Digital signal processing, Multirate signal processing and wavlet filter banks

This paper presents the low complexity design of a non-uniformly spaced digital finite impulse response(FIR) filter bank for digital hearing aid application. Frequency response masking (FRM) technique is used for the implementation of 8 non-uniformly spaced subband filters. FRM technique ensure the drastic reduction in the number of mul-tipliers and adders in linear phase FIR filter. It employs two half-band filters as prototype filters, which leads to significant savings in terms of arithmetic operations. Further reduction can be achieved by using a masking filter with minimum order. The simulation result shows that the proposed filter bank with stop band attenuation of 100 dB can be achieved with 10 multipliers. The audiogram fitting with the selected audiogram gives a maximum matching error of 2.25 dB. From the audiogram matching result, it can be seen that the introduction of non-uniformly spaced filter bank improves the audiogram matching especially for sharp transition in hearing losses at low and high frequencies.

In Pakistan criminals are using different radio systems to convey their hatred messages. It is necessary to stop the propagation of their propaganda by using low cost jamming solutions. For security purposes, various frequency jamming techniques were supposed to be implemented to disturb the radio communication, cell phone, radar and wireless transmission signals. In this research work, these techniques are discussed and low cost solutions are provided to jam the illegal amplitude modulation (AM) and Frequency modulation (FM) based radio systems in laboratory environment. These circuits were implemented with the help of discrete components and integrated circuit (ICS). The power of the purposed system is less than commercially available jammers. The Electronic Workbench (EWB) software and MS Visio were used during the designing process

Development of modern integrated circuit
technologies makes it feasible to develop cheaper, faster
and smaller special purpose signal processing function
circuits. Digital Signal processing functions are generally
implemented either on ASICs with inflexibility, or on
FPGAs with bottlenecks of relatively smaller utilization
factor or lower speed compared to ASIC. Field
Programmable DSP Array (FPDA) is the proposed DSP
dedicated device, redolent to FPGA, but with basic fixed
common modules (CMs) (like adders, subtractors,
multipliers, scaling units, shifters) instead of CLBs. This
paper introduces the development of reconfigurable system
architecture with a focus on FPDA that integrates different
DSP functions like DFT, FFT, DCT, FIR, IIR, and DWT
etc. The switching between DSP functions is occurred by
reconfiguring the interconnection between CMs.
Validation of the proposed architecture has been achieved
on Virtex5 FPGA. The architecture provides sufficient
amount of flexibility, parallelism and scalability.

The information security has become one of the most significant problems in data communication. So it becomes an inseparable part of data communication. In order to address this problem, cryptography and steganography can be combined. This paper proposes a secure communication system. It employs cryptographic algorithm together with steganography. The jointing of these techniques provides a robust and strong communication system that able to withstand against attackers. In this paper, the filter bank cipher is used to encrypt the secret text message, it provide high level of security, scalability and speed. After that, a discrete wavelet transforms (DWT) based steganography is employed to hide the encrypted message in the cover image by modifying the wavelet coefficients. The performance of the proposed system is evaluated using peak signal to noise ratio (PSNR) and histogram analysis. The simulation results show that, the proposed system provides high level of security.

The front propagating problems is an important area of application of Level Set Methods (LSMs) where moving front plays a vital role in evolution of process. In traditional LSMs, the Level Set Function (LSF) defines the evolution of boundary. But it does not apply to the object whose boundary we've to calculate after some time interval i.e. the LSF becomes 'irregular'. As a remedy to this, we've to redefine the LSF so that it'll give the explicit result. The costly re-initialization is a method to do this. But this causes area loss which is one directional. Many ideas were proposed to avoid re-initialization process like Chopp's methods, GDRLSE1, GDRLSE2, and GDRLSE3. But they are not applicable for both the LSMs i.e. PDE based and variational one. We are proposing reaction diffusion (RD) based method which will avoid all the disadvantages of all methods mentioned above, will be stable. There is no need to go for re-initialization of LSF. We'll add a diffusion term into Level Set Evolution (LSE) equation. To have a stable numerical solution of RD based LSE, we'll use two step splitting method (TSSM) which will first iterate the LSE equation and then solve diffusion equation. Solving diffusion equation regularizes the LSF to ensure stability, thus avoiding costly re-initialization completely. The method is applicable for PDE based LSM as well as variational LSM. It gives very good results in terms of high boundary anti-leakage and noise immunity.

The present paper discusses wideband
monitoring and application of polyphase digital filter banks
to dealing with practical tasks in this field. Various ways of
classifying digital filter banks are provided including
uniform DFT-modulated filter banks (direct and polyphase
forms) and we also compare these banks by their
performance. A particular polyphase filter bank with given
characteristics is investigated in the time and frequency
domains. Experimental research illustrates influence of
white Gaussian noise on the quality of signal detection in
filter bank channels. Experiments have been carried out on
the basis of computer simulation in MATLAB and signal
generation with the help of functions from standard
MATLAB toolboxes. Experimental results contain plots of
error of the first and second kinds versus signal-to-noise
ratio at the input of a filter bank. Issues of polyphase filter
bank implementation using CPU and CUDA are also
considered in the paper.

Gen Y places a greater emphasis on mobility and online access. Their expectation must be embraced by the financial services sector, as these groups of future " mass affluent " are the customers to attract, and retain as they travel through life. Banks should adequately understand the demands of the customers and develop products and services that will suit the needs of all ages. The high expectations of these tech-savvy generations must be met by offering technologically advanced self-service features. This paper focuses on the need for Banks to concentrate on achieving customer primacy through digital engagement especially with respect to Gen Y customers. The paper has used a DBAM (Digital Banking Acceptance Model) to identify the key areas which banks should centre their Dig banking efforts.

The sensitivity of human ear is dependent on frequency which is nonlinearly resolved across the audio spectrum .Now to improve the recognition performance in a similar non linear approach requires a front - end design, suggested by empirical evidences. A popular alternative to linear prediction based analysis is therefore filter bank analysis since this provides a much more straightforward route to obtain the desired non-linear frequency resolution. MEL filter bank and BARK filter bank are two popular filter bank analysis
techniques. This paper presents FPGA based implementation of MEL filter bank and BARK filter bank with different bandwidths and different signal spectrum ranges. The designs have been implemented using VHDL, simulated and verified using Xilinx 11.1.For each filter bank, the basic building block is implemented in Spartan 3E. A comparative study among these two mentioned filter banks is also done in
this paper.

This paper presents a new compression technique and image watermarking algorithm based on Contourlet Transform (CT). For image compression, an energy based quantization is used. Scalar quantization is explored for image watermarking. Double filter bank structure is used in CT. The Laplacian Pyramid (LP) is used to capture the point discontinuities, and then followed by a Directional Filter Bank (DFB) to link point
discontinuities. The coefficients of down sampled low pass version of LP decomposed image are re-ordered in a
pre-determined manner and prediction algorithm is used to reduce entropy (bits/pixel). In addition, the coefficients of CT are quantized based on the energy in the particular band. The superiority of proposed algorithm to JPEG is observed in terms of reduced blocking artifacts. The results are also compared with wavelet transform (WT). Superiority of CT to WT is observed when the image contains more contours. The
watermark image is embedded in the low pass image of contourlet decomposition. The watermark can be
extracted with minimum error. In terms of PSNR, the visual quality of the watermarked image is exceptional.
The proposed algorithm is robust to many image attacks and suitable for copyright protection applications.

— This paper presents the low complexity design of a non-uniformly spaced digital finite impulse response (FIR) filter bank for digital hearing aid application. Frequency response masking (FRM) technique is used for the implementation of 12 non-uniformly spaced subband filters. FRM technique ensure the drastic reduction in the number of multipliers and adders in linear phase FIR filter. It employs two half-band filters as prototype filters, which leads to significant savings in terms of arithmetic operations. Further reduction can be achieved by using a masking filter with minimum order. The simulation result shows that the proposed filter bank with stop band attenuation of 80 dB can be achieved with 10 multipliers. The audiogram fitting with the selected audiogram gives a maximum matching error of 1.4 dB. From the audiogram matching result, it can be seen that the introduction of non-uniformly spaced filter bank improves the audiogram matching especially for sharp transition in hearing losses at low and high frequencies

This paper presents a novel method for speech recognition system which then uses the mel frequency cepstrum coefficients (MFCC) and also with the vector quantization (VQ) for processing. The system consists of two phases: the training phase where a database of known speakers are stored, and the testing phase to recognize and identify speech. The MFCC processing is used for feature extraction and the VQ is used for feature matching for the speech recognition system of this paper. Simulation studies have been performed and carried out for speech recognition using MATLAB equipped with DSP based MFCC and VQ to verify the effectiveness of the proposed scheme. Simulation results have shown that the proposed method is remarkably effective for speech recognition. The proposed scheme is less complex, easy to realize, feasible, and accurate.