Professional Documents
Culture Documents
Analog signal
1
Advantages of Digital Over Analog For Communications
2
Analog to Digital Conversion Process (ADC)
Three Step Process
•
amplitude
amplitude
•
amplitude
•• • •
0100101101011001
time • time 1110101010101000
time
• 0100011000100011
• •
• • 0101001111010101
• •
•• • 1110110111010001
•
❖
Sample Quantize Encode
Analog Digital
Signal Captured Quantized Signal
Sampled Data Sampled
Sampling Values Quantizing Data Encoding
selects the chooses the assigns binary
data points amplitude numbers to
we use to Discrete values used Now have those Now have the
Analog signal create the time values: to encode discrete amplitude digital
is continuous digital data few amplitudes Values in values data which
in time & from analog both time & is the final
amplitude signal amplitude result
3
Next Topic – Pulse Code Modulation
Pulse-code modulation (PCM) is used to digitally represent
sampled analog signals. It is the standard form of digital audio
in computers, CDs, digital telephony and other digital audio
applications. The amplitude of the analog signal is sampled at
uniform intervals and each sample is quantized to its nearest
value within a predetermined range of digital levels.
Four-bit coding
(16 discrete levels)
4
Second Step – Quantization I
Quantization is the process of changing a continuous-amplitude
signal into on with discrete amplitudes.
mp
m(t) Quantized samples of mq(t)
Allowed quantization levels
2m p
==
L
m p L = 16 levels 4 bits
To communicate
sampled values, we
send a sequence of bits
that represents the
quantized value.
For 16 quantization levels,
4 bits are required.
PCM can use a
binary
representation of
value.
The PSTN uses PCM
6
Quantization II
We start with a sampled signal {call it m(t)} and now we want to quantize it.
Divide the range (-mp, mp) into L uniformly spaced intervals. The number
intervals is L and the separation between quantized levels is
2 mp
L
The kth sample point of m(t) is designated as m(kTS) and is assigned a value
equal to the midpoint between two adjacent levels. Define:
7
Error Generated by Quantization (Quantization Noise)
Quantization noise q(t ) m(t ) mq (t )
The signal (or message) power S0 is proportional to the square of m(t), thus
mp2
S0 2 m2 (t ) , but if m(t ) is sinusoidal, S0
2
Note: denotes time average
9
Quantization IV
We want a measure of the quality of received signal (that is, the ratio of
the strength of the received signal S0 relative to the strength of the error
Nq due to quantization).
3L2
2
m (t ) 1
It is usually expressed in decibels, Note:
mp2 2
S0 3L2
SQNRdB 10 log 10 10 log 10
Nq 2
Conclusion:
To reduce the quantization error relative to the message signal level, use
smaller quantization steps .
10
The Dilemma of Strong Signals versus Weak Signals
11
Use Compression and Expansion → Companding
Compression Restoration
(m) m(t)
m(t) m(t)
http://www.slideshare.net/91pratham/unit-ipcmvsh 12
Companding Laws
Output (y/ymax)
Input (m/mp) Input (m/mp)
A m m 1
y for 0 1 m m
1 log e A mp mp A y log e 1 for 0 1
log e (1 ) m mp
A Am 1 m p
y 1 log e for 1
1 log e A mp A mp
13
Flattening of the S/N Ratio Using the -Law
S0
Nq
(8 bits)
14
Transmission Bandwidth
BT = nB Hz
15
Example
Solution:
The Nyquist rate is RN = 2 x 3000 Hz = 6000 Hz (samples/second), but the actual
rate is 33⅓ % higher, so that is 6000 Hz + (⅓ x 6000) = 8000 Hz.
16
Example Continued
Solution (continued):
Having chosen n = 8 to guarantee < 0.5% error, to find the bandwidth required
we start with
If 24 such signals are multiplexed onto a single line (known as a T1 Line in the
Bell telephone system), then
¶
A maximum of 2B independent elements of information per second can be
transmitted, error-free, over a noiseless channel of bandwidth B Hz.
17
Exponential Increase of the Output SNR (S/N Ratio)
We start with the SNR (signal-to-noise ratio) equation from slide 10 above:
S0 m 2 (t )
3 L2 denotes time average
Nq mp2
The number of levels L can be expressed as L2 = 22n where n = log2(L) and is
the number of bits to generate L levels. The SNR can now be expressed as
S0 m 2 (t ) 2 n
mp2
3 2
Nq
Using the expression for bandwidth, BT = nB, then we arrive at
S0 m 2 (t ) 2 B /B
3 2 T
Nq mp2
Taking the logarithm gives
S0 S0 m 2 (t )
10 log 10 10 log 10 3 10 2 n log 10 2 6n dB
Nq Nq mp2
dB
18
SNR Example
S0 m 2 (t ) 3P 3
3 (2)2 n 2ave (2)2 n (2)2 n
Nq mp2 mmax 2
L n SNR
S0 32 5 31.8 dB
10 log 10 1.76 6n dB
Nq 64 6 37.8 dB
128 7 43.8 dB
256 8 49.8 dB
19
Bell System’s T1 Carrier System (1962)
The first version, the Transmission System 1 (T1), was introduced in 1962 in
the Bell System, and could transmit up to 24 telephone calls simultaneously
over a single transmission line consisting of copper wire.
20
T1 Carrier – Time Division Multiplexing
21
Comparison of T-Carrier (North America) and E-Carrier (Europe)
22
Worked PCM Example
Using for the quantization noise Nq = [2/12], and taking Pave = 2 W, the
SNR is given by
S Pave 2 24
12 98, 304
N q N q 2 (1.5625 10 2 )2
SNRdB 10 log 10 98, 304 49.93 dB
23
Worked Example for PCM (continued)
The lowest integer number of bits n that will give at least 31.5 levels is n = 5
because 25 = 32 levels. So the answer is 5 bits.
24
Differential Pulse Code Modulation (DPCM)
PCM is not really efficient because it generates so many bits taking up a lot
of bandwidth. Can we improve on this? YES.
25
Differential Pulse Code Modulation (continued)
At the receiver knowing d[k] and the previous value of m[k-1] allows us to
construct the value of m[k].
How do we benefit from doing this?
Suppose mest[k] is the estimate of the kth sample, then the difference d[k]
is defined by
d[k] = m[k] – mest[k]
26
Differential Pulse Code Modulation (continued)
Receiver Concept:
At the receiver we determine the estimate mest[k] from previous sample
values, and then generate m[k] by adding the received d[k] values to the
estimate mest[k]. Thus the reconstruction of the samples is done
iteratively.
27
Digression on Signal Prediction
28
Signal Prediction (continued)
But we can do even better than this. In general,
m[ k ] a1m[ k 1] a2 m[ k 2] . . . aN m[ k N ] mq [ k ]
Note that the input consists of the weighted previous samples m[k-1],
m[k-2], etc. We say that input m[k] gives output mest[k].
29
Linear Predictor Implemented With Transversal Filter
mest [ k ] a1m[ k 1] a2 m[ k 2] . . . aN m[ k N ]
Input
m[k] Delay Delay Delay Delay ... Delay
TS TS TS TS TS
a1 a2 a3 aN
Output mest[k]
30
DPCM Transmitter
Input Output
m[k] d[k] dq[k]
+
Quantizer
+
mest[k]
+
mq[k]
Predictor
The predictor output mest[k] is fed back to the input so the predictor input
mq[k] is given by
mq [ k ] mq [ k ] dq [ k ] m[ k ] d[ k ] dq [ k ] m[ k ] q[ k ]
31
DPCM Receiver
Input Output
dq[k] + mq[k]
+
mest[k]
Predictor
The receiver’s output (which is the predictor’s input) is also the same,
mq[k] = m[k] + q[k].
Hence, we are able to receive the desired signal m[k] plus the quantization
noise, q[k]. It is important to note that from the difference signal d[k] is
much smaller that the noise associated with m[k].
32
DPCM SNR Improvement
33
Adaptive Differential PCM
34
Adaptive Differential PCM Output Example
time
35
Next Topic is Delta Modulation
36