The research described herein was undertaken to determine how masked thresholds changed when individuals wore an active noise reduction (ANR) hearing protection device (HPD), a passive HPD, or no HPD. An ANR earmuff, a passive earmuff, and a user-molded foam earplug were tested in two types of noises (pink and red) at two different noise levels (85 dBA and 100 dBA). The signal used was an industry-standard backup alarm. The experimental design was completely within-subjects. An ascending method of limits was used to obtain 15-20 correct positive responses, which were then averaged to obtain the masked thresholds for each treatment condition. A visual probability monitoring task was incorporated in the experimental design to provide a loading task for the participants. In addition to masked thresholds, comfort and mental workload were assessed. Finally, participants were asked to rank each of the three HPDs with respect to their perceived ability to facilitate hearing the signal in noise. Results indicated that in 85 dBA noise, masked thresholds were lower when hearing protection devices were worn, compared to the unoccluded condition. Additionally, the results indicated that the ANR device provided a significant advantage (lower masked thresholds) over the passive earmuff in the low-frequency biased red noise (across both noise levels) and the 100 dBA noise level (across both noise spectra). However, the ANR earmuff exhibited no significant advantage over the user-molded foam earplug in any of the conditions. Rather, the user-molded foam earplug produced significantly lower masked thresholds at 100 dBA. The results also indicated that there was no difference between the three devices in their perceived ability to facilitate detection of the signal. There was also not a significant difference in comfort ratings between the three HPDs, although there were several complaints about the comfort of the ANR earmuff during the experiment. / Master of Science

Speech processing in the automotive environment is a challenging problem due to the presence of powerful and unpredictable nonstationary noise. This thesis addresses two detection problems involving both nonstationary noise signals and nonstationary desired signals. Two detectors are developed: one to detect passing vehicle noise in the presence of speech and one to detect speech in the presence of passing vehicle noise. The latter is then measured against a state-of-the-art voice activity detector used in telephony. The process of compiling a library of recordings in the automobile to facilitate this research is also detailed.

No description available.

Scale modeling has been commonly used for architectural acoustics but use in other noise control areas is nominal. Acoustic scale modeling theory is first reviewed and then feasibility for small-scale applications, such as is common in the electronics industry, is investigated. Three application cases are used to examine the viability. In the first example, a scale model is used to determine the insertion loss of a rectangular barrier. In the second example, the transmission loss through parallel tubes drilled through a cylinder is measured and results are compared to a 2.85 times scale model with good agreement. The third example is a rectangular cuboid with a smaller cylindrical well bored into it. A point source is placed above the cuboid. The transfer function was measured between positions on the top of the cylinder and inside of the cylindrical well. Treatments were then applied sequentially including a cylindrical barrier around the well, a membrane cover over the opening, and a layer of sound absorption over the well. Results are compared between the full scale and a 5.7 times scale model and correlation between the two is satisfactory.

Mufflers and silencers are commonly used to attenuate noise sources such as internal combustion engines and HVAC systems. Typically, these environments contain mean flow that can affect the acoustic properties of the muffler components and may produce flow generated noise. To characterize the muffler performance, common metrics such as insertion and transmission loss and noise reduction are used in industry. Though transmission loss without flow is often measured and is a relatively simple bench top experiment and useful for model validation purposes, mean flow can significantly affect the muffler performance. There are a few existing and commercial transmission loss rigs that incorporate flow into the measurement procedure. These rigs are useful for model verification including flow but do not predict how the muffler will perform in the system since the source, termination, and pipe lengths significantly impact performance. In this research, the development of an insertion loss test rig is detailed. This testing strategy has the advantage of being simpler, quantifying the self-generated noise due to flow, and taking into account the effect of tailpipe length and a realistic termination. However, the test does not include the actual source and is not as useful for model validation. An electric blower produces the flow and a silencer quiets the flow. Loudspeakers are positioned just downstream of the flow silencer and they are used as the sound source. The low frequency source is a subwoofer installed in a cylindrical enclosure that includes a conical transition from speaker to pipe. Special care is taken to reduce any flow generated noise. Qualification of the system is detailed by comparing the measured transmission loss, noise reduction, and insertion loss to one-dimensional plane wave models. The results demonstrate that the developed rig should be useful as a muffler evaluation tool after a prototype has been constructed. The rig can also be used for transmission loss and noise reduction determination which will prove beneficial for laboratory testing.

Speech is a fundamental means of human communication. In the last several decades, much effort has been devoted to the efficient transmission and storage of speech signals. With advances in technology making mobile communication ubiquitous, communications anywhere has become a reality. The freedom and flexibility offered by mobile technology brings with it new challenges, one of which is robustness to acoustic background noise. Speech enhancement systems form a vital front-end for mobile telephony in noisy environments such as in cars, cafeterias, subway stations, etc., in hearing aids, and to improve the performance of speech recognition systems. In this thesis, which consists of four research articles, we discuss both single and multi-microphone approaches to speech enhancement. The main contribution of this thesis is a framework to exploit available prior knowledge about both speech and noise. The physiology of speech production places a constraint on the possible shapes of the speech spectral envelope, and this information s captured using codebooks of speech linear predictive (LP) coefficients obtained from a large training database. Similarly, information about commonly occurring noise types is captured using a set of noise codebooks, which can be combined with sound environment classi¯cation to treat different environments differently. In paper A, we introduce maximum-likelihood estimation of the speech and noise LP parameters using the codebooks. The codebooks capture only the spectral shape. The speech and noise gain factors are obtained through a frame-by-frame optimization, providing good performance in practical nonstationary noise environments. The estimated parameters are subsequently used in a Wiener filter. Paper B describes Bayesian minimum mean squared error estimation of the speech and noise LP parameters and functions there-of, while retaining the in- stantaneous gain computation. Both memoryless and memory-based estimators are derived. While papers A and B describe single-channel techniques, paper C describes a multi-channel Bayesian speech enhancement approach, where, in addition to temporal processing, the spatial diversity provided by multiple microphones s also exploited. In paper D, we introduce a multi-channel noise reduction technique motivated by blind source separation (BSS) concepts. In contrast to standard BSS approaches, we use the knowledge that one of the signals is speech and that the other is noise, and exploit their different characteristics. / QC 20100929

The ability to reduce flexural vibrations of lightweight structures has been a goal for many researchers. A type of transducer-controller system that accomplishes this is a piezoelectric patch connected to an electrical impedance, or shunt. The piezoelectric patch converts the vibrational strain energy of the structure to which it is bonded into electrical energy. This converted electrical energy is then modified by the shunt to influence to mechanical response. There are many types of shunt circuits which have demonstrated effective control of flexural systems. Of interest in this work is the negative capacitance shunt, which has been shown to produce significant reduction in vibration over a broad frequency range. A negative capacitance circuit produces a current that is 180̊ out of phase from a traditional, passive capacitor. In other words, the voltage of the capacitor decreases as charge is added. The negative capacitance shunt consists of a resistor and an active negative capacitance element. By adding a resistor and negative capacitor to the electrical domain, the shunt acts as a damper and negative spring in the mechanical domain. The performance of the negative capacitance shunt can be increased through proper selection of the shunt's electrical components. Three aspects of component selection are investigated: shunt efficiency, maximum suppression, and stability. First, through electrical modeling of the shunt-patch system, the components can be chosen to increase the efficiency of the shunt for a given impedance. Second, a method is developed that could be utilized to adaptively tune the magnitude of resistance and negative capacitance for maximum control at a given frequency. Third, with regard to stability, as the control gain of the circuit is increased, by adjusting the circuit parameters, there is a point when the shunt will become unstable. A method to predict the stability of the shunt is developed to aid in suppression prediction. The negative capacitance shunt is also combined with a periodic piezoelectric patch array to modify the propagating wave behavior of a vibrating structure. A finite element method is utilized to create models to predict both the propagation constant, which characterizes the reduction in propagating waves, and the velocity frequency response of a full system. Analytical predictions are verified with experimental results for both a 1- and 2-D periodic array. Results show significant attenuation can be achieved with a negative capacitance shunt applied to a piezoelectric patch array. Three electromechanical aspects are developed: design for maximum suppression, more accurate stability prediction, and increased power-output efficiency. First, a method is developed that may be used to adaptively tune the magnitude of resistance and negative capacitance for maximum suppression. Second, with regard to stability, a method is developed to predict the circuit components at which the circuit will obtain a stable output. Third, through electrical modeling of the shunt-patch system, the components are chosen to increase the power output efficiency of the shunt circuit for a given impedance. The negative capacitance shunt is also combined with a periodic piezoelectric patch array to modify the propagating wave behavior of a vibrating structure. Analytical predictions are verified with experimental results for both a 1- and 2-D periodic array. Results show significant attenuation can be achieved with a negative capacitance shunt applied to a piezoelectric patch array.

Contrast-enhanced mammography (CEM) relies on visualizing the growth of new blood vessels (i.e. tumor angiogenesis) to provide sufficient materials for cell proliferation during the development of cancer. Since cancers will accumulate an injected contrast agent more than other tissues, it is possible to use one of several methods to enhance the area of lesions in the x-ray image and remove the contrast of normal tissue. Large area flat panel detectors may be used for CEM wherein the subtraction of two acquired images is used to create the resulting enhanced image. There exist several methods to acquire the images to be subtracted, which include temporal subtraction (pre- and post-contrast images) and dual-energy subtraction (low- and high-energy images), however these methods suffer from artifacts due to patient motion between image acquisitions. In this research the use of a multilayer flat panel detector is examined for CEM that is designed to acquire both (low- and high-energy) images simultaneously, thus avoiding motion artifacts in the resulting subtracted image. For comparison, a dual-energy technique prone to motion artifacts that uses a single-layer detector is also investigated. Both detectors are evaluated and optimized using amorphous selenium as the x-ray to charge conversion material, however the theoretical analysis could be extended to other conversion materials. Experimental results of single pixel prototypes of both multilayer and single-layer detectors are also discussed and compared to theoretical results. For a more comprehensive analysis, the motion artifacts present in dual-exposure techniques are modeled and the performance degradation due to motion artifacts is estimated. The effects of noise reduction techniques are also evaluated to determine potential image quality improvements in CEM images.

Detta examensarbete omfattar akustiska undersökningar utav en desinficerande diskmaskin av typ Getinge 8666/8668 och har utförts på Getinge Desinfection AB i Växjö. Vi har undersökt hur man med enkla medel kan sänka ljudtrycksnivån i maskinens närhet. Mätningar har utförts på diskmaskinens torkprogram, där torkfläktarna är de mest dominerande ljudkällorna och ljudet främst är luftburet. En systematisk analysmetod bestående av införandet av successiva dämpningsåtgärder visade att de dominerande ljudvägarna går via taket, golvet samt till viss del spalter i maskinens väggar. Genom att isolera både tak, golv och spalter kunde ljudtrycksnivån sänkas med totalt hela 5,9 dB. Med enkla produktionsanpassade lösningar, i form utav en isolerad golvsockel och en bättre kåpa runt fläktarna, kan man sänka nivån med så mycket som 3,6 dB. Rapporten ska även kunna användas som ett underlag för vidare akustiska undersökningar på Getinge Desinfection AB. / This dissertation examines acoustic research of a disinfecting dishwasher type Getinge 8666/8668. The research has been performed in collaboration with Getinge Desinfection AB, a world wide known manufacturer of disinfecting equipment for healthcare, located in Växjö, Sweden. We have examined how to reduce noise from the dishwasher with simple measures. Measurements have been performed during the drying program of the machine. The drying fans are the dominating sources of sound, and the sound is air bourn. A systematic method by reducing the sound step by step shows that the major ways of sound is through the ceiling and through the floor. There is also a minor noise leakage through the gaps in the walls of the dishwasher. By isolating the ceiling, the floor and the gaps in the walls, it is possible to reduce the noise with totally 5.9 dB. With more production adapted solutions, in shape of an isolated floor base and a more efficient fan cover, it is possible to reduce the noise with as much as 3.6 dB. This dissertation is thought to be used as a basis for further acoustic research at Getinge Desinfection AB.

Achieving a low noise level is of great interest to the forest machine industry. Traditionally this is obtained by using passive noise reduction, i.e. by using materials for sound isolation and sound absorption. Especially designs to attenuate low frequency noise tend to be bulky and impractical from an installation point of view. An alternative solution to the problem is to use active noise control (ANC). The basic principle of ANC is to generate an anti-noise signal designed to destructively interfere with the unwanted noise. In this thesis two algorithms (Feedback FxLMS and Feedforward FxLMS) are implemented and evaluated for use in the ANC-system. The ANC-system is tuned to the specific environment in the driver’s cabin of a Komatsu forest machine. The algorithms are first tested in a simulated environment and then in real-time inside a forest machine. Simulations are made both in Matlab and in C using both generated signals and recorded signals. The C code is implemented on the Analog Devices Blackfin DSP card BF526. The result showed a significantly reduction of the sound pressure level (SPL) in the driver’s cabin. The noise attenuation obtained using the Feedback FxLMS was approximately 14 dB for a tonal 100 Hz signal and 11 dB using recorded engine noise from a forest machine at 850 rpm.