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Milendrakumar M. Solanki1, Darshankumar C. Dalwadi2, Rajvirsinh Rana3, Tushar Patel4

1Asst. Prof., Electronics dept., B.V.M. Engineering college, V.V.Nagar,2Asst. Prof., Electronics & Telecommunication dept., B.V.M. Engineering College, V.V.Nagar,3Asst. Prof., Electronics & Telecommunication dept., B.V.M. Engineering College, V.V.Nagar,

4Senior Application Engineer, Intel Mobile Communication, Pennsylvania, USA

Abstract—The Speech signal is converted into the electrical signal through the microphone. This analog signal is pass through the codec. The analog signal is converted into the digital sample value through the A/D converter in codec. The digital sample value is pass through the DSP, where sampled value is modified by the particular filter algorithm, which is realized by the DSP processor. Now this modified sample data is send to codec. Codec again convert the digital sample value into the analog form through D/A converter. This filtered analog signal is pass through the audio amplifier and supplied to the speaker.

I. INTRODUCTION

1.1 THE NEED FOR THE SYSTEMAlmost every field of science and engineering such as

acoustics, physics, telecommunication, data communication, control system, and radar deal with signal. In many applications, it is desirable that the frequency spectrum of a signal be modified, reshaped or manipulated according to a desired specification.

The process may include attenuating a range of frequency components and rejecting or isolating one specific frequency component.

Any system or network that exhibits such frequency selective characteristics is called a filter. Filter are used in variety of applications, such as removing noised from signal, removing signal distortion due to the transmission channel, separating two or more distinct signals that were mixed in order to maximize communication channel utilization, demodulating signals and converting discrete time signals into continuous time signals. There are two types of filter analog filter and digital filter.

1.2 ADVANTAGES OF DIGITAL FILTERINGThe term digital filter refers to the computational process

or algorithm by which a digital signal or sequence of numbers termed the output digital sequence. Digital filters involve signal in digital domain, where as analog filter relate signals in the analog domain. The three principles reasons for usingdigital filter are 1) Closer approach to ideal filter approximations, 2) ability to adjust filter characteristics in software rather than by physical tuning, 3) Compatibility of

filter response with sample data. There is some of the advantages of using digital filters over their analog counter parts are high reliability, high accuracy, no effect of component drift on system performance, component tolerances not critical.

II. DSP OVERVIEW

A. What is DSP?

In brief, DSP are processors or microcomputers whose hardware, software and instruction sets an essential for processing digital data representing analog signals in real time. What a DSP does is straightforward. When acting as a digital filter, for example, the DSP receives digital values based on sample of a signal, calculates the results of a filter function operating on these values, and provides digital values that represent the filter output; it can also provides system control signals based on properties of these values. The DSP’s high speed arithmetic and logical hardware is programmed to rapidly execute algorithms modeling the filter transformation.

B. WHY USE A DSP?

To get an idea how analog circuits compares with a DSP system, one could compare two system in terms of filter function. The familiar analog filter use resistors, capacitors, inductors, amplifiers. It can be cheap and easy to assemble but difficult to calibrate, modify and maintain a difficulty that increases exponentially with filter order. For many purposes, one can more easily design, modify, and depend on filters using a DSP because the filter function on the DSP is software based, flexible and repeatable.

C. COMPARISION BETWEEN 90-TAP FIR FILTER AND SHARP CUTOFF CHEBYSHEV FILTERS

Approximating and ideal filter consists of applying a transfer function with approximate coefficients and high enough order, or number of taps. Figure 1 shows the response

Real Time Implementation of processing of sound using DSP based algorithm

13-14 May 2011 B.V.M. Engineering College, V.V.Nagar,Gujarat,India

National Conference on Recent Trends in Engineering & Technology

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of 90 tap FIR filter compared with sharp cutoff chebyshev filters of various orders.

Figure 1 Response of 90 tap FIR filter compared with sharp cutoff Chebyshev’s filters of various orders

The 90 tap example suggests how close the filter can come to approximating an ideal filter. Within a DSP system, programming a 90 tap FIR filter like one figure 1 is not a difficult task. By comparison, it would not be cost effective to attempt this level of approximation with purely analog circuits. Another crucial point in favour of using a DSP to approximate the ideal filter is long term stability. With an FIR the programmable DSP achieves the same response, time after time. Purely analog filter responses of higher order are less stable with time.

Figure 2 Block diagram of the system

III BLOCK DIAGRAM OF THE SYSTEM

From figure 2 it can be seen that codec takes the input from microphone and gives output to the audio amplifier and output of the CODEC given to the ADSP. Output of ADSP is given to the EPROM.

Figure 3 Schematic of the system

IV SCHEMATIC OF THE SYSTEM

Figure 3 shows the schematic of the system. Audio signal come into the board through the stereo jack J1. J1 is connected to the JP2 via audio amplifier. JP2 is used to configure input jack J1 for either line level or microphone input. The center pin in each group of three is connected to one of the AD1847 codec line 1 input pins, 23(UIL) and 17(UIR). Jumper J2 sterio jack bring out line level audio signal from board isconnected to codec pin 30(LDL), 28(LDR) via high pass filter. J4 is jack supply power to the board via regulator which is buck type regulator(LM7805CTB) and gives 5V output. The processor use frame synchronization signals to tell the codec to send and receive data. To transmit data to the codec, it sends a RFS0 pulse to the SDFS input of the codec and then outputs the eight bits on DT0 on the serial clock periods. The codec receives the data on its SDI input. Likewise the processor initiates a data receive operation by sending an TFS) pulse to the codec’s FSX input, which causes the codec to output eight bits on its SDO output on the next eight serial clock periods. The processor receives the data on its DRO input. The ADSP-21xx must be programmed to use normal framing, 8-bit data words and internal, active-high frame sync generation.

V CONCLUSION

We can process the digital sample data value through a different type of filter such as FIR or IIR realize by DSP. Different type of FIR and IIR filters are low pass filter, high pass filter, band pass filter and band reject filter.

REFERENCES

[1] Proakis and Manolakis , Digital Signal Processing[2] Sanjit K. Mitra, Digital Signal Processing

13-14 May 2011 B.V.M. Engineering College, V.V.Nagar,Gujarat,India

National Conference on Recent Trends in Engineering & Technology