134 research outputs found
Array signal processing robust to pointing errors
The objective of this thesis is to design computationally efficient DOA (direction-of-
arrival) estimation algorithms and beamformers robust to pointing errors, by
harnessing the antenna geometrical information and received signals. Initially,
two fast root-MUSIC-type DOA estimation algorithms are developed, which can
be applied in arbitrary arrays. Instead of computing all roots, the first proposed
iterative algorithm calculates the wanted roots only. The second IDFT-based
method obtains the DOAs by scanning a few circles in parallel and thus the
rooting is avoided. Both proposed algorithms, with less computational burden,
have the asymptotically similar performance to the extended root-MUSIC.
The second main contribution in this thesis is concerned with the matched
direction beamformer (MDB), without using the interference subspace. The manifold
vector of the desired signal is modeled as a vector lying in a known linear
subspace, but the associated linear combination vector is otherwise unknown due
to pointing errors. This vector can be found by computing the principal eigen-vector
of a certain rank-one matrix. Then a MDB is constructed which is robust
to both pointing errors and overestimation of the signal subspace dimension.
Finally, an interference cancellation beamformer robust to pointing errors
is considered. By means of vector space projections, much of the pointing error
can be eliminated. A one-step power estimation is derived by using the theory
of covariance fitting. Then an estimate-and-subtract interference canceller beamformer
is proposed, in which the power inversion problem is avoided and the
interferences can be cancelled completely
Estudo de formas de onda e conceção de algoritmos para operação conjunta de sistemas de comunicação e radar
The focus of this thesis is the processing of signals and design of algorithms
that can be used to enable radar functions in communications systems.
Orthogonal frequency division multiplexing (OFDM) is a popular multicarrier
modulation waveform in communication systems. As a wideband
signal, OFDM improves resolution and enables spectral efficiency in radar
systems, while also improving detection performance thanks to its inherent
frequency diversity. This thesis aims to use multicarrier waveforms for radar
systems, to enable the simultaneous operation of radar and communication
functions on the same device. The thesis is divided in two parts. The first
part, studies the adaptation and application of other multicarrier waveforms
to radar functions. At the present time many studies have been carried out
to jointly use the OFDM signal for communication and radar functions, but
other waveforms have shown to be possible candidates for communication
applications. Therefore, studies on the evaluation of the application of these
same signals to radar functions are necessary. In this thesis, to demonstrate
that other multicarrier waveforms can overcome the OFDM waveform
in radar/communication (RadCom) systems, we propose the adaptation of
the filter bank multicarrier (FBMC), generalized frequency division multiplexing
(GFDM) and universal filtering multicarrier (UFMC) waveforms for radar
functions. These alternative waveforms were compared performance-wise
regarding achievable target parameter estimation performance, amount of
residual background noise in the radar image, impact of intersystem interference
and flexibility of parameterization. In the second part of the thesis,
signal processing techniques are explored to solve some of the limitations
of the use of multicarrier waveforms for RadCom systems. Radar systems
based on OFDM are promising candidates for future intelligent transport networks.
Exploring the dual functionality enabled by OFDM, we presents cooperative
methods for high-resolution delay-Doppler and direction-of-arrival
estimation. High-resolution parameter estimation is an important requirement
for automotive radar systems, especially in multi-target scenarios that
require reliable target separation performance. By exploring the cooperation
between vehicles, the studies presented in this thesis also enable the distributed
tracking of targets. The result is a highly accurate multi-target tracking
across the entire cooperative vehicle network, leading to improvements
in transport reliability and safety.O foco desta tese é o processamento de sinais e desenvolvimento de algoritmos
que podem ser utilizados para a habilitar a função de radar nos sistemas
de comunicação. OFDM (Orthogonal Frequency Division Multiplexing)
é uma forma de onda com modulação multi-portadora, popular em sistemas
de comunicação. Para sistemas de radar, O OFDM melhora a resolução e
fornece eficiência espectral, além disso sua diversidade de frequências melhora
o desempenho na detecção do radar. Essa tese tem como objetivo
utilizar formas de onda multi-portadoras para sistemas de radar, possibilitando
a operação simultânea de funções de radar e de comunicação num
mesmo dispositivo. A tese esta dividida em duas partes. Na primeira parte
da tese são realizados estudos da adaptabilidade de outras formas de onda
multi-portadora para funções de radar. Nos dias atuais, muitos estudos sobre
o uso do sinal OFDM para funções de comunicação e radar vêm sendo
realizados, no entanto, outras formas de onda mostram-se possíveis candidatas
a aplicações em sistemas de comunicação, e assim, avaliações para
funções de sistema de radar se tornam necessárias. Nesta tese, com a
intenção de demonstrar que formas de onda multi-portadoras alternativas
podem superar o OFDM nos sistemas de Radar/comunicação (RadCom),
propomos a adaptação das seguintes formas de onda: FBMC (Filter Bank
Multicarrier); GFDM (Generalized Frequency Division Multiplexing); e UFMC
(Universal Filtering Multicarrier) para funções de radar. Também produzimos
uma análise de desempenho dessas formas de onda sobre o aspecto
da estimativa de parâmetros-alvo, ruído de fundo, interferência entre sistemas
e parametrização do sistema. Na segunda parte da tese serão explorados
técnicas de processamento de sinal de forma a solucionar algumas
das limitações do uso de formas de ondas multi-portadora para sistemas
RadCom. Os sistemas de radar baseados no OFDM são candidatos
promissores para futuras redes de transporte inteligentes, porque combinam
funções de estimativa de alvo com funções de rede de comunicação
em um único sistema. Explorando a funcionalidade dupla habilitada pelo
OFDM, nesta tese, apresentamos métodos cooperativos de alta resolução
para estimar o posição, velocidade e direção dos alvos. A estimativa de
parâmetros de alta resolução é um requisito importante para sistemas de
radar automotivo, especialmente em cenários de múltiplos alvos que exigem
melhor desempenho de separação de alvos. Ao explorar a cooperação entre
veículos, os estudos apresentados nesta tese também permitem o rastreamento
distribuído de alvos. O resultado é um rastreamento multi-alvo altamente
preciso em toda a rede de veículos cooperativos, levando a melhorias
na confiabilidade e segurança do transporte.Programa Doutoral em Telecomunicaçõe
Adaptive Interference Removal for Un-coordinated Radar/Communication Co-existence
Most existing approaches to co-existing communication/radar systems assume
that the radar and communication systems are coordinated, i.e., they share
information, such as relative position, transmitted waveforms and channel
state. In this paper, we consider an un-coordinated scenario where a
communication receiver is to operate in the presence of a number of radars, of
which only a sub-set may be active, which poses the problem of estimating the
active waveforms and the relevant parameters thereof, so as to cancel them
prior to demodulation. Two algorithms are proposed for such a joint waveform
estimation/data demodulation problem, both exploiting sparsity of a proper
representation of the interference and of the vector containing the errors of
the data block, so as to implement an iterative joint interference removal/data
demodulation process. The former algorithm is based on classical on-grid
compressed sensing (CS), while the latter forces an atomic norm (AN)
constraint: in both cases the radar parameters and the communication
demodulation errors can be estimated by solving a convex problem. We also
propose a way to improve the efficiency of the AN-based algorithm. The
performance of these algorithms are demonstrated through extensive simulations,
taking into account a variety of conditions concerning both the interferers and
the respective channel states
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MOCZ for Blind Short-Packet Communication: Practical Aspects
We investigate practical aspects of a recently introduced blind (noncoherent) communication scheme, called modulation on conjugate-reciprocal zeros (MOCZ). MOCZ is suitable for a reliable transmission of sporadic and short-packets at ultra-low latency and high spectral efficiency via unknown multipath channels, which are assumed to be static over the receive duration of one packet. The information is modulated on the zeros of the transmitted discrete-time baseband signal’s z− transform. Because of ubiquitous impairments between the transmitter and receiver clocks, a carrier frequency offset occurs after down-conversion to the baseband. This results in a common rotation of the zeros. To identify fractional rotations of the base angle in the zero-pattern, we propose an oversampled direct zero-testing decoder to identify the most likely one. Integer rotations correspond to cyclic shifts of the binary message, which we determine by cyclically permutable codes (CPC). Additionally, the embedding of CPCs into cyclic codes, enables additive error-correction which reduces the bit-error-rate tremendously. Furthermore, we exploit the trident structure in the signal’s autocorrelation for an energy based detector to estimate timing offsets and the effective channel delay spread. We finally demonstrate how this joint data and channel estimation can be largely improved by receive antenna diversity at low SNR
Personal Sound Zones by Subband Filtering and Time Domain Optimization
[EN] Personal Sound Zones (PSZ) systems aim to render independent sound signals to multiple listeners within a room by using arrays of loudspeakers. One of the algorithms used to provide PSZ is Weighted Pressure Matching (wPM), which computes the filters required to render a desired response in the listening zones while reducing the acoustic energy arriving to the quiet zones. This algorithm can be formulated in time and frequency domains. In general, the time-domain formulation (wPM-TD) can obtain good performance with shorter filters and delays than the frequency-domain formulation (wPM-FD). However, wPM-TD requires higher computation for obtaining the optimal filters. In this
article, we propose a novel approach to the wPM algorithm named Weighted Pressure Matching with Subband Decomposition (wPMSD), which formulates an independent time-domain optimization problem for each of the subbands of a Generalized Discrete Fourier Transform (GDFT) filter bank. Solving the optimization independently for each subband has two main advantages: 1) lower computational complexity than wPM-TD to compute the optimal filters; 2) higher versatility than the classic wPM algorithms, as it allows different configurations (sets of loudspeakers, filter lengths, etc.) in each subband. Moreover, filtering the input signals with a GDFT filter bank (as in wPM-SD) requires lower computational effort than broadband filtering (as in wPM-TD and wPM-FD), which is beneficial for practical PSZ systems. We present experimental evaluations showing that wPM-SD offers very similar performance to wPM-TD. In addition, two cases where the versatility of wPM-SD is beneficial for a PSZ system are presented and experimentally validated.This work was supported by Grants RTI2018-098085-B-C41 (MCIU/AEI/FEDER, UE), RED2018-102668-T and PROMETEO/2019/109. The work of Vicent Moles-Cases has been supported by Spanish Ministry of Education under Grant FPU17/01288.Molés-Cases, V.; Piñero, G.; Diego Antón, MD.; Gonzalez, A. (2020). Personal Sound Zones by Subband Filtering and Time Domain Optimization. IEEE/ACM Transactions on Audio Speech and Language Processing. 28:2684-2696. https://doi.org/10.1109/TASLP.2020.3023628S268426962
Filter Optimization for Personal Sound Zones Systems
[ES] Los sistemas de zonas de sonido personal (o sus siglas en inglés PSZ) utilizan altavoces y técnicas de procesado de señal para reproducir sonidos distintos en diferentes zonas de un mismo espacio compartido. Estos sistemas se han popularizado en los últimos años debido a la amplia gama de aplicaciones que podrían verse beneficiadas por la generación de zonas de escucha individuales. El diseño de los filtros utilizados para procesar las señales de sonido es uno de los aspectos más importantes de los sistemas PSZ, al menos para las frecuencias bajas y medias. En la literatura se han propuesto diversos algoritmos para calcular estos filtros, cada uno de ellos con sus ventajas e inconvenientes. En el presente trabajo se revisan los algoritmos para sistemas PSZ propuestos en la literatura y se evalúa experimentalmente su rendimiento en un entorno reverberante. Los distintos algoritmos se comparan teniendo en cuenta aspectos como el aislamiento acústico entre zonas, el error de reproducción, la energía de los filtros y el retardo del sistema. Además, se estudian estrategias computacionalmente eficientes para obtener los filtros y también se compara su complejidad computacional. Los resultados experimentales obtenidos revelan que las soluciones existentes no pueden ofrecer una complejidad computacional baja y al mismo tiempo un buen rendimiento con baja latencia. Por ello se propone un nuevo algoritmo basado en el filtrado subbanda, y se demuestra experimentalmente que este algoritmo mitiga las limitaciones de los algoritmos existentes. Asimismo, este algoritmo ofrece una mayor versatilidad que los algoritmos existentes, ya que se pueden utilizar configuraciones distintas en cada subbanda, como por ejemplo, diferentes longitudes de filtro o distintos conjuntos de altavoces. Por último, se estudia la influencia de las respuestas objetivo en la optimización de los filtros y se propone un nuevo método en el que se aplica una ventana temporal a estas respuestas. El método propuesto se evalúa experimentalmente en dos salas con diferentes tiempos de reverberación y los resultados obtenidos muestran que se puede reducir la energía de las interferencias entre zonas gracias al efecto de la ventana temporal.[CA] Els sistemes de zones de so personal (o les seves sigles en anglés PSZ) fan servir altaveus i tècniques de processament de senyal per a reproduir sons distints en diferents zones d'un mateix espai compartit. Aquests sistemes s'han popularitzat en els últims anys a causa de l'àmplia gamma d'aplicacions que podrien veure's beneficiades per la generació de zones d'escolta individuals. El disseny dels filtres utilitzats per a processar els senyals de so és un dels aspectes més importants dels sistemes PSZ, particularment per a les freqüències baixes i mitjanes. En la literatura s'han proposat diversos algoritmes per a calcular aquests filtres, cadascun d'ells amb els seus avantatges i inconvenients. En aquest treball es revisen els algoritmes proposats en la literatura per a sistemes PSZ i s'avalua experimentalment el seu rendiment en un entorn reverberant. Els distints algoritmes es comparen tenint en compte aspectes com l'aïllament acústic entre zones, l'error de reproducció, l'energia dels filtres i el retard del sistema. A més, s'estudien estratègies de còmput eficient per obtindre els filtres i també es comparen les seves complexitats computacionals. Els resultats experimentals obtinguts revelen que les solucions existents no poder oferir al mateix temps una complexitat computacional baixa i un bon rendiment amb latència baixa. Per això es proposa un nou algoritme basat en el filtrat subbanda que mitiga aquestes limitacions. A més, l'algoritme proposat ofereix una major versatilitat que els algoritmes existents, ja que en cada subbanda el sistema pot utilitzar configuracions diferents, com per exemple, distintes longituds de filtre o distints conjunts d'altaveus. L'algoritme proposat s'avalua experimentalment en un entorn reverberant, i es mostra com pot mitigar satisfactòriament les limitacions dels algoritmes existents. Finalment, s'estudia la influència de les respostes objectiu en l'optimització dels filtres i es proposa un nou mètode en el que s'aplica una finestra temporal a les respostes objectiu. El mètode proposat s'avalua experimentalment en dues sales amb diferents temps de reverberació i els resultats obtinguts mostren que es pot reduir el nivell d'interferència entre zones grècies a l'efecte de la finestra temporal.[EN] Personal Sound Zones (PSZ) systems deliver different sounds to a number of listeners sharing an acoustic space through the use of loudspeakers together with signal processing techniques. These systems have attracted a lot of attention in recent years because of the wide range of applications that would benefit from the generation of individual listening zones, e.g., domestic or automotive audio applications. A key aspect of PSZ systems, at least for low and mid frequencies, is the optimization of the filters used to process the sound signals. Different algorithms have been proposed in the literature for computing those filters, each exhibiting some advantages and disadvantages. In this work, the state-of-the-art algorithms for PSZ systems are reviewed, and their performance in a reverberant environment is evaluated. Aspects such as the acoustic isolation between zones, the reproduction error, the energy of the filters, and the delay of the system are considered in the evaluations. Furthermore, computationally efficient strategies to obtain the filters are studied, and their computational complexity is compared too. The performance and computational evaluations reveal the main limitations of the state-of-the-art algorithms. In particular, the existing solutions can not offer low computational complexity and at the same time good performance for short system delays. Thus, a novel algorithm based on subband filtering that mitigates these limitations is proposed for PSZ systems. In addition, the proposed algorithm offers more versatility than the existing algorithms, since different system configurations, such as different filter lengths or sets of loudspeakers, can be used in each subband. The proposed algorithm is experimentally evaluated and tested in a reverberant environment, and its efficacy to mitigate the limitations of the existing solutions is demonstrated. Finally, the effect of the target responses in the optimization is discussed, and a novel approach that is based on windowing the target responses is proposed. The proposed approach is experimentally evaluated in two rooms with different reverberation levels. The evaluation results reveal that an appropriate windowing of the target responses can reduce the interference level between zones.Molés Cases, V. (2022). Filter Optimization for Personal Sound Zones Systems [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/18611
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