REALIZATION OF FSK MODEM IN MATLAB

AND TIs TMS320C5510 DSP

PROJECT MENTOR
PROF.V.RAJBABU
DEPARTMENT OF ELECTRICAL ENGINEERING
I.I.T BOMBAY



PREPARED BY
ABRAR AHMAD
TRAINEE (August 2010 April 2011)
For PROJECT SUPPORTED BY BHARTI CENTRE
DEPARTMENT OF ELECTRICAL ENGINEERING,
IIT BOMBAY

MATLAB IMPLEMENTATION OF FSK MODEM

Frequency shift keying Modulator/Demodulator also referred as v.21 by
International telecommunication union(ITU) is a system which uses 1650Hz and
1850Hz tone to transmit 1s and 0s respectively .The data rate of the system is
300bits/sec .The sampling rate is 9000Hz.

MODULATOR:
The data to be transmitted is the input by the user in decimal form. Sampling
frequency (Fs) and bit duration are also taken as input .The decimal string is
converted in the binary form and stored in an array. Two sinusoids X and Y of
frequency 1650Hz and 1850Hz respectively are generated. X is transmitted when
the bit is 1 and Y when the bit is 0. The end signal consists of 30 samples per bit
having two frequency components 1650Hz or 1850Hz depending upon the bit
pattern of the input signal.
The final signal is not a continuous sinusoid due to the change in phase after
every bit. This is because the two frequencies 1650Hz and 1850Hz are not
orthogonal frequencies i.e. they are not integer multiples of bit rate.
The modulator function returns sinusoid signal, sampling frequency (Fs) and the
length of the binary array.


The figure above shows the generated signal for [7 7 6] as input from
120
th
sample to 240
th
sample. The change in phase is occurring when
the bit value changes from 0 to 1.





The figure above is the frequency spectrum of the modulated signal.
The dominant frequencies are represented by the bright red color. The
maximum value on the y-axis is equal to half of the sampling frequency
so we can see that the frequencies present in the signal are around
1700 Hz.




DEMODULATOR:
1. MIXER:
The first operation on the input signal to the demodulator is frequency shifting. It
is done by multiplying the signal with complex exponential e
-j2Fct
.The mixer
output will consist of the sum and difference components of the centre frequency
and the two input frequencies. Fc is chosen as 1750Hz so that the two frequencies
1650Hz (F
1
) and 1850Hz (F
2
) are shifted down to +/-100Hz.



The figure above is the frequency spectrum of the mixer output .It consist of the
original frequencies (Fc ,F1, F2) and there sums and differences.

DECIMATOR:
It consists of a down sampler and a low pass filter before the down sampler to
maintain the nyquist criteria. The original signal having 9000 samples per second
is down sampled by a factor of 6 i.e. the sampling frequency is now 1500
samples/sec. In order to prevent the signal from aliasing effect the maximum
frequency component in the signal cannot be greater than half of the sampling
rate (750Hz) .So the filter before the down sampler has a cut off frequency of
750Hz.

Magnitude vs. Frequency plot for the filter described above.


We can see that the cut off frequency is around 0.17 rad/sample which is 750
Hz.

0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
-70
-60
-50
-40
-30
-20
-10
0
Normalized Frequency ( rad/sample)
M
a
g
n
i
t
u
d
e

(
d
B
)
Magnitude Response (dB)
SLICING:
The next operation is to measure the frequency of the received tone. Since we are
looking at one tone at a time we use a faster method for measuring frequency
called slicing. Slicing is defined as follows
s[n] =y[n] (y[n-1])*

If
y[n] =Ae
jwn

Then
s[n]=A e
jwn
A*e
-jw(n-1)
s[n]=|A|
2
e
jwn
Taking the imaginary part of s[n],
imag_s =|A|
2
sin(w)

Now we define and array d[n] such that, if imag_s is a negative value we store a
1 in the array and if imag_s is positive we store a 0 in the array.

So we conclude that when the frequency is negative, |A|
2
sin(w) is also negative
and we have a 1. Similarly when frequency is positive the imaginary part of s[n]
is positive and we have a 0


The figure below is time domain plot of imaginary part of s[n] when the input is
[7 7 6] with an extra 0 transmitted at the end.

The horizontal axis goes till 160 samples i.e. 5 samples per bit and altogether 32
bits. From the figure we can observe that for,
X=8; Y=0.09874 (positive value for symbol 0)
& X=33; Y=-0.1106 (negative value for symbol1)

The reason for this observation is that we had chosen the center frequency (Fc) as
1750Hz at the mixer stage and original frequencies 1650Hz and 1850Hz were
shifted to +/-100 Hz. So when the frequency is negative (i.e. when the symbol is 1)
the imaginary part of the slicer output (used for decoding) is negative and when
the frequency is positive (i.e. when the symbol is 0) the imaginary part of the
slicer output (used for decoding) is positive.
But what if the centre frequency was not 1750Hz .Say for example Fc was 1550Hz.
Then the frequencies 1650Hz and 1850Hz will be shifted to100Hz and 300Hz and
both of them are positive values.
The figure below is the time domain plot of the imaginary part of the slicer output
s[n] for the same input [7 7 6], but the centre frequency (Fc) is 1550Hz at the
mixer stage.

From the figure we observe that for,
X=8; Y=0.2176
X=33; Y=0.1057

Both the 8
th
and 33
rd
samples are positive values .The reason for this is that, since
both the shifted frequencies are positive i.e. 100Hz (for 1) and 300Hz (for 0) the
imaginary part of slicer output will be positive .But we can see that the sample
values (used for decoding) are around 0.2 when a 0 is transmitted and around
0.1 when a1 is transmitted .So in this case, threshold value for the decoding
purpose can be chosen as 0.15 ,instead of 0 as in the previous case for Fc=1750Hz




DECODING OF BITS:
d[n] is an array of 1s and 0s with 5 bits per symbol as the signal was down
sampled by a factor of 6.The centre bit for every 5 bits i.e. 3
rd
,8
th
,13
th
and so on
are taken as the final output as they are least effected by interference from
adjacent symbols.
The stem plot of the final decoded signal is shown below.



The final decoded output is,
[A]= [000001110000011100000110]
i.e. [7 7 6].





REALTIME IMPLEMENTATION OF FSK MODEM

All the specifications described in the mat lab implementation part are same for
real time implementation same except of the sampling frequency which is
changed to 48000Hz

MODULATOR:
The data to be transmitted is modulated by using two sinusoids of different
frequencies. Modulated signal can be generated in two ways which are described
below.
1. Sine table method: Sine tables are arrays with stored values of samples of
one period of a sinusoid of given frequency. Two such tables are used to produce
two sinusoids, one for bit 1 and other for bit 0. But there is a limitation on this
method .Not all frequencies can be generated by using this method .This is
because the sampling rate on the board is fixed (8KHz, 16KHz, 32KHz or 48KHz).So
not all frequencies will have number of samples per period as integer. For
example say the sampling frequency is fixed at 48KHZ .So a frequency of 1700Hz
will have 28.23 samples per period which is not possible as it has to be an integer.

2. Difference equation method: This is a more efficient way of producing a
modulated wave as we can produce wave of any frequency. The sine wave and
cosine wave are described as

sine [n+1] = sine[n] + (w1/fs)* cosine[n]
cosine [n+1] = cosine[n] - (w1/fs)* sine[n+1]

Where w1 is the required angular frequency and fs is the sampling rate.


DEMODULATION:

1. MIXER
Mixing is multiplying the input signal with an exponential of frequency Fc. The
exponential consist of real part i.e. the cosine term and imaginary part i.e. the
sine term. The cosine and the sine waves of frequency Fc can be generated by the
above two described methods. The output of the mixer will consist of real part
and imaginary part. Further processing will be done independently on these two
parts.

//code for the mixer

for(i=11;i>0;i--) /*right shifting and storing the real and imaginary output of mixer which
will be used in filtering part*/
{
x[i]=x[i-1];
y[i]=y[i-1];
}
/*output[] is input to demodulator and mix_cos[] and mix_sine[] are two waves produced
using difference equation*/

x[0]=output[t1]*mix_cos[t2%160]; //real part of mixer output stored in x[]
y[0]=output[t1]*(-mix_sin[t2%160]); //imaginary part of mixer output stored in y[]
.



2. FILTER
Filtering is required to make sure that a nyquist criterion is maintained. A filter of
cut off 750Hz is designed in mat lab and the coefficients are used to implement
filter in real time using difference equation.

//filter

for(i=4;i>0;i--) /*right shifting and storing filtered data to be used in the next step*/

{
fx[i]=fx[i-1];
fy[i]=fy[i-1];
}

/*filter implementation using difference equation*/
// filtering of real part
fx[0]=0.0012200191904*x[0]+0.011597461978*x[1]+ 0.05083764*x[2]+
0.12316145132*x[3]+0.19806837823*x[4]+0.2302300980*x[5]+0.0012200191904*x[6]+0.011
597461978*x[7]+ 0.05083764*x[8]+ 0.12316145132*x[9]+ 0.19806837823*x[10];

// filtering of imaginary part
fy[0]=0.0012200191904*y[0]+0.011597461978*y[1]+ 0.05083764*y[2]+
0.12316145132*y[3]+0.19806837823*y[4]+0.2302300980*y[5]+0.0012200191904*y[6]+0.011
597461978*y[7]+ 0.05083764*y[8]+ 0.12316145132*y[9]+ 0.19806837823*y[10];



DOWNSAMPLER:

Down sampling is reducing the number of samples which will undergo further
processing .In real time it is done by using a while loop which selects every fifth
sample for further processing .Hence sampling rate is reduced to one fifth of the
original rate.



//Code for down sampling
if(counter%5==0) //every fifth sample enters the loop
{
dfx[0]=fx[0];
dfy[0]=fy[0];
dfx[1]=fx[1];
dfy[1]=fy[1];

..


Further processing will be done with in the if loop i.e. on every fifth sample. This
if loop will close at the end.




SLICING:
The next and the final step for demodulation is to measure the incoming frequency.
It is done using a method called Slicing which is described in the mat lab
implementation part. It is defined as
s[n] =y[n] (y[n-1])*
It is basically multiplication of the signal with its delayed conjugate.

// code for slicing
sdfx=(dfx[0]*dfx[1])+(dfy[0]*dfy[1]); //slicing done on real part of signal
sdfy=(dfy[0]*dfx[1])-(dfx[0]*dfy[1]); // slicing done on imaginary part of signal

Decision whether the data is 1 of 0 is done by checking the sign of the
imaginary part of the slicer output. Code for the same is described below.

if(sdfy<0)
{
final [counter1]=1;
sum++;
}
else
final[counter1]=0;

counter1++;

CONCLUSION:

Matlab implementation of the modulator part as well as the demodulator part
worked satisfactorily. Data transmitted and received were the same. Real time
implementation of the modulator part using both the look up table and the
difference equation worked well. But demodulator didnt work well because
reading data sample by sample and then processing it is not possible as during the
processing time, DSP will read any random data and, the data transmitted and
received will not be the same.















FUTURE WORK:

Reading the data sample by sample and then processing it is not possible as during
the processing time, DSP will read any random data as mentioned earlier. So this
problem can be overcome by using two kits one for modulation and other for
demodulation. But using two kits will require synchronization between the two kits
and synchronization of the two kits is tough to achieve as the demodulator kit
needs to start processing as soon as it receives the data which is practically
difficult. So kit will read any random data when the actual data is not coming and
process it. And we need to transmit preamble along with the data to recognize the
starting of the data and a decoding algorithm has to be included in the demodulator
part that recognizes the preamble. Another possible solution is to make use of ping
pong buffers where we process a group of data instead of individual samples. In
this case one buffer (ping) receives the data and stores to its capacity. Once it is
full it transfers the data into the other buffer (pong) and collects the new data
coming in. The data in the pong buffer is processed.
REFERENCES:


1. Sen M Kao, Bob H. Lee and Wenshun Tian , Real time Digital Signal
processing Implementation and Applications , 2
nd
edition , John Wiley &
sons Ltd. , 2006

2. ALAN V.OPPENHEIM ,ALAN S.WILLSKY, SIGNALS & SYSTEMS
published by Prentice Hall , second edition,1997

3. Signal Processing First book ,lab 15:Digital Communication :FSK MODEM