Millions of people across the world currently have noise induced hearing loss, and many are working in conditions with both continuous Gaussian and non-Gaussian noises that could affect their hearing. It was hypothesized that the energy of the noise was the cause of the hearing loss and did not depend on temporal pattern of a noise. This was referred to as the equal energy hypothesis. This hypothesis has been shown to have limitations though. This means that there is a difference in the types of noise a person receives to induce hearing loss and it is necessary to build a system that can easily mimic various conditions to conduct research. This study builds a system that can produce both non-Gaussian impulse/impact noises and continuous Gaussian noise. It was found that the peak sound pressure level of the system could reach well above the needed 120 dB level to represent acoustic trauma and could replicate well above the 85 dB A-weighted sound pressure level to produce conditions of gradual developing hearing loss. The system reached a maximum of 150 dB sound peak pressure level and a maximum of 133 dB A-weighted sound pressure level. Various parameters could easily be adjusted to control the sound, such as the high and low cutoff frequency to center the sound at 4 kHz. The system build can easily be adjusted to create numerous sound conditions and will hopefully be modified and improved in hopes of eventually being used for animal studies to lead to the creation of a method to treat or prevent noise induced hearing loss.

No / Noise amplitude distribution measurements relevant to%satellite-mobile radio systems are reported. The rationale for the%measurements is outlined and the choice of measurement parameters%justified. The measurement equipment and measurement methodology are%described in detail. Results characterizing the elevation angle%distribution of impulsive noise are presented for rural, suburban and%urban environments and also for an arterial road (U.K. motorway)%carrying high density, fast moving traffic. Measurements of the levels%of impulsive noise to be expected in each environment for high- and%low-elevation satellite scenarios using appropriate antenna%configurations are also presented

During the acquisition and transmission of images, it is important that the information is retained with the highest quality. Occasionally noise from various sources can corrupt an image. In this case, various image processing filtering techniques that are effective in removing noise can be used. Noise removal filters are designed to remove specific types of noise. However, they also degrade the image and detail is lost. This is particularly troublesome in cases where there is very little noise. A number of detail preserving filters have been proposed in the past, but most of the work was focused on monochrome images. With today's technological advancements, most of the images used are in full color, and detail preserving filters that have been designed for monochrome, cannot be directly applied for color. In this work, several detail preserving filters that have been designed to remove impulsive noise from color images are first surveyed and compared. We then consider the extension of detail preserving median filters to color images. At the end, we propose new filters that are capable of removing colored impulsive noise with minimum image degradation. Two user adjustable parameters can vary the strength of the filter for best results. Several comparisons with other previously proposed filters will be presented. Finally, in order to make the filter user friendly, an Adobe Photoshop plug-in is to be developed.

Personnel who work on the flight deck of an aircraft carrier are exposed to extreme levels of jet engine noise often in excess of 140 decibels (dB). The current circumaural hearing protective devices (CAHPD) employed by flight deck crewmen are inadequate for the level of protection required for these extreme levels of noise. Fiber-reinforced thermoset polymer composite (FRPC) materials such as aramid fibers used in body armor, have high theoretical values of acoustic impedance due to a fundamentally high modulus of elasticity and may offer a superior level of hearing protection over original equipment (OE) thermoplastic CAHPDs. The objective of this project was to measure and evaluate the attenuation of CAHPD’s constructed from FRPC materials. FRPC CAHPD ear cups were paired with OE thermoplastic CAHPD ear cups of equal shape and thickness, and the protected and unprotected A-weighted sound pressure level (SPL) was measured in continuous and impulse noise environments >80 dBA using a JOLENE manikin. These data were evaluated for paired differences between the protected and unprotected mean SPL, and OE protected and FRPC protected mean SPL and indicates that OE thermoplastic CAHPDs provide greater sound attenuation of continuous noise >80 dBA and aramid FRPC CAHPDs provide greater sound attenuation of impulse noise >80 dBA.

Power line communication has been around in past decades and gained renewed attention thanks to the demand of high-speed Internet access. With the significant advantages of existing infrastructure and accessibility to even remote areas, power grid has become one of the promising competitors for multi-media transmission in household. However, the power line was not oriented for data transmission providing a rather hash environment. To overcome the difficulties, advanced modulation and channel coding schemes should be employed. In the thesis low density parity check code (LDPC) is employed to reduce the loss caused by various kinds of effects in the channel especially the noise since its performance approaches to Shannon capacity limit. Moreover, OFDM multi-carrier transmission technique is involved which could decrease the inter-symbol interference and frequency selective fading. Nevertheless, LDPC decoding process was designed specifically for the common Gaussian white noise condition, combined with OFDM modulation the system still could not provide satisfying and practicable performance so improvements are needed for the system. The main works of the thesis are as follows. Set up an environment of power line transmission investigating and simulating the channel characteristics; employ multi-path channel model and Class‐A noise model for further developing the improvement algorithms to deal with the selective fading and impulse noise. Two algorithms proposed here are from different perspectives: the first one is modifying initial posterior information for LDPC decoding and the second one aims at suppressing the impulse noise after demodulation. Finally, a few simulations are performed to reveal the effectiveness of proposed methods. As a result, the improved scheme shows a great superiority improving the performance by no less than 5dB compared to traditional system.

Millions of people across the world are suffering from noise induced hearing loss (NIHL), especially under working conditions of either continuous Gaussian or non-Gaussian noise that might affect human's hearing function. Impulse noise is a typical non-Gaussian noise exposure in military and industry, and generates severe hearing loss problem. This study mainly focuses on characterization of impulse noise using digital signal analysis method and prediction of the auditory hazard of impulse noise induced hearing loss by the Auditory Hazard Assessment Algorithm for Humans (AHAAH) modeling. A digital noise exposure system has been developed to produce impulse noises with peak sound pressure level (SPL) up to 160 dB. The characterization of impulse noise generated by the system has been investigated and analyzed in both time and frequency domains. Furthermore, the effects of key parameters of impulse noise on auditory risk unit (ARU) are investigated using both simulated and experimental measured impulse noise signals in the AHAAH model. The results showed that the ARUs increased monotonically with the peak pressure (both P+ and P-) increasing. With increasing of the time duration, the ARUs increased first and then decreased, and the peak of ARUs appeared at about t = 0.2 ms (for both t+ and t-). In addition, the auditory hazard of experimental measured impulse noises signals demonstrated a monotonically increasing relationship between ARUs and system voltages.

Despite our understanding of the impact of noise-induced damage to the auditory system, much less is known about the impact of noise exposure on the vestibular system. In this article, we review the anatomical, physiological, and functional evidence for noise-induced damage to peripheral and central vestibular structures. Morphological studies in several animal models have demonstrated cellular damage throughout the peripheral vestibular system and particularly in the otolith organs; however, there is a paucity of data on the effect of noise exposure on human vestibular end organs. Physiological studies have corroborated morphological studies by demonstrating disruption across vestibular pathways with otolith-mediated pathways impacted more than semicircular canal-mediated pathways. Similar to the temporary threshold shifts observed in the auditory system, physiological studies in animals have suggested a capacity for recovery following noise-induced vestibular damage. Human studies have demonstrated that diminished sacculo-collic responses are related to the severity of noise-induced hearing loss, and dose-dependent vestibular deficits following noise exposure have been corroborated in animal models. Further work is needed to better understand the physiological and functional consequences of noise-induced vestibular impairment in animals and humans.

Fusion frames and block sparse recovery are of interest in signal processing and communication applications. In these applications it is required that the fusion frame have some desirable properties. One such requirement is that the fusion frame be tight and its subspaces form an optimal packing in a Grassmannian manifold. Such fusion frames are called Grassmannian fusion frames. Grassmannian frames are known to be optimal dictionaries for sparse recovery as they have minimum coherence. By analogy Grassmannian fusion frames are potential candidates as optimal dictionaries in block sparse processing. The present work intends to study fusion frames in finite dimensional vector spaces assuming a specific structure useful in block sparse signal processing. The main focus of our work is the design of Grassmannian fusion frames and their implication in block sparse recovery. We will consider burst error correction as an application of block sparsity and fusion frame concepts. We propose two new algebraic methods for designing Grassmannian fusion frames. The first method involves use of Fourier matrix and difference sets to obtain a partial Fourier matrix which forms a Grassmannian fusion frame. This fusion frame has a specific structure and the parameters of the fusion frame are determined by the type of difference set used. The second method involves constructing Grassmannian fusion frames from Grassmannian frames which meet the Welch bound. This method uses existing constructions of optimal Grassmannian frames. The method, while fairly general, requires that the dimension of the vector space be divisible by the dimension of the subspaces. A lower bound which is an analog of the Welch bound is derived for the block coherence of dictionaries along with conditions to be satisfied to meet the bound. From these results we conclude that the matrices constructed by us are optimal for block sparse recovery from block coherence viewpoint. There is a strong relation between sparse signal processing and error control coding. It is known that burst errors are block sparse in nature. So, here we attempt to solve the burst error correction problem using block sparse signal recovery methods. The use of Grassmannian fusion frames which we constructed as optimal dictionary allows correction of maximum possible number of errors, when used in conjunction with reconstruction algorithms which exploit block sparsity. We also suggest a modification to improve the applicability of the technique and point out relationship with a method which appeared previously in literature. As an application example, we consider the use of the burst error correction technique for impulse noise cancelation in OFDM system. Impulse noise is bursty in nature and severely degrades OFDM performance. The Grassmannian fusion frames constructed with Fourier matrix and difference sets is ideal for use in this application as it can be easily incorporated into the OFDM system.

Les schémas de Codage Vidéo Linéaire (CVL) inspirés de SoftCast ont émergé dans la dernière décennie comme une alternative aux schémas de codage vidéo classiques. Ces schémas de codage source-canal conjoint exploitent des résultats théoriques montrant qu’une transmission (quasi-)analogique est plus performante dans des situations de multicast que des schémas numériques lorsque les rapports signal-à-bruit des canaux (C-SNR) diffèrent d’un récepteur à l’autre. Dans ce contexte, les schémas de CVL permettent d’obtenir une qualité de vidéo décodée proportionnelle au C-SNR du récepteur.Une première contribution de cette thèse concerne l’optimisation de la matrice de précodage de canal pour une transmission de type OFDM de flux générés par un CVL lorsque les contraintes de puissance diffèrent d’un sous-canal à l’autre. Ce type de contrainte apparait en sur des canaux DSL, ou dans des dispositifs de transmission sur courant porteur en ligne (CPL). Cette thèse propose une solution optimale à ce problème de type multi-level water filling et nécessitant la solution d’un problème de type Structured Hermitian Inverse Eigenvalue. Trois algorithmes sous-optimaux de complexité réduite sont également proposés. Des nombreux résultats de simulation montrent que les algorithmes sous-optimaux ont des performances très proches de l’optimum et réduisent significativement le temps de codage. Le calcul de la matrice de précodage dans une situation de multicast est également abordé. Une seconde contribution principale consiste en la réduction de l’impact du bruit impulsif dans les CVL. Le problème de correction du bruit impulsif est formulé comme un problème d’estimation d’un vecteur creux. Un algorithme de type Fast Bayesian Matching Pursuit (FBMP) est adapté au contexte CVL. Cette approche nécessite de réserver des sous-canaux pour la correction du bruit impulsif, entrainant une diminution de la qualité vidéo en l'absence de bruit impulsif. Un modèle phénoménologique (MP) est proposé pour décrire l’erreur résiduelle après correction du bruit impulsif. Ce modèle permet de d’optimiser le nombre de sous-canaux à réserver en fonction des caractéristiques du bruit impulsif. Les résultats de simulation montrent que le schéma proposé améliore considérablement les performances lorsque le flux CVL est transmis sur un canal sujet à du bruit impulsif. / SoftCast based Linear Video Coding (LVC) schemes have been emerged in the last decade as a quasi analog joint-source-channel alternative to classical video coding schemes. Theoretical analyses have shown that analog coding is better than digital coding in a multicast scenario when the channel signal-to-noise ratios (C-SNR) differ among receivers. LVC schemes provide in such context a decoded video quality at different receivers proportional to their C-SNR.This thesis considers first the channel precoding and decoding matrix design problem for LVC schemes under a per-subchannel power constraint. Such constraint is found, e.g., on Power Line Telecommunication (PLT) channels and is similar to per-antenna power constraints in multi-antenna transmission system. An optimal design approach is proposed, involving a multi-level water filling algorithm and the solution of a structured Hermitian Inverse Eigenvalue problem. Three lower-complexity alternative suboptimal algorithms are also proposed. Extensive experiments show that the suboptimal algorithms perform closely to the optimal one and can reduce significantly the complexity. The precoding matrix design in multicast situations also has been considered.A second main contribution consists in an impulse noise mitigation approach for LVC schemes. Impulse noise identification and correction can be formulated as a sparse vector recovery problem. A Fast Bayesian Matching Pursuit (FBMP) algorithm is adapted to LVC schemes. Subchannels provisioning for impulse noise mitigation is necessary, leading to a nominal video quality decrease in absence of impulse noise. A phenomenological model (PM) is proposed to describe the impulse noise correction residual. Using the PM model, an algorithm to evaluate the optimal number of subchannels to provision is proposed. Simulation results show that the proposed algorithms significantly improve the video quality when transmitted over channels prone to impulse noise.

This work is focused on the subject of the interference modelling for xDSL technologies. First, the xDSL technologies are explained. Following is the presentation and description of the different kinds of the xDSL technologies. The next part deals with the basic parameters of metallic cable lines – especially the primary and secondary parameters. Nowadays wider bandwidths are used for the achievement of higher data transmission rates. During a higher frequency signal transmission a more intensive line attenuation appears. To identify the transfer characteristics of the lines while using an xDSL system, mathematic models of transmission lines are applied. That is why these mathematic models are dealt with in the next chapter. At the end of this section the mathematic models are compared using the modular and phase characteristics. The main aim of the work is to describe the different impacts which influence the efficiency of the xDSL systems. First, the causes interfering from the inside of the cable are deeply explained: Near End Crosstalk (NEXT), Far End Crosstalk (FEXT), Additive White Gaussian Noise (AWGN). Following is the explanation of the external interfering impacts: Radio Frequency Interference (RFI) and Impulse Noise. The next goal of this thesis is a design of a workstation for the tests of spectral features and the efficiency of the xDSL systems. The work also presents a designed GUI application and its description. The GUI application is an instrument for the choice or data entry of the final interference. The last chapter describes a realization of a measurement and shows the measured characteristics which were recorded on the ADSL tester and oscilloscope.