5-IJAEST-Vol-No.4-Issue-No.2-DEVELOPOMENT-OF-PROGRAMMABLE-DEMODULATOR-USING-ARM-PROCESSOR-018-022 by ISERP ISERP

Bhargav Ch, et al. / (IJAEST) INTERNATIONAL JOURNAL OF ADVANCED ENGINEERING SCIENCES AND TECHNOLOGIES Vol No. 4, Issue No. 2, 018 - 022

DEVELOPOMENT OF PROGRAMMABLE DEMODULATOR USING ARM PROCESSOR Div: RC & BSD/SDRS Org: National Remote Sensing Agency Area: Balanagar,Hyderabad,Andhrapradesh Mail id: bhrgvkrshn@gmail.com

Div: RC & BSD/SDRS Org: National Remote Sensing Agency Area: Balanagar,Hyderabad,Andhrapradesh

A. Functional description of S-band down converter S-Band down Converter is a stand-alone unit operating on +15VDC with 500ma current derived from the power supply (control room). The S-Band Synthesized single channel down converter translates input band of frequencies in the range of 2200-2300 MHz to give a fixed IF frequency of 70MHz and tuning of the Down converter local oscillator frequency is derived through phase lock multiplier whose reference frequency is derived from frequency synthesizer unit [1-4]. The technical description of the Down converter is done with reference to the block diagram as shown in fig.1.

Abstract— Many Embedded systems need precise computing, minimum power dissipation at highspeed computations and smaller code size for given computations. MCU based on 32-bit ARM architecture (LPC2378) provide these features. The ARM is a 32-bit Reduced Instruction Set Computer (RISC), Instruction Set Architecture (ISA) developed by ARM Holdings. In the current paper, the Hilbert transform technique is used for digital demodulation, this demodulator facilitates programmability, which includes the selection of the type of the modulation, the necessary input parameters like data rate, carrier frequency etc., are to be provided interactively. The discrete Hilbert transform is a process used to generate complex-valued signals from real-valued signals. The Use of complex signals in lieu of real signals has simplified and improved the performance of signal-processing in the demodulation process.

Name: Radha Nayani,SCI’F’

Name: Bhargav Ch

Keywords: ARM, Complex down Conversion, Hilbert Transform.

I. INTRODUCTION The objective of the paper is to demodulate the modulated signal received from the satellite S-band. The input RF frequency range is 2200-2300MHz and the nominal input

level is -45dBm. The S-band RF signal is down converted to an IF of 70MHz using a Fig 1: Block diagram of S-band Down Converter downconverter as shown in the figure1.

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Signal in the RF band of 2200-2300 MHz is fed to the RF port of mixer thru an isolator (IS1) and band pass filter (BPF1) having an insertion loss of 1dB. The Filter provides flat amplitude characteristics in the pass band and provides very high isolation to the image frequency. Signal frequency for LO port for Mixer (M1) i.e., 2130 to 2230 MHz is derived through a phase lock multiplier, its multiplication factor is 20 and its output power is +15dBm, whose reference frequency in the range of 106.5 to 112 MHz to derive an output frequency of 2130 – 2230 MHz and the reference input level should be of the order of 0+/-3dBm, to give a constant IF of 70 MHz, which is fed to an amplifier with an amplification of 30dB and subsequently through the unit can also be configured remotely through the TCP/IP protocol interface supported by this processor. The Firmware developed also caters to Monitoring and Control of this unit. The Hilbert transform and the ARM processor are used for the programmable demodulation [5-8].

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Bhargav Ch, et al. / (IJAEST) INTERNATIONAL JOURNAL OF ADVANCED ENGINEERING SCIENCES AND TECHNOLOGIES Vol No. 4, Issue No. 2, 018 - 022

To=1/Fo = (2N*Ts)/ (∆ACC) DIRECT DIGITAL SYNTHESIS

Fo = Fo*(∆ACC) = (Fs/2N)*(∆ACC)

(2)

The phase increment, ∆ACC rounded to the nearest integer ([x] is the integer part of x), is given by ∆ ACC = [Fo*(2N/Fs) +0.5]

(3)

Equation (2) is the basic equation of any DDS system. One can infer from (2) the tuning step ∆Fomin, which is the smallest step in frequency, which the DDS can achieve (remember that ∆ACC is an integer) ∆Fomin = Fs/2N

(4)

Equation (4) allows the designer to choose the number of bits (N) of the accumulator ACC. This number N is often referred to as the frequency tuning word length.

The digital signal processing operation we want to perform is to generate a periodic, discrete-time waveform of known frequency Fo. The waveform may be a sine wave. It can also be a saw-tooth wave, a triangle wave, a square wave, or any periodic waveform. We will assume that the sampling frequency Fs is known and constant. Before proceeding with the theory of operation, we summarize why DDS is a valuable technique. 1) The tuning resolution can be made arbitrarily small to satisfy almost any design specification. 2) The phase and the frequency of the waveform can be controlled in one sample period, making phase modulation feasible. 3) The DDS implementation relies upon integer arithmetic, allowing implementation on virtually any microcontroller. 4) The DDS implementation is always stable, even with finite-length control words. There is

We can rewrite (1) in terms of frequency Fo, as a function of ∆ACC:

II.

(1)

no need for an automatic gain control. 5) The phase continuity is preserved whenever the frequency is changed (a valuable tool for tunable waveform generators). A. Theory of Operation and Implementation

The implementation of DDS is divided into two distinct parts as shown in Figure 2: a

discrete time Phase generator (the accumulator) output a phase value ACC, and a phase to waveform converter outputting the desired DDS Fig2: Implementation of DirectDigitalSynthesizer

From a Sampling Frequency to a Phase

The implementation of the DDS relies up on integer arithmetic. The size of the accumulator (or word length) is N b. Assuming that the period of the output signal is 2π rad, the maximum phase is represented by the integer number 2N. Let us denote ∆ACC the phase increment related to the desired output Fo frequency [9-10]. It is decoded as an integer number with N − 1 b. During one sample period Ts, the phase increases by ∆ACC. It thus takes To to reach the maximum phase 2N:

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The minimum frequency Fomin the DDS can generate is given by (2) with ∆ACC = 1, the smallest phase increment which still increases the phase (∆ACC = 0 does not increase the phase). Fomin=Fs/2N (5) The maximum frequency Fomax the DDS can generate is given by the uniform sampling theorem Fomax=Fs/2 (6) From a practical point of view a lower Fomax is often preferred, Fomax= Fs/4 for example. The lower that Fomax is, the easier the analog reconstruction using a low pass filter. C.

From a Phase to a Waveform

The phase is coded with N b in the accumulator. Thus, the waveform can be defined with up to 2N phase values. In case 2N is too large for a realistic implementation, the phase to amplitude converter uses fewer bits than N [11]. Let us note P as the number of bits used as the phase information (with P ≤N). The output waveform values can be stored in a lookup table (LUT) with 2P entries: the output value is computed as: Output = LUT (ACC), This is implemented in the phase to waveform converter in Figure 1. DDS can generate a sine wave with an offset b and a peak amplitude a. The content of the LUT, containing the DDS output values, is computed for the index i ranging from 0 to (2P − 1) using:

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Bhargav Ch, et al. / (IJAEST) INTERNATIONAL JOURNAL OF ADVANCED ENGINEERING SCIENCES AND TECHNOLOGIES Vol No. 4, Issue No. 2, 018 - 022

LUT (i) = [b+a*sin (2*pi*i/2P) +0.5]

(7)

The complex xc(t) signal is known as an analytic signal (beca use it has no negative-frequency spectral components), and its real part is equal to the original real input signal xr(t). If a real sine wave xr(t) is amplitude modulated so its envelope contains information, from an analytic version of the signal we can measure the instantaneous envelope E(t) value using:

(13)

That is, the envelope of the signal is equal to the magnitude of xc(t).We show a simple example of

THE DISCRETE HILBERT TRANSFORM

IV.

Fig7: Functional relationship between the xc(t) and xr(t) signals.

The DDS generates a complex sinusoid at the intermediate to down converting by creating a difference signal at the IF minus the DDS frequency, they also up-convert, generating an unwanted signal at the sum of the two frequencies. Any suitable low-pass filter can be used including FIR, IIR and CIC filters. The most common choice is a FIR filter for low amounts of decimation (less than ten) or a CIC filter followed by a FIR filter for larger down sampling ratios. The DDC is typically used to convert an RF signal down to base band. It does this by digitizing at a high sample rate, and then using purely digital techniques to perform the data reduction. Being digital gives many advantages, including: • Digital stability – not affected by temperature or manufacturing processes. With a DDC, if the system operates at all, it works perfectly – there’s never any tuning or component tolerance to worry about. • Controllability – all aspects of the DDC are controlled from software. The local oscillator can change frequency very rapidly indeed – in many cases a frequency change can take place on the next sample. • Size. A single ADC can feed many DDCs, a boon for multi-carrier applications.

The Hilbert transform (HT) is a mathematical process performed on a real signal xr(t) yielding a new real signal xht(t) the HT of xr(t) as shown in figure6. In words, we can say that all of xht(t)'s positive frequency components

this AM demodulation idea in Figure 8(a).we compute the magnitude of xc(t) to extract the modulating waveform shown as the bold solid curve in Figure 8(b). The |xc(t)| function is a much more accurate representation of the M.W

are equal to xr(t)'s positive frequency components shifted in phase by –90o. Also, all of Fig6: The Hilbert transformer

Fig8: Envelope detection: (a) input xr(t) signal;

xht(t)'s negative frequency components are equal to xr(t)'s negative frequency components shifted in phase by +90o . Mathematically,

Suppose, on the other hand, some real xr(t) sine wave is phase modulated. We can estimate xc(t)'s instantaneous phase ø(t), using:

Xht(ω)=H(ω)Xr(ω)

(11)

If we start with a real time-domain signal xr(t), we can associate with it a complex signal xc(t),as shown in figure7.

xc(t)=xr(t)+jxi(t).

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(12)

(b) Complex envelope detection result |xc(t)|.

Ø (t) =tan-1(xi (t)/xr (t))

(14)

Computing ø(t) is equivalent to phase demodulation of xr(t). Likewise If a real sine wave carrier be frequency modulated we can measure its instantaneous frequency F(t) by

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Bhargav Ch, et al. / (IJAEST) INTERNATIONAL JOURNAL OF ADVANCED ENGINEERING SCIENCES AND TECHNOLOGIES Vol No. 4, Issue No. 2, 018 - 022

F (t) = d/dt (Ă&#x2DC; (t)) = d/dt (tan-1(xi (t)/xr (t))) (15)

Figure 9 shows the overall block diagram.

Fig 9: Overall Block Diagram

Complex down Conversion:

The complex down conversion results in the in-phase and the quadrature phase signals as shown in figure 12 and 13.The figure 11 shows the in-phase signal and quadrature phase signal before low pass filtering. The figure12 shows the characteristics of a low pass filter with cutoff

frequency 1KHz.The figure 13 shows the Inphase component and figure 14 shows the Quadrature phase component after passing through a low pass filter.

1) The modulated signal is received from the satellite S-band. It is a phase modulated signal of the range 2-4GHz.It is down converted to a signal of IF 70MHz. 2) This IF PM signal is digitized using the A/D Converter whose sampling rate is four times that of the maximum frequency of the incoming signal. The A/D converter used is of Successive approximation type and the ADC is of 10-bit resolution. Thus the output of the ADC is digital PM signal. 3) As the PM signal is received from the space, it is mixed up with atmospheric noise. To recover the original signal, raised cosine filtering of the PM signal is done. 4) The digital PM signal is down converted using the complex down conversion scheme which generates an In-phase Component and Quadrature-Phase component. 5) The digital down converted PM Signal is processed in the ARM board. Using the Hilbert transform, the demodulation of PM signal takes place in ARM board. 6) The output can be observed as digital signal in logic analyzer as a digital signal or it can be viewed in a spectrum analyzer or it is converted from digital to analog form using a DAC and viewed as an analog signal using the Oscilloscope. 7) The unit can also be configured remotely through the TCP/IP protocol interface supported by this unit. 8) The firmware developed also caters to the Monitoring and Control of the unit. V. RESULTS AND DISCUSSIONS

calculating the instantaneous time rate of change of xc(t)'s instantaneous phase using:

Input data The Phase modulated input data received from the satellite is shown in figure10. Fig 10: Phase modulated data received from the satellite

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Fig 11: The In-phase and Quadrature phase signals. Fig 12: Low pass filter characteristics Fig 13: Output of a low pass filter: In-phase component

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Bhargav Ch, et al. / (IJAEST) INTERNATIONAL JOURNAL OF ADVANCED ENGINEERING SCIENCES AND TECHNOLOGIES Vol No. 4, Issue No. 2, 018 - 022

[2] Vaclav Cizek,’ Discrete Hilbert Transform’, IEEE Transactions on Audio and Electro acoustics Vol. AU-18, NO. 4 DECEMBER 1970. [3] Ing. Ivo VIŠČOR, Ing. Josef HALÁMEK, CSc ,’DSP and digital down conversion’, Proceeding of the Radioelektronika 99, Brno 1999, pp.225-228. [4] Communication System Design Using DSP Algorithms with Laboratory Experiments for the TMS320C6713 DSK by Steve Tretter.

Fig 14.Output of a low pass filter: Quadrature-

[5] Understanding Digital Signal Processing Richard G Lyons. [6] Andrew N.Sloss,Dominic Symes,Chris Wright,’ARM System Developers Guide-Designing and Optimizing System Software’.

Phase component.

Figure 15 shows the original modulated message signal that is demodulated using the Hilbert transform. VI.

[8] S. Note, J. van Meerbergen, Catthoor, and H. de Man, “Automated Synthesis of a High Speed CORDIC Algorithm with the Cathedral-III Compilation System,” in Proc. IEEE International Symposium on Circuits and Systems (ISCAS), June 1988, pp. 581-584.

[9] P. Nuytkens, and P. V. Broekhoven, "Digital frequency synthesizer," U. S. Pat. 4,933,890, June 12, 1990.

Fig15. Demodulated output using Hilbert transform

[7] Bernard Sklar, ’Digital Communications Fundamentals and Applications 2E’.

CONCLUSION

[10] A. V. Oppenheim, and R. W. Schafer, "Digital Signal Processing," Prentice-Hall, Englewood Cliffs, New Jersey, 1975.

[11] J. Nieznanski, “An alternative approach to the ROMless direct digital synthesis,” IEEE J. of Solid State Circuits, Vol. 33, No. 1, pp. 169 -170, Jan. 1998.

The demodulation using the Hilbert transform yields much more accurate results than the traditional analog demodulation techniques. The unit is designed to be a stand alone unit. The firmware developed also caters to the Monitoring and Control of the unit. The range of the Analog to Digital Converter (ADC) is limited. Its range can be extended using the discrete ADC components. The hardware and firmware requirements to handle higher data rates signals with more complex modulation techniques like BPSK, QPSK and 8-PSK are being worked out. ACKNOWLEDGMENT

The authors like to express their thanks to the management and the Department of ECE, KL University and National Remote Sensing Center (NRSC), ISRO for their moral encouragement and support during this work. REFERENCES [1] Rong Zhang’, Xian-Ci Xiao’, Heng-Ming Tai’ ,’An efficient Digital Down Conversion Method For Multiple Wide Band Signals’, 0-7803-7523-8/02 @2002 IEEE.

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