Development of an Experimental Rig for Investigation of Higher Order Modes in Ducts

Continued progress to reduce fan noise emission from high bypass ratio engine ducts in aircraft increasingly relies on accurate description of the sound propagation in the duct. A project has been undertaken at NASA Langley Research Center to investigate the propagation of higher order modes in ducts with flow. This is a two-pronged approach, including development of analytic models (the subject of a separate paper) and installation of a laboratory-quality test rig. The purposes of the rig are to validate the analytical models and to evaluate novel duct acoustic liner concepts, both passive and active. The dimensions of the experimental rig test section scale to between 25% and 50% of the aft bypass ducts of most modern engines. The duct is of rectangular cross section so as to provide flexibility to design and fabricate test duct liner samples. The test section can accommodate flow paths that are straight through or offset from inlet to discharge, the latter design allowing investigation of the effect of curvature on sound propagation and duct liner performance. The maximum air flow rate through the duct is Mach 0.3. Sound in the duct is generated by an array of 16 high-intensity acoustic drivers. The signals to the loudspeaker array are generated by a multi-input/multi-output feedforward control system that has been developed for this project. The sound is sampled by arrays of flush-mounted microphones and a modal decomposition is performed at the frequency of sound generation. The data acquisition system consists of two arrays of flush-mounted microphones, one upstream of the test section and one downstream. The data are used to determine parameters such as the overall insertion loss of the test section treatment as well as the effect of the treatment on a modal basis such as mode scattering. The methodology used for modal decomposition is described, as is a description of the mode generation control system. Data are presented which demonstrate the performance of the controller to generate the desired mode while suppressing all other cut on modes in the duct

Sound field control with hemi-cylindrical loudspeaker arrays

An acoustical model for the sound field generated by hemi-cylindrical loudspeaker arrays is presented and a method for beamforming with said arrays is derived. The sound field model is obtained by introducing two independent boundary conditions for the sound field of a single impinging plane wave. The model for the radiation from a single loudspeaker in the array is then obtained from the reciprocity principle. Various beam patterns are presented and the theoretically predicted sound field is evaluated as a function of frequency. The results are discussed and an experimental array prototype is presented

Implementation of the Radiation Characteristics of Musical Instruments in Wave Field Synthesis Applications

In this thesis a method to implement the radiation characteristics of musical instruments in wave field synthesis systems is developed. It is applied and tested in two loudspeaker systems.Because the loudspeaker systems have a comparably low number of loudspeakers the wave field is synthesized at discrete listening positions by solving a linear equation system. Thus, for every constellation of listening and source position all loudspeakers can be used for the synthesis. The calculations are done in spectral domain, denying sound propagation velocity at first. This approach causes artefacts in the loudspeaker signals and synthesis errors in the listening area which are compensated by means of psychoacoustic methods. With these methods the aliasing frequency is determined by the extent of the listening area whereas in other wave field synthesis systems it is determined by the distance of adjacent loudspeakers. Musical instruments are simplified as complex point sources to gain, store and propagate their radiation characteristics. This method is the basis of the newly developed “Radiation Method” which improves the matrix conditioning of the equation system and the precision of the wave field synthesis by implementing the radiation characteristics of the driven loudspeakers. In this work, the “Minimum Energy Method” — originally developed for acoustic holography — is applied for matters of wave field synthesis for the first time. It guarantees a robust solution and creates softer loudspeaker driving signals than the Radiation Method but yields a worse approximation of the wave field beyond the discrete listening positions. Psychoacoustic considerations allow for a successfull wave field synthesis: Integration times of the auditory system determine the spatial dimensions in which the wave field synthesis approach works despite different arrival times and directions of wave fronts. By separating the spectrum into frequency bands of the critical band width, masking effects are utilized to reduce the amount of calculations with hardly audible consequances. By applying the “Precedence Fade”, the precedence effect is used to manipulate the perceived source position and improve the reproduction of initial transients of notes. Based on Auditory Scene Analysis principles, “Fading Based Panning” creates precise phantom source positions between the actual loudspeaker positions. Physical measurements, simulations and listening tests prove evidence for the introduced methods and reveal their precision. Furthermore, results of the listening tests show that the perceived spaciousness of instrumental sound not necessarily goes along with distinctness of localization. The introduced methods are compatible to conventional multi channel audio systems as well as other wave field synthesis applications.In dieser Arbeit wird eine Methode entwickelt, um die Abstrahlcharakteristik von Musikinstrumenten in Wellenfeldsynthesesystemen zu implementieren. Diese wird in zwei Lautsprechersystemen umgesetzt und getestet. Aufgrund der vergleichsweise geringen Anzahl an Lautsprechern wird das Schallfeld an diskreten Hörpositionen durch Lösung eines linearen Gleichungssystems resynthetisiert. Dadurch können für jede Konstellation aus Quellen- und Hörposition alle Lautsprecher für die Synthese verwendet werden. Hierzu wird zunächst in Frequenzebene, unter Vernachlässigung der Ausbreitungsgeschwindigkeit des Schalls gerechnet. Dieses Vorgehen sorgt für Artefakte im Schallsignal und Synthesefehler im Hörbereich, die durch psychoakustische Methoden kompensiert werden. Im Vergleich zu anderen Wellenfeldsyntheseverfahren wird bei diesem Vorgehen die Aliasingfrequenz durch die Größe des Hörbereichs und nicht durch den Lautsprecherabstand bestimmt. Musikinstrumente werden als komplexe Punktquellen vereinfacht, wodurch die Abstrahlung erfasst, gespeichert und in den Raum propagiert werden kann. Dieses Vorgehen ist auch die Basis der neu entwickelten “Radiation Method”, die durch Einbeziehung der Abstrahlcharakteristik der verwendeten Lautsprecher die Genauigkeit der Wellenfeldsynthese erhöht und die Konditionierung der Propagierungsmatrix des zu lösenden Gleichungssystems verbessert. In dieser Arbeit wird erstmals die für die akustische Holografie entwickelte “Minimum Energy Method” auf Wellenfeldsynthese angewandt. Sie garantiert eine robuste Lösung und erzeugt leisere Lautsprechersignale und somit mehr konstruktive Interferenz, approximiert das Schallfeld jenseits der diskreten Hörpositionen jedoch schlechter als die Radiation Method. Zahlreiche psychoakustische Überlegungen machen die Umsetzung der Wellenfeldsynthese möglich: Integrationszeiten des Gehörs bestimmen die räumlichen Dimensionen in der die Wellenfeldsynthesemethode — trotz der aus verschiedenen Richtungen und zu unterschiedlichen Zeitpunkten ankommenden Wellenfronten — funktioniert. Durch Teilung des Schallsignals in Frequenzbänder der kritischen Bandbreite wird unter Ausnutzung von Maskierungseffekten die Anzahl an nötigen Rechnungen mit kaum hörbaren Konsequenzen reduziert. Mit dem “Precedence Fade” wird der Präzedenzeffekt genutzt, um die wahrgenommene Schallquellenposition zu beeinflussen. Zudem wird dadurch die Reproduktion transienter Einschwingvorgänge verbessert. Auf Grundlage von Auditory Scene Analysis wird “Fading Based Panning” eingeführt, um darüber hinaus eine präzise Schallquellenlokalisation jenseits der Lautsprecherpositionen zu erzielen. Physikalische Messungen, Simulationen und Hörtests weisen nach, dass die neu eingeführten Methoden funktionieren und zeigen ihre Präzision auf. Auch zeigt sich, dass die wahrgenommene Räumlichkeit eines Instrumentenklangs nicht der Lokalisationssicherheit entspricht. Die eingeführten Methoden sind kompatibel mit konventionellen Mehrkanal-Audiosystemen sowie mit anderen Wellenfeldsynthesesystemen

Active Noise Control at low frequencies for Open Air events

Щороку в місті, поблизу житлових районів, проводяться все більше і більше open air заходів та великих концертів. Головною проблемою цього є шумове забруднення прилежних районів. З іншого боку, невід'ємною частиною таких концертів є високий рівень звукового тиску. Створення звукових зон є одним з можливих шляхів вирішення цієї проблеми. У даній роботі ми розглянемо та охарактеризуємо основні способи створення звукових зон: акустичний контраст, pressure matching та комбінований метод. Розглянуто також класичний метод активного контролю шуму на основі алгоритму найменших квадратів. Для всіх методів отримані кінцеві вирази для розрахунку оптимальних комплексних об'ємних швидкостей гучномовців. Описано значення параметрів регуляції цих методів. Для обчислення звукових зон використовуються виміряні або змодельовані передатні характеристики розповсюдження звуку. У цій роботі вивчається вплив навколишнього середовища та атмосферних умов. Моделювання всіх методів проводилося у середовищі MATLAB. Порівняння результатів проводилося за розрахунковими показниками продуктивності та за частотною характеристикою розрахованих оптимальних ваг. Важливість параметрів регулювання показана при моделюванні різних методів. Для досліджуваної системи обраний оптимальний метод. В цілому метою цієї роботи є порівняння та пошук оптимального методу створення звукових зон, який використовується для управління великими зонами в межах open air події; а також зробити оцінку можливого впливу атмосферних умов на точність і надійність цих методів.Every year more and more open-air events and large concerts are held in the city, near residential areas. The main problem of this is noise pollution in nearby areas. On the other hand, an integral part of such concerts is a high SPL. Creating sound zones is one possible solution to this problem. In this paper, we examine and characterize the main methods of creating sound zones: Acoustic Contrast, Pressure Matching and combined method. The classical method of active noise control based on the LMS algorithm is also considered. For all methods, final expressions are derived for calculating the optimal complex volume velocities of the loudspeakers. The meaning of regularization parameters of these methods is described. To calculate the sound zones, measured or modeled propagation transfer functions are used. The effect of the environment and atmospheric conditions is studied in this work and their impact is evaluated. Simulations of all methods were performed at MATLAB. Comparison of the results was carried out on the calculated performance metrics and on the frequency response of the calculated optimal weights. The importance of the regulation parameters has been shown when simulating various methods. The optimal method was chosen for the system under study. In general, this work aimed to make a comparison and search for the optimal method for creating sound zones, which is used to control large zones within an open air event; and also make an assessment of the possible influence of atmospheric conditions on the accuracy and robustness of these methods.Ежегодно в городе, вблизи жилых районов, проводятся все больше и больше open air мероприятий и больших концертов. Главной проблемой этого является шумовое загрязнение прилегающей районов. С другой стороны, неотъемлемой частью таких концертов является высокий уровень звукового давления. Создание звуковых зон является одним из возможных путей решения этой проблемы. В данной работе мы рассмотрим и охарактеризуем основные способы создание звуковых зон: акустический контраст, pressure matching и комбинированный метод. Рассмотрены также классический метод активного контроля шума на основе алгоритма наименьших квадратов. Для всех методов полученные конечные выражения для расчета оптимальных комплексных объемных скоростей громкоговорителей. Описаны значения параметров регуляции этих методов. Для вычисления звуковых зон используются измеренные или смоделированы передаточные характеристики распространения звука. В этой работе изучается влияние окружающей среды и атмосферных условий. Моделирование всех методов проводилось в среде MATLAB. Сравнение результатов проводилось по расчетным показателям производительности и по частотной характеристикой рассчитанных оптимальных весов. Важность параметров регулирования показана при моделировании различных методов. Для исследуемой системы выбран оптимальный метод. В целом целью этой работы является сравнение и поиск оптимального метода создания звуковых зон, который используется для управления большими зонами в пределах open air события; а также сделать оценку возможного воздействию атмосферных условий на точность и надежность этих методов

On a Simple Technique to Measure the Airborne Noise in a Motor Vehicle using Source Substitution

From various methods of measuring noise of a motor vehicle, there is a lack application on how to obtain the ‘pure’ airborne noise in a vehicle cabin as an important measure to improve the noise control treatment. This project proposes a technique in separating airborne and structure-borne noise in a motor vehicle using sound source substitution method. A cone loudspeaker was used as the substitution source and the overall transmission loss was measured to achieve this purpose. It is found that the structure-borne noise is dominant at low frequencies and airborne noise dominates at high frequency. The intersection frequency between the two is roughly 400 Hz

Spatial noise cancellation inside cars: Performance analysis and experimental results

A loudspeaker array is a key component in active noise cancellation (ANC) systems. Most in-car ANC systems utilize the car’s own integrated loudspeakers to cancel the noise due to engine and other sources. In this paper, we evaluate the integrated loudspeakers’ noise cancelling capabilities by analyzing the in-car noise field and the loudspeaker responses. We show that the average noise power in a spatial region can be expressed using a series of coefficients, and that the noise field can be decomposed into several basis noise patterns. Through analysing the measurements in a car, we show that the car’s built-in loudspeakers are capable of attenuating the driving noise by up to 30 dB for frequencies up to 500 Hz within a spherical region of 10 cm radius

Analysis and control of multi-zone sound field reproduction using modal-domain approach

Multi-zone sound control aims to reproduce multiple sound fields independently and simultaneously over different spatial regions within the same space. This paper investigates the multi-zone sound control problem formulated in the modal domain using the Lagrange cost function and provides a modal-domain analysis of the problem. The Lagrange cost function is formulated to represent a quadratic objective of reproducing a desired sound field within the bright zone and with constraints on sound energy in the dark zone and global region. A fundamental problem in multi-zone reproduction is interzone sound interference, where based on the geometry of the sound zones and the desired sound field within the bright zone the achievable reproduction performance is limited. The modal-domain Lagrangian solution demonstrates the intrinsic ill-posedness of the problem, based on which a parameter, the coefficient of realisability, is developed to evaluate the reproduction limitation. The proposed reproduction method is based on controlling the interference between sound zones and sound leakage outside the sound zones, resulting in a suitable compromise between good bright zone performance and satisfactory dark zone performance. The performance of the proposed design is demonstrated through numerical simulations of two-zone reproduction in free-field and in reverberant environments