bluuebunny/crossref_metadata_2025 huggingface dataset:
Otoacoustic emissions in sheep (Ovis aries): Sex differences and prenatal androgen effects
Authors: McFadden Dennis , Pasanen Edward , Valero Michelle D. , Roberts Eila K. , Lee Theresa M.
Published: May 2008

Otoacoustic emissions (OAEs) were measured in Suffolk sheep (Ovis aries), the first ruminant species tested for OAEs to our knowledge. Some sheep had been administered testosterone or estradiol during prenatal development, some had been gonadectomized after birth, and some had been allowed to develop normally. The click-evoked otoacoustic emissions (CEOAEs) exhibited by the female control group (N=11) were stronger than those in the male control group (N=15), which is the same direction of effect reported for other mammals, including humans. The females administered testosterone prenatally (N=13) had substantially weaker (masculinized) CEOAEs than control females. Both outcomes suggest that prenatal exposure to androgens weakens the cochlear amplifiers. The CEOAEs of males administered testosterone prenatally (N=5) were not different from those of control males, which is contrary to expectation, but in accord with data from similarly treated rhesus monkeys. Seven males (of 24) and seven females (of 34) had no measurable DPOAEs at any frequency or level of primary tones tested, even though all did have normal-appearing CEOAEs. No spontaneous otoacoustic emissions (SOAEs) were found in any ears, a common finding in non-human species. [Supported by NIDCD (DM) and NICHD (TML).]
Multiple looks in spectrotemporal integration
Authors: Hoglund Evelyn , Feth Lawrence
Published: May 2008

The multiple looks hypothesis was originally developed as an explanation for temporal integration and resolution. Our preliminary work supports its extension to spectral integration as well spectrotemporal integration. The current study was designed to measure detection of tonal signals that vary in both spectral and temporal dimensions. Preliminary work began by measuring thresholds in quiet. Elementary signals consisted of 10 ms tone bursts centered on one of eight frequencies spaced 1 ERB apart. Complexes were constructed by adding two to eight individual tone bursts along the temporal or spectral dimension, or both. The signals varied along both dimensions in several ways to focus on the relative salience of each dimension for detection. In quiet, the spectrotemporal integration conditions showed threshold improvement to be limited by spectral integration. Surprisingly, even trial-by-trial randomization of frequencies did not negatively influence detection. For this study, the same tone complexes from the preliminary work were used to measure thresholds in a wide band energetic masker. As expected, threshold improvement was most dependent on increasing the number of tones, with less effect from other variables. Subtle differences between quiet and masked integration performance will be discussed. Research supported by a grant from NIH/NIDCD R01-DC006879.
Estimating the basilar-membrane input/output-function in normal-hearing and hearing-impaired listeners using forward masking
Authors: Jepsen Morten L. , Dau Torsten
Published: May 2008

To partly characterize the function of cochlear processing in humans, the basilar membrane (BM) input-output (I/O) function can be estimated. In recent studies, forward masking has been used to estimate BM compression. If an on-frequency masker is processed compressively, while an off-frequency masker is transformed more linearly, the ratio between the slopes of growth of masking (GOM) functions provides an estimate of BM compression at the signal frequency. In this study, this paradigm is extended to also estimate the knee-point of the I/O-function between linear processing at low levels and compressive processing at medium levels. If a signal can be masked by a low-level on-frequency masker such that signal and masker fall in the linear region of the I/O-function, then a steeper GOM function is expected. The knee-point can then be estimated in the input level region where the GOM changes significantly. Data were collected from eight normal-hearing (NH) and five hearing-impaired (HI) listeners with mild to moderate sensorineural hearing loss. Both groups showed large inter-subject but low intra-subject variability. When the knee-point could be estimated for the HI listeners it was shifted towards higher input levels and compression was similar to that of NH listeners.
Unpredictable interruption can enhance the auditory continuity illusion
Authors: Kobayashi Maori , Kashino Makio
Published: May 2008

It was examined whether the predictability of interruption timing affects the limit of perceiving the continuity illusion. The stimulus consisted of a 500-Hz sinusoidal inducee and a 200-ms, 500-Hz, 1/3-octave noise band inducer, which were alternated repeatedly, with a fixed interval (400 ms) between adjacent inducers (regular condition) or with random intervals (irregular condition). The continuity limit (CL) was measured in terms of the maximum level of the inducee for the illusory continuity to be perceived while maintaining the inducer level at 60 dB SPL. In Experiment 1, the listeners' task was to judge whether the inducee appeared continuous or discontinuous. In Experiment 2, the task was to discriminate illusory and physical continuity in a two-interval forced choice paradigm. In both experiments, the mean CLs in the irregular condition were significantly higher than those in the regular condition. In Experiment 3, when an identical irregular interval pattern was used repeatedly in successive trials, the mean CL gradually declined, and reached the level of the regular condition in approximately 80 trials. These results suggest that unpredictability, not mere irregularity, in interruption timing promotes the continuity illusion, implying the involvement of the short-term plasticity of the auditory system.
Intelligibility of whispered speech interleaved with speech or noise maskers
Authors: Iyer Nandini , Brungart Douglas S. , Simpson Brian D.
Published: May 2008

When an interrupted speech signal is alternated with a noise masker, performance is generally improved relative to the condition where the speech and noise are presented continuously. However, when two interrupted speech signals are alternated, performance declines relative to the continuous presentation condition. One possible explanation for this result is that listeners in the alternated speech and noise condition use the contrast between the periodic temporal structure of voiced speech and the random temporal structure of noise to segregate the time intervals associated with the speech target and noise masker. In the current experiment, a whispered speech signal was either presented continuously in the presence of a noise masker or alternated with a noise masker at an 8-Hz rate. The results show near perfect performance in the alternated whisper and noise condition, suggesting that harmonic structure due to voicing is not necessary to segregate a speech signal from an interleaved random-noise masker. Indeed, when whispered speech was interleaved with voiced speech, performance decreased relative to the continuous condition when the target talker was voiced but not when it was whispered, suggesting that listeners are better at selectively attending to unvoiced intervals and ignoring voiced intervals than the converse.
Effect of sound spatialisation on multitasking in remote meetings
Authors: Wrigley Stuart N. , Tucker Simon , Brown Guy J. , Whittaker Steve
Published: May 2008

Participants in a teleconference often multitask during meetings, e.g., they work on a text-based “foreground” task whilst listening in the 'background' for a topic of interest to appear. Audio material should therefore be presented in a manner that has the smallest possible impact on the foreground task without affecting topic detection. Here, we ask whether spatialised audio presentation of a meeting is less disruptive than the single-channel mixture of talkers that is normally used in teleconference audio. Head-related impulse responses are used to synthesise a stereo signal in which each meeting participant is placed at a spatially distinct location. A number of talker location configurations are used, and we examine how these impact upon a text-based foreground task: finding all letter “e” occurrences in a block of text as quickly as possible. We also examine the effect of listening task (e.g., “listen for keyword x” versus “listen for keyword x spoken by talker on left”) and record listener preferences for audio presentation style. Our results suggest that single-channel and spatialised audio are equally disruptive when listeners are unaware of the audio presentation style. We also report studies in which presentation type and target direction/gender are cued.
The use of virtual acoustics in the evaluation and development of binaural hearing aid algorithms
Authors: Rychtarikova Monika , Van Den Bogaert Tim , Vermeir Gerrit , Eneman Koen , Lauriks Walter , Moonen Marc , Wouters Jan
Published: May 2008

The development of noise reduction algorithms for hearing aids (HA) is not longer only related to the improvement of signal to noise ratio, but also to the quality of hearing, e.g., binaural aspects of hearing. This is very important for the recognition of the localization of sound sources but also for an improved speech intelligibility in noisy situations due to spatial release from masking effects. New design and signal processing algorithms for binaural HA's need to be tested and validated in different acoustical scenarios. As it is too laborious and time consuming to perform sufficient numbers of perceptual evaluations in different rooms with different acoustical parameters, advanced acoustic modeling of different virtual acoustical environments might be needed. Virtual acoustics in our research relates to the convolution of the measured or simulated binaural signals (head related transfer functions - HRTF's) with the impulse response generated from a computer model of a room (using ODEON® software) to simulate binaural sounds. This study investigates the usage of virtual acoustics in the framework of developing algorithms for binaural hearing aids. It evaluates and quantifies the fidelity of binaural signals generated by commercially available virtual acoustics software with respect to the localization of sound and speech intelligibility in different acoustical scenarios.
Auralization of urban soundscaping designs using the Arup SoundLab
Authors: Jouan Seb
Published: May 2008

Noise is unwanted sound (from an urban planning point of view). The standard practice to control noise, while valuable, is a negative process (i.e., reducing unwanted “noise”). In response, there is a drive by several authorities to take a more positive approach to improve and manage soundscapes in cities and civic spaces. Urban soundscape design consists of planning, shaping and managing the sound to fit each area of the masterplan in terms of civic, cultural, and social character. Arup uses its SoundLab, a 3D sound facility (ambisonic), to design and auralise (the sound equivalent to visualisation) soundscapes with planners, architects or artists and demonstrate to Local Planning Authorities or clients the results of the design. An example of urban soundscaping design is Dark Neville Street, Leeds for Leeds City Council where architectural, lighting design, acoustical soundscaping design were integrated together with an artistic sonic piece by Hans-Peter Kuhn which was auralised in the Arup SoundLab. An example of “indoor” sonic art and soundscape design installation is “Harmonic Bridge” by Bill Fontana and Arup Acoustics in the Turbine Hall of the Tate Modern (June-August 2006) where the sonic art piece was composed in the SoundLab.
Salience of spatial attributes on quality evaluation of multichannel audio processing devices
Authors: Choi Inyong , Seo Jung-Hun , Sung Koeng-Mo
Published: May 2008

Objective quality assessment methods for audio signal processing devices, e.g. ITU-R BS.1387-1, measure perceived quality by using various types of test excerpts and incorporating a number of psychoacoustic metrics for evaluating various attributes of quality degradations. As many multichannel audio processing devices are being developed, recent quality assessment methods include not only timbral features but also spatial features, such as analysis on interaural disparities. In this paper, the importance of the newly proposed spatial - interaural - features are studied for different types of multichannel test signals. The spatial features have bigger salience on directly recorded sound sources than artificially manipulated multichannel contents.
Effects of hardware on optimal filter segmentations for the segmented convolution
Authors: Wefers Frank
Published: May 2008

The segmented convolution is a commonly used method for the auralisation in interactive virtual acoustic environments. Typically the auralisation of a virtual scene is done by convolution of the scenes' sound sources signals with impulse responses, that describe the sound transmission paths. The segmented convolution algorithm allows an efficient computation of the convolution in real-time, by segmentation of the impulse response into several parts. In order to allow the realistic simulation of complex acoustical scenes, the convolutions' computational effort must be minimised. It is a fact that the segmentation of the impulse response is a key parameter to the algorithm and has great effects on its computational load, stability and even realisability in real-time. Considerations on optimal filter segmentations mainly founded on a theoretical point, like the runtime complexity of operations or the number of required floating point operations. However, practical systems behave differently: Calculations on real hardware suffer cache losses and memory bandwidth. In this contribution, the influence of real hardware on optimal filter segmentations is examined. An optimisation algorithm is presented, that allows to gain optimal filter segmentations for a dedicated target system. The consequences of efficiency losses to the structure of optimal filter segmentations are discussed.
English speech intelligibility test among persons of Arabic native language
Authors: Abd-Elbasseer Mohammed , Barakat Sanaa
Published: May 2008

English language as a non-native language is the most common language over the world, it is the language of conferences, business and the airports. While the Arabic language is the native language of Egypt, the English language is considered as the second language. This research will conduct to investigate the effect of speech level, rate and repetition on the intelligibility of English language among some highly educated Egyptian persons. 34 males and females subjects are participated in this test. 40 sentences of BKB sentences were used with three scored words per sentence. First, they are subjected to the speech rate of about 220, 190, 160, and 100 words per minute at the speech level of about 60 dBA, which is considered as the normal level. Second,they are subjected to the speech level of 60, 70, and 80 dBA at speech rate of 190 words per minute, which is considered as the normal rate. It was found that the intelligibility score increased with decreasing the rate and increasing the level respectively. It was found that scores of the worst scored sentence for the first 15 subjects did not change, whatever we repeat the tested sentence at normal rate and level.
Comparison of speech vs speech-analogue temporal encoding in human: An auditory brainstem response and categorical perception assessment
Authors: Akhoun Idrick , Richard Céline , Moulin Annie , Veuillet Evelyne , Gallégo Stéphane , Berger-Vachon Christian , Collet Lionel , Thai-Van Hung
Published: May 2008

Temporal speech cues, such as periodicity and voice onset time (VOT), are particularly salient information for speech intelligibility. In addition, speech elicited auditory brainstem response (Speech ABR) to consonant vowels (CV) has been shown to mimic the temporal periodicity of the acoustic signal very accurately. Consequently, Speech ABR has been described as a potential objective tool to explore the temporal integration of speech excerpts in human brainstem. We aimed at putting in evidence the specialization of temporal encoding of speech vs nonspeech signals through clinical and behavioral measurements. A sum of five sinusoids, at the five formant frequencies, modulated by the corresponding CV temporal envelope, was taken as speech-analogue. Accordingly, Speech and Analogue ABRs components were compared, although a priori due to be similar. Furthermore, categorical perception assessment was processed to both analogue and full-speech continuums. If confirmed, these preliminary results (seven normal hearing adults for physiological ABRs and behavioral categorical perception) would bring further evidence of an early specialization of temporal processing of speech, in line with previous research [Kraus and Nicol, Trends in Neuroscience 28(4), 176--81 (2005)]. [Work supported by CNRS, Cochlear & Univ.Lyon 1.]
Factors affecting recognition of vocoded speech: Effect of envelope cutoff frequency and carrier type
Authors: Souza Pamela , Rosen Stuart
Published: May 2008

Multiband vocoded signals are used to study temporal cues as well as to simulate information available to cochlear implant wearers, but different implementations of these signals may affect results. This series of experiments varied envelope cutoff frequency (30 or 300 Hz), carrier type (tone or noise) and number of frequency bands (2-5) for vocoded speech presented to normal-hearing subjects. Dependent measures were consonant recognition (vowel-consonant-vowel syllables), vowel recognition (/b/-vowel-/d/ words), and sentence recognition. Results indicated better performance with a high envelope cutoff for tone-vocoded signals, and no effect of envelope cutoff for noise-vocoded signals. When only low-frequency envelope cues were available, performance was better for noise-vocoded than for tone-vocoded signals. When higher-frequency envelope cues were available, performance was better for tone-vocoded than for noise-vocoded signals. To understand the source of the observed differences, a follow-up study tested the relative contributions of comodulation across bands (as in the 30- and 300-Hz tone-vocoded conditions) and spectral sidebands (as in the 300-Hz tone-vocoded conditions). Those data indicated that comodulation across bands did not contribute to performance. The presence of sidebands which provided more spectral detail improved performance over a sparser spectrum. [Supported by the Bloedel Hearing Research Center and NIH RO1 DC006014.]
When and how disyllables are contracted into monosyllables in Taiwan Mandarin
Authors: Cheng Chierh , Xu Yi
Published: May 2008

In Taiwan Mandarin, a disyllabic word or phrase is often contracted into a monosyllabic unit in conversational speech, just as “do not” is often contracted into “don't” in English. The goal of this research is to a) test the hypothesis that contraction is directly related to time pressure, and so it can be elicited in the laboratory from nonsense disyllabic units, and b) search for the underlying mechanisms of contraction. We devised four groups of nonsense disyllabic combinations according to the level of obstruction in the intervocalic consonant: (1) zero obstruction - CV+V; CV+VN; CV+VV, (2) nasal consonant - CVN+V; CV+NV, (3) non-nasal consonant - CV+CV, where C is a stop, fricative or affricate, and 4) nasal + non-nasal consonant-CVN+CV. These disyllabic units were embedded in carriers that would impose different degrees of time pressure on them. Four speakers of Taiwan Mandarin read aloud these sentences at three articulatory modes: clear, natural and fast. Preliminary examinations showed that contraction occurred more frequently as speech rate increased and as intervocalic obstruction decreased. Detailed acoustic analyses are being conducted to search for evidence of the underlying mechanism of such contraction, and the results will be reported.
Across-ear grouping of speech bands in quiet and in the presence of interference
Authors: Ozmeral Erol , Best Virginia , McCoin Jaime , Shinn-Cunningham Barbara
Published: May 2008

Speech was filtered into 16 frequency bands. Three bands were randomly selected on each trial as the LOW (two low-frequency bands) and HIGH (one high-frequency band) target components. LOW and HIGH were either presented to the same (SAME) or opposite (SPLIT) ears. Four randomly selected, nonoverlapping frequency bands from a different utterance were time reversed, and summed. When present, this masker was either played to the same ear as LOW and HIGH (SAME-MASK), same ear as LOW with HIGH opposite (SPLIT-MASKLOW), or same ear as HIGH with LOW opposite (SPLIT-MASKHIGH). Intelligibility was best in the SAME and SPLIT conditions and much better than with either LOW or HIGH presented alone. Thus, in quiet, listeners can perceptually integrate information across target components with little spectro-temporal correlation, even when spatial cues promote segregation of these components. Performance in the SAME-MASK condition was generally better than either SPLIT-MASKLOW or SPLIT-MASKHIGH conditions (although worse than in SAME and SPLIT conditions). Furthermore, performance in the two SPLIT-MASK conditions was comparable to performance with LOW or HIGH alone. Thus, when spatial cues oppose grouping of the target and there is a competing off-frequency interference, listeners have difficulty integrating information across target components.
The role of combined consonant duration and amplitude processing on speech intelligibility in noise
Authors: Digiovanni Jeffrey J. , Wolfanger Jessica A.
Published: May 2008

Reduced speech intelligibility in noise, especially consonant perception, is a well documented phenomenon. Research in clear speech has revealed many acoustic and phonetic differences between speech spoken in a conversational versus clear-speech mode. In the present study two components were tested in combination, namely increases in consonant duration (and respective decreases in adjacent vowel duration) and selective-consonant amplification. Conditions in which intelligibility improvements were observed for the individual processing of duration and amplitude modifications were chosen and tested in combination. The components were applied to Hearing-in-Noise Test sentences and consonant-vowel (CV) pairs. Stimuli were presented in sound field to 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. Results will be discussed with regard to the greater body of research in clear speech.
Effect of amplification on the intelligibility of speech in hearing impaired children with and without dead regions in the cochlea
Authors: Malicka Alicja N. , Munro Kevin J. , Baer Thomas
Published: May 2008

Adults with high-frequency (HF) sensorineural hearing impairment with and without dead regions (DRs) in the cochlea differ in benefit from amplification of speech presented in quiet [Vickers et al., J. Acoust. Soc. Am. 110, 1164-1175 (2001)]. Subjects with HF DRs showed no improvement in speech intelligibility when spectral components of the speech above about 1.7 times the edge frequency of the DR were amplified according to a hearing-aid prescription formula while performance of those without DRs showed improvement with addition of amplified frequency components up to 7.5 kHz. In the present study we tested a group of six children (8-12 years old) who were experienced hearing-aid users with moderate-to-severe sensorineural hearing impairment. The presence of DRs was diagnosed using the TEN(HL) test and “fast” psychophysical tuning curves. Four children showed evidence for DRs (two unilateral and two bilateral). The vowel-consonant-vowel stimuli (65-dB SPL) were subjected to the frequency-gain characteristic prescribed by the DSL prescription formula then low-pass filtered with various cutoff frequencies and presented via headphones. The results showed that in ears with or without DRs the performance improved with increasing cutoff frequency up to 7.5 kHz.
Acoustic correlates of fricative perception
Authors: Katseff Shira
Published: May 2008

A number of spectral attributes have been proposed as correlates of fricative place of articulation. However, these attributes have difficulty distinguishing fricatives in noncanonical speech, e.g., from children. Because child speech is intelligible, it is possible that some of these characteristics are not used in speech perception. This study collected a corpus of /s/ and /S/-containing words from children between 3 and 5. Adults categorized fricatives extracted from this corpus. A series of regressions are used to compare the ability of spectral attributes to predict (1) human categorization of /s/ and /S/ productions, and (2) the intended fricative production. The analysis reveals that mistakes in perception are driven by overuse of a single spectral attribute, spectral slope. This result suggests caution in equating methods of computational separation with methods of human perception, and provides a simple method for investigating relationships between the two.
Analysis of structure borne transmission in a railway vehicle using Energy Flow Method
Authors: Bouvet Pascal
Published: May 2008

The Energy Flow Method (EFM) is a computationally efficient method, by which the results of a Finite Element Analysis (FEA) may be postprocessed to form energy flow models. It can be used to study structure borne transmission in structures, based on modal descriptions of the structure and the internal cavity. The EFM allows to form a Energy Influence Coefficient model (EIC) or an inverse SEA model. A first issue consists in partitioning the FE model into SEA subsystems, with respect to the SEA assumptions (particularly the weak coupling assumptions). An automatic technique based on cluster analysis, can be used to determine an appropriate SEA substructuring, before calculating the Coupling Loss Factors (CLF) between mechanical subsystem. A second issue is to calculate the CLF between structural and acoustic subsystems. This paper presents an industrial application of these techniques, for the analysis of structure borne transmission inside a tramway vehicle. The EFM technique is used to calculate the vibroacoustic transfer of the car body of the tramway, and comparison with measurements on a vehicle are presented.
Active structural-acoustic control of laminated cylindrical panels using vertically/obliquely reinforced 1-3 piezoelectric composite
Authors: Ray Manas C.
Published: May 2008

The performance of active constrained layer damping (ACLD) treatment for active structural-acoustic control of vibrating thin laminated cylindrical panels has been investigated. The constraining layer of the ACLD treatment has been considered to be made of vertically/obliquely reinforced 1-3 piezoelectric composite material. A finite element model has been developed for the laminated panels integrated with the patches of ACLD treatment to describe the coupled structural-acoustic behavior of the panels enclosing an acoustic cavity. Both velocity and pressure rate feedback controls have been implemented to activate the patches. Symmetric and antisymmetric cross-ply and antisymmetric angle-ply panels have been considered for evaluating the numerical results. Emphasis has also been placed on investigating the effect of the piezoelectric fiber orientation in the constraining layer and the shallowness angle of the cylindrical panels on the performance of the patches.
Experimental tests on smart panels for the reduction of sound radiation
Authors: Gardonio Paolo , Gonzalez Diaz Cristobal , Alujevic Neven , Aoki Yohko
Published: May 2008

This paper presents a comparative experimental study carried out on four types of smart panels for the reduction of sound radiation. The four panels are equipped with decentralized velocity feedback control units that are designed to generate active damping. In this way the frequency averaged response and sound radiation produced by stochastic disturbances can be effectively reduced at low audio frequencies. The first smart panel is composed of a 4×4 array of square piezoelectric patch actuators with accelerometer sensors at their centres. The second smart panel is composed of sixteen triangularly shaped piezoelectric patch actuators with base edges evenly distributed along the perimeter of the panel and accelerometer sensors located on the top vertices of the actuators. The third panel is composed of five electrodynamic inertial actuators with accelerometer sensors located under their footprints. The fourth panel is equipped with a light honeycomb trim panel mounted on four stiff mounts. A set of nine reactive electrodynamic actuators is located in the shallow cavity between the two panels. Accelerometer sensors are mounted at the two footprints of each actuator. The sensor-actuator pairs mounted in the four panels are used to implement analogue local feedback loops.
From the electrical shunting of a loudspeaker to active impedance control
Authors: Lissek Hervé , Sandoz Florian
Published: May 2008

Variable acoustic properties can be obtained on an electroacoustic transducer's voicing face by very basic control strategies, among which are the shunting of the loudspeaker (shortcut, variable electric load, or negative resistance disposals). It is proven that better performances with enhanced controllability can be obtained by way of hybrid feedback control, consisting in a double feedback loop, one on acoustic pressure at the diaphragm and the other on its velocity, leading to a global acoustic impedance control. The present work describes the theory of the hybrid feedback control, by way of block diagrams aiming at visualizing the control principle, starting from the shunted loudspeaker. Simulations of performances obtained on Matlab/Simulink are presented, and compared to experimental results obtained on analog prototypes mounted at the end of a dedicated impedance tube. A last, a discussion on stability issues follows, leading to concluding remarks on the disposal behavior and possible means of enhancements.
Use of peak filter for online secondary path modeling in a feedforward active ship-noise cancellation system
Authors: Bao Xueshan
Published: May 2008

The injected auxiliary noise used in online secondary path modeling method often disturbs the convergence of control filter and increases the residual sound field energy of the active noise cancellation (ANC) system. To mitigate this problem, an improved online secondary path modeling method is proposed, in which the transfer function of secondary path is reconstructed by a peak filter. The frequency response of new transfer function remains unchanged at the frequency band of the noise to be cancelled; while the other bands are attenuated deeply. With this system the simulation experiment of active ship-radiated noise cancellation is accomplished. The result shows that new system's noise cancellation capability is improved and residual sound field energy (including channel response to injected auxiliary noise) is smaller than conventional system without peak filter because of the simplified transfer function of secondary path and reduced energy disturbance between control filter and modeling filter.
Adaptive active noise control incorporating with a transfer function method for reducing acoustic feedback in a duct
Authors: Liu Yun-Hui
Published: May 2008

In most practical applications of active noise control, the acoustic feedback is a major problem that often interferes with the operation of the control system and even renders it unstable. We propose a frequency-domain method based on transfer function to reduce the influence of acoustic feedback based on plane wave transmission theory of sound in a duct. The original signal of primary noise is obtained from the measured signals of reference microphone and error microphone which are transformed to frequency domain by FFT and operated by the proposed method. In this study, Filtered-X LMS algorithm is applied to carry out the active noise controller incorporating with this transfer function method. Experimental results of active noise control show that the system has achieved 18.5 dB maximal attenuation in the frequency band 200-600 Hz. On the contrary, it is only attenuate 3-4 dB if acoustic feedback is present. Therefore, it was to verify that the proposed method of acoustic feedback cancellation incorporating with the Filtered-X LMS algorithm can effectively reduce the influence of acoustic feedback and obtains a better performance of active noise control in a duct.
A two-dimensional ANC system - from simulation to application
Authors: Kleinhenrich Christian , Krahé Detlef , Niepenberg Arndt
Published: May 2008

An existing two-dimensional active noise control (ANC) system attenuates a given noise (primary field) inside a certain area and frequency range by an additionally generated sound (secondary field). The two-dimensional ANC system consists of several modules containing a pair of microphones and a line array of loudspeakers each. The configuration of the modules depends on the acoustical environment and the desired shape of the protection area. Planning this is supported by a tool based on simulating the superposition of both sound fields. This spatial, close to reality simulation for predicting the achievable noise attenuation and determining important system parameters is implemented in MATLAB. With this planning tool it is possible to place all required acoustic sources and sinks into a virtual room and analyze the properties of the resulting sound field concerning real application scenarios. This paper draws a comparison between the results given by simulation and the subsequently realized ANC system within the scope of a typical application.
The question of disguised voice
Authors: Perrot Patrick , Chollet Gerard
Published: May 2008

State of the art automatic speaker recognition systems show very good results in the discrimination between different speakers under controlled recording conditions. In a forensic context, the conditions are uncontrolled and voice can be disguised. In cases of terrorism claim, extortion or kidnapping, it is of great interest for offenders to conceal their identity. Voice disguise is an important constraint to speaker discrimination. Some disguises produce a great variation of parameters and change the perception of an identity. The main risk is to confound a disguised voice and a normal voice and accuse an innocent individual. This paper proposes on one hand to present the impact of voice disguise on automatic speaker recognition and, on the other hand a statistical study in order to detect and identify four disguises among the most common. The first step consists in extracting features and the second step to classify them. MFCC (Mel Frequency Cepstral Coefficient) are considered as features and different classification algorithms have been tested. The studied disguises are based on a deliberated and non electronic way. The proposed analysis of disguised voice classification provides interesting results in detection by the use of SVM (Support Vector Machine) and in identification by the use of GMM (Gaussian mixture models).
Speaker identification on the SCOTUS corpus
Authors: Yuan Jiahong , Liberman Mark
Published: May 2008

This paper reports the results of our experiments on speaker identification in the SCOTUS corpus, which includes oral arguments from the Supreme Court of the United States. Our main findings are as follows: 1) a combination of Gaussian mixture models and monophone HMM models attains near-100% text-independent identification accuracy on utterances that are longer than one second; (2) the sampling rate of 11025 Hz achieves the best performance (higher sampling rates are harmful); and a sampling rate as low as 2000 Hz still achieves more than 90% accuracy; (3) a distance score based on likelihood numbers was used to measure the variability of phones among speakers; we found that the most variable phone is the phone UH (as in good), and the velar nasal NG is more variable than the other two nasal sounds M and N; 4.) our models achieved “perfect” forced alignment on very long speech segments (one hour). These findings and their significance are discussed.
Dependency of recognition rate on number of words for text-independent speaker recognition using vector quantization
Authors: Shimizu Hidenori , Funada Tetsuo
Published: May 2008

In this research, we discuss speaker recognition using the Kohonen feature map. The map is constructed for each speaker, and it is trained by using log-power and fourteenth-order mel-frequency cepstral coefficients (MFCC) and their temporal difference. The quantization distortion is computed between the input speech and a specific vector on the feature map of each speaker. We conduct speaker recognition experiment based on VQ distortion. Utterances of prefectural name in Japan are used as speech data. We examine particularly the dependency of recognition rate on number of words used for recognition. According to our experiments of speaker identification, this system correctly recognizes 98.9% by using a single word for 40 male speakers, while it attains 100% by using more than three words. Moreover, we confirmed superiority of using VQ over HMM under the same experimental conditions.
Sound structure and function of English as a global language
Authors: Bradlow Ann
Published: May 2008

Many English conversations across the globe today involve non-native speakers. To understand the sound structure of English in a global context, we asked native and non-native listeners to rate the similarity of various languages (n=17) to English. In general, the native and non-native listeners' distance-from-English ratings were positively correlated; but, the non-natives tended to rank languages as more distant from English than the natives. Native and non-native listeners then rated the accents of native and non-native speakers. Again, the native and non-native listener ratings were positively correlated, but the non-native listeners generally rated both native and non-native speakers as more foreign-accented than did the native listeners. Together, these data suggest that both native and non-native English generally sound less “marked” to native listeners than to non-native listeners. As a means of understanding how English functions as a global language, we examined communicative efficiency in task-oriented dialogues between various combinations of natives and non-natives. All pairs successfully completed the task; yet, when one or more non-native talkers were involved, the task took about twice as long to complete as when both were native talkers. Together, these studies help characterize the structure and function of English as a twenty-first century lingua franca.
Plurality and plasticity of neural representation for speech sounds
Authors: Zhang Yang
Published: May 2008

High-density event-related potential (ERP) and magnetoencephalography (MEG) data are presented here to illustrate the plurality and plasticity of speech representation in the spatio-temporal dynamics of neural responses. First, there is evidence for both continuous and discrete processing of speech sounds depending on the acoustic parameters and experimental conditions. Speech discrimination and neural sensitivity measures suggest that the perceptual organization of within-category variations is based on both phonetic knowledge and acoustic analysis. Second, attention plays an important role in determining the neural response patterns. Focused attention significantly changes the hemispheric laterality and strength of neural activation for speech processing. Selectively attending to the different aspects (phonetic vs social-indexical information) of the same speech stimuli shows different regional recruitment, hemispheric involvement and temporal characteristics. Third, the nature of language input and experience can significantly affect learning outcome. Enriched exposure alters the neural representations for speech sounds and plasticity of the system. Theoretical and practical implications for language learning and treatment are discussed.
Call early in the evening on a spring day
Authors: Eskenazi Maxine , Raux Antoine
Published: May 2008

The CMU Let's Go Spoken Dialogue System has been used daily for about three years to answer calls to the Pittsburgh Port Authority for bus information in the evening and on weekends. This has resulted in a database of over 50 000 spoken dialogues as of January 2008, one of the largest publicly available sets of this type of data. While retraining the system with part of this data, it became apparent that there are times of the day, of the week and of the year when the average number of successful calls is significantly higher. We will present evidence, using these three measures of time (hour, day of week, month of year) and criteria such as signal-to-noise ratio, estimated success rate, number of turns per dialogue, number of non-understandings per dialogue, and barge-in rate to detect the regular, predictable appearance of high and low success rates and to suggest methods for palliating this effect in order to increase overall dialogue success rates.
Modeling perception of breathy voice quality in vowels using data obtained in an adaptive matching task
Authors: Shrivastav Rahul , Camacho Arturo , Patel Sona A. , Eddins David A.
Published: May 2008

A computational model of breathy voice quality has been developed based on perceptual judgments of synthetic /a/ vowels [Shrivastav, Camacho, and Patel, JASA 120(5), 3248]. This model is based on the ratio of the loudness of aspiration noise (“noise loudness;” NL) to the loudness of the periodic components of the vowel when masked by the aspiration noise (“partial loudness;” PL) [Shrivastav and Sapienza, JASA, 114(1), 2218-2224 (2005)]. Results showed that the model accounted for a large amount of variance in perceptual ratings of training data (R-square=0.92) but not for testing data (R-square=0.59). It was hypothesized that this reduction was partly related to the use of a magnitude estimation task to obtain perceptual judgments, since magnitude estimates are biased by the range of the quantity measured and are prone to inconsistencies in how subjects assign numbers to items [Poulton, 1989; Guilford, 1954]. To minimize such biases, a matching task may be used to obtain ratio level estimates of breathiness [Patel, Shrivastav and Eddins, JASA, 119(5), 3340 (2006)]. The present study describes a model to predict breathy voice quality derived from perceptual judgments obtained using a matching task.
Effectiveness of prosodic features in the perception of mandarin utterance-final particles
Authors: Li Bin , Zhu Jinping
Published: May 2008

Utterance-final particles (UFPs) in Mandarin are regarded as discourse markers, whose “elusive” meanings and functions have been investigated in different perspectives such as pragmatics and semantics. Our study focuses on two most frequently used UFPs: NE and A, both of which can occur in identical utterances. The difference lies in that NE implies a shared common ground between the speaker and the hearer, whereas such commonness lacks for A. A preliminary acoustic analysis found that prosodic features of an utterance were more salient when a UFP was not used. The current study aims at the effectiveness of these prosodic features in perceiving these particles. Native speakers of Mandarin are asked to listen to sentences, which may originally contain a UFP that is removed. Their tasks are to decide whether a UFP, and which one, is needed in a given sentence. It is hypothesized that without textual cues, speakers will reply on prosodic features in their judgment; therefore, they will perform better in sentences where UFPs are not used originally and where prosodic cues are more salient.
The effect of learning on listening to ultra-Fast speech
Authors: Nishimoto Takuya , Kariya Yukika , Watanabe Takayuki
Published: May 2008

We investigated the intelligibility of ultrafast speech which may be used for screen reader for persons with visual disability. The subjects were 35 women who are university students and are not visually/hearing impaired. They were divided into four groups and they listened to 150 words with the speed of approximately 20 morae/s. The vocabulary contained the tasks of high and low familiarity words, and the orders of tasks were different by the groups. Four morae Japanese words from the FW03 database were used as the vocabulary of the recall test. As a result, significant learning effect was observed in cases where the subject listened the high familiarity words in succession. This indicates that the learning effect to the ultrafast speech is promoted when the mental lexical access is easy. We also investigated the mental workloads of the listening task using the NASA-TLX method. As the results, significantly high workload scores were observed at the listening of low familiarity words. The results also suggested that the mental workload decreases when the subject was aware that the mental lexical access was difficult.
Clear speech intelligibility and accentedness ratings for native and non-native talkers and listeners
Authors: Smiljanic Rajka , Bradlow Ann
Published: May 2008

This study investigated how native language background (L1) interacts with speaking style in determining levels of speech intelligibility. In four experiments, we explored whether native and non-native hyper-articulation clear speech strategies provide similar intelligibility benefits for native and high proficiency non-native listeners. The sentence-in-noise perception results revealed that native speech was preferred over non-native speech by both listener groups even when non-native talkers and listeners shared the same L1. Clear speech was shown to be beneficial for both the native and fluent non-native listeners. However, non-native clear speech enhanced intelligibility less than native clear speech, supporting the hypothesis that clear speech production strategies involve enhancement of language-specific phonological contrasts. In order to assess the relationship between objective intelligibility measures and subjective accentedness ratings, we obtained accentedness ratings of native and non-native conversational and clear speech by native and non-native listeners. The results showed that objective intelligibility and subjective accentedness were independent. Overall, these results provide strong evidence that clear speech involves language-specific modifications. Nevertheless, native and high proficiency non-native clear speech modifications are generally helpful for both native and high proficiency non-native listeners even when the objective intelligibility and subjective accentedness levels diverged for various listener and talker groups.
Vietnamese monophthong vowel production by native speakers and American adult learners
Authors: Winn Matthew , Blodgett Allison , Bauman Jessica , Bowles Anita , Charters Lykara , Rytting Anton , Shamoo Jessica
Published: May 2008

This study provides new native speaker data regarding the vowel space and duration contrasts of Vietnamese monophthongs. These data add to existing empirical accounts and raise questions about descriptions aimed at adult language learners. In addition, specific acoustic measures target difficulties that English-speaking adults encounter when producing Vietnamese vowels. These measures include the rounding distinction for back vowels, height distinction for central vowels, and duration distinction between short and long vowels. Results show that adult learners exhibit greater variability and overall less accurate production than native speakers for the high-back unrounded vowel u', with slightly better performance for the mid-back unrounded vowel o', perhaps because it is similar to English / Λ /. Additionally, these learners demonstrate varying levels of success with regard to the production of central vowel height but generally fail to distinguish the two vowel duration categories exhibited by native speakers.