The combination of OFDM with joint pre-processing in adaptive multi-antenna systems offers both an ease of equalization in frequency-selective channels and keeping the signal processing at the mobile stations simple. In addition, the spatial dimension can be efficiently exploited to ensure high system throughput. With the utilization of higher-order modulation the performance of the system is highly sensitive to multiple access interference and nonorthogonal subchannels due to hardware impairments or insufficient adaptation to the current channel conditions. A further source of error in TDD systems are the non-reciprocal transceivers inhibiting the baseband-to-baseband channel reciprocity required for accurate channel state acquisition based on the uplink channel estimate. In this paper, measurement results of a low-cost hardware-based calibration are presented and the drawbacks are discussed leading to the utilization of a recently introduced relative calibration. The latter is applied to an OFDM system and achieves or at least approximates the baseband-to-baseband reciprocity. Thus, it enables the link adaptation using the uplink channel state information. Furthermore, preliminary hardware implementations of the relative calibration running on a real-time system show accurate results.

In this contribution various objective measures that can be used to evaluate speech dereverberation algorithms by means of listening-room compensation (LRC) are compared to subjective listening tests. It is shown that technical measures describing the impulse responses are suitable for evaluation of such algorithms. Most signal-based objective measures fail to judge the specific distortions that may be introduced by LRC algorithms like late reverberation since these artifacts are small in amplitude but perceptually relevant due to the loss of masking of the room impulse response. Only one signal-based measure, the so-called perceptual similarity measure (PSM), showed high correlation with subjective rating for the given test setup.

Distributed MIMO multi-hop relaying is one of the most promising technologies that permits cost-effective improvement of coverage, data rate and end-to-end (e2e) user experience by utilizing distributed low-complexity space-time codes to overcome path losses and deep fades of wireless channels. However, an efficient transmission scheme and resource management are required to exploit these advantages. Specifically, low-complexity adaptive schemes and power control strategies should be designed, thereby achieving robust and cost-efficient e2e communications. In this paper an adaptive transmission scheme is presented, where one relay stops forwarding the message if it is in outage and other nodes adapt to a new space-time code. For this adaptive scheme, optimal as well as sub-optimal closed-form power allocation solutions are derived which minimize the total transmission power while satisfying a given e2e outage probability. The significant power savings due to the proposed approaches in comparison to a non-adaptive scheme is demonstrated by numerical results.

Adaptive coding and modulation is an important topic considering future communication systems. Orthogonal frequency division multiplexing (ODFM) has been identified as a promising technique, which offers the possibility for further enhancements by bit and power loading schemes. Commonly, channel coding has not been considered in the optimization of such algorithms. It is, however, an important component used in nearly every communication system. In this paper we propose a new scheme to adapt code rate, modulation and transmit power by solving a convex optimization problem based on a bisection approach in order to enhance the frame error rate at a fixed target rate.

Distributed MIMO multi-hop relaying can provide cooperative diversity and overcome path losses, hence, boost the end-to-end (e2e) performance. By using a low-complexity adaptive scheme, where one relay stops sending the message if it is in outage and other nodes adapt to a new space-time code, robust communication links can be further achieved. The contribution of this paper is the derivation of near-optimal closed-form solution for joint power and time allocation for such adaptive scheme that minimizes he transmission power while satisfying a given e2e non-ergodic outage probability.

In this paper a soft relaying scheme for coded relay networks combining the benefits of classical Decode-Forward (DF) and Amplify-Forward (AF) is extended to higher order modulation schemes. In order to minimize the mean-squared- error at the destination, the conditional expectation value of the symbol after soft-output channel decoding at the relay is transmitted. The main idea of this scheme called Decode- Estimate-Forward (DEF) is to exploit coding gain like DF while still preserving reliability information as AF. This approach was presented by the authors for BPSK in coded systems, but the extension to arbitrary modulation alphabets presented here allows for more flexible system designs. The performance of the proposed relay function is compared to classical AF and DF in a wide variety of scenarios like AWGN and Rayleigh fading channels as well as different numbers and constellations of relays.

In code division multiple access (CDMA) systems a significant degradation in detection performance due to multiuser interference can be avoided by the utilization of interference cancellation methods. Further enhancement can be obtained by optimizing the power allocation of the users. The resulting constrained single-objective optimization problem is solved here by means of particle swarm optimization (PSO). It is shown that the maximum number of users for a CDMA system can be increased significantly if an optimized power profile is employed. Furthermore, an extensive study of PSO control parameter settings using three different neighborhood topologies is performed on the basis of the power allocation problem, and two constraint-handling techniques are evaluated. Results from the parameter study are compared with examinations from the literature. It is shown that thevon-Neumannneighborhood topology performs consistently better thangbestandlbest. However, strong interaction effects and conflicting recommendations for parameter settings are found that emphasize the need for adaptive approaches.

In this study a self-steering beamformer with binaural output for a head-worn microphone array is investigated in simulated and real- world conditions. The influence of the underlying sound propagation model on the estimation accuracy of the direction of arrival (DOA) estimation algorithm and the overall performance of the combined DOA-beamformer-system is evaluated. For this, technical performance measures as well as objective quality measures based on perceptual models of the auditory system are used. The self-steering beamformer showed better performance than a beamformer with fixed look-direction for SNR values above -2 dB if the propagation model includes at least a coarse head model.

Distributed MIMO multi-hop schemes can provide high data rates through spatially distributed relaying nodes. The relaying nodes allow the deployment of MIMO techniques to enhance the throughput by utilizing uncorrelated sub-channels. However, the spatial farness of geometrically separated relaying nodes results in different path losses from the nodes of one virtual antenna array (VAA) to the nodes of another VAA. In this paper we derive an approximative expression for the end-to-end (e2e) outage probability for such asymmetric networks, where orthogonal space-time block codes (OSTBC) are utilized for transmission. Based on this analytical expression a convex optimization problem that aims to reduce the total transmission power while meeting a given e2e outage level is formulated and an efficient near-optimal power allocation approach with low complexity is proposed. This near-optimum solution leads to the interesting result, that the same power is assigned to each node of one VAA. Thus, the power allocation turns out to be symmetric with respect to the nodes of one VAA also for networks with asymmetrically distributed nodes.

For text-independent speaker identification a prominent combination is to use Gaussian mixture models (GMM) for classification while relying on Mel-frequency cepstral coefficients (MFCC) as features. To take temporal information into account the time difference of features of adjacent speech frames are appended to the initial features. In this paper we investigate the applicability of spectro-temporal features obtained from Gabor-filters and present an algorithm for optimizing the possible parameters. Simulation results on a database show that spectro-temporal features achieve higher recognition rates than purely temporal features for clean speech as well as for disturbed speech.

OFDM is well suited for data-transmission over frequency-selective channels. If the cyclic prefix is chosen sufficiently long, intersymbol interference is avoided. Thus, the linear convolution becomes circular and the channel is rendered into a set of flat-fading channels, which can be easily equalized. But this holds no longer in the case of large Doppler spread, i.e., when the channel is rapidly changing. Then the subcarriers lose their orthogonality giving rise for so-called intercarrier interference which is detrimental for the overall system performance. Among previous proposals to cope with this problem was the use of multiple receive antennas. These are positioned in the moving direction of the receiver such that the individual antennas are receiving delayed replica of the transmit signal via identical paths. By means of interpolation this enables to fabric a virtual non moving point of reception which seemingly experiences no Doppler spread. So far, this technique has been applied in a purely noncoherent setting resorting to differential modulation. We will demonstrate how this approach can be exploited by coherent reception. Our emphasis thereby lies on the use of higher-level modulation as well as least-square (LS) and minimum mean square error (MMSE) channel estimation.

Interleave division multiple access (IDMA) is a promising air interface for future wireless networks. Optimization of access schemes regarding cross user aspects is also a topic of great interest for increasing efficiency. This paper introduces an optimization of the power profile for iterative parallel and successive interference cancellation (PIC/SIC) in IDMA systems. The basic approach is an optimization algorithm called differential evolution (DE). Optimized power allocation is an important means to increase the supportable load as it enhances the convergence behavior of the detector. Furthermore any additional constraint regarding the realistic modeling of communication systems can be implemented easily. In this paper different bit error rate constraints are considered and also the maximum number of iterations is taken into account. Computational complexity is reduced while the required power is barely increased. The precondition for this optimization is the analysis of the iterative detection scheme. This is done by variance transfer charts (VTC).

The performance of evolutionary algorithms is strongly dependent on the setting of control parameters. Not only the convergence speed is influenced, but also if the optimum of a function is reached at all. For differential evolution premature convergence or even stagnation can occur due to certain parameter settings. In this paper a parameter study for differential evolution is conducted. As basis for the examination a real-world problem is employed that consists of optimizing the power allocation for a CDMA (code division multiple access) system. For the CDMA system interference cancellation methods are applied as the detection performance is significantly degraded by multi-user interference. The convergence of the interference cancellation method establishes a constraint for the single-objective optimization problem. Optimization results for both parallel and successive interference cancellation are given. The findings of the parameter study are compared with results from literature, and recommendations concerning settings of DE control parameters are given.

A new way to determine the exact layer-wise SINR distribution for V-BLAST with successive interference cancellation at the receiver is presented. In contrast to previous publications, we do not restrict to zero-forcing, but also consider minimum mean square error interference suppression. It is shown analytically that an optimized detection order has an even larger impact in this case. Numerical examples provide deeper insights into the underlying effects.

In the past few years, a number of publications concentrated on developing soft-in-soft-out algorithms for coded noncoherent detection, thereby enabling iterative processing and avoiding the need of channel estimation. This paper focuses on iterative noncoherent detection of convolutionally coded M- DPSK signals for time-variant Rayleigh flat fading channels without the receiver having channel state information (CSI). Specifically we show that a noncoherent soft demodulator for minimum-shift keying, which was previously reported in the literature, lends itself quite naturally for soft demodulation of DPSK. We present an extension of this receiver in terms of per- survivor processing and examine the influence of different symbol labelings.

Iterative channel estimation and data detection is a useful method to improve the channel estimation quality without sacrificing the bandwidth efficiency. Since both the known training symbols (non-blind) and the unknown data symbols (blind) are used for channel estimation, corresponding techniques are referred to as semiblind. If the channel estimator and data detector are both optimal in the sense of maximum-likelihood criterion, we may call the algorithm as maximum-likelihood (ML) semiblind channel estimation (SBCE). This paper deals with ML-SBCE for frequency-flat multi-input multi-output systems with focus on the channel estimation mean squared error (MSE) analysis. Through semi-analytical efforts, we will show that ML-SBCE is biased at low SNR and tends to be unbiased at high SNR. The reasons of biasing are the erroneous data detection and the correlation between the noise and the detection errors. Besides, we will show that the MSE performance of MLSBCE is also influenced by the noise-error correlation. Based on these analyses, possibilities to compensate the biasing as well as improve the MSE performance will be pointed out.

In recent years, communication systems with multiple antennas at the transmitter and at the receiver have gained considerable interest. Layered architectures like the V-BLAST scheme are a strong candidate to exploit the capacity advantages of multiple antenna systems leading to practical wireless communication schemes with very high data rates. The combination with orthogonal frequency division multiplexing (OFDM), called MIMO-OFDM, with per-antenna-coding is one of the most likely implementations of multilayer architectures in frequency selective environments. In this paper, we present a novel, computational efficient implementation of successive interference cancellation (SIC) for coded MIMO-OFDM. It utilizes a parallelized version of the Sorted QR Decomposition (SQRD) to achieve the same optimized detection order for all subcarriers in order to exploit the error correction capability of the code within the SIC. In comparison to other schemes known from literature our approach requires only a fraction of computational complexity with almost the same performance.

In this contribution, we introduce a novel framework for combining approaches for acoustic echo cancellation and beamforming. Classical schemes of combination incorporate the simple concatenation of both subsystems. However, if the echo cancellers come first, they cannot exploit the noise reduction capabilities of the beamformer. In the other setup, a time-variant beamformer can heavily disturb the succeeding echo canceller's convergence. The new approach establishes a possibility to smoothly switch between two possible target functions: the mean squared errors at the beamformer's inputs and its output, respectively. Thus, one combined system benefits from advantages of both classical schemes. The paper contains a theoretical analysis of possible solutions for preceding echo cancellers, which are adapted using the beamformer output. The appropriate modification of the normalized least mean squares algorithm is derived too.

In this paper, we propose a new algorithm separating blindly the sources of a flat MIMO-communication link, where all sources are independently channel encoded. To this end the proposed method exploits statistical dependencies caused by the channel code for blindly estimating the channel and simultaneously adjusting a linear equalizer. In contrast to most common blind source separation methods the suggested method completely resolves the phase and permutation ambiguity of the estimated channel impulse response and the equalizer. The performance of the presented method is evaluated by numerical results.

In recent publications the use of lattice-reduction for signal detection in multiple antenna systems has been proposed. In this paper, we adopt these lattice-reduction-aided schemes to the MMSE criterion. We show that an obvious way to do this is infeasible and propose an alternative method based on an extended system model, which in conjunction with simple successive interference cancellation nearly reaches the performance of maximum-likelihood detection. Furthermore, we demonstrate that, a sorted QR decomposition can significantly reduce the computational effort associated with lattice-reduction. Thus, the new algorithm clearly outperforms existing methods with comparable complexity.

Theoretical and experimental studies have shown that layered space-time architectures like the BLAST system can exploit the capacity advantage of multiple antenna systems in rich-scattering environments. We present a new efficient algorithm for detecting such architectures with respect to the MMSE criterion. This algorithm utilizes a sorted QR decomposition of the channel matrix and leads to a simple successive detection structure. The algorithm needs only a fraction of the computational effort compared to the standard V-BLAST algorithm and achieves the same bit error performance.

This paper presents two different code acquisition schemes for sectorized multi-antenna CDMA systems in downlink mode. The so-called "separated code optimization" (SCO) decomposes a two dimensional spreading code into its time and space component and treats each part separately. The SCO maximizes the signal to interference and noise ratio at each receiver. In contrast, the "joint code optimization" (JCO) does not distinguish between time and space component and acquires the code simultaneously. Both methods are evaluated by Monte Carlo simulations and their performance are compared.

Post-filters are a powerful extension to improve echo attenuation when combined with the well-known echo canceller. In order to guarantee high quality of the transmitted speech signal, the primary purpose of a post-filtering system is to estimate the power spectral density (PSD) of the residual echo at the output of the echo canceller as accurately as possible. We introduce a novel technique to estimate the residual echo by using a microphone array. The robustness against double-talk and other additive interferences is reached by means of minimum statistics and further enhanced by exploiting spatial information. Simulation results show that the new methods are able to estimate the residual echo even under adverse conditions.

The use of multiple antenna systems provides a great performance advantage in Rayleigh fading environments. To exploit this advantage several schemes like the layered space-time architecture have been proposed, but mostly restricted to narrowband transmission. Recently, Lozano and Papadias have presented a generalization of the well-known V-BLAST detection algorithm for frequency selective fading channels. In order to improve the performance of this receiver, we propose an iterative detection algorithm in this paper.

This paper addresses aspects of channel coding in orthogonal frequency-division multiplexing-code-division multiple access (OFDM-CDMA) uplink systems where each user occupies a bandwidth much larger than the information bit rate. This inherent bandwidth expansion allows the application of powerful low-rate codes under the constraint of low decoding costs. Three different coding strategies are considered: the combination of convolutional and repetition codes, the code-spread system consisting of one single very low-rate convolutional code and a serial concatenation of convolutional, Walsh-Hadamard and repetition code. The latter scheme is improved by combining the Walsh-Hadamard codes with an additional M-phase-shift keying modulation resulting in complex-valued Walsh-Hadamard codes (CWCs). Analytical performance evaluations will be given for these codes for the first time. The application of CWCs as inner codes in a serial code concatenation is also addressed. We derive a symbol-by-symbol maximum a posteriori decoding algorithm in the complex signal space in order to enable iterative decoding for the entire code. A comprehensive performance analysis by simulation of all the proposed coding schemes shows that the Walsh-Hadamard-based schemes are the best choice for low-to-medium system load. Note that even for fully loaded OFDM-CDMA systems, the concatenation with an inner complex-valued Walsh-Hadamard code leads to a bit-error rate less than 10/sup -5/ for an E/sub b//N/sub 0/ of about 6 dB.

The transmission quality of LEO (low Earth orbit) satellite communication systems suffers from shadowed links and the power limitation of the satellites. In order to combat these two impacts, the use of diversity techniques is normally a good solution. Because of the non-frequency selective behavior of the satellite channel, frequency diversity cannot be exploited. Furthermore, time diversity is limited due to the short delay times allowed for speech transmission. Under these circumstances the application of space diversity can improve the transmission quality enormously. A basic model of a LEO satellite concept is the Globalstar system. This system utilizes DS-CDMA technology (direct sequence-code division multiple access), in which space diversity is implemented by combining the convolutionally encoded signals from different satellites with the help of a RAKE receiver. The performance of this concept is compared to a new approach using turbo codes. Regarding the transmission via two satellites, the coded bits of the two coders are split between the two satellite channels. At the receiver iterative decoding can be implemented. This concept needs less processing time and it performs better than the Globalstar system in terms of the bit error rate.

Wireless communication systems such as global system for mobile (GSM) communications are playing a growing role for data transmission. In order to ensure reliable transmission, the channel impulse response has to be estimated accurately. This is a difficult task particularly for fast fading channels caused by high-speed mobile units. We deal with the application of nonblind and blind channel estimation approaches to identify the full rate data Traffic CHannel (TCHF9.6) of GSM. We present a new iterative channel estimation scheme leading to a significant performance improvement especially for high-speed propagation with Doppler frequencies up to 500 Hz. Furthermore, it is shown that blind channel estimation schemes could be as efficient as nonblind methods when regarding bit error rates (BERs) after channel decoding in terms of the E~/sub b//N/sub 0/ ratio. Moreover, a solution for the scalar ambiguity inherent in all blind estimation approaches is suggested.

In this paper, we compare different combinations of a multi-channel non-adaptive noise reduction unit (NRU) and an acoustic echo cancellation unit (AEC) for a standard single-channel voice transmission. The results show that the NRU and the AEC-unit can be interchanged without increasing the computational complexity of the combined system, The length of an AEC's adaptive filter can be greatly reduced, if a multi-channel AEC-unit in front of a multichannel NRU is used. Furthermore, the control information, like the step-size of the NLMS-algorithm, has to be computed only once.

Channel estimation plays a leading role in wireless communication systems such as GSM (Global System for Mobile communications). Putting the main emphasis on investigating the application of non-blind and blind channel estimation approaches to the identification of a full rate data Traffic CHannel (TCH/F9.6), we present a new idea of an iterative channel estimation based upon the capabilities of channel coding. We show that this method leads to a significantly improved performance especially for high speed propagations with Doppler frequencies up to 500 Hz. Giving the bit error rates (BERs) before and after channel decoding in terms of the E~/sub b//N/sub 0/ ratio, we show, furthermore, that blind channel estimation schemes could be as efficient as non-blind methods.

We present an analysis of the generalized sidelobe canceller (GSC). It can be shown that the theoretical limits of the noise reduction performance depend only on the auto- and cross-spectral densities of the input signals. Furthermore, we compute the limits of the noise reduction performance for the theoretically determined diffuse noise field, which is an approximation for reverberant rooms. Our results show that the GSC cannot reduce noise further than 1 dB. These results were verified by simulation of reverberant environments. Only in sound-proof rooms with a reverberation time less than 100 ms the GSC performs well.

Shows that, provided an error-correcting source decoder is used, a higher quality of transmitted image can be achieved by applying a weaker channel code and hence transmitting more source data (finer quantization) in the same bandwidth.

This paper presents an a posteriori channel error correction method, based on subband synthesis. We estimate the position and amplitude of transmission errors using undownsampled subband signals, thus allowing correction of the subband signals and resynthesis of a nearly error-free signal. The method can be extended to multiple dimensions. Examples of images transmitted (compressed) with and without this method are shown. The results are very promising.

We investigate the application of M-ary orthogonal modulation for multi-carrier spread spectrum (MCSS) uplink transmission over a Rayleigh fading indoor channel. Different coherent detection strategies with perfectly known channel coefficients are analyzed. Furthermore, we present a decision-directed estimation receiver to apply channel phase estimation. With decision-directed estimation no redundancy like training data is required to be transmitted. Simulation results show no performance loss for low E~/sub b//N/sub 0/ and only a moderate degradation for high E~/sub b//N/sub 0/ in comparison to ideal phase equalization. Moreover, the results are always contrasted with BPSK performance. Applying M-ary orthogonal modulation is revealed to outperform BPSK with respect to bit error rate and spectral efficiency, even if ideal equalization is considered for BPSK and decision-directed estimation for M-ary orthogonal modulation. The results are generally valid, whereas in this paper, they are based on the European HIPERLAN/2-standardization.

For a multi-carrier spread-spectrum (MC-SS) system, the inner spreading can be optimized by applying M-ary orthogonal modulation. We investigate the concatenation of maximum likelihood (ML) Viterbi (1995) decoding with M-ary orthogonal modulation in a MC-SS system. The system operates over a frequency-selective Rayleigh fading indoor channel in the uplink. We first evaluate the bit specific log-likelihood ratio for ML Viterbi decoding and present an estimation of the ratio exploiting the MC technique. Furthermore, the trade-off between channel coding, M-ary orthogonal modulation and simple spreading is considered by Monte-Carlo simulations. The results are always compared with BPSK performance and they emphasize for the concerned indoor transmission scenario that a moderate bit-error-rate can only be achieved if M-ary orthogonal modulation is employed. All simulations are related to the European Hiperlan/2 standardization whereas the results are generally valid.

In this paper, we discuss applying iterative decoding algorithms (turbo-equalizer) to GSM speech channels. Although the turbo-equalizer under realistic time-variant channel scenarios has a high potential, we only get a minor performance gain when compared to soft-decision algorithms. Certain constraints of the GSM coding scheme prevent better results. In this paper, we discuss the transition from a very simple data transmission model to a full GSM-compatible system and compare the resulting bit-error-rates (BER).

In order to combat transmission errors in digital communication systems, forward error protection based on convolutional codes and interleaving in conjunction with soft decision maximum-likelihood decoding through the Viterbi (193) algorithm is often applied. The expected bit error rate for a given signal-to-noise ratio is generally difficult to foresee for a certain system. In this paper, we introduce a method which vividly treats the error behaviour on particular stages of the receiver, hereby allowing a coarse estimate of the overall bit error rate. Although no close mathematical results are given, several astonishing effects can easily be explained thus providing a better understanding of concatenated systems.

We compare the effect of blind and nonblind channel estimates on the performance of Global System for Mobile communications (GSM) receivers. More precisely, we investigate whether two blind approaches, based on higher order statistics (HOS), can compete with two conventional methods, exploiting training sequences. For blind and nonblind estimates of six fast and slowly fading mobile radio channels, we give simulated bit error rates (BERs), after Viterbi detection, in terms of the signal-to-noise ratio (SNR). We also study the influence of cochannel interferers at different values of the signal-to-interference ratio (SIR). Averaged over the six channel examples, we demonstrate that the blind channel estimation algorithm eigenvector approach to blind identification (EVI) leads to an SNR loss of 1.2-1.3 dB only, while it saves the 22% overhead in GSM data rate caused by the transmission of training sequences. Since just 142 samples are used for blind channel estimation, we consider this performance outstanding for an approach based on HOS.

We investigate the applicability of two algorithms for the blind identification of mixed-phase linear time-invariant FIR systems to the estimation of mobile radio channels on GSM conditions. One approach is based on second order cyclostationary statistics (SOCS), whereas the other exploits higher order stationary statistics (HOSS) of the received signal. While the former class of algorithms suffers from "singular" systems which can not be identified, the latter class is said to require an excessive number of samples of the received signal to achieve comparable performance levels. The purpose of this paper is two-fold: first, we demonstrate that "singular" systems represent a severe limitation to SOCS-based methods when it comes to the estimation of time-variant mobile radio channels from a small number of received samples. Secondly, we reveal that the approach relying on 4th order statistics yields a superior estimation performance: at a signal-to-noise-ratio of 10 dB, all channel examples can be identified from 142 samples of a GSM burst within a normalized mean square error bound of 7 per cent.

Higher-order-statistics (HOS) are being used in many areas of digital signal processing, e.g. in the field of array processing. The main aim is often to suppress Gaussian noise. Mostly, the corresponding algorithms are applied to short data blocks, because only then the stationarity of the data needed for cumulant estimation is given. In many cases, not enough attention is paid to the fact that for short data blocks the suppression of Gaussian noise is small compared to the estimation error made because of the higher order of the cumulants. The property of cumulants used to suppress Gaussian noise is studied in detail. With an algorithm for direction-of-arrival (DOA) estimation in the field of array processing, the estimation errors that occur when using HOS are compared with the estimation errors that occur when using 2nd order statistics. A quantitative result is given to show that for short data blocks the suppression of Gaussian noise with HOS doesn't lead to a better result than using 2nd order statistics.

In this paper the implementation of a broadband beamformer which is built up by several harmonically nested subarrays for each octave band combined with optimal postfiltering is described. This method has the advantage of providing large sensor distances for the postfilter estimation by simultaneously controlling the directivity of the array. The selection of an optimal postfilter is discussed in detail and its estimation based on a Nuttall/Carter method for spectrum estimation is described. The resulting noise reduction system yields improved performance in diffuse noise fields and no distortions in the case of coherent direct path noise. Furthermore, the system is robust to steering misadjustment.

The performance of many communication systems could be improved if the transmission channel was estimated blindly, i.e. without training sequences. As an example, we investigate whether, on GSM conditions, the blind channel estimation method EVI (eigen vector approach to blind identification) can compete with the non-blind least squares scheme based on the cross-correlation. For Gaussian stationary uncorrelated scattering channels, we give simulated bit error rates (BER) after Viterbi detection in terms of the mean signal-to-noise ratio (S~N~R~) for blind, non-blind, and ideal channel estimation. Averaged over three COST-207 propagation environments, EVI leads to an S~N~R~ loss of 1.1 dB only, which is quite remarkable for an approach based on higher order statistics, as just 142 samples can be used for blind channel estimation.

A novel DS-CDMA transmission scheme combining the two well-known techniques of M-ary orthogonal and differential PSK-modulation is presented. This "hybrid" scheme is applicable to the noncoherent uplink of the IS-95 system where up to now only M-ary orthogonal modulation is used. The additional, PSK-modulated data show a slightly increased bit error rate (BER) compared to the data from M-ary modulation. However, the overall degradation caused hereby is far outweighed by the gain achieved by a higher spreading factor. Analytical results confirmed by simulations yield an improvement of approximately 1.0 dB in terms of E/sub b//N/sub 0/ on a Rayleigh fading channel when no channel coding is used. This considerable gain is achieved upon only slight modification of the conventional modulation/demodulation procedure at a low additional complexity. It is demonstrated that the IS-95 system is not properly designed since the full exploitation of noncoherent demodulation techniques results in a gain of nearly 2.0 dB.

The problems of digital transmission over mobile radio channels are investigated under the particular viewpoint of bandwidth efficiency and simple system design. It is shown that a multicarrier system with non-frequency selective and non-time-selective subchannels is a suitable design, which, however, requires an accurate carrier recovery. For this reason a decision feedback loop for tracking a carrier frequency offset is described, which applies to multicarrier systems with coherent or incoherent detection. By using the information of all subsystems at one time instant, the mentioned control loop is not sensitive to incorrect decisions and error propagation, even in the presence of fading channels. The performance of this control loop is investigated under certain mobile radio conditions; some simulation results show that an almost perfect carrier recovery is possible.>

An approach to multipath correction in FM transmission is proposed. It is based on the factorization of the channel transfer function in terms of the roots of the underlying polynomial. A time-recursive algorithm for the identification of the roots of the all-zero channel transfer function at the receiver is presented. It allows an efficient adaptive design of special equalizer structures with low computational burden compared to existing solutions. The multipath correction method was tested in connection with a baseband FM receiver and a modulating signal consisting of a bandlimited stereo-multiplex signal produced by combining sinusoids with random phases distributed uniformly in the interval 0-2 pi . The identification of the channel roots was accomplished by means of a stochastic gradient algorithm with constant step size alpha =0.0005. The trajectories of the identified roots in the z-plane and the corresponding modulus of the equalizer output signal are illustrated for three different channel configurations.>

Multipath propagation is a severe problem in conventional FM broadcasting since this phenomenon causes nonlinear distortions after demodulation. Several proposals for the correction of multipath corrupted FM signals were given in some recent papers, but up to now there exist no satisfactory solutions with sufficiently low hardware complexity for commercial application. The main idea of the present paper is to take advantage of some a priori knowledge of the multipath transfer function with the goal to develop a nonrecursive equalizer structure with minimum hardware expense. For adaptive adjustment of the equalizer, we use a recently introduced constant modulus algorithm (CMA) in a modified form. Instead of direct coefficient adjustment, we regard the parameters of the transmission channel as unknowns which are updated adaptively. The equalizer coefficients are uniquely defined by the channel parameters and can be determined in one step. The advantage of this method is a large improvement of the convergence behavior in comparison to existing solutions. The methods presented here are tested by means of an FM hardware system including FM transmitter, multipath simulation, FM demodulator, and an FIR multipath equalizer, which allows real-time investigations of the equalizer performance under well-defined multipath conditions.

Multipath propagation in conventional FM broadcasting is a severe problem since this phenomenon results in nonlinear distortions after demodulation. In the present paper a solution is given for equalization of the multipath transfer function using FIR digital filters. The main idea is to take advantage of some a-priori knowledge of the multipath channel which reduces the number of the complex equalizer coefficients to 5 or 6. For adaptive adjustment we apply the well-known gradient algorithm using the absolute value of the complex FM envelope as a quality criterion which has to be minimized. Some examples will be given to illustrate the convergence behaviour of the algorithm and the real-time performance of the digital equalizer.