bluuebunny/crossref_metadata_2025 huggingface dataset:
Quantitative assessment of spatial sound distortion by the semi-ideal recording point of a hear-through device
Authors: Hoffmann Pablo , Christensen Flemming , Hammersho/i Dorte
Published: May 2013

A hear-through device combines a microphone and earphone in an earpiece so that when worn, one per ear, it can work as an acoustically transparent system allowing for simultaneous individual binaural recording and playback of the real sound field at the ears. Recognizing the blocked entrance to the ear canal as the ideal recording point—i.e., all directional properties of the incident sound field are recorded without distortion—it is critical for such device to be sufficiently small so that it can be completely inserted into the ear canal. This is not always feasible and the device may stretch out from the ideal position and thus distort the captured spatial information. Here we present measurements that quantify by how much the directional properties of the sound field are distorted by semi-ideal hear-through prototypes built by mounting miniature microphones on the outer part of selected commercial insert earphones. This includes an analysis of the magnitude by which spatial information is distorted and the extent to which these distortions are direction dependent. Potential strategies for compensating these distortions are also considered.
Sequential dependencies in magnitude scaling of loudness
Authors: Joshi Suyash N. , Jesteadt Walt
Published: May 2013

Ten normally hearing listeners used a programmable sone-potentiometer knob to adjust the level of a 1000-Hz sinusoid to match the loudness of numbers presented to them in a magnitude production task. Three different power-law exponents (0.15, 0.30, and 0.60) and a log-law with equal steps in dB were used to program the sone-potentiometer. The knob settings systematically influenced the form of the loudness function. Time series analysis was used to assess the sequential dependencies in the data, which increased with increasing exponent and were greatest for the log-law. It would be possible, therefore, to choose knob properties that minimized these dependencies. When the sequential dependencies were removed from the data, the slope of the loudness functions did not change, but the variability decreased. Sequential dependencies were only present when the level of the tone on the previous trial was higher than on the current trial. According to the attention band hypothesis [Green and Luce, Perception Psychophys., 1974] these dependencies arise from a process similar to selective attention, but observations of rapid adaptation of neurons in the inferior colliculus based on stimulus level statistics [Dean et al., Nature Neurosci. (2005)] would also account for the data. [Work supported by NIH.]
Spatial uncertainty and proximity effects in informational masking identically related to the Simpson–Fitter metric of target-masker separation
Authors: Stamas Jacob , Heo Inseok , Gilbertson Lynn , Chang An-Chieh , Lutfi Robert
Published: May 2013

Further evidence is provided suggesting a primary dependence of informational masking (IM) on the stochastic separation of target and masker given by Simpson-Fitter’s da [Lutfi et al. J. Acoust. Soc. Am. 132, EL109-113 (2012)]. The stimuli were brief bursts of Gaussian noise or words played in sequence as masker-target-masker triads. The apparent position of bursts (words), from left to right, was varied independently and at random on each presentation using KEMAR HTRFs. In the 2IFC procedure, the listener’s task was to choose the target positioned further to the right. The effect on performance of spatial uncertainty regarding the masker was examined by manipulating the position variance of the masker. The effect on performance of spatial proximity of target to masker was examined by manipulating the position mean difference between target and masker. In both cases, the data were well described by a single linear function relating d’ performance to da; intercepts differed across listeners, but slopes were similar. Comparable results presented at this meeting for the effects of spectral uncertainty and similarity of target and masker suggest that the statistical properties of signals may be a more significant determinant of IM than their specific acoustic properties.
Extending Schroeder-phase masking: Influence of direction and shape of masker instantaneous frequency
Authors: Hoglund Evelyn M. , Oh Yonghee , Hribar Joseph F. , Wittum Kelsi J. , Strang Megan L. , Feth Lawrence L.
Published: May 2013

Schroeder (1970) devised an algorithm to produce low peak factor signals. Schroeder signals with equal amplitude spectra, but reversed phase spectra, reveal large differences in masker effectiveness for listeners with normal hearing [Smith et al. (1986)]. Results reported here extend previous work to include detection of multiple bursts of the same frequency, and multiple bursts that increase or decrease in frequency. Signal frequencies were selected to correspond with harmonics in the maskers. Results indicate that changing the frequency of the signal amplifies the difference between the Schroeder-phase maskers, but the direction of the change does not. When a frequency modulated tone is substituted for the Schroeder maskers, masked threshold depends on the shape of the instantaneous frequency (IF) function, as well as the direction of change. For linear FM (with IF similar to the Schroeder maskers), masked thresholds are comparable to the Schroeder-phase maskers. However, for logarithmic FM, IF changes at a constant ERB/s and directional differences are much smaller. A modified channel model [Oh (2012)] shows substantial differences in the basilar membrane response to these different maskers. [Research supported by a grant from the Office of Naval Research #N000140911017.]
Investigating the effects of intensity on the bandwidth of peripheral filtering in an amplitude-modulation notch detection task
Authors: Richardson Matthew L. , Shim Allison I. , Berg Bruce G.
Published: May 2013

The effect of intensity on the effective bandwidth of auditory temporal processing is investigated. Thresholds for detecting sinusoidal amplitude-modulation of a 200-Hz wide band of noise centered at 1000 Hz are measured in the presence of a notched noise masker. The masker consists of two, 200-Hz wide, unmodulated bands of noise placed at frequencies above and below the modulated band. Thresholds for a modulation rate of 10 Hz are estimated for different notch bandwidths ranging from 100 Hz to 2740 Hz. The use of a slow modulation frequency aims to avoid possible central limitations of temporal processing at higher modulation frequencies. Intensity is varied across two conditions, with all three bands of noise presented at either 40 dB SPL or 85 dB SPL. Threshold functions for the two intensity levels are essentially identical. The maximum notch width at which an effect of the masker is observed is approximately 500 Hz. The results are consistent with a hypothesis that the filtering characteristics of temporal processing (e.g., envelope model) and spectral processing (e.g., power spectrum model) are different.
Thresholds of tone pitch contour discrimination for English listeners
Authors: Gilbert Rachael C. , Liu Chang
Published: May 2013

This study aims to provide psychophysical data on English language listeners’ ability to discriminate tone pitch contours. Just noticeable differences (JND) of F0 contour changes were measured in six native listeners of American English. Previous work in our lab found English listener thresholds for offsets of falling tones to be significantly lower than those for onsets of rising tones. To what extent this difference is due to the position of the F0 shift versus the direction of the F0 contour is unclear. In this study, we control for four experimental factors: stimulus type (speech, nonspeech), position of F0 shift (onset, offset), direction of shift (upward, downward), and F0 contour direction (falling, rising). Preliminary results reveal that English listeners had significantly lower psychophysical thresholds for F0 shifts at the offset than at the onset. No significant difference was found for F0 shift direction, F0 contour direction, or stimulus type. The current data suggest that the F0 shift position was the primary determinant in our previous study and replicate other findings showing that English listeners perceive tones on a psychophysical base. Future work will examine these results in relation to those of native tone language listeners.
Infants' ability to perceive the pitch of unresolved harmonics
Authors: Lau Bonnie K. , Werner Lynne A.
Published: May 2013

An important phenomenon for models of pitch perception is that adult listeners’ can extract pitch from complexes containing only unresolved harmonics. Although 3-month-olds discriminate resolved harmonics on the basis of missing fundamental (MF) pitch, their ability to discriminate unresolved harmonics is unknown. This study investigated the ability of adults, 7- and 3-month-olds, to perceive the pitch of unresolved harmonics using an observer-based method. Stimuli were MF complexes that were bandpass filtered with a -12 dB/octave slope, combined in random phase, and presented at 70 dB SPL for 650 ms with a 50 ms rise/fall and with a pink noise to mask distortion products. The experiment consisted of two conditions: (1) “low” unresolved harmonics between 2500 and 4500 Hz based on MFs of 160 Hz (H17–H26) and 200 Hz (H13–H22) and (2) “high” unresolved harmonics between 4000 and 6000 Hz based on MFs of 190 Hz (H22–H31) and 200 Hz (H20–H29). To demonstrate MF pitch discrimination, participants were required to ignore spectral changes in complexes with the same fundamental and to respond only when the fundamental changed. Interestingly, variable performance in the “high” condition was observed with adult participants. However, nearly all infants tested categorized complexes by MF pitch in both conditions, suggesting discrimination of unresolved harmonics at 3 months.
Mistuning detection in a complex of unresolved harmonics: Effects of age
Authors: Mamo Sara K. , Grose John H.
Published: May 2013

Older adults experience speech perception difficulties that are not explained by audiometric thresholds. One hypothesis is that temporal processing deficits contribute to these speech-in-noise difficulties. To test this, threshold for mistuning was measured for a component within a complex of unresolved harmonics—a cue that likely depends on sensitivity to envelope perturbations. The complex comprised harmonics 12–16 of a 100-Hz or 200-Hz fundamental, and the duration was either 170-, 340-, or 680-ms. Presentation level was 70 dB SPL. The starting phases of all components were randomized on each presentation. The 3AFC procedure adaptively varied the mistuning of harmonic 14 to obtain the frequency-shift threshold. Younger and older listeners with audiometrically normal hearing participated. The expectation was that older adults would require greater mistuning to detect changes in envelope periodicity and that this would be more evident at shorter durations. Preliminary results support an overall effect of age as well as effects of fundamental frequency and duration. Results will be considered in the context of parallel speech-evoked ABR studies being undertaken in this population that point to age-related deficits in envelope processing, possibly driven by poor encoding of unresolved harmonics. [Work supported by NIDCD 1-F32-DC012217-01A1 & 5-R01-DC001507.]
The influence of feature detection on working memory in complex auditory fields
Authors: Chiodi AnneMarie , Weaver Aurora , Ries Dennis
Published: May 2013

Research reveals that feature detectors may enhance listener performance in auditory discrimination and detection tasks involving frequency modulation (FM) [Cusack and Carlyon, J. Exp. Psychol. 29, 713–725 (2003)]. The influence of these detectors on retention of auditory information is unknown. This study investigated the impact of FM on listener performance in auditory, delayed-comparison tasks for conditions that differed in the number of background stimuli within two perceptual windows separated by various retention intervals. The background stimuli within both windows were either all modulated sinusoids or pure-tones for a given trial. An additional stimulus was presented in each window that could differ in its modulation state (FM or unmodulated sinusoid) across the two windows. The temporal placement and frequencies of all stimuli within the first window were assigned randomly for a given trial and the second window followed these parameters. Listeners were to determine whether the two windows were the same or different. Preliminary results show that same-different judgments of target modulation state was easier in a field of four unmodulated sinusoids than vice versa. This result occurred regardless of retention interval length. The further influence of field complexity, retention interval length, and working memory span will be discussed.
Visco-thermal dissipations in heterogeneous porous media
Authors: Chevillotte Fabien , Jaouen Luc , Bécot François-Xavier
Published: May 2013

Semi-phenomenological models have been widely used since the 1990’s for modeling visco-thermal dissipations of acoustical energy through porous media. These dissipations are taken into account by two complex frequency-dependent functions (the dynamic density ρeq (ω) and the dynamic bulk modulus Keq (ω)), which are analytically derived from macroscopic parameters. Other models were derived for modeling perforated plates [J. Sound Vib. 303 (2007)], double porosity media [J. Acoust. Soc. Am. 114(1) (2003)] or, more recently, porous composites made of porous inclusions in a substrate porous media [Acta Acoust. 96 (2010)]. So far, this latter model is not able to consider the shape of the host and the client media. This model can neither be extended to limiting cases of perforated plate model nor double porosity model. Based on a modiﬁed equivalent ﬂuid model, this work proposes a uniﬁed model which accounts, analytically, for the shape of the inclusions, might they be porous or not. This model enables to describe the acoustic behavior of any kind of composite media from perforated plates to arbitrarily shaped porous composites including configurations of porous inclusions in solid matrix or double porosity media. In addition, possible pressure interactions between the substrate material and the inclusions are accounted for.
Prediction of acoustic properties of parallel assemblies by means of transfer matrix method
Authors: Verdière Kévin , Panneton Raymond , Elkoun Saïd , Dupont Thomas , Leclaire Philippe
Published: May 2013

The Transfer Matrix Method (TMM) is used conventionally to predict the acoustic properties of laterally infinite homogeneous layers assembled in series to form a multilayer. In this work, a parallel assembly process of transfer matrices is used to model heterogeneous materials such as patchworks, acoustic mosaics, or a collection of acoustic elements in parallel. In this method, it is assumed that each parallel element can be modeled by a 2x2 transfer matrix, and no diffusion exists between elements. The method is validated by comparison with finite element (FE) simulations and acoustical tube measurements on different configurations at normal and oblique incidence. Then, an overview of the possibilities, such as the combination of series and parallel matrices, the sound absorption coefficient, and the transmission loss of a parallel array of resonators or three-dimensional geometries is presented and discussed.
Investigations on the sensitivity of the relationships between sound absorption characteristics and microstructure related parameters for polyurethane foams
Authors: Ouisse Morvan , Doutres Olivier , Atalla Noureddine , Ichchou Mohamed
Published: May 2013

Straightforward semi-phenomenological models have been developed for highly porous polyurethane foams to predict the macroscopic non-acoustic parameters involved in the classical Johnson-Champoux-Allard model (i.e., porosity, airflow resistivity…) from microstructure properties (i.e., strut length, strut thickness, and reticulation rate). These microstructure properties are measured using sophisticated optical methods (i.e., optical microscope, SEM) and a large variability can be observed due to great complexity of the 3D microstructure; variability also depends on the precision of the measurement device. This work investigates how the variability associated with the model inputs affects the model outputs (i.e., non-acoustic parameters, surface impedance, and sound absorption coefficient). The sensitivity analysis is based on the Fourier Amplitude Sensitivity Test (FAST). It helps quantify the correlation between the input parameters and identify the parameters contributing the most to output variability, thus requiring precise measurement. This study illustrates the preponderant impact of the reticulation rate (i.e., open pore content) on acoustic performances and guides the user on the required optical measurement device.
Modeling of the acoustic absorption of bi-modal polylactide foams
Authors: Ghaffari Mosanenzadeh Shahrzad , Doutres Olivier , Naguib Hani E. , Park Chul B. , Atalla Noureddine
Published: May 2013

In this study, highly porous bi-modal structures were designed and fabricated from polylactide (PLA) as the main structure and utilizing polyethylene glycol (PEG) to form micro pores by compression molding combined with particulate leaching technique. The pore size of the foam structure was controlled by salt particulates and higher interconnectivity was achieved by the co-continuous blending morphology of PLA matrix with water-soluble PEG. This fabrication method makes it possible to control pore geometry and interconnectivity closely and therefore is an ideal approach to study the relation between microstructure and acoustic properties of the foams. PLA is a bio-based thermoplastic polymer derived from renewable resources. Therefore, the resulting acoustic foams are benign and environmentally friendly. Fabricated foams were characterized based on cellular, acoustic, and mechanical properties. The acoustic performance of the foams was studied by measuring the normal incident absorption coefficient in accordance with the ASTM E1050 standard. An analytical model based on Johnson–Champoux–Allard model was used to numerically simulate the acoustic performance of foams under study. Numerical results predict the absorption behavior of PLA foams with high accuracy. Through this research, open porosities close to 90% were achieved, and the effect of water soluble polymer on cellular properties, acoustic and mechanical performance of polylactide foams was studied.
Reflexion of flexural waves at the end of a tapered beam of quadratic profile covered with a thin viscoelastic layer
Authors: Denis Vivien , Poittevin Julien , Pelat Adrien , Elie Benjamin , Gautier Francois
Published: May 2013

Flexural waves propagating in a beam can be efficiently absorbed if one extremity is tapered with a power law profile and covered by a very thin viscoelastic layer [Krylov, JSV 274, 605–619 (2004)]. Such a terminaison induces an effect known as “the acoustic black hole effect” (ABH), which is resulting from properties of propagation of flexural wave in beams having non homogeneous thicknesses: if the thickness decreases locally, flexural waves slow down and the amplitude of the displacement field increases, leading to efficient energy dissipation if an absorbing layer is placed where the thickness is minimum [Georgiev et al., JSV 330, 2497–2508 (2011)]. Absorption of the ABH terminaison is estimated, thanks to the direct measurement of the reflexion coefficient, using a wave decomposition technique. Experimental modal analysis of a ABH beam can be performed using a “high resolution” technique, which permits to estimate the modal density. Analysis of these experimental results is performed, thanks to a model based on the finite difference method. It is shown that local transverse modes are playing an important role in the absorption properties of ABH.
Multilevel models, covariates, and controlled factors in experimental speech research: Unified analyses of highly structured data
Authors: Silbert Noah H. , Linck Jared A. , VanDam Mark
Published: May 2013

Experimental speech research often makes use of complex experimental designs, but even when multiple experimental factors are manipulated, measured outcomes may be influenced by non-controlled and incompletely controlled factors. Multilevel models (of which mixed-effect models are a special case) enable unified analysis of the relationships between, on the one hand, trial-level data and, on the other, experimental factors and potentially important non-controlled variables. Fitted multilevel models allow us to draw inferences simultaneously about group-level experimental effects and covariates (the typical focus of experimental work) as well as individual subject and item properties (both of which can be important in applied research). The utility of multilevel models will be illustrated with analyses of data from a number of studies. We present models of phonological structure, gender differences, and within-gender subject variability in the acoustics of spoken English consonants; simultaneous modeling of experimental factors, subject and item variability, and second language proficiency in bilingual lexical processing; and modeling of the effects of age, hearing loss, phonological/lexical properties, subject and item variability, and multiple vocabulary-related covariates in early language development.
Synthesis of the speech signals by using autocorrelation function
Authors: Sato Shin-ichi , Bidondo Alejandro
Published: May 2013

The running autocorrelation function (r-ACF) is obtained by the FFT method based on the Wiener–Khinchine theorem after obtaining the power density spectrum for a signal. This study attempted to reconstruct the original speech signal by using a part of its r-ACF. First, the stationary part of the vowel signals were investigated to determine until which delay time of the ACF (maximum time lag) is necessary to recognize the reconstructed signals as the original ones. Then, the continuous speech signals were investigated to determine the appropriate integration interval as well as the maximum time lag.
Relationship between Intelligibility and autocorrelation factors of Japanese monosyllables
Authors: Shimokura Ryota , Akasaka Sakie , Hosoi Hiroshi , Matsui Toshie
Published: May 2013

Amplified speech sound by a hearing aid can improve the speech intelligibility for conductive hearing loss, while improve partly for sensorineural hearing loss. Although the indiscernible consonants in Japanese monosyllables have been identified, it is still unclear which acoustical feature in the consonants causes the difficulty on hearing. Therefore, the aim of this study is to clarify an influential factor on the speech intelligibility using autocorrelation function (ACF), which can capture temporal features of signals. The ACF factors were compared with percent articulations obtained by a Japanese speech intelligibility test (n = 144). The time-series factors were calculated by running ACF along a monosyllable, and the median values of the ACF factors were represented. As results, an effective duration (τe [ms]) of the ACF was correlated with the averaged percent articulations among the consonants (r = 0.87, p < 0.01). The τe indicates temporal fluctuation of speech signals including its fundamental frequency. The deteriorated perceptual function for temporal fluctuation may reduce the recognition ability of the consonants, so the application of the ACF analysis for a hearing aid may help the hearing of patients with sensorineural hearing loss.
Speaker recognition analysis using running autocorrelation function parameters
Authors: Bidondo Alejandro , Sato Shin-ichi , Kinigsberg Ezequiel , Arouxet Mariano , Andrés Sabater , Arias Agustín , Saavedra Adrián , Groisman Ariel
Published: May 2013

The human brain's process associated with the recognition and identification of acoustic signals is regarded as the calculation of the distances between “sound vectors;“ the ones listened in the present with the memorized vectors in previous listening. Matching vectors (minimal distance between them) would indicate sounds come from similar sound sources or same sound source. In this study, the statistical values of r-ACF (running autocorrelation function) microscopic parameters of 10 spoken words recordings (in Spanish) by the same and different speakers were calculated, and the aforementioned vector's distances were constructed by using the distances between the r-ACF parameters, being able to predict the degree of similarity between the speakers.
Mismatched distances from speakers to telephone in a forensic-voice-comparison case
Authors: Enzinger Ewald
Published: May 2013

In a forensic-voice-comparison (FVC) case, one speaker (A) was talking on a mobile telephone, and another (B) was standing a short distance away. Later, B moved closer to the telephone. Shortly thereafter, there was a section of speech where the identity of the speaker was disputed. All material for training an FVC-system could be extracted from this single recording, but there was a near-far mismatch: Training data for A were near, training data for B were far, and the disputed speech was near. We describe a procedure for addressing the degree of validity and reliability of an FVC system under such conditions, prior to it being applied to the casework recording: Sections of recordings of pairs of speakers of known identity are used to train an A and a B model; multiple other sections from each of the A and B recordings are used as test data; a likelihood ratio is calculated for each test section; and system validity and reliability are assessed. Prior to training and testing, the A and B recordings were played through loudspeakers and rerecorded via a mobile-telephone network, B was rerecorded twice, once with the loudspeaker near and once with it far from the telephone.
Exploring duration and spectral parameters of English /m/ for forensic speaker comparison
Authors: Kavanagh Colleen
Published: May 2013

The speaker-specificity of five acoustic features of British English /m/ was explored from a forensic speaker comparison perspective. Normalized duration, center of gravity (COG), standard deviation (SD), and frequencies at peak and minimum amplitudes were measured for 30 adult male Standard Southern British English and Leeds English speakers. Spectral measurements were made in each of five frequency bands (0–0.5 kHz, 0.5–1 kHz, 1–2 kHz, 2–3 kHz, and 3–4 kHz) and calculated from a 40-ms window at the midpoint of each token. ANOVAs showed Speaker to be a highly significant factor for all variables. Discriminant analysis (DA) and likelihood ratio (LR) estimation assessed speaker discrimination with individual predictors and combinations thereof. Sample sizes limited the number of predictors in DA to eight; F-ratios were used to select the best predictors for analysis. The COG+SD (bands 1, 3, 4, 5) and Best 8 F-ratios (COG bands 1, 4, 5 + SD 1, 3, 4 + Peak 1, 4) tests achieved 53% and 49% correct classification, respectively. The Best 8 F-ratios and COG+SD tests also produced the best LR results, while COG+Peak performed similarly. DA and LR results for all predictor combinations will be presented and the most promising speaker comparison parameters highlighted.
Examining long-term formant distributions as a discriminant in forensic speaker comparisons under a likelihood ratio framework
Authors: Gold Erica , French Peter , Harrison Philip
Published: May 2013

This study investigates the use of long-term formant distributions (LTFD) as a discriminant in forensic speaker comparisons. LTFD are the distributions calculated for all values of each formant for a speaker in a single recording. Spontaneous speech recordings from 100 male speakers of Southern Standard British English, aged 18–25 were analyzed from the DyViS Database (Nolan 2009). The recordings were auto-segmented to obtain a minimum of 50 s of vowels per speaker. The iCAbS (iterative cepstral analysis by synthesis) formant tracker was used to automatically extract and measure F1-F4 every 5 ms. To assess the evidential value of the LTFDs, likelihood ratios (LRs) were computed using a MatLab implementation of Aitken and Lucy’s (2004) Multivariate Kernel-Density formula (Morrison 2007). It was found that LTFD performs well overall, but much better with different speaker comparisons than same speaker comparisons (97.76 % compared to 78% of comparisons providing correct support; Cllr = 0.9072 and EER = 5.47%). LTFD appears to be a good discriminant to include in forensic speaker comparison analyses and offers the added attraction of avoiding potential correlation problems between vowel phonemes.
Fusion of multiple formant-trajectory- and fundamental-frequency-based forensic-voice-comparison systems: Chinese /ei1/, /ai2/, and /iau1/
Authors: Zhang Cuiling , Enzinger Ewald
Published: May 2013

This study investigates the fusion of multiple formant-trajectory- and fundamental-frequency-trajectory-based (f0-trajectory-based) forensic-voice-comparison systems. Each system was based on tokens of a single phoneme: tokens of Chinese /ei1/, /ai2/, and /iau1/ (numbers indicate tones). Human-supervised formant-trajectory and f0-trajectory measurements were made on tokens from a database of recordings of 60 female speakers of Chinese. Discrete cosine transforms (DCT) were fitted to the trajectories and the DCT coefficients used to calculate likelihood ratios via the multivariate kernel density (MVKD) formula. The individual-phoneme systems were fused with each other and with a baseline mel-frequency cepstral-coefficient (MFCC) Gaussian-mixture-model universal-background-model (GMM-UBM). The latter made use of the entire speech-active portion of the recordings. Tests were conducted using high-quality recordings as nominal suspect samples and mobile-to-landline transmitted recordings as nominal offender samples. Fusion of the phoneme-systems with the baseline system via logistic regression did not lead to any substantial improvement in validity and reliability deteriorated.
Articulatory-based speaker recognition
Authors: Rodrigues Luis , Kroeker John
Published: May 2013

This paper presents a new methodology for computational speaker recognition based on a mathematical model of articulatory speech production. The method, based on articulatory phonology is tested on the MOCHA database for recognizing a male speaker and a female speaker. From an engineering perspective, in articulatory phonology one is interested in the trajectories over time of a set of articulators. These time trajectories are associated with the production of speech. The basic phonological unit in articulatory phonology is the articulatory gesture, which is defined as a dynamic system specified by a characteristic set of parameters. This dynamic system receives as inputs a target state and a set of parameters that tune the system to the desired action. The output is the solution of the state equation, i.e., the state trajectory, where the sate is formed by the x-y positions of the important articulators that describe human speech. The state trajectory is then mapped to the output speech waveform by emulating the human vocal tract, through the observation equation and the MFCCs frequency description. A simplification of this model will be used in this paper for speaker recognition with 100% success in recognizing a male and a female speaker.
A dynamic automatic noisy speech recognition system for a single-channel hybrid noisy industrial environment
Authors: Paul Sheuli
Published: May 2013

A dynamic noisy speech recognition system is developed to recognize single-channel small spoken commands in a hybrid noisy industrial environment. This hybrid system has three parts: (a) hybrid pre-processing to enhance noisy speech, (b) feature extraction for perceptual speech features, (c) classification and recognition for the DANSR's result. Here, the single-channel is only one microphone, and the hybrid noise is environmental mixed noise distinguished as: (i) strong, (ii) time varying steady-unsteady, and (iii) mild. A new adaptive feature extraction technique based on local trigonometric transformation (LTT) is introduced and examined. This is adapted with psychoacoustic quantities such as Bark scaled critical band spectrum, loudness scale, and perceptual entropy. Here the spectral analysis is done by rising cut-off function, folding operation, and discrete cosine transformation (DCT-IV) instead of Fourier transform. Then, inverse DCT-IV and unfolding operation result in perceptual LTT (PLTT) features. These are recognized by hidden Markov model (HMM). The new PLTT features are more efficient and perceptually meaningful than the standard feature extraction techniques. The DANSR system is a novel solution for small commands to a long existing hybrid noise problem.
Causal binary mask estimation for speech enhancement using sparsity constraints
Authors: Kressner Abigail A. , Anderson David V. , Rozell Christopher J.
Published: May 2013

While most single-channel noise reduction algorithms fail to improve speech intelligibility, the ideal binary mask (IBM) has demonstrated substantial intelligibility improvements for both normal- and impaired-hearing listeners. However, this approach exploits oracle knowledge of the target and interferer signals to preserve only the time-frequency regions that are target-dominated. Single-channel noise suppression algorithms trying to approximate the IBM using locally estimated signal-to-noise ratios without oracle knowledge have had limited success. Thought of in another way, the IBM exploits the disjoint placement of the target and interferer in time and frequency to create a time-frequency signal representation that is more sparse (i.e., has fewer non-zeros). In recent work (in preparation for ICASSP 2013), we have introduced a novel time-frequency masking algorithm based on a sparse approximation algorithm from the signal processing literature. However, the algorithm employs a non-causal estimator. The present work introduces an improved de-noising algorithm that uses more realistic frame-based (causal) computations to estimate a binary mask.
Objective and subjective evaluation of complementary Wiener filter for speech dereverberation
Authors: Ohtani Kento , Komatsu Tatsuya , Kondo Kazunobu , Nishino Takanori , Takeda Kazuya
Published: May 2013

Acoustic distortion caused by reverberation can degrade speech quality and performance of speech-based systems. Several dereverberation techniques have been proposed in the literature. For example, a dereverberation method using a complementary Wiener filter can suppress late reverberation with few computational resources. As a method for dereverberation, the method using a complementary Wiener filter has been proposed, and for the exponential decay impulse response model, it is shown theoretically that we can suppress reverberation with few computational resources. In this report, we approximate expectation of the power spectrum. which is necessary to calculate a complementary Wiener filter as exponential moving average. We conducted dereverberation experiments using actual environment room impulse response. The results of the objective evaluation show that the suppression performances of the actual environment room impulse response can approximate from the results of the exponential decay impulse response model. Additionally, we investigated the relationship between the results of objective evaluation and the results of subjective evaluation. In a small reverberation environment, we can see strong correlation between the results of objective and subjective evaluation.
Investigations into the human pinna shapes on head-related transfer functions in the median plane
Authors: Komatsu Hajime , Nakano Kota , Nakayama Masato , Nishiura Takanobu
Published: May 2013

The binaural reproduction system requires many accurate measurements of head-related transfer functions (HRTFs) to achieve the high-precision sound localization. However, the actual measurement of HRTFs has a heavy burden for subjects. To solve this problem, personalize HRTFs have been proposed. In the personalize HRTFs, the interaural level difference (ILD) and the interaural time difference (ITD) are utilized on the sound localization in the horizontal plane, and the spectral envelope of HRTFs is utilized on the sound localization in the median plane. In the present paper, we focus on the human pinna shapes as listener’s anthropometric parameters on the sound localization in the median plane. In order to reveal the effect of human pinna shapes on HRTFs in the median plane, we investigate the relationship between human pinna shapes and the spectrum envelope of HRTFs. More specifically, we crafted the dummy pinna for the dummy head. Also, we investigated the spectrum envelope of HRTFs in various shape conditions of the dummy pinna in the median plane. As a result of investigations, we confirmed correspondence relationship between human pinna shapes and the spectrum envelope of HRTFs.
Acoustic echo cancelation in discrete Fourier transform domain based on adaptive combination of adaptive filters
Authors: Azpicueta-Ruiz Luis A. , Figueiras-Vidal Anibal , Arenas-Garcia Jeronimo
Published: May 2013

Acoustic echo cancelers (AECs) are vital to many of communication systems, including hands-free telephone and videoconference, among others. Recently, adaptive combination of adaptive filters has been presented as an easy but effective method to improve its performance, alleviating different compromises inherent to adaptive filters (responsible to identify the room impulse response). The most important tradeoff is related with the selection of the step size, which involves speed of convergence and residual echo. However, AECs require long adaptive filters, forcing to employ frequency-domain techniques to reduce the computational load and to accelerate the algorithms convergence when colored inputs—such speech—are presents. In this paper, we present an AEC based on combination of filters in discrete Fourier transform domain. Considering that both the input signal and the cancelation scenario make the behavior of adaptive filters is frequency dependent, our proposal exploits the combination capabilities employing different mixing parameters to separately combine independent spectral regions of two frequency-domain adaptive filters with different step sizes. In this way, our scheme outperforms recent algorithms where only a single combining parameter mixes the overall outputs of two frequency-domain adaptive filters. These advantages are illustrated by means of realistic experiments.
A study of the reflection coefficients and backscattering effects of one-dimensional rough poroelastic surfaces using the finite element method
Authors: Bonomo Anthony L. , Isakson Marcia J. , Chotiros Nicholas
Published: May 2013

Acoustic reflection and scattering effects of one-dimensional rough poroelastic surfaces are studied using the finite element method. The poroelastic sediment layer is modeled following the classical work of Biot as extended by Stoll, which assumes that two attenuating compressional waves and one attenuating shear wave propagate in the sediment. The rough surfaces are generated using power-law type spectra and the incident wave used is a Gaussian tapered plane wave. This work seeks to assess how the reflection coefficients and backscattering effects of a poroelastic bottom vary as a function of frequency, roughness, and sediment type. Special consideration is given to the mesh required to accurately resolve the effects of the slow compressional and shear waves, which often have wave speeds slower than the fast compressional wave by an order of magnitude or more. [Work sponsored by the Office of Naval Research, Ocean Acoustics.]
Non-invasive characterization of fluid mud from scalar and vector noise fields due to a small boat
Authors: Hermand Jean-Pierre , Ren Qunyan
Published: May 2013

The passive geoacoustic characterization of sediment using the ratio between pressure and vector field that are measured by an easily deployable system is discussed. The ratio is very sensitive to environmental properties but independent of unknown source spectrogram, e.g., boat noise that exhibits complex spectral shape. Noise data sets due to different runs with a small boat were recorded on two closely adjacent hydrophones offshore the Amazon Rio mouth and processed by a nonlinear inversion scheme. Global optimization based on genetic algorithms provides the geoacoustic parameters of the fluid mud and underlying mud sediment, and the marginal posterior probabilities. Good consistency among the respective inversion results demonstrates the feasibility of the method under far from ideal conditions for environmental characterization, in term of unknown range, source ship navigation data, unknown source spectra, uncertain receiver depth, tilt, etc.
Determination of grain size distribution in water-saturated granular medium using p-wave attenuation dispersion
Authors: Yang Haesang , Lee Keunhwa , Seong Woojae
Published: May 2013

P-wave attenuation in the water-saturated granular medium depends on both the frequency and the grain size. In this study, the use of the attenuation dispersion for the determination of grain size distribution in the water-saturated granular medium is discussed. For the dense granular medium, mathematical model considering multiple scattering is used for regression algorithm by fitting model predictions to the measured attenuation data. Inversion of grain size distribution is carried out numerically, and the results are discussed and compared to measured data for the water-saturated glass beads with unimodal and bimodal distributions.
Electronically variable room acoustics—Motivations and challenges
Authors: Hoover K. Anthony , Ellison Steve
Published: May 2013

Electronically variable room acoustics, or “active acoustics,” has become an effective solution to a variety of room acoustics challenges. The motivations for considering such systems often go beyond acoustics. Energy conservation, financial considerations, historic preservation, and balancing the needs of a venue's constituencies all can play a role in the determination to employ active acoustics. This paper will discuss some practical examples, including planned renovations to the Santa Monica Civic Auditorium, home of the Academy Awards during the 1960s, a resident orchestra, legendary rock concerts, and a unique hydraulic floor to convert the Civic from a performance space to an exhibit space. The active acoustic system objectives, design strategies, and challenges will be discussed.
Adapting, enhancing, and fictionalizing room acoustics
Authors: Ohl Ted , Adelman-Larsen Niels
Published: May 2013

The need for adjustable acoustics extends far beyond the Performing Arts Center or world class concert hall. The small scale of the existing market for adjustable acoustics impedes the development of products economical enough to expand the market. Expand the market by: Raising awareness of the options available for adjusting the acoustics in their room will expand the market. User and designer feedback regarding desirable features also will contribute. Broadening the concept of variable acoustics to include noise control creates opportunities in the industrial sector. Focusing on cost saving efficiencies in design and production of products making the cost of adjustable acoustic solutions accessible to more clients; developing products suitable for retrofit as well as new construction Product Development: identify critical and desirable features in collaboration with users and designers. Develop product performance data to enable acousticians to incorporate variable acoustics into their designs and predict performance for clients. Incorporate other design team members’ input to anticipate conflicts such as space use, look, cost, and other systems. Available Products Available product types will be quickly reviewed. In-situ test data from existing projects will be presented. More innovation to reach a broader market will be encouraged
National Gypsum Company's acoustically enhanced gypsum board—SoundBreak XP
Authors: Cusa Stephen A.
Published: May 2013

Gold Bond® BRAND SoundBreak XP Gypsum Board is an acoustically enhanced gypsum board used in the construction of high STC wall assemblies. This innovative gypsum board allows for construction of high STC wall assemblies that are thinner, cost effective, and more reliable than traditional methods. In addition to achieving a higher performance wall relative to acoustics, SoundBreak XP helps to achieve a higher performance wall in general. SoundBreak XP comes standard with mold and mildew resistance and is manufactured with an abrasion resistant paper. Both the mold resistance and the abrasion resistance achieve the best possible scores relative to the standard ASTM test methods. SoundBreak XP installs and finishes like standard gypsum board.
Simulation of normal incidence sound absorption coefficients of perforated panels with/without glass wool by transmission line parameters in a two- port network
Authors: Nakai Takayoshi , Yoshida Kota
Published: May 2013

This paper describes simulation of normal incidence sound absorption coefficients of perforated panels by ABCD-parameters in a two-port network. Maa and Sakagami have investigated micro perforated panels, MPP. But their theories can treat only near 1% perforation rates of perforated panels with back cavities. If sound propagates as a plane wave, sound propagation can be represented as ABCD-parameters in a two-port network. Perforated panels, back cavities, and glass wool absorption materials are represented as matrix of ABCD-parameters, respectively. ABCD-parameters of a perforated panel with a back cavity are calculated as multiplication of their matrices. An input impedance can be calculated from the calculated ABCD-parameters. A normal incident absorption coefficient is calculated from the input impedance. Holes of the perforated panels have losses of viscous friction and thermal conduction at their walls. Simulations are done in the condition of 0.25 to 5 mm diameters of holes, 0.25% to 25% perforation rates, 0.5 to 5 mm thickness of the perforated panels with back cavities in which there are or are not glass wool absorption materials. The results of these simulations are good agreements with the results of our measurements by transfer function method.
A new approach to ultrasonic cleaning
Authors: Leighton Tim , Birkin Peter R. , Offin Doug
Published: May 2013

Traditional ultrasonic cleaning baths are limited in that they cannot clean objects that are too large to fit in the bath, and cannot be taken to objects with complex geometries in order to “clean in place.” Furthermore, the object to be cleaned sits in a “soup” of contaminated liquid, and while cavitation fields can be set up under test conditions, immersion of the object to be cleaned can significantly degrade the bath’s performance by disrupting the sound field. An alternative technique, which does not use ultrasound is the commercial pressure- or -power washer, where high speed jets of water and cleaning agent are pumped onto a surface. Although these can “clean in place,” they pump large volumes of water, and produce significant volumes of contaminated run-off and contaminated aerosol, both of which are hazards for secondary contamination of users and water supplies. The momentum of the water and pump requirements mean they are difficult to scale up. This paper presents a low volume flow technique for ultrasonic cleaning in place, benefits being that it operates with low flow rates (1–2 L/min), and there is no need to expend energy on heating the water.