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Journal articles on the topic 'Speech waveform analysis'

Author: Grafiati

Published: 4 June 2021

Last updated: 6 February 2022

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1

Askenfelt, Anders G., and Britta Hammarberg. "Speech Waveform Perturbation Analysis." Journal of Speech, Language, and Hearing Research 29, no. 1 (1986): 50–64. http://dx.doi.org/10.1044/jshr.2901.50.

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The performance of seven acoustic measures of cycle-to-cycle variations (perturbations) in the speech waveform was compared. All measures were calculated automatically and applied on running speech. Three of the measures refer to the frequency of occurrence and severity of waveform perturbations in special selected parts of the speech, identified by means of the rate of change in the fundamental frequency. Three other measures refer to statistical properties of the distribution of the relative frequency differences between adjacent pitch periods. One perturbation measure refers to the percenta

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2

Yohanes, Banu W. "Linear Prediction and Long Term Predictor Analysis and Synthesis." Techné : Jurnal Ilmiah Elektroteknika 16, no. 01 (2017): 49–58. http://dx.doi.org/10.31358/techne.v16i01.158.

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Spectral analysis may not provide an accurate description of speech articulation. This article presents an experimental setup of representing speech waveform directly in terms of timevarying parameters. It is related to the transfer function of the vocal tract. Linear Prediction, Long Term Predictor Analysis, and Synthesis filters are designed and implemented, as well as the theory behind introduced. The workflows of the filters are explained by detailed and codes of those filters. Original waveform files are framed with Hamming window and for each frames the filters are applied, and the recon

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Tadic, Predrag, Zeljko Djurovic, and Branko Kovacevic. "Analysis of speech waveform quantization methods." Journal of Automatic Control 18, no. 1 (2008): 19–22. http://dx.doi.org/10.2298/jac0801019t.

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Digitalization, consisting of sampling and quantization, is the first step in any digital signal processing algorithm. In most cases, the quantization is uniform. However, having knowledge of certain stochastic attributes of the signal (namely, the probability density function, or pdf), quantization can be made more efficient, in the sense of achieving a greater signal to quantization noise ratio. This means that narrower channel bandwidths are required for transmitting a signal of the same quality. Alternatively, if signal storage is of interest, rather than transmission, considerable savings

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Read, Charles, Eugene H. Buder, and Raymond D. Kent. "Speech Analysis Systems." Journal of Speech, Language, and Hearing Research 35, no. 2 (1992): 314–32. http://dx.doi.org/10.1044/jshr.3502.314.

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Performance characteristics are reviewed for seven systems marketed for acoustic speech analysis: CSpeech, CSRE, ILS-PC, Kay Elemetrics model 5500 Sona-Graph, MacSpeech Lab II, MSL, and Signalyze. The characteristics reviewed include system components, basic capabilities (signal acquisition, waveform operations, analysis, and other functions), documentation, user interface, data formats and journaling, speed and precision of spectral analysis, and speed and precision of fundamental frequency analysis. Basic capabilities are also tabulated for three recently introduced systems: the Sensimetrics

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Debruyne, F., P. Delaere, J. Wouters, and P. Uwents. "Acoustic analysis of tracheo-oesophageal versus oesophageal speech." Journal of Laryngology & Otology 108, no. 4 (1994): 325–28. http://dx.doi.org/10.1017/s0022215100126660.

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AbstractIn order to evaluate the vocal quality of tracheo-oesophageal and oesophageal speech, several objective acoustic parameters were measured in the acoustic waveform (fundamental frequency, waveform perturbation) and in the frequency spectrum (harmonic prominence, spectral slope). Twelve patients using tracheo-oesophageal speech (with the Provox® valve) and 12 patients using oesophageal speech for at least two months, participated.The main results were that tracheo-oesophageal voices more often showed a detectable fundamental frequency, and that this fundamental frequency was fairly stabl

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Boggs, George J., and Michael D. Connelly. "WFORM: A graphical speech-waveform editing and analysis system." Behavior Research Methods, Instruments, & Computers 18, no. 1 (1986): 25–31. http://dx.doi.org/10.3758/bf03200989.

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7

Balakrishnan Sivakumar and Praveen Kadakola Biligirirangaiah. "Analysis of vowel addition or deletion in Continuous Speech." Global Journal of Engineering and Technology Advances 7, no. 3 (2021): 136–43. http://dx.doi.org/10.30574/gjeta.2021.7.3.0084.

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In order to improve the recognition performance, the articulation of the transcription is very important in the process of training. For continuous speech, the essential characteristics of various speakers are pronunciation variation, over focused or inadequately highlighted words can results the waveform misalignment in the sub word unit margin. Because of the deviation in the articulation leads into misalignment when this is compared with articulation dictionary. So the deletion or insertion of the sub word is necessary. This happens because for each expression, the transcription is not prec

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Zhang, Fawen, Chelsea Benson, and Steven J. Cahn. "Cortical Encoding of Timbre Changes in Cochlear Implant Users." Journal of the American Academy of Audiology 24, no. 01 (2013): 046–58. http://dx.doi.org/10.3766/jaaa.24.1.6.

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Background: Most cochlear implant (CI) users describe music as a noise-like and unpleasant sound. Using behavioral tests, most prior studies have shown that perception of pitch-based melody and timbre is poor in CI users. Purpose: This article will focus on cortical encoding of timbre changes in CI users, which may allow us to find solutions to further improve CI benefits. Furthermore, the value of using objective measures to reveal neural encoding of timbre changes may be reflected in this study. Research Design: A case-control study of the mismatch negativity (MMN) using electrophysiological

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Ingram, Kelly, Ferenc Bunta, and David Ingram. "Digital Data Collection and Analysis." Language, Speech, and Hearing Services in Schools 35, no. 2 (2004): 112–21. http://dx.doi.org/10.1044/0161-1461(2004/013).

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Technology for digital speech recording and speech analysis is now readily available for all clinicians who use a computer. This article discusses some advantages of moving from analog to digital recordings and outlines basic recording procedures. The purpose of this article is to familiarize speech-language pathologists with computerized audio files and the benefits of working with those sound files as opposed to using analog recordings. This article addresses transcription issues and offers practical examples of various functions, such as playback, editing sound files, using waveform display

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10

Chi-Sang Jung, Young-Sun Joo, and Hong-Goo Kang. "Waveform Interpolation-Based Speech Analysis/Synthesis for HMM-Based TTS Systems." IEEE Signal Processing Letters 19, no. 12 (2012): 809–12. http://dx.doi.org/10.1109/lsp.2012.2221703.

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11

Heo, Hee-Soo, Byung-Min So, IL-Ho Yang, and Ha-Jin Yu. "A Speech Waveform Forgery Detection Algorithm Based on Frequency Distribution Analysis." Phonetics and Speech Sciences 7, no. 4 (2015): 35–40. http://dx.doi.org/10.13064/ksss.2015.7.4.035.

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Palmer, Shannon B., and Frank E. Musiek. "N1-P2 Recordings to Gaps in Broadband Noise." Journal of the American Academy of Audiology 24, no. 01 (2013): 037–45. http://dx.doi.org/10.3766/jaaa.24.1.5.

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Background: Normal temporal processing is important for the perception of speech in quiet and in difficult listening situations. Temporal resolution is commonly measured using a behavioral gap detection task, where the patient or subject must participate in the evaluation process. This is difficult to achieve with subjects who cannot reliably complete a behavioral test. However, recent research has investigated the use of evoked potential measures to evaluate gap detection. Purpose: The purpose of the current study was to record N1-P2 responses to gaps in broadband noise in normal hearing youn

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Enders, Jörg, Weihua Geng, Peijun Li, Michael W. Frazier, and David J. Scholl. "The shift-invariant discrete wavelet transform and application to speech waveform analysis." Journal of the Acoustical Society of America 117, no. 4 (2005): 2122–33. http://dx.doi.org/10.1121/1.1869732.

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Ikuma, Takeshi, Melda Kunduk, and Andrew J. McWhorter. "Advanced Waveform Decomposition for High-Speed Videoendoscopy Analysis." Journal of Voice 27, no. 3 (2013): 369–75. http://dx.doi.org/10.1016/j.jvoice.2013.01.004.

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Nisha, K. V., and U. Ajith Kumar. "Pre-Attentive Neural Signatures of Auditory Spatial Processing in Listeners With Normal Hearing and Sensorineural Hearing Impairment: A Comparative Study." American Journal of Audiology 28, no. 2S (2019): 437–49. http://dx.doi.org/10.1044/2018_aja-ind50-18-0099.

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Purpose This study was carried out to understand the neural intricacies of auditory spatial processing in listeners with sensorineural hearing impairment (SNHI) and compare it with normal hearing (NH) listeners using both local and global measures of waveform analyses. Method A standard group comparison research design was adopted in this study. Participants were assigned to 2 groups. Group I consisted of 13 participants with mild–moderate flat or sloping SNHI, while Group II consisted of 13 participants with NH sensitivity. Electroencephalographic data using virtual acoustic stimuli (spatiall

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Davis, Tara, Nicholas Stanley, and Lori Foran. "Age-Related Effects of Dichotic Attentional Mode on Interaural Asymmetry: An AERP Study with Independent Component Analysis." Journal of the American Academy of Audiology 26, no. 05 (2015): 461–77. http://dx.doi.org/10.3766/jaaa.14094.

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Background: The degree of interaural asymmetry (IA) obtained on a dichotic listening task is affected by attentional demands attributable to the mode of test administration. Previous research has shown that IA in the elderly is more influenced by increased attentional demands than young adults (YAs), but the effect of attentional mode on IA in middle-aged adults (MAs) has not been established. Auditory event-related potentials (AERPs), such as the N400, allow the evaluation of subtle differences in linguistic and cognitive processing between YAs and MAs that are imperceptible by behavioral mea

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17

Saloni, Saloni, Rajender K. Sharma, and Anil K. Gupta. "Human Voice Waveform Analysis for Categorization of Healthy and Parkinson Subjects." International Journal of Healthcare Information Systems and Informatics 11, no. 1 (2016): 21–35. http://dx.doi.org/10.4018/ijhisi.2016010102.

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Parkinson disease is a neurological disorder. In this disease control over body muscles get disturbed. In almost 90% of the cases, people suffering from Parkinson disease (PD) have speech disorders. The goal of the paper is to differentiate healthy and PD affected persons using voice analysis. There are no well-developed lab techniques available for Parkinson detection. Parkinson detection using voice analysis is a noninvasive, reliable and economic method. Using this technique patient need not to visit the clinic. In this paper the authors have recorded 155 phonations from 25 healthy and 22 P

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18

Childers, D. G., D. M. Hicks, G. P. Moore, L. Eskenazi, and A. L. Lalwani. "Electroglottography and Vocal Fold Physiology." Journal of Speech, Language, and Hearing Research 33, no. 2 (1990): 245–54. http://dx.doi.org/10.1044/jshr.3302.245.

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The electroglottogram (EGG) is known to be related to vocal fold motion. A major hypothesis undergoing examination in several research centers is that the EGG is related to the area of contact of the vocal folds. This hypothesis is difficult to substantiate with direct measurements using human subjects. However, other supporting evidence can be offered. For this study we made measurements from synchronized ultra high-speed laryngeal films and from EGG waveforms collected from subjects with normal larynges and patients with vocal disorders. We compare certain features of the EGG waveform to (a)

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Pan, Hui, Mei Gao, and Yan Pan. "Research on Signal Acquisition and Analysis System Based on Virtual Instrument Technology." Applied Mechanics and Materials 556-562 (May 2014): 4321–24. http://dx.doi.org/10.4028/www.scientific.net/amm.556-562.4321.

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In this paper, we used Lab VIEW development platform, designed a virtual signal acquisition and analysis system based on the soundcard. In addition we studied the signal acquisition performance of it. This system has functions of signal acquisition, analysis, waveform display and storage. It is easy to use with good interface, and function appending is available on user requirement. It offers a kind of DAQ solution with low cost. It has broad application prospect, can be extended to the field of speech recognition, ambient noise monitoring and laboratory measurement etc.

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20

Gonzalez, Jennifer E., and Frank E. Musiek. "The Onset–Offset N1–P2 Auditory Evoked Response in Individuals With High-Frequency Sensorineural Hearing Loss: Responses to Broadband Noise." American Journal of Audiology 30, no. 2 (2021): 423–32. http://dx.doi.org/10.1044/2021_aja-20-00113.

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Purpose Clinical use of electrophysiologic measures has been limited to use of brief stimuli to evoke responses. While brief stimuli elicit onset responses in individuals with normal hearing and normal central auditory nervous system (CANS) function, responses represent the integrity of a fraction of the mainly excitatory central auditory neurons. Longer stimuli could provide information regarding excitatory and inhibitory CANS function. Our goal was to measure the onset–offset N1–P2 auditory evoked response in subjects with normal hearing and subjects with moderate high-frequency sensorineura

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21

He, Dong Kang, You Cai Xu, Xin Shi Li, et al. "Research on Applied-Information Technology in Digital Speech Based on LMD Algorithm." Advanced Materials Research 1014 (July 2014): 447–51. http://dx.doi.org/10.4028/www.scientific.net/amr.1014.447.

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As a new nonlinear and non-stationary signal analysis method,local mean decomposition (LMD) has a good adaptability. We decompose the original non-stationary acceleration vibration signals into several stationary production function (PF).But performing LMD will produce end effects which make results distorted. A hidden Markov model (HMM)-based speech recognition system for Chinese spell.After analyzing reasons for end effects of LMD in detail,a new method based on weighted matching similar waveform was proposed.Experiments in speech recognition to the production function as the training model,

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22

Carney, Arlene Earley. "Vibrotactile Perception of Segmental Features of Speech." Journal of Speech, Language, and Hearing Research 31, no. 3 (1988): 438–48. http://dx.doi.org/10.1044/jshr.3103.438.

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This experiment compared the recognition performance of artificially deafened listeners for segmental stimuli presented through a single-channel tactile device and through a 24-channel vocoder. Both consonant and vowel stimuli were tested under visual only, tactile only, and visual + tactile conditions. Each subject received a pretest, eight 2-hr training sessions, and a posttest. Results indicated no significant differences between subjects' overall recognition performance with two different tactile devices. Analysis of consonant confusions showed that both devices transmit the features of vo

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23

Nishiguchi, Masayuki. "Method and apparatus for speech encoding and decoding by sinusoidal analysis and waveform encoding with phase reproducibility." Journal of the Acoustical Society of America 125, no. 5 (2009): 3486. http://dx.doi.org/10.1121/1.3139576.

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24

Milenkovic, Paul, and Feng Mo. "Effect of the vocal tract yielding sidewall on inverse filter analysis of the glottal waveform." Journal of Voice 2, no. 4 (1988): 271–78. http://dx.doi.org/10.1016/s0892-1997(88)80019-5.

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25

Jacob, Agnes, and P. Mythili. "Developing a Child Friendly Text-to-Speech System." Advances in Human-Computer Interaction 2008 (2008): 1–6. http://dx.doi.org/10.1155/2008/597971.

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This paper discusses the implementation details of a child friendly, good quality, English text-to-speech (TTS) system that is phoneme-based, concatenative, easy to set up and use with little memory. Direct waveform concatenation and linear prediction coding (LPC) are used. Most existing TTS systems are unit-selection based, which use standard speech databases available in neutral adult voices. Here reduced memory is achieved by the concatenation of phonemes and by replacing phonetic wave files with their LPC coefficients. Linguistic analysis was used to reduce the algorithmic complexity inste

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Xu, Xiaona, Li Yang, Yue Zhao, and Hui Wang. "End-to-End Speech Synthesis for Tibetan Multidialect." Complexity 2021 (January 25, 2021): 1–8. http://dx.doi.org/10.1155/2021/6682871.

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The research on Tibetan speech synthesis technology has been mainly focusing on single dialect, and thus there is a lack of research on Tibetan multidialect speech synthesis technology. This paper presents an end-to-end Tibetan multidialect speech synthesis model to realize a speech synthesis system which can be used to synthesize different Tibetan dialects. Firstly, Wylie transliteration scheme is used to convert the Tibetan text into the corresponding Latin letters, which effectively reduces the size of training corpus and the workload of front-end text processing. Secondly, a shared feature

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Dahmani, Mohamed, and Mhania Guerti. "Recurrence Quantification Analysis of Glottal Signal as non Linear Tool for Pathological Voice Assessment and Classification." International Arab Journal of Information Technology 17, no. 6 (2020): 857–66. http://dx.doi.org/10.34028/iajit/17/6/4.

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Automatic detection and assessment of Vocal Folds pathologies using signal processing techniques knows an extensively challenge use in the voice or speech research community. This paper contributes the application of the Recurrence Quantification Analysis (RQA) to a glottal signal waveform in order to evaluate the dynamic process of Vocal Folds (VFs) for diagnosis and classify the voice disorders. The proposed solution starts by extracting the glottal signal waveform from the voice signal through an inverse filtering algorithm. In the next step, the parameters of RQA are determined via the Rec

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Davis, Tara M., James Jerger, and Jeffrey Martin. "Electrophysiological Evidence of Augmented Interaural Asymmetry in Middle-Aged Listeners." Journal of the American Academy of Audiology 24, no. 03 (2013): 159–73. http://dx.doi.org/10.3766/jaaa.24.3.3.

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Background: Various dimensions of auditory processing, especially the perception of speech in the presence of background competition, have been shown to deteriorate with age. A persistent problem in the assessment of these age-related changes has been the high prevalence of age-related high-frequency hearing loss in elderly persons. Some investigators have suggested that a more fruitful approach to the study of age-related decline might be to study middle-aged, rather than elderly, persons, where confounding high-frequency hearing loss is less prevalent. Purpose: To determine whether an increa

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Tan, Jingqian, Jia Luo, Xin Wang, et al. "Analysis of Click and Swept-Tone Auditory Brainstem Response Results for Moderate and Severe Sensorineural Hearing Loss." Audiology and Neurotology 25, no. 6 (2020): 336–44. http://dx.doi.org/10.1159/000507691.

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<b><i>Introduction:</i></b> Auditory brainstem response (ABR) is one of the commonly used methods in clinical settings to evaluate the hearing sensitivity and auditory function. The current ABR measurement usually adopts click sound as the stimuli. However, there may be partial ABR amplitude attenuation due to the delay characteristics of the cochlear traveling wave along the basilar membrane. To solve that problem, a swept-tone method was proposed, in which the show-up time of different frequency components was adjusted to compensate the delay characteristics of the co

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Miller, Christi W., Joshua G. W. Bernstein, Xuyang Zhang, Yu-Hsiang Wu, Ruth A. Bentler, and Kelly Tremblay. "The Effects of Static and Moving Spectral Ripple Sensitivity on Unaided and Aided Speech Perception in Noise." Journal of Speech, Language, and Hearing Research 61, no. 12 (2018): 3113–26. http://dx.doi.org/10.1044/2018_jslhr-h-17-0373.

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Purpose This study evaluated whether certain spectral ripple conditions were more informative than others in predicting ecologically relevant unaided and aided speech outcomes. Method A quasi-experimental study design was used to evaluate 67 older adult hearing aid users with bilateral, symmetrical hearing loss. Speech perception in noise was tested under conditions of unaided and aided, auditory-only and auditory–visual, and 2 types of noise. Predictors included age, audiometric thresholds, audibility, hearing aid compression, and modulation depth detection thresholds for moving (4-Hz) or sta

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Perez, Ana, Karin Ziliotto, and Liliane Pereira. "Test-Retest of Long Latency Auditory Evoked Potentials (P300) with Pure Tone and Speech Stimuli." International Archives of Otorhinolaryngology 21, no. 02 (2016): 134–39. http://dx.doi.org/10.1055/s-0036-1583527.

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Introduction Long latency auditory evoked potentials, especially P300, have been used for clinical evaluation of mental processing. Many factors can interfere with Auditory Evoked Potential - P300 results, suggesting large intra and inter-subject variations. Objective The objective of the study was to identify the reliability of P3 components (latency and amplitude) over 4–6 weeks and the most stable auditory stimulus with the best test-retest agreement. Methods Ten normal-hearing women participated in the study. Only subjects without auditory processing problems were included. To determine th

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Davis, Tara M., and James Jerger. "The Effect of Middle Age on the Late Positive Component of the Auditory Event-Related Potential." Journal of the American Academy of Audiology 25, no. 02 (2014): 199–209. http://dx.doi.org/10.3766/jaaa.25.2.8.

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Background: We recently described a research study in which age-related changes in interaural asymmetry were elicited using the N400 of the auditory event-related potentials (AERP) (Davis et al, 2013). The N400 was the primary focus due to its sensitivity to various aspects of semantic processing (Kutas and Hillyard, 1984), which we measured using a quasi-dichotic semantic category judgment task in competing speech. In this article, we describe age-related changes that occurred in the late positive component (LPC) of the AERP in the same study. The LPC peak occurs subsequent to the N400 peak o

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Mozaffarilegha, Marjan, Ali Esteki, Mohsen Ahadi, and Ahmadreza Nazeri. "Identification of Dynamic Patterns of Speech-Evoked Auditory Brainstem Response Based on Ensemble Empirical Mode Decomposition and Nonlinear Time Series Analysis Methods." International Journal of Bifurcation and Chaos 26, no. 12 (2016): 1650202. http://dx.doi.org/10.1142/s0218127416502023.

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The speech-evoked auditory brainstem response (sABR) shows how complex sounds such as speech and music are processed in the auditory system. Speech-ABR could be used to evaluate particular impairments and improvements in auditory processing system. Many researchers used linear approaches for characterizing different components of sABR signal, whereas nonlinear techniques are not applied so commonly. The primary aim of the present study is to examine the underlying dynamics of normal sABR signals. The secondary goal is to evaluate whether some chaotic features exist in this signal. We have pres

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Dzulkarnain, Ahmad Aidil Arafat, Wan Mahirah Wan Mhd Pandi, Wayne J. Wilson, Andrew P. Bradley, and Faizah Sapian. "A preliminary investigation into the use of an auditory brainstem response (ABR) simulator for training audiology students in waveform analysis." International Journal of Audiology 53, no. 8 (2014): 514–21. http://dx.doi.org/10.3109/14992027.2014.897763.

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Sussman, Harvey M., David Fruchter, Jon Hilbert, and Joseph Sirosh. "Linear correlates in the speech signal: The orderly output constraint." Behavioral and Brain Sciences 21, no. 2 (1998): 241–59. http://dx.doi.org/10.1017/s0140525x98001174.

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Neuroethological investigations of mammalian and avian auditory systems have documented species-specific specializations for processing complex acoustic signals that could, if viewed in abstract terms, have an intriguing and striking relevance for human speech sound categorization and representation. Each species forms biologically relevant categories based on combinatorial analysis of information-bearing parameters within the complex input signal. This target article uses known neural models from the mustached bat and barn owl to develop, by analogy, a conceptualization of human processing of

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Ringelienė, Živilė, and Mark Filipovič. "Žodžių atpažinimo, grįsto paslėptaisiais Markovo modeliais, vizualizavimo ir analizės programinė įranga." Informacijos mokslai 56 (January 1, 2011): 63–72. http://dx.doi.org/10.15388/im.2011.0.3150.

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Straipsnyje aprašomas atpažinimo, grįsto paslėptaisiais Markovo modeliais, sistemos prototipo veikimas. Ši programinė įranga skirta lietuvių kalbos žodžių atpažinimui tirti. Nagrinėjama, kaip sistemos pateikiama informacija apie žodžių atpažinimo procesą ir rezultatus padeda analizuoti klaidų priežastis. Žodžio atpažinimas priklauso nuo žodžio ribų nustatymo tikslumo. Signalo, energijos, žodžio ribų vizualizavimas leidžia lengviau įvertinti, ar sistema teisingai nustatė ribas. Jei žodis atpažintasklaidingai dėl to, kad buvo blogai nustatytos ribos, galima keisti sistemos parametrų, darančių įt

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Barker, Matthew D., Abin Kuruvilla-Mathew, and Suzanne C. Purdy. "Cortical Auditory-Evoked Potential and Behavioral Evidence for Differences in Auditory Processing between Good and Poor Readers." Journal of the American Academy of Audiology 28, no. 06 (2017): 534–45. http://dx.doi.org/10.3766/jaaa.16054.

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Background: The relationship between auditory processing (AP) and reading is thought to be significant; however our understanding of this relationship is somewhat limited. Previous studies have investigated the relation between certain electrophysiological and behavioral measures of AP and reading abilities in children. This study attempts to further understand that relation. Purpose: Differences in AP between good and poor readers were investigated using electrophysiological and behavioral measures. Study Sample: Thirty-two children (15 female) aged 9–11 yr were placed in either a good reader

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Kist, Andreas M., Pablo Gómez, Denis Dubrovskiy, et al. "A Deep Learning Enhanced Novel Software Tool for Laryngeal Dynamics Analysis." Journal of Speech, Language, and Hearing Research 64, no. 6 (2021): 1889–903. http://dx.doi.org/10.1044/2021_jslhr-20-00498.

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Purpose High-speed videoendoscopy (HSV) is an emerging, but barely used, endoscopy technique in the clinic to assess and diagnose voice disorders because of the lack of dedicated software to analyze the data. HSV allows to quantify the vocal fold oscillations by segmenting the glottal area. This challenging task has been tackled by various studies; however, the proposed approaches are mostly limited and not suitable for daily clinical routine. Method We developed a user-friendly software in C# that allows the editing, motion correction, segmentation, and quantitative analysis of HSV data. We f

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Atcherson, Samuel R., and Page C. Moore. "Are Chirps Better than Clicks and Tonebursts for Evoking Middle Latency Responses?" Journal of the American Academy of Audiology 25, no. 06 (2014): 576–83. http://dx.doi.org/10.3766/jaaa.25.6.7.

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Background: The middle latency response (MLR) is considered a valid clinical tool for assessing the integrity of cortical and subcortical structures. Several investigators have demonstrated that a rising frequency chirp stimulus is capable of eliciting not only larger wave V amplitudes but larger MLR components as well. However, the chirp has never been specifically examined in a hemispheric electrode montage setup that is typical for neurodiagnostic application and site-of-lesion testing. Purpose: The purpose of this study was to examine the effect of chirp, click, and toneburst stimuli on ML

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Bae, In-Ho, Soo-Geun Wang, Soon-Bok Kwon, Seong-Tae Kim, Eui-Suk Sung, and Jin-Choon Lee. "Clinical Application of Two-Dimensional Scanning Digital Kymography in Discrimination of Diplophonia." Journal of Speech, Language, and Hearing Research 62, no. 10 (2019): 3643–54. http://dx.doi.org/10.1044/2019_jslhr-s-18-0175.

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Purpose The purpose of this study was to investigate the characteristics of diplophonia using an auditory perception and multimodal simultaneous examination, which included sound waveform analysis, electroglottography (EGG), digital kymography (DKG), and 2-dimensional scanning digital kymography (2D DKG). Additionally, we compared the diagnostic accuracy of each method using a binary classifier in confusion matrix and convenience of discrimination, based on the time required for interpretation. Method One normophonic male, 12 patients with diplophonia, and 12 dysphonia patients without diploph

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Walliker, J. R., and A. J. Fourcin. "Signal-Processing Hearing Aids for the Totally and Profoundly Deaf." Annals of Otology, Rhinology & Laryngology 96, no. 1_suppl (1987): 74–76. http://dx.doi.org/10.1177/00034894870960s137.

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We have developed a family of single-channel signal-processing aids for the profoundly and totally deaf. Common to them all are the analysis of speech into the components most important to the deaf lipreader; the synthesis of stimuli which make the best use of the patient's sensory abilities; and facilities to ensure accurate matching of the aid to the patient. The totally deaf are electrically stimulated by electrodes on the promontory or on the round window of the cochlea using charge-balanced controlled current square waves automatically adjusted to be at a comfortable level. Many potential

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Rong, Panying. "Automated Acoustic Analysis of Oral Diadochokinesis to Assess Bulbar Motor Involvement in Amyotrophic Lateral Sclerosis." Journal of Speech, Language, and Hearing Research 63, no. 1 (2020): 59–73. http://dx.doi.org/10.1044/2019_jslhr-19-00178.

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Purpose The purpose of this article was to validate a novel acoustic analysis of oral diadochokinesis (DDK) in assessing bulbar motor involvement in amyotrophic lateral sclerosis (ALS). Method An automated acoustic DDK analysis was developed, which filtered out the voice features and extracted the envelope of the acoustic waveform reflecting the temporal pattern of syllable repetitions during an oral DDK task (i.e., repetitions of /tɑ/ at the maximum rate on 1 breath). Cycle-to-cycle temporal variability (cTV) of envelope fluctuations and syllable repetition rate (sylRate) were derived from th

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Bhargava, Saurabh, Florian Blättler, Sepp Kollmorgen, Shih-Chii Liu, and Richard H. R. Hahnloser. "Linear Methods for Efficient and Fast Separation of Two Sources Recorded with a Single Microphone." Neural Computation 27, no. 10 (2015): 2231–59. http://dx.doi.org/10.1162/neco_a_00776.

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This letter addresses the problem of separating two speakers from a single microphone recording. Three linear methods are tested for source separation, all of which operate directly on sound spectrograms: (1) eigenmode analysis of covariance difference to identify spectro-temporal features associated with large variance for one source and small variance for the other source; (2) maximum likelihood demixing in which the mixture is modeled as the sum of two gaussian signals and maximum likelihood is used to identify the most likely sources; and (3) suppression-regression, in which autoregressive

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Durai, Mithila, Mary G. O’Keeffe, and Grant D. Searchfield. "A Review of Auditory Prediction and Its Potential Role in Tinnitus Perception." Journal of the American Academy of Audiology 29, no. 06 (2018): 533–47. http://dx.doi.org/10.3766/jaaa.17025.

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AbstractThe precise mechanisms underlying tinnitus perception and distress are still not fully understood. A recent proposition is that auditory prediction errors and related memory representations may play a role in driving tinnitus perception. It is of interest to further explore this.To obtain a comprehensive narrative synthesis of current research in relation to auditory prediction and its potential role in tinnitus perception and severity.A narrative review methodological framework was followed.The key words Prediction Auditory, Memory Prediction Auditory, Tinnitus AND Memory, Tinnitus AN

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Fong, Simon, Kun Lan, and Raymond Wong. "Classifying Human Voices by Using Hybrid SFX Time-Series Preprocessing and Ensemble Feature Selection." BioMed Research International 2013 (2013): 1–27. http://dx.doi.org/10.1155/2013/720834.

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Voice biometrics is one kind of physiological characteristics whose voice is different for each individual person. Due to this uniqueness, voice classification has found useful applications in classifying speakers’ gender, mother tongue or ethnicity (accent), emotion states, identity verification, verbal command control, and so forth. In this paper, we adopt a new preprocessing method named Statistical Feature Extraction (SFX) for extracting important features in training a classification model, based on piecewise transformation treating an audio waveform as a time-series. Using SFX we can fai

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Gottermeier, Linda, and Carol De Filippo. "Patterns of Aided Loudness Growth in Experienced Adult Listeners with Early-Onset Severe–Profound Hearing Loss." Journal of the American Academy of Audiology 29, no. 06 (2018): 457–76. http://dx.doi.org/10.3766/jaaa.16060.

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AbstractIndividuals with early-onset severe–profound bilateral hearing loss (S/PHL) manifest diverse levels of benefit and satisfaction with hearing aids (HAs), even with prescriptive HA fitting. Such fittings incorporate normal loudness values, but little is known about aided loudness outcomes in this population and how those outcomes affect benefit or satisfaction.To describe aided loudness growth and satisfaction with aided listening in experienced adult HA users with S/PHL.The Contour Test of loudness perception was administered to listeners with S/PHL in the aided sound field using broadb

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Henkin, Yael, Yifat Yaar-Soffer, Lihi Givon, and Minka Hildesheimer. "Hearing with Two Ears: Evidence for Cortical Binaural Interaction during Auditory Processing." Journal of the American Academy of Audiology 26, no. 04 (2015): 384–92. http://dx.doi.org/10.3766/jaaa.26.4.6.

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Background: Integration of information presented to the two ears has been shown to manifest in binaural interaction components (BICs) that occur along the ascending auditory pathways. In humans, BICs have been studied predominantly at the brainstem and thalamocortical levels; however, understanding of higher cortically driven mechanisms of binaural hearing is limited. Purpose: To explore whether BICs are evident in auditory event-related potentials (AERPs) during the advanced perceptual and postperceptual stages of cortical processing. Research Design: The AERPs N1, P3, and a late negative com

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Wilson, Richard H. "Amplitude (vu and rms) and Temporal (msec) Measures of Two Northwestern University Auditory Test No. 6 Recordings." Journal of the American Academy of Audiology 26, no. 04 (2015): 346–54. http://dx.doi.org/10.3766/jaaa.26.4.3.

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Background: In 1940, a cooperative effort by the radio networks and Bell Telephone produced the volume unit (vu) meter that has been the mainstay instrument for monitoring the level of speech signals in commercial broadcasting and research laboratories. With the use of computers, today the amplitude of signals can be quantified easily using the root mean square (rms) algorithm. Researchers had previously reported that amplitude estimates of sentences and running speech were 4.8 dB higher when measured with a vu meter than when calculated with rms. This study addresses the vu–rms relation as ap

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Terzopoulos, D. "Co-occurrence analysis of speech waveforms." IEEE Transactions on Acoustics, Speech, and Signal Processing 33, no. 1 (1985): 5–30. http://dx.doi.org/10.1109/tassp.1985.1164511.

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Wilson, Richard H. "Variables that Influence the Recognition Performance of Interrupted Words: Rise-Fall Shape and Temporal Location of the Interruptions." Journal of the American Academy of Audiology 25, no. 07 (2014): 688–96. http://dx.doi.org/10.3766/jaaa.25.7.8.

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Background: The abrupt transition of a signal from off to on and vice versa typically produces spectral splatter that can mask other signals that are spectrally removed from the nominal signal frequency. Both the Miller and Licklider (1950) and Cherry (1953) studies of interrupted speech and alternated speech, respectively, acknowledged the generation of extraneous noise by the rapid on and off characteristics of their unshaped signals but noted for slower interruption rates (e.g., 10 interruptions per second); the masking effects were minimal. Recent studies of interrupted speech have avoided

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