We present a novel power-efficient wireless sensor network for continuously monitoring and analyzing seismic vibrations with sensor nodes and forwarding the retrieved information with low-cost relay nodes to backend applications. The applied vibration sensing algorithms are derived from the DIN 4150-3 standard. All nodes in the network are battery-powered and equipped with an IEEE 802.15.4 compatible radio transceiver. The nodes communicate with each other by executing a novel power-efficient protocol stack, which provides all network functions required by the vibration-sensing application and uses a publish/subscribe messaging protocol for communicating between the network nodes and the backend applications. Results obtained in certification and field tests show that the proposed vibration-sensing solution is standard-compliant, and that the wireless vibration sensor network (WVSN) exhibits excellent performance in terms of packet delivery rate, latency, and power efficiency.

We present a simple but powerful dynamic link adaptation mechanism for wireless LANs, which are based on the IEEE 802.11a standard. Using this method, a transmitter is able to detect whether the quality of the link is improving or deteriorating, and based on this information to switch to a higher or lower transmission rate, respectively. For determining the link quality, the transmitter employs only information that is available locally. Therefore, the proposed method can be implemented without changes or enhancements to the current IEEE 802.11 standard. The efficiency of our method is investigated and evaluated by means of simulations.

The wireless communication landscape will continue to develop at a rapid pace over the next years. New systems and standards are on the horizon, which enable broadband wireless communication in the office, at home, and "on the move". They will finally bring the information services provided by the Internet and the World-Wide-Web also to the mobile users, together with a variety of multimedia entertainment services. In this paper we give an overview of key developments in wireless communication systems. The emerging 3rd generation cellular systems and the growing importance of wireless local-area and personal-area networks are discussed. The physical and medium-access control (MAC) layer of the new broadband wireless LANs which enable "hot-spot" data and multimedia services in airports, hotel lobbies, and a variety of other "public" places are presented. We give an outlook towards a future 4th-generation wireless communication system, which can be integrated with the IP backbone network and provides Quality of Service (QoS) support for multimedia applications. It supports dynamic scheduling, link adaptation, and frequency selection in addition to full mobility support. Finally, we illuminate some of the underlying technology developments driving the wireless evolution.

A novel receiver for data-transmission systems using trellis-coded modulation is investigated. It comprises a whitened-matched filter and a trellis decoder which combines the previously separated functions of equalization and trellis-coded modulation (TCM) decoding. TCM encoder, transmission channel, and whitened-matched filter are modeled by a single finite-state machine with combined intersymbol interference and code states. Using ISI-state truncation techniques and the set-partitioning principles inherent in TCM, a systematic method is then developed for reducing the state complexity of the corresponding ISI and code trellis. A modified branch metric is used for canceling those ISI terms which are not represented by the trellis states. The approach leads to a family of Viterbi decoders which offer a tradeoff between decoding complexity and performance. An adaptive version of the proposed receiver is discussed, and an efficient structure for reduced-state decoding is given. Simulation results are presented for channels with severe amplitude and phase distortion. It is shown that the proposed receiver achieves a significant gain in noise margin over a conventional receiver which uses separate linear equalization and TCM decoding.>

Fast initial training of a voiceband data-modem receiver which employs an equalizer with fractionl-Tspaced coefficients is presented. The modem receiver performs rapid gain acquisition, and monitors the equalizer delay line for the presence of a periodic training signal whose period is equal to the length of the equalizer delay line. Testing for periodicity is combined with estimating carrier-frequency offset. Once a full period of the training signal has been obtained, the equalizer coefficients are computed by spectral division, and coefficient centering is performed. The receiver then proceeds in a reference-directed mode until the end of the training signal is detected. In the paper, algorithms for rapid gain acquisition and the detection of a cyclic training signal in the presence of noise and carrier-frequency offset are presented. Cyclic training of an equalizer with fractional-Tspaced coefficients is derived. It is shown that the equalizer coefficients can be computed efficiently by spectral division. Coefficient centering, carrier-phase and symbol synchronization, reference-directed training, and the transition to data mode are described. Finally, measurement results for an experimental 2400 baud modem using an equalizer with 64T/2-spaced coefficients are given. The modem achieves a startup time of 20 ms on a Bell 3002 unconditioned leased line.

The SP16 processor executes typical signal-processing tasks efficiently by overlapping in each processor cycle four functions: instruction prefetching, data-memory access and transfer of one data word over a single data bus, ALU operation, and multiplication. Despite SP16 being a one-address machine with a single data bus, it can perform filter operations with a multiplication and ALU operation in every processor cycle.

The paper deals with the design of digital transmitter and receiver filters with finite impulse response (FIR) for data transmission over band-limited channels. The filters are matched and satisfy a zero intersymbol interference constraint when cascaded. For baseband transmission, the filters achieve optimum spectral concentration in the frequency range[-(1+\beta)/2T, (1+\beta)/2T]. Mathematically, the filter design leads to a generalized eigenvalue problem which is solved numerically by a projected gradient Procedure. For transmission over bandpass channels by combined amplitude and phase modulation, the design technique is modified so that filters with complex-valued impulse response and optimum spectral concentration in the range of positive bandpass frequencies[f_{c} - (1+\beta)/2T, f_{c} + (1+\beta)/2T]are obtained. In addition, the complex formulation allows the design of impulse responses with enhanced spectral attenuation in the corresponding range of negative frequencies in order to minimize imageband interference. Results are shown in terms of filter coefficients, signal spectra, and spectral concentrations obtained. For example, filters designed for a voiceband data modem operating at a symbol rate of 2400 baud achieve a spectral concentration of 98.5 percent with 24 coefficients and\beta = 0.1, and with only 0.001 percent of the total energy in the imageband region.

Trellis coding is investigated for a class ofN-user multiple-access channels. It is shown thatN-user trellis coding, in conjunction with Viterbi or sequential decoding, permits the multiple-access function to be combined with forward error correction. The distance and rate properties are analyzed for linear (convolutional)N-user trellis codes, and the use of nonlinearN-user trellis coding is discussed. A procedure is given for constructingN-user convolutional-codeN-tuples with large free distance. Finally the error performance is analyzed, and coding gains of several decibels over uncoded time sharing are verified by simulation results.

Number Theoretic Transforms (NTT's), unlike the Discrete Fourier Transform (DFT), are defined in finite rings and fields rather than in the field of complex numbers. Some NTT's have a transform structure like the Fast Fourier Transform (FFT) and can be used for fast digital signal processing. The computational effort and the signal-to-noise ratio (SNR) performance of linear filtering in finite rings and fields are investigated. In particular, the effect of limited word lengths, i.e.,b \leq 16, and long transform lengths on the SNR is analyzed. It is shown that for small word lengths and/or moderate to large transform lengths NTT filtering achieves a better SNR than FFT filtering with fixed-point arithmetic. Some new NTT's with a single- or mixed-radix fast transform structure are presented. While these NTT's may require special modulo arithmetic they achieve optimum transform length for any given word length b in the range8 \leq b \leq 16.

A new analysis of the computational effort and the error probability of sequential decoding is presented, which is based entirely on the distance properties of a particular convolutional code and employs no random-coding arguments. An upper bound on the computational distributionP(C_{t}>N_{t})for a specific time-invariant code is derived, which decreases exponentially with the column distance of the code. It is proved that rapid column-distance growth minimizes the decoding effort and therefore also the probability of decoding failure or erasure. In an analogous way, the undetected error probability of sequential decoding with a particular fixed code is proved to decrease exponentially with the free distance and to increase linearly with the number of minimum free-weight codewords. This analysis proves that code construction for sequential decoding should maximize column-distance growth and free distance in order to guarantee fast decoding, a minimum erasure probability, and a low undetected error probability.

A new algorithm for erasurefree sequential decoding of convolutional codes is introduced which achieves low error probabilities at substantially higher decoding speeds than the Viterbi decoding algorithm. The algorithmic properties of the Multiple Stack Algorithm (MSA) are investigated and it is demonstrated that the MSA reaches a decision with an exponentially rather than Pareto distributed computational effort. The MSA's error probability on the binary symmetric channel is studied as a function of its parameters and its performance and complexity compared to that of the Viterbi algorithm. The MSA is seen to achieve equal and lower error probabilities with a significantly lower average decoding effort. The new algorithm can thus be considered an attractive alternative to the Viterbi algorithm where low error probabilities and high decoding speeds are required simultaneously.

The relationship between the column distance function and the computational effort of sequential decoding is studied and the results of computer simulations are reported. A table ofR = 1/2codes having good free distance and optimum average column distance function (CDF) is presented.