bluuebunny/crossref_metadata_2025 huggingface dataset:
Focusing sound in coastal environments.
Authors: Kankey Andrew T. , Koopmann Gary H. , Becker Kyle M. , Bradley David L.
Published: October 2013

An investigation of the ability to focus low-frequency acoustic energy in very shallow water was carried out in a semiprotected harbor in June 2008. From the end of a finger pier, a linear array of seven acoustic sources was suspended 2 m from the harbor bottom. The mean water depth in the harbor was 11 m with surficial sediments characterized by silts, sandy silts, and clay. The acoustic field was measured by an array of 11 hydrophones oriented in a cross pattern in the harbor. The phones ranged from 100 to 325 m from the center of the array. The objective of this work was to explore alternative methods to time-delay beamforming for focusing the sound field in the harbor. The approach is based on optimizing the phase of the individual sources to provide maximum sound pressure levels at a particular location in the harbor. The optimization scheme requires a priori interrogation of the harbor with a number of hydrophones in the area of interest. However, it is designed to be insensitive to vertical boundaries and other obstructions that can be found in typical working harbors. Results from recent field work are discussed.
Acoustic research in the Hudson River Estuary at the Stevens Maritime Security Laboratory.
Authors: Sutin Alexander , Bunin Barry
Published: October 2013

Stevens Institute of Technology has established a Maritime Security Laboratory (MSL) as a national laboratory resource for government, industry, and universities to advance technologies for the protection of USN maritime infrastructure. Experiment instrumentation includes research vessels, a multiplicity of hydrophones and emitters, stand alone acoustic buoys, diver acoustic simulators, unmanned underwater vehicles (UUVs), and precision instrumentation placement capabilities. The in-river experiments are controlled remotely from a Visualization Center on campus. Acoustic research is supported by sound speed profile measurements, integrated video and acoustic tracking of surface events, and global positionlng system tracking of live divers. Recent results include determination of parameters defining the detection distance of a threat: source level, transmission loss, and ambient noise. The combination of acoustic noise with video data for different kinds of ships in the Hudson River enables estimation of sound attenuation in a wide frequency band. The establishment of a library of various estuarine signatures, including divers, boats, ships, UUVs, construction equipment, and so forth, is underway. This knowledge can be used in a variety of intruder detection scenarios and for optimal methods of threat detection. [This work was supported by ONR Project N00014-05-1-0632: Navy Force Protection Technology Assessment Project.]
Automated passive acoustic intruder detection based on correlation approach.
Authors: Sedunov Alexander , Fillinger Laurent , Sutin Alexander
Published: October 2013

We consider the problem of passive acoustic detection of intruders in shallow waters using the signals acquired from multiple hydrophones. The suggested approach uses correlation of the acoustic signals recorded by different hydrophones as well as other features such as periodicity. Cross correlation of signals is conducted within a sliding window after passing through a multiband filter bank. This multiband filtering allows leveraging the intrinsic spectral properties of emitted signals to discriminate intruder emission from acoustic noise sources such as ship traffic typically present in the river. Further heuristic processing exploits the periodic nature of signals such as human breathing, by detecting the beginnings and endings of the diver’s inhalation signals. The detector assesses whether the recognized features are consistent with previously documented breathing rates in order to automatically and robustly detect the presence of a diver. The developed approach was applied in diver detection tests conducted in the Hudson River estuary where cross-correlation technique allowed finding the line of bearing to a diver. The challenges to implementation of such method in a standalone acoustic buoy under development at Stevens are also discussed. [This work was supported by ONR Project No. N00014-05-1-0632: Navy Force Protection Technology Assessment Project].
Development of special acoustic materials for the Experimental Media and Performing Arts Center at Rensselaer Polytechnic Institute: Part 2.
Authors: Belanger Zackery
Published: October 2013

Modern facilities for experimental performance create rigorous acoustical challenges because of the extreme and peripheral nature of their use. Spaces firmly rooted in tradition have acoustic criteria, which are largely known, but experimental spaces must accommodate the traditional, the contemporary, and the not yet conceived. The Experimental Media and Performing Arts Center at Rensselaer Polytechnic Institute, opening this year in Troy, NY, is one such facility. Five major venues—a concert hall, a visual performance studio, a musical performance studio, a theater, and a production suite—are implemented in a complementary harmony of flexibility. The concert hall includes a lightweight Nomex ceiling, which selectively reflects and transmits sound, and the studios utilize adjustable diffusers and absorbers of custom design. Isolation of the critical venues is extensive in anticipation of the use of extreme levels and low frequency sound.
Case study: Exposed structure acoustical solutions.
Authors: Good Kenneth , Browne Sean
Published: October 2013

Many new buildings employ an exposed structure design aesthetic and traditional construction elements such as ceilings and walls are being eliminated. These materials perform an acoustical function that is sacrificed in order to achieve visual and sustainability requirements. This presentation will explore several problematic exposed structure spaces and the acoustical solutions that restored proper performance.
Acoustical optimization of shapes and materials used in modern architecture.
Authors: D’Antonio Peter
Published: October 2013

Classic architecture or classically inspired architecture benefits from the fact that scattering surface, in the form of columns, statuary, and relief ornamentation, is an integral part of the architecture. As architecture evolved into using less ornate surfaces in smooth rectilinear and more recently curvilinear forms, this created a need to design scattering surfaces that complement modern architecture. When contemporary scattering surfaces are required, shape optimization, using any reflective material, has proven to be mutually useful for the acoustician and the architect. A shape drawn by the architect is parametrized and then optimized maintaining the desired motif. An iterative computer program that combines the benefits of boundary element and multidimensional minimization techniques will be described and examples will be shown. When absorption is required, microperforation and microslit designs offer a novel solution, using light transmitting plastics and metals. Microperforated wood veneer panels are also available in which the perforations are barely visible at normal viewing distances. A summary of the theory and application of these ideas can be found in work of Cox and D’Antonio [Acoustic Absorbers and Diffusers: Theory, Design and Application (Spon, 2004)].
Innovative green acoustic techniques to control noise ingress and sound propagation and to reduce energy consumption.
Authors: Biziorek Ryan
Published: October 2013

The current LEED rating system currently does not give credit for any elements of acoustic design within the point system. Too often, projects can become too focused on obtaining the points within the defined categories while sacrificing the basic elements of acoustic design (i.e., noise ingress from natural ventilation). The acoustic consultant must often find innovative ways to incorporate acoustic design into the building that will support the sustainability goals of the building and can be used to support and/or obtain a LEED point. Three innovative methods of green acoustic design in project case studies will be presented: natural ventilation techniques to maintain privacy, acoustic comfort in indoor working environments, and energy saving techniques for performing arts facilities. If applied to a project in an innovative method, the acoustics of a project need not be compromised and could be used as a LEED credit for innovation in design.
Spatial unmasking of pure tones in broadband noise by zebra finches (Taeniopygia guttata).
Authors: Dent Micheal , Cone Jarrod
Published: October 2013

Detecting a signal embedded in noise is known to be enhanced by spatially separating the signal and noise in humans and other animals. This process is known as spatial unmasking and is a part of the larger phenomenon of the cocktail party problem. The exact mechanisms of unmasking are unknown, but binaural processes are thought to be at least partially involved. Most animals that exhibit unmasking are fairly adept at localizing pure tones in space. We wished to study spatial unmasking in an animal that is very poor at sound localization: the zebra finch. Zebra finches were trained using operant conditioning techniques and the psychophysical method of constant stimuli to peck keys for food reinforcement when they detected a tone embedded in a broadband noise masker. Thresholds were obtained for pure tones ranging from 500 Hz to 4000 Hz when the signal and the noise were emitted from the same speaker and when they were emitted from speaker locations separated by 180 deg. Zebra finches showed relatively little unmasking and there was large variation across subjects and frequencies, suggesting that the mechanisms underlying sound localization are related to those that result in spatial unmasking.
Transmission fidelity in rhesus monkey (Macaca mulatta) “coos” and “screams”.
Authors: Tarkington Eric , Heimbauer Lisa , Owren Michael
Published: October 2013

In animal communication, studies of environmental acoustics examine issues such as habitat-specific propagation effects on vocalizations and likely impact on psychologically significant aspects (e.g., localizability and identifiability). The current work compared transmission fidelity in harmonically structured “coos” versus noisy “screams” produced by female rhesus monkeys (Macaca mulatta). Calls were broadcast in a temperate, mixed forest using a loudspeaker (Genelec 1029A) positioned either 0.5 m or 1.0 m above the ground, and re-recorded using two microphones (Sennheiser MKH106T) and a digital audio deck (TAS-CAM DA-P1). The “near” microphone remained at a constant distance of 2.5 m from the loudspeaker, while the “far” microphone was positioned at 10, 20, 30, 40, or 50 m. Eight different coos and eight different screams were played four times each at every distance, with cross-correlation values calculated for each pair of near- and far-microphone recordings. Results included that correlations decreased monotonically with far-microphone distance and were higher for calls broadcast at 1.0 m than at 0.5 m. Furthermore, however, correlations were significantly higher for coos than for screams, consistent with other evidence that harmonically structured calls are better suited to functions such as long-distance transmission of caller identity than are noise-based vocalizations.
Ultrasonic characterization of insect wing reflectivity and wing-beat motion at 200 kHz.
Authors: Swanson David , Baker Tom , terMeulen Ryan
Published: October 2013

A bistatic active ultrasonic sensor is used to detect and identify an insect attracted to a pheromone lure in the beam intersection. The ultrasound beam is a continuous waveform (CW) of approximately 200 kHz to avoid detection by the insect auditory system, which is very sensitive near 40 kHz and below to detect predatory bats. Both amplitude modulation (AM) and frequency modulation (FM) of the CW beam are examined. The reflection factor of the wing material is measured using ultrasonic pulses. The near-field interference nulls of the transmitting beam are modeled and measured to ensure that the lure is not placed in one of these zones. While good AM and FM signals are seen in the received waveform, the pivoting and flapping motions of the wings along with the insect movements create a complex pulse-train like waveform. This is seen to be inherently due to the narrow beam-width for the transducers and a lack of nonspecular scattering from the moving wings. The data show that a simple AM detector can be used to detect the presence of an insect at the lure and to measure the wing-beat frequency. [Work supported by USDA].
Relative saliency of envelope and fine structure cues in zebra finch song.
Authors: Vernaleo Beth , Dooling Robert
Published: October 2013

Birdsong provides a useful model for communication and vocal development, and zebra finch song in particular is attractive for its acoustical complexity and repetitive nature. Males sing one song for the purpose of mating display and territory defense, whereas females do not sing. In this study, we are particularly interested in which acoustic features of a male’s song are most perceptually salient. Using a repeating background of a single song motif, zebra finches were trained to discriminate changes to two cues in song: increases in intersyllable interval duration (envelope cue) and reversals of single syllables within a song motif (fine structure cue). Results show that zebra finches are able to discriminate changes to fine structure of syllables much more easily than changes to the overall envelope of song, specifically intersyllable intervals. Further experiments have been done using a noise burst modeled song in which song syllables were replaced by frozen random noise bursts of the same duration. Results show that zebra finches are able to discriminate single noise burst reversals, suggesting that they are able to attend to and follow fine structure on a very small time scale. [Work supported by NIH/NIDCD 5R01DC000198 and 2P30DC004664.]
A radiation force technique for acoustic power calibration of high intensity focused ultrasound transducers.
Authors: Maruvada Subha
Published: October 2013

It is essential to know the acoustic power radiated by transducers used in high intensity focused ultrasound (HIFU) surgery devices for both safety and effectiveness considerations. The power radiated by medical ultrasound transducers usually is measured via radiation force balance (RFB) methods. However, for the high power focused fields encountered in HIFU applications, such measurements can be difficult due to the need for short measurement times to prevent transducer damage, heating of the RFB target, and bubble formation. To address these challenges, a procedure based on pulsed measurements was formulated. High output focused ultrasound transducers were characterized in terms of an effective duty factor, which was then used to calculate the power during the pulse at high drive levels. Two absorbing target designs were used, and both gave reliable results and displayed no damage and minimal temperature rise if placed near the HIFU transducer and away from the focus. The procedure was reproducible up to the maximum power generated of approximately 230 W, representative of the HIFU range, thus allowing the radiated power to be calibrated in terms of the peak-to-peak voltage applied to the transducer.
Reducing measurement errors: Taking the design of microphones to the limit.
Authors: Gramtorp Johan , Olsen Erling Sandermann
Published: October 2013

It is impossible to make one microphone that covers all possible applications. In the real world, the properties of a microphone are the result of a number of carefully chosen compromises based on the anticipated measurement situation and the measurement equipment. The electroacoustic properties of condenser measurement microphones are rigidly bound to the physical properties of the microphone cartridge and the electrical properties of the preamplifier. This means that in the design of a measurement microphone the challenge is to find the right balance between the various physical properties of the microphone. However, recent development in simulation models, construction technology, and knowledge of material properties has opened for the design of microphones that has pushed the limits. A new class of general purpose measurement microphones has been born. In this paper, the physical properties that determine the electroacoustic properties of a condenser microphone are reviewed, and the various compromises that must be chosen are discussed. The possibilities and limitations in microphone design are illustrated with examples of the newest microphone designs from Brüel & Kjær.
Phase frequency responses of measurement microphones and their calibration.
Authors: Frederiksen Erling
Published: October 2013

For more than 20 years, comparison calibration of microphone phase responses has been a topic of interest. The interest grew up in the 1980s, when it became technically possible to design instruments for measurement of sound intensity and particle velocity. At that time, the calibration frequency range was limited by the first generation of comparison couplers, which typically worked up to 5 kHz. Since then, new applications and needs for wider frequency range or higher accuracy have occurred. Today phase response comparison calibration of half-inch and quarter-inch microphones can be made with couplers from below 20 Hz up to about 20 kHz. The uncertainty is estimated to about 0.010 deg between 20 and 500 Hz from where it increases to some tenths of a degree at 20 kHz. Microphone array applications require large numbers of microphones with essentially equal phase responses. This has increased the need for reference standards, whose absolute phase responses are known. The Danish Primary Laboratory of Acoustics calibrates such reference standards by the reciprocity method. This paper gives an overview of the spread of microphone phase responses, commonly required tolerances, phase calibration methods, and their uncertainties.
Acoustical impedance measurement.
Authors: King Charles
Published: October 2013

Many components of specialty transducers have extremely tight acoustic impedance tolerances and become difficult to measure in a production environment. Comparing the device to the acoustical impedance of a cavity is one method of determining its acoustical impedance. A precise method of determining the acoustical impedance is described. The technique applies methods normally found in defining cavities for pressure reciprocity calibration to obtain a precision acoustical reference. Pressure reciprocity standards also define how environmental factors influence acoustical properties and these are included into the system to maintain the precision regardless of the environmental conditions or the frequency of operation. Further improvements are obtained by employing differential measurements throughout the measurement and calibration process to eliminate many common errors common to field calibration of microphones. Ultimately the measured data are curve fitted to an equivalent circuit model to extract the desired components from the measurement artifacts. This talk will discuss the method of defining the acoustical components and extracting the desired value from the equivalent circuit model.
Demonstration of an impedance based method for sonar calibration and monitoring.
Authors: Bachand Corey , Brown David A. , Aronov Boris
Published: October 2013

It is well known that impedance based methods can be used to monitor the performance degradation of sonar transducers. A new approach that involves precise measurements of the voltage (V), current (I) and phase (P), termed the VIP-impedance method, has been developed and implemented by the authors for characterizing individual transducer and array performance in sonar systems. Examples of predicted transducer sensitivities and beam patterns on both cylindrical and tonpiltz sonar arrays using the new method are presented and compared with experimental data.
A Helmholtz resonator experiment for the Listen Up project.
Authors: Greene Chad A. , Argo Theodore F. , Wilson Preston S.
Published: October 2013

The Listen Up project seeks to develop an educational booklet and apparatus that will be used to foster interest in acoustics and teach basic acoustical concepts to middle school students. The packaged kit must be low cost; hence this experiment uses a 16 oz plastic water bottle as a Helmholtz resonator, a disposable syringe, and an inexpensive pitch pipe. Understanding the Helmholtz resonator is fundamental to many applications in acoustics. A simple algebraic model of the device, derived from a mechanical mass-spring analogy, relates the volume of air in the bottle to the resonance frequency of the system, and hence the pitch that is produced when one blows over the bottle opening. The volume of air inside the bottle can be easily and accurately controlled by adding water with a graduated syringe. The dependence of pitch on the volume of air in the bottle can be systematically demonstrated, and the validity of the model can be checked by comparing the sound produced to musical notes from the pitch pipe. This experiment yields a physical understanding of a common acoustical phenomenon and inspires further interest and study of more complex phenomena.
Tuning forks, resonators, and beats: Demonstrations for “Project Listen Up”.
Authors: Korman Murray S.
Published: October 2013

Gearing to an audience of all ages, a demonstration apparatus is presented including student versions of Hermann von Helmholtz’s tuning fork sounder, circa 1859 (where an intermittent current in an electromagnetic coil drives the tuning fork, which is located near a resonator), and Alexander Graham Bell’s tuning fork experiment, circa 1876 (where a variable resistance circuit due to a slight tuning fork contact in a conductive liquid causes a relay to vibrate at the same frequency as the fork). Using the popularity of a basic mechanical acoustical apparatus and briefly unfolding the histories behind Helmholtz combining several of his tuning fork sounders to generate vowel sounds and Bell’s first telephone, students are then given careful explanations of the demonstrations. Next, the focus shifts to well developed basic experiments that students can perform in a laboratory setting with emphasis placed on presenting the details of a carefully thought out scientific approach. “Project Listen Up” (an initiative of the Education in Acoustics Committee) hopes to develop an educational apparatus for a broad range of learners as one of their goals. [Work supported by USNA.]
Introducing perceptual coding using a double-blind A/B testing demonstration.
Authors: Porter Scott
Published: October 2013

Many of today’s college students experience music in compressed formats on a regular basis. While most are familiar with the term “MP3,” only a few know of the psychoacoustic principles and perceptual coding schemes which this and other compressed music formats rely upon. In fact, it is not uncommon that they do not realize the audible differences that exist between the CD-quality audio files and those compressed at common commercial bit rates. To illustrate this, the author used freely available software (WINABX and AUDACITY) to demonstrate the differences with double-blind A/B testing as part of an introductory course in acoustics offered at the Pennsylvania State University. The software will be briefly explained, the questionnaire used to acquire the students’ responses will be shown, and an interactive demonstration will be included.
Visualization of harmonic starting process for flute sound experiment.
Authors: Takahashi Yoshinori , Tohyama Mikio , Miyoshi Kazunori
Published: October 2013

Signal analysis based on spectral accumulation has great potential for monitoring the condition of structures using random vibration records excited by natural forces. This article describes cumulative harmonic analysis (CHA) by introducing a spectral accumulation function into Berman and Fincham’s conventional cumulative analysis, thus exploring a potential new area in cumulative analysis. CHA effectively visualizes the feedback process from the flow of air against an edge. In this work, we visualized the harmonic starting process for understanding of physical experiment of musical instrument study especially the flute.
Audio analysis using sparse representations.
Authors: Plumbley Mark D. , Abdallah Samer A. , Jafari Maria G. , Nesbit Andrew
Published: October 2013

The method of “sparse representations,” based on the idea that observations should be represented by only a few items chosen from a large number of possible items, has emerged recently as an interesting approach to the analysis of images and audio. New theoretical advances and practical algorithms mean that the sparse representations approach is becoming a potentially powerful signal processing and analysis method. Some of the key concepts in sparse representations will be introduced, including algorithms to find sparse representations of data. An overview of some applications of sparse representations in audio will be described, including for automatic music transcription and audio source separation, and pointers will be given for possible future directions in this area. [This work has been supported by grants and studentships from the UK Engineering and Physical Sciences Research Council.]
Toward automatic music transcription from audio input.
Authors: Sagayama Shigeki , Kameoka Hirokazu , Takeda Haruto
Published: October 2013

Transcribing audio input to obtain a music score has been a basic but yet hard problem in music signal and information processing. It can be paralleled with continuous speech recognition comprising of acoustic analysis, acoustic model, language model, and decoder. This presentation discusses functional modules for the music case, i.e., estimation of multiple fundamental frequencies, rhythm models, and chord modeling. For multi-F0 estimation, a computational auditory scene analysis-motivated approach is taken to model human hearing where each acoustic object is modeled with Gaussian mixture both along frequency and time axis [Kameoka (2005)]. An extended EM-algorithm is applied to iterative estimation of fundamental frequencies and onset/offset timings of music notes contained in the given spectrogram of the audio input. As for rhythm estimation, HMM is used to model the sequence of note lengths where tempo is treated in the model as a varying hidden variable [Takeda (2006)]. Chord progression is also modeled with HMM [Kawakami (2000)] where transition probabilities between chords and emission probabilities of notes from the hypothesized chord have been trained with a music database. Related issues are also discussed including other approaches to multiple fundamental frequencies with specmurt analysis [Sagayama (2005)], non-negative matrix factorization [Raczynski (2007)], and timber modeling [Miyamoto (2007)].
Prior structures for non-negative matrix factorization based audio source separation.
Authors: Virtanen Tuomas , Cemgil Ali Taylan
Published: October 2013

A generative signal model corresponding to non-negative matrix factorization algorithms is presented, and the model is extended by using priors for the matrices to be estimated. In the analysis of audio signals, this approach allows using prior information about the sounds, while allowing adaptation of the model parameters to the exact characteristics of the observed signal. Specifically, conjugate priors lead to fast inference algorithms. Methods for learning the parameters of the prior distributions from training material are described. The presented approach is evaluated on various sound source separation tasks, including standard data sets of music and speech.
Inferring missing spectral data.
Authors: Smaragdis Paris , Raj Bhiksha
Published: October 2013

In this talk we will present a methodology that allows us to infer missing portions of spectrograms. We will present an approach that constructs models of sounds by observing either examples or the existing portions of a spectrogram with missing data. Once the model is learned we can use it to reconstruct missing areas of a spectrogram with inaudible artifacts. This process is very useful when trying to correct errors from spectral editing or when dealing with processes that corrupt a signal in the time/frequency domain. We will show how this approach is appropriate for polyphonic audio signals and that it significantly outperforms approaches using generic statistical models which are ill suited for spectral data.
AudioDB: Scalable approximate nearest-neighbor search with automatic radius-bounded indexing.
Authors: Casey Michael
Published: October 2013

This talk describes new approximate nearest-neighbor methods employed in a scalable audio-feature database system called “AudioDB.” This open-source system is designed to scale to storing and searching hundreds of millions of feature vectors on standard UNIX workstation platforms. A radius-bounded nearest-neighbor vector-sequence search algorithm, based on locality sensitive hashing (LSH), achieves sublinear retrieval times at this scale. The performance of the LSH-based algorithm depends critically on the choice of radius bound supplied—the wrong value impacts retrieval accuracy or retrieval time. An optimal radius estimator is derived by modeling the minimum value distribution of a random sample of a data set’s pairwise distance distribution. When used with LSH this yields accurate search results with retrieval times several orders of magnitude faster than exhaustive search methods and space-partitioning methods. The same statistical sampling method is used to perform retrieval tasks at successively higher levels of specificity on labeled or unlabeled audio collections. The result is a system that (a) unifies audio retrieval tasks across a range of specificities, using the statistical framework of background distance-distribution sampling and hypothesis testing (b) is as accurate as exhaustive search methods and (c) is three orders of magnitude faster than exhaustive search methods.
Combining prior-knowledge and grouping cues using a spectral clustering approach.
Authors: Tzanetakis George , MartinsI Luis Gustavo
Published: October 2013

Learning happens at the boundary interactions beween prior knowledge and incoming data. The same interplay takes place when trying to analyze and separate complex mixtures of sound sources such as music. Many approaches to this problem can be broadly categorized as either model based or grouping based. Although it is known that our perceptual system utilizes both of these types of processing, building such systems computationally has been challenging. As a result most existing systems either rely on prior source models or are solely based on grouping cues. In this work it is argued that formulating this integration problem as clustering based on similarities between time-frequency atoms provides an expressive but discipined approach to building sound source characterization and separation systems and to evaluating their performance. After describing the main components of such an architecture, we describe a concrete realization that is based on spectral clustering of a sinusoidal representation. We show how this approach can be used to model both traditional grouping cues such as frequency and amplitude continuity as well as other types of information and prior knowledge such as stereo panning, onsets, harmonicity, and timbre models for specific integration will also be described.
A theoretical frequency band limitation for analog recordings.
Authors: Zucker Michael
Published: October 2013

Proponents of analog recording techniques have described their superiority over digitally sampled signals in part by referencing a theoretically infinite frequency band response. This is a typical argument used in favoring audiophile grade phonograph records to digitally sampled music. The actual frequency response of any individual transducer cartridge or phonograph record is determined by empirical measurements and is often bounded by electrical noise considerations as well as the physical condition of the record being played. These measured and predicted frequency responses, however, still hold the contention that there is theoretically infinite frequency band content available on perfect analog recordings and that the actual limit on frequency content is simply a result of an imperfect implementation of recording and playback. In this study an equation analogous to the Nyquist sampling theorem is derived using the geometry of a phonograph pickup system and simple calculus to obtain an absolute theoretical frequency limit for analog recordings based on the size of the pickup and the speed at which the time signal is written and read. This equation may prove useful in designing analog recording systems specific to a desired frequency band response.
Outdoor to indoor A-weighted sound level reduction of typical modular classrooms and assessment of potential performance improvements based on the outdoor-indoor transmission class spectrum.
Authors: Stewart Noral
Published: October 2013

Several different designs for modular classrooms from various parts of the United States were examined for their expected A-weighted outdoor to indoor sound level reduction based on the sound spectrum used for the ASTM E1332 Outdoor Indoor Transmission Class (OITC). Analysis was simplified by using an average absorption effect and the OITC rating of walls and roofs as estimated using the INSUL computer program. Results showed a significant variation among designs with the better designs providing a good foundation for further improvement.
Flight systems aeroelastic-acoustics simulation.
Authors: Gupta Kajal K. , Choi Sangbum , Ibrahim Adem
Published: October 2013

Many practical problems, such as flight vehicles, are often characterized by complex interactions among a number of primary disciplines as structures, fluids, controls, and propulsion, among others. In some critical flight regimes as transonic flow, the dynamics behavior of fluids tend to be highly complex, needing computational fluid dynamics (CFD)-based modeling, rather than linear aerodynamic methods, for accurate prediction of unsteady flow. Such unsteady pressure values, generated by the interaction of fluids and elastic structures, could then be conveniently used to compute acoustic wave frequencies and sound levels. For accurate simulation of complex engineering problems such as advanced aircraft, it is necessary to employ unstructured grids to model the fluid discipline, such being mostly the case for structure modeling. Effective CFD-based aeroelastic modeling, using finite element discretization of both fluids and structures, employing unstructured grids, have been efficiently utilized for simulation of Hyper-X and F-18 aircraft. In connection with a current ongoing project Stratospheric Observatory For Infrared Astronomy, evaluation of acoustic activities in the cavity of the modified Boeing 747SP aircraft, housing the telescope, became a crucial issue. The multidisciplinary code STARS7 was recently extended to include the acoustic effects.
Removing unwanted signals from wall pressure and vibration measurements on a structure subjected to turbulent boundary layer excitation.
Authors: Bonness William , Jenkins David , Capone Dean
Published: October 2013

Fluid-structure interaction experiments typically involve measurements of the excitation force (or pressure) and the corresponding vibration response to the excitation. In addition to desired flow information, measured turbulent boundary layer wall pressure data often include unwanted signals such as acoustic pressures and vibration induced pressures. Measured vibration data on a structure can also include unwanted electrical noise and vibration energy from adjacent structures. A noise removal technique is presented, which allows one to remove an unlimited number of unwanted correlated signals from a set of measured data. In its simplest form, this technique is related to the coherent output power. However, the more general technique provides an ability to remove multiple signals and to retain complex values (magnitude and phase). These advantages can yield significantly greater information of the flow field and structure under investigation. This technique is demonstrated using measurements from an aluminum cylinder internally filled with water flowing at 20 ft/s.
A theoretical model to simulate two-phase vibroacoustical frequency and damping effects in a pipe.
Authors: Debut Vincent , Antunes Jose
Published: October 2013

In order to avoid excessive flow-induced vibrations in industrial components operating in two-phase flow, the analysis and understanding of energy dissipation mechanisms are of prime importance. Several experimental studies since the classic work [L.N. Carlucci, “Damping and hydrodynamic mass of a cylinder in simulated two-phase flow,” J. Mech. Des. 102, 597–602 (1980)] revealed the strong dependence of two-phase damping on the characteristics of the two-phase flow, particularly the void fraction and fluid used. All these investigations agreed in the complexity of the dissipation processes involved. To tackle such an intricate problem, we consider in this paper the fluid-structure dynamics of a pipe filled with a bubbly liquid which interacts with two single-degree of freedom piston terminations. Several formulations for the coupled problem are stated and compared, the two-phase acoustics being based on a homogeneous mixture formulation [L. van Wijngaarden, “One-dimensional of liquids containing small gas bubbles,” Annu. Rev. Fluid. Mech. 4, 369–396 (1972)]. Then, extensive numerical computations are performed. Plots of the evolution of the modal frequencies and damping as a function of the mixture void-fraction are obtained, and the corresponding vibroacoustical transfer functions and numerical time-domain simulations are presented.
Performance comparison of viscoelastic materials typically used for acoustical damping of aircraft fuselage structures.
Authors: Cintosun Esen , Atalla Noureddine , Stecenko Tatjana , Bolduc Maxime
Published: October 2013

Vibration tests were performed to compare typical viscoelastic damping materials attached to two distinct representative aircraft fuselage structures. The fuselage structures were a 19×20-in2 Al ribbed panel and a 20×21-in2 carbon composite ribbed panel. A shaker was used to excite each panel with and without viscoelastic material treatment. Each test structure along with the shaker was situated in an environmental chamber to collect measurements at low temperatures of −20, −30, and −40 °C. A laser vibrometer was used to collect velocity measurements at 15–26 random locations on the surface of the panels. The parameters that were compared included input mobility, damping loss factor (DLF), and space averaged squared velocity. The DLF values that were obtained using the decay rate, power input, and half-power bandwidth methods were presented as averages in 1/3 octave frequency bandwidths (from 100 to 2500 Hz). The space and frequency averaged squared velocity was used to quantitatively categorize the viscoelastic damping materials. The effect of viscoelastic damping material coverage was also evaluated by comparing measurements collected at 50% and 80% coverages. The results of this experimental study are being utilized in finite element modeling, optimization of viscoelastic material treatment, and development of alternatives to viscoelastic damping. [Work supported by the MTI Polyfab Inc.]
Harmonic control of vibration and sound radiation of a plate using a virtual impedance approach.
Authors: Quaegebeur Nicolas , Micheau Philippe , Berry Alain
Published: October 2013

The present work focuses on the harmonic control of plate vibration and radiation using distributed collocated piezoceramic units. The objective is to add virtual impedance units on the plate and to optimize its complex value for each frequency. In active damping (velocity feedback controllers), the added virtual impedance is equivalent to a resistive effect. In the present study, the added virtual impedance is chosen arbitrarily in the complex plane. Each added impedance unit is composed of a collocated PZT actuator and a PVDF sensor. This approach allows obtaining dual variables for harmonic control. In order to tune the PZT voltage at each frequency, the virtual impedance approach is implemented with a harmonic adaptive control. The stability is always ensured with a centralized processing, and for certain frequencies, a decentralized processing can be used. Preliminary theoretical study is carried out to assess the behavior of the smart panel for different added virtual impedances for a mechanical excitation by a shaker. Those numerical results are compared to experimental measurements performed in an anechoic chamber. Our proposed strategy compare favorably to common strategy of active damping to reduce harmonic sound radiation. Hence, a reduction of 20 dB can be achieved at certain frequencies.
Noncontact surface wave testing of pavements using microphones.
Authors: Ryden Nils , Lowe Michael J. S. , Cawley Peter
Published: October 2013

Pavements are constructed using several layers of materials, and their durability depends mainly on the stiffness modulus and thickness of these strata. Surface wave testing is an effective tool to measure the stiffness and thickness of pavement layers. However, measurements are still based on spot testing with fixed receivers and source. The typical large size of pavements and the cost of closing down roads to make stationary testing makes these measurements impractical. We present experiments where a multichannel array of microphones and an automatic source are attached on a small trolley so that measurements can be taken almost continuously while moving. Measurements on asphalt or concrete pavement layers are based on supersonic leaky air-coupled surface waves. We also demonstrate that the same approach can be applied to softer granular pavement layers utilizing the seismic-to-acoustic coupling in poroelastic materials. Results show that microphones can be successfully used to measure correct surface wave dispersion curves even while moving along the surface. This opens up the possibility for faster on-the-fly surface wave testing of pavement layers since surface contact is no longer required. The theoretical background along with experimental results of the application to nondestructive testing of pavements will be presented.
Singing sands, musical grains, and booming sand dunes.
Authors: Patitsas Tom
Published: October 2013

The origin of the seismic and acoustic emissions from a bed of singing sands or musical grains, impacted by a pestle, is sought in a boundary layer, several millimeters thick, at the leading front end of the pestle. It is assumed that such a layer is the result of the fluidization of the grain asperities due to the high stress level at the pestle front end. Such a fluidization results in a very low modulus of rigidity and in a shear phase velocity about 1 m/s in the boundary layer. The frequencies of the shear modes of vibration, in such a layer, are compared with those determined experimentally. The frequencies depend weakly on the geometry of the leading pestle front. The same concept of the boundary layer can account for the emissions from plates of sand sliding on a dune surface and from grains shaken in a glass jar.
Variable resolution spectral/temporal features for automatic speech recognition.
Authors: Zahorian Stephen , Wu James
Published: October 2013

Many studies from speech science have shown that the mel frequency scale more closely matches speech perception than the linear frequency scale. Automatic speech recognition engineers have empirically demonstrated that the use of the mel scale results in more accurate speech recognition than that obtainable with features computed with respect to a linear frequency scale. The features most typically used for automatic speech recognition are mel frequency cepstral coefficients (MFCCs), along with delta and acceleration terms that represent the temporal evolution of MFCC over very short time intervals. However, the MFCC features do not encode the better temporal resolution that is possible at higher frequencies with low-frequency resolution. In this paper, a two-dimensional feature set is presented that incorporates good frequency resolution and low-time resolution at low-frequencies, and low frequency resolution and good time resolution at high frequencies. These features are computed from overlapping blocks with an effective length of approximately 100 ms at low frequencies and approximately 20 ms at high frequencies. Experimental results are given for phonetic recognition using the TIMIT database. The implications are that features that encode the temporal evolution of speech spectra are important for automatic speech recognition.
Comparison of a new frequency domain periodicity detector derived from a temporally stable power spectral representation and zero frequency filtering based periodicity detector.
Authors: Kawahara Hideki , Itagaki Hanae , Nisimura Ryuichi , Irino Toshio
Published: October 2013

Recently, an event based periodicity detection method was proposed using zero frequency filtering method [Yegnanarayana et al., ITRW Aalborg 2008]. The authors also introduced a new periodicity detector based on TANDEM-STRAIGHT [Kawahara et al., ICASSP2008 (2008)], a combination of temporally stable power spectrum estimation method for periodic signals and a spectral envelope recovery method based on consistent sampling [Unser, Proc. IEEE (2000)]. These methods provide a complementary set of information on excitation source signals of speech sounds. MATLAB implementation of these methods was evaluated using publicly available EGG databases and compared with popular existing methods. It was found that the zero frequency filtering based method runs eight times faster than real time even with MATLAB implementation and yielded comparable performance to popular methods. The TANDEM-STRAIGHT based methods also yielded comparable performance and additional information that is useful to represent diplophonia, for example. Detailed analysis examples and their application for TANDEM-STRAIGHT will also be discussed.
Vowel and speaker classification based on multivowel linear discriminant classfication.
Authors: Yonovitz Al , Anderson Ryan , Van Orden Sarah
Published: October 2013

Accurate and automated voice or speaker identification has been a major goal for those involved in forensic issues. In addition, voice and speaker identification has many applications in security and business. Numerous previous efforts to derive features for speaker classification have failed to achieve a sufficiently low-error rate. In this study a linear discriminant analysis was independently performed for vowel formants (F1, F2, F3) for each of ten vowels. The standardized canonical discriminant coefficients (SCDCs) weight each of the formants toward the group separation. These SCDC values are then linearly combined to form a single scalar for each vowel. An analysis that considered the multivariate data vector composed of the ten vowel scalars was then used for classification. All analysis was accomplished with the classic archived data set of Peterson and Barney (1952) and an additional data set derived for this study. The classification was over 98% accurate in separating males (M), females (F), and children (C) into subgroups, greater than any other study has provided. The methodology was used in a manner which tested specific speaker identification.