately
the signal need to be recovered.

• If amplitude of signal fluctuates between two values,


then only two levels are needed.
• If signal is voice signal, which has many amplitude
values, need more quantization levels.

• In audio digitizing, L=256 and in video it is in


thousands.
Quantization Error
• Lower value of L increases quantization error.
• I/P to quantizer are real values. O/P values are approximated
values.
• The O/P values are chosen to be the middle value in the zone.
• If i/p value is also at the middle of the zone, there is no
quantization error; otherwise there is an error.
• Quantization error changes the SNR of the signal.
• It can be proven that contribution of quantization error to SNR
dB depends on number of quantization levels L, or the bits per
sample n b, as shown in the formula below
SNRdB = 6.02 nb +1.76dB
Example
What is the SNRdB in the example above

Solution
Use the formula to find the quantization.
 Eight levels in the example, so bits/ sample.
There fore SNRdB = 6.02(3) + 1.76 = 19.82dB.
Increasing the number of levels increases the SNR.
Example
A telephone subscriber line must have an SNR dB above 40.
What is the minimum number of bits per sample?

Solution
We can calculate the number of bits as

40=6.02nb+1.76dB
6.02nb=40-1.76dB
nb=40-1.76dB/6.02
nb=6.35

Telephone companies usually assign 7 or 8 bits per sample.


Uniform Vs Non-Uniform quantization
 For many applications, the distribution of the
instantaneous amplitudes in the analog signal is not
uniform.
• Change in amplitude occurs frequently in lower
amplitude than in higher one.
• For these type of applications, non uniform zones are
used i.e height of ∆ is not fixed.
• It is greater near lower amplitudes and less near higher
amplitudes.
Uniform Vs Non-Uniform quantization
• Non uniform quantization can be achieved by using a process
called Companding and expanding.
• Companding means reducing the instantaneous voltage
amplitude for large values.
• Expanding is a opposite process.
• The signal is companded at the sender before conversion and
expanded near receiver after conversion.
• Companding gives greater weight to strong signals and less
weight to weak ones.
• It is proved that non uniform quantization effectively reduces
the SNRdB of quantization.
Example
Encoding
 After each sample is quantized and the number of bits per
sample is decided.
 Each sample can be changed to nb-bit code word which is
shown in the last row.
 Number of bits for each sample is determined from the
number of quantization levels.
 If number of quantization levels is L, the number of bits is

 Bit rate can be determined from the formula


Bit rate(N)=sampling rate * number of bits/sample
= fs * nb
Example
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 bit rate are calculated as
follows:
Components of a PCM decoder
Signal recovery
• Recovery of original signal requires PCM decoder.
• The decoder uses a circuitry to convert the code words into a pulse
that holds the amplitude until the next pulse.
• Stair case signal is generated.
• It is passed through a low-pass filter to eliminate high frequency
component signals present in i/p analog signal (i.e To smooth the
staircase signal into an analog signal.)
• Filter has the same cut off frequency as the original signal at the
sender.
• If the signal is sampled at the Nyquist sampling rate and if there
are enough quantization levels, the original signal will be
recreated.
Note: The Maximum and minimum values of the original signal can
be achieved by using amplification.
PCM bandwidth
 Suppose the BW of a low-pass analog signal is given, What is the
[Link] of the channel that can pass this digitized signal.??
 Band width(Range of frequencies) is proportional to signal rate(Baud
Rate)
 Minimum BW of a line encoded signal is
Bmin = C* N*(1/r)
Substitute the value of N in the formula (Bit rate(N)=nb*fs)
Bmin =C*N*(1/r)
=C*nb*fs*1/r
=C*nb*2*Banalog*1/r
 when 1/r=1(NRZ or bipolar) and c=1/2(avg. case)
minimum BW is Bmin = nb*Banalog
 Means minimum band width of the digital signal is nb times greater
Maximum Data Rate of a channel
 Nyquist theorem which gives the data rate of a
channel(low pass channel with band width B) as
Nmax = 2*B*Log2 L bps

 Minimum required band width if the data rate and the


number of signal levels are fixed is
Example
We have a low-pass analog signal of 4 kHz. If we send the
analog signal, need a channel with minimum bandwidth of 4
kHz.

If we digitize the signal and send 8 bits/sample,


Minimum band width of the channel required is
8 × 4 kHz = 32 kHz.
4-3 TRANSMISSION MODES

Primary concern when data is transmitted from one


device to another is the wiring & data stream on the wire.

 Do we send 1 bit at a time; or do we group bits into


larger groups and send , if so, how?

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 one bit is sent with each clock tick.
Data transmission modes
Parallel Transmission
• Binary data 1’s and 0’s may be organized into groups of n bits each.
Computers produce and consume data in groups of bits.
• By grouping n bits of data at a time and sending is Parallel
transmission.
•Use n wires to send n bits at a time.

•Each bit has its own wire, and all n bits of one group can be
transmitted with each clock tick from one device to another.

Advantage: Speed Can increase the transfer speed by a factor


of n over serial transmission.
Disadvantage: Requires n communication lines to transmit the data
stream. Therefore, limited to short distance
It is expensive.
Parallel transmission
Serial Transmission
•In serial transmission one bit follows another.
•Only one communication channel required rather
than n to transmit data between two communicating
devices.

•Since communication within devices is parallel,


conversion devices are required at the interface
between the sender and the line(P-to-S) and between
Line and the receiver(S-to-P).
Serial transmission
Three ways of serial transmission
 Asynchronous transmission:
• Timing of signal is unimportant.
• Information is received and translated by agreed upon
patterns.
• As long as patterns are followed, the receiving devices
can retrieve information.
• Patterns are based on grouping bit streams into bytes.
• A group of 8 bits is sent on the link as a unit.
• The sending device handles each group independently,
relaying it to the link whenever ready, without regard to a
timer.
Continued..
• Without synchronization, the receiver cannot use timing
to predict when the next group will arrive.
• To alert the receiver, each new group is added with extra
bits in the beginning which is 0, called start bit and 1 bit
at the end to inform receiver the end of byte, called stop
bits, increasing to size of information to 10 bits.
• Transmission of each byte may then be followed by a gap
of varying duration, represented by an idle channel or by a
stream of additional stop bits.
• This mechanism is called asynchronous because, at byte
level, the sender and receiver do not have to be
synchronized.
Continued..
• Within each byte, the receiver must still be synchronized with the
incoming bit stream.
• When the receiver detects a new bit, it sets timer and starts counting
bits as they come in.
• After n bits, the receiver looks for stop bit. As soon it is detected, it
waits until it detects next start bit.
Advantages:
1. Cheap and effective

2. Suitable for low speed communication(Eg: Connection of


keyboard to a computer)
Disadvantage:
1. Addition of start, stop and gaps in to bit streams make this
transmission mode slower.
Asynchronous transmission
Synchronous transmission
• Bit streams are combined to form frames which contains multiple
bytes.
• Each byte is introduced onto the transmission medium without a
gap between the one and next one.
• The receiver separates the strings into the bytes or characters, to
reconstruct the information.
• The sender puts its data onto the line as one long string.
Continued..
• If sender wishes to send data in separate bursts, the gap between
bursts must be filled with a special sequences of 0’s and 1’s, which
means idle.
• The receiver counts the bits as they arrive and group them into 8-
bit units.
• Without gaps, start and stop bits, there is no built in mechanism to
help the receiving device to adjust its bit synchronization
midstream.
• Timing is very important because the accuracy of the received
information is completely dependent on the ability of the receiving
device to count the bits as they come in.
 Advantages
• Faster transmission(no gaps, no extra bits)
• Useful for high speed applications
Isochronous
• Synchronous transmission fails in real time A/V, because
uneven delay between frames are not acceptable.
Eg:- TV images are broadcast at the rate of 30 images/sec.
They must be viewed at the same rate.
--If each image is sent by using one or more frames, there
should not be any delay between the frames.
--For these type of applications, entire stream of bits must
be synchronized.
--Isochronous, guarantees that data arrives at fixed rate.
Difference between serial & parallel
transmission

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