This dissertation presents beamforming designs. Using novel techniques and methods, the performance of the beamforming is improved on dual-band, tri-band, flexible function, tunable function in THz, and dynamic controllability on incident wave.

The increased use of personal communication devices, personal computers and wireless cellular telephones enables the development of new inter-personal communication systems. The merge between computers and telephony technologies brings up the demand for convenient hands-free communications. In such systems the users wish to lead a conversation in much the same way as in a normal person-to-person conversation. The advantages of hands-free telephones are safety, convenience and greater flexibility. In many countries and regions, hand held telephony in cars is prohibited by legislation. By placing the microphone far away from the user a number of disadvantages are introduced, which results in substantial speech distortion and poor sound quality. These disturbances are mainly caused by room reverberation and background noise. Furthermore, acoustic feedback generated at the near-end side is a problem for the far-end side talker, who will hear his/her own voice echoed with 100-200 ms delay, making speech conversation substantially more difficult. Digital filtering may be used to obtain a similar sound quality as for hand held telephony. Three major tasks must be addressed in order to improve the quality of hands-free communication systems; noise suppression, room reverberation suppression, and acoustic feedback cancellation of the hands-free loudspeaker. The filtering operation must perform the above mentioned tasks without causing severe near-end speech distortion. A properly designed broad-band microphone array is able to perform all the given tasks, i.e. speech enhancement, echo cancellation and reverberation suppression, in a concise and effective manner. This is due to the fact that the spatial domain may be utilized as well as the temporal domain. This thesis deals with the problem of specification and design of beamformers used to extract the source signal information. A new subband adaptive beamforming algorithm is proposed, where many of the drawbacks embedded in conventional adaptive beamforming are eliminated. Evaluation in a car hands-free situation show the benefits of the proposed method. Blind signal separation is discussed and a new structure based on frequency domain inverse channel identification and time domain separation, is proposed. Further, filter-bank properties and design are discussed together with performance limitations in subband beamforming structures. / Avhandlingen behandlar specifikation och konstruktion av mikrofon-arrayer för att extrahera talinformation. En ny adaptiv delbands beamforming-algoritm föreslås där många av nackdelarna hos konventionella adaptiva beamformers är eliminerade. En utvärdering i en bil med ett frihands-system bekräftar fördelarna med den föreslagna metoden. Blind signal-separation diskuteras och en ny struktur föreslås, baserad på en inverterande kanalidentifiering utförd i frekvensdomän med en kontinuerlig separation utförd i tidsdomän. Filterbanks-egenskaper och designmetoder diskuteras tillsammans med begränsningar som finns i beamforming-strukturer utförda i delband.

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Doctor of Philosophy / The next generation mobile networks are expected to provide multimedia applications with a high quality of service. On the other hand, interference among multiple base stations (BS) that co-exist in the same location limits the capacity of wireless networks. In conventional wireless networks, the base stations do not cooperate with each other. The BSs transmit individually to their respective mobile stations (MS) and treat the transmission from other BSs as interference. An alternative to this structure is a network cooperation structure. Here, BSs cooperate with other BSs to simultaneously transmit to their respective MSs using the same frequency band at a given time slot. By doing this, we significantly increase the capacity of the networks. This thesis presents novel research results on a noncooperative transmission scheme and a cooperative transmission scheme for multi-user multiple-input-multiple-output orthogonal frequency division multiplexing (MIMO-OFDM). We first consider the performance limit of a noncooperative transmission scheme. Here, we propose a method to reduce the interference and increase the throughput of orthogonal frequency division multiplexing (OFDM) systems in co-working wireless local area networks (WLANs) by using joint adaptive multiple antennas(AMA) and adaptive modulation (AM) with acknowledgement (ACK) Eigen-steering. The calculation of AMA and AM are performed at the receiver. The AMA is used to suppress interference and to maximize the signal-to-interference-plus-noise ratio (SINR). The AM scheme is used to allocate OFDM sub-carriers, power, and modulation mode subject to the constraints of power, discrete modulation, and the bit error rate (BER). The transmit weights, the allocation of power, and the allocation of sub-carriers are obtained at the transmitter using ACK Eigen-steering. The derivations of AMA, AM, and ACK Eigen-steering are shown. The performance of joint AMA and AM for various AMA configurations is evaluated through the simulations of BER and spectral efficiency (SE) against SIR. To improve the performance of the system further, we propose a practical cooperative transmission scheme to mitigate against the interference in co-working WLANs. Here, we consider a network coordination among BSs. We employ Tomlinson Harashima precoding (THP), joint transmit-receive beamforming based on SINR (signal-to-interference-plus-noise-ratio) maximization, and an adaptive precoding order to eliminate co-working interference and achieve bit error rate (BER) fairness among different users. We also consider the design of the system when partial channel state information (CSI) (where each user only knows its own CSI) and full CSI (where each user knows CSI of all users) are available at the receiver respectively. We prove analytically and by simulation that the performance of our proposed scheme will not be degraded under partial CSI. The simulation results show that the proposed scheme considerably outperforms both the existing noncooperative and cooperative transmission schemes. A method to design a spectrally efficient cooperative downlink transmission scheme employing precoding and beamforming is also proposed. The algorithm eliminates the interference and achieves symbol error rate (SER) fairness among different users. To eliminate the interference, Tomlinson Harashima precoding (THP) is used to cancel part of the interference while the transmit-receive antenna weights cancel the remaining one. A new novel iterative method is applied to generate the transmit-receive antenna weights. To achieve SER fairness among different users and further improve the performance of MIMO systems, we develop algorithms that provide equal SINR across all users and order the users so that the minimum SINR for each user is maximized. The simulation results show that the proposed scheme considerably outperforms existing cooperative transmission schemes in terms of the SER performance and complexity and approaches an interference free performance under the same configuration. We could improve the performance of the proposed interference cancellation further. This is because the proposed interference cancellation does not consider receiver noise when calculating the transmit-receive weight antennas. In addition, the proposed scheme mentioned above is designed specifically for a single-stream multi-user transmission. Here, we employ THP precoding and an iterative method based on the uplink-downlink duality principle to generate the transmit-receive antenna weights. The algorithm provides an equal SINR across all users. A simpler method is then proposed by trading off the complexity with a slight performance degradation. The proposed methods are extended to also work when the receiver does not have complete Channel State Informations (CSIs). A new method of setting the user precoding order, which has a much lower complexity than the VBLAST type ordering scheme but with almost the same performance, is also proposed. The simulation results show that the proposed schemes considerably outperform existing cooperative transmission schemes in terms of SER performance and approach an interference free performance. In all the cooperative transmission schemes proposed above, we use THP to cancel part of the interference. In this thesis, we also consider an alternative approach that bypasses the use of THP. The task of cancelling the interference from other users now lies solely within the transmit-receive antenna weights. We consider multiuser Gaussian broadcast channels with multiple antennas at both transmitter and receivers. An iterative multiple beamforming (IMB) algorithm is proposed, which is flexible in the antenna configuration and performs well in low to moderate data rates. Its capacity and bit error rate performance are compared with the ones achieved by the traditional zero-forcing method.

In FDD amplify-and-forward distributed relay network, codebook techniques are utilized to feedback quantized CSI with limited cost. First, this thesis focuses on the phaseonly codebook and with-power-control codebook design methods under individual relay power constraints. Phase-only codebooks can be generated off-line with the Grassmannian beamforming criterion. Due to non-uniform distribution of the optimal beamforming vector in the vector space, The Lloyd’s algorithm is proposed for with-power-control codebook designs. To reduce search complexity, a suboptimal method for the codebook update stage in the Lloyd’s algorithm is proposed. Its performance is compared to the performance of the global search method which provides the optimal solution but incurs high computation complexity. Second, this thesis investigates the performance difference between phaseonly and with-power-control codebooks. It is found that the power control gain is tightly related to the relay locations. When the relays are close to the source node, the gain from power control is negligible and using phase-only codebooks becomes a viable choice for feedback due to its simple implantation and off-line computation. Finally, the problem of codebook design extends to the total relay power constraint case, and the Lloyd’s algorithm with primary eigenvector method is proposed to design a suboptimal codebook. / UOIT

This work explored the design considerations on the real-time medical ultrasound adaptive beamformer implementations using different computing platforms: CPU, GPU and FPGA. Adaptive beamforming has been well considered as an advanced solution for improving the image quality of medical ultrasound imaging machines. Although it provides promising improvements in lateral resolution, image contrast and imaging penetration depth, the use of adaptive beamforming is substantially more computationally demanding than conventional delay-and-sum beamformers. In order not to compromise the real-time performance of medical ultrasound systems, an accelerated solution is desirable. In this work, CPU implementation was used as a baseline implementation, based on which the intrinsic characteristics of the algorithm were analyzed. After the analysis of a particular adaptive beamforming algorithm, minimum-variance adaptive beamforming, two design parameters M and L were found to affect the implementation performance in two aspects: computational demand and image quality. The trends of the two aspects were contradictory with respect to the increment of M and L values. In our experiments, when M and L increased, the computational demand increased in a cubic curve; meanwhile, the image quality did not have much improvement when the increased values of M and L entered certain ranges. Since we targeted at a real-time solution without sacrificing the good image quality that adaptive beamforming proposed, a tradeoff was made on the selection of M and L values to balance the two contradictory requirements. Built upon the theoretical algorithmic analysis of the real-time adaptive beamformer realization, the implementations were developed with FPGA and GPU. While a dedicated hardware solution might be able to address the computational demand of the particular design, the need for an efficient algorithm exploration framework demanded a reprogrammable platform solution that was high-performance and easily reconfigurable. Besides, although a simple processor could provide convenient algorithm exploration via software development environment, real-time performance was usually not achievable. As a result, a reprogrammable medical ultrasound research platform for investigating advanced imaging algorithms was constructed in our project. The use of FPGA and GPU for implementing the real-time adaptive beamformer on our platform was explored. In our test cases, both FPGA- and GPUbased solutions achieved real-time throughput exceeding 80 frames-per-second, and over 38x improvement when compared to our baseline CPU implementation. Moreover, the implementations were also evaluated in terms of portability, data accuracy, programmability, and system integration. Due to its high power consumption, high-performance GPU solution is best suited for bedside applications, while FPGAs are more suitable for portable and hand-held medical ultrasound machines. Besides, while the development time on GPU platform remains much lower than its FPGA counterpart, the FPGA solution is effective in providing the necessary I/O bandwidth to enable an end-to-end real-time reconfigurable medical ultrasound image formation system. / published_or_final_version / Electrical and Electronic Engineering / Doctoral / Doctor of Philosophy

Sensor arrays have been successfully applied to many engineering fields and the theoretical as well as applied aspects of senor array processing have received intensive research interest. Practically, sensor array systems usually suffer from uncertainties such as unknown gains and phases, mutual coupling, and look direction mismatch. In this thesis, problems of direction finding and beamforming in the presence of array uncertainties are addressed, and new algorithms to tackle these problems are developed. In certain applications, senor arrays are only partly calibrated. Hence, the exact responses of some sensors are unknown, which degrades the performance of traditional direction finding techniques. To tackle this problem, a new method for direction finding with partly calibrated uniform linear arrays (ULAs) is proposed. It generalizes the estimation of signal parameters via rotational invariance techniques (ESPRIT) by modeling the imperfection of the ULA as gain and phase uncertainties. For a fully calibrated array, it reduces to the standard ESPRIT algorithm. In this method, the direction-of-arrivals (DOAs), unknown gains and phases of the uncalibrated sensors can be estimated in closed-form without performing spectral grid search. Moreover, it can be further improved by a refining scheme proposed. Its superiority over existing methods is demonstrated by simulation results. Apart from unknown gains and phases, the mutual coupling caused by interactions among sensors also seriously deteriorate the performance of array processing techniques. In ULAs, the mutual coupling matrix (MCM) can be approximated as a banded symmetric Toeplitz matrix. Using this specific property, a new parameterization of the steering vector is proposed and the corresponding method for joint estimation of DOAs and MCM is derived. Compared with the conventional subarray-based method, the proposed one makes use of the whole array and achieves better performance especially for weak signals. On the other hand, the specific property is further employed to develop a new approach to calibrate the steering vector. By incorporating the calibrated steering vector with a diagonally loaded robust beamformer, a new adaptive beamformer for ULAs with mutual coupling is obtained. It is found that the resultant steering vector estimate considerably improves the robustness of the beamformer against mutual coupling. Another common uncertainty in sensor array systems is the look direction mismatch. Though numerous robust beamformers have been developed accordingly, most of them cannot offer sufficient robustness against large look direction errors. To this end, a new robust beamforming method which can flexibly control the magnitude response in the look direction is proposed. By linearizing the nonconvex constraints in the original problem, the resultant problem is convex and can be solved using second-order cone programming (SOCP). Moreover, to further improve the robustness against array covariance uncertainties, this method is extended by optimizing its worst-case performance. Unlike some conventional methods restricted to specific arrays, the proposed method is applicable to arbitrary array geometries. Simulation results show that the proposed method offers comparable performance to the optimal solution for uniform linear arrays, and also achieves good results under different array specifications and geometries. / published_or_final_version / Electrical and Electronic Engineering / Doctoral / Doctor of Philosophy

The application of adaptive beamforming to biomedical ultrasound imaging has been an active research area in recent years. Adaptive beamforming techniques have the capability of achieving excellent resolution and sidelobe suppression, thus improving the quality of the ultrasound images. This quality improvement, however, comes at a high computational cost. The work presented in this thesis aims to answer the following basic question: Can we reduce the computational complexity of adaptive beamforming without a significant degradation of the image quality? Our objective is to explore a combination of low-complexity non-adaptive beamforming, such as the conventional Delay-and-Sum (DAS) method, with high-complexity adaptive beamforming, such as the standard Minimum-Variance Distortionless Response (MVDR) method implemented using the Generalized Sidelobe Canceller (GSC). Such a combination should have the lower computational complexity than adaptive beamforming, but it should also offer the image quality comparable to that obtained using adaptive beamforming. In addition to the adaptive GSC-based MVDR beamforming method, we also investigate the performance of the so-called Adaptive Single Snapshot Beamformer (ASSB), which is relatively unexplored in the ultrasound imaging literature. The main idea behind our approach to combining a non-adaptive beamformer with an adaptive one is based on the use of the data-dependent variable known as the coherence factor. The resulting hybrid beamforming method can be summarized as follows: For each input snapshot to be beamformed, calculate the corresponding coherence factor; if the coherence factor is below a certain threshold, use non-adaptive DAS beamforming, otherwise use adaptive (GSC-based or ASSB-based) beamforming. We have applied this simple switching scheme to the simulated B-mode ultrasound images of the 12-point and point-scatterer-cyst phantoms that are commonly used in the ultrasound imaging literature to evaluate the image quality. Our simulation results show that, in comparison to optimal high-complexity always-adaptive beamforming, our hybrid beamformer can yield significant computational savings that range from 59% to 99%, while maintaining the image quality (measured in terms of resolution and contrast) within a 5% degradation margin. / Graduate