In this paper, we propose a new family of wireless networks-the multimodal wireless networks. These networks offer multiple functionalities realized on the same infrastructure. A multimodal wireless network has two modes of operation: 1) the communication mode, when the network is used as a traditional wireless communication network, and 2) the surveillance mode, when the network is used as a distributed sensor network that can detect illegal intrusion. The surveillance functionality is realized by analyzing the properties of the received signals, and the change of the propagation environment caused by the intruder serves as the basis for intrusion detection. We derive maximum likelihood estimators and detectors based on the generalized likelihood ratio test that detects changes in the propagation environment: the single-cycle detector, which makes decisions at the end of each scanning cycle, and the multicycle detector, which combines information from multiple scanning cycles prior to detection. We also analyze the performance of these detectors by deriving the Cramer-Rao lower bound on the variance of the parameter estimators and determining the distribution of the log-likelihood ratio under both detection hypotheses. This will allow us to compare the theoretical performance of the single-cycle and the multicycle detectors and obtain analytical results to determine the decision threshold for a given probability of false alarm. To prevent performance degradation due to slowly drifting parameter values, we propose different strategies to track and update the parameter estimates. The experimental results obtained from an implemented prototype surveillance system show very promising detection capabilities. For example, the state of a door (open or closed) could be detected with a probability of 0.99 at a probability of false alarm 10-5.

Multimodal wireless networks are wireless networks that offer multiple functionalities realized on the same infrastructure. In this paper, we consider a multimodal network that has two modes of operation: the communication mode, when the network is used as a traditional wireless communication network, and the surveillance mode, when the network is used as a distributed sensor network that can detect illegal intrusion by detecting changes in the propagation environment caused by the intruder. We address the problem of distributed surveillance in a network consisting of multiple nodes: we develop a multi-sensor model for the received signal parameters of interest and derive a detector capable of detecting changes in these parameters. The experimental results demonstrate that combining the information from different nodes with the appropriate fusion function results in considerably improved detection performance compared to that of single-node detectors.

The signal subspace approach for non-parametric speech enhancement is considered. Several algorithms have been proposed in the literature but only partly analyzed. Here, the different algorithms are compared, and the emphasis is put onto the limiting factors and practical behavior of the estimators. Experimental results show that the signal subspace approach may lead to a significant enhancement of the signal to noise ratio of the output signal.

The initial design considerations and research goals for an ATM network based virtual seminar room with five sites are presented. The basic observation behind the design of the virtual seminar room is, that besides the constant growth in available bandwidth for transmission in communication networks, many networks either already offer or are developing technologies to give quality of service (QoS) guarantees. This means that applications not only will support transmission of coded audio and video, but can be designed in a known and well-controlled network environment, which enables the applications to provide, in a broad sense, a high and reliable quality presented at the human computer interface.

A new approach to the localization of high impedance ground faults in compensated radial power distribution networks is presented. The total size of such networks is often very large and a major part of the monitoring of these is carried out manually. The increasing complexity of industrial processes and communication systems lead to demands for improved monitoring of power distribution networks so that the quality of power delivery can be kept at a controlled level. The ground fault localization method for each feeder in a network is based on the centralized frequency broadband measurement of three phase voltages and currents. The method consists of a feature extractor, based on a grid description of the feeder by impulse responses, and a neural network for ground fault localization. The emphasis of this paper is the feature extractor, and the detection of the time instance of a ground fault.

Digital telecommunication networks may involve a multiple number of public switched telephone networks (PSTN), cellular and mobile systems and to some extent also satellite systems. Most of these networks contain non-linear speech coders and other speech algorithms which may degrade the overall end-to-end quality of speech. An important problem is how to assess the speech quality of such compounded systems. The object of this paper is to describe the first stage of the construction of a proposed three-layer model for speech quality assessment. A subjective test of the speech quality of 16 different compounded transmission paths (mixtures of PCM, GSM full and half rate, DECT, CELP, LD CELP, FS10-16) is carried out by 40 subjects using 21 different rating scales. The main result of this paper is the test results which lead to the definition of four main perceptual dimensions to be used in the second layer of the proposed model.

Digital telecommunication networks involve multiple number of public switched telephone networks (PSTN), cellular and mobile systems and to some extent also satellite systems. Most of these networks contain non-linear speech coders which may degrade the overall end-to-end quality of speech. An important problem is how to assess the speech quality of such compounded systems. The object of this paper is to describe the development of a proposed three-layer model for objective speech quality assessment. The design of the objective measure is based on a subjective test of the speech quality of 16 compounded transmission paths and involves 40 subjects using 21 different rating scales. The subjective test supports the definition of four main perceptual dimensions to be used in the second layer of the proposed model. These defined perceptual elements form the basis of designing an objective measure for predicting the total quality of a given telecommunication connection.

Simultaneous design of multiplier-free recursive filters (IIR filters) and their hardware implementation in Xilinx field programmable gate array (XC4000) is presented. The hardware design methodology leads to high performance recursive filters with sampling frequencies in the interval 15-21 MHz (17 bits internal data representation). It is demonstrated that the time-area efficiency and performance of the architectures are considerably above any known approach.

Simultaneous design of multiplier-free filters and their hardware implementation in Xilinx field programmable gate array (XC4000) is presented. The filter synthesis method is a new approach based on cascade coupling of low order sections. The complexity of the design algorithm is /spl Oscr/ (filter order). The hardware design methodology leads to high performance filters with sampling frequencies in the interval 20-50 MHz. Time-area efficiency and performance of the architectures are considerably above any known approach.

A time signal prediction algorithm based on relative neighborhood graph (RNG) localized FIR filters is defined. The RNG connects two nodes, of input space dimension D, if their lune does not contain any other node. The FIR filters associated with the nodes, are used for local approximation of the training vectors belonging to the lunes formed by the nodes. The predictor training is carried out by iteration through 3 stages: initialization of the RNG of the training signal by vector quantization, LS estimation of the FIR filters localized in the input space by RNG nodes and adaptation of the RNG nodes by equalizing the LS approximation error among the lunes formed by the nodes of the RNG. The training properties of the predictor is exemplified on a burst signal and characterized by the normalized mean square error (NMSE) and the mean valence of the RNG nodes through the adaptation.

We consider an algorithm for reduction of broadband noise in speech based on signal subspaces. The algorithm is formulated by means of the quotient singular value decomposition (QSVD). With this formulation, a prewhitening operation becomes an integral part of the algorithm. We demonstrate that this is essential in connection with updating issues in real-time recursive applications. We also illustrate by examples that we are able to achieve a satisfactory quality of the reconstructed signal.

The problem of achieving robust speech transmission on a digital mobile radio channel subjected to severe fading is addressed. The study includes a performance comparison between the Reed-Solomon and the punctured convolutional types of channel coders, which were used in conjunction with speech coders with bit rates ranging from 6.6 kb/s down to 5.4 kb/s. Speech coding within this range is of particular interest for the standardization of the GSM half-rate system, where the gross bit rate is limited to 11.4 kb/s. The properties of the speech coders and the channel coders are described, as well as results of a series of experiments comparing speech quality and robustness against the transmission errors of a typical GSM half-rate channel.>

A family of quantization-based piecewise linear filter networks is proposed. For stationary signals, a filter network from this family is a generalization of the classical Wiener filter with an input signal and a desired response. The construction of the filter network is based on quantization of the input signal x(n) into quantization classes. With each quantization class is associated a linear filter. The filtering at time n is carried out by the filter belonging to the actual quantization class of x(n) and the filters belonging to the neighbor quantization classes of x(n) (regularization). This construction leads to a three-layer filter network. The first layer consists of the quantization class filters for the input signal. The second layer carries out the regularization between neighbor quantization classes, and the third layer constitutes a decision of quantization class from where the resulting output is obtained.>

The authors describe their contribution to the ETSI (European Telecommunications Standards Institute) effort for a definition of a GSM (Group Special Mobile) standard half-rate coding system for the pan-European digital mobile cellular radio system. The system consists of a speech coder and a channel coder. An overview of the speech coding algorithm is given. The results from a study of the different versions of the speech coder are described. A two-stage channel coding scheme is presented, and the performance test results are shown. An overview of the DSP implementation, computational requirements, and memory usage is also presented.>

We report on the initial development of a phonetic vocoder operating at 100 b/s. With each phoneme, the vocoder transmits the duration and a single pitch value. The synthesizer uses a large inventory of diphone templates to synthesize a desired phoneme string. To determine a phoneme string from input speech, the analyzer takes into account the synthesis model by using the same inventory of diphone templates, augmented by additional diphone templates to account for alternate pronunciations. The phoneme string is chosen to minimize the difference between the diphone templates and the input speech according to a distance measure.