Multiuser MIMO techniques with feedback

Kooperative Antennenanlagen haben vor kurzem einen heißen Forschungsthema geworden, da Sie deutlich höhere spektrale Effizienz als herkömmliche zelluläre Systeme versprechen. Der Gewinn wird durch die Eliminierung von Inter-Zelle Störungen (ICI) durch Koordinierung der-Antenne Übertragungen erworben. Vor kurzem, verteilte Organisation Methoden vorgeschlagen. Eine der größten Herausforderungen für das Dezentrale kooperative Antennensystem ist Kanalschätzung für den Downlink Kanal besonders wenn FDD verwendet wird. Alle zugehörigen Basisstationen im genossenschaftlichen Bereich müssen die vollständige Kanal Informationen zu Wissen, die entsprechenden precoding Gewicht Matrix zu berechnen. Diese Information ist von mobilen Stationen übertragen werden Stationen mit Uplink Ressourcen zu stützen. Wird als mehrere Basisstationen und mehreren mobilen Stationen in kooperativen Antennensysteme und jede Basisstation und Mobilstation beteiligt sind, können mit mehreren Antennen ausgestattet sein, die Anzahl der Kanal Parameter wieder gefüttert werden erwartet, groß zu sein. In dieser Arbeit wird ein effizientes Feedback Techniken der downlink Kanal Informationen sind für die Multi-user Multiple Input Multiple Output Fall vorgeschlagen, der insbesondere auf verteilte kooperative Antennensysteme zielt. Zuerst wird ein Unterraum-basiertes Kanalquantisierungsverfahren vorgeschlagen, das ein vorbestimmtes Codebuch verwendet. Ein iterativer Codebuchentwurfsalgorithmus wird vorgeschlagen, der zu einem lokalen optimalen Codebuch konvergiert. Darüber hinaus werden Feedback-Overhead-Reduktionsverfahren entwickelt, die die zeitliche Korrelation des Kanals ausnutzen. Es wird gezeigt, dass das vorgeschlagene adaptive Codebuchverfahren in Verbindung mit einem Datenkomprimierungsschema eine Leistung nahe an dem perfekten Kanalfall erzielt, was viel weniger Rückkopplungsoverhead im Vergleich zu anderen Techniken erfordert. Das auf dem Unterraum basierende Kanalquantisierungsverfahren wird erweitert, indem mehrere Antennen auf der Senderseite und/oder auf der Empfängerseite eingeführt werden, und die Leistung eines Vorcodierungs- (/Decodierungs-) Schemas mit regulierter Blockdiagonalisierung (RBD) wurde untersucht. Es wird ein kosteneffizientes Decodierungsmatrixquantisierungsverfahren vorgeschlagen, dass eine komplexe Berechnung an der Mobilstation vermeiden kann, während es nur eine leichte Verschlechterung zeigt. Die Arbeit wird abgeschlossen, indem die vorgeschlagenen Feedback-Methoden hinsichtlich ihrer Leistung, ihres erforderlichen Feedback-Overheads und ihrer Rechenkomplexität verglichen werden.Cooperative antenna systems have recently become a hot research topic, as they promise significantly higher spectral efficiency than conventional cellular systems. The gain is acquired by eliminating inter-cell interference (ICI) through coordination of the base antenna transmissions. Recently, distributed organization methods have been suggested. One of the main challenges of the distributed cooperative antenna system is channel estimation for the downlink channel especially when FDD is used. All of the associated base stations in the cooperative area need to know the full channel state information to calculate the corresponding precoding weight matrix. This information has to be transferred from mobile stations to base stations by using uplink resources. As several base stations and several mobile stations are involved in cooperative antenna systems and each base station and mobile station may be equipped with multiple antennas, the number of channel state parameters to be fed back is expected to be big. In this thesis, efficient feedback techniques of the downlink channel state information are proposed for the multi-user multiple-input multiple-output case, targeting distributed cooperative antenna systems in particular. First, a subspace based channel quantization method is proposed which employs a predefined codebook. An iterative codebook design algorithm is proposed which converges to a local optimum codebook. Furthermore, feedback overhead reduction methods are devised exploiting temporal correlation of the channel. It is shown that the proposed adaptive codebook method in conjunction with a data compression scheme achieves a performance close to the perfect channel case, requiring much less feedback overhead compared with other techniques. The subspace based channel quantization method is extended by introducing multiple antennas at the transmitter side and/or at the receiver side and the performance of a regularized block diagonalization (RBD) precoding(/decoding) scheme has been investigated as well as a zero-forcing (ZF) precoding scheme. A cost-efficient decoding matrix quantization method is proposed which can avoid a complex computation at the mobile station while showing only a slight degradation. The thesis is concluded by comparing the proposed feedback methods in terms of their performance, their required feedback overhead, and their computational complexity. The techniques that are developed in this thesis can be useful and applicable for 5G, which is envisioned to support the high granularity/resolution codebook and its efficient deployment schemes. Keywords: MU-MIMO, COOPA, limited feedback, CSI, CQ, feedback overhead reduction, Givens rotatio

Limited feedback MIMO techniques for temporally correlated channels and linear receivers

Advanced mobile wireless networks will make extensive use of multiantenna (MIMO) transceivers to comply with high requirements of data rates and reliability. The use of feedback channels is of paramount importance to achieve this goal in systems employing frequency division duplexing (FDD). The design of the feedback mechanism is challenging due to the severe constraints imposed by computational complexity and feedback bandwidth restrictions. This thesis addresses the design of transmission strategies in both single-user and multi-user MIMO systems, based on compact feedback messages. First, recursive feedback mechanisms for single-user transmission scenarios are proposed, including stochastic gradient techniques, deterministic updates based on Givens rotations and low computational complexity schemes based on partial update filtering concepts. Thereafter, channel feedback algorithms are proposed, and a convergence analysis for static channels is presented. These algorithms can be used to provide channel side information to any multi-user MIMO solution. A limited-feedback decentralized multi-user MIMO solution is proposed, which avoids the need for explicit channel feedback. A feed-forward technique is proposed, which allows our methods to operate in presence of feedback errors. The performance of all the proposed algorithms is illustrated via link-level simulations, where the effect of different parameter values is assessed. Our results show that the proposed methods outperform existing limited-feedback counterparts over a range of low to medium mobile speeds, for moderate antenna array sizes that are deemed practical for commercial deployment. The computational complexity reduction of some of the proposed algorithms is also shown to be considerable, when compared to existing techniques

Binaural Audio Signal Processing Using Interaural Coherence Matching

Binaural room impulse responses (BRIRs) characterize the transfer of sound from a source in a room to the left and right ear entrances of a listener. Applying BRIRs to sound source signals enables headphone listening with the perception of a three dimensional auditory image. BRIRs are usually linear filters of several hundred milliseconds to several seconds length. The waveforms of the BRIRs contain therefore a vast amount of information. This thesis studies the modeling of BRIRs with a reduced set of parameters. It is shown that late BRIR tails can be modeled perceptually accurately by considering only the time-frequency energy decay relief and frequency dependent interaural coherence (IC). This insight on BRIR modeling enables a number of algorithms with advantages over the previous state of the art. Three such algorithms are proposed: The first algorithm makes it possible to obtain BRIRs by measuring room properties and listener properties separately, vastly reducing the number of measurements necessary to measure listener-specific BRIRs for a number of listeners and rooms. The listener properties are measured as a head related transfer function (HRTF) set and the room properties are measured as a B-format1 room impulse response (RIR). It is shown how to combine the HRTF set of the listener with a B-format RIR to obtain BRIRs for that room individualized for the listener. This technique uses the insight on BRIR perception by computing the BRIR tail as a frequency dependent, linear combination of B-format channels, designed to obtain the desired energy decay relief and interaural coherence. A serious problem related to convolving sound source signals with BRIRs is the computational complexity of implementing long BRIRs as finite impulse response (FIR) filters. Inspired by the perceptual experiments on BRIR tails, a modified Jot reverberator is proposed, simulating BRIR tails with the desired frequency dependent interaural coherence, requiring significantly less computational power than direct application of BRIRs. Also inspired by the perception of BRIRs, an extension of this reverberator is proposed, modeling efficiently the reverberation tail with the correct coherence and also distinct early reflections using two parallel feedback delay networks. If stereo signals are played back using headphones, unnatural binaural cues are given to the listener, e.g. interaural level difference (ILD) changes not accompanied by corresponding interaural time difference (ITD) changes or diffuse sound with unnatural IC. In order to simulate stereo listening in a room and to avoid these unnatural cues, BRIRs can be applied to the left and right stereo channels. Besides the computational complexity associated with applying the BRIR filters, this technique has a number of disadvantages. The room associated with the used BRIRs is imposed on the stereo signal, which usually already contains reverberation and applying BRIRs leads to a change in reverberation time and to coloration. A technique is proposed in which the direct sound is rendered using data extracted from HRTFs and the ambient sound contained in the stereo signal is modified such that its coherence is matched to the coherence of a binaural recording of diffuse sound, without modifying its spectrum. Implementations of reverberators based on general feedback-delay networks (e.g. Jot reverberators) can require a high number of operations for implementing the so-called feedback matrix. For certain applications where the number of channels needs to be high, such as decorrelators, this can pose a real problem. Special types of matrices are known which can be implemented efficiently due to matrix elements having the same magnitude. However, the complexity can also be reduced by introducing many zero elements. Different types of such sparse feedback matrices are proposed and tested for their suitability in Jot reverberators. A highly efficient feedback matrix is obtained by combining both approaches, choosing the nonzero elements of a sparse matrix from efficiently implementable Hadamard matrices. ______________________________ 1 B-format refers to a 4-channel signal recorded with four coincident microphones: one omni and three dipole microphones pointing in orthogonal directions

The Application of Adaptive Linear and N on-Linear Filters to Fringe Order Identification in White-Light Interferometry Systems

Conventional optical interferometry systems driven by highly coherent light sources have a very short unambiguous operating range, a direct consequence of the flatness of the interference fringes visibility profile at the output of the system. The range can be extended by using a white-light interferometer (WU), which is driven by a low-coherence source and produces a Gaussian visibility profile with a unique maximum in correspondence of the central fringe. Due to system and/or measurement noise, however, the position of the maximum (from which an accurate measurement of the measurand - displacement, temperature, pressure, flow, etc. - can be derived) is not easily detectable, and can lead to large measurement errors. This is especially true in a multiplexing scheme, where the source power is distributed evenly among various sensors, with a corresponding drop in the overall signal-to-noise ratio. The inclusion of a signal processing scheme at the receiver end is thus a necessity. As the fringe pattern at the output of a WLI system is basically a noisy sine wave amplitude modulated by a Gaussian envelope, it can be classified as a non-stationary, narrow-band, linear but non-Gaussian signa\. So far, no attempt has been made to apply digital filtering techniques, as understood in the signal processing community, to the output signal of a WLI system. This thesis constitutes a first step in that direction. Since the only measurable information given by the system is contained in the output signal, the system is modelled as a "black box" driven by the system and measurement noise processes and containing an unknown set of parameters. Standard least squares techniques can then be applied to estimate the parameters of the model, as is usually done in the field of system identification when only noisy output measurements are available. It is shown that identification of the model parameters is equivalent to finding a set of coefficients for an inverse filter which takes the WU signal at its input and delivers the unknown noise process at the output. The non-stationarity of the signal is accounted for by allowing for time variations of the model parameters; this justifies the use of adaptive filters with time-varying coefficients. A new central fringe identification scheme is proposed, based on a modification of the standard least mean square (LMS) adaptive filtering algorithm in combination with amplitude thresholding of the fringe pattern. The new scheme is shown to offer considerable improvement in the identification rate when tested against current schemes over comparable operating ranges, while retaining the computational simplicity and operational speed of the standard LMS. Its performance is also shown to be largely independent of the step-size parameter controlling the rate of convergence and tracking in the standard LMS, which is known to be the main obstacle for a successful application of the algorithm in a practical setting. The non-Gaussianity of the signal is explored and an attempt is made to apply higher-order statistics (HOS) algorithms to central fringe identification. The effectiveness of Gaussianity tests on pilot Gaussian data is seen to depend not only on the number and length of records available but, perhaps more importantly, on the bandwidth of the process. Violation of the stationarity assumption is shown to lead to mis-classification of a seemingly non-Gaussian signal into a Gaussian one, as the visibility profile may alter the distribution of the underlying sinusoid making it appear Gaussian, even when beam diffraction and wavefront aberrations combine to produce a nonGaussian profile. HOS-based adaptive algorithms may still be of some benefit, however, if processing is confined to that region of the fringe pattern where sufficient non-Gaussianity is allowed to develop. Non-linear adaptive filters based on the Volterra theories are finally applied to compensate for possible non-linearities introduced by mismatches in optical components, chromatic aberrations, and analogue-to-digital converters. It is shown that although a Volterra filter is able to reproduce the low-amplitude distortions of the fringe pattern better than a linear filter does, the identification rate does not improve. Reasons are given for such behaviour

Underwater acoustic communications

The underwater acoustic medium poses unique challenges to the design of robust, high throughput digital communications. The aim of this work is to identify modulation and receiver processing techniques to enable the reliable transfer of data at high rate, at range between two, potentially mobile parties using acoustics. More generally, this work seeks to investigate techniques to effectively communicate between two or more parties over a wide range of channel conditions where data rate is a key but not always the absolute performance requirement. Understanding the intrinsic ocean mechanisms that influence signal coherence, the relationship between signal coherence and optimum signal design, and the development of robust modulation and receiver processing techniques are the main areas of study within this work. New and established signal design, modulation, synchronisation, equalisation and spatial processing techniques are investigated. Several new, innovative techniques are presented which seek to improve the robustness of ‘classical’ solutions to the underwater acoustic communications problem. The performance of these techniques to mitigate the severe temporal dispersion of the underwater channel and its unique temporal variability are assessed. A candidate modulation, synchronisation and equalisation architecture is proposed based on a spatial-temporal adaptive signal processing (STAP) receiver. Comprehensive simulation results are presented to demonstrate the performance of the candidate receiver to time selective, frequency selective and spatially selective channel behaviour. Several innovative techniques are presented which maximise system performance over a wider range of operational and environmental conditions. Field trials results are presented based on system evaluation over a wide range of geographically distinct environments demonstrating system performance over a diverse range of ocean bathymetry, topography and background noise conditions. A real time implementation of the system is reported and field trials results presented demonstrating the capability of the system to support a wide range of data formats including video at useful frame rates. Within this work, several novel techniques have been developed which have extended the state of the art in high data rate underwater communications:- • Robust, high fidelity open loop synchronisation techniques capable of operating at marginal signal-to-noise ratios over a wide range of severely time spread environments. These high probability of synchronisation, low probability of false alarm techniques, provide the means for ‘burst’ open loop synchronisation in time, Doppler and space (bearing). The techniques have been demonstrated in communication and position fixing/navigation systems to provide repeatable range accuracy’s to centimetric order. • Novel closed loop synchronisation compensation for STAP receiver architectures. Specifically, this work has demonstrated the performance benefits of including both delay lock loop (DLL) and phase lock loop (PLL) support for acoustic adaptive receivers to offload tracking effort from the fractional feedforward equaliser section. It has been shown that the addition of a DLL/PLL outperforms the PLL only case for Doppler errors exceeding a few fractions of a knot. • Recycling of training data has been demonstrated as a potentially useful means to improve equaliser convergence in difficult acoustic channels. With suitable processing power, training data recycling introduces no additional transmission time overhead, which may be a limiting factor in battery powered applications. • Forward and time reverse decoding of packet data has been demonstrated as an effective means to overcome some non-minimum phase channel conditions. It has also been shown that there may be further benefits in terms of improved bit error performance, by exploiting concurrent forward and backward symbol data under modest channel conditions. • Several wideband techniques have been developed and demonstrated to be effective at resolving and coherently tracking difficult doubly spread acoustic channels. In particular, wideband spread spectrum techniques have been shown to be effective at resolving acoustic multipath, and with the aid of independent delay lock loops, track individual path arrivals. Techniques have been developed which can effect coherent or non-coherent recombination of these paths with a view to improving the robustness of an acoustic link operating at very low signal-to-noise levels. • Demonstrated throughputs of up to 41kbps in a difficult, tropical environment, featuring significant biological noise levels for mobile platforms at range up to 1.5km. • Demonstrated throughputs of between 300bps and 1600bps in a shallow, reverberant environment, at a range up to 21km at LF. • Implemented and demonstrated all algorithms in real time systems

Recursive Estimation of Structure and Motion from Monocular Images

The determination of the 3D motion of a camera and the 3D structure of the scene in which the camera is moving, known as the Structure from Motion (SFM) problem, is a central problem in computer vision. Specifically, the recursive (online) estimation is of major interest for robotics applications such as navigation and mapping. Many problems still hinder the deployment of SFM in real-life applications namely, (1) the robustness to noise, outliers and ambiguous motions, (2) the numerical tractability with a large number of features and (3) the cases of rapidly varying camera velocities. Towards solving those problems, this research presents the following four contributions that can be used individually, together, or combined with other approaches. A motion-only filter is devised by capitalizing on algebraic threading constraints. This filter efficiently integrates information over multiple frames achieving a performance comparable to the best state of the art filters. However, unlike other filter based approaches, it is not affected by large baselines (displacement between camera centers). An approach is introduced to incorporate, with only a small computational overhead, a large number of frame-to-frame features (i.e., features that are matched only in pairs of consecutive frames) in any analytic filter. The computational overhead grows linearly with the number of added frame-to-frame features and the experimental results show an increased accuracy and consistency. A novel filtering approach scalable to accommodate a large number of features is proposed. This approach achieves both the scalability of the state of the art filter in scalability and the accuracy of the state of the art filter in accuracy. A solution to the problem of prediction over large baselines in monocular Bayesian filters is presented. This problem is due to the fact that a simple prediction, using constant velocity models for example, is not suitable for large baselines, and the projections of the 3D points that are in the state vector can not be used in the prediction due to the need of preserving the statistical independence of the prediction and update steps

Proceedings of the 35th WIC Symposium on Information Theory in the Benelux and the 4th joint WIC/IEEE Symposium on Information Theory and Signal Processing in the Benelux, Eindhoven, the Netherlands May 12-13, 2014

Compressive sensing (CS) as an approach for data acquisition has recently received much attention. In CS, the signal recovery problem from the observed data requires the solution of a sparse vector from an underdetermined system of equations. The underlying sparse signal recovery problem is quite general with many applications and is the focus of this talk. The main emphasis will be on Bayesian approaches for sparse signal recovery. We will examine sparse priors such as the super-Gaussian and student-t priors and appropriate MAP estimation methods. In particular, re-weighted l2 and re-weighted l1 methods developed to solve the optimization problem will be discussed. The talk will also examine a hierarchical Bayesian framework and then study in detail an empirical Bayesian method, the Sparse Bayesian Learning (SBL) method. If time permits, we will also discuss Bayesian methods for sparse recovery problems with structure; Intra-vector correlation in the context of the block sparse model and inter-vector correlation in the context of the multiple measurement vector problem