Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

DOEpatents

Holzrichter, John F.; Ng, Lawrence C.

1998-01-01

The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used for purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching.

Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

DOEpatents

Holzrichter, J.F.; Ng, L.C.

1998-03-17

The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used for purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching. 35 figs.

Speech coding, reconstruction and recognition using acoustics and electromagnetic waves

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

Holzrichter, J.F.; Ng, L.C.

The use of EM radiation in conjunction with simultaneously recorded acoustic speech information enables a complete mathematical coding of acoustic speech. The methods include the forming of a feature vector for each pitch period of voiced speech and the forming of feature vectors for each time frame of unvoiced, as well as for combined voiced and unvoiced speech. The methods include how to deconvolve the speech excitation function from the acoustic speech output to describe the transfer function each time frame. The formation of feature vectors defining all acoustic speech units over well defined time frames can be used formore » purposes of speech coding, speech compression, speaker identification, language-of-speech identification, speech recognition, speech synthesis, speech translation, speech telephony, and speech teaching. 35 figs.« less

Methods and apparatus for non-acoustic speech characterization and recognition

DOEpatents

Holzrichter, John F.

1999-01-01

By simultaneously recording EM wave reflections and acoustic speech information, the positions and velocities of the speech organs as speech is articulated can be defined for each acoustic speech unit. Well defined time frames and feature vectors describing the speech, to the degree required, can be formed. Such feature vectors can uniquely characterize the speech unit being articulated each time frame. The onset of speech, rejection of external noise, vocalized pitch periods, articulator conditions, accurate timing, the identification of the speaker, acoustic speech unit recognition, and organ mechanical parameters can be determined.

Methods and apparatus for non-acoustic speech characterization and recognition

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

Holzrichter, J.F.

By simultaneously recording EM wave reflections and acoustic speech information, the positions and velocities of the speech organs as speech is articulated can be defined for each acoustic speech unit. Well defined time frames and feature vectors describing the speech, to the degree required, can be formed. Such feature vectors can uniquely characterize the speech unit being articulated each time frame. The onset of speech, rejection of external noise, vocalized pitch periods, articulator conditions, accurate timing, the identification of the speaker, acoustic speech unit recognition, and organ mechanical parameters can be determined.

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.

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.

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.

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

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

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.

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.

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

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.

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.

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

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,…

Role of the middle ear muscle apparatus in mechanisms of speech signal discrimination

NASA Technical Reports Server (NTRS)

Moroz, B. S.; Bazarov, V. G.; Sachenko, S. V.

1980-01-01

A method of impedance reflexometry was used to examine 101 students with hearing impairment in order to clarify the interrelation between speech discrimination and the state of the middle ear muscles. Ability to discriminate speech signals depends to some extent on the functional state of intraaural muscles. Speech discrimination was greatly impaired in the absence of stapedial muscle acoustic reflex, in the presence of low thresholds of stimulation and in very small values of reflex amplitude increase. Discrimination was not impeded in positive AR, high values of relative thresholds and normal increase of reflex amplitude in response to speech signals with augmenting intensity.

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.

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

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.

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

[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

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.

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

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.

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.

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.

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.

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

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

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

Learning to perceptually organize speech signals in native fashion.

PubMed

Nittrouer, Susan; Lowenstein, Joanna H

2010-03-01

The ability to recognize speech involves sensory, perceptual, and cognitive processes. For much of the history of speech perception research, investigators have focused on the first and third of these, asking how much and what kinds of sensory information are used by normal and impaired listeners, as well as how effective amounts of that information are altered by "top-down" cognitive processes. This experiment focused on perceptual processes, asking what accounts for how the sensory information in the speech signal gets organized. Two types of speech signals processed to remove properties that could be considered traditional acoustic cues (amplitude envelopes and sine wave replicas) were presented to 100 listeners in five groups: native English-speaking (L1) adults, 7-, 5-, and 3-year-olds, and native Mandarin-speaking adults who were excellent second-language (L2) users of English. The L2 adults performed more poorly than L1 adults with both kinds of signals. Children performed more poorly than L1 adults but showed disproportionately better performance for the sine waves than for the amplitude envelopes compared to both groups of adults. Sentence context had similar effects across groups, so variability in recognition was attributed to differences in perceptual organization of the sensory information, presumed to arise from native language experience.

Acoustic assessment of speech privacy curtains in two nursing units

PubMed Central

Pope, Diana S.; Miller-Klein, Erik T.

2016-01-01

Hospitals have complex soundscapes that create challenges to patient care. Extraneous noise and high reverberation rates impair speech intelligibility, which leads to raised voices. In an unintended spiral, the increasing noise may result in diminished speech privacy, as people speak loudly to be heard over the din. The products available to improve hospital soundscapes include construction materials that absorb sound (acoustic ceiling tiles, carpet, wall insulation) and reduce reverberation rates. Enhanced privacy curtains are now available and offer potential for a relatively simple way to improve speech privacy and speech intelligibility by absorbing sound at the hospital patient's bedside. Acoustic assessments were performed over 2 days on two nursing units with a similar design in the same hospital. One unit was built with the 1970s’ standard hospital construction and the other was newly refurbished (2013) with sound-absorbing features. In addition, we determined the effect of an enhanced privacy curtain versus standard privacy curtains using acoustic measures of speech privacy and speech intelligibility indexes. Privacy curtains provided auditory protection for the patients. In general, that protection was increased by the use of enhanced privacy curtains. On an average, the enhanced curtain improved sound absorption from 20% to 30%; however, there was considerable variability, depending on the configuration of the rooms tested. Enhanced privacy curtains provide measureable improvement to the acoustics of patient rooms but cannot overcome larger acoustic design issues. To shorten reverberation time, additional absorption, and compact and more fragmented nursing unit floor plate shapes should be considered. PMID:26780959

Acoustic assessment of speech privacy curtains in two nursing units.

PubMed

Pope, Diana S; Miller-Klein, Erik T

2016-01-01

Hospitals have complex soundscapes that create challenges to patient care. Extraneous noise and high reverberation rates impair speech intelligibility, which leads to raised voices. In an unintended spiral, the increasing noise may result in diminished speech privacy, as people speak loudly to be heard over the din. The products available to improve hospital soundscapes include construction materials that absorb sound (acoustic ceiling tiles, carpet, wall insulation) and reduce reverberation rates. Enhanced privacy curtains are now available and offer potential for a relatively simple way to improve speech privacy and speech intelligibility by absorbing sound at the hospital patient's bedside. Acoustic assessments were performed over 2 days on two nursing units with a similar design in the same hospital. One unit was built with the 1970s' standard hospital construction and the other was newly refurbished (2013) with sound-absorbing features. In addition, we determined the effect of an enhanced privacy curtain versus standard privacy curtains using acoustic measures of speech privacy and speech intelligibility indexes. Privacy curtains provided auditory protection for the patients. In general, that protection was increased by the use of enhanced privacy curtains. On an average, the enhanced curtain improved sound absorption from 20% to 30%; however, there was considerable variability, depending on the configuration of the rooms tested. Enhanced privacy curtains provide measureable improvement to the acoustics of patient rooms but cannot overcome larger acoustic design issues. To shorten reverberation time, additional absorption, and compact and more fragmented nursing unit floor plate shapes should be considered.

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

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.

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

Speech perception of sine-wave signals by children with cochlear implants

PubMed Central

Nittrouer, Susan; Kuess, Jamie; Lowenstein, Joanna H.

2015-01-01

Children need to discover linguistically meaningful structures in the acoustic speech signal. Being attentive to recurring, time-varying formant patterns helps in that process. However, that kind of acoustic structure may not be available to children with cochlear implants (CIs), thus hindering development. The major goal of this study was to examine whether children with CIs are as sensitive to time-varying formant structure as children with normal hearing (NH) by asking them to recognize sine-wave speech. The same materials were presented as speech in noise, as well, to evaluate whether any group differences might simply reflect general perceptual deficits on the part of children with CIs. Vocabulary knowledge, phonemic awareness, and “top-down” language effects were all also assessed. Finally, treatment factors were examined as possible predictors of outcomes. Results showed that children with CIs were as accurate as children with NH at recognizing sine-wave speech, but poorer at recognizing speech in noise. Phonemic awareness was related to that recognition. Top-down effects were similar across groups. Having had a period of bimodal stimulation near the time of receiving a first CI facilitated these effects. Results suggest that children with CIs have access to the important time-varying structure of vocal-tract formants. PMID:25994709

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.

Language identification from visual-only speech signals

PubMed Central

Ronquest, Rebecca E.; Levi, Susannah V.; Pisoni, David B.

2010-01-01

Our goal in the present study was to examine how observers identify English and Spanish from visual-only displays of speech. First, we replicated the recent findings of Soto-Faraco et al. (2007) with Spanish and English bilingual and monolingual observers using different languages and a different experimental paradigm (identification). We found that prior linguistic experience affected response bias but not sensitivity (Experiment 1). In two additional experiments, we investigated the visual cues that observers use to complete the language-identification task. The results of Experiment 2 indicate that some lexical information is available in the visual signal but that it is limited. Acoustic analyses confirmed that our Spanish and English stimuli differed acoustically with respect to linguistic rhythmic categories. In Experiment 3, we tested whether this rhythmic difference could be used by observers to identify the language when the visual stimuli is temporally reversed, thereby eliminating lexical information but retaining rhythmic differences. The participants performed above chance even in the backward condition, suggesting that the rhythmic differences between the two languages may aid language identification in visual-only speech signals. The results of Experiments 3A and 3B also confirm previous findings that increased stimulus length facilitates language identification. Taken together, the results of these three experiments replicate earlier findings and also show that prior linguistic experience, lexical information, rhythmic structure, and utterance length influence visual-only language identification. PMID:20675804

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.

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.

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

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.

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.

Amplitude Modulations of Acoustic Communication Signals

NASA Astrophysics Data System (ADS)

Turesson, Hjalmar K.

2011-12-01

In human speech, amplitude modulations at 3 -- 8 Hz are important for discrimination and detection. Two different neurophysiological theories have been proposed to explain this effect. The first theory proposes that, as a consequence of neocortical synaptic dynamics, signals that are amplitude modulated at 3 -- 8 Hz are propagated better than un-modulated signals, or signals modulated above 8 Hz. This suggests that neural activity elicited by vocalizations modulated at 3 -- 8 Hz is optimally transmitted, and the vocalizations better discriminated and detected. The second theory proposes that 3 -- 8 Hz amplitude modulations interact with spontaneous neocortical oscillations. Specifically, vocalizations modulated at 3 -- 8 Hz entrain local populations of neurons, which in turn, modulate the amplitude of high frequency gamma oscillations. This suggests that vocalizations modulated at 3 -- 8 Hz should induce stronger cross-frequency coupling. Similar to human speech, we found that macaque monkey vocalizations also are amplitude modulated between 3 and 8 Hz. Humans and macaque monkeys share similarities in vocal production, implying that the auditory systems subserving perception of acoustic communication signals also share similarities. Based on the similarities between human speech and macaque monkey vocalizations, we addressed how amplitude modulated vocalizations are processed in the auditory cortex of macaque monkeys, and what behavioral relevance modulations may have. Recording single neuron activity, as well as, the activity of local populations of neurons allowed us to test both of the neurophysiological theories presented above. We found that single neuron responses to vocalizations amplitude modulated at 3 -- 8 Hz resulted in better stimulus discrimination than vocalizations lacking 3 -- 8 Hz modulations, and that the effect most likely was mediated by synaptic dynamics. In contrast, we failed to find support for the oscillation-based model proposing a

Attentional modulation of informational masking on early cortical representations of speech signals.

PubMed

Zhang, Changxin; Arnott, Stephen R; Rabaglia, Cristina; Avivi-Reich, Meital; Qi, James; Wu, Xihong; Li, Liang; Schneider, Bruce A

2016-01-01

To recognize speech in a noisy auditory scene, listeners need to perceptually segregate the target talker's voice from other competing sounds (stream segregation). A number of studies have suggested that the attentional demands placed on listeners increase as the acoustic properties and informational content of the competing sounds become more similar to that of the target voice. Hence we would expect attentional demands to be considerably greater when speech is masked by speech than when it is masked by steady-state noise. To investigate the role of attentional mechanisms in the unmasking of speech sounds, event-related potentials (ERPs) were recorded to a syllable masked by noise or competing speech under both active (the participant was asked to respond when the syllable was presented) or passive (no response was required) listening conditions. The results showed that the long-latency auditory response to a syllable (/bi/), presented at different signal-to-masker ratios (SMRs), was similar in both passive and active listening conditions, when the masker was a steady-state noise. In contrast, a switch from the passive listening condition to the active one, when the masker was two-talker speech, significantly enhanced the ERPs to the syllable. These results support the hypothesis that the need to engage attentional mechanisms in aid of scene analysis increases as the similarity (both acoustic and informational) between the target speech and the competing background sounds increases. Copyright © 2015 Elsevier B.V. All rights reserved.

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

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…

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…

[The endpoint detection of cough signal in continuous speech].

PubMed

Yang, Guoqing; Mo, Hongqiang; Li, Wen; Lian, Lianfang; Zheng, Zeguang

2010-06-01

The endpoint detection of cough signal in continuous speech has been researched in order to improve the efficiency and veracity of manual recognition or computer-based automatic recognition. First, using the short time zero crossing ratio(ZCR) for identifying the suspicious coughs and getting the threshold of short time energy based on acoustic characteristics of cough. Then, the short time energy is combined with short time ZCR in order to implement the endpoint detection of cough in continuous speech. To evaluate the effect of the method, first, the virtual number of coughs in each recording was identified by two experienced doctors using the graphical user interface (GUI). Second, the recordings were analyzed by automatic endpoint detection program under Matlab7.0. Finally, the comparison between these two results showed: The error rate of undetected cough is 2.18%, and 98.13% of noise, silence and speech were removed. The way of setting short time energy threshold is robust. The endpoint detection program can remove most speech and noise, thus maintaining a lower rate of error.

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.

[Influence of human personal features on acoustic correlates of speech emotional intonation characteristics].

PubMed

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

2009-01-01

Comparative study of acoustic correlates of emotional intonation was conducted on two types of speech material: sensible speech utterances and short meaningless words. The corpus of speech signals of different emotional intonations (happy, angry, frightened, sad and neutral) was created using the actor's method of simulation of emotions. Native Russian 20-70-year-old speakers (both professional actors and non-actors) participated in the study. In the corpus, the following characteristics were analyzed: mean values and standard deviations of the power, fundamental frequency, frequencies of the first and second formants, and utterance duration. Comparison of each emotional intonation with "neutral" utterances showed the greatest deviations of the fundamental frequency and frequencies of the first formant. The direction of these deviations was independent of the semantic content of speech utterance and its duration, age, gender, and being actor or non-actor, though the personal features of the speakers affected the absolute values of these frequencies.

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.…

Detection of Obstructive sleep apnea in awake subjects by exploiting body posture effects on the speech signal.

PubMed

Kriboy, M; Tarasiuk, A; Zigel, Y

2014-01-01

Obstructive sleep apnea (OSA) is a common sleep disorder. OSA is associated with several anatomical and functional abnormalities of the upper airway. It was shown that these abnormalities in the upper airway are also likely to be the reason for increased rate of apneic events in the supine position. Functional and structural changes in the vocal tract can affect the acoustic properties of speech. We hypothesize that acoustic properties of speech that are affected by body position may aid in distinguishing between OSA and non-OSA patients. We aimed to explore the possibility to differentiate OSA and non-OSA patients by analyzing the acoustic properties of their speech signal in upright sitting and supine positions. 35 awake patients were recorded while pronouncing sustained vowels in the upright sitting and supine positions. Using linear discriminant analysis (LDA) classifier, accuracy of 84.6%, sensitivity of 92.7%, and specificity of 80.0% were achieved. This study provides the proof of concept that it is possible to screen for OSA by analyzing and comparing speech properties acquired in upright sitting vs. supine positions. An acoustic-based screening system during wakefulness may address the growing needs for a reliable OSA screening tool; further studies are needed to support these findings.

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)…

Predicting speech intelligibility based on the signal-to-noise envelope power ratio after modulation-frequency selective processing.

PubMed

Jørgensen, Søren; Dau, Torsten

2011-09-01

A model for predicting the intelligibility of processed noisy speech is proposed. The speech-based envelope power spectrum model has a similar structure as the model of Ewert and Dau [(2000). J. Acoust. Soc. Am. 108, 1181-1196], developed to account for modulation detection and masking data. The model estimates the speech-to-noise envelope power ratio, SNR(env), at the output of a modulation filterbank and relates this metric to speech intelligibility using the concept of an ideal observer. Predictions were compared to data on the intelligibility of speech presented in stationary speech-shaped noise. The model was further tested in conditions with noisy speech subjected to reverberation and spectral subtraction. Good agreement between predictions and data was found in all cases. For spectral subtraction, an analysis of the model's internal representation of the stimuli revealed that the predicted decrease of intelligibility was caused by the estimated noise envelope power exceeding that of the speech. The classical concept of the speech transmission index fails in this condition. The results strongly suggest that the signal-to-noise ratio at the output of a modulation frequency selective process provides a key measure of speech intelligibility. © 2011 Acoustical Society of America

[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.

Does experimental pain affect auditory processing of speech-relevant signals? A study in healthy young adults.

PubMed

Sapir, Shimon; Pud, Dorit

2008-01-01

To assess the effect of tonic pain stimulation on auditory processing of speech-relevant acoustic signals in healthy pain-free volunteers. Sixty university students, randomly assigned to either a thermal pain stimulation (46 degrees C/6 min) group (PS) or no pain stimulation group (NPS), performed a rate change detection task (RCDT) involving sinusoidally frequency-modulated vowel-like signals. Task difficulty was manipulated by changing the rate of the modulated signals (henceforth rate). Perceived pain intensity was evaluated using a visual analog scale (VAS) (0-100). Mean pain rating was approximately 33 in the PS group and approximately 3 in the NPS group. Pain stimulation was associated with poorer performance on the RCDT, but this trend was not statistically significant. Performance worsened with increasing rate of signal modulation in both groups (p < 0.0001), with no pain by rate interaction. The present findings indicate a trend whereby mild or moderate pain appears to affect auditory processing of speech-relevant acoustic signals. This trend, however, was not statistically significant. It is possible that more intense pain would yield more pronounced (deleterious) effects on auditory processing, but this needs to be verified empirically.

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.

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

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

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.

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…

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.

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.

Method and apparatus for obtaining complete speech signals for speech recognition applications

NASA Technical Reports Server (NTRS)

Abrash, Victor (Inventor); Cesari, Federico (Inventor); Franco, Horacio (Inventor); George, Christopher (Inventor); Zheng, Jing (Inventor)

2009-01-01

The present invention relates to a method and apparatus for obtaining complete speech signals for speech recognition applications. In one embodiment, the method continuously records an audio stream comprising a sequence of frames to a circular buffer. When a user command to commence or terminate speech recognition is received, the method obtains a number of frames of the audio stream occurring before or after the user command in order to identify an augmented audio signal for speech recognition processing. In further embodiments, the method analyzes the augmented audio signal in order to locate starting and ending speech endpoints that bound at least a portion of speech to be processed for recognition. At least one of the speech endpoints is located using a Hidden Markov Model.

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…

Differential effects of speech situations on mothers' and fathers' infant-directed and dog-directed speech: An acoustic analysis.

PubMed

Gergely, Anna; Faragó, Tamás; Galambos, Ágoston; Topál, József

2017-10-23

There is growing evidence that dog-directed and infant-directed speech have similar acoustic characteristics, like high overall pitch, wide pitch range, and attention-getting devices. However, it is still unclear whether dog- and infant-directed speech have gender or context-dependent acoustic features. In the present study, we collected comparable infant-, dog-, and adult directed speech samples (IDS, DDS, and ADS) in four different speech situations (Storytelling, Task solving, Teaching, and Fixed sentences situations); we obtained the samples from parents whose infants were younger than 30 months of age and also had pet dog at home. We found that ADS was different from IDS and DDS, independently of the speakers' gender and the given situation. Higher overall pitch in DDS than in IDS during free situations was also found. Our results show that both parents hyperarticulate their vowels when talking to children but not when addressing dogs: this result is consistent with the goal of hyperspeech in language tutoring. Mothers, however, exaggerate their vowels for their infants under 18 months more than fathers do. Our findings suggest that IDS and DDS have context-dependent features and support the notion that people adapt their prosodic features to the acoustic preferences and emotional needs of their audience.

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.

Speech as a breakthrough signaling resource in the cognitive evolution of biological complex adaptive systems.

PubMed

Mattei, Tobias A

2014-12-01

In self-adapting dynamical systems, a significant improvement in the signaling flow among agents constitutes one of the most powerful triggering events for the emergence of new complex behaviors. Ackermann and colleagues' comprehensive phylogenetic analysis of the brain structures involved in acoustic communication provides further evidence of the essential role which speech, as a breakthrough signaling resource, has played in the evolutionary development of human cognition viewed from the standpoint of complex adaptive system analysis.

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.

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.

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.

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

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.

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.

Empirical mode decomposition for analyzing acoustical signals

NASA Technical Reports Server (NTRS)

Huang, Norden E. (Inventor)

2005-01-01

The present invention discloses a computer implemented signal analysis method through the Hilbert-Huang Transformation (HHT) for analyzing acoustical signals, which are assumed to be nonlinear and nonstationary. The Empirical Decomposition Method (EMD) and the Hilbert Spectral Analysis (HSA) are used to obtain the HHT. Essentially, the acoustical signal will be decomposed into the Intrinsic Mode Function Components (IMFs). Once the invention decomposes the acoustic signal into its constituting components, all operations such as analyzing, identifying, and removing unwanted signals can be performed on these components. Upon transforming the IMFs into Hilbert spectrum, the acoustical signal may be compared with other acoustical signals.

Acoustic analysis of trill sounds.

PubMed

Dhananjaya, N; Yegnanarayana, B; Bhaskararao, Peri

2012-04-01

In this paper, the acoustic-phonetic characteristics of steady apical trills--trill sounds produced by the periodic vibration of the apex of the tongue--are studied. Signal processing methods, namely, zero-frequency filtering and zero-time liftering of speech signals, are used to analyze the excitation source and the resonance characteristics of the vocal tract system, respectively. Although it is natural to expect the effect of trilling on the resonances of the vocal tract system, it is interesting to note that trilling influences the glottal source of excitation as well. The excitation characteristics derived using zero-frequency filtering of speech signals are glottal epochs, strength of impulses at the glottal epochs, and instantaneous fundamental frequency of the glottal vibration. Analysis based on zero-time liftering of speech signals is used to study the dynamic resonance characteristics of vocal tract system during the production of trill sounds. Qualitative analysis of trill sounds in different vowel contexts, and the acoustic cues that may help spotting trills in continuous speech are discussed.

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.

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.

The Relationship Between Acoustic Signal Typing and Perceptual Evaluation of Tracheoesophageal Voice Quality for Sustained Vowels.

PubMed

Clapham, Renee P; van As-Brooks, Corina J; van Son, Rob J J H; Hilgers, Frans J M; van den Brekel, Michiel W M

2015-07-01

To investigate the relationship between acoustic signal typing and perceptual evaluation of sustained vowels produced by tracheoesophageal (TE) speakers and the use of signal typing in the clinical setting. Two evaluators independently categorized 1.75-second segments of narrow-band spectrograms according to acoustic signal typing and independently evaluated the recording of the same segments on a visual analog scale according to overall perceptual acoustic voice quality. The relationship between acoustic signal typing and overall voice quality (as a continuous scale and as a four-point ordinal scale) was investigated and the proportion of inter-rater agreement as well as the reliability between the two measures is reported. The agreement between signal type (I-IV) and ordinal voice quality (four-point scale) was low but significant, and there was a significant linear relationship between the variables. Signal type correctly predicted less than half of the voice quality data. There was a significant main effect of signal type on continuous voice quality scores with significant differences in median quality scores between signal types I-IV, I-III, and I-II. Signal typing can be used as an adjunct to perceptual and acoustic evaluation of the same stimuli for TE speech as part of a multidimensional evaluation protocol. Signal typing in its current form provides limited predictive information on voice quality, and there is significant overlap between signal types II and III and perceptual categories. Future work should consider whether the current four signal types could be refined. Copyright © 2015 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Predicting couple therapy outcomes based on speech acoustic features

PubMed Central

Nasir, Md; Baucom, Brian Robert; Narayanan, Shrikanth

2017-01-01

Automated assessment and prediction of marital outcome in couples therapy is a challenging task but promises to be a potentially useful tool for clinical psychologists. Computational approaches for inferring therapy outcomes using observable behavioral information obtained from conversations between spouses offer objective means for understanding relationship dynamics. In this work, we explore whether the acoustics of the spoken interactions of clinically distressed spouses provide information towards assessment of therapy outcomes. The therapy outcome prediction task in this work includes detecting whether there was a relationship improvement or not (posed as a binary classification) as well as discerning varying levels of improvement or decline in the relationship status (posed as a multiclass recognition task). We use each interlocutor’s acoustic speech signal characteristics such as vocal intonation and intensity, both independently and in relation to one another, as cues for predicting the therapy outcome. We also compare prediction performance with one obtained via standardized behavioral codes characterizing the relationship dynamics provided by human experts as features for automated classification. Our experiments, using data from a longitudinal clinical study of couples in distressed relations, showed that predictions of relationship outcomes obtained directly from vocal acoustics are comparable or superior to those obtained using human-rated behavioral codes as prediction features. In addition, combining direct signal-derived features with manually coded behavioral features improved the prediction performance in most cases, indicating the complementarity of relevant information captured by humans and machine algorithms. Additionally, considering the vocal properties of the interlocutors in relation to one another, rather than in isolation, showed to be important for improving the automatic prediction. This finding supports the notion that behavioral

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…

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.

Infant Perception of Atypical Speech Signals

ERIC Educational Resources Information Center

Vouloumanos, Athena; Gelfand, Hanna M.

2013-01-01

The ability to decode atypical and degraded speech signals as intelligible is a hallmark of speech perception. Human adults can perceive sounds as speech even when they are generated by a variety of nonhuman sources including computers and parrots. We examined how infants perceive the speech-like vocalizations of a parrot. Further, we examined how…

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.

Negative blood oxygen level dependent signals during speech comprehension.

PubMed

Rodriguez Moreno, Diana; Schiff, Nicholas D; Hirsch, Joy

2015-05-01

Speech comprehension studies have generally focused on the isolation and function of regions with positive blood oxygen level dependent (BOLD) signals with respect to a resting baseline. Although regions with negative BOLD signals in comparison to a resting baseline have been reported in language-related tasks, their relationship to regions of positive signals is not fully appreciated. Based on the emerging notion that the negative signals may represent an active function in language tasks, the authors test the hypothesis that negative BOLD signals during receptive language are more associated with comprehension than content-free versions of the same stimuli. Regions associated with comprehension of speech were isolated by comparing responses to passive listening to natural speech to two incomprehensible versions of the same speech: one that was digitally time reversed and one that was muffled by removal of high frequencies. The signal polarity was determined by comparing the BOLD signal during each speech condition to the BOLD signal during a resting baseline. As expected, stimulation-induced positive signals relative to resting baseline were observed in the canonical language areas with varying signal amplitudes for each condition. Negative BOLD responses relative to resting baseline were observed primarily in frontoparietal regions and were specific to the natural speech condition. However, the BOLD signal remained indistinguishable from baseline for the unintelligible speech conditions. Variations in connectivity between brain regions with positive and negative signals were also specifically related to the comprehension of natural speech. These observations of anticorrelated signals related to speech comprehension are consistent with emerging models of cooperative roles represented by BOLD signals of opposite polarity.

Negative Blood Oxygen Level Dependent Signals During Speech Comprehension

PubMed Central

Rodriguez Moreno, Diana; Schiff, Nicholas D.

2015-01-01

Abstract Speech comprehension studies have generally focused on the isolation and function of regions with positive blood oxygen level dependent (BOLD) signals with respect to a resting baseline. Although regions with negative BOLD signals in comparison to a resting baseline have been reported in language-related tasks, their relationship to regions of positive signals is not fully appreciated. Based on the emerging notion that the negative signals may represent an active function in language tasks, the authors test the hypothesis that negative BOLD signals during receptive language are more associated with comprehension than content-free versions of the same stimuli. Regions associated with comprehension of speech were isolated by comparing responses to passive listening to natural speech to two incomprehensible versions of the same speech: one that was digitally time reversed and one that was muffled by removal of high frequencies. The signal polarity was determined by comparing the BOLD signal during each speech condition to the BOLD signal during a resting baseline. As expected, stimulation-induced positive signals relative to resting baseline were observed in the canonical language areas with varying signal amplitudes for each condition. Negative BOLD responses relative to resting baseline were observed primarily in frontoparietal regions and were specific to the natural speech condition. However, the BOLD signal remained indistinguishable from baseline for the unintelligible speech conditions. Variations in connectivity between brain regions with positive and negative signals were also specifically related to the comprehension of natural speech. These observations of anticorrelated signals related to speech comprehension are consistent with emerging models of cooperative roles represented by BOLD signals of opposite polarity. PMID:25412406

Dimension-Based Statistical Learning Affects Both Speech Perception and Production

ERIC Educational Resources Information Center

Lehet, Matthew; Holt, Lori L.

2017-01-01

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…

Speech Music Discrimination Using Class-Specific Features

DTIC Science & Technology

2004-08-01

Speech Music Discrimination Using Class-Specific Features Thomas Beierholm...between speech and music . Feature extraction is class-specific and can therefore be tailored to each class meaning that segment size, model orders...interest. Some of the applications of audio signal classification are speech/ music classification [1], acoustical environmental classification [2][3

Compensation for Coarticulation: Disentangling Auditory and Gestural Theories of Perception of Coarticulatory Effects in Speech

ERIC Educational Resources Information Center

Viswanathan, Navin; Magnuson, James S.; Fowler, Carol A.

2010-01-01

According to one approach to speech perception, listeners perceive speech by applying general pattern matching mechanisms to the acoustic signal (e.g., Diehl, Lotto, & Holt, 2004). An alternative is that listeners perceive the phonetic gestures that structured the acoustic signal (e.g., Fowler, 1986). The two accounts have offered different…

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.

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…

Cross-modal interactions during perception of audiovisual speech and nonspeech signals: an fMRI study.

PubMed

Hertrich, Ingo; Dietrich, Susanne; Ackermann, Hermann

2011-01-01

During speech communication, visual information may interact with the auditory system at various processing stages. Most noteworthy, recent magnetoencephalography (MEG) data provided first evidence for early and preattentive phonetic/phonological encoding of the visual data stream--prior to its fusion with auditory phonological features [Hertrich, I., Mathiak, K., Lutzenberger, W., & Ackermann, H. Time course of early audiovisual interactions during speech and non-speech central-auditory processing: An MEG study. Journal of Cognitive Neuroscience, 21, 259-274, 2009]. Using functional magnetic resonance imaging, the present follow-up study aims to further elucidate the topographic distribution of visual-phonological operations and audiovisual (AV) interactions during speech perception. Ambiguous acoustic syllables--disambiguated to /pa/ or /ta/ by the visual channel (speaking face)--served as test materials, concomitant with various control conditions (nonspeech AV signals, visual-only and acoustic-only speech, and nonspeech stimuli). (i) Visual speech yielded an AV-subadditive activation of primary auditory cortex and the anterior superior temporal gyrus (STG), whereas the posterior STG responded both to speech and nonspeech motion. (ii) The inferior frontal and the fusiform gyrus of the right hemisphere showed a strong phonetic/phonological impact (differential effects of visual /pa/ vs. /ta/) upon hemodynamic activation during presentation of speaking faces. Taken together with the previous MEG data, these results point at a dual-pathway model of visual speech information processing: On the one hand, access to the auditory system via the anterior supratemporal “what" path may give rise to direct activation of "auditory objects." On the other hand, visual speech information seems to be represented in a right-hemisphere visual working memory, providing a potential basis for later interactions with auditory information such as the McGurk effect.

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

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

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

Intelligent acoustic data fusion technique for information security analysis

NASA Astrophysics Data System (ADS)

Jiang, Ying; Tang, Yize; Lu, Wenda; Wang, Zhongfeng; Wang, Zepeng; Zhang, Luming

2017-08-01

Tone is an essential component of word formation in all tonal languages, and it plays an important role in the transmission of information in speech communication. Therefore, tones characteristics study can be applied into security analysis of acoustic signal by the means of language identification, etc. In speech processing, fundamental frequency (F0) is often viewed as representing tones by researchers of speech synthesis. However, regular F0 values may lead to low naturalness in synthesized speech. Moreover, F0 and tone are not equivalent linguistically; F0 is just a representation of a tone. Therefore, the Electroglottography (EGG) signal is collected for deeper tones characteristics study. In this paper, focusing on the Northern Kam language, which has nine tonal contours and five level tone types, we first collected EGG and speech signals from six natural male speakers of the Northern Kam language, and then achieved the clustering distributions of the tone curves. After summarizing the main characteristics of tones of Northern Kam, we analyzed the relationship between EGG and speech signal parameters, and laid the foundation for further security analysis of acoustic signal.

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.

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.

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.

Effect of signal to noise ratio on the speech perception ability of older adults

PubMed Central

Shojaei, Elahe; Ashayeri, Hassan; Jafari, Zahra; Zarrin Dast, Mohammad Reza; Kamali, Koorosh

2016-01-01

Background: Speech perception ability depends on auditory and extra-auditory elements. The signal- to-noise ratio (SNR) is an extra-auditory element that has an effect on the ability to normally follow speech and maintain a conversation. Speech in noise perception difficulty is a common complaint of the elderly. In this study, the importance of SNR magnitude as an extra-auditory effect on speech perception in noise was examined in the elderly. Methods: The speech perception in noise test (SPIN) was conducted on 25 elderly participants who had bilateral low–mid frequency normal hearing thresholds at three SNRs in the presence of ipsilateral white noise. These participants were selected by available sampling method. Cognitive screening was done using the Persian Mini Mental State Examination (MMSE) test. Results: Independent T- test, ANNOVA and Pearson Correlation Index were used for statistical analysis. There was a significant difference in word discrimination scores at silence and at three SNRs in both ears (p≤0.047). Moreover, there was a significant difference in word discrimination scores for paired SNRs (0 and +5, 0 and +10, and +5 and +10 (p≤0.04)). No significant correlation was found between age and word recognition scores at silence and at three SNRs in both ears (p≥0.386). Conclusion: Our results revealed that decreasing the signal level and increasing the competing noise considerably reduced the speech perception ability in normal hearing at low–mid thresholds in the elderly. These results support the critical role of SNRs for speech perception ability in the elderly. Furthermore, our results revealed that normal hearing elderly participants required compensatory strategies to maintain normal speech perception in challenging acoustic situations. PMID:27390712

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.

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.

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.

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.

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.

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.

Articulatory Mediation of Speech Perception: A Causal Analysis of Multi-Modal Imaging Data

ERIC Educational Resources Information Center

Gow, David W., Jr.; Segawa, Jennifer A.

2009-01-01

The inherent confound between the organization of articulation and the acoustic-phonetic structure of the speech signal makes it exceptionally difficult to evaluate the competing claims of motor and acoustic-phonetic accounts of how listeners recognize coarticulated speech. Here we use Granger causation analysis of high spatiotemporal resolution…

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.

Automatic Speech Recognition from Neural Signals: A Focused Review.

PubMed

Herff, Christian; Schultz, Tanja

2016-01-01

Speech interfaces have become widely accepted and are nowadays integrated in various real-life applications and devices. They have become a part of our daily life. However, speech interfaces presume the ability to produce intelligible speech, which might be impossible due to either loud environments, bothering bystanders or incapabilities to produce speech (i.e., patients suffering from locked-in syndrome). For these reasons it would be highly desirable to not speak but to simply envision oneself to say words or sentences. Interfaces based on imagined speech would enable fast and natural communication without the need for audible speech and would give a voice to otherwise mute people. This focused review analyzes the potential of different brain imaging techniques to recognize speech from neural signals by applying Automatic Speech Recognition technology. We argue that modalities based on metabolic processes, such as functional Near Infrared Spectroscopy and functional Magnetic Resonance Imaging, are less suited for Automatic Speech Recognition from neural signals due to low temporal resolution but are very useful for the investigation of the underlying neural mechanisms involved in speech processes. In contrast, electrophysiologic activity is fast enough to capture speech processes and is therefor better suited for ASR. Our experimental results indicate the potential of these signals for speech recognition from neural data with a focus on invasively measured brain activity (electrocorticography). As a first example of Automatic Speech Recognition techniques used from neural signals, we discuss the Brain-to-text system.

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).

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…

Improving the speech intelligibility in classrooms

NASA Astrophysics Data System (ADS)

Lam, Choi Ling Coriolanus

of the reverberation time, the indoor ambient noise (or background noise level), the signal-to-noise ratio, and the speech transmission index, it aims to establish a guideline for improving the speech intelligibility in classrooms for any countries and any environmental conditions. The study showed that the acoustical conditions of most of the measured classrooms in Hong Kong are unsatisfactory. The selection of materials inside a classroom is important for improving speech intelligibility at design stage, especially the acoustics ceiling, to shorten the reverberation time inside the classroom. The signal-to-noise should be higher than 11dB(A) for over 70% of speech perception, either tonal or non-tonal languages, without the usage of address system. The unexpected results bring out a call to revise the standard design and to devise acceptable standards for classrooms in Hong Kong. It is also demonstrated a method for assessment on the classroom in other cities with similar environmental conditions.

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

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.

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

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…

Informational approach to the analysis of acoustic signals

NASA Astrophysics Data System (ADS)

Senkevich, Yuriy; Dyuk, Vyacheslav; Mishchenko, Mikhail; Solodchuk, Alexandra

2017-10-01

The example of linguistic processing of acoustic signals of a seismic event would be an information approach to the processing of non-stationary signals. The method for converting an acoustic signal into an information message is described by identifying repetitive self-similar patterns. The definitions of the event selection indicators in the symbolic recording of the acoustic signal are given. The results of processing an acoustic signal by a computer program realizing the processing of linguistic data are shown. Advantages and disadvantages of using software algorithms are indicated.

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

Associations between tongue movement pattern consistency and formant movement pattern consistency in response to speech behavioral modificationsa)

PubMed Central

Mefferd, Antje S.

2016-01-01

The degree of speech movement pattern consistency can provide information about speech motor control. Although tongue motor control is particularly important because of the tongue's primary contribution to the speech acoustic signal, capturing tongue movements during speech remains difficult and costly. This study sought to determine if formant movements could be used to estimate tongue movement pattern consistency indirectly. Two age groups (seven young adults and seven older adults) and six speech conditions (typical, slow, loud, clear, fast, bite block speech) were selected to elicit an age- and task-dependent performance range in tongue movement pattern consistency. Kinematic and acoustic spatiotemporal indexes (STI) were calculated based on sentence-length tongue movement and formant movement signals, respectively. Kinematic and acoustic STI values showed strong associations across talkers and moderate to strong associations for each talker across speech tasks; although, in cases where task-related tongue motor performance changes were relatively small, the acoustic STI values were poorly associated with kinematic STI values. These findings suggest that, depending on the sensitivity needs, formant movement pattern consistency could be used in lieu of direct kinematic analysis to indirectly examine speech motor control. PMID:27908069

Ultrasonic speech translator and communications system

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

Akerman, M.A.; Ayers, C.W.; Haynes, H.D.

1996-07-23

A wireless communication system undetectable by radio frequency methods for converting audio signals, including human voice, to electronic signals in the ultrasonic frequency range, transmitting the ultrasonic signal by way of acoustical pressure waves across a carrier medium, including gases, liquids, or solids, and reconverting the ultrasonic acoustical pressure waves back to the original audio signal. The ultrasonic speech translator and communication system includes an ultrasonic transmitting device and an ultrasonic receiving device. The ultrasonic transmitting device accepts as input an audio signal such as human voice input from a microphone or tape deck. The ultrasonic transmitting device frequency modulatesmore » an ultrasonic carrier signal with the audio signal producing a frequency modulated ultrasonic carrier signal, which is transmitted via acoustical pressure waves across a carrier medium such as gases, liquids or solids. The ultrasonic receiving device converts the frequency modulated ultrasonic acoustical pressure waves to a frequency modulated electronic signal, demodulates the audio signal from the ultrasonic carrier signal, and conditions the demodulated audio signal to reproduce the original audio signal at its output. 7 figs.« less

Ultrasonic speech translator and communications system

DOEpatents

Akerman, M.A.; Ayers, C.W.; Haynes, H.D.

1996-07-23

A wireless communication system undetectable by radio frequency methods for converting audio signals, including human voice, to electronic signals in the ultrasonic frequency range, transmitting the ultrasonic signal by way of acoustical pressure waves across a carrier medium, including gases, liquids, or solids, and reconverting the ultrasonic acoustical pressure waves back to the original audio signal. The ultrasonic speech translator and communication system includes an ultrasonic transmitting device and an ultrasonic receiving device. The ultrasonic transmitting device accepts as input an audio signal such as human voice input from a microphone or tape deck. The ultrasonic transmitting device frequency modulates an ultrasonic carrier signal with the audio signal producing a frequency modulated ultrasonic carrier signal, which is transmitted via acoustical pressure waves across a carrier medium such as gases, liquids or solids. The ultrasonic receiving device converts the frequency modulated ultrasonic acoustical pressure waves to a frequency modulated electronic signal, demodulates the audio signal from the ultrasonic carrier signal, and conditions the demodulated audio signal to reproduce the original audio signal at its output. 7 figs.

Ultrasonic speech translator and communications system

DOEpatents

Akerman, M. Alfred; Ayers, Curtis W.; Haynes, Howard D.

1996-01-01

A wireless communication system undetectable by radio frequency methods for converting audio signals, including human voice, to electronic signals in the ultrasonic frequency range, transmitting the ultrasonic signal by way of acoustical pressure waves across a carrier medium, including gases, liquids, or solids, and reconverting the ultrasonic acoustical pressure waves back to the original audio signal. The ultrasonic speech translator and communication system (20) includes an ultrasonic transmitting device (100) and an ultrasonic receiving device (200). The ultrasonic transmitting device (100) accepts as input (115) an audio signal such as human voice input from a microphone (114) or tape deck. The ultrasonic transmitting device (100) frequency modulates an ultrasonic carrier signal with the audio signal producing a frequency modulated ultrasonic carrier signal, which is transmitted via acoustical pressure waves across a carrier medium such as gases, liquids or solids. The ultrasonic receiving device (200) converts the frequency modulated ultrasonic acoustical pressure waves to a frequency modulated electronic signal, demodulates the audio signal from the ultrasonic carrier signal, and conditions the demodulated audio signal to reproduce the original audio signal at its output (250).

Hot topics: Signal processing in acoustics

NASA Astrophysics Data System (ADS)

Gaumond, Charles F.

2005-09-01

Signal processing in acoustics is a multidisciplinary group of people that work in many areas of acoustics. We have chosen two areas that have shown exciting new applications of signal processing to acoustics or have shown exciting and important results from the use of signal processing. In this session, two hot topics are shown: the use of noiselike acoustic fields to determine sound propagation structure and the use of localization to determine animal behaviors. The first topic shows the application of correlation on geo-acoustic fields to determine the Greens function for propagation through the Earth. These results can then be further used to solve geo-acoustic inverse problems. The first topic also shows the application of correlation using oceanic noise fields to determine the Greens function through the ocean. These results also have utility for oceanic inverse problems. The second topic shows exciting results from the detection, localization, and tracking of marine mammals by two different groups. Results from detection and localization of bullfrogs are shown, too. Each of these studies contributed to the knowledge of animal behavior. [Work supported by ONR.

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.

Analysis of Acoustic Features in Speakers with Cognitive Disorders and Speech Impairments

NASA Astrophysics Data System (ADS)

Saz, Oscar; Simón, Javier; Rodríguez, W. Ricardo; Lleida, Eduardo; Vaquero, Carlos

2009-12-01

This work presents the results in the analysis of the acoustic features (formants and the three suprasegmental features: tone, intensity and duration) of the vowel production in a group of 14 young speakers suffering different kinds of speech impairments due to physical and cognitive disorders. A corpus with unimpaired children's speech is used to determine the reference values for these features in speakers without any kind of speech impairment within the same domain of the impaired speakers; this is 57 isolated words. The signal processing to extract the formant and pitch values is based on a Linear Prediction Coefficients (LPCs) analysis of the segments considered as vowels in a Hidden Markov Model (HMM) based Viterbi forced alignment. Intensity and duration are also based in the outcome of the automated segmentation. As main conclusion of the work, it is shown that intelligibility of the vowel production is lowered in impaired speakers even when the vowel is perceived as correct by human labelers. The decrease in intelligibility is due to a 30% of increase in confusability in the formants map, a reduction of 50% in the discriminative power in energy between stressed and unstressed vowels and to a 50% increase of the standard deviation in the length of the vowels. On the other hand, impaired speakers keep good control of tone in the production of stressed and unstressed vowels.

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 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.

Hierarchical Organization of Auditory and Motor Representations in Speech Perception: Evidence from Searchlight Similarity Analysis

PubMed Central

Evans, Samuel; Davis, Matthew H.

2015-01-01

How humans extract the identity of speech sounds from highly variable acoustic signals remains unclear. Here, we use searchlight representational similarity analysis (RSA) to localize and characterize neural representations of syllables at different levels of the hierarchically organized temporo-frontal pathways for speech perception. We asked participants to listen to spoken syllables that differed considerably in their surface acoustic form by changing speaker and degrading surface acoustics using noise-vocoding and sine wave synthesis while we recorded neural responses with functional magnetic resonance imaging. We found evidence for a graded hierarchy of abstraction across the brain. At the peak of the hierarchy, neural representations in somatomotor cortex encoded syllable identity but not surface acoustic form, at the base of the hierarchy, primary auditory cortex showed the reverse. In contrast, bilateral temporal cortex exhibited an intermediate response, encoding both syllable identity and the surface acoustic form of speech. Regions of somatomotor cortex associated with encoding syllable identity in perception were also engaged when producing the same syllables in a separate session. These findings are consistent with a hierarchical account of how variable acoustic signals are transformed into abstract representations of the identity of speech sounds. PMID:26157026

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,…

Multistage audiovisual integration of speech: dissociating identification and detection.

PubMed

Eskelund, Kasper; Tuomainen, Jyrki; Andersen, Tobias S

2011-02-01

Speech perception integrates auditory and visual information. This is evidenced by the McGurk illusion where seeing the talking face influences the auditory phonetic percept and by the audiovisual detection advantage where seeing the talking face influences the detectability of the acoustic speech signal. Here, we show that identification of phonetic content and detection can be dissociated as speech-specific and non-specific audiovisual integration effects. To this end, we employed synthetically modified stimuli, sine wave speech (SWS), which is an impoverished speech signal that only observers informed of its speech-like nature recognize as speech. While the McGurk illusion only occurred for informed observers, the audiovisual detection advantage occurred for naïve observers as well. This finding supports a multistage account of audiovisual integration of speech in which the many attributes of the audiovisual speech signal are integrated by separate integration processes.

Random Deep Belief Networks for Recognizing Emotions from Speech Signals.

PubMed

Wen, Guihua; Li, Huihui; Huang, Jubing; Li, Danyang; Xun, Eryang

2017-01-01

Now the human emotions can be recognized from speech signals using machine learning methods; however, they are challenged by the lower recognition accuracies in real applications due to lack of the rich representation ability. Deep belief networks (DBN) can automatically discover the multiple levels of representations in speech signals. To make full of its advantages, this paper presents an ensemble of random deep belief networks (RDBN) method for speech emotion recognition. It firstly extracts the low level features of the input speech signal and then applies them to construct lots of random subspaces. Each random subspace is then provided for DBN to yield the higher level features as the input of the classifier to output an emotion label. All outputted emotion labels are then fused through the majority voting to decide the final emotion label for the input speech signal. The conducted experimental results on benchmark speech emotion databases show that RDBN has better accuracy than the compared methods for speech emotion recognition.

Random Deep Belief Networks for Recognizing Emotions from Speech Signals

PubMed Central

Li, Huihui; Huang, Jubing; Li, Danyang; Xun, Eryang

2017-01-01

Now the human emotions can be recognized from speech signals using machine learning methods; however, they are challenged by the lower recognition accuracies in real applications due to lack of the rich representation ability. Deep belief networks (DBN) can automatically discover the multiple levels of representations in speech signals. To make full of its advantages, this paper presents an ensemble of random deep belief networks (RDBN) method for speech emotion recognition. It firstly extracts the low level features of the input speech signal and then applies them to construct lots of random subspaces. Each random subspace is then provided for DBN to yield the higher level features as the input of the classifier to output an emotion label. All outputted emotion labels are then fused through the majority voting to decide the final emotion label for the input speech signal. The conducted experimental results on benchmark speech emotion databases show that RDBN has better accuracy than the compared methods for speech emotion recognition. PMID:28356908

Effects of human fatigue on speech signals

NASA Astrophysics Data System (ADS)

Stamoulis, Catherine

2004-05-01

Cognitive performance may be significantly affected by fatigue. In the case of critical personnel, such as pilots, monitoring human fatigue is essential to ensure safety and success of a given operation. One of the modalities that may be used for this purpose is speech, which is sensitive to respiratory changes and increased muscle tension of vocal cords, induced by fatigue. Age, gender, vocal tract length, physical and emotional state may significantly alter speech intensity, duration, rhythm, and spectral characteristics. In addition to changes in speech rhythm, fatigue may also affect the quality of speech, such as articulation. In a noisy environment, detecting fatigue-related changes in speech signals, particularly subtle changes at the onset of fatigue, may be difficult. Therefore, in a performance-monitoring system, speech parameters which are significantly affected by fatigue need to be identified and extracted from input signals. For this purpose, a series of experiments was performed under slowly varying cognitive load conditions and at different times of the day. The results of the data analysis are presented here.

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…

Noise-immune multisensor transduction of speech

NASA Astrophysics Data System (ADS)

Viswanathan, Vishu R.; Henry, Claudia M.; Derr, Alan G.; Roucos, Salim; Schwartz, Richard M.

1986-08-01

Two types of configurations of multiple sensors were developed, tested and evaluated in speech recognition application for robust performance in high levels of acoustic background noise: One type combines the individual sensor signals to provide a single speech signal input, and the other provides several parallel inputs. For single-input systems, several configurations of multiple sensors were developed and tested. Results from formal speech intelligibility and quality tests in simulated fighter aircraft cockpit noise show that each of the two-sensor configurations tested outperforms the constituent individual sensors in high noise. Also presented are results comparing the performance of two-sensor configurations and individual sensors in speaker-dependent, isolated-word speech recognition tests performed using a commercial recognizer (Verbex 4000) in simulated fighter aircraft cockpit noise.

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…

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

An ALE meta-analysis on the audiovisual integration of speech signals.

PubMed

Erickson, Laura C; Heeg, Elizabeth; Rauschecker, Josef P; Turkeltaub, Peter E

2014-11-01

The brain improves speech processing through the integration of audiovisual (AV) signals. Situations involving AV speech integration may be crudely dichotomized into those where auditory and visual inputs contain (1) equivalent, complementary signals (validating AV speech) or (2) inconsistent, different signals (conflicting AV speech). This simple framework may allow the systematic examination of broad commonalities and differences between AV neural processes engaged by various experimental paradigms frequently used to study AV speech integration. We conducted an activation likelihood estimation metaanalysis of 22 functional imaging studies comprising 33 experiments, 311 subjects, and 347 foci examining "conflicting" versus "validating" AV speech. Experimental paradigms included content congruency, timing synchrony, and perceptual measures, such as the McGurk effect or synchrony judgments, across AV speech stimulus types (sublexical to sentence). Colocalization of conflicting AV speech experiments revealed consistency across at least two contrast types (e.g., synchrony and congruency) in a network of dorsal stream regions in the frontal, parietal, and temporal lobes. There was consistency across all contrast types (synchrony, congruency, and percept) in the bilateral posterior superior/middle temporal cortex. Although fewer studies were available, validating AV speech experiments were localized to other regions, such as ventral stream visual areas in the occipital and inferior temporal cortex. These results suggest that while equivalent, complementary AV speech signals may evoke activity in regions related to the corroboration of sensory input, conflicting AV speech signals recruit widespread dorsal stream areas likely involved in the resolution of conflicting sensory signals. Copyright © 2014 Wiley Periodicals, Inc.

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…

Hierarchical Organization of Auditory and Motor Representations in Speech Perception: Evidence from Searchlight Similarity Analysis.

PubMed

Evans, Samuel; Davis, Matthew H

2015-12-01

How humans extract the identity of speech sounds from highly variable acoustic signals remains unclear. Here, we use searchlight representational similarity analysis (RSA) to localize and characterize neural representations of syllables at different levels of the hierarchically organized temporo-frontal pathways for speech perception. We asked participants to listen to spoken syllables that differed considerably in their surface acoustic form by changing speaker and degrading surface acoustics using noise-vocoding and sine wave synthesis while we recorded neural responses with functional magnetic resonance imaging. We found evidence for a graded hierarchy of abstraction across the brain. At the peak of the hierarchy, neural representations in somatomotor cortex encoded syllable identity but not surface acoustic form, at the base of the hierarchy, primary auditory cortex showed the reverse. In contrast, bilateral temporal cortex exhibited an intermediate response, encoding both syllable identity and the surface acoustic form of speech. Regions of somatomotor cortex associated with encoding syllable identity in perception were also engaged when producing the same syllables in a separate session. These findings are consistent with a hierarchical account of how variable acoustic signals are transformed into abstract representations of the identity of speech sounds. © The Author 2015. Published by Oxford University Press.

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

Acoustic Signal Processing in Photorefractive Optical Systems.

NASA Astrophysics Data System (ADS)

Zhou, Gan

This thesis discusses applications of the photorefractive effect in the context of acoustic signal processing. The devices and systems presented here illustrate the ideas and optical principles involved in holographic processing of acoustic information. The interest in optical processing stems from the similarities between holographic optical systems and contemporary models for massively parallel computation, in particular, neural networks. An initial step in acoustic processing is the transformation of acoustic signals into relevant optical forms. A fiber-optic transducer with photorefractive readout transforms acoustic signals into optical images corresponding to their short-time spectrum. The device analyzes complex sound signals and interfaces them with conventional optical correlators. The transducer consists of 130 multimode optical fibers sampling the spectral range of 100 Hz to 5 kHz logarithmically. A physical model of the human cochlea can help us understand some characteristics of human acoustic transduction and signal representation. We construct a life-sized cochlear model using elastic membranes coupled with two fluid-filled chambers, and use a photorefractive novelty filter to investigate its response. The detection sensitivity is determined to be 0.3 angstroms per root Hz at 2 kHz. Qualitative agreement is found between the model response and physiological data. Delay lines map time-domain signals into space -domain and permit holographic processing of temporal information. A parallel optical delay line using dynamic beam coupling in a rotating photorefractive crystal is presented. We experimentally demonstrate a 64 channel device with 0.5 seconds of time-delay and 167 Hz bandwidth. Acoustic signal recognition is described in a photorefractive system implementing the time-delay neural network model. The system consists of a photorefractive optical delay-line and a holographic correlator programmed in a LiNbO_3 crystal. We demonstrate the recognition

Integration of auditory and somatosensory error signals in the neural control of speech movements.

PubMed

Feng, Yongqiang; Gracco, Vincent L; Max, Ludo

2011-08-01

We investigated auditory and somatosensory feedback contributions to the neural control of speech. In task I, sensorimotor adaptation was studied by perturbing one of these sensory modalities or both modalities simultaneously. The first formant (F1) frequency in the auditory feedback was shifted up by a real-time processor and/or the extent of jaw opening was increased or decreased with a force field applied by a robotic device. All eight subjects lowered F1 to compensate for the up-shifted F1 in the feedback signal regardless of whether or not the jaw was perturbed. Adaptive changes in subjects' acoustic output resulted from adjustments in articulatory movements of the jaw or tongue. Adaptation in jaw opening extent in response to the mechanical perturbation occurred only when no auditory feedback perturbation was applied or when the direction of adaptation to the force was compatible with the direction of adaptation to a simultaneous acoustic perturbation. In tasks II and III, subjects' auditory and somatosensory precision and accuracy were estimated. Correlation analyses showed that the relationships 1) between F1 adaptation extent and auditory acuity for F1 and 2) between jaw position adaptation extent and somatosensory acuity for jaw position were weak and statistically not significant. Taken together, the combined findings from this work suggest that, in speech production, sensorimotor adaptation updates the underlying control mechanisms in such a way that the planning of vowel-related articulatory movements takes into account a complex integration of error signals from previous trials but likely with a dominant role for the auditory modality.

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.

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.

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.

EEG-based auditory attention decoding using unprocessed binaural signals in reverberant and noisy conditions?

PubMed

Aroudi, Ali; Doclo, Simon

2017-07-01

To decode auditory attention from single-trial EEG recordings in an acoustic scenario with two competing speakers, a least-squares method has been recently proposed. This method however requires the clean speech signals of both the attended and the unattended speaker to be available as reference signals. Since in practice only the binaural signals consisting of a reverberant mixture of both speakers and background noise are available, in this paper we explore the potential of using these (unprocessed) signals as reference signals for decoding auditory attention in different acoustic conditions (anechoic, reverberant, noisy, and reverberant-noisy). In addition, we investigate whether it is possible to use these signals instead of the clean attended speech signal for filter training. The experimental results show that using the unprocessed binaural signals for filter training and for decoding auditory attention is feasible with a relatively large decoding performance, although for most acoustic conditions the decoding performance is significantly lower than when using the clean speech signals.

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…

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

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.

A Comparison of Signal Enhancement Methods for Extracting Tonal Acoustic Signals

NASA Technical Reports Server (NTRS)

Jones, Michael G.

1998-01-01

The measurement of pure tone acoustic pressure signals in the presence of masking noise, often generated by mean flow, is a continual problem in the field of passive liner duct acoustics research. In support of the Advanced Subsonic Technology Noise Reduction Program, methods were investigated for conducting measurements of advanced duct liner concepts in harsh, aeroacoustic environments. This report presents the results of a comparison study of three signal extraction methods for acquiring quality acoustic pressure measurements in the presence of broadband noise (used to simulate the effects of mean flow). The performance of each method was compared to a baseline measurement of a pure tone acoustic pressure 3 dB above a uniform, broadband noise background.

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…

Spatial acoustic signal processing for immersive communication

NASA Astrophysics Data System (ADS)

Atkins, Joshua

Computing is rapidly becoming ubiquitous as users expect devices that can augment and interact naturally with the world around them. In these systems it is necessary to have an acoustic front-end that is able to capture and reproduce natural human communication. Whether the end point is a speech recognizer or another human listener, the reduction of noise, reverberation, and acoustic echoes are all necessary and complex challenges. The focus of this dissertation is to provide a general method for approaching these problems using spherical microphone and loudspeaker arrays.. In this work, a theory of capturing and reproducing three-dimensional acoustic fields is introduced from a signal processing perspective. In particular, the decomposition of the spatial part of the acoustic field into an orthogonal basis of spherical harmonics provides not only a general framework for analysis, but also many processing advantages. The spatial sampling error limits the upper frequency range with which a sound field can be accurately captured or reproduced. In broadband arrays, the cost and complexity of using multiple transducers is an issue. This work provides a flexible optimization method for determining the location of array elements to minimize the spatial aliasing error. The low frequency array processing ability is also limited by the SNR, mismatch, and placement error of transducers. To address this, a robust processing method is introduced and used to design a reproduction system for rendering over arbitrary loudspeaker arrays or binaurally over headphones. In addition to the beamforming problem, the multichannel acoustic echo cancellation (MCAEC) issue is also addressed. A MCAEC must adaptively estimate and track the constantly changing loudspeaker-room-microphone response to remove the sound field presented over the loudspeakers from that captured by the microphones. In the multichannel case, the system is overdetermined and many adaptive schemes fail to converge to

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.

Audiovisual Speech Perception in Children with Developmental Language Disorder in Degraded Listening Conditions

ERIC Educational Resources Information Center

Meronen, Auli; Tiippana, Kaisa; Westerholm, Jari; Ahonen, Timo

2013-01-01

Purpose: The effect of the signal-to-noise ratio (SNR) on the perception of audiovisual speech in children with and without developmental language disorder (DLD) was investigated by varying the noise level and the sound intensity of acoustic speech. The main hypotheses were that the McGurk effect (in which incongruent visual speech alters the…

[Detection of Weak Speech Signals from Strong Noise Background Based on Adaptive Stochastic Resonance].

PubMed

Lu, Huanhuan; Wang, Fuzhong; Zhang, Huichun

2016-04-01

Traditional speech detection methods regard the noise as a jamming signal to filter,but under the strong noise background,these methods lost part of the original speech signal while eliminating noise.Stochastic resonance can use noise energy to amplify the weak signal and suppress the noise.According to stochastic resonance theory,a new method based on adaptive stochastic resonance to extract weak speech signals is proposed.This method,combined with twice sampling,realizes the detection of weak speech signals from strong noise.The parameters of the systema,b are adjusted adaptively by evaluating the signal-to-noise ratio of the output signal,and then the weak speech signal is optimally detected.Experimental simulation analysis showed that under the background of strong noise,the output signal-to-noise ratio increased from the initial value-7dB to about 0.86 dB,with the gain of signalto-noise ratio is 7.86 dB.This method obviously raises the signal-to-noise ratio of the output speech signals,which gives a new idea to detect the weak speech signals in strong noise environment.

A novel radar sensor for the non-contact detection of speech signals.

PubMed

Jiao, Mingke; Lu, Guohua; Jing, Xijing; Li, Sheng; Li, Yanfeng; Wang, Jianqi

2010-01-01

Different speech detection sensors have been developed over the years but they are limited by the loss of high frequency speech energy, and have restricted non-contact detection due to the lack of penetrability. This paper proposes a novel millimeter microwave radar sensor to detect speech signals. The utilization of a high operating frequency and a superheterodyne receiver contributes to the high sensitivity of the radar sensor for small sound vibrations. In addition, the penetrability of microwaves allows the novel sensor to detect speech signals through nonmetal barriers. Results show that the novel sensor can detect high frequency speech energies and that the speech quality is comparable to traditional microphone speech. Moreover, the novel sensor can detect speech signals through a nonmetal material of a certain thickness between the sensor and the subject. Thus, the novel speech sensor expands traditional speech detection techniques and provides an exciting alternative for broader application prospects.

A Novel Radar Sensor for the Non-Contact Detection of Speech Signals

PubMed Central

Jiao, Mingke; Lu, Guohua; Jing, Xijing; Li, Sheng; Li, Yanfeng; Wang, Jianqi

2010-01-01

Different speech detection sensors have been developed over the years but they are limited by the loss of high frequency speech energy, and have restricted non-contact detection due to the lack of penetrability. This paper proposes a novel millimeter microwave radar sensor to detect speech signals. The utilization of a high operating frequency and a superheterodyne receiver contributes to the high sensitivity of the radar sensor for small sound vibrations. In addition, the penetrability of microwaves allows the novel sensor to detect speech signals through nonmetal barriers. Results show that the novel sensor can detect high frequency speech energies and that the speech quality is comparable to traditional microphone speech. Moreover, the novel sensor can detect speech signals through a nonmetal material of a certain thickness between the sensor and the subject. Thus, the novel speech sensor expands traditional speech detection techniques and provides an exciting alternative for broader application prospects. PMID:22399895

Massively-Parallel Architectures for Automatic Recognition of Visual Speech Signals

DTIC Science & Technology

1988-10-12

Secusrity Clamifieation, Nlassively-Parallel Architectures for Automa ic Recognitio of Visua, Speech Signals 12. PERSONAL AUTHOR(S) Terrence J...characteristics of speech from tJhe, visual speech signals. Neural networks have been trained on a database of vowels. The rqw images of faces , aligned and...images of faces , aligned and preprocessed, were used as input to these network which were trained to estimate the corresponding envelope of the

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.

Dimensional analysis of acoustically propagated signals

NASA Technical Reports Server (NTRS)

Hansen, Scott D.; Thomson, Dennis W.

1993-01-01

Traditionally, long term measurements of atmospherically propagated sound signals have consisted of time series of multiminute averages. Only recently have continuous measurements with temporal resolution corresponding to turbulent time scales been available. With modern digital data acquisition systems we now have the capability to simultaneously record both acoustical and meteorological parameters with sufficient temporal resolution to allow us to examine in detail relationships between fluctuating sound and the meteorological variables, particularly wind and temperature, which locally determine the acoustic refractive index. The atmospheric acoustic propagation medium can be treated as a nonlinear dynamical system, a kind of signal processor whose innards depend on thermodynamic and turbulent processes in the atmosphere. The atmosphere is an inherently nonlinear dynamical system. In fact one simple model of atmospheric convection, the Lorenz system, may well be the most widely studied of all dynamical systems. In this paper we report some results of our having applied methods used to characterize nonlinear dynamical systems to study the characteristics of acoustical signals propagated through the atmosphere. For example, we investigate whether or not it is possible to parameterize signal fluctuations in terms of fractal dimensions. For time series one such parameter is the limit capacity dimension. Nicolis and Nicolis were among the first to use the kind of methods we have to study the properties of low dimension global attractors.

Collaborative Signaling of Informational Structures by Dynamic Speech Rate.

ERIC Educational Resources Information Center

Koiso, Hanae; Shimojima, Atsushi; Katagiri, Yasuhiro

1998-01-01

Investigated the functions of dynamic speech rates as contextualization cues in conversational Japanese, examining five spontaneous task-oriented dialogs and analyzing the potential of speech-rate changes in signaling the structure of the information being exchanged. Results found a correlation between speech decelerations and the openings of new…

Speech intelligibility and speech quality of modified loudspeaker announcements examined in a simulated aircraft cabin.

PubMed

Pennig, Sibylle; Quehl, Julia; Wittkowski, Martin

2014-01-01

Acoustic modifications of loudspeaker announcements were investigated in a simulated aircraft cabin to improve passengers' speech intelligibility and quality of communication in this specific setting. Four experiments with 278 participants in total were conducted in an acoustic laboratory using a standardised speech test and subjective rating scales. In experiments 1 and 2 the sound pressure level (SPL) of the announcements was varied (ranging from 70 to 85 dB(A)). Experiments 3 and 4 focused on frequency modification (octave bands) of the announcements. All studies used a background noise with the same SPL (74 dB(A)), but recorded at different seat positions in the aircraft cabin (front, rear). The results quantify speech intelligibility improvements with increasing signal-to-noise ratio and amplification of particular octave bands, especially the 2 kHz and the 4 kHz band. Thus, loudspeaker power in an aircraft cabin can be reduced by using appropriate filter settings in the loudspeaker system.

Listening Effort: How the Cognitive Consequences of Acoustic Challenge Are Reflected in Brain and Behavior

PubMed Central

2018-01-01

Everyday conversation frequently includes challenges to the clarity of the acoustic speech signal, including hearing impairment, background noise, and foreign accents. Although an obvious problem is the increased risk of making word identification errors, extracting meaning from a degraded acoustic signal is also cognitively demanding, which contributes to increased listening effort. The concepts of cognitive demand and listening effort are critical in understanding the challenges listeners face in comprehension, which are not fully predicted by audiometric measures. In this article, the authors review converging behavioral, pupillometric, and neuroimaging evidence that understanding acoustically degraded speech requires additional cognitive support and that this cognitive load can interfere with other operations such as language processing and memory for what has been heard. Behaviorally, acoustic challenge is associated with increased errors in speech understanding, poorer performance on concurrent secondary tasks, more difficulty processing linguistically complex sentences, and reduced memory for verbal material. Measures of pupil dilation support the challenge associated with processing a degraded acoustic signal, indirectly reflecting an increase in neural activity. Finally, functional brain imaging reveals that the neural resources required to understand degraded speech extend beyond traditional perisylvian language networks, most commonly including regions of prefrontal cortex, premotor cortex, and the cingulo-opercular network. Far from being exclusively an auditory problem, acoustic degradation presents listeners with a systems-level challenge that requires the allocation of executive cognitive resources. An important point is that a number of dissociable processes can be engaged to understand degraded speech, including verbal working memory and attention-based performance monitoring. The specific resources required likely differ as a function of the

Listening Effort: How the Cognitive Consequences of Acoustic Challenge Are Reflected in Brain and Behavior.

PubMed

Peelle, Jonathan E

Everyday conversation frequently includes challenges to the clarity of the acoustic speech signal, including hearing impairment, background noise, and foreign accents. Although an obvious problem is the increased risk of making word identification errors, extracting meaning from a degraded acoustic signal is also cognitively demanding, which contributes to increased listening effort. The concepts of cognitive demand and listening effort are critical in understanding the challenges listeners face in comprehension, which are not fully predicted by audiometric measures. In this article, the authors review converging behavioral, pupillometric, and neuroimaging evidence that understanding acoustically degraded speech requires additional cognitive support and that this cognitive load can interfere with other operations such as language processing and memory for what has been heard. Behaviorally, acoustic challenge is associated with increased errors in speech understanding, poorer performance on concurrent secondary tasks, more difficulty processing linguistically complex sentences, and reduced memory for verbal material. Measures of pupil dilation support the challenge associated with processing a degraded acoustic signal, indirectly reflecting an increase in neural activity. Finally, functional brain imaging reveals that the neural resources required to understand degraded speech extend beyond traditional perisylvian language networks, most commonly including regions of prefrontal cortex, premotor cortex, and the cingulo-opercular network. Far from being exclusively an auditory problem, acoustic degradation presents listeners with a systems-level challenge that requires the allocation of executive cognitive resources. An important point is that a number of dissociable processes can be engaged to understand degraded speech, including verbal working memory and attention-based performance monitoring. The specific resources required likely differ as a function of the

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

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

Articulatory speech synthesis and speech production modelling

NASA Astrophysics Data System (ADS)

Huang, Jun

This dissertation addresses the problem of speech synthesis and speech production modelling based on the fundamental principles of human speech production. Unlike the conventional source-filter model, which assumes the independence of the excitation and the acoustic filter, we treat the entire vocal apparatus as one system consisting of a fluid dynamic aspect and a mechanical part. We model the vocal tract by a three-dimensional moving geometry. We also model the sound propagation inside the vocal apparatus as a three-dimensional nonplane-wave propagation inside a viscous fluid described by Navier-Stokes equations. In our work, we first propose a combined minimum energy and minimum jerk criterion to estimate the dynamic vocal tract movements during speech production. Both theoretical error bound analysis and experimental results show that this method can achieve very close match at the target points and avoid the abrupt change in articulatory trajectory at the same time. Second, a mechanical vocal fold model is used to compute the excitation signal of the vocal tract. The advantage of this model is that it is closely coupled with the vocal tract system based on fundamental aerodynamics. As a result, we can obtain an excitation signal with much more detail than the conventional parametric vocal fold excitation model. Furthermore, strong evidence of source-tract interaction is observed. Finally, we propose a computational model of the fricative and stop types of sounds based on the physical principles of speech production. The advantage of this model is that it uses an exogenous process to model the additional nonsteady and nonlinear effects due to the flow mode, which are ignored by the conventional source- filter speech production model. A recursive algorithm is used to estimate the model parameters. Experimental results show that this model is able to synthesize good quality fricative and stop types of sounds. Based on our dissertation work, we carefully argue

Acoustic Localization with Infrasonic Signals

NASA Astrophysics Data System (ADS)

Threatt, Arnesha; Elbing, Brian

2015-11-01

Numerous geophysical and anthropogenic events emit infrasonic frequencies (<20 Hz), including volcanoes, hurricanes, wind turbines and tornadoes. These sounds, which cannot be heard by the human ear, can be detected from large distances (in excess of 100 miles) due to low frequency acoustic signals having a very low decay rate in the atmosphere. Thus infrasound could be used for long-range, passive monitoring and detection of these events. An array of microphones separated by known distances can be used to locate a given source, which is known as acoustic localization. However, acoustic localization with infrasound is particularly challenging due to contamination from other signals, sensitivity to wind noise and producing a trusted source for system development. The objective of the current work is to create an infrasonic source using a propane torch wand or a subwoofer and locate the source using multiple infrasonic microphones. This presentation will present preliminary results from various microphone configurations used to locate the source.

Expertise with artificial non-speech sounds recruits speech-sensitive cortical regions

PubMed Central

Leech, Robert; Holt, Lori L.; Devlin, Joseph T.; Dick, Frederic

2009-01-01

Regions of the human temporal lobe show greater activation for speech than for other sounds. These differences may reflect intrinsically specialized domain-specific adaptations for processing speech, or they may be driven by the significant expertise we have in listening to the speech signal. To test the expertise hypothesis, we used a video-game-based paradigm that tacitly trained listeners to categorize acoustically complex, artificial non-linguistic sounds. Before and after training, we used functional MRI to measure how expertise with these sounds modulated temporal lobe activation. Participants’ ability to explicitly categorize the non-speech sounds predicted the change in pre- to post-training activation in speech-sensitive regions of the left posterior superior temporal sulcus, suggesting that emergent auditory expertise may help drive this functional regionalization. Thus, seemingly domain-specific patterns of neural activation in higher cortical regions may be driven in part by experience-based restructuring of high-dimensional perceptual space. PMID:19386919

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.

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

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.

The Role of the Listener's State in Speech Perception

ERIC Educational Resources Information Center

Viswanathan, Navin

2009-01-01

Accounts of speech perception disagree on whether listeners perceive the acoustic signal (Diehl, Lotto, & Holt, 2004) or the vocal tract gestures that produce the signal (e.g., Fowler, 1986). In this dissertation, I outline a research program using a phenomenon called "perceptual compensation for coarticulation" (Mann, 1980) to examine this…

[Acoustic conditions in open plan office - Application of technical measures in a typical room].

PubMed

Mikulski, Witold

2018-03-09

Noise in open plan offices should not exceed acceptable levels for the hearing protection. Its major negative effects on employees are nuisance and impediment in execution of work. Specific technical solutions should be introduced to provide proper acoustic conditions for work performance. Acoustic evaluation of a typical open plan office was presented in the article published in "Medycyna Pracy" 5/2016. None of the rooms meets all the criteria, therefore, in this article one of the rooms was chosen to apply different technical solutions to check the possibility of reaching proper acoustic conditions. Acoustic effectiveness of those solutions was verified by means of digital simulation. The model was checked by comparing the results of measurements and calculations before using simulation. The analyzis revealed that open plan offices supplemented with signals for masking speech signals can meet all the required criteria. It is relatively easy to reach proper reverberation time (i.e., sound absorption). It is more difficult to reach proper values of evaluation parameters determined from A-weighted sound pressure level (SPLA) of speech. The most difficult is to provide proper values of evaluation parameters determined from speech transmission index (STI). Finally, it is necessary (besides acoustic treatment) to use devices for speech masking. The study proved that it is technically possible to reach proper acoustic condition. Main causes of employees complaints in open plan office are inadequate acoustic work conditions. Therefore, it is necessary to apply specific technical solutions - not only sound absorbing suspended ceiling and high acoustic barriers, but also devices for speech masking. Med Pr 2018;69(2):153-165. This work is available in Open Access model and licensed under a CC BY-NC 3.0 PL license.

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.

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

Time-frequency feature representation using multi-resolution texture analysis and acoustic activity detector for real-life speech emotion recognition.

PubMed

Wang, Kun-Ching

2015-01-14

The classification of emotional speech is mostly considered in speech-related research on human-computer interaction (HCI). In this paper, the purpose is to present a novel feature extraction based on multi-resolutions texture image information (MRTII). The MRTII feature set is derived from multi-resolution texture analysis for characterization and classification of different emotions in a speech signal. The motivation is that we have to consider emotions have different intensity values in different frequency bands. In terms of human visual perceptual, the texture property on multi-resolution of emotional speech spectrogram should be a good feature set for emotion classification in speech. Furthermore, the multi-resolution analysis on texture can give a clearer discrimination between each emotion than uniform-resolution analysis on texture. In order to provide high accuracy of emotional discrimination especially in real-life, an acoustic activity detection (AAD) algorithm must be applied into the MRTII-based feature extraction. Considering the presence of many blended emotions in real life, in this paper make use of two corpora of naturally-occurring dialogs recorded in real-life call centers. Compared with the traditional Mel-scale Frequency Cepstral Coefficients (MFCC) and the state-of-the-art features, the MRTII features also can improve the correct classification rates of proposed systems among different language databases. Experimental results show that the proposed MRTII-based feature information inspired by human visual perception of the spectrogram image can provide significant classification for real-life emotional recognition in speech.

Time-Frequency Feature Representation Using Multi-Resolution Texture Analysis and Acoustic Activity Detector for Real-Life Speech Emotion Recognition

PubMed Central

Wang, Kun-Ching

2015-01-01

The classification of emotional speech is mostly considered in speech-related research on human-computer interaction (HCI). In this paper, the purpose is to present a novel feature extraction based on multi-resolutions texture image information (MRTII). The MRTII feature set is derived from multi-resolution texture analysis for characterization and classification of different emotions in a speech signal. The motivation is that we have to consider emotions have different intensity values in different frequency bands. In terms of human visual perceptual, the texture property on multi-resolution of emotional speech spectrogram should be a good feature set for emotion classification in speech. Furthermore, the multi-resolution analysis on texture can give a clearer discrimination between each emotion than uniform-resolution analysis on texture. In order to provide high accuracy of emotional discrimination especially in real-life, an acoustic activity detection (AAD) algorithm must be applied into the MRTII-based feature extraction. Considering the presence of many blended emotions in real life, in this paper make use of two corpora of naturally-occurring dialogs recorded in real-life call centers. Compared with the traditional Mel-scale Frequency Cepstral Coefficients (MFCC) and the state-of-the-art features, the MRTII features also can improve the correct classification rates of proposed systems among different language databases. Experimental results show that the proposed MRTII-based feature information inspired by human visual perception of the spectrogram image can provide significant classification for real-life emotional recognition in speech. PMID:25594590

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

50 years of progress in microphone arrays for speech processing

NASA Astrophysics Data System (ADS)

Elko, Gary W.; Frisk, George V.

2004-10-01

In the early 1980s, Jim Flanagan had a dream of covering the walls of a room with microphones. He occasionally referred to this concept as acoustic wallpaper. Being a new graduate in the field of acoustics and signal processing, it was fortunate that Bell Labs was looking for someone to investigate this area of microphone arrays for telecommunication. The job interview was exciting, with all of the big names in speech signal processing and acoustics sitting in the audience, many of whom were the authors of books and articles that were seminal contributions to the fields of acoustics and signal processing. If there ever was an opportunity of a lifetime, this was it. Fortunately, some of the work had already begun, and Sessler and West had already laid the groundwork for directional electret microphones. This talk will describe some of the very early work done at Bell Labs on microphone arrays and reflect on some of the many systems, from large 400-element arrays, to small two-microphone arrays. These microphone array systems were built under Jim Flanagan's leadership in an attempt to realize his vision of seamless hands-free speech communication between people and the communication of people with machines.

Integrated Spacesuit Audio System Enhances Speech Quality and Reduces Noise

NASA Technical Reports Server (NTRS)

Huang, Yiteng Arden; Chen, Jingdong; Chen, Shaoyan Sharyl

2009-01-01

A new approach has been proposed for increasing astronaut comfort and speech capture. Currently, the special design of a spacesuit forms an extreme acoustic environment making it difficult to capture clear speech without compromising comfort. The proposed Integrated Spacesuit Audio (ISA) system is to incorporate the microphones into the helmet and use software to extract voice signals from background noise.

Speech perception at positive signal-to-noise ratios using adaptive adjustment of time compression.

PubMed

Schlueter, Anne; Brand, Thomas; Lemke, Ulrike; Nitzschner, Stefan; Kollmeier, Birger; Holube, Inga

2015-11-01

Positive signal-to-noise ratios (SNRs) characterize listening situations most relevant for hearing-impaired listeners in daily life and should therefore be considered when evaluating hearing aid algorithms. For this, a speech-in-noise test was developed and evaluated, in which the background noise is presented at fixed positive SNRs and the speech rate (i.e., the time compression of the speech material) is adaptively adjusted. In total, 29 younger and 12 older normal-hearing, as well as 24 older hearing-impaired listeners took part in repeated measurements. Younger normal-hearing and older hearing-impaired listeners conducted one of two adaptive methods which differed in adaptive procedure and step size. Analysis of the measurements with regard to list length and estimation strategy for thresholds resulted in a practical method measuring the time compression for 50% recognition. This method uses time-compression adjustment and step sizes according to Versfeld and Dreschler [(2002). J. Acoust. Soc. Am. 111, 401-408], with sentence scoring, lists of 30 sentences, and a maximum likelihood method for threshold estimation. Evaluation of the procedure showed that older participants obtained higher test-retest reliability compared to younger participants. Depending on the group of listeners, one or two lists are required for training prior to data collection.

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.

Spectral and temporal resolutions of information-bearing acoustic changes for understanding vocoded sentencesa)

PubMed Central

Stilp, Christian E.; Goupell, Matthew J.

2015-01-01

Short-time spectral changes in the speech signal are important for understanding noise-vocoded sentences. These information-bearing acoustic changes, measured using cochlea-scaled entropy in cochlear implant simulations [CSECI; Stilp et al. (2013). J. Acoust. Soc. Am. 133(2), EL136–EL141; Stilp (2014). J. Acoust. Soc. Am. 135(3), 1518–1529], may offer better understanding of speech perception by cochlear implant (CI) users. However, perceptual importance of CSECI for normal-hearing listeners was tested at only one spectral resolution and one temporal resolution, limiting generalizability of results to CI users. Here, experiments investigated the importance of these informational changes for understanding noise-vocoded sentences at different spectral resolutions (4–24 spectral channels; Experiment 1), temporal resolutions (4–64 Hz cutoff for low-pass filters that extracted amplitude envelopes; Experiment 2), or when both parameters varied (6–12 channels, 8–32 Hz; Experiment 3). Sentence intelligibility was reduced more by replacing high-CSECI intervals with noise than replacing low-CSECI intervals, but only when sentences had sufficient spectral and/or temporal resolution. High-CSECI intervals were more important for speech understanding as spectral resolution worsened and temporal resolution improved. Trade-offs between CSECI and intermediate spectral and temporal resolutions were minimal. These results suggest that signal processing strategies that emphasize information-bearing acoustic changes in speech may improve speech perception for CI users. PMID:25698018

Acoustic and Perceptual Effects of Dysarthria in Greek with a Focus on Lexical Stress

NASA Astrophysics Data System (ADS)

Papakyritsis, Ioannis

The field of motor speech disorders in Greek is substantially underresearched. Additionally, acoustic studies on lexical stress in dysarthria are generally very rare (Kim et al. 2010). This dissertation examined the acoustic and perceptual effects of Greek dysarthria focusing on lexical stress. Additional possibly deviant speech characteristics were acoustically analyzed. Data from three dysarthric participants and matched controls was analyzed using a case study design. The analysis of lexical stress was based on data drawn from a single word repetition task that included pairs of disyllabic words differentiated by stress location. This data was acoustically analyzed in terms of the use of the acoustic cues for Greek stress. The ability of the dysarthric participants to signal stress in single words was further assessed in a stress identification task carried out by 14 naive Greek listeners. Overall, the acoustic and perceptual data indicated that, although all three dysarthric speakers presented with some difficulty in the patterning of stressed and unstressed syllables, each had different underlying problems that gave rise to quite distinct patterns of deviant speech characteristics. The atypical use of lexical stress cues in Anna's data obscured the prominence relations of stressed and unstressed syllables to the extent that the position of lexical stress was usually not perceptually transparent. Chris and Maria on the other hand, did not have marked difficulties signaling lexical stress location, although listeners were not 100% successful in the stress identification task. For the most part, Chris' atypical phonation patterns and Maria's very slow rate of speech did not interfere with lexical stress signaling. The acoustic analysis of the lexical stress cues was generally in agreement with the participants' performance in the stress identification task. Interestingly, in all three dysarthric participants, but more so in Anna, targets stressed on the 1st

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

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

Use of amplitude modulation cues recovered from frequency modulation for cochlear implant users when original speech cues are severely degraded.

PubMed

Won, Jong Ho; Shim, Hyun Joon; Lorenzi, Christian; Rubinstein, Jay T

2014-06-01

Won et al. (J Acoust Soc Am 132:1113-1119, 2012) reported that cochlear implant (CI) speech processors generate amplitude-modulation (AM) cues recovered from broadband speech frequency modulation (FM) and that CI users can use these cues for speech identification in quiet. The present study was designed to extend this finding for a wide range of listening conditions, where the original speech cues were severely degraded by manipulating either the acoustic signals or the speech processor. The manipulation of the acoustic signals included the presentation of background noise, simulation of reverberation, and amplitude compression. The manipulation of the speech processor included changing the input dynamic range and the number of channels. For each of these conditions, multiple levels of speech degradation were tested. Speech identification was measured for CI users and compared for stimuli having both AM and FM information (intact condition) or FM information only (FM condition). Each manipulation degraded speech identification performance for both intact and FM conditions. Performance for the intact and FM conditions became similar for stimuli having the most severe degradations. Identification performance generally overlapped for the intact and FM conditions. Moreover, identification performance for the FM condition was better than chance performance even at the maximum level of distortion. Finally, significant correlations were found between speech identification scores for the intact and FM conditions. Altogether, these results suggest that despite poor frequency selectivity, CI users can make efficient use of AM cues recovered from speech FM in difficult listening situations.

Intelligent processing of acoustic emission signals

NASA Astrophysics Data System (ADS)

Sachse, Wolfgang; Grabec, Igor

1992-07-01

Recent developments in applying neural-like signal-processing procedures for analyzing acoustic emission signals are summarized. These procedures employ a set of learning signals to develop a memory that can subsequently be utilized to process other signals to recover information about an unknown source. A majority of the current applications to process ultrasonic waveforms are based on multilayered, feed-forward neural networks, trained with some type of back-error propagation rule.

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

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…

Perception and the temporal properties of speech

NASA Astrophysics Data System (ADS)

Gordon, Peter C.

1991-11-01

Four experiments addressing the role of attention in phonetic perception are reported. The first experiment shows that the relative importance of two cues to the voicing distinction changes when subjects must perform an arithmetic distractor task at the same time as identifying a speech stimulus. The voice onset time cue loses phonetic significance when subjects are distracted, while the F0 onset frequency cue does not. The second experiment shows a similar pattern for two cues to the distinction between the vowels /i/ (as in 'beat') and /I/ (as in 'bit'). Together these experiments indicate that careful attention to speech perception is necessary for strong acoustic cues to achieve their full phonetic impact, while weaker acoustic cues achieve their full phonetic impact without close attention. Experiment 3 shows that this pattern is obtained when the distractor task places little demand on verbal short term memory. Experiment 4 provides a large data set for testing formal models of the role of attention in speech perception. Attention is shown to influence the signal to noise ratio in phonetic encoding. This principle is instantiated in a network model in which the role of attention is to reduce noise in the phonetic encoding of acoustic cues. Implications of this work for understanding speech perception and general theories of the role of attention in perception are discussed.

Imitative Production of Rising Speech Intonation in Pediatric Cochlear Implant Recipients

PubMed Central

Peng, Shu-Chen; Tomblin, J. Bruce; Spencer, Linda J.; Hurtig, Richard R.

2011-01-01

Purpose This study investigated the acoustic characteristics of pediatric cochlear implant (CI) recipients' imitative production of rising speech intonation, in relation to the perceptual judgments by listeners with normal hearing (NH). Method Recordings of a yes–no interrogative utterance imitated by 24 prelingually deafened children with a CI were extracted from annual evaluation sessions. These utterances were perceptually judged by adult NH listeners in regard with intonation contour type (non-rise, partial-rise, or full-rise) and contour appropriateness (on a 5-point scale). Fundamental frequency, intensity, and duration properties of each utterance were also acoustically analyzed. Results Adult NH listeners' judgments of intonation contour type and contour appropriateness for each CI participant 's utterances were highly positively correlated. The pediatric CI recipients did not consistently use appropriate intonation contours when imitating a yes–no question. Acoustic properties of speech intonation produced by these individuals were discernible among utterances of different intonation contour types according to NH listeners' perceptual judgments. Conclusions These findings delineated the perceptual and acoustic characteristics of speech intonation imitated by prelingually deafened children and young adults with a CI. Future studies should address whether the degraded signals these individuals perceive via a CI contribute to their difficulties with speech intonation production. PMID:17905907

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

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

Classroom Acoustics: The Problem, Impact, and Solution.

ERIC Educational Resources Information Center

Berg, Frederick S.; And Others

1996-01-01

This article describes aspects of classroom acoustics that interfere with the ability of listeners to understand speech. It considers impacts on students and teachers and offers four possible solutions: noise control, signal control without amplification, individual amplification systems, and sound field amplification systems. (Author/DB)

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.…

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

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.

Real-Time Control of an Articulatory-Based Speech Synthesizer for Brain Computer Interfaces

PubMed Central

Bocquelet, Florent; Hueber, Thomas; Girin, Laurent; Savariaux, Christophe; Yvert, Blaise

2016-01-01

Restoring natural speech in paralyzed and aphasic people could be achieved using a Brain-Computer Interface (BCI) controlling a speech synthesizer in real-time. To reach this goal, a prerequisite is to develop a speech synthesizer producing intelligible speech in real-time with a reasonable number of control parameters. We present here an articulatory-based speech synthesizer that can be controlled in real-time for future BCI applications. This synthesizer converts movements of the main speech articulators (tongue, jaw, velum, and lips) into intelligible speech. The articulatory-to-acoustic mapping is performed using a deep neural network (DNN) trained on electromagnetic articulography (EMA) data recorded on a reference speaker synchronously with the produced speech signal. This DNN is then used in both offline and online modes to map the position of sensors glued on different speech articulators into acoustic parameters that are further converted into an audio signal using a vocoder. In offline mode, highly intelligible speech could be obtained as assessed by perceptual evaluation performed by 12 listeners. Then, to anticipate future BCI applications, we further assessed the real-time control of the synthesizer by both the reference speaker and new speakers, in a closed-loop paradigm using EMA data recorded in real time. A short calibration period was used to compensate for differences in sensor positions and articulatory differences between new speakers and the reference speaker. We found that real-time synthesis of vowels and consonants was possible with good intelligibility. In conclusion, these results open to future speech BCI applications using such articulatory-based speech synthesizer. PMID:27880768

Key considerations in designing a speech brain-computer interface.

PubMed

Bocquelet, Florent; Hueber, Thomas; Girin, Laurent; Chabardès, Stéphan; Yvert, Blaise

2016-11-01

Restoring communication in case of aphasia is a key challenge for neurotechnologies. To this end, brain-computer strategies can be envisioned to allow artificial speech synthesis from the continuous decoding of neural signals underlying speech imagination. Such speech brain-computer interfaces do not exist yet and their design should consider three key choices that need to be made: the choice of appropriate brain regions to record neural activity from, the choice of an appropriate recording technique, and the choice of a neural decoding scheme in association with an appropriate speech synthesis method. These key considerations are discussed here in light of (1) the current understanding of the functional neuroanatomy of cortical areas underlying overt and covert speech production, (2) the available literature making use of a variety of brain recording techniques to better characterize and address the challenge of decoding cortical speech signals, and (3) the different speech synthesis approaches that can be considered depending on the level of speech representation (phonetic, acoustic or articulatory) envisioned to be decoded at the core of a speech BCI paradigm. Copyright © 2017 The Author(s). Published by Elsevier Ltd.. All rights reserved.

Cochlear Implant Microphone Location Affects Speech Recognition in Diffuse Noise

PubMed Central

Kolberg, Elizabeth R.; Sheffield, Sterling W.; Davis, Timothy J.; Sunderhaus, Linsey W.; Gifford, René H.

2015-01-01

Background Despite improvements in cochlear implants (CIs), CI recipients continue to experience significant communicative difficulty in background noise. Many potential solutions have been proposed to help increase signal-to-noise ratio in noisy environments, including signal processing and external accessories. To date, however, the effect of microphone location on speech recognition in noise has focused primarily on hearing aid users. Purpose The purpose of this study was to (1) measure physical output for the T-Mic as compared with the integrated behind-the-ear(BTE) processor mic for various source azimuths, and (2) to investigate the effect of CI processor mic location for speech recognition in semi-diffuse noise with speech originating from various source azimuths as encountered in everyday communicative environments. Research Design A repeated-measures, within-participant design was used to compare performance across listening conditions. Study Sample A total of 11 adults with Advanced Bionics CIs were recruited for this study. Data Collection and Analysis Physical acoustic output was measured on a Knowles Experimental Mannequin for Acoustic Research (KEMAR) for the T-Mic and BTE mic, with broadband noise presented at 0 and 90° (directed toward the implant processor). In addition to physical acoustic measurements, we also assessed recognition of sentences constructed by researchers at Texas Instruments, the Massachusetts Institute of Technology, and the Stanford Research Institute (TIMIT sentences) at 60 dBA for speech source azimuths of 0, 90, and 270°. Sentences were presented in a semi-diffuse restaurant noise originating from the R-SPACE 8-loudspeaker array. Signal-to-noise ratio was determined individually to achieve approximately 50% correct in the unilateral implanted listening condition with speech at 0°. Performance was compared across the T-Mic, 50/50, and the integrated BTE processor mic. Results The integrated BTE mic provided approximately 5

Cochlear implant microphone location affects speech recognition in diffuse noise.

PubMed

Kolberg, Elizabeth R; Sheffield, Sterling W; Davis, Timothy J; Sunderhaus, Linsey W; Gifford, René H

2015-01-01

Despite improvements in cochlear implants (CIs), CI recipients continue to experience significant communicative difficulty in background noise. Many potential solutions have been proposed to help increase signal-to-noise ratio in noisy environments, including signal processing and external accessories. To date, however, the effect of microphone location on speech recognition in noise has focused primarily on hearing aid users. The purpose of this study was to (1) measure physical output for the T-Mic as compared with the integrated behind-the-ear (BTE) processor mic for various source azimuths, and (2) to investigate the effect of CI processor mic location for speech recognition in semi-diffuse noise with speech originating from various source azimuths as encountered in everyday communicative environments. A repeated-measures, within-participant design was used to compare performance across listening conditions. A total of 11 adults with Advanced Bionics CIs were recruited for this study. Physical acoustic output was measured on a Knowles Experimental Mannequin for Acoustic Research (KEMAR) for the T-Mic and BTE mic, with broadband noise presented at 0 and 90° (directed toward the implant processor). In addition to physical acoustic measurements, we also assessed recognition of sentences constructed by researchers at Texas Instruments, the Massachusetts Institute of Technology, and the Stanford Research Institute (TIMIT sentences) at 60 dBA for speech source azimuths of 0, 90, and 270°. Sentences were presented in a semi-diffuse restaurant noise originating from the R-SPACE 8-loudspeaker array. Signal-to-noise ratio was determined individually to achieve approximately 50% correct in the unilateral implanted listening condition with speech at 0°. Performance was compared across the T-Mic, 50/50, and the integrated BTE processor mic. The integrated BTE mic provided approximately 5 dB attenuation from 1500-4500 Hz for signals presented at 0° as compared with 90�

Speech research: Studies on the nature of speech, instrumentation for its investigation, and practical applications

NASA Astrophysics Data System (ADS)

Liberman, A. M.

1982-03-01

This report 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: Speech perception and memory coding in relation to reading ability; The use of orthographic structure by deaf adults: Recognition of finger-spelled letters; Exploring the information support for speech; The stream of speech; Using the acoustic signal to make inferences about place and duration of tongue-palate contact. Patterns of human interlimb coordination emerge from the the properties of nonlinear limit cycle oscillatory processes: Theory and data; Motor control: Which themes do we orchestrate? Exploring the nature of motor control in Down's syndrome; Periodicity and auditory memory: A pilot study; Reading skill and language skill: On the role of sign order and morphological structure in memory for American Sign Language sentences; Perception of nasal consonants with special reference to Catalan; and Speech production Characteristics of the hearing impaired.

The role of first formant information in simulated electro-acoustic hearing.

PubMed

Verschuur, Carl; Boland, Conor; Frost, Emily; Constable, Jack

2013-06-01

Cochlear implant (CI) recipients with residual hearing show improved performance with the addition of low-frequency acoustic stimulation (electro-acoustic stimulation, EAS). The present study sought to determine whether a synthesized first formant (F1) signal provided benefit to speech recognition in simulated EAS hearing and to compare such benefit with that from other low-frequency signals. A further aim was to determine if F1 amplitude or frequency was more important in determining benefit and if F1 benefit varied with formant bandwidth. In two experiments, sentence recordings from a male speaker were processed via a simulation of a partial insertion CI, and presented to normal hearing listeners in combination with various low-frequency signals, including a tone tracking fundamental frequency (F0), low-pass filtered speech, and signals based on F1 estimation. A simulated EAS benefit was found with F1 signals, and was similar to the benefit from F0 or low-pass filtered speech. The benefit did not differ significantly with the narrowing or widening of the F1 bandwidth. The benefit from low-frequency envelope signals was significantly less than the benefit from any low-frequency signal containing fine frequency information. Results indicate that F1 provides a benefit in simulated EAS hearing but low frequency envelope information is less important than low frequency fine structure in determining such benefit.

Hidden Markov models in automatic speech recognition

NASA Astrophysics Data System (ADS)

Wrzoskowicz, Adam

1993-11-01

This article describes a method for constructing an automatic speech recognition system based on hidden Markov models (HMMs). The author discusses the basic concepts of HMM theory and the application of these models to the analysis and recognition of speech signals. The author provides algorithms which make it possible to train the ASR system and recognize signals on the basis of distinct stochastic models of selected speech sound classes. The author describes the specific components of the system and the procedures used to model and recognize speech. The author discusses problems associated with the choice of optimal signal detection and parameterization characteristics and their effect on the performance of the system. The author presents different options for the choice of speech signal segments and their consequences for the ASR process. The author gives special attention to the use of lexical, syntactic, and semantic information for the purpose of improving the quality and efficiency of the system. The author also describes an ASR system developed by the Speech Acoustics Laboratory of the IBPT PAS. The author discusses the results of experiments on the effect of noise on the performance of the ASR system and describes methods of constructing HMM's designed to operate in a noisy environment. The author also describes a language for human-robot communications which was defined as a complex multilevel network from an HMM model of speech sounds geared towards Polish inflections. The author also added mandatory lexical and syntactic rules to the system for its communications vocabulary.

Auditory perception bias in speech imitation

PubMed Central

Postma-Nilsenová, Marie; Postma, Eric

2013-01-01

In an experimental study, we explored the role of auditory perception bias in vocal pitch imitation. Psychoacoustic tasks involving a missing fundamental indicate that some listeners are attuned to the relationship between all the higher harmonics present in the signal, which supports their perception of the fundamental frequency (the primary acoustic correlate of pitch). Other listeners focus on the lowest harmonic constituents of the complex sound signal which may hamper the perception of the fundamental. These two listener types are referred to as fundamental and spectral listeners, respectively. We hypothesized that the individual differences in speakers' capacity to imitate F0 found in earlier studies, may at least partly be due to the capacity to extract information about F0 from the speech signal. Participants' auditory perception bias was determined with a standard missing fundamental perceptual test. Subsequently, speech data were collected in a shadowing task with two conditions, one with a full speech signal and one with high-pass filtered speech above 300 Hz. The results showed that perception bias toward fundamental frequency was related to the degree of F0 imitation. The effect was stronger in the condition with high-pass filtered speech. The experimental outcomes suggest advantages for fundamental listeners in communicative situations where F0 imitation is used as a behavioral cue. Future research needs to determine to what extent auditory perception bias may be related to other individual properties known to improve imitation, such as phonetic talent. PMID:24204361

Prediction and constraint in audiovisual speech perception

PubMed Central

Peelle, Jonathan E.; Sommers, Mitchell S.

2015-01-01

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 precision of prediction. Electrophysiological studies demonstrate 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 auditory 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, supported

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…

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

Signal Restoration of Non-stationary Acoustic Signals in the Time Domain

NASA Technical Reports Server (NTRS)

Babkin, Alexander S.

1988-01-01

Signal restoration is a method of transforming a nonstationary signal acquired by a ground based microphone to an equivalent stationary signal. The benefit of the signal restoration is a simplification of the flight test requirements because it could dispense with the need to acquire acoustic data with another aircraft flying in concert with the rotorcraft. The data quality is also generally improved because the contamination of the signal by the propeller and wind noise is not present. The restoration methodology can also be combined with other data acquisition methods, such as a multiple linear microphone array for further improvement of the test results. The methodology and software are presented for performing the signal restoration in the time domain. The method has no restrictions on flight path geometry or flight regimes. Only requirement is that the aircraft spatial position be known relative to the microphone location and synchronized with the acoustic data. The restoration process assumes that the moving source radiates a stationary signal, which is then transformed into a nonstationary signal by various modulation processes. The restoration contains only the modulation due to the source motion.

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.

Envelope and intensity based prediction of psychoacoustic masking and speech intelligibility.

PubMed

Biberger, Thomas; Ewert, Stephan D

2016-08-01

Human auditory perception and speech intelligibility have been successfully described based on the two concepts of spectral masking and amplitude modulation (AM) masking. The power-spectrum model (PSM) [Patterson and Moore (1986). Frequency Selectivity in Hearing, pp. 123-177] accounts for effects of spectral masking and critical bandwidth, while the envelope power-spectrum model (EPSM) [Ewert and Dau (2000). J. Acoust. Soc. Am. 108, 1181-1196] has been successfully applied to AM masking and discrimination. Both models extract the long-term (envelope) power to calculate signal-to-noise ratios (SNR). Recently, the EPSM has been applied to speech intelligibility (SI) considering the short-term envelope SNR on various time scales (multi-resolution speech-based envelope power-spectrum model; mr-sEPSM) to account for SI in fluctuating noise [Jørgensen, Ewert, and Dau (2013). J. Acoust. Soc. Am. 134, 436-446]. Here, a generalized auditory model is suggested combining the classical PSM and the mr-sEPSM to jointly account for psychoacoustics and speech intelligibility. The model was extended to consider the local AM depth in conditions with slowly varying signal levels, and the relative role of long-term and short-term SNR was assessed. The suggested generalized power-spectrum model is shown to account for a large variety of psychoacoustic data and to predict speech intelligibility in various types of background noise.

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.

Speech Recognition in Nonnative versus Native English-Speaking College Students in a Virtual Classroom.

PubMed

Neave-DiToro, Dorothy; Rubinstein, Adrienne; Neuman, Arlene C

2017-05-01

Limited attention has been given to the effects of classroom acoustics at the college level. Many studies have reported that nonnative speakers of English are more likely to be affected by poor room acoustics than native speakers. An important question is how classroom acoustics affect speech perception of nonnative college students. The combined effect of noise and reverberation on the speech recognition performance of college students who differ in age of English acquisition was evaluated under conditions simulating classrooms with reverberation times (RTs) close to ANSI recommended RTs. A mixed design was used in this study. Thirty-six native and nonnative English-speaking college students with normal hearing, ages 18-28 yr, participated. Two groups of nine native participants (native monolingual [NM] and native bilingual) and two groups of nine nonnative participants (nonnative early and nonnative late) were evaluated in noise under three reverberant conditions (0.03, 0.06, and 0.08 sec). A virtual test paradigm was used, which represented a signal reaching a student at the back of a classroom. Speech recognition in noise was measured using the Bamford-Kowal-Bench Speech-in-Noise (BKB-SIN) test and signal-to-noise ratio required for correct repetition of 50% of the key words in the stimulus sentences (SNR-50) was obtained for each group in each reverberant condition. A mixed-design analysis of variance was used to determine statistical significance as a function of listener group and RT. SNR-50 was significantly higher for nonnative listeners as compared to native listeners, and a more favorable SNR-50 was needed as RT increased. The most dramatic effect on SNR-50 was found in the group with later acquisition of English, whereas the impact of early introduction of a second language was subtler. At the ANSI standard's maximum recommended RT (0.6 sec), all groups except the NM group exhibited a mild signal-to-noise ratio (SNR) loss. At the 0.8 sec RT, all groups

Very low-frequency signals support perceptual organization of implant-simulated speech for adults and children

PubMed Central

Nittrouer, Susan; Tarr, Eric; Bolster, Virginia; Caldwell-Tarr, Amanda; Moberly, Aaron C.; Lowenstein, Joanna H.

2014-01-01

Objective Using signals processed to simulate speech received through cochlear implants and low-frequency extended hearing aids, this study examined the proposal that low-frequency signals facilitate the perceptual organization of broader, spectrally degraded signals. Design In two experiments, words and sentences were presented in diotic and dichotic configurations as four-channel noise-vocoded signals (VOC-only), and as those signals combined with the acoustic signal below 250 Hz (LOW-plus). Dependent measures were percent correct recognition scores, and the difference between scores for the two processing conditions given as proportions of recognition scores for VOC-only. The influence of linguistic context was also examined. Study Sample Participants had normal hearing. In all, 40 adults, 40 7-year-olds, and 20 5-year-olds participated. Results Participants of all ages showed benefits of adding the low-frequency signal. The effect was greater for sentences than words, but no effect of configuration was found. The influence of linguistic context was similar across age groups, and did not contribute to the low-frequency effect. Listeners who scored more poorly with VOC-only stimuli showed greater low-frequency effects. Conclusion The benefit of adding a very low-frequency signal to a broader, spectrally degraded signal seems to derive from its facilitative influence on perceptual organization of the sensory input. PMID:24456179

Decoding Speech With Integrated Hybrid Signals Recorded From the Human Ventral Motor Cortex.

PubMed

Ibayashi, Kenji; Kunii, Naoto; Matsuo, Takeshi; Ishishita, Yohei; Shimada, Seijiro; Kawai, Kensuke; Saito, Nobuhito

2018-01-01

Restoration of speech communication for locked-in patients by means of brain computer interfaces (BCIs) is currently an important area of active research. Among the neural signals obtained from intracranial recordings, single/multi-unit activity (SUA/MUA), local field potential (LFP), and electrocorticography (ECoG) are good candidates for an input signal for BCIs. However, the question of which signal or which combination of the three signal modalities is best suited for decoding speech production remains unverified. In order to record SUA, LFP, and ECoG simultaneously from a highly localized area of human ventral sensorimotor cortex (vSMC), we fabricated an electrode the size of which was 7 by 13 mm containing sparsely arranged microneedle and conventional macro contacts. We determined which signal modality is the most capable of decoding speech production, and tested if the combination of these signals could improve the decoding accuracy of spoken phonemes. Feature vectors were constructed from spike frequency obtained from SUAs and event-related spectral perturbation derived from ECoG and LFP signals, then input to the decoder. The results showed that the decoding accuracy for five spoken vowels was highest when features from multiple signals were combined and optimized for each subject, and reached 59% when averaged across all six subjects. This result suggests that multi-scale signals convey complementary information for speech articulation. The current study demonstrated that simultaneous recording of multi-scale neuronal activities could raise decoding accuracy even though the recording area is limited to a small portion of cortex, which is advantageous for future implementation of speech-assisting BCIs.

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.

Micropower Mixed-signal VLSI Independent Component Analysis for Gradient Flow Acoustic Source Separation.

PubMed

Stanaćević, Milutin; Li, Shuo; Cauwenberghs, Gert

2016-07-01

A parallel micro-power mixed-signal VLSI implementation of independent component analysis (ICA) with reconfigurable outer-product learning rules is presented. With the gradient sensing of the acoustic field over a miniature microphone array as a pre-processing method, the proposed ICA implementation can separate and localize up to 3 sources in mild reverberant environment. The ICA processor is implemented in 0.5 µm CMOS technology and occupies 3 mm × 3 mm area. At 16 kHz sampling rate, ASIC consumes 195 µW power from a 3 V supply. The outer-product implementation of natural gradient and Herault-Jutten ICA update rules demonstrates comparable performance to benchmark FastICA algorithm in ideal conditions and more robust performance in noisy and reverberant environment. Experiments demonstrate perceptually clear separation and precise localization over wide range of separation angles of two speech sources presented through speakers positioned at 1.5 m from the array on a conference room table. The presented ASIC leads to a extreme small form factor and low power consumption microsystem for source separation and localization required in applications like intelligent hearing aids and wireless distributed acoustic sensor arrays.

Hybrid Speaker Recognition Using Universal Acoustic Model

NASA Astrophysics Data System (ADS)

Nishimura, Jun; Kuroda, Tadahiro

We propose a novel speaker recognition approach using a speaker-independent universal acoustic model (UAM) for sensornet applications. In sensornet applications such as “Business Microscope”, interactions among knowledge workers in an organization can be visualized by sensing face-to-face communication using wearable sensor nodes. In conventional studies, speakers are detected by comparing energy of input speech signals among the nodes. However, there are often synchronization errors among the nodes which degrade the speaker recognition performance. By focusing on property of the speaker's acoustic channel, UAM can provide robustness against the synchronization error. The overall speaker recognition accuracy is improved by combining UAM with the energy-based approach. For 0.1s speech inputs and 4 subjects, speaker recognition accuracy of 94% is achieved at the synchronization error less than 100ms.

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

Experimental investigation of the effects of the acoustical conditions in a simulated classroom on speech recognition and learning in children a

PubMed Central

Valente, Daniel L.; Plevinsky, Hallie M.; Franco, John M.; Heinrichs-Graham, Elizabeth C.; Lewis, Dawna E.

2012-01-01

The potential effects of acoustical environment on speech understanding are especially important as children enter school where students’ ability to hear and understand complex verbal information is critical to learning. However, this ability is compromised because of widely varied and unfavorable classroom acoustics. The extent to which unfavorable classroom acoustics affect children’s performance on longer learning tasks is largely unknown as most research has focused on testing children using words, syllables, or sentences as stimuli. In the current study, a simulated classroom environment was used to measure comprehension performance of two classroom learning activities: a discussion and lecture. Comprehension performance was measured for groups of elementary-aged students in one of four environments with varied reverberation times and background noise levels. The reverberation time was either 0.6 or 1.5 s, and the signal-to-noise level was either +10 or +7 dB. Performance is compared to adult subjects as well as to sentence-recognition in the same condition. Significant differences were seen in comprehension scores as a function of age and condition; both increasing background noise and reverberation degraded performance in comprehension tasks compared to minimal differences in measures of sentence-recognition. PMID:22280587

Communication in a noisy environment: Perception of one's own voice and speech enhancement

NASA Astrophysics Data System (ADS)

Le Cocq, Cecile

Workers in noisy industrial environments are often confronted to communication problems. Lost of workers complain about not being able to communicate easily with their coworkers when they wear hearing protectors. In consequence, they tend to remove their protectors, which expose them to the risk of hearing loss. In fact this communication problem is a double one: first the hearing protectors modify one's own voice perception; second they interfere with understanding speech from others. This double problem is examined in this thesis. When wearing hearing protectors, the modification of one's own voice perception is partly due to the occlusion effect which is produced when an earplug is inserted in the car canal. This occlusion effect has two main consequences: first the physiological noises in low frequencies are better perceived, second the perception of one's own voice is modified. In order to have a better understanding of this phenomenon, the literature results are analyzed systematically, and a new method to quantify the occlusion effect is developed. Instead of stimulating the skull with a bone vibrator or asking the subject to speak as is usually done in the literature, it has been decided to excite the buccal cavity with an acoustic wave. The experiment has been designed in such a way that the acoustic wave which excites the buccal cavity does not excite the external car or the rest of the body directly. The measurement of the hearing threshold in open and occluded car has been used to quantify the subjective occlusion effect for an acoustic wave in the buccal cavity. These experimental results as well as those reported in the literature have lead to a better understanding of the occlusion effect and an evaluation of the role of each internal path from the acoustic source to the internal car. The speech intelligibility from others is altered by both the high sound levels of noisy industrial environments and the speech signal attenuation due to hearing

Signal processing for passive detection and classification of underwater acoustic signals

NASA Astrophysics Data System (ADS)

Chung, Kil Woo

2011-12-01

This dissertation examines signal processing for passive detection, classification and tracking of underwater acoustic signals for improving port security and the security of coastal and offshore operations. First, we consider the problem of passive acoustic detection of a diver in a shallow water environment. A frequency-domain multi-band matched-filter approach to swimmer detection is presented. The idea is to break the frequency contents of the hydrophone signals into multiple narrow frequency bands, followed by time averaged (about half of a second) energy calculation over each band. Then, spectra composed of such energy samples over the chosen frequency bands are correlated to form a decision variable. The frequency bands with highest Signal/Noise ratio are used for detection. The performance of the proposed approach is demonstrated for experimental data collected for a diver in the Hudson River. We also propose a new referenceless frequency-domain multi-band detector which, unlike other reference-based detectors, does not require a diver specific signature. Instead, our detector matches to a general feature of the diver spectrum in the high frequency range: the spectrum is roughly periodic in time and approximately flat when the diver exhales. The performance of the proposed approach is demonstrated by using experimental data collected from the Hudson River. Moreover, we present detection, classification and tracking of small vessel signals. Hydroacoustic sensors can be applied for the detection of noise generated by vessels, and this noise can be used for vessel detection, classification and tracking. This dissertation presents recent improvements aimed at the measurement and separation of ship DEMON (Detection of Envelope Modulation on Noise) acoustic signatures in busy harbor conditions. Ship signature measurements were conducted in the Hudson River and NY Harbor. The DEMON spectra demonstrated much better temporal stability compared with the full ship

Decoding Speech With Integrated Hybrid Signals Recorded From the Human Ventral Motor Cortex

PubMed Central

Ibayashi, Kenji; Kunii, Naoto; Matsuo, Takeshi; Ishishita, Yohei; Shimada, Seijiro; Kawai, Kensuke; Saito, Nobuhito

2018-01-01

Restoration of speech communication for locked-in patients by means of brain computer interfaces (BCIs) is currently an important area of active research. Among the neural signals obtained from intracranial recordings, single/multi-unit activity (SUA/MUA), local field potential (LFP), and electrocorticography (ECoG) are good candidates for an input signal for BCIs. However, the question of which signal or which combination of the three signal modalities is best suited for decoding speech production remains unverified. In order to record SUA, LFP, and ECoG simultaneously from a highly localized area of human ventral sensorimotor cortex (vSMC), we fabricated an electrode the size of which was 7 by 13 mm containing sparsely arranged microneedle and conventional macro contacts. We determined which signal modality is the most capable of decoding speech production, and tested if the combination of these signals could improve the decoding accuracy of spoken phonemes. Feature vectors were constructed from spike frequency obtained from SUAs and event-related spectral perturbation derived from ECoG and LFP signals, then input to the decoder. The results showed that the decoding accuracy for five spoken vowels was highest when features from multiple signals were combined and optimized for each subject, and reached 59% when averaged across all six subjects. This result suggests that multi-scale signals convey complementary information for speech articulation. The current study demonstrated that simultaneous recording of multi-scale neuronal activities could raise decoding accuracy even though the recording area is limited to a small portion of cortex, which is advantageous for future implementation of speech-assisting BCIs. PMID:29674950

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

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.

Prediction Errors but Not Sharpened Signals Simulate Multivoxel fMRI Patterns during Speech Perception

PubMed Central

Davis, Matthew H.

2016-01-01

Successful perception depends on combining sensory input with prior knowledge. However, the underlying mechanism by which these two sources of information are combined is unknown. In speech perception, as in other domains, two functionally distinct coding schemes have been proposed for how expectations influence representation of sensory evidence. Traditional models suggest that expected features of the speech input are enhanced or sharpened via interactive activation (Sharpened Signals). Conversely, Predictive Coding suggests that expected features are suppressed so that unexpected features of the speech input (Prediction Errors) are processed further. The present work is aimed at distinguishing between these two accounts of how prior knowledge influences speech perception. By combining behavioural, univariate, and multivariate fMRI measures of how sensory detail and prior expectations influence speech perception with computational modelling, we provide evidence in favour of Prediction Error computations. Increased sensory detail and informative expectations have additive behavioural and univariate neural effects because they both improve the accuracy of word report and reduce the BOLD signal in lateral temporal lobe regions. However, sensory detail and informative expectations have interacting effects on speech representations shown by multivariate fMRI in the posterior superior temporal sulcus. When prior knowledge was absent, increased sensory detail enhanced the amount of speech information measured in superior temporal multivoxel patterns, but with informative expectations, increased sensory detail reduced the amount of measured information. Computational simulations of Sharpened Signals and Prediction Errors during speech perception could both explain these behavioural and univariate fMRI observations. However, the multivariate fMRI observations were uniquely simulated by a Prediction Error and not a Sharpened Signal model. The interaction between prior

Graph-based sensor fusion for classification of transient acoustic signals.

PubMed

Srinivas, Umamahesh; Nasrabadi, Nasser M; Monga, Vishal

2015-03-01

Advances in acoustic sensing have enabled the simultaneous acquisition of multiple measurements of the same physical event via co-located acoustic sensors. We exploit the inherent correlation among such multiple measurements for acoustic signal classification, to identify the launch/impact of munition (i.e., rockets, mortars). Specifically, we propose a probabilistic graphical model framework that can explicitly learn the class conditional correlations between the cepstral features extracted from these different measurements. Additionally, we employ symbolic dynamic filtering-based features, which offer improvements over the traditional cepstral features in terms of robustness to signal distortions. Experiments on real acoustic data sets show that our proposed algorithm outperforms conventional classifiers as well as the recently proposed joint sparsity models for multisensor acoustic classification. Additionally our proposed algorithm is less sensitive to insufficiency in training samples compared to competing approaches.

Detection and tracking of drones using advanced acoustic cameras

NASA Astrophysics Data System (ADS)

Busset, Joël.; Perrodin, Florian; Wellig, Peter; Ott, Beat; Heutschi, Kurt; Rühl, Torben; Nussbaumer, Thomas

2015-10-01

Recent events of drones flying over city centers, official buildings and nuclear installations stressed the growing threat of uncontrolled drone proliferation and the lack of real countermeasure. Indeed, detecting and tracking them can be difficult with traditional techniques. A system to acoustically detect and track small moving objects, such as drones or ground robots, using acoustic cameras is presented. The described sensor, is completely passive, and composed of a 120-element microphone array and a video camera. The acoustic imaging algorithm determines in real-time the sound power level coming from all directions, using the phase of the sound signals. A tracking algorithm is then able to follow the sound sources. Additionally, a beamforming algorithm selectively extracts the sound coming from each tracked sound source. This extracted sound signal can be used to identify sound signatures and determine the type of object. The described techniques can detect and track any object that produces noise (engines, propellers, tires, etc). It is a good complementary approach to more traditional techniques such as (i) optical and infrared cameras, for which the object may only represent few pixels and may be hidden by the blooming of a bright background, and (ii) radar or other echo-localization techniques, suffering from the weakness of the echo signal coming back to the sensor. The distance of detection depends on the type (frequency range) and volume of the noise emitted by the object, and on the background noise of the environment. Detection range and resilience to background noise were tested in both, laboratory environments and outdoor conditions. It was determined that drones can be tracked up to 160 to 250 meters, depending on their type. Speech extraction was also experimentally investigated: the speech signal of a person being 80 to 100 meters away can be captured with acceptable speech intelligibility.

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

PREDICTIVE MODELING OF ACOUSTIC SIGNALS FROM THERMOACOUSTIC POWER SENSORS (TAPS)

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

Dumm, Christopher M.; Vipperman, Jeffrey S.

2016-06-30

Thermoacoustic Power Sensor (TAPS) technology offers the potential for self-powered, wireless measurement of nuclear reactor core operating conditions. TAPS are based on thermoacoustic engines, which harness thermal energy from fission reactions to generate acoustic waves by virtue of gas motion through a porous stack of thermally nonconductive material. TAPS can be placed in the core, where they generate acoustic waves whose frequency and amplitude are proportional to the local temperature and radiation flux, respectively. TAPS acoustic signals are not measured directly at the TAPS; rather, they propagate wirelessly from an individual TAPS through the reactor, and ultimately to a low-powermore » receiver network on the vessel’s exterior. In order to rely on TAPS as primary instrumentation, reactor-specific models which account for geometric/acoustic complexities in the signal propagation environment must be used to predict the amplitude and frequency of TAPS signals at receiver locations. The reactor state may then be derived by comparing receiver signals to the reference levels established by predictive modeling. In this paper, we develop and experimentally benchmark a methodology for predictive modeling of the signals generated by a TAPS system, with the intent of subsequently extending these efforts to modeling of TAPS in a liquid sodium environmen« less

Facial expressions and the evolution of the speech rhythm.

PubMed

Ghazanfar, Asif A; Takahashi, Daniel Y

2014-06-01

In primates, different vocalizations are produced, at least in part, by making different facial expressions. Not surprisingly, humans, apes, and monkeys all recognize the correspondence between vocalizations and the facial postures associated with them. However, one major dissimilarity between monkey vocalizations and human speech is that, in the latter, the acoustic output and associated movements of the mouth are both rhythmic (in the 3- to 8-Hz range) and tightly correlated, whereas monkey vocalizations have a similar acoustic rhythmicity but lack the concommitant rhythmic facial motion. This raises the question of how we evolved from a presumptive ancestral acoustic-only vocal rhythm to the one that is audiovisual with improved perceptual sensitivity. According to one hypothesis, this bisensory speech rhythm evolved through the rhythmic facial expressions of ancestral primates. If this hypothesis has any validity, we expect that the extant nonhuman primates produce at least some facial expressions with a speech-like rhythm in the 3- to 8-Hz frequency range. Lip smacking, an affiliative signal observed in many genera of primates, satisfies this criterion. We review a series of studies using developmental, x-ray cineradiographic, EMG, and perceptual approaches with macaque monkeys producing lip smacks to further investigate this hypothesis. We then explore its putative neural basis and remark on important differences between lip smacking and speech production. Overall, the data support the hypothesis that lip smacking may have been an ancestral expression that was linked to vocal output to produce the original rhythmic audiovisual speech-like utterances in the human lineage.

Tolerance for audiovisual asynchrony is enhanced by the spectrotemporal fidelity of the speaker's mouth movements and speech.

PubMed

Shahin, Antoine J; Shen, Stanley; Kerlin, Jess R

2017-01-01

We examined the relationship between tolerance for audiovisual onset asynchrony (AVOA) and the spectrotemporal fidelity of the spoken words and the speaker's mouth movements. In two experiments that only varied in the temporal order of sensory modality, visual speech leading (exp1) or lagging (exp2) acoustic speech, participants watched intact and blurred videos of a speaker uttering trisyllabic words and nonwords that were noise vocoded with 4-, 8-, 16-, and 32-channels. They judged whether the speaker's mouth movements and the speech sounds were in-sync or out-of-sync . Individuals perceived synchrony (tolerated AVOA) on more trials when the acoustic speech was more speech-like (8 channels and higher vs. 4 channels), and when visual speech was intact than blurred (exp1 only). These findings suggest that enhanced spectrotemporal fidelity of the audiovisual (AV) signal prompts the brain to widen the window of integration promoting the fusion of temporally distant AV percepts.

A software tool for analyzing multichannel cochlear implant signals.

PubMed

Lai, Wai Kong; Bögli, Hans; Dillier, Norbert

2003-10-01

A useful and convenient means to analyze the radio frequency (RF) signals being sent by a speech processor to a cochlear implant would be to actually capture and display them with appropriate software. This is particularly useful for development or diagnostic purposes. sCILab (Swiss Cochlear Implant Laboratory) is such a PC-based software tool intended for the Nucleus family of Multichannel Cochlear Implants. Its graphical user interface provides a convenient and intuitive means for visualizing and analyzing the signals encoding speech information. Both numerical and graphic displays are available for detailed examination of the captured CI signals, as well as an acoustic simulation of these CI signals. sCILab has been used in the design and verification of new speech coding strategies, and has also been applied as an analytical tool in studies of how different parameter settings of existing speech coding strategies affect speech perception. As a diagnostic tool, it is also useful for troubleshooting problems with the external equipment of the cochlear implant systems.

The ``listener'' in the modeling of speech prosody

NASA Astrophysics Data System (ADS)

Kohler, Klaus J.

2004-05-01

Autosegmental-metrical modeling of speech prosody is principally speaker-oriented. The production of pitch patterns, in systematic lab speech experiments as well as in spontaneous speech corpora, is analyzed in f0 tracings, from which sequences of H(igh) and L(ow) are abstracted. The perceptual relevance of these pitch categories in the transmission from speakers to listeners is largely not conceptualized; thus their modeling in speech communication lacks an essential component. In the metalinguistic task of labeling speech data with the annotation system ToBI, the ``listener'' plays a subordinate role as well: H and L, being suggestive of signal values, are allocated with reference to f0 curves and little or no concern for perceptual classification by the trained labeler. The seriousness of this theoretical gap in the modeling of speech prosody is demonstrated by experimental data concerning f0-peak alignment. A number of papers in JASA have dealt with this topic from the point of synchronizing f0 with the vocal tract time course in acoustic output. However, perceptual experiments within the Kiel intonation model show that ``early,'' ``medial'' and ``late'' peak alignments need to be defined perceptually and that in doing so microprosodic variation has to be filtered out from the surface signal.

Spectral analysis method and sample generation for real time visualization of speech

NASA Astrophysics Data System (ADS)

Hobohm, Klaus

A method for translating speech signals into optical models, characterized by high sound discrimination and learnability and designed to provide to deaf persons a feedback towards control of their way of speaking, is presented. Important properties of speech production and perception processes and organs involved in these mechanisms are recalled in order to define requirements for speech visualization. It is established that the spectral representation of time, frequency and amplitude resolution of hearing must be fair and continuous variations of acoustic parameters of speech signal must be depicted by a continuous variation of images. A color table was developed for dynamic illustration and sonograms were generated with five spectral analysis methods such as Fourier transformations and linear prediction coding. For evaluating sonogram quality, test persons had to recognize consonant/vocal/consonant words and an optimized analysis method was achieved with a fast Fourier transformation and a postprocessor. A hardware concept of a real time speech visualization system, based on multiprocessor technology in a personal computer, is presented.

Study of environmental sound source identification based on hidden Markov model for robust speech recognition

NASA Astrophysics Data System (ADS)

Nishiura, Takanobu; Nakamura, Satoshi

2003-10-01

Humans communicate with each other through speech by focusing on the target speech among environmental sounds in real acoustic environments. We can easily identify the target sound from other environmental sounds. For hands-free speech recognition, the identification of the target speech from environmental sounds is imperative. This mechanism may also be important for a self-moving robot to sense the acoustic environments and communicate with humans. Therefore, this paper first proposes hidden Markov model (HMM)-based environmental sound source identification. Environmental sounds are modeled by three states of HMMs and evaluated using 92 kinds of environmental sounds. The identification accuracy was 95.4%. This paper also proposes a new HMM composition method that composes speech HMMs and an HMM of categorized environmental sounds for robust environmental sound-added speech recognition. As a result of the evaluation experiments, we confirmed that the proposed HMM composition outperforms the conventional HMM composition with speech HMMs and a noise (environmental sound) HMM trained using noise periods prior to the target speech in a captured signal. [Work supported by Ministry of Public Management, Home Affairs, Posts and Telecommunications of Japan.

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

Primary acoustic signal structure during free falling drop collision with a water surface

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

Chashechkin, Yu. D., E-mail: chakin@ipmnet.ru; Prokhorov, V. E., E-mail: prohorov@ipmnet.ru

2016-04-15

Consistent optical and acoustic techniques have been used to study the structure of hydrodynamic disturbances and acoustic signals generated as a free falling drop penetrates water. The relationship between the structures of hydrodynamic and acoustic perturbations arising as a result of a falling drop contacting with the water surface and subsequent immersion into water is traced. The primary acoustic signal is characterized, in addition to stably reproduced features (steep leading edge followed by long decay with local pressure maxima), by irregular high-frequency packets, which are studied for the first time. Reproducible experimental data are used to recognize constant and variablemore » components of the primary acoustic signal.« less

Children's need for favorable acoustics in schools

NASA Astrophysics Data System (ADS)

Nelson, Peggy B.

2003-10-01

Children continue to improve their understanding of speech in noise and reverberation throughout childhood and adolescence. They do not typically achieve adult performance levels until their late teenage years. As a result, schools that are designed to be acoustically adequate for adult understanding may be insufficient for full understanding by young children. In addition, children with hearing loss, those with attention problems, and those learning in a non-native language require even more favorable signal-to-noise ratios. This tutorial will review the literature gathered by the ANSl/ASA working group on classroom acoustics that shaped the recommendations of the working group. Special topics will include speech perception data from typically developing infants and children, from children with hearing loss, and from adults and children listening in a non-native language. In addition, the tutorial will overview recommendations contained within ANSI standard 12.60-2002: Acoustical Performance Criteria, Design Requirements, and Guidelines for Schools. The discussion will also include issues related to designing quiet classrooms and working with local schools and professionals.

Speech Rate Normalization and Phonemic Boundary Perception in Cochlear-Implant Users.

PubMed

Jaekel, Brittany N; Newman, Rochelle S; Goupell, Matthew J

2017-05-24

Normal-hearing (NH) listeners rate normalize, temporarily remapping phonemic category boundaries to account for a talker's speech rate. It is unknown if adults who use auditory prostheses called cochlear implants (CI) can rate normalize, as CIs transmit degraded speech signals to the auditory nerve. Ineffective adjustment to rate information could explain some of the variability in this population's speech perception outcomes. Phonemes with manipulated voice-onset-time (VOT) durations were embedded in sentences with different speech rates. Twenty-three CI and 29 NH participants performed a phoneme identification task. NH participants heard the same unprocessed stimuli as the CI participants or stimuli degraded by a sine vocoder, simulating aspects of CI processing. CI participants showed larger rate normalization effects (6.6 ms) than the NH participants (3.7 ms) and had shallower (less reliable) category boundary slopes. NH participants showed similarly shallow slopes when presented acoustically degraded vocoded signals, but an equal or smaller rate effect in response to reductions in available spectral and temporal information. CI participants can rate normalize, despite their degraded speech input, and show a larger rate effect compared to NH participants. CI participants may particularly rely on rate normalization to better maintain perceptual constancy of the speech signal.

Speech Rate Normalization and Phonemic Boundary Perception in Cochlear-Implant Users

PubMed Central

Newman, Rochelle S.; Goupell, Matthew J.

2017-01-01

Purpose Normal-hearing (NH) listeners rate normalize, temporarily remapping phonemic category boundaries to account for a talker's speech rate. It is unknown if adults who use auditory prostheses called cochlear implants (CI) can rate normalize, as CIs transmit degraded speech signals to the auditory nerve. Ineffective adjustment to rate information could explain some of the variability in this population's speech perception outcomes. Method Phonemes with manipulated voice-onset-time (VOT) durations were embedded in sentences with different speech rates. Twenty-three CI and 29 NH participants performed a phoneme identification task. NH participants heard the same unprocessed stimuli as the CI participants or stimuli degraded by a sine vocoder, simulating aspects of CI processing. Results CI participants showed larger rate normalization effects (6.6 ms) than the NH participants (3.7 ms) and had shallower (less reliable) category boundary slopes. NH participants showed similarly shallow slopes when presented acoustically degraded vocoded signals, but an equal or smaller rate effect in response to reductions in available spectral and temporal information. Conclusion CI participants can rate normalize, despite their degraded speech input, and show a larger rate effect compared to NH participants. CI participants may particularly rely on rate normalization to better maintain perceptual constancy of the speech signal. PMID:28395319

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.

Modeling of acoustic emission signal propagation in waveguides.

PubMed

Zelenyak, Andreea-Manuela; Hamstad, Marvin A; Sause, Markus G R

2015-05-21

Acoustic emission (AE) testing is a widely used nondestructive testing (NDT) method to investigate material failure. When environmental conditions are harmful for the operation of the sensors, waveguides are typically mounted in between the inspected structure and the sensor. Such waveguides can be built from different materials or have different designs in accordance with the experimental needs. All these variations can cause changes in the acoustic emission signals in terms of modal conversion, additional attenuation or shift in frequency content. A finite element method (FEM) was used to model acoustic emission signal propagation in an aluminum plate with an attached waveguide and was validated against experimental data. The geometry of the waveguide is systematically changed by varying the radius and height to investigate the influence on the detected signals. Different waveguide materials were implemented and change of material properties as function of temperature were taken into account. Development of the option of modeling different waveguide options replaces the time consuming and expensive trial and error alternative of experiments. Thus, the aim of this research has important implications for those who use waveguides for AE testing.

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.

The effect of habitat acoustics on common marmoset vocal signal transmission.

PubMed

Morrill, Ryan J; Thomas, A Wren; Schiel, Nicola; Souto, Antonio; Miller, Cory T

2013-09-01

Noisy acoustic environments present several challenges for the evolution of acoustic communication systems. Among the most significant is the need to limit degradation of spectro-temporal signal structure in order to maintain communicative efficacy. This can be achieved by selecting for several potentially complementary processes. Selection can act on behavioral mechanisms permitting signalers to control the timing and occurrence of signal production to avoid acoustic interference. Likewise, the signal itself may be the target of selection, biasing the evolution of its structure to comprise acoustic features that avoid interference from ambient noise or degrade minimally in the habitat. Here, we address the latter topic for common marmoset (Callithrix jacchus) long-distance contact vocalizations, known as phee calls. Our aim was to test whether this vocalization is specifically adapted for transmission in a species-typical forest habitat, the Atlantic forests of northeastern Brazil. We combined seasonal analyses of ambient habitat acoustics with experiments in which pure tones, clicks, and vocalizations were broadcast and rerecorded at different distances to characterize signal degradation in the habitat. Ambient sound was analyzed from intervals throughout the day and over rainy and dry seasons, showing temporal regularities across varied timescales. Broadcast experiment results indicated that the tone and click stimuli showed the typically inverse relationship between frequency and signaling efficacy. Although marmoset phee calls degraded over distance with marked predictability compared with artificial sounds, they did not otherwise appear to be specially designed for increased transmission efficacy or minimal interference in this habitat. We discuss these data in the context of other similar studies and evidence of potential behavioral mechanisms for avoiding acoustic interference in order to maintain effective vocal communication in common marmosets. © 2013

The Effect of Habitat Acoustics on Common Marmoset Vocal Signal Transmission

PubMed Central

MORRILL, RYAN J.; THOMAS, A. WREN; SCHIEL, NICOLA; SOUTO, ANTONIO; MILLER, CORY T.

2013-01-01

Noisy acoustic environments present several challenges for the evolution of acoustic communication systems. Among the most significant is the need to limit degradation of spectro-temporal signal structure in order to maintain communicative efficacy. This can be achieved by selecting for several potentially complementary processes. Selection can act on behavioral mechanisms permitting signalers to control the timing and occurrence of signal production to avoid acoustic interference. Likewise, the signal itself may be the target of selection, biasing the evolution of its structure to comprise acoustic features that avoid interference from ambient noise or degrade minimally in the habitat. Here, we address the latter topic for common marmoset (Callithrix jacchus) long-distance contact vocalizations, known as phee calls. Our aim was to test whether this vocalization is specifically adapted for transmission in a species-typical forest habitat, the Atlantic forests of northeastern Brazil. We combined seasonal analyses of ambient habitat acoustics with experiments in which pure tones, clicks, and vocalizations were broadcast and rerecorded at different distances to characterize signal degradation in the habitat. Ambient sound was analyzed from intervals throughout the day and over rainy and dry seasons, showing temporal regularities across varied timescales. Broadcast experiment results indicated that the tone and click stimuli showed the typically inverse relationship between frequency and signaling efficacy. Although marmoset phee calls degraded over distance with marked predictability compared with artificial sounds, they did not otherwise appear to be specially designed for increased transmission efficacy or minimal interference in this habitat. We discuss these data in the context of other similar studies and evidence of potential behavioral mechanisms for avoiding acoustic interference in order to maintain effective vocal communication in common marmosets. PMID

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

Cochlear implantation with hearing preservation yields significant benefit for speech recognition in complex listening environments.

PubMed

Gifford, René H; Dorman, Michael F; Skarzynski, Henryk; Lorens, Artur; Polak, Marek; Driscoll, Colin L W; Roland, Peter; Buchman, Craig A

2013-01-01

The aim of this study was to assess the benefit of having preserved acoustic hearing in the implanted ear for speech recognition in complex listening environments. The present study included a within-subjects, repeated-measures design including 21 English-speaking and 17 Polish-speaking cochlear implant (CI) recipients with preserved acoustic hearing in the implanted ear. The patients were implanted with electrodes that varied in insertion depth from 10 to 31 mm. Mean preoperative low-frequency thresholds (average of 125, 250, and 500 Hz) in the implanted ear were 39.3 and 23.4 dB HL for the English- and Polish-speaking participants, respectively. In one condition, speech perception was assessed in an eight-loudspeaker environment in which the speech signals were presented from one loudspeaker and restaurant noise was presented from all loudspeakers. In another condition, the signals were presented in a simulation of a reverberant environment with a reverberation time of 0.6 sec. The response measures included speech reception thresholds (SRTs) and percent correct sentence understanding for two test conditions: CI plus low-frequency hearing in the contralateral ear (bimodal condition) and CI plus low-frequency hearing in both ears (best-aided condition). A subset of six English-speaking listeners were also assessed on measures of interaural time difference thresholds for a 250-Hz signal. Small, but significant, improvements in performance (1.7-2.1 dB and 6-10 percentage points) were found for the best-aided condition versus the bimodal condition. Postoperative thresholds in the implanted ear were correlated with the degree of electric and acoustic stimulation (EAS) benefit for speech recognition in diffuse noise. There was no reliable relationship among measures of audiometric threshold in the implanted ear nor elevation in threshold after surgery and improvement in speech understanding in reverberation. There was a significant correlation between interaural time

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

Cued Speech for Enhancing Speech Perception and First Language Development of Children With Cochlear Implants

PubMed Central

Leybaert, Jacqueline; LaSasso, Carol J.

2010-01-01

Nearly 300 million people worldwide have moderate to profound hearing loss. Hearing impairment, if not adequately managed, has strong socioeconomic and affective impact on individuals. Cochlear implants have become the most effective vehicle for helping profoundly deaf children and adults to understand spoken language, to be sensitive to environmental sounds, and, to some extent, to listen to music. The auditory information delivered by the cochlear implant remains non-optimal for speech perception because it delivers a spectrally degraded signal and lacks some of the fine temporal acoustic structure. In this article, we discuss research revealing the multimodal nature of speech perception in normally-hearing individuals, with important inter-subject variability in the weighting of auditory or visual information. We also discuss how audio-visual training, via Cued Speech, can improve speech perception in cochlear implantees, particularly in noisy contexts. Cued Speech is a system that makes use of visual information from speechreading combined with hand shapes positioned in different places around the face in order to deliver completely unambiguous information about the syllables and the phonemes of spoken language. We support our view that exposure to Cued Speech before or after the implantation could be important in the aural rehabilitation process of cochlear implantees. We describe five lines of research that are converging to support the view that Cued Speech can enhance speech perception in individuals with cochlear implants. PMID:20724357

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…

Neural mechanisms underlying auditory feedback control of speech

PubMed Central

Reilly, Kevin J.; Guenther, Frank H.

2013-01-01

The neural substrates underlying auditory feedback control of speech were investigated using a combination of functional magnetic resonance imaging (fMRI) and computational modeling. Neural responses were measured while subjects spoke monosyllabic words under two conditions: (i) normal auditory feedback of their speech, and (ii) auditory feedback in which the first formant frequency of their speech was unexpectedly shifted in real time. Acoustic measurements showed compensation to the shift within approximately 135 ms of onset. Neuroimaging revealed increased activity in bilateral superior temporal cortex during shifted feedback, indicative of neurons coding mismatches between expected and actual auditory signals, as well as right prefrontal and Rolandic cortical activity. Structural equation modeling revealed increased influence of bilateral auditory cortical areas on right frontal areas during shifted speech, indicating that projections from auditory error cells in posterior superior temporal cortex to motor correction cells in right frontal cortex mediate auditory feedback control of speech. PMID:18035557

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

Channel noise enhances signal detectability in a model of acoustic neuron through the stochastic resonance paradigm.

PubMed

Liberti, M; Paffi, A; Maggio, F; De Angelis, A; Apollonio, F; d'Inzeo, G

2009-01-01

A number of experimental investigations have evidenced the extraordinary sensitivity of neuronal cells to weak input stimulations, including electromagnetic (EM) fields. Moreover, it has been shown that biological noise, due to random channels gating, acts as a tuning factor in neuronal processing, according to the stochastic resonant (SR) paradigm. In this work the attention is focused on noise arising from the stochastic gating of ionic channels in a model of Ranvier node of acoustic fibers. The small number of channels gives rise to a high noise level, which is able to cause a spike train generation even in the absence of stimulations. A SR behavior has been observed in the model for the detection of sinusoidal signals at frequencies typical of the speech.

Intense acoustic bursts as a signal-enhancement mechanism in ultrasound-modulated optical tomography.

PubMed

Kim, Chulhong; Zemp, Roger J; Wang, Lihong V

2006-08-15

Biophotonic imaging with ultrasound-modulated optical tomography (UOT) promises ultrasonically resolved imaging in biological tissues. A key challenge in this imaging technique is a low signal-to-noise ratio (SNR). We show significant UOT signal enhancement by using intense time-gated acoustic bursts. A CCD camera captured the speckle pattern from a laser-illuminated tissue phantom. Differences in speckle contrast were observed when ultrasonic bursts were applied, compared with when no ultrasound was applied. When CCD triggering was synchronized with burst initiation, acoustic-radiation-force-induced displacements were detected. To avoid mechanical contrast in UOT images, the CCD camera acquisition was delayed several milliseconds until transient effects of acoustic radiation force attenuated to a satisfactory level. The SNR of our system was sufficiently high to provide an image pixel per acoustic burst without signal averaging. Because of the substantially improved SNR, the use of intense acoustic bursts is a promising signal enhancement strategy for UOT.

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.

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…

Speech perception as an active cognitive process

PubMed Central

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

2014-01-01

One view of speech perception is that acoustic signals are transformed into representations for pattern matching to determine linguistic structure. This process can be taken as a statistical pattern-matching problem, assuming realtively stable linguistic categories are characterized by neural representations related to auditory properties of speech that can be compared to speech input. This kind of pattern matching can be termed a passive process which implies rigidity of processing with few demands on cognitive processing. An alternative view is that speech recognition, even in early stages, is an active process in which speech analysis is attentionally guided. Note that this does not mean consciously guided but that information-contingent changes in early auditory encoding can occur as a function of context and experience. Active processing assumes that attention, plasticity, and listening goals are important in considering how listeners cope with adverse circumstances that impair hearing by masking noise in the environment or hearing loss. Although theories of speech perception have begun to incorporate some active processing, they seldom treat early speech encoding as plastic and attentionally guided. Recent research has suggested that speech perception is the product of both feedforward and feedback interactions between a number of brain regions that include descending projections perhaps as far downstream as the cochlea. It is important to understand how the ambiguity of the speech signal and constraints of context dynamically determine cognitive resources recruited during perception including focused attention, learning, and working memory. Theories of speech perception need to go beyond the current corticocentric approach in order to account for the intrinsic dynamics of the auditory encoding of speech. In doing so, this may provide new insights into ways in which hearing disorders and loss may be treated either through augementation or therapy. PMID

Characterizing, synthesizing, and/or canceling out acoustic signals from sound sources

DOEpatents

Holzrichter, John F [Berkeley, CA; Ng, Lawrence C [Danville, CA

2007-03-13

A system for characterizing, synthesizing, and/or canceling out acoustic signals from inanimate and animate sound sources. Electromagnetic sensors monitor excitation sources in sound producing systems, such as animate sound sources such as the human voice, or from machines, musical instruments, and various other structures. Acoustical output from these sound producing systems is also monitored. From such information, a transfer function characterizing the sound producing system is generated. From the transfer function, acoustical output from the sound producing system may be synthesized or canceled. The systems disclosed enable accurate calculation of transfer functions relating specific excitations to specific acoustical outputs. Knowledge of such signals and functions can be used to effect various sound replication, sound source identification, and sound cancellation applications.

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.

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.

Acoustic change detection algorithm using an FM radio

NASA Astrophysics Data System (ADS)

Goldman, Geoffrey H.; Wolfe, Owen

2012-06-01

The U.S. Army is interested in developing low-cost, low-power, non-line-of-sight sensors for monitoring human activity. One modality that is often overlooked is active acoustics using sources of opportunity such as speech or music. Active acoustics can be used to detect human activity by generating acoustic images of an area at different times, then testing for changes among the imagery. A change detection algorithm was developed to detect physical changes in a building, such as a door changing positions or a large box being moved using acoustics sources of opportunity. The algorithm is based on cross correlating the acoustic signal measured from two microphones. The performance of the algorithm was shown using data generated with a hand-held FM radio as a sound source and two microphones. The algorithm could detect a door being opened in a hallway.

Subauditory Speech Recognition based on EMG/EPG Signals

NASA Technical Reports Server (NTRS)

Jorgensen, Charles; Lee, Diana Dee; Agabon, Shane; Lau, Sonie (Technical Monitor)

2003-01-01

Sub-vocal electromyogram/electro palatogram (EMG/EPG) signal classification is demonstrated as a method for silent speech recognition. Recorded electrode signals from the larynx and sublingual areas below the jaw are noise filtered and transformed into features using complex dual quad tree wavelet transforms. Feature sets for six sub-vocally pronounced words are trained using a trust region scaled conjugate gradient neural network. Real time signals for previously unseen patterns are classified into categories suitable for primitive control of graphic objects. Feature construction, recognition accuracy and an approach for extension of the technique to a variety of real world application areas are presented.

Effects of Compression on Speech Acoustics, Intelligibility, and Sound Quality

PubMed Central

Souza, Pamela E.

2002-01-01

The topic of compression has been discussed quite extensively in the last 20 years (eg, Braida et al., 1982; Dillon, 1996, 2000; Dreschler, 1992; Hickson, 1994; Kuk, 2000 and 2002; Kuk and Ludvigsen, 1999; Moore, 1990; Van Tasell, 1993; Venema, 2000; Verschuure et al., 1996; Walker and Dillon, 1982). However, the latest comprehensive update by this journal was published in 1996 (Kuk, 1996). Since that time, use of compression hearing aids has increased dramatically, from half of hearing aids dispensed only 5 years ago to four out of five hearing aids dispensed today (Strom, 2002b). Most of today's digital and digitally programmable hearing aids are compression devices (Strom, 2002a). It is probable that within a few years, very few patients will be fit with linear hearing aids. Furthermore, compression has increased in complexity, with greater numbers of parameters under the clinician's control. Ideally, these changes will translate to greater flexibility and precision in fitting and selection. However, they also increase the need for information about the effects of compression amplification on speech perception and speech quality. As evidenced by the large number of sessions at professional conferences on fitting compression hearing aids, clinicians continue to have questions about compression technology and when and how it should be used. How does compression work? Who are the best candidates for this technology? How should adjustable parameters be set to provide optimal speech recognition? What effect will compression have on speech quality? These and other questions continue to drive our interest in this technology. This article reviews the effects of compression on the speech signal and the implications for speech intelligibility, quality, and design of clinical procedures. PMID:25425919

The interaction of acoustic and linguistic grouping cues in auditory object formation

NASA Astrophysics Data System (ADS)

Shapley, Kathy; Carrell, Thomas

2005-09-01

One of the earliest explanations for good speech intelligibility in poor listening situations was context [Miller et al., J. Exp. Psychol. 41 (1951)]. Context presumably allows listeners to group and predict speech appropriately and is known as a top-down listening strategy. Amplitude comodulation is another mechanism that has been shown to improve sentence intelligibility. Amplitude comodulation provides acoustic grouping information without changing the linguistic content of the desired signal [Carrell and Opie, Percept. Psychophys. 52 (1992); Hu and Wang, Proceedings of ICASSP-02 (2002)] and is considered a bottom-up process. The present experiment investigated how amplitude comodulation and semantic information combined to improve speech intelligibility. Sentences with high- and low-predictability word sequences [Boothroyd and Nittrouer, J. Acoust. Soc. Am. 84 (1988)] were constructed in two different formats: time-varying sinusoidal sentences (TVS) and reduced-channel sentences (RC). The stimuli were chosen because they minimally represent the traditionally defined speech cues and therefore emphasized the importance of the high-level context effects and low-level acoustic grouping cues. Results indicated that semantic information did not influence intelligibility levels of TVS and RC sentences. In addition amplitude modulation aided listeners' intelligibility scores in the TVS condition but hindered listeners' intelligibility scores in the RC condition.

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.

[A modified speech enhancement algorithm for electronic cochlear implant and its digital signal processing realization].

PubMed

Wang, Yulin; Tian, Xuelong

2014-08-01

In order to improve the speech quality and auditory perceptiveness of electronic cochlear implant under strong noise background, a speech enhancement system used for electronic cochlear implant front-end was constructed. Taking digital signal processing (DSP) as the core, the system combines its multi-channel buffered serial port (McBSP) data transmission channel with extended audio interface chip TLV320AIC10, so speech signal acquisition and output with high speed are realized. Meanwhile, due to the traditional speech enhancement method which has the problems as bad adaptability, slow convergence speed and big steady-state error, versiera function and de-correlation principle were used to improve the existing adaptive filtering algorithm, which effectively enhanced the quality of voice communications. Test results verified the stability of the system and the de-noising performance of the algorithm, and it also proved that they could provide clearer speech signals for the deaf or tinnitus patients.

Calculation of selective filters of a device for primary analysis of speech signals

NASA Astrophysics Data System (ADS)

Chudnovskii, L. S.; Ageev, V. M.

2014-07-01

The amplitude-frequency responses of filters for primary analysis of speech signals, which have a low quality factor and a high rolloff factor in the high-frequency range, are calculated using the linear theory of speech production and psychoacoustic measurement data. The frequency resolution of the filter system for a sinusoidal signal is 40-200 Hz. The modulation-frequency resolution of amplitude- and frequency-modulated signals is 3-6 Hz. The aforementioned features of the calculated filters are close to the amplitudefrequency responses of biological auditory systems at the level of the eighth nerve.

Frontal top-down signals increase coupling of auditory low-frequency oscillations to continuous speech in human listeners.

PubMed

Park, Hyojin; Ince, Robin A A; Schyns, Philippe G; Thut, Gregor; Gross, Joachim

2015-06-15

Humans show a remarkable ability to understand continuous speech even under adverse listening conditions. This ability critically relies on dynamically updated predictions of incoming sensory information, but exactly how top-down predictions improve speech processing is still unclear. Brain oscillations are a likely mechanism for these top-down predictions [1, 2]. Quasi-rhythmic components in speech are known to entrain low-frequency oscillations in auditory areas [3, 4], and this entrainment increases with intelligibility [5]. We hypothesize that top-down signals from frontal brain areas causally modulate the phase of brain oscillations in auditory cortex. We use magnetoencephalography (MEG) to monitor brain oscillations in 22 participants during continuous speech perception. We characterize prominent spectral components of speech-brain coupling in auditory cortex and use causal connectivity analysis (transfer entropy) to identify the top-down signals driving this coupling more strongly during intelligible speech than during unintelligible speech. We report three main findings. First, frontal and motor cortices significantly modulate the phase of speech-coupled low-frequency oscillations in auditory cortex, and this effect depends on intelligibility of speech. Second, top-down signals are significantly stronger for left auditory cortex than for right auditory cortex. Third, speech-auditory cortex coupling is enhanced as a function of stronger top-down signals. Together, our results suggest that low-frequency brain oscillations play a role in implementing predictive top-down control during continuous speech perception and that top-down control is largely directed at left auditory cortex. This suggests a close relationship between (left-lateralized) speech production areas and the implementation of top-down control in continuous speech perception. Copyright © 2015 The Authors. Published by Elsevier Ltd.. All rights reserved.

Frontal Top-Down Signals Increase Coupling of Auditory Low-Frequency Oscillations to Continuous Speech in Human Listeners

PubMed Central

Park, Hyojin; Ince, Robin A.A.; Schyns, Philippe G.; Thut, Gregor; Gross, Joachim

2015-01-01

Summary Humans show a remarkable ability to understand continuous speech even under adverse listening conditions. This ability critically relies on dynamically updated predictions of incoming sensory information, but exactly how top-down predictions improve speech processing is still unclear. Brain oscillations are a likely mechanism for these top-down predictions [1, 2]. Quasi-rhythmic components in speech are known to entrain low-frequency oscillations in auditory areas [3, 4], and this entrainment increases with intelligibility [5]. We hypothesize that top-down signals from frontal brain areas causally modulate the phase of brain oscillations in auditory cortex. We use magnetoencephalography (MEG) to monitor brain oscillations in 22 participants during continuous speech perception. We characterize prominent spectral components of speech-brain coupling in auditory cortex and use causal connectivity analysis (transfer entropy) to identify the top-down signals driving this coupling more strongly during intelligible speech than during unintelligible speech. We report three main findings. First, frontal and motor cortices significantly modulate the phase of speech-coupled low-frequency oscillations in auditory cortex, and this effect depends on intelligibility of speech. Second, top-down signals are significantly stronger for left auditory cortex than for right auditory cortex. Third, speech-auditory cortex coupling is enhanced as a function of stronger top-down signals. Together, our results suggest that low-frequency brain oscillations play a role in implementing predictive top-down control during continuous speech perception and that top-down control is largely directed at left auditory cortex. This suggests a close relationship between (left-lateralized) speech production areas and the implementation of top-down control in continuous speech perception. PMID:26028433

Negative Effect of Acoustic Panels on Listening Effort in a Classroom Environment.

PubMed

Amlani, Amyn M; Russo, Timothy A

Acoustic panels are used to lessen the pervasive effects of noise and reverberation on speech understanding in a classroom environment. These panels, however, predominately absorb high-frequency energy important to speech understanding. Therefore, a classroom environment treated with acoustic panels might negatively influence the transmission of the target signal, resulting in an increase in listening effort exerted by the listener. Acoustic panels were installed in a public school environment that did not meet the ANSI-recommended guidelines for classroom design. We assessed the modifications to the acoustic climate by quantifying the effect of (1) acoustic panel (i.e., without, with) on the transmission of a standardized target signal at different seat positions (i.e., A-D) using the Speech Transmission Index (STI) and (2) acoustic panel and seat position on listening-effort performance in a group of third-grade students having normal-hearing sensitivity using a dual-task paradigm. STI measurements are described qualitatively. We used a repeated-measures randomized design to assess listening-effort performance of monosyllabic words in a primary task and digit recall in a secondary task for the independent variables of acoustic panel and seat position. Twenty-seven, third-grade students (12 males, 15 females), ranging in age from 8.3 to 9.4 yr (mean = 8.7 yr, standard deviation = 0.7), participated in this study. Qualitatively, we performed STI measurements under both testing conditions (i.e., panel and seat location). For the primary task of the dual-task paradigm, participants heard a ten-item list of monosyllabic words (i.e., ten words per list) recorded through a manikin in the classroom environment without and with acoustic panels and at different seat positions. Participants were asked to repeat each word exactly as it was heard. During the secondary task, participants were shown a single, random string of five digits before the presentation of the

Real-time GMAW quality classification using an artificial neural network with airborne acoustic signals as inputs

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

Matteson, A.; Morris, R.; Tate, R.

1993-12-31

The acoustic signal produced by the gas metal arc welding (GMAW) arc contains information about the behavior of the arc column, the molten pool and droplet transfer. It is possible to detect some defect producing conditions from the acoustic signal from the GMAW arc. An intelligent sensor, called the Weld Acoustic Monitor (WAM) has been developed to take advantage of this acoustic information in order to provide real-time quality assessment information for process control. The WAM makes use of an Artificial Neural Network (ANN) to classify the characteristic arc acoustic signals of acceptable and unacceptable welds. The ANN used inmore » the Weld Acoustic Monitor developed its own set of rules for this classification problem by learning a data base of known GMAW acoustic signals.« less

Effects of hearing aid settings for electric-acoustic stimulation.

PubMed

Dillon, Margaret T; Buss, Emily; Pillsbury, Harold C; Adunka, Oliver F; Buchman, Craig A; Adunka, Marcia C

2014-02-01

Cochlear implant (CI) recipients with postoperative hearing preservation may utilize an ipsilateral bimodal listening condition known as electric-acoustic stimulation (EAS). Studies on EAS have reported significant improvements in speech perception abilities over CI-alone listening conditions. Adjustments to the hearing aid (HA) settings to match prescription targets routinely used in the programming of conventional amplification may provide additional gains in speech perception abilities. Investigate the difference in users' speech perception scores when listening with the recommended HA settings for EAS patients versus HA settings adjusted to match National Acoustic Laboratories' nonlinear fitting procedure version 1 (NAL-NL1) targets. Prospective analysis of the influence of HA settings. Nine EAS recipients with greater than 12 mo of listening experience with the DUET speech processor. Subjects were tested in the EAS listening condition with two different HA setting configurations. Speech perception materials included consonant-nucleus-consonant (CNC) words in quiet, AzBio sentences in 10-talker speech babble at a signal-to-noise ratio (SNR) of +10, and the Bamford-Kowal-Bench sentences in noise (BKB-SIN) test. The speech perception performance on each test measure was compared between the two HA configurations. Subjects experienced a significant improvement in speech perception abilities with the HA settings adjusted to match NAL-NL1 targets over the recommended HA settings. EAS subjects have been shown to experience improvements in speech perception abilities when listening to ipsilateral combined stimulation. This population's abilities may be underestimated with current HA settings. Tailoring the HA output to the patient's individual hearing loss offers improved outcomes on speech perception measures. American Academy of Audiology.

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.

On the Perception of Speech Sounds as Biologically Significant Signals1,2

PubMed Central

Pisoni, David B.

2012-01-01

This paper reviews some of the major evidence and arguments currently available to support the view that human speech perception may require the use of specialized neural mechanisms for perceptual analysis. Experiments using synthetically produced speech signals with adults are briefly summarized and extensions of these results to infants and other organisms are reviewed with an emphasis towards detailing those aspects of speech perception that may require some need for specialized species-specific processors. Finally, some comments on the role of early experience in perceptual development are provided as an attempt to identify promising areas of new research in speech perception. PMID:399200

Coupled Research in Ocean Acoustics and Signal Processing for the Next Generation of Underwater Acoustic Communication Systems

DTIC Science & Technology

2016-08-05

JPAnalytics LLC CC: DCMA Boston DTIC Director, NRL Progress Report #8 Coupled Research in Ocean Acoustics and Signal Processing for the Next...Generation of Underwater Acoustic Communication Systems Principal Investigator’s Name: Dr. James Preisig Period Covered By Report: 1/20/2016 to 4/19/2016...Technical work this period has spanned two areas. The first of these is VHF Acoustics . During this time period, the Principle Investigator worked with Dr

Applications of sub-audible speech recognition based upon electromyographic signals

NASA Technical Reports Server (NTRS)

Jorgensen, C. Charles (Inventor); Betts, Bradley J. (Inventor)

2009-01-01

Method and system for generating electromyographic or sub-audible signals (''SAWPs'') and for transmitting and recognizing the SAWPs that represent the original words and/or phrases. The SAWPs may be generated in an environment that interferes excessively with normal speech or that requires stealth communications, and may be transmitted using encoded, enciphered or otherwise transformed signals that are less subject to signal distortion or degradation in the ambient environment.

Speech Enhancement based on the Dominant Classification Between Speech and Noise Using Feature Data in Spectrogram of Observation Signal

NASA Astrophysics Data System (ADS)

Nomura, Yukihiro; Lu, Jianming; Sekiya, Hiroo; Yahagi, Takashi

This paper presents a speech enhancement using the classification between the dominants of speech and noise. In our system, a new classification scheme between the dominants of speech and noise is proposed. The proposed classifications use the standard deviation of the spectrum of observation signal in each band. We introduce two oversubtraction factors for the dominants of speech and noise, respectively. And spectral subtraction is carried out after the classification. The proposed method is tested on several noise types from the Noisex-92 database. From the investigation of segmental SNR, Itakura-Saito distance measure, inspection of spectrograms and listening tests, the proposed system is shown to be effective to reduce background noise. Moreover, the enhanced speech using our system generates less musical noise and distortion than that of conventional systems.

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

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

"Who" is saying "what"? Brain-based decoding of human voice and speech.

PubMed

Formisano, Elia; De Martino, Federico; Bonte, Milene; Goebel, Rainer

2008-11-07

Can we decipher speech content ("what" is being said) and speaker identity ("who" is saying it) from observations of brain activity of a listener? Here, we combine functional magnetic resonance imaging with a data-mining algorithm and retrieve what and whom a person is listening to from the neural fingerprints that speech and voice signals elicit in the listener's auditory cortex. These cortical fingerprints are spatially distributed and insensitive to acoustic variations of the input so as to permit the brain-based recognition of learned speech from unknown speakers and of learned voices from previously unheard utterances. Our findings unravel the detailed cortical layout and computational properties of the neural populations at the basis of human speech recognition and speaker identification.

Perceptual weighting of the envelope and fine structure across frequency bands for sentence intelligibility: Effect of interruption at the syllabic-rate and periodic-rate of speech

PubMed Central

Fogerty, Daniel

2011-01-01

Listeners often only have fragments of speech available to understand the intended message due to competing background noise. In order to maximize successful speech recognition, listeners must allocate their perceptual resources to the most informative acoustic properties. The speech signal contains temporally-varying acoustics in the envelope and fine structure that are present across the frequency spectrum. Understanding how listeners perceptually weigh these acoustic properties in different frequency regions during interrupted speech is essential for the design of assistive listening devices. This study measured the perceptual weighting of young normal-hearing listeners for the envelope and fine structure in each of three frequency bands for interrupted sentence materials. Perceptual weights were obtained during interruption at the syllabic rate (i.e., 4 Hz) and the periodic rate (i.e., 128 Hz) of speech. Potential interruption interactions with fundamental frequency information were investigated by shifting the natural pitch contour higher relative to the interruption rate. The availability of each acoustic property was varied independently by adding noise at different levels. Perceptual weights were determined by correlating a listener’s performance with the availability of each acoustic property on a trial-by-trial basis. Results demonstrated similar relative weights across the interruption conditions, with emphasis on the envelope in high-frequencies. PMID:21786914

Speech perception in noise with a harmonic complex excited vocoder.

PubMed

Churchill, Tyler H; Kan, Alan; Goupell, Matthew J; Ihlefeld, Antje; Litovsky, Ruth Y

2014-04-01

A cochlear implant (CI) presents band-pass-filtered acoustic envelope information by modulating current pulse train levels. Similarly, a vocoder presents envelope information by modulating an acoustic carrier. By studying how normal hearing (NH) listeners are able to understand degraded speech signals with a vocoder, the parameters that best simulate electric hearing and factors that might contribute to the NH-CI performance difference may be better understood. A vocoder with harmonic complex carriers (fundamental frequency, f0 = 100 Hz) was used to study the effect of carrier phase dispersion on speech envelopes and intelligibility. The starting phases of the harmonic components were randomly dispersed to varying degrees prior to carrier filtering and modulation. NH listeners were tested on recognition of a closed set of vocoded words in background noise. Two sets of synthesis filters simulated different amounts of current spread in CIs. Results showed that the speech vocoded with carriers whose starting phases were maximally dispersed was the most intelligible. Superior speech understanding may have been a result of the flattening of the dispersed-phase carrier's intrinsic temporal envelopes produced by the large number of interacting components in the high-frequency channels. Cross-correlogram analyses of auditory nerve model simulations confirmed that randomly dispersing the carrier's component starting phases resulted in better neural envelope representation. However, neural metrics extracted from these analyses were not found to accurately predict speech recognition scores for all vocoded speech conditions. It is possible that central speech understanding mechanisms are insensitive to the envelope-fine structure dichotomy exploited by vocoders.

Acoustic/seismic signal propagation and sensor performance modeling

NASA Astrophysics Data System (ADS)

Wilson, D. Keith; Marlin, David H.; Mackay, Sean

2007-04-01

Performance, optimal employment, and interpretation of data from acoustic and seismic sensors depend strongly and in complex ways on the environment in which they operate. Software tools for guiding non-expert users of acoustic and seismic sensors are therefore much needed. However, such tools require that many individual components be constructed and correctly connected together. These components include the source signature and directionality, representation of the atmospheric and terrain environment, calculation of the signal propagation, characterization of the sensor response, and mimicking of the data processing at the sensor. Selection of an appropriate signal propagation model is particularly important, as there are significant trade-offs between output fidelity and computation speed. Attenuation of signal energy, random fading, and (for array systems) variations in wavefront angle-of-arrival should all be considered. Characterization of the complex operational environment is often the weak link in sensor modeling: important issues for acoustic and seismic modeling activities include the temporal/spatial resolution of the atmospheric data, knowledge of the surface and subsurface terrain properties, and representation of ambient background noise and vibrations. Design of software tools that address these challenges is illustrated with two examples: a detailed target-to-sensor calculation application called the Sensor Performance Evaluator for Battlefield Environments (SPEBE) and a GIS-embedded approach called Battlefield Terrain Reasoning and Awareness (BTRA).

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

Acoustic and perceptual effects of magnifying interaural difference cues in a simulated "binaural" hearing aid.

PubMed

de Taillez, Tobias; Grimm, Giso; Kollmeier, Birger; Neher, Tobias

2018-06-01

To investigate the influence of an algorithm designed to enhance or magnify interaural difference cues on speech signals in noisy, spatially complex conditions using both technical and perceptual measurements. To also investigate the combination of interaural magnification (IM), monaural microphone directionality (DIR), and binaural coherence-based noise reduction (BC). Speech-in-noise stimuli were generated using virtual acoustics. A computational model of binaural hearing was used to analyse the spatial effects of IM. Predicted speech quality changes and signal-to-noise-ratio (SNR) improvements were also considered. Additionally, a listening test was carried out to assess speech intelligibility and quality. Listeners aged 65-79 years with and without sensorineural hearing loss (N = 10 each). IM increased the horizontal separation of concurrent directional sound sources without introducing any major artefacts. In situations with diffuse noise, however, the interaural difference cues were distorted. Preprocessing the binaural input signals with DIR reduced distortion. IM influenced neither speech intelligibility nor speech quality. The IM algorithm tested here failed to improve speech perception in noise, probably because of the dispersion and inconsistent magnification of interaural difference cues in complex environments.

Inferring imagined speech using EEG signals: a new approach using Riemannian manifold features

NASA Astrophysics Data System (ADS)

Nguyen, Chuong H.; Karavas, George K.; Artemiadis, Panagiotis

2018-02-01

Objective. In this paper, we investigate the suitability of imagined speech for brain-computer interface (BCI) applications. Approach. A novel method based on covariance matrix descriptors, which lie in Riemannian manifold, and the relevance vector machines classifier is proposed. The method is applied on electroencephalographic (EEG) signals and tested in multiple subjects. Main results. The method is shown to outperform other approaches in the field with respect to accuracy and robustness. The algorithm is validated on various categories of speech, such as imagined pronunciation of vowels, short words and long words. The classification accuracy of our methodology is in all cases significantly above chance level, reaching a maximum of 70% for cases where we classify three words and 95% for cases of two words. Significance. The results reveal certain aspects that may affect the success of speech imagery classification from EEG signals, such as sound, meaning and word complexity. This can potentially extend the capability of utilizing speech imagery in future BCI applications. The dataset of speech imagery collected from total 15 subjects is also published.

Fault diagnosis of helical gearbox using acoustic signal and wavelets

NASA Astrophysics Data System (ADS)

Pranesh, SK; Abraham, Siju; Sugumaran, V.; Amarnath, M.

2017-05-01

The efficient transmission of power in machines is needed and gears are an appropriate choice. Faults in gears result in loss of energy and money. The monitoring and fault diagnosis are done by analysis of the acoustic and vibrational signals which are generally considered to be unwanted by products. This study proposes the usage of machine learning algorithm for condition monitoring of a helical gearbox by using the sound signals produced by the gearbox. Artificial faults were created and subsequently signals were captured by a microphone. An extensive study using different wavelet transformations for feature extraction from the acoustic signals was done, followed by waveletselection and feature selection using J48 decision tree and feature classification was performed using K star algorithm. Classification accuracy of 100% was obtained in the study

System and method for investigating sub-surface features of a rock formation with acoustic sources generating coded signals

DOEpatents

Vu, Cung Khac; Nihei, Kurt; Johnson, Paul A; Guyer, Robert; Ten Cate, James A; Le Bas, Pierre-Yves; Larmat, Carene S

2014-12-30

A system and a method for investigating rock formations includes generating, by a first acoustic source, a first acoustic signal comprising a first plurality of pulses, each pulse including a first modulated signal at a central frequency; and generating, by a second acoustic source, a second acoustic signal comprising a second plurality of pulses. A receiver arranged within the borehole receives a detected signal including a signal being generated by a non-linear mixing process from the first-and-second acoustic signal in a non-linear mixing zone within the intersection volume. The method also includes-processing the received signal to extract the signal generated by the non-linear mixing process over noise or over signals generated by a linear interaction process, or both.

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…

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

Approximated mutual information training for speech recognition using myoelectric signals.

PubMed

Guo, Hua J; Chan, A D C

2006-01-01

A new training algorithm called the approximated maximum mutual information (AMMI) is proposed to improve the accuracy of myoelectric speech recognition using hidden Markov models (HMMs). Previous studies have demonstrated that automatic speech recognition can be performed using myoelectric signals from articulatory muscles of the face. Classification of facial myoelectric signals can be performed using HMMs that are trained using the maximum likelihood (ML) algorithm; however, this algorithm maximizes the likelihood of the observations in the training sequence, which is not directly associated with optimal classification accuracy. The AMMI training algorithm attempts to maximize the mutual information, thereby training the HMMs to optimize their parameters for discrimination. Our results show that AMMI training consistently reduces the error rates compared to these by the ML training, increasing the accuracy by approximately 3% on average.

Two Dimensional Processing Of Speech And Ecg Signals Using The Wigner-Ville Distribution

NASA Astrophysics Data System (ADS)

Boashash, Boualem; Abeysekera, Saman S.

1986-12-01

The Wigner-Ville Distribution (WVD) has been shown to be a valuable tool for the analysis of non-stationary signals such as speech and Electrocardiogram (ECG) data. The one-dimensional real data are first transformed into a complex analytic signal using the Hilbert Transform and then a 2-dimensional image is formed using the Wigner-Ville Transform. For speech signals, a contour plot is determined and used as a basic feature. for a pattern recognition algorithm. This method is compared with the classical Short Time Fourier Transform (STFT) and is shown, to be able to recognize isolated words better in a noisy environment. The same method together with the concept of instantaneous frequency of the signal is applied to the analysis of ECG signals. This technique allows one to classify diseased heart-beat signals. Examples are shown.

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.

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.

Bird population density estimated from acoustic signals

USGS Publications Warehouse

Dawson, D.K.; Efford, M.G.

2009-01-01

Many animal species are detected primarily by sound. Although songs, calls and other sounds are often used for population assessment, as in bird point counts and hydrophone surveys of cetaceans, there are few rigorous methods for estimating population density from acoustic data. 2. The problem has several parts - distinguishing individuals, adjusting for individuals that are missed, and adjusting for the area sampled. Spatially explicit capture-recapture (SECR) is a statistical methodology that addresses jointly the second and third parts of the problem. We have extended SECR to use uncalibrated information from acoustic signals on the distance to each source. 3. We applied this extension of SECR to data from an acoustic survey of ovenbird Seiurus aurocapilla density in an eastern US deciduous forest with multiple four-microphone arrays. We modelled average power from spectrograms of ovenbird songs measured within a window of 0??7 s duration and frequencies between 4200 and 5200 Hz. 4. The resulting estimates of the density of singing males (0??19 ha -1 SE 0??03 ha-1) were consistent with estimates of the adult male population density from mist-netting (0??36 ha-1 SE 0??12 ha-1). The fitted model predicts sound attenuation of 0??11 dB m-1 (SE 0??01 dB m-1) in excess of losses from spherical spreading. 5.Synthesis and applications. Our method for estimating animal population density from acoustic signals fills a gap in the census methods available for visually cryptic but vocal taxa, including many species of bird and cetacean. The necessary equipment is simple and readily available; as few as two microphones may provide adequate estimates, given spatial replication. The method requires that individuals detected at the same place are acoustically distinguishable and all individuals vocalize during the recording interval, or that the per capita rate of vocalization is known. We believe these requirements can be met, with suitable field methods, for a significant

Digital Signal Processing in Acoustics--Part 2.

ERIC Educational Resources Information Center

Davies, H.; McNeill, D. J.

1986-01-01

Reviews the potential of a data acquisition system for illustrating the nature and significance of ideas in digital signal processing. Focuses on the fast Fourier transform and the utility of its two-channel format, emphasizing cross-correlation and its two-microphone technique of acoustic intensity measurement. Includes programing format. (ML)

Cochlear implantation with hearing preservation yields significant benefit for speech recognition in complex listening environments

PubMed Central

Gifford, René H.; Dorman, Michael F.; Skarzynski, Henryk; Lorens, Artur; Polak, Marek; Driscoll, Colin L. W.; Roland, Peter; Buchman, Craig A.

2012-01-01

Objective The aim of this study was to assess the benefit of having preserved acoustic hearing in the implanted ear for speech recognition in complex listening environments. Design The current study included a within subjects, repeated-measures design including 21 English speaking and 17 Polish speaking cochlear implant recipients with preserved acoustic hearing in the implanted ear. The patients were implanted with electrodes that varied in insertion depth from 10 to 31 mm. Mean preoperative low-frequency thresholds (average of 125, 250 and 500 Hz) in the implanted ear were 39.3 and 23.4 dB HL for the English- and Polish-speaking participants, respectively. In one condition, speech perception was assessed in an 8-loudspeaker environment in which the speech signals were presented from one loudspeaker and restaurant noise was presented from all loudspeakers. In another condition, the signals were presented in a simulation of a reverberant environment with a reverberation time of 0.6 sec. The response measures included speech reception thresholds (SRTs) and percent correct sentence understanding for two test conditions: cochlear implant (CI) plus low-frequency hearing in the contralateral ear (bimodal condition) and CI plus low-frequency hearing in both ears (best aided condition). A subset of 6 English-speaking listeners were also assessed on measures of interaural time difference (ITD) thresholds for a 250-Hz signal. Results Small, but significant, improvements in performance (1.7 – 2.1 dB and 6 – 10 percentage points) were found for the best-aided condition vs. the bimodal condition. Postoperative thresholds in the implanted ear were correlated with the degree of EAS benefit for speech recognition in diffuse noise. There was no reliable relationship among measures of audiometric threshold in the implanted ear nor elevation in threshold following surgery and improvement in speech understanding in reverberation. There was a significant correlation between ITD

Recognizing speech under a processing load: dissociating energetic from informational factors.

PubMed

Mattys, Sven L; Brooks, Joanna; Cooke, Martin

2009-11-01

Effects of perceptual and cognitive loads on spoken-word recognition have so far largely escaped investigation. This study lays the foundations of a psycholinguistic approach to speech recognition in adverse conditions that draws upon the distinction between energetic masking, i.e., listening environments leading to signal degradation, and informational masking, i.e., listening environments leading to depletion of higher-order, domain-general processing resources, independent of signal degradation. We show that severe energetic masking, such as that produced by background speech or noise, curtails reliance on lexical-semantic knowledge and increases relative reliance on salient acoustic detail. In contrast, informational masking, induced by a resource-depleting competing task (divided attention or a memory load), results in the opposite pattern. Based on this clear dissociation, we propose a model of speech recognition that addresses not only the mapping between sensory input and lexical representations, as traditionally advocated, but also the way in which this mapping interfaces with general cognition and non-linguistic processes.

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

The effects of Thalamic Deep Brain Stimulation on speech dynamics in patients with Essential Tremor: An articulographic study.

PubMed

Mücke, Doris; Hermes, Anne; Roettger, Timo B; Becker, Johannes; Niemann, Henrik; Dembek, Till A; Timmermann, Lars; Visser-Vandewalle, Veerle; Fink, Gereon R; Grice, Martine; Barbe, Michael T

2018-01-01

Acoustic studies have revealed that patients with Essential Tremor treated with thalamic Deep Brain Stimulation (DBS) may suffer from speech deterioration in terms of imprecise oral articulation and reduced voicing control. Based on the acoustic signal one cannot infer, however, whether this deterioration is due to a general slowing down of the speech motor system (e.g., a target undershoot of a desired articulatory goal resulting from being too slow) or disturbed coordination (e.g., a target undershoot caused by problems with the relative phasing of articulatory movements). To elucidate this issue further, we here investigated both acoustics and articulatory patterns of the labial and lingual system using Electromagnetic Articulography (EMA) in twelve Essential Tremor patients treated with thalamic DBS and twelve age- and sex-matched controls. By comparing patients with activated (DBS-ON) and inactivated stimulation (DBS-OFF) with control speakers, we show that critical changes in speech dynamics occur on two levels: With inactivated stimulation (DBS-OFF), patients showed coordination problems of the labial and lingual system in terms of articulatory imprecision and slowness. These effects of articulatory discoordination worsened under activated stimulation, accompanied by an additional overall slowing down of the speech motor system. This leads to a poor performance of syllables on the acoustic surface, reflecting an aggravation either of pre-existing cerebellar deficits and/or the affection of the upper motor fibers of the internal capsule.

Perceptual evaluation and acoustic analysis of pneumatic artificial larynx.

PubMed

Xu, Jie Jie; Chen, Xi; Lu, Mei Ping; Qiao, Ming Zhe

2009-12-01

To investigate the perceptual and acoustic characteristics of the pneumatic artificial larynx (PAL) and evaluate its speech ability and clinical value. Prospective study. The study was conducted in the Voice Lab, Department of Otorhinolaryngology, The First Affiliated Hospital of Nanjing Medical University. Forty-six laryngectomy patients using the PAL were rated for intelligibility and fluency of speech. The voice signals of sustained vowel /a/ for 40 healthy controls and 42 successful patients using the PAL were measured by a computer system. The acoustic parameters and sound spectrographs were analyzed and compared between the two groups. Forty-two of 46 patients using the PAL (91.3%) acquired successful speech capability. The intelligibility scores of 42 successful PAL speakers ranged from 71 to 95 percent, and the intelligibility range of four unsuccessful speakers was 30 to 50 percent. The fluency was judged as good or excellent in 42 successful patients, and poor or fair in four unsuccessful patients. There was no significant difference in average fundamental frequency, maximum intensity, jitter, shimmer, and normalized noise energy (NNE) between 42 successful PAL speakers and 40 healthy controls, while the maximum phonation time (MPT) of PAL speakers was slightly lower than that of the controls. The sound spectrographs of the patients using the PAL approximated those of the healthy controls. The PAL has the advantage of a high percentage of successful vocal rehabilitation. PAL speech is fluent and intelligible. The acoustic characteristics of the PAL are similar to those of a normal voice.

Can you hear me yet? An intracranial investigation of speech and non-speech audiovisual interactions in human cortex.

PubMed

Rhone, Ariane E; Nourski, Kirill V; Oya, Hiroyuki; Kawasaki, Hiroto; Howard, Matthew A; McMurray, Bob

In everyday conversation, viewing a talker's face can provide information about the timing and content of an upcoming speech signal, resulting in improved intelligibility. Using electrocorticography, we tested whether human auditory cortex in Heschl's gyrus (HG) and on superior temporal gyrus (STG) and motor cortex on precentral gyrus (PreC) were responsive to visual/gestural information prior to the onset of sound and whether early stages of auditory processing were sensitive to the visual content (speech syllable versus non-speech motion). Event-related band power (ERBP) in the high gamma band was content-specific prior to acoustic onset on STG and PreC, and ERBP in the beta band differed in all three areas. Following sound onset, we found with no evidence for content-specificity in HG, evidence for visual specificity in PreC, and specificity for both modalities in STG. These results support models of audio-visual processing in which sensory information is integrated in non-primary cortical areas.

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.

Call recognition and individual identification of fish vocalizations based on automatic speech recognition: An example with the Lusitanian toadfish.

PubMed

Vieira, Manuel; Fonseca, Paulo J; Amorim, M Clara P; Teixeira, Carlos J C

2015-12-01

The study of acoustic communication in animals often requires not only the recognition of species specific acoustic signals but also the identification of individual subjects, all in a complex acoustic background. Moreover, when very long recordings are to be analyzed, automatic recognition and identification processes are invaluable tools to extract the relevant biological information. A pattern recognition methodology based on hidden Markov models is presented inspired by successful results obtained in the most widely known and complex acoustical communication signal: human speech. This methodology was applied here for the first time to the detection and recognition of fish acoustic signals, specifically in a stream of round-the-clock recordings of Lusitanian toadfish (Halobatrachus didactylus) in their natural estuarine habitat. The results show that this methodology is able not only to detect the mating sounds (boatwhistles) but also to identify individual male toadfish, reaching an identification rate of ca. 95%. Moreover this method also proved to be a powerful tool to assess signal durations in large data sets. However, the system failed in recognizing other sound types.

High visual resolution matters in audiovisual speech perception, but only for some.

PubMed

Alsius, Agnès; Wayne, Rachel V; Paré, Martin; Munhall, Kevin G

2016-07-01

The basis for individual differences in the degree to which visual speech input enhances comprehension of acoustically degraded speech is largely unknown. Previous research indicates that fine facial detail is not critical for visual enhancement when auditory information is available; however, these studies did not examine individual differences in ability to make use of fine facial detail in relation to audiovisual speech perception ability. Here, we compare participants based on their ability to benefit from visual speech information in the presence of an auditory signal degraded with noise, modulating the resolution of the visual signal through low-pass spatial frequency filtering and monitoring gaze behavior. Participants who benefited most from the addition of visual information (high visual gain) were more adversely affected by the removal of high spatial frequency information, compared to participants with low visual gain, for materials with both poor and rich contextual cues (i.e., words and sentences, respectively). Differences as a function of gaze behavior between participants with the highest and lowest visual gains were observed only for words, with participants with the highest visual gain fixating longer on the mouth region. Our results indicate that the individual variance in audiovisual speech in noise performance can be accounted for, in part, by better use of fine facial detail information extracted from the visual signal and increased fixation on mouth regions for short stimuli. Thus, for some, audiovisual speech perception may suffer when the visual input (in addition to the auditory signal) is less than perfect.

Acoustic and laryngographic measures of the laryngeal reflexes of linguistic prominence and vocal effort in German1

PubMed Central

Mooshammer, Christine

2010-01-01

This study uses acoustic and physiological measures to compare laryngeal reflexes of global changes in vocal effort to the effects of modulating such aspects of linguistic prominence as sentence accent, induced by focus variation, and word stress. Seven speakers were recorded by using a laryngograph. The laryngographic pulses were preprocessed to normalize time and amplitude. The laryngographic pulse shape was quantified using open and skewness quotients and also by applying a functional version of the principal component analysis. Acoustic measures included the acoustic open quotient and spectral balance in the vowel ∕e∕ during the test syllable. The open quotient and the laryngographic pulse shape indicated a significantly shorter open phase for loud speech than for soft speech. Similar results were found for lexical stress, suggesting that lexical stress and loud speech are produced with a similar voice source mechanism. Stressed syllables were distinguished from unstressed syllables by their open phase and pulse shape, even in the absence of sentence accent. Evidence for laryngeal involvement in signaling focus, independent of fundamental frequency changes, was not as consistent across speakers. Acoustic results on various spectral balance measures were generally much less consistent compared to results from laryngographic data. PMID:20136226

Speaker compensation for local perturbation of fricative acoustic feedback.

PubMed

Casserly, Elizabeth D

2011-04-01

Feedback perturbation studies of speech acoustics have revealed a great deal about how speakers monitor and control their productions of segmental (e.g., formant frequencies) and non-segmental (e.g., pitch) linguistic elements. The majority of previous work, however, overlooks the role of acoustic feedback in consonant production and makes use of acoustic manipulations that effect either entire utterances or the entire acoustic signal, rather than more temporally and phonetically restricted alterations. This study, therefore, seeks to expand the feedback perturbation literature by examining perturbation of consonant acoustics that is applied in a time-restricted and phonetically specific manner. The spectral center of the alveopalatal fricative [∫] produced in vowel-fricative-vowel nonwords was incrementally raised until it reached the potential for [s]-like frequencies, but the characteristics of high-frequency energy outside the target fricative remained unaltered. An "offline," more widely accessible signal processing method was developed to perform this manipulation. The local feedback perturbation resulted in changes to speakers' fricative production that were more variable, idiosyncratic, and restricted than the compensation seen in more global acoustic manipulations reported in the literature. Implications and interpretations of the results, as well as future directions for research based on the findings, are discussed.

Leak detection in gas pipeline by acoustic and signal processing - A review

NASA Astrophysics Data System (ADS)

Adnan, N. F.; Ghazali, M. F.; Amin, M. M.; Hamat, A. M. A.

2015-12-01

The pipeline system is the most important part in media transport in order to deliver fluid to another station. The weak maintenance and poor safety will contribute to financial losses in term of fluid waste and environmental impacts. There are many classifications of techniques to make it easier to show their specific method and application. This paper's discussion about gas leak detection in pipeline system using acoustic method will be presented in this paper. The wave propagation in the pipeline is a key parameter in acoustic method when the leak occurs and the pressure balance of the pipe will generated by the friction between wall in the pipe. The signal processing is used to decompose the raw signal and show in time- frequency. Findings based on the acoustic method can be used for comparative study in the future. Acoustic signal and HHT is the best method to detect leak in gas pipelines. More experiments and simulation need to be carried out to get the fast result of leaking and estimation of their location.

Perception of temporally modified speech in auditory neuropathy.

PubMed

Hassan, Dalia Mohamed

2011-01-01

Disrupted auditory nerve activity in auditory neuropathy (AN) significantly impairs the sequential processing of auditory information, resulting in poor speech perception. This study investigated the ability of AN subjects to perceive temporally modified consonant-vowel (CV) pairs and shed light on their phonological awareness skills. Four Arabic CV pairs were selected: /ki/-/gi/, /to/-/do/, /si/-/sti/ and /so/-/zo/. The formant transitions in consonants and the pauses between CV pairs were prolonged. Rhyming, segmentation and blending skills were tested using words at a natural rate of speech and with prolongation of the speech stream. Fourteen adult AN subjects were compared to a matched group of cochlear-impaired patients in their perception of acoustically processed speech. The AN group distinguished the CV pairs at a low speech rate, in particular with modification of the consonant duration. Phonological awareness skills deteriorated in adult AN subjects but improved with prolongation of the speech inter-syllabic time interval. A rehabilitation program for AN should consider temporal modification of speech, training for auditory temporal processing and the use of devices with innovative signal processing schemes. Verbal modifications as well as visual imaging appear to be promising compensatory strategies for remediating the affected phonological processing skills.

Sparse gammatone signal model optimized for English speech does not match the human auditory filters.

PubMed

Strahl, Stefan; Mertins, Alfred

2008-07-18

Evidence that neurosensory systems use sparse signal representations as well as improved performance of signal processing algorithms using sparse signal models raised interest in sparse signal coding in the last years. For natural audio signals like speech and environmental sounds, gammatone atoms have been derived as expansion functions that generate a nearly optimal sparse signal model (Smith, E., Lewicki, M., 2006. Efficient auditory coding. Nature 439, 978-982). Furthermore, gammatone functions are established models for the human auditory filters. Thus far, a practical application of a sparse gammatone signal model has been prevented by the fact that deriving the sparsest representation is, in general, computationally intractable. In this paper, we applied an accelerated version of the matching pursuit algorithm for gammatone dictionaries allowing real-time and large data set applications. We show that a sparse signal model in general has advantages in audio coding and that a sparse gammatone signal model encodes speech more efficiently in terms of sparseness than a sparse modified discrete cosine transform (MDCT) signal model. We also show that the optimal gammatone parameters derived for English speech do not match the human auditory filters, suggesting for signal processing applications to derive the parameters individually for each applied signal class instead of using psychometrically derived parameters. For brain research, it means that care should be taken with directly transferring findings of optimality for technical to biological systems.

Real-time, in situ monitoring of nanoporation using electric field-induced acoustic signal

NASA Astrophysics Data System (ADS)

Zarafshani, Ali; Faiz, Rowzat; Samant, Pratik; Zheng, Bin; Xiang, Liangzhong

2018-02-01

The use of nanoporation in reversible or irreversible electroporation, e.g. cancer ablation, is rapidly growing. This technique uses an ultra-short and intense electric pulse to increase the membrane permeability, allowing non-permeant drugs and genes access to the cytosol via nanopores in the plasma membrane. It is vital to create a real-time in situ monitoring technique to characterize this process and answer the need created by the successful electroporation procedure of cancer treatment. All suggested monitoring techniques for electroporation currently are for pre-and post-stimulation exposure with no real-time monitoring during electric field exposure. This study was aimed at developing an innovative technology for real-time in situ monitoring of electroporation based on the typical cell exposure-induced acoustic emissions. The acoustic signals are the result of the electric field, which itself can be used in realtime to characterize the process of electroporation. We varied electric field distribution by varying the electric pulse from 1μ - 100ns and varying the voltage intensity from 0 - 1.2ܸ݇ to energize two electrodes in a bi-polar set-up. An ultrasound transducer was used for collecting acoustic signals around the subject under test. We determined the relative location of the acoustic signals by varying the position of the electrodes relative to the transducer and varying the electric field distribution between the electrodes to capture a variety of acoustic signals. Therefore, the electric field that is utilized in the nanoporation technique also produces a series of corresponding acoustic signals. This offers a novel imaging technique for the real-time in situ monitoring of electroporation that may directly improve treatment efficiency.