Near-Instantaneously Adaptive HSDPA-Style OFDM Versus MC-CDMA Transceivers for WIFI, WIMAX, and Next-Generation Cellular Systems

Burts-by-burst (BbB) adaptive high-speed downlink packet access (HSDPA) style multicarrier systems are reviewed, identifying their most critical design aspects. These systems exhibit numerous attractive features, rendering them eminently eligible for employment in next-generation wireless systems. It is argued that BbB-adaptive or symbol-by-symbol adaptive orthogonal frequency division multiplex (OFDM) modems counteract the near instantaneous channel quality variations and hence attain an increased throughput or robustness in comparison to their fixed-mode counterparts. Although they act quite differently, various diversity techniques, such as Rake receivers and space-time block coding (STBC) are also capable of mitigating the channel quality variations in their effort to reduce the bit error ratio (BER), provided that the individual antenna elements experience independent fading. By contrast, in the presence of correlated fading imposed by shadowing or time-variant multiuser interference, the benefits of space-time coding erode and it is unrealistic to expect that a fixed-mode space-time coded system remains capable of maintaining a near-constant BER

Discrete Wavelet Transforms

The discrete wavelet transform (DWT) algorithms have a firm position in processing of signals in several areas of research and industry. As DWT provides both octave-scale frequency and spatial timing of the analyzed signal, it is constantly used to solve and treat more and more advanced problems. The present book: Discrete Wavelet Transforms: Algorithms and Applications reviews the recent progress in discrete wavelet transform algorithms and applications. The book covers a wide range of methods (e.g. lifting, shift invariance, multi-scale analysis) for constructing DWTs. The book chapters are organized into four major parts. Part I describes the progress in hardware implementations of the DWT algorithms. Applications include multitone modulation for ADSL and equalization techniques, a scalable architecture for FPGA-implementation, lifting based algorithm for VLSI implementation, comparison between DWT and FFT based OFDM and modified SPIHT codec. Part II addresses image processing algorithms such as multiresolution approach for edge detection, low bit rate image compression, low complexity implementation of CQF wavelets and compression of multi-component images. Part III focuses watermaking DWT algorithms. Finally, Part IV describes shift invariant DWTs, DC lossless property, DWT based analysis and estimation of colored noise and an application of the wavelet Galerkin method. The chapters of the present book consist of both tutorial and highly advanced material. Therefore, the book is intended to be a reference text for graduate students and researchers to obtain state-of-the-art knowledge on specific applications

Enhanced carrierless amplitude and phase modulation for optical communication systems

This thesis develops and investigates enhanced techniques for carrierless amplitude and phase modulation (CAP) in optical communication systems. The CAP scheme is studied as the physical layer modulation technique due to its implementation simplicity and versatility, that enables its implementation as a single carrier (CAP) or multi-carrier technique (m-CAP). The effect of timing jitter on the error performance of CAP is first investigated. The investigation indicates that synchronization is a critical requirement for CAP receiver and as a result, a novel low-complexity synchronization algorithm is developed with experimental demonstration for CAP-based visible light communication (VLC) systems. To further reduce the overall link complexity, a fractionally-spaced equalizer (FSE) is considered to mitigate the effects of inter-symbol interference (ISI) and timing jitter. The FSE implementation, which eliminates the need for a separate synchronization block, is shown through simulation and VLC experimental demonstration to outperform symbol-spaced equalizers (SSE) that are reported in literature for CAP-based VLC systems. Furthermore, in this thesis, spectrally-efficient index modulation techniques are developed for CAP. The proposed techniques can be divided into two broad groups, namely spatial index CAP (S-CAP) and subband index CAP (SI-CAP). The proposed spatial index techniques leverage the fact that in VLC, multiple optical sources are often required. The spatial CAP (S-CAP) transmits CAP signal through one of Nt available LEDs. It is developed to reduce equalization requirement and improve the spectral efficiency of the conventional CAP. In addition to the bits transmitted through the CAP symbol, the S-CAP encodes additional bits on the indexing/spatial location of the LEDs. The generalised S-CAP (GS-CAP) is further developed to relax the S-CAP limitation of using a single LED per symbol duration. In addition to the S-CAP scheme, multiple-input multiple-output (MIMO) techniques of repetitive-coded CAP (RC-CAP) and spatial multiplexing CAP (SMux-CAP) are investigated for CAP. Low-complexity detectors are also developed for the MIMO schemes. A key challenge of the MIMO schemes is that they suffer power penalty when channel gains are similar, which occur when the optical sources are closely located. The use of multiple receivers and power factor imbalance (PFI) techniques are proposed to mitigate this power penalty. The techniques result in significant improvement in the power efficiency of the MIMO schemes and ensure that the spectral efficiency gain is obtained with little power penalty. Finally, subband index CAP (SI-CAP) is developed to improve the spectral efficiency of m-CAP and reduce its peak-to-average power ratio (PAPR). The SI-CAP encodes additional information bits on the selection of ‘active’ subbands of m-CAP and only modulate data symbols on these ‘active’ subbands. The error performance of the proposed SI-CAP is evaluated analytically and verified with computer-based simulations. The SI-CAP technique is also experimented for both VLC and step-index plastic optical fibre (SI-POF) communication links. The experimental results show that for a fixed power efficiency, SI-CAP achieves higher data rate compared tom-CAP. For example, at a representative bit error rate (BER) of 10-5, the SI-CAP achieves a data rate and power efficiency gain of 26:5 Mb/s and 2:5 dB, respectively when compared to m-CAP. In addition, an enhanced SI-CAP (eSI-CAP) is developed to address the complexity that arises in SI-CAP at higher modulation order. The results of the experimental demonstrations in VLC and 10 m SI-POF link shows that when compared with m-CAP, eSI-CAP consistently yields a data rate improvement of between 7% and 13% for varying values of the SNR

Compensation of fibre impairments in coherent optical systems

Tese de mestrado integrado. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 201

Optimal channel equalization for filterbank transceivers in presence of white noise

Filterbank transceivers are widely employed in data communication networks to cope with inter-symbol-interference (ISI) through the use of redundancies. This dissertation studies the design of the optimal channel equalizer for both time-invariant and time-varying channels, and wide-sense stationary (WSS) and possible non-stationary white noise processes. Channel equalization is investigated via the filterbank transceivers approach. All perfect reconstruction (PR) or zero-forcing (ZF) receiver filterbanks are parameterized in an affine form, which eliminate completely the ISI. The optimal channel equalizer is designed through minimization of the mean-squared-error (MSE) between the detected signals and the transmitted signals. Our main results show that the optimal channel equalizer has the form of state estimators, and is a modified Kalman filter. The results in this dissertation are applicable to discrete wavelet multitone (DWMT) systems, multirate transmultiplexers, orthogonal frequency division multiplexing (OFDM), and direct-sequence/spread-spectrum (DS/SS) based code division multiple access (CDMA) networks. Design algorithms for the optimal channel equalizers are developed for different channel models, and white noise processes, and simulation examples are worked out to illustrate the proposed design algorithms

Channelization for Multi-Standard Software-Defined Radio Base Stations

As the number of radio standards increase and spectrum resources come under more pressure, it becomes ever less efficient to reserve bands of spectrum for exclusive use by a single radio standard. Therefore, this work focuses on channelization structures compatible with spectrum sharing among multiple wireless standards and dynamic spectrum allocation in particular. A channelizer extracts independent communication channels from a wideband signal, and is one of the most computationally expensive components in a communications receiver. This work specifically focuses on non-uniform channelizers suitable for multi-standard Software-Defined Radio (SDR) base stations in general and public mobile radio base stations in particular. A comprehensive evaluation of non-uniform channelizers (existing and developed during the course of this work) shows that parallel and recombined variants of the Generalised Discrete Fourier Transform Modulated Filter Bank (GDFT-FB) represent the best trade-off between computational load and flexibility for dynamic spectrum allocation. Nevertheless, for base station applications (with many channels) very high filter orders may be required, making the channelizers difficult to physically implement. To mitigate this problem, multi-stage filtering techniques are applied to the GDFT-FB. It is shown that these multi-stage designs can significantly reduce the filter orders and number of operations required by the GDFT-FB. An alternative approach, applying frequency response masking techniques to the GDFT-FB prototype filter design, leads to even bigger reductions in the number of coefficients, but computational load is only reduced for oversampled configurations and then not as much as for the multi-stage designs. Both techniques render the implementation of GDFT-FB based non-uniform channelizers more practical. Finally, channelization solutions for some real-world spectrum sharing use cases are developed before some final physical implementation issues are considered

Low-Complexity Multicarrier Waveform Processing Schemes fo Future Wireless Communications

Wireless communication systems deliver enormous variety of services and applications. Nowa- days, wireless communications play a key-role in many fields, such as industry, social life, education, and home automation. The growing demand for wireless services and applications has motivated the development of the next generation cellular radio access technology called fifth-generation new radio (5G-NR). The future networks are required to magnify the delivered user data rates to gigabits per second, reduce the communication latency below 1 ms, and en- able communications for massive number of simple devices. Those main features of the future networks come with new demands for the wireless communication systems, such as enhancing the efficiency of the radio spectrum use at below 6 GHz frequency bands, while supporting various services with quite different requirements for the waveform related key parameters. The current wireless systems lack the capabilities to handle those requirements. For exam- ple, the long-term evolution (LTE) employs the cyclic-prefix orthogonal frequency-division multiplexing (CP-OFDM) waveform, which has critical drawbacks in the 5G-NR context. The basic drawback of CP-OFDM waveform is the lack of spectral localization. Therefore, spectrally enhanced variants of CP-OFDM or other multicarrier waveforms with well localized spectrum should be considered. This thesis investigates spectrally enhanced CP-OFDM (E-OFDM) schemes to suppress the out-of-band (OOB) emissions, which are normally produced by CP-OFDM. Commonly, the weighted overlap-and-add (WOLA) scheme applies smooth time-domain window on the CP- OFDM waveform, providing spectrally enhanced subcarriers and reducing the OOB emissions with very low additional computational complexity. Nevertheless, the suppression perfor- mance of WOLA-OFDM is not sufficient near the active subband. Another technique is based on filtering the CP-OFDM waveform, which is referred to as F-OFDM. F-OFDM is able to provide well-localized spectrum, however, with significant increase in the computational com- plexity in the basic scheme with time-domain filters. Also filter-bank multicarrier (FBMC) waveforms are included in this study. FBMC has been widely studied as a potential post- OFDM scheme with nearly ideal subcarrier spectrum localization. However, this scheme has quite high computational complexity while being limited to uniformly distributed sub- bands. Anyway, filter-bank based waveform processing is one of the main topics of this work. Instead of traditional polyphase network (PPN) based uniform filter banks, the focus is on fast-convolution filter banks (FC-FBs), which utilize fast Fourier transform (FFT) domain processing to realize effectively filter-banks with high flexibility in terms of subcarrier bandwidths and center frequencies. FC-FBs are applied for both FBMC and F-OFDM waveform genera- tion and processing with greatly increased flexibility and significantly reduced computational complexity. This study proposes novel structures for FC-FB processing based on decomposition of the FC-FB structure consisting of forward and inverse discrete Fourier transforms (DFT and IDFT). The decomposition of multirate FC provides means of reducing the computational complexity in some important specific scenarios. A generic FC decomposition model is proposed and analyzed. This scheme is mathematically equivalent to the corresponding direct FC imple- mentation, with exactly the same performance. The benefits of the optimized decomposition structure appear mainly in communication scenarios with relatively narrow active transmis- sion band, resulting in significantly reduced computational complexity compared to the direct FC structure. The narrowband scenarios find their places in the recent 3GPP specification of cellular low- power wide-area (LPWA) access technology called narrowband internet-of-things (NB-IoT). NB-IoT aims at introducing the IoT to LTE and GSM frequency bands in coexistence with those technologies. NB-IoT uses CP-OFDM based waveforms with parameters compatible with the LTE. However, additional means are needed also for NB-IoT transmitters to improve the spec- trum localization. For NB-IoT user devices, it is important to consider ultra-low complexity solutions, and a look-up table (LUT) based approach is proposed to implement NB-IoT uplink transmitters with filtered waveforms. This approach provides completely multiplication-free digital baseband implementations and the addition rates are similar or smaller than in the basic NB-IoT waveform generation without the needed elements for spectrum enhancement. The basic idea includes storing full or partial waveforms for all possible data symbol combinations. Then the transmitted waveform is composed through summation of needed stored partial waveforms and trivial phase rotations. The LUT based scheme is developed with different vari- ants tackling practical implementations issues of NB-IoT device transmitters, considering also the effects of nonlinear power amplifier. Moreover, a completely multiplication and addition- free LUT variant is proposed and found to be feasible for very narrowband transmission, with up to 3 subcarriers. The finite-wordlength performance of LUT variants is evaluated through simulations

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