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Presented by : Radhika Ravikrindi(1V B-TECH, ECE) Shanti Nalluri (IV B-TECH, ECE ) Contact us at:
radhi_487@yahoo.co.in Shantinalluri@yahoo.com
ABSTRACT
Digital signal is a numerical representation of the analog signal. It may be easier and more cost effective to process these signals in the digital world. One of the main advantages of digital signal is that it can be stored and manipulated easily. To do this several processes are in use. Of these, Digital Signal Processor plays a leading role.This is used in various applications like instrumentation & measurement, communications , audio & video processing ,graphics , image enhancement , navigation , radars ,control-robotics, machine vision ,guidance etc.
The objective is to encrypt the speech signal. That is to convert the speech signal into an unknown form and then the scrambled speech signal is transmitted. The person who know, how the speech signal is
encrypted can decrypt (i.e.,converting the scrambled speech signal into original signal) the signal. Thus security of sending speech message is done.
For encryption and decryption of the signal, we develop a software program in assembly language programming techniques of ADSP 2181. In this paper, encryption and decryption of the speech signal is done using the Digital Signal Processor ADSP 2181.
CONTENTS
1. INTRODUCTION 2. DIGITAL SIGNAL PROCESSING 3. DIGITAL SIGNAL PROCESSOR 4. ADVANTAGES 5. ENCRYPTION & DECRYPTION 6. CONCLUSION 7. REFERENCES
INTRODUCTION
In recent trends, cell phone, voice chat, etc., is the main topic of the communication. But it is not possible to speak secretly through them. Our objective is to speak secretly, i.e., to maintain security of certain important information. So for that we are going for a technology using digital signal processor. Our paper also deals with storing and retrieval of speech signal using digital signal processor. We have used ADSP processor 2181. This is 16 bit fixed point processor. Here the voice signals is acquired and then digitally coded and stored in the memory of processor. Using FFT (Fast Fourier Transform) and IFFT (Inverse Fast Fourier Transform) techniques, speech signal is scrambled. Then it is decoded to obtain the same speech signal and now the original speech is confused. Then it is transmitted using transmitter. The coding and decoding is done with CODEC. In the receiver side, if the receiver knows how the speech signal is scrambled, then the receiver would descramble the scrambled signal and get the original speech signal. The programming is done on the assembly language of the ADSP 2181 processor
The basic explanation of DSP is the processing of signal by digital systems, to improve the signal. The improvement may be clearer sound, sharper images, or faster data. The digital systems are software and hardware. In digital processing of signal consists of a number of mathematical operations as specified by a software program.
CLOCK 24.576Mhz
ADSP-2181
CODEC
MIC speaker
OFFLINE PROCESSING: In this processing, the entire input signal resides in the computer at the time of manipulation, for example, in medical imaging such as tomography, the data is acquired while the patient is inside the machine, but the image reconstruction may be delayed until a later time. The key point is all the information is simultaneously available to the processing program. REAL TIME PROCESSING: Signal is produced at the same time that the input signal is being acquired. This real time capability makes the DSP perfect for applications that cannot tolerate any delays. For example, did you ever talk on a cell phone where two people couldnt talk at once? You had to wait until the other person finished talking. If you both spoke simultaneously, the signal will be cut. You couldnt hear the other person. With todays digital cell phones, which use DSP, you can talk normally. The DSP processors inside cell phones process sounds so rapidly you hear them as quickly as you can speak in real time.
CIRCULAR BUFFER :
To calculate the output sample, we must have access to a certain number of the most recent samples from the input. Let a0,a1,,a7 be the value of the 8 most recent samples from the input signal. These 8 samples must be stored in memory and x[n],x[n-1],,x[n-7] continually updated as new samples are acquired. The best way to manage the stored samples is circular buffering.
MEMORY STORED ADDRESS VALUES 20041 20042 20043 20044 20045 20046 20047 Circular buffers are used to store the most recent values of a continually updated signal. 20048 0.225768 0.210768 0.105768 0.225108 0.225768 0.235768 0.224568 0.345768
The fig shows how an eight sample circular buffer might appear at some instant of time. Figure illustrates an eight sample circular buffer. Consider that we have placed this circular buffer in eight consecutive memory locations, 20041 to 20048. Fig shows how the eight samples from the input might be stored at one particular instant of time. The idea of circular buffering is that the end of this linear array is connected to its beginning; memory location 20041 is viewed as being next to 20048, just as 20044 is next to 20045. You keep track of the array by a pointer (a variable whose value is an address) that indicates where the most recent sample resides. For instance in the fig the pointer contains the address 20044, while it contains 20045 after the arrival of next sample. When a new sample is acquired, it replaces the oldest sample in the array, and the pointer is moved one address ahead.
Circular buffers are efficient because only one value needs to be changed when a new sample is acquired. Four parameters are needed to manage the circular buffer. There must be a pointer that indicates the start of the circular buffer in memory (in this example, 20041). There must be a pointer indicating the end of the array (e.g. 20048) or a variable that holds its length (e.g. 8). The step size of the memory addressing must be one. Example: address 20043 contains one sample; address 20044 contains the next sample and so on. The above three values define the size and configuration of the circular buffer, and will not change during the program operation. The pointer to the most recent sample must be modified as each new sample is acquired. In other words, there must be program logic that controls how this fourth value is updated based on the value of the first three values. While this logic is quite simple, it must be very fast. This is the whole point of this discussion; DSPs must be optimized at managing circular buffers to achieve the highest possible execution speed.
SPEECH ENCRYPTION :
Speech signal
scrambled signal
ADC
FFT
SCRAMBLING
IFFT
DAC
ADC
FFT
DESCRAMBLING
IFFT
DAC
decimation can be done in two ways namely Decimation In Time (DIT) and Decimation In Frequency (DIF). Here we use Radix-2 DIT FFT algorithm.
GRAPHICAL REPRESENTATION:
SPEECH SIGNAL SCRAMBLED SIGNAL
a bcd
pqrs
pqrs
abcd
FLOW CHART:
START
Return
FFT
IFFT
Output to CODEC
Return
ALGORITHM :
ENCRYPTION : 1) First the speech signal (analog signal) is converted into digital signal by passing through the CODEC. Now the signal is in time domain and it is sent to the ADSP 2181 processor.
2) By running 128-point FFT algorithm using computer (system is connected to the processor) on ADSP 2181, the signal which is in time domain is converted into frequency domain. Now the signal is sampled into 128 samples and it is stored in the processor memory. 3) By doing assembly language program on ADSP 2181, first 64 samples are displaced into next 64 samples position and vice versa. This is called scrambling. 4) Now the original signal is scrambled and by doing IFFT algorithm on ADSP 2181, the scrambled signal, which is in frequency domain, is converted into time domain. 5)By CODEC scrambled signal which is a digital signal is converted into analog signal and then it is transmitted.
DECRYPTION : Received signal (scrambled signal) is converted into digital signal by CODEC and it is sent to the processor. 1) By 128-point FFT algorithm, the signal in time domain is converted into frequency domain as done in the encryption part. 2) By doing assembly language program on ADSP 2181, the displaced 64 samples are placed to its original position. This is called descrambling. 3) Now the scrambled signal is descrambled and we get the original signal. 4) By doing IFFT, the original signal, which is in frequency domain, is converted into time domain and by CODEC we get original analog signal (speech signal). The assembly language program was executed for the speech signal and the output was checked. Encryption and Decryption is highly realized. We have stored the speech signal in the memory in encoded form. We decode the same to get the analog output. For coding and decoding, CODEC is used.
CONCLUSION :
We can use this technique in military for sending secret messages. We can use this technique wherever we want security. What we have done is storing the speech signal for a period of few milli seconds. This can be increased if we extend the memory size and select that through a decoding switch.
REFERENCES :
Sanjit K. Mitra, Digital signal processing Alan V.Oppenheiim and Allan Willsky, Signals and Systems P.Ramesbabu , Digital signal processing ADSP-218X DSP Instruction Set Reference WEBSITES: www.analogdevice.com www.DSPguide.com