This thesis provides analytical and experimental proof-of-concept for a new feedback-controlled sound transmission control module for use in an active segmented partition (ASP) array. The objective of such a module is to provide high transmission loss down to low audible frequencies while minimizing the overall mass of the module. This objective is accomplished in the new module by using actively controlled panels in conjunction with analog feedback controllers. The new module also overcomes two limitations that exist in current ASP modules: the inability to control broadband random-noise and the lack of bidirectional control through the module. Overcoming these limitations represents an important advancement in the research area of actively controlled partitions and broadens the number of potential applications for ASP arrays. Analogous circuit models were developed and used to predict the performance of the new ASP module under feedback control. The preliminary design consists of two loudspeaker drivers mounted back-to-back in a duct, with two decoupled analog feedback controllers connected to reduce the vibration of the loudspeaker cones. It was found that the classical analogous circuit model of a loudspeaker proved inadequate for modeling the low- and mid-frequency transmission loss due to resonance effects of the loudspeaker surround. An enhanced model of a loudspeaker was then used to account for this phenomenon and more accurately predict the transmission loss behavior. An experimental proof-of-concept module was constructed using two 10 cm diameter loudspeaker drivers, two accelerometers, and other off-the-shelf materials. The two analog feedback controllers used in the module were designed and built using measured frequency response function techniques. The passive and active transmission loss of the module was measured using a plane-wave tube. Transmission loss of broadband random-noise in excess of 50 dB was achieved between 100 Hz and 2 kHz. The experimental transmission loss results validated the numerical model and showcased the transmission loss performance of the new module design.

No description available.

<p dir="ltr">Active noise control has been implemented in various applications as a highly flexible, customizable and adaptive lightweight noise control technique which also serves as effective complementary counterpart to passive noise control techniques (such as sound absorbing packages). As on-chip computing power advances, low-cost implementation of active noise control algorithms targeting at controlling noise in large spatial regions is made more possible than ever before, which also excited another wave of active research on this topic in recent years after the emerging and flourishing active noise control research era in the 1990's. To control larger space, the use of a multi-input and output (MIMO) system is necessary, since the controller needs to be designed based on the measured sound information in the targeted control region (these sensors are referred to error microphones). However, it is not practical to add a limitless number of error microphones to populate the whole control region, and it is sometimes not even possible to locate the error microphone directly at targeted locations when the system is in operation due to practical constraints (e.g., in a car cabin, it is not possible to place microphones in people's ears). Therefore, the virtual sensing technique have been to predict the sound at targeted locations from remote measurements. One of the challenges in virtual sensing is its performance robustness under a time-varying acoustic environment. The purpose of this work is mainly to use the time-varying acoustic environment introduced by a person's head motion as an example case study to explore the possibility of virtual sensing the sound at the person's two ears for different head positions based on acoustic data measured at a small-sized microphone array located behind the head without any auxiliary motion tracking devices. More specifically, it is to develop a machine learning based data-driven model that uses the cross-spectral matrix of sound signals measured at the remote microphones to predict the frequency response functions between remote microphone measurements and sound at ears (i.e., the virtual sensing frequency response functions) under different head positions. </p><p> </p><p dir="ltr">To get the data to train a neural network model, a measurement setup was suggested in the paper. A HATS dummy system that mimics the human hearing system with two microphones at the ear location was placed between the noise source and the reference microphone array composed of five microphones. Treating two ear microphone locations as the desired location of virtual sensors and microphone arrays as reference microphones, different measurements were taken by slightly changing the location and angle of the HATS dummy. A cross-spectrum density matrix was calculated with the measured data, and a frequency response matrix was calculated between the microphone array and the ear microphones, which would be used to make input data and target data for the neural network, respectively. With the cross-spectrum data, Dimension reduction was processed. A covariance matrix with the vectorized cross-spectrum density matrix was calculated, and power variation was evaluated to understand which frequency bands are sensitive to the change in the acoustic environment. In the hyperparameter choice, log-cosh was used for the loss function, LeakyRelu was used for the activation function, and Adam optimizer was selected. After comparing different learning rate strategies, a cosine decay with an initial learning rate of 0.003 was used for the learning rate setup. Frequency response with the target range from 51 Hz to 2000 Hz was estimated successfully with the listed neural networks setting with mean square error as 0.1205 and mean absolute error as 0.2025. Its error was compared with the standard deviation of the frequency response across the measurements. The error from the estimation was significantly lower than the standard deviation, which shows that the frequency response estimation using a neural network could increase the performance of active noise control with a virtual sensor even with the change in the acoustic environment.</p>

In the last decade acoustic noise has become more and more regarded as a problem. In cars, boats, trains and aircraft, low-frequency noise reduces comfort. Lightweight materials and more powerful engines are used in high-speed vehicles, resulting in a general increase in interior noise levels. Low-frequency noise is annoying and during periods of long exposure it causes fatigue and discomfort. The masking effect which low-frequency noise has on speech reduces speech intelligibility. Low-frequency noise is sought to be attenuated in a wide range of applications in order to improve comfort and speech intelligibility. The use of conventional passive methods to attenuate low-frequency noise is often impractical since considerable bulk and weight are required; in transportation large weight is associated with high fuel consumption. In order to overcome the problems of ineffective passive suppression of low-frequency noise, the technique of active noise control has become of considerable interest. The fundamental principle of active noise control is based on secondary sources producing ``anti-noise.'' Destructive interference between the generated and the primary sound fields results in noise attenuation. Active noise control systems significantly increase the capacity for attenuating low-frequency noise without major increase in volume and weight. This doctoral dissertation deals with the topic of active noise control within the passenger cabin in aircraft, and within headsets. The work focuses on methods, controller structures and adaptive algorithms for attenuating tonal low-frequency noise produced by synchronized or moderately synchronized propellers generating beating sound fields. The control algorithm is a central part of an active noise control system. A multiple-reference feedforward controller based on the novel actuator-individual normalized Filtered-X Least-Mean-Squares algorithm is introduced, yielding significant attenuation of such period noise. This algorithm is of the LMS-type, and owing to the novel normalization it can also be regarded as a Newton-type algorithm. The new algorithm combines low computational complexity with high performance. For that reason the algorithm is suitable for use in systems with a large number of control sources and control sensors in order to reduce the computional power required by the control system. The computational power of the DSP hardware is limited, and therefore algorithms with high computational complexity allow fewer control sources and sensors to be used, often with reduced noise attenuation as a result. In applications, such as controlling aircraft cabin noise, where a large multiple-channel system is needed to control the relative complex interior sound field, it is of great importance to keep down the computational complexity of the algorithm so that a large number of loudspeakers and microphones can be used. The dissertation presents theoretical work, off-line computer experiments and practical real-time experiments using the actuator-individual normalized algorithm. The computer experiments are principally based on real-life cabin noise data recorded during flight in a twin-engine propeller aircraft and in a helicopter. The practical experiments were carried out in a full-scale fuselage section from a propeller aircraft. / Buller i vår dagliga miljö kan ha en negativ inverkan på vår hälsa. I många sammanhang, i tex bilar, båtar och flygplan, förekommer lågfrekvent buller. Lågfrekvent buller är oftast inte skadligt för hörseln, men kan vara tröttande och försvåra konversationen mellan personer som vistas i en utsatt miljö. En dämpning av bullernivån medför en förbättrad taluppfattbarhet samt en komfortökning. Att dämpa lågfrekvent buller med traditionella passiva metoder, tex absorbenter och reflektorer, är oftast ineffektivt. Det krävs stora, skrymmande absorbenter för att dämpa denna typ av buller samt tunga skiljeväggar för att förhindra att bullret transmitteras vidare från ett utrymme till ett annat. Metoder som är mera lämpade vid dämpning av lågfrekvent buller är de aktiva. De aktiva metoderna baseras på att en vågrörelse som ligger i motfas med en annan överlagras och de släcker ut varandra. Bullerdämpningen erhålls genom att ett ljudfält genereras som är lika starkt som bullret men i motfas med detta. De aktiva bullerdämpningsmetoderna medför en effektiv dämpning av lågfrekvent buller samtidigt som volymen, tex hos bilkupen eller båt/flygplanskabinen ej påverkas nämnvärt. Dessutom kan fordonets/farkostens vikt reduceras vilket är tacksamt för bränsleförbrukningen. I de flesta tillämpningar varierar bullrets karaktär, dvs styrka och frekvensinnehåll. För att följa dessa variationer krävs ett adaptivt (självinställande) reglersystem som styr genereringen av motljudet. I propellerflygplan är de dominerande frekvenserna i kabinbullret relaterat till propellrarnas varvtal, man känner alltså till frekvenserna som skall dämpas. Man utnyttjar en varvtalssignal för att generera signaler, så kallade referenssignaler, med de frekvenser som skall dämpas. Dessa bearbetas av ett reglersystem som generar signaler till högtalarna som i sin tur generar motljudet. För att ställa in högtalarsignalerna så att en effektiv dämpning erhålls, används mikrofoner utplacerade i kabinen som mäter bullret. För att åstadkomma en effektiv bullerdämpning i ett rum, tex i en flygplanskabin, behövs flera högtalare och mikrofoner, vilket kräver ett avancerat reglersystem. I doktorsavhandlingen ''Active Control of Propeller-Induced Noise in Aircraft'' behandlas olika metoder för att reducera kabinbuller härrörande från propellrarna. Här presenteras olika strukturer på reglersystem samt beräkningsalgoritmer för att ställa in systemet. För stora system där många högtalare och mikrofoner används, samt flera frekvenser skall dämpas, är det viktigt att systemet inte behöver för stor beräkningskapacitet för att generera motljudet. Metoderna som behandlas ger en effektiv dämpning till låg beräkningskostnad. Delar av materialet som presenteras i avhandlingen har ingått i ett EU-projekt med inriktning mot bullerundertryckning i propellerflygplan. I projektet har flera europeiska flygplanstillverkare deltagit. Avhandlingen behandlar även aktiv bullerdämpning i headset, som används av helikopterpiloter. I denna tillämpning har aktiv bullerdämpning används för att öka taluppfattbarheten.

Several measurements may be used as error signals to determine how to appropriately control a sound field. These include pressure, particle velocity, energy density and intensity. In this thesis, numerical models are used to show which signals perform best in is free-field active noise control (ANC) using error sensors located in the near field of the sound sources. The second is equalization in a free field and a semi-free field. Minimized energy density total power output (MEDToPO) plots are developed; these indicate the maximum achievable attenuation for a chosen error sensor as a function of location. A global listening area equalization coefficient (GLAEC) is found to evaluate the performance of the equalization methods. It is calculated by finding the average of the spectral standard deviation of several frequency response measurements in a specified listening area. For free-field ANC employing error sensors located in the near field, pressure-based measurements perform the best. For free-field equalization over an extended listening region, total energy density performs best. Equalization of an extended listening region is more successful over a limited low-frequency bandwidth.

A control metric, the weighted sum of spatial gradients (WSSG), has been developed for use in active structural acoustic control (ASAC). Previous development of WSSG [1] showed that it was an effective control metric on simply supported plates, while being simpler to measure than other control metrics, such as volume velocity. The purpose of the current work is to demonstrate that the previous research can be generalized to plates with a wider variety of boundary conditions and on less ideal plates. Two classes of plates have been considered: clamped flat plates, and ribbed plates. On clamped flat plates an analytical model has been developed for use in WSSG that assumes the mode shapes are the product of clamped-clamped beam mode shapes. The boundary condition specific weights for use in WSSG have been derived from this formulation and provide a relatively uniform measurement field, as in the case of the simply supported plate. Using this control metric, control of radiated sound power has been simulated. The results show that WSSG provides comparable control to volume velocity on the clamped plate. Results also show, through random placement of the sensors on the plate, that similar control can be achieved regardless of sensor location. This demonstrates that WSSG is an effective control metric on a variety of boundary conditions. Ribbed plates were considered because of their wide use in aircraft and ships. In this case, a finite-element model of the plate has been used to obtain the displacement field on the plate under a variety of boundary conditions. Due to the discretized model involved, a numerical, as opposed to analytical, formulation for WSSG has been developed. Simulations using this model show that ASAC can be performed effectively on ribbed plates. In particular WSSG was found to perform comparable to or better than volume velocity on all boundary conditions examined. The sensor insensitivity property was found to hold within each section (divided by the ribs) of the plate, a slightly modified form of the flat plate insensitivity property where the plates have been shown to be relatively insensitive to sensor location over the entire surface of the plate. Improved control at natural frequencies can be achieved by applying a second control force. This confirms that ASAC is a viable option for the control of radiated sound power on non-ideal physical systems similar to ribbed plates.

This work investigates the application of active noise control (ANC) to speech. ANC has had success reducing tonal noise. In this work, that success was extended to noise that is not completely tonal but has some tonal elements such as speech. Limitations such as causality were established on the active control of human speech. An optimal configuration for control actuators was developed for a sphere using a genetic algorithm. The optimal error sensor location was found from exploring the nulls associated with the magnitude of the radiated pressure with reference to the primary pressure field. Both numerically predicted and experimentally validated results for the attenuation of single frequency tones were shown. The differences between the numerically predicted results for attenuation with a sphere present in the pressure field and monopoles in the free-field are also discussed.The attenuation from ANC of both monotone and natural speech is shown and a discussion about the effect of causality on the results is given. The sentence “Joe took father’s shoe bench out” was used for both monotone and natural speech. Over this entire monotone speech sentence, the average attenuation was 8.6 dB with a peak attenuation of 10.6 dB for the syllable “Joe”. Natural speech attenuation was 1.1 dB for the sentence average with a peak attenuation on the syllable “bench” of 2.4 dB. In addition to the lower attenuation values for natural speech, the pressure level for the word “took” was increased by 2.3 dB. Also, the harmonic at 420 Hz in the word “father’s” of monotone speech was reduced globally up to 20 dB. Based on the results of the attenuation of monotone and natural speech, it was concluded that a reasonable amount of attenuation could be achieved on natural speech if its correlation could approach that of monotone speech.

In this work we explore online secondary path modelling (SPM) in the context of spatial active noise control (ANC). Specifically, we are interested in the reduction of broadband noise over a three-dimensional region, without restrictions on microphone and loudspeaker array placement. As spatial ANC generally requires many channels, both ANC and SPM methods must have low computational cost. The SPM methods are intended to be used with a specific spatial ANC algorithm based on kernel interpolation. By incorporating SPM, the spatial ANC method is enabled to operate under timevarying secondary paths. Four SPM methods are considered in detail, of which three are based on the auxiliary noise technique. Descriptions of the algorithms are presented for the multichannel case, in addition to block-based implementations taking advantage of the fast Fourier transform to drastically reduce computational cost. Impulse responses to simulate a soundfield are recorded using a programmable robot arm. The algorithms are evaluated through simulations to show their respective strengths and weaknesses. It is found that the auxiliary noise based SPM methods have good convergence properties for both control filter and secondary path estimate, although the auxiliary noise’s degrading effect on the residual noise leads to a similar total noise reduction as the auxiliary noise free method. For all algorithms, the noise control performance worsens, and the convergence time increases by more than an order of magnitude, compared to when the secondary paths are known. It is verified that the kernel interpolation based spatial ANC method successfully reduces noise over a region even when used with online SPM. / I detta projekt undersöks sekundärvägsmodellering för spatial aktiv brusreducering. Fokus ligger på minskning av brus över en tredimensionell region, för metoder utan några restriktioner när det gäller mikrofon- och högtalarplacering. Efterssom spatial brusreducering generellt kräver många kanaler, behöver både sekundärvägsmodellering samt brusreducering ha mycket låg beräkningskostnad. Metoderna för sekundärvägsmodellering är menade att användas tillsammans med en specifik spatial brusreduceringsalgoritm baserad på kärninterpolation. Genom att inkludera sekundärvägsmodellering kan den spatiala brusreduceringsmetoden operera även då sekundärvägarna är tidsvarierande. Fyra metoder för sekundärvägsmodellering är undersökta i detalj, tre av vilka är baserade på auxiliärbrusprincipen. Dessa algoritmer är beskrivna för multikanalsfallet, tillsammans med blockbaserade implementationer som utnyttjar den snabba Fouriertransformen för att drastiskt minska sina beräkningskostader. Impulssvar som kan användas för att simulera ett ljudfält är inspelade med hjälp av en programmerbar robotarm. Algoritmerna är utvärderade genom simuleringar för att visa deras respektive styrkor och svagheter. Experimenten visade att de algoritmer som använder sig av auxiliärbrus har bra konvergenskaraktäristik för både kontrollfilter och sekundärvägsestimat. Däremot, på grund av auxiliärbrusets negativa inverkan på residualbruset i rummet, är den totala brusreduceringen snarlik det den auxiliärbrusfria metoden ger. För alla algoritmer blir brusreduceringen försämrad och konvergenstiden ökad med mer än en storleksordning när sekundärvägsmodellering används, jämfört med när sekundärvägarna är kända. Det verifierades också att den spatiala brusreduceringsmetoden baserad på kärninterpolation kan reducera brus över en region även när den används tillsammans med sekundärvägsmodellering.

This study presents comprehensive active cancellation of booming noise caused by the engine and the driveline inside a passenger car. In modern noise control systems for vehicles, booming noise caused by engine harmonics could be effectively suppressed by employing active noise control. However, practical attempts or studies for the active suppression of driveline booming noise are scarce. One of the reasons may be that since the booming noise caused by the driveline is not harmonic with the engine speed, reference signals cannot be generated conventionally. Thus, passive approaches are generally employed to improve the driveline noise. To address this limitation, we propose a method for generating reference signals from engine revolution speed to suppress the driveline noise, such as propeller shaft and tire noise. Reference signals for driveline noise suppression were generated using the information from the torque converter, gear ratio, and final drive ratio. A practical active noise control system was implemented in a six-cylindered large sedan to validate the proposed method. The experimental results showed that the engine firing order was suppressed by 8.0 dB. Moreover, the first order of the propeller shaft and the second and third orders of the tires were suppressed by 5.5 dB, 3.9 dB, and 2.3 dB for entire seat positions. Furthermore, the results presented in this study were considered effective for improving annoyance perception through subjective evaluation.

This study shows that a concurrent active noise control strategy for engine harmonics and road noise has a complementary effect. In particular, we found that engine booming noise is additionally attenuated when road noise control is concurrently used with engine harmonics control; an additional attenuation of 2.08 dB and 1.25 dB for the C1.5 and C2.0 orders, respectively, was achieved. A parallel multichannel feedforward controller for non-stationary narrowband engine harmonics and broadband road noise was designed and implemented to reduce noise in all four seats. Two control signals were considered independent because the reference signals, engine revolution speed for the engine harmonic controller, and acceleration signal for the road noise controller are uncorrelated. However, if the reference sensor for the road noise controller is installed along the overlapping transfer path between the engine noise and road noise, the engine noise may also be suppressed by the control signal for the road noise attenuation. Based on transfer path analyses for both engine harmonics and road noise, the optimal positions for the reference sensors were selected. In addition, we identified several overlapping transfer paths between the engine booming noise and road noise. A practical active noise control system combined with a remote microphone technique was implemented for a large six-cylinder sedan using a vehicle audio system to evaluate the noise attenuation performance. The experiments showed that the interior noise from the engine and road excitation was effectively suppressed by the proposed concurrent control strategy..