2 research outputs found

    Design and Hardware Implementation of a Speech Cipher System

    Get PDF
    Digital ciphering of speech signals based on one of modern cryptography algorithms, called the Rijndael algorithm, is studied and presented in this paper. The algorithm meets most of the requirements of security level in recent applications. A system to encrypt speech files recorded with Sound Blaster Card of a personal computer was proposed and simulated successfully using MATLAB® language.  Subjective measure and objective measure using segmental spectral signal-to-noise ratio, were used to test the proposed system performance. In these tests residual intelligibility of the encrypted speech and quality of the recovered speech were calculated and assessed. Finally, a hardware implementation of the above cipher system has been proposed using the TMS320-C30. The real time requirements from the speech cipher system have been computed in terms of execution time together with factors affecting such implementation. The results show the capability of the cipher system to be implemented using the DSP device suggested. Furthermore, the results of hardware implementation also show the security of the system is very close to that of the simulated version

    Adaptive Discrete Filters for Telephone Channels Based on the Wavelet Packet Transform

    Get PDF
    The wavelet transform provides good and in many times excellent results when used as a basic block transform in many systems such as electronic, communication, medical and even chemical systems. The paper uses the wavelet packet transform to adjust the tap gains of the adaptive filter used in channel equalization and estimation. The results using the wavelet technique achieve good improvements in convergence time over the ordinary LMS algorithm. The two systems were compared on full mathematical and simulation basis. Learning curves for adaptive channel equalization and adaptive channel estimation using wavelet packet transform with different mother functions, different level decompositions, different step sizes, different levels of signal to noise ratio, different telephone channels and different filter sizes were compared with conventional LMS adaptive channel equalization and channel estimation. The simulation results carried out using the MATLAB package version 6.1, demonstrate the efficiency of the proposed technique
    corecore