No description available.

É apresentado um amplificador de potência (PA) com controle digital da potência de saída, operando na banda S de frequência (2,2 GHz). Este PA utiliza um transformador de entrada para reduzir as flutuações dos sinais de terra. Um estágio de excitação oferece uma impedância apropriada para a fonte de entrada e ganho para o próximo estágio. O estágio de controle é usado para melhorar a eficiência do PA, composto por quatro ramos paralelos de chaves, onde os estados (ligado ou desligado) são separadamente ativados por uma palavra de controle de 4 bits. O estágio de saída implementa um amplificador classe E, usando uma topologia cascode para minimizar o estresse de tensão sobre os transistores, permitindo sua utilização sob tensão de alimentação de 3,3 V para se atingir uma potência de saída máxima em torno de 1 W, em um processo CMOS 130 nm, cuja tensão típica de alimentação é 1,2 V. O PA proposto foi projetado em uma tecnologia CMOS 130 nm para RF, ocupa uma área de 1,900 x 0,875 mm2 e os resultados das simulações em leiaute extraído obtidos demonstram uma potência de saída máxima de 28,5 dBm (707 mW), com PAE (Power- Added Efficiency) correspondente de 49,7%, para uma tensão de alimentação de 3,3 V. O controle de 4 bits permite um ajuste dentro da faixa dinâmica da potência de saída entre 13,6 a 28,5 dBm (22,9 a 707 mW), divididos em 15 passos, com o PAE variando de 9,1% a 49,7%. O PA proposto permite redução do consumo de potência quando este não está transmitindo na potência máxima. A potência consumida atinge um mínimo de 0,21Wquando a potência de saída é de 13,6 dBm (22,9 mW) e um máximo de 1,4 W quando a potência de saída é de 28,5 dBm (707 mW), o que representa 1,19 W de economia, aumentando a vida da bateria. A linearidade obtida neste circuito mostrou-se suficiente para atender os requisitos da máscara de emissão de espúrios de um padrão de comunicação com envoltória constante largamente utilizado, apresentando desempenho adequado para atender as especificações dos sistemas de comunicações modernos. / A power amplifier with digital power control useful to S-Band (2.2 GHz) applications and with an output power around 1 W is presented. It uses an input transformer to reduce ground bounce effects. A tuned driver stage provides impedance matching to the input signal source and proper gain to the next stage. A control stage is used for efficiency improvement, composed by four parallel branches where the state (on or off) is separately activated by a 4-bit input. The class-E power stage uses a cascode topology to minimize the voltage stress over the power transistors, allowing higher supply voltages. The PA was designed in a 130 nm RF CMOS process and the layout has a total area of 1.900 x 0.875 mm2, post-layout simulations resulted a peak output power of 28.5 dBm with a maximum power added efficiency (PAE) around 49.7% under 3.3 V of supply voltage. The 4-bit control allows a total output power dynamic range adjustment of 14.9 dB, divided in 15 steps, with the PAE changing from 9.1% to 49.7%. The proposed PA allows reduce the power consumption when it isn’t transmitting at the maximum output power. Where the power consumption is only 0.21 W when the PA is at the minimum output power level of 13.6 dBm (22.9 mW), which is 1.19 W smaller than the power consumption at full mode (1.4 W), increasing the battery life. The linearity in this circuit meet the emission mask requirements for a widely used communication standard with constant envelope. Post-layout simulation results indicate an overall performance adequate to fulfill the specifications of modern wireless communication systems.

RF fingerprinting can solve the indoor positioning problem with satisfactory accuracy, but the methodology depends on the so-called radio map calibrated in the offline phase via manual site-survey, which is costly, time-consuming and somewhat error-prone. It also assumes the RF fingerprint’s signal-spatial correlations to remain static throughout the online positioning phase, which generally does not hold in practice. This is because indoor environments constantly experience dynamic changes, causing the radio signal strengths to fluctuate over time, which weakens the signal-spatial correlations of the RF fingerprints. State-of-the-arts have proposed adaptive RF fingerprint methodology capable of calibrating the radio map in real-time and on-demand to address these drawbacks. However, existing implementations are highly server-centric, which is less robust, does not scale well, and not privacy-friendly. This thesis aims to address these drawbacks by exploring the feasibility of implementing an adaptive RF fingerprint indoor positioning system in a distributed and client-centric architecture using only commodity Wi-Fi hardware, so it can seamlessly integrate with existing Wi-Fi network and allow it to offer both networking and positioning services. Such approach has not been explored in previous works, which forms the basis of this thesis’ main contribution. The proposed methodology utilizes a network of distributed location beacons as its reference infrastructure; hence the system is more robust since it does not have any single point-of-failure. Each location beacon periodically broadcasts its coordinate to announce its presence in the area, plus coefficients that model its real-time RSS distribution around the transmitting antenna. These coefficients are constantly self-calibrated by the location beacon using empirical RSS measurements obtained from neighbouring location beacons in a collaborative fashion, and fitting the values using path loss with log-normal shadowing model as a function of inter-beacon distances while minimizing the error in a least-squared sense. By self-modelling its RSS distribution in real-time, the location beacon becomes aware of its dynamically fluctuating signal levels caused by physical, environmental and temporal characteristics of the indoor environment. The implementation of this self-modelling feature on commodity Wi-Fi hardware is another original contribution of this thesis. Location discovery is managed locally by the clients, which means the proposed system can support unlimited number of client devices simultaneously while also protect user’s privacy because no information is shared with external parties. It starts by listening for beacon frames broadcasted by nearby location beacons and measuring their RSS values to establish the RF fingerprint of the unknown point. Next, it simulates the reference RF fingerprints of predetermined points inside the target area, effectively calibrating the site’s radio map, by computing the RSS values of all detected location beacons using their respective coordinates and path loss coefficients embedded inside the received beacon frames. Note that the coefficients model the real-time RSS distribution of each location beacon around its transmitting antenna; hence, the radio map is able to adapt itself to the dynamic fluctuations of the radio signal to maintain its signal-spatial correlations. The final step is to search the radio map to find the reference RF fingerprint that most closely resembles the unknown sample, where its coordinate is returned as the location result. One positioning approach would be to first construct a full radio map by computing the RSS of all detected location beacons at all predetermined calibration points, then followed by an exhaustive search over all reference RF fingerprints to find the best match. Generally, RF fingerprint algorithm performs better with higher number of calibration points per unit area since more locations can be classified, while extra RSS components can help to better distinguish between nearby calibration points. However, to calibrate and search many RF fingerprints will incur substantial computing costs, which is unsuitable for power and resource limited client devices. To address this challenge, this thesis introduces a novel algorithm suitable for client-centric positioning as another contribution. Given an unknown RF fingerprint to solve for location, the proposed algorithm first sorts the RSS in descending order. It then iterates over this list, first selecting the location beacon with the strongest RSS because this implies the unknown location is closest to the said location beacon. Next, it computes the beacon’s RSS using its path loss coefficients and coordinate information one calibration point at a time while simultaneously compares the result with the measured value. If they are similar, the algorithm keeps this location for subsequent processing; else it is removed because distant points relative to the unknown location would exhibit vastly different RSS values due to the different site-specific obstructions encountered by the radio signal propagation. The algorithm repeats the process by selecting the next strongest location beacon, but this time it only computes its RSS for those points identified in the previous iteration. After the last iteration completes, the average coordinate of remaining calibration points is returned as the location result. Matlab simulation shows the proposed algorithm only takes about half of the time to produce a location estimate with similar positioning accuracy compared to conventional algorithm that does a full radio map calibration and exhaustive RF fingerprint search. As part of the thesis’ contribution, a prototype of the proposed indoor positioning system is developed using only commodity Wi-Fi hardware and open-source software to evaluate its usability in real-world settings and to demonstrate possible implementation on existing Wi-Fi installations. Experimental results verify the proposed system yields consistent positioning accuracy, even in highly dynamic indoor environments and changing location beacon topologies.

Distortion product otoacoustic emissions (DPOAEs) arise within the cochlea in response to two stimulus tones (f1 and f2) at frequencies such as 2f1 − f2 and 2f2 − f1. Each DPOAE derives from two contributing mechanisms within the cochlea: a distributed distortion source and a reflection source. They are used for hearing screening, but a better understanding of their cochlear origin and transmission could potentially extend their clinical application to facilitate objective hearing loss assessment, differential diagnosis of sensorineural hearing losses and improved auditory rehabilitation using hearing aids. In this thesis a numerical model of the human cochlea is developed to study the generation of DPOAEs. It is based on a pre-existing active nonlinear model, the micromechanics of which are carefully re-tuned to simulate the response of the human cochlea to single- and two- tone stimulation. Particular attention is paid to the form and position of the nonlinearity within the model to best match experimental results. The model is also reformulated to verify its stability and ensure computational convergence of the iterative frequency domain solution method. Its predictions are validated against estimated time domain simulations and documented experimental DPOAE measurements. Additionally a novel method is developed for decomposing each frequency component of the cochlear response into forward and backward travelling waves, which is applied to investigate the multiple sources of both the 2f1 − f2 and 2f2 − f1 DPOAEs. The model is used to explain and predict a variety of phenomena observed in experimental DPOAE studies. It also confirms for the 2f1 − f2 emission, that the two source mechanisms are spatially separated and that the only significant reflection contribution is associated with the 2f1 − f2 travelling wave. In contrast, it predicts that the two source mechanisms will overlap in the case of the 2f2 − f1 DPOAE, which can be influenced by reflection of both the primary and 2f2 − f1 travelling waves.

No description available.

Normal-hearing adults have no difficulty in recognising consonants accurately, even in moderately adverse listening conditions. By contrast, users of multichannel cochlear implants have difficulty with the accurate perception of consonants, even in good listening conditions. Cochlear implant users are known to show systematic deficits in recognition of consonant features, with perception of the place feature, which relies on spectral information, being worst. These deficits may be attributed both to signal distortions introduced by the processing of the implants and to other factors, in particular the spectrotemporal distortions which occur at the interface between electrode array and auditory nervous system, including cross-channel interaction. The objective of the work reported here was to attempt to partial out the relative contribution of these different factors to consonant recognition. This was achieved by comparing cochlear implant users’ perceptual errors, analysed in terms of information transmission, with errors made by normal-hearing subjects listening to acoustic models of implant processing, in various conditions. Two initial experiments were undertaken to develop and refine an acoustic model of the Nucleus 24 cochlear implant. Findings from these two experiments informed the design of the main acoustic model experiment, which was undertaken in parallel with a further experiment involving users of the Nucleus 24 device. In both experiments, subjects listened to nonsense syllables with and without the addition of stationary background noise, in three different configurations of implant processing parameters. Additionally, in the acoustic model experiment, a simulation of cross-channel spread of excitation, or “channel interaction”, was varied. Results showed that acoustic model experiments were predictive of the pattern of consonant feature transmission in cochlear implant users with better baseline consonant recognition scores. Deficits in consonant recognition in this subgroup could be explained by the loss of phonemically relevant acoustic information in speech due to the nature of cochlear implant processing, while channel interaction appeared to play a smaller role in accounting for problems in consonant recognition. The work also evaluated the effect of changes in channel number and stimulation rate and failed to find any changes in consonant recognition as these parameters were varied. The lack of a stimulation rate effect was consistent with acoustic measurements of the temporal modulation transfer function of the processor, which showed almost no change across stimulation rates.

No description available.

It has been argued that apparent masking-level differences (MLDs) in users of bilateral bone-anchored hearing aids (BAHAs) provide evidence of binaural hearing. However, there is considerably less acoustical isolation between the two ears with bone conduction (BC) compared to air conduction (AC). The apparent MLDs may have arisen, at least in part, from inter-cranial interference between signals arising from the two BAHAs (i.e. monaural effect). That might also explain some of the inter-individual variation in both the magnitude and the direction of the MLDs reported in BAHA users. The present study was composed of three experimental stages with the main aim to investigate the influence of interference in normal hearing participants by measuring masking level difference in AC and BC to explore the conditions contributing to the reported variation. An additional aim was to investigate the performance of a newly designed BC transducer; the balanced electromagnetic separation transducer (BEST), for bone conduction research as well as more general clinical use. Stage 1 evaluated the performance of the BEST in comparison to the clinically used RadioEar B71 in a series of acoustical (sensitivity and harmonic distortion) and psychoacoustical (hearing thresholds and vibrotactile thresholds) measurements. The results from these studies led to the use of the BEST in the second and third stages because they produced significantly lower harmonic distortion at low frequencies (mainly 250 Hz). The psychoacoustic measurements alluded to the need to use different calibration values with the BESTs. Stage 2 was a preliminary investigation comparing the MLDs with standard bilateral configurations between the AC and BC in nine normal-hearing participants. Signals were pure tones at one of three frequencies (250, 500, 1000 Hz), presented via AC or BC. Broadband noise (100- 5000 Hz) was always presented via AC at 70 dB SPL. Thresholds were estimated using a three-alternative forced choice procedure combined with an adaptive staircase. Transducers used were insert earphones and the BESTs for BC testing. The results from this stage showed a statistical significant difference between AC and BC MLDs at 250, 500 and 1000 Hz (mean difference is 9.4, 6.6 and 3.5 dB respectively). Evidence of the change in the MLDs direction is observed at 250 Hz in three participants. Stage 3 consisted of the investigation of inter-cranial interference in eighteen normal hearing participants. This stage was composed of three main measurements. The first measurement compared the AC and BC MLDs at three test frequencies. The second measurement evaluated the transcranial attenuation (TA). The third measurement was the novel feature of the study it evaluated the monaural interference effect through the measurement of the diotic and dichotic conditions in one test ear. A significant discrepancy was found between the AC and BC MLDs of approximately 6, 1.5 and 2.5 dB at 500, 1000 and 2000 Hz, respectively. The TA was found to be lower than 10 dB at the three test frequencies. Measurable MTLDs were reported in some of the participants, high inter-subject variability was observed in the direction of the MTLDs. The BEST can reliably replace the B71 in clinical setup. Formal adjustment of the reference equivalent threshold force levels is advised. Binaural hearing was achieved through bilateral BC stimulation to a lesser magnitude compared to AC MLDs in normal hearing participants. The discrepancy between the AC and BC MLDs was reduced with the increase in the frequency. The discrepancy can partially be explained by the cross-talk of the signal in one ear. The results showed that in some participants the magnitude of the monaural tone level difference was similar to the magnitude of the BC MLD. Further investigation is recommended to investigate the association of the transcranial delay with the discrepancy between the AC and BC MLDs. This investigation also recommends the investigation of the AC and BC MLDs in patients fitted with bilateral BAHAs.

É apresentado um amplificador de potência (PA) com controle digital da potência de saída, operando na banda S de frequência (2,2 GHz). Este PA utiliza um transformador de entrada para reduzir as flutuações dos sinais de terra. Um estágio de excitação oferece uma impedância apropriada para a fonte de entrada e ganho para o próximo estágio. O estágio de controle é usado para melhorar a eficiência do PA, composto por quatro ramos paralelos de chaves, onde os estados (ligado ou desligado) são separadamente ativados por uma palavra de controle de 4 bits. O estágio de saída implementa um amplificador classe E, usando uma topologia cascode para minimizar o estresse de tensão sobre os transistores, permitindo sua utilização sob tensão de alimentação de 3,3 V para se atingir uma potência de saída máxima em torno de 1 W, em um processo CMOS 130 nm, cuja tensão típica de alimentação é 1,2 V. O PA proposto foi projetado em uma tecnologia CMOS 130 nm para RF, ocupa uma área de 1,900 x 0,875 mm2 e os resultados das simulações em leiaute extraído obtidos demonstram uma potência de saída máxima de 28,5 dBm (707 mW), com PAE (Power- Added Efficiency) correspondente de 49,7%, para uma tensão de alimentação de 3,3 V. O controle de 4 bits permite um ajuste dentro da faixa dinâmica da potência de saída entre 13,6 a 28,5 dBm (22,9 a 707 mW), divididos em 15 passos, com o PAE variando de 9,1% a 49,7%. O PA proposto permite redução do consumo de potência quando este não está transmitindo na potência máxima. A potência consumida atinge um mínimo de 0,21Wquando a potência de saída é de 13,6 dBm (22,9 mW) e um máximo de 1,4 W quando a potência de saída é de 28,5 dBm (707 mW), o que representa 1,19 W de economia, aumentando a vida da bateria. A linearidade obtida neste circuito mostrou-se suficiente para atender os requisitos da máscara de emissão de espúrios de um padrão de comunicação com envoltória constante largamente utilizado, apresentando desempenho adequado para atender as especificações dos sistemas de comunicações modernos. / A power amplifier with digital power control useful to S-Band (2.2 GHz) applications and with an output power around 1 W is presented. It uses an input transformer to reduce ground bounce effects. A tuned driver stage provides impedance matching to the input signal source and proper gain to the next stage. A control stage is used for efficiency improvement, composed by four parallel branches where the state (on or off) is separately activated by a 4-bit input. The class-E power stage uses a cascode topology to minimize the voltage stress over the power transistors, allowing higher supply voltages. The PA was designed in a 130 nm RF CMOS process and the layout has a total area of 1.900 x 0.875 mm2, post-layout simulations resulted a peak output power of 28.5 dBm with a maximum power added efficiency (PAE) around 49.7% under 3.3 V of supply voltage. The 4-bit control allows a total output power dynamic range adjustment of 14.9 dB, divided in 15 steps, with the PAE changing from 9.1% to 49.7%. The proposed PA allows reduce the power consumption when it isn’t transmitting at the maximum output power. Where the power consumption is only 0.21 W when the PA is at the minimum output power level of 13.6 dBm (22.9 mW), which is 1.19 W smaller than the power consumption at full mode (1.4 W), increasing the battery life. The linearity in this circuit meet the emission mask requirements for a widely used communication standard with constant envelope. Post-layout simulation results indicate an overall performance adequate to fulfill the specifications of modern wireless communication systems.

abstract: Traditional wireless communication systems operate in duplexed modes i.e. using time division duplexing or frequency division duplexing. These methods can respectively emulate full duplex mode operation or realize full duplex mode operation with decreased spectral efficiency. This thesis presents a novel method of achieving full duplex operation by actively cancelling out the transmitted signal in pseudo-real time. With appropriate hardware, the algorithms and techniques used in this work can be implemented in real time without any knowledge of the channel or any training sequence. Convergence times of down to 1 ms can be achieved which is adequate for the coherence bandwidths associated with an indoor environment. By utilizing adaptive cancellation, additional overhead for re-calibrating the system in other open-loop methods is not needed. / Dissertation/Thesis / Masters Thesis Electrical Engineering 2016