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Keying (FSK)
Steven A. Tretter
May 22, 2004
The process of converting data blocks to sinusoids with the desired frequencies can be
modeled as shown in the figure. Assuming transmission starts at t = 0, the “D/A” converter
generates the staircase signal
∞
∆f X
m(t) = I(n)p(t − nTs ) (2)
2 n=0
1
which has the pre-envelope
Rt
jωc t j2π m(τ ) dτ jφ0
s+ (t) = Ac e e 0 e (5)
∆f k−1
X k−1
X
φm (kTs ) = π I(n) = πh I(n) (9)
fs n=0 n=0
Therefore,
(t − kTs )
φm (t) = φm (kTs ) + πh I(k) for kTs ≤ t < (k + 1)Ts (10)
Ts
EXAMPLE: AMPS Wideband Analog Data Signals
The IS-54-B standard in Section 2.1.3.2 Wideband Analog Data Signals specifies that
data bits should be sent using continuous-phase binary (M = 2) FSK with a frequency
deviation of ±8 kHz. The bit rate is specified to be 10 kHz and the bits are Manchester or
split-phase encoded. The exact words in the standard are:
2.1.3.2.1 Encoding
The reverse control channel (RECC) and reverse voice channel (RVC) wideband data
streams (see 2.7) must be further encoded such that each nonreturn-to-zero binary
2
one is transformed to a zero-to-one transition, and each nonreturn-to-zero binary zero
is transformed to a one-to-zero transition.
The filtered wideband data stream must then be used to modulate the transmitter
carrier using direct binary frequency shift keying. A one (i.e., high state) into the
modulator must correspond to a nominal peak frequency deviation 8 kHz above the
carrier frequency, and a zero into the modulator must correspond to a nominal peak
frequency deviation 8 kHz below the carrier frequency
When rectangular pulses of width Ts /2 are used for the levels of the Manchester encoded
bit stream, the spectrum of the FSK signal is wider than the allowed 30 kHz channel band-
width. The IS-54 standard does not say how to solve this problem, but manufacturers have
solved it by lowpass filtering the Manchester encoded bit stream before it is applied to the
FM modulator. For example, 50% raised cosine shaping of the pulses for each half baud
could be used.
The phase change caused by this frequency deviation during the baud when time is taken
relative to the start of the baud is
Z t
bn (t) = π∆f I(n) p(τ ) dτ for 0 ≤ t < Ts (13)
0
3
It is convenient to define the following functions:
F (f ) = E {Fn (f )} (17)
3. n o
F̄b (f ) = E F̄n (f )ejBn (18)
4. The average squared magnitude of a pulse transform
n o
P (f ) = E |Fn (f )|2 (19)
where
sin π[f Ts − (2n − 1 − M ) h2 ] sin πTs [f − (2n − 1 − M ) ∆f
2
]
An (f ) = h = ∆f (22)
[f Ts − (2n − 1 − M ) 2 ] πTs [f − (2n − 1 − M ) 2 ]
cos(2πf Ts − αn,m ) − ψ cos αn,m
Bn,m (f ) = (23)
1 + ψ 2 − 2ψ cos 2πf Ts
αn,m = πh(m + n − 1 − M ) (24)
sin M πh
ψ = (25)
M sin πh
2
John G. Proakis, Digital Communications, 2nd Ed., McGraw-Hill, 1989, p. 200.
4
When h is an integer, S(f ) also contains Dirac impulses indicating discrete spectral compo-
nents as well as the distributed part.
Notice that the An (f ) sinc function has its peak at the tone frequency (2n − 1 − M )∆f /2
and zeros at multiples of the symbol rate, fs from the tone frequency. This is exactly what
would be expected for a burst of duration Ts of a sinusoid at the tone frequency. The
remaining terms account for the continuous phase property and give a narrower spectrum
than if the the phase were discontinuous.
The phase deviations over a symbol are the triangular shaped functions
(
−π∆f I(n)t for 0 ≤ t < Ts /2
bn (t) = (27)
π∆f I(n)(t − Ts ) for Ts /2 ≤ t < Ts
∆f
I(n)] T2s sin π[f + ∆f I(n)] T2s
= e−jπ[f + 2 2
π[f + ∆f
2
I(n)]
∆f
I(n)] 3T2s −jπ∆f Ts I(n) sin π[f − ∆f I(n)] T2s
+ e−jπ[f − 2 e 2
(28)
π[f − ∆f
2
I(n)]
The transform of a modulated pulse essentially is the sum of two sinc functions, one
centered at the frequency ∆f /2 and the other at −∆f /2. The zeros of the sinc functions
occur at multiples of 2fs away from the peaks. Therefore, the spectrum for Manchester
coding is wider than for the previous example with rectangular frequency pulses.
Let the unshifted sinc function be denoted by
sin πf T2s
A(f ) = (29)
πf
and the pulse transform for I(n) = 1 be
∆f
) T2s ∆f
) 3T2s −jπ∆f Ts
G(f ) = Fn (f )|I(n)=1 = e−jπ(f + 2 A(f + ∆f
2
) + e−jπ(f − 2 e A(f − ∆f
2
) (30)
5
The phase deviation b(n) for I(n) = −1 is the negative of that for I(n) = 1 so the modulated
pulses are complex conjugates. It can be shown by direct computation or by the transform
of conjugates property that the transform for I(n) = −1 is
The following identity will be useful in computing the squared magnitudes of several
complex sums of functions. Let z1 and z2 be complex numbers. Then
Also,
|G(−f )|2 = |G(−f )|2 = |G(f )|2 (34)
The following formula can be derived with some algebra and will be required in computations
below:
∆f
<e{G(f )G(−f )} = A2 (f + 2
) cos π(f − ∆f
2
)Ts + A2 (f − ∆f2
) cos π(f + ∆f
2
)Ts
+ 2A(f + ∆f2
)A(f − ∆f2
) cos π∆f T2s (35)
6
Using the formulas derived above, this result can be expressed as
∆f
Ts Sv (f ) = A2 (f + 2
) sin2 π(f − ∆f 2
) T2s + A2 (f − ∆f 2
) sin2 π(f + ∆f 2
) T2s
−2A(f + ∆f 2
)A(f − ∆f2
) sin π(f + ∆f 2
) T2s sin π(f − ∆f 2
) T2s
∞
1
+ |F (f )|2
X
δ (f − nfs ) (42)
Ts n=−∞
∆f Ts 2
h i
= A(f + ∆f
2
) sin π(f − ∆f Ts
2
) 2
− A(f − ∆f
2
) sin π(f + 2
) 2
∞
1
+ |F (f )|2
X
δ (f − nfs ) (43)
Ts n=−∞
3 FSK Demodulation
Continuous phase FSK signals can be demodulated using any of the methods described in the
Frequency Modulation notes. These include the frequency discriminator, the approximate
discriminator, and the phase-locked loop. In a wireless environment with rapid and deep
fading, it will be difficult to keep a PLL locked. Also, with TDMA transmission where
the signals are sent in bursts, a PLL will not have time to reliably acquire the signal. A
discriminator will be more effective in these cases. A high quality discrete-time discriminator
implementation is shown in the FM lecture notes. The approximate discriminator discussed
in the FM notes with a sampling rate of several samples per half symbol should be adequate
for the IS-54-B Manchester encoded signal.
Another FSK demodulator is shown in Figure 2 for the binary FSK case. In the more
general M -ary case, the input FSK signal is applied to M bandpass filters tuned to the
tone frequencies. The energies of the filter outputs are measured. At the end of each baud,
the frequency corresponding to the filter with the largest output energy is selected as the
estimate of what was transmitted. This process can be implemented in the binary case as
shown in Figure 2. The outputs of the upper and lower bandpass tone filters are squared and
subtracted. The result is passed through a lowpass filter whose output is a measure of the
difference in the energies of the tone filter outputs. When the upper tone is transmitted the
result will be positive and when the lower tone is transmitted it will be negative. Therefore,
the decision can be made by simply slicing (quantizing) the lowpass filter output to a 1 or 0.
The filters must be carefully designed to have adequate bandwidths and time constants. In
my experience, the discrete-time discriminator shown if the FM lecture notes works better
for signals with small modulation index.
7
Upper
- Tone - (·)2
Filter
+
s(t)
?
Lowpass
+ - - Slicer -
Filter
−
6
Lower
- Tone - (·)2
Filter