Department Publications

Real-time scanned-array direct-form-I hardware implementations of two-dimensional (2D) infinite impulse response (IIR) frequency-planar beam plane-wave (PW) filters have potentially wide applications in the directional enhancement of spatio-temporal broadband PWs based on their directions of arrival (DOAs). The proposed prototypes consist of a microphone sensor array, low-noise-amplifiers (LNAs), multiplexers (MUXs), a programmable gain amplifier (PGA), an analog to digital converter (ADC), a digital to analog converter (DAC), and a field programmable gate array (FPGA) circuit based 2D IIR spatio-temporal beam filter implemented on a single Xilinx Virtex2P xc2vp30-7ff896 FPGA chip. Starting from published 1 st -order designs, novel FPGA architectures for highly-selective 2 nd - and 3 rd -order beam PW filters are proposed, simulated, implemented on FPGA, and verified on-chip.

• Digital image processing helps us enhance images to make them visually pleasing, or accentuate regions or features of an image to better represent the content.

Beamforming, in brief, is a signal processing technique which has potential applications in the areas such as sonar, ultrasound, seismic, microwave imaging, radio astronomy and communications. In this report we are presenting a prototype that filters audio signals of interest, based on their direction of arrival (DOA). The prototype described in this report consists of both hardware and software. A microphone sensor array, low-noise-amplifiers, multiplexer, adjustable gain controller, FPGA development board and speakers constitute the hardware. FPGA processor is embedded with a 2D IIR beam filter. The hardware take the audio inputs, multiplex them, digitize them and finally fed them in to FPGA processor. The digital filter, filters the asynchronously sampled broadband spatio-temporal plane-wave signals based on DOA and yields a processed signal as the output. Further the report describes the synthesis of 2D IIR …

We saw that a given image may have a peaked histogram. Such a peaked or non-flat histogram shows that pixels do not equally occupy all the gray levels. This results in a low contrast image. We can remedy this by carrying out the operation called histogram equalization. Histogram equalization is a gray-level transformation that results in an image with a more or less flat histogram. Consider, for now, that continuous intensity values of an image are to be processed. We assume that r∈[0, L− 1]. Lets consider the intensity transformation s= T (r) 0≤ r≤ L− 1(3) that produces an output intensity level s for every pixel in the input image having intensity r. We assume that

Let Z be a set of integers. The sampling process used to produce the digital image may be viewed as a partitioning in the xy-plane in to a grid, with coordinates of the center of each grid being a pair of elements from the Cartesian product Z2. In the terminology of set theory the function f (x, y) is said to be a digital image if (x, y) are integers from Z2 and f is a mapping that assigns an intensity value in the set of real numbers, R, to each distinct pair of coordinates (x, y). If the elements of R also are integers, a digital image then becomes a two-dimensional function, whose coordinates and the amplitude values are integers. Let A be a set in Z2, the elements of which are pixel coordinates (x, y). If w=(x, y) is an element of A, then we write w∈ A.

Bandwidth scarcity is a common problem faced in multimedia transmission over broadband wireless networks. From a consumer's perspective, the effectiveness of multimedia delivery suffers significantly in such networks. Further, in a mobile environment, when the number of users increases, they experience interruptions in the received multimedia streams. Adaptive streaming is one solution that can be adopted in any network environment regardless of the nature of the network. Adaptive streaming can be achieved using transcoding techniques. Transcoding is a mechanism that converts a video to a form that has less information so that the resulting data volume is appropriate for streaming over a low bandwidth network. However, the quality of the video stream drops due to transcoding. The advantage is that when there is a large number of users sharing the network, users could watch a video at a lower quality …

The development and expansion of broadband technologies has stimulated the multimedia streaming market for the delivery of live streaming services which can be used for a variety of applications. The design of a generic audio and video streaming platform that can be employed for a wide range of applications is presented in this paper. A key feature of the platform is the ‘subscription framework’ which enables the service users to subscribe to a set of live feeds with their application specific requirements. The platform is capable of adjusting its service functionalities and the extent to which the media streams are secured for a particular user based on the requirements given at the service subscription. The platform is designed based on a layered architecture and a prototype implementation is carried out within the University network.

The advent of low-cost, photo-quality inkjet printers has raised the need for an objective means of determining print quality that is consistent with what the end-user perceives. High-level quality metrics have been specified in the International Association for Standardization/International Electrotechnical Commission (ISO/IEC) guidelines on hardcopy print assessment [1]. These guidelines include metrics for four distinct categories of printed areas: line and character metrics, solid fill metrics, tint solid metrics, and background field metrics. Other metrics that enable the quantification of performance aspects relevant to inkjet printers have also been proposed. These aspects include color registration, color consistency, modulation transfer function (MTF), text quality, sharpness [2], dot quality, and line quality [3, 4]. Multiple efforts have been made to automate the process of image quality assessment both during product …

The performance of nonlinear adaptive filters in modeling devices with mild and moderately strong nonlinearities such as hearing aids was investigated. The existence on nonlinearities in hearing aids and that nonlinear adaptive filters were better suited for hearing aid modeling compared to their linear counterparts with equal number of filter coefficients. The signal to noise ratio improvement in modeling showed improvements in the range of 5% to 50%. The results suggested that as the nonlinearity of the device becomes stronger, a larger number of nonlinear coefficients should be used.

5.1 Discussion: One of the most cumbersome and difficult problems associated with obtaining solutions for electromagnetic integral equations has been the manipulation of large matrices resulting from numerical methods such as MoM. Unfortunately, such situations are unavoidable in practical problems where the analysis (such as electro-magnetic wave scattering or radiation characteristics) has to be made in respect of objects of complex shapes and sizes. As mentioned in the introductory chapter, several techniques have been developed over the years in order to address this problem. Objective of this research work has been, to investigate the possibility of addressing the above-mentioned problem through wavelet transform techniques that started gaining popularity among the research community over a decade ago. Though there is evidence that wavelet transform techniques have been invented in early 20t h …

The paper describes several digital signal processing (DSP) techniques used in the development of a computer-based speech trainer for hearing impaired children. Children with congenital hearing impairments have difficulties in speaking, and even in making the basic sounds associated with speech. Speech therapists use specialized training methods to train such children. The dearth of qualified speech therapists, and other facilities hinder the speech development of many children in need of such training in Sri Lanka. The speech trainer described in this paper was developed as an alleviation to the above problem. The training tool developed, will aid a child with initial guidance from an adult, to master the pronunciation of initial sounds taught in a speech therapy programme, in a game-like environment. In its current status, the training tool can guide children in pronouncing the five vowel sounds, the first step in …

The paper describes the development of a speech trainer based on digital signal processing (DSP) techniques, for hearing impaired children. The DSP techniques investigated in the implementation of this software-based speech trainer, the final" product'and its degree of success in actual use are highlighted. Children with congenital hearing impairments have difficulties in speaking, and even in making the basic sounds associated with speech. Speech therapists use specialized training methods to train such children. The first step in such programs is the training of the pronunciation of common sounds. The dearth of qualified speech therapists, and other facilities hinder the speech development of many children in need of such training.

The trainer provides facilities for two modes of signal processing: real time and non-real time. The microphone picks up the input signal which is amplified, converted to digital in the analog front end, and fed to the microcontroller. The analog front end also conditions the signal to be analysed as required.

An array processor design methodology suitable for hard real-time systems is presented. Scheduling and projection of the dependence graph (DG) are solved using integer programming. By exploring the regularity of the DG it is possible to solve the necessary IP problems in an efficient manner. This methodology provides a unified approach for linear and nonlinear projection of regular and partially regular DGs.<>

Frequency Domain Adaptive Filters (FDAFs) are an important subject of current research in digital signal processing. One of the main reasons is that FDAFs have the ability to reduce complexity, especially when the filter length becomes very large. In Adaptive Filter (AF) configurations a convolution and a correlation have to be carried out. These operations can be done by using overlap methods in combination with Discrete Fourier Transforms (DFTs). Two well-known overlap methods are the so-called overlap-save and the overlap-add method. In literature 111 it is stated that for AF configurations the overlap-save method is to be preferred to the overlap-add method for complexity reasons. By introducing a new FDAF structure we will show that this statement is not correct and that an FDAF using the overlap-add method can be realized with the same number of DFTs as the FDAF using the overlap-save method