bluuebunny/crossref_metadata_2025 huggingface dataset:
An effect of top-down processing on intensity discrimination in the sensory-trace mode
Authors: Hafter Ervin R. , Bonnel Anne-Marie , Gandhi Swapan
Published: May 2008

A traditional measure of intensity discrimination presents a standard stimulus followed by a test. In one perceptual strategy, the subject compares the test to a coded version of the standard in long-term “context-coded memory” while, in another, the comparison is to a copy of the standard held in short-term “trace memory” [Durlach and Braida, J. Acoust. Soc. Amer, 46, 372-383 (1969)]. Subjects can be induced to use trace memory by roving the level of the standard randomly from trial to trial. It is well known that roving lowers overall performance, but by parsing individual standards we find that the loss is not uniform. Rather, an inverse correlation with the roved levels shows a severe departure from Weber's Law, with slopes of about -6 dB over 20 dB of roving. This can be modeled in terms of an additive noise that, unlike the multiplicative noise implied by Weber's Law, is constant across levels. We believe this represents top-down processes involved in the use of stimuli held in rehearsal memory.
Contribution of very low modulation frequencies to speech intelligibility in a competing speech task
Authors: Füllgrabe Christian , Stone Michael A. , Moore Brian
Published: May 2008

Vocoder studies have demonstrated that low-rate temporal-envelope cues (below ∼ 50 Hz) in a few frequency bands are sufficient for good speech identification in quiet. From the effects of low pass and high pass filtering of the temporal envelopes extracted from band pass filtered sentences, Drullman et al. [J. Acoust. Soc. Am. 95, 2670-2680 (1994); J. Acoust. Soc. Am. 95, 1053-1064 (1994)] argued that modulation frequencies between 4 and 16 Hz are the most important for speech identification in a steady background noise. However, Stone and Moore (JASA, in press) found that modulation frequencies below 2 Hz were also beneficial when the speech signal was processed through an 8-channel (but not a 12-channel) noise vocoder. The present study investigated speech identification for nine normal-hearing listeners using IEEE sentences presented with a competing talker; the mixed signal was processed using a 6-, 9-, or 15-channel noise vocoder. Low-rate temporal-envelope cues were manipulated using a compressor for each channel. Compression speed varied from slow to fast, so as to progressively degrade modulation frequencies below 4 Hz. Independent of the spectral resolution of the vocoder, mean intelligibility decreased as the compression speed increased; even modulation frequencies below 1.3 Hz contributed significantly to speech intelligibility.
Amplitude modulation depth discrimination in hearing-impaired and normal-hearing listeners
Authors: Ewert Stephan D. , Volmer Jutta , Dau Torsten , Verhey Jesko
Published: May 2008

The processing of amplitude modulations (AM) of sounds is assumed to be crucial for decoding and understanding of speech in humans. Since hearing-impaired (HI) listeners often suffer from severely hampered speech intelligibility, particularly in reverberant or noisy environments, they might also show degraded performance in AM processing tasks. However, several studies indicated a similar or even better performance in AM detection tasks for sensorineural HI listeners than for normal hearing (NH) listeners when reduced audibility was compensated. In addition to AM detection, this study investigates the differential processing of amplitude modulation depth in HI and NH listeners. AM-depth discrimination of a 4-, 8-, and 30-Hz sinusoidal AM, imposed on a 1- or 4-kHz pure-tone carrier, was measured. The AM of the standard ranged from being well detectable to near threshold. AM-depth discrimination thresholds strongly varied among HI listeners and were elevated in comparison to NH for high standard depths. A model of AM processing is suggested incorporating an individually adjusted simulation of the auditory periphery. To account for the data of HI listeners, however, the key element appeared to be an increased internal noise in the AM-depth domain. Consequences for speech perception are discussed.
The role of consonant duration and amplitude processing on speech intelligibility in noise
Authors: Digiovanni Jeffrey J. , Stover Ashley K.
Published: May 2008

The effects of hearing loss on speech intelligibility have led researchers to search for methods to ameliorate these effects. Clear speech has been studied extensively to determine the specific acoustic and phonetic differences between speech spoken in a conversational versus a clear-speech mode. Two characteristics of clear speech, increased consonant duration and selective-consonant amplification were investigated. These components were applied individually to Hearing-in-Noise Test sentences and consonant-vowel (CV) pairs. Stimuli were presented in sound field to hearing-impaired and normal-hearing individuals in the presence of speech-shaped noise. Percent correct was measured for HINT sentences and confusion matrices were developed for the CV pair intelligibility tests. An information transmission analysis was performed. During HINT sentence presentations, subjects benefited most from lengthening the duration of the consonant, maximally in the 30 ms condition, beyond which significant degradation in intelligibility was observed. Consonant specific amplification of 10 dB resulted in improved intelligibility for sentences. These data suggest that the two processing components are viable to improve speech intelligibility and they, however, also suggest that there is a limit to the processing whereby benefits are no longer observed.
Resolution, spectral weighting, and integration of information across tonotopically remote cochlear regions: hearing-sensitivity, sensation level, and training effects
Authors: Espinoza-Varas Blas
Published: May 2008

This paper examines how listeners resolve, weight, and integrate redundant sensory information from tonotopically remote cochlear regions, one in the mid (m) the other in the high (h) frequencies. Subjects listened to two-component complexes and attempted to resolve and integrate simultaneous differences in frequency (DFm) at ≈1000 Hz and in duration (DTh) at ≈3500 Hz. Discrimination performance was studied as a function of hearing sensitivity, sensation level (SL) and training. Normal hearing listeners tend to resolve both DFm and DTh and integrate the information if the component SLs are equal, but resolve only the louder component difference if the SLs are very unequal; i.e., spectral weighting is biased and integration is limited or nil. Much the same pattern obtains when the low SL in H results from high-frequency impaired sensitivity (IS). Once established, equating the component SLs (e.g., by high-frequency amplification) is often ineffective to rectify the weighting bias; doing so requires extensive discrimination training with the low SL component. With IS, an anomalous trend obtains when the SL is much lower for (m) than for (h): the resolution of DFm is equal to or better than that of DTh, and integration is quite efficient, possibly reflecting tonotopic map reorganization.
Pitch discrimination of simultaneous and nonsimultaneous complexes across spectral regions
Authors: Borchert Elizabeth , Oxenham Andrew J. , Micheyl Christophe
Published: May 2008

Detecting differences in fundamental frequency (F0) across different spectral regions may be important in the perceptual organization of both simultaneous and sequential sounds, but it is unknown whether the same mechanisms are involved in these two cases. Here we directly compare the detection of pitch differences between pairs of complex tones presented either simultaneously or sequentially. 28 normal-hearing listeners heard two pairs of complex tones in each trial and indicated the pair in which the pitches of the two tones differed. The complex tones in each pair were bandpass filtered to have nonoverlapping spectra, but at least some resolved harmonics in all cases. Performance was generally better in the simultaneous than in the sequential condition. In a follow-up condition, it was found that introducing an onset asynchrony between the two spectral regions in the simultaneous condition resulted in impaired performance, approaching that found in the sequential case, despite the duration of the simultaneous portion remaining the same. The results suggest that perceived fusion, rather than an explicit F0 extraction and comparison may underlie the detection of F0 differences between simultaneous groups of harmonics in distinct spectral regions. [Work supported by NIH grant R01 DC 05216.]
Phoneme recognition as a function of the number of auditory filter outputs
Authors: Apoux Frederic , Healy Eric W.
Published: May 2008

It has been proposed that listeners take advantage of brief “coups d'oeil” when processing speech in noise. These glimpses can be characterized both in time and frequency. The obligatory role of the auditory filters in determining the nature of any further processing suggests that the frequency extent of a glimpse should be equivalent to that of an auditory filter. Accordingly, it is hypothesized that the spectral characteristics of glimpses primarily relate to the available number of auditory channels. The present study investigated the number of auditory filter outputs needed to identify phonemes in quiet. Stimuli were first restricted to 80-8000 Hz and then split into 30 contiguous auditory filter width bands. Normal-hearing listeners were presented with N bands whose spectral location was selected randomly from trial to trial. No signal was presented in the other bands. Consistent with previous studies, performance gradually increased with the number of bands. An asymptote was reached with 24 and 16 bands for vowels and consonants, respectively. While high levels of speech understanding are typically observed with as few as 4 channels of spectral information, our results indicate that accurate phoneme recognition requires combination of a much larger number of auditory filter outputs.
Critical assessment of operational path analysis: mathematical problems of transmissibility estimation
Authors: Gajdatsy Peter , Janssens Karl , Gielen Ludo , Mas Peter , Van Der Auweraer Herman
Published: May 2008

Classical transfer path analysis (TPA) is a widely used and reliable method for tackling noise and vibration problems. But due to its complexity and time-consuming measurement procedure there is an ongoing research for simpler and faster methods. One such method, most often referred to as operational path analysis (OPA), was presented two years ago and has quickly gained popularity in the NVH field for its speed and ease of use. To estimate path contributions transmissibilities are calculated from the measured operational reference and target accelerations and sound pressures. This new method is now critically examined and compared to a reliable classical TPA measurement. The results of this examination reveal three significant weaknesses. This paper focuses on the problems related to the estimation of transmissibilities which mostly arise from the limited amount of orders present in the signal and the coherence between inputs. It is shown that despite the advantages of the method, it is not applicable in many situations and has to be used with care for it can easily give misleading results.
Decentralized control of sound radiation from an aircraft-style panel using iterative loop recovery
Authors: Schiller Noah H. , Cabell Randolph H. , Fuller Chris R.
Published: May 2008

A decentralized LQG-based control strategy is designed to reduce low-frequency sound transmission through periodically stiffened panels. While modern control strategies have been used to reduce sound radiation from relatively simple structural acoustic systems, significant implementation issues have to be addressed before these control strategies can be extended to large systems such as the fuselage of an aircraft. For instance, centralized approaches typically require a high level of connectivity and are computationally intensive, while decentralized strategies face stability problems caused by the unmodeled interaction between neighboring control units. Since accurate uncertainty bounds are not known a priori, it is difficult to ensure the decentralized control system will be robust without making the controller overly conservative. Therefore, an iterative approach is suggested, which utilizes frequency-shaped loop recovery. The approach accounts for modeling error introduced by neighboring control loops, requires no communication between subsystems, and is relatively simple. The control strategy is validated using real-time control experiments performed on a built-up aluminum test structure representative of the fuselage of an aircraft. Experiments demonstrate that the iterative approach is capable of achieving 12 dB peak reductions and a 3.6 dB integrated reduction in radiated sound power from the stiffened panel.
Concept of active reverberation box
Authors: Suzuki Hisaharu , Omoto Akira
Published: May 2008

A quite small reverberation box with active controller system is proposed. If we try to measure the characteristics of acoustic materials, such as glass wool, in a small sized enclosure, there are few dominant modes which cause the condition that the incident wave to target material is biased toward the specific direction. As a result, the measured result shows the irregular distribution which depends on measuring point or evaluating method such as the reverberation time reading. The proposed system which utilized the active control enables us to create a plane wave with arbitrary direction in the enclosure. Therefore it has possibilities of solving the problems caused by insufficient incident condition. Also, in this study, the two evaluating strategies that the traditional reverberation time method and the steady state analysis method are concerned.
Reproduction of loud low-frequency soundfields with Boundary Surface Control
Authors: Friot Emmanuel , Guillermin Régine , Pinhède Cédric
Published: May 2008

Many research works have focused on multichannel soundfield reproduction during the past decade, and complete systems are now commercially available for 3D sound control e.g., with Wave Field Synthesis. However, the usual underlying approximation of free-field propagation is not valid when reproducing high level low frequency noise, such as a sonic boom, inside a closet; a specific sound control strategy is required in this case. In this context the paper presents an experiment of low-frequency soundfield reproduction with open-loop control of the acoustic pressure at a set of 30 microphones enclosing a listening area. It is shown that this so-called Boundary Surface Control strategy allows an accurate reproduction of 3D soundfields inside the LMA sonic boom closet, equipped with 16 wall-embedded noise sources, at frequencies from 3 to 200 Hz.
Experimental assessment of the performance of a “smart foam” absorber
Authors: Leroy Pierre M. , Herzog Philippe , Berry Alain , Atalla Noureddine
Published: May 2008

The signal processing implementation of a hybrid passive/active absorber (smart foam) made up from the combination of a passive absorbent (poroelastic melamine foam) typically efficient at the high frequencies, and a bonded, curved PVDF film, to enhance low frequency performance is considered. Three methods for obtaining the control signal are experimentally compared in the case of a plane wave excitation. A prototype of such smart foam has been built and tested in a waveguide (rectangular impedance tube) at frequencies between 100 Hz and 1000 Hz. The performances in term of the ratio of reflected to incident sound pressure are presented and discussed, comparing the three methods. The first method uses estimations of the transfer functions between the sources and two microphones in the tube to calculate off-line an optimal filter in the frequency domain. It is then compared with numerical simulations, based on a 3D FE model of the smart foam. The two other methods are based on a real-time adaptive control using a FXLMS algorithm, a unidirectional microphone as error sensor, and the primary source signal as reference signal. The control filter is obtained after the initial convergence, using either pure tone disturbance (secnd method) or broad band disturbance (third method).
Active vibration reduction applied to the compressor of an air-conditioning unit for trams
Authors: Bös Joachim , Janssen Enrico , Kauba Michael , Mayer Dirk
Published: May 2008

Within the framework of the European Integrated Project InMAR (intelligent materials for active noise reduction) active vibration control and active structural acoustic control approaches are applied to an HVAC (heating, ventilation, and air conditioning) unit used to air-condition the driver's cab of a tram. Measurements previously performed by the manufacturer of the unit indicated that annoying tonal noise in the 50 Hz and 100 Hz 1/3 octave bands inside the driver's cab is mainly caused by the vibration of the compressor mounted in the HVAC unit on the roof of the tram. Therefore, two different concepts for the design of active compressor mounts were developed that are used to reduce the vibration excitation of the HVAC unit's housing. The first one is an active tuned vibration absorber whose natural frequency can be adapted by means of piezoelectric patch actuators and which behaves as a vibration compensator at higher frequencies. The second one is an active mount based on four piezoelectric stack actuators and an elastomer part that deflects the effective direction of the piezo stack displacement by 90 degrees while amplifying the displacement by a factor of approximately 30. These two concepts are compared with each other.
Training by research - program of, and experiences in, the Smart Structures Research Training Network project
Authors: Augusztinovicz Fülöp
Published: May 2008

Smart Structures is a Marie Curie Research Training network (RTN) named “A Computer Aided Engineering Approach to Smart Structures Design”. The overall objective of the RTN is to develop an educational and research framework for developing macroscale noise control applications based on intelligent material systems. The scientific and technological objective of the network is to advance the state-of-the-art in smart materials research to the level of system integration and industrial applicability, with a specific focus on solutions for noise and vibration reduction and for vibration-based damage detection. The paper gives a short overview of the project partners' activity and summarizes the experiences of the researchers and their tutors after the first 18 months of the network.
New multichannel modified filtered-x algorithms for active noise control using the dichotomous coordinate descent method
Authors: Albu Felix , Paleologu Constantin
Published: May 2008

In this paper, several multichannel modified filtered-x algorithms for active noise control or acoustic equalization systems using the dichotomous coordinate descent method (DCD) are introduced. This multiplierless and divisionless method is used for avoiding the matrix inversion that appears in adaptive algorithms such us fast transversal filters (FTF), recursive least square (RLS) based algorithms, affine projection (AP) or its fast versions. The study is focused on the important computational savings given by the use of DCD method, the effect on the convergence properties and stability of the investigated algorithms. A comparison of their convergence performance in case of using non-ideal noisy acoustic plants is also given. It is proved by simulations that the use of the dichotomous coordinate descent method can be an interesting option for reducing the computational cost of practical multichannel algorithms for ANC or acoustic equalization algorithms.
Estimation of target-to-interferer ratio using the Auditory Image Model
Authors: Carlin Michael
Published: May 2008

Performance in speech processing applications such as speaker recognition becomes severely degraded when the short-time audio being analyzed contains more than one speaker. However, it has been shown that if the audio is only minimally-corrupted by the interfering speech, i.e., the target-to-interferer ratio (TIR) is sufficiently large, then accurate recognition results can still be achieved. During phonation, estimation of TIR is especially critical since uncorrupted vowel sounds contain important speaker-discriminating information. This research investigates a method to estimate the relative intensity of interfering speech using the Auditory Image Model (AIM) of Patterson et al. (J. Acoust. Soc. Am., 98, 1890-1894 (1995)). The proposed TIR estimator attempts to exploit both the apparent high resolution in the simulated Neural Activity Pattern and variation in cross-channel Strobe Point correlation when observing overlapping vowel sounds. Experiments were conducted for five canonical male vowels which were perceptually-scaled using the STRAIGHT algorithm (Chapter in Speech Separation by Humans and Machines, P. Divenyi, ed., Kluwer Academic Publishers, 2005) and superimposed at varying levels of TIR. Results suggest that the proposed approach is a promising step towards both detecting the presence and relative intensity of an interfering speaker.
Extraction of likelihood-ratio forensic evidence from the formant trajectories of diphthongs
Authors: Morrison Geoffrey Stewart , Rose Phil , Kinoshita Yuko
Published: May 2008

The likelihood-ratio approach to forensic speaker recognition seeks to determine the likelihood that one would observe the evidence, the acoustic difference between suspect and offender speech samples, under the hypothesis that they were produced by the same speaker versus under the hypothesis that they were produced by different speakers. Before the results of a scientific forensic technique can be presented in court, it is necessary to demonstrate its efficacy. This presentation tests the efficacy of extracting information from the formant trajectories of diphthongs. Differences in physiology and learned motor patterns could potentially lead to different speakers producing quite different formant trajectories which could in turn lead to strong forensic evidence. The data tested were /aI/, /aυ/, /eI/, /"open o"I/, /oυ/, /i"schwa"/, and /ε"schwa"/ tokens produced in several phonetic contexts by 27 male speakers of Australian English. Cubic polynomials were fitted to each vowel token, and the coefficient values were used in a multivariate-kernel-density procedure which calculated likelihood ratios. Cross-validated same-speaker and different-speaker comparisons were made, resulting in a series of same-speaker and different-speaker likelihood ratios for each vowel phoneme. Results indicated that substantial strength of evidence with respect to speaker identity can be extracted from diphthong formant trajectories.
Cross-language auditory-visual speech perception development
Authors: Burnham Denis , Sekiyama Kaoru , Erdener Dogu
Published: May 2008

Speech perception is auditory-visual (AV), but there is more visual information use by English than Japanese language perceivers (Sekiyama & Tohkura, 1993). This raises two questions: (1) when and (2) why do such differences emerge? In Experiment 1 with English- and Japanese-language children (6, 8, 11 years) and adults, AV speech perception (AVSP) was equivalent in English and Japanese 6-year-olds, significantly increased between 6 and 8 years in English- but not Japanese-language children, and remained stable over all ages for the Japanese. The origin of the Japanese-English adult difference in English language children's dramatic rise in AVSP between 6 and 8 years was investigated in Experiment 2. English-language children (5 to 8 years) and adults were tested for AVSP, reading, articulation, and language-specific speech perception (focus on native and away from non-native speech sounds, a known correlate of reading, Burnham, 2003). Regressions showed that language specific speech perception reliably predicted AVSP. It appears children use extra information (AVSP) in the face of challenging linguistic hurdles (learning to read), but only when the language is phonetically complex and the phoneme-to-grapheme conversion rules of the script are not transparent (English, but not Japanese).
Duration modeling for English letters embedded in Chinese speech
Authors: Lai Wen-Hsing
Published: May 2008

A review of existing multilingual TTS (text-to-speech) systems shows that the secondary language inserted into the primary language sounds more like isolated individual words in an alien language environment and not congruous with the primary language's prosody. Since the letter-by-letter spelling of English words or acronyms appears in Chinese speech quite often, a duration modeling approach for English letters embedded in Chinese speech is proposed to make the English congruous with the primary language's tempo. It takes several major factors as additive factors and estimates all model parameters by an EM (expectation-maximization) algorithm. Experimental results showed that the standard deviation of the duration from the test set was greatly reduced from 59.82 to 9.37 ms by the duration modeling while eliminating effects from factors. The root mean squared error between the original and estimated durations was 9.35 ms for the open tests. Experimental results have confirmed its effectiveness on isolating several main factors that seriously affects the duration. Moreover, the estimated value of the factors agreed well to our prior linguistic knowledge. Besides, the hidden state labels produced by the EM algorithm were linguistically meaningful.
Speech segmentation in monolingual and bilingual infant learners of Canadian English and Canadian French
Authors: Polka Linda , Proulx Jennifer , Sundara Megha
Published: May 2008

Speech segmentation skills emerging in infancy are influenced by the infant's native language. English infants favor a stress-based strategy (Jusczyk et al., 1999) whereas French infants favor a syllable-based strategy (Nazzi et al., 2006). Cross-linguistic findings from our lab, based on between group comparisons, show that monolingual 8-month-olds learning either Canadian English or Canadian French segment bisyllabic words in their native language but not in a rhythmically-different non-native language. Canadian French infants fail to segment Canadian English; Canadian English infants fail to segment Canadian French. Bilingual infants (exposed to both languages) were tested in each language on separate days. Although they appear to segment in both languages, attrition has high in the second test session. To assess cross-language segmentation more directly we tested monolingual and bilingual 8-month-olds using a task designed to assess segmentation in both languages in the same infant within a single test session. Findings confirm that monolingual 8-month-olds fail to segment bisyllabic words in a rhythmically-different non-native language. Preliminary data show that bilingual 8-month-olds segment only in the language that is favored in their language input. Thus, in early stages of speech processing all infants appear to develop speech segmentation strategies that are optimal for one language.
Some acoustic cues in the detection of the Nepalese aspiration
Authors: Khatiwada Rajesh
Published: May 2008

In terms of laryngeal setting, most languages have only two types of stops. In the Nepalese sound inventory there are four types of stops: voiceless unaspirated, voiced unaspirated, voiceless aspirated, and voiced aspirated (or breathy voiced). The main objective of this paper is to present the prominent and consistent acoustic cues that differentiate the aspirated segments from their unaspirated counterparts of the Nepalese coronals - the dental-alveolar (t,th,d,dh), retroflex (T,Th,D,Dh) and affricates (ts,tsh,dz,dzh). VOT (Voice Onset Time) (Lisker and Abramson 1964) is the familiar model used to characterize three of these phonation types - voiced, voiceless, and aspiration stops. Some authors have pointed out that this method is not able to distinguish the plain voiced and voiced aspirated stops. An alternative segmentation model proposed by Mikuteit and Reetz (2007) is adopted here. Beside the analysis of the closure duration of segments, measuring the beginning of the burst or its release, we use other acoustic cues to characterize Nepalese aspiration such as F0 lowering, the structure of the lower harmonics and spectral tilt (Hanson 1997, Stevens 1998). This experimental work is based on the corpus based on four native speakers of Nepali.
Prosodic realization of focus on verbal phrases in two Chinese dialects
Authors: Chen Yiya
Published: May 2008

A well-established observation in West-Germanic languages is that when a verbal phrase (e.g., likes movies) is uttered as new information (i.e., focused), speakers often accent (via F0 changes) the noun only (e.g., Ladd 1980 and 1996). Listeners, in turn, find that a single accent on noun sounds as appropriate as when both verb and object are accented (e.g., Gussenhoven 1987, Birch and Clifton 1995, Welby 2003). This study will report results of an experiment which was designed to examine the prosodic realization of focus on verbal phrases in two tonal dialects - Beijing Mandarin and Shanghai Chinese. Both dialects use F0 changes to indicate word meanings, but they differ in tone sandhi patterns. That is, they differ in how words group into phrasal tonal domains and how lexical tones undergo changes within the domain. Two types of focus were elicited: One provides new information to Wh-questions and the other corrections of wrong information in the preceding discourse. The goal of the study is to contribute to the development of a cross-linguistic theory of the prosodic realization of focus. [Work supported by the VENI research grant from the Netherlands Organization for Scientific Research.]
Determining acoustical parameters using cochlear modeling and auditory masking
Authors: Van Dorp Schuitman Jasper , De Vries Diemer
Published: May 2008

The acoustical qualities of a concert hall or any other room are generally expressed using acoustical parameters determined from impulse responses. From microphone array measurements it turned out that these parameters can fluctuate severely over small distances, whereas the perceptual cues for which these parameters are supposed to be a measure remain constant. This means that a local parameter value has a very low predictive value for acoustic quality. In this research, cochlear modeling techniques and simulations of auditory masking effects have been applied to model human hearing. These techniques together model various stages in the auditory path, like the movement of the basilar membrane inside the cochlea and mechanisms inside the brains. It turns out that determining acoustical parameters using this representation leads to results which show much less spatial fluctuations, and are closer to human perception.
Diffuseness and intensity analysis of spatial impulse responses
Authors: Lokki Tapio
Published: May 2008

Spatial impulse responses, meaning responses measured with a microphone grid, were measured in seven concert halls. The microphone array consisted of 12 omni-microphones, enabling a construction of three intensity pairs (in x, y, and z directions) with 1 cm spacing and three intensity pairs with 10 cm spacing. In each hall impulse responses were measured with at least three loudspeaker and four microphone positions. They were analyzed with directional audio coding methodology, which enables analysis of diffuseness and instantaneous intensity as a function of time and frequency. In other words, with this analysis it is possible to analyze the directions of early reflections and to estimate diffuseness of sound field in a measurement position. Preliminary results indicate that diffuseness is quite similar in different positions in one hall, but it varies more between halls. The directions of early reflections are hard to visualize, however some example videos are shown to get an idea about the possibilities of such an analysis technique.
Relevance of acoustic parameters for musician communication
Authors: Guthrie Anne , Braasch Jonas
Published: May 2008

In situations of indeterminate musical performance (particularly in telepresence, where acoustic degradation is a frequent concern), autonomous musical communication, both practical and artistic, forms the crux of the musical material. The relevance of stage acoustic and psychoacoustic parameters to contemporary performance situations must be re-examined with regards to the heightened importance of communication. Parameters developed by A. C. Gade, J. Meyer, and D. Brungart are starting points for this examination. Experiments are conducted with four instrumentalists playing excerpts from a composition by Christian Wolff (open notation allows for measurable variations depending on communication quality), communicating telematically between two virtual environments. Parameters determined by questionnaire to have the strongest effect on quality and efficiency of communication are varied at intervals and evaluated by the performers. Five parameters are tested: Self-to-Others Ratio, Commonality of Aural Space, Masking of Individual Voices, Visual-Audio Synchrony, and Position/Directivity. The performances are recorded and analyzed for variations in musical content, such as dynamics, rhythm, register, and density. The three sets of data (objective parameters, performer evaluation, and musical analysis) are compared to determine the effects of the selected parameters on musical communication.
On the use of diffusion equations to model the acoustics of coupled rooms
Authors: Billon Alexis , Valeau Vincent , Picaut Judicaël , Foy Cédric , Sakout Anas
Published: May 2008

The acoustics of coupled rooms are characterized by energy exchanges through apertures and/or partition walls. The use of systems of diffusion equations allows to predict the temporal and spatial energy distributions in these configurations quite accurately. In this presentation, the diffusion formalism for room acoustics-prediction is summarized. The systems of equations to be solved in the case of coupling through an aperture and through a partition wall are presented. For two rooms coupled through an aperture (two classrooms connected through an open door), the results obtained with the diffusion model are compared to experimental data, in terms of sound pressure levels and sound decays. On the other hand, for the case of two classrooms connected through a partition wall, the diffusion model is compared to experimental data in terms of sound pressure level difference only. Finally, an engineering application is presented in the configuration involving a workroom including multiple sound sources (e.g., machines) connected to offices through open and closed doors.
The application of acoustically coupled spaces in concert hall design
Authors: Jaffe Christopher
Published: May 2008

At mid-twentieth century, a hall having a reverberation decay curve with a late arriving extended second slope was considered an acoustical failure. It was assumed that that the level of the extended reverberation in the hall would interfere with the ongoing running music of the ensemble and reduce orchestral clarity, definition and transparency. To my knowledge, only two practitioners pioneered the utilization of physical acoustic coupling in concert shell and concert hall design during those early years. One was Russell Johnson, when he was with Bolt Beranek and Newman and later with his own firm Artec. The other was me with both my firms Stagecraft Corporation and Jaffe Acoustics. This paper discusses how the need to improve the concert hall environment of American multi-use theatres led to the application of coupling techniques in these halls and how both Mr. Johnson and I later applied physical acoustic coupling in single room concert spaces.
Using Seagliders for acoustic receiving and communication
Authors: Howe Bruce M. , Boyd Michael L.
Published: May 2008

Underwater gliders are beginning to be used as tools in ocean acoustics and acoustical oceanography. Results from several experiments conducted in summer 2006 with Seagliders equipped with acoustic modems and receivers are described. Off Kauai, a glider received signals from the Acoustic Thermometry of Ocean Climate/North Pacific Acoustic Laboratory 75 Hz source; subsequent coherent processing showed close to theoretical gain for 12 min records while moving away from the source at ranges >100 km with velocity 20 cm/s (measured by travel time, Doppler, and dead reckoning). In the Monterey Bay MB06 experiment, two-way communications between other subsea platforms and shore via the acoustic modem-equipped glider was demonstrated (albeit with latency). The results support the future use of gliders as precision navigated platforms, communication and time distribution nodes, and thermometry/tomography mobile receivers. Work supported by the Office of Naval Research.
Speech perception in classroom noise and reverberation by children with typical and impaired hearing - Preliminary results
Authors: Iglehart Frank
Published: May 2008

A child's ability to perceive speech in the classroom influences academic progress. In this study, students with a various degrees of hearing loss perceive spoken words in sentences in a classroom. The room contains multiple noise levels and reverberation times (RTs). Classroom reverberation is measured using standardized procedures (with slight modifications to ASTM C423-02a: X2). RTs are controlled by quantities of acoustic panels in the room. Using the BKB-SIN Test, the speech-to-noise ratio at which the students perceive 50% words correctly (i.e., SNR-50) are measured in three reverberant conditions. Data collected to date indicate that students with severe-to-profound hearing loss (n=15; ages 8-16 years) demonstrated average SNR-50s of +12 dB (SD=4), +13 dB (SD=4), and +17 dB (SD=4) for conditions of 0.3, 0.6 and 0.9 s RT, respectively. Students with typical hearing (n=14; ages 8-16 years) had SNR-50s that averaged -4 dB (SD=2), -3 dB (SD=2) and -2 dB (SD=2), respectively. Performance/intensity curves are also calculated in order to estimate minimal optimal listening conditions for each RT. This is an ongoing study. The session presentation will include emerging data on children with other degrees of hearing loss.
An Ambiophonics system for the study of intelligibility in the virtual classrooms
Authors: Prodi Nicola , Farnetani Andrea , Fausti Patrizio , Pompoli Roberto
Published: May 2008

In recent years the investigations on the intelligibility has profited by the increasing efficiency of the simulation and rendering technologies of virtual acoustics. In particular the latest systems are well suited to reproduce sound fields whose tridimensional characteristics are quite close to the real ones both objectively and subjectively. In this framework the Ambiophonics system, consisting of the merging of a double stereo-dipole and of an first order Ambisonics system, has gained particular attention. In the configuration used in this work the system employs twelve loudspeakers and is fed either with measured binaural and B-format data or by simulated ones. Based on this apparatus listening tests have been prepared in the Italian language to investigate the acoustics of classrooms and to focus on the effect of the directional characteristics of noise (i.e., fan-coils or tapping from the upper floor) on the intelligibility of words. The whole apparatus is driven by a Labview application which has also the aim of collecting the responses by the testers.
The effect of amplification on children's performance in the classroom
Authors: Shield Bridget M. , Dockrell Julie E.
Published: May 2008

The use of amplification systems in the classroom has the potential to reduce the impact of poor classroom acoustics for typically developing pupils and those with special educational needs (SENs). The immediate benefits of amplified acoustic signals on the performance of 253 primary school children, including 24 children with special needs, were examined. All participants were familiar with the use of the amplification systems. Children's performance was assessed on two verbal measures (spelling and oral comprehension) and one non-verbal measure, using a balanced repeated measures design. It was predicted that the effects of amplification would be evident for spelling and oral language comprehension, but that there would be no discernable effect on speed of processing. Children with SEN were expected to have an added advantage with amplification. The predictions were partially supported. Both gain score analysis and ANOVAs of performance scores with amplification, controlling for performance without amplification, revealed an effect on spelling only for the typically developing children, while children with SENs benefited in both the spelling and the oral comprehension tasks. When installed appropriately, amplification improves children's ability to decode single words; SEN children also benefit in terms of their ability to process oral input.
Education vs cost - Difficulties in implementing acoustical design standards in classrooms
Authors: Scanlon Patricia M. , Perry James , Cerami Victoria J.
Published: May 2008

Good acoustical design standards for classrooms are established - quiet background noise levels due to mechancial systems and exterior sources, use of absorptive finishes to control reverberation, walls, doors and floors able to control airborne and impact noise transmission. So why do so many schools fail to follow these guidelines? Are private schools even less likely to adhere to these standards? What are the construction and equipment options? We will discuss various educational standards, as well as the reticence of school planners to comply. We will discuss what schools often build, versus what one might say they should build, together with cost comparison. We will review case studies where guidelines were, and were not, followed - examples that reflect the acoustician's mission to educate educators on the value of acoustical design considerations in classrooms.
Mean vs event sound-level prediction: obtaining consistency between atmospheric data inputs, propagation models, and the predictand
Authors: Wilson David K. , Pettit Chris L. , Ostashev Vladimir E. , Lewis Matthew S.
Published: May 2008

The following, deceptively challenging, questions are addressed: What are the most suitable atmospheric data resources and propagation models for predicting event (explosion and other short duration) sound-exposure levels? Do these differ from those most suitable for predicting mean sound levels? Atmospheric data typically consist either of single, “snapshot” samples of the vertical profiles, as from a weather balloon, or average vertical profiles, as from climatology or a numerical weather model. Recent statistical studies, based on high-resolution atmospheric simulation, demonstrate the superiority of mean profiles for prediction of both mean and event sound levels, even when single-sample profiles are synchronized to and collected along the path of the propagation event. Running propagation models “blind” to the nature of the atmospheric inputs is shown to be hazardous: predictions from mean profiles lack turbulent scattering, thus underestimating sound levels near the ground, whereas predictions from single-sample profiles implicitly assume the turbulence has infinite horizontal extent, thus overestimating sound levels. Some desirable consistency results from numerically solving parabolic equations (PEs) for statistical moments of the sound pressure, rather than conventional deterministic PEs. The moment PEs directly predict mean sound levels or the expected value and variability of event sound-exposure levels.
Characteristics of road traffic noise level statistics for shielded areas
Authors: Forssén Jens , Hornikx Maarten
Published: May 2008

For noise immission, it is of interest to study other noise level statistics besides the long-term equivalent levels and maximum levels. By further analysis of the time variations of the noise level, an improved description of the negative effects of the noise may be achieved, for instance concerning perceived annoyance. Here, noise level histograms, i.e., probability density functions of sound pressure levels, from controlled recordings have been investigated. This has been made for a situation of special interest, which is a courtyard shielded from a dominating road traffic noise source. It has been reported previously that many shielded urban areas show levels that are considerably higher than the equivalent level, described as an upward tail of the histogram, which is not a usual characteristic of directly exposed areas. From the analysis made here, it is shown that the upward tail, i.e., the higher levels, of the shielded area is caused by locally occurring, unshielded road traffic events. It is concluded that the upward tail as a common characteristic of shielded urban areas may well be due to locally occurring noise events, for instance due to local road traffic.
A cellular automaton for urban traffic noise
Authors: Salomons Erik
Published: May 2008

Propagation of traffic noise in a city is a complex phenomenon, due to multiple reflection, diffraction, and scattering at irregular facades of buildings. These effects may be calculated with computer models based on numerical integration of the basic acoustic equations, but in practice these models can be applied only to small urban regions due to limited computer power. Here we propose a new approach for simulating urban traffic noise: a cellular automaton (CA) based on simple update rules for the configuration of cars in a city, and simple rules for propagation of sound to receivers. An example is presented for a square model city of 25 km2, consisting of 106 square cells. The CA employs a time integration step of 0.3 s, and includes noise contributions from all cars in the city. The fluctuating sound level is computed for a period of 24 h, both for a receiver along a street and for a receiver that is screened by buildings. While the sound level at the first receiver shows sharp peaks corresponding to passages of cars, the sound level fluctuations at the screened receiver are much smaller as most of the sound energy comes from distant cars in this case.
Elaboration of a sacttering matrix measurement procedure using the p-v probe
Authors: Kchaou Yamen , Taktak Mohamed , Ville Jean Michel , Haddar Mohamed , Foucart Félix
Published: May 2008

The scattering matrix which relates traveling waves amplitudes as state variables has been shown to be more attractive than transfer or mobility matrices since it reflects the fundamental duct nature: it gives a more complete description of the transmission, reflection, and conversion properties of the duct. In the University of Technology of Compiègne, an experimental procedure was developed to measure this matrix: a p-p probe mounted on a setup designed during DUCAT project is used to measure pressures at two cross sections on the both side of the test lined duct, then by using a modal decomposition and separation techniques, the scattering matrix is deduced. In this paper, a method to measure the multimodal scattering matrix based on the use of a p-v probe getting simultaneously the acoustic pressure and velocity at one section on the both side of the test duct is developed. A comparison of some acoustics values (scattering matrix coefficients, acoustic powersœ) of a hard wall duct straight duct obtained by each technique with the theory is presented to evaluate its advantages and limitations.