SDPLL-Based Frequency Estimation of a Sinusoid in Colored Noise

The problem of frequency estimation of a single sinusoid observed in colored noise is addressed. Our estimator is based on the operation of the sinusoidal digital phase-locked loop (SDPLL) which carries the frequency information in its phase error after the noisy sinusoid has been acquired by the SDPLL. We show by computer simulations that this frequency estimator beats the Cramer-Rao bound (CRB) on the frequency error variance for moderate and high SNRs when the colored noise has a general low-pass filtered (LPF) characteristic, thereby outperforming, in terms of frequency error variance, several existing techniques some of which are, in addition, computationally demanding. Moreover, the present approach generalizes on existing work that addresses different methods of sinusoid frequency estimation involving specific colored noise models such as the moving average (MA) noise model. An insightful theoretical analysis is presented to support the practical findings

Simulated Assessment of Interference Effects in Direct Sequence SpreadSpectrum (DSSS) QPSK Receiver

This research developed and validated a generic simulation for a direct sequence spread spectrum (DSSS), using differential phase shift keying (DPSK) and phase shift keying (PSK) modulations, providing the flexibility for assessing intentional interference effect using DSSS quadrature phase shift keying receiver (QPSK) with matched filtering as a reference. The evaluation compares a comprehensive pool of jamming waveforms at pass-band that include continuous wave (CW) interference, broad-band jamming, partial-band interference and pulsed interference. The methodology for jamming assessment included comparing the bit error rate (BER) versus required jamming to signal ratio (JSR) for different interferers using the Monte Carlo approach. This thesis also analyzes the effect of varying the jammer bandwidth for broad-band jammers including broad-band noise (BBN), frequency hopping interference (FHI), comb- spectrum interference (CSI), multi-tone jamming (MTJ), random frequency modulated interference (RFMI) and linear frequency modulated interference (LFMI). Also, the effect of changing the duty cycle for pulsed CW waveforms is compared with the worst case pulsed jamming equation. After the evaluation of different interferers, the research concludes that pulsed binary phase shift keying (BPSK) jamming is the most effective technique, whereas the CW tone jamming and CW BPSK interference result are least effective. It is also concluded that by finding an optimum bandwidth, FHI and BBN improves the required JSR by approximately 2.1 dB, RFMI and LFMI interference by 0.9 and 1.5 dB respectively. Alternately, MTJ and CSI improves their effectiveness in 4.1 dB and 3.6 dB respectively, matching the performance of the pulsed BPSK jammer

Bias Removal Approach in System Identification and Arma Spectral Estimation

Electrical Engineerin

Digital neuromorphic auditory systems

This dissertation presents several digital neuromorphic auditory systems. Neuromorphic systems are capable of running in real-time at a smaller computing cost and consume lower power than on widely available general computers. These auditory systems are considered neuromorphic as they are modelled after computational models of the mammalian auditory pathway and are capable of running on digital hardware, or more specifically on a field-programmable gate array (FPGA). The models introduced are categorised into three parts: a cochlear model, an auditory pitch model, and a functional primary auditory cortical (A1) model. The cochlear model is the primary interface of an input sound signal and transmits the 2D time-frequency representation of the sound to the pitch models as well as to the A1 model. In the pitch model, pitch information is extracted from the sound signal in the form of a fundamental frequency. From the A1 model, timbre information in the form of time-frequency envelope information of the sound signal is extracted. Since the computational auditory models mentioned above are required to be implemented on FPGAs that possess fewer computational resources than general-purpose computers, the algorithms in the models are optimised so that they fit on a single FPGA. The optimisation includes using simplified hardware-implementable signal processing algorithms. Computational resource information of each model on FPGA is extracted to understand the minimum computational resources required to run each model. This information includes the quantity of logic modules, register quantity utilised, and power consumption. Similarity comparisons are also made between the output responses of the computational auditory models on software and hardware using pure tones, chirp signals, frequency-modulated signal, moving ripple signals, and musical signals as input. The limitation of the responses of the models to musical signals at multiple intensity levels is also presented along with the use of an automatic gain control algorithm to alleviate such limitations. With real-world musical signals as their inputs, the responses of the models are also tested using classifiers – the response of the auditory pitch model is used for the classification of monophonic musical notes, and the response of the A1 model is used for the classification of musical instruments with their respective monophonic signals. Classification accuracy results are shown for model output responses on both software and hardware. With the hardware implementable auditory pitch model, the classification score stands at 100% accuracy for musical notes from the 4th and 5th octaves containing 24 classes of notes. With the hardware implementation auditory timbre model, the classification score is 92% accuracy for 12 classes musical instruments. Also presented is the difference in memory requirements of the model output responses on both software and hardware – pitch and timbre responses used for the classification exercises use 24 and 2 times less memory space for hardware than software

Frequency estimation for single-carrier and OFDM signals in communication and radar systems

Eine der klassischen Problemstellungen in der Signalverarbeitung ist die Schaetzung der Frequenz eines Signals, das von weissem Rauschen additiv ueberlagert ist. Diese bedeutende Aufgabe stellt sich in vielen verschiedenen Anwendungsbereichen wie der Kommunikationstechnik, beim Doppler-Radar, beim Radar mit synthetischer Apertur (SAR), beim Array Processing, bei Radio-Frequency-IDentification (RFID), bei Resonanz-Sensoren usw. Die Anforderungen bezueglich der Leistungsfaehigkeit des Frequenzschaetzers haengen von der Anwendung ab. Die Leistungsfaehigkeit ist dabei oft unter Beruecksichtigung der folgenden 4 Punkte definiert: i) Genauigkeit, Richtigkeit der Schaetzung, ii) Arbeitsbereich (estimation range), iii) Grenzwerte der Schaetzung (im Vergleich zu einer theoretisch moeglichen Schwelle) und iv) Komplexitaet der Implementierung. Diese Anforderungen koennen nicht unabhaengig voneinander betrachtet werden und stehen sich teilweise gegenueber. Beispielsweise erfordert die Erzielung von Ergebnissen nahe an der theoretisch moeglichen Schwelle eine hohe Komplexitaet. Ebenso kann ein Schaetz-ergebnis von hoher Genauigkeit oftmals nur fuer einen stark eingeschraenkten Arbeitsbereich erzielt werden. Die Frequenzschaetzung ist im Falle von durch Fading hervorgerufenem multiplikativem Rauschen noch herausfordernder. Es handelt sich dann um den allgemeinen Fall der Frequenzschaetzung. Bisher hat man bereits viel Arbeit in die Ableitung eines Schaetzers für diesen allgemeinen Fall investiert. Ein Schaetzer, der optimal bezueglich aller oben genannten Kriterien ist, duerfte allerdings nur schwer zu finden sein. In dieser Dissertation wird mit Blick auf Kommunikationstechnik und Radaranwendungen ein verallgemeinerter, in geschlossener Form vorliegender, Frequenzschaetzer eingefuehrt, der alle genannten Kriterien der Leistungs-faehigkeit beruecksichtigt. Die Herleitung des Schaetzers beruht auf dem Prinzip der kleinsten Fehlerquadrate fuer den nichtlinearen Fall in Verbindung mit der Abelschen partiellen Summation. Zudem werden verschiedene modifizierte Frequenzschaetzer vorgestellt, die sich fuer Faelle in denen kein Fading oder nur sehr geringes Fading auftritt, eignen.Estimating the frequency of a signal embedded in additive white Gaussian noise is one of the classical problems in signal processing. It is of fundamental importance in various applications such as in communications, Doppler radar, synthetic aperture radar (SAR), array processing, radio frequency identification (RFID), resonance sensor, etc. The requirement on the performance of the frequency estimator varies with the application. The performance is often defined using four indexes: i). estimation accuracy, ii). estimation range, iii). estimation threshold, and iv). implementation complexity. These indexes may be in contrast with each other. For example, achieving a low threshold usually implies a high complexity. Likewise, good estimation accuracy is often obtained at the price of a narrow estimation range. The estimation becomes even more difficult in the presence of fading-induced multiplicative noise which is considered to be the general case of the frequency estimation problem. There have been a lot of efforts in deriving the estimator for the general case, however, a generalized estimator that fulfills all indexes can be hardly obtained. Focusing on communications and radar applications, this thesis proposes a new generalized closed-form frequency estimator that compromises all performance indexes. The derivation of the proposed estimator relies on the nonlinear least-squares principle in conjunction with the well known summation-by-parts formula. In addition to this, several modified frequency estimators suitable for non-fading or very slow fading scenarios, are also introduced in this thesis

Residual acceleration data on IML-1: Development of a data reduction and dissemination plan

The research performed consisted of three stages: (1) identification of sensitive IML-1 experiments and sensitivity ranges by order of magnitude estimates, numerical modeling, and investigator input; (2) research and development towards reduction, supplementation, and dissemination of residual acceleration data; and (3) implementation of the plan on existing acceleration databases

Sinusoidal frequency estimation with applications to ultrasound

This thesis comprises two parts. The first part deals with single carrier and multiple-carrier based frequency estimation. The second part is concerned with the application of ultrasound using the proposed estimators and introduces a novel efficient implementation of a subspace tracking technique. In the first part, the problem of single frequency estimation is initially examined, and a hybrid single tone estimator is proposed, comprising both coarse and refined estimates. The coarse estimate of the unknown frequency is obtained using the unweighted linear prediction method, and is used to remove the frequency dependence of the signal-to-noise ratio (SNR) threshold. The SNR threshold is then further reduced via a combination of using an aver aging filter and an outlier removal scheme. Finally, a refined frequency estimate is formed using a weighted phase average technique. The hybrid estimator outperforms other recently developed estimators and is found to be independent of the underlying frequency. A second topic considered in the first part of this thesis is multiple-carrier based frequency estimation. Based on this idea, three novel velocity estimators are proposed by exploiting the velocity dependence of the backscattered carriers using synthetic data, all three proposed estimators are found to exhibit the capability of mitigating the poor high velocity performance of the conventional correlation based techniques and thereby provide usable performance beyond the conventional Nyquist velocity limit. To evaluate these methods statistically, the Cramer-Rao lower bound for the velocity estimation is derived. In the second part, the fundamentals of ultrasound are briefly reviewed. An efficient subspace tracking technique is introduced as a way to implement clutter eigenfilters, greatly reducing the computation complexity as compared to conventional eigenfilters which are based on the evaluation of the block singular value decomposition technique. Finally, the hybrid estimator and the multiple-carrier based velocity estimators proposed in the first part of the thesis are examined with realistic radio frequency data, illustrating the usefulness of these methods in solving practical problems

Robust synchronization for PSK (DVB-S2) and OFDM systems

The advent of high data rate (broadband) applications and user mobility into modern wireless communications presents new challenges for synchronization in digital receivers. These include low operating signal-to-noise ratios, wideband channel effects, Doppler effects and local oscillator instabilities. In this thesis, we investigate robust synchronization for DVB-S2 (Digital Video Broadcasting via Satellite) and OFDM (Orthogonal Frequency Division Multiplexing) systems, as these technologies are well-suited for the provision of broadband services in the satellite and terrestrial channels respectively.EThOS - Electronic Theses Online ServiceGBUnited Kingdo