The major application of active noise control (ANC) has been focused on using a single reference signal; the work on multiple reference ANC is very scarce. Here, the behavior of multiple reference ANC is analyzed in both the frequency and time domain, and the coherence functions are provided to evaluate the effectiveness of multiple reference ANC. When there are multiple noise sources, multiple reference sensors are needed to generate complete reference signals. A simplified method combines those signals from multiple reference sensors into a single reference signal. Although this method could result in satisfactory noise control effects under special circumstances, the performance is generally compromised. A widely adopted method feeds each reference signal into a different control filter. This approach suffers from the problem of ill-conditioning when the reference signals are correlated. The problem of ill-conditioning results in slow convergence rate and high sensitivity to measurement error especially when the FXLMS algorithm is applied. To handle this particular problem, the decorrelated Filtered-X LMS (DFXLMS) algorithm is developed and studied in this thesis. Both simulations and experiments have been conducted to verify the DFXLMS algorithm and other issues associated with multiple reference ANC. The results presented herein are consistent with the theoretical analysis, and favorably indicate that the DFXLMS algorithm is effective in improving the convergence speed. To take the maximum advantage of the TMS320C30 DSP board used to implement the controller, several DSP programming issues are discussed, and assembly routines are given in the appendix. Furthermore, a graphical user interface (GUI) running under Windows' environment is introduced. The main purpose of the GUI is to facilitate parameters modification, real time data monitoring and DSP process control. / Master of Science

Brainstem Auditory Evoked Potentials (BAEP) contain valuable information about the condition of the neural fibers associated with the auditory pathways. Extraction of this information is a difficult task due to contamination by on-going scalp EEG. This thesis reviews the current processing techniques and introduces adaptive noise cancellation (ANC) using systolic arrays as an alternative to existing technology. Q-R decomposition theory is reviewed and an explanation of the mechanics of systolic adaptive noise cancellation (SANC) is presented. A modified Given's rotation algorithm is derived resulting in a saving of up to 2/3 in memory requirements. Real data were collected in the laboratory. Real and simulated data were processed to determine the characteristics and effectiveness of adaptive noise cancellation strategies. Successful ANC of BAEP was performed on simulated data using a number or signal-to-noise ratios (S/N), data sequence lengths, reference signals and filter parameter values. We conclude that systolic arrays are a very powerful and appropriate technique for the extraction or BAEPs. Correlation studies indicated that the pre-stimulus EEG signal is inadequately correlated to the primary signal for successful ANC or BAEP in real data. A multi-channel collection scheme is outlined for future collection or Evoked Potential data. A summary or experimental results is presented to address the problem or data collection and signal processing optimization. / Thesis / Master of Engineering (MEngr)

It is well known that common-mode (CM) conducted electromagnetic interference (EMI) is caused by the common-mode current flowing through the parasitic capacitance of transistors, diodes, and transformers to ground in the power circuit. Because of the potential for interference with other systems as well as governmental regulations, it is necessary to attenuate this noise. Ordinarily this must be accomplished by using a magnetic choke on the input power lines, which can result in large penalties to the overall size, weight, and cost of the completed system. In order to lessen the requirement for this magnetic choke, there has been in recent years a desire to introduce noise cancellation techniques to the area of EMI. This text introduces a method of canceling the common-mode EMI by using a compensating transformer winding and a capacitor. Compared with active cancellation techniques, it is much simpler and requires no additional transistors and gate-drive circuitry since it merely adds a small copper winding and a small capacitor. By using this technique the size of the EMI filter can be reduced, especially for applications requiring high currents. In this thesis a survey of CM noise reduction techniques is presented, encompassing conventional and active cancellation techniques. The new method for passive noise cancellation is presented, which is then applied to families of isolated DC/DC converters, non-isolated DC/DC converters, and DC/AC inverters and motor drives. The method, results, and ramifications of this technique are presented in order of appearance. / Master of Science

Use of mobile equipment has increased exponentially over the last decade. As use becomes more widespread so too does the demand for new functionalities. The limited memory and computational power of many mobile devices has proven to be a challenge resulting in many innovative solutions and a number of new standards. Despite this, there is often a requirement for additional enhancement to improve quality. The focus of this thesis work has been to perform enhancement within two different areas; audio or speech encoding and video encoding/decoding. The audio enhancement section of this thesis addresses the well known problem in the GSM system with an interfering signal generated by the switching nature of TDMA cellular telephony. Two different solutions are given to suppress such interference internally in the mobile handset. The first method involves the use of subtractive noise cancellation employing correlators, the second uses a structure of IIR noth filters. Both solutions use control algorithms based on the state of the communication between the mobile handset and the base station. The video section of this thesis presents two post-filters and one pre-filter. The two post-filters are designed to improve visual quality of highly compressed video streams from standard, block-based video codecs by combating both blocking and ringing artifacts. The second post-filter also performs sharpening. The pre-filter is designed to increase the coding efficiency of a standard block based video codec. By introducing a pre-processing algorithm before the encoder, the amount of camera disturbance and the complexity of the sequence can be decreased, thereby increasing coding efficiency.

A wireless, battery operated pulse oximeter system with a forehead mounted optical sensor was designed in our laboratory. This wireless pulse oximeter (WPO) would enable field medics to monitor arterial oxygen saturation (SpO2) and heart rate (HR) information accurately following injuries, thereby help to prioritize life saving medical interventions when resources are limited. Pulse oximeters developed for field-based applications must be resistant to motion artifacts since motion artifacts degrade the signal quality of the photoplethysmographic (PPG) signals from which measurements are derived. This study was undertaken to investigate if accelerometer-based adaptive noise cancellation (ANC) can be used to reduce SpO2 and HR errors induced by motion artifacts typically encountered during field applications. Preliminary studies conducted offline showed that ANC can minimize SpO2 and HR errors during jogging, running, and staircase climbing. An 8th order LMS filter with ì = 0.01 was successfully implemented in the WPO's embedded microcontroller. After real-time adaptive filtering of motion corrupted PPG signals, errors for HR values ranging between 60 - 180BPM were reduced from 12BPM to 6BPM. Similarly, ambient breathing SpO2 errors were reduced from 5% to 2%.

"A wireless wearable battery-operated pulse oximeter has been developed in our laboratory for field triage applications. The wearable pulse oximeter, which uses a forehead-mounted sensor to provide arterial oxygen saturation (SpO2) and heart rate (HR) information, would enable field medics to monitor vital physiological information following critical injuries, thereby helping to prioritize life saving medical interventions. This study was undertaken to investigate if accelerometry (ACC)-based adaptive noise cancellation (ANC) is effective in minimizing SpO2 and HR errors induced during jogging to simulate certain motion artifacts expected to occur in the field. Preliminary tests confirmed that processing the motion corrupted photoplethysmographic (PPG) signals by simple Least-Mean-Square (LMS) and Recursive Least-Squares (RLS) ANC algorithms can help to improve the signal-to-noise ratio of motion-corrupted PPG signals, thereby reducing SpO2 and HR errors during jogging. The study showed also that the degree of improvement depends on filter order. In addition, we found that it would be more feasible to implement an LMS adaptive filter within an embedded microcontroller environment since the LMS algorithm requires significantly less operations."

Because of the need to move large amounts of data effienciently, optical based communications are a critical component of modern telecommunications. And as a key enabler of optical communications, electrical components play a critical role in optical data links. Optoelectronic integrated circuits provide the bridge between the optical and electrical realms. Electronic integrated circuits are also integral parts of the optical link, interfacing with post processing circuitry and compensating for any limitations along the link. In this investigation, three circuits for optical data link applications are studied. Two optoelectronic integrated circuit front-ends for freespace and long haul applications, respectively and an active filter for near end cross talk cancellation associated with high data rate transmission. The first circuit is an 8x8 monolithic receiver array for a Spatial Division Multiplexing optical link. A compact and low power 8x8 array was designed and demonstrated a channel that received data at rates of 1Gb/s. It is the first completely monolithic demonstration of a 2D receiver array within a conventional ion implanted GaAs MESFET process. The second circuit demonstrated a long wavelength (1.55 m) optoelectronic receiver for long haul applications. The circuit utilized a TWA topology, which maximizes the available bandwidth from the GaAs MESFET process. It incorporated a thin-film inverted MSM photodetector to achieve nearly monolithic integration. The final circuit is a tunable high pass active filter in 0.18 m CMOS technology. As part of a NEXT noise canceller architecture, it will provide the means to extend data transmission in FR-4 legacy backplanes into the tens of Gb/s datarate.

This thesis presents a feedback design for an active noise cancellation (ANC) headphone. The designed ANC headphone consists of an analog controller, an audio power amplifier, a headphone speaker, a mini microphone, and a microphone amplifier, which constitute a feedback loop. The controller design follows the method of feedback sensitivity shaping with degree constraint introduced by R. Nagamune and A. Blomqvist in 2005. The advantage of this method is that it eliminates the needs for choosing an analytic weighting function and performing model reduction to yield a lower-order controller, as commonly required in conventional H2/H¡Û optimizations. A fifth-order analog controller for the ANC headphone is designed. The experimental result shows a maximum acoustic noise reduction of 19.7 dB near 200 Hz and an overall noise reduction of more than 10 dB in the control frequency band from 107 Hz to 523 Hz. Moreover, the out-of-band noise amplification is limited to a barely noticeable level of 4.26 dB.

Η παρούσα διπλωματική εργασία αποτελεί βιβλιογραφική έρευνα στο επιστημονικό πεδίο της προσαρμοστικής ακύρωσης θορύβου (ANC). Αρχικά γίνεται μια αναφορά στη γενική φιλοσοφία της ANC, παρατίθεται μια σύντομη ιστορική αναδρομή και αναφέρονται τα πεδία εφαρμογής της ANC. Γίνεται ανάλυση των βασικών κατηγοριών προσαρμοστικής ακύρωσης θορύβου βασισμένες στον εμπροσθοτροφοδοτούμενο και στον ανατροφοδοτούμενο έλεγχο. Πρώτα συζητείται η δομή του ευρυζωνικού προσαρμοστικού εμπροσθοτροφοδοτούμενου ελέγχου, με την εξαγωγή και την ανάλυση του αλγόριθμου FXLMS. Στη συνέχεια αναλύονται τα στενής ζώνης εμπροσθοτροφοδοτούμενα συστήματα, εισάγοντας τη μέθοδο σύνθεσης κυματομορφής, τα προσαρμοστικά φίλτρα αποκοπής και την ANC πολλαπλών συχνοτήτων. Έπειτα αναπτύσσεται η έννοια της προσαρμοστικής ακύρωσης μέσω ανατροφοδοτούμενου έλεγχου από τη σκοπιά των σημάτων αναφοράς προσδίδοντας μια συσχέτιση με τα συστήματα εμπροσθοτροφοδότησης. Εν συνεχεία, η ανάλυση των μονοκαναλικών συστημάτων επεκτείνεται στα πολυκαναλικά συστήματα ANC. Παρουσιάζονται διάφορες online τεχνικές μοντελοποίησης δευτερεύουσας διαδρομής. Επιπρόσθετα, παρουσιάζονται διάφοροι ειδικοί ANC αλγόριθμοι όπως δικτυωτή ANC, ANC στο πεδίο της συχνότητας, ANC υποζώνης και ο αναδρομικός αλγόριθμος ελαχίστων τετραγώνων (RLS). Τέλος παρουσιάζονται αναλυτικά οι εφαρμογές της προσαρμοστικής ακύρωσης σε πρακτικό και πειραματικό επίπεδο, ενώ δίνεται μια ποσοτική συνεισφορά στη μείωση θορύβου. / This thesis is a literature research in the scientific field of adaptive noise cancellation (ANC). Originally becomes a reference to the general philosophy of the ANC, given a brief historical overview and a reference to the fields of application of ANC . An analysis of the main categories of adaptive noise cancellation based on feed-forward and feedback control. First a discussion is made, of the structure of broadband adaptive feed-forward control by extracting and analyzing the FXLMS algorithm. Then an analysis of narrowband feedforward systems is given, introducing the waveform synthesis method, adaptive notch filters and multiple frequency ANC. Then we develope the concept of adaptive cancellation via feedback control from the perspective of reference signals giving a correlation to feedforward systems. Subsequently, the analysis of single channel systems extends in multi-channel ANC. Various online secondary path modeling techniques are presented. Additionally, several special ANC algorithms are presented, such as lattice ANC, ANC in the frequency domain, subband ANC and the recursive least squares algorithm (RLS). Finally applications of adaptive cancellation are presented in detail, on practical and experimental level, given a qualitative contribution to noise reduction.

Content removed due to copyright: Conference proceedings (I) J. Jeong, and T.J. Moir, "Kepstrum approach to real-time speech enhancement methods using two microphones", Proceedings of the International Conference on Sensing Technology (ICST), pp 691-695, November 21-23, 2005, Palmerston North, New Zealand Conference proceedings (II) J. Jeong and T. J. Moir, "Two-microphone kepstrum approach to real-time speech enhancement methods" Proceedings of the IEEE International Conference on Engineering of Intelligent Systems (ICEIS), pp 392-397, April 22-23, 2006, Islamabad, Pakistan Conference proceedings (III) T. J. Moir and J. Jeong, "Identification of non-minimum phase transfer function components" Proceedings of the IEEE International Symposium on Signal Processing and Information Technology (ISSPIT), pp 380-384, August 27-30, 2006, Vancouver, Canada / This research is mainly concerned with a robust method for an improved performance of a real-time speech enhancement and noise cancellation in a real reverberant environment. Therefore, the thesis titled, "A Kepstrum Approach to Real-Time Speech Enhancement" presents an application technique of a kepstrum method to a speech enhancement method. The kepstrum approach is based on a fundamental theory of kepstrum analysis, which gives a mathematical construct to the application of a speech enhancement. kepstrum analysis is applied to the system identification application of unknown acoustic transfer functions between two microphones. This kepstrum method provides a mathematical representation with FFT based processing and is independent of acoustic path model order. The front-end application of the kepstrum method to speech enhancement methods provides an improved performance in speech enhancement and noise cancellation with several favourable effects.