Electrocorticographic representations of segmental features in continuous speech

PubMed Central

Lotte, Fabien; Brumberg, Jonathan S.; Brunner, Peter; Gunduz, Aysegul; Ritaccio, Anthony L.; Guan, Cuntai; Schalk, Gerwin

2015-01-01

Acoustic speech output results from coordinated articulation of dozens of muscles, bones and cartilages of the vocal mechanism. While we commonly take the fluency and speed of our speech productions for granted, the neural mechanisms facilitating the requisite muscular control are not completely understood. Previous neuroimaging and electrophysiology studies of speech sensorimotor control has typically concentrated on speech sounds (i.e., phonemes, syllables and words) in isolation; sentence-length investigations have largely been used to inform coincident linguistic processing. In this study, we examined the neural representations of segmental features (place and manner of articulation, and voicing status) in the context of fluent, continuous speech production. We used recordings from the cortical surface [electrocorticography (ECoG)] to simultaneously evaluate the spatial topography and temporal dynamics of the neural correlates of speech articulation that may mediate the generation of hypothesized gestural or articulatory scores. We found that the representation of place of articulation involved broad networks of brain regions during all phases of speech production: preparation, execution and monitoring. In contrast, manner of articulation and voicing status were dominated by auditory cortical responses after speech had been initiated. These results provide a new insight into the articulatory and auditory processes underlying speech production in terms of their motor requirements and acoustic correlates. PMID:25759647

Shared acoustic codes underlie emotional communication in music and speech-Evidence from deep transfer learning.

PubMed

Coutinho, Eduardo; Schuller, Björn

2017-01-01

Music and speech exhibit striking similarities in the communication of emotions in the acoustic domain, in such a way that the communication of specific emotions is achieved, at least to a certain extent, by means of shared acoustic patterns. From an Affective Sciences points of view, determining the degree of overlap between both domains is fundamental to understand the shared mechanisms underlying such phenomenon. From a Machine learning perspective, the overlap between acoustic codes for emotional expression in music and speech opens new possibilities to enlarge the amount of data available to develop music and speech emotion recognition systems. In this article, we investigate time-continuous predictions of emotion (Arousal and Valence) in music and speech, and the Transfer Learning between these domains. We establish a comparative framework including intra- (i.e., models trained and tested on the same modality, either music or speech) and cross-domain experiments (i.e., models trained in one modality and tested on the other). In the cross-domain context, we evaluated two strategies-the direct transfer between domains, and the contribution of Transfer Learning techniques (feature-representation-transfer based on Denoising Auto Encoders) for reducing the gap in the feature space distributions. Our results demonstrate an excellent cross-domain generalisation performance with and without feature representation transfer in both directions. In the case of music, cross-domain approaches outperformed intra-domain models for Valence estimation, whereas for Speech intra-domain models achieve the best performance. This is the first demonstration of shared acoustic codes for emotional expression in music and speech in the time-continuous domain.

System And Method For Characterizing Voiced Excitations Of Speech And Acoustic Signals, Removing Acoustic Noise From Speech, And Synthesizi

DOEpatents

Burnett, Greg C.; Holzrichter, John F.; Ng, Lawrence C.

2006-04-25

The present invention is a system and method for characterizing human (or animate) speech voiced excitation functions and acoustic signals, for removing unwanted acoustic noise which often occurs when a speaker uses a microphone in common environments, and for synthesizing personalized or modified human (or other animate) speech upon command from a controller. A low power EM sensor is used to detect the motions of windpipe tissues in the glottal region of the human speech system before, during, and after voiced speech is produced by a user. From these tissue motion measurements, a voiced excitation function can be derived. Further, the excitation function provides speech production information to enhance noise removal from human speech and it enables accurate transfer functions of speech to be obtained. Previously stored excitation and transfer functions can be used for synthesizing personalized or modified human speech. Configurations of EM sensor and acoustic microphone systems are described to enhance noise cancellation and to enable multiple articulator measurements.

Acoustic Sources of Accent in Second Language Japanese Speech.

PubMed

Idemaru, Kaori; Wei, Peipei; Gubbins, Lucy

2018-05-01

This study reports an exploratory analysis of the acoustic characteristics of second language (L2) speech which give rise to the perception of a foreign accent. Japanese speech samples were collected from American English and Mandarin Chinese speakers ( n = 16 in each group) studying Japanese. The L2 participants and native speakers ( n = 10) provided speech samples modeling after six short sentences. Segmental (vowels and stops) and prosodic features (rhythm, tone, and fluency) were examined. Native Japanese listeners ( n = 10) rated the samples with regard to degrees of foreign accent. The analyses predicting accent ratings based on the acoustic measurements indicated that one of the prosodic features in particular, tone (defined as high and low patterns of pitch accent and intonation in this study), plays an important role in robustly predicting accent rating in L2 Japanese across the two first language (L1) backgrounds. These results were consistent with the prediction based on phonological and phonetic comparisons between Japanese and English, as well as Japanese and Mandarin Chinese. The results also revealed L1-specific predictors of perceived accent in Japanese. The findings of this study contribute to the growing literature that examines sources of perceived foreign accent.

Decoding Articulatory Features from fMRI Responses in Dorsal Speech Regions.

PubMed

Correia, Joao M; Jansma, Bernadette M B; Bonte, Milene

2015-11-11

The brain's circuitry for perceiving and producing speech may show a notable level of overlap that is crucial for normal development and behavior. The extent to which sensorimotor integration plays a role in speech perception remains highly controversial, however. Methodological constraints related to experimental designs and analysis methods have so far prevented the disentanglement of neural responses to acoustic versus articulatory speech features. Using a passive listening paradigm and multivariate decoding of single-trial fMRI responses to spoken syllables, we investigated brain-based generalization of articulatory features (place and manner of articulation, and voicing) beyond their acoustic (surface) form in adult human listeners. For example, we trained a classifier to discriminate place of articulation within stop syllables (e.g., /pa/ vs /ta/) and tested whether this training generalizes to fricatives (e.g., /fa/ vs /sa/). This novel approach revealed generalization of place and manner of articulation at multiple cortical levels within the dorsal auditory pathway, including auditory, sensorimotor, motor, and somatosensory regions, suggesting the representation of sensorimotor information. Additionally, generalization of voicing included the right anterior superior temporal sulcus associated with the perception of human voices as well as somatosensory regions bilaterally. Our findings highlight the close connection between brain systems for speech perception and production, and in particular, indicate the availability of articulatory codes during passive speech perception. Sensorimotor integration is central to verbal communication and provides a link between auditory signals of speech perception and motor programs of speech production. It remains highly controversial, however, to what extent the brain's speech perception system actively uses articulatory (motor), in addition to acoustic/phonetic, representations. In this study, we examine the role of

Effects and modeling of phonetic and acoustic confusions in accented speech.

PubMed

Fung, Pascale; Liu, Yi

2005-11-01

Accented speech recognition is more challenging than standard speech recognition due to the effects of phonetic and acoustic confusions. Phonetic confusion in accented speech occurs when an expected phone is pronounced as a different one, which leads to erroneous recognition. Acoustic confusion occurs when the pronounced phone is found to lie acoustically between two baseform models and can be equally recognized as either one. We propose that it is necessary to analyze and model these confusions separately in order to improve accented speech recognition without degrading standard speech recognition. Since low phonetic confusion units in accented speech do not give rise to automatic speech recognition errors, we focus on analyzing and reducing phonetic and acoustic confusability under high phonetic confusion conditions. We propose using likelihood ratio test to measure phonetic confusion, and asymmetric acoustic distance to measure acoustic confusion. Only accent-specific phonetic units with low acoustic confusion are used in an augmented pronunciation dictionary, while phonetic units with high acoustic confusion are reconstructed using decision tree merging. Experimental results show that our approach is effective and superior to methods modeling phonetic confusion or acoustic confusion alone in accented speech, with a significant 5.7% absolute WER reduction, without degrading standard speech recognition.

Study of acoustic correlates associate with emotional speech

NASA Astrophysics Data System (ADS)

Yildirim, Serdar; Lee, Sungbok; Lee, Chul Min; Bulut, Murtaza; Busso, Carlos; Kazemzadeh, Ebrahim; Narayanan, Shrikanth

2004-10-01

This study investigates the acoustic characteristics of four different emotions expressed in speech. The aim is to obtain detailed acoustic knowledge on how a speech signal is modulated by changes from neutral to a certain emotional state. Such knowledge is necessary for automatic emotion recognition and classification and emotional speech synthesis. Speech data obtained from two semi-professional actresses are analyzed and compared. Each subject produces 211 sentences with four different emotions; neutral, sad, angry, happy. We analyze changes in temporal and acoustic parameters such as magnitude and variability of segmental duration, fundamental frequency and the first three formant frequencies as a function of emotion. Acoustic differences among the emotions are also explored with mutual information computation, multidimensional scaling and acoustic likelihood comparison with normal speech. Results indicate that speech associated with anger and happiness is characterized by longer duration, shorter interword silence, higher pitch and rms energy with wider ranges. Sadness is distinguished from other emotions by lower rms energy and longer interword silence. Interestingly, the difference in formant pattern between [happiness/anger] and [neutral/sadness] are better reflected in back vowels such as /a/(/father/) than in front vowels. Detailed results on intra- and interspeaker variability will be reported.

Acoustic correlates of sexual orientation and gender-role self-concept in women's speech.

PubMed

Kachel, Sven; Simpson, Adrian P; Steffens, Melanie C

2017-06-01

Compared to studies of male speakers, relatively few studies have investigated acoustic correlates of sexual orientation in women. The present investigation focuses on shedding more light on intra-group variability in lesbians and straight women by using a fine-grained analysis of sexual orientation and collecting data on psychological characteristics (e.g., gender-role self-concept). For a large-scale women's sample (overall n = 108), recordings of spontaneous and read speech were analyzed for median fundamental frequency and acoustic vowel space features. Two studies showed no acoustic differences between lesbians and straight women, but there was evidence of acoustic differences within sexual orientation groups. Intra-group variability in median f0 was found to depend on the exclusivity of sexual orientation; F1 and F2 in /iː/ (study 1) and median f0 (study 2) were acoustic correlates of gender-role self-concept, at least for lesbians. Other psychological characteristics (e.g., sexual orientation of female friends) were also reflected in lesbians' speech. Findings suggest that acoustic features indexicalizing sexual orientation can only be successfully interpreted in combination with a fine-grained analysis of psychological characteristics.

Speech recognition: Acoustic phonetic and lexical knowledge representation

NASA Astrophysics Data System (ADS)

Zue, V. W.

1983-02-01

The purpose of this program is to develop a speech data base facility under which the acoustic characteristics of speech sounds in various contexts can be studied conveniently; investigate the phonological properties of a large lexicon of, say 10,000 words, and determine to what extent the phontactic constraints can be utilized in speech recognition; study the acoustic cues that are used to mark work boundaries; develop a test bed in the form of a large-vocabulary, IWR system to study the interactions of acoustic, phonetic and lexical knowledge; and develop a limited continuous speech recognition system with the goal of recognizing any English word from its spelling in order to assess the interactions of higher-level knowledge sources.

Intelligibility Evaluation of Pathological Speech through Multigranularity Feature Extraction and Optimization.

PubMed

Fang, Chunying; Li, Haifeng; Ma, Lin; Zhang, Mancai

2017-01-01

Pathological speech usually refers to speech distortion resulting from illness or other biological insults. The assessment of pathological speech plays an important role in assisting the experts, while automatic evaluation of speech intelligibility is difficult because it is usually nonstationary and mutational. In this paper, we carry out an independent innovation of feature extraction and reduction, and we describe a multigranularity combined feature scheme which is optimized by the hierarchical visual method. A novel method of generating feature set based on S -transform and chaotic analysis is proposed. There are BAFS (430, basic acoustics feature), local spectral characteristics MSCC (84, Mel S -transform cepstrum coefficients), and chaotic features (12). Finally, radar chart and F -score are proposed to optimize the features by the hierarchical visual fusion. The feature set could be optimized from 526 to 96 dimensions based on NKI-CCRT corpus and 104 dimensions based on SVD corpus. The experimental results denote that new features by support vector machine (SVM) have the best performance, with a recognition rate of 84.4% on NKI-CCRT corpus and 78.7% on SVD corpus. The proposed method is thus approved to be effective and reliable for pathological speech intelligibility evaluation.

Optimizing acoustical conditions for speech intelligibility in classrooms

NASA Astrophysics Data System (ADS)

Yang, Wonyoung

High speech intelligibility is imperative in classrooms where verbal communication is critical. However, the optimal acoustical conditions to achieve a high degree of speech intelligibility have previously been investigated with inconsistent results, and practical room-acoustical solutions to optimize the acoustical conditions for speech intelligibility have not been developed. This experimental study validated auralization for speech-intelligibility testing, investigated the optimal reverberation for speech intelligibility for both normal and hearing-impaired listeners using more realistic room-acoustical models, and proposed an optimal sound-control design for speech intelligibility based on the findings. The auralization technique was used to perform subjective speech-intelligibility tests. The validation study, comparing auralization results with those of real classroom speech-intelligibility tests, found that if the room to be auralized is not very absorptive or noisy, speech-intelligibility tests using auralization are valid. The speech-intelligibility tests were done in two different auralized sound fields---approximately diffuse and non-diffuse---using the Modified Rhyme Test and both normal and hearing-impaired listeners. A hybrid room-acoustical prediction program was used throughout the work, and it and a 1/8 scale-model classroom were used to evaluate the effects of ceiling barriers and reflectors. For both subject groups, in approximately diffuse sound fields, when the speech source was closer to the listener than the noise source, the optimal reverberation time was zero. When the noise source was closer to the listener than the speech source, the optimal reverberation time was 0.4 s (with another peak at 0.0 s) with relative output power levels of the speech and noise sources SNS = 5 dB, and 0.8 s with SNS = 0 dB. In non-diffuse sound fields, when the noise source was between the speaker and the listener, the optimal reverberation time was 0.6 s with

Infant-Directed Visual Prosody: Mothers’ Head Movements and Speech Acoustics

PubMed Central

Smith, Nicholas A.; Strader, Heather L.

2014-01-01

Acoustical changes in the prosody of mothers’ speech to infants are distinct and near universal. However, less is known about the visible properties mothers’ infant-directed (ID) speech, and their relation to speech acoustics. Mothers’ head movements were tracked as they interacted with their infants using ID speech, and compared to movements accompanying their adult-directed (AD) speech. Movement measures along three dimensions of head translation, and three axes of head rotation were calculated. Overall, more head movement was found for ID than AD speech, suggesting that mothers exaggerate their visual prosody in a manner analogous to the acoustical exaggerations in their speech. Regression analyses examined the relation between changing head position and changing acoustical pitch (F0) over time. Head movements and voice pitch were more strongly related in ID speech than in AD speech. When these relations were examined across time windows of different durations, stronger relations were observed for shorter time windows (< 5 sec). However, the particular form of these more local relations did not extend or generalize to longer time windows. This suggests that the multimodal correspondences in speech prosody are variable in form, and occur within limited time spans. PMID:25242907

Acoustic landmarks contain more information about the phone string than other frames for automatic speech recognition with deep neural network acoustic model

NASA Astrophysics Data System (ADS)

He, Di; Lim, Boon Pang; Yang, Xuesong; Hasegawa-Johnson, Mark; Chen, Deming

2018-06-01

Most mainstream Automatic Speech Recognition (ASR) systems consider all feature frames equally important. However, acoustic landmark theory is based on a contradictory idea, that some frames are more important than others. Acoustic landmark theory exploits quantal non-linearities in the articulatory-acoustic and acoustic-perceptual relations to define landmark times at which the speech spectrum abruptly changes or reaches an extremum; frames overlapping landmarks have been demonstrated to be sufficient for speech perception. In this work, we conduct experiments on the TIMIT corpus, with both GMM and DNN based ASR systems and find that frames containing landmarks are more informative for ASR than others. We find that altering the level of emphasis on landmarks by re-weighting acoustic likelihood tends to reduce the phone error rate (PER). Furthermore, by leveraging the landmark as a heuristic, one of our hybrid DNN frame dropping strategies maintained a PER within 0.44% of optimal when scoring less than half (45.8% to be precise) of the frames. This hybrid strategy out-performs other non-heuristic-based methods and demonstrate the potential of landmarks for reducing computation.

Acoustics of Clear Speech: Effect of Instruction

ERIC Educational Resources Information Center

Lam, Jennifer; Tjaden, Kris; Wilding, Greg

2012-01-01

Purpose: This study investigated how different instructions for eliciting clear speech affected selected acoustic measures of speech. Method: Twelve speakers were audio-recorded reading 18 different sentences from the Assessment of Intelligibility of Dysarthric Speech (Yorkston & Beukelman, 1984). Sentences were produced in habitual, clear,…

Vocal Features of Song and Speech: Insights from Schoenberg's Pierrot Lunaire.

PubMed

Merrill, Julia; Larrouy-Maestri, Pauline

2017-01-01

Similarities and differences between speech and song are often examined. However, the perceptual definition of these two types of vocalization is challenging. Indeed, the prototypical characteristics of speech or song support top-down processes, which influence listeners' perception of acoustic information. In order to examine vocal features associated with speaking and singing, we propose an innovative approach designed to facilitate bottom-up mechanisms in perceiving vocalizations by using material situated between speech and song: Speechsong. 25 participants were asked to evaluate 20 performances of a speechsong composition by Arnold Schoenberg, "Pierrot lunaire" op. 21 from 1912, evaluating 20 features of vocal-articulatory expression. Raters provided reliable judgments concerning the vocal features used by the performers and did not show strong appeal or specific expectations in reference to Schoenberg's piece. By examining the relationship between the vocal features and the impression of song or speech, the results confirm the importance of pitch (height, contour, range), but also point to the relevance of register, timbre, tension and faucal distance. Besides highlighting vocal features associated with speech and song, this study supports the relevance of the present approach of focusing on a theoretical middle category in order to better understand vocal expression in song and speech.

Vocal Features of Song and Speech: Insights from Schoenberg's Pierrot Lunaire

PubMed Central

Merrill, Julia; Larrouy-Maestri, Pauline

2017-01-01

Similarities and differences between speech and song are often examined. However, the perceptual definition of these two types of vocalization is challenging. Indeed, the prototypical characteristics of speech or song support top-down processes, which influence listeners' perception of acoustic information. In order to examine vocal features associated with speaking and singing, we propose an innovative approach designed to facilitate bottom-up mechanisms in perceiving vocalizations by using material situated between speech and song: Speechsong. 25 participants were asked to evaluate 20 performances of a speechsong composition by Arnold Schoenberg, “Pierrot lunaire” op. 21 from 1912, evaluating 20 features of vocal-articulatory expression. Raters provided reliable judgments concerning the vocal features used by the performers and did not show strong appeal or specific expectations in reference to Schoenberg's piece. By examining the relationship between the vocal features and the impression of song or speech, the results confirm the importance of pitch (height, contour, range), but also point to the relevance of register, timbre, tension and faucal distance. Besides highlighting vocal features associated with speech and song, this study supports the relevance of the present approach of focusing on a theoretical middle category in order to better understand vocal expression in song and speech. PMID:28744233

Effect of classroom acoustics on the speech intelligibility of students.

PubMed

Rabelo, Alessandra Terra Vasconcelos; Santos, Juliana Nunes; Oliveira, Rafaella Cristina; Magalhães, Max de Castro

2014-01-01

To analyze the acoustic parameters of classrooms and the relationship among equivalent sound pressure level (Leq), reverberation time (T₃₀), the Speech Transmission Index (STI), and the performance of students in speech intelligibility testing. A cross-sectional descriptive study, which analyzed the acoustic performance of 18 classrooms in 9 public schools in Belo Horizonte, Minas Gerais, Brazil, was conducted. The following acoustic parameters were measured: Leq, T₃₀, and the STI. In the schools evaluated, a speech intelligibility test was performed on 273 students, 45.4% of whom were boys, with an average age of 9.4 years. The results of the speech intelligibility test were compared to the values of the acoustic parameters with the help of Student's t-test. The Leq, T₃₀, and STI tests were conducted in empty and furnished classrooms. Children showed better results in speech intelligibility tests conducted in classrooms with less noise, a lower T₃₀, and greater STI values. The majority of classrooms did not meet the recommended regulatory standards for good acoustic performance. Acoustic parameters have a direct effect on the speech intelligibility of students. Noise contributes to a decrease in their understanding of information presented orally, which can lead to negative consequences in their education and their social integration as future professionals.

Identification of four class emotion from Indonesian spoken language using acoustic and lexical features

NASA Astrophysics Data System (ADS)

Kasyidi, Fatan; Puji Lestari, Dessi

2018-03-01

One of the important aspects in human to human communication is to understand emotion of each party. Recently, interactions between human and computer continues to develop, especially affective interaction where emotion recognition is one of its important components. This paper presents our extended works on emotion recognition of Indonesian spoken language to identify four main class of emotions: Happy, Sad, Angry, and Contentment using combination of acoustic/prosodic features and lexical features. We construct emotion speech corpus from Indonesia television talk show where the situations are as close as possible to the natural situation. After constructing the emotion speech corpus, the acoustic/prosodic and lexical features are extracted to train the emotion model. We employ some machine learning algorithms such as Support Vector Machine (SVM), Naive Bayes, and Random Forest to get the best model. The experiment result of testing data shows that the best model has an F-measure score of 0.447 by using only the acoustic/prosodic feature and F-measure score of 0.488 by using both acoustic/prosodic and lexical features to recognize four class emotion using the SVM RBF Kernel.

Acoustic richness modulates the neural networks supporting intelligible speech processing.

PubMed

Lee, Yune-Sang; Min, Nam Eun; Wingfield, Arthur; Grossman, Murray; Peelle, Jonathan E

2016-03-01

The information contained in a sensory signal plays a critical role in determining what neural processes are engaged. Here we used interleaved silent steady-state (ISSS) functional magnetic resonance imaging (fMRI) to explore how human listeners cope with different degrees of acoustic richness during auditory sentence comprehension. Twenty-six healthy young adults underwent scanning while hearing sentences that varied in acoustic richness (high vs. low spectral detail) and syntactic complexity (subject-relative vs. object-relative center-embedded clause structures). We manipulated acoustic richness by presenting the stimuli as unprocessed full-spectrum speech, or noise-vocoded with 24 channels. Importantly, although the vocoded sentences were spectrally impoverished, all sentences were highly intelligible. These manipulations allowed us to test how intelligible speech processing was affected by orthogonal linguistic and acoustic demands. Acoustically rich speech showed stronger activation than acoustically less-detailed speech in a bilateral temporoparietal network with more pronounced activity in the right hemisphere. By contrast, listening to sentences with greater syntactic complexity resulted in increased activation of a left-lateralized network including left posterior lateral temporal cortex, left inferior frontal gyrus, and left dorsolateral prefrontal cortex. Significant interactions between acoustic richness and syntactic complexity occurred in left supramarginal gyrus, right superior temporal gyrus, and right inferior frontal gyrus, indicating that the regions recruited for syntactic challenge differed as a function of acoustic properties of the speech. Our findings suggest that the neural systems involved in speech perception are finely tuned to the type of information available, and that reducing the richness of the acoustic signal dramatically alters the brain's response to spoken language, even when intelligibility is high. Copyright © 2015 Elsevier

An acoustic comparison of two women's infant- and adult-directed speech

NASA Astrophysics Data System (ADS)

Andruski, Jean; Katz-Gershon, Shiri

2003-04-01

In addition to having prosodic characteristics that are attractive to infant listeners, infant-directed (ID) speech shares certain characteristics of adult-directed (AD) clear speech, such as increased acoustic distance between vowels, that might be expected to make ID speech easier for adults to perceive in noise than AD conversational speech. However, perceptual tests of two women's ID productions by Andruski and Bessega [J. Acoust. Soc. Am. 112, 2355] showed that is not always the case. In a word identification task that compared ID speech with AD clear and conversational speech, one speaker's ID productions were less well-identified than AD clear speech, but better identified than AD conversational speech. For the second woman, ID speech was the least accurately identified of the three speech registers. For both speakers, hard words (infrequent words with many lexical neighbors) were also at an increased disadvantage relative to easy words (frequent words with few lexical neighbors) in speech registers that were less accurately perceived. This study will compare several acoustic properties of these women's productions, including pitch and formant-frequency characteristics. Results of the acoustic analyses will be examined with the original perceptual results to suggest reasons for differences in listener's accuracy in identifying these two women's ID speech in noise.

Specific acoustic models for spontaneous and dictated style in indonesian speech recognition

NASA Astrophysics Data System (ADS)

Vista, C. B.; Satriawan, C. H.; Lestari, D. P.; Widyantoro, D. H.

2018-03-01

The performance of an automatic speech recognition system is affected by differences in speech style between the data the model is originally trained upon and incoming speech to be recognized. In this paper, the usage of GMM-HMM acoustic models for specific speech styles is investigated. We develop two systems for the experiments; the first employs a speech style classifier to predict the speech style of incoming speech, either spontaneous or dictated, then decodes this speech using an acoustic model specifically trained for that speech style. The second system uses both acoustic models to recognise incoming speech and decides upon a final result by calculating a confidence score of decoding. Results show that training specific acoustic models for spontaneous and dictated speech styles confers a slight recognition advantage as compared to a baseline model trained on a mixture of spontaneous and dictated training data. In addition, the speech style classifier approach of the first system produced slightly more accurate results than the confidence scoring employed in the second system.

Fluid-acoustic interactions and their impact on pathological voiced speech

NASA Astrophysics Data System (ADS)

Erath, Byron D.; Zanartu, Matias; Peterson, Sean D.; Plesniak, Michael W.

2011-11-01

Voiced speech is produced by vibration of the vocal fold structures. Vocal fold dynamics arise from aerodynamic pressure loadings, tissue properties, and acoustic modulation of the driving pressures. Recent speech science advancements have produced a physiologically-realistic fluid flow solver (BLEAP) capable of prescribing asymmetric intraglottal flow attachment that can be easily assimilated into reduced order models of speech. The BLEAP flow solver is extended to incorporate acoustic loading and sound propagation in the vocal tract by implementing a wave reflection analog approach for sound propagation based on the governing BLEAP equations. This enhanced physiological description of the physics of voiced speech is implemented into a two-mass model of speech. The impact of fluid-acoustic interactions on vocal fold dynamics is elucidated for both normal and pathological speech through linear and nonlinear analysis techniques. Supported by NSF Grant CBET-1036280.

Major depressive disorder discrimination using vocal acoustic features.

PubMed

Taguchi, Takaya; Tachikawa, Hirokazu; Nemoto, Kiyotaka; Suzuki, Masayuki; Nagano, Toru; Tachibana, Ryuki; Nishimura, Masafumi; Arai, Tetsuaki

2018-01-01

The voice carries various information produced by vibrations of the vocal cords and the vocal tract. Though many studies have reported a relationship between vocal acoustic features and depression, including mel-frequency cepstrum coefficients (MFCCs) which applied to speech recognition, there have been few studies in which acoustic features allowed discrimination of patients with depressive disorder. Vocal acoustic features as biomarker of depression could make differential diagnosis of patients with depressive state. In order to achieve differential diagnosis of depression, in this preliminary study, we examined whether vocal acoustic features could allow discrimination between depressive patients and healthy controls. Subjects were 36 patients who met the criteria for major depressive disorder and 36 healthy controls with no current or past psychiatric disorders. Voices of reading out digits before and after verbal fluency task were recorded. Voices were analyzed using OpenSMILE. The extracted acoustic features, including MFCCs, were used for group comparison and discriminant analysis between patients and controls. The second dimension of MFCC (MFCC 2) was significantly different between groups and allowed the discrimination between patients and controls with a sensitivity of 77.8% and a specificity of 86.1%. The difference in MFCC 2 between the two groups reflected an energy difference of frequency around 2000-3000Hz. The MFCC 2 was significantly different between depressive patients and controls. This feature could be a useful biomarker to detect major depressive disorder. Sample size was relatively small. Psychotropics could have a confounding effect on voice. Copyright © 2017 Elsevier B.V. All rights reserved.

Suppressed Alpha Oscillations Predict Intelligibility of Speech and its Acoustic Details

PubMed Central

Weisz, Nathan

2012-01-01

Modulations of human alpha oscillations (8–13 Hz) accompany many cognitive processes, but their functional role in auditory perception has proven elusive: Do oscillatory dynamics of alpha reflect acoustic details of the speech signal and are they indicative of comprehension success? Acoustically presented words were degraded in acoustic envelope and spectrum in an orthogonal design, and electroencephalogram responses in the frequency domain were analyzed in 24 participants, who rated word comprehensibility after each trial. First, the alpha power suppression during and after a degraded word depended monotonically on spectral and, to a lesser extent, envelope detail. The magnitude of this alpha suppression exhibited an additional and independent influence on later comprehension ratings. Second, source localization of alpha suppression yielded superior parietal, prefrontal, as well as anterior temporal brain areas. Third, multivariate classification of the time–frequency pattern across participants showed that patterns of late posterior alpha power allowed best for above-chance classification of word intelligibility. Results suggest that both magnitude and topography of late alpha suppression in response to single words can indicate a listener's sensitivity to acoustic features and the ability to comprehend speech under adverse listening conditions. PMID:22100354

Varying acoustic-phonemic ambiguity reveals that talker normalization is obligatory in speech processing.

PubMed

Choi, Ja Young; Hu, Elly R; Perrachione, Tyler K

2018-04-01

The nondeterministic relationship between speech acoustics and abstract phonemic representations imposes a challenge for listeners to maintain perceptual constancy despite the highly variable acoustic realization of speech. Talker normalization facilitates speech processing by reducing the degrees of freedom for mapping between encountered speech and phonemic representations. While this process has been proposed to facilitate the perception of ambiguous speech sounds, it is currently unknown whether talker normalization is affected by the degree of potential ambiguity in acoustic-phonemic mapping. We explored the effects of talker normalization on speech processing in a series of speeded classification paradigms, parametrically manipulating the potential for inconsistent acoustic-phonemic relationships across talkers for both consonants and vowels. Listeners identified words with varying potential acoustic-phonemic ambiguity across talkers (e.g., beet/boat vs. boot/boat) spoken by single or mixed talkers. Auditory categorization of words was always slower when listening to mixed talkers compared to a single talker, even when there was no potential acoustic ambiguity between target sounds. Moreover, the processing cost imposed by mixed talkers was greatest when words had the most potential acoustic-phonemic overlap across talkers. Models of acoustic dissimilarity between target speech sounds did not account for the pattern of results. These results suggest (a) that talker normalization incurs the greatest processing cost when disambiguating highly confusable sounds and (b) that talker normalization appears to be an obligatory component of speech perception, taking place even when the acoustic-phonemic relationships across sounds are unambiguous.

Speech-feature discrimination in children with Asperger syndrome as determined with the multi-feature mismatch negativity paradigm.

PubMed

Kujala, T; Kuuluvainen, S; Saalasti, S; Jansson-Verkasalo, E; von Wendt, L; Lepistö, T

2010-09-01

Asperger syndrome, belonging to the autistic spectrum of disorders, involves deficits in social interaction and prosodic use of language but normal development of formal language abilities. Auditory processing involves both hyper- and hypoactive reactivity to acoustic changes. Responses composed of mismatch negativity (MMN) and obligatory components were recorded for five types of deviations in syllables (vowel, vowel duration, consonant, syllable frequency, syllable intensity) with the multi-feature paradigm from 8-12-year old children with Asperger syndrome. Children with Asperger syndrome had larger MMNs for intensity and smaller MMNs for frequency changes than typically developing children, whereas no MMN group differences were found for the other deviant stimuli. Furthermore, children with Asperger syndrome performed more poorly than controls in Comprehension of Instructions subtest of a language test battery. Cortical speech-sound discrimination is aberrant in children with Asperger syndrome. This is evident both as hypersensitive and depressed neural reactions to speech-sound changes, and is associated with features (frequency, intensity) which are relevant for prosodic processing. The multi-feature MMN paradigm, which includes variation and thereby resembles natural speech hearing circumstances, suggests abnormal pattern of speech discrimination in Asperger syndrome, including both hypo- and hypersensitive responses for speech features. 2010 International Federation of Clinical Neurophysiology. Published by Elsevier Ireland Ltd. All rights reserved.

Automated Intelligibility Assessment of Pathological Speech Using Phonological Features

NASA Astrophysics Data System (ADS)

Middag, Catherine; Martens, Jean-Pierre; Van Nuffelen, Gwen; De Bodt, Marc

2009-12-01

It is commonly acknowledged that word or phoneme intelligibility is an important criterion in the assessment of the communication efficiency of a pathological speaker. People have therefore put a lot of effort in the design of perceptual intelligibility rating tests. These tests usually have the drawback that they employ unnatural speech material (e.g., nonsense words) and that they cannot fully exclude errors due to listener bias. Therefore, there is a growing interest in the application of objective automatic speech recognition technology to automate the intelligibility assessment. Current research is headed towards the design of automated methods which can be shown to produce ratings that correspond well with those emerging from a well-designed and well-performed perceptual test. In this paper, a novel methodology that is built on previous work (Middag et al., 2008) is presented. It utilizes phonological features, automatic speech alignment based on acoustic models that were trained on normal speech, context-dependent speaker feature extraction, and intelligibility prediction based on a small model that can be trained on pathological speech samples. The experimental evaluation of the new system reveals that the root mean squared error of the discrepancies between perceived and computed intelligibilities can be as low as 8 on a scale of 0 to 100.

Acoustic properties of naturally produced clear speech at normal speaking rates

NASA Astrophysics Data System (ADS)

Krause, Jean C.; Braida, Louis D.

2004-01-01

Sentences spoken ``clearly'' are significantly more intelligible than those spoken ``conversationally'' for hearing-impaired listeners in a variety of backgrounds [Picheny et al., J. Speech Hear. Res. 28, 96-103 (1985); Uchanski et al., ibid. 39, 494-509 (1996); Payton et al., J. Acoust. Soc. Am. 95, 1581-1592 (1994)]. While producing clear speech, however, talkers often reduce their speaking rate significantly [Picheny et al., J. Speech Hear. Res. 29, 434-446 (1986); Uchanski et al., ibid. 39, 494-509 (1996)]. Yet speaking slowly is not solely responsible for the intelligibility benefit of clear speech (over conversational speech), since a recent study [Krause and Braida, J. Acoust. Soc. Am. 112, 2165-2172 (2002)] showed that talkers can produce clear speech at normal rates with training. This finding suggests that clear speech has inherent acoustic properties, independent of rate, that contribute to improved intelligibility. Identifying these acoustic properties could lead to improved signal processing schemes for hearing aids. To gain insight into these acoustical properties, conversational and clear speech produced at normal speaking rates were analyzed at three levels of detail (global, phonological, and phonetic). Although results suggest that talkers may have employed different strategies to achieve clear speech at normal rates, two global-level properties were identified that appear likely to be linked to the improvements in intelligibility provided by clear/normal speech: increased energy in the 1000-3000-Hz range of long-term spectra and increased modulation depth of low frequency modulations of the intensity envelope. Other phonological and phonetic differences associated with clear/normal speech include changes in (1) frequency of stop burst releases, (2) VOT of word-initial voiceless stop consonants, and (3) short-term vowel spectra.

Preserved Acoustic Hearing in Cochlear Implantation Improves Speech Perception

PubMed Central

Sheffield, Sterling W.; Jahn, Kelly; Gifford, René H.

2015-01-01

Background With improved surgical techniques and electrode design, an increasing number of cochlear implant (CI) recipients have preserved acoustic hearing in the implanted ear, thereby resulting in bilateral acoustic hearing. There are currently no guidelines, however, for clinicians with respect to audio-metric criteria and the recommendation of amplification in the implanted ear. The acoustic bandwidth necessary to obtain speech perception benefit from acoustic hearing in the implanted ear is unknown. Additionally, it is important to determine if, and in which listening environments, acoustic hearing in both ears provides more benefit than hearing in just one ear, even with limited residual hearing. Purpose The purposes of this study were to (1) determine whether acoustic hearing in an ear with a CI provides as much speech perception benefit as an equivalent bandwidth of acoustic hearing in the non-implanted ear, and (2) determine whether acoustic hearing in both ears provides more benefit than hearing in just one ear. Research Design A repeated-measures, within-participant design was used to compare performance across listening conditions. Study Sample Seven adults with CIs and bilateral residual acoustic hearing (hearing preservation) were recruited for the study. Data Collection and Analysis Consonant-nucleus-consonant word recognition was tested in four conditions: CI alone, CI + acoustic hearing in the nonimplanted ear, CI + acoustic hearing in the implanted ear, and CI + bilateral acoustic hearing. A series of low-pass filters were used to examine the effects of acoustic bandwidth through an insert earphone with amplification. Benefit was defined as the difference among conditions. The benefit of bilateral acoustic hearing was tested in both diffuse and single-source background noise. Results were analyzed using repeated-measures analysis of variance. Results Similar benefit was obtained for equivalent acoustic frequency bandwidth in either ear. Acoustic

Preliminary study of acoustic analysis for evaluating speech-aid oral prostheses: Characteristic dips in octave spectrum for comparison of nasality.

PubMed

Chang, Yen-Liang; Hung, Chao-Ho; Chen, Po-Yueh; Chen, Wei-Chang; Hung, Shih-Han

2015-10-01

Acoustic analysis is often used in speech evaluation but seldom for the evaluation of oral prostheses designed for reconstruction of surgical defect. This study aimed to introduce the application of acoustic analysis for patients with velopharyngeal insufficiency (VPI) due to oral surgery and rehabilitated with oral speech-aid prostheses. The pre- and postprosthetic rehabilitation acoustic features of sustained vowel sounds from two patients with VPI were analyzed and compared with the acoustic analysis software Praat. There were significant differences in the octave spectrum of sustained vowel speech sound between the pre- and postprosthetic rehabilitation. Acoustic measurements of sustained vowels for patients before and after prosthetic treatment showed no significant differences for all parameters of fundamental frequency, jitter, shimmer, noise-to-harmonics ratio, formant frequency, F1 bandwidth, and band energy difference. The decrease in objective nasality perceptions correlated very well with the decrease in dips of the spectra for the male patient with a higher speech bulb height. Acoustic analysis may be a potential technique for evaluating the functions of oral speech-aid prostheses, which eliminates dysfunctions due to the surgical defect and contributes to a high percentage of intelligible speech. Octave spectrum analysis may also be a valuable tool for detecting changes in nasality characteristics of the voice during prosthetic treatment of VPI. Copyright © 2014. Published by Elsevier B.V.

Neural Tuning to Low-Level Features of Speech throughout the Perisylvian Cortex.

PubMed

Berezutskaya, Julia; Freudenburg, Zachary V; Güçlü, Umut; van Gerven, Marcel A J; Ramsey, Nick F

2017-08-16

Despite a large body of research, we continue to lack a detailed account of how auditory processing of continuous speech unfolds in the human brain. Previous research showed the propagation of low-level acoustic features of speech from posterior superior temporal gyrus toward anterior superior temporal gyrus in the human brain (Hullett et al., 2016). In this study, we investigate what happens to these neural representations past the superior temporal gyrus and how they engage higher-level language processing areas such as inferior frontal gyrus. We used low-level sound features to model neural responses to speech outside of the primary auditory cortex. Two complementary imaging techniques were used with human participants (both males and females): electrocorticography (ECoG) and fMRI. Both imaging techniques showed tuning of the perisylvian cortex to low-level speech features. With ECoG, we found evidence of propagation of the temporal features of speech sounds along the ventral pathway of language processing in the brain toward inferior frontal gyrus. Increasingly coarse temporal features of speech spreading from posterior superior temporal cortex toward inferior frontal gyrus were associated with linguistic features such as voice onset time, duration of the formant transitions, and phoneme, syllable, and word boundaries. The present findings provide the groundwork for a comprehensive bottom-up account of speech comprehension in the human brain. SIGNIFICANCE STATEMENT We know that, during natural speech comprehension, a broad network of perisylvian cortical regions is involved in sound and language processing. Here, we investigated the tuning to low-level sound features within these regions using neural responses to a short feature film. We also looked at whether the tuning organization along these brain regions showed any parallel to the hierarchy of language structures in continuous speech. Our results show that low-level speech features propagate throughout the

Acoustic Analysis of Speech of Cochlear Implantees and Its Implications

PubMed Central

Patadia, Rajesh; Govale, Prajakta; Rangasayee, R.; Kirtane, Milind

2012-01-01

Objectives Cochlear implantees have improved speech production skills compared with those using hearing aids, as reflected in their acoustic measures. When compared to normal hearing controls, implanted children had fronted vowel space and their /s/ and /∫/ noise frequencies overlapped. Acoustic analysis of speech provides an objective index of perceived differences in speech production which can be precursory in planning therapy. The objective of this study was to compare acoustic characteristics of speech in cochlear implantees with those of normal hearing age matched peers to understand implications. Methods Group 1 consisted of 15 children with prelingual bilateral severe-profound hearing loss (age, 5-11 years; implanted between 4-10 years). Prior to an implant behind the ear, hearing aids were used; prior & post implantation subjects received at least 1 year of aural intervention. Group 2 consisted of 15 normal hearing age matched peers. Sustained productions of vowels and words with selected consonants were recorded. Using Praat software for acoustic analysis, digitized speech tokens were measured for F1, F2, and F3 of vowels; centre frequency (Hz) and energy concentration (dB) in burst; voice onset time (VOT in ms) for stops; centre frequency (Hz) of noise in /s/; rise time (ms) for affricates. A t-test was used to find significant differences between groups. Results Significant differences were found in VOT for /b/, F1 and F2 of /e/, and F3 of /u/. No significant differences were found for centre frequency of burst, energy concentration for stops, centre frequency of noise in /s/, or rise time for affricates. These findings suggest that auditory feedback provided by cochlear implants enable subjects to monitor production of speech sounds. Conclusion Acoustic analysis of speech is an essential method for discerning characteristics which have or have not been improved by cochlear implantation and thus for planning intervention. PMID:22701768

Acoustic foundations of the speech-to-song illusion.

PubMed

Tierney, Adam; Patel, Aniruddh D; Breen, Mara

2018-06-01

In the "speech-to-song illusion," certain spoken phrases are heard as highly song-like when isolated from context and repeated. This phenomenon occurs to a greater degree for some stimuli than for others, suggesting that particular cues prompt listeners to perceive a spoken phrase as song. Here we investigated the nature of these cues across four experiments. In Experiment 1, participants were asked to rate how song-like spoken phrases were after each of eight repetitions. Initial ratings were correlated with the consistency of an underlying beat and within-syllable pitch slope, while rating change was linked to beat consistency, within-syllable pitch slope, and melodic structure. In Experiment 2, the within-syllable pitch slope of the stimuli was manipulated, and this manipulation changed the extent to which participants heard certain stimuli as more musical than others. In Experiment 3, the extent to which the pitch sequences of a phrase fit a computational model of melodic structure was altered, but this manipulation did not have a significant effect on musicality ratings. In Experiment 4, the consistency of intersyllable timing was manipulated, but this manipulation did not have an effect on the change in perceived musicality after repetition. Our methods provide a new way of studying the causal role of specific acoustic features in the speech-to-song illusion via subtle acoustic manipulations of speech, and show that listeners can rapidly (and implicitly) assess the degree to which nonmusical stimuli contain musical structure. (PsycINFO Database Record (c) 2018 APA, all rights reserved).

Talker Differences in Clear and Conversational Speech: Acoustic Characteristics of Vowels

ERIC Educational Resources Information Center

Ferguson, Sarah Hargus; Kewley-Port, Diane

2007-01-01

Purpose: To determine the specific acoustic changes that underlie improved vowel intelligibility in clear speech. Method: Seven acoustic metrics were measured for conversational and clear vowels produced by 12 talkers--6 who previously were found (S. H. Ferguson, 2004) to produce a large clear speech vowel intelligibility effect for listeners with…

Speech waveform perturbation analysis: a perceptual-acoustical comparison of seven measures.

PubMed

Askenfelt, A G; Hammarberg, B

1986-03-01

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 percentage of consecutive pitch period differences with alternating signs. The acoustic measures were tested on tape recorded speech samples from 41 voice patients, before and after successful therapy. Scattergrams of acoustic waveform perturbation data versus an average of perceived deviant voice qualities, as rated by voice clinicians, are presented. The perturbation measures were compared with regard to the acoustic-perceptual correlation and their ability to discriminate between normal and pathological voice status. The standard deviation of the distribution of the relative frequency differences was suggested as the most useful acoustic measure of waveform perturbations for clinical applications.

EEG oscillations entrain their phase to high-level features of speech sound.

PubMed

Zoefel, Benedikt; VanRullen, Rufin

2016-01-01

Phase entrainment of neural oscillations, the brain's adjustment to rhythmic stimulation, is a central component in recent theories of speech comprehension: the alignment between brain oscillations and speech sound improves speech intelligibility. However, phase entrainment to everyday speech sound could also be explained by oscillations passively following the low-level periodicities (e.g., in sound amplitude and spectral content) of auditory stimulation-and not by an adjustment to the speech rhythm per se. Recently, using novel speech/noise mixture stimuli, we have shown that behavioral performance can entrain to speech sound even when high-level features (including phonetic information) are not accompanied by fluctuations in sound amplitude and spectral content. In the present study, we report that neural phase entrainment might underlie our behavioral findings. We observed phase-locking between electroencephalogram (EEG) and speech sound in response not only to original (unprocessed) speech but also to our constructed "high-level" speech/noise mixture stimuli. Phase entrainment to original speech and speech/noise sound did not differ in the degree of entrainment, but rather in the actual phase difference between EEG signal and sound. Phase entrainment was not abolished when speech/noise stimuli were presented in reverse (which disrupts semantic processing), indicating that acoustic (rather than linguistic) high-level features play a major role in the observed neural entrainment. Our results provide further evidence for phase entrainment as a potential mechanism underlying speech processing and segmentation, and for the involvement of high-level processes in the adjustment to the rhythm of speech. Copyright © 2015 Elsevier Inc. All rights reserved.

Toward a model for lexical access based on acoustic landmarks and distinctive features

NASA Astrophysics Data System (ADS)

Stevens, Kenneth N.

2002-04-01

This article describes a model in which the acoustic speech signal is processed to yield a discrete representation of the speech stream in terms of a sequence of segments, each of which is described by a set (or bundle) of binary distinctive features. These distinctive features specify the phonemic contrasts that are used in the language, such that a change in the value of a feature can potentially generate a new word. This model is a part of a more general model that derives a word sequence from this feature representation, the words being represented in a lexicon by sequences of feature bundles. The processing of the signal proceeds in three steps: (1) Detection of peaks, valleys, and discontinuities in particular frequency ranges of the signal leads to identification of acoustic landmarks. The type of landmark provides evidence for a subset of distinctive features called articulator-free features (e.g., [vowel], [consonant], [continuant]). (2) Acoustic parameters are derived from the signal near the landmarks to provide evidence for the actions of particular articulators, and acoustic cues are extracted by sampling selected attributes of these parameters in these regions. The selection of cues that are extracted depends on the type of landmark and on the environment in which it occurs. (3) The cues obtained in step (2) are combined, taking context into account, to provide estimates of ``articulator-bound'' features associated with each landmark (e.g., [lips], [high], [nasal]). These articulator-bound features, combined with the articulator-free features in (1), constitute the sequence of feature bundles that forms the output of the model. Examples of cues that are used, and justification for this selection, are given, as well as examples of the process of inferring the underlying features for a segment when there is variability in the signal due to enhancement gestures (recruited by a speaker to make a contrast more salient) or due to overlap of gestures from

Speech Perception in Complex Acoustic Environments: Developmental Effects

ERIC Educational Resources Information Center

Leibold, Lori J.

2017-01-01

Purpose: The ability to hear and understand speech in complex acoustic environments follows a prolonged time course of development. The purpose of this article is to provide a general overview of the literature describing age effects in susceptibility to auditory masking in the context of speech recognition, including a summary of findings related…

From prosodic structure to acoustic saliency: A fMRI investigation of speech rate, clarity, and emphasis

NASA Astrophysics Data System (ADS)

Golfinopoulos, Elisa

Acoustic variability in fluent speech can arise at many stages in speech production planning and execution. For example, at the phonological encoding stage, the grouping of phonemes into syllables determines which segments are coarticulated and, by consequence, segment-level acoustic variation. Likewise phonetic encoding, which determines the spatiotemporal extent of articulatory gestures, will affect the acoustic detail of segments. Functional magnetic resonance imaging (fMRI) was used to measure brain activity of fluent adult speakers in four speaking conditions: fast, normal, clear, and emphatic (or stressed) speech. These speech manner changes typically result in acoustic variations that do not change the lexical or semantic identity of productions but do affect the acoustic saliency of phonemes, syllables and/or words. Acoustic responses recorded inside the scanner were assessed quantitatively using eight acoustic measures and sentence duration was used as a covariate of non-interest in the neuroimaging analysis. Compared to normal speech, emphatic speech was characterized acoustically by a greater difference between stressed and unstressed vowels in intensity, duration, and fundamental frequency, and neurally by increased activity in right middle premotor cortex and supplementary motor area, and bilateral primary sensorimotor cortex. These findings are consistent with right-lateralized motor planning of prosodic variation in emphatic speech. Clear speech involved an increase in average vowel and sentence durations and average vowel spacing, along with increased activity in left middle premotor cortex and bilateral primary sensorimotor cortex. These findings are consistent with an increased reliance on feedforward control, resulting in hyper-articulation, under clear as compared to normal speech. Fast speech was characterized acoustically by reduced sentence duration and average vowel spacing, and neurally by increased activity in left anterior frontal

Exploration of Acoustic Features for Automatic Vowel Discrimination in Spontaneous Speech

ERIC Educational Resources Information Center

Tyson, Na'im R.

2012-01-01

In an attempt to understand what acoustic/auditory feature sets motivated transcribers towards certain labeling decisions, I built machine learning models that were capable of discriminating between canonical and non-canonical vowels excised from the Buckeye Corpus. Specifically, I wanted to model when the dictionary form and the transcribed-form…

A Hybrid Acoustic and Pronunciation Model Adaptation Approach for Non-native Speech Recognition

NASA Astrophysics Data System (ADS)

Oh, Yoo Rhee; Kim, Hong Kook

In this paper, we propose a hybrid model adaptation approach in which pronunciation and acoustic models are adapted by incorporating the pronunciation and acoustic variabilities of non-native speech in order to improve the performance of non-native automatic speech recognition (ASR). Specifically, the proposed hybrid model adaptation can be performed at either the state-tying or triphone-modeling level, depending at which acoustic model adaptation is performed. In both methods, we first analyze the pronunciation variant rules of non-native speakers and then classify each rule as either a pronunciation variant or an acoustic variant. The state-tying level hybrid method then adapts pronunciation models and acoustic models by accommodating the pronunciation variants in the pronunciation dictionary and by clustering the states of triphone acoustic models using the acoustic variants, respectively. On the other hand, the triphone-modeling level hybrid method initially adapts pronunciation models in the same way as in the state-tying level hybrid method; however, for the acoustic model adaptation, the triphone acoustic models are then re-estimated based on the adapted pronunciation models and the states of the re-estimated triphone acoustic models are clustered using the acoustic variants. From the Korean-spoken English speech recognition experiments, it is shown that ASR systems employing the state-tying and triphone-modeling level adaptation methods can relatively reduce the average word error rates (WERs) by 17.1% and 22.1% for non-native speech, respectively, when compared to a baseline ASR system.

Speech recognition: Acoustic-phonetic knowledge acquisition and representation

NASA Astrophysics Data System (ADS)

Zue, Victor W.

1988-09-01

The long-term research goal is to develop and implement speaker-independent continuous speech recognition systems. It is believed that the proper utilization of speech-specific knowledge is essential for such advanced systems. This research is thus directed toward the acquisition, quantification, and representation, of acoustic-phonetic and lexical knowledge, and the application of this knowledge to speech recognition algorithms. In addition, we are exploring new speech recognition alternatives based on artificial intelligence and connectionist techniques. We developed a statistical model for predicting the acoustic realization of stop consonants in various positions in the syllable template. A unification-based grammatical formalism was developed for incorporating this model into the lexical access algorithm. We provided an information-theoretic justification for the hierarchical structure of the syllable template. We analyzed segmented duration for vowels and fricatives in continuous speech. Based on contextual information, we developed durational models for vowels and fricatives that account for over 70 percent of the variance, using data from multiple, unknown speakers. We rigorously evaluated the ability of human spectrogram readers to identify stop consonants spoken by many talkers and in a variety of phonetic contexts. Incorporating the declarative knowledge used by the readers, we developed a knowledge-based system for stop identification. We achieved comparable system performance to that to the readers.

Clear Speech Variants: An Acoustic Study in Parkinson's Disease

ERIC Educational Resources Information Center

Lam, Jennifer; Tjaden, Kris

2016-01-01

Purpose: The authors investigated how different variants of clear speech affect segmental and suprasegmental acoustic measures of speech in speakers with Parkinson's disease and a healthy control group. Method: A total of 14 participants with Parkinson's disease and 14 control participants served as speakers. Each speaker produced 18 different…

A novel probabilistic framework for event-based speech recognition

NASA Astrophysics Data System (ADS)

Juneja, Amit; Espy-Wilson, Carol

2003-10-01

One of the reasons for unsatisfactory performance of the state-of-the-art automatic speech recognition (ASR) systems is the inferior acoustic modeling of low-level acoustic-phonetic information in the speech signal. An acoustic-phonetic approach to ASR, on the other hand, explicitly targets linguistic information in the speech signal, but such a system for continuous speech recognition (CSR) is not known to exist. A probabilistic and statistical framework for CSR based on the idea of the representation of speech sounds by bundles of binary valued articulatory phonetic features is proposed. Multiple probabilistic sequences of linguistically motivated landmarks are obtained using binary classifiers of manner phonetic features-syllabic, sonorant and continuant-and the knowledge-based acoustic parameters (APs) that are acoustic correlates of those features. The landmarks are then used for the extraction of knowledge-based APs for source and place phonetic features and their binary classification. Probabilistic landmark sequences are constrained using manner class language models for isolated or connected word recognition. The proposed method could overcome the disadvantages encountered by the early acoustic-phonetic knowledge-based systems that led the ASR community to switch to systems highly dependent on statistical pattern analysis methods and probabilistic language or grammar models.

Clear Speech Variants: An Acoustic Study in Parkinson's Disease.

PubMed

Lam, Jennifer; Tjaden, Kris

2016-08-01

The authors investigated how different variants of clear speech affect segmental and suprasegmental acoustic measures of speech in speakers with Parkinson's disease and a healthy control group. A total of 14 participants with Parkinson's disease and 14 control participants served as speakers. Each speaker produced 18 different sentences selected from the Sentence Intelligibility Test (Yorkston & Beukelman, 1996). All speakers produced stimuli in 4 speaking conditions (habitual, clear, overenunciate, and hearing impaired). Segmental acoustic measures included vowel space area and first moment (M1) coefficient difference measures for consonant pairs. Second formant slope of diphthongs and measures of vowel and fricative durations were also obtained. Suprasegmental measures included fundamental frequency, sound pressure level, and articulation rate. For the majority of adjustments, all variants of clear speech instruction differed from the habitual condition. The overenunciate condition elicited the greatest magnitude of change for segmental measures (vowel space area, vowel durations) and the slowest articulation rates. The hearing impaired condition elicited the greatest fricative durations and suprasegmental adjustments (fundamental frequency, sound pressure level). Findings have implications for a model of speech production for healthy speakers as well as for speakers with dysarthria. Findings also suggest that particular clear speech instructions may target distinct speech subsystems.

Speech intelligibility in noise using throat and acoustic microphones.

PubMed

Acker-Mills, Barbara E; Houtsma, Adrianus J M; Ahroon, William A

2006-01-01

Helicopter cockpits are very noisy and this noise must be reduced for effective communication. The standard U.S. Army aviation helmet is equipped with a noise-canceling acoustic microphone, but some ambient noise still is transmitted. Throat microphones are not sensitive to air molecule vibrations and thus, transmittal of ambient noise is reduced. It is possible that throat microphones could enhance speech communication in helicopters, but speech intelligibility with the devices must first be assessed. In the current study, speech intelligibility of signals generated by an acoustic microphone, a throat microphone, and by the combined output of the two microphones was assessed using the Modified Rhyme Test (MRT). Stimulus words were recorded in a reverberant chamber with ambient broadband noise intensity at 90 and 106 dBA. Listeners completed the MRT task in the same settings, thus simulating the typical environment of a rotary-wing aircraft. Results show that speech intelligibility is significantly worse for the throat microphone (average percent correct = 55.97) than for the acoustic microphone (average percent correct = 69.70), particularly for the higher noise level. In addition, no benefit is gained by simultaneously using both microphones. A follow-up experiment evaluated different consonants using the Diagnostic Rhyme Test and replicated the MRT results. The current results show that intelligibility using throat microphones is poorer than with the use of boom microphones in noisy and in quiet environments. Therefore, throat microphones are not recommended for use in any situation where fast and accurate speech intelligibility is essential.

Acoustic Differences between Humorous and Sincere Communicative Intentions

ERIC Educational Resources Information Center

Hoicka, Elena; Gattis, Merideth

2012-01-01

Previous studies indicate that the acoustic features of speech discriminate between positive and negative communicative intentions, such as approval and prohibition. Two studies investigated whether acoustic features of speech can discriminate between two positive communicative intentions: humour and sweet-sincerity, where sweet-sincerity involved…

Tongue-Palate Contact Pressure, Oral Air Pressure, and Acoustics of Clear Speech

ERIC Educational Resources Information Center

Searl, Jeff; Evitts, Paul M.

2013-01-01

Purpose: The authors compared articulatory contact pressure (ACP), oral air pressure (Po), and speech acoustics for conversational versus clear speech. They also assessed the relationship of these measures to listener perception. Method: Twelve adults with normal speech produced monosyllables in a phrase using conversational and clear speech.…

Emotionally conditioning the target-speech voice enhances recognition of the target speech under "cocktail-party" listening conditions.

PubMed

Lu, Lingxi; Bao, Xiaohan; Chen, Jing; Qu, Tianshu; Wu, Xihong; Li, Liang

2018-05-01

Under a noisy "cocktail-party" listening condition with multiple people talking, listeners can use various perceptual/cognitive unmasking cues to improve recognition of the target speech against informational speech-on-speech masking. One potential unmasking cue is the emotion expressed in a speech voice, by means of certain acoustical features. However, it was unclear whether emotionally conditioning a target-speech voice that has none of the typical acoustical features of emotions (i.e., an emotionally neutral voice) can be used by listeners for enhancing target-speech recognition under speech-on-speech masking conditions. In this study we examined the recognition of target speech against a two-talker speech masker both before and after the emotionally neutral target voice was paired with a loud female screaming sound that has a marked negative emotional valence. The results showed that recognition of the target speech (especially the first keyword in a target sentence) was significantly improved by emotionally conditioning the target speaker's voice. Moreover, the emotional unmasking effect was independent of the unmasking effect of the perceived spatial separation between the target speech and the masker. Also, (skin conductance) electrodermal responses became stronger after emotional learning when the target speech and masker were perceptually co-located, suggesting an increase of listening efforts when the target speech was informationally masked. These results indicate that emotionally conditioning the target speaker's voice does not change the acoustical parameters of the target-speech stimuli, but the emotionally conditioned vocal features can be used as cues for unmasking target speech.

Speech privacy and annoyance considerations in the acoustic environment of passenger cars of high-speed trains.

PubMed

Jeon, Jin Yong; Hong, Joo Young; Jang, Hyung Suk; Kim, Jae Hyeon

2015-12-01

It is necessary to consider not only annoyance of interior noises but also speech privacy to achieve acoustic comfort in a passenger car of a high-speed train because speech from other passengers can be annoying. This study aimed to explore an optimal acoustic environment to satisfy speech privacy and reduce annoyance in a passenger car. Two experiments were conducted using speech sources and compartment noise of a high speed train with varying speech-to-noise ratios (SNRA) and background noise levels (BNL). Speech intelligibility was tested in experiment I, and in experiment II, perceived speech privacy, annoyance, and acoustic comfort of combined sounds with speech and background noise were assessed. The results show that speech privacy and annoyance were significantly influenced by the SNRA. In particular, the acoustic comfort was evaluated as acceptable when the SNRA was less than -6 dB for both speech privacy and noise annoyance. In addition, annoyance increased significantly as the BNL exceeded 63 dBA, whereas the effect of the background-noise level on the speech privacy was not significant. These findings suggest that an optimal level of interior noise in a passenger car might exist between 59 and 63 dBA, taking normal speech levels into account.

Acoustic Analysis of the Voiced-Voiceless Distinction in Dutch Tracheoesophageal Speech

ERIC Educational Resources Information Center

Jongmans, Petra; Wempe, Ton G.; van Tinteren, Harm; Hilgers, Frans J. M.; Pols, Louis C. W.; van As-Brooks, Corina J.

2010-01-01

Purpose: Confusions between voiced and voiceless plosives and voiced and voiceless fricatives are common in Dutch tracheoesophageal (TE) speech. This study investigates (a) which acoustic measures are found to convey a correct voicing contrast in TE speech and (b) whether different measures are found in TE speech than in normal laryngeal (NL)…

Software for objective comparison of vocal acoustic features over weeks of audio recording: KLFromRecordingDays

NASA Astrophysics Data System (ADS)

Soderstrom, Ken; Alalawi, Ali

KLFromRecordingDays allows measurement of Kullback-Leibler (KL) distances between 2D probability distributions of vocal acoustic features. Greater KL distance measures reflect increased phonological divergence across the vocalizations compared. The software has been used to compare *.wav file recordings made by Sound Analysis Recorder 2011 of songbird vocalizations pre- and post-drug and surgical manipulations. Recordings from individual animals in *.wav format are first organized into subdirectories by recording day and then segmented into individual syllables uttered and acoustic features of these syllables using Sound Analysis Pro 2011 (SAP). KLFromRecordingDays uses syllable acoustic feature data output by SAP to a MySQL table to generate and compare "template" (typically pre-treatment) and "target" (typically post-treatment) probability distributions. These distributions are a series of virtual 2D plots of the duration of each syllable (as x-axis) to each of 13 other acoustic features measured by SAP for that syllable (as y-axes). Differences between "template" and "target" probability distributions for each acoustic feature are determined by calculating KL distance, a measure of divergence of the target 2D distribution pattern from that of the template. KL distances and the mean KL distance across all acoustic features are calculated for each recording day and output to an Excel spreadsheet. Resulting data for individual subjects may then be pooled across treatment groups and graphically summarized and used for statistical comparisons. Because SAP-generated MySQL files are accessed directly, data limits associated with spreadsheet output are avoided, and the totality of vocal output over weeks may be objectively analyzed all at once. The software has been useful for measuring drug effects on songbird vocalizations and assessing recovery from damage to regions of vocal motor cortex. It may be useful in studies employing other species, and as part of speech

Quantified acoustic-optical speech signal incongruity identifies cortical sites of audiovisual speech processing

PubMed Central

Bernstein, Lynne E.; Lu, Zhong-Lin; Jiang, Jintao

2008-01-01

A fundamental question about human perception is how the speech perceiving brain combines auditory and visual phonetic stimulus information. We assumed that perceivers learn the normal relationship between acoustic and optical signals. We hypothesized that when the normal relationship is perturbed by mismatching the acoustic and optical signals, cortical areas responsible for audiovisual stimulus integration respond as a function of the magnitude of the mismatch. To test this hypothesis, in a previous study, we developed quantitative measures of acoustic-optical speech stimulus incongruity that correlate with perceptual measures. In the current study, we presented low incongruity (LI, matched), medium incongruity (MI, moderately mismatched), and high incongruity (HI, highly mismatched) audiovisual nonsense syllable stimuli during fMRI scanning. Perceptual responses differed as a function of the incongruity level, and BOLD measures were found to vary regionally and quantitatively with perceptual and quantitative incongruity levels. Each increase in level of incongruity resulted in an increase in overall levels of cortical activity and in additional activations. However, the only cortical region that demonstrated differential sensitivity to the three stimulus incongruity levels (HI > MI > LI) was a subarea of the left supramarginal gyrus (SMG). The left SMG might support a fine-grained analysis of the relationship between audiovisual phonetic input in comparison with stored knowledge, as hypothesized here. The methods here show that quantitative manipulation of stimulus incongruity is a new and powerful tool for disclosing the system that processes audiovisual speech stimuli. PMID:18495091

The contrast between alveolar and velar stops with typical speech data: acoustic and articulatory analyses.

PubMed

Melo, Roberta Michelon; Mota, Helena Bolli; Berti, Larissa Cristina

2017-06-08

This study used acoustic and articulatory analyses to characterize the contrast between alveolar and velar stops with typical speech data, comparing the parameters (acoustic and articulatory) of adults and children with typical speech development. The sample consisted of 20 adults and 15 children with typical speech development. The analyzed corpus was organized through five repetitions of each target-word (/'kap ə/, /'tapə/, /'galo/ e /'daɾə/). These words were inserted into a carrier phrase and the participant was asked to name them spontaneously. Simultaneous audio and video data were recorded (tongue ultrasound images). The data was submitted to acoustic analyses (voice onset time; spectral peak and burst spectral moments; vowel/consonant transition and relative duration measures) and articulatory analyses (proportion of significant axes of the anterior and posterior tongue regions and description of tongue curves). Acoustic and articulatory parameters were effective to indicate the contrast between alveolar and velar stops, mainly in the adult group. Both speech analyses showed statistically significant differences between the two groups. The acoustic and articulatory parameters provided signals to characterize the phonic contrast of speech. One of the main findings in the comparison between adult and child speech was evidence of articulatory refinement/maturation even after the period of segment acquisition.

Acoustic evidence for phonologically mismatched speech errors.

PubMed

Gormley, Andrea

2015-04-01

Speech errors are generally said to accommodate to their new phonological context. This accommodation has been validated by several transcription studies. The transcription methodology is not the best choice for detecting errors at this level, however, as this type of error can be difficult to perceive. This paper presents an acoustic analysis of speech errors that uncovers non-accommodated or mismatch errors. A mismatch error is a sub-phonemic error that results in an incorrect surface phonology. This type of error could arise during the processing of phonological rules or they could be made at the motor level of implementation. The results of this work have important implications for both experimental and theoretical research. For experimentalists, it validates the tools used for error induction and the acoustic determination of errors free of the perceptual bias. For theorists, this methodology can be used to test the nature of the processes proposed in language production.

Visual Cortical Entrainment to Motion and Categorical Speech Features during Silent Lipreading

PubMed Central

O’Sullivan, Aisling E.; Crosse, Michael J.; Di Liberto, Giovanni M.; Lalor, Edmund C.

2017-01-01

Speech is a multisensory percept, comprising an auditory and visual component. While the content and processing pathways of audio speech have been well characterized, the visual component is less well understood. In this work, we expand current methodologies using system identification to introduce a framework that facilitates the study of visual speech in its natural, continuous form. Specifically, we use models based on the unheard acoustic envelope (E), the motion signal (M) and categorical visual speech features (V) to predict EEG activity during silent lipreading. Our results show that each of these models performs similarly at predicting EEG in visual regions and that respective combinations of the individual models (EV, MV, EM and EMV) provide an improved prediction of the neural activity over their constituent models. In comparing these different combinations, we find that the model incorporating all three types of features (EMV) outperforms the individual models, as well as both the EV and MV models, while it performs similarly to the EM model. Importantly, EM does not outperform EV and MV, which, considering the higher dimensionality of the V model, suggests that more data is needed to clarify this finding. Nevertheless, the performance of EMV, and comparisons of the subject performances for the three individual models, provides further evidence to suggest that visual regions are involved in both low-level processing of stimulus dynamics and categorical speech perception. This framework may prove useful for investigating modality-specific processing of visual speech under naturalistic conditions. PMID:28123363

On the Acoustics of Emotion in Audio: What Speech, Music, and Sound have in Common

PubMed Central

Weninger, Felix; Eyben, Florian; Schuller, Björn W.; Mortillaro, Marcello; Scherer, Klaus R.

2013-01-01

Without doubt, there is emotional information in almost any kind of sound received by humans every day: be it the affective state of a person transmitted by means of speech; the emotion intended by a composer while writing a musical piece, or conveyed by a musician while performing it; or the affective state connected to an acoustic event occurring in the environment, in the soundtrack of a movie, or in a radio play. In the field of affective computing, there is currently some loosely connected research concerning either of these phenomena, but a holistic computational model of affect in sound is still lacking. In turn, for tomorrow’s pervasive technical systems, including affective companions and robots, it is expected to be highly beneficial to understand the affective dimensions of “the sound that something makes,” in order to evaluate the system’s auditory environment and its own audio output. This article aims at a first step toward a holistic computational model: starting from standard acoustic feature extraction schemes in the domains of speech, music, and sound analysis, we interpret the worth of individual features across these three domains, considering four audio databases with observer annotations in the arousal and valence dimensions. In the results, we find that by selection of appropriate descriptors, cross-domain arousal, and valence regression is feasible achieving significant correlations with the observer annotations of up to 0.78 for arousal (training on sound and testing on enacted speech) and 0.60 for valence (training on enacted speech and testing on music). The high degree of cross-domain consistency in encoding the two main dimensions of affect may be attributable to the co-evolution of speech and music from multimodal affect bursts, including the integration of nature sounds for expressive effects. PMID:23750144

On the Acoustics of Emotion in Audio: What Speech, Music, and Sound have in Common.

PubMed

Weninger, Felix; Eyben, Florian; Schuller, Björn W; Mortillaro, Marcello; Scherer, Klaus R

2013-01-01

WITHOUT DOUBT, THERE IS EMOTIONAL INFORMATION IN ALMOST ANY KIND OF SOUND RECEIVED BY HUMANS EVERY DAY: be it the affective state of a person transmitted by means of speech; the emotion intended by a composer while writing a musical piece, or conveyed by a musician while performing it; or the affective state connected to an acoustic event occurring in the environment, in the soundtrack of a movie, or in a radio play. In the field of affective computing, there is currently some loosely connected research concerning either of these phenomena, but a holistic computational model of affect in sound is still lacking. In turn, for tomorrow's pervasive technical systems, including affective companions and robots, it is expected to be highly beneficial to understand the affective dimensions of "the sound that something makes," in order to evaluate the system's auditory environment and its own audio output. This article aims at a first step toward a holistic computational model: starting from standard acoustic feature extraction schemes in the domains of speech, music, and sound analysis, we interpret the worth of individual features across these three domains, considering four audio databases with observer annotations in the arousal and valence dimensions. In the results, we find that by selection of appropriate descriptors, cross-domain arousal, and valence regression is feasible achieving significant correlations with the observer annotations of up to 0.78 for arousal (training on sound and testing on enacted speech) and 0.60 for valence (training on enacted speech and testing on music). The high degree of cross-domain consistency in encoding the two main dimensions of affect may be attributable to the co-evolution of speech and music from multimodal affect bursts, including the integration of nature sounds for expressive effects.

Acoustical features of two Mayan monuments at Chichen Itza: Accident or design?

NASA Astrophysics Data System (ADS)

Lubman, David

2002-11-01

Chichen Itza dominated the early postclassic Maya world, ca. 900-1200 C.E. Two of its colossal monuments, the Great Ball Court and the temple of Kukulkan, reflect the sophisticated, hybrid culture of a Mexicanized Maya civilization. The architecture seems intended for ceremony and ritual drama. Deducing ritual practices will advance the understanding of a lost civilization, but what took place there is largely unknown. Perhaps acoustical science can add value. Unexpected and unusual acoustical features can be interpreted as intriguing clues or irrelevant accidents. Acoustical advocates believe that, when combined with an understanding of the Maya worldview, acoustical features can provide unique insights into how the Maya designed and used theater spaces. At Chichen Itza's monuments, sound reinforcement features improve rulers and priests ability to address large crowds, and Ball Court whispering galleries permit speech communication over unexpectedly large distances. Handclaps at Kukulkan stimulate chirps that mimic a revered bird (''Kukul''), thus reinforcing cultic beliefs. A ball striking playing field wall stimulates flutter echoes at the Great Ball Court; their strength and duration arguably had dramatic, mythic, and practical significance. Interpretations of the possible mythic, magic, and political significance of sound phenomena at these Maya monuments strongly suggests intentional design.

Acoustical conditions for speech communication in active elementary school classrooms

NASA Astrophysics Data System (ADS)

Sato, Hiroshi; Bradley, John

2005-04-01

Detailed acoustical measurements were made in 34 active elementary school classrooms with typical rectangular room shape in schools near Ottawa, Canada. There was an average of 21 students in classrooms. The measurements were made to obtain accurate indications of the acoustical quality of conditions for speech communication during actual teaching activities. Mean speech and noise levels were determined from the distribution of recorded sound levels and the average speech-to-noise ratio was 11 dBA. Measured mid-frequency reverberation times (RT) during the same occupied conditions varied from 0.3 to 0.6 s, and were a little less than for the unoccupied rooms. RT values were not related to noise levels. Octave band speech and noise levels, useful-to-detrimental ratios, and Speech Transmission Index values were also determined. Key results included: (1) The average vocal effort of teachers corresponded to louder than Pearsons Raised voice level; (2) teachers increase their voice level to overcome ambient noise; (3) effective speech levels can be enhanced by up to 5 dB by early reflection energy; and (4) student activity is seen to be the dominant noise source, increasing average noise levels by up to 10 dBA during teaching activities. [Work supported by CLLRnet.

Perceiving speech in context: Compensation for contextual variability during acoustic cue encoding and categorization

NASA Astrophysics Data System (ADS)

Toscano, Joseph Christopher

Several fundamental questions about speech perception concern how listeners understand spoken language despite considerable variability in speech sounds across different contexts (the problem of lack of invariance in speech). This contextual variability is caused by several factors, including differences between individual talkers' voices, variation in speaking rate, and effects of coarticulatory context. A number of models have been proposed to describe how the speech system handles differences across contexts. Critically, these models make different predictions about (1) whether contextual variability is handled at the level of acoustic cue encoding or categorization, (2) whether it is driven by feedback from category-level processes or interactions between cues, and (3) whether listeners discard fine-grained acoustic information to compensate for contextual variability. Separating the effects of cue- and category-level processing has been difficult because behavioral measures tap processes that occur well after initial cue encoding and are influenced by task demands and linguistic information. Recently, we have used the event-related brain potential (ERP) technique to examine cue encoding and online categorization. Specifically, we have looked at differences in the auditory N1 as a measure of acoustic cue encoding and the P3 as a measure of categorization. This allows us to examine multiple levels of processing during speech perception and can provide a useful tool for studying effects of contextual variability. Here, I apply this approach to determine the point in processing at which context has an effect on speech perception and to examine whether acoustic cues are encoded continuously. Several types of contextual variability (talker gender, speaking rate, and coarticulation), as well as several acoustic cues (voice onset time, formant frequencies, and bandwidths), are examined in a series of experiments. The results suggest that (1) at early stages of speech

A speech processing study using an acoustic model of a multiple-channel cochlear implant

NASA Astrophysics Data System (ADS)

Xu, Ying

1998-10-01

A cochlear implant is an electronic device designed to provide sound information for adults and children who have bilateral profound hearing loss. The task of representing speech signals as electrical stimuli is central to the design and performance of cochlear implants. Studies have shown that the current speech- processing strategies provide significant benefits to cochlear implant users. However, the evaluation and development of speech-processing strategies have been complicated by hardware limitations and large variability in user performance. To alleviate these problems, an acoustic model of a cochlear implant with the SPEAK strategy is implemented in this study, in which a set of acoustic stimuli whose psychophysical characteristics are as close as possible to those produced by a cochlear implant are presented on normal-hearing subjects. To test the effectiveness and feasibility of this acoustic model, a psychophysical experiment was conducted to match the performance of a normal-hearing listener using model- processed signals to that of a cochlear implant user. Good agreement was found between an implanted patient and an age-matched normal-hearing subject in a dynamic signal discrimination experiment, indicating that this acoustic model is a reasonably good approximation of a cochlear implant with the SPEAK strategy. The acoustic model was then used to examine the potential of the SPEAK strategy in terms of its temporal and frequency encoding of speech. It was hypothesized that better temporal and frequency encoding of speech can be accomplished by higher stimulation rates and a larger number of activated channels. Vowel and consonant recognition tests were conducted on normal-hearing subjects using speech tokens processed by the acoustic model, with different combinations of stimulation rate and number of activated channels. The results showed that vowel recognition was best at 600 pps and 8 activated channels, but further increases in stimulation rate and

Perceptual and Acoustic Reliability Estimates for the Speech Disorders Classification System (SDCS)

ERIC Educational Resources Information Center

Shriberg, Lawrence D.; Fourakis, Marios; Hall, Sheryl D.; Karlsson, Heather B.; Lohmeier, Heather L.; McSweeny, Jane L.; Potter, Nancy L.; Scheer-Cohen, Alison R.; Strand, Edythe A.; Tilkens, Christie M.; Wilson, David L.

2010-01-01

A companion paper describes three extensions to a classification system for paediatric speech sound disorders termed the Speech Disorders Classification System (SDCS). The SDCS uses perceptual and acoustic data reduction methods to obtain information on a speaker's speech, prosody, and voice. The present paper provides reliability estimates for…

Vowels in clear and conversational speech: Talker differences in acoustic characteristics and intelligibility for normal-hearing listeners

NASA Astrophysics Data System (ADS)

Hargus Ferguson, Sarah; Kewley-Port, Diane

2002-05-01

Several studies have shown that when a talker is instructed to speak as though talking to a hearing-impaired person, the resulting ``clear'' speech is significantly more intelligible than typical conversational speech. Recent work in this lab suggests that talkers vary in how much their intelligibility improves when they are instructed to speak clearly. The few studies examining acoustic characteristics of clear and conversational speech suggest that these differing clear speech effects result from different acoustic strategies on the part of individual talkers. However, only two studies to date have directly examined differences among talkers producing clear versus conversational speech, and neither included acoustic analysis. In this project, clear and conversational speech was recorded from 41 male and female talkers aged 18-45 years. A listening experiment demonstrated that for normal-hearing listeners in noise, vowel intelligibility varied widely among the 41 talkers for both speaking styles, as did the magnitude of the speaking style effect. Acoustic analyses using stimuli from a subgroup of talkers shown to have a range of speaking style effects will be used to assess specific acoustic correlates of vowel intelligibility in clear and conversational speech. [Work supported by NIHDCD-02229.

Unvoiced Speech Recognition Using Tissue-Conductive Acoustic Sensor

NASA Astrophysics Data System (ADS)

Heracleous, Panikos; Kaino, Tomomi; Saruwatari, Hiroshi; Shikano, Kiyohiro

2006-12-01

We present the use of stethoscope and silicon NAM (nonaudible murmur) microphones in automatic speech recognition. NAM microphones are special acoustic sensors, which are attached behind the talker's ear and can capture not only normal (audible) speech, but also very quietly uttered speech (nonaudible murmur). As a result, NAM microphones can be applied in automatic speech recognition systems when privacy is desired in human-machine communication. Moreover, NAM microphones show robustness against noise and they might be used in special systems (speech recognition, speech transform, etc.) for sound-impaired people. Using adaptation techniques and a small amount of training data, we achieved for a 20 k dictation task a[InlineEquation not available: see fulltext.] word accuracy for nonaudible murmur recognition in a clean environment. In this paper, we also investigate nonaudible murmur recognition in noisy environments and the effect of the Lombard reflex on nonaudible murmur recognition. We also propose three methods to integrate audible speech and nonaudible murmur recognition using a stethoscope NAM microphone with very promising results.

Improving Understanding of Emotional Speech Acoustic Content

NASA Astrophysics Data System (ADS)

Tinnemore, Anna

Children with cochlear implants show deficits in identifying emotional intent of utterances without facial or body language cues. A known limitation to cochlear implants is the inability to accurately portray the fundamental frequency contour of speech which carries the majority of information needed to identify emotional intent. Without reliable access to the fundamental frequency, other methods of identifying vocal emotion, if identifiable, could be used to guide therapies for training children with cochlear implants to better identify vocal emotion. The current study analyzed recordings of adults speaking neutral sentences with a set array of emotions in a child-directed and adult-directed manner. The goal was to identify acoustic cues that contribute to emotion identification that may be enhanced in child-directed speech, but are also present in adult-directed speech. Results of this study showed that there were significant differences in the variation of the fundamental frequency, the variation of intensity, and the rate of speech among emotions and between intended audiences.

Acoustic Evidence for Phonologically Mismatched Speech Errors

ERIC Educational Resources Information Center

Gormley, Andrea

2015-01-01

Speech errors are generally said to accommodate to their new phonological context. This accommodation has been validated by several transcription studies. The transcription methodology is not the best choice for detecting errors at this level, however, as this type of error can be difficult to perceive. This paper presents an acoustic analysis of…

The a priori SDR Estimation Techniques with Reduced Speech Distortion for Acoustic Echo and Noise Suppression

NASA Astrophysics Data System (ADS)

Thoonsaengngam, Rattapol; Tangsangiumvisai, Nisachon

This paper proposes an enhanced method for estimating the a priori Signal-to-Disturbance Ratio (SDR) to be employed in the Acoustic Echo and Noise Suppression (AENS) system for full-duplex hands-free communications. The proposed a priori SDR estimation technique is modified based upon the Two-Step Noise Reduction (TSNR) algorithm to suppress the background noise while preserving speech spectral components. In addition, a practical approach to determine accurately the Echo Spectrum Variance (ESV) is presented based upon the linear relationship assumption between the power spectrum of far-end speech and acoustic echo signals. The ESV estimation technique is then employed to alleviate the acoustic echo problem. The performance of the AENS system that employs these two proposed estimation techniques is evaluated through the Echo Attenuation (EA), Noise Attenuation (NA), and two speech distortion measures. Simulation results based upon real speech signals guarantee that our improved AENS system is able to mitigate efficiently the problem of acoustic echo and background noise, while preserving the speech quality and speech intelligibility.

Acoustic-Emergent Phonology in the Amplitude Envelope of Child-Directed Speech

PubMed Central

Leong, Victoria; Goswami, Usha

2015-01-01

When acquiring language, young children may use acoustic spectro-temporal patterns in speech to derive phonological units in spoken language (e.g., prosodic stress patterns, syllables, phonemes). Children appear to learn acoustic-phonological mappings rapidly, without direct instruction, yet the underlying developmental mechanisms remain unclear. Across different languages, a relationship between amplitude envelope sensitivity and phonological development has been found, suggesting that children may make use of amplitude modulation (AM) patterns within the envelope to develop a phonological system. Here we present the Spectral Amplitude Modulation Phase Hierarchy (S-AMPH) model, a set of algorithms for deriving the dominant AM patterns in child-directed speech (CDS). Using Principal Components Analysis, we show that rhythmic CDS contains an AM hierarchy comprising 3 core modulation timescales. These timescales correspond to key phonological units: prosodic stress (Stress AM, ~2 Hz), syllables (Syllable AM, ~5 Hz) and onset-rime units (Phoneme AM, ~20 Hz). We argue that these AM patterns could in principle be used by naïve listeners to compute acoustic-phonological mappings without lexical knowledge. We then demonstrate that the modulation statistics within this AM hierarchy indeed parse the speech signal into a primitive hierarchically-organised phonological system comprising stress feet (proto-words), syllables and onset-rime units. We apply the S-AMPH model to two other CDS corpora, one spontaneous and one deliberately-timed. The model accurately identified 72–82% (freely-read CDS) and 90–98% (rhythmically-regular CDS) stress patterns, syllables and onset-rime units. This in-principle demonstration that primitive phonology can be extracted from speech AMs is termed Acoustic-Emergent Phonology (AEP) theory. AEP theory provides a set of methods for examining how early phonological development is shaped by the temporal modulation structure of speech across

Acoustic-Emergent Phonology in the Amplitude Envelope of Child-Directed Speech.

PubMed

Leong, Victoria; Goswami, Usha

2015-01-01

When acquiring language, young children may use acoustic spectro-temporal patterns in speech to derive phonological units in spoken language (e.g., prosodic stress patterns, syllables, phonemes). Children appear to learn acoustic-phonological mappings rapidly, without direct instruction, yet the underlying developmental mechanisms remain unclear. Across different languages, a relationship between amplitude envelope sensitivity and phonological development has been found, suggesting that children may make use of amplitude modulation (AM) patterns within the envelope to develop a phonological system. Here we present the Spectral Amplitude Modulation Phase Hierarchy (S-AMPH) model, a set of algorithms for deriving the dominant AM patterns in child-directed speech (CDS). Using Principal Components Analysis, we show that rhythmic CDS contains an AM hierarchy comprising 3 core modulation timescales. These timescales correspond to key phonological units: prosodic stress (Stress AM, ~2 Hz), syllables (Syllable AM, ~5 Hz) and onset-rime units (Phoneme AM, ~20 Hz). We argue that these AM patterns could in principle be used by naïve listeners to compute acoustic-phonological mappings without lexical knowledge. We then demonstrate that the modulation statistics within this AM hierarchy indeed parse the speech signal into a primitive hierarchically-organised phonological system comprising stress feet (proto-words), syllables and onset-rime units. We apply the S-AMPH model to two other CDS corpora, one spontaneous and one deliberately-timed. The model accurately identified 72-82% (freely-read CDS) and 90-98% (rhythmically-regular CDS) stress patterns, syllables and onset-rime units. This in-principle demonstration that primitive phonology can be extracted from speech AMs is termed Acoustic-Emergent Phonology (AEP) theory. AEP theory provides a set of methods for examining how early phonological development is shaped by the temporal modulation structure of speech across

Noise-robust speech recognition through auditory feature detection and spike sequence decoding.

PubMed

Schafer, Phillip B; Jin, Dezhe Z

2014-03-01

Speech recognition in noisy conditions is a major challenge for computer systems, but the human brain performs it routinely and accurately. Automatic speech recognition (ASR) systems that are inspired by neuroscience can potentially bridge the performance gap between humans and machines. We present a system for noise-robust isolated word recognition that works by decoding sequences of spikes from a population of simulated auditory feature-detecting neurons. Each neuron is trained to respond selectively to a brief spectrotemporal pattern, or feature, drawn from the simulated auditory nerve response to speech. The neural population conveys the time-dependent structure of a sound by its sequence of spikes. We compare two methods for decoding the spike sequences--one using a hidden Markov model-based recognizer, the other using a novel template-based recognition scheme. In the latter case, words are recognized by comparing their spike sequences to template sequences obtained from clean training data, using a similarity measure based on the length of the longest common sub-sequence. Using isolated spoken digits from the AURORA-2 database, we show that our combined system outperforms a state-of-the-art robust speech recognizer at low signal-to-noise ratios. Both the spike-based encoding scheme and the template-based decoding offer gains in noise robustness over traditional speech recognition methods. Our system highlights potential advantages of spike-based acoustic coding and provides a biologically motivated framework for robust ASR development.

Speech Intelligibility Advantages using an Acoustic Beamformer Display

NASA Technical Reports Server (NTRS)

Begault, Durand R.; Sunder, Kaushik; Godfroy, Martine; Otto, Peter

2015-01-01

A speech intelligibility test conforming to the Modified Rhyme Test of ANSI S3.2 "Method for Measuring the Intelligibility of Speech Over Communication Systems" was conducted using a prototype 12-channel acoustic beamformer system. The target speech material (signal) was identified against speech babble (noise), with calculated signal-noise ratios of 0, 5 and 10 dB. The signal was delivered at a fixed beam orientation of 135 deg (re 90 deg as the frontal direction of the array) and the noise at 135 deg (co-located) and 0 deg (separated). A significant improvement in intelligibility from 57% to 73% was found for spatial separation for the same signal-noise ratio (0 dB). Significant effects for improved intelligibility due to spatial separation were also found for higher signal-noise ratios (5 and 10 dB).

Psychoacoustic cues to emotion in speech prosody and music.

PubMed

Coutinho, Eduardo; Dibben, Nicola

2013-01-01

There is strong evidence of shared acoustic profiles common to the expression of emotions in music and speech, yet relatively limited understanding of the specific psychoacoustic features involved. This study combined a controlled experiment and computational modelling to investigate the perceptual codes associated with the expression of emotion in the acoustic domain. The empirical stage of the study provided continuous human ratings of emotions perceived in excerpts of film music and natural speech samples. The computational stage created a computer model that retrieves the relevant information from the acoustic stimuli and makes predictions about the emotional expressiveness of speech and music close to the responses of human subjects. We show that a significant part of the listeners' second-by-second reported emotions to music and speech prosody can be predicted from a set of seven psychoacoustic features: loudness, tempo/speech rate, melody/prosody contour, spectral centroid, spectral flux, sharpness, and roughness. The implications of these results are discussed in the context of cross-modal similarities in the communication of emotion in the acoustic domain.

DARPA TIMIT acoustic-phonetic continous speech corpus CD-ROM. NIST speech disc 1-1.1

NASA Astrophysics Data System (ADS)

Garofolo, J. S.; Lamel, L. F.; Fisher, W. M.; Fiscus, J. G.; Pallett, D. S.

1993-02-01

The Texas Instruments/Massachusetts Institute of Technology (TIMIT) corpus of read speech has been designed to provide speech data for the acquisition of acoustic-phonetic knowledge and for the development and evaluation of automatic speech recognition systems. TIMIT contains speech from 630 speakers representing 8 major dialect divisions of American English, each speaking 10 phonetically-rich sentences. The TIMIT corpus includes time-aligned orthographic, phonetic, and word transcriptions, as well as speech waveform data for each spoken sentence. The release of TIMIT contains several improvements over the Prototype CD-ROM released in December, 1988: (1) full 630-speaker corpus, (2) checked and corrected transcriptions, (3) word-alignment transcriptions, (4) NIST SPHERE-headered waveform files and header manipulation software, (5) phonemic dictionary, (6) new test and training subsets balanced for dialectal and phonetic coverage, and (7) more extensive documentation.

a Comparative Analysis of Fluent and Cerebral Palsied Speech.

NASA Astrophysics Data System (ADS)

van Doorn, Janis Lee

Several features of the acoustic waveforms of fluent and cerebral palsied speech were compared, using six fluent and seven cerebral palsied subjects, with a major emphasis being placed on an investigation of the trajectories of the first three formants (vocal tract resonances). To provide an overall picture which included other acoustic features, fundamental frequency, intensity, speech timing (speech rate and syllable duration), and prevocalization (vocalization prior to initial stop consonants found in cerebral palsied speech) were also investigated. Measurements were made using repetitions of a test sentence which was chosen because it required large excursions of the speech articulators (lips, tongue and jaw), so that differences in the formant trajectories for the fluent and cerebral palsied speakers would be emphasized. The acoustic features were all extracted from the digitized speech waveform (10 kHz sampling rate): the fundamental frequency contours were derived manually, the intensity contours were measured using the signal covariance, speech rate and syllable durations were measured manually, as were the prevocalization durations, while the formant trajectories were derived from short time spectra which were calculated for each 10 ms of speech using linear prediction analysis. Differences which were found in the acoustic features can be summarized as follows. For cerebral palsied speakers, the fundamental frequency contours generally showed inappropriate exaggerated fluctuations, as did some of the intensity contours; the mean fundamental frequencies were either higher or the same as for the fluent subjects; speech rates were reduced, and syllable durations were longer; prevocalization was consistently present at the beginning of the test sentence; formant trajectories were found to have overall reduced frequency ranges, and to contain anomalous transitional features, but it is noteworthy that for any one cerebral palsied subject, the inappropriate

Acoustic landmarks drive delta-theta oscillations to enable speech comprehension by facilitating perceptual parsing

PubMed Central

Doelling, Keith; Arnal, Luc; Ghitza, Oded; Poeppel, David

2013-01-01

A growing body of research suggests that intrinsic neuronal slow (< 10 Hz) oscillations in auditory cortex appear to track incoming speech and other spectro-temporally complex auditory signals. Within this framework, several recent studies have identified critical-band temporal envelopes as the specific acoustic feature being reflected by the phase of these oscillations. However, how this alignment between speech acoustics and neural oscillations might underpin intelligibility is unclear. Here we test the hypothesis that the ‘sharpness’ of temporal fluctuations in the critical band envelope acts as a temporal cue to speech syllabic rate, driving delta-theta rhythms to track the stimulus and facilitate intelligibility. We interpret our findings as evidence that sharp events in the stimulus cause cortical rhythms to re-align and parse the stimulus into syllable-sized chunks for further decoding. Using magnetoencephalographic recordings, we show that by removing temporal fluctuations that occur at the syllabic rate, envelope-tracking activity is reduced. By artificially reinstating these temporal fluctuations, envelope-tracking activity is regained. These changes in tracking correlate with intelligibility of the stimulus. Together, the results suggest that the sharpness of fluctuations in the stimulus, as reflected in the cochlear output, drive oscillatory activity to track and entrain to the stimulus, at its syllabic rate. This process likely facilitates parsing of the stimulus into meaningful chunks appropriate for subsequent decoding, enhancing perception and intelligibility. PMID:23791839

Monaural room acoustic parameters from music and speech.

PubMed

Kendrick, Paul; Cox, Trevor J; Li, Francis F; Zhang, Yonggang; Chambers, Jonathon A

2008-07-01

This paper compares two methods for extracting room acoustic parameters from reverberated speech and music. An approach which uses statistical machine learning, previously developed for speech, is extended to work with music. For speech, reverberation time estimations are within a perceptual difference limen of the true value. For music, virtually all early decay time estimations are within a difference limen of the true value. The estimation accuracy is not good enough in other cases due to differences between the simulated data set used to develop the empirical model and real rooms. The second method carries out a maximum likelihood estimation on decay phases at the end of notes or speech utterances. This paper extends the method to estimate parameters relating to the balance of early and late energies in the impulse response. For reverberation time and speech, the method provides estimations which are within the perceptual difference limen of the true value. For other parameters such as clarity, the estimations are not sufficiently accurate due to the natural reverberance of the excitation signals. Speech is a better test signal than music because of the greater periods of silence in the signal, although music is needed for low frequency measurement.

Acoustic diagnosis of pulmonary hypertension: automated speech- recognition-inspired classification algorithm outperforms physicians

NASA Astrophysics Data System (ADS)

Kaddoura, Tarek; Vadlamudi, Karunakar; Kumar, Shine; Bobhate, Prashant; Guo, Long; Jain, Shreepal; Elgendi, Mohamed; Coe, James Y.; Kim, Daniel; Taylor, Dylan; Tymchak, Wayne; Schuurmans, Dale; Zemp, Roger J.; Adatia, Ian

2016-09-01

We hypothesized that an automated speech- recognition-inspired classification algorithm could differentiate between the heart sounds in subjects with and without pulmonary hypertension (PH) and outperform physicians. Heart sounds, electrocardiograms, and mean pulmonary artery pressures (mPAp) were recorded simultaneously. Heart sound recordings were digitized to train and test speech-recognition-inspired classification algorithms. We used mel-frequency cepstral coefficients to extract features from the heart sounds. Gaussian-mixture models classified the features as PH (mPAp ≥ 25 mmHg) or normal (mPAp < 25 mmHg). Physicians blinded to patient data listened to the same heart sound recordings and attempted a diagnosis. We studied 164 subjects: 86 with mPAp ≥ 25 mmHg (mPAp 41 ± 12 mmHg) and 78 with mPAp < 25 mmHg (mPAp 17 ± 5 mmHg) (p  < 0.005). The correct diagnostic rate of the automated speech-recognition-inspired algorithm was 74% compared to 56% by physicians (p = 0.005). The false positive rate for the algorithm was 34% versus 50% (p = 0.04) for clinicians. The false negative rate for the algorithm was 23% and 68% (p = 0.0002) for physicians. We developed an automated speech-recognition-inspired classification algorithm for the acoustic diagnosis of PH that outperforms physicians that could be used to screen for PH and encourage earlier specialist referral.

An acoustic feature-based similarity scoring system for speech rehabilitation assistance.

PubMed

Syauqy, Dahnial; Wu, Chao-Min; Setyawati, Onny

2016-08-01

The purpose of this study is to develop a tool to assist speech therapy and rehabilitation, which focused on automatic scoring based on the comparison of the patient's speech with another normal speech on several aspects including pitch, vowel, voiced-unvoiced segments, strident fricative and sound intensity. The pitch estimation employed the use of cepstrum-based algorithm for its robustness; the vowel classification used multilayer perceptron (MLP) to classify vowel from pitch and formants; and the strident fricative detection was based on the major peak spectral intensity, location and the pitch existence in the segment. In order to evaluate the performance of the system, this study analyzed eight patient's speech recordings (four males, four females; 4-58-years-old), which had been recorded in previous study in cooperation with Taipei Veterans General Hospital and Taoyuan General Hospital. The experiment result on pitch algorithm showed that the cepstrum method had 5.3% of gross pitch error from a total of 2086 frames. On the vowel classification algorithm, MLP method provided 93% accuracy (men), 87% (women) and 84% (children). In total, the overall results showed that 156 tool's grading results (81%) were consistent compared to 192 audio and visual observations done by four experienced respondents. Implication for Rehabilitation Difficulties in communication may limit the ability of a person to transfer and exchange information. The fact that speech is one of the primary means of communication has encouraged the needs of speech diagnosis and rehabilitation. The advances of technology in computer-assisted speech therapy (CAST) improve the quality, time efficiency of the diagnosis and treatment of the disorders. The present study attempted to develop tool to assist speech therapy and rehabilitation, which provided simple interface to let the assessment be done even by the patient himself without the need of particular knowledge of speech processing while at the

Classifying acoustic signals into phoneme categories: average and dyslexic readers make use of complex dynamical patterns and multifractal scaling properties of the speech signal

PubMed Central

2015-01-01

Several competing aetiologies of developmental dyslexia suggest that the problems with acquiring literacy skills are causally entailed by low-level auditory and/or speech perception processes. The purpose of this study is to evaluate the diverging claims about the specific deficient peceptual processes under conditions of strong inference. Theoretically relevant acoustic features were extracted from a set of artificial speech stimuli that lie on a /bAk/-/dAk/ continuum. The features were tested on their ability to enable a simple classifier (Quadratic Discriminant Analysis) to reproduce the observed classification performance of average and dyslexic readers in a speech perception experiment. The ‘classical’ features examined were based on component process accounts of developmental dyslexia such as the supposed deficit in Envelope Rise Time detection and the deficit in the detection of rapid changes in the distribution of energy in the frequency spectrum (formant transitions). Studies examining these temporal processing deficit hypotheses do not employ measures that quantify the temporal dynamics of stimuli. It is shown that measures based on quantification of the dynamics of complex, interaction-dominant systems (Recurrence Quantification Analysis and the multifractal spectrum) enable QDA to classify the stimuli almost identically as observed in dyslexic and average reading participants. It seems unlikely that participants used any of the features that are traditionally associated with accounts of (impaired) speech perception. The nature of the variables quantifying the temporal dynamics of the speech stimuli imply that the classification of speech stimuli cannot be regarded as a linear aggregate of component processes that each parse the acoustic signal independent of one another, as is assumed by the ‘classical’ aetiologies of developmental dyslexia. It is suggested that the results imply that the differences in speech perception performance between

Intensity Accents in French 2 Year Olds' Speech.

ERIC Educational Resources Information Center

Allen, George D.

The acoustic features and functions of accentuation in French are discussed, and features of accentuation in the speech of French 2-year-olds are explored. The four major acoustic features used to signal accentual distinctions are fundamental frequency of voicing, duration of segments and syllables, intensity of segments and syllables, and…

The Interaction of Temporal and Spectral Acoustic Information with Word Predictability on Speech Intelligibility

NASA Astrophysics Data System (ADS)

Shahsavarani, Somayeh Bahar

High-level, top-down information such as linguistic knowledge is a salient cortical resource that influences speech perception under most listening conditions. But, are all listeners able to exploit these resources for speech facilitation to the same extent? It was found that children with cochlear implants showed different patterns of benefit from contextual information in speech perception compared with their normal-haring peers. Previous studies have discussed the role of non-acoustic factors such as linguistic and cognitive capabilities to account for this discrepancy. Given the fact that the amount of acoustic information encoded and processed by auditory nerves of listeners with cochlear implants differs from normal-hearing listeners and even varies across individuals with cochlear implants, it is important to study the interaction of specific acoustic properties of the speech signal with contextual cues. This relationship has been mostly neglected in previous research. In this dissertation, we aimed to explore how different acoustic dimensions interact to affect listeners' abilities to combine top-down information with bottom-up information in speech perception beyond the known effects of linguistic and cognitive capacities shown previously. Specifically, the present study investigated whether there were any distinct context effects based on the resolution of spectral versus slowly-varying temporal information in perception of spectrally impoverished speech. To that end, two experiments were conducted. In both experiments, a noise-vocoded technique was adopted to generate spectrally-degraded speech to approximate acoustic cues delivered to listeners with cochlear implants. The frequency resolution was manipulated by varying the number of frequency channels. The temporal resolution was manipulated by low-pass filtering of amplitude envelope with varying low-pass cutoff frequencies. The stimuli were presented to normal-hearing native speakers of American

Postlingual deaf speech and the role of audition in speech production: comments on Waldstein's paper [R.S. Waldstein, J. Acoust. Soc. Am. 88, 2099-2114 (1990)].

PubMed

Sapir, S; Canter, G J

1991-09-01

Using acoustic analysis techniques, Waldstein [J. Acoust. Soc. Am. 88, 2099-2114 (1990] reported abnormal speech findings in postlingual deaf speakers. She interpreted her findings to suggest that auditory feedback is important in motor speech control. However, it is argued here that Waldstein's interpretation may be unwarranted without addressing the possibility of neurologic deficits (e.g., dysarthria) as confounding (or even primary) causes of the abnormal speech in her subjects.

A physiologically-inspired model reproducing the speech intelligibility benefit in cochlear implant listeners with residual acoustic hearing.

PubMed

Zamaninezhad, Ladan; Hohmann, Volker; Büchner, Andreas; Schädler, Marc René; Jürgens, Tim

2017-02-01

This study introduces a speech intelligibility model for cochlear implant users with ipsilateral preserved acoustic hearing that aims at simulating the observed speech-in-noise intelligibility benefit when receiving simultaneous electric and acoustic stimulation (EA-benefit). The model simulates the auditory nerve spiking in response to electric and/or acoustic stimulation. The temporally and spatially integrated spiking patterns were used as the final internal representation of noisy speech. Speech reception thresholds (SRTs) in stationary noise were predicted for a sentence test using an automatic speech recognition framework. The model was employed to systematically investigate the effect of three physiologically relevant model factors on simulated SRTs: (1) the spatial spread of the electric field which co-varies with the number of electrically stimulated auditory nerves, (2) the "internal" noise simulating the deprivation of auditory system, and (3) the upper bound frequency limit of acoustic hearing. The model results show that the simulated SRTs increase monotonically with increasing spatial spread for fixed internal noise, and also increase with increasing the internal noise strength for a fixed spatial spread. The predicted EA-benefit does not follow such a systematic trend and depends on the specific combination of the model parameters. Beyond 300 Hz, the upper bound limit for preserved acoustic hearing is less influential on speech intelligibility of EA-listeners in stationary noise. The proposed model-predicted EA-benefits are within the range of EA-benefits shown by 18 out of 21 actual cochlear implant listeners with preserved acoustic hearing. Copyright © 2016 Elsevier B.V. All rights reserved.

Automatic detection of obstructive sleep apnea using speech signals.

PubMed

Goldshtein, Evgenia; Tarasiuk, Ariel; Zigel, Yaniv

2011-05-01

Obstructive sleep apnea (OSA) is a common disorder associated with anatomical abnormalities of the upper airways that affects 5% of the population. Acoustic parameters may be influenced by the vocal tract structure and soft tissue properties. We hypothesize that speech signal properties of OSA patients will be different than those of control subjects not having OSA. Using speech signal processing techniques, we explored acoustic speech features of 93 subjects who were recorded using a text-dependent speech protocol and a digital audio recorder immediately prior to polysomnography study. Following analysis of the study, subjects were divided into OSA (n=67) and non-OSA (n=26) groups. A Gaussian mixture model-based system was developed to model and classify between the groups; discriminative features such as vocal tract length and linear prediction coefficients were selected using feature selection technique. Specificity and sensitivity of 83% and 79% were achieved for the male OSA and 86% and 84% for the female OSA patients, respectively. We conclude that acoustic features from speech signals during wakefulness can detect OSA patients with good specificity and sensitivity. Such a system can be used as a basis for future development of a tool for OSA screening. © 2011 IEEE

Effect of Reflective Practice on Student Recall of Acoustics for Speech Science

ERIC Educational Resources Information Center

Walden, Patrick R.; Bell-Berti, Fredericka

2013-01-01

Researchers have developed models of learning through experience; however, these models are rarely named as a conceptual frame for educational research in the sciences. This study examined the effect of reflective learning responses on student recall of speech acoustics concepts. Two groups of undergraduate students enrolled in a speech science…

[Perception of emotional intonation of noisy speech signal with different acoustic parameters by adults of different age and gender].

PubMed

Dmitrieva, E S; Gel'man, V Ia

2011-01-01

The listener-distinctive features of recognition of different emotional intonations (positive, negative and neutral) of male and female speakers in the presence or absence of background noise were studied in 49 adults aged 20-79 years. In all the listeners noise produced the most pronounced decrease in recognition accuracy for positive emotional intonation ("joy") as compared to other intonations, whereas it did not influence the recognition accuracy of "anger" in 65-79-year-old listeners. The higher emotion recognition rates of a noisy signal were observed for speech emotional intonations expressed by female speakers. Acoustic characteristics of noisy and clear speech signals underlying perception of speech emotional prosody were found for adult listeners of different age and gender.

The Effectiveness of Clear Speech as a Masker

ERIC Educational Resources Information Center

Calandruccio, Lauren; Van Engen, Kristin; Dhar, Sumitrajit; Bradlow, Ann R.

2010-01-01

Purpose: It is established that speaking clearly is an effective means of enhancing intelligibility. Because any signal-processing scheme modeled after known acoustic-phonetic features of clear speech will likely affect both target and competing speech, it is important to understand how speech recognition is affected when a competing speech signal…

Examining Acoustic and Kinematic Measures of Articulatory Working Space: Effects of Speech Intensity.

PubMed

Whitfield, Jason A; Dromey, Christopher; Palmer, Panika

2018-05-17

The purpose of this study was to examine the effect of speech intensity on acoustic and kinematic vowel space measures and conduct a preliminary examination of the relationship between kinematic and acoustic vowel space metrics calculated from continuously sampled lingual marker and formant traces. Young adult speakers produced 3 repetitions of 2 different sentences at 3 different loudness levels. Lingual kinematic and acoustic signals were collected and analyzed. Acoustic and kinematic variants of several vowel space metrics were calculated from the formant frequencies and the position of 2 lingual markers. Traditional metrics included triangular vowel space area and the vowel articulation index. Acoustic and kinematic variants of sentence-level metrics based on the articulatory-acoustic vowel space and the vowel space hull area were also calculated. Both acoustic and kinematic variants of the sentence-level metrics significantly increased with an increase in loudness, whereas no statistically significant differences in traditional vowel-point metrics were observed for either the kinematic or acoustic variants across the 3 loudness conditions. In addition, moderate-to-strong relationships between the acoustic and kinematic variants of the sentence-level vowel space metrics were observed for the majority of participants. These data suggest that both kinematic and acoustic vowel space metrics that reflect the dynamic contributions of both consonant and vowel segments are sensitive to within-speaker changes in articulation associated with manipulations of speech intensity.

Language Comprehension in Language-Learning Impaired Children Improved with Acoustically Modified Speech

NASA Astrophysics Data System (ADS)

Tallal, Paula; Miller, Steve L.; Bedi, Gail; Byma, Gary; Wang, Xiaoqin; Nagarajan, Srikantan S.; Schreiner, Christoph; Jenkins, William M.; Merzenich, Michael M.

1996-01-01

A speech processing algorithm was developed to create more salient versions of the rapidly changing elements in the acoustic waveform of speech that have been shown to be deficiently processed by language-learning impaired (LLI) children. LLI children received extensive daily training, over a 4-week period, with listening exercises in which all speech was translated into this synthetic form. They also received daily training with computer "games" designed to adaptively drive improvements in temporal processing thresholds. Significant improvements in speech discrimination and language comprehension abilities were demonstrated in two independent groups of LLI children.

A magnetic resonance imaging study on the articulatory and acoustic speech parameters of Malay vowels

PubMed Central

2014-01-01

The phonetic properties of six Malay vowels are investigated using magnetic resonance imaging (MRI) to visualize the vocal tract in order to obtain dynamic articulatory parameters during speech production. To resolve image blurring due to the tongue movement during the scanning process, a method based on active contour extraction is used to track tongue contours. The proposed method efficiently tracks tongue contours despite the partial blurring of MRI images. Consequently, the articulatory parameters that are effectively measured as tongue movement is observed, and the specific shape of the tongue and its position for all six uttered Malay vowels are determined. Speech rehabilitation procedure demands some kind of visual perceivable prototype of speech articulation. To investigate the validity of the measured articulatory parameters based on acoustic theory of speech production, an acoustic analysis based on the uttered vowels by subjects has been performed. As the acoustic speech and articulatory parameters of uttered speech were examined, a correlation between formant frequencies and articulatory parameters was observed. The experiments reported a positive correlation between the constriction location of the tongue body and the first formant frequency, as well as a negative correlation between the constriction location of the tongue tip and the second formant frequency. The results demonstrate that the proposed method is an effective tool for the dynamic study of speech production. PMID:25060583

A magnetic resonance imaging study on the articulatory and acoustic speech parameters of Malay vowels.

PubMed

Zourmand, Alireza; Mirhassani, Seyed Mostafa; Ting, Hua-Nong; Bux, Shaik Ismail; Ng, Kwan Hoong; Bilgen, Mehmet; Jalaludin, Mohd Amin

2014-07-25

The phonetic properties of six Malay vowels are investigated using magnetic resonance imaging (MRI) to visualize the vocal tract in order to obtain dynamic articulatory parameters during speech production. To resolve image blurring due to the tongue movement during the scanning process, a method based on active contour extraction is used to track tongue contours. The proposed method efficiently tracks tongue contours despite the partial blurring of MRI images. Consequently, the articulatory parameters that are effectively measured as tongue movement is observed, and the specific shape of the tongue and its position for all six uttered Malay vowels are determined.Speech rehabilitation procedure demands some kind of visual perceivable prototype of speech articulation. To investigate the validity of the measured articulatory parameters based on acoustic theory of speech production, an acoustic analysis based on the uttered vowels by subjects has been performed. As the acoustic speech and articulatory parameters of uttered speech were examined, a correlation between formant frequencies and articulatory parameters was observed. The experiments reported a positive correlation between the constriction location of the tongue body and the first formant frequency, as well as a negative correlation between the constriction location of the tongue tip and the second formant frequency. The results demonstrate that the proposed method is an effective tool for the dynamic study of speech production.

Articulatory-acoustic vowel space: application to clear speech in individuals with Parkinson's disease.

PubMed

Whitfield, Jason A; Goberman, Alexander M

2014-01-01

Individuals with Parkinson disease (PD) often exhibit decreased range of movement secondary to the disease process, which has been shown to affect articulatory movements. A number of investigations have failed to find statistically significant differences between control and disordered groups, and between speaking conditions, using traditional vowel space area measures. The purpose of the current investigation was to evaluate both between-group (PD versus control) and within-group (habitual versus clear) differences in articulatory function using a novel vowel space measure, the articulatory-acoustic vowel space (AAVS). The novel AAVS is calculated from continuously sampled formant trajectories of connected speech. In the current study, habitual and clear speech samples from twelve individuals with PD along with habitual control speech samples from ten neurologically healthy adults were collected and acoustically analyzed. In addition, a group of listeners completed perceptual rating of speech clarity for all samples. Individuals with PD were perceived to exhibit decreased speech clarity compared to controls. Similarly, the novel AAVS measure was significantly lower in individuals with PD. In addition, the AAVS measure significantly tracked changes between the habitual and clear conditions that were confirmed by perceptual ratings. In the current study, the novel AAVS measure is shown to be sensitive to disease-related group differences and within-person changes in articulatory function of individuals with PD. Additionally, these data confirm that individuals with PD can modulate the speech motor system to increase articulatory range of motion and speech clarity when given a simple prompt. The reader will be able to (i) describe articulatory behavior observed in the speech of individuals with Parkinson disease; (ii) describe traditional measures of vowel space area and how they relate to articulation; (iii) describe a novel measure of vowel space, the articulatory-acoustic

Speech Recognition Using Multiple Features and Multiple Recognizers

DTIC Science & Technology

1991-12-03

6 2.1 Introduction ............................................... 6 2.2 Human Speech Communication Process...119 How to Setup ASRT.......................................... 119 How to Use Interactive Menus .................................. 120...recognize a word from an acoustic signal. The human ear and brain perform this type of recognition with incredible speed and precision. Even though

Perceptual centres in speech - an acoustic analysis

NASA Astrophysics Data System (ADS)

Scott, Sophie Kerttu

Perceptual centres, or P-centres, represent the perceptual moments of occurrence of acoustic signals - the 'beat' of a sound. P-centres underlie the perception and production of rhythm in perceptually regular speech sequences. P-centres have been modelled both in speech and non speech (music) domains. The three aims of this thesis were toatest out current P-centre models to determine which best accounted for the experimental data bto identify a candidate parameter to map P-centres onto (a local approach) as opposed to the previous global models which rely upon the whole signal to determine the P-centre the final aim was to develop a model of P-centre location which could be applied to speech and non speech signals. The first aim was investigated by a series of experiments in which a) speech from different speakers was investigated to determine whether different models could account for variation between speakers b) whether rendering the amplitude time plot of a speech signal affects the P-centre of the signal c) whether increasing the amplitude at the offset of a speech signal alters P-centres in the production and perception of speech. The second aim was carried out by a) manipulating the rise time of different speech signals to determine whether the P-centre was affected, and whether the type of speech sound ramped affected the P-centre shift b) manipulating the rise time and decay time of a synthetic vowel to determine whether the onset alteration was had more affect on P-centre than the offset manipulation c) and whether the duration of a vowel affected the P-centre, if other attributes (amplitude, spectral contents) were held constant. The third aim - modelling P-centres - was based on these results. The Frequency dependent Amplitude Increase Model of P-centre location (FAIM) was developed using a modelling protocol, the APU GammaTone Filterbank and the speech from different speakers. The P-centres of the stimuli corpus were highly predicted by attributes of

Neural source dynamics of brain responses to continuous stimuli: Speech processing from acoustics to comprehension.

PubMed

Brodbeck, Christian; Presacco, Alessandro; Simon, Jonathan Z

2018-05-15

Human experience often involves continuous sensory information that unfolds over time. This is true in particular for speech comprehension, where continuous acoustic signals are processed over seconds or even minutes. We show that brain responses to such continuous stimuli can be investigated in detail, for magnetoencephalography (MEG) data, by combining linear kernel estimation with minimum norm source localization. Previous research has shown that the requirement to average data over many trials can be overcome by modeling the brain response as a linear convolution of the stimulus and a kernel, or response function, and estimating a kernel that predicts the response from the stimulus. However, such analysis has been typically restricted to sensor space. Here we demonstrate that this analysis can also be performed in neural source space. We first computed distributed minimum norm current source estimates for continuous MEG recordings, and then computed response functions for the current estimate at each source element, using the boosting algorithm with cross-validation. Permutation tests can then assess the significance of individual predictor variables, as well as features of the corresponding spatio-temporal response functions. We demonstrate the viability of this technique by computing spatio-temporal response functions for speech stimuli, using predictor variables reflecting acoustic, lexical and semantic processing. Results indicate that processes related to comprehension of continuous speech can be differentiated anatomically as well as temporally: acoustic information engaged auditory cortex at short latencies, followed by responses over the central sulcus and inferior frontal gyrus, possibly related to somatosensory/motor cortex involvement in speech perception; lexical frequency was associated with a left-lateralized response in auditory cortex and subsequent bilateral frontal activity; and semantic composition was associated with bilateral temporal and

Modification of computational auditory scene analysis (CASA) for noise-robust acoustic feature

NASA Astrophysics Data System (ADS)

Kwon, Minseok

While there have been many attempts to mitigate interferences of background noise, the performance of automatic speech recognition (ASR) still can be deteriorated by various factors with ease. However, normal hearing listeners can accurately perceive sounds of their interests, which is believed to be a result of Auditory Scene Analysis (ASA). As a first attempt, the simulation of the human auditory processing, called computational auditory scene analysis (CASA), was fulfilled through physiological and psychological investigations of ASA. CASA comprised of Zilany-Bruce auditory model, followed by tracking fundamental frequency for voice segmentation and detecting pairs of onset/offset at each characteristic frequency (CF) for unvoiced segmentation. The resulting Time-Frequency (T-F) representation of acoustic stimulation was converted into acoustic feature, gammachirp-tone frequency cepstral coefficients (GFCC). 11 keywords with various environmental conditions are used and the robustness of GFCC was evaluated by spectral distance (SD) and dynamic time warping distance (DTW). In "clean" and "noisy" conditions, the application of CASA generally improved noise robustness of the acoustic feature compared to a conventional method with or without noise suppression using MMSE estimator. The intial study, however, not only showed the noise-type dependency at low SNR, but also called the evaluation methods in question. Some modifications were made to capture better spectral continuity from an acoustic feature matrix, to obtain faster processing speed, and to describe the human auditory system more precisely. The proposed framework includes: 1) multi-scale integration to capture more accurate continuity in feature extraction, 2) contrast enhancement (CE) of each CF by competition with neighboring frequency bands, and 3) auditory model modifications. The model modifications contain the introduction of higher Q factor, middle ear filter more analogous to human auditory system

Acoustic analysis of speech under stress.

PubMed

Sondhi, Savita; Khan, Munna; Vijay, Ritu; Salhan, Ashok K; Chouhan, Satish

2015-01-01

When a person is emotionally charged, stress could be discerned in his voice. This paper presents a simplified and a non-invasive approach to detect psycho-physiological stress by monitoring the acoustic modifications during a stressful conversation. Voice database consists of audio clips from eight different popular FM broadcasts wherein the host of the show vexes the subjects who are otherwise unaware of the charade. The audio clips are obtained from real-life stressful conversations (no simulated emotions). Analysis is done using PRAAT software to evaluate mean fundamental frequency (F0) and formant frequencies (F1, F2, F3, F4) both in neutral and stressed state. Results suggest that F0 increases with stress; however, formant frequency decreases with stress. Comparison of Fourier and chirp spectra of short vowel segment shows that for relaxed speech, the two spectra are similar; however, for stressed speech, they differ in the high frequency range due to increased pitch modulation.

Fifty years of progress in acoustic phonetics

NASA Astrophysics Data System (ADS)

Stevens, Kenneth N.

2004-10-01

Three events that occurred 50 or 60 years ago shaped the study of acoustic phonetics, and in the following few decades these events influenced research and applications in speech disorders, speech development, speech synthesis, speech recognition, and other subareas in speech communication. These events were: (1) the source-filter theory of speech production (Chiba and Kajiyama; Fant); (2) the development of the sound spectrograph and its interpretation (Potter, Kopp, and Green; Joos); and (3) the birth of research that related distinctive features to acoustic patterns (Jakobson, Fant, and Halle). Following these events there has been systematic exploration of the articulatory, acoustic, and perceptual bases of phonological categories, and some quantification of the sources of variability in the transformation of this phonological representation of speech into its acoustic manifestations. This effort has been enhanced by studies of how children acquire language in spite of this variability and by research on speech disorders. Gaps in our knowledge of this inherent variability in speech have limited the directions of applications such as synthesis and recognition of speech, and have led to the implementation of data-driven techniques rather than theoretical principles. Some examples of advances in our knowledge, and limitations of this knowledge, are reviewed.

Perceptually relevant speech tracking in auditory and motor cortex reflects distinct linguistic features

PubMed Central

Gross, Joachim; Kayser, Christoph

2018-01-01

During online speech processing, our brain tracks the acoustic fluctuations in speech at different timescales. Previous research has focused on generic timescales (for example, delta or theta bands) that are assumed to map onto linguistic features such as prosody or syllables. However, given the high intersubject variability in speaking patterns, such a generic association between the timescales of brain activity and speech properties can be ambiguous. Here, we analyse speech tracking in source-localised magnetoencephalographic data by directly focusing on timescales extracted from statistical regularities in our speech material. This revealed widespread significant tracking at the timescales of phrases (0.6–1.3 Hz), words (1.8–3 Hz), syllables (2.8–4.8 Hz), and phonemes (8–12.4 Hz). Importantly, when examining its perceptual relevance, we found stronger tracking for correctly comprehended trials in the left premotor (PM) cortex at the phrasal scale as well as in left middle temporal cortex at the word scale. Control analyses using generic bands confirmed that these effects were specific to the speech regularities in our stimuli. Furthermore, we found that the phase at the phrasal timescale coupled to power at beta frequency (13–30 Hz) in motor areas. This cross-frequency coupling presumably reflects top-down temporal prediction in ongoing speech perception. Together, our results reveal specific functional and perceptually relevant roles of distinct tracking and cross-frequency processes along the auditory–motor pathway. PMID:29529019

Acoustics in Halls for Speech and Music

NASA Astrophysics Data System (ADS)

Gade, Anders C.

This chapter deals specifically with concepts, tools, and architectural variables of importance when designing auditoria for speech and music. The focus will be on cultivating the useful components of the sound in the room rather than on avoiding noise from outside or from installations, which is dealt with in Chap. 11. The chapter starts by presenting the subjective aspects of the room acoustic experience according to consensus at the time of writing. Then follows a description of their objective counterparts, the objective room acoustic parameters, among which the classical reverberation time measure is only one of many, but still of fundamental value. After explanations on how these parameters can be measured and predicted during the design phase, the remainder of the chapter deals with how the acoustic properties can be controlled by the architectural design of auditoria. This is done by presenting the influence of individual design elements as well as brief descriptions of halls designed for specific purposes, such as drama, opera, and symphonic concerts. Finally, some important aspects of loudspeaker installations in auditoria are briefly touched upon.

Digital signal processing at Bell Labs-Foundations for speech and acoustics research

NASA Astrophysics Data System (ADS)

Rabiner, Lawrence R.

2004-05-01

Digital signal processing (DSP) is a fundamental tool for much of the research that has been carried out of Bell Labs in the areas of speech and acoustics research. The fundamental bases for DSP include the sampling theorem of Nyquist, the method for digitization of analog signals by Shannon et al., methods of spectral analysis by Tukey, the cepstrum by Bogert et al., and the FFT by Tukey (and Cooley of IBM). Essentially all of these early foundations of DSP came out of the Bell Labs Research Lab in the 1930s, 1940s, 1950s, and 1960s. This fundamental research was motivated by fundamental applications (mainly in the areas of speech, sonar, and acoustics) that led to novel design methods for digital filters (Kaiser, Golden, Rabiner, Schafer), spectrum analysis methods (Rabiner, Schafer, Allen, Crochiere), fast convolution methods based on the FFT (Helms, Bergland), and advanced digital systems used to implement telephony channel banks (Jackson, McDonald, Freeny, Tewksbury). This talk summarizes the key contributions to DSP made at Bell Labs, and illustrates how DSP was utilized in the areas of speech and acoustics research. It also shows the vast, worldwide impact of this DSP research on modern consumer electronics.

Perceptual, auditory and acoustic vocal analysis of speech and singing in choir conductors.

PubMed

Rehder, Maria Inês Beltrati Cornacchioni; Behlau, Mara

2008-01-01

the voice of choir conductors. to evaluate the vocal quality of choir conductors based on the production of a sustained vowel during singing and when speaking in order to observe auditory and acoustic differences. participants of this study were 100 choir conductors, with an equal distribution between genders. Participants were asked to produce the sustained vowel "é" using a singing and speaking voice. Speech samples were analyzed based on auditory-perceptive and acoustic parameters. The auditory-perceptive analysis was carried out by two speech-language pathologist, specialists in this field of knowledge. The acoustic analysis was carried out with the support of the computer software Doctor Speech (Tiger Electronics, SRD, USA, version 4.0), using the Real Analysis module. the auditory-perceptive analysis of the vocal quality indicated that most conductors have adapted voices, presenting more alterations in their speaking voice. The acoustic analysis indicated different values between genders and between the different production modalities. The fundamental frequency was higher in the singing voice, as well as the values for the first formant; the second formant presented lower values in the singing voice, with statistically significant results only for women. the voice of choir conductors is adapted, presenting fewer deviations in the singing voice when compared to the speaking voice. Productions differ based the voice modality, singing or speaking.

A multimodal spectral approach to characterize rhythm in natural speech.

PubMed

Alexandrou, Anna Maria; Saarinen, Timo; Kujala, Jan; Salmelin, Riitta

2016-01-01

Human utterances demonstrate temporal patterning, also referred to as rhythm. While simple oromotor behaviors (e.g., chewing) feature a salient periodical structure, conversational speech displays a time-varying quasi-rhythmic pattern. Quantification of periodicity in speech is challenging. Unimodal spectral approaches have highlighted rhythmic aspects of speech. However, speech is a complex multimodal phenomenon that arises from the interplay of articulatory, respiratory, and vocal systems. The present study addressed the question of whether a multimodal spectral approach, in the form of coherence analysis between electromyographic (EMG) and acoustic signals, would allow one to characterize rhythm in natural speech more efficiently than a unimodal analysis. The main experimental task consisted of speech production at three speaking rates; a simple oromotor task served as control. The EMG-acoustic coherence emerged as a sensitive means of tracking speech rhythm, whereas spectral analysis of either EMG or acoustic amplitude envelope alone was less informative. Coherence metrics seem to distinguish and highlight rhythmic structure in natural speech.

Voice Modulations in German Ironic Speech

ERIC Educational Resources Information Center

Scharrer, Lisa; Christmann, Ursula; Knoll, Monja

2011-01-01

Previous research has shown that in different languages ironic speech is acoustically modulated compared to literal speech, and these modulations are assumed to aid the listener in the comprehension process by acting as cues that mark utterances as ironic. The present study was conducted to identify paraverbal features of German "ironic…

Formant Centralization Ratio: A Proposal for a New Acoustic Measure of Dysarthric Speech

ERIC Educational Resources Information Center

Sapir, Shimon; Ramig, Lorraine O.; Spielman, Jennifer L.; Fox, Cynthia

2010-01-01

Purpose: The vowel space area (VSA) has been used as an acoustic metric of dysarthric speech, but with varying degrees of success. In this study, the authors aimed to test an alternative metric to the VSA--the "formant centralization ratio" (FCR), which is hypothesized to more effectively differentiate dysarthric from healthy speech and register…

Examining Acoustic and Kinematic Measures of Articulatory Working Space: Effects of Speech Intensity

ERIC Educational Resources Information Center

Whitfield, Jason A.; Dromey, Christopher; Palmer, Panika

2018-01-01

Purpose: The purpose of this study was to examine the effect of speech intensity on acoustic and kinematic vowel space measures and conduct a preliminary examination of the relationship between kinematic and acoustic vowel space metrics calculated from continuously sampled lingual marker and formant traces. Method: Young adult speakers produced 3…

[Acoustic voice analysis using the Praat program: comparative study with the Dr. Speech program].

PubMed

Núñez Batalla, Faustino; González Márquez, Rocío; Peláez González, M Belén; González Laborda, Irene; Fernández Fernández, María; Morato Galán, Marta

2014-01-01

The European Laryngological Society (ELS) basic protocol for functional assessment of voice pathology includes 5 different approaches: perception, videostroboscopy, acoustics, aerodynamics and subjective rating by the patient. In this study we focused on acoustic voice analysis. The purpose of the present study was to correlate the results obtained by the commercial software Dr. Speech and the free software Praat in 2 fields: 1. Narrow-band spectrogram (the presence of noise according to Yanagihara, and the presence of subharmonics) (semi-quantitative). 2. Voice acoustic parameters (jitter, shimmer, harmonics-to-noise ratio, fundamental frequency) (quantitative). We studied a total of 99 voice samples from individuals with Reinke's oedema diagnosed using videostroboscopy. One independent observer used Dr. Speech 3.0 and a second one used the Praat program (Phonetic Sciences, University of Amsterdam). The spectrographic analysis consisted of obtaining a narrow-band spectrogram from the previous digitalised voice samples by the 2 independent observers. They then determined the presence of noise in the spectrogram, using the Yanagihara grades, as well as the presence of subharmonics. As a final result, the acoustic parameters of jitter, shimmer, harmonics-to-noise ratio and fundamental frequency were obtained from the 2 acoustic analysis programs. The results indicated that the sound spectrogram and the numerical values obtained for shimmer and jitter were similar for both computer programs, even though types 1, 2 and 3 voice samples were analysed. The Praat and Dr. Speech programs provide similar results in the acoustic analysis of pathological voices. Copyright © 2013 Elsevier España, S.L. All rights reserved.

Influence of compact disk recording protocols on reliability and comparability of speech audiometry outcomes: acoustic analysis.

PubMed

Di Berardino, F; Tognola, G; Paglialonga, A; Alpini, D; Grandori, F; Cesarani, A

2010-08-01

To assess whether different compact disk recording protocols, used to prepare speech test material, affect the reliability and comparability of speech audiometry testing. We conducted acoustic analysis of compact disks used in clinical practice, to determine whether speech material had been recorded using similar procedures. To assess the impact of different recording procedures on speech test outcomes, normal hearing subjects were tested using differently prepared compact disks, and their psychometric curves compared. Acoustic analysis revealed that speech material had been recorded using different protocols. The major difference was the gain between the levels at which the speech material and the calibration signal had been recorded. Although correct calibration of the audiometer was performed for each compact disk before testing, speech recognition thresholds and maximum intelligibility thresholds differed significantly between compact disks (p < 0.05), and were influenced by the gain between the recording level of the speech material and the calibration signal. To ensure the reliability and comparability of speech test outcomes obtained using different compact disks, it is recommended to check for possible differences in the recording gains used to prepare the compact disks, and then to compensate for any differences before testing.

The Speech multi features fusion perceptual hash algorithm based on tensor decomposition

NASA Astrophysics Data System (ADS)

Huang, Y. B.; Fan, M. H.; Zhang, Q. Y.

2018-03-01

With constant progress in modern speech communication technologies, the speech data is prone to be attacked by the noise or maliciously tampered. In order to make the speech perception hash algorithm has strong robustness and high efficiency, this paper put forward a speech perception hash algorithm based on the tensor decomposition and multi features is proposed. This algorithm analyses the speech perception feature acquires each speech component wavelet packet decomposition. LPCC, LSP and ISP feature of each speech component are extracted to constitute the speech feature tensor. Speech authentication is done by generating the hash values through feature matrix quantification which use mid-value. Experimental results showing that the proposed algorithm is robust for content to maintain operations compared with similar algorithms. It is able to resist the attack of the common background noise. Also, the algorithm is highly efficiency in terms of arithmetic, and is able to meet the real-time requirements of speech communication and complete the speech authentication quickly.

Differences in early speech patterns between Parkinson variant of multiple system atrophy and Parkinson's disease.

PubMed

Huh, Young Eun; Park, Jongkyu; Suh, Mee Kyung; Lee, Sang Eun; Kim, Jumin; Jeong, Yuri; Kim, Hee-Tae; Cho, Jin Whan

2015-08-01

In Parkinson variant of multiple system atrophy (MSA-P), patterns of early speech impairment and their distinguishing features from Parkinson's disease (PD) require further exploration. Here, we compared speech data among patients with early-stage MSA-P, PD, and healthy subjects using quantitative acoustic and perceptual analyses. Variables were analyzed for men and women in view of gender-specific features of speech. Acoustic analysis revealed that male patients with MSA-P exhibited more profound speech abnormalities than those with PD, regarding increased voice pitch, prolonged pause time, and reduced speech rate. This might be due to widespread pathology of MSA-P in nigrostriatal or extra-striatal structures related to speech production. Although several perceptual measures were mildly impaired in MSA-P and PD patients, none of these parameters showed a significant difference between patient groups. Detailed speech analysis using acoustic measures may help distinguish between MSA-P and PD early in the disease process. Copyright © 2015 Elsevier Inc. All rights reserved.

[Simulation of speech perception with cochlear implants : Influence of frequency and level of fundamental frequency components with electronic acoustic stimulation].

PubMed

Rader, T; Fastl, H; Baumann, U

2017-03-01

After implantation of cochlear implants with hearing preservation for combined electronic acoustic stimulation (EAS), the residual acoustic hearing ability relays fundamental speech frequency information in the low frequency range. With the help of acoustic simulation of EAS hearing perception the impact of frequency and level fine structure of speech signals can be systematically examined. The aim of this study was to measure the speech reception threshold (SRT) under various noise conditions with acoustic EAS simulation by variation of the frequency and level information of the fundamental frequency f0 of speech. The study was carried out to determine to what extent the SRT is impaired by modification of the f0 fine structure. Using partial tone time pattern analysis an acoustic EAS simulation of the speech material from the Oldenburg sentence test (OLSA) was generated. In addition, determination of the f0 curve of the speech material was conducted. Subsequently, either the parameter frequency or level of f0 was fixed in order to remove one of the two fine contour information of the speech signal. The processed OLSA sentences were used to determine the SRT in background noise under various test conditions. The conditions "f0 fixed frequency" and "f0 fixed level" were tested under two different situations, under "amplitude modulated background noise" and "continuous background noise" conditions. A total of 24 subjects with normal hearing participated in the study. The SRT in background noise for the condition "f0 fixed frequency" was more favorable in continuous noise with 2.7 dB and in modulated noise with 0.8 dB compared to the condition "f0 fixed level" with 3.7 dB and 2.9 dB, respectively. In the simulation of speech perception with cochlear implants and acoustic components, the level information of the fundamental frequency had a stronger impact on speech intelligibility than the frequency information. The method of simulation of transmission of

Effects of Age, Acoustic Challenge, and Verbal Working Memory on Recall of Narrative Speech.

PubMed

Ward, Caitlin M; Rogers, Chad S; Van Engen, Kristin J; Peelle, Jonathan E

2016-01-01

A common goal during speech comprehension is to remember what we have heard. Encoding speech into long-term memory frequently requires processes such as verbal working memory that may also be involved in processing degraded speech. Here the authors tested whether young and older adult listeners' memory for short stories was worse when the stories were acoustically degraded, or whether the additional contextual support provided by a narrative would protect against these effects. The authors tested 30 young adults (aged 18-28 years) and 30 older adults (aged 65-79 years) with good self-reported hearing. Participants heard short stories that were presented as normal (unprocessed) speech or acoustically degraded using a noise vocoding algorithm with 24 or 16 channels. The degraded stories were still fully intelligible. Following each story, participants were asked to repeat the story in as much detail as possible. Recall was scored using a modified idea unit scoring approach, which included separately scoring hierarchical levels of narrative detail. Memory for acoustically degraded stories was significantly worse than for normal stories at some levels of narrative detail. Older adults' memory for the stories was significantly worse overall, but there was no interaction between age and acoustic clarity or level of narrative detail. Verbal working memory (assessed by reading span) significantly correlated with recall accuracy for both young and older adults, whereas hearing ability (better ear pure tone average) did not. The present findings are consistent with a framework in which the additional cognitive demands caused by a degraded acoustic signal use resources that would otherwise be available for memory encoding for both young and older adults. Verbal working memory is a likely candidate for supporting both of these processes.

Acoustics of Clear and Noise-Adapted Speech in Children, Young, and Older Adults

ERIC Educational Resources Information Center

Smiljanic, Rajka; Gilbert, Rachael C.

2017-01-01

Purpose: This study investigated acoustic-phonetic modifications produced in noise-adapted speech (NAS) and clear speech (CS) by children, young adults, and older adults. Method: Ten children (11-13 years of age), 10 young adults (18-29 years of age), and 10 older adults (60-84 years of age) read sentences in conversational and clear speaking…

The Queen's English: an alternative, biosocial hypothesis for the distinctive features of "gay speech".

PubMed

Rendall, Drew; Vasey, Paul L; McKenzie, Jared

2008-02-01

Popular stereotypes concerning the speech of homosexuals typically attribute speech patterns characteristic of the opposite-sex, i.e., broadly feminized speech in gay men and broadly masculinized speech in lesbian women. A small body of recent empirical research has begun to address the subject more systematically and to consider specific mechanistic hypotheses to account for the potentially distinctive features of homosexual speech. Results do not yet fully endorse the stereotypes but they do not entirely discount them either; nor do they cleanly favor any single mechanistic hypothesis. To contribute to this growing body of research, we report acoustic analyses of 2,875 vowel sounds from a balanced set of 125 speakers representing heterosexual and homosexual individuals of each sex from southern Alberta, Canada. Analyses focused on voice pitch and formant frequencies which together determine the principle perceptual features of vowels. There was no significant difference in mean voice pitch between heterosexual and homosexual men or between heterosexual and homosexual women, but there were significant differences in the formant frequencies of vowels produced by both homosexual groups compared to their heterosexual counterparts. Formant frequency differences were specific to only certain vowel sounds and some could be attributed to basic differences in body size between heterosexual and homosexual speakers. The remaining formant frequency differences were not obviously due to differences in vocal tract anatomy between heterosexual and homosexual speakers, nor did they reflect global feminization or masculinization of vowel production patterns in homosexual men and women, respectively. The vowel-specific differences observed could reflect social modeling processes in which only certain speech patterns of the opposite-sex, or of same-sex homosexuals, are selectively adopted. However, we introduce an alternative biosocial hypothesis, specifically that the distinctive

Accuracy of Perceptual and Acoustic Methods for the Detection of Inspiratory Loci in Spontaneous Speech

PubMed Central

Wang, Yu-Tsai; Nip, Ignatius S. B.; Green, Jordan R.; Kent, Ray D.; Kent, Jane Finley; Ullman, Cara

2012-01-01

The current study investigates the accuracy of perceptually and acoustically determined inspiratory loci in spontaneous speech for the purpose of identifying breath groups. Sixteen participants were asked to talk about simple topics in daily life at a comfortable speaking rate and loudness while connected to a pneumotach and audio microphone. The locations of inspiratory loci were determined based on the aerodynamic signal, which served as a reference for loci identified perceptually and acoustically. Signal detection theory was used to evaluate the accuracy of the methods. The results showed that the greatest accuracy in pause detection was achieved (1) perceptually based on the agreement between at least 2 of the 3 judges; (2) acoustically using a pause duration threshold of 300 ms. In general, the perceptually-based method was more accurate than was the acoustically-based method. Inconsistencies among perceptually-determined, acoustically-determined, and aerodynamically-determined inspiratory loci for spontaneous speech should be weighed in selecting a method of breath-group determination. PMID:22362007

Adaptation to an electropalatograph palate: acoustic, impressionistic, and perceptual data.

PubMed

McLeod, Sharynne; Searl, Jeff

2006-05-01

The purpose of this study was to evaluate adaptation to the electropalatograph (EPG) from the perspective of consonant acoustics, listener perceptions, and speaker ratings. Seven adults with typical speech wore an EPG and pseudo-EPG palate over 2 days and produced syllables, read a passage, counted, and rated their adaptation to the palate. Consonant acoustics, listener ratings, and speaker ratings were analyzed. The spectral mean for the burst (/t/) and frication (/s/) was reduced for the first 60-120 min of wearing the pseudo-EPG palate. Temporal features (stop gap, frication, and syllable duration) were unaffected by wearing the pseudo-EPG palate. The EPG palate had a similar effect on consonant acoustics as the pseudo-EPG palate. Expert listener ratings indicated minimal to no change in speech naturalness or distortion from the pseudo-EPG or EPG palate. The sounds [see text] were most likely to be affected. Speaker self-ratings related to oral comfort, speech, tongue movement, appearance, and oral sensation were negatively affected by the presence of the palatal devices. Speakers detected a substantial difference when wearing a palatal device, but the effects on speech were minimal based on listener ratings. Spectral features of consonants were initially affected, although adaptation occurred. Wearing an EPG or pseudo-EPG palate for approximately 2 hr results in relatively normal-sounding speech with acoustic features similar to a no-palate condition.

Musical ability and non-native speech-sound processing are linked through sensitivity to pitch and spectral information.

PubMed

Kempe, Vera; Bublitz, Dennis; Brooks, Patricia J

2015-05-01

Is the observed link between musical ability and non-native speech-sound processing due to enhanced sensitivity to acoustic features underlying both musical and linguistic processing? To address this question, native English speakers (N = 118) discriminated Norwegian tonal contrasts and Norwegian vowels. Short tones differing in temporal, pitch, and spectral characteristics were used to measure sensitivity to the various acoustic features implicated in musical and speech processing. Musical ability was measured using Gordon's Advanced Measures of Musical Audiation. Results showed that sensitivity to specific acoustic features played a role in non-native speech-sound processing: Controlling for non-verbal intelligence, prior foreign language-learning experience, and sex, sensitivity to pitch and spectral information partially mediated the link between musical ability and discrimination of non-native vowels and lexical tones. The findings suggest that while sensitivity to certain acoustic features partially mediates the relationship between musical ability and non-native speech-sound processing, complex tests of musical ability also tap into other shared mechanisms. © 2014 The British Psychological Society.

Logopenic and Nonfluent Variants of Primary Progressive Aphasia Are Differentiated by Acoustic Measures of Speech Production

PubMed Central

Ballard, Kirrie J.; Savage, Sharon; Leyton, Cristian E.; Vogel, Adam P.; Hornberger, Michael; Hodges, John R.

2014-01-01

Differentiation of logopenic (lvPPA) and nonfluent/agrammatic (nfvPPA) variants of Primary Progressive Aphasia is important yet remains challenging since it hinges on expert based evaluation of speech and language production. In this study acoustic measures of speech in conjunction with voxel-based morphometry were used to determine the success of the measures as an adjunct to diagnosis and to explore the neural basis of apraxia of speech in nfvPPA. Forty-one patients (21 lvPPA, 20 nfvPPA) were recruited from a consecutive sample with suspected frontotemporal dementia. Patients were diagnosed using the current gold-standard of expert perceptual judgment, based on presence/absence of particular speech features during speaking tasks. Seventeen healthy age-matched adults served as controls. MRI scans were available for 11 control and 37 PPA cases; 23 of the PPA cases underwent amyloid ligand PET imaging. Measures, corresponding to perceptual features of apraxia of speech, were periods of silence during reading and relative vowel duration and intensity in polysyllable word repetition. Discriminant function analyses revealed that a measure of relative vowel duration differentiated nfvPPA cases from both control and lvPPA cases (r 2 = 0.47) with 88% agreement with expert judgment of presence of apraxia of speech in nfvPPA cases. VBM analysis showed that relative vowel duration covaried with grey matter intensity in areas critical for speech motor planning and programming: precentral gyrus, supplementary motor area and inferior frontal gyrus bilaterally, only affected in the nfvPPA group. This bilateral involvement of frontal speech networks in nfvPPA potentially affects access to compensatory mechanisms involving right hemisphere homologues. Measures of silences during reading also discriminated the PPA and control groups, but did not increase predictive accuracy. Findings suggest that a measure of relative vowel duration from of a polysyllable word repetition task

A chimpanzee recognizes synthetic speech with significantly reduced acoustic cues to phonetic content.

PubMed

Heimbauer, Lisa A; Beran, Michael J; Owren, Michael J

2011-07-26

A long-standing debate concerns whether humans are specialized for speech perception, which some researchers argue is demonstrated by the ability to understand synthetic speech with significantly reduced acoustic cues to phonetic content. We tested a chimpanzee (Pan troglodytes) that recognizes 128 spoken words, asking whether she could understand such speech. Three experiments presented 48 individual words, with the animal selecting a corresponding visuographic symbol from among four alternatives. Experiment 1 tested spectrally reduced, noise-vocoded (NV) synthesis, originally developed to simulate input received by human cochlear-implant users. Experiment 2 tested "impossibly unspeechlike" sine-wave (SW) synthesis, which reduces speech to just three moving tones. Although receiving only intermittent and noncontingent reward, the chimpanzee performed well above chance level, including when hearing synthetic versions for the first time. Recognition of SW words was least accurate but improved in experiment 3 when natural words in the same session were rewarded. The chimpanzee was more accurate with NV than SW versions, as were 32 human participants hearing these items. The chimpanzee's ability to spontaneously recognize acoustically reduced synthetic words suggests that experience rather than specialization is critical for speech-perception capabilities that some have suggested are uniquely human. Copyright © 2011 Elsevier Ltd. All rights reserved.

A Cross-Language Study of Acoustic Predictors of Speech Intelligibility in Individuals With Parkinson's Disease

PubMed Central

Choi, Yaelin

2017-01-01

Purpose The present study aimed to compare acoustic models of speech intelligibility in individuals with the same disease (Parkinson's disease [PD]) and presumably similar underlying neuropathologies but with different native languages (American English [AE] and Korean). Method A total of 48 speakers from the 4 speaker groups (AE speakers with PD, Korean speakers with PD, healthy English speakers, and healthy Korean speakers) were asked to read a paragraph in their native languages. Four acoustic variables were analyzed: acoustic vowel space, voice onset time contrast scores, normalized pairwise variability index, and articulation rate. Speech intelligibility scores were obtained from scaled estimates of sentences extracted from the paragraph. Results The findings indicated that the multiple regression models of speech intelligibility were different in Korean and AE, even with the same set of predictor variables and with speakers matched on speech intelligibility across languages. Analysis of the descriptive data for the acoustic variables showed the expected compression of the vowel space in speakers with PD in both languages, lower normalized pairwise variability index scores in Korean compared with AE, and no differences within or across language in articulation rate. Conclusions The results indicate that the basis of an intelligibility deficit in dysarthria is likely to depend on the native language of the speaker and listener. Additional research is required to explore other potential predictor variables, as well as additional language comparisons to pursue cross-linguistic considerations in classification and diagnosis of dysarthria types. PMID:28821018

Speech endpoint detection with non-language speech sounds for generic speech processing applications

NASA Astrophysics Data System (ADS)

McClain, Matthew; Romanowski, Brian

2009-05-01

Non-language speech sounds (NLSS) are sounds produced by humans that do not carry linguistic information. Examples of these sounds are coughs, clicks, breaths, and filled pauses such as "uh" and "um" in English. NLSS are prominent in conversational speech, but can be a significant source of errors in speech processing applications. Traditionally, these sounds are ignored by speech endpoint detection algorithms, where speech regions are identified in the audio signal prior to processing. The ability to filter NLSS as a pre-processing step can significantly enhance the performance of many speech processing applications, such as speaker identification, language identification, and automatic speech recognition. In order to be used in all such applications, NLSS detection must be performed without the use of language models that provide knowledge of the phonology and lexical structure of speech. This is especially relevant to situations where the languages used in the audio are not known apriori. We present the results of preliminary experiments using data from American and British English speakers, in which segments of audio are classified as language speech sounds (LSS) or NLSS using a set of acoustic features designed for language-agnostic NLSS detection and a hidden-Markov model (HMM) to model speech generation. The results of these experiments indicate that the features and model used are capable of detection certain types of NLSS, such as breaths and clicks, while detection of other types of NLSS such as filled pauses will require future research.

Advancements in robust algorithm formulation for speaker identification of whispered speech

NASA Astrophysics Data System (ADS)

Fan, Xing

Whispered speech is an alternative speech production mode from neutral speech, which is used by talkers intentionally in natural conversational scenarios to protect privacy and to avoid certain content from being overheard/made public. Due to the profound differences between whispered and neutral speech in production mechanism and the absence of whispered adaptation data, the performance of speaker identification systems trained with neutral speech degrades significantly. This dissertation therefore focuses on developing a robust closed-set speaker recognition system for whispered speech by using no or limited whispered adaptation data from non-target speakers. This dissertation proposes the concept of "High''/"Low'' performance whispered data for the purpose of speaker identification. A variety of acoustic properties are identified that contribute to the quality of whispered data. An acoustic analysis is also conducted to compare the phoneme/speaker dependency of the differences between whispered and neutral data in the feature domain. The observations from those acoustic analysis are new in this area and also serve as a guidance for developing robust speaker identification systems for whispered speech. This dissertation further proposes two systems for speaker identification of whispered speech. One system focuses on front-end processing. A two-dimensional feature space is proposed to search for "Low''-quality performance based whispered utterances and separate feature mapping functions are applied to vowels and consonants respectively in order to retain the speaker's information shared between whispered and neutral speech. The other system focuses on speech-mode-independent model training. The proposed method generates pseudo whispered features from neutral features by using the statistical information contained in a whispered Universal Background model (UBM) trained from extra collected whispered data from non-target speakers. Four modeling methods are proposed

Speech intelligibility in complex acoustic environments in young children

NASA Astrophysics Data System (ADS)

Litovsky, Ruth

2003-04-01

While the auditory system undergoes tremendous maturation during the first few years of life, it has become clear that in complex scenarios when multiple sounds occur and when echoes are present, children's performance is significantly worse than their adult counterparts. The ability of children (3-7 years of age) to understand speech in a simulated multi-talker environment and to benefit from spatial separation of the target and competing sounds was investigated. In these studies, competing sources vary in number, location, and content (speech, modulated or unmodulated speech-shaped noise and time-reversed speech). The acoustic spaces were also varied in size and amount of reverberation. Finally, children with chronic otitis media who received binaural training were tested pre- and post-training on a subset of conditions. Results indicated the following. (1) Children experienced significantly more masking than adults, even in the simplest conditions tested. (2) When the target and competing sounds were spatially separated speech intelligibility improved, but the amount varied with age, type of competing sound, and number of competitors. (3) In a large reverberant classroom there was no benefit of spatial separation. (4) Binaural training improved speech intelligibility performance in children with otitis media. Future work includes similar studies in children with unilateral and bilateral cochlear implants. [Work supported by NIDCD, DRF, and NOHR.

Speech Adaptation to Kinematic Recording Sensors: Perceptual and Acoustic Findings

ERIC Educational Resources Information Center

Dromey, Christopher; Hunter, Elise; Nissen, Shawn L.

2018-01-01

Purpose: This study used perceptual and acoustic measures to examine the time course of speech adaptation after the attachment of electromagnetic sensor coils to the tongue, lips, and jaw. Method: Twenty native English speakers read aloud stimulus sentences before the attachment of the sensors, immediately after attachment, and again 5, 10, 15,…

Speech acoustic markers of early stage and prodromal Huntington's disease: a marker of disease onset?

PubMed

Vogel, Adam P; Shirbin, Christopher; Churchyard, Andrew J; Stout, Julie C

2012-12-01

Speech disturbances (e.g., altered prosody) have been described in symptomatic Huntington's Disease (HD) individuals, however, the extent to which speech changes in gene positive pre-manifest (PreHD) individuals is largely unknown. The speech of individuals carrying the mutant HTT gene is a behavioural/motor/cognitive marker demonstrating some potential as an objective indicator of early HD onset and disease progression. Speech samples were acquired from 30 individuals carrying the mutant HTT gene (13 PreHD, 17 early stage HD) and 15 matched controls. Participants read a passage, produced a monologue and said the days of the week. Data were analysed acoustically for measures of timing, frequency and intensity. There was a clear effect of group across most acoustic measures, so that speech performance differed in-line with disease progression. Comparisons across groups revealed significant differences between the control and the early stage HD group on measures of timing (e.g., speech rate). Participants carrying the mutant HTT gene presented with slower rates of speech, took longer to say words and produced greater silences between and within words compared to healthy controls. Importantly, speech rate showed a significant correlation to burden of disease scores. The speech of early stage HD differed significantly from controls. The speech of PreHD, although not reaching significance, tended to lie between the performance of controls and early stage HD. This suggests that changes in speech production appear to be developing prior to diagnosis. Copyright © 2012 Elsevier Ltd. All rights reserved.

Predicting Speech Intelligibility with a Multiple Speech Subsystems Approach in Children with Cerebral Palsy

ERIC Educational Resources Information Center

Lee, Jimin; Hustad, Katherine C.; Weismer, Gary

2014-01-01

Purpose: Speech acoustic characteristics of children with cerebral palsy (CP) were examined with a multiple speech subsystems approach; speech intelligibility was evaluated using a prediction model in which acoustic measures were selected to represent three speech subsystems. Method: Nine acoustic variables reflecting different subsystems, and…

Temporal modulations in speech and music.

PubMed

Ding, Nai; Patel, Aniruddh D; Chen, Lin; Butler, Henry; Luo, Cheng; Poeppel, David

2017-10-01

Speech and music have structured rhythms. Here we discuss a major acoustic correlate of spoken and musical rhythms, the slow (0.25-32Hz) temporal modulations in sound intensity and compare the modulation properties of speech and music. We analyze these modulations using over 25h of speech and over 39h of recordings of Western music. We show that the speech modulation spectrum is highly consistent across 9 languages (including languages with typologically different rhythmic characteristics). A different, but similarly consistent modulation spectrum is observed for music, including classical music played by single instruments of different types, symphonic, jazz, and rock. The temporal modulations of speech and music show broad but well-separated peaks around 5 and 2Hz, respectively. These acoustically dominant time scales may be intrinsic features of speech and music, a possibility which should be investigated using more culturally diverse samples in each domain. Distinct modulation timescales for speech and music could facilitate their perceptual analysis and its neural processing. Copyright © 2017 Elsevier Ltd. All rights reserved.

Disordered speech disrupts conversational entrainment: a study of acoustic-prosodic entrainment and communicative success in populations with communication challenges

PubMed Central

Borrie, Stephanie A.; Lubold, Nichola; Pon-Barry, Heather

2015-01-01

Conversational entrainment, a pervasive communication phenomenon in which dialogue partners adapt their behaviors to align more closely with one another, is considered essential for successful spoken interaction. While well-established in other disciplines, this phenomenon has received limited attention in the field of speech pathology and the study of communication breakdowns in clinical populations. The current study examined acoustic-prosodic entrainment, as well as a measure of communicative success, in three distinctly different dialogue groups: (i) healthy native vs. healthy native speakers (Control), (ii) healthy native vs. foreign-accented speakers (Accented), and (iii) healthy native vs. dysarthric speakers (Disordered). Dialogue group comparisons revealed significant differences in how the groups entrain on particular acoustic–prosodic features, including pitch, intensity, and jitter. Most notably, the Disordered dialogues were characterized by significantly less acoustic-prosodic entrainment than the Control dialogues. Further, a positive relationship between entrainment indices and communicative success was identified. These results suggest that the study of conversational entrainment in speech pathology will have essential implications for both scientific theory and clinical application in this domain. PMID:26321996

Acoustic Features Influence Musical Choices Across Multiple Genres.

PubMed

Barone, Michael D; Bansal, Jotthi; Woolhouse, Matthew H

2017-01-01

Based on a large behavioral dataset of music downloads, two analyses investigate whether the acoustic features of listeners' preferred musical genres influence their choice of tracks within non-preferred, secondary musical styles. Analysis 1 identifies feature distributions for pairs of genre-defined subgroups that are distinct. Using correlation analysis, these distributions are used to test the degree of similarity between subgroups' main genres and the other music within their download collections. Analysis 2 explores the issue of main-to-secondary genre influence through the production of 10 feature-influence matrices, one per acoustic feature, in which cell values indicate the percentage change in features for genres and subgroups compared to overall population averages. In total, 10 acoustic features and 10 genre-defined subgroups are explored within the two analyses. Results strongly indicate that the acoustic features of people's main genres influence the tracks they download within non-preferred, secondary musical styles. The nature of this influence and its possible actuating mechanisms are discussed with respect to research on musical preference, personality, and statistical learning.

Acoustic Features Influence Musical Choices Across Multiple Genres

PubMed Central

Barone, Michael D.; Bansal, Jotthi; Woolhouse, Matthew H.

2017-01-01

Based on a large behavioral dataset of music downloads, two analyses investigate whether the acoustic features of listeners' preferred musical genres influence their choice of tracks within non-preferred, secondary musical styles. Analysis 1 identifies feature distributions for pairs of genre-defined subgroups that are distinct. Using correlation analysis, these distributions are used to test the degree of similarity between subgroups' main genres and the other music within their download collections. Analysis 2 explores the issue of main-to-secondary genre influence through the production of 10 feature-influence matrices, one per acoustic feature, in which cell values indicate the percentage change in features for genres and subgroups compared to overall population averages. In total, 10 acoustic features and 10 genre-defined subgroups are explored within the two analyses. Results strongly indicate that the acoustic features of people's main genres influence the tracks they download within non-preferred, secondary musical styles. The nature of this influence and its possible actuating mechanisms are discussed with respect to research on musical preference, personality, and statistical learning. PMID:28725200

Are you a good mimic? Neuro-acoustic signatures for speech imitation ability

PubMed Central

Reiterer, Susanne M.; Hu, Xiaochen; Sumathi, T. A.; Singh, Nandini C.

2013-01-01

We investigated individual differences in speech imitation ability in late bilinguals using a neuro-acoustic approach. One hundred and thirty-eight German-English bilinguals matched on various behavioral measures were tested for “speech imitation ability” in a foreign language, Hindi, and categorized into “high” and “low ability” groups. Brain activations and speech recordings were obtained from 26 participants from the two extreme groups as they performed a functional neuroimaging experiment which required them to “imitate” sentences in three conditions: (A) German, (B) English, and (C) German with fake English accent. We used recently developed novel acoustic analysis, namely the “articulation space” as a metric to compare speech imitation abilities of the two groups. Across all three conditions, direct comparisons between the two groups, revealed brain activations (FWE corrected, p < 0.05) that were more widespread with significantly higher peak activity in the left supramarginal gyrus and postcentral areas for the low ability group. The high ability group, on the other hand showed significantly larger articulation space in all three conditions. In addition, articulation space also correlated positively with imitation ability (Pearson's r = 0.7, p < 0.01). Our results suggest that an expanded articulation space for high ability individuals allows access to a larger repertoire of sounds, thereby providing skilled imitators greater flexibility in pronunciation and language learning. PMID:24155739

Perception of Sentence Stress in Speech Correlates With the Temporal Unpredictability of Prosodic Features.

PubMed

Kakouros, Sofoklis; Räsänen, Okko

2016-09-01

Numerous studies have examined the acoustic correlates of sentential stress and its underlying linguistic functionality. However, the mechanism that connects stress cues to the listener's attentional processing has remained unclear. Also, the learnability versus innateness of stress perception has not been widely discussed. In this work, we introduce a novel perspective to the study of sentential stress and put forward the hypothesis that perceived sentence stress in speech is related to the unpredictability of prosodic features, thereby capturing the attention of the listener. As predictability is based on the statistical structure of the speech input, the hypothesis also suggests that stress perception is a result of general statistical learning mechanisms. To study this idea, computational simulations are performed where temporal prosodic trajectories are modeled with an n-gram model. Probabilities of the feature trajectories are subsequently evaluated on a set of novel utterances and compared to human perception of stress. The results show that the low-probability regions of F0 and energy trajectories are strongly correlated with stress perception, giving support to the idea that attention and unpredictability of sensory stimulus are mutually connected. Copyright © 2015 Cognitive Science Society, Inc.

Acoustic constituents of prosodic typology

NASA Astrophysics Data System (ADS)

Komatsu, Masahiko

Different languages sound different, and considerable part of it derives from the typological difference of prosody. Although such difference is often referred to as lexical accent types (stress accent, pitch accent, and tone; e.g. English, Japanese, and Chinese respectively) and rhythm types (stress-, syllable-, and mora-timed rhythms; e.g. English, Spanish, and Japanese respectively), it is unclear whether these types are determined in terms of acoustic properties, The thesis intends to provide a potential basis for the description of prosody in terms of acoustics. It argues for the hypothesis that the source component of the source-filter model (acoustic features) approximately corresponds to prosody (linguistic features) through several experimental-phonetic studies. The study consists of four parts. (1) Preliminary experiment: Perceptual language identification tests were performed using English and Japanese speech samples whose frequency spectral information (i.e. non-source component) is heavily reduced. The results indicated that humans can discriminate languages with such signals. (2) Discussion on the linguistic information that the source component contains: This part constitutes the foundation of the argument of the thesis. Perception tests of consonants with the source signal indicated that the source component carries the information on broad categories of phonemes that contributes to the creation of rhythm. (3) Acoustic analysis: The speech samples of Chinese, English, Japanese, and Spanish, differing in prosodic types, were analyzed. These languages showed difference in acoustic characteristics of the source component. (4) Perceptual experiment: A language identification test for the above four languages was performed using the source signal with its acoustic features parameterized. It revealed that humans can discriminate prosodic types solely with the source features and that the discrimination is easier as acoustic information increases. The

An acoustical assessment of pitch-matching accuracy in relation to speech frequency, speech frequency range, age and gender in preschool children

NASA Astrophysics Data System (ADS)

Trollinger, Valerie L.

This study investigated the relationship between acoustical measurement of singing accuracy in relationship to speech fundamental frequency, speech fundamental frequency range, age and gender in preschool-aged children. Seventy subjects from Southeastern Pennsylvania; the San Francisco Bay Area, California; and Terre Haute, Indiana, participated in the study. Speech frequency was measured by having the subjects participate in spontaneous and guided speech activities with the researcher, with 18 diverse samples extracted from each subject's recording for acoustical analysis for fundamental frequency in Hz with the CSpeech computer program. The fundamental frequencies were averaged together to derive a mean speech frequency score for each subject. Speech range was calculated by subtracting the lowest fundamental frequency produced from the highest fundamental frequency produced, resulting in a speech range measured in increments of Hz. Singing accuracy was measured by having the subjects each echo-sing six randomized patterns using the pitches Middle C, D, E, F♯, G and A (440), using the solfege syllables of Do and Re, which were recorded by a 5-year-old female model. For each subject, 18 samples of singing were recorded. All samples were analyzed by the CSpeech for fundamental frequency. For each subject, deviation scores in Hz were derived by calculating the difference between what the model sang in Hz and what the subject sang in response in Hz. Individual scores for each child consisted of an overall mean total deviation frequency, mean frequency deviations for each pattern, and mean frequency deviation for each pitch. Pearson correlations, MANOVA and ANOVA analyses, Multiple Regressions and Discriminant Analysis revealed the following findings: (1) moderate but significant (p < .001) relationships emerged between mean speech frequency and the ability to sing the pitches E, F♯, G and A in the study; (2) mean speech frequency also emerged as the strongest

Particle Swarm Optimization Based Feature Enhancement and Feature Selection for Improved Emotion Recognition in Speech and Glottal Signals

PubMed Central

Muthusamy, Hariharan; Polat, Kemal; Yaacob, Sazali

2015-01-01

In the recent years, many research works have been published using speech related features for speech emotion recognition, however, recent studies show that there is a strong correlation between emotional states and glottal features. In this work, Mel-frequency cepstralcoefficients (MFCCs), linear predictive cepstral coefficients (LPCCs), perceptual linear predictive (PLP) features, gammatone filter outputs, timbral texture features, stationary wavelet transform based timbral texture features and relative wavelet packet energy and entropy features were extracted from the emotional speech (ES) signals and its glottal waveforms(GW). Particle swarm optimization based clustering (PSOC) and wrapper based particle swarm optimization (WPSO) were proposed to enhance the discerning ability of the features and to select the discriminating features respectively. Three different emotional speech databases were utilized to gauge the proposed method. Extreme learning machine (ELM) was employed to classify the different types of emotions. Different experiments were conducted and the results show that the proposed method significantly improves the speech emotion recognition performance compared to previous works published in the literature. PMID:25799141

A Speech Recognition-based Solution for the Automatic Detection of Mild Cognitive Impairment from Spontaneous Speech

PubMed Central

Tóth, László; Hoffmann, Ildikó; Gosztolya, Gábor; Vincze, Veronika; Szatlóczki, Gréta; Bánréti, Zoltán; Pákáski, Magdolna; Kálmán, János

2018-01-01

Background: Even today the reliable diagnosis of the prodromal stages of Alzheimer’s disease (AD) remains a great challenge. Our research focuses on the earliest detectable indicators of cognitive de-cline in mild cognitive impairment (MCI). Since the presence of language impairment has been reported even in the mild stage of AD, the aim of this study is to develop a sensitive neuropsychological screening method which is based on the analysis of spontaneous speech production during performing a memory task. In the future, this can form the basis of an Internet-based interactive screening software for the recognition of MCI. Methods: Participants were 38 healthy controls and 48 clinically diagnosed MCI patients. The provoked spontaneous speech by asking the patients to recall the content of 2 short black and white films (one direct, one delayed), and by answering one question. Acoustic parameters (hesitation ratio, speech tempo, length and number of silent and filled pauses, length of utterance) were extracted from the recorded speech sig-nals, first manually (using the Praat software), and then automatically, with an automatic speech recogni-tion (ASR) based tool. First, the extracted parameters were statistically analyzed. Then we applied machine learning algorithms to see whether the MCI and the control group can be discriminated automatically based on the acoustic features. Results: The statistical analysis showed significant differences for most of the acoustic parameters (speech tempo, articulation rate, silent pause, hesitation ratio, length of utterance, pause-per-utterance ratio). The most significant differences between the two groups were found in the speech tempo in the delayed recall task, and in the number of pauses for the question-answering task. The fully automated version of the analysis process – that is, using the ASR-based features in combination with machine learning - was able to separate the two classes with an F1-score of 78

A Speech Recognition-based Solution for the Automatic Detection of Mild Cognitive Impairment from Spontaneous Speech.

PubMed

Toth, Laszlo; Hoffmann, Ildiko; Gosztolya, Gabor; Vincze, Veronika; Szatloczki, Greta; Banreti, Zoltan; Pakaski, Magdolna; Kalman, Janos

2018-01-01

Even today the reliable diagnosis of the prodromal stages of Alzheimer's disease (AD) remains a great challenge. Our research focuses on the earliest detectable indicators of cognitive decline in mild cognitive impairment (MCI). Since the presence of language impairment has been reported even in the mild stage of AD, the aim of this study is to develop a sensitive neuropsychological screening method which is based on the analysis of spontaneous speech production during performing a memory task. In the future, this can form the basis of an Internet-based interactive screening software for the recognition of MCI. Participants were 38 healthy controls and 48 clinically diagnosed MCI patients. The provoked spontaneous speech by asking the patients to recall the content of 2 short black and white films (one direct, one delayed), and by answering one question. Acoustic parameters (hesitation ratio, speech tempo, length and number of silent and filled pauses, length of utterance) were extracted from the recorded speech signals, first manually (using the Praat software), and then automatically, with an automatic speech recognition (ASR) based tool. First, the extracted parameters were statistically analyzed. Then we applied machine learning algorithms to see whether the MCI and the control group can be discriminated automatically based on the acoustic features. The statistical analysis showed significant differences for most of the acoustic parameters (speech tempo, articulation rate, silent pause, hesitation ratio, length of utterance, pause-per-utterance ratio). The most significant differences between the two groups were found in the speech tempo in the delayed recall task, and in the number of pauses for the question-answering task. The fully automated version of the analysis process - that is, using the ASR-based features in combination with machine learning - was able to separate the two classes with an F1-score of 78.8%. The temporal analysis of spontaneous speech

Prosodic Contrasts in Ironic Speech

ERIC Educational Resources Information Center

Bryant, Gregory A.

2010-01-01

Prosodic features in spontaneous speech help disambiguate implied meaning not explicit in linguistic surface structure, but little research has examined how these signals manifest themselves in real conversations. Spontaneously produced verbal irony utterances generated between familiar speakers in conversational dyads were acoustically analyzed…

Executives' speech expressiveness: analysis of perceptive and acoustic aspects of vocal dynamics.

PubMed

Marquezin, Daniela Maria Santos Serrano; Viola, Izabel; Ghirardi, Ana Carolina de Assis Moura; Madureira, Sandra; Ferreira, Léslie Piccolotto

2015-01-01

To analyze speech expressiveness in a group of executives based on perceptive and acoustic aspects of vocal dynamics. Four male subjects participated in the research study (S1, S2, S3, and S4). The assessments included the Kingdomality test to obtain the keywords of communicative attitudes; perceptive-auditory assessment to characterize vocal quality and dynamics, performed by three judges who are speech language pathologists; perceptiveauditory assessment to judge the chosen keywords; speech acoustics to assess prosodic elements (Praat software); and a statistical analysis. According to the perceptive-auditory analysis of vocal dynamics, S1, S2, S3, and S4 did not show vocal alterations and all of them were considered with lowered habitual pitch. S1: pointed out as insecure, nonobjective, nonempathetic, and unconvincing with inappropriate use of pauses that are mainly formed by hesitations; inadequate separation of prosodic groups with breaking of syntagmatic constituents. S2: regular use of pauses for respiratory reload, organization of sentences, and emphasis, which is considered secure, little objective, empathetic, and convincing. S3: pointed out as secure, objective, empathetic, and convincing with regular use of pauses for respiratory reload and organization of sentences and hesitations. S4: the most secure, objective, empathetic, and convincing, with proper use of pauses for respiratory reload, planning, and emphasis; prosodic groups agreed with the statement, without separating the syntagmatic constituents. The speech characteristics and communicative attitudes were highlighted in two subjects in a different manner, in such a way that the slow rate of speech and breaks of the prosodic groups transmitted insecurity, little objectivity, and nonpersuasion.

Cross-Channel Amplitude Sweeps Are Crucial to Speech Intelligibility

ERIC Educational Resources Information Center

Prendergast, Garreth; Green, Gary G. R.

2012-01-01

Classical views of speech perception argue that the static and dynamic characteristics of spectral energy peaks (formants) are the acoustic features that underpin phoneme recognition. Here we use representations where the amplitude modulations of sub-band filtered speech are described, precisely, in terms of co-sinusoidal pulses. These pulses are…

A Model for Speech Processing in Second Language Listening Activities

ERIC Educational Resources Information Center

Zoghbor, Wafa Shahada

2016-01-01

Teachers' understanding of the process of speech perception could inform practice in listening classrooms. Catford (1950) developed a model for speech perception taking into account the influence of the acoustic features of the linguistic forms used by the speaker, whereby the listener "identifies" and "interprets" these…

Combined electric and acoustic hearing performance with Zebra® speech processor: speech reception, place, and temporal coding evaluation.

PubMed

Vaerenberg, Bart; Péan, Vincent; Lesbros, Guillaume; De Ceulaer, Geert; Schauwers, Karen; Daemers, Kristin; Gnansia, Dan; Govaerts, Paul J

2013-06-01

To assess the auditory performance of Digisonic(®) cochlear implant users with electric stimulation (ES) and electro-acoustic stimulation (EAS) with special attention to the processing of low-frequency temporal fine structure. Six patients implanted with a Digisonic(®) SP implant and showing low-frequency residual hearing were fitted with the Zebra(®) speech processor providing both electric and acoustic stimulation. Assessment consisted of monosyllabic speech identification tests in quiet and in noise at different presentation levels, and a pitch discrimination task using harmonic and disharmonic intonating complex sounds ( Vaerenberg et al., 2011 ). These tests investigate place and time coding through pitch discrimination. All tasks were performed with ES only and with EAS. Speech results in noise showed significant improvement with EAS when compared to ES. Whereas EAS did not yield better results in the harmonic intonation test, the improvements in the disharmonic intonation test were remarkable, suggesting better coding of pitch cues requiring phase locking. These results suggest that patients with residual hearing in the low-frequency range still have good phase-locking capacities, allowing them to process fine temporal information. ES relies mainly on place coding but provides poor low-frequency temporal coding, whereas EAS also provides temporal coding in the low-frequency range. Patients with residual phase-locking capacities can make use of these cues.

The Use of Artificial Neural Networks to Estimate Speech Intelligibility from Acoustic Variables: A Preliminary Analysis.

ERIC Educational Resources Information Center

Metz, Dale Evan; And Others

1992-01-01

A preliminary scheme for estimating the speech intelligibility of hearing-impaired speakers from acoustic parameters, using a computerized artificial neural network to process mathematically the acoustic input variables, is outlined. Tests with 60 hearing-impaired speakers found the scheme to be highly accurate in identifying speakers separated by…

The Relationship Between Speech Production and Speech Perception Deficits in Parkinson's Disease.

PubMed

De Keyser, Kim; Santens, Patrick; Bockstael, Annelies; Botteldooren, Dick; Talsma, Durk; De Vos, Stefanie; Van Cauwenberghe, Mieke; Verheugen, Femke; Corthals, Paul; De Letter, Miet

2016-10-01

This study investigated the possible relationship between hypokinetic speech production and speech intensity perception in patients with Parkinson's disease (PD). Participants included 14 patients with idiopathic PD and 14 matched healthy controls (HCs) with normal hearing and cognition. First, speech production was objectified through a standardized speech intelligibility assessment, acoustic analysis, and speech intensity measurements. Second, an overall estimation task and an intensity estimation task were addressed to evaluate overall speech perception and speech intensity perception, respectively. Finally, correlation analysis was performed between the speech characteristics of the overall estimation task and the corresponding acoustic analysis. The interaction between speech production and speech intensity perception was investigated by an intensity imitation task. Acoustic analysis and speech intensity measurements demonstrated significant differences in speech production between patients with PD and the HCs. A different pattern in the auditory perception of speech and speech intensity was found in the PD group. Auditory perceptual deficits may influence speech production in patients with PD. The present results suggest a disturbed auditory perception related to an automatic monitoring deficit in PD.

Automatic measurement and representation of prosodic features

NASA Astrophysics Data System (ADS)

Ying, Goangshiuan Shawn

Effective measurement and representation of prosodic features of the acoustic signal for use in automatic speech recognition and understanding systems is the goal of this work. Prosodic features-stress, duration, and intonation-are variations of the acoustic signal whose domains are beyond the boundaries of each individual phonetic segment. Listeners perceive prosodic features through a complex combination of acoustic correlates such as intensity, duration, and fundamental frequency (F0). We have developed new tools to measure F0 and intensity features. We apply a probabilistic global error correction routine to an Average Magnitude Difference Function (AMDF) pitch detector. A new short-term frequency-domain Teager energy algorithm is used to measure the energy of a speech signal. We have conducted a series of experiments performing lexical stress detection on words in continuous English speech from two speech corpora. We have experimented with two different approaches, a segment-based approach and a rhythm unit-based approach, in lexical stress detection. The first approach uses pattern recognition with energy- and duration-based measurements as features to build Bayesian classifiers to detect the stress level of a vowel segment. In the second approach we define rhythm unit and use only the F0-based measurement and a scoring system to determine the stressed segment in the rhythm unit. A duration-based segmentation routine was developed to break polysyllabic words into rhythm units. The long-term goal of this work is to develop a system that can effectively detect the stress pattern for each word in continuous speech utterances. Stress information will be integrated as a constraint for pruning the word hypotheses in a word recognition system based on hidden Markov models.

The Auditory-Brainstem Response to Continuous, Non-repetitive Speech Is Modulated by the Speech Envelope and Reflects Speech Processing

PubMed Central

Reichenbach, Chagit S.; Braiman, Chananel; Schiff, Nicholas D.; Hudspeth, A. J.; Reichenbach, Tobias

2016-01-01

The auditory-brainstem response (ABR) to short and simple acoustical signals is an important clinical tool used to diagnose the integrity of the brainstem. The ABR is also employed to investigate the auditory brainstem in a multitude of tasks related to hearing, such as processing speech or selectively focusing on one speaker in a noisy environment. Such research measures the response of the brainstem to short speech signals such as vowels or words. Because the voltage signal of the ABR has a tiny amplitude, several hundred to a thousand repetitions of the acoustic signal are needed to obtain a reliable response. The large number of repetitions poses a challenge to assessing cognitive functions due to neural adaptation. Here we show that continuous, non-repetitive speech, lasting several minutes, may be employed to measure the ABR. Because the speech is not repeated during the experiment, the precise temporal form of the ABR cannot be determined. We show, however, that important structural features of the ABR can nevertheless be inferred. In particular, the brainstem responds at the fundamental frequency of the speech signal, and this response is modulated by the envelope of the voiced parts of speech. We accordingly introduce a novel measure that assesses the ABR as modulated by the speech envelope, at the fundamental frequency of speech and at the characteristic latency of the response. This measure has a high signal-to-noise ratio and can hence be employed effectively to measure the ABR to continuous speech. We use this novel measure to show that the ABR is weaker to intelligible speech than to unintelligible, time-reversed speech. The methods presented here can be employed for further research on speech processing in the auditory brainstem and can lead to the development of future clinical diagnosis of brainstem function. PMID:27303286

Acoustic Correlates of Emphatic Stress in Central Catalan

ERIC Educational Resources Information Center

Nadeu, Marianna; Hualde, Jose Ignacio

2012-01-01

A common feature of public speech in Catalan is the placement of prominence on lexically unstressed syllables ("emphatic stress"). This paper presents an acoustic study of radio speech data. Instances of emphatic stress were perceptually identified. Within-word comparison between vowels with emphatic stress and vowels with primary lexical stress…

Decoding spectrotemporal features of overt and covert speech from the human cortex

PubMed Central

Martin, Stéphanie; Brunner, Peter; Holdgraf, Chris; Heinze, Hans-Jochen; Crone, Nathan E.; Rieger, Jochem; Schalk, Gerwin; Knight, Robert T.; Pasley, Brian N.

2014-01-01

Auditory perception and auditory imagery have been shown to activate overlapping brain regions. We hypothesized that these phenomena also share a common underlying neural representation. To assess this, we used electrocorticography intracranial recordings from epileptic patients performing an out loud or a silent reading task. In these tasks, short stories scrolled across a video screen in two conditions: subjects read the same stories both aloud (overt) and silently (covert). In a control condition the subject remained in a resting state. We first built a high gamma (70–150 Hz) neural decoding model to reconstruct spectrotemporal auditory features of self-generated overt speech. We then evaluated whether this same model could reconstruct auditory speech features in the covert speech condition. Two speech models were tested: a spectrogram and a modulation-based feature space. For the overt condition, reconstruction accuracy was evaluated as the correlation between original and predicted speech features, and was significant in each subject (p < 10−5; paired two-sample t-test). For the covert speech condition, dynamic time warping was first used to realign the covert speech reconstruction with the corresponding original speech from the overt condition. Reconstruction accuracy was then evaluated as the correlation between original and reconstructed speech features. Covert reconstruction accuracy was compared to the accuracy obtained from reconstructions in the baseline control condition. Reconstruction accuracy for the covert condition was significantly better than for the control condition (p < 0.005; paired two-sample t-test). The superior temporal gyrus, pre- and post-central gyrus provided the highest reconstruction information. The relationship between overt and covert speech reconstruction depended on anatomy. These results provide evidence that auditory representations of covert speech can be reconstructed from models that are built from an overt speech

Acoustic Changes in the Speech of Children with Cerebral Palsy Following an Intensive Program of Dysarthria Therapy

ERIC Educational Resources Information Center

Pennington, Lindsay; Lombardo, Eftychia; Steen, Nick; Miller, Nick

2018-01-01

Background: The speech intelligibility of children with dysarthria and cerebral palsy has been observed to increase following therapy focusing on respiration and phonation. Aims: To determine if speech intelligibility change following intervention is associated with change in acoustic measures of voice. Methods & Procedures: We recorded 16…

A comparison of recordings of sentences and spontaneous speech: perceptual and acoustic measures in preschool children's voices.

PubMed

McAllister, Anita; Brandt, Signe Kofoed

2012-09-01

A well-controlled recording in a studio is fundamental in most voice rehabilitation. However, this laboratory like recording method has been questioned because voice use in a natural environment may be quite different. In children's natural environment, high background noise levels are common and are an important factor contributing to voice problems. The primary noise source in day-care centers is the children themselves. The aim of the present study was to compare perceptual evaluations of voice quality and acoustic measures from a controlled recording with recordings of spontaneous speech in children's natural environment in a day-care setting. Eleven 5-year-old children were recorded three times during a day at the day care. The controlled speech material consisted of repeated sentences. Matching sentences were selected from the spontaneous speech. All sentences were repeated three times. Recordings were randomized and analyzed acoustically and perceptually. Statistic analyses showed that fundamental frequency was significantly higher in spontaneous speech (P<0.01) as was hyperfunction (P<0.001). The only characteristic the controlled sentences shared with spontaneous speech was degree of hoarseness (Spearman's rho=0.564). When data for boys and girls were analyzed separately, a correlation was found for the parameter breathiness (rho=0.551) for boys, and for girls the correlation for hoarseness remained (rho=0.752). Regarding acoustic data, none of the measures correlated across recording conditions for the whole group. Copyright © 2012 The Voice Foundation. Published by Mosby, Inc. All rights reserved.

Cortical Tracking of Global and Local Variations of Speech Rhythm during Connected Natural Speech Perception.

PubMed

Alexandrou, Anna Maria; Saarinen, Timo; Kujala, Jan; Salmelin, Riitta

2018-06-19

During natural speech perception, listeners must track the global speaking rate, that is, the overall rate of incoming linguistic information, as well as transient, local speaking rate variations occurring within the global speaking rate. Here, we address the hypothesis that this tracking mechanism is achieved through coupling of cortical signals to the amplitude envelope of the perceived acoustic speech signals. Cortical signals were recorded with magnetoencephalography (MEG) while participants perceived spontaneously produced speech stimuli at three global speaking rates (slow, normal/habitual, and fast). Inherently to spontaneously produced speech, these stimuli also featured local variations in speaking rate. The coupling between cortical and acoustic speech signals was evaluated using audio-MEG coherence. Modulations in audio-MEG coherence spatially differentiated between tracking of global speaking rate, highlighting the temporal cortex bilaterally and the right parietal cortex, and sensitivity to local speaking rate variations, emphasizing the left parietal cortex. Cortical tuning to the temporal structure of natural connected speech thus seems to require the joint contribution of both auditory and parietal regions. These findings suggest that cortical tuning to speech rhythm operates on two functionally distinct levels: one encoding the global rhythmic structure of speech and the other associated with online, rapidly evolving temporal predictions. Thus, it may be proposed that speech perception is shaped by evolutionary tuning, a preference for certain speaking rates, and predictive tuning, associated with cortical tracking of the constantly changing rate of linguistic information in a speech stream.

Predicting speech intelligibility with a multiple speech subsystems approach in children with cerebral palsy.

PubMed

Lee, Jimin; Hustad, Katherine C; Weismer, Gary

2014-10-01

Speech acoustic characteristics of children with cerebral palsy (CP) were examined with a multiple speech subsystems approach; speech intelligibility was evaluated using a prediction model in which acoustic measures were selected to represent three speech subsystems. Nine acoustic variables reflecting different subsystems, and speech intelligibility, were measured in 22 children with CP. These children included 13 with a clinical diagnosis of dysarthria (speech motor impairment [SMI] group) and 9 judged to be free of dysarthria (no SMI [NSMI] group). Data from children with CP were compared to data from age-matched typically developing children. Multiple acoustic variables reflecting the articulatory subsystem were different in the SMI group, compared to the NSMI and typically developing groups. A significant speech intelligibility prediction model was obtained with all variables entered into the model (adjusted R2 = .801). The articulatory subsystem showed the most substantial independent contribution (58%) to speech intelligibility. Incremental R2 analyses revealed that any single variable explained less than 9% of speech intelligibility variability. Children in the SMI group had articulatory subsystem problems as indexed by acoustic measures. As in the adult literature, the articulatory subsystem makes the primary contribution to speech intelligibility variance in dysarthria, with minimal or no contribution from other systems.

Separable spectro-temporal Gabor filter bank features: Reducing the complexity of robust features for automatic speech recognition.

PubMed

Schädler, Marc René; Kollmeier, Birger

2015-04-01

To test if simultaneous spectral and temporal processing is required to extract robust features for automatic speech recognition (ASR), the robust spectro-temporal two-dimensional-Gabor filter bank (GBFB) front-end from Schädler, Meyer, and Kollmeier [J. Acoust. Soc. Am. 131, 4134-4151 (2012)] was de-composed into a spectral one-dimensional-Gabor filter bank and a temporal one-dimensional-Gabor filter bank. A feature set that is extracted with these separate spectral and temporal modulation filter banks was introduced, the separate Gabor filter bank (SGBFB) features, and evaluated on the CHiME (Computational Hearing in Multisource Environments) keywords-in-noise recognition task. From the perspective of robust ASR, the results showed that spectral and temporal processing can be performed independently and are not required to interact with each other. Using SGBFB features permitted the signal-to-noise ratio (SNR) to be lowered by 1.2 dB while still performing as well as the GBFB-based reference system, which corresponds to a relative improvement of the word error rate by 12.8%. Additionally, the real time factor of the spectro-temporal processing could be reduced by more than an order of magnitude. Compared to human listeners, the SNR needed to be 13 dB higher when using Mel-frequency cepstral coefficient features, 11 dB higher when using GBFB features, and 9 dB higher when using SGBFB features to achieve the same recognition performance.

Acoustic analysis in Mudejar-Gothic churches: Experimental results

NASA Astrophysics Data System (ADS)

Galindo, Miguel; Zamarreño, Teófilo; Girón, Sara

2005-05-01

This paper describes the preliminary results of research work in acoustics, conducted in a set of 12 Mudejar-Gothic churches in the city of Seville in the south of Spain. Despite common architectural style, the churches feature individual characteristics and have volumes ranging from 3947 to 10 708 m3. Acoustic parameters were measured in unoccupied churches according to the ISO-3382 standard. An extensive experimental study was carried out using impulse response analysis through a maximum length sequence measurement system in each church. It covered aspects such as reverberation (reverberation times, early decay times), distribution of sound levels (sound strength); early to late sound energy parameters derived from the impulse responses (center time, clarity for speech, clarity, definition, lateral energy fraction), and speech intelligibility (rapid speech transmission index), which all take both spectral and spatial distribution into account. Background noise was also measured to obtain the NR indices. The study describes the acoustic field inside each temple and establishes a discussion for each one of the acoustic descriptors mentioned by using the theoretical models available and the principles of architectural acoustics. Analysis of the quality of the spaces for music and speech is carried out according to the most widespread criteria for auditoria. .

Acoustic analysis in Mudejar-Gothic churches: experimental results.

PubMed

Galindo, Miguel; Zamarreño, Teófilo; Girón, Sara

2005-05-01

This paper describes the preliminary results of research work in acoustics, conducted in a set of 12 Mudejar-Gothic churches in the city of Seville in the south of Spain. Despite common architectural style, the churches feature individual characteristics and have volumes ranging from 3947 to 10 708 m3. Acoustic parameters were measured in unoccupied churches according to the ISO-3382 standard. An extensive experimental study was carried out using impulse response analysis through a maximum length sequence measurement system in each church. It covered aspects such as reverberation (reverberation times, early decay times), distribution of sound levels (sound strength); early to late sound energy parameters derived from the impulse responses (center time, clarity for speech, clarity, definition, lateral energy fraction), and speech intelligibility (rapid speech transmission index), which all take both spectral and spatial distribution into account. Background noise was also measured to obtain the NR indices. The study describes the acoustic field inside each temple and establishes a discussion for each one of the acoustic descriptors mentioned by using the theoretical models available and the principles of architectural acoustics. Analysis of the quality of the spaces for music and speech is carried out according to the most widespread criteria for auditoria.

[Speech perception with electric-acoustic stimulation : Comparison with bilateral cochlear implant users in different noise conditions].

PubMed

Rader, T

2015-02-01

Cochlear implantation with the aim of hearing preservation for combined electric-acoustic stimulation (EAS) is the therapy of choice for patients with residual low-frequency hearing. Preserved residual acoustic hearing has a positive effect on speech intelligibility in difficult noise conditions. The goal of this study was to assess speech reception thresholds in various complex noise conditions for patients with EAS in comparison with patients using bilateral cochlear implants (CI). Speech perception in noise was measured for bilateral CI and EAS patient groups. A total of 22 listeners with normal hearing served as a control group. Speech reception thresholds (SRT) were measured using a closed-set sentence matrix test. Speech was presented with a single source in frontal position; noise was presented in frontal position or in a multisource noise field (MSNF) consisting of a four-loudspeaker array with independent noise sources. Modulated speech-simulating noise and pseudocontinuous noise served respectively as interference signal with different temporal characteristics. The average SRTs in the EAS group were significantly better in all test conditions than those of the group with bilateral CI. Both user groups showed significant improvement in the MSNF condition compared with the frontal noise condition as a result of bilateral interaction. The normal-hearing control group was able to use short temporal gaps in modulated noise to improve speech perception in noise (gap listening). This effect was absent in both implanted user groups. Patients with combined EAS in one ear and a hearing aid in the contralateral ear show significantly improved speech perception in complex noise conditions compared with bilateral CI recipients.

Discriminative analysis of lip motion features for speaker identification and speech-reading.

PubMed

Cetingül, H Ertan; Yemez, Yücel; Erzin, Engin; Tekalp, A Murat

2006-10-01

There have been several studies that jointly use audio, lip intensity, and lip geometry information for speaker identification and speech-reading applications. This paper proposes using explicit lip motion information, instead of or in addition to lip intensity and/or geometry information, for speaker identification and speech-reading within a unified feature selection and discrimination analysis framework, and addresses two important issues: 1) Is using explicit lip motion information useful, and, 2) if so, what are the best lip motion features for these two applications? The best lip motion features for speaker identification are considered to be those that result in the highest discrimination of individual speakers in a population, whereas for speech-reading, the best features are those providing the highest phoneme/word/phrase recognition rate. Several lip motion feature candidates have been considered including dense motion features within a bounding box about the lip, lip contour motion features, and combination of these with lip shape features. Furthermore, a novel two-stage, spatial, and temporal discrimination analysis is introduced to select the best lip motion features for speaker identification and speech-reading applications. Experimental results using an hidden-Markov-model-based recognition system indicate that using explicit lip motion information provides additional performance gains in both applications, and lip motion features prove more valuable in the case of speech-reading application.

Predicting Speech Intelligibility with A Multiple Speech Subsystems Approach in Children with Cerebral Palsy

PubMed Central

Lee, Jimin; Hustad, Katherine C.; Weismer, Gary

2014-01-01

Purpose Speech acoustic characteristics of children with cerebral palsy (CP) were examined with a multiple speech subsystem approach; speech intelligibility was evaluated using a prediction model in which acoustic measures were selected to represent three speech subsystems. Method Nine acoustic variables reflecting different subsystems, and speech intelligibility, were measured in 22 children with CP. These children included 13 with a clinical diagnosis of dysarthria (SMI), and nine judged to be free of dysarthria (NSMI). Data from children with CP were compared to data from age-matched typically developing children (TD). Results Multiple acoustic variables reflecting the articulatory subsystem were different in the SMI group, compared to the NSMI and TD groups. A significant speech intelligibility prediction model was obtained with all variables entered into the model (Adjusted R-squared = .801). The articulatory subsystem showed the most substantial independent contribution (58%) to speech intelligibility. Incremental R-squared analyses revealed that any single variable explained less than 9% of speech intelligibility variability. Conclusions Children in the SMI group have articulatory subsystem problems as indexed by acoustic measures. As in the adult literature, the articulatory subsystem makes the primary contribution to speech intelligibility variance in dysarthria, with minimal or no contribution from other systems. PMID:24824584

Audio-video feature correlation: faces and speech

NASA Astrophysics Data System (ADS)

Durand, Gwenael; Montacie, Claude; Caraty, Marie-Jose; Faudemay, Pascal

1999-08-01

This paper presents a study of the correlation of features automatically extracted from the audio stream and the video stream of audiovisual documents. In particular, we were interested in finding out whether speech analysis tools could be combined with face detection methods, and to what extend they should be combined. A generic audio signal partitioning algorithm as first used to detect Silence/Noise/Music/Speech segments in a full length movie. A generic object detection method was applied to the keyframes extracted from the movie in order to detect the presence or absence of faces. The correlation between the presence of a face in the keyframes and of the corresponding voice in the audio stream was studied. A third stream, which is the script of the movie, is warped on the speech channel in order to automatically label faces appearing in the keyframes with the name of the corresponding character. We naturally found that extracted audio and video features were related in many cases, and that significant benefits can be obtained from the joint use of audio and video analysis methods.

Detection of Clinical Depression in Adolescents’ Speech During Family Interactions

PubMed Central

Low, Lu-Shih Alex; Maddage, Namunu C.; Lech, Margaret; Sheeber, Lisa B.; Allen, Nicholas B.

2013-01-01

The properties of acoustic speech have previously been investigated as possible cues for depression in adults. However, these studies were restricted to small populations of patients and the speech recordings were made during patients’ clinical interviews or fixed-text reading sessions. Symptoms of depression often first appear during adolescence at a time when the voice is changing, in both males and females, suggesting that specific studies of these phenomena in adolescent populations are warranted. This study investigated acoustic correlates of depression in a large sample of 139 adolescents (68 clinically depressed and 71 controls). Speech recordings were made during naturalistic interactions between adolescents and their parents. Prosodic, cepstral, spectral, and glottal features, as well as features derived from the Teager energy operator (TEO), were tested within a binary classification framework. Strong gender differences in classification accuracy were observed. The TEO-based features clearly outperformed all other features and feature combinations, providing classification accuracy ranging between 81%–87% for males and 72%–79% for females. Close, but slightly less accurate, results were obtained by combining glottal features with prosodic and spectral features (67%–69% for males and 70%–75% for females). These findings indicate the importance of nonlinear mechanisms associated with the glottal flow formation as cues for clinical depression. PMID:21075715

Dimensional feature weighting utilizing multiple kernel learning for single-channel talker location discrimination using the acoustic transfer function.

PubMed

Takashima, Ryoichi; Takiguchi, Tetsuya; Ariki, Yasuo

2013-02-01

This paper presents a method for discriminating the location of the sound source (talker) using only a single microphone. In a previous work, the single-channel approach for discriminating the location of the sound source was discussed, where the acoustic transfer function from a user's position is estimated by using a hidden Markov model of clean speech in the cepstral domain. In this paper, each cepstral dimension of the acoustic transfer function is newly weighted, in order to obtain the cepstral dimensions having information that is useful for classifying the user's position. Then, this paper proposes a feature-weighting method for the cepstral parameter using multiple kernel learning, defining the base kernels for each cepstral dimension of the acoustic transfer function. The user's position is trained and classified by support vector machine. The effectiveness of this method has been confirmed by sound source (talker) localization experiments performed in different room environments.

[Prosody, speech input and language acquisition].

PubMed

Jungheim, M; Miller, S; Kühn, D; Ptok, M

2014-04-01

In order to acquire language, children require speech input. The prosody of the speech input plays an important role. In most cultures adults modify their code when communicating with children. Compared to normal speech this code differs especially with regard to prosody. For this review a selective literature search in PubMed and Scopus was performed. Prosodic characteristics are a key feature of spoken language. By analysing prosodic features, children gain knowledge about underlying grammatical structures. Child-directed speech (CDS) is modified in a way that meaningful sequences are highlighted acoustically so that important information can be extracted from the continuous speech flow more easily. CDS is said to enhance the representation of linguistic signs. Taking into consideration what has previously been described in the literature regarding the perception of suprasegmentals, CDS seems to be able to support language acquisition due to the correspondence of prosodic and syntactic units. However, no findings have been reported, stating that the linguistically reduced CDS could hinder first language acquisition.

The cortical representation of the speech envelope is earlier for audiovisual speech than audio speech.

PubMed

Crosse, Michael J; Lalor, Edmund C

2014-04-01

Visual speech can greatly enhance a listener's comprehension of auditory speech when they are presented simultaneously. Efforts to determine the neural underpinnings of this phenomenon have been hampered by the limited temporal resolution of hemodynamic imaging and the fact that EEG and magnetoencephalographic data are usually analyzed in response to simple, discrete stimuli. Recent research has shown that neuronal activity in human auditory cortex tracks the envelope of natural speech. Here, we exploit this finding by estimating a linear forward-mapping between the speech envelope and EEG data and show that the latency at which the envelope of natural speech is represented in cortex is shortened by >10 ms when continuous audiovisual speech is presented compared with audio-only speech. In addition, we use a reverse-mapping approach to reconstruct an estimate of the speech stimulus from the EEG data and, by comparing the bimodal estimate with the sum of the unimodal estimates, find no evidence of any nonlinear additive effects in the audiovisual speech condition. These findings point to an underlying mechanism that could account for enhanced comprehension during audiovisual speech. Specifically, we hypothesize that low-level acoustic features that are temporally coherent with the preceding visual stream may be synthesized into a speech object at an earlier latency, which may provide an extended period of low-level processing before extraction of semantic information.

Real-Time Speech/Music Classification With a Hierarchical Oblique Decision Tree

DTIC Science & Technology

2008-04-01

REAL-TIME SPEECH/ MUSIC CLASSIFICATION WITH A HIERARCHICAL OBLIQUE DECISION TREE Jun Wang, Qiong Wu, Haojiang Deng, Qin Yan Institute of Acoustics...time speech/ music classification with a hierarchical oblique decision tree. A set of discrimination features in frequency domain are selected...handle signals without discrimination and can not work properly in the existence of multimedia signals. This paper proposes a real-time speech/ music

Segregation of Whispered Speech Interleaved with Noise or Speech Maskers

DTIC Science & Technology

2011-08-01

range over which the talker can be heard. Whispered speech is produced by modulating the flow of air through partially open vocal folds. Because the...source of excitation is turbulent air flow , the acoustic characteristics of whispered speech differs from voiced speech [1, 2]. Despite the acoustic...signals provided by cochlear implants. Two studies investigated the segregation of simultaneously presented whispered vowels [7, 8] in a standard

A screening approach for classroom acoustics using web-based listening tests and subjective ratings.

PubMed

Persson Waye, Kerstin; Magnusson, Lennart; Fredriksson, Sofie; Croy, Ilona

2015-01-01

Perception of speech is crucial in school where speech is the main mode of communication. The aim of the study was to evaluate whether a web based approach including listening tests and questionnaires could be used as a screening tool for poor classroom acoustics. The prime focus was the relation between pupils' comprehension of speech, the classroom acoustics and their description of the acoustic qualities of the classroom. In total, 1106 pupils aged 13-19, from 59 classes and 38 schools in Sweden participated in a listening study using Hagerman's sentences administered via Internet. Four listening conditions were applied: high and low background noise level and positions close and far away from the loudspeaker. The pupils described the acoustic quality of the classroom and teachers provided information on the physical features of the classroom using questionnaires. In 69% of the classes, at least three pupils described the sound environment as adverse and in 88% of the classes one or more pupil reported often having difficulties concentrating due to noise. The pupils' comprehension of speech was strongly influenced by the background noise level (p<0.001) and distance to the loudspeakers (p<0.001). Of the physical classroom features, presence of suspended acoustic panels (p<0.05) and length of the classroom (p<0.01) predicted speech comprehension. Of the pupils' descriptions of acoustic qualities, clattery significantly (p<0.05) predicted speech comprehension. Clattery was furthermore associated to difficulties understanding each other, while the description noisy was associated to concentration difficulties. The majority of classrooms do not seem to have an optimal sound environment. The pupil's descriptions of acoustic qualities and listening tests can be one way of predicting sound conditions in the classroom.

Acoustic differences between humorous and sincere communicative intentions.

PubMed

Hoicka, Elena; Gattis, Merideth

2012-11-01

Previous studies indicate that the acoustic features of speech discriminate between positive and negative communicative intentions, such as approval and prohibition. Two studies investigated whether acoustic features of speech can discriminate between two positive communicative intentions: humour and sweet-sincerity, where sweet-sincerity involved being sincere in a positive, warm-hearted way. In Study 1, 22 mothers read a book containing humorous, sweet-sincere, and neutral-sincere images to their 19- to 24-month-olds. In Study 2, 41 mothers read a book containing humorous or sweet-sincere sentences and images to their 18- to 24-month-olds. Mothers used a higher mean F0 to communicate visual humour as compared to visual sincerity. Mothers used greater F0 mean, range, and standard deviation; greater intensity mean, range, and standard deviation; and a slower speech rate to communicate verbal humour as compared to verbal sweet-sincerity. Mothers used a rising linear contour to communicate verbal humour, but used no specific contour to express verbal sweet-sincerity. We conclude that speakers provide acoustic cues enabling listeners to distinguish between positive communicative intentions. ©2011 The British Psychological Society.

LANDMARK-BASED SPEECH RECOGNITION: REPORT OF THE 2004 JOHNS HOPKINS SUMMER WORKSHOP.

PubMed

Hasegawa-Johnson, Mark; Baker, James; Borys, Sarah; Chen, Ken; Coogan, Emily; Greenberg, Steven; Juneja, Amit; Kirchhoff, Katrin; Livescu, Karen; Mohan, Srividya; Muller, Jennifer; Sonmez, Kemal; Wang, Tianyu

2005-01-01

Three research prototype speech recognition systems are described, all of which use recently developed methods from artificial intelligence (specifically support vector machines, dynamic Bayesian networks, and maximum entropy classification) in order to implement, in the form of an automatic speech recognizer, current theories of human speech perception and phonology (specifically landmark-based speech perception, nonlinear phonology, and articulatory phonology). All three systems begin with a high-dimensional multiframe acoustic-to-distinctive feature transformation, implemented using support vector machines trained to detect and classify acoustic phonetic landmarks. Distinctive feature probabilities estimated by the support vector machines are then integrated using one of three pronunciation models: a dynamic programming algorithm that assumes canonical pronunciation of each word, a dynamic Bayesian network implementation of articulatory phonology, or a discriminative pronunciation model trained using the methods of maximum entropy classification. Log probability scores computed by these models are then combined, using log-linear combination, with other word scores available in the lattice output of a first-pass recognizer, and the resulting combination score is used to compute a second-pass speech recognition output.

Quantitative acoustic measurements for characterization of speech and voice disorders in early untreated Parkinson's disease.

PubMed

Rusz, J; Cmejla, R; Ruzickova, H; Ruzicka, E

2011-01-01

An assessment of vocal impairment is presented for separating healthy people from persons with early untreated Parkinson's disease (PD). This study's main purpose was to (a) determine whether voice and speech disorder are present from early stages of PD before starting dopaminergic pharmacotherapy, (b) ascertain the specific characteristics of the PD-related vocal impairment, (c) identify PD-related acoustic signatures for the major part of traditional clinically used measurement methods with respect to their automatic assessment, and (d) design new automatic measurement methods of articulation. The varied speech data were collected from 46 Czech native speakers, 23 with PD. Subsequently, 19 representative measurements were pre-selected, and Wald sequential analysis was then applied to assess the efficiency of each measure and the extent of vocal impairment of each subject. It was found that measurement of the fundamental frequency variations applied to two selected tasks was the best method for separating healthy from PD subjects. On the basis of objective acoustic measures, statistical decision-making theory, and validation from practicing speech therapists, it has been demonstrated that 78% of early untreated PD subjects indicate some form of vocal impairment. The speech defects thus uncovered differ individually in various characteristics including phonation, articulation, and prosody.

Shared acoustic codes underlie emotional communication in music and speech—Evidence from deep transfer learning

PubMed Central

Schuller, Björn

2017-01-01

Music and speech exhibit striking similarities in the communication of emotions in the acoustic domain, in such a way that the communication of specific emotions is achieved, at least to a certain extent, by means of shared acoustic patterns. From an Affective Sciences points of view, determining the degree of overlap between both domains is fundamental to understand the shared mechanisms underlying such phenomenon. From a Machine learning perspective, the overlap between acoustic codes for emotional expression in music and speech opens new possibilities to enlarge the amount of data available to develop music and speech emotion recognition systems. In this article, we investigate time-continuous predictions of emotion (Arousal and Valence) in music and speech, and the Transfer Learning between these domains. We establish a comparative framework including intra- (i.e., models trained and tested on the same modality, either music or speech) and cross-domain experiments (i.e., models trained in one modality and tested on the other). In the cross-domain context, we evaluated two strategies—the direct transfer between domains, and the contribution of Transfer Learning techniques (feature-representation-transfer based on Denoising Auto Encoders) for reducing the gap in the feature space distributions. Our results demonstrate an excellent cross-domain generalisation performance with and without feature representation transfer in both directions. In the case of music, cross-domain approaches outperformed intra-domain models for Valence estimation, whereas for Speech intra-domain models achieve the best performance. This is the first demonstration of shared acoustic codes for emotional expression in music and speech in the time-continuous domain. PMID:28658285

Factors Affecting Acoustics and Speech Intelligibility in the Operating Room: Size Matters.

PubMed

McNeer, Richard R; Bennett, Christopher L; Horn, Danielle Bodzin; Dudaryk, Roman

2017-06-01

Noise in health care settings has increased since 1960 and represents a significant source of dissatisfaction among staff and patients and risk to patient safety. Operating rooms (ORs) in which effective communication is crucial are particularly noisy. Speech intelligibility is impacted by noise, room architecture, and acoustics. For example, sound reverberation time (RT60) increases with room size, which can negatively impact intelligibility, while room objects are hypothesized to have the opposite effect. We explored these relationships by investigating room construction and acoustics of the surgical suites at our institution. We studied our ORs during times of nonuse. Room dimensions were measured to calculate room volumes (VR). Room content was assessed by estimating size and assigning items into 5 volume categories to arrive at an adjusted room content volume (VC) metric. Psychoacoustic analyses were performed by playing sweep tones from a speaker and recording the impulse responses (ie, resulting sound fields) from 3 locations in each room. The recordings were used to calculate 6 psychoacoustic indices of intelligibility. Multiple linear regression was performed using VR and VC as predictor variables and each intelligibility index as an outcome variable. A total of 40 ORs were studied. The surgical suites were characterized by a large degree of construction and surface finish heterogeneity and varied in size from 71.2 to 196.4 m (average VR = 131.1 [34.2] m). An insignificant correlation was observed between VR and VC (Pearson correlation = 0.223, P = .166). Multiple linear regression model fits and β coefficients for VR were highly significant for each of the intelligibility indices and were best for RT60 (R = 0.666, F(2, 37) = 39.9, P < .0001). For Dmax (maximum distance where there is <15% loss of consonant articulation), both VR and VC β coefficients were significant. For RT60 and Dmax, after controlling for VC, partial correlations were 0.825 (P

Factors Affecting Acoustics and Speech Intelligibility in the Operating Room: Size Matters

PubMed Central

Bennett, Christopher L.; Horn, Danielle Bodzin; Dudaryk, Roman

2017-01-01

INTRODUCTION: Noise in health care settings has increased since 1960 and represents a significant source of dissatisfaction among staff and patients and risk to patient safety. Operating rooms (ORs) in which effective communication is crucial are particularly noisy. Speech intelligibility is impacted by noise, room architecture, and acoustics. For example, sound reverberation time (RT60) increases with room size, which can negatively impact intelligibility, while room objects are hypothesized to have the opposite effect. We explored these relationships by investigating room construction and acoustics of the surgical suites at our institution. METHODS: We studied our ORs during times of nonuse. Room dimensions were measured to calculate room volumes (VR). Room content was assessed by estimating size and assigning items into 5 volume categories to arrive at an adjusted room content volume (VC) metric. Psychoacoustic analyses were performed by playing sweep tones from a speaker and recording the impulse responses (ie, resulting sound fields) from 3 locations in each room. The recordings were used to calculate 6 psychoacoustic indices of intelligibility. Multiple linear regression was performed using VR and VC as predictor variables and each intelligibility index as an outcome variable. RESULTS: A total of 40 ORs were studied. The surgical suites were characterized by a large degree of construction and surface finish heterogeneity and varied in size from 71.2 to 196.4 m3 (average VR = 131.1 [34.2] m3). An insignificant correlation was observed between VR and VC (Pearson correlation = 0.223, P = .166). Multiple linear regression model fits and β coefficients for VR were highly significant for each of the intelligibility indices and were best for RT60 (R2 = 0.666, F(2, 37) = 39.9, P < .0001). For Dmax (maximum distance where there is <15% loss of consonant articulation), both VR and VC β coefficients were significant. For RT60 and Dmax, after controlling for VC

Effect of body position on vocal tract acoustics: Acoustic pharyngometry and vowel formants.

PubMed

Vorperian, Houri K; Kurtzweil, Sara L; Fourakis, Marios; Kent, Ray D; Tillman, Katelyn K; Austin, Diane

2015-08-01

The anatomic basis and articulatory features of speech production are often studied with imaging studies that are typically acquired in the supine body position. It is important to determine if changes in body orientation to the gravitational field alter vocal tract dimensions and speech acoustics. The purpose of this study was to assess the effect of body position (upright versus supine) on (1) oral and pharyngeal measurements derived from acoustic pharyngometry and (2) acoustic measurements of fundamental frequency (F0) and the first four formant frequencies (F1-F4) for the quadrilateral point vowels. Data were obtained for 27 male and female participants, aged 17 to 35 yrs. Acoustic pharyngometry showed a statistically significant effect of body position on volumetric measurements, with smaller values in the supine than upright position, but no changes in length measurements. Acoustic analyses of vowels showed significantly larger values in the supine than upright position for the variables of F0, F3, and the Euclidean distance from the centroid to each corner vowel in the F1-F2-F3 space. Changes in body position affected measurements of vocal tract volume but not length. Body position also affected the aforementioned acoustic variables, but the main vowel formants were preserved.

[Combining speech sample and feature bilateral selection algorithm for classification of Parkinson's disease].

PubMed

Zhang, Xiaoheng; Wang, Lirui; Cao, Yao; Wang, Pin; Zhang, Cheng; Yang, Liuyang; Li, Yongming; Zhang, Yanling; Cheng, Oumei

2018-02-01

Diagnosis of Parkinson's disease (PD) based on speech data has been proved to be an effective way in recent years. However, current researches just care about the feature extraction and classifier design, and do not consider the instance selection. Former research by authors showed that the instance selection can lead to improvement on classification accuracy. However, no attention is paid on the relationship between speech sample and feature until now. Therefore, a new diagnosis algorithm of PD is proposed in this paper by simultaneously selecting speech sample and feature based on relevant feature weighting algorithm and multiple kernel method, so as to find their synergy effects, thereby improving classification accuracy. Experimental results showed that this proposed algorithm obtained apparent improvement on classification accuracy. It can obtain mean classification accuracy of 82.5%, which was 30.5% higher than the relevant algorithm. Besides, the proposed algorithm detected the synergy effects of speech sample and feature, which is valuable for speech marker extraction.

Neural Oscillations Carry Speech Rhythm through to Comprehension

PubMed Central

Peelle, Jonathan E.; Davis, Matthew H.

2012-01-01

A key feature of speech is the quasi-regular rhythmic information contained in its slow amplitude modulations. In this article we review the information conveyed by speech rhythm, and the role of ongoing brain oscillations in listeners’ processing of this content. Our starting point is the fact that speech is inherently temporal, and that rhythmic information conveyed by the amplitude envelope contains important markers for place and manner of articulation, segmental information, and speech rate. Behavioral studies demonstrate that amplitude envelope information is relied upon by listeners and plays a key role in speech intelligibility. Extending behavioral findings, data from neuroimaging – particularly electroencephalography (EEG) and magnetoencephalography (MEG) – point to phase locking by ongoing cortical oscillations to low-frequency information (~4–8 Hz) in the speech envelope. This phase modulation effectively encodes a prediction of when important events (such as stressed syllables) are likely to occur, and acts to increase sensitivity to these relevant acoustic cues. We suggest a framework through which such neural entrainment to speech rhythm can explain effects of speech rate on word and segment perception (i.e., that the perception of phonemes and words in connected speech is influenced by preceding speech rate). Neuroanatomically, acoustic amplitude modulations are processed largely bilaterally in auditory cortex, with intelligible speech resulting in differential recruitment of left-hemisphere regions. Notable among these is lateral anterior temporal cortex, which we propose functions in a domain-general fashion to support ongoing memory and integration of meaningful input. Together, the reviewed evidence suggests that low-frequency oscillations in the acoustic speech signal form the foundation of a rhythmic hierarchy supporting spoken language, mirrored by phase-locked oscillations in the human brain. PMID:22973251

Obstructive sleep apnea severity estimation: Fusion of speech-based systems.

PubMed

Ben Or, D; Dafna, E; Tarasiuk, A; Zigel, Y

2016-08-01

Obstructive sleep apnea (OSA) is a common sleep-related breathing disorder. Previous studies associated OSA with anatomical abnormalities of the upper respiratory tract that may be reflected in the acoustic characteristics of speech. We tested the hypothesis that the speech signal carries essential information that can assist in early assessment of OSA severity by estimating apnea-hypopnea index (AHI). 198 men referred to routine polysomnography (PSG) were recorded shortly prior to sleep onset while reading a one-minute speech protocol. The different parts of the speech recordings, i.e., sustained vowels, short-time frames of fluent speech, and the speech recording as a whole, underwent separate analyses, using sustained vowels features, short-term features, and long-term features, respectively. Applying support vector regression and regression trees, these features were used in order to estimate AHI. The fusion of the outputs of the three subsystems resulted in a diagnostic agreement of 67.3% between the speech-estimated AHI and the PSG-determined AHI, and an absolute error rate of 10.8 events/hr. Speech signal analysis may assist in the estimation of AHI, thus allowing the development of a noninvasive tool for OSA screening.

Revisiting Neil Armstrongs Moon-Landing Quote: Implications for Speech Perception, Function Word Reduction, and Acoustic Ambiguity

PubMed Central

Baese-Berk, Melissa M.; Dilley, Laura C.; Schmidt, Stephanie; Morrill, Tuuli H.; Pitt, Mark A.

2016-01-01

Neil Armstrong insisted that his quote upon landing on the moon was misheard, and that he had said one small step for a man, instead of one small step for man. What he said is unclear in part because function words like a can be reduced and spectrally indistinguishable from the preceding context. Therefore, their presence can be ambiguous, and they may disappear perceptually depending on the rate of surrounding speech. Two experiments are presented examining production and perception of reduced tokens of for and for a in spontaneous speech. Experiment 1 investigates the distributions of several acoustic features of for and for a. The results suggest that the distributions of for and for a overlap substantially, both in terms of temporal and spectral characteristics. Experiment 2 examines perception of these same tokens when the context speaking rate differs. The perceptibility of the function word a varies as a function of this context speaking rate. These results demonstrate that substantial ambiguity exists in the original quote from Armstrong, and that this ambiguity may be understood through context speaking rate. PMID:27603209

Revisiting Neil Armstrongs Moon-Landing Quote: Implications for Speech Perception, Function Word Reduction, and Acoustic Ambiguity.

PubMed

Baese-Berk, Melissa M; Dilley, Laura C; Schmidt, Stephanie; Morrill, Tuuli H; Pitt, Mark A

2016-01-01

Neil Armstrong insisted that his quote upon landing on the moon was misheard, and that he had said one small step for a man, instead of one small step for man. What he said is unclear in part because function words like a can be reduced and spectrally indistinguishable from the preceding context. Therefore, their presence can be ambiguous, and they may disappear perceptually depending on the rate of surrounding speech. Two experiments are presented examining production and perception of reduced tokens of for and for a in spontaneous speech. Experiment 1 investigates the distributions of several acoustic features of for and for a. The results suggest that the distributions of for and for a overlap substantially, both in terms of temporal and spectral characteristics. Experiment 2 examines perception of these same tokens when the context speaking rate differs. The perceptibility of the function word a varies as a function of this context speaking rate. These results demonstrate that substantial ambiguity exists in the original quote from Armstrong, and that this ambiguity may be understood through context speaking rate.

Speech feature discrimination in deaf children following cochlear implantation

NASA Astrophysics Data System (ADS)

Bergeson, Tonya R.; Pisoni, David B.; Kirk, Karen Iler

2002-05-01

Speech feature discrimination is a fundamental perceptual skill that is often assumed to underlie word recognition and sentence comprehension performance. To investigate the development of speech feature discrimination in deaf children with cochlear implants, we conducted a retrospective analysis of results from the Minimal Pairs Test (Robbins et al., 1988) selected from patients enrolled in a longitudinal study of speech perception and language development. The MP test uses a 2AFC procedure in which children hear a word and select one of two pictures (bat-pat). All 43 children were prelingually deafened, received a cochlear implant before 6 years of age or between ages 6 and 9, and used either oral or total communication. Children were tested once every 6 months to 1 year for 7 years; not all children were tested at each interval. By 2 years postimplant, the majority of these children achieved near-ceiling levels of discrimination performance for vowel height, vowel place, and consonant manner. Most of the children also achieved plateaus but did not reach ceiling performance for consonant place and voicing. The relationship between speech feature discrimination, spoken word recognition, and sentence comprehension will be discussed. [Work supported by NIH/NIDCD Research Grant No. R01DC00064 and NIH/NIDCD Training Grant No. T32DC00012.

Common cues to emotion in the dynamic facial expressions of speech and song.

PubMed

Livingstone, Steven R; Thompson, William F; Wanderley, Marcelo M; Palmer, Caroline

2015-01-01

Speech and song are universal forms of vocalization that may share aspects of emotional expression. Research has focused on parallels in acoustic features, overlooking facial cues to emotion. In three experiments, we compared moving facial expressions in speech and song. In Experiment 1, vocalists spoke and sang statements each with five emotions. Vocalists exhibited emotion-dependent movements of the eyebrows and lip corners that transcended speech-song differences. Vocalists' jaw movements were coupled to their acoustic intensity, exhibiting differences across emotion and speech-song. Vocalists' emotional movements extended beyond vocal sound to include large sustained expressions, suggesting a communicative function. In Experiment 2, viewers judged silent videos of vocalists' facial expressions prior to, during, and following vocalization. Emotional intentions were identified accurately for movements during and after vocalization, suggesting that these movements support the acoustic message. Experiment 3 compared emotional identification in voice-only, face-only, and face-and-voice recordings. Emotion judgements for voice-only singing were poorly identified, yet were accurate for all other conditions, confirming that facial expressions conveyed emotion more accurately than the voice in song, yet were equivalent in speech. Collectively, these findings highlight broad commonalities in the facial cues to emotion in speech and song, yet highlight differences in perception and acoustic-motor production.

Exploiting Acoustic and Syntactic Features for Automatic Prosody Labeling in a Maximum Entropy Framework

PubMed Central

Sridhar, Vivek Kumar Rangarajan; Bangalore, Srinivas; Narayanan, Shrikanth S.

2009-01-01

In this paper, we describe a maximum entropy-based automatic prosody labeling framework that exploits both language and speech information. We apply the proposed framework to both prominence and phrase structure detection within the Tones and Break Indices (ToBI) annotation scheme. Our framework utilizes novel syntactic features in the form of supertags and a quantized acoustic–prosodic feature representation that is similar to linear parameterizations of the prosodic contour. The proposed model is trained discriminatively and is robust in the selection of appropriate features for the task of prosody detection. The proposed maximum entropy acoustic–syntactic model achieves pitch accent and boundary tone detection accuracies of 86.0% and 93.1% on the Boston University Radio News corpus, and, 79.8% and 90.3% on the Boston Directions corpus. The phrase structure detection through prosodic break index labeling provides accuracies of 84% and 87% on the two corpora, respectively. The reported results are significantly better than previously reported results and demonstrate the strength of maximum entropy model in jointly modeling simple lexical, syntactic, and acoustic features for automatic prosody labeling. PMID:19603083

Acoustic features of objects matched by an echolocating bottlenose dolphin.

PubMed

Delong, Caroline M; Au, Whitlow W L; Lemonds, David W; Harley, Heidi E; Roitblat, Herbert L

2006-03-01

The focus of this study was to investigate how dolphins use acoustic features in returning echolocation signals to discriminate among objects. An echolocating dolphin performed a match-to-sample task with objects that varied in size, shape, material, and texture. After the task was completed, the features of the object echoes were measured (e.g., target strength, peak frequency). The dolphin's error patterns were examined in conjunction with the between-object variation in acoustic features to identify the acoustic features that the dolphin used to discriminate among the objects. The present study explored two hypotheses regarding the way dolphins use acoustic information in echoes: (1) use of a single feature, or (2) use of a linear combination of multiple features. The results suggested that dolphins do not use a single feature across all object sets or a linear combination of six echo features. Five features appeared to be important to the dolphin on four or more sets: the echo spectrum shape, the pattern of changes in target strength and number of highlights as a function of object orientation, and peak and center frequency. These data suggest that dolphins use multiple features and integrate information across echoes from a range of object orientations.

Recognizing Whispered Speech Produced by an Individual with Surgically Reconstructed Larynx Using Articulatory Movement Data

PubMed Central

Cao, Beiming; Kim, Myungjong; Mau, Ted; Wang, Jun

2017-01-01

Individuals with larynx (vocal folds) impaired have problems in controlling their glottal vibration, producing whispered speech with extreme hoarseness. Standard automatic speech recognition using only acoustic cues is typically ineffective for whispered speech because the corresponding spectral characteristics are distorted. Articulatory cues such as the tongue and lip motion may help in recognizing whispered speech since articulatory motion patterns are generally not affected. In this paper, we investigated whispered speech recognition for patients with reconstructed larynx using articulatory movement data. A data set with both acoustic and articulatory motion data was collected from a patient with surgically reconstructed larynx using an electromagnetic articulograph. Two speech recognition systems, Gaussian mixture model-hidden Markov model (GMM-HMM) and deep neural network-HMM (DNN-HMM), were used in the experiments. Experimental results showed adding either tongue or lip motion data to acoustic features such as mel-frequency cepstral coefficient (MFCC) significantly reduced the phone error rates on both speech recognition systems. Adding both tongue and lip data achieved the best performance. PMID:29423453

Rapid tuning shifts in human auditory cortex enhance speech intelligibility

PubMed Central

Holdgraf, Christopher R.; de Heer, Wendy; Pasley, Brian; Rieger, Jochem; Crone, Nathan; Lin, Jack J.; Knight, Robert T.; Theunissen, Frédéric E.

2016-01-01

Experience shapes our perception of the world on a moment-to-moment basis. This robust perceptual effect of experience parallels a change in the neural representation of stimulus features, though the nature of this representation and its plasticity are not well-understood. Spectrotemporal receptive field (STRF) mapping describes the neural response to acoustic features, and has been used to study contextual effects on auditory receptive fields in animal models. We performed a STRF plasticity analysis on electrophysiological data from recordings obtained directly from the human auditory cortex. Here, we report rapid, automatic plasticity of the spectrotemporal response of recorded neural ensembles, driven by previous experience with acoustic and linguistic information, and with a neurophysiological effect in the sub-second range. This plasticity reflects increased sensitivity to spectrotemporal features, enhancing the extraction of more speech-like features from a degraded stimulus and providing the physiological basis for the observed ‘perceptual enhancement' in understanding speech. PMID:27996965

Acoustic analysis of speech variables during depression and after improvement.

PubMed

Nilsonne, A

1987-09-01

Speech recordings were made of 16 depressed patients during depression and after clinical improvement. The recordings were analyzed using a computer program which extracts acoustic parameters from the fundamental frequency contour of the voice. The percent pause time, the standard deviation of the voice fundamental frequency distribution, the standard deviation of the rate of change of the voice fundamental frequency and the average speed of voice change were found to correlate to the clinical state of the patient. The mean fundamental frequency, the total reading time and the average rate of change of the voice fundamental frequency did not differ between the depressed and the improved group. The acoustic measures were more strongly correlated to the clinical state of the patient as measured by global depression scores than to single depressive symptoms such as retardation or agitation.

Speech Acquisition and Automatic Speech Recognition for Integrated Spacesuit Audio Systems

NASA Technical Reports Server (NTRS)

Huang, Yiteng; Chen, Jingdong; Chen, Shaoyan

2010-01-01

A voice-command human-machine interface system has been developed for spacesuit extravehicular activity (EVA) missions. A multichannel acoustic signal processing method has been created for distant speech acquisition in noisy and reverberant environments. This technology reduces noise by exploiting differences in the statistical nature of signal (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, the automatic speech recognition (ASR) accuracy can be improved to the level at which crewmembers would find the speech interface useful. The developed speech human/machine interface will enable both crewmember usability and operational efficiency. It can enjoy a fast rate of data/text entry, small overall size, and can be lightweight. In addition, this design will free the hands and eyes of a suited crewmember. The system components and steps include beam forming/multi-channel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, model adaption, ASR HMM (Hidden Markov Model) training, and ASR decoding. A state-of-the-art phoneme recognizer can obtain an accuracy rate of 65 percent when the training and testing data are free of noise. When it is used in spacesuits, the rate drops to about 33 percent. With the developed microphone array speech-processing technologies, the performance is improved and the phoneme recognition accuracy rate rises to 44 percent. The recognizer can be further improved by combining the microphone array and HMM model adaptation techniques and using speech samples collected from inside spacesuits. In addition, arithmetic complexity models for the major HMMbased ASR components were developed. They can help real-time ASR system designers select proper tasks when in the face of constraints in computational resources.

Deep bottleneck features for spoken language identification.

PubMed

Jiang, Bing; Song, Yan; Wei, Si; Liu, Jun-Hua; McLoughlin, Ian Vince; Dai, Li-Rong

2014-01-01

A key problem in spoken language identification (LID) is to design effective representations which are specific to language information. For example, in recent years, representations based on both phonotactic and acoustic features have proven their effectiveness for LID. Although advances in machine learning have led to significant improvements, LID performance is still lacking, especially for short duration speech utterances. With the hypothesis that language information is weak and represented only latently in speech, and is largely dependent on the statistical properties of the speech content, existing representations may be insufficient. Furthermore they may be susceptible to the variations caused by different speakers, specific content of the speech segments, and background noise. To address this, we propose using Deep Bottleneck Features (DBF) for spoken LID, motivated by the success of Deep Neural Networks (DNN) in speech recognition. We show that DBFs can form a low-dimensional compact representation of the original inputs with a powerful descriptive and discriminative capability. To evaluate the effectiveness of this, we design two acoustic models, termed DBF-TV and parallel DBF-TV (PDBF-TV), using a DBF based i-vector representation for each speech utterance. Results on NIST language recognition evaluation 2009 (LRE09) show significant improvements over state-of-the-art systems. By fusing the output of phonotactic and acoustic approaches, we achieve an EER of 1.08%, 1.89% and 7.01% for 30 s, 10 s and 3 s test utterances respectively. Furthermore, various DBF configurations have been extensively evaluated, and an optimal system proposed.

Improving on hidden Markov models: An articulatorily constrained, maximum likelihood approach to speech recognition and speech coding

DOE Office of Scientific and Technical Information (OSTI.GOV)

Hogden, J.

The goal of the proposed research is to test a statistical model of speech recognition that incorporates the knowledge that speech is produced by relatively slow motions of the tongue, lips, and other speech articulators. This model is called Maximum Likelihood Continuity Mapping (Malcom). Many speech researchers believe that by using constraints imposed by articulator motions, we can improve or replace the current hidden Markov model based speech recognition algorithms. Unfortunately, previous efforts to incorporate information about articulation into speech recognition algorithms have suffered because (1) slight inaccuracies in our knowledge or the formulation of our knowledge about articulation maymore » decrease recognition performance, (2) small changes in the assumptions underlying models of speech production can lead to large changes in the speech derived from the models, and (3) collecting measurements of human articulator positions in sufficient quantity for training a speech recognition algorithm is still impractical. The most interesting (and in fact, unique) quality of Malcom is that, even though Malcom makes use of a mapping between acoustics and articulation, Malcom can be trained to recognize speech using only acoustic data. By learning the mapping between acoustics and articulation using only acoustic data, Malcom avoids the difficulties involved in collecting articulator position measurements and does not require an articulatory synthesizer model to estimate the mapping between vocal tract shapes and speech acoustics. Preliminary experiments that demonstrate that Malcom can learn the mapping between acoustics and articulation are discussed. Potential applications of Malcom aside from speech recognition are also discussed. Finally, specific deliverables resulting from the proposed research are described.« less

Speech Perception With Combined Electric-Acoustic Stimulation: A Simulation and Model Comparison.

PubMed

Rader, Tobias; Adel, Youssef; Fastl, Hugo; Baumann, Uwe

2015-01-01

The aim of this study is to simulate speech perception with combined electric-acoustic stimulation (EAS), verify the advantage of combined stimulation in normal-hearing (NH) subjects, and then compare it with cochlear implant (CI) and EAS user results from the authors' previous study. Furthermore, an automatic speech recognition (ASR) system was built to examine the impact of low-frequency information and is proposed as an applied model to study different hypotheses of the combined-stimulation advantage. Signal-detection-theory (SDT) models were applied to assess predictions of subject performance without the need to assume any synergistic effects. Speech perception was tested using a closed-set matrix test (Oldenburg sentence test), and its speech material was processed to simulate CI and EAS hearing. A total of 43 NH subjects and a customized ASR system were tested. CI hearing was simulated by an aurally adequate signal spectrum analysis and representation, the part-tone-time-pattern, which was vocoded at 12 center frequencies according to the MED-EL DUET speech processor. Residual acoustic hearing was simulated by low-pass (LP)-filtered speech with cutoff frequencies 200 and 500 Hz for NH subjects and in the range from 100 to 500 Hz for the ASR system. Speech reception thresholds were determined in amplitude-modulated noise and in pseudocontinuous noise. Previously proposed SDT models were lastly applied to predict NH subject performance with EAS simulations. NH subjects tested with EAS simulations demonstrated the combined-stimulation advantage. Increasing the LP cutoff frequency from 200 to 500 Hz significantly improved speech reception thresholds in both noise conditions. In continuous noise, CI and EAS users showed generally better performance than NH subjects tested with simulations. In modulated noise, performance was comparable except for the EAS at cutoff frequency 500 Hz where NH subject performance was superior. The ASR system showed similar behavior

Emotional recognition from the speech signal for a virtual education agent

NASA Astrophysics Data System (ADS)

Tickle, A.; Raghu, S.; Elshaw, M.

2013-06-01

This paper explores the extraction of features from the speech wave to perform intelligent emotion recognition. A feature extract tool (openSmile) was used to obtain a baseline set of 998 acoustic features from a set of emotional speech recordings from a microphone. The initial features were reduced to the most important ones so recognition of emotions using a supervised neural network could be performed. Given that the future use of virtual education agents lies with making the agents more interactive, developing agents with the capability to recognise and adapt to the emotional state of humans is an important step.

Multi-voxel Patterns Reveal Functionally Differentiated Networks Underlying Auditory Feedback Processing of Speech

PubMed Central

Zheng, Zane Z.; Vicente-Grabovetsky, Alejandro; MacDonald, Ewen N.; Munhall, Kevin G.; Cusack, Rhodri; Johnsrude, Ingrid S.

2013-01-01

The everyday act of speaking involves the complex processes of speech motor control. An important component of control is monitoring, detection and processing of errors when auditory feedback does not correspond to the intended motor gesture. Here we show, using fMRI and converging operations within a multi-voxel pattern analysis framework, that this sensorimotor process is supported by functionally differentiated brain networks. During scanning, a real-time speech-tracking system was employed to deliver two acoustically different types of distorted auditory feedback or unaltered feedback while human participants were vocalizing monosyllabic words, and to present the same auditory stimuli while participants were passively listening. Whole-brain analysis of neural-pattern similarity revealed three functional networks that were differentially sensitive to distorted auditory feedback during vocalization, compared to during passive listening. One network of regions appears to encode an ‘error signal’ irrespective of acoustic features of the error: this network, including right angular gyrus, right supplementary motor area, and bilateral cerebellum, yielded consistent neural patterns across acoustically different, distorted feedback types, only during articulation (not during passive listening). In contrast, a fronto-temporal network appears sensitive to the speech features of auditory stimuli during passive listening; this preference for speech features was diminished when the same stimuli were presented as auditory concomitants of vocalization. A third network, showing a distinct functional pattern from the other two, appears to capture aspects of both neural response profiles. Taken together, our findings suggest that auditory feedback processing during speech motor control may rely on multiple, interactive, functionally differentiated neural systems. PMID:23467350

Department of Cybernetic Acoustics

NASA Astrophysics Data System (ADS)

The development of the theory, instrumentation and applications of methods and systems for the measurement, analysis, processing and synthesis of acoustic signals within the audio frequency range, particularly of the speech signal and the vibro-acoustic signal emitted by technical and industrial equipments treated as noise and vibration sources was discussed. The research work, both theoretical and experimental, aims at applications in various branches of science, and medicine, such as: acoustical diagnostics and phoniatric rehabilitation of pathological and postoperative states of the speech organ; bilateral ""man-machine'' speech communication based on the analysis, recognition and synthesis of the speech signal; vibro-acoustical diagnostics and continuous monitoring of the state of machines, technical equipments and technological processes.

Designing acoustics for linguistically diverse classrooms: Effects of background noise, reverberation and talker foreign accent on speech comprehension by native and non-native English-speaking listeners

NASA Astrophysics Data System (ADS)

Peng, Zhao Ellen

The current classroom acoustics standard (ANSI S12.60-2010) recommends core learning spaces not to exceed background noise level (BNL) of 35 dBA and reverberation time (RT) of 0.6 second, based on speech intelligibility performance mainly by the native English-speaking population. Existing literature has not correlated these recommended values well with student learning outcomes. With a growing population of non-native English speakers in American classrooms, the special needs for perceiving degraded speech among non-native listeners, either due to realistic room acoustics or talker foreign accent, have not been addressed in the current standard. This research seeks to investigate the effects of BNL and RT on the comprehension of English speech from native English and native Mandarin Chinese talkers as perceived by native and non-native English listeners, and to provide acoustic design guidelines to supplement the existing standard. This dissertation presents two studies on the effects of RT and BNL on more realistic classroom learning experiences. How do native and non-native English-speaking listeners perform on speech comprehension tasks under adverse acoustic conditions, if the English speech is produced by talkers of native English (Study 1) versus native Mandarin Chinese (Study 2)? Speech comprehension materials were played back in a listening chamber to individual listeners: native and non-native English-speaking in Study 1; native English, native Mandarin Chinese, and other non-native English-speaking in Study 2. Each listener was screened for baseline English proficiency level, and completed dual tasks simultaneously involving speech comprehension and adaptive dot-tracing under 15 acoustic conditions, comprised of three BNL conditions (RC-30, 40, and 50) and five RT scenarios (0.4 to 1.2 seconds). The results show that BNL and RT negatively affect both objective performance and subjective perception of speech comprehension, more severely for non

System and method for characterizing voiced excitations of speech and acoustic signals, removing acoustic noise from speech, and synthesizing speech

DOEpatents

Burnett, Greg C.; Holzrichter, John F.; Ng, Lawrence C.

2002-01-01

Low power EM waves are used to detect motions of vocal tract tissues of the human speech system before, during, and after voiced speech. A voiced excitation function is derived. The excitation function provides speech production information to enhance speech characterization and to enable noise removal from human speech.

Tongue- and Jaw-Specific Contributions to Acoustic Vowel Contrast Changes in the Diphthong /ai/ in Response to Slow, Loud, And Clear Speech

ERIC Educational Resources Information Center

Mefferd, Antje S.

2017-01-01

Purpose: This study sought to determine decoupled tongue and jaw displacement changes and their specific contributions to acoustic vowel contrast changes during slow, loud, and clear speech. Method: Twenty typical talkers repeated "see a kite again" 5 times in 4 speech conditions (typical, slow, loud, clear). Speech kinematics were…

Pathological speech signal analysis and classification using empirical mode decomposition.

PubMed

Kaleem, Muhammad; Ghoraani, Behnaz; Guergachi, Aziz; Krishnan, Sridhar

2013-07-01

Automated classification of normal and pathological speech signals can provide an objective and accurate mechanism for pathological speech diagnosis, and is an active area of research. A large part of this research is based on analysis of acoustic measures extracted from sustained vowels. However, sustained vowels do not reflect real-world attributes of voice as effectively as continuous speech, which can take into account important attributes of speech such as rapid voice onset and termination, changes in voice frequency and amplitude, and sudden discontinuities in speech. This paper presents a methodology based on empirical mode decomposition (EMD) for classification of continuous normal and pathological speech signals obtained from a well-known database. EMD is used to decompose randomly chosen portions of speech signals into intrinsic mode functions, which are then analyzed to extract meaningful temporal and spectral features, including true instantaneous features which can capture discriminative information in signals hidden at local time-scales. A total of six features are extracted, and a linear classifier is used with the feature vector to classify continuous speech portions obtained from a database consisting of 51 normal and 161 pathological speakers. A classification accuracy of 95.7 % is obtained, thus demonstrating the effectiveness of the methodology.

Speech Perception in the Classroom.

ERIC Educational Resources Information Center

Smaldino, Joseph J.; Crandell, Carl C.

1999-01-01

This article discusses how poor room acoustics can make speech inaudible and presents a speech-perception model demonstrating the linkage between adequacy of classroom acoustics and the development of a speech and language systems. It argues both aspects must be considered when evaluating barriers to listening and learning in a classroom.…

Inconsistency of speech in children with childhood apraxia of speech, phonological disorders, and typical speech

NASA Astrophysics Data System (ADS)

Iuzzini, Jenya

There is a lack of agreement on the features used to differentiate Childhood Apraxia of Speech (CAS) from Phonological Disorders (PD). One criterion which has gained consensus is lexical inconsistency of speech (ASHA, 2007); however, no accepted measure of this feature has been defined. Although lexical assessment provides information about consistency of an item across repeated trials, it may not capture the magnitude of inconsistency within an item. In contrast, segmental analysis provides more extensive information about consistency of phoneme usage across multiple contexts and word-positions. The current research compared segmental and lexical inconsistency metrics in preschool-aged children with PD, CAS, and typical development (TD) to determine how inconsistency varies with age in typical and disordered speakers, and whether CAS and PD were differentiated equally well by both assessment levels. Whereas lexical and segmental analyses may be influenced by listener characteristics or speaker intelligibility, the acoustic signal is less vulnerable to these factors. In addition, the acoustic signal may reveal information which is not evident in the perceptual signal. A second focus of the current research was motivated by Blumstein et al.'s (1980) classic study on voice onset time (VOT) in adults with acquired apraxia of speech (AOS) which demonstrated a motor impairment underlying AOS. In the current study, VOT analyses were conducted to determine the relationship between age and group with the voicing distribution for bilabial and alveolar plosives. Findings revealed that 3-year-olds evidenced significantly higher inconsistency than 5-year-olds; segmental inconsistency approached 0% in 5-year-olds with TD, whereas it persisted in children with PD and CAS suggesting that for child in this age-range, inconsistency is a feature of speech disorder rather than typical development (Holm et al., 2007). Likewise, whereas segmental and lexical inconsistency were

Trimodal speech perception: how residual acoustic hearing supplements cochlear-implant consonant recognition in the presence of visual cues.

PubMed

Sheffield, Benjamin M; Schuchman, Gerald; Bernstein, Joshua G W

2015-01-01

As cochlear implant (CI) acceptance increases and candidacy criteria are expanded, these devices are increasingly recommended for individuals with less than profound hearing loss. As a result, many individuals who receive a CI also retain acoustic hearing, often in the low frequencies, in the nonimplanted ear (i.e., bimodal hearing) and in some cases in the implanted ear (i.e., hybrid hearing) which can enhance the performance achieved by the CI alone. However, guidelines for clinical decisions pertaining to cochlear implantation are largely based on expectations for postsurgical speech-reception performance with the CI alone in auditory-only conditions. A more comprehensive prediction of postimplant performance would include the expected effects of residual acoustic hearing and visual cues on speech understanding. An evaluation of auditory-visual performance might be particularly important because of the complementary interaction between the speech information relayed by visual cues and that contained in the low-frequency auditory signal. The goal of this study was to characterize the benefit provided by residual acoustic hearing to consonant identification under auditory-alone and auditory-visual conditions for CI users. Additional information regarding the expected role of residual hearing in overall communication performance by a CI listener could potentially lead to more informed decisions regarding cochlear implantation, particularly with respect to recommendations for or against bilateral implantation for an individual who is functioning bimodally. Eleven adults 23 to 75 years old with a unilateral CI and air-conduction thresholds in the nonimplanted ear equal to or better than 80 dB HL for at least one octave frequency between 250 and 1000 Hz participated in this study. Consonant identification was measured for conditions involving combinations of electric hearing (via the CI), acoustic hearing (via the nonimplanted ear), and speechreading (visual cues

Influence of Architectural Features and Styles on Various Acoustical Measures in Churches

NASA Astrophysics Data System (ADS)

Carvalho, Antonio Pedro Oliveira De.

This work reports on acoustical field measurements made in a major survey of 41 Catholic churches in Portugal that were built in the last 14 centuries. A series of monaural and binaural acoustical measurements was taken at multiple source/receiver positions in each church using the impulse response with noise burst method. The acoustical measures were Reverberation Time (RT), Early Decay Time (EDT), Clarity (C80), Definition (D), Center Time (TS), Loudness (L), Bass Ratios based on the Reverberation Time and Loudness rm (BR_-RT and rm BR_-L), Rapid Speech Transmission Index (RASTI), and the binaural Coherence (COH). The scope of this research is to investigate how the acoustical performance of Catholic churches relates to their architectural features and to determine simple formulas to predict acoustical measures by the use of elementary architectural parameters. Prediction equations were defined among the acoustical measures to estimate values at individual locations within each room as well as the mean values in each church. Best fits with rm R^2~0.9 were not uncommon among many of the measures. Within and interchurch differences in the data for the acoustical measures were also analyzed. The variations of RT and EDT were identified as much smaller than the variations of the other measures. The churches tested were grouped in eight architectural styles, and the effect of their evolution through time on these acoustical measures was investigated. Statistically significant differences were found regarding some architectural styles that can be traced to historical changes in Church history, especially to the Reformation period. Prediction equations were defined to estimate mean acoustical measures by the use of fifteen simple architectural parameters. The use of the Sabine and Eyring reverberation time equations was tested. The effect of coupled spaces was analyzed, and a new algorithm for the application of the Sabine equation was developed, achieving an average of

Population responses in primary auditory cortex simultaneously represent the temporal envelope and periodicity features in natural speech.

PubMed

Abrams, Daniel A; Nicol, Trent; White-Schwoch, Travis; Zecker, Steven; Kraus, Nina

2017-05-01

Speech perception relies on a listener's ability to simultaneously resolve multiple temporal features in the speech signal. Little is known regarding neural mechanisms that enable the simultaneous coding of concurrent temporal features in speech. Here we show that two categories of temporal features in speech, the low-frequency speech envelope and periodicity cues, are processed by distinct neural mechanisms within the same population of cortical neurons. We measured population activity in primary auditory cortex of anesthetized guinea pig in response to three variants of a naturally produced sentence. Results show that the envelope of population responses closely tracks the speech envelope, and this cortical activity more closely reflects wider bandwidths of the speech envelope compared to narrow bands. Additionally, neuronal populations represent the fundamental frequency of speech robustly with phase-locked responses. Importantly, these two temporal features of speech are simultaneously observed within neuronal ensembles in auditory cortex in response to clear, conversation, and compressed speech exemplars. Results show that auditory cortical neurons are adept at simultaneously resolving multiple temporal features in extended speech sentences using discrete coding mechanisms. Copyright © 2017 Elsevier B.V. All rights reserved.

Speech after Radial Forearm Free Flap Reconstruction of the Tongue: A Longitudinal Acoustic Study of Vowel and Diphthong Sounds

ERIC Educational Resources Information Center

Laaksonen, Juha-Pertti; Rieger, Jana; Happonen, Risto-Pekka; Harris, Jeffrey; Seikaly, Hadi

2010-01-01

The purpose of this study was to use acoustic analyses to describe speech outcomes over the course of 1 year after radial forearm free flap (RFFF) reconstruction of the tongue. Eighteen Canadian English-speaking females and males with reconstruction for oral cancer had speech samples recorded (pre-operative, and 1 month, 6 months, and 1 year…

Distinct Cortical Pathways for Music and Speech Revealed by Hypothesis-Free Voxel Decomposition

PubMed Central

Norman-Haignere, Sam

2015-01-01

SUMMARY The organization of human auditory cortex remains unresolved, due in part to the small stimulus sets common to fMRI studies and the overlap of neural populations within voxels. To address these challenges, we measured fMRI responses to 165 natural sounds and inferred canonical response profiles (“components”) whose weighted combinations explained voxel responses throughout auditory cortex. This analysis revealed six components, each with interpretable response characteristics despite being unconstrained by prior functional hypotheses. Four components embodied selectivity for particular acoustic features (frequency, spectrotemporal modulation, pitch). Two others exhibited pronounced selectivity for music and speech, respectively, and were not explainable by standard acoustic features. Anatomically, music and speech selectivity concentrated in distinct regions of non-primary auditory cortex. However, music selectivity was weak in raw voxel responses, and its detection required a decomposition method. Voxel decomposition identifies primary dimensions of response variation across natural sounds, revealing distinct cortical pathways for music and speech. PMID:26687225

Distinct Cortical Pathways for Music and Speech Revealed by Hypothesis-Free Voxel Decomposition.

PubMed

Norman-Haignere, Sam; Kanwisher, Nancy G; McDermott, Josh H

2015-12-16

The organization of human auditory cortex remains unresolved, due in part to the small stimulus sets common to fMRI studies and the overlap of neural populations within voxels. To address these challenges, we measured fMRI responses to 165 natural sounds and inferred canonical response profiles ("components") whose weighted combinations explained voxel responses throughout auditory cortex. This analysis revealed six components, each with interpretable response characteristics despite being unconstrained by prior functional hypotheses. Four components embodied selectivity for particular acoustic features (frequency, spectrotemporal modulation, pitch). Two others exhibited pronounced selectivity for music and speech, respectively, and were not explainable by standard acoustic features. Anatomically, music and speech selectivity concentrated in distinct regions of non-primary auditory cortex. However, music selectivity was weak in raw voxel responses, and its detection required a decomposition method. Voxel decomposition identifies primary dimensions of response variation across natural sounds, revealing distinct cortical pathways for music and speech. Copyright © 2015 Elsevier Inc. All rights reserved.

Predicting Intelligibility Gains in Individuals with Dysarthria from Baseline Speech Features

ERIC Educational Resources Information Center

Fletcher, Annalise R.; McAuliffe, Megan J.; Lansford, Kaitlin L.; Sinex, Donal G.; Liss, Julie M.

2017-01-01

Purpose: Across the treatment literature, behavioral speech modifications have produced variable intelligibility changes in speakers with dysarthria. This study is the first of two articles exploring whether measurements of baseline speech features can predict speakers' responses to these modifications. Method: Fifty speakers (7 older individuals…

Detecting Parkinson's disease from sustained phonation and speech signals.

PubMed

Vaiciukynas, Evaldas; Verikas, Antanas; Gelzinis, Adas; Bacauskiene, Marija

2017-01-01

This study investigates signals from sustained phonation and text-dependent speech modalities for Parkinson's disease screening. Phonation corresponds to the vowel /a/ voicing task and speech to the pronunciation of a short sentence in Lithuanian language. Signals were recorded through two channels simultaneously, namely, acoustic cardioid (AC) and smart phone (SP) microphones. Additional modalities were obtained by splitting speech recording into voiced and unvoiced parts. Information in each modality is summarized by 18 well-known audio feature sets. Random forest (RF) is used as a machine learning algorithm, both for individual feature sets and for decision-level fusion. Detection performance is measured by the out-of-bag equal error rate (EER) and the cost of log-likelihood-ratio. Essentia audio feature set was the best using the AC speech modality and YAAFE audio feature set was the best using the SP unvoiced modality, achieving EER of 20.30% and 25.57%, respectively. Fusion of all feature sets and modalities resulted in EER of 19.27% for the AC and 23.00% for the SP channel. Non-linear projection of a RF-based proximity matrix into the 2D space enriched medical decision support by visualization.

Dynamic Encoding of Speech Sequence Probability in Human Temporal Cortex

PubMed Central

Leonard, Matthew K.; Bouchard, Kristofer E.; Tang, Claire

2015-01-01

Sensory processing involves identification of stimulus features, but also integration with the surrounding sensory and cognitive context. Previous work in animals and humans has shown fine-scale sensitivity to context in the form of learned knowledge about the statistics of the sensory environment, including relative probabilities of discrete units in a stream of sequential auditory input. These statistics are a defining characteristic of one of the most important sequential signals humans encounter: speech. For speech, extensive exposure to a language tunes listeners to the statistics of sound sequences. To address how speech sequence statistics are neurally encoded, we used high-resolution direct cortical recordings from human lateral superior temporal cortex as subjects listened to words and nonwords with varying transition probabilities between sound segments. In addition to their sensitivity to acoustic features (including contextual features, such as coarticulation), we found that neural responses dynamically encoded the language-level probability of both preceding and upcoming speech sounds. Transition probability first negatively modulated neural responses, followed by positive modulation of neural responses, consistent with coordinated predictive and retrospective recognition processes, respectively. Furthermore, transition probability encoding was different for real English words compared with nonwords, providing evidence for online interactions with high-order linguistic knowledge. These results demonstrate that sensory processing of deeply learned stimuli involves integrating physical stimulus features with their contextual sequential structure. Despite not being consciously aware of phoneme sequence statistics, listeners use this information to process spoken input and to link low-level acoustic representations with linguistic information about word identity and meaning. PMID:25948269

Speech masking and cancelling and voice obscuration

DOE Office of Scientific and Technical Information (OSTI.GOV)

Holzrichter, John F.

A non-acoustic sensor is used to measure a user's speech and then broadcasts an obscuring acoustic signal diminishing the user's vocal acoustic output intensity and/or distorting the voice sounds making them unintelligible to persons nearby. The non-acoustic sensor is positioned proximate or contacting a user's neck or head skin tissue for sensing speech production information.

Speech reception with different bilateral directional processing schemes: Influence of binaural hearing, audiometric asymmetry, and acoustic scenario.

PubMed

Neher, Tobias; Wagener, Kirsten C; Latzel, Matthias

2017-09-01

Hearing aid (HA) users can differ markedly in their benefit from directional processing (or beamforming) algorithms. The current study therefore investigated candidacy for different bilateral directional processing schemes. Groups of elderly listeners with symmetric (N = 20) or asymmetric (N = 19) hearing thresholds for frequencies below 2 kHz, a large spread in the binaural intelligibility level difference (BILD), and no difference in age, overall degree of hearing loss, or performance on a measure of selective attention took part. Aided speech reception was measured using virtual acoustics together with a simulation of a linked pair of completely occluding behind-the-ear HAs. Five processing schemes and three acoustic scenarios were used. The processing schemes differed in the tradeoff between signal-to-noise ratio (SNR) improvement and binaural cue preservation. The acoustic scenarios consisted of a frontal target talker presented against two speech maskers from ±60° azimuth or spatially diffuse cafeteria noise. For both groups, a significant interaction between BILD, processing scheme, and acoustic scenario was found. This interaction implied that, in situations with lateral speech maskers, HA users with BILDs larger than about 2 dB profited more from preserved low-frequency binaural cues than from greater SNR improvement, whereas for smaller BILDs the opposite was true. Audiometric asymmetry reduced the influence of binaural hearing. In spatially diffuse noise, the maximal SNR improvement was generally beneficial. N 0 S π detection performance at 500 Hz predicted the benefit from low-frequency binaural cues. Together, these findings provide a basis for adapting bilateral directional processing to individual and situational influences. Further research is needed to investigate their generalizability to more realistic HA conditions (e.g., with low-frequency vent-transmitted sound). Copyright © 2017 Elsevier B.V. All rights reserved.

Common cues to emotion in the dynamic facial expressions of speech and song

PubMed Central

Livingstone, Steven R.; Thompson, William F.; Wanderley, Marcelo M.; Palmer, Caroline

2015-01-01

Speech and song are universal forms of vocalization that may share aspects of emotional expression. Research has focused on parallels in acoustic features, overlooking facial cues to emotion. In three experiments, we compared moving facial expressions in speech and song. In Experiment 1, vocalists spoke and sang statements each with five emotions. Vocalists exhibited emotion-dependent movements of the eyebrows and lip corners that transcended speech–song differences. Vocalists’ jaw movements were coupled to their acoustic intensity, exhibiting differences across emotion and speech–song. Vocalists’ emotional movements extended beyond vocal sound to include large sustained expressions, suggesting a communicative function. In Experiment 2, viewers judged silent videos of vocalists’ facial expressions prior to, during, and following vocalization. Emotional intentions were identified accurately for movements during and after vocalization, suggesting that these movements support the acoustic message. Experiment 3 compared emotional identification in voice-only, face-only, and face-and-voice recordings. Emotion judgements for voice-only singing were poorly identified, yet were accurate for all other conditions, confirming that facial expressions conveyed emotion more accurately than the voice in song, yet were equivalent in speech. Collectively, these findings highlight broad commonalities in the facial cues to emotion in speech and song, yet highlight differences in perception and acoustic-motor production. PMID:25424388

The acoustic features of human laughter

NASA Astrophysics Data System (ADS)

Bachorowski, Jo-Anne; Owren, Michael J.

2002-05-01

Remarkably little is known about the acoustic features of laughter, despite laughter's ubiquitous role in human vocal communication. Outcomes are described for 1024 naturally produced laugh bouts recorded from 97 young adults. Acoustic analysis focused on temporal characteristics, production modes, source- and filter-related effects, and indexical cues to laugher sex and individual identity. The results indicate that laughter is a remarkably complex vocal signal, with evident diversity in both production modes and fundamental frequency characteristics. Also of interest was finding a consistent lack of articulation effects in supralaryngeal filtering. Outcomes are compared to previously advanced hypotheses and conjectures about this species-typical vocal signal.

The auditory representation of speech sounds in human motor cortex

PubMed Central

Cheung, Connie; Hamilton, Liberty S; Johnson, Keith; Chang, Edward F

2016-01-01

In humans, listening to speech evokes neural responses in the motor cortex. This has been controversially interpreted as evidence that speech sounds are processed as articulatory gestures. However, it is unclear what information is actually encoded by such neural activity. We used high-density direct human cortical recordings while participants spoke and listened to speech sounds. Motor cortex neural patterns during listening were substantially different than during articulation of the same sounds. During listening, we observed neural activity in the superior and inferior regions of ventral motor cortex. During speaking, responses were distributed throughout somatotopic representations of speech articulators in motor cortex. The structure of responses in motor cortex during listening was organized along acoustic features similar to auditory cortex, rather than along articulatory features as during speaking. Motor cortex does not contain articulatory representations of perceived actions in speech, but rather, represents auditory vocal information. DOI: http://dx.doi.org/10.7554/eLife.12577.001 PMID:26943778

Recognition of Emotions in Mexican Spanish Speech: An Approach Based on Acoustic Modelling of Emotion-Specific Vowels

PubMed Central

Caballero-Morales, Santiago-Omar

2013-01-01

An approach for the recognition of emotions in speech is presented. The target language is Mexican Spanish, and for this purpose a speech database was created. The approach consists in the phoneme acoustic modelling of emotion-specific vowels. For this, a standard phoneme-based Automatic Speech Recognition (ASR) system was built with Hidden Markov Models (HMMs), where different phoneme HMMs were built for the consonants and emotion-specific vowels associated with four emotional states (anger, happiness, neutral, sadness). Then, estimation of the emotional state from a spoken sentence is performed by counting the number of emotion-specific vowels found in the ASR's output for the sentence. With this approach, accuracy of 87–100% was achieved for the recognition of emotional state of Mexican Spanish speech. PMID:23935410

Speech Emotion Feature Selection Method Based on Contribution Analysis Algorithm of Neural Network

DOE Office of Scientific and Technical Information (OSTI.GOV)

Wang Xiaojia; Mao Qirong; Zhan Yongzhao

There are many emotion features. If all these features are employed to recognize emotions, redundant features may be existed. Furthermore, recognition result is unsatisfying and the cost of feature extraction is high. In this paper, a method to select speech emotion features based on contribution analysis algorithm of NN is presented. The emotion features are selected by using contribution analysis algorithm of NN from the 95 extracted features. Cluster analysis is applied to analyze the effectiveness for the features selected, and the time of feature extraction is evaluated. Finally, 24 emotion features selected are used to recognize six speech emotions.more » The experiments show that this method can improve the recognition rate and the time of feature extraction.« less

Speech perception and production in severe environments

NASA Astrophysics Data System (ADS)

Pisoni, David B.

1990-09-01

The goal was to acquire new knowledge about speech perception and production in severe environments such as high masking noise, increased cognitive load or sustained attentional demands. Changes were examined in speech production under these adverse conditions through acoustic analysis techniques. One set of studies focused on the effects of noise on speech production. The experiments in this group were designed to generate a database of speech obtained in noise and in quiet. A second set of experiments was designed to examine the effects of cognitive load on the acoustic-phonetic properties of speech. Talkers were required to carry out a demanding perceptual motor task while they read lists of test words. A final set of experiments explored the effects of vocal fatigue on the acoustic-phonetic properties of speech. Both cognitive load and vocal fatigue are present in many applications where speech recognition technology is used, yet their influence on speech production is poorly understood.

Combined Electric and Acoustic Stimulation With Hearing Preservation: Effect of Cochlear Implant Low-Frequency Cutoff on Speech Understanding and Perceived Listening Difficulty.

PubMed

Gifford, René H; Davis, Timothy J; Sunderhaus, Linsey W; Menapace, Christine; Buck, Barbara; Crosson, Jillian; O'Neill, Lori; Beiter, Anne; Segel, Phil

The primary objective of this study was to assess the effect of electric and acoustic overlap for speech understanding in typical listening conditions using semidiffuse noise. This study used a within-subjects, repeated measures design including 11 experienced adult implant recipients (13 ears) with functional residual hearing in the implanted and nonimplanted ear. The aided acoustic bandwidth was fixed and the low-frequency cutoff for the cochlear implant (CI) was varied systematically. Assessments were completed in the R-SPACE sound-simulation system which includes a semidiffuse restaurant noise originating from eight loudspeakers placed circumferentially about the subject's head. AzBio sentences were presented at 67 dBA with signal to noise ratio varying between +10 and 0 dB determined individually to yield approximately 50 to 60% correct for the CI-alone condition with full CI bandwidth. Listening conditions for all subjects included CI alone, bimodal (CI + contralateral hearing aid), and bilateral-aided electric and acoustic stimulation (EAS; CI + bilateral hearing aid). Low-frequency cutoffs both below and above the original "clinical software recommendation" frequency were tested for all patients, in all conditions. Subjects estimated listening difficulty for all conditions using listener ratings based on a visual analog scale. Three primary findings were that (1) there was statistically significant benefit of preserved acoustic hearing in the implanted ear for most overlap conditions, (2) the default clinical software recommendation rarely yielded the highest level of speech recognition (1 of 13 ears), and (3) greater EAS overlap than that provided by the clinical recommendation yielded significant improvements in speech understanding. For standard-electrode CI recipients with preserved hearing, spectral overlap of acoustic and electric stimuli yielded significantly better speech understanding and less listening effort in a laboratory-based, restaurant

Auditory-Perceptual and Acoustic Methods in Measuring Dysphonia Severity of Korean Speech.

PubMed

Maryn, Youri; Kim, Hyung-Tae; Kim, Jaeock

2016-09-01

The purpose of this study was to explore the criterion-related concurrent validity of two standardized auditory-perceptual rating protocols and the Acoustic Voice Quality Index (AVQI) for measuring dysphonia severity in Korean speech. Sixty native Korean subjects with various voice disorders were asked to sustain the vowel [a:] and to read aloud the Korean text "Walk." A 3-second midvowel portion of the sustained vowel and two sentences (with 25 syllables) were edited, concatenated, and analyzed according to methods described elsewhere. From 56 participants, both continuous speech and sustained vowel recordings had sufficiently high signal-to-noise ratios (35.5 dB and 37 dB on average, respectively) and were therefore subjected to further dysphonia severity analysis with (1) "G" or Grade from the GRBAS protocol, (2) "OS" or Overall Severity from the Consensus Auditory-Perceptual Evaluation of Voice protocol, and (3) AVQI. First, high correlations were found between G and OS (rS = 0.955 for sustained vowels; rS = 0.965 for continuous speech). Second, the AVQI showed a strong correlation with G (rS = 0.911) as well as OS (rP = 0.924). These findings are in agreement with similar studies dealing with continuous speech in other languages. The present study highlights the criterion-related concurrent validity of these methods in Korean speech. Furthermore, it supports the cross-linguistic robustness of the AVQI as a valid and objective marker of overall dysphonia severity. Copyright © 2016 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Discriminating between auditory and motor cortical responses to speech and non-speech mouth sounds

PubMed Central

Agnew, Z.K.; McGettigan, C.; Scott, S.K.

2012-01-01

Several perspectives on speech perception posit a central role for the representation of articulations in speech comprehension, supported by evidence for premotor activation when participants listen to speech. However no experiments have directly tested whether motor responses mirror the profile of selective auditory cortical responses to native speech sounds, or whether motor and auditory areas respond in different ways to sounds. We used fMRI to investigate cortical responses to speech and non-speech mouth (ingressive click) sounds. Speech sounds activated bilateral superior temporal gyri more than other sounds, a profile not seen in motor and premotor cortices. These results suggest that there are qualitative differences in the ways that temporal and motor areas are activated by speech and click sounds: anterior temporal lobe areas are sensitive to the acoustic/phonetic properties while motor responses may show more generalised responses to the acoustic stimuli. PMID:21812557

Effects of intelligibility on working memory demand for speech perception.

PubMed

Francis, Alexander L; Nusbaum, Howard C

2009-08-01

Understanding low-intelligibility speech is effortful. In three experiments, we examined the effects of intelligibility on working memory (WM) demands imposed by perception of synthetic speech. In all three experiments, a primary speeded word recognition task was paired with a secondary WM-load task designed to vary the availability of WM capacity during speech perception. Speech intelligibility was varied either by training listeners to use available acoustic cues in a more diagnostic manner (as in Experiment 1) or by providing listeners with more informative acoustic cues (i.e., better speech quality, as in Experiments 2 and 3). In the first experiment, training significantly improved intelligibility and recognition speed; increasing WM load significantly slowed recognition. A significant interaction between training and load indicated that the benefit of training on recognition speed was observed only under low memory load. In subsequent experiments, listeners received no training; intelligibility was manipulated by changing synthesizers. Improving intelligibility without training improved recognition accuracy, and increasing memory load still decreased it, but more intelligible speech did not produce more efficient use of available WM capacity. This suggests that perceptual learning modifies the way available capacity is used, perhaps by increasing the use of more phonetically informative features and/or by decreasing use of less informative ones.

A keyword spotting model using perceptually significant energy features

NASA Astrophysics Data System (ADS)

Umakanthan, Padmalochini

The task of a keyword recognition system is to detect the presence of certain words in a conversation based on the linguistic information present in human speech. Such keyword spotting systems have applications in homeland security, telephone surveillance and human-computer interfacing. General procedure of a keyword spotting system involves feature generation and matching. In this work, new set of features that are based on the psycho-acoustic masking nature of human speech are proposed. After developing these features a time aligned pattern matching process was implemented to locate the words in a set of unknown words. A word boundary detection technique based on frame classification using the nonlinear characteristics of speech is also addressed in this work. Validation of this keyword spotting model was done using widely acclaimed Cepstral features. The experimental results indicate the viability of using these perceptually significant features as an augmented feature set in keyword spotting.

Speech disorders in Parkinson's disease: early diagnostics and effects of medication and brain stimulation.

PubMed

Brabenec, L; Mekyska, J; Galaz, Z; Rektorova, Irena

2017-03-01

Hypokinetic dysarthria (HD) occurs in 90% of Parkinson's disease (PD) patients. It manifests specifically in the areas of articulation, phonation, prosody, speech fluency, and faciokinesis. We aimed to systematically review papers on HD in PD with a special focus on (1) early PD diagnosis and monitoring of the disease progression using acoustic voice and speech analysis, and (2) functional imaging studies exploring neural correlates of HD in PD, and (3) clinical studies using acoustic analysis to evaluate effects of dopaminergic medication and brain stimulation. A systematic literature search of articles written in English before March 2016 was conducted in the Web of Science, PubMed, SpringerLink, and IEEE Xplore databases using and combining specific relevant keywords. Articles were categorized into three groups: (1) articles focused on neural correlates of HD in PD using functional imaging (n = 13); (2) articles dealing with the acoustic analysis of HD in PD (n = 52); and (3) articles concerning specifically dopaminergic and brain stimulation-related effects as assessed by acoustic analysis (n = 31); the groups were then reviewed. We identified 14 combinations of speech tasks and acoustic features that can be recommended for use in describing the main features of HD in PD. While only a few acoustic parameters correlate with limb motor symptoms and can be partially relieved by dopaminergic medication, HD in PD seems to be mainly related to non-dopaminergic deficits and associated particularly with non-motor symptoms. Future studies should combine non-invasive brain stimulation with voice behavior approaches to achieve the best treatment effects by enhancing auditory-motor integration.

Experimental comparison between speech transmission index, rapid speech transmission index, and speech intelligibility index.

PubMed

Larm, Petra; Hongisto, Valtteri

2006-02-01

During the acoustical design of, e.g., auditoria or open-plan offices, it is important to know how speech can be perceived in various parts of the room. Different objective methods have been developed to measure and predict speech intelligibility, and these have been extensively used in various spaces. In this study, two such methods were compared, the speech transmission index (STI) and the speech intelligibility index (SII). Also the simplification of the STI, the room acoustics speech transmission index (RASTI), was considered. These quantities are all based on determining an apparent speech-to-noise ratio on selected frequency bands and summing them using a specific weighting. For comparison, some data were needed on the possible differences of these methods resulting from the calculation scheme and also measuring equipment. Their prediction accuracy was also of interest. Measurements were made in a laboratory having adjustable noise level and absorption, and in a real auditorium. It was found that the measurement equipment, especially the selection of the loudspeaker, can greatly affect the accuracy of the results. The prediction accuracy of the RASTI was found acceptable, if the input values for the prediction are accurately known, even though the studied space was not ideally diffuse.

Acoustic changes in the speech of children with cerebral palsy following an intensive program of dysarthria therapy.

PubMed

Pennington, Lindsay; Lombardo, Eftychia; Steen, Nick; Miller, Nick

2018-01-01

The speech intelligibility of children with dysarthria and cerebral palsy has been observed to increase following therapy focusing on respiration and phonation. To determine if speech intelligibility change following intervention is associated with change in acoustic measures of voice. We recorded 16 young people with cerebral palsy and dysarthria (nine girls; mean age 14 years, SD = 2; nine spastic type, two dyskinetic, four mixed; one Worster-Drought) producing speech in two conditions (single words, connected speech) twice before and twice after therapy focusing on respiration, phonation and rate. In both single-word and connected speech we measured vocal intensity (root mean square-RMS), period-to-period variability (Shimmer APQ, Jitter RAP and PPQ) and harmonics-to-noise ratio (HNR). In connected speech we also measured mean fundamental frequency, utterance duration in seconds and speech and articulation rate (syllables/s with and without pauses respectively). All acoustic measures were made using Praat. Intelligibility was calculated in previous research. In single words statistically significant but very small reductions were observed in period-to-period variability following therapy: Shimmer APQ -0.15 (95% CI = -0.21 to -0.09); Jitter RAP -0.08 (95% CI = -0.14 to -0.01); Jitter PPQ -0.08 (95% CI = -0.15 to -0.01). No changes in period-to-period perturbation across phrases in connected speech were detected. However, changes in connected speech were observed in phrase length, rate and intensity. Following therapy, mean utterance duration increased by 1.11 s (95% CI = 0.37-1.86) when measured with pauses and by 1.13 s (95% CI = 0.40-1.85) when measured without pauses. Articulation rate increased by 0.07 syllables/s (95% CI = 0.02-0.13); speech rate increased by 0.06 syllables/s (95% CI = < 0.01-0.12); and intensity increased by 0.03 Pascals (95% CI = 0.02-0.04). There was a gradual reduction in mean fundamental frequency across all time points (-11.85 Hz, 95

Joint Spatial-Spectral Feature Space Clustering for Speech Activity Detection from ECoG Signals

PubMed Central

Kanas, Vasileios G.; Mporas, Iosif; Benz, Heather L.; Sgarbas, Kyriakos N.; Bezerianos, Anastasios; Crone, Nathan E.

2014-01-01

Brain machine interfaces for speech restoration have been extensively studied for more than two decades. The success of such a system will depend in part on selecting the best brain recording sites and signal features corresponding to speech production. The purpose of this study was to detect speech activity automatically from electrocorticographic signals based on joint spatial-frequency clustering of the ECoG feature space. For this study, the ECoG signals were recorded while a subject performed two different syllable repetition tasks. We found that the optimal frequency resolution to detect speech activity from ECoG signals was 8 Hz, achieving 98.8% accuracy by employing support vector machines (SVM) as a classifier. We also defined the cortical areas that held the most information about the discrimination of speech and non-speech time intervals. Additionally, the results shed light on the distinct cortical areas associated with the two syllable repetition tasks and may contribute to the development of portable ECoG-based communication. PMID:24658248

Multilevel Analysis in Analyzing Speech Data

ERIC Educational Resources Information Center

Guddattu, Vasudeva; Krishna, Y.

2011-01-01

The speech produced by human vocal tract is a complex acoustic signal, with diverse applications in phonetics, speech synthesis, automatic speech recognition, speaker identification, communication aids, speech pathology, speech perception, machine translation, hearing research, rehabilitation and assessment of communication disorders and many…

Speech adjustments for room acoustics and their effects on vocal effort

PubMed Central

Bottalico, Pasquale

2016-01-01

Objectives The aims of the present study are: (1) to analyze the effects of the acoustical environment and the voice style on time dose (Dt_p,) and fundamental frequency (mean fo and standard deviation std_fo), while taking into account the effect of short term vocal fatigue; (2) to predict the self-reported vocal effort from the voice acoustical parameters. Methods Ten male and ten female subjects were recorded while reading a text in normal and loud styles, in three rooms - anechoic, semi-reverberant and reverberant –with and without acrylic glass panels 0.5 m from the mouth, which increased external auditory feedback. Subjects quantified how much effort was required to speak in each condition on a visual analogue scale after each task. Results (Aim1) In the loud style, Dt_p, fo and std_fo increased. The Dt_p was higher in the reverberant room compared to the other two rooms. Both genders tended to increase fo in less reverberant environments, while a more monotonous speech was produced in rooms with greater reverberation. All three voice parameters increased with short-term vocal fatigue. (Aim2) A model of the vocal effort to acoustic vocal parameters is proposed. The SPL (Sound Pressure Level) contributed to 66% of the variance explained by the model, followed by the fundamental frequency (30%) and the modulation in amplitude (4%). Conclusions The results provide insight into how voice acoustical parameters can predict vocal effort. In particular, it increased when SPL and fo increased and when the amplitude voice modulation (std_ΔSPL) decreased. PMID:28029555

Obstructive Sleep Apnea in Women: Study of Speech and Craniofacial Characteristics

PubMed Central

Tyan, Marina; Fernández Pozo, Rubén; Toledano, Doroteo; Lopez Gonzalo, Eduardo; Alcazar Ramirez, Jose Daniel; Hernandez Gomez, Luis Alfonso

2017-01-01

Background Obstructive sleep apnea (OSA) is a common sleep disorder characterized by frequent cessation of breathing lasting 10 seconds or longer. The diagnosis of OSA is performed through an expensive procedure, which requires an overnight stay at the hospital. This has led to several proposals based on the analysis of patients’ facial images and speech recordings as an attempt to develop simpler and cheaper methods to diagnose OSA. Objective The objective of this study was to analyze possible relationships between OSA and speech and facial features on a female population and whether these possible connections may be affected by the specific clinical characteristics in OSA population and, more specifically, to explore how the connection between OSA and speech and facial features can be affected by gender. Methods All the subjects are Spanish subjects suspected to suffer from OSA and referred to a sleep disorders unit. Voice recordings and photographs were collected in a supervised but not highly controlled way, trying to test a scenario close to a realistic clinical practice scenario where OSA is assessed using an app running on a mobile device. Furthermore, clinical variables such as weight, height, age, and cervical perimeter, which are usually reported as predictors of OSA, were also gathered. Acoustic analysis is centered in sustained vowels. Facial analysis consists of a set of local craniofacial features related to OSA, which were extracted from images after detecting facial landmarks by using the active appearance models. To study the probable OSA connection with speech and craniofacial features, correlations among apnea-hypopnea index (AHI), clinical variables, and acoustic and facial measurements were analyzed. Results The results obtained for female population indicate mainly weak correlations (r values between .20 and .39). Correlations between AHI, clinical variables, and speech features show the prevalence of formant frequencies over bandwidths

High-frequency neural activity predicts word parsing in ambiguous speech streams

PubMed Central

Basirat, Anahita; Azizi, Leila; van Wassenhove, Virginie

2016-01-01

During speech listening, the brain parses a continuous acoustic stream of information into computational units (e.g., syllables or words) necessary for speech comprehension. Recent neuroscientific hypotheses have proposed that neural oscillations contribute to speech parsing, but whether they do so on the basis of acoustic cues (bottom-up acoustic parsing) or as a function of available linguistic representations (top-down linguistic parsing) is unknown. In this magnetoencephalography study, we contrasted acoustic and linguistic parsing using bistable speech sequences. While listening to the speech sequences, participants were asked to maintain one of the two possible speech percepts through volitional control. We predicted that the tracking of speech dynamics by neural oscillations would not only follow the acoustic properties but also shift in time according to the participant's conscious speech percept. Our results show that the latency of high-frequency activity (specifically, beta and gamma bands) varied as a function of the perceptual report. In contrast, the phase of low-frequency oscillations was not strongly affected by top-down control. Whereas changes in low-frequency neural oscillations were compatible with the encoding of prelexical segmentation cues, high-frequency activity specifically informed on an individual's conscious speech percept. PMID:27605528

A Cross-Language Study of Acoustic Predictors of Speech Intelligibility in Individuals with Parkinson's Disease

ERIC Educational Resources Information Center

Kim, Yunjung; Choi, Yaelin

2017-01-01

Purpose: The present study aimed to compare acoustic models of speech intelligibility in individuals with the same disease (Parkinson's disease [PD]) and presumably similar underlying neuropathologies but with different native languages (American English [AE] and Korean). Method: A total of 48 speakers from the 4 speaker groups (AE speakers with…

Contributions of local speech encoding and functional connectivity to audio-visual speech perception

PubMed Central

Giordano, Bruno L; Ince, Robin A A; Gross, Joachim; Schyns, Philippe G; Panzeri, Stefano; Kayser, Christoph

2017-01-01

Seeing a speaker’s face enhances speech intelligibility in adverse environments. We investigated the underlying network mechanisms by quantifying local speech representations and directed connectivity in MEG data obtained while human participants listened to speech of varying acoustic SNR and visual context. During high acoustic SNR speech encoding by temporally entrained brain activity was strong in temporal and inferior frontal cortex, while during low SNR strong entrainment emerged in premotor and superior frontal cortex. These changes in local encoding were accompanied by changes in directed connectivity along the ventral stream and the auditory-premotor axis. Importantly, the behavioral benefit arising from seeing the speaker’s face was not predicted by changes in local encoding but rather by enhanced functional connectivity between temporal and inferior frontal cortex. Our results demonstrate a role of auditory-frontal interactions in visual speech representations and suggest that functional connectivity along the ventral pathway facilitates speech comprehension in multisensory environments. DOI: http://dx.doi.org/10.7554/eLife.24763.001 PMID:28590903

Effect of Acoustic Spectrographic Instruction on Production of English /i/ and /I/ by Spanish Pre-Service English Teachers

ERIC Educational Resources Information Center

Quintana-Lara, Marcela

2014-01-01

This study investigates the effects of Acoustic Spectrographic Instruction on the production of the English phonological contrast /i/ and / I /. Acoustic Spectrographic Instruction is based on the assumption that physical representations of speech sounds and spectrography allow learners to objectively see and modify those non-accurate features in…

Long short-term memory for speaker generalization in supervised speech separation

PubMed Central

Chen, Jitong; Wang, DeLiang

2017-01-01

Speech separation can be formulated as learning to estimate a time-frequency mask from acoustic features extracted from noisy speech. For supervised speech separation, generalization to unseen noises and unseen speakers is a critical issue. Although deep neural networks (DNNs) have been successful in noise-independent speech separation, DNNs are limited in modeling a large number of speakers. To improve speaker generalization, a separation model based on long short-term memory (LSTM) is proposed, which naturally accounts for temporal dynamics of speech. Systematic evaluation shows that the proposed model substantially outperforms a DNN-based model on unseen speakers and unseen noises in terms of objective speech intelligibility. Analyzing LSTM internal representations reveals that LSTM captures long-term speech contexts. It is also found that the LSTM model is more advantageous for low-latency speech separation and it, without future frames, performs better than the DNN model with future frames. The proposed model represents an effective approach for speaker- and noise-independent speech separation. PMID:28679261

The aprosody of schizophrenia: Computationally derived acoustic phonetic underpinnings of monotone speech.

PubMed

Compton, Michael T; Lunden, Anya; Cleary, Sean D; Pauselli, Luca; Alolayan, Yazeed; Halpern, Brooke; Broussard, Beth; Crisafio, Anthony; Capulong, Leslie; Balducci, Pierfrancesco Maria; Bernardini, Francesco; Covington, Michael A

2018-02-12

Acoustic phonetic methods are useful in examining some symptoms of schizophrenia; we used such methods to understand the underpinnings of aprosody. We hypothesized that, compared to controls and patients without clinically rated aprosody, patients with aprosody would exhibit reduced variability in: pitch (F0), jaw/mouth opening and tongue height (formant F1), tongue front/back position and/or lip rounding (formant F2), and intensity/loudness. Audiorecorded speech was obtained from 98 patients (including 25 with clinically rated aprosody and 29 without) and 102 unaffected controls using five tasks: one describing a drawing, two based on spontaneous speech elicited through a question (Tasks 2 and 3), and two based on reading prose excerpts (Tasks 4 and 5). We compared groups on variation in pitch (F0), formant F1 and F2, and intensity/loudness. Regarding pitch variation, patients with aprosody differed significantly from controls in Task 5 in both unadjusted tests and those adjusted for sociodemographics. For the standard deviation (SD) of F1, no significant differences were found in adjusted tests. Regarding SD of F2, patients with aprosody had lower values than controls in Task 3, 4, and 5. For variation in intensity/loudness, patients with aprosody had lower values than patients without aprosody and controls across the five tasks. Findings could represent a step toward developing new methods for measuring and tracking the severity of this specific negative symptom using acoustic phonetic parameters; such work is relevant to other psychiatric and neurological disorders. Copyright © 2018 Elsevier B.V. All rights reserved.

The interaction between awareness of one's own speech disorder with linguistics variables: distinctive features and severity of phonological disorder.

PubMed

Dias, Roberta Freitas; Melo, Roberta Michelon; Mezzomo, Carolina Lisbôa; Mota, Helena Bolli

2013-01-01

To analyze the possible relationship among the awareness of one's own speech disorder and some aspects of the phonological system, as the number and the type of changed distinctive features, as well as the interaction among the severity of the disorder and the non-specification of distinctive features. The analyzed group has 23 children with diagnosis of speech disorder, aged 5:0 to 7:7. The speech data were analyzed through the Distinctive Features Analysis and classified by the Percentage of Correct Consonants. One also applied the Awareness of one's own speech disorder test. The children were separated in two groups: with awareness of their own speech disorder established (more than 50% of correct identification) and without awareness of their own speech disorder established (less than 50% of correct identification). Finally, the variables of this research were submitted to analysis using descriptive and inferential statistics. The type of changed distinctive features weren't different between the groups, as well as the total of changed features and the severity disorder. However, a correlation between the severity disorder and the non-specification of distinctive features was verified, because the more severe disorders have more changes in these linguistic variables. The awareness of one's own speech disorder doesn't seem to be directly influenced by the type and by the number of changed distinctive features, neither by the speech disorder severity. Moreover, one verifies that the greater phonological disorder severity, the greater the number of changed distinctive features.

Transient Auditory Storage of Acoustic Details Is Associated with Release of Speech from Informational Masking in Reverberant Conditions

ERIC Educational Resources Information Center

Huang, Ying; Huang, Qiang; Chen, Xun; Wu, Xihong; Li, Liang

2009-01-01

Perceptual integration of the sound directly emanating from the source with reflections needs both temporal storage and correlation computation of acoustic details. We examined whether the temporal storage is frequency dependent and associated with speech unmasking. In Experiment 1, a break in correlation (BIC) between interaurally correlated…

Temporal processing of speech in a time-feature space

NASA Astrophysics Data System (ADS)

Avendano, Carlos

1997-09-01

The performance of speech communication systems often degrades under realistic environmental conditions. Adverse environmental factors include additive noise sources, room reverberation, and transmission channel distortions. This work studies the processing of speech in the temporal-feature or modulation spectrum domain, aiming for alleviation of the effects of such disturbances. Speech reflects the geometry of the vocal organs, and the linguistically dominant component is in the shape of the vocal tract. At any given point in time, the shape of the vocal tract is reflected in the short-time spectral envelope of the speech signal. The rate of change of the vocal tract shape appears to be important for the identification of linguistic components. This rate of change, or the rate of change of the short-time spectral envelope can be described by the modulation spectrum, i.e. the spectrum of the time trajectories described by the short-time spectral envelope. For a wide range of frequency bands, the modulation spectrum of speech exhibits a maximum at about 4 Hz, the average syllabic rate. Disturbances often have modulation frequency components outside the speech range, and could in principle be attenuated without significantly affecting the range with relevant linguistic information. Early efforts for exploiting the modulation spectrum domain (temporal processing), such as the dynamic cepstrum or the RASTA processing, used ad hoc designed processing and appear to be suboptimal. As a major contribution, in this dissertation we aim for a systematic data-driven design of temporal processing. First we analytically derive and discuss some properties and merits of temporal processing for speech signals. We attempt to formalize the concept and provide a theoretical background which has been lacking in the field. In the experimental part we apply temporal processing to a number of problems including adaptive noise reduction in cellular telephone environments, reduction of

Room Acoustics

NASA Astrophysics Data System (ADS)

Kuttruff, Heinrich; Mommertz, Eckard

The traditional task of room acoustics is to create or formulate conditions which ensure the best possible propagation of sound in a room from a sound source to a listener. Thus, objects of room acoustics are in particular assembly halls of all kinds, such as auditoria and lecture halls, conference rooms, theaters, concert halls or churches. Already at this point, it has to be pointed out that these conditions essentially depend on the question if speech or music should be transmitted; in the first case, the criterion for transmission quality is good speech intelligibility, in the other case, however, the success of room-acoustical efforts depends on other factors that cannot be quantified that easily, not least it also depends on the hearing habits of the listeners. In any case, absolutely "good acoustics" of a room do not exist.

High-frequency neural activity predicts word parsing in ambiguous speech streams.

PubMed

Kösem, Anne; Basirat, Anahita; Azizi, Leila; van Wassenhove, Virginie

2016-12-01

During speech listening, the brain parses a continuous acoustic stream of information into computational units (e.g., syllables or words) necessary for speech comprehension. Recent neuroscientific hypotheses have proposed that neural oscillations contribute to speech parsing, but whether they do so on the basis of acoustic cues (bottom-up acoustic parsing) or as a function of available linguistic representations (top-down linguistic parsing) is unknown. In this magnetoencephalography study, we contrasted acoustic and linguistic parsing using bistable speech sequences. While listening to the speech sequences, participants were asked to maintain one of the two possible speech percepts through volitional control. We predicted that the tracking of speech dynamics by neural oscillations would not only follow the acoustic properties but also shift in time according to the participant's conscious speech percept. Our results show that the latency of high-frequency activity (specifically, beta and gamma bands) varied as a function of the perceptual report. In contrast, the phase of low-frequency oscillations was not strongly affected by top-down control. Whereas changes in low-frequency neural oscillations were compatible with the encoding of prelexical segmentation cues, high-frequency activity specifically informed on an individual's conscious speech percept. Copyright © 2016 the American Physiological Society.

Recognition of speaker-dependent continuous speech with KEAL

NASA Astrophysics Data System (ADS)

Mercier, G.; Bigorgne, D.; Miclet, L.; Le Guennec, L.; Querre, M.

1989-04-01

A description of the speaker-dependent continuous speech recognition system KEAL is given. An unknown utterance, is recognized by means of the followng procedures: acoustic analysis, phonetic segmentation and identification, word and sentence analysis. The combination of feature-based, speaker-independent coarse phonetic segmentation with speaker-dependent statistical classification techniques is one of the main design features of the acoustic-phonetic decoder. The lexical access component is essentially based on a statistical dynamic programming technique which aims at matching a phonemic lexical entry containing various phonological forms, against a phonetic lattice. Sentence recognition is achieved by use of a context-free grammar and a parsing algorithm derived from Earley's parser. A speaker adaptation module allows some of the system parameters to be adjusted by matching known utterances with their acoustical representation. The task to be performed, described by its vocabulary and its grammar, is given as a parameter of the system. Continuously spoken sentences extracted from a 'pseudo-Logo' language are analyzed and results are presented.

Speech Adjustments for Room Acoustics and Their Effects on Vocal Effort.

PubMed

Bottalico, Pasquale

2017-05-01

The aims of the present study are (1) to analyze the effects of the acoustical environment and the voice style on time dose (D t_p ) and fundamental frequency (mean f 0 and standard deviation std_f 0 ) while taking into account the effect of short-term vocal fatigue and (2) to predict the self-reported vocal effort from the voice acoustical parameters. Ten male and ten female subjects were recorded while reading a text in normal and loud styles, in three rooms-anechoic, semi-reverberant, and reverberant-with and without acrylic glass panels 0.5 m from the mouth, which increased external auditory feedback. Subjects quantified how much effort was required to speak in each condition on a visual analogue scale after each task. (Aim1) In the loud style, D t_p , f 0 , and std_f 0 increased. The D t_p was higher in the reverberant room compared to the other two rooms. Both genders tended to increase f 0 in less reverberant environments, whereas a more monotonous speech was produced in rooms with greater reverberation. All three voice parameters increased with short-term vocal fatigue. (Aim2) A model of the vocal effort to acoustic vocal parameters is proposed. The sound pressure level contributed to 66% of the variance explained by the model, followed by the f 0 (30%) and the modulation in amplitude (4%). The results provide insight into how voice acoustical parameters can predict vocal effort. In particular, it increased when SPL and f 0 increased and when the amplitude voice modulation decreased. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Associations between speech features and phenotypic severity in Treacher Collins syndrome

PubMed Central

2014-01-01

Background Treacher Collins syndrome (TCS, OMIM 154500) is a rare congenital disorder of craniofacial development. Characteristic hypoplastic malformations of the ears, zygomatic arch, mandible and pharynx have been described in detail. However, reports on the impact of these malformations on speech are few. Exploring speech features and investigating if speech function is related to phenotypic severity are essential for optimizing follow-up and treatment. Methods Articulation, nasal resonance, voice and intelligibility were examined in 19 individuals (5–74 years, median 34 years) divided into three groups comprising children 5–10 years (n = 4), adolescents 11–18 years (n = 4) and adults 29 years and older (n = 11). A speech composite score (0–6) was calculated to reflect the variability of speech deviations. TCS severity scores of phenotypic expression and total scores of Nordic Orofacial Test-Screening (NOT-S) measuring orofacial dysfunction were used in analyses of correlation with speech characteristics (speech composite scores). Results Children and adolescents presented with significantly higher speech composite scores (median 4, range 1–6) than adults (median 1, range 0–5). Nearly all children and adolescents (6/8) displayed speech deviations of articulation, nasal resonance and voice, while only three adults were identified with multiple speech aberrations. The variability of speech dysfunction in TCS was exhibited by individual combinations of speech deviations in 13/19 participants. The speech composite scores correlated with TCS severity scores and NOT-S total scores. Speech composite scores higher than 4 were associated with cleft palate. The percent of intelligible words in connected speech was significantly lower in children and adolescents (median 77%, range 31–99) than in adults (98%, range 93–100). Intelligibility of speech among the children was markedly inconsistent and clearly affecting the understandability

Associations between speech features and phenotypic severity in Treacher Collins syndrome.

PubMed

Asten, Pamela; Akre, Harriet; Persson, Christina

2014-04-28

Treacher Collins syndrome (TCS, OMIM 154500) is a rare congenital disorder of craniofacial development. Characteristic hypoplastic malformations of the ears, zygomatic arch, mandible and pharynx have been described in detail. However, reports on the impact of these malformations on speech are few. Exploring speech features and investigating if speech function is related to phenotypic severity are essential for optimizing follow-up and treatment. Articulation, nasal resonance, voice and intelligibility were examined in 19 individuals (5-74 years, median 34 years) divided into three groups comprising children 5-10 years (n = 4), adolescents 11-18 years (n = 4) and adults 29 years and older (n = 11). A speech composite score (0-6) was calculated to reflect the variability of speech deviations. TCS severity scores of phenotypic expression and total scores of Nordic Orofacial Test-Screening (NOT-S) measuring orofacial dysfunction were used in analyses of correlation with speech characteristics (speech composite scores). Children and adolescents presented with significantly higher speech composite scores (median 4, range 1-6) than adults (median 1, range 0-5). Nearly all children and adolescents (6/8) displayed speech deviations of articulation, nasal resonance and voice, while only three adults were identified with multiple speech aberrations. The variability of speech dysfunction in TCS was exhibited by individual combinations of speech deviations in 13/19 participants. The speech composite scores correlated with TCS severity scores and NOT-S total scores. Speech composite scores higher than 4 were associated with cleft palate. The percent of intelligible words in connected speech was significantly lower in children and adolescents (median 77%, range 31-99) than in adults (98%, range 93-100). Intelligibility of speech among the children was markedly inconsistent and clearly affecting the understandability. Multiple speech deviations were identified in

Processing changes when listening to foreign-accented speech

PubMed Central

Romero-Rivas, Carlos; Martin, Clara D.; Costa, Albert

2015-01-01

This study investigates the mechanisms responsible for fast changes in processing foreign-accented speech. Event Related brain Potentials (ERPs) were obtained while native speakers of Spanish listened to native and foreign-accented speakers of Spanish. We observed a less positive P200 component for foreign-accented speech relative to native speech comprehension. This suggests that the extraction of spectral information and other important acoustic features was hampered during foreign-accented speech comprehension. However, the amplitude of the N400 component for foreign-accented speech comprehension decreased across the experiment, suggesting the use of a higher level, lexical mechanism. Furthermore, during native speech comprehension, semantic violations in the critical words elicited an N400 effect followed by a late positivity. During foreign-accented speech comprehension, semantic violations only elicited an N400 effect. Overall, our results suggest that, despite a lack of improvement in phonetic discrimination, native listeners experience changes at lexical-semantic levels of processing after brief exposure to foreign-accented speech. Moreover, these results suggest that lexical access, semantic integration and linguistic re-analysis processes are permeable to external factors, such as the accent of the speaker. PMID:25859209

Fluid Dynamics of Human Phonation and Speech

NASA Astrophysics Data System (ADS)

Mittal, Rajat; Erath, Byron D.; Plesniak, Michael W.

2013-01-01

This article presents a review of the fluid dynamics, flow-structure interactions, and acoustics associated with human phonation and speech. Our voice is produced through the process of phonation in the larynx, and an improved understanding of the underlying physics of this process is essential to advancing the treatment of voice disorders. Insights into the physics of phonation and speech can also contribute to improved vocal training and the development of new speech compression and synthesis schemes. This article introduces the key biomechanical features of the laryngeal physiology, reviews the basic principles of voice production, and summarizes the progress made over the past half-century in understanding the flow physics of phonation and speech. Laryngeal pathologies, which significantly enhance the complexity of phonatory dynamics, are discussed. After a thorough examination of the state of the art in computational modeling and experimental investigations of phonatory biomechanics, we present a synopsis of the pacing issues in this arena and an outlook for research in this fascinating subject.

Improving the speech intelligibility in classrooms

NASA Astrophysics Data System (ADS)

Lam, Choi Ling Coriolanus

One of the major acoustical concerns in classrooms is the establishment of effective verbal communication between teachers and students. Non-optimal acoustical conditions, resulting in reduced verbal communication, can cause two main problems. First, they can lead to reduce learning efficiency. Second, they can also cause fatigue, stress, vocal strain and health problems, such as headaches and sore throats, among teachers who are forced to compensate for poor acoustical conditions by raising their voices. Besides, inadequate acoustical conditions can induce the usage of public address system. Improper usage of such amplifiers or loudspeakers can lead to impairment of students' hearing systems. The social costs of poor classroom acoustics will be large to impair the learning of children. This invisible problem has far reaching implications for learning, but is easily solved. Many researches have been carried out that they have accurately and concisely summarized the research findings on classrooms acoustics. Though, there is still a number of challenging questions remaining unanswered. Most objective indices for speech intelligibility are essentially based on studies of western languages. Even several studies of tonal languages as Mandarin have been conducted, there is much less on Cantonese. In this research, measurements have been done in unoccupied rooms to investigate the acoustical parameters and characteristics of the classrooms. The speech intelligibility tests, which based on English, Mandarin and Cantonese, and the survey were carried out on students aged from 5 years old to 22 years old. It aims to investigate the differences in intelligibility between English, Mandarin and Cantonese of the classrooms in Hong Kong. The significance on speech transmission index (STI) related to Phonetically Balanced (PB) word scores will further be developed. Together with developed empirical relationship between the speech intelligibility in classrooms with the variations

Acoustic Analysis of PD Speech

PubMed Central

Chenausky, Karen; MacAuslan, Joel; Goldhor, Richard

2011-01-01

According to the U.S. National Institutes of Health, approximately 500,000 Americans have Parkinson's disease (PD), with roughly another 50,000 receiving new diagnoses each year. 70%–90% of these people also have the hypokinetic dysarthria associated with PD. Deep brain stimulation (DBS) substantially relieves motor symptoms in advanced-stage patients for whom medication produces disabling dyskinesias. This study investigated speech changes as a result of DBS settings chosen to maximize motor performance. The speech of 10 PD patients and 12 normal controls was analyzed for syllable rate and variability, syllable length patterning, vowel fraction, voice-onset time variability, and spirantization. These were normalized by the controls' standard deviation to represent distance from normal and combined into a composite measure. Results show that DBS settings relieving motor symptoms can improve speech, making it up to three standard deviations closer to normal. However, the clinically motivated settings evaluated here show greater capacity to impair, rather than improve, speech. A feedback device developed from these findings could be useful to clinicians adjusting DBS parameters, as a means for ensuring they do not unwittingly choose DBS settings which impair patients' communication. PMID:21977333

Automatic intelligibility classification of sentence-level pathological speech

PubMed Central

Kim, Jangwon; Kumar, Naveen; Tsiartas, Andreas; Li, Ming; Narayanan, Shrikanth S.

2014-01-01

Pathological speech usually refers to the condition of speech distortion resulting from atypicalities in voice and/or in the articulatory mechanisms owing to disease, illness or other physical or biological insult to the production system. Although automatic evaluation of speech intelligibility and quality could come in handy in these scenarios to assist experts in diagnosis and treatment design, the many sources and types of variability often make it a very challenging computational processing problem. In this work we propose novel sentence-level features to capture abnormal variation in the prosodic, voice quality and pronunciation aspects in pathological speech. In addition, we propose a post-classification posterior smoothing scheme which refines the posterior of a test sample based on the posteriors of other test samples. Finally, we perform feature-level fusions and subsystem decision fusion for arriving at a final intelligibility decision. The performances are tested on two pathological speech datasets, the NKI CCRT Speech Corpus (advanced head and neck cancer) and the TORGO database (cerebral palsy or amyotrophic lateral sclerosis), by evaluating classification accuracy without overlapping subjects’ data among training and test partitions. Results show that the feature sets of each of the voice quality subsystem, prosodic subsystem, and pronunciation subsystem, offer significant discriminating power for binary intelligibility classification. We observe that the proposed posterior smoothing in the acoustic space can further reduce classification errors. The smoothed posterior score fusion of subsystems shows the best classification performance (73.5% for unweighted, and 72.8% for weighted, average recalls of the binary classes). PMID:25414544

Effects of programming threshold and maplaw settings on acoustic thresholds and speech discrimination with the MED-EL COMBI 40+ cochlear implant.

PubMed

Boyd, Paul J

2006-12-01

The principal task in the programming of a cochlear implant (CI) speech processor is the setting of the electrical dynamic range (output) for each electrode, to ensure that a comfortable loudness percept is obtained for a range of input levels. This typically involves separate psychophysical measurement of electrical threshold ([theta] e) and upper tolerance levels using short current bursts generated by the fitting software. Anecdotal clinical experience and some experimental studies suggest that the measurement of [theta]e is relatively unimportant and that the setting of upper tolerance limits is more critical for processor programming. The present study aims to test this hypothesis and examines in detail how acoustic thresholds and speech recognition are affected by setting of the lower limit of the output ("Programming threshold" or "PT") to understand better the influence of this parameter and how it interacts with certain other programming parameters. Test programs (maps) were generated with PT set to artificially high and low values and tested on users of the MED-EL COMBI 40+ CI system. Acoustic thresholds and speech recognition scores (sentence tests) were measured for each of the test maps. Acoustic thresholds were also measured using maps with a range of output compression functions ("maplaws"). In addition, subjective reports were recorded regarding the presence of "background threshold stimulation" which is occasionally reported by CI users if PT is set to relatively high values when using the CIS strategy. Manipulation of PT was found to have very little effect. Setting PT to minimum produced a mean 5 dB (S.D. = 6.25) increase in acoustic thresholds, relative to thresholds with PT set normally, and had no statistically significant effect on speech recognition scores on a sentence test. On the other hand, maplaw setting was found to have a significant effect on acoustic thresholds (raised as maplaw is made more linear), which provides some theoretical

Obstructive Sleep Apnea in Women: Study of Speech and Craniofacial Characteristics.

PubMed

Tyan, Marina; Espinoza-Cuadros, Fernando; Fernández Pozo, Rubén; Toledano, Doroteo; Lopez Gonzalo, Eduardo; Alcazar Ramirez, Jose Daniel; Hernandez Gomez, Luis Alfonso

2017-11-06

Obstructive sleep apnea (OSA) is a common sleep disorder characterized by frequent cessation of breathing lasting 10 seconds or longer. The diagnosis of OSA is performed through an expensive procedure, which requires an overnight stay at the hospital. This has led to several proposals based on the analysis of patients' facial images and speech recordings as an attempt to develop simpler and cheaper methods to diagnose OSA. The objective of this study was to analyze possible relationships between OSA and speech and facial features on a female population and whether these possible connections may be affected by the specific clinical characteristics in OSA population and, more specifically, to explore how the connection between OSA and speech and facial features can be affected by gender. All the subjects are Spanish subjects suspected to suffer from OSA and referred to a sleep disorders unit. Voice recordings and photographs were collected in a supervised but not highly controlled way, trying to test a scenario close to a realistic clinical practice scenario where OSA is assessed using an app running on a mobile device. Furthermore, clinical variables such as weight, height, age, and cervical perimeter, which are usually reported as predictors of OSA, were also gathered. Acoustic analysis is centered in sustained vowels. Facial analysis consists of a set of local craniofacial features related to OSA, which were extracted from images after detecting facial landmarks by using the active appearance models. To study the probable OSA connection with speech and craniofacial features, correlations among apnea-hypopnea index (AHI), clinical variables, and acoustic and facial measurements were analyzed. The results obtained for female population indicate mainly weak correlations (r values between .20 and .39). Correlations between AHI, clinical variables, and speech features show the prevalence of formant frequencies over bandwidths, with F2/i/ being the most appropriate

Subcortical Contributions to Motor Speech: Phylogenetic, Developmental, Clinical.

PubMed

Ziegler, W; Ackermann, H

2017-08-01

Vocal learning is an exclusively human trait among primates. However, songbirds demonstrate behavioral features resembling human speech learning. Two circuits have a preeminent role in this human behavior; namely, the corticostriatal and the cerebrocerebellar motor loops. While the striatal contribution can be traced back to the avian anterior forebrain pathway (AFP), the sensorimotor adaptation functions of the cerebellum appear to be human specific in acoustic communication. This review contributes to an ongoing discussion on how birdsong translates into human speech. While earlier approaches were focused on higher linguistic functions, we place the motor aspects of speaking at center stage. Genetic data are brought together with clinical and developmental evidence to outline the role of cerebrocerebellar and corticostriatal interactions in human speech. Copyright © 2017 Elsevier Ltd. All rights reserved.

Speech perception with combined electric-acoustic stimulation and bilateral cochlear implants in a multisource noise field.

PubMed

Rader, Tobias; Fastl, Hugo; Baumann, Uwe

2013-01-01

The aim of the study was to measure and compare speech perception in users of electric-acoustic stimulation (EAS) supported by a hearing aid in the unimplanted ear and in bilateral cochlear implant (CI) users under different noise and sound field conditions. Gap listening was assessed by comparing performance in unmodulated and modulated Comité Consultatif International Téléphonique et Télégraphique (CCITT) noise conditions, and binaural interaction was investigated by comparing single source and multisource sound fields. Speech perception in noise was measured using a closed-set sentence test (Oldenburg Sentence Test, OLSA) in a multisource noise field (MSNF) consisting of a four-loudspeaker array with independent noise sources and a single source in frontal position (S0N0). Speech simulating noise (Fastl-noise), CCITT-noise (continuous), and OLSA-noise (pseudo continuous) served as noise sources with different temporal patterns. Speech tests were performed in two groups of subjects who were using either EAS (n = 12) or bilateral CIs (n = 10). All subjects in the EAS group were fitted with a high-power hearing aid in the opposite ear (bimodal EAS). The average group score on monosyllable in quiet was 68.8% (EAS) and 80.5% (bilateral CI). A group of 22 listeners with normal hearing served as controls to compare and evaluate potential gap listening effects in implanted patients. Average speech reception thresholds in the EAS group were significantly lower than those for the bilateral CI group in all test conditions (CCITT 6.1 dB, p = 0.001; Fastl-noise 5.4 dB, p < 0.01; Oldenburg-(OL)-noise 1.6 dB, p < 0.05). Bilateral CI and EAS user groups showed a significant improvement of 4.3 dB (p = 0.004) and 5.4 dB (p = 0.002) between S0N0 and MSNF sound field conditions respectively, which signifies advantages caused by bilateral interaction in both groups. Performance in the control group showed a significant gap listening effect with a difference of 6.5 dB between

'Who's a good boy?!' Dogs prefer naturalistic dog-directed speech.

PubMed

Benjamin, Alex; Slocombe, Katie

2018-05-01

Infant-directed speech (IDS) is a special speech register thought to aid language acquisition and improve affiliation in human infants. Although IDS shares some of its properties with dog-directed speech (DDS), it is unclear whether the production of DDS is functional, or simply an overgeneralisation of IDS within Western cultures. One recent study found that, while puppies attended more to a script read with DDS compared with adult-directed speech (ADS), adult dogs displayed no preference. In contrast, using naturalistic speech and a more ecologically valid set-up, we found that adult dogs attended to and showed more affiliative behaviour towards a speaker of DDS than of ADS. To explore whether this preference for DDS was modulated by the dog-specific words typically used in DDS, the acoustic features (prosody) of DDS or a combination of the two, we conducted a second experiment. Here the stimuli from experiment 1 were produced with reversed prosody, meaning the prosody and content of ADS and DDS were mismatched. The results revealed no significant effect of speech type, or content, suggesting that it is maybe the combination of the acoustic properties and the dog-related content of DDS that modulates the preference shown for naturalistic DDS. Overall, the results of this study suggest that naturalistic DDS, comprising of both dog-directed prosody and dog-relevant content words, improves dogs' attention and may strengthen the affiliative bond between humans and their pets.

Underwater speech communications with a modulated laser

NASA Astrophysics Data System (ADS)

Woodward, B.; Sari, H.

2008-04-01

A novel speech communications system using a modulated laser beam has been developed for short-range applications in which high directionality is an exploitable feature. Although it was designed for certain underwater applications, such as speech communications between divers or between a diver and the surface, it may equally be used for air applications. With some modification it could be used for secure diver-to-diver communications in the situation where untethered divers are swimming close together and do not want their conversations monitored by intruders. Unlike underwater acoustic communications, where the transmitted speech may be received at ranges of hundreds of metres omnidirectionally, a laser communication link is very difficult to intercept and also obviates the need for cables that become snagged or broken. Further applications include the transmission of speech and data, including the short message service (SMS), from a fixed installation such as a sea-bed habitat; and data transmission to and from an autonomous underwater vehicle (AUV), particularly during docking manoeuvres. The performance of the system has been assessed subjectively by listening tests, which revealed that the speech was intelligible, although of poor quality due to the speech algorithm used.

Acoustic and Perceptual Measurement of Expressive Prosody in High-Functioning Autism: Increased Pitch Range and What it Means to Listeners

ERIC Educational Resources Information Center

Nadig, Aparna; Shaw, Holly

2012-01-01

Are there consistent markers of atypical prosody in speakers with high functioning autism (HFA) compared to typically-developing speakers? We examined: (1) acoustic measurements of pitch range, mean pitch and speech rate in conversation, (2) perceptual ratings of conversation for these features and overall prosody, and (3) acoustic measurements of…

Effects of Additional Low-Pass-Filtered Speech on Listening Effort for Noise-Band-Vocoded Speech in Quiet and in Noise.

PubMed

Pals, Carina; Sarampalis, Anastasios; van Dijk, Mart; Başkent, Deniz

2018-05-11

Residual acoustic hearing in electric-acoustic stimulation (EAS) can benefit cochlear implant (CI) users in increased sound quality, speech intelligibility, and improved tolerance to noise. The goal of this study was to investigate whether the low-pass-filtered acoustic speech in simulated EAS can provide the additional benefit of reducing listening effort for the spectrotemporally degraded signal of noise-band-vocoded speech. Listening effort was investigated using a dual-task paradigm as a behavioral measure, and the NASA Task Load indeX as a subjective self-report measure. The primary task of the dual-task paradigm was identification of sentences presented in three experiments at three fixed intelligibility levels: at near-ceiling, 50%, and 79% intelligibility, achieved by manipulating the presence and level of speech-shaped noise in the background. Listening effort for the primary intelligibility task was reflected in the performance on the secondary, visual response time task. Experimental speech processing conditions included monaural or binaural vocoder, with added low-pass-filtered speech (to simulate EAS) or without (to simulate CI). In Experiment 1, in quiet with intelligibility near-ceiling, additional low-pass-filtered speech reduced listening effort compared with binaural vocoder, in line with our expectations, although not compared with monaural vocoder. In Experiments 2 and 3, for speech in noise, added low-pass-filtered speech allowed the desired intelligibility levels to be reached at less favorable speech-to-noise ratios, as expected. It is interesting that this came without the cost of increased listening effort usually associated with poor speech-to-noise ratios; at 50% intelligibility, even a reduction in listening effort on top of the increased tolerance to noise was observed. The NASA Task Load indeX did not capture these differences. The dual-task results provide partial evidence for a potential decrease in listening effort as a result of

Speech Recognition: Acoustic-Phonetic Knowledge Acquisition and Representation.

DTIC Science & Technology

1987-09-25

the release duration is the voice onset time, or VOT. For the purpose of this investigation, alveolar flaps ( as in "butter’) and and glottalized /t/’s...Cambridge, Massachusetts 02139 Abstract females and 8 males. The other sentence was said by 7 females We discuss a framework for an acoustic-phonetic...tarned a number of semivowels. One sentence was said by 6 vowels + + "jpporte.d by a Xerox Fellowhsp Table It Features which characterite

ON THE NATURE OF SPEECH SCIENCE.

ERIC Educational Resources Information Center

PETERSON, GORDON E.

IN THIS ARTICLE THE NATURE OF THE DISCIPLINE OF SPEECH SCIENCE IS CONSIDERED AND THE VARIOUS BASIC AND APPLIED AREAS OF THE DISCIPLINE ARE DISCUSSED. THE BASIC AREAS ENCOMPASS THE VARIOUS PROCESSES OF THE PHYSIOLOGY OF SPEECH PRODUCTION, THE ACOUSTICAL CHARACTERISTICS OF SPEECH, INCLUDING THE SPEECH WAVE TYPES AND THE INFORMATION-BEARING ACOUSTIC…

Speech versus non-speech as irrelevant sound: controlling acoustic variation.

PubMed

Little, Jason S; Martin, Frances Heritage; Thomson, Richard H S

2010-09-01

Functional differences between speech and non-speech within the irrelevant sound effect were investigated using repeated and changing formats of irrelevant sounds in the form of intelligible words and unintelligible signal correlated noise (SCN) versions of the words. Event-related potentials were recorded from 25 females aged between 18 and 25 while they completed a serial order recall task in the presence of irrelevant sound or silence. As expected and in line with the changing-state hypothesis both words and SCN produced robust changing-state effects. However, words produced a greater changing-state effect than SCN indicating that the spectral detail inherent within speech accounts for the greater irrelevant sound effect and changing-state effect typically observed with speech. ERP data in the form of N1 amplitude was modulated within some irrelevant sound conditions suggesting that attentional aspects are involved in the elicitation of the irrelevant sound effect. Copyright (c) 2010 Elsevier B.V. All rights reserved.

Infant-Mother Acoustic-Prosodic Alignment and Developmental Risk.

PubMed

Seidl, Amanda; Cristia, Alejandrina; Soderstrom, Melanie; Ko, Eon-Suk; Abel, Emily A; Kellerman, Ashleigh; Schwichtenberg, A J

2018-06-19

One promising early marker for autism and other communicative and language disorders is early infant speech production. Here we used daylong recordings of high- and low-risk infant-mother dyads to examine whether acoustic-prosodic alignment as well as two automated measures of infant vocalization are related to developmental risk status indexed via familial risk and developmental progress at 36 months of age. Automated analyses of the acoustics of daylong real-world interactions were used to examine whether pitch characteristics of one vocalization by the mother or the child predicted those of the vocalization response by the other speaker and whether other features of infants' speech in daylong recordings were associated with developmental risk status or outcomes. Low-risk and high-risk dyads did not differ in the level of acoustic-prosodic alignment, which was overall not significant. Further analyses revealed that acoustic-prosodic alignment did not predict infants' later developmental progress, which was, however, associated with two automated measures of infant vocalizations (daily vocalizations and conversational turns). Although further research is needed, these findings suggest that automated measures of vocalizations drawn from daylong recordings are a possible early identification tool for later developmental progress/concerns. https://osf.io/cdn3v/.

Distributed neural signatures of natural audiovisual speech and music in the human auditory cortex.

PubMed

Salmi, Juha; Koistinen, Olli-Pekka; Glerean, Enrico; Jylänki, Pasi; Vehtari, Aki; Jääskeläinen, Iiro P; Mäkelä, Sasu; Nummenmaa, Lauri; Nummi-Kuisma, Katarina; Nummi, Ilari; Sams, Mikko

2017-08-15

During a conversation or when listening to music, auditory and visual information are combined automatically into audiovisual objects. However, it is still poorly understood how specific type of visual information shapes neural processing of sounds in lifelike stimulus environments. Here we applied multi-voxel pattern analysis to investigate how naturally matching visual input modulates supratemporal cortex activity during processing of naturalistic acoustic speech, singing and instrumental music. Bayesian logistic regression classifiers with sparsity-promoting priors were trained to predict whether the stimulus was audiovisual or auditory, and whether it contained piano playing, speech, or singing. The predictive performances of the classifiers were tested by leaving one participant at a time for testing and training the model using the remaining 15 participants. The signature patterns associated with unimodal auditory stimuli encompassed distributed locations mostly in the middle and superior temporal gyrus (STG/MTG). A pattern regression analysis, based on a continuous acoustic model, revealed that activity in some of these MTG and STG areas were associated with acoustic features present in speech and music stimuli. Concurrent visual stimulus modulated activity in bilateral MTG (speech), lateral aspect of right anterior STG (singing), and bilateral parietal opercular cortex (piano). Our results suggest that specific supratemporal brain areas are involved in processing complex natural speech, singing, and piano playing, and other brain areas located in anterior (facial speech) and posterior (music-related hand actions) supratemporal cortex are influenced by related visual information. Those anterior and posterior supratemporal areas have been linked to stimulus identification and sensory-motor integration, respectively. Copyright © 2017 Elsevier Inc. All rights reserved.

Acoustic changes in voice after tonsillectomy.

PubMed

Saida, H; Hirose, H

1996-01-01

The vocal tract from the glottis to the lips is considered to he a resonator and the voice is changeable depending upon the shape of the vocal tract. In this report, we examined the change in pharyngeal size and acoustic feature of voice after tonsillectomy. Subjects were 20 patients. The distance between both anterior pillars (glossopalatine arches), and between both posterior pillars (pharyngopalatine arches) was measured weekly. For acoustic measurements, the five Japanese vowels and Japanese conversational sentences were recorded and analyzed. The distance between both anterior pillars became wider 2 weeks postoperatively, and tended to become narrower thereafter. The distance between both posterior pillars became wider even after 4 weeks postoperatively. No consistent changes in F0, F1 and F2 were found after surgery. Although there was a tendency for a decrease in F3, tonsillectomy did not appear to change the acoustical features of the Japanese vowels remarkably. It was assumed that the subject may adjust the shape of the vocal tract to produce consistent speech sounds after the surgery using auditory feedback.

Acoustic Analysis of Nasal Vowels in Monguor Language

NASA Astrophysics Data System (ADS)

Zhang, Hanbin

2017-09-01

The purpose of the study is to analyze the spectrum characteristics and acoustic features for the nasal vowels [ɑ˜] and [ɔ˜] in Monguor language. On the base of acoustic parameter database of the Monguor speech, the study finds out that there are five main zero-pole pairs appearing for the nasal vowel [ɔ˜] and two zero-pole pairs appear for the nasal vowel [ɔ˜]. The results of regression analysis demonstrate that the duration of the nasal vowel [ɔ˜] or the nasal vowel [ɔ˜] can be predicted by its F1, F2 and F3 respectively.

Words from spontaneous conversational speech can be recognized with human-like accuracy by an error-driven learning algorithm that discriminates between meanings straight from smart acoustic features, bypassing the phoneme as recognition unit.

PubMed

Arnold, Denis; Tomaschek, Fabian; Sering, Konstantin; Lopez, Florence; Baayen, R Harald

2017-01-01

Sound units play a pivotal role in cognitive models of auditory comprehension. The general consensus is that during perception listeners break down speech into auditory words and subsequently phones. Indeed, cognitive speech recognition is typically taken to be computationally intractable without phones. Here we present a computational model trained on 20 hours of conversational speech that recognizes word meanings within the range of human performance (model 25%, native speakers 20-44%), without making use of phone or word form representations. Our model also generates successfully predictions about the speed and accuracy of human auditory comprehension. At the heart of the model is a 'wide' yet sparse two-layer artificial neural network with some hundred thousand input units representing summaries of changes in acoustic frequency bands, and proxies for lexical meanings as output units. We believe that our model holds promise for resolving longstanding theoretical problems surrounding the notion of the phone in linguistic theory.

Objective speech quality evaluation of real-time speech coders

NASA Astrophysics Data System (ADS)

Viswanathan, V. R.; Russell, W. H.; Huggins, A. W. F.

1984-02-01

This report describes the work performed in two areas: subjective testing of a real-time 16 kbit/s adaptive predictive coder (APC) and objective speech quality evaluation of real-time coders. The speech intelligibility of the APC coder was tested using the Diagnostic Rhyme Test (DRT), and the speech quality was tested using the Diagnostic Acceptability Measure (DAM) test, under eight operating conditions involving channel error, acoustic background noise, and tandem link with two other coders. The test results showed that the DRT and DAM scores of the APC coder equalled or exceeded the corresponding test scores fo the 32 kbit/s CVSD coder. In the area of objective speech quality evaluation, the report describes the development, testing, and validation of a procedure for automatically computing several objective speech quality measures, given only the tape-recordings of the input speech and the corresponding output speech of a real-time speech coder.

Speech recognition features for EEG signal description in detection of neonatal seizures.

PubMed

Temko, A; Boylan, G; Marnane, W; Lightbody, G

2010-01-01

In this work, features which are usually employed in automatic speech recognition (ASR) are used for the detection of neonatal seizures in newborn EEG. Three conventional ASR feature sets are compared to the feature set which has been previously developed for this task. The results indicate that the thoroughly-studied spectral envelope based ASR features perform reasonably well on their own. Additionally, the SVM Recursive Feature Elimination routine is applied to all extracted features pooled together. It is shown that ASR features consistently appear among the top-rank features.

Alternating motion rate as an index of speech motor disorder in traumatic brain injury.

PubMed

Wang, Yu-Tsai; Kent, Ray D; Duffy, Joseph R; Thomas, Jack E; Weismer, Gary

2004-01-01

The task of syllable alternating motion rate (AMR) (also called diadochokinesis) is suitable for examining speech disorders of varying degrees of severity and in individuals with varying levels of linguistic and cognitive ability. However, very limited information on this task has been published for subjects with traumatic brain injury (TBI). This study is a quantitative and qualitative acoustic analysis of AMR in seven subjects with TBI. The primary goal was to use acoustic analyses to assess speech motor control disturbances for the group as a whole and for individual patients. Quantitative analyses included measures of syllable rate, syllable and intersyllable gap durations, energy maxima, and voice onset time (VOT). Qualitative analyses included classification of features evident in spectrograms and waveforms to provide a more detailed description. The TBI group had (1) a slowed syllable rate due mostly to lengthened syllables and, to a lesser degree, lengthened intersyllable gaps, (2) highly correlated syllable rates between AMR and conversation, (3) temporal and energy maxima irregularities within repetition sequences, (4) normal median VOT values but with large variation, and (5) a number of speech production abnormalities revealed by qualitative analysis, including explosive speech quality, breathy voice quality, phonatory instability, multiple or missing stop bursts, continuous voicing, and spirantization. The relationships between these findings and TBI speakers' neurological status and dysarthria types are also discussed. It was concluded that acoustic analyses of the AMR task provides specific information on motor speech limitations in individuals with TBI.

Irregular Speech Rate Dissociates Auditory Cortical Entrainment, Evoked Responses, and Frontal Alpha

PubMed Central

Kayser, Stephanie J.; Ince, Robin A.A.; Gross, Joachim

2015-01-01

The entrainment of slow rhythmic auditory cortical activity to the temporal regularities in speech is considered to be a central mechanism underlying auditory perception. Previous work has shown that entrainment is reduced when the quality of the acoustic input is degraded, but has also linked rhythmic activity at similar time scales to the encoding of temporal expectations. To understand these bottom-up and top-down contributions to rhythmic entrainment, we manipulated the temporal predictive structure of speech by parametrically altering the distribution of pauses between syllables or words, thereby rendering the local speech rate irregular while preserving intelligibility and the envelope fluctuations of the acoustic signal. Recording EEG activity in human participants, we found that this manipulation did not alter neural processes reflecting the encoding of individual sound transients, such as evoked potentials. However, the manipulation significantly reduced the fidelity of auditory delta (but not theta) band entrainment to the speech envelope. It also reduced left frontal alpha power and this alpha reduction was predictive of the reduced delta entrainment across participants. Our results show that rhythmic auditory entrainment in delta and theta bands reflect functionally distinct processes. Furthermore, they reveal that delta entrainment is under top-down control and likely reflects prefrontal processes that are sensitive to acoustical regularities rather than the bottom-up encoding of acoustic features. SIGNIFICANCE STATEMENT The entrainment of rhythmic auditory cortical activity to the speech envelope is considered to be critical for hearing. Previous work has proposed divergent views in which entrainment reflects either early evoked responses related to sound encoding or high-level processes related to expectation or cognitive selection. Using a manipulation of speech rate, we dissociated auditory entrainment at different time scales. Specifically, our

Musical melody and speech intonation: singing a different tune.

PubMed

Zatorre, Robert J; Baum, Shari R

2012-01-01

Music and speech are often cited as characteristically human forms of communication. Both share the features of hierarchical structure, complex sound systems, and sensorimotor sequencing demands, and both are used to convey and influence emotions, among other functions [1]. Both music and speech also prominently use acoustical frequency modulations, perceived as variations in pitch, as part of their communicative repertoire. Given these similarities, and the fact that pitch perception and production involve the same peripheral transduction system (cochlea) and the same production mechanism (vocal tract), it might be natural to assume that pitch processing in speech and music would also depend on the same underlying cognitive and neural mechanisms. In this essay we argue that the processing of pitch information differs significantly for speech and music; specifically, we suggest that there are two pitch-related processing systems, one for more coarse-grained, approximate analysis and one for more fine-grained accurate representation, and that the latter is unique to music. More broadly, this dissociation offers clues about the interface between sensory and motor systems, and highlights the idea that multiple processing streams are a ubiquitous feature of neuro-cognitive architectures.

The Effect of Uni- and Bilateral Thalamic Deep Brain Stimulation on Speech in Patients With Essential Tremor: Acoustics and Intelligibility.

PubMed

Becker, Johannes; Barbe, Michael T; Hartinger, Mariam; Dembek, Till A; Pochmann, Jil; Wirths, Jochen; Allert, Niels; Mücke, Doris; Hermes, Anne; Meister, Ingo G; Visser-Vandewalle, Veerle; Grice, Martine; Timmermann, Lars

2017-04-01

Deep brain stimulation (DBS) of the ventral intermediate nucleus (VIM) is performed to suppress medically-resistant essential tremor (ET). However, stimulation induced dysarthria (SID) is a common side effect, limiting the extent to which tremor can be suppressed. To date, the exact pathogenesis of SID in VIM-DBS treated ET patients is unknown. We investigate the effect of inactivated, uni- and bilateral VIM-DBS on speech production in patients with ET. We employ acoustic measures, tempo, and intelligibility ratings and patient's self-estimated speech to quantify SID, with a focus on comparing bilateral to unilateral stimulation effects and the effect of electrode position on speech. Sixteen German ET patients participated in this study. Each patient was acoustically recorded with DBS-off, unilateral-right-hemispheric-DBS-on, unilateral-left-hemispheric-DBS-on, and bilateral-DBS-on during an oral diadochokinesis task and a read German standard text. To capture the extent of speech impairment, we measured syllable duration and intensity ratio during the DDK task. Naïve listeners rated speech tempo and speech intelligibility of the read text on a 5-point-scale. Patients had to rate their "ability to speak". We found an effect of bilateral compared to unilateral and inactivated stimulation on syllable durations and intensity ratio, as well as on external intelligibility ratings and patients' VAS scores. Additionally, VAS scores are associated with more laterally located active contacts. For speech ratings, we found an effect of syllable duration such that tempo and intelligibility was rated worse for speakers exhibiting greater syllable durations. Our data confirms that SID is more pronounced under bilateral compared to unilateral stimulation. Laterally located electrodes are associated with more severe SID according to patient's self-ratings. We can confirm the relation between diadochokinetic rate and SID in that listener's tempo and intelligibility ratings can be

Amygdala and auditory cortex exhibit distinct sensitivity to relevant acoustic features of auditory emotions.

PubMed

Pannese, Alessia; Grandjean, Didier; Frühholz, Sascha

2016-12-01

Discriminating between auditory signals of different affective value is critical to successful social interaction. It is commonly held that acoustic decoding of such signals occurs in the auditory system, whereas affective decoding occurs in the amygdala. However, given that the amygdala receives direct subcortical projections that bypass the auditory cortex, it is possible that some acoustic decoding occurs in the amygdala as well, when the acoustic features are relevant for affective discrimination. We tested this hypothesis by combining functional neuroimaging with the neurophysiological phenomena of repetition suppression (RS) and repetition enhancement (RE) in human listeners. Our results show that both amygdala and auditory cortex responded differentially to physical voice features, suggesting that the amygdala and auditory cortex decode the affective quality of the voice not only by processing the emotional content from previously processed acoustic features, but also by processing the acoustic features themselves, when these are relevant to the identification of the voice's affective value. Specifically, we found that the auditory cortex is sensitive to spectral high-frequency voice cues when discriminating vocal anger from vocal fear and joy, whereas the amygdala is sensitive to vocal pitch when discriminating between negative vocal emotions (i.e., anger and fear). Vocal pitch is an instantaneously recognized voice feature, which is potentially transferred to the amygdala by direct subcortical projections. These results together provide evidence that, besides the auditory cortex, the amygdala too processes acoustic information, when this is relevant to the discrimination of auditory emotions. Copyright © 2016 Elsevier Ltd. All rights reserved.

Finding the music of speech: Musical knowledge influences pitch processing in speech.

PubMed

Vanden Bosch der Nederlanden, Christina M; Hannon, Erin E; Snyder, Joel S

2015-10-01

Few studies comparing music and language processing have adequately controlled for low-level acoustical differences, making it unclear whether differences in music and language processing arise from domain-specific knowledge, acoustic characteristics, or both. We controlled acoustic characteristics by using the speech-to-song illusion, which often results in a perceptual transformation to song after several repetitions of an utterance. Participants performed a same-different pitch discrimination task for the initial repetition (heard as speech) and the final repetition (heard as song). Better detection was observed for pitch changes that violated rather than conformed to Western musical scale structure, but only when utterances transformed to song, indicating that music-specific pitch representations were activated and influenced perception. This shows that music-specific processes can be activated when an utterance is heard as song, suggesting that the high-level status of a stimulus as either language or music can be behaviorally dissociated from low-level acoustic factors. Copyright © 2015 Elsevier B.V. All rights reserved.

Lexico-semantic and acoustic-phonetic processes in the perception of noise-vocoded speech: implications for cochlear implantation

PubMed Central

McGettigan, Carolyn; Rosen, Stuart; Scott, Sophie K.

2014-01-01

Noise-vocoding is a transformation which, when applied to speech, severely reduces spectral resolution and eliminates periodicity, yielding a stimulus that sounds “like a harsh whisper” (Scott et al., 2000, p. 2401). This process simulates a cochlear implant, where the activity of many thousand hair cells in the inner ear is replaced by direct stimulation of the auditory nerve by a small number of tonotopically-arranged electrodes. Although a cochlear implant offers a powerful means of restoring some degree of hearing to profoundly deaf individuals, the outcomes for spoken communication are highly variable (Moore and Shannon, 2009). Some variability may arise from differences in peripheral representation (e.g., the degree of residual nerve survival) but some may reflect differences in higher-order linguistic processing. In order to explore this possibility, we used noise-vocoding to explore speech recognition and perceptual learning in normal-hearing listeners tested across several levels of the linguistic hierarchy: segments (consonants and vowels), single words, and sentences. Listeners improved significantly on all tasks across two test sessions. In the first session, individual differences analyses revealed two independently varying sources of variability: one lexico-semantic in nature and implicating the recognition of words and sentences, and the other an acoustic-phonetic factor associated with words and segments. However, consequent to learning, by the second session there was a more uniform covariance pattern concerning all stimulus types. A further analysis of phonetic feature recognition allowed greater insight into learning-related changes in perception and showed that, surprisingly, participants did not make full use of cues that were preserved in the stimuli (e.g., vowel duration). We discuss these findings in relation cochlear implantation, and suggest auditory training strategies to maximize speech recognition performance in the absence of

Discrimination of speech stimuli based on neuronal response phase patterns depends on acoustics but not comprehension.

PubMed

Howard, Mary F; Poeppel, David

2010-11-01

Speech stimuli give rise to neural activity in the listener that can be observed as waveforms using magnetoencephalography. Although waveforms vary greatly from trial to trial due to activity unrelated to the stimulus, it has been demonstrated that spoken sentences can be discriminated based on theta-band (3-7 Hz) phase patterns in single-trial response waveforms. Furthermore, manipulations of the speech signal envelope and fine structure that reduced intelligibility were found to produce correlated reductions in discrimination performance, suggesting a relationship between theta-band phase patterns and speech comprehension. This study investigates the nature of this relationship, hypothesizing that theta-band phase patterns primarily reflect cortical processing of low-frequency (<40 Hz) modulations present in the acoustic signal and required for intelligibility, rather than processing exclusively related to comprehension (e.g., lexical, syntactic, semantic). Using stimuli that are quite similar to normal spoken sentences in terms of low-frequency modulation characteristics but are unintelligible (i.e., their time-inverted counterparts), we find that discrimination performance based on theta-band phase patterns is equal for both types of stimuli. Consistent with earlier findings, we also observe that whereas theta-band phase patterns differ across stimuli, power patterns do not. We use a simulation model of the single-trial response to spoken sentence stimuli to demonstrate that phase-locked responses to low-frequency modulations of the acoustic signal can account not only for the phase but also for the power results. The simulation offers insight into the interpretation of the empirical results with respect to phase-resetting and power-enhancement models of the evoked response.

Changes in acoustic features and their conjunctions are processed by separate neuronal populations.

PubMed

Takegata, R; Huotilainen, M; Rinne, T; Näätänen, R; Winkler, I

2001-03-05

We investigated the relationship between the neuronal populations involved in detecting change in two acoustic features and their conjunction. Equivalent current dipole (ECD) models of the magnetic mismatch negativity (MMNm) generators were calculated for infrequent changes in pitch, perceived sound source location, and the conjunction of these two features. All of these three changes elicited MMNms that were generated in the vicinity of auditory cortex. The location of the ECD best describing the MMNm to the conjunction deviant was anterior to those for the MMNm responses elicited by either one of the constituent features. The present data thus suggest that at least partially separate neuronal populations are involved in detecting change in acoustic features and feature conjunctions.

Prediction and constraint in audiovisual speech perception.

PubMed

Peelle, Jonathan E; Sommers, Mitchell S

2015-07-01

During face-to-face conversational speech listeners must efficiently process a rapid and complex stream of multisensory information. Visual speech can serve as a critical complement to auditory information because it provides cues to both the timing of the incoming acoustic signal (the amplitude envelope, influencing attention and perceptual sensitivity) and its content (place and manner of articulation, constraining lexical selection). Here we review behavioral and neurophysiological evidence regarding listeners' use of visual speech information. Multisensory integration of audiovisual speech cues improves recognition accuracy, particularly for speech in noise. Even when speech is intelligible based solely on auditory information, adding visual information may reduce the cognitive demands placed on listeners through increasing the precision of prediction. Electrophysiological studies demonstrate that oscillatory cortical entrainment to speech in auditory cortex is enhanced when visual speech is present, increasing sensitivity to important acoustic cues. Neuroimaging studies also suggest increased activity in auditory cortex when congruent visual information is available, but additionally emphasize the involvement of heteromodal regions of posterior superior temporal sulcus as playing a role in integrative processing. We interpret these findings in a framework of temporally-focused lexical competition in which visual speech information affects auditory processing to increase sensitivity to acoustic information through an early integration mechanism, and a late integration stage that incorporates specific information about a speaker's articulators to constrain the number of possible candidates in a spoken utterance. Ultimately it is words compatible with both auditory and visual information that most strongly determine successful speech perception during everyday listening. Thus, audiovisual speech perception is accomplished through multiple stages of integration

Asynchronous sampling of speech with some vocoder experimental results

NASA Technical Reports Server (NTRS)

Babcock, M. L.

1972-01-01

The method of asynchronously sampling speech is based upon the derivatives of the acoustical speech signal. The following results are apparent from experiments to date: (1) It is possible to represent speech by a string of pulses of uniform amplitude, where the only information contained in the string is the spacing of the pulses in time; (2) the string of pulses may be produced in a simple analog manner; (3) the first derivative of the original speech waveform is the most important for the encoding process; (4) the resulting pulse train can be utilized to control an acoustical signal production system to regenerate the intelligence of the original speech.

Speech enhancement based on modified phase-opponency detectors

NASA Astrophysics Data System (ADS)

Deshmukh, Om D.; Espy-Wilson, Carol Y.

2005-09-01

A speech enhancement algorithm based on a neural model was presented by Deshmukh et al., [149th meeting of the Acoustical Society America, 2005]. The algorithm consists of a bank of Modified Phase Opponency (MPO) filter pairs tuned to different center frequencies. This algorithm is able to enhance salient spectral features in speech signals even at low signal-to-noise ratios. However, the algorithm introduces musical noise and sometimes misses a spectral peak that is close in frequency to a stronger spectral peak. Refinement in the design of the MPO filters was recently made that takes advantage of the falling spectrum of the speech signal in sonorant regions. The modified set of filters leads to better separation of the noise and speech signals, and more accurate enhancement of spectral peaks. The improvements also lead to a significant reduction in musical noise. Continuity algorithms based on the properties of speech signals are used to further reduce the musical noise effect. The efficiency of the proposed method in enhancing the speech signal when the level of the background noise is fluctuating will be demonstrated. The performance of the improved speech enhancement method will be compared with various spectral subtraction-based methods. [Work supported by NSF BCS0236707.

Detection and Classification of Whale Acoustic Signals

NASA Astrophysics Data System (ADS)

Xian, Yin

vocalization data set. The word error rate of the DCTNet feature is similar to the MFSC in speech recognition tasks, suggesting that the convolutional network is able to reveal acoustic content of speech signals.

Extensions to the Speech Disorders Classification System (SDCS)

PubMed Central

Shriberg, Lawrence D.; Fourakis, Marios; Hall, Sheryl D.; Karlsson, Heather B.; Lohmeier, Heather L.; McSweeny, Jane L.; Potter, Nancy L.; Scheer-Cohen, Alison R.; Strand, Edythe A.; Tilkens, Christie M.; Wilson, David L.

2010-01-01

This report describes three extensions to a classification system for pediatric speech sound disorders termed the Speech Disorders Classification System (SDCS). Part I describes a classification extension to the SDCS to differentiate motor speech disorders from speech delay and to differentiate among three subtypes of motor speech disorders. Part II describes the Madison Speech Assessment Protocol (MSAP), an approximately two-hour battery of 25 measures that includes 15 speech tests and tasks. Part III describes the Competence, Precision, and Stability Analytics (CPSA) framework, a current set of approximately 90 perceptual- and acoustic-based indices of speech, prosody, and voice used to quantify and classify subtypes of Speech Sound Disorders (SSD). A companion paper, Shriberg, Fourakis, et al. (2010) provides reliability estimates for the perceptual and acoustic data reduction methods used in the SDCS. The agreement estimates in the companion paper support the reliability of SDCS methods and illustrate the complementary roles of perceptual and acoustic methods in diagnostic analyses of SSD of unknown origin. Examples of research using the extensions to the SDCS described in the present report include diagnostic findings for a sample of youth with motor speech disorders associated with galactosemia (Shriberg, Potter, & Strand, 2010) and a test of the hypothesis of apraxia of speech in a group of children with autism spectrum disorders (Shriberg, Paul, Black, & van Santen, 2010). All SDCS methods and reference databases running in the PEPPER (Programs to Examine Phonetic and Phonologic Evaluation Records; [Shriberg, Allen, McSweeny, & Wilson, 2001]) environment will be disseminated without cost when complete. PMID:20831378

Neuroscience-inspired computational systems for speech recognition under noisy conditions

NASA Astrophysics Data System (ADS)

Schafer, Phillip B.

Humans routinely recognize speech in challenging acoustic environments with background music, engine sounds, competing talkers, and other acoustic noise. However, today's automatic speech recognition (ASR) systems perform poorly in such environments. In this dissertation, I present novel methods for ASR designed to approach human-level performance by emulating the brain's processing of sounds. I exploit recent advances in auditory neuroscience to compute neuron-based representations of speech, and design novel methods for decoding these representations to produce word transcriptions. I begin by considering speech representations modeled on the spectrotemporal receptive fields of auditory neurons. These representations can be tuned to optimize a variety of objective functions, which characterize the response properties of a neural population. I propose an objective function that explicitly optimizes the noise invariance of the neural responses, and find that it gives improved performance on an ASR task in noise compared to other objectives. The method as a whole, however, fails to significantly close the performance gap with humans. I next consider speech representations that make use of spiking model neurons. The neurons in this method are feature detectors that selectively respond to spectrotemporal patterns within short time windows in speech. I consider a number of methods for training the response properties of the neurons. In particular, I present a method using linear support vector machines (SVMs) and show that this method produces spikes that are robust to additive noise. I compute the spectrotemporal receptive fields of the neurons for comparison with previous physiological results. To decode the spike-based speech representations, I propose two methods designed to work on isolated word recordings. The first method uses a classical ASR technique based on the hidden Markov model. The second method is a novel template-based recognition scheme that takes

Some behavioral and neurobiological constraints on theories of audiovisual speech integration: a review and suggestions for new directions.

PubMed

Altieri, Nicholas; Pisoni, David B; Townsend, James T

2011-01-01

Summerfield (1987) proposed several accounts of audiovisual speech perception, a field of research that has burgeoned in recent years. The proposed accounts included the integration of discrete phonetic features, vectors describing the values of independent acoustical and optical parameters, the filter function of the vocal tract, and articulatory dynamics of the vocal tract. The latter two accounts assume that the representations of audiovisual speech perception are based on abstract gestures, while the former two assume that the representations consist of symbolic or featural information obtained from visual and auditory modalities. Recent converging evidence from several different disciplines reveals that the general framework of Summerfield's feature-based theories should be expanded. An updated framework building upon the feature-based theories is presented. We propose a processing model arguing that auditory and visual brain circuits provide facilitatory information when the inputs are correctly timed, and that auditory and visual speech representations do not necessarily undergo translation into a common code during information processing. Future research on multisensory processing in speech perception should investigate the connections between auditory and visual brain regions, and utilize dynamic modeling tools to further understand the timing and information processing mechanisms involved in audiovisual speech integration.

A Deep Ensemble Learning Method for Monaural Speech Separation.

PubMed

Zhang, Xiao-Lei; Wang, DeLiang

2016-03-01

Monaural speech separation is a fundamental problem in robust speech processing. Recently, deep neural network (DNN)-based speech separation methods, which predict either clean speech or an ideal time-frequency mask, have demonstrated remarkable performance improvement. However, a single DNN with a given window length does not leverage contextual information sufficiently, and the differences between the two optimization objectives are not well understood. In this paper, we propose a deep ensemble method, named multicontext networks, to address monaural speech separation. The first multicontext network averages the outputs of multiple DNNs whose inputs employ different window lengths. The second multicontext network is a stack of multiple DNNs. Each DNN in a module of the stack takes the concatenation of original acoustic features and expansion of the soft output of the lower module as its input, and predicts the ratio mask of the target speaker; the DNNs in the same module employ different contexts. We have conducted extensive experiments with three speech corpora. The results demonstrate the effectiveness of the proposed method. We have also compared the two optimization objectives systematically and found that predicting the ideal time-frequency mask is more efficient in utilizing clean training speech, while predicting clean speech is less sensitive to SNR variations.

Analysis of False Starts in Spontaneous Speech.

ERIC Educational Resources Information Center

O'Shaughnessy, Douglas

A primary difference between spontaneous speech and read speech concerns the use of false starts, where a speaker interrupts the flow of speech to restart his or her utterance. A study examined the acoustic aspects of such restarts in a widely-used speech database, examining approximately 1000 utterances, about 10% of which contained a restart.…

Dimension-based statistical learning affects both speech perception and production

PubMed Central

Lehet, Matthew; Holt, Lori L.

2016-01-01

Multiple acoustic dimensions signal speech categories. However, dimensions vary in their informativeness; some are more diagnostic of category membership than others. Speech categorization reflects these dimensional regularities such that diagnostic dimensions carry more “perceptual weight” and more effectively signal category membership to native listeners. Yet, perceptual weights are malleable. When short-term experience deviates from long-term language norms, such as in a foreign accent, the perceptual weight of acoustic dimensions in signaling speech category membership rapidly adjusts. The present study investigated whether rapid adjustments in listeners’ perceptual weights in response to speech that deviates from the norms also affects listeners’ own speech productions. In a word recognition task, the correlation between two acoustic dimensions signaling consonant categories, fundamental frequency (F0) and voice onset time (VOT), matched the correlation typical of English, then shifted to an “artificial accent” that reversed the relationship, and then shifted back. Brief, incidental exposure to the artificial accent caused participants to down-weight perceptual reliance on F0, consistent with previous research. Throughout the task, participants were intermittently prompted with pictures to produce these same words. In the block in which listeners heard the artificial accent with a reversed F0 x VOT correlation, F0 was a less robust cue to voicing in listeners’ own speech productions. The statistical regularities of short-term speech input affect both speech perception and production, as evidenced via shifts in how acoustic dimensions are weighted. PMID:27666146

Deep Brain Stimulation of the Subthalamic Nucleus Parameter Optimization for Vowel Acoustics and Speech Intelligibility in Parkinson's Disease

ERIC Educational Resources Information Center

Knowles, Thea; Adams, Scott; Abeyesekera, Anita; Mancinelli, Cynthia; Gilmore, Greydon; Jog, Mandar

2018-01-01

Purpose: The settings of 3 electrical stimulation parameters were adjusted in 12 speakers with Parkinson's disease (PD) with deep brain stimulation of the subthalamic nucleus (STN-DBS) to examine their effects on vowel acoustics and speech intelligibility. Method: Participants were tested under permutations of low, mid, and high STN-DBS frequency,…

Characterizing Articulation in Apraxic Speech Using Real-Time Magnetic Resonance Imaging.

PubMed

Hagedorn, Christina; Proctor, Michael; Goldstein, Louis; Wilson, Stephen M; Miller, Bruce; Gorno-Tempini, Maria Luisa; Narayanan, Shrikanth S

2017-04-14

Real-time magnetic resonance imaging (MRI) and accompanying analytical methods are shown to capture and quantify salient aspects of apraxic speech, substantiating and expanding upon evidence provided by clinical observation and acoustic and kinematic data. Analysis of apraxic speech errors within a dynamic systems framework is provided and the nature of pathomechanisms of apraxic speech discussed. One adult male speaker with apraxia of speech was imaged using real-time MRI while producing spontaneous speech, repeated naming tasks, and self-paced repetition of word pairs designed to elicit speech errors. Articulatory data were analyzed, and speech errors were detected using time series reflecting articulatory activity in regions of interest. Real-time MRI captured two types of apraxic gestural intrusion errors in a word pair repetition task. Gestural intrusion errors in nonrepetitive speech, multiple silent initiation gestures at the onset of speech, and covert (unphonated) articulation of entire monosyllabic words were also captured. Real-time MRI and accompanying analytical methods capture and quantify many features of apraxic speech that have been previously observed using other modalities while offering high spatial resolution. This patient's apraxia of speech affected the ability to select only the appropriate vocal tract gestures for a target utterance, suppressing others, and to coordinate them in time.

Recovering With Acquired Apraxia of Speech: The First 2 Years.

PubMed

Haley, Katarina L; Shafer, Jennifer N; Harmon, Tyson G; Jacks, Adam

2016-12-01

This study was intended to document speech recovery for 1 person with acquired apraxia of speech quantitatively and on the basis of her lived experience. The second author sustained a traumatic brain injury that resulted in acquired apraxia of speech. Over a 2-year period, she documented her recovery through 22 video-recorded monologues. We analyzed these monologues using a combination of auditory perceptual, acoustic, and qualitative methods. Recovery was evident for all quantitative variables examined. For speech sound production, the recovery was most prominent during the first 3 months, but slower improvement was evident for many months. Measures of speaking rate, fluency, and prosody changed more gradually throughout the entire period. A qualitative analysis of topics addressed in the monologues was consistent with the quantitative speech recovery and indicated a subjective dynamic relationship between accuracy and rate, an observation that several factors made speech sound production variable, and a persisting need for cognitive effort while speaking. Speech features improved over an extended time, but the recovery trajectories differed, indicating dynamic reorganization of the underlying speech production system. The relationship among speech dimensions should be examined in other cases and in population samples. The combination of quantitative and qualitative analysis methods offers advantages for understanding clinically relevant aspects of recovery.

Neurophysiological Influence of Musical Training on Speech Perception

PubMed Central

Shahin, Antoine J.

2011-01-01

Does musical training affect our perception of speech? For example, does learning to play a musical instrument modify the neural circuitry for auditory processing in a way that improves one's ability to perceive speech more clearly in noisy environments? If so, can speech perception in individuals with hearing loss (HL), who struggle in noisy situations, benefit from musical training? While music and speech exhibit some specialization in neural processing, there is evidence suggesting that skills acquired through musical training for specific acoustical processes may transfer to, and thereby improve, speech perception. The neurophysiological mechanisms underlying the influence of musical training on speech processing and the extent of this influence remains a rich area to be explored. A prerequisite for such transfer is the facilitation of greater neurophysiological overlap between speech and music processing following musical training. This review first establishes a neurophysiological link between musical training and speech perception, and subsequently provides further hypotheses on the neurophysiological implications of musical training on speech perception in adverse acoustical environments and in individuals with HL. PMID:21716639

Neurophysiological influence of musical training on speech perception.

PubMed

Shahin, Antoine J

2011-01-01

Does musical training affect our perception of speech? For example, does learning to play a musical instrument modify the neural circuitry for auditory processing in a way that improves one's ability to perceive speech more clearly in noisy environments? If so, can speech perception in individuals with hearing loss (HL), who struggle in noisy situations, benefit from musical training? While music and speech exhibit some specialization in neural processing, there is evidence suggesting that skills acquired through musical training for specific acoustical processes may transfer to, and thereby improve, speech perception. The neurophysiological mechanisms underlying the influence of musical training on speech processing and the extent of this influence remains a rich area to be explored. A prerequisite for such transfer is the facilitation of greater neurophysiological overlap between speech and music processing following musical training. This review first establishes a neurophysiological link between musical training and speech perception, and subsequently provides further hypotheses on the neurophysiological implications of musical training on speech perception in adverse acoustical environments and in individuals with HL.

Hearing speech in music.

PubMed

Ekström, Seth-Reino; Borg, Erik

2011-01-01

The masking effect of a piano composition, played at different speeds and in different octaves, on speech-perception thresholds was investigated in 15 normal-hearing and 14 moderately-hearing-impaired subjects. Running speech (just follow conversation, JFC) testing and use of hearing aids increased the everyday validity of the findings. A comparison was made with standard audiometric noises [International Collegium of Rehabilitative Audiology (ICRA) noise and speech spectrum-filtered noise (SPN)]. All masking sounds, music or noise, were presented at the same equivalent sound level (50 dBA). The results showed a significant effect of piano performance speed and octave (P<.01). Low octave and fast tempo had the largest effect; and high octave and slow tempo, the smallest. Music had a lower masking effect than did ICRA noise with two or six speakers at normal vocal effort (P<.01) and SPN (P<.05). Subjects with hearing loss had higher masked thresholds than the normal-hearing subjects (P<.01), but there were smaller differences between masking conditions (P<.01). It is pointed out that music offers an interesting opportunity for studying masking under realistic conditions, where spectral and temporal features can be varied independently. The results have implications for composing music with vocal parts, designing acoustic environments and creating a balance between speech perception and privacy in social settings.

Lexical effects on speech production and intelligibility in Parkinson's disease

NASA Astrophysics Data System (ADS)

Chiu, Yi-Fang

Individuals with Parkinson's disease (PD) often have speech deficits that lead to reduced speech intelligibility. Previous research provides a rich database regarding the articulatory deficits associated with PD including restricted vowel space (Skodda, Visser, & Schlegel, 2011) and flatter formant transitions (Tjaden & Wilding, 2004; Walsh & Smith, 2012). However, few studies consider the effect of higher level structural variables of word usage frequency and the number of similar sounding words (i.e. neighborhood density) on lower level articulation or on listeners' perception of dysarthric speech. The purpose of the study is to examine the interaction of lexical properties and speech articulation as measured acoustically in speakers with PD and healthy controls (HC) and the effect of lexical properties on the perception of their speech. Individuals diagnosed with PD and age-matched healthy controls read sentences with words that varied in word frequency and neighborhood density. Acoustic analysis was performed to compare second formant transitions in diphthongs, an indicator of the dynamics of tongue movement during speech production, across different lexical characteristics. Young listeners transcribed the spoken sentences and the transcription accuracy was compared across lexical conditions. The acoustic results indicate that both PD and HC speakers adjusted their articulation based on lexical properties but the PD group had significant reductions in second formant transitions compared to HC. Both groups of speakers increased second formant transitions for words with low frequency and low density, but the lexical effect is diphthong dependent. The change in second formant slope was limited in the PD group when the required formant movement for the diphthong is small. The data from listeners' perception of the speech by PD and HC show that listeners identified high frequency words with greater accuracy suggesting the use of lexical knowledge during the

Phrase-level speech simulation with an airway modulation model of speech production

PubMed Central

Story, Brad H.

2012-01-01

Artificial talkers and speech synthesis systems have long been used as a means of understanding both speech production and speech perception. The development of an airway modulation model is described that simulates the time-varying changes of the glottis and vocal tract, as well as acoustic wave propagation, during speech production. The result is a type of artificial talker that can be used to study various aspects of how sound is generated by humans and how that sound is perceived by a listener. The primary components of the model are introduced and simulation of words and phrases are demonstrated. PMID:23503742

[Vocal effectiveness in speech and singing: acoustical, physiological and perceptive aspects. applications in speech therapy].

PubMed

Pillot, C; Vaissière, J

2006-01-01

What is vocal effectiveness in lyrical singing in comparison to speech? Our study tries to answer this question, using vocal efficiency and spectral vocal effectiveness. Vocal efficiency was mesured for a trained and untrained subject. According to these invasive measures, it appears that the trained singer uses her larynx less efficiently. Efficiency of the larynx in terms of energy then appears to be secondary to the desired voice quality. The acoustic measures of spectral vocal effectiveness of vowels and sentences, spoken and sung by 23 singers, reveal two complementary markers: The "singing power ratio" and the difference in amplitude between the singing formant and the spectral minimum that follows it. Magnetic resonance imaging and simulations of [a], [i] and [o] spoken and sung show laryngeal lowering and the role of the piriform sinuses as the physiological foundations of spectral vocal effectiveness, perceptively related to carrying power. These scientifical aspects allow applications in voice therapy, such as physiological and perceptual foundations allowing patients to recuperate voice carrying power with or without background noise.

Is Birdsong More Like Speech or Music?

PubMed

Shannon, Robert V

2016-04-01

Music and speech share many acoustic cues but not all are equally important. For example, harmonic pitch is essential for music but not for speech. When birds communicate is their song more like speech or music? A new study contrasting pitch and spectral patterns shows that birds perceive their song more like humans perceive speech. Copyright © 2016 Elsevier Ltd. All rights reserved.

Articulatory-to-Acoustic Relations in Response to Speaking Rate and Loudness Manipulations

ERIC Educational Resources Information Center

Mefferd, Antje S.; Green, Jordan R.

2010-01-01

Purpose: In this investigation, the authors determined the strength of association between tongue kinematic and speech acoustics changes in response to speaking rate and loudness manipulations. Performance changes in the kinematic and acoustic domains were measured using two aspects of speech production presumably affecting speech clarity:…

A survey of acoustic conditions in semi-open plan classrooms in the United Kingdom.

PubMed

Greenland, Emma E; Shield, Bridget M

2011-09-01

This paper reports the results of a large scale, detailed acoustic survey of 42 open plan classrooms of varying design in the UK each of which contained between 2 and 14 teaching areas or classbases. The objective survey procedure, which was designed specifically for use in open plan classrooms, is described. The acoustic measurements relating to speech intelligibility within a classbase, including ambient noise level, intrusive noise level, speech to noise ratio, speech transmission index, and reverberation time, are presented. The effects on speech intelligibility of critical physical design variables, such as the number of classbases within an open plan unit and the selection of acoustic finishes for control of reverberation, are examined. This analysis enables limitations of open plan classrooms to be discussed and acoustic design guidelines to be developed to ensure good listening conditions. The types of teaching activity to provide adequate acoustic conditions, plus the speech intelligibility requirements of younger children, are also discussed. © 2011 Acoustical Society of America

Improved segregation of simultaneous talkers differentially affects perceptual and cognitive capacity demands for recognizing speech in competing speech.

PubMed

Francis, Alexander L

2010-02-01

Perception of speech in competing speech is facilitated by spatial separation of the target and distracting speech, but this benefit may arise at either a perceptual or a cognitive level of processing. Load theory predicts different effects of perceptual and cognitive (working memory) load on selective attention in flanker task contexts, suggesting that this paradigm may be used to distinguish levels of interference. Two experiments examined interference from competing speech during a word recognition task under different perceptual and working memory loads in a dual-task paradigm. Listeners identified words produced by a talker of one gender while ignoring a talker of the other gender. Perceptual load was manipulated using a nonspeech response cue, with response conditional upon either one or two acoustic features (pitch and modulation). Memory load was manipulated with a secondary task consisting of one or six visually presented digits. In the first experiment, the target and distractor were presented at different virtual locations (0 degrees and 90 degrees , respectively), whereas in the second, all the stimuli were presented from the same apparent location. Results suggest that spatial cues improve resistance to distraction in part by reducing working memory demand.

Formant trajectory characteristics in speakers with dysarthria and homogeneous speech intelligibility scores: Further data

NASA Astrophysics Data System (ADS)

Kim, Yunjung; Weismer, Gary; Kent, Ray D.

2005-09-01

In previous work [J. Acoust. Soc. Am. 117, 2605 (2005)], we reported on formant trajectory characteristics of a relatively large number of speakers with dysarthria and near-normal speech intelligibility. The purpose of that analysis was to begin a documentation of the variability, within relatively homogeneous speech-severity groups, of acoustic measures commonly used to predict across-speaker variation in speech intelligibility. In that study we found that even with near-normal speech intelligibility (90%-100%), many speakers had reduced formant slopes for some words and distributional characteristics of acoustic measures that were different than values obtained from normal speakers. In the current report we extend those findings to a group of speakers with dysarthria with somewhat poorer speech intelligibility than the original group. Results are discussed in terms of the utility of certain acoustic measures as indices of speech intelligibility, and as explanatory data for theories of dysarthria. [Work supported by NIH Award R01 DC00319.

Normal Aspects of Speech, Hearing, and Language.

ERIC Educational Resources Information Center

Minifie, Fred. D., Ed.; And Others

This book is written as a guide to the understanding of the processes involved in human speech communication. Ten authorities contributed material to provide an introduction to the physiological aspects of speech production and reception, the acoustical aspects of speech production and transmission, the psychophysics of sound reception, the nature…

Tutorial on architectural acoustics

NASA Astrophysics Data System (ADS)

Shaw, Neil; Talaske, Rick; Bistafa, Sylvio

2002-11-01

This tutorial is intended to provide an overview of current knowledge and practice in architectural acoustics. Topics covered will include basic concepts and history, acoustics of small rooms (small rooms for speech such as classrooms and meeting rooms, music studios, small critical listening spaces such as home theatres) and the acoustics of large rooms (larger assembly halls, auditoria, and performance halls).

Human emotions track changes in the acoustic environment.

PubMed

Ma, Weiyi; Thompson, William Forde

2015-11-24

Emotional responses to biologically significant events are essential for human survival. Do human emotions lawfully track changes in the acoustic environment? Here we report that changes in acoustic attributes that are well known to interact with human emotions in speech and music also trigger systematic emotional responses when they occur in environmental sounds, including sounds of human actions, animal calls, machinery, or natural phenomena, such as wind and rain. Three changes in acoustic attributes known to signal emotional states in speech and music were imposed upon 24 environmental sounds. Evaluations of stimuli indicated that human emotions track such changes in environmental sounds just as they do for speech and music. Such changes not only influenced evaluations of the sounds themselves, they also affected the way accompanying facial expressions were interpreted emotionally. The findings illustrate that human emotions are highly attuned to changes in the acoustic environment, and reignite a discussion of Charles Darwin's hypothesis that speech and music originated from a common emotional signal system based on the imitation and modification of environmental sounds.

Human emotions track changes in the acoustic environment

PubMed Central

Ma, Weiyi; Thompson, William Forde

2015-01-01

Emotional responses to biologically significant events are essential for human survival. Do human emotions lawfully track changes in the acoustic environment? Here we report that changes in acoustic attributes that are well known to interact with human emotions in speech and music also trigger systematic emotional responses when they occur in environmental sounds, including sounds of human actions, animal calls, machinery, or natural phenomena, such as wind and rain. Three changes in acoustic attributes known to signal emotional states in speech and music were imposed upon 24 environmental sounds. Evaluations of stimuli indicated that human emotions track such changes in environmental sounds just as they do for speech and music. Such changes not only influenced evaluations of the sounds themselves, they also affected the way accompanying facial expressions were interpreted emotionally. The findings illustrate that human emotions are highly attuned to changes in the acoustic environment, and reignite a discussion of Charles Darwin’s hypothesis that speech and music originated from a common emotional signal system based on the imitation and modification of environmental sounds. PMID:26553987

Abnormal laughter-like vocalisations replacing speech in primary progressive aphasia.

PubMed

Rohrer, Jonathan D; Warren, Jason D; Rossor, Martin N

2009-09-15

We describe ten patients with a clinical diagnosis of primary progressive aphasia (PPA) (pathologically confirmed in three cases) who developed abnormal laughter-like vocalisations in the context of progressive speech output impairment leading to mutism. Failure of speech output was accompanied by increasing frequency of the abnormal vocalisations until ultimately they constituted the patient's only extended utterance. The laughter-like vocalisations did not show contextual sensitivity but occurred as an automatic vocal output that replaced speech. Acoustic analysis of the vocalisations in two patients revealed abnormal motor features including variable note duration and inter-note interval, loss of temporal symmetry of laugh notes and loss of the normal decrescendo. Abnormal laughter-like vocalisations may be a hallmark of a subgroup in the PPA spectrum with impaired control and production of nonverbal vocal behaviour due to disruption of fronto-temporal networks mediating vocalisation.

Dimension-Based Statistical Learning Affects Both Speech Perception and Production.

PubMed

Lehet, Matthew; Holt, Lori L

2017-04-01

Multiple acoustic dimensions signal speech categories. However, dimensions vary in their informativeness; some are more diagnostic of category membership than others. Speech categorization reflects these dimensional regularities such that diagnostic dimensions carry more "perceptual weight" and more effectively signal category membership to native listeners. Yet perceptual weights are malleable. When short-term experience deviates from long-term language norms, such as in a foreign accent, the perceptual weight of acoustic dimensions in signaling speech category membership rapidly adjusts. The present study investigated whether rapid adjustments in listeners' perceptual weights in response to speech that deviates from the norms also affects listeners' own speech productions. In a word recognition task, the correlation between two acoustic dimensions signaling consonant categories, fundamental frequency (F0) and voice onset time (VOT), matched the correlation typical of English, and then shifted to an "artificial accent" that reversed the relationship, and then shifted back. Brief, incidental exposure to the artificial accent caused participants to down-weight perceptual reliance on F0, consistent with previous research. Throughout the task, participants were intermittently prompted with pictures to produce these same words. In the block in which listeners heard the artificial accent with a reversed F0 × VOT correlation, F0 was a less robust cue to voicing in listeners' own speech productions. The statistical regularities of short-term speech input affect both speech perception and production, as evidenced via shifts in how acoustic dimensions are weighted. Copyright © 2016 Cognitive Science Society, Inc.

Rating, ranking, and understanding acoustical quality in university classrooms

NASA Astrophysics Data System (ADS)

Hodgson, Murray

2002-08-01

Nonoptimal classroom acoustical conditions directly affect speech perception and, thus, learning by students. Moreover, they may lead to voice problems for the instructor, who is forced to raise his/her voice when lecturing to compensate for poor acoustical conditions. The project applied previously developed simplified methods to predict speech intelligibility in occupied classrooms from measurements in unoccupied and occupied university classrooms. The methods were used to predict the speech intelligibility at various positions in 279 University of British Columbia (UBC) classrooms, when 70% occupied, and for four instructor voice levels. Classrooms were classified and rank ordered by acoustical quality, as determined by the room-average speech intelligibility. This information was used by UBC to prioritize classrooms for renovation. Here, the statistical results are reported to illustrate the range of acoustical qualities found at a typical university. Moreover, the variations of quality with relevant classroom acoustical parameters were studied to better understand the results. In particular, the factors leading to the best and worst conditions were studied. It was found that 81% of the 279 classrooms have "good," "very good," or "excellent" acoustical quality with a "typical" (average-male) instructor. However, 50 (18%) of the classrooms had "fair" or "poor" quality, and two had "bad" quality, due to high ventilation-noise levels. Most rooms were "very good" or "excellent" at the front, and "good" or "very good" at the back. Speech quality varied strongly with the instructor voice level. In the worst case considered, with a quiet female instructor, most of the classrooms were "bad" or "poor." Quality also varies with occupancy, with decreased occupancy resulting in decreased quality. The research showed that a new classroom acoustical design and renovation should focus on limiting background noise. They should promote high instructor speech levels at the back

Acoustic analysis of changes in articulation proficiency in patients with advanced head and neck cancer treated with chemoradiotherapy.

PubMed

Jacobi, Irene; van Rossum, Maya A; van der Molen, Lisette; Hilgers, Frans J M; van den Brekel, Michiel W M

2013-12-01

Our aim was to characterize articulation proficiency and differences between tumor sites before and after chemoradiotherapy for advanced head and neck cancer with the help of acoustic measures. Our further goal was to improve objective speech measures and gain insight into muscle functioning before and after treatment. In 34 patients with laryngeal or hypopharyngeal, nasal or nasopharyngeal, or oral or oropharyngeal cancer, we acoustically analyzed nasality, vowel space, precision, and strength of articulation in 12 speech sounds (/a/, /i/, /u/, /p/, /s/, /z/, /1/, /t/, /tj/, /k/, /x/, /r/) before treatment and 10 weeks and 1 year after treatment. Outcomes were compared between assessment points and between tumor sites. Nasality in nonlaryngeal sites was significantly reduced by treatment. Most affected in articulation were the oral or oropharyngeal cancer sites, followed by the nasal or nasopharyngeal sites. One year after treatment, vowel space had not recovered and consonant articulation had weakened. Laryngeal sites were less affected in articulation by tumor or treatment. Analyses of articulatory-acoustic features are a useful instrument for assessing articulation and speech quality objectively. Assessment of a number of sounds representing various articulation manners, places, and tongue shapes revealed patterns of speech deterioration after chemoradiotherapy. The results suggest that patients' speech could benefit from articulation exercises to address changes in muscle coordination and/or sensitivity and to counteract side effects and "underexercise" atrophy.

Phase-Locked Responses to Speech in Human Auditory Cortex are Enhanced During Comprehension

PubMed Central

Peelle, Jonathan E.; Gross, Joachim; Davis, Matthew H.

2013-01-01

A growing body of evidence shows that ongoing oscillations in auditory cortex modulate their phase to match the rhythm of temporally regular acoustic stimuli, increasing sensitivity to relevant environmental cues and improving detection accuracy. In the current study, we test the hypothesis that nonsensory information provided by linguistic content enhances phase-locked responses to intelligible speech in the human brain. Sixteen adults listened to meaningful sentences while we recorded neural activity using magnetoencephalography. Stimuli were processed using a noise-vocoding technique to vary intelligibility while keeping the temporal acoustic envelope consistent. We show that the acoustic envelopes of sentences contain most power between 4 and 7 Hz and that it is in this frequency band that phase locking between neural activity and envelopes is strongest. Bilateral oscillatory neural activity phase-locked to unintelligible speech, but this cerebro-acoustic phase locking was enhanced when speech was intelligible. This enhanced phase locking was left lateralized and localized to left temporal cortex. Together, our results demonstrate that entrainment to connected speech does not only depend on acoustic characteristics, but is also affected by listeners’ ability to extract linguistic information. This suggests a biological framework for speech comprehension in which acoustic and linguistic cues reciprocally aid in stimulus prediction. PMID:22610394

Phase-locked responses to speech in human auditory cortex are enhanced during comprehension.

PubMed

Peelle, Jonathan E; Gross, Joachim; Davis, Matthew H

2013-06-01

A growing body of evidence shows that ongoing oscillations in auditory cortex modulate their phase to match the rhythm of temporally regular acoustic stimuli, increasing sensitivity to relevant environmental cues and improving detection accuracy. In the current study, we test the hypothesis that nonsensory information provided by linguistic content enhances phase-locked responses to intelligible speech in the human brain. Sixteen adults listened to meaningful sentences while we recorded neural activity using magnetoencephalography. Stimuli were processed using a noise-vocoding technique to vary intelligibility while keeping the temporal acoustic envelope consistent. We show that the acoustic envelopes of sentences contain most power between 4 and 7 Hz and that it is in this frequency band that phase locking between neural activity and envelopes is strongest. Bilateral oscillatory neural activity phase-locked to unintelligible speech, but this cerebro-acoustic phase locking was enhanced when speech was intelligible. This enhanced phase locking was left lateralized and localized to left temporal cortex. Together, our results demonstrate that entrainment to connected speech does not only depend on acoustic characteristics, but is also affected by listeners' ability to extract linguistic information. This suggests a biological framework for speech comprehension in which acoustic and linguistic cues reciprocally aid in stimulus prediction.

Sensitivity to Structure in the Speech Signal by Children with Speech Sound Disorder and Reading Disability

PubMed Central

Johnson, Erin Phinney; Pennington, Bruce F.; Lowenstein, Joanna H.; Nittrouer, Susan

2011-01-01

Purpose Children with speech sound disorder (SSD) and reading disability (RD) have poor phonological awareness, a problem believed to arise largely from deficits in processing the sensory information in speech, specifically individual acoustic cues. However, such cues are details of acoustic structure. Recent theories suggest that listeners also need to be able to integrate those details to perceive linguistically relevant form. This study examined abilities of children with SSD, RD, and SSD+RD not only to process acoustic cues but also to recover linguistically relevant form from the speech signal. Method Ten- to 11-year-olds with SSD (n = 17), RD (n = 16), SSD+RD (n = 17), and Controls (n = 16) were tested to examine their sensitivity to (1) voice onset times (VOT); (2) spectral structure in fricative-vowel syllables; and (3) vocoded sentences. Results Children in all groups performed similarly with VOT stimuli, but children with disorders showed delays on other tasks, although the specifics of their performance varied. Conclusion Children with poor phonemic awareness not only lack sensitivity to acoustic details, but are also less able to recover linguistically relevant forms. This is contrary to one of the main current theories of the relation between spoken and written language development. PMID:21329941

Histogram equalization with Bayesian estimation for noise robust speech recognition.

PubMed

Suh, Youngjoo; Kim, Hoirin

2018-02-01

The histogram equalization approach is an efficient feature normalization technique for noise robust automatic speech recognition. However, it suffers from performance degradation when some fundamental conditions are not satisfied in the test environment. To remedy these limitations of the original histogram equalization methods, class-based histogram equalization approach has been proposed. Although this approach showed substantial performance improvement under noise environments, it still suffers from performance degradation due to the overfitting problem when test data are insufficient. To address this issue, the proposed histogram equalization technique employs the Bayesian estimation method in the test cumulative distribution function estimation. It was reported in a previous study conducted on the Aurora-4 task that the proposed approach provided substantial performance gains in speech recognition systems based on the acoustic modeling of the Gaussian mixture model-hidden Markov model. In this work, the proposed approach was examined in speech recognition systems with deep neural network-hidden Markov model (DNN-HMM), the current mainstream speech recognition approach where it also showed meaningful performance improvement over the conventional maximum likelihood estimation-based method. The fusion of the proposed features with the mel-frequency cepstral coefficients provided additional performance gains in DNN-HMM systems, which otherwise suffer from performance degradation in the clean test condition.

Top-down Processes in Simulated Electric-Acoustic Hearing: The Effect of Linguistic Context on Bimodal Benefit for Temporally Interrupted Speech

PubMed Central

Oh, Soo Hee; Donaldson, Gail S.; Kong, Ying-Yee

2016-01-01

Objectives Previous studies have documented the benefits of bimodal hearing as compared with a CI alone, but most have focused on the importance of bottom-up, low-frequency cues. The purpose of the present study was to evaluate the role of top-down processing in bimodal hearing by measuring the effect of sentence context on bimodal benefit for temporally interrupted sentences. It was hypothesized that low-frequency acoustic cues would facilitate the use of contextual information in the interrupted sentences, resulting in greater bimodal benefit for the higher context (CUNY) sentences than for the lower context (IEEE) sentences. Design Young normal-hearing listeners were tested in simulated bimodal listening conditions in which noise band vocoded sentences were presented to one ear with or without low-pass (LP) filtered speech or LP harmonic complexes (LPHCs) presented to the contralateral ear. Speech recognition scores were measured in three listening conditions: vocoder-alone, vocoder combined with LP speech, and vocoder combined with LPHCs. Temporally interrupted versions of the CUNY and IEEE sentences were used to assess listeners’ ability to fill in missing segments of speech by using top-down linguistic processing. Sentences were square-wave gated at a rate of 5 Hz with a 50 percent duty cycle. Three vocoder channel conditions were tested for each type of sentence (8, 12, and 16 channels for CUNY; 12, 16, and 32 channels for IEEE) and bimodal benefit was compared for similar amounts of spectral degradation (matched-channel comparisons) and similar ranges of baseline performance. Two gain measures, percentage-point gain and normalized gain, were examined. Results Significant effects of context on bimodal benefit were observed when LP speech was presented to the residual-hearing ear. For the matched-channel comparisons, CUNY sentences showed significantly higher normalized gains than IEEE sentences for both the 12-channel (20 points higher) and 16-channel (18

Perceived gender in clear and conversational speech

NASA Astrophysics Data System (ADS)

Booz, Jaime A.

Although many studies have examined acoustic and sociolinguistic differences between male and female speech, the relationship between talker speaking style and perceived gender has not yet been explored. The present study attempts to determine whether clear speech, a style adopted by talkers who perceive some barrier to effective communication, shifts perceptions of femininity for male and female talkers. Much of our understanding of gender perception in voice and speech is based on sustained vowels or single words, eliminating temporal, prosodic, and articulatory cues available in more naturalistic, connected speech. Thus, clear and conversational sentence stimuli, selected from the 41 talkers of the Ferguson Clear Speech Database (Ferguson, 2004) were presented to 17 normal-hearing listeners, aged 18 to 30. They rated the talkers' gender using a visual analog scale with "masculine" and "feminine" endpoints. This response method was chosen to account for within-category shifts of gender perception by allowing nonbinary responses. Mixed-effects regression analysis of listener responses revealed a small but significant effect of speaking style, and this effect was larger for male talkers than female talkers. Because of the high degree of talker variability observed for talker gender, acoustic analyses of these sentences were undertaken to determine the relationship between acoustic changes in clear and conversational speech and perceived femininity. Results of these analyses showed that mean fundamental frequency (fo) and f o standard deviation were significantly correlated to perceived gender for both male and female talkers, and vowel space was significantly correlated only for male talkers. Speaking rate and breathiness measures (CPPS) were not significantly related for either group. Outcomes of this study indicate that adopting a clear speaking style is correlated with increases in perceived femininity. Although the increase was small, some changes associated

Some Behavioral and Neurobiological Constraints on Theories of Audiovisual Speech Integration: A Review and Suggestions for New Directions

PubMed Central

Altieri, Nicholas; Pisoni, David B.; Townsend, James T.

2012-01-01

Summerfield (1987) proposed several accounts of audiovisual speech perception, a field of research that has burgeoned in recent years. The proposed accounts included the integration of discrete phonetic features, vectors describing the values of independent acoustical and optical parameters, the filter function of the vocal tract, and articulatory dynamics of the vocal tract. The latter two accounts assume that the representations of audiovisual speech perception are based on abstract gestures, while the former two assume that the representations consist of symbolic or featural information obtained from visual and auditory modalities. Recent converging evidence from several different disciplines reveals that the general framework of Summerfield’s feature-based theories should be expanded. An updated framework building upon the feature-based theories is presented. We propose a processing model arguing that auditory and visual brain circuits provide facilitatory information when the inputs are correctly timed, and that auditory and visual speech representations do not necessarily undergo translation into a common code during information processing. Future research on multisensory processing in speech perception should investigate the connections between auditory and visual brain regions, and utilize dynamic modeling tools to further understand the timing and information processing mechanisms involved in audiovisual speech integration. PMID:21968081

Speech interference and transmission on residential balconies with road traffic noise.

PubMed

Naish, Daniel A; Tan, Andy C C; Nur Demirbilek, F

2013-01-01

Balcony acoustic treatments can mitigate the effects of community road traffic noise. To further investigate, a theoretical study into the effects of balcony acoustic treatment combinations on speech interference and transmission is conducted for various street geometries. Nine different balcony types are investigated using a combined specular and diffuse reflection computer model. Diffusion in the model is calculated using the radiosity technique. The balcony types include a standard balcony with or without a ceiling and with various combinations of parapet, ceiling absorption and ceiling shield. A total of 70 balcony and street geometrical configurations are analyzed with each balcony type, resulting in 630 scenarios. In each scenario the reverberation time, speech interference level (SIL) and speech transmission index (STI) are calculated. These indicators are compared to determine trends based on the effects of propagation path, inclusion of opposite buildings and difference with a reference position outside the balcony. The results demonstrate trends in SIL and STI with different balcony types. It is found that an acoustically treated balcony reduces speech interference. A parapet provides the largest improvement, followed by absorption on the ceiling. The largest reductions in speech interference arise when a combination of balcony acoustic treatments are applied.

Applications for Subvocal Speech

NASA Technical Reports Server (NTRS)

Jorgensen, Charles; Betts, Bradley

2007-01-01

A research and development effort now underway is directed toward the use of subvocal speech for communication in settings in which (1) acoustic noise could interfere excessively with ordinary vocal communication and/or (2) acoustic silence or secrecy of communication is required. By "subvocal speech" is meant sub-audible electromyographic (EMG) signals, associated with speech, that are acquired from the surface of the larynx and lingual areas of the throat. Topics addressed in this effort include recognition of the sub-vocal EMG signals that represent specific original words or phrases; transformation (including encoding and/or enciphering) of the signals into forms that are less vulnerable to distortion, degradation, and/or interception; and reconstruction of the original words or phrases at the receiving end of a communication link. Potential applications include ordinary verbal communications among hazardous- material-cleanup workers in protective suits, workers in noisy environments, divers, and firefighters, and secret communications among law-enforcement officers and military personnel in combat and other confrontational situations.

A new feature constituting approach to detection of vocal fold pathology

NASA Astrophysics Data System (ADS)

Hariharan, M.; Polat, Kemal; Yaacob, Sazali

2014-08-01

In the last two decades, non-invasive methods through acoustic analysis of voice signal have been proved to be excellent and reliable tool to diagnose vocal fold pathologies. This paper proposes a new feature vector based on the wavelet packet transform and singular value decomposition for the detection of vocal fold pathology. k-means clustering based feature weighting is proposed to increase the distinguishing performance of the proposed features. In this work, two databases Massachusetts Eye and Ear Infirmary (MEEI) voice disorders database and MAPACI speech pathology database are used. Four different supervised classifiers such as k-nearest neighbour (k-NN), least-square support vector machine, probabilistic neural network and general regression neural network are employed for testing the proposed features. The experimental results uncover that the proposed features give very promising classification accuracy of 100% for both MEEI database and MAPACI speech pathology database.

Talker variability in audio-visual speech perception

PubMed Central

Heald, Shannon L. M.; Nusbaum, Howard C.

2014-01-01

A change in talker is a change in the context for the phonetic interpretation of acoustic patterns of speech. Different talkers have different mappings between acoustic patterns and phonetic categories and listeners need to adapt to these differences. Despite this complexity, listeners are adept at comprehending speech in multiple-talker contexts, albeit at a slight but measurable performance cost (e.g., slower recognition). So far, this talker variability cost has been demonstrated only in audio-only speech. Other research in single-talker contexts have shown, however, that when listeners are able to see a talker’s face, speech recognition is improved under adverse listening (e.g., noise or distortion) conditions that can increase uncertainty in the mapping between acoustic patterns and phonetic categories. Does seeing a talker’s face reduce the cost of word recognition in multiple-talker contexts? We used a speeded word-monitoring task in which listeners make quick judgments about target word recognition in single- and multiple-talker contexts. Results show faster recognition performance in single-talker conditions compared to multiple-talker conditions for both audio-only and audio-visual speech. However, recognition time in a multiple-talker context was slower in the audio-visual condition compared to audio-only condition. These results suggest that seeing a talker’s face during speech perception may slow recognition by increasing the importance of talker identification, signaling to the listener a change in talker has occurred. PMID:25076919

Talker variability in audio-visual speech perception.

PubMed

Heald, Shannon L M; Nusbaum, Howard C

2014-01-01

A change in talker is a change in the context for the phonetic interpretation of acoustic patterns of speech. Different talkers have different mappings between acoustic patterns and phonetic categories and listeners need to adapt to these differences. Despite this complexity, listeners are adept at comprehending speech in multiple-talker contexts, albeit at a slight but measurable performance cost (e.g., slower recognition). So far, this talker variability cost has been demonstrated only in audio-only speech. Other research in single-talker contexts have shown, however, that when listeners are able to see a talker's face, speech recognition is improved under adverse listening (e.g., noise or distortion) conditions that can increase uncertainty in the mapping between acoustic patterns and phonetic categories. Does seeing a talker's face reduce the cost of word recognition in multiple-talker contexts? We used a speeded word-monitoring task in which listeners make quick judgments about target word recognition in single- and multiple-talker contexts. Results show faster recognition performance in single-talker conditions compared to multiple-talker conditions for both audio-only and audio-visual speech. However, recognition time in a multiple-talker context was slower in the audio-visual condition compared to audio-only condition. These results suggest that seeing a talker's face during speech perception may slow recognition by increasing the importance of talker identification, signaling to the listener a change in talker has occurred.

Classroom Acoustics: Understanding Barriers to Learning.

ERIC Educational Resources Information Center

Crandell, Carl C., Ed.; Smaldino, Joseph J., Ed.

2001-01-01

This booklet explores classroom acoustics and their importance on the learning potential of children with hearing loss and related disabilities. The booklet also reviews research on classroom acoustics and the need for the development of classroom acoustics standards. Chapters examine: 1) a speech-perception model demonstrating the linkage between…

Auditory-tactile echo-reverberating stuttering speech corrector

NASA Astrophysics Data System (ADS)

Kuniszyk-Jozkowiak, Wieslawa; Adamczyk, Bogdan

1997-02-01

The work presents the construction of a device, which transforms speech sounds into acoustical and tactile signals of echo and reverberation. Research has been done on the influence of the echo and reverberation, which are transmitted as acoustic and tactile stimuli, on speech fluency. Introducing the echo or reverberation into the auditory feedback circuit results in a reduction of stuttering. A bit less, but still significant corrective effects are observed while using the tactile channel for transmitting the signals. The use of joined auditory and tactile channels increases the effects of their corrective influence on the stutterers' speech. The results of the experiment justify the use of the tactile channel in the stutterers' therapy.

Predicting word-recognition performance in noise by young listeners with normal hearing using acoustic, phonetic, and lexical variables.

PubMed

McArdle, Rachel; Wilson, Richard H

2008-06-01

To analyze the 50% correct recognition data that were from the Wilson et al (this issue) study and that were obtained from 24 listeners with normal hearing; also to examine whether acoustic, phonetic, or lexical variables can predict recognition performance for monosyllabic words presented in speech-spectrum noise. The specific variables are as follows: (a) acoustic variables (i.e., effective root-mean-square sound pressure level, duration), (b) phonetic variables (i.e., consonant features such as manner, place, and voicing for initial and final phonemes; vowel phonemes), and (c) lexical variables (i.e., word frequency, word familiarity, neighborhood density, neighborhood frequency). The descriptive, correlational study will examine the influence of acoustic, phonetic, and lexical variables on speech recognition in noise performance. Regression analysis demonstrated that 45% of the variance in the 50% point was accounted for by acoustic and phonetic variables whereas only 3% of the variance was accounted for by lexical variables. These findings suggest that monosyllabic word-recognition-in-noise is more dependent on bottom-up processing than on top-down processing. The results suggest that when speech-in-noise testing is used in a pre- and post-hearing-aid-fitting format, the use of monosyllabic words may be sensitive to changes in audibility resulting from amplification.

Auditory-Perceptual Learning Improves Speech Motor Adaptation in Children

PubMed Central

Shiller, Douglas M.; Rochon, Marie-Lyne

2015-01-01

Auditory feedback plays an important role in children’s speech development by providing the child with information about speech outcomes that is used to learn and fine-tune speech motor plans. The use of auditory feedback in speech motor learning has been extensively studied in adults by examining oral motor responses to manipulations of auditory feedback during speech production. Children are also capable of adapting speech motor patterns to perceived changes in auditory feedback, however it is not known whether their capacity for motor learning is limited by immature auditory-perceptual abilities. Here, the link between speech perceptual ability and the capacity for motor learning was explored in two groups of 5–7-year-old children who underwent a period of auditory perceptual training followed by tests of speech motor adaptation to altered auditory feedback. One group received perceptual training on a speech acoustic property relevant to the motor task while a control group received perceptual training on an irrelevant speech contrast. Learned perceptual improvements led to an enhancement in speech motor adaptation (proportional to the perceptual change) only for the experimental group. The results indicate that children’s ability to perceive relevant speech acoustic properties has a direct influence on their capacity for sensory-based speech motor adaptation. PMID:24842067

[Perception features of emotional intonation of short pseudowords].

PubMed

Dmitrieva, E S; Gel'man, V Ia; Zaĭtseva, K A; Orlov, A M

2012-01-01

Reaction time and recognition accuracy of speech emotional intonations in short meaningless words that differed only in one phoneme with background noise and without it were studied in 49 adults of 20-79 years old. The results were compared with the same parameters of emotional intonations in intelligent speech utterances under similar conditions. Perception of emotional intonations at different linguistic levels (phonological and lexico-semantic) was found to have both common features and certain peculiarities. Recognition characteristics of emotional intonations depending on gender and age of listeners appeared to be invariant with regard to linguistic levels of speech stimuli. Phonemic composition of pseudowords was found to influence the emotional perception, especially against the background noise. The most significant stimuli acoustic characteristic responsible for the perception of speech emotional prosody in short meaningless words under the two experimental conditions, i.e. with and without background noise, was the fundamental frequency variation.

Spectral identification of sperm whales from Littoral Acoustic Demonstration Center passive acoustic recordings

NASA Astrophysics Data System (ADS)

Sidorovskaia, Natalia A.; Richard, Blake; Ioup, George E.; Ioup, Juliette W.

2005-09-01

The Littoral Acoustic Demonstration Center (LADC) made a series of passive broadband acoustic recordings in the Gulf of Mexico and Ligurian Sea to study noise and marine mammal phonations. The collected data contain a large amount of various types of sperm whale phonations, such as isolated clicks and communication codas. It was previously reported that the spectrograms of the extracted clicks and codas contain well-defined null patterns that seem to be unique for individuals. The null pattern is formed due to individual features of the sound production organs of an animal. These observations motivated the present studies of adapting human speech identification techniques for deep-diving marine mammal phonations. A three-state trained hidden Markov model (HMM) was used with the phonation spectra of sperm whales. The HHM-algorithm gave 75% accuracy in identifying individuals when it had been initially tested for the acoustic data set correlated with visual observations of sperm whales. A comparison of the identification accuracy based on null-pattern similarity analysis and the HMM-algorithm is presented. The results can establish the foundation for developing an acoustic identification database for sperm whales and possibly other deep-diving marine mammals that would be difficult to observe visually. [Research supported by ONR.

The semantics of prosody: acoustic and perceptual evidence of prosodic correlates to word meaning.

PubMed

Nygaard, Lynne C; Herold, Debora S; Namy, Laura L

2009-01-01

This investigation examined whether speakers produce reliable prosodic correlates to meaning across semantic domains and whether listeners use these cues to derive word meaning from novel words. Speakers were asked to produce phrases in infant-directed speech in which novel words were used to convey one of two meanings from a set of antonym pairs (e.g., big/small). Acoustic analyses revealed that some acoustic features were correlated with overall valence of the meaning. However, each word meaning also displayed a unique acoustic signature, and semantically related meanings elicited similar acoustic profiles. In two perceptual tests, listeners either attempted to identify the novel words with a matching meaning dimension (picture pair) or with mismatched meaning dimensions. Listeners inferred the meaning of the novel words significantly more often when prosody matched the word meaning choices than when prosody mismatched. These findings suggest that speech contains reliable prosodic markers to word meaning and that listeners use these prosodic cues to differentiate meanings. That prosody is semantic suggests a reconceptualization of traditional distinctions between linguistic and nonlinguistic properties of spoken language. Copyright © 2009 Cognitive Science Society, Inc.

Classifier Subset Selection for the Stacked Generalization Method Applied to Emotion Recognition in Speech

PubMed Central

Álvarez, Aitor; Sierra, Basilio; Arruti, Andoni; López-Gil, Juan-Miguel; Garay-Vitoria, Nestor

2015-01-01

In this paper, a new supervised classification paradigm, called classifier subset selection for stacked generalization (CSS stacking), is presented to deal with speech emotion recognition. The new approach consists of an improvement of a bi-level multi-classifier system known as stacking generalization by means of an integration of an estimation of distribution algorithm (EDA) in the first layer to select the optimal subset from the standard base classifiers. The good performance of the proposed new paradigm was demonstrated over different configurations and datasets. First, several CSS stacking classifiers were constructed on the RekEmozio dataset, using some specific standard base classifiers and a total of 123 spectral, quality and prosodic features computed using in-house feature extraction algorithms. These initial CSS stacking classifiers were compared to other multi-classifier systems and the employed standard classifiers built on the same set of speech features. Then, new CSS stacking classifiers were built on RekEmozio using a different set of both acoustic parameters (extended version of the Geneva Minimalistic Acoustic Parameter Set (eGeMAPS)) and standard classifiers and employing the best meta-classifier of the initial experiments. The performance of these two CSS stacking classifiers was evaluated and compared. Finally, the new paradigm was tested on the well-known Berlin Emotional Speech database. We compared the performance of single, standard stacking and CSS stacking systems using the same parametrization of the second phase. All of the classifications were performed at the categorical level, including the six primary emotions plus the neutral one. PMID:26712757

Towards Contactless Silent Speech Recognition Based on Detection of Active and Visible Articulators Using IR-UWB Radar

PubMed Central

Shin, Young Hoon; Seo, Jiwon

2016-01-01

People with hearing or speaking disabilities are deprived of the benefits of conventional speech recognition technology because it is based on acoustic signals. Recent research has focused on silent speech recognition systems that are based on the motions of a speaker’s vocal tract and articulators. Because most silent speech recognition systems use contact sensors that are very inconvenient to users or optical systems that are susceptible to environmental interference, a contactless and robust solution is hence required. Toward this objective, this paper presents a series of signal processing algorithms for a contactless silent speech recognition system using an impulse radio ultra-wide band (IR-UWB) radar. The IR-UWB radar is used to remotely and wirelessly detect motions of the lips and jaw. In order to extract the necessary features of lip and jaw motions from the received radar signals, we propose a feature extraction algorithm. The proposed algorithm noticeably improved speech recognition performance compared to the existing algorithm during our word recognition test with five speakers. We also propose a speech activity detection algorithm to automatically select speech segments from continuous input signals. Thus, speech recognition processing is performed only when speech segments are detected. Our testbed consists of commercial off-the-shelf radar products, and the proposed algorithms are readily applicable without designing specialized radar hardware for silent speech processing. PMID:27801867

Towards Contactless Silent Speech Recognition Based on Detection of Active and Visible Articulators Using IR-UWB Radar.

PubMed

Shin, Young Hoon; Seo, Jiwon

2016-10-29

People with hearing or speaking disabilities are deprived of the benefits of conventional speech recognition technology because it is based on acoustic signals. Recent research has focused on silent speech recognition systems that are based on the motions of a speaker's vocal tract and articulators. Because most silent speech recognition systems use contact sensors that are very inconvenient to users or optical systems that are susceptible to environmental interference, a contactless and robust solution is hence required. Toward this objective, this paper presents a series of signal processing algorithms for a contactless silent speech recognition system using an impulse radio ultra-wide band (IR-UWB) radar. The IR-UWB radar is used to remotely and wirelessly detect motions of the lips and jaw. In order to extract the necessary features of lip and jaw motions from the received radar signals, we propose a feature extraction algorithm. The proposed algorithm noticeably improved speech recognition performance compared to the existing algorithm during our word recognition test with five speakers. We also propose a speech activity detection algorithm to automatically select speech segments from continuous input signals. Thus, speech recognition processing is performed only when speech segments are detected. Our testbed consists of commercial off-the-shelf radar products, and the proposed algorithms are readily applicable without designing specialized radar hardware for silent speech processing.

Abnormal laughter-like vocalisations replacing speech in primary progressive aphasia

PubMed Central

Rohrer, Jonathan D.; Warren, Jason D.; Rossor, Martin N.

2009-01-01

We describe ten patients with a clinical diagnosis of primary progressive aphasia (PPA) (pathologically confirmed in three cases) who developed abnormal laughter-like vocalisations in the context of progressive speech output impairment leading to mutism. Failure of speech output was accompanied by increasing frequency of the abnormal vocalisations until ultimately they constituted the patient's only extended utterance. The laughter-like vocalisations did not show contextual sensitivity but occurred as an automatic vocal output that replaced speech. Acoustic analysis of the vocalisations in two patients revealed abnormal motor features including variable note duration and inter-note interval, loss of temporal symmetry of laugh notes and loss of the normal decrescendo. Abnormal laughter-like vocalisations may be a hallmark of a subgroup in the PPA spectrum with impaired control and production of nonverbal vocal behaviour due to disruption of fronto-temporal networks mediating vocalisation. PMID:19435636

[Nature of speech disorders in Parkinson disease].

PubMed

Pawlukowska, W; Honczarenko, K; Gołąb-Janowska, M

2013-01-01

The aim of the study was to discuss physiology and pathology of speech and review of the literature on speech disorders in Parkinson disease. Additionally, the most effective methods to diagnose the speech disorders in Parkinson disease were also stressed. Afterward, articulatory, respiratory, acoustic and pragmatic factors contributing to the exacerbation of the speech disorders were discussed. Furthermore, the study dealt with the most important types of speech treatment techniques available (pharmacological and behavioral) and a significance of Lee Silverman Voice Treatment was highlighted.

Subcortical processing of speech regularities underlies reading and music aptitude in children.

PubMed

Strait, Dana L; Hornickel, Jane; Kraus, Nina

2011-10-17

Neural sensitivity to acoustic regularities supports fundamental human behaviors such as hearing in noise and reading. Although the failure to encode acoustic regularities in ongoing speech has been associated with language and literacy deficits, how auditory expertise, such as the expertise that is associated with musical skill, relates to the brainstem processing of speech regularities is unknown. An association between musical skill and neural sensitivity to acoustic regularities would not be surprising given the importance of repetition and regularity in music. Here, we aimed to define relationships between the subcortical processing of speech regularities, music aptitude, and reading abilities in children with and without reading impairment. We hypothesized that, in combination with auditory cognitive abilities, neural sensitivity to regularities in ongoing speech provides a common biological mechanism underlying the development of music and reading abilities. We assessed auditory working memory and attention, music aptitude, reading ability, and neural sensitivity to acoustic regularities in 42 school-aged children with a wide range of reading ability. Neural sensitivity to acoustic regularities was assessed by recording brainstem responses to the same speech sound presented in predictable and variable speech streams. Through correlation analyses and structural equation modeling, we reveal that music aptitude and literacy both relate to the extent of subcortical adaptation to regularities in ongoing speech as well as with auditory working memory and attention. Relationships between music and speech processing are specifically driven by performance on a musical rhythm task, underscoring the importance of rhythmic regularity for both language and music. These data indicate common brain mechanisms underlying reading and music abilities that relate to how the nervous system responds to regularities in auditory input. Definition of common biological underpinnings

Subcortical processing of speech regularities underlies reading and music aptitude in children

PubMed Central

2011-01-01

Background Neural sensitivity to acoustic regularities supports fundamental human behaviors such as hearing in noise and reading. Although the failure to encode acoustic regularities in ongoing speech has been associated with language and literacy deficits, how auditory expertise, such as the expertise that is associated with musical skill, relates to the brainstem processing of speech regularities is unknown. An association between musical skill and neural sensitivity to acoustic regularities would not be surprising given the importance of repetition and regularity in music. Here, we aimed to define relationships between the subcortical processing of speech regularities, music aptitude, and reading abilities in children with and without reading impairment. We hypothesized that, in combination with auditory cognitive abilities, neural sensitivity to regularities in ongoing speech provides a common biological mechanism underlying the development of music and reading abilities. Methods We assessed auditory working memory and attention, music aptitude, reading ability, and neural sensitivity to acoustic regularities in 42 school-aged children with a wide range of reading ability. Neural sensitivity to acoustic regularities was assessed by recording brainstem responses to the same speech sound presented in predictable and variable speech streams. Results Through correlation analyses and structural equation modeling, we reveal that music aptitude and literacy both relate to the extent of subcortical adaptation to regularities in ongoing speech as well as with auditory working memory and attention. Relationships between music and speech processing are specifically driven by performance on a musical rhythm task, underscoring the importance of rhythmic regularity for both language and music. Conclusions These data indicate common brain mechanisms underlying reading and music abilities that relate to how the nervous system responds to regularities in auditory input

Acoustic Processing of Temporally Modulated Sounds in Infants: Evidence from a Combined Near-Infrared Spectroscopy and EEG Study

PubMed Central

Telkemeyer, Silke; Rossi, Sonja; Nierhaus, Till; Steinbrink, Jens; Obrig, Hellmuth; Wartenburger, Isabell

2010-01-01

Speech perception requires rapid extraction of the linguistic content from the acoustic signal. The ability to efficiently process rapid changes in auditory information is important for decoding speech and thereby crucial during language acquisition. Investigating functional networks of speech perception in infancy might elucidate neuronal ensembles supporting perceptual abilities that gate language acquisition. Interhemispheric specializations for language have been demonstrated in infants. How these asymmetries are shaped by basic temporal acoustic properties is under debate. We recently provided evidence that newborns process non-linguistic sounds sharing temporal features with language in a differential and lateralized fashion. The present study used the same material while measuring brain responses of 6 and 3 month old infants using simultaneous recordings of electroencephalography (EEG) and near-infrared spectroscopy (NIRS). NIRS reveals that the lateralization observed in newborns remains constant over the first months of life. While fast acoustic modulations elicit bilateral neuronal activations, slow modulations lead to right-lateralized responses. Additionally, auditory-evoked potentials and oscillatory EEG responses show differential responses for fast and slow modulations indicating a sensitivity for temporal acoustic variations. Oscillatory responses reveal an effect of development, that is, 6 but not 3 month old infants show stronger theta-band desynchronization for slowly modulated sounds. Whether this developmental effect is due to increasing fine-grained perception for spectrotemporal sounds in general remains speculative. Our findings support the notion that a more general specialization for acoustic properties can be considered the basis for lateralization of speech perception. The results show that concurrent assessment of vascular based imaging and electrophysiological responses have great potential in the research on language acquisition

An Acoustic Study of the Relationships among Neurologic Disease, Dysarthria Type, and Severity of Dysarthria

ERIC Educational Resources Information Center

Kim, Yunjung; Kent, Raymond D.; Weismer, Gary

2011-01-01

Purpose: This study examined acoustic predictors of speech intelligibility in speakers with several types of dysarthria secondary to different diseases and conducted classification analysis solely by acoustic measures according to 3 variables (disease, speech severity, and dysarthria type). Method: Speech recordings from 107 speakers with…

Modifying Speech to Children based on their Perceived Phonetic Accuracy

PubMed Central

Julien, Hannah M.; Munson, Benjamin

2014-01-01

Purpose We examined the relationship between adults' perception of the accuracy of children's speech, and acoustic detail in their subsequent productions to children. Methods Twenty-two adults participated in a task in which they rated the accuracy of 2- and 3-year-old children's word-initial /s/and /∫/ using a visual analog scale (VAS), then produced a token of the same word as if they were responding to the child whose speech they had just rated. Result The duration of adults' fricatives varied as a function of their perception of the accuracy of children's speech: longer fricatives were produced following productions that they rated as inaccurate. This tendency to modify duration in response to perceived inaccurate tokens was mediated by measures of self-reported experience interacting with children. However, speakers did not increase the spectral distinctiveness of their fricatives following the perception of inaccurate tokens. Conclusion These results suggest that adults modify temporal features of their speech in response to perceiving children's inaccurate productions. These longer fricatives are potentially both enhanced input to children, and an error-corrective signal. PMID:22744140

On the importance of early reflections for speech in rooms.

PubMed

Bradley, J S; Sato, H; Picard, M

2003-06-01

This paper presents the results of new studies based on speech intelligibility tests in simulated sound fields and analyses of impulse response measurements in rooms used for speech communication. The speech intelligibility test results confirm the importance of early reflections for achieving good conditions for speech in rooms. The addition of early reflections increased the effective signal-to-noise ratio and related speech intelligibility scores for both impaired and nonimpaired listeners. The new results also show that for common conditions where the direct sound is reduced, it is only possible to understand speech because of the presence of early reflections. Analyses of measured impulse responses in rooms intended for speech show that early reflections can increase the effective signal-to-noise ratio by up to 9 dB. A room acoustics computer model is used to demonstrate that the relative importance of early reflections can be influenced by the room acoustics design.

Lip Movement Exaggerations During Infant-Directed Speech

PubMed Central

Green, Jordan R.; Nip, Ignatius S. B.; Wilson, Erin M.; Mefferd, Antje S.; Yunusova, Yana

2011-01-01

Purpose Although a growing body of literature has indentified the positive effects of visual speech on speech and language learning, oral movements of infant-directed speech (IDS) have rarely been studied. This investigation used 3-dimensional motion capture technology to describe how mothers modify their lip movements when talking to their infants. Method Lip movements were recorded from 25 mothers as they spoke to their infants and other adults. Lip shapes were analyzed for differences across speaking conditions. The maximum fundamental frequency, duration, acoustic intensity, and first and second formant frequency of each vowel also were measured. Results Lip movements were significantly larger during IDS than during adult-directed speech, although the exaggerations were vowel specific. All of the vowels produced during IDS were characterized by an elevated vocal pitch and a slowed speaking rate when compared with vowels produced during adult-directed speech. Conclusion The pattern of lip-shape exaggerations did not provide support for the hypothesis that mothers produce exemplar visual models of vowels during IDS. Future work is required to determine whether the observed increases in vertical lip aperture engender visual and acoustic enhancements that facilitate the early learning of speech. PMID:20699342

Virtual acoustics displays

NASA Astrophysics Data System (ADS)

Wenzel, Elizabeth M.; Fisher, Scott S.; Stone, Philip K.; Foster, Scott H.

1991-03-01

The real time acoustic display capabilities are described which were developed for the Virtual Environment Workstation (VIEW) Project at NASA-Ames. The acoustic display is capable of generating localized acoustic cues in real time over headphones. An auditory symbology, a related collection of representational auditory 'objects' or 'icons', can be designed using ACE (Auditory Cue Editor), which links both discrete and continuously varying acoustic parameters with information or events in the display. During a given display scenario, the symbology can be dynamically coordinated in real time with 3-D visual objects, speech, and gestural displays. The types of displays feasible with the system range from simple warnings and alarms to the acoustic representation of multidimensional data or events.

Virtual acoustics displays

NASA Technical Reports Server (NTRS)

Wenzel, Elizabeth M.; Fisher, Scott S.; Stone, Philip K.; Foster, Scott H.

1991-01-01

The real time acoustic display capabilities are described which were developed for the Virtual Environment Workstation (VIEW) Project at NASA-Ames. The acoustic display is capable of generating localized acoustic cues in real time over headphones. An auditory symbology, a related collection of representational auditory 'objects' or 'icons', can be designed using ACE (Auditory Cue Editor), which links both discrete and continuously varying acoustic parameters with information or events in the display. During a given display scenario, the symbology can be dynamically coordinated in real time with 3-D visual objects, speech, and gestural displays. The types of displays feasible with the system range from simple warnings and alarms to the acoustic representation of multidimensional data or events.

Speech research: A report on the status and progress of studies on the nature of speech, instrumentation for its investigation, and practical applications

NASA Astrophysics Data System (ADS)

Liberman, A. M.

1980-06-01

This report (1 April - 30 June) is one of a regular series on the status and progress of studies on the nature of speech, instrumentation for its investigation, and practical applications. Manuscripts cover the following topics: The perceptual equivalance of two acoustic cues for a speech contrast is specific to phonetic perception; Duplex perception of acoustic patterns as speech and nonspeech; Evidence for phonetic processing of cues to place of articulation: Perceived manner affects perceived place; Some articulatory correlates of perceptual isochrony; Effects of utterance continuity on phonetic judgments; Laryngeal adjustments in stuttering: A glottographic observation using a modified reaction paradigm; Missing -ing in reading: Letter detection errors on word endings; Speaking rate; syllable stress, and vowel identity; Sonority and syllabicity: Acoustic correlates of perception, Influence of vocalic context on perception of the (S)-(s) distinction.

A Musical Approach to Speech Melody

PubMed Central

Chow, Ivan; Brown, Steven

2018-01-01

We present here a musical approach to speech melody, one that takes advantage of the intervallic precision made possible with musical notation. Current phonetic and phonological approaches to speech melody either assign localized pitch targets that impoverish the acoustic details of the pitch contours and/or merely highlight a few salient points of pitch change, ignoring all the rest of the syllables. We present here an alternative model using musical notation, which has the advantage of representing the pitch of all syllables in a sentence as well as permitting a specification of the intervallic excursions among syllables and the potential for group averaging of pitch use across speakers. We tested the validity of this approach by recording native speakers of Canadian English reading unfamiliar test items aloud, spanning from single words to full sentences containing multiple intonational phrases. The fundamental-frequency trajectories of the recorded items were converted from hertz into semitones, averaged across speakers, and transcribed into musical scores of relative pitch. Doing so allowed us to quantify local and global pitch-changes associated with declarative, imperative, and interrogative sentences, and to explore the melodic dynamics of these sentence types. Our basic observation is that speech is atonal. The use of a musical score ultimately has the potential to combine speech rhythm and melody into a unified representation of speech prosody, an important analytical feature that is not found in any current linguistic approach to prosody. PMID:29556206

Watch what you say, your computer might be listening: A review of automated speech recognition

NASA Technical Reports Server (NTRS)

Degennaro, Stephen V.

1991-01-01

Spoken language is the most convenient and natural means by which people interact with each other and is, therefore, a promising candidate for human-machine interactions. Speech also offers an additional channel for hands-busy applications, complementing the use of motor output channels for control. Current speech recognition systems vary considerably across a number of important characteristics, including vocabulary size, speaking mode, training requirements for new speakers, robustness to acoustic environments, and accuracy. Algorithmically, these systems range from rule-based techniques through more probabilistic or self-learning approaches such as hidden Markov modeling and neural networks. This tutorial begins with a brief summary of the relevant features of current speech recognition systems and the strengths and weaknesses of the various algorithmic approaches.

Classroom acoustics and intervention strategies to enhance the learning environment

NASA Astrophysics Data System (ADS)

Savage, Christal

The classroom environment can be an acoustically difficult atmosphere for students to learn effectively, sometimes due in part to poor acoustical properties. Noise and reverberation have a substantial influence on room acoustics and subsequently intelligibility of speech. The American Speech-Language-Hearing Association (ASHA, 1995) developed minimal standards for noise and reverberation in a classroom for the purpose of providing an adequate listening environment. A lack of adherence to these standards may have undesirable consequences, which may lead to poor academic performance. The purpose of this capstone project is to develop a protocol to measure the acoustical properties of reverberation time and noise levels in elementary classrooms and present the educators with strategies to improve the learning environment. Noise level and reverberation will be measured and recorded in seven, unoccupied third grade classrooms in Lincoln Parish in North Louisiana. The recordings will occur at six specific distances in the classroom to simulate teacher and student positions. The recordings will be compared to the American Speech-Language-Hearing Association standards for noise and reverberation. If discrepancies are observed, the primary investigator will serve as an auditory consultant for the school and educators to recommend remediation and intervention strategies to improve these acoustical properties. The hypothesis of the study is that the classroom acoustical properties of noise and reverberation will exceed the American Speech-Language-Hearing Association standards; therefore, the auditory consultant will provide strategies to improve those acoustical properties.

The influence of selective attention to auditory and visual speech on the integration of audiovisual speech information.

PubMed

Buchan, Julie N; Munhall, Kevin G

2011-01-01

Conflicting visual speech information can influence the perception of acoustic speech, causing an illusory percept of a sound not present in the actual acoustic speech (the McGurk effect). We examined whether participants can voluntarily selectively attend to either the auditory or visual modality by instructing participants to pay attention to the information in one modality and to ignore competing information from the other modality. We also examined how performance under these instructions was affected by weakening the influence of the visual information by manipulating the temporal offset between the audio and video channels (experiment 1), and the spatial frequency information present in the video (experiment 2). Gaze behaviour was also monitored to examine whether attentional instructions influenced the gathering of visual information. While task instructions did have an influence on the observed integration of auditory and visual speech information, participants were unable to completely ignore conflicting information, particularly information from the visual stream. Manipulating temporal offset had a more pronounced interaction with task instructions than manipulating the amount of visual information. Participants' gaze behaviour suggests that the attended modality influences the gathering of visual information in audiovisual speech perception.

Acoustic Quality Levels of Mosques in Batu Pahat

NASA Astrophysics Data System (ADS)

Azizah Adnan, Nor; Nafida Raja Shahminan, Raja; Khair Ibrahim, Fawazul; Tami, Hannifah; Yusuff, M. Rizal M.; Murniwaty Samsudin, Emedya; Ismail, Isham

2018-04-01

Every Friday, Muslims has been required to perform a special prayer known as the Friday prayers which involve the delivery of a brief lecture (Khutbah). Speech intelligibility in oral communications presented by the preacher affected all the congregation and determined the level of acoustic quality in the interior of the mosque. Therefore, this study intended to assess the level of acoustic quality of three public mosques in Batu Pahat. Good acoustic quality is essential in contributing towards appreciation in prayers and increasing khusyu’ during the worship, which is closely related to the speech intelligibility corresponding to the actual function of the mosque according to Islam. Acoustic parameters measured includes noise criteria (NC), reverberation time (RT) and speech transmission index (STI), and was performed using the sound level meter and sound measurement instruments. This test is carried out through the physical observation with the consideration of space and volume design as a factor affecting acoustic parameters. Results from all 3 mosques as the showed that the acoustic quality level inside these buildings are slightly poor which is at below 0.45 coefficients based on the standard. Among the factors that influencing the low acoustical quality are location, building materials, installation of sound absorption material and the number of occupants inside the mosque. As conclusion, the acoustic quality level of a mosque is highly depends on physical factors of the mosque such as the architectural design and space volume besides other factors as been identified by this study.

Start/End Delays of Voiced and Unvoiced Speech Signals

DOE Office of Scientific and Technical Information (OSTI.GOV)

Herrnstein, A

Recent experiments using low power EM-radar like sensors (e.g, GEMs) have demonstrated a new method for measuring vocal fold activity and the onset times of voiced speech, as vocal fold contact begins to take place. Similarly the end time of a voiced speech segment can be measured. Secondly it appears that in most normal uses of American English speech, unvoiced-speech segments directly precede or directly follow voiced-speech segments. For many applications, it is useful to know typical duration times of these unvoiced speech segments. A corpus, assembled earlier of spoken ''Timit'' words, phrases, and sentences and recorded using simultaneously measuredmore » acoustic and EM-sensor glottal signals, from 16 male speakers, was used for this study. By inspecting the onset (or end) of unvoiced speech, using the acoustic signal, and the onset (or end) of voiced speech using the EM sensor signal, the average duration times for unvoiced segments preceding onset of vocalization were found to be 300ms, and for following segments, 500ms. An unvoiced speech period is then defined in time, first by using the onset of the EM-sensed glottal signal, as the onset-time marker for the voiced speech segment and end marker for the unvoiced segment. Then, by subtracting 300ms from the onset time mark of voicing, the unvoiced speech segment start time is found. Similarly, the times for a following unvoiced speech segment can be found. While data of this nature have proven to be useful for work in our laboratory, a great deal of additional work remains to validate such data for use with general populations of users. These procedures have been useful for applying optimal processing algorithms over time segments of unvoiced, voiced, and non-speech acoustic signals. For example, these data appear to be of use in speaker validation, in vocoding, and in denoising algorithms.« less