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Taura et al.: A New Approach to VHF/FM Broadcast Receiver Using Digital Signal Processing Techniques

75 1

A NEW APPROACH TO VHFlFM BROADCAST RECEIVER USING

DIGITAL SIGNAL PROCESSING TECHNIQUES
Kenichi Taura', Masahiro Tsujishita', Masayuki Tsuji', Eiji Asano' and Masayuki Ishida'
'Imaging Systems Laboratory, 'Sanda Works, Mitsubishi Electric Corporation, Japan

ABSTRACT
This paper describes the development of a VHFIFM
broadcast receiver using digital signal processing
techniques. The developed method gives improved
receiver performance, and enables it to be integrated
into general digital signal processing.

We have also developed an extended sampling

rate conversion technique to synchronize the
sampling rate with the broadcast pilot signal utilizing
interpolation-decimation filtering, while the stereo
decoder process itself has been based on a
synchronous sampling type operation [2].

1. INTRODUCTION
We have developed a new VHF/FM broadcast
receiver that realizes FM demodulation, stereo
decoder and noise reduction by using digital signal
processing. We adopted a quadrature type FM
demodulator [I] for its simple structure, and
combined it with a newly developed AM (amplitude
modulation) cancellation process, to reduce the
demodulated signal distortion.

2. RECEIVER STRUCTURE
Figure 1 shows a block diagram of the FM
broadcast receiver. A double conversion system is
adopted for analog signal processing to make a 608
kHz IF (intermediate frequency) signal suitable for
input to the FM demodulator. An anti-alias analog
filter is placed before the AD converter to suppress
the higher harmonic component of the IF signal.

Fig. 1 FM Receiver Signal Processing Block Diagram

Original manuscript received June 19,2000

0098 3063/00 $10.00 2000 IEEE

IEEE Transactions on Consumer Electronics, Vol. 46, No. 3, AUGUST 2000

152

3. FM DEMODULATION
Figure 2 shows a detailed block diagram of the
FM demodulator. The demodulator is composed of a
basic quadrature demodulator and an AM
cancellation process. An arcsine compensation
process is also incorporated for the further reduction
of demodulation signal distortion.
3.1 Basic Quadrature Demodulator
The basic demodulation process comprises a
one-sample timing delay, the multiplication of two
sample data items before and after the delay and a
low pass filter for the products, as shown in the
figure. The quadrature demodulator operation can be
expressed as follows, by denoting the sample data
before and after the delay, by x0 and x l ,
respectively.
X O = AC * ~ 0 ~ { 2 n f c +
k Tp ( k T ) }

~ l AC
= ' ~ 0 ~ { 2 n f ~l)T
( k+-p ( ( k - 1)T))
Where, the p(kT) and p((k-l)T) denote the phase
angle due to the modulation and the Ac denotes the
input signal amplitude. The required condition
between the delay time T and the FM carrier
frequency fc, to form a quadrature demodulator, is
known as, 2 n f c T = n / 2 + m . n . Where, m = 0, 1,
2, ...
BY applying the condition, with setting m = 0
corresponding to Our system, the multiplication
product becomes,
~ 0 . =~Ac'
1 *sin(p(kT)-p((k-l)T)}/2

+AcZ+sin{4nfckT+p(kT)
+ p ( ( k - I)T)}/2
The low frequency component of the product yo can
be expressed as follows.

y o = AcZ .sin{p(kT) - p ( ( k - 1)T))

s Ac'

T{dp(t)/dt}/2

- - - (I)

Since dp(t)/dt indicates a time differential of the

modulation phase, this indicates the modulation
frequency itself. Therefore yo gives the demodulated
output of the modulation signal dp(t)/dt. To fulfill the
conditions of quadrature demodulation, we selected
T = 112.432 microsecond and fc = 608 kHz.
From Equation 1, it can be understood that the
following two problems are implied in the quadrature
type demodulator.
1 . The demodulated output is directly proportional to
the input signal power Ac'. This means that any
fluctuation of the amplitude Ac would result the
amplitude fluctuation of the demodulated signal in
proportion to be square of the input amplitude.
2. The demodulation characteristic is sine function
shaped, to be precise. The non-linear demodulation
characteristics cause distortion in the demodulated
sound signal, especially when the modulation index
is large.
Since a demodulator input signal usually
contains higher harmonic components and other
spurious components, it is necessary to filter out
these spurious components before the demodulation
processing. This filtering process unavoidably
induces secondary AM on the signal, according to its
amplitude-frequency characteristics, and this AM of
the signal is often the primary factor of the
demodulated signal distortion. To resolve this
problem, we have adopted a newly developed AM
cancellation process to accompany the basic
demodulator.

Quadrature Demodulator

FM

Timing
Adjust.

4
L.-----

1Ix Calculation
(Polynomial
----....---------.-------~-------~.---~~-.
~1
Approx.)
Input Signal Power Detection

Fig. 2 FM Demodulation Process Block Diagram

Taura et al.: A New Approach to VHFFM Broadcast Receiver Using Digital Signal Processing Techniques

3.2. AM cancellation Process

The AM cancellation process comprises (a)
detection of the input signal power Ac', (b)
calculation of the compensation coefficient inversely
proportional to Ac', (c) adjustment of the
compensation timing and (d) multiplication of the
coefficient by the demodulated signal to cancel out
the AM that appears on the demodulated signal.
-10

-20

+Witout AM
Cancellation (Mono)
th A M Cancellation

* -30

.!

-40

-50

-60
-70
600

700
800
900
1000
Filter Cut-off Frequency (kHz)

1100

Fig. 3 Effect of the AM Cancellation Process

(1 kHz sine wave 30% modulation)
To detect the input signal power, it is
advantageous to calculate the square sum of the
sample data x0 and x l , before and after the delay of
the demodulator. That is,
x 0 2 + xlz = Ac2[1 - sin(26p) .sin{4nfckT

+ 2pKk - 1)T)l!21
where, 6p = p ( k T ) - p ( ( k - 1)T)
As seen in the expression, the second doublefrequency term of the carrier has small value of
multiplier, sin ( 2 delta p)/2, depending on the
modulation index. Therefore the first signal power
term can easily be obtained by using rather simple
low-pass filter.
The calculation of compensation coefficient (b)
has been performed by second-order polynomial
approximation. We have confirmed that the
approximation gives 0.2% accuracy for +/-20% of
the input range, that corresponding I O % of AM on
the input signal. After adjusting the timing between
the demodulated signal and the coefficient, by
multiplying the signal proportional to input signal

153

power by the coefficient inversely proportional to

input signal power, the influence from the input AM
is removed.
Figure 3 shows the results of the computer
simulation of FM demodulation operation, with and
without the AM cancellation process, calculated with
changing the cut-off frequency of the input anti-alias
filter. It is evident, from the results, that the AM
cancellation gives significant improvement in
reducing the demodulation signal distortion.

3.3 Arcsine Compensation

An arcsine compensation process is also
incorporated for the further reduction of
demodulation signal distortion. We investigated a
process that compensates for the sine function
shaped demodulator characteristics by applying an
arcsine function to the demodulation signal. In
practice the process is based on the polynomial series
exmession of an arcsine function.
x3 3 5 5 7
arcsin(x)=x+-+-x
+-x
+...
6 40
112
Figure 4 shows the computer simulation results
for the signal distortion, with and without the arcsine
compensation, by changing the modulation index.
Two cases of the compensation are shown in the
figure, one is an approximation up to the third order,
and the other is an approximation up to the fifth
order. From the results, we confirmed that the third
order approximation is good enough for the
demodulation condition.
-20
-30

-40

-50
-60

E
3

-70
-80
20

40

60
80
100
Modulation Index (%)

120

Fig. 4 Effect of the Arcsine Compensation

(1 kHz sine wave monaural modulation)

140

154

IEEE Transactions on Consumer Electronics, Vol. 46, No. 3, AUGUST 2000

fs=
38KHr

fS =

FM Composite

fs

FM L-channel
Signal

Pilot Cancel

fs = 608 kHr
I-

FM R-channel
Signal

Control
Block ;

Signal

Coefficient

.. . _ _ _ _ _Phase Error Signal

_ _ _ _ _ _ A

19 kHz
Sine Table

Fig. 5 Stereo Decoder Process Block Diagram

4. STEREO DECODER
4.1. Stereo Decoder Principles
An FM composite stereo signal comprises a
base band audio signal (L + R)/2, a sub-channel and
a pilot signal. The sub-channel is a double sideband
modulated signal of a 38 kHz modulated by an audio
(L -R)/2 signal, and the pilot signal is a phase-locked
signal at half the frequency (19 kHz) of the subcarrier. It is well known that if we could sample the
composite signal every time the sub-carrier phase
reached d 2 , or the positive peak point, the sampled
data would give the L-channel decoded signal, and
the R-channel signal could be obtained by the
sampled data every time the sub-carrier phase
reached 3 d 2 , or the negative peak point.
The stereo decoder of the proposed system was
designed according to this principle. In this type of
stereo decoder, it is essential to ensure that the
sampling timing accurately coincides with the peak
point of the sub-carrier to achieve an accurate stereo
decoder and obtain good channel separation. For this
purpose, we have incorporated a phase-locked loop
operation to the pilot signal to ensure accurate
timing.
4.2. Conventional Method
In conventional systems, the sampling
frequency of analog to digital (AD) conversion is
usually controlled to synchronize it with the pilot
signal for the stereo decoder. In an analog control
system with a voltage controlled oscillator (VCO), or
a voltage controlled crystal oscillator (VCXO), to
adjust the AD conversion frequency, there must be a

digital to analog converter for the phase error signal.

This system could be easy to realize, but because of
the necessity of special analog components, and the
high cost of the product, it is not suitable for general
purpose digital signal processing.
An alternative method that controls the AD
conversion timing by using digital means, like a
programmable-counter input high-speed clock which
gives variable timing output according to the control
through its division ratio, can be employed to realize
fully digital system. However in this method,
variations of the A D conversion time interval, at least
equal to the high-speed clock period, unavoidably
occur because of the control which synchronizes the
pilot signal. The sample time interval variation At,
due to the AD conversion time interval variation,
adds a rather big step wise phase variation of 2dcAt
between the sample data before and after the one
sample delay in the FM demodulator, and the phase
variation makes a clicking noise on the output signal
because of the demodulation process. Consequently
this method was not considered feasible for our
system.

4.3 Developed Stereo Decoder

4.3.1 Structure of Developed System
Because of the problems outlined above we
adopted a newly developed synchronization method
for the pilot signal. In our method the sample timing
is controlled by using an interpolation-decimation
filtering technique applied to the FM demodulated
sound signal. Figure 5 shows a block diagram of the
stereo decoder process. As shown in Figure 5 , the

Taura et al.: A New Approach to VHFFM Broadcast Receiver Using Digital Signal Processing Techniques

stereo decoder process is composed of pilot signal

phase detection, a interpolation-decimation filter to
synchronize the output data timing with the pilot
signal phase, and a stereo decoder filter that outputs
two decimated signals, as the left and right audio
channels.

4.3.2 Operation of Interpolation-Decimation

Filter
It is well known that the interpolationdecimation filter process can be realized by a single
finite impulse response (FIR) filter calculation [4].
The basic FIR filter is designed to operate at the
sampling frequency L f s i n , and with a lower cut-off
frequency than f s i n I 2 or fs-outl2, where L denotes
the interpolation ratio, fs-in denotes the sampling
frequency of the input data, and fs-out denotes the
sampling frequency of the output data. In fact the
filter coefficient used for the calculation is one of the
L subsets of the coefficient of the basic FIR filter
taken every interval L.
Figure 6 shows an example of the input and
output signals of the interpolation-decimation filter.
The input signal waveform and sample data are
shown in Figure 6a. Figure 6b shows the L-time
over-sampled data of input in (a). Figure 6c shows
the M-time decimated data of the over-sampled data
in (b).
Since a specific L over-sampled data points,
which corresponds specific input data timing, can be
obtained by an interpolation filter operation using
one of the L coefficient sets, the M-time decimation
operation 6c can be performed by the repetition of
the filter calculation for the M interval of oversampled data, by selecting the proper input data
timing and the proper coefficient set. In the same
way, the decimation ratio M can be changed with
changing the filter calculation timing and/or the
coefficient set selection. We have used this aspect to
control the filter output data timing to synchronize
the pilot signal phase by using a phase error signal.
The phase error signal is the low frequency
component of the product of the pilot signal and an
internal 19 kHz reference signal, which is generated
by reading the sine lookup table at every filter
outputs timing. The phase error signal is then input to
the control block of the interpolation-decimation
filter. Since the control block operates to reduce the
input phase error by controlling the interpolation-

755

decimation filter, a phase-locked loop operation with

the pilot signal is performed by this loop.
In actual fact, a periodical change of the
decimation ratio M to M+l or M-1 is performed in
the interpolation-decimation filter, and the period of
the decimation ratio changes and the selection of an
increment or decrement in the decimation ratio is
controlled according to the phase error signal, by the
coefficient selection block operation. In our system
the operation parameters are set to L = 64,M = 128
and f s i n = 608kHz. Since the L fs-in become
38.912 MHz, this is enough to perform fine control
of the sampling timing.
level

lewl

I
lwei

M/(L k i n )

; time

.*

t
lime

time

Fig. 6 Interpolation-Decimation Filter Input and

Output Signals
4.4 Stereo Decoder Filter

The stereo decoder filter operates as two 1 I4

decimation filters, and it gives two individual outputs,
one for L-channel, and one for R-channel audio
output. The main purpose of the stereo decoder filter
is to avoid the possible alias from the upper
frequency band (approximately 61 kHz (76 minus
15)). It also has the effect of avoiding interference in
the audio signals from FM multiplex signals in the
higher frequency band than the stereo sub-channel
(23 to 53 kHz), and of improving the signal to noise
ratio by suppressing the alias of the high frequency
band noise.

IEEE Transactions on Consumer Electronics, Vol. 46, No. 3, AUGUST 2000

756

4.5 Pilot Cancellation Process

The pilot cancellation process first detects the
amplitude of the pilot signal on the stereo composite
signal, then adjusts the cancellation signal amplitude
generated from the sine look up table, and finally
subtracts the cancellation signal from the composite
signal. It removes the remaining pilot in the audio
output and is effective in improving the performance
of the succeeding noise cancellation process.

4.6 Stereo Decoder Performance

Figure 7 shows the computer simulation results
for the signal to noise ratio, and stereo separation and
distortion, by changing the master clock frequency
error. The simulations were done with a stereo
decoder input sampling frequency of 608 kHz by
using fixed-point arithmetic using 16-bit data
expressions. It was confirmed that the process is
tolerant of +/-300 ppm error of the master clock
frequency.

IO0
110

arithmetic using 16-bit data expressions.

Figure 8 shows the computer simulation results
of the total operation performance of the signal to
noise ratio, for various master clock frequency errors.
It was confirmed that the developed system gives
more than a 79 dB signal to noise over the master
clock frequency deviation of +/-300 ppm.

Modulation
(1 kHz sine wave)

I
I
I
I 100% I

30%
1100%
Monaural1 30%
Stereo

S/N
(dB)

Distortion
(dB)

74.0
84.4
76.3

1
I
I
I

~~

86.8

1
I
I
1

-65.7
-58.9

-67.7
-64.1

Stereo
Separation
(dB)
62.6
59. I

90

LEl=a=a
80

r.

70

iii

70

60

+Monaural
-Stereo

50

+Distortion
-3 S/N

60-300

20

-200

-100

100

200

300

Master Crock Frequency Error (ppm)

I 0o
-300

-200

-100

100

200

300

MCLK Frequency Error (ppm)

Fig. 7 Stereo Decoder Performances

( 1 kHz sine wave 100%stereo modulation,
the signs of the distortion values are inverted.)

4.7 Total Performance

Table 1 shows the computer simulation results
of the total operation performance from the FM
demodulator to the sound signal output using a 10-bit
accuracy AD converter to sample the FM input signal.
The simulations were done with an FM demodulator
input fs of 4.864 MHz by using fixed-point

Fig. 8 FM Receiver Process Performances

(1 kHz sine wave 100% modulation)

5. NOISE REDUCTION
It is almost mandatory to fit car-use receivers
with noise reduction systems. The two main causes
of system noise generation in car-use FM receivers
are the impulsive electromagnetic noise caused by
the engine ignition system, electric motors, switches
etc. and the multi-path noise caused by the multiple
propagation paths of radio waves and interference
due to the phase differences between the signals.
In our system we adopted a pulse noise

Taura et al.: A New Approach to VHF/FM Broadcast Receiver Using Digital Signal Processing Techniques

canceller that removes the noise and interpolates the

audio signal waveform when the noise has been
detected. We also adopted a stereo reception control
(SRC) process to reduce the multi-path noise. Since
the noise cancellation process operates on each audio
signal channel, it enables the secondary noise
generation that is inevitable in the conventional
method, which operates on a composite stereo signal
because of the sub-carrier loss during the noise
cancellation operation, to be avoided.

6. CONCLUSIONS
The proposed method can provide an
adjustment-free FM demodulation process with very
low signal distortion. It also allows a stereo decoder
for general purpose digital signal processing to be
realized by eliminating the necessity of a voltage
controlled oscillator to synchronize the sampling
with the pilot signal. We believe that the proposed
method can facilitate the FM radio receiver process
to be integrated into a digital LSI, and make it easier
to integrate it with various digital broadcast receiver
processes.

REFERENCES
[I] K. Kobayashi, "Performance of Quadrature Type
FM Demodulator Using Digital Signal Processing",
The Transactions of IEICE Vol. J65-B, No. 7, July
1982, pp. 890 - 897.
[2] J. E. Haug, et al., "A DSP Based Stereo Decoder
for Automotive Radio", SAE Technical Paper Series,
#900244, February 1990.
131 M. Hagiwara and M. Nakagawa, "Digital Signal
Processing Type Stereo FM Receiver", IEEE Trans.
on Consumer Electronics, Vol. CE-32, No. 1,
February 1986, pp. 37 - 43.
141 R. E. Crochiere, L. R. Rabiner, "Multirate Digital
Signal Processing", Prentice-Hall, 1983.
BIOGRAPHIES
Kenichi Taura graduated in Electrical Engineering
from Sasebo Technical College in 1971. Since 1971,
he has been with Mitsubishi Electric Corporation,
and is engaged in the research and development of
audio equipment for car use. He is a member of
IEICE.

151

Masahiro Tsujishita received his M.E. degree in

Electronics Engineering from Nippon University in
1986. Since 1986, he has been with Mitsubishi
Electric Corporation, and is engaged in the research
and development of audio equipment for car use. He
is a member of IEICE.
Masayuki Tsuji received his B.E. degree in
Electrical Engineering from Doshisha University in
1986. Since 1986, he has been with Mitsubishi
Electric Corporation, and is engaged in the research
and development of audio-visual equipment and
LSIS.
Eiji Asano graduated in Electrical Engineering from
Matsuyama Technical High School in 1965. Since
1965, he has been with Mitsubishi Electric
Corporation, and is engaged in the design and
development of audio equipment for car use.
Masayuki Ishida received his B.E. degree in
Electrical Engineering from Tokyo Institute of
Technology in 1975. Since 1975, he has been with
Mitsubishi Electric Corporation, and is engaged in
the research and development of digital audio
equipment. He is a member of IEICE.