Incorporating Auditory Models in Speech/Audio Applications

NASA Astrophysics Data System (ADS)

Krishnamoorthi, Harish

2011-12-01

Following the success in incorporating perceptual models in audio coding algorithms, their application in other speech/audio processing systems is expanding. In general, all perceptual speech/audio processing algorithms involve minimization of an objective function that directly/indirectly incorporates properties of human perception. This dissertation primarily investigates the problems associated with directly embedding an auditory model in the objective function formulation and proposes possible solutions to overcome high complexity issues for use in real-time speech/audio algorithms. Specific problems addressed in this dissertation include: 1) the development of approximate but computationally efficient auditory model implementations that are consistent with the principles of psychoacoustics, 2) the development of a mapping scheme that allows synthesizing a time/frequency domain representation from its equivalent auditory model output. The first problem is aimed at addressing the high computational complexity involved in solving perceptual objective functions that require repeated application of auditory model for evaluation of different candidate solutions. In this dissertation, a frequency pruning and a detector pruning algorithm is developed that efficiently implements the various auditory model stages. The performance of the pruned model is compared to that of the original auditory model for different types of test signals in the SQAM database. Experimental results indicate only a 4-7% relative error in loudness while attaining up to 80-90 % reduction in computational complexity. Similarly, a hybrid algorithm is developed specifically for use with sinusoidal signals and employs the proposed auditory pattern combining technique together with a look-up table to store representative auditory patterns. The second problem obtains an estimate of the auditory representation that minimizes a perceptual objective function and transforms the auditory pattern back to

Audio-visual speech processing in age-related hearing loss: Stronger integration and increased frontal lobe recruitment.

PubMed

Rosemann, Stephanie; Thiel, Christiane M

2018-07-15

Hearing loss is associated with difficulties in understanding speech, especially under adverse listening conditions. In these situations, seeing the speaker improves speech intelligibility in hearing-impaired participants. On the neuronal level, previous research has shown cross-modal plastic reorganization in the auditory cortex following hearing loss leading to altered processing of auditory, visual and audio-visual information. However, how reduced auditory input effects audio-visual speech perception in hearing-impaired subjects is largely unknown. We here investigated the impact of mild to moderate age-related hearing loss on processing audio-visual speech using functional magnetic resonance imaging. Normal-hearing and hearing-impaired participants performed two audio-visual speech integration tasks: a sentence detection task inside the scanner and the McGurk illusion outside the scanner. Both tasks consisted of congruent and incongruent audio-visual conditions, as well as auditory-only and visual-only conditions. We found a significantly stronger McGurk illusion in the hearing-impaired participants, which indicates stronger audio-visual integration. Neurally, hearing loss was associated with an increased recruitment of frontal brain areas when processing incongruent audio-visual, auditory and also visual speech stimuli, which may reflect the increased effort to perform the task. Hearing loss modulated both the audio-visual integration strength measured with the McGurk illusion and brain activation in frontal areas in the sentence task, showing stronger integration and higher brain activation with increasing hearing loss. Incongruent compared to congruent audio-visual speech revealed an opposite brain activation pattern in left ventral postcentral gyrus in both groups, with higher activation in hearing-impaired participants in the incongruent condition. Our results indicate that already mild to moderate hearing loss impacts audio-visual speech processing

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.

A video, text, and speech-driven realistic 3-d virtual head for human-machine interface.

PubMed

Yu, Jun; Wang, Zeng-Fu

2015-05-01

A multiple inputs-driven realistic facial animation system based on 3-D virtual head for human-machine interface is proposed. The system can be driven independently by video, text, and speech, thus can interact with humans through diverse interfaces. The combination of parameterized model and muscular model is used to obtain a tradeoff between computational efficiency and high realism of 3-D facial animation. The online appearance model is used to track 3-D facial motion from video in the framework of particle filtering, and multiple measurements, i.e., pixel color value of input image and Gabor wavelet coefficient of illumination ratio image, are infused to reduce the influence of lighting and person dependence for the construction of online appearance model. The tri-phone model is used to reduce the computational consumption of visual co-articulation in speech synchronized viseme synthesis without sacrificing any performance. The objective and subjective experiments show that the system is suitable for human-machine interaction.

Mistaking minds and machines: How speech affects dehumanization and anthropomorphism.

PubMed

Schroeder, Juliana; Epley, Nicholas

2016-11-01

Treating a human mind like a machine is an essential component of dehumanization, whereas attributing a humanlike mind to a machine is an essential component of anthropomorphism. Here we tested how a cue closely connected to a person's actual mental experience-a humanlike voice-affects the likelihood of mistaking a person for a machine, or a machine for a person. We predicted that paralinguistic cues in speech are particularly likely to convey the presence of a humanlike mind, such that removing voice from communication (leaving only text) would increase the likelihood of mistaking the text's creator for a machine. Conversely, adding voice to a computer-generated script (resulting in speech) would increase the likelihood of mistaking the text's creator for a human. Four experiments confirmed these hypotheses, demonstrating that people are more likely to infer a human (vs. computer) creator when they hear a voice expressing thoughts than when they read the same thoughts in text. Adding human visual cues to text (i.e., seeing a person perform a script in a subtitled video clip), did not increase the likelihood of inferring a human creator compared with only reading text, suggesting that defining features of personhood may be conveyed more clearly in speech (Experiments 1 and 2). Removing the naturalistic paralinguistic cues that convey humanlike capacity for thinking and feeling, such as varied pace and intonation, eliminates the humanizing effect of speech (Experiment 4). We discuss implications for dehumanizing others through text-based media, and for anthropomorphizing machines through speech-based media. (PsycINFO Database Record (c) 2016 APA, all rights reserved).

Audio-visual speech perception: a developmental ERP investigation

PubMed Central

Knowland, Victoria CP; Mercure, Evelyne; Karmiloff-Smith, Annette; Dick, Fred; Thomas, Michael SC

2014-01-01

Being able to see a talking face confers a considerable advantage for speech perception in adulthood. However, behavioural data currently suggest that children fail to make full use of these available visual speech cues until age 8 or 9. This is particularly surprising given the potential utility of multiple informational cues during language learning. We therefore explored this at the neural level. The event-related potential (ERP) technique has been used to assess the mechanisms of audio-visual speech perception in adults, with visual cues reliably modulating auditory ERP responses to speech. Previous work has shown congruence-dependent shortening of auditory N1/P2 latency and congruence-independent attenuation of amplitude in the presence of auditory and visual speech signals, compared to auditory alone. The aim of this study was to chart the development of these well-established modulatory effects over mid-to-late childhood. Experiment 1 employed an adult sample to validate a child-friendly stimulus set and paradigm by replicating previously observed effects of N1/P2 amplitude and latency modulation by visual speech cues; it also revealed greater attenuation of component amplitude given incongruent audio-visual stimuli, pointing to a new interpretation of the amplitude modulation effect. Experiment 2 used the same paradigm to map cross-sectional developmental change in these ERP responses between 6 and 11 years of age. The effect of amplitude modulation by visual cues emerged over development, while the effect of latency modulation was stable over the child sample. These data suggest that auditory ERP modulation by visual speech represents separable underlying cognitive processes, some of which show earlier maturation than others over the course of development. PMID:24176002

The Study and Implementation of Text-to-Speech System for Agricultural Information

NASA Astrophysics Data System (ADS)

Zheng, Huoguo; Hu, Haiyan; Liu, Shihong; Meng, Hong

The Broadcast and Television coverage has increased to more than 98% in china. Information services by radio have wide coverage, low cost, easy-to-grass-roots farmers to accept etc. characteristics. In order to play the better role of broadcast information service, as well as aim at the problem of lack of information resource in rural, we R & D the text-to-speech system. The system includes two parts, software and hardware device, both of them can translate text into audio file. The software subsystem was implemented basic on third-part middleware, and the hardware subsystem was realized with microelectronics technology. Results indicate that the hardware is better than software. The system has been applied in huailai city hebei province, which has conversed more than 8000 audio files as programming materials for the local radio station.

Infant Perception of Audio-Visual Speech Synchrony

ERIC Educational Resources Information Center

Lewkowicz, David J.

2010-01-01

Three experiments investigated perception of audio-visual (A-V) speech synchrony in 4- to 10-month-old infants. Experiments 1 and 2 used a convergent-operations approach by habituating infants to an audiovisually synchronous syllable (Experiment 1) and then testing for detection of increasing degrees of A-V asynchrony (366, 500, and 666 ms) or by…

Listening to an Audio Drama Activates Two Processing Networks, One for All Sounds, Another Exclusively for Speech

PubMed Central

Boldt, Robert; Malinen, Sanna; Seppä, Mika; Tikka, Pia; Savolainen, Petri; Hari, Riitta; Carlson, Synnöve

2013-01-01

Earlier studies have shown considerable intersubject synchronization of brain activity when subjects watch the same movie or listen to the same story. Here we investigated the across-subjects similarity of brain responses to speech and non-speech sounds in a continuous audio drama designed for blind people. Thirteen healthy adults listened for ∼19 min to the audio drama while their brain activity was measured with 3 T functional magnetic resonance imaging (fMRI). An intersubject-correlation (ISC) map, computed across the whole experiment to assess the stimulus-driven extrinsic brain network, indicated statistically significant ISC in temporal, frontal and parietal cortices, cingulate cortex, and amygdala. Group-level independent component (IC) analysis was used to parcel out the brain signals into functionally coupled networks, and the dependence of the ICs on external stimuli was tested by comparing them with the ISC map. This procedure revealed four extrinsic ICs of which two–covering non-overlapping areas of the auditory cortex–were modulated by both speech and non-speech sounds. The two other extrinsic ICs, one left-hemisphere-lateralized and the other right-hemisphere-lateralized, were speech-related and comprised the superior and middle temporal gyri, temporal poles, and the left angular and inferior orbital gyri. In areas of low ISC four ICs that were defined intrinsic fluctuated similarly as the time-courses of either the speech-sound-related or all-sounds-related extrinsic ICs. These ICs included the superior temporal gyrus, the anterior insula, and the frontal, parietal and midline occipital cortices. Taken together, substantial intersubject synchronization of cortical activity was observed in subjects listening to an audio drama, with results suggesting that speech is processed in two separate networks, one dedicated to the processing of speech sounds and the other to both speech and non-speech sounds. PMID:23734202

Listening to an audio drama activates two processing networks, one for all sounds, another exclusively for speech.

PubMed

Boldt, Robert; Malinen, Sanna; Seppä, Mika; Tikka, Pia; Savolainen, Petri; Hari, Riitta; Carlson, Synnöve

2013-01-01

Earlier studies have shown considerable intersubject synchronization of brain activity when subjects watch the same movie or listen to the same story. Here we investigated the across-subjects similarity of brain responses to speech and non-speech sounds in a continuous audio drama designed for blind people. Thirteen healthy adults listened for ∼19 min to the audio drama while their brain activity was measured with 3 T functional magnetic resonance imaging (fMRI). An intersubject-correlation (ISC) map, computed across the whole experiment to assess the stimulus-driven extrinsic brain network, indicated statistically significant ISC in temporal, frontal and parietal cortices, cingulate cortex, and amygdala. Group-level independent component (IC) analysis was used to parcel out the brain signals into functionally coupled networks, and the dependence of the ICs on external stimuli was tested by comparing them with the ISC map. This procedure revealed four extrinsic ICs of which two-covering non-overlapping areas of the auditory cortex-were modulated by both speech and non-speech sounds. The two other extrinsic ICs, one left-hemisphere-lateralized and the other right-hemisphere-lateralized, were speech-related and comprised the superior and middle temporal gyri, temporal poles, and the left angular and inferior orbital gyri. In areas of low ISC four ICs that were defined intrinsic fluctuated similarly as the time-courses of either the speech-sound-related or all-sounds-related extrinsic ICs. These ICs included the superior temporal gyrus, the anterior insula, and the frontal, parietal and midline occipital cortices. Taken together, substantial intersubject synchronization of cortical activity was observed in subjects listening to an audio drama, with results suggesting that speech is processed in two separate networks, one dedicated to the processing of speech sounds and the other to both speech and non-speech sounds.

Transitioning from Analog to Digital Audio Recording in Childhood Speech Sound Disorders

ERIC Educational Resources Information Center

Shriberg, Lawrence D.; Mcsweeny, Jane L.; Anderson, Bruce E.; Campbell, Thomas F.; Chial, Michael R.; Green, Jordan R.; Hauner, Katherina K.; Moore, Christopher A.; Rusiewicz, Heather L.; Wilson, David L.

2005-01-01

Few empirical findings or technical guidelines are available on the current transition from analog to digital audio recording in childhood speech sound disorders. Of particular concern in the present context was whether a transition from analog- to digital-based transcription and coding of prosody and voice features might require re-standardizing…

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.

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

PubMed

Crosse, Michael J; Lalor, Edmund C

2014-04-01

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

The effect of simultaneous text on the recall of noise-degraded speech.

PubMed

Grossman, Irina; Rajan, Ramesh

2017-05-01

Written and spoken language utilize the same processing system, enabling text to modulate speech processing. We investigated how simultaneously presented text affected speech recall in babble noise using a retrospective recall task. Participants were presented with text-speech sentence pairs in multitalker babble noise and then prompted to recall what they heard or what they read. In Experiment 1, sentence pairs were either congruent or incongruent and they were presented in silence or at 1 of 4 noise levels. Audio and Visual control groups were also tested with sentences presented in only 1 modality. Congruent text facilitated accurate recall of degraded speech; incongruent text had no effect. Text and speech were seldom confused for each other. A consideration of the effects of the language background found that monolingual English speakers outperformed early multilinguals at recalling degraded speech; however the effects of text on speech processing were analogous. Experiment 2 considered if the benefit provided by matching text was maintained when the congruency of the text and speech becomes more ambiguous because of the addition of partially mismatching text-speech sentence pairs that differed only on their final keyword and because of the use of low signal-to-noise ratios. The experiment focused on monolingual English speakers; the results showed that even though participants commonly confused text-for-speech during incongruent text-speech pairings, these confusions could not fully account for the benefit provided by matching text. Thus, we uniquely demonstrate that congruent text benefits the recall of noise-degraded speech. (PsycINFO Database Record (c) 2017 APA, all rights reserved).

Audio-visual speech cue combination.

PubMed

Arnold, Derek H; Tear, Morgan; Schindel, Ryan; Roseboom, Warrick

2010-04-16

Different sources of sensory information can interact, often shaping what we think we have seen or heard. This can enhance the precision of perceptual decisions relative to those made on the basis of a single source of information. From a computational perspective, there are multiple reasons why this might happen, and each predicts a different degree of enhanced precision. Relatively slight improvements can arise when perceptual decisions are made on the basis of multiple independent sensory estimates, as opposed to just one. These improvements can arise as a consequence of probability summation. Greater improvements can occur if two initially independent estimates are summated to form a single integrated code, especially if the summation is weighted in accordance with the variance associated with each independent estimate. This form of combination is often described as a Bayesian maximum likelihood estimate. Still greater improvements are possible if the two sources of information are encoded via a common physiological process. Here we show that the provision of simultaneous audio and visual speech cues can result in substantial sensitivity improvements, relative to single sensory modality based decisions. The magnitude of the improvements is greater than can be predicted on the basis of either a Bayesian maximum likelihood estimate or a probability summation. Our data suggest that primary estimates of speech content are determined by a physiological process that takes input from both visual and auditory processing, resulting in greater sensitivity than would be possible if initially independent audio and visual estimates were formed and then subsequently combined.

Machine Translation from Text

NASA Astrophysics Data System (ADS)

Habash, Nizar; Olive, Joseph; Christianson, Caitlin; McCary, John

Machine translation (MT) from text, the topic of this chapter, is perhaps the heart of the GALE project. Beyond being a well defined application that stands on its own, MT from text is the link between the automatic speech recognition component and the distillation component. The focus of MT in GALE is on translating from Arabic or Chinese to English. The three languages represent a wide range of linguistic diversity and make the GALE MT task rather challenging and exciting.

Audio-visual speech intelligibility benefits with bilateral cochlear implants when talker location varies.

PubMed

van Hoesel, Richard J M

2015-04-01

One of the key benefits of using cochlear implants (CIs) in both ears rather than just one is improved localization. It is likely that in complex listening scenes, improved localization allows bilateral CI users to orient toward talkers to improve signal-to-noise ratios and gain access to visual cues, but to date, that conjecture has not been tested. To obtain an objective measure of that benefit, seven bilateral CI users were assessed for both auditory-only and audio-visual speech intelligibility in noise using a novel dynamic spatial audio-visual test paradigm. For each trial conducted in spatially distributed noise, first, an auditory-only cueing phrase that was spoken by one of four talkers was selected and presented from one of four locations. Shortly afterward, a target sentence was presented that was either audio-visual or, in another test configuration, audio-only and was spoken by the same talker and from the same location as the cueing phrase. During the target presentation, visual distractors were added at other spatial locations. Results showed that in terms of speech reception thresholds (SRTs), the average improvement for bilateral listening over the better performing ear alone was 9 dB for the audio-visual mode, and 3 dB for audition-alone. Comparison of bilateral performance for audio-visual and audition-alone showed that inclusion of visual cues led to an average SRT improvement of 5 dB. For unilateral device use, no such benefit arose, presumably due to the greatly reduced ability to localize the target talker to acquire visual information. The bilateral CI speech intelligibility advantage over the better ear in the present study is much larger than that previously reported for static talker locations and indicates greater everyday speech benefits and improved cost-benefit than estimated to date.

Effects of aging on audio-visual speech integration.

PubMed

Huyse, Aurélie; Leybaert, Jacqueline; Berthommier, Frédéric

2014-10-01

This study investigated the impact of aging on audio-visual speech integration. A syllable identification task was presented in auditory-only, visual-only, and audio-visual congruent and incongruent conditions. Visual cues were either degraded or unmodified. Stimuli were embedded in stationary noise alternating with modulated noise. Fifteen young adults and 15 older adults participated in this study. Results showed that older adults had preserved lipreading abilities when the visual input was clear but not when it was degraded. The impact of aging on audio-visual integration also depended on the quality of the visual cues. In the visual clear condition, the audio-visual gain was similar in both groups and analyses in the framework of the fuzzy-logical model of perception confirmed that older adults did not differ from younger adults in their audio-visual integration abilities. In the visual reduction condition, the audio-visual gain was reduced in the older group, but only when the noise was stationary, suggesting that older participants could compensate for the loss of lipreading abilities by using the auditory information available in the valleys of the noise. The fuzzy-logical model of perception confirmed the significant impact of aging on audio-visual integration by showing an increased weight of audition in the older group.

Audio Steganography with Embedded Text

NASA Astrophysics Data System (ADS)

Teck Jian, Chua; Chai Wen, Chuah; Rahman, Nurul Hidayah Binti Ab.; Hamid, Isredza Rahmi Binti A.

2017-08-01

Audio steganography is about hiding the secret message into the audio. It is a technique uses to secure the transmission of secret information or hide their existence. It also may provide confidentiality to secret message if the message is encrypted. To date most of the steganography software such as Mp3Stego and DeepSound use block cipher such as Advanced Encryption Standard or Data Encryption Standard to encrypt the secret message. It is a good practice for security. However, the encrypted message may become too long to embed in audio and cause distortion of cover audio if the secret message is too long. Hence, there is a need to encrypt the message with stream cipher before embedding the message into the audio. This is because stream cipher provides bit by bit encryption meanwhile block cipher provide a fixed length of bits encryption which result a longer output compare to stream cipher. Hence, an audio steganography with embedding text with Rivest Cipher 4 encryption cipher is design, develop and test in this project.

Investigating Perceptual Biases, Data Reliability, and Data Discovery in a Methodology for Collecting Speech Errors From Audio Recordings.

PubMed

Alderete, John; Davies, Monica

2018-04-01

This work describes a methodology of collecting speech errors from audio recordings and investigates how some of its assumptions affect data quality and composition. Speech errors of all types (sound, lexical, syntactic, etc.) were collected by eight data collectors from audio recordings of unscripted English speech. Analysis of these errors showed that: (i) different listeners find different errors in the same audio recordings, but (ii) the frequencies of error patterns are similar across listeners; (iii) errors collected "online" using on the spot observational techniques are more likely to be affected by perceptual biases than "offline" errors collected from audio recordings; and (iv) datasets built from audio recordings can be explored and extended in a number of ways that traditional corpus studies cannot be.

Relating dynamic brain states to dynamic machine states: Human and machine solutions to the speech recognition problem

PubMed Central

Liu, Xunying; Zhang, Chao; Woodland, Phil; Fonteneau, Elisabeth

2017-01-01

There is widespread interest in the relationship between the neurobiological systems supporting human cognition and emerging computational systems capable of emulating these capacities. Human speech comprehension, poorly understood as a neurobiological process, is an important case in point. Automatic Speech Recognition (ASR) systems with near-human levels of performance are now available, which provide a computationally explicit solution for the recognition of words in continuous speech. This research aims to bridge the gap between speech recognition processes in humans and machines, using novel multivariate techniques to compare incremental ‘machine states’, generated as the ASR analysis progresses over time, to the incremental ‘brain states’, measured using combined electro- and magneto-encephalography (EMEG), generated as the same inputs are heard by human listeners. This direct comparison of dynamic human and machine internal states, as they respond to the same incrementally delivered sensory input, revealed a significant correspondence between neural response patterns in human superior temporal cortex and the structural properties of ASR-derived phonetic models. Spatially coherent patches in human temporal cortex responded selectively to individual phonetic features defined on the basis of machine-extracted regularities in the speech to lexicon mapping process. These results demonstrate the feasibility of relating human and ASR solutions to the problem of speech recognition, and suggest the potential for further studies relating complex neural computations in human speech comprehension to the rapidly evolving ASR systems that address the same problem domain. PMID:28945744

Semantic Indexing of Multimedia Content Using Visual, Audio, and Text Cues

NASA Astrophysics Data System (ADS)

Adams, W. H.; Iyengar, Giridharan; Lin, Ching-Yung; Naphade, Milind Ramesh; Neti, Chalapathy; Nock, Harriet J.; Smith, John R.

2003-12-01

We present a learning-based approach to the semantic indexing of multimedia content using cues derived from audio, visual, and text features. We approach the problem by developing a set of statistical models for a predefined lexicon. Novel concepts are then mapped in terms of the concepts in the lexicon. To achieve robust detection of concepts, we exploit features from multiple modalities, namely, audio, video, and text. Concept representations are modeled using Gaussian mixture models (GMM), hidden Markov models (HMM), and support vector machines (SVM). Models such as Bayesian networks and SVMs are used in a late-fusion approach to model concepts that are not explicitly modeled in terms of features. Our experiments indicate promise in the proposed classification and fusion methodologies: our proposed fusion scheme achieves more than 10% relative improvement over the best unimodal concept detector.

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.

Off the ear with no loss in speech understanding: comparing the RONDO and the OPUS 2 cochlear implant audio processors.

PubMed

Dazert, Stefan; Thomas, Jan Peter; Büchner, Andreas; Müller, Joachim; Hempel, John Martin; Löwenheim, Hubert; Mlynski, Robert

2017-03-01

The RONDO is a single-unit cochlear implant audio processor, which omits the need for a behind-the-ear (BTE) audio processor. The primary aim was to compare speech perception results in quiet and in noise with the RONDO and the OPUS 2, a BTE audio processor. Secondary aims were to determine subjects' self-assessed levels of sound quality and gather subjective feedback on RONDO use. All speech perception tests were performed with the RONDO and the OPUS 2 behind-the-ear audio processor at 3 test intervals. Subjects were required to use the RONDO between test intervals. Subjects were tested at upgrade from the OPUS 2 to the RONDO and at 1 and 6 months after upgrade. Speech perception was determined using the Freiburg Monosyllables in quiet test and the Oldenburg Sentence Test (OLSA) in noise. Subjective perception was determined using the Hearing Implant Sound Quality Index (HISQUI 19 ), and a RONDO device-specific questionnaire. 50 subjects participated in the study. Neither speech perception scores nor self-perceived sound quality scores were significantly different at any interval between the RONDO and the OPUS 2. Subjects reported high levels of satisfaction with the RONDO. The RONDO provides comparable speech perception to the OPUS 2 while providing users with high levels of satisfaction and comfort without increasing health risk. The RONDO is a suitable and safe alternative to traditional BTE audio processors.

Choosing and Using Text-to-Speech Software

ERIC Educational Resources Information Center

Peters, Tom; Bell, Lori

2007-01-01

This article describes a computer-based technology for generating speech called text-to-speech (TTS). This software is ready for widespread use by libraries, other organizations, and individual users. It offers the affordable ability to turn just about any electronic text that is not image-based into an artificially spoken communication. The…

Speech entrainment enables patients with Broca’s aphasia to produce fluent speech

PubMed Central

Hubbard, H. Isabel; Hudspeth, Sarah Grace; Holland, Audrey L.; Bonilha, Leonardo; Fromm, Davida; Rorden, Chris

2012-01-01

A distinguishing feature of Broca’s aphasia is non-fluent halting speech typically involving one to three words per utterance. Yet, despite such profound impairments, some patients can mimic audio-visual speech stimuli enabling them to produce fluent speech in real time. We call this effect ‘speech entrainment’ and reveal its neural mechanism as well as explore its usefulness as a treatment for speech production in Broca’s aphasia. In Experiment 1, 13 patients with Broca’s aphasia were tested in three conditions: (i) speech entrainment with audio-visual feedback where they attempted to mimic a speaker whose mouth was seen on an iPod screen; (ii) speech entrainment with audio-only feedback where patients mimicked heard speech; and (iii) spontaneous speech where patients spoke freely about assigned topics. The patients produced a greater variety of words using audio-visual feedback compared with audio-only feedback and spontaneous speech. No difference was found between audio-only feedback and spontaneous speech. In Experiment 2, 10 of the 13 patients included in Experiment 1 and 20 control subjects underwent functional magnetic resonance imaging to determine the neural mechanism that supports speech entrainment. Group results with patients and controls revealed greater bilateral cortical activation for speech produced during speech entrainment compared with spontaneous speech at the junction of the anterior insula and Brodmann area 47, in Brodmann area 37, and unilaterally in the left middle temporal gyrus and the dorsal portion of Broca’s area. Probabilistic white matter tracts constructed for these regions in the normal subjects revealed a structural network connected via the corpus callosum and ventral fibres through the extreme capsule. Unilateral areas were connected via the arcuate fasciculus. In Experiment 3, all patients included in Experiment 1 participated in a 6-week treatment phase using speech entrainment to improve speech production

Blind speech separation system for humanoid robot with FastICA for audio filtering and separation

NASA Astrophysics Data System (ADS)

Budiharto, Widodo; Santoso Gunawan, Alexander Agung

2016-07-01

Nowadays, there are many developments in building intelligent humanoid robot, mainly in order to handle voice and image. In this research, we propose blind speech separation system using FastICA for audio filtering and separation that can be used in education or entertainment. Our main problem is to separate the multi speech sources and also to filter irrelevant noises. After speech separation step, the results will be integrated with our previous speech and face recognition system which is based on Bioloid GP robot and Raspberry Pi 2 as controller. The experimental results show the accuracy of our blind speech separation system is about 88% in command and query recognition cases.

Audio-visual speech experience with age influences perceived audio-visual asynchrony in speech.

PubMed

Alm, Magnus; Behne, Dawn

2013-10-01

Previous research indicates that perception of audio-visual (AV) synchrony changes in adulthood. Possible explanations for these age differences include a decline in hearing acuity, a decline in cognitive processing speed, and increased experience with AV binding. The current study aims to isolate the effect of AV experience by comparing synchrony judgments from 20 young adults (20 to 30 yrs) and 20 normal-hearing middle-aged adults (50 to 60 yrs), an age range for which a decline of cognitive processing speed is expected to be minimal. When presented with AV stop consonant syllables with asynchronies ranging from 440 ms audio-lead to 440 ms visual-lead, middle-aged adults showed significantly less tolerance for audio-lead than young adults. Middle-aged adults also showed a greater shift in their point of subjective simultaneity than young adults. Natural audio-lead asynchronies are arguably more predictable than natural visual-lead asynchronies, and this predictability may render audio-lead thresholds more prone to experience-related fine-tuning.

Applying Spatial Audio to Human Interfaces: 25 Years of NASA Experience

NASA Technical Reports Server (NTRS)

Begault, Durand R.; Wenzel, Elizabeth M.; Godfrey, Martine; Miller, Joel D.; Anderson, Mark R.

2010-01-01

From the perspective of human factors engineering, the inclusion of spatial audio within a human-machine interface is advantageous from several perspectives. Demonstrated benefits include the ability to monitor multiple streams of speech and non-speech warning tones using a cocktail party advantage, and for aurally-guided visual search. Other potential benefits include the spatial coordination and interaction of multimodal events, and evaluation of new communication technologies and alerting systems using virtual simulation. Many of these technologies were developed at NASA Ames Research Center, beginning in 1985. This paper reviews examples and describes the advantages of spatial sound in NASA-related technologies, including space operations, aeronautics, and search and rescue. The work has involved hardware and software development as well as basic and applied research.

Audio-visual speech perception in adult readers with dyslexia: an fMRI study.

PubMed

Rüsseler, Jascha; Ye, Zheng; Gerth, Ivonne; Szycik, Gregor R; Münte, Thomas F

2018-04-01

Developmental dyslexia is a specific deficit in reading and spelling that often persists into adulthood. In the present study, we used slow event-related fMRI and independent component analysis to identify brain networks involved in perception of audio-visual speech in a group of adult readers with dyslexia (RD) and a group of fluent readers (FR). Participants saw a video of a female speaker saying a disyllabic word. In the congruent condition, audio and video input were identical whereas in the incongruent condition, the two inputs differed. Participants had to respond to occasionally occurring animal names. The independent components analysis (ICA) identified several components that were differently modulated in FR and RD. Two of these components including fusiform gyrus and occipital gyrus showed less activation in RD compared to FR possibly indicating a deficit to extract face information that is needed to integrate auditory and visual information in natural speech perception. A further component centered on the superior temporal sulcus (STS) also exhibited less activation in RD compared to FR. This finding is corroborated in the univariate analysis that shows less activation in STS for RD compared to FR. These findings suggest a general impairment in recruitment of audiovisual processing areas in dyslexia during the perception of natural speech.

Evaluating Text-to-Speech Synthesizers

ERIC Educational Resources Information Center

Cardoso, Walcir; Smith, George; Fuentes, Cesar Garcia

2015-01-01

Text-To-Speech (TTS) synthesizers have piqued the interest of researchers for their potential to enhance the L2 acquisition of writing (Kirstein, 2006), vocabulary and reading (Proctor, Dalton, & Grisham, 2007) and pronunciation (Cardoso, Collins, & White, 2012; Soler-Urzua, 2011). Despite their proven effectiveness, there is a need for…

Steganalysis of recorded speech

NASA Astrophysics Data System (ADS)

Johnson, Micah K.; Lyu, Siwei; Farid, Hany

2005-03-01

Digital audio provides a suitable cover for high-throughput steganography. At 16 bits per sample and sampled at a rate of 44,100 Hz, digital audio has the bit-rate to support large messages. In addition, audio is often transient and unpredictable, facilitating the hiding of messages. Using an approach similar to our universal image steganalysis, we show that hidden messages alter the underlying statistics of audio signals. Our statistical model begins by building a linear basis that captures certain statistical properties of audio signals. A low-dimensional statistical feature vector is extracted from this basis representation and used by a non-linear support vector machine for classification. We show the efficacy of this approach on LSB embedding and Hide4PGP. While no explicit assumptions about the content of the audio are made, our technique has been developed and tested on high-quality recorded speech.

Do gender differences in audio-visual benefit and visual influence in audio-visual speech perception emerge with age?

PubMed Central

Alm, Magnus; Behne, Dawn

2015-01-01

Gender and age have been found to affect adults’ audio-visual (AV) speech perception. However, research on adult aging focuses on adults over 60 years, who have an increasing likelihood for cognitive and sensory decline, which may confound positive effects of age-related AV-experience and its interaction with gender. Observed age and gender differences in AV speech perception may also depend on measurement sensitivity and AV task difficulty. Consequently both AV benefit and visual influence were used to measure visual contribution for gender-balanced groups of young (20–30 years) and middle-aged adults (50–60 years) with task difficulty varied using AV syllables from different talkers in alternative auditory backgrounds. Females had better speech-reading performance than males. Whereas no gender differences in AV benefit or visual influence were observed for young adults, visually influenced responses were significantly greater for middle-aged females than middle-aged males. That speech-reading performance did not influence AV benefit may be explained by visual speech extraction and AV integration constituting independent abilities. Contrastingly, the gender difference in visually influenced responses in middle adulthood may reflect an experience-related shift in females’ general AV perceptual strategy. Although young females’ speech-reading proficiency may not readily contribute to greater visual influence, between young and middle-adulthood recurrent confirmation of the contribution of visual cues induced by speech-reading proficiency may gradually shift females AV perceptual strategy toward more visually dominated responses. PMID:26236274

pyAudioAnalysis: An Open-Source Python Library for Audio Signal Analysis.

PubMed

Giannakopoulos, Theodoros

2015-01-01

Audio information plays a rather important role in the increasing digital content that is available today, resulting in a need for methodologies that automatically analyze such content: audio event recognition for home automations and surveillance systems, speech recognition, music information retrieval, multimodal analysis (e.g. audio-visual analysis of online videos for content-based recommendation), etc. This paper presents pyAudioAnalysis, an open-source Python library that provides a wide range of audio analysis procedures including: feature extraction, classification of audio signals, supervised and unsupervised segmentation and content visualization. pyAudioAnalysis is licensed under the Apache License and is available at GitHub (https://github.com/tyiannak/pyAudioAnalysis/). Here we present the theoretical background behind the wide range of the implemented methodologies, along with evaluation metrics for some of the methods. pyAudioAnalysis has been already used in several audio analysis research applications: smart-home functionalities through audio event detection, speech emotion recognition, depression classification based on audio-visual features, music segmentation, multimodal content-based movie recommendation and health applications (e.g. monitoring eating habits). The feedback provided from all these particular audio applications has led to practical enhancement of the library.

pyAudioAnalysis: An Open-Source Python Library for Audio Signal Analysis

PubMed Central

Giannakopoulos, Theodoros

2015-01-01

Audio information plays a rather important role in the increasing digital content that is available today, resulting in a need for methodologies that automatically analyze such content: audio event recognition for home automations and surveillance systems, speech recognition, music information retrieval, multimodal analysis (e.g. audio-visual analysis of online videos for content-based recommendation), etc. This paper presents pyAudioAnalysis, an open-source Python library that provides a wide range of audio analysis procedures including: feature extraction, classification of audio signals, supervised and unsupervised segmentation and content visualization. pyAudioAnalysis is licensed under the Apache License and is available at GitHub (https://github.com/tyiannak/pyAudioAnalysis/). Here we present the theoretical background behind the wide range of the implemented methodologies, along with evaluation metrics for some of the methods. pyAudioAnalysis has been already used in several audio analysis research applications: smart-home functionalities through audio event detection, speech emotion recognition, depression classification based on audio-visual features, music segmentation, multimodal content-based movie recommendation and health applications (e.g. monitoring eating habits). The feedback provided from all these particular audio applications has led to practical enhancement of the library. PMID:26656189

Texting while driving: is speech-based text entry less risky than handheld text entry?

PubMed

He, J; Chaparro, A; Nguyen, B; Burge, R J; Crandall, J; Chaparro, B; Ni, R; Cao, S

2014-11-01

Research indicates that using a cell phone to talk or text while maneuvering a vehicle impairs driving performance. However, few published studies directly compare the distracting effects of texting using a hands-free (i.e., speech-based interface) versus handheld cell phone, which is an important issue for legislation, automotive interface design and driving safety training. This study compared the effect of speech-based versus handheld text entries on simulated driving performance by asking participants to perform a car following task while controlling the duration of a secondary text-entry task. Results showed that both speech-based and handheld text entries impaired driving performance relative to the drive-only condition by causing more variation in speed and lane position. Handheld text entry also increased the brake response time and increased variation in headway distance. Text entry using a speech-based cell phone was less detrimental to driving performance than handheld text entry. Nevertheless, the speech-based text entry task still significantly impaired driving compared to the drive-only condition. These results suggest that speech-based text entry disrupts driving, but reduces the level of performance interference compared to text entry with a handheld device. In addition, the difference in the distraction effect caused by speech-based and handheld text entry is not simply due to the difference in task duration. Copyright © 2014 Elsevier Ltd. All rights reserved.

Machine Learning Through Signature Trees. Applications to Human Speech.

ERIC Educational Resources Information Center

White, George M.

A signature tree is a binary decision tree used to classify unknown patterns. An attempt was made to develop a computer program for manipulating signature trees as a general research tool for exploring machine learning and pattern recognition. The program was applied to the problem of speech recognition to test its effectiveness for a specific…

Communicating headings and preview sentences in text and speech.

PubMed

Lorch, Robert F; Chen, Hung-Tao; Lemarié, Julie

2012-09-01

Two experiments tested the effects of preview sentences and headings on the quality of college students' outlines of informational texts. Experiment 1 found that performance was much better in the preview sentences condition than in a no-signals condition for both printed text and text-to-speech (TTS) audio rendering of the printed text. In contrast, performance in the headings condition was good for the printed text but poor for the auditory presentation because the TTS software failed to communicate nonverbal information carried by the visual headings. Experiment 2 compared outlining performance for five headings conditions during TTS presentation. Using a theoretical framework, "signaling available, relevant, accessible" (SARA) information, to provide an analysis of the information content of headings in the printed text, the manipulation of the headings systematically restored information that was omitted by the TTS application in Experiment 1. The result was that outlining performance improved to levels similar to the visual headings condition of Experiment 1. It is argued that SARA is a useful framework for guiding future development of TTS software for a wide variety of text signaling devices, not just headings.

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.

Involvement of Right STS in Audio-Visual Integration for Affective Speech Demonstrated Using MEG

PubMed Central

Hagan, Cindy C.; Woods, Will; Johnson, Sam; Green, Gary G. R.; Young, Andrew W.

2013-01-01

Speech and emotion perception are dynamic processes in which it may be optimal to integrate synchronous signals emitted from different sources. Studies of audio-visual (AV) perception of neutrally expressed speech demonstrate supra-additive (i.e., where AV>[unimodal auditory+unimodal visual]) responses in left STS to crossmodal speech stimuli. However, emotions are often conveyed simultaneously with speech; through the voice in the form of speech prosody and through the face in the form of facial expression. Previous studies of AV nonverbal emotion integration showed a role for right (rather than left) STS. The current study therefore examined whether the integration of facial and prosodic signals of emotional speech is associated with supra-additive responses in left (cf. results for speech integration) or right (due to emotional content) STS. As emotional displays are sometimes difficult to interpret, we also examined whether supra-additive responses were affected by emotional incongruence (i.e., ambiguity). Using magnetoencephalography, we continuously recorded eighteen participants as they viewed and heard AV congruent emotional and AV incongruent emotional speech stimuli. Significant supra-additive responses were observed in right STS within the first 250 ms for emotionally incongruent and emotionally congruent AV speech stimuli, which further underscores the role of right STS in processing crossmodal emotive signals. PMID:23950977

Involvement of right STS in audio-visual integration for affective speech demonstrated using MEG.

PubMed

Hagan, Cindy C; Woods, Will; Johnson, Sam; Green, Gary G R; Young, Andrew W

2013-01-01

Speech and emotion perception are dynamic processes in which it may be optimal to integrate synchronous signals emitted from different sources. Studies of audio-visual (AV) perception of neutrally expressed speech demonstrate supra-additive (i.e., where AV>[unimodal auditory+unimodal visual]) responses in left STS to crossmodal speech stimuli. However, emotions are often conveyed simultaneously with speech; through the voice in the form of speech prosody and through the face in the form of facial expression. Previous studies of AV nonverbal emotion integration showed a role for right (rather than left) STS. The current study therefore examined whether the integration of facial and prosodic signals of emotional speech is associated with supra-additive responses in left (cf. results for speech integration) or right (due to emotional content) STS. As emotional displays are sometimes difficult to interpret, we also examined whether supra-additive responses were affected by emotional incongruence (i.e., ambiguity). Using magnetoencephalography, we continuously recorded eighteen participants as they viewed and heard AV congruent emotional and AV incongruent emotional speech stimuli. Significant supra-additive responses were observed in right STS within the first 250 ms for emotionally incongruent and emotionally congruent AV speech stimuli, which further underscores the role of right STS in processing crossmodal emotive signals.

Automatic Speech Acquisition and Recognition for Spacesuit Audio Systems

NASA Technical Reports Server (NTRS)

Ye, Sherry

2015-01-01

NASA has a widely recognized but unmet need for novel human-machine interface technologies that can facilitate communication during astronaut extravehicular activities (EVAs), when loud noises and strong reverberations inside spacesuits make communication challenging. WeVoice, Inc., has developed a multichannel signal-processing method for speech acquisition in noisy and reverberant environments that enables automatic speech recognition (ASR) technology inside spacesuits. The technology reduces noise by exploiting differences between the statistical nature of signals (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, ASR accuracy can be improved to the level at which crewmembers will find the speech interface useful. System components and features include beam forming/multichannel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, and ASR decoding. Arithmetic complexity models were developed and will help designers of real-time ASR systems select proper tasks when confronted with constraints in computational resources. In Phase I of the project, WeVoice validated the technology. The company further refined the technology in Phase II and developed a prototype for testing and use by suited astronauts.

Anthropomorphic Coding of Speech and Audio: A Model Inversion Approach

NASA Astrophysics Data System (ADS)

Feldbauer, Christian; Kubin, Gernot; Kleijn, W. Bastiaan

2005-12-01

Auditory modeling is a well-established methodology that provides insight into human perception and that facilitates the extraction of signal features that are most relevant to the listener. The aim of this paper is to provide a tutorial on perceptual speech and audio coding using an invertible auditory model. In this approach, the audio signal is converted into an auditory representation using an invertible auditory model. The auditory representation is quantized and coded. Upon decoding, it is then transformed back into the acoustic domain. This transformation converts a complex distortion criterion into a simple one, thus facilitating quantization with low complexity. We briefly review past work on auditory models and describe in more detail the components of our invertible model and its inversion procedure, that is, the method to reconstruct the signal from the output of the auditory model. We summarize attempts to use the auditory representation for low-bit-rate coding. Our approach also allows the exploitation of the inherent redundancy of the human auditory system for the purpose of multiple description (joint source-channel) coding.

Perception of synthetic speech produced automatically by rule: Intelligibility of eight text-to-speech systems.

PubMed

Greene, Beth G; Logan, John S; Pisoni, David B

1986-03-01

We present the results of studies designed to measure the segmental intelligibility of eight text-to-speech systems and a natural speech control, using the Modified Rhyme Test (MRT). Results indicated that the voices tested could be grouped into four categories: natural speech, high-quality synthetic speech, moderate-quality synthetic speech, and low-quality synthetic speech. The overall performance of the best synthesis system, DECtalk-Paul, was equivalent to natural speech only in terms of performance on initial consonants. The findings are discussed in terms of recent work investigating the perception of synthetic speech under more severe conditions. Suggestions for future research on improving the quality of synthetic speech are also considered.

Perception of synthetic speech produced automatically by rule: Intelligibility of eight text-to-speech systems

PubMed Central

GREENE, BETH G.; LOGAN, JOHN S.; PISONI, DAVID B.

2012-01-01

We present the results of studies designed to measure the segmental intelligibility of eight text-to-speech systems and a natural speech control, using the Modified Rhyme Test (MRT). Results indicated that the voices tested could be grouped into four categories: natural speech, high-quality synthetic speech, moderate-quality synthetic speech, and low-quality synthetic speech. The overall performance of the best synthesis system, DECtalk-Paul, was equivalent to natural speech only in terms of performance on initial consonants. The findings are discussed in terms of recent work investigating the perception of synthetic speech under more severe conditions. Suggestions for future research on improving the quality of synthetic speech are also considered. PMID:23225916

Paper-Based Textbooks with Audio Support for Print-Disabled Students.

PubMed

Fujiyoshi, Akio; Ohsawa, Akiko; Takaira, Takuya; Tani, Yoshiaki; Fujiyoshi, Mamoru; Ota, Yuko

2015-01-01

Utilizing invisible 2-dimensional codes and digital audio players with a 2-dimensional code scanner, we developed paper-based textbooks with audio support for students with print disabilities, called "multimodal textbooks." Multimodal textbooks can be read with the combination of the two modes: "reading printed text" and "listening to the speech of the text from a digital audio player with a 2-dimensional code scanner." Since multimodal textbooks look the same as regular textbooks and the price of a digital audio player is reasonable (about 30 euro), we think multimodal textbooks are suitable for students with print disabilities in ordinary classrooms.

Laboratory and in-flight experiments to evaluate 3-D audio display technology

NASA Technical Reports Server (NTRS)

Ericson, Mark; Mckinley, Richard; Kibbe, Marion; Francis, Daniel

1994-01-01

Laboratory and in-flight experiments were conducted to evaluate 3-D audio display technology for cockpit applications. A 3-D audio display generator was developed which digitally encodes naturally occurring direction information onto any audio signal and presents the binaural sound over headphones. The acoustic image is stabilized for head movement by use of an electromagnetic head-tracking device. In the laboratory, a 3-D audio display generator was used to spatially separate competing speech messages to improve the intelligibility of each message. Up to a 25 percent improvement in intelligibility was measured for spatially separated speech at high ambient noise levels (115 dB SPL). During the in-flight experiments, pilots reported that spatial separation of speech communications provided a noticeable improvement in intelligibility. The use of 3-D audio for target acquisition was also investigated. In the laboratory, 3-D audio enabled the acquisition of visual targets in about two seconds average response time at 17 degrees accuracy. During the in-flight experiments, pilots correctly identified ground targets 50, 75, and 100 percent of the time at separation angles of 12, 20, and 35 degrees, respectively. In general, pilot performance in the field with the 3-D audio display generator was as expected, based on data from laboratory experiments.

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.

Audio in Courseware: Design Knowledge Issues.

ERIC Educational Resources Information Center

Aarntzen, Diana

1993-01-01

Considers issues that need to be addressed when incorporating audio in courseware design. Topics discussed include functions of audio in courseware; the relationship between auditive and visual information; learner characteristics in relation to audio; events of instruction; and audio characteristics, including interactivity and speech technology.…

Text as a Supplement to Speech in Young and Older Adults a)

PubMed Central

Krull, Vidya; Humes, Larry E.

2015-01-01

Objective The purpose of this experiment was to quantify the contribution of visual text to auditory speech recognition in background noise. Specifically, we tested the hypothesis that partially accurate visual text from an automatic speech recognizer could be used successfully to supplement speech understanding in difficult listening conditions in older adults, with normal or impaired hearing. Our working hypotheses were based on what is known regarding audiovisual speech perception in the elderly from speechreading literature. We hypothesized that: 1) combining auditory and visual text information will result in improved recognition accuracy compared to auditory or visual text information alone; 2) benefit from supplementing speech with visual text (auditory and visual enhancement) in young adults will be greater than that in older adults; and 3) individual differences in performance on perceptual measures would be associated with cognitive abilities. Design Fifteen young adults with normal hearing, fifteen older adults with normal hearing, and fifteen older adults with hearing loss participated in this study. All participants completed sentence recognition tasks in auditory-only, text-only, and combined auditory-text conditions. The auditory sentence stimuli were spectrally shaped to restore audibility for the older participants with impaired hearing. All participants also completed various cognitive measures, including measures of working memory, processing speed, verbal comprehension, perceptual and cognitive speed, processing efficiency, inhibition, and the ability to form wholes from parts. Group effects were examined for each of the perceptual and cognitive measures. Audiovisual benefit was calculated relative to performance on auditory-only and visual-text only conditions. Finally, the relationship between perceptual measures and other independent measures were examined using principal-component factor analyses, followed by regression analyses. Results

Audio Classification in Speech and Music: A Comparison between a Statistical and a Neural Approach

NASA Astrophysics Data System (ADS)

Bugatti, Alessandro; Flammini, Alessandra; Migliorati, Pierangelo

2002-12-01

We focus the attention on the problem of audio classification in speech and music for multimedia applications. In particular, we present a comparison between two different techniques for speech/music discrimination. The first method is based on Zero crossing rate and Bayesian classification. It is very simple from a computational point of view, and gives good results in case of pure music or speech. The simulation results show that some performance degradation arises when the music segment contains also some speech superimposed on music, or strong rhythmic components. To overcome these problems, we propose a second method, that uses more features, and is based on neural networks (specifically a multi-layer Perceptron). In this case we obtain better performance, at the expense of a limited growth in the computational complexity. In practice, the proposed neural network is simple to be implemented if a suitable polynomial is used as the activation function, and a real-time implementation is possible even if low-cost embedded systems are used.

Use of Computer Speech Technologies To Enhance Learning.

ERIC Educational Resources Information Center

Ferrell, Joe

1999-01-01

Discusses the design of an innovative learning system that uses new technologies for the man-machine interface, incorporating a combination of Automatic Speech Recognition (ASR) and Text To Speech (TTS) synthesis. Highlights include using speech technologies to mimic the attributes of the ideal tutor and design features. (AEF)

Advances in audio source seperation and multisource audio content retrieval

NASA Astrophysics Data System (ADS)

Vincent, Emmanuel

2012-06-01

Audio source separation aims to extract the signals of individual sound sources from a given recording. In this paper, we review three recent advances which improve the robustness of source separation in real-world challenging scenarios and enable its use for multisource content retrieval tasks, such as automatic speech recognition (ASR) or acoustic event detection (AED) in noisy environments. We present a Flexible Audio Source Separation Toolkit (FASST) and discuss its advantages compared to earlier approaches such as independent component analysis (ICA) and sparse component analysis (SCA). We explain how cues as diverse as harmonicity, spectral envelope, temporal fine structure or spatial location can be jointly exploited by this toolkit. We subsequently present the uncertainty decoding (UD) framework for the integration of audio source separation and audio content retrieval. We show how the uncertainty about the separated source signals can be accurately estimated and propagated to the features. Finally, we explain how this uncertainty can be efficiently exploited by a classifier, both at the training and the decoding stage. We illustrate the resulting performance improvements in terms of speech separation quality and speaker recognition accuracy.

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.

Fuzzy Logic-Based Audio Pattern Recognition

NASA Astrophysics Data System (ADS)

Malcangi, M.

2008-11-01

Audio and audio-pattern recognition is becoming one of the most important technologies to automatically control embedded systems. Fuzzy logic may be the most important enabling methodology due to its ability to rapidly and economically model such application. An audio and audio-pattern recognition engine based on fuzzy logic has been developed for use in very low-cost and deeply embedded systems to automate human-to-machine and machine-to-machine interaction. This engine consists of simple digital signal-processing algorithms for feature extraction and normalization, and a set of pattern-recognition rules manually tuned or automatically tuned by a self-learning process.

Comparing Learning Gains: Audio Versus Text-based Instructor Communication in a Blended Online Learning Environment

NASA Astrophysics Data System (ADS)

Shimizu, Dominique

Though blended course audio feedback has been associated with several measures of course satisfaction at the postsecondary and graduate levels compared to text feedback, it may take longer to prepare and positive results are largely unverified in K-12 literature. The purpose of this quantitative study was to investigate the time investment and learning impact of audio communications with 228 secondary students in a blended online learning biology unit at a central Florida public high school. A short, individualized audio message regarding the student's progress was given to each student in the audio group; similar text-based messages were given to each student in the text-based group on the same schedule; a control got no feedback. A pretest and posttest were employed to measure learning gains in the three groups. To compare the learning gains in two types of feedback with each other and to no feedback, a controlled, randomized, experimental design was implemented. In addition, the creation and posting of audio and text feedback communications were timed in order to assess whether audio feedback took longer to produce than text only feedback. While audio feedback communications did take longer to create and post, there was no difference between learning gains as measured by posttest scores when student received audio, text-based, or no feedback. Future studies using a similar randomized, controlled experimental design are recommended to verify these results and test whether the trend holds in a broader range of subjects, over different time frames, and using a variety of assessment types to measure student learning.

Effects of Text, Audio and Learner Control on Text-Sound Association and Cognitive Load of EFL Learners

ERIC Educational Resources Information Center

Enciso Bernal, Ana Maria

2014-01-01

This study investigated the effects of concurrent audio and equivalent onscreen text on the ability of learners of English as a foreign language (EFL) to form associations between textual and aural forms of target vocabulary words. The study also looked at the effects of learner control over an audio sequence on the association of textual and…

Development and testing of an audio forensic software for enhancing speech signals masked by loud music

NASA Astrophysics Data System (ADS)

Dobre, Robert A.; Negrescu, Cristian; Stanomir, Dumitru

2016-12-01

In many situations audio recordings can decide the fate of a trial when accepted as evidence. But until they can be taken into account they must be authenticated at first, but also the quality of the targeted content (speech in most cases) must be good enough to remove any doubt. In this scope two main directions of multimedia forensics come into play: content authentication and noise reduction. This paper presents an application that is included in the latter. If someone would like to conceal their conversation, the easiest way to do it would be to turn loud the nearest audio system. In this situation, if a microphone was placed close by, the recorded signal would be apparently useless because the speech signal would be masked by the loud music signal. The paper proposes an adaptive filters based solution to remove the musical content from a previously described signal mixture in order to recover the masked vocal signal. Two adaptive filtering algorithms were tested in the proposed solution: the Normalised Least Mean Squares (NLMS) and Recursive Least Squares (RLS). Their performances in the described situation were evaluated using Simulink, compared and included in the paper.

Vowel Imagery Decoding toward Silent Speech BCI Using Extreme Learning Machine with Electroencephalogram

PubMed Central

Kim, Jongin; Park, Hyeong-jun

2016-01-01

The purpose of this study is to classify EEG data on imagined speech in a single trial. We recorded EEG data while five subjects imagined different vowels, /a/, /e/, /i/, /o/, and /u/. We divided each single trial dataset into thirty segments and extracted features (mean, variance, standard deviation, and skewness) from all segments. To reduce the dimension of the feature vector, we applied a feature selection algorithm based on the sparse regression model. These features were classified using a support vector machine with a radial basis function kernel, an extreme learning machine, and two variants of an extreme learning machine with different kernels. Because each single trial consisted of thirty segments, our algorithm decided the label of the single trial by selecting the most frequent output among the outputs of the thirty segments. As a result, we observed that the extreme learning machine and its variants achieved better classification rates than the support vector machine with a radial basis function kernel and linear discrimination analysis. Thus, our results suggested that EEG responses to imagined speech could be successfully classified in a single trial using an extreme learning machine with a radial basis function and linear kernel. This study with classification of imagined speech might contribute to the development of silent speech BCI systems. PMID:28097128

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

Audio-visual onset differences are used to determine syllable identity for ambiguous audio-visual stimulus pairs

PubMed Central

ten Oever, Sanne; Sack, Alexander T.; Wheat, Katherine L.; Bien, Nina; van Atteveldt, Nienke

2013-01-01

Content and temporal cues have been shown to interact during audio-visual (AV) speech identification. Typically, the most reliable unimodal cue is used more strongly to identify specific speech features; however, visual cues are only used if the AV stimuli are presented within a certain temporal window of integration (TWI). This suggests that temporal cues denote whether unimodal stimuli belong together, that is, whether they should be integrated. It is not known whether temporal cues also provide information about the identity of a syllable. Since spoken syllables have naturally varying AV onset asynchronies, we hypothesize that for suboptimal AV cues presented within the TWI, information about the natural AV onset differences can aid in speech identification. To test this, we presented low-intensity auditory syllables concurrently with visual speech signals, and varied the stimulus onset asynchronies (SOA) of the AV pair, while participants were instructed to identify the auditory syllables. We revealed that specific speech features (e.g., voicing) were identified by relying primarily on one modality (e.g., auditory). Additionally, we showed a wide window in which visual information influenced auditory perception, that seemed even wider for congruent stimulus pairs. Finally, we found a specific response pattern across the SOA range for syllables that were not reliably identified by the unimodal cues, which we explained as the result of the use of natural onset differences between AV speech signals. This indicates that temporal cues not only provide information about the temporal integration of AV stimuli, but additionally convey information about the identity of AV pairs. These results provide a detailed behavioral basis for further neuro-imaging and stimulation studies to unravel the neurofunctional mechanisms of the audio-visual-temporal interplay within speech perception. PMID:23805110

Audio-visual onset differences are used to determine syllable identity for ambiguous audio-visual stimulus pairs.

PubMed

Ten Oever, Sanne; Sack, Alexander T; Wheat, Katherine L; Bien, Nina; van Atteveldt, Nienke

2013-01-01

Content and temporal cues have been shown to interact during audio-visual (AV) speech identification. Typically, the most reliable unimodal cue is used more strongly to identify specific speech features; however, visual cues are only used if the AV stimuli are presented within a certain temporal window of integration (TWI). This suggests that temporal cues denote whether unimodal stimuli belong together, that is, whether they should be integrated. It is not known whether temporal cues also provide information about the identity of a syllable. Since spoken syllables have naturally varying AV onset asynchronies, we hypothesize that for suboptimal AV cues presented within the TWI, information about the natural AV onset differences can aid in speech identification. To test this, we presented low-intensity auditory syllables concurrently with visual speech signals, and varied the stimulus onset asynchronies (SOA) of the AV pair, while participants were instructed to identify the auditory syllables. We revealed that specific speech features (e.g., voicing) were identified by relying primarily on one modality (e.g., auditory). Additionally, we showed a wide window in which visual information influenced auditory perception, that seemed even wider for congruent stimulus pairs. Finally, we found a specific response pattern across the SOA range for syllables that were not reliably identified by the unimodal cues, which we explained as the result of the use of natural onset differences between AV speech signals. This indicates that temporal cues not only provide information about the temporal integration of AV stimuli, but additionally convey information about the identity of AV pairs. These results provide a detailed behavioral basis for further neuro-imaging and stimulation studies to unravel the neurofunctional mechanisms of the audio-visual-temporal interplay within speech perception.

Audio Visual Integration with Competing Sources in the Framework of Audio Visual Speech Scene Analysis.

PubMed

Ganesh, Attigodu Chandrashekara; Berthommier, Frédéric; Schwartz, Jean-Luc

2016-01-01

We introduce "Audio-Visual Speech Scene Analysis" (AVSSA) as an extension of the two-stage Auditory Scene Analysis model towards audiovisual scenes made of mixtures of speakers. AVSSA assumes that a coherence index between the auditory and the visual input is computed prior to audiovisual fusion, enabling to determine whether the sensory inputs should be bound together. Previous experiments on the modulation of the McGurk effect by audiovisual coherent vs. incoherent contexts presented before the McGurk target have provided experimental evidence supporting AVSSA. Indeed, incoherent contexts appear to decrease the McGurk effect, suggesting that they produce lower audiovisual coherence hence less audiovisual fusion. The present experiments extend the AVSSA paradigm by creating contexts made of competing audiovisual sources and measuring their effect on McGurk targets. The competing audiovisual sources have respectively a high and a low audiovisual coherence (that is, large vs. small audiovisual comodulations in time). The first experiment involves contexts made of two auditory sources and one video source associated to either the first or the second audio source. It appears that the McGurk effect is smaller after the context made of the visual source associated to the auditory source with less audiovisual coherence. In the second experiment with the same stimuli, the participants are asked to attend to either one or the other source. The data show that the modulation of fusion depends on the attentional focus. Altogether, these two experiments shed light on audiovisual binding, the AVSSA process and the role of attention.

Cortical Integration of Audio-Visual Information

PubMed Central

Vander Wyk, Brent C.; Ramsay, Gordon J.; Hudac, Caitlin M.; Jones, Warren; Lin, David; Klin, Ami; Lee, Su Mei; Pelphrey, Kevin A.

2013-01-01

We investigated the neural basis of audio-visual processing in speech and non-speech stimuli. Physically identical auditory stimuli (speech and sinusoidal tones) and visual stimuli (animated circles and ellipses) were used in this fMRI experiment. Relative to unimodal stimuli, each of the multimodal conjunctions showed increased activation in largely non-overlapping areas. The conjunction of Ellipse and Speech, which most resembles naturalistic audiovisual speech, showed higher activation in the right inferior frontal gyrus, fusiform gyri, left posterior superior temporal sulcus, and lateral occipital cortex. The conjunction of Circle and Tone, an arbitrary audio-visual pairing with no speech association, activated middle temporal gyri and lateral occipital cortex. The conjunction of Circle and Speech showed activation in lateral occipital cortex, and the conjunction of Ellipse and Tone did not show increased activation relative to unimodal stimuli. Further analysis revealed that middle temporal regions, although identified as multimodal only in the Circle-Tone condition, were more strongly active to Ellipse-Speech or Circle-Speech, but regions that were identified as multimodal for Ellipse-Speech were always strongest for Ellipse-Speech. Our results suggest that combinations of auditory and visual stimuli may together be processed by different cortical networks, depending on the extent to which speech or non-speech percepts are evoked. PMID:20709442

Text-to-audiovisual speech synthesizer for children with learning disabilities.

PubMed

Mendi, Engin; Bayrak, Coskun

2013-01-01

Learning disabilities affect the ability of children to learn, despite their having normal intelligence. Assistive tools can highly increase functional capabilities of children with learning disorders such as writing, reading, or listening. In this article, we describe a text-to-audiovisual synthesizer that can serve as an assistive tool for such children. The system automatically converts an input text to audiovisual speech, providing synchronization of the head, eye, and lip movements of the three-dimensional face model with appropriate facial expressions and word flow of the text. The proposed system can enhance speech perception and help children having learning deficits to improve their chances of success.

Auditory Support in Linguistically Diverse Classrooms: Factors Related to Bilingual Text-to-Speech Use

ERIC Educational Resources Information Center

Van Laere, E.; Braak, J.

2017-01-01

Text-to-speech technology can act as an important support tool in computer-based learning environments (CBLEs) as it provides auditory input, next to on-screen text. Particularly for students who use a language at home other than the language of instruction (LOI) applied at school, text-to-speech can be useful. The CBLE E-Validiv offers content in…

Highlight summarization in golf videos using audio signals

NASA Astrophysics Data System (ADS)

Kim, Hyoung-Gook; Kim, Jin Young

2008-01-01

In this paper, we present an automatic summarization of highlights in golf videos based on audio information alone without video information. The proposed highlight summarization system is carried out based on semantic audio segmentation and detection on action units from audio signals. Studio speech, field speech, music, and applause are segmented by means of sound classification. Swing is detected by the methods of impulse onset detection. Sounds like swing and applause form a complete action unit, while studio speech and music parts are used to anchor the program structure. With the advantage of highly precise detection of applause, highlights are extracted effectively. Our experimental results obtain high classification precision on 18 golf games. It proves that the proposed system is very effective and computationally efficient to apply the technology to embedded consumer electronic devices.

Audio-Tutorial Instruction in Medicine.

ERIC Educational Resources Information Center

Boyle, Gloria J.; Herrick, Merlyn C.

This progress report concerns an audio-tutorial approach used at the University of Missouri-Columbia School of Medicine. Instructional techniques such as slide-tape presentations, compressed speech audio tapes, computer-assisted instruction (CAI), motion pictures, television, microfiche, and graphic and printed materials have been implemented,…

Audio Frequency Analysis in Mobile Phones

ERIC Educational Resources Information Center

Aguilar, Horacio Munguía

2016-01-01

A new experiment using mobile phones is proposed in which its audio frequency response is analyzed using the audio port for inputting external signal and getting a measurable output. This experiment shows how the limited audio bandwidth used in mobile telephony is the main cause of the poor speech quality in this service. A brief discussion is…

Speech recognition technology: an outlook for human-to-machine interaction.

PubMed

Erdel, T; Crooks, S

2000-01-01

Speech recognition, as an enabling technology in healthcare-systems computing, is a topic that has been discussed for quite some time, but is just now coming to fruition. Traditionally, speech-recognition software has been constrained by hardware, but improved processors and increased memory capacities are starting to remove some of these limitations. With these barriers removed, companies that create software for the healthcare setting have the opportunity to write more successful applications. Among the criticisms of speech-recognition applications are the high rates of error and steep training curves. However, even in the face of such negative perceptions, there remains significant opportunities for speech recognition to allow healthcare providers and, more specifically, physicians, to work more efficiently and ultimately spend more time with their patients and less time completing necessary documentation. This article will identify opportunities for inclusion of speech-recognition technology in the healthcare setting and examine major categories of speech-recognition software--continuous speech recognition, command and control, and text-to-speech. We will discuss the advantages and disadvantages of each area, the limitations of the software today, and how future trends might affect them.

Audio-Visual Speaker Diarization Based on Spatiotemporal Bayesian Fusion.

PubMed

Gebru, Israel D; Ba, Sileye; Li, Xiaofei; Horaud, Radu

2018-05-01

Speaker diarization consists of assigning speech signals to people engaged in a dialogue. An audio-visual spatiotemporal diarization model is proposed. The model is well suited for challenging scenarios that consist of several participants engaged in multi-party interaction while they move around and turn their heads towards the other participants rather than facing the cameras and the microphones. Multiple-person visual tracking is combined with multiple speech-source localization in order to tackle the speech-to-person association problem. The latter is solved within a novel audio-visual fusion method on the following grounds: binaural spectral features are first extracted from a microphone pair, then a supervised audio-visual alignment technique maps these features onto an image, and finally a semi-supervised clustering method assigns binaural spectral features to visible persons. The main advantage of this method over previous work is that it processes in a principled way speech signals uttered simultaneously by multiple persons. The diarization itself is cast into a latent-variable temporal graphical model that infers speaker identities and speech turns, based on the output of an audio-visual association process, executed at each time slice, and on the dynamics of the diarization variable itself. The proposed formulation yields an efficient exact inference procedure. A novel dataset, that contains audio-visual training data as well as a number of scenarios involving several participants engaged in formal and informal dialogue, is introduced. The proposed method is thoroughly tested and benchmarked with respect to several state-of-the art diarization algorithms.

Using speech recognition to enhance the Tongue Drive System functionality in computer access.

PubMed

Huo, Xueliang; Ghovanloo, Maysam

2011-01-01

Tongue Drive System (TDS) is a wireless tongue operated assistive technology (AT), which can enable people with severe physical disabilities to access computers and drive powered wheelchairs using their volitional tongue movements. TDS offers six discrete commands, simultaneously available to the users, for pointing and typing as a substitute for mouse and keyboard in computer access, respectively. To enhance the TDS performance in typing, we have added a microphone, an audio codec, and a wireless audio link to its readily available 3-axial magnetic sensor array, and combined it with a commercially available speech recognition software, the Dragon Naturally Speaking, which is regarded as one of the most efficient ways for text entry. Our preliminary evaluations indicate that the combined TDS and speech recognition technologies can provide end users with significantly higher performance than using each technology alone, particularly in completing tasks that require both pointing and text entry, such as web surfing.

Using Speech Recognition to Enhance the Tongue Drive System Functionality in Computer Access

PubMed Central

Huo, Xueliang; Ghovanloo, Maysam

2013-01-01

Tongue Drive System (TDS) is a wireless tongue operated assistive technology (AT), which can enable people with severe physical disabilities to access computers and drive powered wheelchairs using their volitional tongue movements. TDS offers six discrete commands, simultaneously available to the users, for pointing and typing as a substitute for mouse and keyboard in computer access, respectively. To enhance the TDS performance in typing, we have added a microphone, an audio codec, and a wireless audio link to its readily available 3-axial magnetic sensor array, and combined it with a commercially available speech recognition software, the Dragon Naturally Speaking, which is regarded as one of the most efficient ways for text entry. Our preliminary evaluations indicate that the combined TDS and speech recognition technologies can provide end users with significantly higher performance than using each technology alone, particularly in completing tasks that require both pointing and text entry, such as web surfing. PMID:22255801

Learning diagnostic models using speech and language measures.

PubMed

Peintner, Bart; Jarrold, William; Vergyriy, Dimitra; Richey, Colleen; Tempini, Maria Luisa Gorno; Ogar, Jennifer

2008-01-01

We describe results that show the effectiveness of machine learning in the automatic diagnosis of certain neurodegenerative diseases, several of which alter speech and language production. We analyzed audio from 9 control subjects and 30 patients diagnosed with one of three subtypes of Frontotemporal Lobar Degeneration. From this data, we extracted features of the audio signal and the words the patient used, which were obtained using our automated transcription technologies. We then automatically learned models that predict the diagnosis of the patient using these features. Our results show that learned models over these features predict diagnosis with accuracy significantly better than random. Future studies using higher quality recordings will likely improve these results.

"Look What I Did!": Student Conferences with Text-to-Speech Software

ERIC Educational Resources Information Center

Young, Chase; Stover, Katie

2014-01-01

The authors describe a strategy that empowers students to edit and revise their own writing. Students input their writing in to text-to-speech software that rereads the text aloud. While listening, students make necessary revisions and edits.

A centralized audio presentation manager

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

Papp, A.L. III; Blattner, M.M.

1994-05-16

The centralized audio presentation manager addresses the problems which occur when multiple programs running simultaneously attempt to use the audio output of a computer system. Time dependence of sound means that certain auditory messages must be scheduled simultaneously, which can lead to perceptual problems due to psychoacoustic phenomena. Furthermore, the combination of speech and nonspeech audio is examined; each presents its own problems of perceptibility in an acoustic environment composed of multiple auditory streams. The centralized audio presentation manager receives abstract parameterized message requests from the currently running programs, and attempts to create and present a sonic representation in themore » most perceptible manner through the use of a theoretically and empirically designed rule set.« less

Structuring Broadcast Audio for Information Access

NASA Astrophysics Data System (ADS)

Gauvain, Jean-Luc; Lamel, Lori

2003-12-01

One rapidly expanding application area for state-of-the-art speech recognition technology is the automatic processing of broadcast audiovisual data for information access. Since much of the linguistic information is found in the audio channel, speech recognition is a key enabling technology which, when combined with information retrieval techniques, can be used for searching large audiovisual document collections. Audio indexing must take into account the specificities of audio data such as needing to deal with the continuous data stream and an imperfect word transcription. Other important considerations are dealing with language specificities and facilitating language portability. At Laboratoire d'Informatique pour la Mécanique et les Sciences de l'Ingénieur (LIMSI), broadcast news transcription systems have been developed for seven languages: English, French, German, Mandarin, Portuguese, Spanish, and Arabic. The transcription systems have been integrated into prototype demonstrators for several application areas such as audio data mining, structuring audiovisual archives, selective dissemination of information, and topic tracking for media monitoring. As examples, this paper addresses the spoken document retrieval and topic tracking tasks.

Audio-Visual and Meaningful Semantic Context Enhancements in Older and Younger Adults.

PubMed

Smayda, Kirsten E; Van Engen, Kristin J; Maddox, W Todd; Chandrasekaran, Bharath

2016-01-01

Speech perception is critical to everyday life. Oftentimes noise can degrade a speech signal; however, because of the cues available to the listener, such as visual and semantic cues, noise rarely prevents conversations from continuing. The interaction of visual and semantic cues in aiding speech perception has been studied in young adults, but the extent to which these two cues interact for older adults has not been studied. To investigate the effect of visual and semantic cues on speech perception in older and younger adults, we recruited forty-five young adults (ages 18-35) and thirty-three older adults (ages 60-90) to participate in a speech perception task. Participants were presented with semantically meaningful and anomalous sentences in audio-only and audio-visual conditions. We hypothesized that young adults would outperform older adults across SNRs, modalities, and semantic contexts. In addition, we hypothesized that both young and older adults would receive a greater benefit from a semantically meaningful context in the audio-visual relative to audio-only modality. We predicted that young adults would receive greater visual benefit in semantically meaningful contexts relative to anomalous contexts. However, we predicted that older adults could receive a greater visual benefit in either semantically meaningful or anomalous contexts. Results suggested that in the most supportive context, that is, semantically meaningful sentences presented in the audiovisual modality, older adults performed similarly to young adults. In addition, both groups received the same amount of visual and meaningful benefit. Lastly, across groups, a semantically meaningful context provided more benefit in the audio-visual modality relative to the audio-only modality, and the presence of visual cues provided more benefit in semantically meaningful contexts relative to anomalous contexts. These results suggest that older adults can perceive speech as well as younger adults when both

Audio-Visual and Meaningful Semantic Context Enhancements in Older and Younger Adults

PubMed Central

Smayda, Kirsten E.; Van Engen, Kristin J.; Maddox, W. Todd; Chandrasekaran, Bharath

2016-01-01

Speech perception is critical to everyday life. Oftentimes noise can degrade a speech signal; however, because of the cues available to the listener, such as visual and semantic cues, noise rarely prevents conversations from continuing. The interaction of visual and semantic cues in aiding speech perception has been studied in young adults, but the extent to which these two cues interact for older adults has not been studied. To investigate the effect of visual and semantic cues on speech perception in older and younger adults, we recruited forty-five young adults (ages 18–35) and thirty-three older adults (ages 60–90) to participate in a speech perception task. Participants were presented with semantically meaningful and anomalous sentences in audio-only and audio-visual conditions. We hypothesized that young adults would outperform older adults across SNRs, modalities, and semantic contexts. In addition, we hypothesized that both young and older adults would receive a greater benefit from a semantically meaningful context in the audio-visual relative to audio-only modality. We predicted that young adults would receive greater visual benefit in semantically meaningful contexts relative to anomalous contexts. However, we predicted that older adults could receive a greater visual benefit in either semantically meaningful or anomalous contexts. Results suggested that in the most supportive context, that is, semantically meaningful sentences presented in the audiovisual modality, older adults performed similarly to young adults. In addition, both groups received the same amount of visual and meaningful benefit. Lastly, across groups, a semantically meaningful context provided more benefit in the audio-visual modality relative to the audio-only modality, and the presence of visual cues provided more benefit in semantically meaningful contexts relative to anomalous contexts. These results suggest that older adults can perceive speech as well as younger adults when

Detecting Abnormal Word Utterances in Children With Autism Spectrum Disorders: Machine-Learning-Based Voice Analysis Versus Speech Therapists.

PubMed

Nakai, Yasushi; Takiguchi, Tetsuya; Matsui, Gakuyo; Yamaoka, Noriko; Takada, Satoshi

2017-10-01

Abnormal prosody is often evident in the voice intonations of individuals with autism spectrum disorders. We compared a machine-learning-based voice analysis with human hearing judgments made by 10 speech therapists for classifying children with autism spectrum disorders ( n = 30) and typical development ( n = 51). Using stimuli limited to single-word utterances, machine-learning-based voice analysis was superior to speech therapist judgments. There was a significantly higher true-positive than false-negative rate for machine-learning-based voice analysis but not for speech therapists. Results are discussed in terms of some artificiality of clinician judgments based on single-word utterances, and the objectivity machine-learning-based voice analysis adds to judging abnormal prosody.

Study of an Audio Playback Machine Storage, Distribution, and Repair System. Options for Machine Operation. Study II, Part 1, Phase 2, Final Report.

ERIC Educational Resources Information Center

ManTech Technical Services Corp., Fairfax, VA.

This report presents the results of a management study of audio playback equipment operations conducted by the National Library Service, Library of Congress, its associated network of state and local machine lending agencies (MLA), and other parties that play a role in current operations. The objectives were to document current operations,…

Synchronized and noise-robust audio recordings during realtime magnetic resonance imaging scans.

PubMed

Bresch, Erik; Nielsen, Jon; Nayak, Krishna; Narayanan, Shrikanth

2006-10-01

This letter describes a data acquisition setup for recording, and processing, running speech from a person in a magnetic resonance imaging (MRI) scanner. The main focus is on ensuring synchronicity between image and audio acquisition, and in obtaining good signal to noise ratio to facilitate further speech analysis and modeling. A field-programmable gate array based hardware design for synchronizing the scanner image acquisition to other external data such as audio is described. The audio setup itself features two fiber optical microphones and a noise-canceling filter. Two noise cancellation methods are described including a novel approach using a pulse sequence specific model of the gradient noise of the MRI scanner. The setup is useful for scientific speech production studies. Sample results of speech and singing data acquired and processed using the proposed method are given.

Brain-to-text: decoding spoken phrases from phone representations in the brain.

PubMed

Herff, Christian; Heger, Dominic; de Pesters, Adriana; Telaar, Dominic; Brunner, Peter; Schalk, Gerwin; Schultz, Tanja

2015-01-01

It has long been speculated whether communication between humans and machines based on natural speech related cortical activity is possible. Over the past decade, studies have suggested that it is feasible to recognize isolated aspects of speech from neural signals, such as auditory features, phones or one of a few isolated words. However, until now it remained an unsolved challenge to decode continuously spoken speech from the neural substrate associated with speech and language processing. Here, we show for the first time that continuously spoken speech can be decoded into the expressed words from intracranial electrocorticographic (ECoG) recordings.Specifically, we implemented a system, which we call Brain-To-Text that models single phones, employs techniques from automatic speech recognition (ASR), and thereby transforms brain activity while speaking into the corresponding textual representation. Our results demonstrate that our system can achieve word error rates as low as 25% and phone error rates below 50%. Additionally, our approach contributes to the current understanding of the neural basis of continuous speech production by identifying those cortical regions that hold substantial information about individual phones. In conclusion, the Brain-To-Text system described in this paper represents an important step toward human-machine communication based on imagined speech.

Brain-to-text: decoding spoken phrases from phone representations in the brain

PubMed Central

Herff, Christian; Heger, Dominic; de Pesters, Adriana; Telaar, Dominic; Brunner, Peter; Schalk, Gerwin; Schultz, Tanja

2015-01-01

It has long been speculated whether communication between humans and machines based on natural speech related cortical activity is possible. Over the past decade, studies have suggested that it is feasible to recognize isolated aspects of speech from neural signals, such as auditory features, phones or one of a few isolated words. However, until now it remained an unsolved challenge to decode continuously spoken speech from the neural substrate associated with speech and language processing. Here, we show for the first time that continuously spoken speech can be decoded into the expressed words from intracranial electrocorticographic (ECoG) recordings.Specifically, we implemented a system, which we call Brain-To-Text that models single phones, employs techniques from automatic speech recognition (ASR), and thereby transforms brain activity while speaking into the corresponding textual representation. Our results demonstrate that our system can achieve word error rates as low as 25% and phone error rates below 50%. Additionally, our approach contributes to the current understanding of the neural basis of continuous speech production by identifying those cortical regions that hold substantial information about individual phones. In conclusion, the Brain-To-Text system described in this paper represents an important step toward human-machine communication based on imagined speech. PMID:26124702

(abstract) Synthesis of Speaker Facial Movements to Match Selected Speech Sequences

NASA Technical Reports Server (NTRS)

Scott, Kenneth C.

1994-01-01

We are developing a system for synthesizing image sequences the simulate the facial motion of a speaker. To perform this synthesis, we are pursuing two major areas of effort. We are developing the necessary computer graphics technology to synthesize a realistic image sequence of a person speaking selected speech sequences. Next, we are developing a model that expresses the relation between spoken phonemes and face/mouth shape. A subject is video taped speaking an arbitrary text that contains expression of the full list of desired database phonemes. The subject is video taped from the front speaking normally, recording both audio and video detail simultaneously. Using the audio track, we identify the specific video frames on the tape relating to each spoken phoneme. From this range we digitize the video frame which represents the extreme of mouth motion/shape. Thus, we construct a database of images of face/mouth shape related to spoken phonemes. A selected audio speech sequence is recorded which is the basis for synthesizing a matching video sequence; the speaker need not be the same as used for constructing the database. The audio sequence is analyzed to determine the spoken phoneme sequence and the relative timing of the enunciation of those phonemes. Synthesizing an image sequence corresponding to the spoken phoneme sequence is accomplished using a graphics technique known as morphing. Image sequence keyframes necessary for this processing are based on the spoken phoneme sequence and timing. We have been successful in synthesizing the facial motion of a native English speaker for a small set of arbitrary speech segments. Our future work will focus on advancement of the face shape/phoneme model and independent control of facial features.

Implementing Audio-CASI on Windows’ Platforms

PubMed Central

Cooley, Philip C.; Turner, Charles F.

2011-01-01

Audio computer-assisted self interviewing (Audio-CASI) technologies have recently been shown to provide important and sometimes dramatic improvements in the quality of survey measurements. This is particularly true for measurements requiring respondents to divulge highly sensitive information such as their sexual, drug use, or other sensitive behaviors. However, DOS-based Audio-CASI systems that were designed and adopted in the early 1990s have important limitations. Most salient is the poor control they provide for manipulating the video presentation of survey questions. This article reports our experiences adapting Audio-CASI to Microsoft Windows 3.1 and Windows 95 platforms. Overall, our Windows-based system provided the desired control over video presentation and afforded other advantages including compatibility with a much wider array of audio devices than our DOS-based Audio-CASI technologies. These advantages came at the cost of increased system requirements --including the need for both more RAM and larger hard disks. While these costs will be an issue for organizations converting large inventories of PCS to Windows Audio-CASI today, this will not be a serious constraint for organizations and individuals with small inventories of machines to upgrade or those purchasing new machines today. PMID:22081743

Synchronized and noise-robust audio recordings during realtime magnetic resonance imaging scans (L)

PubMed Central

Bresch, Erik; Nielsen, Jon; Nayak, Krishna; Narayanan, Shrikanth

2007-01-01

This letter describes a data acquisition setup for recording, and processing, running speech from a person in a magnetic resonance imaging (MRI) scanner. The main focus is on ensuring synchronicity between image and audio acquisition, and in obtaining good signal to noise ratio to facilitate further speech analysis and modeling. A field-programmable gate array based hardware design for synchronizing the scanner image acquisition to other external data such as audio is described. The audio setup itself features two fiber optical microphones and a noise-canceling filter. Two noise cancellation methods are described including a novel approach using a pulse sequence specific model of the gradient noise of the MRI scanner. The setup is useful for scientific speech production studies. Sample results of speech and singing data acquired and processed using the proposed method are given. PMID:17069275

Influences of selective adaptation on perception of audiovisual speech

PubMed Central

Dias, James W.; Cook, Theresa C.; Rosenblum, Lawrence D.

2016-01-01

Research suggests that selective adaptation in speech is a low-level process dependent on sensory-specific information shared between the adaptor and test-stimuli. However, previous research has only examined how adaptors shift perception of unimodal test stimuli, either auditory or visual. In the current series of experiments, we investigated whether adaptation to cross-sensory phonetic information can influence perception of integrated audio-visual phonetic information. We examined how selective adaptation to audio and visual adaptors shift perception of speech along an audiovisual test continuum. This test-continuum consisted of nine audio-/ba/-visual-/va/ stimuli, ranging in visual clarity of the mouth. When the mouth was clearly visible, perceivers “heard” the audio-visual stimulus as an integrated “va” percept 93.7% of the time (e.g., McGurk & MacDonald, 1976). As visibility of the mouth became less clear across the nine-item continuum, the audio-visual “va” percept weakened, resulting in a continuum ranging in audio-visual percepts from /va/ to /ba/. Perception of the test-stimuli was tested before and after adaptation. Changes in audiovisual speech perception were observed following adaptation to visual-/va/ and audiovisual-/va/, but not following adaptation to auditory-/va/, auditory-/ba/, or visual-/ba/. Adaptation modulates perception of integrated audio-visual speech by modulating the processing of sensory-specific information. The results suggest that auditory and visual speech information are not completely integrated at the level of selective adaptation. PMID:27041781

Visually Impaired Persons' Comprehension of Text Presented with Speech Synthesis.

ERIC Educational Resources Information Center

Hjelmquist, E.; And Others

1992-01-01

This study of 48 individuals with visual impairments (16 middle-aged with experience in synthetic speech, 16 middle-aged inexperienced, and 16 older inexperienced) found that speech synthesis, compared to natural speech, generally yielded lower results with respect to memory and understanding of texts. Experience had no effect on performance.…

Cue Integration in Categorical Tasks: Insights from Audio-Visual Speech Perception

PubMed Central

Bejjanki, Vikranth Rao; Clayards, Meghan; Knill, David C.; Aslin, Richard N.

2011-01-01

Previous cue integration studies have examined continuous perceptual dimensions (e.g., size) and have shown that human cue integration is well described by a normative model in which cues are weighted in proportion to their sensory reliability, as estimated from single-cue performance. However, this normative model may not be applicable to categorical perceptual dimensions (e.g., phonemes). In tasks defined over categorical perceptual dimensions, optimal cue weights should depend not only on the sensory variance affecting the perception of each cue but also on the environmental variance inherent in each task-relevant category. Here, we present a computational and experimental investigation of cue integration in a categorical audio-visual (articulatory) speech perception task. Our results show that human performance during audio-visual phonemic labeling is qualitatively consistent with the behavior of a Bayes-optimal observer. Specifically, we show that the participants in our task are sensitive, on a trial-by-trial basis, to the sensory uncertainty associated with the auditory and visual cues, during phonemic categorization. In addition, we show that while sensory uncertainty is a significant factor in determining cue weights, it is not the only one and participants' performance is consistent with an optimal model in which environmental, within category variability also plays a role in determining cue weights. Furthermore, we show that in our task, the sensory variability affecting the visual modality during cue-combination is not well estimated from single-cue performance, but can be estimated from multi-cue performance. The findings and computational principles described here represent a principled first step towards characterizing the mechanisms underlying human cue integration in categorical tasks. PMID:21637344

Review of Speech-to-Text Recognition Technology for Enhancing Learning

ERIC Educational Resources Information Center

Shadiev, Rustam; Hwang, Wu-Yuin; Chen, Nian-Shing; Huang, Yueh-Min

2014-01-01

This paper reviewed literature from 1999 to 2014 inclusively on how Speech-to-Text Recognition (STR) technology has been applied to enhance learning. The first aim of this review is to understand how STR technology has been used to support learning over the past fifteen years, and the second is to analyze all research evidence to understand how…

Speech sound classification and detection of articulation disorders with support vector machines and wavelets.

PubMed

Georgoulas, George; Georgopoulos, Voula C; Stylios, Chrysostomos D

2006-01-01

This paper proposes a novel integrated methodology to extract features and classify speech sounds with intent to detect the possible existence of a speech articulation disorder in a speaker. Articulation, in effect, is the specific and characteristic way that an individual produces the speech sounds. A methodology to process the speech signal, extract features and finally classify the signal and detect articulation problems in a speaker is presented. The use of support vector machines (SVMs), for the classification of speech sounds and detection of articulation disorders is introduced. The proposed method is implemented on a data set where different sets of features and different schemes of SVMs are tested leading to satisfactory performance.

Exploring expressivity and emotion with artificial voice and speech technologies.

PubMed

Pauletto, Sandra; Balentine, Bruce; Pidcock, Chris; Jones, Kevin; Bottaci, Leonardo; Aretoulaki, Maria; Wells, Jez; Mundy, Darren P; Balentine, James

2013-10-01

Emotion in audio-voice signals, as synthesized by text-to-speech (TTS) technologies, was investigated to formulate a theory of expression for user interface design. Emotional parameters were specified with markup tags, and the resulting audio was further modulated with post-processing techniques. Software was then developed to link a selected TTS synthesizer with an automatic speech recognition (ASR) engine, producing a chatbot that could speak and listen. Using these two artificial voice subsystems, investigators explored both artistic and psychological implications of artificial speech emotion. Goals of the investigation were interdisciplinary, with interest in musical composition, augmentative and alternative communication (AAC), commercial voice announcement applications, human-computer interaction (HCI), and artificial intelligence (AI). The work-in-progress points towards an emerging interdisciplinary ontology for artificial voices. As one study output, HCI tools are proposed for future collaboration.

Fidelity of Automatic Speech Processing for Adult and Child Talker Classifications.

PubMed

VanDam, Mark; Silbert, Noah H

2016-01-01

Automatic speech processing (ASP) has recently been applied to very large datasets of naturalistically collected, daylong recordings of child speech via an audio recorder worn by young children. The system developed by the LENA Research Foundation analyzes children's speech for research and clinical purposes, with special focus on of identifying and tagging family speech dynamics and the at-home acoustic environment from the auditory perspective of the child. A primary issue for researchers, clinicians, and families using the Language ENvironment Analysis (LENA) system is to what degree the segment labels are valid. This classification study evaluates the performance of the computer ASP output against 23 trained human judges who made about 53,000 judgements of classification of segments tagged by the LENA ASP. Results indicate performance consistent with modern ASP such as those using HMM methods, with acoustic characteristics of fundamental frequency and segment duration most important for both human and machine classifications. Results are likely to be important for interpreting and improving ASP output.

Fidelity of Automatic Speech Processing for Adult and Child Talker Classifications

PubMed Central

2016-01-01

Automatic speech processing (ASP) has recently been applied to very large datasets of naturalistically collected, daylong recordings of child speech via an audio recorder worn by young children. The system developed by the LENA Research Foundation analyzes children's speech for research and clinical purposes, with special focus on of identifying and tagging family speech dynamics and the at-home acoustic environment from the auditory perspective of the child. A primary issue for researchers, clinicians, and families using the Language ENvironment Analysis (LENA) system is to what degree the segment labels are valid. This classification study evaluates the performance of the computer ASP output against 23 trained human judges who made about 53,000 judgements of classification of segments tagged by the LENA ASP. Results indicate performance consistent with modern ASP such as those using HMM methods, with acoustic characteristics of fundamental frequency and segment duration most important for both human and machine classifications. Results are likely to be important for interpreting and improving ASP output. PMID:27529813

McGurk stimuli for the investigation of multisensory integration in cochlear implant users: The Oldenburg Audio Visual Speech Stimuli (OLAVS).

PubMed

Stropahl, Maren; Schellhardt, Sebastian; Debener, Stefan

2017-06-01

The concurrent presentation of different auditory and visual syllables may result in the perception of a third syllable, reflecting an illusory fusion of visual and auditory information. This well-known McGurk effect is frequently used for the study of audio-visual integration. Recently, it was shown that the McGurk effect is strongly stimulus-dependent, which complicates comparisons across perceivers and inferences across studies. To overcome this limitation, we developed the freely available Oldenburg audio-visual speech stimuli (OLAVS), consisting of 8 different talkers and 12 different syllable combinations. The quality of the OLAVS set was evaluated with 24 normal-hearing subjects. All 96 stimuli were characterized based on their stimulus disparity, which was obtained from a probabilistic model (cf. Magnotti & Beauchamp, 2015). Moreover, the McGurk effect was studied in eight adult cochlear implant (CI) users. By applying the individual, stimulus-independent parameters of the probabilistic model, the predicted effect of stronger audio-visual integration in CI users could be confirmed, demonstrating the validity of the new stimulus material.

[Ventriloquism and audio-visual integration of voice and face].

PubMed

Yokosawa, Kazuhiko; Kanaya, Shoko

2012-07-01

Presenting synchronous auditory and visual stimuli in separate locations creates the illusion that the sound originates from the direction of the visual stimulus. Participants' auditory localization bias, called the ventriloquism effect, has revealed factors affecting the perceptual integration of audio-visual stimuli. However, many studies on audio-visual processes have focused on performance in simplified experimental situations, with a single stimulus in each sensory modality. These results cannot necessarily explain our perceptual behavior in natural scenes, where various signals exist within a single sensory modality. In the present study we report the contributions of a cognitive factor, that is, the audio-visual congruency of speech, although this factor has often been underestimated in previous ventriloquism research. Thus, we investigated the contribution of speech congruency on the ventriloquism effect using a spoken utterance and two videos of a talking face. The salience of facial movements was also manipulated. As a result, when bilateral visual stimuli are presented in synchrony with a single voice, cross-modal speech congruency was found to have a significant impact on the ventriloquism effect. This result also indicated that more salient visual utterances attracted participants' auditory localization. The congruent pairing of audio-visual utterances elicited greater localization bias than did incongruent pairing, whereas previous studies have reported little dependency on the reality of stimuli in ventriloquism. Moreover, audio-visual illusory congruency, owing to the McGurk effect, caused substantial visual interference to auditory localization. This suggests that a greater flexibility in responding to multi-sensory environments exists than has been previously considered.

Visual speech segmentation: using facial cues to locate word boundaries in continuous speech

PubMed Central

Mitchel, Aaron D.; Weiss, Daniel J.

2014-01-01

Speech is typically a multimodal phenomenon, yet few studies have focused on the exclusive contributions of visual cues to language acquisition. To address this gap, we investigated whether visual prosodic information can facilitate speech segmentation. Previous research has demonstrated that language learners can use lexical stress and pitch cues to segment speech and that learners can extract this information from talking faces. Thus, we created an artificial speech stream that contained minimal segmentation cues and paired it with two synchronous facial displays in which visual prosody was either informative or uninformative for identifying word boundaries. Across three familiarisation conditions (audio stream alone, facial streams alone, and paired audiovisual), learning occurred only when the facial displays were informative to word boundaries, suggesting that facial cues can help learners solve the early challenges of language acquisition. PMID:25018577

Text to Speech (TTS) Capabilities for the Common Driver Trainer (CDT)

DTIC Science & Technology

2010-10-01

harnessing in’leigle jalClpeno jocelyn linu ~ los angeles lottery margarine mathematlze mathematized mathematized meme memes memol...including Julie, Kate, and Paul . Based upon the names of the voices, it may be that the VoiceText capability is the technology being used currently on...DFTTSExportToFileEx(O, " Paul ", 1, 1033, "Testing the Digital Future Text-to-Speech SDK.", -1, -1, -1, -1, -1, DFTTS_ TEXT_ TYPE_ XML, "test.wav", 0, "", -1

An Algorithm for Controlled Integration of Sound and Text.

ERIC Educational Resources Information Center

Wohlert, Harry S.; McCormick, Martin

1985-01-01

A serious drawback in introducing sound into computer programs for teaching foreign language speech has been the lack of an algorithm to turn off the cassette recorder immediately to keep screen text and audio in synchronization. This article describes a program which solves that problem. (SED)

Evaluating the Effort Expended to Understand Speech in Noise Using a Dual-Task Paradigm: The Effects of Providing Visual Speech Cues

ERIC Educational Resources Information Center

Fraser, Sarah; Gagne, Jean-Pierre; Alepins, Majolaine; Dubois, Pascale

2010-01-01

Purpose: Using a dual-task paradigm, 2 experiments (Experiments 1 and 2) were conducted to assess differences in the amount of listening effort expended to understand speech in noise in audiovisual (AV) and audio-only (A-only) modalities. Experiment 1 had equivalent noise levels in both modalities, and Experiment 2 equated speech recognition…

Use of speech-to-text technology for documentation by healthcare providers.

PubMed

Ajami, Sima

2016-01-01

Medical records are a critical component of a patient's treatment. However, documentation of patient-related information is considered a secondary activity in the provision of healthcare services, often leading to incomplete medical records and patient data of low quality. Advances in information technology (IT) in the health system and registration of information in electronic health records (EHR) using speechto- text conversion software have facilitated service delivery. This narrative review is a literature search with the help of libraries, books, conference proceedings, databases of Science Direct, PubMed, Proquest, Springer, SID (Scientific Information Database), and search engines such as Yahoo, and Google. I used the following keywords and their combinations: speech recognition, automatic report documentation, voice to text software, healthcare, information, and voice recognition. Due to lack of knowledge of other languages, I searched all texts in English or Persian with no time limits. Of a total of 70, only 42 articles were selected. Speech-to-text conversion technology offers opportunities to improve the documentation process of medical records, reduce cost and time of recording information, enhance the quality of documentation, improve the quality of services provided to patients, and support healthcare providers in legal matters. Healthcare providers should recognize the impact of this technology on service delivery.

Ontology-based structured cosine similarity in document summarization: with applications to mobile audio-based knowledge management.

PubMed

Yuan, Soe-Tsyr; Sun, Jerry

2005-10-01

Development of algorithms for automated text categorization in massive text document sets is an important research area of data mining and knowledge discovery. Most of the text-clustering methods were grounded in the term-based measurement of distance or similarity, ignoring the structure of the documents. In this paper, we present a novel method named structured cosine similarity (SCS) that furnishes document clustering with a new way of modeling on document summarization, considering the structure of the documents so as to improve the performance of document clustering in terms of quality, stability, and efficiency. This study was motivated by the problem of clustering speech documents (of no rich document features) attained from the wireless experience oral sharing conducted by mobile workforce of enterprises, fulfilling audio-based knowledge management. In other words, this problem aims to facilitate knowledge acquisition and sharing by speech. The evaluations also show fairly promising results on our method of structured cosine similarity.

Audio-visual imposture

NASA Astrophysics Data System (ADS)

Karam, Walid; Mokbel, Chafic; Greige, Hanna; Chollet, Gerard

2006-05-01

A GMM based audio visual speaker verification system is described and an Active Appearance Model with a linear speaker transformation system is used to evaluate the robustness of the verification. An Active Appearance Model (AAM) is used to automatically locate and track a speaker's face in a video recording. A Gaussian Mixture Model (GMM) based classifier (BECARS) is used for face verification. GMM training and testing is accomplished on DCT based extracted features of the detected faces. On the audio side, speech features are extracted and used for speaker verification with the GMM based classifier. Fusion of both audio and video modalities for audio visual speaker verification is compared with face verification and speaker verification systems. To improve the robustness of the multimodal biometric identity verification system, an audio visual imposture system is envisioned. It consists of an automatic voice transformation technique that an impostor may use to assume the identity of an authorized client. Features of the transformed voice are then combined with the corresponding appearance features and fed into the GMM based system BECARS for training. An attempt is made to increase the acceptance rate of the impostor and to analyzing the robustness of the verification system. Experiments are being conducted on the BANCA database, with a prospect of experimenting on the newly developed PDAtabase developed within the scope of the SecurePhone project.

A Digital Liquid State Machine With Biologically Inspired Learning and Its Application to Speech Recognition.

PubMed

Zhang, Yong; Li, Peng; Jin, Yingyezhe; Choe, Yoonsuck

2015-11-01

This paper presents a bioinspired digital liquid-state machine (LSM) for low-power very-large-scale-integration (VLSI)-based machine learning applications. To the best of the authors' knowledge, this is the first work that employs a bioinspired spike-based learning algorithm for the LSM. With the proposed online learning, the LSM extracts information from input patterns on the fly without needing intermediate data storage as required in offline learning methods such as ridge regression. The proposed learning rule is local such that each synaptic weight update is based only upon the firing activities of the corresponding presynaptic and postsynaptic neurons without incurring global communications across the neural network. Compared with the backpropagation-based learning, the locality of computation in the proposed approach lends itself to efficient parallel VLSI implementation. We use subsets of the TI46 speech corpus to benchmark the bioinspired digital LSM. To reduce the complexity of the spiking neural network model without performance degradation for speech recognition, we study the impacts of synaptic models on the fading memory of the reservoir and hence the network performance. Moreover, we examine the tradeoffs between synaptic weight resolution, reservoir size, and recognition performance and present techniques to further reduce the overhead of hardware implementation. Our simulation results show that in terms of isolated word recognition evaluated using the TI46 speech corpus, the proposed digital LSM rivals the state-of-the-art hidden Markov-model-based recognizer Sphinx-4 and outperforms all other reported recognizers including the ones that are based upon the LSM or neural networks.

Reduced efficiency of audiovisual integration for nonnative speech.

PubMed

Yi, Han-Gyol; Phelps, Jasmine E B; Smiljanic, Rajka; Chandrasekaran, Bharath

2013-11-01

The role of visual cues in native listeners' perception of speech produced by nonnative speakers has not been extensively studied. Native perception of English sentences produced by native English and Korean speakers in audio-only and audiovisual conditions was examined. Korean speakers were rated as more accented in audiovisual than in the audio-only condition. Visual cues enhanced word intelligibility for native English speech but less so for Korean-accented speech. Reduced intelligibility of Korean-accented audiovisual speech was associated with implicit visual biases, suggesting that listener-related factors partially influence the efficiency of audiovisual integration for nonnative speech perception.

Accessible Text-to-Speech Options for Students Who Struggle with Reading

ERIC Educational Resources Information Center

Bone, Erin K.; Bouck, Emily C.

2017-01-01

As students progress through school they spend more time reading to obtain information. Reading to learn can be a struggle for any student, but it tends to be a bigger obstacle for students with disabilities. Using text-to-speech applications and extensions is one way to assist students with disabilities who struggle to independently complete…

The enhancement of beneficial effects following audio feedback by cognitive preparation in the treatment of social anxiety: a single-session experiment.

PubMed

Nilsson, Jan-Erik; Lundh, Lars-Gunnar; Faghihi, Shahriar; Roth-Andersson, Gun

2011-12-01

According to cognitive models, negatively biased processing of the publicly observable self is an important aspect of social phobia; if this is true, effective methods for producing corrective feedback concerning the public self should be strived for. Video feedback is proven effective, but since one's voice represents another aspect of the self, audio feedback should produce equivalent results. This is the first study to assess the enhancement of audio feedback by cognitive preparation in a single-session randomized controlled experiment. Forty socially anxious participants were asked to give a speech, then to listen to and evaluate a taped recording of their performance. Half of the sample was given cognitive preparation prior to the audio feedback and the remainder received audio feedback only. Cognitive preparation involved asking participants to (1) predict in detail what they would hear on the audiotape, (2) form an image of themselves giving the speech and (3) listen to the audio recording as though they were listening to a stranger. To assess generalization effects all participants were asked to give a second speech. Audio feedback with cognitive preparation was shown to produce less negative ratings after the first speech, and effects generalized to the evaluation of the second speech. More positive speech evaluations were associated with corresponding reductions of state anxiety. Social anxiety as indexed by the Implicit Association Test was reduced in participants given cognitive preparation. Small sample size; analogue study. Audio feedback with cognitive preparation may be utilized as a treatment intervention for social phobia. Copyright © 2011 Elsevier Ltd. All rights reserved.

Video-assisted segmentation of speech and audio track

NASA Astrophysics Data System (ADS)

Pandit, Medha; Yusoff, Yusseri; Kittler, Josef; Christmas, William J.; Chilton, E. H. S.

1999-08-01

Video database research is commonly concerned with the storage and retrieval of visual information invovling sequence segmentation, shot representation and video clip retrieval. In multimedia applications, video sequences are usually accompanied by a sound track. The sound track contains potential cues to aid shot segmentation such as different speakers, background music, singing and distinctive sounds. These different acoustic categories can be modeled to allow for an effective database retrieval. In this paper, we address the problem of automatic segmentation of audio track of multimedia material. This audio based segmentation can be combined with video scene shot detection in order to achieve partitioning of the multimedia material into semantically significant segments.

Kernel-Based Sensor Fusion With Application to Audio-Visual Voice Activity Detection

NASA Astrophysics Data System (ADS)

Dov, David; Talmon, Ronen; Cohen, Israel

2016-12-01

In this paper, we address the problem of multiple view data fusion in the presence of noise and interferences. Recent studies have approached this problem using kernel methods, by relying particularly on a product of kernels constructed separately for each view. From a graph theory point of view, we analyze this fusion approach in a discrete setting. More specifically, based on a statistical model for the connectivity between data points, we propose an algorithm for the selection of the kernel bandwidth, a parameter, which, as we show, has important implications on the robustness of this fusion approach to interferences. Then, we consider the fusion of audio-visual speech signals measured by a single microphone and by a video camera pointed to the face of the speaker. Specifically, we address the task of voice activity detection, i.e., the detection of speech and non-speech segments, in the presence of structured interferences such as keyboard taps and office noise. We propose an algorithm for voice activity detection based on the audio-visual signal. Simulation results show that the proposed algorithm outperforms competing fusion and voice activity detection approaches. In addition, we demonstrate that a proper selection of the kernel bandwidth indeed leads to improved performance.

Use of Video and Audio Texts in EFL Listening Test

ERIC Educational Resources Information Center

Basal, Ahmet; Gülözer, Kaine; Demir, Ibrahim

2015-01-01

The study aims to discover whether audio or video modality in a listening test is more beneficial to test takers. In this study, the posttest-only control group design was utilized and quantitative data were collected in order to measure participant performances concerning two types of modality (audio or video) in a listening test. The…

A Study of Text-to-Speech (TTS) in Children's English Learning

ERIC Educational Resources Information Center

Huang, Yi-Ching; Liao, Lung-Chuan

2015-01-01

The purpose of this study was to explore the effects of the digital material incorporated into Text-to- Speech system for students' English spelling. The digital material was made on the basis of the Spelling Bee vocabulary list (approximately 300 words) issued by the selected school. 21 third graders from a private bilingual school in Taiwan were…

Neural Entrainment to Rhythmically Presented Auditory, Visual, and Audio-Visual Speech in Children

PubMed Central

Power, Alan James; Mead, Natasha; Barnes, Lisa; Goswami, Usha

2012-01-01

Auditory cortical oscillations have been proposed to play an important role in speech perception. It is suggested that the brain may take temporal “samples” of information from the speech stream at different rates, phase resetting ongoing oscillations so that they are aligned with similar frequency bands in the input (“phase locking”). Information from these frequency bands is then bound together for speech perception. To date, there are no explorations of neural phase locking and entrainment to speech input in children. However, it is clear from studies of language acquisition that infants use both visual speech information and auditory speech information in learning. In order to study neural entrainment to speech in typically developing children, we use a rhythmic entrainment paradigm (underlying 2 Hz or delta rate) based on repetition of the syllable “ba,” presented in either the auditory modality alone, the visual modality alone, or as auditory-visual speech (via a “talking head”). To ensure attention to the task, children aged 13 years were asked to press a button as fast as possible when the “ba” stimulus violated the rhythm for each stream type. Rhythmic violation depended on delaying the occurrence of a “ba” in the isochronous stream. Neural entrainment was demonstrated for all stream types, and individual differences in standardized measures of language processing were related to auditory entrainment at the theta rate. Further, there was significant modulation of the preferred phase of auditory entrainment in the theta band when visual speech cues were present, indicating cross-modal phase resetting. The rhythmic entrainment paradigm developed here offers a method for exploring individual differences in oscillatory phase locking during development. In particular, a method for assessing neural entrainment and cross-modal phase resetting would be useful for exploring developmental learning difficulties thought to involve temporal

A Selective Deficit in Phonetic Recalibration by Text in Developmental Dyslexia.

PubMed

Keetels, Mirjam; Bonte, Milene; Vroomen, Jean

2018-01-01

Upon hearing an ambiguous speech sound, listeners may adjust their perceptual interpretation of the speech input in accordance with contextual information, like accompanying text or lipread speech (i.e., phonetic recalibration; Bertelson et al., 2003). As developmental dyslexia (DD) has been associated with reduced integration of text and speech sounds, we investigated whether this deficit becomes manifest when text is used to induce this type of audiovisual learning. Adults with DD and normal readers were exposed to ambiguous consonants halfway between /aba/ and /ada/ together with text or lipread speech. After this audiovisual exposure phase, they categorized auditory-only ambiguous test sounds. Results showed that individuals with DD, unlike normal readers, did not use text to recalibrate their phoneme categories, whereas their recalibration by lipread speech was spared. Individuals with DD demonstrated similar deficits when ambiguous vowels (halfway between /wIt/ and /wet/) were recalibrated by text. These findings indicate that DD is related to a specific letter-speech sound association deficit that extends over phoneme classes (vowels and consonants), but - as lipreading was spared - does not extend to a more general audio-visual integration deficit. In particular, these results highlight diminished reading-related audiovisual learning in addition to the commonly reported phonological problems in developmental dyslexia.

Impact of Audio-Visual Asynchrony on Lip-Reading Effects -Neuromagnetic and Psychophysical Study-

PubMed Central

Yahata, Izumi; Kanno, Akitake; Sakamoto, Shuichi; Takanashi, Yoshitaka; Takata, Shiho; Nakasato, Nobukazu; Kawashima, Ryuta; Katori, Yukio

2016-01-01

The effects of asynchrony between audio and visual (A/V) stimuli on the N100m responses of magnetoencephalography in the left hemisphere were compared with those on the psychophysical responses in 11 participants. The latency and amplitude of N100m were significantly shortened and reduced in the left hemisphere by the presentation of visual speech as long as the temporal asynchrony between A/V stimuli was within 100 ms, but were not significantly affected with audio lags of -500 and +500 ms. However, some small effects were still preserved on average with audio lags of 500 ms, suggesting similar asymmetry of the temporal window to that observed in psychophysical measurements, which tended to be more robust (wider) for audio lags; i.e., the pattern of visual-speech effects as a function of A/V lag observed in the N100m in the left hemisphere grossly resembled that in psychophysical measurements on average, although the individual responses were somewhat varied. The present results suggest that the basic configuration of the temporal window of visual effects on auditory-speech perception could be observed from the early auditory processing stage. PMID:28030631

Advancements in text-to-speech technology and implications for AAC applications

NASA Astrophysics Data System (ADS)

Syrdal, Ann K.

2003-10-01

Intelligibility was the initial focus in text-to-speech (TTS) research, since it is clearly a necessary condition for the application of the technology. Sufficiently high intelligibility (approximating human speech) has been achieved in the last decade by the better formant-based and concatenative TTS systems. This led to commercially available TTS systems for highly motivated users, particularly the blind and vocally impaired. Some unnatural qualities of TTS were exploited by these users, such as very fast speaking rates and altered pitch ranges for flagging relevant information. Recently, the focus in TTS research has turned to improving naturalness, so that synthetic speech sounds more human and less robotic. Unit selection approaches to concatenative synthesis have dramatically improved TTS quality, although at the cost of larger and more complex systems. This advancement in naturalness has made TTS technology more acceptable to the general public. The vocally impaired appreciate a more natural voice with which to represent themselves when communicating with others. Unit selection TTS does not achieve such high speaking rates as the earlier TTS systems, however, which is a disadvantage to some AAC device users. An important new research emphasis is to improve and increase the range of emotional expressiveness of TTS.

Speech information retrieval: a review

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

Hafen, Ryan P.; Henry, Michael J.

Audio is an information-rich component of multimedia. Information can be extracted from audio in a number of different ways, and thus there are several established audio signal analysis research fields. These fields include speech recognition, speaker recognition, audio segmentation and classification, and audio finger-printing. The information that can be extracted from tools and methods developed in these fields can greatly enhance multimedia systems. In this paper, we present the current state of research in each of the major audio analysis fields. The goal is to introduce enough back-ground for someone new in the field to quickly gain high-level understanding andmore » to provide direction for further study.« less

A technology prototype system for rating therapist empathy from audio recordings in addiction counseling.

PubMed

Xiao, Bo; Huang, Chewei; Imel, Zac E; Atkins, David C; Georgiou, Panayiotis; Narayanan, Shrikanth S

2016-04-01

Scaling up psychotherapy services such as for addiction counseling is a critical societal need. One challenge is ensuring quality of therapy, due to the heavy cost of manual observational assessment. This work proposes a speech technology-based system to automate the assessment of therapist empathy-a key therapy quality index-from audio recordings of the psychotherapy interactions. We designed a speech processing system that includes voice activity detection and diarization modules, and an automatic speech recognizer plus a speaker role matching module to extract the therapist's language cues. We employed Maximum Entropy models, Maximum Likelihood language models, and a Lattice Rescoring method to characterize high vs. low empathic language. We estimated therapy-session level empathy codes using utterance level evidence obtained from these models. Our experiments showed that the fully automated system achieved a correlation of 0.643 between expert annotated empathy codes and machine-derived estimations, and an accuracy of 81% in classifying high vs. low empathy, in comparison to a 0.721 correlation and 86% accuracy in the oracle setting using manual transcripts. The results show that the system provides useful information that can contribute to automatic quality insurance and therapist training.

A technology prototype system for rating therapist empathy from audio recordings in addiction counseling

PubMed Central

Xiao, Bo; Huang, Chewei; Imel, Zac E.; Atkins, David C.; Georgiou, Panayiotis; Narayanan, Shrikanth S.

2016-01-01

Scaling up psychotherapy services such as for addiction counseling is a critical societal need. One challenge is ensuring quality of therapy, due to the heavy cost of manual observational assessment. This work proposes a speech technology-based system to automate the assessment of therapist empathy—a key therapy quality index—from audio recordings of the psychotherapy interactions. We designed a speech processing system that includes voice activity detection and diarization modules, and an automatic speech recognizer plus a speaker role matching module to extract the therapist's language cues. We employed Maximum Entropy models, Maximum Likelihood language models, and a Lattice Rescoring method to characterize high vs. low empathic language. We estimated therapy-session level empathy codes using utterance level evidence obtained from these models. Our experiments showed that the fully automated system achieved a correlation of 0.643 between expert annotated empathy codes and machine-derived estimations, and an accuracy of 81% in classifying high vs. low empathy, in comparison to a 0.721 correlation and 86% accuracy in the oracle setting using manual transcripts. The results show that the system provides useful information that can contribute to automatic quality insurance and therapist training. PMID:28286867

A Bit Stream Scalable Speech/Audio Coder Combining Enhanced Regular Pulse Excitation and Parametric Coding

NASA Astrophysics Data System (ADS)

Riera-Palou, Felip; den Brinker, Albertus C.

2007-12-01

This paper introduces a new audio and speech broadband coding technique based on the combination of a pulse excitation coder and a standardized parametric coder, namely, MPEG-4 high-quality parametric coder. After presenting a series of enhancements to regular pulse excitation (RPE) to make it suitable for the modeling of broadband signals, it is shown how pulse and parametric codings complement each other and how they can be merged to yield a layered bit stream scalable coder able to operate at different points in the quality bit rate plane. The performance of the proposed coder is evaluated in a listening test. The major result is that the extra functionality of the bit stream scalability does not come at the price of a reduced performance since the coder is competitive with standardized coders (MP3, AAC, SSC).

Orthographic learning and the role of text-to-speech software in Dutch disabled readers.

PubMed

Staels, Eva; Van den Broeck, Wim

2015-01-01

In this study, we examined whether orthographic learning can be demonstrated in disabled readers learning to read in a transparent orthography (Dutch). In addition, we tested the effect of the use of text-to-speech software, a new form of direct instruction, on orthographic learning. Both research goals were investigated by replicating Share's self-teaching paradigm. A total of 65 disabled Dutch readers were asked to read eight stories containing embedded homophonic pseudoword targets (e.g., Blot/Blod), with or without the support of text-to-speech software. The amount of orthographic learning was assessed 3 or 7 days later by three measures of orthographic learning. First, the results supported the presence of orthographic learning during independent silent reading by demonstrating that target spellings were correctly identified more often, named more quickly, and spelled more accurately than their homophone foils. Our results support the hypothesis that all readers, even poor readers of transparent orthographies, are capable of developing word-specific knowledge. Second, a negative effect of text-to-speech software on orthographic learning was demonstrated in this study. This negative effect was interpreted as the consequence of passively listening to the auditory presentation of the text. We clarify how these results can be interpreted within current theoretical accounts of orthographic learning and briefly discuss implications for remedial interventions. © Hammill Institute on Disabilities 2013.

Audio-Visual Speech Perception: A Developmental ERP Investigation

ERIC Educational Resources Information Center

Knowland, Victoria C. P.; Mercure, Evelyne; Karmiloff-Smith, Annette; Dick, Fred; Thomas, Michael S. C.

2014-01-01

Being able to see a talking face confers a considerable advantage for speech perception in adulthood. However, behavioural data currently suggest that children fail to make full use of these available visual speech cues until age 8 or 9. This is particularly surprising given the potential utility of multiple informational cues during language…

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.

Audio-visual affective expression recognition

NASA Astrophysics Data System (ADS)

Huang, Thomas S.; Zeng, Zhihong

2007-11-01

Automatic affective expression recognition has attracted more and more attention of researchers from different disciplines, which will significantly contribute to a new paradigm for human computer interaction (affect-sensitive interfaces, socially intelligent environments) and advance the research in the affect-related fields including psychology, psychiatry, and education. Multimodal information integration is a process that enables human to assess affective states robustly and flexibly. In order to understand the richness and subtleness of human emotion behavior, the computer should be able to integrate information from multiple sensors. We introduce in this paper our efforts toward machine understanding of audio-visual affective behavior, based on both deliberate and spontaneous displays. Some promising methods are presented to integrate information from both audio and visual modalities. Our experiments show the advantage of audio-visual fusion in affective expression recognition over audio-only or visual-only approaches.

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

Fall Detection Using Smartphone Audio Features.

PubMed

Cheffena, Michael

2016-07-01

An automated fall detection system based on smartphone audio features is developed. The spectrogram, mel frequency cepstral coefficents (MFCCs), linear predictive coding (LPC), and matching pursuit (MP) features of different fall and no-fall sound events are extracted from experimental data. Based on the extracted audio features, four different machine learning classifiers: k-nearest neighbor classifier (k-NN), support vector machine (SVM), least squares method (LSM), and artificial neural network (ANN) are investigated for distinguishing between fall and no-fall events. For each audio feature, the performance of each classifier in terms of sensitivity, specificity, accuracy, and computational complexity is evaluated. The best performance is achieved using spectrogram features with ANN classifier with sensitivity, specificity, and accuracy all above 98%. The classifier also has acceptable computational requirement for training and testing. The system is applicable in home environments where the phone is placed in the vicinity of the user.

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,…

The sweet-home project: audio technology in smart homes to improve well-being and reliance.

PubMed

Vacher, Michel; Istrate, Dan; Portet, François; Joubert, Thierry; Chevalier, Thierry; Smidtas, Serge; Meillon, Brigitte; Lecouteux, Benjamin; Sehili, Mohamed; Chahuara, Pedro; Méniard, Sylvain

2011-01-01

The Sweet-Home project aims at providing audio-based interaction technology that lets the user have full control over their home environment, at detecting distress situations and at easing the social inclusion of the elderly and frail population. This paper presents an overview of the project focusing on the multimodal sound corpus acquisition and labelling and on the investigated techniques for speech and sound recognition. The user study and the recognition performances show the interest of this audio technology.

Does Use of Text-to-Speech and Related Read-Aloud Tools Improve Reading Comprehension for Students with Reading Disabilities? A Meta-Analysis

ERIC Educational Resources Information Center

Wood, Sarah G.; Moxley, Jerad H.; Tighe, Elizabeth L.; Wagner, Richard K.

2018-01-01

Text-to-speech and related read-aloud tools are being widely implemented in an attempt to assist students' reading comprehension skills. Read-aloud software, including text-to-speech, is used to translate written text into spoken text, enabling one to listen to written text while reading along. It is not clear how effective text-to-speech is at…

Integrating Text-to-Speech Software into Pedagogically Sound Teaching and Learning Scenarios

ERIC Educational Resources Information Center

Rughooputh, S. D. D. V.; Santally, M. I.

2009-01-01

This paper presents a new technique of delivery of classes--an instructional technique which will no doubt revolutionize the teaching and learning, whether for on-campus, blended or online modules. This is based on the simple task of instructionally incorporating text-to-speech software embedded in the lecture slides that will simulate exactly the…

Sensitivity to audio-visual synchrony and its relation to language abilities in children with and without ASD.

PubMed

Righi, Giulia; Tenenbaum, Elena J; McCormick, Carolyn; Blossom, Megan; Amso, Dima; Sheinkopf, Stephen J

2018-04-01

Autism Spectrum Disorder (ASD) is often accompanied by deficits in speech and language processing. Speech processing relies heavily on the integration of auditory and visual information, and it has been suggested that the ability to detect correspondence between auditory and visual signals helps to lay the foundation for successful language development. The goal of the present study was to examine whether young children with ASD show reduced sensitivity to temporal asynchronies in a speech processing task when compared to typically developing controls, and to examine how this sensitivity might relate to language proficiency. Using automated eye tracking methods, we found that children with ASD failed to demonstrate sensitivity to asynchronies of 0.3s, 0.6s, or 1.0s between a video of a woman speaking and the corresponding audio track. In contrast, typically developing children who were language-matched to the ASD group, were sensitive to both 0.6s and 1.0s asynchronies. We also demonstrated that individual differences in sensitivity to audiovisual asynchronies and individual differences in orientation to relevant facial features were both correlated with scores on a standardized measure of language abilities. Results are discussed in the context of attention to visual language and audio-visual processing as potential precursors to language impairment in ASD. Autism Res 2018, 11: 645-653. © 2018 International Society for Autism Research, Wiley Periodicals, Inc. Speech processing relies heavily on the integration of auditory and visual information, and it has been suggested that the ability to detect correspondence between auditory and visual signals helps to lay the foundation for successful language development. The goal of the present study was to explore whether children with ASD process audio-visual synchrony in ways comparable to their typically developing peers, and the relationship between preference for synchrony and language ability. Results showed that

Research in speech communication.

PubMed

Flanagan, J

1995-10-24

Advances in digital speech processing are now supporting application and deployment of a variety of speech technologies for human/machine communication. In fact, new businesses are rapidly forming about these technologies. But these capabilities are of little use unless society can afford them. Happily, explosive advances in microelectronics over the past two decades have assured affordable access to this sophistication as well as to the underlying computing technology. The research challenges in speech processing remain in the traditionally identified areas of recognition, synthesis, and coding. These three areas have typically been addressed individually, often with significant isolation among the efforts. But they are all facets of the same fundamental issue--how to represent and quantify the information in the speech signal. This implies deeper understanding of the physics of speech production, the constraints that the conventions of language impose, and the mechanism for information processing in the auditory system. In ongoing research, therefore, we seek more accurate models of speech generation, better computational formulations of language, and realistic perceptual guides for speech processing--along with ways to coalesce the fundamental issues of recognition, synthesis, and coding. Successful solution will yield the long-sought dictation machine, high-quality synthesis from text, and the ultimate in low bit-rate transmission of speech. It will also open the door to language-translating telephony, where the synthetic foreign translation can be in the voice of the originating talker.

Audio-vocal responses of vocal fundamental frequency and formant during sustained vowel vocalizations in different noises.

PubMed

Lee, Shao-Hsuan; Hsiao, Tzu-Yu; Lee, Guo-She

2015-06-01

Sustained vocalizations of vowels [a], [i], and syllable [mə] were collected in twenty normal-hearing individuals. On vocalizations, five conditions of different audio-vocal feedback were introduced separately to the speakers including no masking, wearing supra-aural headphones only, speech-noise masking, high-pass noise masking, and broad-band-noise masking. Power spectral analysis of vocal fundamental frequency (F0) was used to evaluate the modulations of F0 and linear-predictive-coding was used to acquire first two formants. The results showed that while the formant frequencies were not significantly shifted, low-frequency modulations (<3 Hz) of F0 significantly increased with reduced audio-vocal feedback across speech sounds and were significantly correlated with auditory awareness of speakers' own voices. For sustained speech production, the motor speech controls on F0 may depend on a feedback mechanism while articulation should rely more on a feedforward mechanism. Power spectral analysis of F0 might be applied to evaluate audio-vocal control for various hearing and neurological disorders in the future. Copyright © 2015 Elsevier B.V. All rights reserved.

The Effect of Speech-to-Text Technology on Learning a Writing Strategy

ERIC Educational Resources Information Center

Haug, Katrina N.; Klein, Perry D.

2018-01-01

Previous research has shown that speech-to-text (STT) software can support students in producing a given piece of writing. This is the 1st study to investigate the use of STT to teach a writing strategy. We pretested 45 Grade 5 students on argument writing and trained them to use STT. Students participated in 4 lessons on an argument writing…

Semantic Context Detection Using Audio Event Fusion

NASA Astrophysics Data System (ADS)

Chu, Wei-Ta; Cheng, Wen-Huang; Wu, Ja-Ling

2006-12-01

Semantic-level content analysis is a crucial issue in achieving efficient content retrieval and management. We propose a hierarchical approach that models audio events over a time series in order to accomplish semantic context detection. Two levels of modeling, audio event and semantic context modeling, are devised to bridge the gap between physical audio features and semantic concepts. In this work, hidden Markov models (HMMs) are used to model four representative audio events, that is, gunshot, explosion, engine, and car braking, in action movies. At the semantic context level, generative (ergodic hidden Markov model) and discriminative (support vector machine (SVM)) approaches are investigated to fuse the characteristics and correlations among audio events, which provide cues for detecting gunplay and car-chasing scenes. The experimental results demonstrate the effectiveness of the proposed approaches and provide a preliminary framework for information mining by using audio characteristics.

Audio-Visual Speech in Noise Perception in Dyslexia

ERIC Educational Resources Information Center

van Laarhoven, Thijs; Keetels, Mirjam; Schakel, Lemmy; Vroomen, Jean

2018-01-01

Individuals with developmental dyslexia (DD) may experience, besides reading problems, other speech-related processing deficits. Here, we examined the influence of visual articulatory information (lip-read speech) at various levels of background noise on auditory word recognition in children and adults with DD. We found that children with a…

Estimation of Teacher Practices Based on Text Transcripts of Teacher Speech Using a Support Vector Machine Algorithm

ERIC Educational Resources Information Center

Araya, Roberto; Plana, Francisco; Dartnell, Pablo; Soto-Andrade, Jorge; Luci, Gina; Salinas, Elena; Araya, Marylen

2012-01-01

Teacher practice is normally assessed by observers who watch classes or videos of classes. Here, we analyse an alternative strategy that uses text transcripts and a support vector machine classifier. For each one of the 710 videos of mathematics classes from the 2005 Chilean National Teacher Assessment Programme, a single 4-minute slice was…

SNR-adaptive stream weighting for audio-MES ASR.

PubMed

Lee, Ki-Seung

2008-08-01

Myoelectric signals (MESs) from the speaker's mouth region have been successfully shown to improve the noise robustness of automatic speech recognizers (ASRs), thus promising to extend their usability in implementing noise-robust ASR. In the recognition system presented herein, extracted audio and facial MES features were integrated by a decision fusion method, where the likelihood score of the audio-MES observation vector was given by a linear combination of class-conditional observation log-likelihoods of two classifiers, using appropriate weights. We developed a weighting process adaptive to SNRs. The main objective of the paper involves determining the optimal SNR classification boundaries and constructing a set of optimum stream weights for each SNR class. These two parameters were determined by a method based on a maximum mutual information criterion. Acoustic and facial MES data were collected from five subjects, using a 60-word vocabulary. Four types of acoustic noise including babble, car, aircraft, and white noise were acoustically added to clean speech signals with SNR ranging from -14 to 31 dB. The classification accuracy of the audio ASR was as low as 25.5%. Whereas, the classification accuracy of the MES ASR was 85.2%. The classification accuracy could be further improved by employing the proposed audio-MES weighting method, which was as high as 89.4% in the case of babble noise. A similar result was also found for the other types of noise.

Combining Machine Learning and Natural Language Processing to Assess Literary Text Comprehension

ERIC Educational Resources Information Center

Balyan, Renu; McCarthy, Kathryn S.; McNamara, Danielle S.

2017-01-01

This study examined how machine learning and natural language processing (NLP) techniques can be leveraged to assess the interpretive behavior that is required for successful literary text comprehension. We compared the accuracy of seven different machine learning classification algorithms in predicting human ratings of student essays about…

End-to-End ASR-Free Keyword Search From Speech

NASA Astrophysics Data System (ADS)

Audhkhasi, Kartik; Rosenberg, Andrew; Sethy, Abhinav; Ramabhadran, Bhuvana; Kingsbury, Brian

2017-12-01

End-to-end (E2E) systems have achieved competitive results compared to conventional hybrid hidden Markov model (HMM)-deep neural network based automatic speech recognition (ASR) systems. Such E2E systems are attractive due to the lack of dependence on alignments between input acoustic and output grapheme or HMM state sequence during training. This paper explores the design of an ASR-free end-to-end system for text query-based keyword search (KWS) from speech trained with minimal supervision. Our E2E KWS system consists of three sub-systems. The first sub-system is a recurrent neural network (RNN)-based acoustic auto-encoder trained to reconstruct the audio through a finite-dimensional representation. The second sub-system is a character-level RNN language model using embeddings learned from a convolutional neural network. Since the acoustic and text query embeddings occupy different representation spaces, they are input to a third feed-forward neural network that predicts whether the query occurs in the acoustic utterance or not. This E2E ASR-free KWS system performs respectably despite lacking a conventional ASR system and trains much faster.

An Introduction to Boiler Water Chemistry for the Marine Engineer: A Text of Audio-Tutorial Instruction.

ERIC Educational Resources Information Center

Schlenker, Richard M.; And Others

Presented is a manuscript for an introductory boiler water chemistry course for marine engineer education. The course is modular, self-paced, audio-tutorial, contract graded and combined lecture-laboratory instructed. Lectures are presented to students individually via audio-tapes and 35 mm slides. The course consists of a total of 17 modules -…

MPEG-7 audio-visual indexing test-bed for video retrieval

NASA Astrophysics Data System (ADS)

Gagnon, Langis; Foucher, Samuel; Gouaillier, Valerie; Brun, Christelle; Brousseau, Julie; Boulianne, Gilles; Osterrath, Frederic; Chapdelaine, Claude; Dutrisac, Julie; St-Onge, Francis; Champagne, Benoit; Lu, Xiaojian

2003-12-01

This paper reports on the development status of a Multimedia Asset Management (MAM) test-bed for content-based indexing and retrieval of audio-visual documents within the MPEG-7 standard. The project, called "MPEG-7 Audio-Visual Document Indexing System" (MADIS), specifically targets the indexing and retrieval of video shots and key frames from documentary film archives, based on audio-visual content like face recognition, motion activity, speech recognition and semantic clustering. The MPEG-7/XML encoding of the film database is done off-line. The description decomposition is based on a temporal decomposition into visual segments (shots), key frames and audio/speech sub-segments. The visible outcome will be a web site that allows video retrieval using a proprietary XQuery-based search engine and accessible to members at the Canadian National Film Board (NFB) Cineroute site. For example, end-user will be able to ask to point on movie shots in the database that have been produced in a specific year, that contain the face of a specific actor who tells a specific word and in which there is no motion activity. Video streaming is performed over the high bandwidth CA*net network deployed by CANARIE, a public Canadian Internet development organization.

Detecting double compression of audio signal

NASA Astrophysics Data System (ADS)

Yang, Rui; Shi, Yun Q.; Huang, Jiwu

2010-01-01

MP3 is the most popular audio format nowadays in our daily life, for example music downloaded from the Internet and file saved in the digital recorder are often in MP3 format. However, low bitrate MP3s are often transcoded to high bitrate since high bitrate ones are of high commercial value. Also audio recording in digital recorder can be doctored easily by pervasive audio editing software. This paper presents two methods for the detection of double MP3 compression. The methods are essential for finding out fake-quality MP3 and audio forensics. The proposed methods use support vector machine classifiers with feature vectors formed by the distributions of the first digits of the quantized MDCT (modified discrete cosine transform) coefficients. Extensive experiments demonstrate the effectiveness of the proposed methods. To the best of our knowledge, this piece of work is the first one to detect double compression of audio signal.

Research in speech communication.

PubMed Central

Flanagan, J

1995-01-01

Advances in digital speech processing are now supporting application and deployment of a variety of speech technologies for human/machine communication. In fact, new businesses are rapidly forming about these technologies. But these capabilities are of little use unless society can afford them. Happily, explosive advances in microelectronics over the past two decades have assured affordable access to this sophistication as well as to the underlying computing technology. The research challenges in speech processing remain in the traditionally identified areas of recognition, synthesis, and coding. These three areas have typically been addressed individually, often with significant isolation among the efforts. But they are all facets of the same fundamental issue--how to represent and quantify the information in the speech signal. This implies deeper understanding of the physics of speech production, the constraints that the conventions of language impose, and the mechanism for information processing in the auditory system. In ongoing research, therefore, we seek more accurate models of speech generation, better computational formulations of language, and realistic perceptual guides for speech processing--along with ways to coalesce the fundamental issues of recognition, synthesis, and coding. Successful solution will yield the long-sought dictation machine, high-quality synthesis from text, and the ultimate in low bit-rate transmission of speech. It will also open the door to language-translating telephony, where the synthetic foreign translation can be in the voice of the originating talker. Images Fig. 1 Fig. 2 Fig. 5 Fig. 8 Fig. 11 Fig. 12 Fig. 13 PMID:7479806

Large Vocabulary Audio-Visual Speech Recognition

DTIC Science & Technology

2002-06-12

www.is.cs.cmu.edu Email: waibel(a)cs.cmu~edu Inttractive Systenms Labs ttoctis Ssstms Labs Meeting Browser - -- Interpreting Human Communication "Why did...Speech Interacti Stams Labs t-cive Systms Focus of Attention Tracking Conclusion - Complete Model of Human Communication is Needed - Include all

Multisensory and modality specific processing of visual speech in different regions of the premotor cortex

PubMed Central

Callan, Daniel E.; Jones, Jeffery A.; Callan, Akiko

2014-01-01

Behavioral and neuroimaging studies have demonstrated that brain regions involved with speech production also support speech perception, especially under degraded conditions. The premotor cortex (PMC) has been shown to be active during both observation and execution of action (“Mirror System” properties), and may facilitate speech perception by mapping unimodal and multimodal sensory features onto articulatory speech gestures. For this functional magnetic resonance imaging (fMRI) study, participants identified vowels produced by a speaker in audio-visual (saw the speaker's articulating face and heard her voice), visual only (only saw the speaker's articulating face), and audio only (only heard the speaker's voice) conditions with varying audio signal-to-noise ratios in order to determine the regions of the PMC involved with multisensory and modality specific processing of visual speech gestures. The task was designed so that identification could be made with a high level of accuracy from visual only stimuli to control for task difficulty and differences in intelligibility. The results of the functional magnetic resonance imaging (fMRI) analysis for visual only and audio-visual conditions showed overlapping activity in inferior frontal gyrus and PMC. The left ventral inferior premotor cortex (PMvi) showed properties of multimodal (audio-visual) enhancement with a degraded auditory signal. The left inferior parietal lobule and right cerebellum also showed these properties. The left ventral superior and dorsal premotor cortex (PMvs/PMd) did not show this multisensory enhancement effect, but there was greater activity for the visual only over audio-visual conditions in these areas. The results suggest that the inferior regions of the ventral premotor cortex are involved with integrating multisensory information, whereas, more superior and dorsal regions of the PMC are involved with mapping unimodal (in this case visual) sensory features of the speech signal with

Linguistic experience and audio-visual perception of non-native fricatives.

PubMed

Wang, Yue; Behne, Dawn M; Jiang, Haisheng

2008-09-01

This study examined the effects of linguistic experience on audio-visual (AV) perception of non-native (L2) speech. Canadian English natives and Mandarin Chinese natives differing in degree of English exposure [long and short length of residence (LOR) in Canada] were presented with English fricatives of three visually distinct places of articulation: interdentals nonexistent in Mandarin and labiodentals and alveolars common in both languages. Stimuli were presented in quiet and in a cafe-noise background in four ways: audio only (A), visual only (V), congruent AV (AVc), and incongruent AV (AVi). Identification results showed that overall performance was better in the AVc than in the A or V condition and better in quiet than in cafe noise. While the Mandarin long LOR group approximated the native English patterns, the short LOR group showed poorer interdental identification, more reliance on visual information, and greater AV-fusion with the AVi materials, indicating the failure of L2 visual speech category formation with the short LOR non-natives and the positive effects of linguistic experience with the long LOR non-natives. These results point to an integrated network in AV speech processing as a function of linguistic background and provide evidence to extend auditory-based L2 speech learning theories to the visual domain.

Solid State Audio/Speech Processor Analysis.

DTIC Science & Technology

1980-03-01

techniques. The techniques were demonstrated to be worthwhile in an efficient realtime AWR system. Finally, microprocessor architectures were designed to...do not include custom chip development, detailed hardware design , construction or testing. ITTDCD is very encouraged by the results obtained in this...California, Berkley, was responsible for furnishing the simulation data of OD speech analysis techniques and for the design and development of the hardware OD

Presentation video retrieval using automatically recovered slide and spoken text

NASA Astrophysics Data System (ADS)

Cooper, Matthew

2013-03-01

Video is becoming a prevalent medium for e-learning. Lecture videos contain text information in both the presentation slides and lecturer's speech. This paper examines the relative utility of automatically recovered text from these sources for lecture video retrieval. To extract the visual information, we automatically detect slides within the videos and apply optical character recognition to obtain their text. Automatic speech recognition is used similarly to extract spoken text from the recorded audio. We perform controlled experiments with manually created ground truth for both the slide and spoken text from more than 60 hours of lecture video. We compare the automatically extracted slide and spoken text in terms of accuracy relative to ground truth, overlap with one another, and utility for video retrieval. Results reveal that automatically recovered slide text and spoken text contain different content with varying error profiles. Experiments demonstrate that automatically extracted slide text enables higher precision video retrieval than automatically recovered spoken text.

A Diagnostic Marker to Discriminate Childhood Apraxia of Speech From Speech Delay: I. Development and Description of the Pause Marker

PubMed Central

Strand, Edythe A.; Fourakis, Marios; Jakielski, Kathy J.; Hall, Sheryl D.; Karlsson, Heather B.; Mabie, Heather L.; McSweeny, Jane L.; Tilkens, Christie M.; Wilson, David L.

2017-01-01

Purpose The goal of this article (PM I) is to describe the rationale for and development of the Pause Marker (PM), a single-sign diagnostic marker proposed to discriminate early or persistent childhood apraxia of speech from speech delay. Method The authors describe and prioritize 7 criteria with which to evaluate the research and clinical utility of a diagnostic marker for childhood apraxia of speech, including evaluation of the present proposal. An overview is given of the Speech Disorders Classification System, including extensions completed in the same approximately 3-year period in which the PM was developed. Results The finalized Speech Disorders Classification System includes a nosology and cross-classification procedures for childhood and persistent speech disorders and motor speech disorders (Shriberg, Strand, & Mabie, 2017). A PM is developed that provides procedural and scoring information, and citations to papers and technical reports that include audio exemplars of the PM and reference data used to standardize PM scores are provided. Conclusions The PM described here is an acoustic-aided perceptual sign that quantifies one aspect of speech precision in the linguistic domain of phrasing. This diagnostic marker can be used to discriminate early or persistent childhood apraxia of speech from speech delay. PMID:28384779

Audio stream classification for multimedia database search

NASA Astrophysics Data System (ADS)

Artese, M.; Bianco, S.; Gagliardi, I.; Gasparini, F.

2013-03-01

Search and retrieval of huge archives of Multimedia data is a challenging task. A classification step is often used to reduce the number of entries on which to perform the subsequent search. In particular, when new entries of the database are continuously added, a fast classification based on simple threshold evaluation is desirable. In this work we present a CART-based (Classification And Regression Tree [1]) classification framework for audio streams belonging to multimedia databases. The database considered is the Archive of Ethnography and Social History (AESS) [2], which is mainly composed of popular songs and other audio records describing the popular traditions handed down generation by generation, such as traditional fairs, and customs. The peculiarities of this database are that it is continuously updated; the audio recordings are acquired in unconstrained environment; and for the non-expert human user is difficult to create the ground truth labels. In our experiments, half of all the available audio files have been randomly extracted and used as training set. The remaining ones have been used as test set. The classifier has been trained to distinguish among three different classes: speech, music, and song. All the audio files in the dataset have been previously manually labeled into the three classes above defined by domain experts.

An Evaluation of Text-to-Speech Synthesizers in the Foreign Language Classroom: Learners' Perceptions

ERIC Educational Resources Information Center

Bione, Tiago; Grimshaw, Jennica; Cardoso, Walcir

2016-01-01

As stated in Cardoso, Smith, and Garcia Fuentes (2015), second language researchers and practitioners have explored the pedagogical capabilities of Text-To-Speech synthesizers (TTS) for their potential to enhance the acquisition of writing (e.g. Kirstein, 2006), vocabulary and reading (e.g. Proctor, Dalton, & Grisham, 2007), and pronunciation…

Auditory and audio-visual processing in patients with cochlear, auditory brainstem, and auditory midbrain implants: An EEG study.

PubMed

Schierholz, Irina; Finke, Mareike; Kral, Andrej; Büchner, Andreas; Rach, Stefan; Lenarz, Thomas; Dengler, Reinhard; Sandmann, Pascale

2017-04-01

There is substantial variability in speech recognition ability across patients with cochlear implants (CIs), auditory brainstem implants (ABIs), and auditory midbrain implants (AMIs). To better understand how this variability is related to central processing differences, the current electroencephalography (EEG) study compared hearing abilities and auditory-cortex activation in patients with electrical stimulation at different sites of the auditory pathway. Three different groups of patients with auditory implants (Hannover Medical School; ABI: n = 6, CI: n = 6; AMI: n = 2) performed a speeded response task and a speech recognition test with auditory, visual, and audio-visual stimuli. Behavioral performance and cortical processing of auditory and audio-visual stimuli were compared between groups. ABI and AMI patients showed prolonged response times on auditory and audio-visual stimuli compared with NH listeners and CI patients. This was confirmed by prolonged N1 latencies and reduced N1 amplitudes in ABI and AMI patients. However, patients with central auditory implants showed a remarkable gain in performance when visual and auditory input was combined, in both speech and non-speech conditions, which was reflected by a strong visual modulation of auditory-cortex activation in these individuals. In sum, the results suggest that the behavioral improvement for audio-visual conditions in central auditory implant patients is based on enhanced audio-visual interactions in the auditory cortex. Their findings may provide important implications for the optimization of electrical stimulation and rehabilitation strategies in patients with central auditory prostheses. Hum Brain Mapp 38:2206-2225, 2017. © 2017 Wiley Periodicals, Inc. © 2017 Wiley Periodicals, Inc.

Vector adaptive predictive coder for speech and audio

NASA Technical Reports Server (NTRS)

Chen, Juin-Hwey (Inventor); Gersho, Allen (Inventor)

1990-01-01

A real-time vector adaptive predictive coder which approximates each vector of K speech samples by using each of M fixed vectors in a first codebook to excite a time-varying synthesis filter and picking the vector that minimizes distortion. Predictive analysis for each frame determines parameters used for computing from vectors in the first codebook zero-state response vectors that are stored at the same address (index) in a second codebook. Encoding of input speech vectors s.sub.n is then carried out using the second codebook. When the vector that minimizes distortion is found, its index is transmitted to a decoder which has a codebook identical to the first codebook of the decoder. There the index is used to read out a vector that is used to synthesize an output speech vector s.sub.n. The parameters used in the encoder are quantized, for example by using a table, and the indices are transmitted to the decoder where they are decoded to specify transfer characteristics of filters used in producing the vector s.sub.n from the receiver codebook vector selected by the vector index transmitted.

Parametric Representation of the Speaker's Lips for Multimodal Sign Language and Speech Recognition

NASA Astrophysics Data System (ADS)

Ryumin, D.; Karpov, A. A.

2017-05-01

In this article, we propose a new method for parametric representation of human's lips region. The functional diagram of the method is described and implementation details with the explanation of its key stages and features are given. The results of automatic detection of the regions of interest are illustrated. A speed of the method work using several computers with different performances is reported. This universal method allows applying parametrical representation of the speaker's lipsfor the tasks of biometrics, computer vision, machine learning, and automatic recognition of face, elements of sign languages, and audio-visual speech, including lip-reading.

The Sweet-Home project: audio processing and decision making in smart home to improve well-being and reliance.

PubMed

Vacher, Michel; Chahuara, Pedro; Lecouteux, Benjamin; Istrate, Dan; Portet, Francois; Joubert, Thierry; Sehili, Mohamed; Meillon, Brigitte; Bonnefond, Nicolas; Fabre, Sébastien; Roux, Camille; Caffiau, Sybille

2013-01-01

The Sweet-Home project aims at providing audio-based interaction technology that lets the user have full control over their home environment, at detecting distress situations and at easing the social inclusion of the elderly and frail population. This paper presents an overview of the project focusing on the implemented techniques for speech and sound recognition as context-aware decision making with uncertainty. A user experiment in a smart home demonstrates the interest of this audio-based technology.

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

PubMed

Buchan, Julie N; Munhall, Kevin G

2011-01-01

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

NaturalReader: A New Generation Text Reader

ERIC Educational Resources Information Center

Flood, Jacqueline

2007-01-01

NaturalReader (http://www.naturalreaders.com/) is a new generation text reader, which means that it reads any machine readable text using synthesized speech without having to copy and paste the selected text into the NaturalReader application window. It installs a toolbar directly into all of the Microsoft Office[TM] programs and uses a mini-board…

Lip movements affect infants' audiovisual speech perception.

PubMed

Yeung, H Henny; Werker, Janet F

2013-05-01

Speech is robustly audiovisual from early in infancy. Here we show that audiovisual speech perception in 4.5-month-old infants is influenced by sensorimotor information related to the lip movements they make while chewing or sucking. Experiment 1 consisted of a classic audiovisual matching procedure, in which two simultaneously displayed talking faces (visual [i] and [u]) were presented with a synchronous vowel sound (audio /i/ or /u/). Infants' looking patterns were selectively biased away from the audiovisual matching face when the infants were producing lip movements similar to those needed to produce the heard vowel. Infants' looking patterns returned to those of a baseline condition (no lip movements, looking longer at the audiovisual matching face) when they were producing lip movements that did not match the heard vowel. Experiment 2 confirmed that these sensorimotor effects interacted with the heard vowel, as looking patterns differed when infants produced these same lip movements while seeing and hearing a talking face producing an unrelated vowel (audio /a/). These findings suggest that the development of speech perception and speech production may be mutually informative.

Visual speech information: a help or hindrance in perceptual processing of dysarthric speech.

PubMed

Borrie, Stephanie A

2015-03-01

This study investigated the influence of visual speech information on perceptual processing of neurologically degraded speech. Fifty listeners identified spastic dysarthric speech under both audio (A) and audiovisual (AV) conditions. Condition comparisons revealed that the addition of visual speech information enhanced processing of the neurologically degraded input in terms of (a) acuity (percent phonemes correct) of vowels and consonants and (b) recognition (percent words correct) of predictive and nonpredictive phrases. Listeners exploited stress-based segmentation strategies more readily in AV conditions, suggesting that the perceptual benefit associated with adding visual speech information to the auditory signal-the AV advantage-has both segmental and suprasegmental origins. Results also revealed that the magnitude of the AV advantage can be predicted, to some degree, by the extent to which an individual utilizes syllabic stress cues to inform word recognition in AV conditions. Findings inform the development of a listener-specific model of speech perception that applies to processing of dysarthric speech in everyday communication contexts.

Audiovisual cues benefit recognition of accented speech in noise but not perceptual adaptation

PubMed Central

Banks, Briony; Gowen, Emma; Munro, Kevin J.; Adank, Patti

2015-01-01

Perceptual adaptation allows humans to recognize different varieties of accented speech. We investigated whether perceptual adaptation to accented speech is facilitated if listeners can see a speaker’s facial and mouth movements. In Study 1, participants listened to sentences in a novel accent and underwent a period of training with audiovisual or audio-only speech cues, presented in quiet or in background noise. A control group also underwent training with visual-only (speech-reading) cues. We observed no significant difference in perceptual adaptation between any of the groups. To address a number of remaining questions, we carried out a second study using a different accent, speaker and experimental design, in which participants listened to sentences in a non-native (Japanese) accent with audiovisual or audio-only cues, without separate training. Participants’ eye gaze was recorded to verify that they looked at the speaker’s face during audiovisual trials. Recognition accuracy was significantly better for audiovisual than for audio-only stimuli; however, no statistical difference in perceptual adaptation was observed between the two modalities. Furthermore, Bayesian analysis suggested that the data supported the null hypothesis. Our results suggest that although the availability of visual speech cues may be immediately beneficial for recognition of unfamiliar accented speech in noise, it does not improve perceptual adaptation. PMID:26283946

Audiovisual cues benefit recognition of accented speech in noise but not perceptual adaptation.

PubMed

Banks, Briony; Gowen, Emma; Munro, Kevin J; Adank, Patti

2015-01-01

Perceptual adaptation allows humans to recognize different varieties of accented speech. We investigated whether perceptual adaptation to accented speech is facilitated if listeners can see a speaker's facial and mouth movements. In Study 1, participants listened to sentences in a novel accent and underwent a period of training with audiovisual or audio-only speech cues, presented in quiet or in background noise. A control group also underwent training with visual-only (speech-reading) cues. We observed no significant difference in perceptual adaptation between any of the groups. To address a number of remaining questions, we carried out a second study using a different accent, speaker and experimental design, in which participants listened to sentences in a non-native (Japanese) accent with audiovisual or audio-only cues, without separate training. Participants' eye gaze was recorded to verify that they looked at the speaker's face during audiovisual trials. Recognition accuracy was significantly better for audiovisual than for audio-only stimuli; however, no statistical difference in perceptual adaptation was observed between the two modalities. Furthermore, Bayesian analysis suggested that the data supported the null hypothesis. Our results suggest that although the availability of visual speech cues may be immediately beneficial for recognition of unfamiliar accented speech in noise, it does not improve perceptual adaptation.

Orthographic Learning and the Role of Text-to-Speech Software in Dutch Disabled Readers

ERIC Educational Resources Information Center

Staels, Eva; Van den Broeck, Wim

2015-01-01

In this study, we examined whether orthographic learning can be demonstrated in disabled readers learning to read in a transparent orthography (Dutch). In addition, we tested the effect of the use of text-to-speech software, a new form of direct instruction, on orthographic learning. Both research goals were investigated by replicating Share's…

Noise-Canceling Helmet Audio System

NASA Technical Reports Server (NTRS)

Seibert, Marc A.; Culotta, Anthony J.

2007-01-01

A prototype helmet audio system has been developed to improve voice communication for the wearer in a noisy environment. The system was originally intended to be used in a space suit, wherein noise generated by airflow of the spacesuit life-support system can make it difficult for remote listeners to understand the astronaut s speech and can interfere with the astronaut s attempt to issue vocal commands to a voice-controlled robot. The system could be adapted to terrestrial use in helmets of protective suits that are typically worn in noisy settings: examples include biohazard, fire, rescue, and diving suits. The system (see figure) includes an array of microphones and small loudspeakers mounted at fixed positions in a helmet, amplifiers and signal-routing circuitry, and a commercial digital signal processor (DSP). Notwithstanding the fixed positions of the microphones and loudspeakers, the system can accommodate itself to any normal motion of the wearer s head within the helmet. The system operates in conjunction with a radio transceiver. An audio signal arriving via the transceiver intended to be heard by the wearer is adjusted in volume and otherwise conditioned and sent to the loudspeakers. The wearer s speech is collected by the microphones, the outputs of which are logically combined (phased) so as to form a microphone- array directional sensitivity pattern that discriminates in favor of sounds coming from vicinity of the wearer s mouth and against sounds coming from elsewhere. In the DSP, digitized samples of the microphone outputs are processed to filter out airflow noise and to eliminate feedback from the loudspeakers to the microphones. The resulting conditioned version of the wearer s speech signal is sent to the transceiver.

An experimental version of the MZT (speech-from-text) system with external F(sub 0) control

NASA Astrophysics Data System (ADS)

Nowak, Ignacy

1994-12-01

The version of a Polish speech from text system described in this article was developed using the speech-from-text system. The new system has additional functions which make it possible to enter commands in edited orthographic text to control the phrase component and accentuation parameters. This makes it possible to generate a series of modified intonation contours in the texts spoken by the system. The effects obtained are made easier to control by a graphic illustration of the base frequency pattern in phrases that were last 'spoken' by the system. This version of the system was designed as a test prototype which will help us expand and refine our set of rules for automatic generation of intonation contours, which in turn will enable the fully automated speech-from-text system to generate speech with a more varied and precisely formed fundamental frequency pattern.

Audio-based queries for video retrieval over Java enabled mobile devices

NASA Astrophysics Data System (ADS)

Ahmad, Iftikhar; Cheikh, Faouzi Alaya; Kiranyaz, Serkan; Gabbouj, Moncef

2006-02-01

In this paper we propose a generic framework for efficient retrieval of audiovisual media based on its audio content. This framework is implemented in a client-server architecture where the client application is developed in Java to be platform independent whereas the server application is implemented for the PC platform. The client application adapts to the characteristics of the mobile device where it runs such as screen size and commands. The entire framework is designed to take advantage of the high-level segmentation and classification of audio content to improve speed and accuracy of audio-based media retrieval. Therefore, the primary objective of this framework is to provide an adaptive basis for performing efficient video retrieval operations based on the audio content and types (i.e. speech, music, fuzzy and silence). Experimental results approve that such an audio based video retrieval scheme can be used from mobile devices to search and retrieve video clips efficiently over wireless networks.

Implementation of Three Text to Speech Systems for Kurdish Language

NASA Astrophysics Data System (ADS)

Bahrampour, Anvar; Barkhoda, Wafa; Azami, Bahram Zahir

Nowadays, concatenative method is used in most modern TTS systems to produce artificial speech. The most important challenge in this method is choosing appropriate unit for creating database. This unit must warranty smoothness and high quality speech, and also, creating database for it must reasonable and inexpensive. For example, syllable, phoneme, allophone, and, diphone are appropriate units for all-purpose systems. In this paper, we implemented three synthesis systems for Kurdish language based on syllable, allophone, and diphone and compare their quality using subjective testing.

Reading's SLiCK with New Audio Texts and Strategies.

ERIC Educational Resources Information Center

Boyle, Elizabeth A.; Washburn, Shari Gallin; Rosenberg, Michael S.; Connelly, Vincent J.; Brinckerhoff, Loring C.; Banerjee, Manju

2002-01-01

This article discusses challenges for secondary students with disabilities and alternative instructional methods that teachers of students with poor reading skills can use to convey content information effectively and efficiently. The use of audio textbooks on CD-ROMs is emphasized and the SLiCK strategy is explained as a support for the CD-ROM.…

Visual-Auditory Integration during Speech Imitation in Autism

ERIC Educational Resources Information Center

Williams, Justin H. G.; Massaro, Dominic W.; Peel, Natalie J.; Bosseler, Alexis; Suddendorf, Thomas

2004-01-01

Children with autistic spectrum disorder (ASD) may have poor audio-visual integration, possibly reflecting dysfunctional "mirror neuron" systems which have been hypothesised to be at the core of the condition. In the present study, a computer program, utilizing speech synthesizer software and a "virtual" head (Baldi), delivered speech stimuli for…

Measuring Implicit and Explicit Attitudes toward Foreign-Accented Speech

ERIC Educational Resources Information Center

Pantos, Andrew J.

2010-01-01

The purpose of this research was to investigate the nature of listeners' attitudes toward foreign-accented speech and the manner in which those attitudes are formed. This study measured 165 participants' implicit and explicit attitudes toward US- and foreign-accented audio stimuli. Implicit attitudes were measured with an audio Implicit…

Investigating an Application of Speech-to-Text Recognition: A Study on Visual Attention and Learning Behaviour

ERIC Educational Resources Information Center

Huang, Y-M.; Liu, C-J.; Shadiev, Rustam; Shen, M-H.; Hwang, W-Y.

2015-01-01

One major drawback of previous research on speech-to-text recognition (STR) is that most findings showing the effectiveness of STR for learning were based upon subjective evidence. Very few studies have used eye-tracking techniques to investigate visual attention of students on STR-generated text. Furthermore, not much attention was paid to…

A hybrid technique for speech segregation and classification using a sophisticated deep neural network

PubMed Central

Nawaz, Tabassam; Mehmood, Zahid; Rashid, Muhammad; Habib, Hafiz Adnan

2018-01-01

Recent research on speech segregation and music fingerprinting has led to improvements in speech segregation and music identification algorithms. Speech and music segregation generally involves the identification of music followed by speech segregation. However, music segregation becomes a challenging task in the presence of noise. This paper proposes a novel method of speech segregation for unlabelled stationary noisy audio signals using the deep belief network (DBN) model. The proposed method successfully segregates a music signal from noisy audio streams. A recurrent neural network (RNN)-based hidden layer segregation model is applied to remove stationary noise. Dictionary-based fisher algorithms are employed for speech classification. The proposed method is tested on three datasets (TIMIT, MIR-1K, and MusicBrainz), and the results indicate the robustness of proposed method for speech segregation. The qualitative and quantitative analysis carried out on three datasets demonstrate the efficiency of the proposed method compared to the state-of-the-art speech segregation and classification-based methods. PMID:29558485

Fifty years of progress in speech and speaker recognition

NASA Astrophysics Data System (ADS)

Furui, Sadaoki

2004-10-01

Speech and speaker recognition technology has made very significant progress in the past 50 years. The progress can be summarized by the following changes: (1) from template matching to corpus-base statistical modeling, e.g., HMM and n-grams, (2) from filter bank/spectral resonance to Cepstral features (Cepstrum + DCepstrum + DDCepstrum), (3) from heuristic time-normalization to DTW/DP matching, (4) from gdistanceh-based to likelihood-based methods, (5) from maximum likelihood to discriminative approach, e.g., MCE/GPD and MMI, (6) from isolated word to continuous speech recognition, (7) from small vocabulary to large vocabulary recognition, (8) from context-independent units to context-dependent units for recognition, (9) from clean speech to noisy/telephone speech recognition, (10) from single speaker to speaker-independent/adaptive recognition, (11) from monologue to dialogue/conversation recognition, (12) from read speech to spontaneous speech recognition, (13) from recognition to understanding, (14) from single-modality (audio signal only) to multi-modal (audio/visual) speech recognition, (15) from hardware recognizer to software recognizer, and (16) from no commercial application to many practical commercial applications. Most of these advances have taken place in both the fields of speech recognition and speaker recognition. The majority of technological changes have been directed toward the purpose of increasing robustness of recognition, including many other additional important techniques not noted above.

Audio-guided audiovisual data segmentation, indexing, and retrieval

NASA Astrophysics Data System (ADS)

Zhang, Tong; Kuo, C.-C. Jay

1998-12-01

While current approaches for video segmentation and indexing are mostly focused on visual information, audio signals may actually play a primary role in video content parsing. In this paper, we present an approach for automatic segmentation, indexing, and retrieval of audiovisual data, based on audio content analysis. The accompanying audio signal of audiovisual data is first segmented and classified into basic types, i.e., speech, music, environmental sound, and silence. This coarse-level segmentation and indexing step is based upon morphological and statistical analysis of several short-term features of the audio signals. Then, environmental sounds are classified into finer classes, such as applause, explosions, bird sounds, etc. This fine-level classification and indexing step is based upon time- frequency analysis of audio signals and the use of the hidden Markov model as the classifier. On top of this archiving scheme, an audiovisual data retrieval system is proposed. Experimental results show that the proposed approach has an accuracy rate higher than 90 percent for the coarse-level classification, and higher than 85 percent for the fine-level classification. Examples of audiovisual data segmentation and retrieval are also provided.

Cepstral domain modification of audio signals for data embedding: preliminary results

NASA Astrophysics Data System (ADS)

Gopalan, Kaliappan

2004-06-01

A method of embedding data in an audio signal using cepstral domain modification is described. Based on successful embedding in the spectral points of perceptually masked regions in each frame of speech, first the technique was extended to embedding in the log spectral domain. This extension resulted at approximately 62 bits /s of embedding with less than 2 percent of bit error rate (BER) for a clean cover speech (from the TIMIT database), and about 2.5 percent for a noisy speech (from an air traffic controller database), when all frames - including silence and transition between voiced and unvoiced segments - were used. Bit error rate increased significantly when the log spectrum in the vicinity of a formant was modified. In the next procedure, embedding by altering the mean cepstral values of two ranges of indices was studied. Tests on both a noisy utterance and a clean utterance indicated barely noticeable perceptual change in speech quality when lower range of cepstral indices - corresponding to vocal tract region - was modified in accordance with data. With an embedding capacity of approximately 62 bits/s - using one bit per each frame regardless of frame energy or type of speech - initial results showed a BER of less than 1.5 percent for a payload capacity of 208 embedded bits using the clean cover speech. BER of less than 1.3 percent resulted for the noisy host with a capacity was 316 bits. When the cepstrum was modified in the region of excitation, BER increased to over 10 percent. With quantization causing no significant problem, the technique warrants further studies with different cepstral ranges and sizes. Pitch-synchronous cepstrum modification, for example, may be more robust to attacks. In addition, cepstrum modification in regions of speech that are perceptually masked - analogous to embedding in frequency masked regions - may yield imperceptible stego audio with low BER.

Machine Learning Methods for Articulatory Data

ERIC Educational Resources Information Center

Berry, Jeffrey James

2012-01-01

Humans make use of more than just the audio signal to perceive speech. Behavioral and neurological research has shown that a person's knowledge of how speech is produced influences what is perceived. With methods for collecting articulatory data becoming more ubiquitous, methods for extracting useful information are needed to make this data…

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).

Dialogue enabling speech-to-text user assistive agent system for hearing-impaired person.

PubMed

Lee, Seongjae; Kang, Sunmee; Han, David K; Ko, Hanseok

2016-06-01

A novel approach for assisting bidirectional communication between people of normal hearing and hearing-impaired is presented. While the existing hearing-impaired assistive devices such as hearing aids and cochlear implants are vulnerable in extreme noise conditions or post-surgery side effects, the proposed concept is an alternative approach wherein spoken dialogue is achieved by means of employing a robust speech recognition technique which takes into consideration of noisy environmental factors without any attachment into human body. The proposed system is a portable device with an acoustic beamformer for directional noise reduction and capable of performing speech-to-text transcription function, which adopts a keyword spotting method. It is also equipped with an optimized user interface for hearing-impaired people, rendering intuitive and natural device usage with diverse domain contexts. The relevant experimental results confirm that the proposed interface design is feasible for realizing an effective and efficient intelligent agent for hearing-impaired.

Neural entrainment to rhythmic speech in children with developmental dyslexia

PubMed Central

Power, Alan J.; Mead, Natasha; Barnes, Lisa; Goswami, Usha

2013-01-01

A rhythmic paradigm based on repetition of the syllable “ba” was used to study auditory, visual, and audio-visual oscillatory entrainment to speech in children with and without dyslexia using EEG. Children pressed a button whenever they identified a delay in the isochronous stimulus delivery (500 ms; 2 Hz delta band rate). Response power, strength of entrainment and preferred phase of entrainment in the delta and theta frequency bands were compared between groups. The quality of stimulus representation was also measured using cross-correlation of the stimulus envelope with the neural response. The data showed a significant group difference in the preferred phase of entrainment in the delta band in response to the auditory and audio-visual stimulus streams. A different preferred phase has significant implications for the quality of speech information that is encoded neurally, as it implies enhanced neuronal processing (phase alignment) at less informative temporal points in the incoming signal. Consistent with this possibility, the cross-correlogram analysis revealed superior stimulus representation by the control children, who showed a trend for larger peak r-values and significantly later lags in peak r-values compared to participants with dyslexia. Significant relationships between both peak r-values and peak lags were found with behavioral measures of reading. The data indicate that the auditory temporal reference frame for speech processing is atypical in developmental dyslexia, with low frequency (delta) oscillations entraining to a different phase of the rhythmic syllabic input. This would affect the quality of encoding of speech, and could underlie the cognitive impairments in phonological representation that are the behavioral hallmark of this developmental disorder across languages. PMID:24376407

Audio-visual speech perception in prelingually deafened Japanese children following sequential bilateral cochlear implantation.

PubMed

Yamamoto, Ryosuke; Naito, Yasushi; Tona, Risa; Moroto, Saburo; Tamaya, Rinko; Fujiwara, Keizo; Shinohara, Shogo; Takebayashi, Shinji; Kikuchi, Masahiro; Michida, Tetsuhiko

2017-11-01

An effect of audio-visual (AV) integration is observed when the auditory and visual stimuli are incongruent (the McGurk effect). In general, AV integration is helpful especially in subjects wearing hearing aids or cochlear implants (CIs). However, the influence of AV integration on spoken word recognition in individuals with bilateral CIs (Bi-CIs) has not been fully investigated so far. In this study, we investigated AV integration in children with Bi-CIs. The study sample included thirty one prelingually deafened children who underwent sequential bilateral cochlear implantation. We assessed their responses to congruent and incongruent AV stimuli with three CI-listening modes: only the 1st CI, only the 2nd CI, and Bi-CIs. The responses were assessed in the whole group as well as in two sub-groups: a proficient group (syllable intelligibility ≥80% with the 1st CI) and a non-proficient group (syllable intelligibility < 80% with the 1st CI). We found evidence of the McGurk effect in each of the three CI-listening modes. AV integration responses were observed in a subset of incongruent AV stimuli, and the patterns observed with the 1st CI and with Bi-CIs were similar. In the proficient group, the responses with the 2nd CI were not significantly different from those with the 1st CI whereas in the non-proficient group the responses with the 2nd CI were driven by visual stimuli more than those with the 1st CI. Our results suggested that prelingually deafened Japanese children who underwent sequential bilateral cochlear implantation exhibit AV integration abilities, both in monaural listening as well as in binaural listening. We also observed a higher influence of visual stimuli on speech perception with the 2nd CI in the non-proficient group, suggesting that Bi-CIs listeners with poorer speech recognition rely on visual information more compared to the proficient subjects to compensate for poorer auditory input. Nevertheless, poorer quality auditory input with the 2nd

Perceptual congruency of audio-visual speech affects ventriloquism with bilateral visual stimuli.

PubMed

Kanaya, Shoko; Yokosawa, Kazuhiko

2011-02-01

Many studies on multisensory processes have focused on performance in simplified experimental situations, with a single stimulus in each sensory modality. However, these results cannot necessarily be applied to explain our perceptual behavior in natural scenes where various signals exist within one sensory modality. We investigated the role of audio-visual syllable congruency on participants' auditory localization bias or the ventriloquism effect using spoken utterances and two videos of a talking face. Salience of facial movements was also manipulated. Results indicated that more salient visual utterances attracted participants' auditory localization. Congruent pairing of audio-visual utterances elicited greater localization bias than incongruent pairing, while previous studies have reported little dependency on the reality of stimuli in ventriloquism. Moreover, audio-visual illusory congruency, owing to the McGurk effect, caused substantial visual interference on auditory localization. Multisensory performance appears more flexible and adaptive in this complex environment than in previous studies.

Toward diagnostic and phenotype markers for genetically transmitted speech delay.

PubMed

Shriberg, Lawrence D; Lewis, Barbara A; Tomblin, J Bruce; McSweeny, Jane L; Karlsson, Heather B; Scheer, Alison R

2005-08-01

Converging evidence supports the hypothesis that the most common subtype of childhood speech sound disorder (SSD) of currently unknown origin is genetically transmitted. We report the first findings toward a set of diagnostic markers to differentiate this proposed etiological subtype (provisionally termed speech delay-genetic) from other proposed subtypes of SSD of unknown origin. Conversational speech samples from 72 preschool children with speech delay of unknown origin from 3 research centers were selected from an audio archive. Participants differed on the number of biological, nuclear family members (0 or 2+) classified as positive for current and/or prior speech-language disorder. Although participants in the 2 groups were found to have similar speech competence, as indexed by their Percentage of Consonants Correct scores, their speech error patterns differed significantly in 3 ways. Compared with children who may have reduced genetic load for speech delay (no affected nuclear family members), children with possibly higher genetic load (2+ affected members) had (a) a significantly higher proportion of relative omission errors on the Late-8 consonants; (b) a significantly lower proportion of relative distortion errors on these consonants, particularly on the sibilant fricatives /s/, /z/, and //; and (c) a significantly lower proportion of backed /s/ distortions, as assessed by both perceptual and acoustic methods. Machine learning routines identified a 3-part classification rule that included differential weightings of these variables. The classification rule had diagnostic accuracy value of 0.83 (95% confidence limits = 0.74-0.92), with positive and negative likelihood ratios of 9.6 (95% confidence limits = 3.1-29.9) and 0.40 (95% confidence limits = 0.24-0.68), respectively. The diagnostic accuracy findings are viewed as promising. The error pattern for this proposed subtype of SSD is viewed as consistent with the cognitive-linguistic processing deficits

The Downside of Greater Lexical Influences: Selectively Poorer Speech Perception in Noise

PubMed Central

Xie, Zilong; Tessmer, Rachel; Chandrasekaran, Bharath

2017-01-01

Purpose Although lexical information influences phoneme perception, the extent to which reliance on lexical information enhances speech processing in challenging listening environments is unclear. We examined the extent to which individual differences in lexical influences on phonemic processing impact speech processing in maskers containing varying degrees of linguistic information (2-talker babble or pink noise). Method Twenty-nine monolingual English speakers were instructed to ignore the lexical status of spoken syllables (e.g., gift vs. kift) and to only categorize the initial phonemes (/g/ vs. /k/). The same participants then performed speech recognition tasks in the presence of 2-talker babble or pink noise in audio-only and audiovisual conditions. Results Individuals who demonstrated greater lexical influences on phonemic processing experienced greater speech processing difficulties in 2-talker babble than in pink noise. These selective difficulties were present across audio-only and audiovisual conditions. Conclusion Individuals with greater reliance on lexical processes during speech perception exhibit impaired speech recognition in listening conditions in which competing talkers introduce audible linguistic interferences. Future studies should examine the locus of lexical influences/interferences on phonemic processing and speech-in-speech processing. PMID:28586824

Machine printed text and handwriting identification in noisy document images.

PubMed

Zheng, Yefeng; Li, Huiping; Doermann, David

2004-03-01

In this paper, we address the problem of the identification of text in noisy document images. We are especially focused on segmenting and identifying between handwriting and machine printed text because: 1) Handwriting in a document often indicates corrections, additions, or other supplemental information that should be treated differently from the main content and 2) the segmentation and recognition techniques requested for machine printed and handwritten text are significantly different. A novel aspect of our approach is that we treat noise as a separate class and model noise based on selected features. Trained Fisher classifiers are used to identify machine printed text and handwriting from noise and we further exploit context to refine the classification. A Markov Random Field-based (MRF) approach is used to model the geometrical structure of the printed text, handwriting, and noise to rectify misclassifications. Experimental results show that our approach is robust and can significantly improve page segmentation in noisy document collections.

Investigating the Effectiveness of Speech-To-Text Recognition Applications on Learning Performance, Attention, and Meditation

ERIC Educational Resources Information Center

Shadiev, Rustam; Huang, Yueh-Min; Hwang, Jan-Pan

2017-01-01

In this study, the effectiveness of the application of speech-to-text recognition (STR) technology on enhancing learning and concentration in a calm state of mind, hereafter referred to as meditation (An intentional and self-regulated focusing of attention in order to relax and calm the mind), was investigated. This effectiveness was further…

Using Text-to-Speech Reading Support for an Adult with Mild Aphasia and Cognitive Impairment

ERIC Educational Resources Information Center

Harvey, Judy; Hux, Karen; Snell, Jeffry

2013-01-01

This single case study served to examine text-to-speech (TTS) effects on reading rate and comprehension in an individual with mild aphasia and cognitive impairment. Findings showed faster reading, given TTS presented at a normal speaking rate, but no significant comprehension changes. TTS may support reading in people with aphasia when time…

Language/Culture Modulates Brain and Gaze Processes in Audiovisual Speech Perception.

PubMed

Hisanaga, Satoko; Sekiyama, Kaoru; Igasaki, Tomohiko; Murayama, Nobuki

2016-10-13

Several behavioural studies have shown that the interplay between voice and face information in audiovisual speech perception is not universal. Native English speakers (ESs) are influenced by visual mouth movement to a greater degree than native Japanese speakers (JSs) when listening to speech. However, the biological basis of these group differences is unknown. Here, we demonstrate the time-varying processes of group differences in terms of event-related brain potentials (ERP) and eye gaze for audiovisual and audio-only speech perception. On a behavioural level, while congruent mouth movement shortened the ESs' response time for speech perception, the opposite effect was observed in JSs. Eye-tracking data revealed a gaze bias to the mouth for the ESs but not the JSs, especially before the audio onset. Additionally, the ERP P2 amplitude indicated that ESs processed multisensory speech more efficiently than auditory-only speech; however, the JSs exhibited the opposite pattern. Taken together, the ESs' early visual attention to the mouth was likely to promote phonetic anticipation, which was not the case for the JSs. These results clearly indicate the impact of language and/or culture on multisensory speech processing, suggesting that linguistic/cultural experiences lead to the development of unique neural systems for audiovisual speech perception.

Contributions of speech science to the technology of man-machine voice interactions

NASA Technical Reports Server (NTRS)

Lea, Wayne A.

1977-01-01

Research in speech understanding was reviewed. Plans which include prosodics research, phonological rules for speech understanding systems, and continued interdisciplinary phonetics research are discussed. Improved acoustic phonetic analysis capabilities in speech recognizers are suggested.

Intentional Voice Command Detection for Trigger-Free Speech Interface

NASA Astrophysics Data System (ADS)

Obuchi, Yasunari; Sumiyoshi, Takashi

In this paper we introduce a new framework of audio processing, which is essential to achieve a trigger-free speech interface for home appliances. If the speech interface works continually in real environments, it must extract occasional voice commands and reject everything else. It is extremely important to reduce the number of false alarms because the number of irrelevant inputs is much larger than the number of voice commands even for heavy users of appliances. The framework, called Intentional Voice Command Detection, is based on voice activity detection, but enhanced by various speech/audio processing techniques such as emotion recognition. The effectiveness of the proposed framework is evaluated using a newly-collected large-scale corpus. The advantages of combining various features were tested and confirmed, and the simple LDA-based classifier demonstrated acceptable performance. The effectiveness of various methods of user adaptation is also discussed.

Introduction to Human Services, Chapter III. Video Script Package, Text, and Audio Script Package.

ERIC Educational Resources Information Center

Miami-Dade Community Coll., FL.

Video, textual, and audio components of the third module of a multi-media, introductory course on Human Services are presented. The module packages, developed at Miami-Dade Community College, deal with technology, social change, and problem dependencies. A video cassette script is first provided that explores the "traditional,""inner," and "other…

Audio-Visual Temporal Recalibration Can be Constrained by Content Cues Regardless of Spatial Overlap.

PubMed

Roseboom, Warrick; Kawabe, Takahiro; Nishida, Shin'ya

2013-01-01

It has now been well established that the point of subjective synchrony for audio and visual events can be shifted following exposure to asynchronous audio-visual presentations, an effect often referred to as temporal recalibration. Recently it was further demonstrated that it is possible to concurrently maintain two such recalibrated estimates of audio-visual temporal synchrony. However, it remains unclear precisely what defines a given audio-visual pair such that it is possible to maintain a temporal relationship distinct from other pairs. It has been suggested that spatial separation of the different audio-visual pairs is necessary to achieve multiple distinct audio-visual synchrony estimates. Here we investigated if this is necessarily true. Specifically, we examined whether it is possible to obtain two distinct temporal recalibrations for stimuli that differed only in featural content. Using both complex (audio visual speech; see Experiment 1) and simple stimuli (high and low pitch audio matched with either vertically or horizontally oriented Gabors; see Experiment 2) we found concurrent, and opposite, recalibrations despite there being no spatial difference in presentation location at any point throughout the experiment. This result supports the notion that the content of an audio-visual pair alone can be used to constrain distinct audio-visual synchrony estimates regardless of spatial overlap.

Audio-Visual Temporal Recalibration Can be Constrained by Content Cues Regardless of Spatial Overlap

PubMed Central

Roseboom, Warrick; Kawabe, Takahiro; Nishida, Shin’Ya

2013-01-01

It has now been well established that the point of subjective synchrony for audio and visual events can be shifted following exposure to asynchronous audio-visual presentations, an effect often referred to as temporal recalibration. Recently it was further demonstrated that it is possible to concurrently maintain two such recalibrated estimates of audio-visual temporal synchrony. However, it remains unclear precisely what defines a given audio-visual pair such that it is possible to maintain a temporal relationship distinct from other pairs. It has been suggested that spatial separation of the different audio-visual pairs is necessary to achieve multiple distinct audio-visual synchrony estimates. Here we investigated if this is necessarily true. Specifically, we examined whether it is possible to obtain two distinct temporal recalibrations for stimuli that differed only in featural content. Using both complex (audio visual speech; see Experiment 1) and simple stimuli (high and low pitch audio matched with either vertically or horizontally oriented Gabors; see Experiment 2) we found concurrent, and opposite, recalibrations despite there being no spatial difference in presentation location at any point throughout the experiment. This result supports the notion that the content of an audio-visual pair alone can be used to constrain distinct audio-visual synchrony estimates regardless of spatial overlap. PMID:23658549

The Role of Speech Prosody and Text Reading Prosody in Children's Reading Comprehension

ERIC Educational Resources Information Center

Veenendaal, Nathalie J.; Groen, Margriet A.; Verhoeven, Ludo

2014-01-01

Background: Text reading prosody has been associated with reading comprehension. However, text reading prosody is a reading-dependent measure that relies heavily on decoding skills. Investigation of the contribution of speech prosody--which is independent from reading skills--in addition to text reading prosody, to reading comprehension could…

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…

Comparison of Acoustic and Kinematic Approaches to Measuring Utterance-Level Speech Variability

ERIC Educational Resources Information Center

Howell, Peter; Anderson, Andrew J.; Bartrip, Jon; Bailey, Eleanor

2009-01-01

Purpose: The spatiotemporal index (STI) is one measure of variability. As currently implemented, kinematic data are used, requiring equipment that cannot be used with some patient groups or in scanners. An experiment is reported that addressed whether STI can be extended to an audio measure of sound pressure of the speech envelope over time that…

Visual contribution to the multistable perception of speech.

PubMed

Sato, Marc; Basirat, Anahita; Schwartz, Jean-Luc

2007-11-01

The multistable perception of speech, or verbal transformation effect, refers to perceptual changes experienced while listening to a speech form that is repeated rapidly and continuously. In order to test whether visual information from the speaker's articulatory gestures may modify the emergence and stability of verbal auditory percepts, subjects were instructed to report any perceptual changes during unimodal, audiovisual, and incongruent audiovisual presentations of distinct repeated syllables. In a first experiment, the perceptual stability of reported auditory percepts was significantly modulated by the modality of presentation. In a second experiment, when audiovisual stimuli consisting of a stable audio track dubbed with a video track that alternated between congruent and incongruent stimuli were presented, a strong correlation between the timing of perceptual transitions and the timing of video switches was found. Finally, a third experiment showed that the vocal tract opening onset event provided by the visual input could play the role of a bootstrap mechanism in the search for transformations. Altogether, these results demonstrate the capacity of visual information to control the multistable perception of speech in its phonetic content and temporal course. The verbal transformation effect thus provides a useful experimental paradigm to explore audiovisual interactions in speech perception.

"Look who's talking!" Gaze Patterns for Implicit and Explicit Audio-Visual Speech Synchrony Detection in Children With High-Functioning Autism.

PubMed

Grossman, Ruth B; Steinhart, Erin; Mitchell, Teresa; McIlvane, William

2015-06-01

Conversation requires integration of information from faces and voices to fully understand the speaker's message. To detect auditory-visual asynchrony of speech, listeners must integrate visual movements of the face, particularly the mouth, with auditory speech information. Individuals with autism spectrum disorder may be less successful at such multisensory integration, despite their demonstrated preference for looking at the mouth region of a speaker. We showed participants (individuals with and without high-functioning autism (HFA) aged 8-19) a split-screen video of two identical individuals speaking side by side. Only one of the speakers was in synchrony with the corresponding audio track and synchrony switched between the two speakers every few seconds. Participants were asked to watch the video without further instructions (implicit condition) or to specifically watch the in-synch speaker (explicit condition). We recorded which part of the screen and face their eyes targeted. Both groups looked at the in-synch video significantly more with explicit instructions. However, participants with HFA looked at the in-synch video less than typically developing (TD) peers and did not increase their gaze time as much as TD participants in the explicit task. Importantly, the HFA group looked significantly less at the mouth than their TD peers, and significantly more at non-face regions of the image. There were no between-group differences for eye-directed gaze. Overall, individuals with HFA spend less time looking at the crucially important mouth region of the face during auditory-visual speech integration, which is maladaptive gaze behavior for this type of task. © 2015 International Society for Autism Research, Wiley Periodicals, Inc.

Computational validation of the motor contribution to speech perception.

PubMed

Badino, Leonardo; D'Ausilio, Alessandro; Fadiga, Luciano; Metta, Giorgio

2014-07-01

Action perception and recognition are core abilities fundamental for human social interaction. A parieto-frontal network (the mirror neuron system) matches visually presented biological motion information onto observers' motor representations. This process of matching the actions of others onto our own sensorimotor repertoire is thought to be important for action recognition, providing a non-mediated "motor perception" based on a bidirectional flow of information along the mirror parieto-frontal circuits. State-of-the-art machine learning strategies for hand action identification have shown better performances when sensorimotor data, as opposed to visual information only, are available during learning. As speech is a particular type of action (with acoustic targets), it is expected to activate a mirror neuron mechanism. Indeed, in speech perception, motor centers have been shown to be causally involved in the discrimination of speech sounds. In this paper, we review recent neurophysiological and machine learning-based studies showing (a) the specific contribution of the motor system to speech perception and (b) that automatic phone recognition is significantly improved when motor data are used during training of classifiers (as opposed to learning from purely auditory data). Copyright © 2014 Cognitive Science Society, Inc.

Reading Machines for Blind People.

ERIC Educational Resources Information Center

Fender, Derek H.

1983-01-01

Ten stages of developing reading machines for blind people are analyzed: handling of text material; optics; electro-optics; pattern recognition; character recognition; storage; speech synthesizers; browsing and place finding; computer indexing; and other sources of input. Cost considerations of the final product are emphasized. (CL)

Mandarin Visual Speech Information

ERIC Educational Resources Information Center

Chen, Trevor H.

2010-01-01

While the auditory-only aspects of Mandarin speech are heavily-researched and well-known in the field, this dissertation addresses its lesser-known aspects: The visual and audio-visual perception of Mandarin segmental information and lexical-tone information. Chapter II of this dissertation focuses on the audiovisual perception of Mandarin…

Machine learning based sample extraction for automatic speech recognition using dialectal Assamese speech.

PubMed

Agarwalla, Swapna; Sarma, Kandarpa Kumar

2016-06-01

Automatic Speaker Recognition (ASR) and related issues are continuously evolving as inseparable elements of Human Computer Interaction (HCI). With assimilation of emerging concepts like big data and Internet of Things (IoT) as extended elements of HCI, ASR techniques are found to be passing through a paradigm shift. Oflate, learning based techniques have started to receive greater attention from research communities related to ASR owing to the fact that former possess natural ability to mimic biological behavior and that way aids ASR modeling and processing. The current learning based ASR techniques are found to be evolving further with incorporation of big data, IoT like concepts. Here, in this paper, we report certain approaches based on machine learning (ML) used for extraction of relevant samples from big data space and apply them for ASR using certain soft computing techniques for Assamese speech with dialectal variations. A class of ML techniques comprising of the basic Artificial Neural Network (ANN) in feedforward (FF) and Deep Neural Network (DNN) forms using raw speech, extracted features and frequency domain forms are considered. The Multi Layer Perceptron (MLP) is configured with inputs in several forms to learn class information obtained using clustering and manual labeling. DNNs are also used to extract specific sentence types. Initially, from a large storage, relevant samples are selected and assimilated. Next, a few conventional methods are used for feature extraction of a few selected types. The features comprise of both spectral and prosodic types. These are applied to Recurrent Neural Network (RNN) and Fully Focused Time Delay Neural Network (FFTDNN) structures to evaluate their performance in recognizing mood, dialect, speaker and gender variations in dialectal Assamese speech. The system is tested under several background noise conditions by considering the recognition rates (obtained using confusion matrices and manually) and computation time

Speech enhancement on smartphone voice recording

NASA Astrophysics Data System (ADS)

Tris Atmaja, Bagus; Nur Farid, Mifta; Arifianto, Dhany

2016-11-01

Speech enhancement is challenging task in audio signal processing to enhance the quality of targeted speech signal while suppress other noises. In the beginning, the speech enhancement algorithm growth rapidly from spectral subtraction, Wiener filtering, spectral amplitude MMSE estimator to Non-negative Matrix Factorization (NMF). Smartphone as revolutionary device now is being used in all aspect of life including journalism; personally and professionally. Although many smartphones have two microphones (main and rear) the only main microphone is widely used for voice recording. This is why the NMF algorithm widely used for this purpose of speech enhancement. This paper evaluate speech enhancement on smartphone voice recording by using some algorithms mentioned previously. We also extend the NMF algorithm to Kulback-Leibler NMF with supervised separation. The last algorithm shows improved result compared to others by spectrogram and PESQ score evaluation.

New Measures of Masked Text Recognition in Relation to Speech-in-Noise Perception and Their Associations with Age and Cognitive Abilities

ERIC Educational Resources Information Center

Besser, Jana; Zekveld, Adriana A.; Kramer, Sophia E.; Ronnberg, Jerker; Festen, Joost M.

2012-01-01

Purpose: In this research, the authors aimed to increase the analogy between Text Reception Threshold (TRT; Zekveld, George, Kramer, Goverts, & Houtgast, 2007) and Speech Reception Threshold (SRT; Plomp & Mimpen, 1979) and to examine the TRT's value in estimating cognitive abilities that are important for speech comprehension in noise. Method: The…

Emotion to emotion speech conversion in phoneme level

NASA Astrophysics Data System (ADS)

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

2004-10-01

Having an ability to synthesize emotional speech can make human-machine interaction more natural in spoken dialogue management. This study investigates the effectiveness of prosodic and spectral modification in phoneme level on emotion-to-emotion speech conversion. The prosody modification is performed with the TD-PSOLA algorithm (Moulines and Charpentier, 1990). We also transform the spectral envelopes of source phonemes to match those of target phonemes using LPC-based spectral transformation approach (Kain, 2001). Prosodic speech parameters (F0, duration, and energy) for target phonemes are estimated from the statistics obtained from the analysis of an emotional speech database of happy, angry, sad, and neutral utterances collected from actors. Listening experiments conducted with native American English speakers indicate that the modification of prosody only or spectrum only is not sufficient to elicit targeted emotions. The simultaneous modification of both prosody and spectrum results in higher acceptance rates of target emotions, suggesting that not only modeling speech prosody but also modeling spectral patterns that reflect underlying speech articulations are equally important to synthesize emotional speech with good quality. We are investigating suprasegmental level modifications for further improvement in speech quality and expressiveness.

An Introduction to Topic Modeling as an Unsupervised Machine Learning Way to Organize Text Information

ERIC Educational Resources Information Center

Snyder, Robin M.

2015-01-01

The field of topic modeling has become increasingly important over the past few years. Topic modeling is an unsupervised machine learning way to organize text (or image or DNA, etc.) information such that related pieces of text can be identified. This paper/session will present/discuss the current state of topic modeling, why it is important, and…

Speech Synthesis Applied to Language Teaching.

ERIC Educational Resources Information Center

Sherwood, Bruce

1981-01-01

The experimental addition of speech output to computer-based Esperanto lessons using speech synthesized from text is described. Because of Esperanto's phonetic spelling and simple rhythm, it is particularly easy to describe the mechanisms of Esperanto synthesis. Attention is directed to how the text-to-speech conversion is performed and the ways…

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.

Expanding Audio Access to Mathematics Expressions by Students with Visual Impairments via MathML. Research Report. ETS RR-17-13

ERIC Educational Resources Information Center

Frankel, Lois; Brownstein, Beth; Soiffer, Neil

2017-01-01

This report describes the pilot conducted in the final phase of a project, Expanding Audio Access to Mathematics Expressions by Students With Visual Impairments via MathML, to provide easy-to-use tools for authoring and rendering secondary-school algebra-level math expressions in synthesized speech that is useful for students with blindness or low…

Audio-vocal interaction in single neurons of the monkey ventrolateral prefrontal cortex.

PubMed

Hage, Steffen R; Nieder, Andreas

2015-05-06

Complex audio-vocal integration systems depend on a strong interconnection between the auditory and the vocal motor system. To gain cognitive control over audio-vocal interaction during vocal motor control, the PFC needs to be involved. Neurons in the ventrolateral PFC (VLPFC) have been shown to separately encode the sensory perceptions and motor production of vocalizations. It is unknown, however, whether single neurons in the PFC reflect audio-vocal interactions. We therefore recorded single-unit activity in the VLPFC of rhesus monkeys (Macaca mulatta) while they produced vocalizations on command or passively listened to monkey calls. We found that 12% of randomly selected neurons in VLPFC modulated their discharge rate in response to acoustic stimulation with species-specific calls. Almost three-fourths of these auditory neurons showed an additional modulation of their discharge rates either before and/or during the monkeys' motor production of vocalization. Based on these audio-vocal interactions, the VLPFC might be well positioned to combine higher order auditory processing with cognitive control of the vocal motor output. Such audio-vocal integration processes in the VLPFC might constitute a precursor for the evolution of complex learned audio-vocal integration systems, ultimately giving rise to human speech. Copyright © 2015 the authors 0270-6474/15/357030-11$15.00/0.

Interactions between Text Chat and Audio Modalities for L2 Communication and Feedback in the Synthetic World "Second Life"

ERIC Educational Resources Information Center

Wigham, Ciara R.; Chanier, Thierry

2015-01-01

This paper reports on a study of the interactions between text chat and audio modalities in L2 communication in a synthetic (virtual) world and observes whether the text chat modality was used for corrective feedback and the characteristics of the latter. This is examined within the context of a hybrid content and language integrated learning…

Hearing impaired speech in noisy classrooms

NASA Astrophysics Data System (ADS)

Shahin, Kimary; McKellin, William H.; Jamieson, Janet; Hodgson, Murray; Pichora-Fuller, M. Kathleen

2005-04-01

Noisy classrooms have been shown to induce among students patterns of interaction similar to those used by hearing impaired people [W. H. McKellin et al., GURT (2003)]. In this research, the speech of children in a noisy classroom setting was investigated to determine if noisy classrooms have an effect on students' speech. Audio recordings were made of the speech of students during group work in their regular classrooms (grades 1-7), and of the speech of the same students in a sound booth. Noise level readings in the classrooms were also recorded. Each student's noisy and quiet environment speech samples were acoustically analyzed for prosodic and segmental properties (f0, pitch range, pitch variation, phoneme duration, vowel formants), and compared. The analysis showed that the students' speech in the noisy classrooms had characteristics of the speech of hearing-impaired persons [e.g., R. O'Halpin, Clin. Ling. and Phon. 15, 529-550 (2001)]. Some educational implications of our findings were identified. [Work supported by the Peter Wall Institute for Advanced Studies, University of British Columbia.

Automatic detection of obstructive sleep apnea using speech signals.

PubMed

Goldshtein, Evgenia; Tarasiuk, Ariel; Zigel, Yaniv

2011-05-01

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

Speech Intelligibility and Psychosocial Functioning in Deaf Children and Teens with Cochlear Implants

ERIC Educational Resources Information Center

Freeman, Valerie; Pisoni, David B.; Kronenberger, William G.; Castellanos, Irina

2017-01-01

Deaf children with cochlear implants (CIs) are at risk for psychosocial adjustment problems, possibly due to delayed speech-language skills. This study investigated associations between a core component of spoken-language ability--speech intelligibility--and the psychosocial development of prelingually deaf CI users. Audio-transcription measures…

Automated Desensitization for the Clinical Treatment of Speech Anxiety

ERIC Educational Resources Information Center

McManus, Marianne; Lohr, James

1976-01-01

A self-guided, audio-tape, desensitization treatment procedure, using standard cassette recorders in a counseling service office, is an effective means for modifying self-report of speech anxiety. (MB)

Crossmodal and incremental perception of audiovisual cues to emotional speech.

PubMed

Barkhuysen, Pashiera; Krahmer, Emiel; Swerts, Marc

2010-01-01

In this article we report on two experiments about the perception of audiovisual cues to emotional speech. The article addresses two questions: 1) how do visual cues from a speaker's face to emotion relate to auditory cues, and (2) what is the recognition speed for various facial cues to emotion? Both experiments reported below are based on tests with video clips of emotional utterances collected via a variant of the well-known Velten method. More specifically, we recorded speakers who displayed positive or negative emotions, which were congruent or incongruent with the (emotional) lexical content of the uttered sentence. In order to test this, we conducted two experiments. The first experiment is a perception experiment in which Czech participants, who do not speak Dutch, rate the perceived emotional state of Dutch speakers in a bimodal (audiovisual) or a unimodal (audio- or vision-only) condition. It was found that incongruent emotional speech leads to significantly more extreme perceived emotion scores than congruent emotional speech, where the difference between congruent and incongruent emotional speech is larger for the negative than for the positive conditions. Interestingly, the largest overall differences between congruent and incongruent emotions were found for the audio-only condition, which suggests that posing an incongruent emotion has a particularly strong effect on the spoken realization of emotions. The second experiment uses a gating paradigm to test the recognition speed for various emotional expressions from a speaker's face. In this experiment participants were presented with the same clips as experiment I, but this time presented vision-only. The clips were shown in successive segments (gates) of increasing duration. Results show that participants are surprisingly accurate in their recognition of the various emotions, as they already reach high recognition scores in the first gate (after only 160 ms). Interestingly, the recognition scores

Speech-Act and Text-Act Theory: "Theme-ing" in Freshman Composition.

ERIC Educational Resources Information Center

Horner, Winifred B.

In contrast to a speech-act theory that is limited by a simple speaker/hearer relationship, a text-act theory of written language allows for the historical or personal context of a writer and reader, both in the written work itself and in the act of reading. This theory can be applied to theme writing, essay examinations, and revision in the…

Using ultrasound visual biofeedback to treat persistent primary speech sound disorders.

PubMed

Cleland, Joanne; Scobbie, James M; Wrench, Alan A

2015-01-01

Growing evidence suggests that speech intervention using visual biofeedback may benefit people for whom visual skills are stronger than auditory skills (for example, the hearing-impaired population), especially when the target articulation is hard to describe or see. Diagnostic ultrasound can be used to image the tongue and has recently become more compact and affordable leading to renewed interest in it as a practical, non-invasive visual biofeedback tool. In this study, we evaluate its effectiveness in treating children with persistent speech sound disorders that have been unresponsive to traditional therapy approaches. A case series of seven different children (aged 6-11) with persistent speech sound disorders were evaluated. For each child, high-speed ultrasound (121 fps), audio and lip video recordings were made while probing each child's specific errors at five different time points (before, during and after intervention). After intervention, all the children made significant progress on targeted segments, evidenced by both perceptual measures and changes in tongue-shape.

Objective voice and speech analysis of persons with chronic hoarseness by prosodic analysis of speech samples.

PubMed

Haderlein, Tino; Döllinger, Michael; Matoušek, Václav; Nöth, Elmar

2016-10-01

Automatic voice assessment is often performed using sustained vowels. In contrast, speech analysis of read-out texts can be applied to voice and speech assessment. Automatic speech recognition and prosodic analysis were used to find regression formulae between automatic and perceptual assessment of four voice and four speech criteria. The regression was trained with 21 men and 62 women (average age 49.2 years) and tested with another set of 24 men and 49 women (48.3 years), all suffering from chronic hoarseness. They read the text 'Der Nordwind und die Sonne' ('The North Wind and the Sun'). Five voice and speech therapists evaluated the data on 5-point Likert scales. Ten prosodic and recognition accuracy measures (features) were identified which describe all the examined criteria. Inter-rater correlation within the expert group was between r = 0.63 for the criterion 'match of breath and sense units' and r = 0.87 for the overall voice quality. Human-machine correlation was between r = 0.40 for the match of breath and sense units and r = 0.82 for intelligibility. The perceptual ratings of different criteria were highly correlated with each other. Likewise, the feature sets modeling the criteria were very similar. The automatic method is suitable for assessing chronic hoarseness in general and for subgroups of functional and organic dysphonia. In its current version, it is almost as reliable as a randomly picked rater from a group of voice and speech therapists.

Audio feature extraction using probability distribution function

NASA Astrophysics Data System (ADS)

Suhaib, A.; Wan, Khairunizam; Aziz, Azri A.; Hazry, D.; Razlan, Zuradzman M.; Shahriman A., B.

2015-05-01

Voice recognition has been one of the popular applications in robotic field. It is also known to be recently used for biometric and multimedia information retrieval system. This technology is attained from successive research on audio feature extraction analysis. Probability Distribution Function (PDF) is a statistical method which is usually used as one of the processes in complex feature extraction methods such as GMM and PCA. In this paper, a new method for audio feature extraction is proposed which is by using only PDF as a feature extraction method itself for speech analysis purpose. Certain pre-processing techniques are performed in prior to the proposed feature extraction method. Subsequently, the PDF result values for each frame of sampled voice signals obtained from certain numbers of individuals are plotted. From the experimental results obtained, it can be seen visually from the plotted data that each individuals' voice has comparable PDF values and shapes.

Wavelet-based audio embedding and audio/video compression

NASA Astrophysics Data System (ADS)

Mendenhall, Michael J.; Claypoole, Roger L., Jr.

2001-12-01

Watermarking, traditionally used for copyright protection, is used in a new and exciting way. An efficient wavelet-based watermarking technique embeds audio information into a video signal. Several effective compression techniques are applied to compress the resulting audio/video signal in an embedded fashion. This wavelet-based compression algorithm incorporates bit-plane coding, index coding, and Huffman coding. To demonstrate the potential of this audio embedding and audio/video compression algorithm, we embed an audio signal into a video signal and then compress. Results show that overall compression rates of 15:1 can be achieved. The video signal is reconstructed with a median PSNR of nearly 33 dB. Finally, the audio signal is extracted from the compressed audio/video signal without error.

78 FR 49693 - Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services...

Federal Register 2010, 2011, 2012, 2013, 2014

2013-08-15

...] Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services...: This is a summary of the Commission's Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech...), Internet Protocol Relay (IP Relay), and IP captioned telephone service (IP CTS) as compensable forms of TRS...

Non-fluent speech following stroke is caused by impaired efference copy.

PubMed

Feenaughty, Lynda; Basilakos, Alexandra; Bonilha, Leonardo; den Ouden, Dirk-Bart; Rorden, Chris; Stark, Brielle; Fridriksson, Julius

2017-09-01

Efference copy is a cognitive mechanism argued to be critical for initiating and monitoring speech: however, the extent to which breakdown of efference copy mechanisms impact speech production is unclear. This study examined the best mechanistic predictors of non-fluent speech among 88 stroke survivors. Objective speech fluency measures were subjected to a principal component analysis (PCA). The primary PCA factor was then entered into a multiple stepwise linear regression analysis as the dependent variable, with a set of independent mechanistic variables. Participants' ability to mimic audio-visual speech ("speech entrainment response") was the best independent predictor of non-fluent speech. We suggest that this "speech entrainment" factor reflects integrity of internal monitoring (i.e., efference copy) of speech production, which affects speech initiation and maintenance. Results support models of normal speech production and suggest that therapy focused on speech initiation and maintenance may improve speech fluency for individuals with chronic non-fluent aphasia post stroke.

Modeling Driving Performance Using In-Vehicle Speech Data From a Naturalistic Driving Study.

PubMed

Kuo, Jonny; Charlton, Judith L; Koppel, Sjaan; Rudin-Brown, Christina M; Cross, Suzanne

2016-09-01

We aimed to (a) describe the development and application of an automated approach for processing in-vehicle speech data from a naturalistic driving study (NDS), (b) examine the influence of child passenger presence on driving performance, and (c) model this relationship using in-vehicle speech data. Parent drivers frequently engage in child-related secondary behaviors, but the impact on driving performance is unknown. Applying automated speech-processing techniques to NDS audio data would facilitate the analysis of in-vehicle driver-child interactions and their influence on driving performance. Speech activity detection and speaker diarization algorithms were applied to audio data from a Melbourne-based NDS involving 42 families. Multilevel models were developed to evaluate the effect of speech activity and the presence of child passengers on driving performance. Speech activity was significantly associated with velocity and steering angle variability. Child passenger presence alone was not associated with changes in driving performance. However, speech activity in the presence of two child passengers was associated with the most variability in driving performance. The effects of in-vehicle speech on driving performance in the presence of child passengers appear to be heterogeneous, and multiple factors may need to be considered in evaluating their impact. This goal can potentially be achieved within large-scale NDS through the automated processing of observational data, including speech. Speech-processing algorithms enable new perspectives on driving performance to be gained from existing NDS data, and variables that were once labor-intensive to process can be readily utilized in future research. © 2016, Human Factors and Ergonomics Society.

Computationally Efficient Clustering of Audio-Visual Meeting Data

NASA Astrophysics Data System (ADS)

Hung, Hayley; Friedland, Gerald; Yeo, Chuohao

This chapter presents novel computationally efficient algorithms to extract semantically meaningful acoustic and visual events related to each of the participants in a group discussion using the example of business meeting recordings. The recording setup involves relatively few audio-visual sensors, comprising a limited number of cameras and microphones. We first demonstrate computationally efficient algorithms that can identify who spoke and when, a problem in speech processing known as speaker diarization. We also extract visual activity features efficiently from MPEG4 video by taking advantage of the processing that was already done for video compression. Then, we present a method of associating the audio-visual data together so that the content of each participant can be managed individually. The methods presented in this article can be used as a principal component that enables many higher-level semantic analysis tasks needed in search, retrieval, and navigation.

Divergent neural responses to narrative speech in disorders of consciousness.

PubMed

Iotzov, Ivan; Fidali, Brian C; Petroni, Agustin; Conte, Mary M; Schiff, Nicholas D; Parra, Lucas C

2017-11-01

Clinical assessment of auditory attention in patients with disorders of consciousness is often limited by motor impairment. Here, we employ intersubject correlations among electroencephalography responses to naturalistic speech in order to assay auditory attention among patients and healthy controls. Electroencephalographic data were recorded from 20 subjects with disorders of consciousness and 14 healthy controls during of two narrative audio stimuli, presented both forwards and time-reversed. Intersubject correlation of evoked electroencephalography signals were calculated, comparing responses of both groups to those of the healthy control subjects. This analysis was performed blinded and subsequently compared to the diagnostic status of each patient based on the Coma Recovery Scale-Revised. Subjects with disorders of consciousness exhibit significantly lower intersubject correlation than healthy controls during narrative speech. Additionally, while healthy subjects had higher intersubject correlation values in forwards versus backwards presentation, neural responses did not vary significantly with the direction of playback in subjects with disorders of consciousness. Increased intersubject correlation values in the backward speech condition were noted with improving disorder of consciousness diagnosis, both in cross-sectional analysis and in a subset of patients with longitudinal data. Intersubject correlation of neural responses to narrative speech audition differentiates healthy controls from patients and appears to index clinical diagnoses in disorders of consciousness.

Audio Visual Technology and the Teaching of Foreign Languages.

ERIC Educational Resources Information Center

Halbig, Michael C.

Skills in comprehending the spoken language source are becoming increasingly important due to the audio-visual orientation of our culture. It would seem natural, therefore, to adjust the learning goals and environment accordingly. The video-cassette machine is an ideal means for creating this learning environment and developing the listening…

Comparisons of Audio and Audiovisual Measures of Stuttering Frequency and Severity in Preschool-Age Children

ERIC Educational Resources Information Center

Rousseau, Isabelle; Onslow, Mark; Packman, Ann; Jones, Mark

2008-01-01

Purpose: To determine whether measures of stuttering frequency and measures of overall stuttering severity in preschoolers differ when made from audio-only recordings compared with audiovisual recordings. Method: Four blinded speech-language pathologists who had extensive experience with preschoolers who stutter measured stuttering frequency and…

Reconstruction of audio waveforms from spike trains of artificial cochlea models

PubMed Central

Zai, Anja T.; Bhargava, Saurabh; Mesgarani, Nima; Liu, Shih-Chii

2015-01-01

Spiking cochlea models describe the analog processing and spike generation process within the biological cochlea. Reconstructing the audio input from the artificial cochlea spikes is therefore useful for understanding the fidelity of the information preserved in the spikes. The reconstruction process is challenging particularly for spikes from the mixed signal (analog/digital) integrated circuit (IC) cochleas because of multiple non-linearities in the model and the additional variance caused by random transistor mismatch. This work proposes an offline method for reconstructing the audio input from spike responses of both a particular spike-based hardware model called the AEREAR2 cochlea and an equivalent software cochlea model. This method was previously used to reconstruct the auditory stimulus based on the peri-stimulus histogram of spike responses recorded in the ferret auditory cortex. The reconstructed audio from the hardware cochlea is evaluated against an analogous software model using objective measures of speech quality and intelligibility; and further tested in a word recognition task. The reconstructed audio under low signal-to-noise (SNR) conditions (SNR < –5 dB) gives a better classification performance than the original SNR input in this word recognition task. PMID:26528113

TongueToSpeech (TTS): Wearable wireless assistive device for augmented speech.

PubMed

Marjanovic, Nicholas; Piccinini, Giacomo; Kerr, Kevin; Esmailbeigi, Hananeh

2017-07-01

Speech is an important aspect of human communication; individuals with speech impairment are unable to communicate vocally in real time. Our team has developed the TongueToSpeech (TTS) device with the goal of augmenting speech communication for the vocally impaired. The proposed device is a wearable wireless assistive device that incorporates a capacitive touch keyboard interface embedded inside a discrete retainer. This device connects to a computer, tablet or a smartphone via Bluetooth connection. The developed TTS application converts text typed by the tongue into audible speech. Our studies have concluded that an 8-contact point configuration between the tongue and the TTS device would yield the best user precision and speed performance. On average using the TTS device inside the oral cavity takes 2.5 times longer than the pointer finger using a T9 (Text on 9 keys) keyboard configuration to type the same phrase. In conclusion, we have developed a discrete noninvasive wearable device that allows the vocally impaired individuals to communicate in real time.

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

Speech vs. singing: infants choose happier sounds

PubMed Central

Corbeil, Marieve; Trehub, Sandra E.; Peretz, Isabelle

2013-01-01

Infants prefer speech to non-vocal sounds and to non-human vocalizations, and they prefer happy-sounding speech to neutral speech. They also exhibit an interest in singing, but there is little knowledge of their relative interest in speech and singing. The present study explored infants' attention to unfamiliar audio samples of speech and singing. In Experiment 1, infants 4–13 months of age were exposed to happy-sounding infant-directed speech vs. hummed lullabies by the same woman. They listened significantly longer to the speech, which had considerably greater acoustic variability and expressiveness, than to the lullabies. In Experiment 2, infants of comparable age who heard the lyrics of a Turkish children's song spoken vs. sung in a joyful/happy manner did not exhibit differential listening. Infants in Experiment 3 heard the happily sung lyrics of the Turkish children's song vs. a version that was spoken in an adult-directed or affectively neutral manner. They listened significantly longer to the sung version. Overall, happy voice quality rather than vocal mode (speech or singing) was the principal contributor to infant attention, regardless of age. PMID:23805119

Audio Spectrogram Representations for Processing with Convolutional Neural Networks

NASA Astrophysics Data System (ADS)

Wyse, L.

2017-05-01

One of the decisions that arise when designing a neural network for any application is how the data should be represented in order to be presented to, and possibly generated by, a neural network. For audio, the choice is less obvious than it seems to be for visual images, and a variety of representations have been used for different applications including the raw digitized sample stream, hand-crafted features, machine discovered features, MFCCs and variants that include deltas, and a variety of spectral representations. This paper reviews some of these representations and issues that arise, focusing particularly on spectrograms for generating audio using neural networks for style transfer.

Listeners' Perceptions of Speech and Language Disorders

ERIC Educational Resources Information Center

Allard, Emily R.; Williams, Dale F.

2008-01-01

Using semantic differential scales with nine trait pairs, 445 adults rated five audio-taped speech samples, one depicting an individual without a disorder and four portraying communication disorders. Statistical analyses indicated that the no disorder sample was rated higher with respect to the trait of employability than were the articulation,…

78 FR 49717 - Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services...

Federal Register 2010, 2011, 2012, 2013, 2014

2013-08-15

...] Speech-to-Speech and Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services... Internet Protocol (IP) Speech-to-Speech Telecommunications Relay Services; Telecommunications Relay... (IP Relay) and video relay service (VRS), the Commission should bundle national STS outreach efforts...

Virtual Acoustic Displays for Teleconferencing: Intelligibility Advantage for "Telephone Grade" Audio

NASA Technical Reports Server (NTRS)

Begault, Durand R.; Null, Cynthia H. (Technical Monitor)

1994-01-01

Speech intelligibility was evaluated using a virtual acoustic ("3-D audio") display using the method specified by ANSI. Ten subjects were evaluated with stimuli either unfiltered or low-pass filtered at 4 kHz. Results show virtual acoustic techniques are advantageous for both full-bandwidth (44.1 kHz srate) and low (8 kHz srate) bandwidth "telephone-grade" teleconferencing systems.

Zipf's Law in Short-Time Timbral Codings of Speech, Music, and Environmental Sound Signals

PubMed Central

Haro, Martín; Serrà, Joan; Herrera, Perfecto; Corral, Álvaro

2012-01-01

Timbre is a key perceptual feature that allows discrimination between different sounds. Timbral sensations are highly dependent on the temporal evolution of the power spectrum of an audio signal. In order to quantitatively characterize such sensations, the shape of the power spectrum has to be encoded in a way that preserves certain physical and perceptual properties. Therefore, it is common practice to encode short-time power spectra using psychoacoustical frequency scales. In this paper, we study and characterize the statistical properties of such encodings, here called timbral code-words. In particular, we report on rank-frequency distributions of timbral code-words extracted from 740 hours of audio coming from disparate sources such as speech, music, and environmental sounds. Analogously to text corpora, we find a heavy-tailed Zipfian distribution with exponent close to one. Importantly, this distribution is found independently of different encoding decisions and regardless of the audio source. Further analysis on the intrinsic characteristics of most and least frequent code-words reveals that the most frequent code-words tend to have a more homogeneous structure. We also find that speech and music databases have specific, distinctive code-words while, in the case of the environmental sounds, this database-specific code-words are not present. Finally, we find that a Yule-Simon process with memory provides a reasonable quantitative approximation for our data, suggesting the existence of a common simple generative mechanism for all considered sound sources. PMID:22479497

International Meeting To Discuss Audio Technology as Applied to Library Services for Blind Individuals (3rd, Toronto, Ontario, Canada, April 20-22, 1995). Volumes 1-3.

ERIC Educational Resources Information Center

Library of Congress, Washington, DC. National Library Service for the Blind and Physically Handicapped.

This three-day conference on the subject of audio technology for the production of materials for the blind, takes the court reporter approach to recording the speeches and discussions of the meeting. The result is a three volume set of complete transcripts, one volume for each day of the meeting, but continuous in form. The highlights of each…

Cortical processing during smartphone text messaging.

PubMed

Tatum, William O; DiCiaccio, Benedetto; Yelvington, Kirsten H

2016-06-01

The objective of this study was to report the EEG features of text messaging using smartphones. One hundred twenty-nine patients were prospectively evaluated during video-EEG monitoring (VEM) over 16months. A reproducible texting rhythm (TR) present during active text messaging with a smartphone was compared with passive and forced audio telephone use, thumb/finger movements, cognitive testing/calculation, scanning eye movements, and speech/language tasks in patients with and without epilepsy. Statistical significance was set at p<0.05. Twenty-seven patients with a TR were identified from a cohort of 129 (93 female, mean age: 36; range: 18-71) unselected VEM patients. Fifty-three out of 129 patients had epileptic seizures (ES), 74/129 had nonepileptic seizures (NES), and 2/129 were dual-diagnosed. A reproducible TR was present in 27/129 (20.9%) specific to text messaging (p<0.0001) and present in 28% of patients with ES and 16% of patients with NES (p=NS). The TR was absent during independent tasks and audio cellular telephone use (p<0.0001). Age, gender, epilepsy type, MRI results, and EEG lateralization in patients with focal seizures were unrelated (p=NS). Our results suggest that the TR on scalp EEG represents a novel technology-specific neurophysiological alteration of brain networks. We propose that cortical processing in the contemporary brain is uniquely activated by the use of PEDs. These findings have practical implications that could impact industry and research in nonverbal communication. Copyright © 2016 Elsevier Inc. All rights reserved.

An introduction to quantum machine learning

NASA Astrophysics Data System (ADS)

Schuld, Maria; Sinayskiy, Ilya; Petruccione, Francesco

2015-04-01

Machine learning algorithms learn a desired input-output relation from examples in order to interpret new inputs. This is important for tasks such as image and speech recognition or strategy optimisation, with growing applications in the IT industry. In the last couple of years, researchers investigated if quantum computing can help to improve classical machine learning algorithms. Ideas range from running computationally costly algorithms or their subroutines efficiently on a quantum computer to the translation of stochastic methods into the language of quantum theory. This contribution gives a systematic overview of the emerging field of quantum machine learning. It presents the approaches as well as technical details in an accessible way, and discusses the potential of a future theory of quantum learning.

Deep learning

NASA Astrophysics Data System (ADS)

Lecun, Yann; Bengio, Yoshua; Hinton, Geoffrey

2015-05-01

Deep learning allows computational models that are composed of multiple processing layers to learn representations of data with multiple levels of abstraction. These methods have dramatically improved the state-of-the-art in speech recognition, visual object recognition, object detection and many other domains such as drug discovery and genomics. Deep learning discovers intricate structure in large data sets by using the backpropagation algorithm to indicate how a machine should change its internal parameters that are used to compute the representation in each layer from the representation in the previous layer. Deep convolutional nets have brought about breakthroughs in processing images, video, speech and audio, whereas recurrent nets have shone light on sequential data such as text and speech.

Deep learning.

PubMed

LeCun, Yann; Bengio, Yoshua; Hinton, Geoffrey

2015-05-28

Deep learning allows computational models that are composed of multiple processing layers to learn representations of data with multiple levels of abstraction. These methods have dramatically improved the state-of-the-art in speech recognition, visual object recognition, object detection and many other domains such as drug discovery and genomics. Deep learning discovers intricate structure in large data sets by using the backpropagation algorithm to indicate how a machine should change its internal parameters that are used to compute the representation in each layer from the representation in the previous layer. Deep convolutional nets have brought about breakthroughs in processing images, video, speech and audio, whereas recurrent nets have shone light on sequential data such as text and speech.

Speech Processing and Recognition (SPaRe)

DTIC Science & Technology

2011-01-01

results in the areas of automatic speech recognition (ASR), speech processing, machine translation (MT), natural language processing ( NLP ), and...Processing ( NLP ), Information Retrieval (IR) 16. SECURITY CLASSIFICATION OF: UNCLASSIFED 17. LIMITATION OF ABSTRACT 18. NUMBER OF PAGES 19a. NAME...Figure 9, the IOC was only expected to provide document submission and search; automatic speech recognition (ASR) for English, Spanish, Arabic , and

Effects of Dictation, Speech to Text, and Handwriting on the Written Composition of Elementary School English Language Learners

ERIC Educational Resources Information Center

Arcon, Nina; Klein, Perry D.; Dombroski, Jill D.

2017-01-01

Previous research has shown that both dictation and speech-to-text (STT) software can increase the quality of writing for native English speakers. The purpose of this study was to investigate the effect of these modalities on the written composition and cognitive load of elementary school English language learners (ELLs). In a within-subjects…

Emerging Realities of Text-to-Speech Software for Nonnative-English-Speaking Community College Students in the Freshman Year

ERIC Educational Resources Information Center

Baker, Fiona S.

2015-01-01

This study explores the expectations and early and subsequent realities of text-to-speech software for 24 nonnative-English-speaking college students who were experiencing reading difficulties in their freshman year of college. The study took place over two semesters in one academic year (from September to June) at a community college on the…

Real-time speech-driven animation of expressive talking faces

NASA Astrophysics Data System (ADS)

Liu, Jia; You, Mingyu; Chen, Chun; Song, Mingli

2011-05-01

In this paper, we present a real-time facial animation system in which speech drives mouth movements and facial expressions synchronously. Considering five basic emotions, a hierarchical structure with an upper layer of emotion classification is established. Based on the recognized emotion label, the under-layer classification at sub-phonemic level has been modelled on the relationship between acoustic features of frames and audio labels in phonemes. Using certain constraint, the predicted emotion labels of speech are adjusted to gain the facial expression labels which are combined with sub-phonemic labels. The combinations are mapped into facial action units (FAUs), and audio-visual synchronized animation with mouth movements and facial expressions is generated by morphing between FAUs. The experimental results demonstrate that the two-layer structure succeeds in both emotion and sub-phonemic classifications, and the synthesized facial sequences reach a comparative convincing quality.

A machine learning based approach to identify protected health information in Chinese clinical text.

PubMed

Du, Liting; Xia, Chenxi; Deng, Zhaohua; Lu, Gary; Xia, Shuxu; Ma, Jingdong

2018-08-01

With the increasing application of electronic health records (EHRs) in the world, protecting private information in clinical text has drawn extensive attention from healthcare providers to researchers. De-identification, the process of identifying and removing protected health information (PHI) from clinical text, has been central to the discourse on medical privacy since 2006. While de-identification is becoming the global norm for handling medical records, there is a paucity of studies on its application on Chinese clinical text. Without efficient and effective privacy protection algorithms in place, the use of indispensable clinical information would be confined. We aimed to (i) describe the current process for PHI in China, (ii) propose a machine learning based approach to identify PHI in Chinese clinical text, and (iii) validate the effectiveness of the machine learning algorithm for de-identification in Chinese clinical text. Based on 14,719 discharge summaries from regional health centers in Ya'an City, Sichuan province, China, we built a conditional random fields (CRF) model to identify PHI in clinical text, and then used the regular expressions to optimize the recognition results of the PHI categories with fewer samples. We constructed a Chinese clinical text corpus with PHI tags through substantial manual annotation, wherein the descriptive statistics of PHI manifested its wide range and diverse categories. The evaluation showed with a high F-measure of 0.9878 that our CRF-based model had a good performance for identifying PHI in Chinese clinical text. The rapid adoption of EHR in the health sector has created an urgent need for tools that can parse patient specific information from Chinese clinical text. Our application of CRF algorithms for de-identification has shown the potential to meet this need by offering a highly accurate and flexible solution to analyzing Chinese clinical text. Copyright © 2018 Elsevier B.V. All rights reserved.

The Development of the Text Reception Threshold Test: A Visual Analogue of the Speech Reception Threshold Test

ERIC Educational Resources Information Center

Zekveld, Adriana A.; George, Erwin L. J.; Kramer, Sophia E.; Goverts, S. Theo; Houtgast, Tammo

2007-01-01

Purpose: In this study, the authors aimed to develop a visual analogue of the widely used Speech Reception Threshold (SRT; R. Plomp & A. M. Mimpen, 1979b) test. The Text Reception Threshold (TRT) test, in which visually presented sentences are masked by a bar pattern, enables the quantification of modality-aspecific variance in speech-in-noise…

Audio spectrum and sound pressure levels vary between pulse oximeters.

PubMed

Chandra, Deven; Tessler, Michael J; Usher, John

2006-01-01

The variable-pitch pulse oximeter is an important intraoperative patient monitor. Our ability to hear its auditory signal depends on its acoustical properties and our hearing. This study quantitatively describes the audio spectrum and sound pressure levels of the monitoring tones produced by five variable-pitch pulse oximeters. We compared the Datex-Ohmeda Capnomac Ultima, Hewlett-Packard M1166A, Datex-Engstrom AS/3, Ohmeda Biox 3700, and Datex-Ohmeda 3800 oximeters. Three machines of each of the five models were assessed for sound pressure levels (using a precision sound level meter) and audio spectrum (using a hanning windowed fast Fourier trans-form of three beats at saturations of 99%, 90%, and 85%). The widest range of sound pressure levels was produced by the Hewlett-Packard M1166A (46.5 +/- 1.74 dB to 76.9 +/- 2.77 dB). The loudest model was the Datex-Engstrom AS/3 (89.2 +/- 5.36 dB). Three oximeters, when set to the lower ranges of their volume settings, were indistinguishable from background operating room noise. Each model produced sounds with different audio spectra. Although each model produced a fundamental tone with multiple harmonic overtones, the number of harmonics varied with each model; from three harmonic tones on the Hewlett-Packard M1166A, to 12 on the Ohmeda Biox 3700. There were variations between models, and individual machines of the same model with respect to the fundamental tone associated with a given saturation. There is considerable variance in the sound pressure and audio spectrum of commercially-available pulse oximeters. Further studies are warranted in order to establish standards.

Neurophysiological evidence for the interplay of speech segmentation and word-referent mapping during novel word learning.

PubMed

François, Clément; Cunillera, Toni; Garcia, Enara; Laine, Matti; Rodriguez-Fornells, Antoni

2017-04-01

Learning a new language requires the identification of word units from continuous speech (the speech segmentation problem) and mapping them onto conceptual representation (the word to world mapping problem). Recent behavioral studies have revealed that the statistical properties found within and across modalities can serve as cues for both processes. However, segmentation and mapping have been largely studied separately, and thus it remains unclear whether both processes can be accomplished at the same time and if they share common neurophysiological features. To address this question, we recorded EEG of 20 adult participants during both an audio alone speech segmentation task and an audiovisual word-to-picture association task. The participants were tested for both the implicit detection of online mismatches (structural auditory and visual semantic violations) as well as for the explicit recognition of words and word-to-picture associations. The ERP results from the learning phase revealed a delayed learning-related fronto-central negativity (FN400) in the audiovisual condition compared to the audio alone condition. Interestingly, while online structural auditory violations elicited clear MMN/N200 components in the audio alone condition, visual-semantic violations induced meaning-related N400 modulations in the audiovisual condition. The present results support the idea that speech segmentation and meaning mapping can take place in parallel and act in synergy to enhance novel word learning. Copyright © 2016 Elsevier Ltd. All rights reserved.

A scheme for racquet sports video analysis with the combination of audio-visual information

NASA Astrophysics Data System (ADS)

Xing, Liyuan; Ye, Qixiang; Zhang, Weigang; Huang, Qingming; Yu, Hua

2005-07-01

As a very important category in sports video, racquet sports video, e.g. table tennis, tennis and badminton, has been paid little attention in the past years. Considering the characteristics of this kind of sports video, we propose a new scheme for structure indexing and highlight generating based on the combination of audio and visual information. Firstly, a supervised classification method is employed to detect important audio symbols including impact (ball hit), audience cheers, commentator speech, etc. Meanwhile an unsupervised algorithm is proposed to group video shots into various clusters. Then, by taking advantage of temporal relationship between audio and visual signals, we can specify the scene clusters with semantic labels including rally scenes and break scenes. Thirdly, a refinement procedure is developed to reduce false rally scenes by further audio analysis. Finally, an exciting model is proposed to rank the detected rally scenes from which many exciting video clips such as game (match) points can be correctly retrieved. Experiments on two types of representative racquet sports video, table tennis video and tennis video, demonstrate encouraging results.

Dual Key Speech Encryption Algorithm Based Underdetermined BSS

PubMed Central

Zhao, Huan; Chen, Zuo; Zhang, Xixiang

2014-01-01

When the number of the mixed signals is less than that of the source signals, the underdetermined blind source separation (BSS) is a significant difficult problem. Due to the fact that the great amount data of speech communications and real-time communication has been required, we utilize the intractability of the underdetermined BSS problem to present a dual key speech encryption method. The original speech is mixed with dual key signals which consist of random key signals (one-time pad) generated by secret seed and chaotic signals generated from chaotic system. In the decryption process, approximate calculation is used to recover the original speech signals. The proposed algorithm for speech signals encryption can resist traditional attacks against the encryption system, and owing to approximate calculation, decryption becomes faster and more accurate. It is demonstrated that the proposed method has high level of security and can recover the original signals quickly and efficiently yet maintaining excellent audio quality. PMID:24955430

Are the Literacy Difficulties That Characterize Developmental Dyslexia Associated with a Failure to Integrate Letters and Speech Sounds?

ERIC Educational Resources Information Center

Nash, Hannah M.; Gooch, Debbie; Hulme, Charles; Mahajan, Yatin; McArthur, Genevieve; Steinmetzger, Kurt; Snowling, Margaret J.

2017-01-01

The "automatic letter-sound integration hypothesis" (Blomert, [Blomert, L., 2011]) proposes that dyslexia results from a failure to fully integrate letters and speech sounds into automated audio-visual objects. We tested this hypothesis in a sample of English-speaking children with dyslexic difficulties (N = 13) and samples of…

Multisensory integration of speech sounds with letters vs. visual speech: only visual speech induces the mismatch negativity.

PubMed

Stekelenburg, Jeroen J; Keetels, Mirjam; Vroomen, Jean

2018-05-01

Numerous studies have demonstrated that the vision of lip movements can alter the perception of auditory speech syllables (McGurk effect). While there is ample evidence for integration of text and auditory speech, there are only a few studies on the orthographic equivalent of the McGurk effect. Here, we examined whether written text, like visual speech, can induce an illusory change in the perception of speech sounds on both the behavioural and neural levels. In a sound categorization task, we found that both text and visual speech changed the identity of speech sounds from an /aba/-/ada/ continuum, but the size of this audiovisual effect was considerably smaller for text than visual speech. To examine at which level in the information processing hierarchy these multisensory interactions occur, we recorded electroencephalography in an audiovisual mismatch negativity (MMN, a component of the event-related potential reflecting preattentive auditory change detection) paradigm in which deviant text or visual speech was used to induce an illusory change in a sequence of ambiguous sounds halfway between /aba/ and /ada/. We found that only deviant visual speech induced an MMN, but not deviant text, which induced a late P3-like positive potential. These results demonstrate that text has much weaker effects on sound processing than visual speech does, possibly because text has different biological roots than visual speech. © 2018 The Authors. European Journal of Neuroscience published by Federation of European Neuroscience Societies and John Wiley & Sons Ltd.

Effects of Audio-Visual Integration on the Detection of Masked Speech and Non-Speech Sounds

ERIC Educational Resources Information Center

Eramudugolla, Ranmalee; Henderson, Rachel; Mattingley, Jason B.

2011-01-01

Integration of simultaneous auditory and visual information about an event can enhance our ability to detect that event. This is particularly evident in the perception of speech, where the articulatory gestures of the speaker's lips and face can significantly improve the listener's detection and identification of the message, especially when that…

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

PubMed Central

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

2017-01-01

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

Perception of the Multisensory Coherence of Fluent Audiovisual Speech in Infancy: Its Emergence & the Role of Experience

PubMed Central

Lewkowicz, David J.; Minar, Nicholas J.; Tift, Amy H.; Brandon, Melissa

2014-01-01

To investigate the developmental emergence of the ability to perceive the multisensory coherence of native and non-native audiovisual fluent speech, we tested 4-, 8–10, and 12–14 month-old English-learning infants. Infants first viewed two identical female faces articulating two different monologues in silence and then in the presence of an audible monologue that matched the visible articulations of one of the faces. Neither the 4-month-old nor the 8–10 month-old infants exhibited audio-visual matching in that neither group exhibited greater looking at the matching monologue. In contrast, the 12–14 month-old infants exhibited matching and, consistent with the emergence of perceptual expertise for the native language, they perceived the multisensory coherence of native-language monologues earlier in the test trials than of non-native language monologues. Moreover, the matching of native audible and visible speech streams observed in the 12–14 month olds did not depend on audio-visual synchrony whereas the matching of non-native audible and visible speech streams did depend on synchrony. Overall, the current findings indicate that the perception of the multisensory coherence of fluent audiovisual speech emerges late in infancy, that audio-visual synchrony cues are more important in the perception of the multisensory coherence of non-native than native audiovisual speech, and that the emergence of this skill most likely is affected by perceptual narrowing. PMID:25462038

Can you hear my age? Influences of speech rate and speech spontaneity on estimation of speaker age

PubMed Central

Skoog Waller, Sara; Eriksson, Mårten; Sörqvist, Patrik

2015-01-01

Cognitive hearing science is mainly about the study of how cognitive factors contribute to speech comprehension, but cognitive factors also partake in speech processing to infer non-linguistic information from speech signals, such as the intentions of the talker and the speaker’s age. Here, we report two experiments on age estimation by “naïve” listeners. The aim was to study how speech rate influences estimation of speaker age by comparing the speakers’ natural speech rate with increased or decreased speech rate. In Experiment 1, listeners were presented with audio samples of read speech from three different speaker age groups (young, middle aged, and old adults). They estimated the speakers as younger when speech rate was faster than normal and as older when speech rate was slower than normal. This speech rate effect was slightly greater in magnitude for older (60–65 years) speakers in comparison with younger (20–25 years) speakers, suggesting that speech rate may gain greater importance as a perceptual age cue with increased speaker age. This pattern was more pronounced in Experiment 2, in which listeners estimated age from spontaneous speech. Faster speech rate was associated with lower age estimates, but only for older and middle aged (40–45 years) speakers. Taken together, speakers of all age groups were estimated as older when speech rate decreased, except for the youngest speakers in Experiment 2. The absence of a linear speech rate effect in estimates of younger speakers, for spontaneous speech, implies that listeners use different age estimation strategies or cues (possibly vocabulary) depending on the age of the speaker and the spontaneity of the speech. Potential implications for forensic investigations and other applied domains are discussed. PMID:26236259

The contribution of visual information to the perception of speech in noise with and without informative temporal fine structure

PubMed Central

Stacey, Paula C.; Kitterick, Pádraig T.; Morris, Saffron D.; Sumner, Christian J.

2017-01-01

Understanding what is said in demanding listening situations is assisted greatly by looking at the face of a talker. Previous studies have observed that normal-hearing listeners can benefit from this visual information when a talker's voice is presented in background noise. These benefits have also been observed in quiet listening conditions in cochlear-implant users, whose device does not convey the informative temporal fine structure cues in speech, and when normal-hearing individuals listen to speech processed to remove these informative temporal fine structure cues. The current study (1) characterised the benefits of visual information when listening in background noise; and (2) used sine-wave vocoding to compare the size of the visual benefit when speech is presented with or without informative temporal fine structure. The accuracy with which normal-hearing individuals reported words in spoken sentences was assessed across three experiments. The availability of visual information and informative temporal fine structure cues was varied within and across the experiments. The results showed that visual benefit was observed using open- and closed-set tests of speech perception. The size of the benefit increased when informative temporal fine structure cues were removed. This finding suggests that visual information may play an important role in the ability of cochlear-implant users to understand speech in many everyday situations. Models of audio-visual integration were able to account for the additional benefit of visual information when speech was degraded and suggested that auditory and visual information was being integrated in a similar way in all conditions. The modelling results were consistent with the notion that audio-visual benefit is derived from the optimal combination of auditory and visual sensory cues. PMID:27085797

Mining protein function from text using term-based support vector machines

PubMed Central

Rice, Simon B; Nenadic, Goran; Stapley, Benjamin J

2005-01-01

Background Text mining has spurred huge interest in the domain of biology. The goal of the BioCreAtIvE exercise was to evaluate the performance of current text mining systems. We participated in Task 2, which addressed assigning Gene Ontology terms to human proteins and selecting relevant evidence from full-text documents. We approached it as a modified form of the document classification task. We used a supervised machine-learning approach (based on support vector machines) to assign protein function and select passages that support the assignments. As classification features, we used a protein's co-occurring terms that were automatically extracted from documents. Results The results evaluated by curators were modest, and quite variable for different problems: in many cases we have relatively good assignment of GO terms to proteins, but the selected supporting text was typically non-relevant (precision spanning from 3% to 50%). The method appears to work best when a substantial set of relevant documents is obtained, while it works poorly on single documents and/or short passages. The initial results suggest that our approach can also mine annotations from text even when an explicit statement relating a protein to a GO term is absent. Conclusion A machine learning approach to mining protein function predictions from text can yield good performance only if sufficient training data is available, and significant amount of supporting data is used for prediction. The most promising results are for combined document retrieval and GO term assignment, which calls for the integration of methods developed in BioCreAtIvE Task 1 and Task 2. PMID:15960835

Interactive Activation Model of Speech Perception.

DTIC Science & Technology

1984-11-01

contract. 0 Elar, .l... & .McC’lelland .1.1. Speech perception a, a cognitive proces,: The interactive act ia- %e., tion model of speech perception. In...attempts to provide a machine solution to the problem of speech perception. A second kind of model, growing out of Cognitive Psychology, attempts to...architectures to cognitive and perceptual problems. We also owe a debt to what we might call the computational connectionists -- those who have applied highly

Unsupervised Decoding of Long-Term, Naturalistic Human Neural Recordings with Automated Video and Audio Annotations

PubMed Central

Wang, Nancy X. R.; Olson, Jared D.; Ojemann, Jeffrey G.; Rao, Rajesh P. N.; Brunton, Bingni W.

2016-01-01

Fully automated decoding of human activities and intentions from direct neural recordings is a tantalizing challenge in brain-computer interfacing. Implementing Brain Computer Interfaces (BCIs) outside carefully controlled experiments in laboratory settings requires adaptive and scalable strategies with minimal supervision. Here we describe an unsupervised approach to decoding neural states from naturalistic human brain recordings. We analyzed continuous, long-term electrocorticography (ECoG) data recorded over many days from the brain of subjects in a hospital room, with simultaneous audio and video recordings. We discovered coherent clusters in high-dimensional ECoG recordings using hierarchical clustering and automatically annotated them using speech and movement labels extracted from audio and video. To our knowledge, this represents the first time techniques from computer vision and speech processing have been used for natural ECoG decoding. Interpretable behaviors were decoded from ECoG data, including moving, speaking and resting; the results were assessed by comparison with manual annotation. Discovered clusters were projected back onto the brain revealing features consistent with known functional areas, opening the door to automated functional brain mapping in natural settings. PMID:27148018

Hearing Lips and Seeing Voices: How Cortical Areas Supporting Speech Production Mediate Audiovisual Speech Perception

PubMed Central

Skipper, Jeremy I.; van Wassenhove, Virginie; Nusbaum, Howard C.; Small, Steven L.

2009-01-01

Observing a speaker’s mouth profoundly influences speech perception. For example, listeners perceive an “illusory” “ta” when the video of a face producing /ka/ is dubbed onto an audio /pa/. Here, we show how cortical areas supporting speech production mediate this illusory percept and audiovisual (AV) speech perception more generally. Specifically, cortical activity during AV speech perception occurs in many of the same areas that are active during speech production. We find that different perceptions of the same syllable and the perception of different syllables are associated with different distributions of activity in frontal motor areas involved in speech production. Activity patterns in these frontal motor areas resulting from the illusory “ta” percept are more similar to the activity patterns evoked by AV/ta/ than they are to patterns evoked by AV/pa/ or AV/ka/. In contrast to the activity in frontal motor areas, stimulus-evoked activity for the illusory “ta” in auditory and somatosensory areas and visual areas initially resembles activity evoked by AV/pa/ and AV/ka/, respectively. Ultimately, though, activity in these regions comes to resemble activity evoked by AV/ta/. Together, these results suggest that AV speech elicits in the listener a motor plan for the production of the phoneme that the speaker might have been attempting to produce, and that feedback in the form of efference copy from the motor system ultimately influences the phonetic interpretation. PMID:17218482

Effects of audio-visual presentation of target words in word translation training

NASA Astrophysics Data System (ADS)

Akahane-Yamada, Reiko; Komaki, Ryo; Kubo, Rieko

2004-05-01

Komaki and Akahane-Yamada (Proc. ICA2004) used 2AFC translation task in vocabulary training, in which the target word is presented visually in orthographic form of one language, and the appropriate meaning in another language has to be chosen between two choices. Present paper examined the effect of audio-visual presentation of target word when native speakers of Japanese learn to translate English words into Japanese. Pairs of English words contrasted in several phonemic distinctions (e.g., /r/-/l/, /b/-/v/, etc.) were used as word materials, and presented in three conditions; visual-only (V), audio-only (A), and audio-visual (AV) presentations. Identification accuracy of those words produced by two talkers was also assessed. During pretest, the accuracy for A stimuli was lowest, implying that insufficient translation ability and listening ability interact with each other when aurally presented word has to be translated. However, there was no difference in accuracy between V and AV stimuli, suggesting that participants translate the words depending on visual information only. The effect of translation training using AV stimuli did not transfer to identification ability, showing that additional audio information during translation does not help improve speech perception. Further examination is necessary to determine the effective L2 training method. [Work supported by TAO, Japan.

Efficient audio signal processing for embedded systems

NASA Astrophysics Data System (ADS)

Chiu, Leung Kin

As mobile platforms continue to pack on more computational power, electronics manufacturers start to differentiate their products by enhancing the audio features. However, consumers also demand smaller devices that could operate for longer time, hence imposing design constraints. In this research, we investigate two design strategies that would allow us to efficiently process audio signals on embedded systems such as mobile phones and portable electronics. In the first strategy, we exploit properties of the human auditory system to process audio signals. We designed a sound enhancement algorithm to make piezoelectric loudspeakers sound ”richer" and "fuller." Piezoelectric speakers have a small form factor but exhibit poor response in the low-frequency region. In the algorithm, we combine psychoacoustic bass extension and dynamic range compression to improve the perceived bass coming out from the tiny speakers. We also developed an audio energy reduction algorithm for loudspeaker power management. The perceptually transparent algorithm extends the battery life of mobile devices and prevents thermal damage in speakers. This method is similar to audio compression algorithms, which encode audio signals in such a ways that the compression artifacts are not easily perceivable. Instead of reducing the storage space, however, we suppress the audio contents that are below the hearing threshold, therefore reducing the signal energy. In the second strategy, we use low-power analog circuits to process the signal before digitizing it. We designed an analog front-end for sound detection and implemented it on a field programmable analog array (FPAA). The system is an example of an analog-to-information converter. The sound classifier front-end can be used in a wide range of applications because programmable floating-gate transistors are employed to store classifier weights. Moreover, we incorporated a feature selection algorithm to simplify the analog front-end. A machine

Nasal and Oral Inspiration during Natural Speech Breathing

ERIC Educational Resources Information Center

Lester, Rosemary A.; Hoit, Jeannette D.

2014-01-01

Purpose: The purpose of this study was to determine the typical pattern for inspiration during speech breathing in healthy adults, as well as the factors that might influence it. Method: Ten healthy adults, 18-45 years of age, performed a variety of speaking tasks while nasal ram pressure, audio, and video recordings were obtained. Inspirations…

Speech-Message Extraction from Interference Introduced by External Distributed Sources

NASA Astrophysics Data System (ADS)

Kanakov, V. A.; Mironov, N. A.

2017-08-01

The problem of this study involves the extraction of a speech signal originating from a certain spatial point and calculation of the intelligibility of the extracted voice message. It is solved by the method of decreasing the influence of interference from the speech-message sources on the extracted signal. This method is based on introducing the time delays, which depend on the spatial coordinates, to the recording channels. Audio records of the voices of eight different people were used as test objects during the studies. It is proved that an increase in the number of microphones improves intelligibility of the speech message which is extracted from interference.

Young Autistic Children's Listening Preferences in Regard to Speech: A Possible Characterization of the Symptom of Social Withdrawal.

ERIC Educational Resources Information Center

Klin, Ami

1991-01-01

Twelve autistic children (ages 4-6) were given a choice between their mothers' speech and the noise of superimposed voices. In contrast to comparison groups of mentally retarded and normally developing children, the autistic children actively preferred the superimposed voices or showed a lack of preference for either audio segment. (Author/JDD)

HomeBank: An Online Repository of Daylong Child-Centered Audio Recordings

PubMed Central

VanDam, Mark; Warlaumont, Anne S.; Bergelson, Elika; Cristia, Alejandrina; Soderstrom, Melanie; De Palma, Paul; MacWhinney, Brian

2017-01-01

HomeBank is introduced here. It is a public, permanent, extensible, online database of daylong audio recorded in naturalistic environments. HomeBank serves two primary purposes. First, it is a repository for raw audio and associated files: one database requires special permissions, and another redacted database allows unrestricted public access. Associated files include metadata such as participant demographics and clinical diagnostics, automated annotations, and human-generated transcriptions and annotations. Many recordings use the child-perspective LENA recorders (LENA Research Foundation, Boulder, Colorado, United States), but various recordings and metadata can be accommodated. The HomeBank database can have both vetted and unvetted recordings, with different levels of accessibility. Additionally, HomeBank is an open repository for processing and analysis tools for HomeBank or similar data sets. HomeBank is flexible for users and contributors, making primary data available to researchers, especially those in child development, linguistics, and audio engineering. HomeBank facilitates researchers’ access to large-scale data and tools, linking the acoustic, auditory, and linguistic characteristics of children’s environments with a variety of variables including socioeconomic status, family characteristics, language trajectories, and disorders. Automated processing applied to daylong home audio recordings is now becoming widely used in early intervention initiatives, helping parents to provide richer speech input to at-risk children. PMID:27111272

Facilities to assist people to research into stammered speech

PubMed Central

Howell, Peter; Huckvale, Mark

2008-01-01

The purpose of this article is to indicate how access can be obtained, through Stammering Research, to audio recordings and transcriptions of spontaneous speech data from speakers who stammer. Selections of the first author’s data are available in several formats. We describe where to obtain free software for manipulation and analysis of the data in their respective formats. Papers reporting analyses of these data are invited as submissions to this section of Stammering Research. It is intended that subsequent analyses that employ these data will be published in Stammering Research on an on-going basis. Plans are outlined to provide similar data from young speakers (ones developing fluently and ones who stammer), follow-up data from speakers who stammer, data from speakers who stammer who do not speak English and from speakers who have other speech disorders, for comparison, all through the pages of Stammering Research. The invitation is extended to those promulgating evidence-based practice approaches (see the Journal of Fluency Disorders, volume 28, number 4 which is a special issue devoted to this topic) and anyone with other interesting data related to stammering to prepare them in a form that can be made accessible to others via Stammering Research. PMID:18418475

One approach to design of speech emotion database

NASA Astrophysics Data System (ADS)

Uhrin, Dominik; Chmelikova, Zdenka; Tovarek, Jaromir; Partila, Pavol; Voznak, Miroslav

2016-05-01

This article describes a system for evaluating the credibility of recordings with emotional character. Sound recordings form Czech language database for training and testing systems of speech emotion recognition. These systems are designed to detect human emotions in his voice. The emotional state of man is useful in the security forces and emergency call service. Man in action (soldier, police officer and firefighter) is often exposed to stress. Information about the emotional state (his voice) will help to dispatch to adapt control commands for procedure intervention. Call agents of emergency call service must recognize the mental state of the caller to adjust the mood of the conversation. In this case, the evaluation of the psychological state is the key factor for successful intervention. A quality database of sound recordings is essential for the creation of the mentioned systems. There are quality databases such as Berlin Database of Emotional Speech or Humaine. The actors have created these databases in an audio studio. It means that the recordings contain simulated emotions, not real. Our research aims at creating a database of the Czech emotional recordings of real human speech. Collecting sound samples to the database is only one of the tasks. Another one, no less important, is to evaluate the significance of recordings from the perspective of emotional states. The design of a methodology for evaluating emotional recordings credibility is described in this article. The results describe the advantages and applicability of the developed method.

Animation, audio, and spatial ability: Optimizing multimedia for scientific explanations

NASA Astrophysics Data System (ADS)

Koroghlanian, Carol May

This study investigated the effects of audio, animation and spatial ability in a computer based instructional program for biology. The program presented instructional material via text or audio with lean text and included eight instructional sequences presented either via static illustrations or animations. High school students enrolled in a biology course were blocked by spatial ability and randomly assigned to one of four treatments (Text-Static Illustration Audio-Static Illustration, Text-Animation, Audio-Animation). The study examined the effects of instructional mode (Text vs. Audio), illustration mode (Static Illustration vs. Animation) and spatial ability (Low vs. High) on practice and posttest achievement, attitude and time. Results for practice achievement indicated that high spatial ability participants achieved more than low spatial ability participants. Similar results for posttest achievement and spatial ability were not found. Participants in the Static Illustration treatments achieved the same as participants in the Animation treatments on both the practice and posttest. Likewise, participants in the Text treatments achieved the same as participants in the Audio treatments on both the practice and posttest. In terms of attitude, participants responded favorably to the computer based instructional program. They found the program interesting, felt the static illustrations or animations made the explanations easier to understand and concentrated on learning the material. Furthermore, participants in the Animation treatments felt the information was easier to understand than participants in the Static Illustration treatments. However, no difference for any attitude item was found for participants in the Text as compared to those in the Audio treatments. Significant differences were found by Spatial Ability for three attitude items concerning concentration and interest. In all three items, the low spatial ability participants responded more positively

Impact of Language on Development of Auditory-Visual Speech Perception

ERIC Educational Resources Information Center

Sekiyama, Kaoru; Burnham, Denis

2008-01-01

The McGurk effect paradigm was used to examine the developmental onset of inter-language differences between Japanese and English in auditory-visual speech perception. Participants were asked to identify syllables in audiovisual (with congruent or discrepant auditory and visual components), audio-only, and video-only presentations at various…

Comparing the Impact of Rates of Text-to-Speech Software on Reading Fluency and Comprehension for Adults with Reading Difficulties

ERIC Educational Resources Information Center

Coleman, Mari Beth; Killdare, Laura K.; Bell, Sherry Mee; Carter, Amanda M.

2014-01-01

The purpose of this study was to determine the impact of text-to-speech software on reading fluency and comprehension for four postsecondary students with below average reading fluency and comprehension including three students diagnosed with learning disabilities and concomitant conditions (e.g., attention deficit hyperactivity disorder, seizure…

MARTI: man-machine animation real-time interface

NASA Astrophysics Data System (ADS)

Jones, Christian M.; Dlay, Satnam S.

1997-05-01

The research introduces MARTI (man-machine animation real-time interface) for the realization of natural human-machine interfacing. The system uses simple vocal sound-tracks of human speakers to provide lip synchronization of computer graphical facial models. We present novel research in a number of engineering disciplines, which include speech recognition, facial modeling, and computer animation. This interdisciplinary research utilizes the latest, hybrid connectionist/hidden Markov model, speech recognition system to provide very accurate phone recognition and timing for speaker independent continuous speech, and expands on knowledge from the animation industry in the development of accurate facial models and automated animation. The research has many real-world applications which include the provision of a highly accurate and 'natural' man-machine interface to assist user interactions with computer systems and communication with one other using human idiosyncrasies; a complete special effects and animation toolbox providing automatic lip synchronization without the normal constraints of head-sets, joysticks, and skilled animators; compression of video data to well below standard telecommunication channel bandwidth for video communications and multi-media systems; assisting speech training and aids for the handicapped; and facilitating player interaction for 'video gaming' and 'virtual worlds.' MARTI has introduced a new level of realism to man-machine interfacing and special effect animation which has been previously unseen.

Perception of audio-visual speech synchrony in Spanish-speaking children with and without specific language impairment

PubMed Central

PONS, FERRAN; ANDREU, LLORENC.; SANZ-TORRENT, MONICA; BUIL-LEGAZ, LUCIA; LEWKOWICZ, DAVID J.

2014-01-01

Speech perception involves the integration of auditory and visual articulatory information and, thus, requires the perception of temporal synchrony between this information. There is evidence that children with Specific Language Impairment (SLI) have difficulty with auditory speech perception but it is not known if this is also true for the integration of auditory and visual speech. Twenty Spanish-speaking children with SLI, twenty typically developing age-matched Spanish-speaking children, and twenty Spanish-speaking children matched for MLU-w participated in an eye-tracking study to investigate the perception of audiovisual speech synchrony. Results revealed that children with typical language development perceived an audiovisual asynchrony of 666ms regardless of whether the auditory or visual speech attribute led the other one. Children with SLI only detected the 666 ms asynchrony when the auditory component followed the visual component. None of the groups perceived an audiovisual asynchrony of 366ms. These results suggest that the difficulty of speech processing by children with SLI would also involve difficulties in integrating auditory and visual aspects of speech perception. PMID:22874648

Perception of audio-visual speech synchrony in Spanish-speaking children with and without specific language impairment.

PubMed

Pons, Ferran; Andreu, Llorenç; Sanz-Torrent, Monica; Buil-Legaz, Lucía; Lewkowicz, David J

2013-06-01

Speech perception involves the integration of auditory and visual articulatory information, and thus requires the perception of temporal synchrony between this information. There is evidence that children with specific language impairment (SLI) have difficulty with auditory speech perception but it is not known if this is also true for the integration of auditory and visual speech. Twenty Spanish-speaking children with SLI, twenty typically developing age-matched Spanish-speaking children, and twenty Spanish-speaking children matched for MLU-w participated in an eye-tracking study to investigate the perception of audiovisual speech synchrony. Results revealed that children with typical language development perceived an audiovisual asynchrony of 666 ms regardless of whether the auditory or visual speech attribute led the other one. Children with SLI only detected the 666 ms asynchrony when the auditory component preceded [corrected] the visual component. None of the groups perceived an audiovisual asynchrony of 366 ms. These results suggest that the difficulty of speech processing by children with SLI would also involve difficulties in integrating auditory and visual aspects of speech perception.

Integrating voice evaluation: correlation between acoustic and audio-perceptual measures.

PubMed

Vaz Freitas, Susana; Melo Pestana, Pedro; Almeida, Vítor; Ferreira, Aníbal

2015-05-01

This article aims to establish correlations between acoustic and audio-perceptual measures using the GRBAS scale with respect to four different voice analysis software programs. Exploratory, transversal. A total of 90 voice records were collected and analyzed with the Dr. Speech (Tiger Electronics, Seattle, WA), Multidimensional Voice Program (Kay Elemetrics, NJ, USA), PRAAT (University of Amsterdam, The Netherlands), and Voice Studio (Seegnal, Oporto, Portugal) software programs. The acoustic measures were correlated to the audio-perceptual parameters of the GRBAS and rated by 10 experts. The predictive value of the acoustic measurements related to the audio-perceptual parameters exhibited magnitudes ranging from weak (R(2)a=0.17) to moderate (R(2)a=0.71). The parameter exhibiting the highest correlation magnitude is B (Breathiness), whereas the weaker correlation magnitudes were found to be for A (Asthenia) and S (Strain). The acoustic measures with stronger predictive values were local Shimmer, harmonics-to-noise ratio, APQ5 shimmer, and PPQ5 jitter, with different magnitudes for each one of the studied software programs. Some acoustic measures are pointed as significant predictors of GRBAS parameters, but they differ among software programs. B (Breathiness) was the parameter exhibiting the highest correlation magnitude. Copyright © 2015 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Texts as Metaphoric Machines and the Challenge of the Digital

ERIC Educational Resources Information Center

Kouppanou, Anna

2016-01-01

In this essay Anna Kouppanou expands the notion of metaphor from its received meaning to refer to an embodied and material process of connectedness that transforms the domains that it brings together. Because of metaphor's reliance on materiality and exteriority Kouppanou turns to literary texts, which she calls "metaphoric machines." In…

Pitch-Based Segregation of Reverberant Speech

DTIC Science & Technology

2005-02-01

speaker recognition in real environments, audio information retrieval and hearing prosthesis. Second, although binaural listening improves the...intelligibility of target speech under anechoic conditions (Bronkhorst, 2000), this binaural advantage is largely eliminated by reverberation (Plomp, 1976...Brown and Cooke, 1994; Wang and Brown, 1999; Hu and Wang, 2004) as well as in binaural separation (e.g., Roman et al., 2003; Palomaki et al., 2004

A study on nonlinear characteristics of speech sound with reference to some languages of North East region

NASA Astrophysics Data System (ADS)

Dutta, Rashmi

parametric model, which has been most useful in practical applications. Developing a system that can understand natural language has been a continuing goal of speech researchers. Fully automatic high quality machine translation systems are extremely difficult to build. The difficulty arises from the following reasons: In any natural language text, only part of the information to be conveyed is explicitly expressed. It is the human mind which fills up and supplements the details using contextual.

Visual-auditory integration during speech imitation in autism.

PubMed

Williams, Justin H G; Massaro, Dominic W; Peel, Natalie J; Bosseler, Alexis; Suddendorf, Thomas

2004-01-01

Children with autistic spectrum disorder (ASD) may have poor audio-visual integration, possibly reflecting dysfunctional 'mirror neuron' systems which have been hypothesised to be at the core of the condition. In the present study, a computer program, utilizing speech synthesizer software and a 'virtual' head (Baldi), delivered speech stimuli for identification in auditory, visual or bimodal conditions. Children with ASD were poorer than controls at recognizing stimuli in the unimodal conditions, but once performance on this measure was controlled for, no group difference was found in the bimodal condition. A group of participants with ASD were also trained to develop their speech-reading ability. Training improved visual accuracy and this also improved the children's ability to utilize visual information in their processing of speech. Overall results were compared to predictions from mathematical models based on integration and non-integration, and were most consistent with the integration model. We conclude that, whilst they are less accurate in recognizing stimuli in the unimodal condition, children with ASD show normal integration of visual and auditory speech stimuli. Given that training in recognition of visual speech was effective, children with ASD may benefit from multi-modal approaches in imitative therapy and language training.

Scientific bases of human-machine communication by voice.

PubMed Central

Schafer, R W

1995-01-01

The scientific bases for human-machine communication by voice are in the fields of psychology, linguistics, acoustics, signal processing, computer science, and integrated circuit technology. The purpose of this paper is to highlight the basic scientific and technological issues in human-machine communication by voice and to point out areas of future research opportunity. The discussion is organized around the following major issues in implementing human-machine voice communication systems: (i) hardware/software implementation of the system, (ii) speech synthesis for voice output, (iii) speech recognition and understanding for voice input, and (iv) usability factors related to how humans interact with machines. PMID:7479802

A social feedback loop for speech development and its reduction in autism

PubMed Central

Warlaumont, Anne S.; Richards, Jeffrey A.; Gilkerson, Jill; Oller, D. Kimbrough

2014-01-01

We analyze the microstructure of child-adult interaction during naturalistic, daylong, automatically labeled audio recordings (13,836 hours total) of children (8- to 48-month-olds) with and without autism. We find that adult responses are more likely when child vocalizations are speech-related. In turn, a child vocalization is more likely to be speech-related if the previous speech-related child vocalization received an immediate adult response. Taken together, these results are consistent with the idea that there is a social feedback loop between child and caregiver that promotes speech-language development. Although this feedback loop applies in both typical development and autism, children with autism produce proportionally fewer speech-related vocalizations and the responses they receive are less contingent on whether their vocalizations are speech-related. We argue that such differences will diminish the strength of the social feedback loop with cascading effects on speech development over time. Differences related to socioeconomic status are also reported. PMID:24840717

Comparison of audio and audiovisual measures of adult stuttering: Implications for clinical trials.

PubMed

O'Brian, Sue; Jones, Mark; Onslow, Mark; Packman, Ann; Menzies, Ross; Lowe, Robyn

2015-04-15

This study investigated whether measures of percentage syllables stuttered (%SS) and stuttering severity ratings with a 9-point scale differ when made from audiovisual compared with audio-only recordings. Four experienced speech-language pathologists measured %SS and assigned stuttering severity ratings to 10-minute audiovisual and audio-only recordings of 36 adults. There was a mean 18% increase in %SS scores when samples were presented in audiovisual compared with audio-only mode. This result was consistent across both higher and lower %SS scores and was found to be directly attributable to counts of stuttered syllables rather than the total number of syllables. There was no significant difference between stuttering severity ratings made from the two modes. In clinical trials research, when using %SS as the primary outcome measure, audiovisual samples would be preferred as long as clear, good quality, front-on images can be easily captured. Alternatively, stuttering severity ratings may be a more valid measure to use as they correlate well with %SS and values are not influenced by the presentation mode.

The Neural Basis of Speech Perception through Lipreading and Manual Cues: Evidence from Deaf Native Users of Cued Speech

PubMed Central

Aparicio, Mario; Peigneux, Philippe; Charlier, Brigitte; Balériaux, Danielle; Kavec, Martin; Leybaert, Jacqueline

2017-01-01

We present here the first neuroimaging data for perception of Cued Speech (CS) by deaf adults who are native users of CS. CS is a visual mode of communicating a spoken language through a set of manual cues which accompany lipreading and disambiguate it. With CS, sublexical units of the oral language are conveyed clearly and completely through the visual modality without requiring hearing. The comparison of neural processing of CS in deaf individuals with processing of audiovisual (AV) speech in normally hearing individuals represents a unique opportunity to explore the similarities and differences in neural processing of an oral language delivered in a visuo-manual vs. an AV modality. The study included deaf adult participants who were early CS users and native hearing users of French who process speech audiovisually. Words were presented in an event-related fMRI design. Three conditions were presented to each group of participants. The deaf participants saw CS words (manual + lipread), words presented as manual cues alone, and words presented to be lipread without manual cues. The hearing group saw AV spoken words, audio-alone and lipread-alone. Three findings are highlighted. First, the middle and superior temporal gyrus (excluding Heschl’s gyrus) and left inferior frontal gyrus pars triangularis constituted a common, amodal neural basis for AV and CS perception. Second, integration was inferred in posterior parts of superior temporal sulcus for audio and lipread information in AV speech, but in the occipito-temporal junction, including MT/V5, for the manual cues and lipreading in CS. Third, the perception of manual cues showed a much greater overlap with the regions activated by CS (manual + lipreading) than lipreading alone did. This supports the notion that manual cues play a larger role than lipreading for CS processing. The present study contributes to a better understanding of the role of manual cues as support of visual speech perception in the framework

Foreign Subtitles Help but Native-Language Subtitles Harm Foreign Speech Perception

PubMed Central

Mitterer, Holger; McQueen, James M.

2009-01-01

Understanding foreign speech is difficult, in part because of unusual mappings between sounds and words. It is known that listeners in their native language can use lexical knowledge (about how words ought to sound) to learn how to interpret unusual speech-sounds. We therefore investigated whether subtitles, which provide lexical information, support perceptual learning about foreign speech. Dutch participants, unfamiliar with Scottish and Australian regional accents of English, watched Scottish or Australian English videos with Dutch, English or no subtitles, and then repeated audio fragments of both accents. Repetition of novel fragments was worse after Dutch-subtitle exposure but better after English-subtitle exposure. Native-language subtitles appear to create lexical interference, but foreign-language subtitles assist speech learning by indicating which words (and hence sounds) are being spoken. PMID:19918371

Auditory cross-modal reorganization in cochlear implant users indicates audio-visual integration.

PubMed

Stropahl, Maren; Debener, Stefan

2017-01-01

There is clear evidence for cross-modal cortical reorganization in the auditory system of post-lingually deafened cochlear implant (CI) users. A recent report suggests that moderate sensori-neural hearing loss is already sufficient to initiate corresponding cortical changes. To what extend these changes are deprivation-induced or related to sensory recovery is still debated. Moreover, the influence of cross-modal reorganization on CI benefit is also still unclear. While reorganization during deafness may impede speech recovery, reorganization also has beneficial influences on face recognition and lip-reading. As CI users were observed to show differences in multisensory integration, the question arises if cross-modal reorganization is related to audio-visual integration skills. The current electroencephalography study investigated cortical reorganization in experienced post-lingually deafened CI users ( n  = 18), untreated mild to moderately hearing impaired individuals (n = 18) and normal hearing controls ( n  = 17). Cross-modal activation of the auditory cortex by means of EEG source localization in response to human faces and audio-visual integration, quantified with the McGurk illusion, were measured. CI users revealed stronger cross-modal activations compared to age-matched normal hearing individuals. Furthermore, CI users showed a relationship between cross-modal activation and audio-visual integration strength. This may further support a beneficial relationship between cross-modal activation and daily-life communication skills that may not be fully captured by laboratory-based speech perception tests. Interestingly, hearing impaired individuals showed behavioral and neurophysiological results that were numerically between the other two groups, and they showed a moderate relationship between cross-modal activation and the degree of hearing loss. This further supports the notion that auditory deprivation evokes a reorganization of the auditory system

The Effect of Audio and Animation in Multimedia Instruction

ERIC Educational Resources Information Center

Koroghlanian, Carol; Klein, James D.

2004-01-01

This study investigated the effects of audio, animation, and spatial ability in a multimedia computer program for high school biology. Participants completed a multimedia program that presented content by way of text or audio with lean text. In addition, several instructional sequences were presented either with static illustrations or animations.…

Is talking to an automated teller machine natural and fun?

PubMed

Chan, F Y; Khalid, H M

Usability and affective issues of using automatic speech recognition technology to interact with an automated teller machine (ATM) are investigated in two experiments. The first uncovered dialogue patterns of ATM users for the purpose of designing the user interface for a simulated speech ATM system. Applying the Wizard-of-Oz methodology, multiple mapping and word spotting techniques, the speech driven ATM accommodates bilingual users of Bahasa Melayu and English. The second experiment evaluates the usability of a hybrid speech ATM, comparing it with a simulated manual ATM. The aim is to investigate how natural and fun can talking to a speech ATM be for these first-time users. Subjects performed the withdrawal and balance enquiry tasks. The ANOVA was performed on the usability and affective data. The results showed significant differences between systems in the ability to complete the tasks as well as in transaction errors. Performance was measured on the time taken by subjects to complete the task and the number of speech recognition errors that occurred. On the basis of user emotions, it can be said that the hybrid speech system enabled pleasurable interaction. Despite the limitations of speech recognition technology, users are set to talk to the ATM when it becomes available for public use.

A haptic-inspired audio approach for structural health monitoring decision-making

NASA Astrophysics Data System (ADS)

Mao, Zhu; Todd, Michael; Mascareñas, David

2015-03-01

Haptics is the field at the interface of human touch (tactile sensation) and classification, whereby tactile feedback is used to train and inform a decision-making process. In structural health monitoring (SHM) applications, haptic devices have been introduced and applied in a simplified laboratory scale scenario, in which nonlinearity, representing the presence of damage, was encoded into a vibratory manual interface. In this paper, the "spirit" of haptics is adopted, but here ultrasonic guided wave scattering information is transformed into audio (rather than tactile) range signals. After sufficient training, the structural damage condition, including occurrence and location, can be identified through the encoded audio waveforms. Different algorithms are employed in this paper to generate the transformed audio signals and the performance of each encoding algorithms is compared, and also compared with standard machine learning classifiers. In the long run, the haptic decision-making is aiming to detect and classify structural damages in a more rigorous environment, and approaching a baseline-free fashion with embedded temperature compensation.

Collusion-Resistant Audio Fingerprinting System in the Modulated Complex Lapped Transform Domain

PubMed Central

Garcia-Hernandez, Jose Juan; Feregrino-Uribe, Claudia; Cumplido, Rene

2013-01-01

Collusion-resistant fingerprinting paradigm seems to be a practical solution to the piracy problem as it allows media owners to detect any unauthorized copy and trace it back to the dishonest users. Despite the billionaire losses in the music industry, most of the collusion-resistant fingerprinting systems are devoted to digital images and very few to audio signals. In this paper, state-of-the-art collusion-resistant fingerprinting ideas are extended to audio signals and the corresponding parameters and operation conditions are proposed. Moreover, in order to carry out fingerprint detection using just a fraction of the pirate audio clip, block-based embedding and its corresponding detector is proposed. Extensive simulations show the robustness of the proposed system against average collusion attack. Moreover, by using an efficient Fast Fourier Transform core and standard computer machines it is shown that the proposed system is suitable for real-world scenarios. PMID:23762455

Using machine learning to disentangle homonyms in large text corpora.

PubMed

Roll, Uri; Correia, Ricardo A; Berger-Tal, Oded

2018-06-01

Systematic reviews are an increasingly popular decision-making tool that provides an unbiased summary of evidence to support conservation action. These reviews bridge the gap between researchers and managers by presenting a comprehensive overview of all studies relating to a particular topic and identify specifically where and under which conditions an effect is present. However, several technical challenges can severely hinder the feasibility and applicability of systematic reviews, for example, homonyms (terms that share spelling but differ in meaning). Homonyms add noise to search results and cannot be easily identified or removed. We developed a semiautomated approach that can aid in the classification of homonyms among narratives. We used a combination of automated content analysis and artificial neural networks to quickly and accurately sift through large corpora of academic texts and classify them to distinct topics. As an example, we explored the use of the word reintroduction in academic texts. Reintroduction is used within the conservation context to indicate the release of organisms to their former native habitat; however, a Web of Science search for this word returned thousands of publications in which the term has other meanings and contexts. Using our method, we automatically classified a sample of 3000 of these publications with over 99% accuracy, relative to a manual classification. Our approach can be used easily with other homonyms and can greatly facilitate systematic reviews or similar work in which homonyms hinder the harnessing of large text corpora. Beyond homonyms we see great promise in combining automated content analysis and machine-learning methods to handle and screen big data for relevant information in conservation science. © 2017 Society for Conservation Biology.

Internet Video Telephony Allows Speech Reading by Deaf Individuals and Improves Speech Perception by Cochlear Implant Users

PubMed Central

Mantokoudis, Georgios; Dähler, Claudia; Dubach, Patrick; Kompis, Martin; Caversaccio, Marco D.; Senn, Pascal

2013-01-01

Objective To analyze speech reading through Internet video calls by profoundly hearing-impaired individuals and cochlear implant (CI) users. Methods Speech reading skills of 14 deaf adults and 21 CI users were assessed using the Hochmair Schulz Moser (HSM) sentence test. We presented video simulations using different video resolutions (1280×720, 640×480, 320×240, 160×120 px), frame rates (30, 20, 10, 7, 5 frames per second (fps)), speech velocities (three different speakers), webcameras (Logitech Pro9000, C600 and C500) and image/sound delays (0–500 ms). All video simulations were presented with and without sound and in two screen sizes. Additionally, scores for live Skype™ video connection and live face-to-face communication were assessed. Results Higher frame rate (>7 fps), higher camera resolution (>640×480 px) and shorter picture/sound delay (<100 ms) were associated with increased speech perception scores. Scores were strongly dependent on the speaker but were not influenced by physical properties of the camera optics or the full screen mode. There is a significant median gain of +8.5%pts (p = 0.009) in speech perception for all 21 CI-users if visual cues are additionally shown. CI users with poor open set speech perception scores (n = 11) showed the greatest benefit under combined audio-visual presentation (median speech perception +11.8%pts, p = 0.032). Conclusion Webcameras have the potential to improve telecommunication of hearing-impaired individuals. PMID:23359119

Internet video telephony allows speech reading by deaf individuals and improves speech perception by cochlear implant users.

PubMed

Mantokoudis, Georgios; Dähler, Claudia; Dubach, Patrick; Kompis, Martin; Caversaccio, Marco D; Senn, Pascal

2013-01-01

To analyze speech reading through Internet video calls by profoundly hearing-impaired individuals and cochlear implant (CI) users. Speech reading skills of 14 deaf adults and 21 CI users were assessed using the Hochmair Schulz Moser (HSM) sentence test. We presented video simulations using different video resolutions (1280 × 720, 640 × 480, 320 × 240, 160 × 120 px), frame rates (30, 20, 10, 7, 5 frames per second (fps)), speech velocities (three different speakers), webcameras (Logitech Pro9000, C600 and C500) and image/sound delays (0-500 ms). All video simulations were presented with and without sound and in two screen sizes. Additionally, scores for live Skype™ video connection and live face-to-face communication were assessed. Higher frame rate (>7 fps), higher camera resolution (>640 × 480 px) and shorter picture/sound delay (<100 ms) were associated with increased speech perception scores. Scores were strongly dependent on the speaker but were not influenced by physical properties of the camera optics or the full screen mode. There is a significant median gain of +8.5%pts (p = 0.009) in speech perception for all 21 CI-users if visual cues are additionally shown. CI users with poor open set speech perception scores (n = 11) showed the greatest benefit under combined audio-visual presentation (median speech perception +11.8%pts, p = 0.032). Webcameras have the potential to improve telecommunication of hearing-impaired individuals.

Natural speech algorithm applied to baseline interview data can predict which patients will respond to psilocybin for treatment-resistant depression.

PubMed

Carrillo, Facundo; Sigman, Mariano; Fernández Slezak, Diego; Ashton, Philip; Fitzgerald, Lily; Stroud, Jack; Nutt, David J; Carhart-Harris, Robin L

2018-04-01

Natural speech analytics has seen some improvements over recent years, and this has opened a window for objective and quantitative diagnosis in psychiatry. Here, we used a machine learning algorithm applied to natural speech to ask whether language properties measured before psilocybin for treatment-resistant can predict for which patients it will be effective and for which it will not. A baseline autobiographical memory interview was conducted and transcribed. Patients with treatment-resistant depression received 2 doses of psilocybin, 10 mg and 25 mg, 7 days apart. Psychological support was provided before, during and after all dosing sessions. Quantitative speech measures were applied to the interview data from 17 patients and 18 untreated age-matched healthy control subjects. A machine learning algorithm was used to classify between controls and patients and predict treatment response. Speech analytics and machine learning successfully differentiated depressed patients from healthy controls and identified treatment responders from non-responders with a significant level of 85% of accuracy (75% precision). Automatic natural language analysis was used to predict effective response to treatment with psilocybin, suggesting that these tools offer a highly cost-effective facility for screening individuals for treatment suitability and sensitivity. The sample size was small and replication is required to strengthen inferences on these results. Copyright © 2018 Elsevier B.V. All rights reserved.

Data-driven analysis of functional brain interactions during free listening to music and speech.

PubMed

Fang, Jun; Hu, Xintao; Han, Junwei; Jiang, Xi; Zhu, Dajiang; Guo, Lei; Liu, Tianming

2015-06-01

Natural stimulus functional magnetic resonance imaging (N-fMRI) such as fMRI acquired when participants were watching video streams or listening to audio streams has been increasingly used to investigate functional mechanisms of the human brain in recent years. One of the fundamental challenges in functional brain mapping based on N-fMRI is to model the brain's functional responses to continuous, naturalistic and dynamic natural stimuli. To address this challenge, in this paper we present a data-driven approach to exploring functional interactions in the human brain during free listening to music and speech streams. Specifically, we model the brain responses using N-fMRI by measuring the functional interactions on large-scale brain networks with intrinsically established structural correspondence, and perform music and speech classification tasks to guide the systematic identification of consistent and discriminative functional interactions when multiple subjects were listening music and speech in multiple categories. The underlying premise is that the functional interactions derived from N-fMRI data of multiple subjects should exhibit both consistency and discriminability. Our experimental results show that a variety of brain systems including attention, memory, auditory/language, emotion, and action networks are among the most relevant brain systems involved in classic music, pop music and speech differentiation. Our study provides an alternative approach to investigating the human brain's mechanism in comprehension of complex natural music and speech.

Influence of speech sample on perceptual rating of hypernasality.

PubMed

Medeiros, Maria Natália Leite de; Fukushiro, Ana Paula; Yamashita, Renata Paciello

2016-07-07

To investigate the influence of speech sample of spontaneous conversation or sentences repetition on intra and inter-rater hypernasality reliability. One hundred and twenty audio recorded speech samples (60 containing spontaneous conversation and 60 containing repeated sentences) of individuals with repaired cleft palate±lip, both genders, aged between 6 and 52 years old (mean=21±10) were selected and edited. Three experienced speech and language pathologists rated hypernasality according to their own criteria using 4-point scale: 1=absence of hypernasality, 2=mild hypernasality, 3=moderate hypernasality and 4=severe hypernasality, first in spontaneous speech samples and 30 days after, in sentences repetition samples. Intra- and inter-rater agreements were calculated for both speech samples and were statistically compared by the Z test at a significance level of 5%. Comparison of intra-rater agreements between both speech samples showed an increase of the coefficients obtained in the analysis of sentences repetition compared to those obtained in spontaneous conversation. Comparison between inter-rater agreement showed no significant difference among the three raters for the two speech samples. Sentences repetition improved intra-raters reliability of perceptual judgment of hypernasality. However, the speech sample had no influence on reliability among different raters.

A study of speech interfaces for the vehicle environment.

DOT National Transportation Integrated Search

2013-05-01

Over the past few years, there has been a shift in automotive human machine interfaces from : visual-manual interactions (pushing buttons and rotating knobs) to speech interaction. In terms of : distraction, the industry views speech interaction as a...

Two Stage Data Augmentation for Low Resourced Speech Recognition (Author’s Manuscript)

DTIC Science & Technology

2016-09-12

speech recognition, deep neural networks, data augmentation 1. Introduction When training data is limited—whether it be audio or text—the obvious...Schwartz, and S. Tsakalidis, “Enhancing low resource keyword spotting with au- tomatically retrieved web documents,” in Interspeech, 2015, pp. 839–843. [2...and F. Seide, “Feature learning in deep neural networks - a study on speech recognition tasks,” in International Conference on Learning Representations

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

PubMed

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

2005-01-01

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

Digital Audio Application to Short Wave Broadcasting

NASA Technical Reports Server (NTRS)

Chen, Edward Y.

1997-01-01

Digital audio is becoming prevalent not only in consumer electornics, but also in different broadcasting media. Terrestrial analog audio broadcasting in the AM and FM bands will be eventually be replaced by digital systems.

A social feedback loop for speech development and its reduction in autism.

PubMed

Warlaumont, Anne S; Richards, Jeffrey A; Gilkerson, Jill; Oller, D Kimbrough

2014-07-01

We analyzed the microstructure of child-adult interaction during naturalistic, daylong, automatically labeled audio recordings (13,836 hr total) of children (8- to 48-month-olds) with and without autism. We found that an adult was more likely to respond when the child's vocalization was speech related rather than not speech related. In turn, a child's vocalization was more likely to be speech related if the child's previous speech-related vocalization had received an immediate adult response rather than no response. Taken together, these results are consistent with the idea that there is a social feedback loop between child and caregiver that promotes speech development. Although this feedback loop applies in both typical development and autism, children with autism produced proportionally fewer speech-related vocalizations, and the responses they received were less contingent on whether their vocalizations were speech related. We argue that such differences will diminish the strength of the social feedback loop and have cascading effects on speech development over time. Differences related to socioeconomic status are also reported. © The Author(s) 2014.

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.

Aircraft noise and speech intelligibility in an outdoor living space.

PubMed

Alvarsson, Jesper J; Nordström, Henrik; Lundén, Peter; Nilsson, Mats E

2014-06-01

Studies of effects on speech intelligibility from aircraft noise in outdoor places are currently lacking. To explore these effects, first-order ambisonic recordings of aircraft noise were reproduced outdoors in a pergola. The average background level was 47 dB LA eq. Lists of phonetically balanced words (LAS max,word = 54 dB) were reproduced simultaneously with aircraft passage noise (LAS max,noise = 72-84 dB). Twenty individually tested listeners wrote down each presented word while seated in the pergola. The main results were (i) aircraft noise negatively affects speech intelligibility at sound pressure levels that exceed those of the speech sound (signal-to-noise ratio, S/N < 0), and (ii) the simple A-weighted S/N ratio was nearly as good an indicator of speech intelligibility as were two more advanced models, the Speech Intelligibility Index and Glasberg and Moore's [J. Audio Eng. Soc. 53, 906-918 (2005)] partial loudness model. This suggests that any of these indicators is applicable for predicting effects of aircraft noise on speech intelligibility outdoors.

Applications of Hilbert Spectral Analysis for Speech and Sound Signals

NASA Technical Reports Server (NTRS)

Huang, Norden E.

2003-01-01

A new method for analyzing nonlinear and nonstationary data has been developed, and the natural applications are to speech and sound signals. The key part of the method is the Empirical Mode Decomposition method with which any complicated data set can be decomposed into a finite and often small number of Intrinsic Mode Functions (IMF). An IMF is defined as any function having the same numbers of zero-crossing and extrema, and also having symmetric envelopes defined by the local maxima and minima respectively. The IMF also admits well-behaved Hilbert transform. This decomposition method is adaptive, and, therefore, highly efficient. Since the decomposition is based on the local characteristic time scale of the data, it is applicable to nonlinear and nonstationary processes. With the Hilbert transform, the Intrinsic Mode Functions yield instantaneous frequencies as functions of time, which give sharp identifications of imbedded structures. This method invention can be used to process all acoustic signals. Specifically, it can process the speech signals for Speech synthesis, Speaker identification and verification, Speech recognition, and Sound signal enhancement and filtering. Additionally, as the acoustical signals from machinery are essentially the way the machines are talking to us. Therefore, the acoustical signals, from the machines, either from sound through air or vibration on the machines, can tell us the operating conditions of the machines. Thus, we can use the acoustic signal to diagnosis the problems of machines.

Musical examination to bridge audio data and sheet music

NASA Astrophysics Data System (ADS)

Pan, Xunyu; Cross, Timothy J.; Xiao, Liangliang; Hei, Xiali

2015-03-01

The digitalization of audio is commonly implemented for the purpose of convenient storage and transmission of music and songs in today's digital age. Analyzing digital audio for an insightful look at a specific musical characteristic, however, can be quite challenging for various types of applications. Many existing musical analysis techniques can examine a particular piece of audio data. For example, the frequency of digital sound can be easily read and identified at a specific section in an audio file. Based on this information, we could determine the musical note being played at that instant, but what if you want to see a list of all the notes played in a song? While most existing methods help to provide information about a single piece of the audio data at a time, few of them can analyze the available audio file on a larger scale. The research conducted in this work considers how to further utilize the examination of audio data by storing more information from the original audio file. In practice, we develop a novel musical analysis system Musicians Aid to process musical representation and examination of audio data. Musicians Aid solves the previous problem by storing and analyzing the audio information as it reads it rather than tossing it aside. The system can provide professional musicians with an insightful look at the music they created and advance their understanding of their work. Amateur musicians could also benefit from using it solely for the purpose of obtaining feedback about a song they were attempting to play. By comparing our system's interpretation of traditional sheet music with their own playing, a musician could ensure what they played was correct. More specifically, the system could show them exactly where they went wrong and how to adjust their mistakes. In addition, the application could be extended over the Internet to allow users to play music with one another and then review the audio data they produced. This would be particularly

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.

Reference-free automatic quality assessment of tracheoesophageal speech.

PubMed

Huang, Andy; Falk, Tiago H; Chan, Wai-Yip; Parsa, Vijay; Doyle, Philip

2009-01-01

Evaluation of the quality of tracheoesophageal (TE) speech using machines instead of human experts can enhance the voice rehabilitation process for patients who have undergone total laryngectomy and voice restoration. Towards the goal of devising a reference-free TE speech quality estimation algorithm, we investigate the efficacy of speech signal features that are used in standard telephone-speech quality assessment algorithms, in conjunction with a recently introduced speech modulation spectrum measure. Tests performed on two TE speech databases demonstrate that the modulation spectral measure and a subset of features in the standard ITU-T P.563 algorithm estimate TE speech quality with better correlation (up to 0.9) than previously proposed features.

The influence of visual speech information on the intelligibility of English consonants produced by non-native speakers.

PubMed

Kawase, Saya; Hannah, Beverly; Wang, Yue

2014-09-01

This study examines how visual speech information affects native judgments of the intelligibility of speech sounds produced by non-native (L2) speakers. Native Canadian English perceivers as judges perceived three English phonemic contrasts (/b-v, θ-s, l-ɹ/) produced by native Japanese speakers as well as native Canadian English speakers as controls. These stimuli were presented under audio-visual (AV, with speaker voice and face), audio-only (AO), and visual-only (VO) conditions. The results showed that, across conditions, the overall intelligibility of Japanese productions of the native (Japanese)-like phonemes (/b, s, l/) was significantly higher than the non-Japanese phonemes (/v, θ, ɹ/). In terms of visual effects, the more visually salient non-Japanese phonemes /v, θ/ were perceived as significantly more intelligible when presented in the AV compared to the AO condition, indicating enhanced intelligibility when visual speech information is available. However, the non-Japanese phoneme /ɹ/ was perceived as less intelligible in the AV compared to the AO condition. Further analysis revealed that, unlike the native English productions, the Japanese speakers produced /ɹ/ without visible lip-rounding, indicating that non-native speakers' incorrect articulatory configurations may decrease the degree of intelligibility. These results suggest that visual speech information may either positively or negatively affect L2 speech intelligibility.

Visual face-movement sensitive cortex is relevant for auditory-only speech recognition.

PubMed

Riedel, Philipp; Ragert, Patrick; Schelinski, Stefanie; Kiebel, Stefan J; von Kriegstein, Katharina

2015-07-01

It is commonly assumed that the recruitment of visual areas during audition is not relevant for performing auditory tasks ('auditory-only view'). According to an alternative view, however, the recruitment of visual cortices is thought to optimize auditory-only task performance ('auditory-visual view'). This alternative view is based on functional magnetic resonance imaging (fMRI) studies. These studies have shown, for example, that even if there is only auditory input available, face-movement sensitive areas within the posterior superior temporal sulcus (pSTS) are involved in understanding what is said (auditory-only speech recognition). This is particularly the case when speakers are known audio-visually, that is, after brief voice-face learning. Here we tested whether the left pSTS involvement is causally related to performance in auditory-only speech recognition when speakers are known by face. To test this hypothesis, we applied cathodal transcranial direct current stimulation (tDCS) to the pSTS during (i) visual-only speech recognition of a speaker known only visually to participants and (ii) auditory-only speech recognition of speakers they learned by voice and face. We defined the cathode as active electrode to down-regulate cortical excitability by hyperpolarization of neurons. tDCS to the pSTS interfered with visual-only speech recognition performance compared to a control group without pSTS stimulation (tDCS to BA6/44 or sham). Critically, compared to controls, pSTS stimulation additionally decreased auditory-only speech recognition performance selectively for voice-face learned speakers. These results are important in two ways. First, they provide direct evidence that the pSTS is causally involved in visual-only speech recognition; this confirms a long-standing prediction of current face-processing models. Secondly, they show that visual face-sensitive pSTS is causally involved in optimizing auditory-only speech recognition. These results are in line

Action Unit Models of Facial Expression of Emotion in the Presence of Speech

PubMed Central

Shah, Miraj; Cooper, David G.; Cao, Houwei; Gur, Ruben C.; Nenkova, Ani; Verma, Ragini

2014-01-01

Automatic recognition of emotion using facial expressions in the presence of speech poses a unique challenge because talking reveals clues for the affective state of the speaker but distorts the canonical expression of emotion on the face. We introduce a corpus of acted emotion expression where speech is either present (talking) or absent (silent). The corpus is uniquely suited for analysis of the interplay between the two conditions. We use a multimodal decision level fusion classifier to combine models of emotion from talking and silent faces as well as from audio to recognize five basic emotions: anger, disgust, fear, happy and sad. Our results strongly indicate that emotion prediction in the presence of speech from action unit facial features is less accurate when the person is talking. Modeling talking and silent expressions separately and fusing the two models greatly improves accuracy of prediction in the talking setting. The advantages are most pronounced when silent and talking face models are fused with predictions from audio features. In this multi-modal prediction both the combination of modalities and the separate models of talking and silent facial expression of emotion contribute to the improvement. PMID:25525561

Nasal and Oral Inspiration During Natural Speech Breathing

PubMed Central

Lester, Rosemary A.; Hoit, Jeannette D.

2015-01-01

Purpose The purpose of this study was to determine the typical pattern for inspiration during speech breathing in healthy adults, as well as the factors that might influence it. Method Ten healthy adults, 18–45 years of age, performed a variety of speaking tasks while nasal ram pressure, audio, and video recordings were obtained. Inspirations were categorized as a nasal only, oral only, simultaneous nasal and oral, or alternating nasal and oral inspiration. The method was validated using nasal airflow, oral airflow, audio, and video recordings for two participants. Results The predominant pattern was simultaneous nasal and oral inspirations for all speaking tasks. This pattern was not affected by the nature of the speaking task or by the phonetic context surrounding the inspiration. The validation procedure confirmed that nearly all inspirations during counting and paragraph reading were simultaneous nasal and oral inspirations; whereas for sentence reading, the predominant pattern was alternating nasal and oral inspirations across the three phonetic contexts. Conclusions Healthy adults inspire through both the nose and mouth during natural speech breathing. This pattern of inspiration is likely beneficial in reducing pathway resistance while preserving some of the benefits of nasal breathing. PMID:24129013

Advances to the development of a basic Mexican sign-to-speech and text language translator

NASA Astrophysics Data System (ADS)

Garcia-Bautista, G.; Trujillo-Romero, F.; Diaz-Gonzalez, G.

2016-09-01

Sign Language (SL) is the basic alternative communication method between deaf people. However, most of the hearing people have trouble understanding the SL, making communication with deaf people almost impossible and taking them apart from daily activities. In this work we present an automatic basic real-time sign language translator capable of recognize a basic list of Mexican Sign Language (MSL) signs of 10 meaningful words, letters (A-Z) and numbers (1-10) and translate them into speech and text. The signs were collected from a group of 35 MSL signers executed in front of a Microsoft Kinect™ Sensor. The hand gesture recognition system use the RGB-D camera to build and storage data point clouds, color and skeleton tracking information. In this work we propose a method to obtain the representative hand trajectory pattern information. We use Euclidean Segmentation method to obtain the hand shape and Hierarchical Centroid as feature extraction method for images of numbers and letters. A pattern recognition method based on a Back Propagation Artificial Neural Network (ANN) is used to interpret the hand gestures. Finally, we use K-Fold Cross Validation method for training and testing stages. Our results achieve an accuracy of 95.71% on words, 98.57% on numbers and 79.71% on letters. In addition, an interactive user interface was designed to present the results in voice and text format.

A Prospectus for the Future Development of a Speech Lab: Hypertext Applications.

ERIC Educational Resources Information Center

Berube, David M.

This paper presents a plan for the next generation of speech laboratories which integrates technologies of modern communication in order to improve and modernize the instructional process. The paper first examines the application of intermediate technologies including audio-video recording and playback, computer assisted instruction and testing…

Timing of Gestures: Gestures Anticipating or Simultaneous with Speech as Indexes of Text Comprehension in Children and Adults

ERIC Educational Resources Information Center

Ianì, Francesco; Cutica, Ilaria; Bucciarelli, Monica

2017-01-01

The deep comprehension of a text is tantamount to the construction of an articulated mental model of that text. The number of correct recollections is an index of a learner's mental model of a text. We assume that another index of comprehension is the timing of the gestures produced during text recall; gestures are simultaneous with speech when…

Speech emotion recognition methods: A literature review

NASA Astrophysics Data System (ADS)

Basharirad, Babak; Moradhaseli, Mohammadreza

2017-10-01

Recently, attention of the emotional speech signals research has been boosted in human machine interfaces due to availability of high computation capability. There are many systems proposed in the literature to identify the emotional state through speech. Selection of suitable feature sets, design of a proper classifications methods and prepare an appropriate dataset are the main key issues of speech emotion recognition systems. This paper critically analyzed the current available approaches of speech emotion recognition methods based on the three evaluating parameters (feature set, classification of features, accurately usage). In addition, this paper also evaluates the performance and limitations of available methods. Furthermore, it highlights the current promising direction for improvement of speech emotion recognition systems.

The Automation System Censor Speech for the Indonesian Rude Swear Words Based on Support Vector Machine and Pitch Analysis

NASA Astrophysics Data System (ADS)

Endah, S. N.; Nugraheni, D. M. K.; Adhy, S.; Sutikno

2017-04-01

According to Law No. 32 of 2002 and the Indonesian Broadcasting Commission Regulation No. 02/P/KPI/12/2009 & No. 03/P/KPI/12/2009, stated that broadcast programs should not scold with harsh words, not harass, insult or demean minorities and marginalized groups. However, there are no suitable tools to censor those words automatically. Therefore, researches to develop a system of intelligent software to censor the words automatically are needed. To conduct censor, the system must be able to recognize the words in question. This research proposes the classification of speech divide into two classes using Support Vector Machine (SVM), first class is set of rude words and the second class is set of properly words. The speech pitch values as an input in SVM, it used for the development of the system for the Indonesian rude swear word. The results of the experiment show that SVM is good for this system.

Calibration of Clinical Audio Recording and Analysis Systems for Sound Intensity Measurement.

PubMed

Maryn, Youri; Zarowski, Andrzej

2015-11-01

Sound intensity is an important acoustic feature of voice/speech signals. Yet recordings are performed with different microphone, amplifier, and computer configurations, and it is therefore crucial to calibrate sound intensity measures of clinical audio recording and analysis systems on the basis of output of a sound-level meter. This study was designed to evaluate feasibility, validity, and accuracy of calibration methods, including audiometric speech noise signals and human voice signals under typical speech conditions. Calibration consisted of 3 comparisons between data from 29 measurement microphone-and-computer systems and data from the sound-level meter: signal-specific comparison with audiometric speech noise at 5 levels, signal-specific comparison with natural voice at 3 levels, and cross-signal comparison with natural voice at 3 levels. Intensity measures from recording systems were then linearly converted into calibrated data on the basis of these comparisons, and validity and accuracy of calibrated sound intensity were investigated. Very strong correlations and quasisimilarity were found between calibrated data and sound-level meter data across calibration methods and recording systems. Calibration of clinical sound intensity measures according to this method is feasible, valid, accurate, and representative for a heterogeneous set of microphones and data acquisition systems in real-life circumstances with distinct noise contexts.

Design Foundations for Content-Rich Acoustic Interfaces: Investigating Audemes as Referential Non-Speech Audio Cues

ERIC Educational Resources Information Center

Ferati, Mexhid Adem

2012-01-01

To access interactive systems, blind and visually impaired users can leverage their auditory senses by using non-speech sounds. The current structure of non-speech sounds, however, is geared toward conveying user interface operations (e.g., opening a file) rather than large theme-based information (e.g., a history passage) and, thus, is ill-suited…

The development of co-speech gesture in the communication of children with autism spectrum disorders.

PubMed

Sowden, Hannah; Clegg, Judy; Perkins, Michael

2013-12-01

Co-speech gestures have a close semantic relationship to speech in adult conversation. In typically developing children co-speech gestures which give additional information to speech facilitate the emergence of multi-word speech. A difficulty with integrating audio-visual information is known to exist for individuals with Autism Spectrum Disorder (ASD), which may affect development of the speech-gesture system. A longitudinal observational study was conducted with four children with ASD, aged 2;4 to 3;5 years. Participants were video-recorded for 20 min every 2 weeks during their attendance on an intervention programme. Recording continued for up to 8 months, thus affording a rich analysis of gestural practices from pre-verbal to multi-word speech across the group. All participants combined gesture with either speech or vocalisations. Co-speech gestures providing additional information to speech were observed to be either absent or rare. Findings suggest that children with ASD do not make use of the facilitating communicative effects of gesture in the same way as typically developing children.

Hierarchical vs non-hierarchical audio indexation and classification for video genres

NASA Astrophysics Data System (ADS)

Dammak, Nouha; BenAyed, Yassine

2018-04-01

In this paper, Support Vector Machines (SVMs) are used for segmenting and indexing video genres based on only audio features extracted at block level, which has a prominent asset by capturing local temporal information. The main contribution of our study is to show the wide effect on the classification accuracies while using an hierarchical categorization structure based on Mel Frequency Cepstral Coefficients (MFCC) audio descriptor. In fact, the classification consists in three common video genres: sports videos, music clips and news scenes. The sub-classification may divide each genre into several multi-speaker and multi-dialect sub-genres. The validation of this approach was carried out on over 360 minutes of video span yielding a classification accuracy of over 99%.

Audio-Vision: Audio-Visual Interaction in Desktop Multimedia.

ERIC Educational Resources Information Center

Daniels, Lee

Although sophisticated multimedia authoring applications are now available to amateur programmers, the use of audio in of these programs has been inadequate. Due to the lack of research in the use of audio in instruction, there are few resources to assist the multimedia producer in using sound effectively and efficiently. This paper addresses the…

Techniques for decoding speech phonemes and sounds: A concept

NASA Technical Reports Server (NTRS)

Lokerson, D. C.; Holby, H. G.

1975-01-01

Techniques studied involve conversion of speech sounds into machine-compatible pulse trains. (1) Voltage-level quantizer produces number of output pulses proportional to amplitude characteristics of vowel-type phoneme waveforms. (2) Pulses produced by quantizer of first speech formants are compared with pulses produced by second formants.

Characteristics of audio and sub-audio telluric signals

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

Telford, W.M.

1977-06-01

Telluric current measurements in the audio and sub-audio frequency range, made in various parts of Canada and South America over the past four years, indicate that the signal amplitude is relatively uniform over 6 to 8 midday hours (LMT) except in Chile and that the signal anisotropy is reasonably constant in azimuth.

Individual Differences in Audio-Vocal Speech Imitation Aptitude in Late Bilinguals: Functional Neuro-Imaging and Brain Morphology

PubMed Central

Reiterer, Susanne Maria; Hu, Xiaochen; Erb, Michael; Rota, Giuseppina; Nardo, Davide; Grodd, Wolfgang; Winkler, Susanne; Ackermann, Hermann

2011-01-01

An unanswered question in adult language learning or late bi and multilingualism is why individuals show marked differences in their ability to imitate foreign accents. While recent research acknowledges that more adults than previously assumed can still acquire a “native” foreign accent, very little is known about the neuro-cognitive correlates of this special ability. We investigated 140 German-speaking individuals displaying varying degrees of “mimicking” capacity, based on natural language text, sentence, and word imitations either in their second language English or in Hindi and Tamil, languages they had never been exposed to. The large subject pool was strictly controlled for previous language experience prior to magnetic resonance imaging. The late-onset (around 10 years) bilinguals showed significant individual differences as to how they employed their left-hemisphere speech areas: higher hemodynamic activation in a distinct fronto-parietal network accompanied low ability, while high ability paralleled enhanced gray matter volume in these areas concomitant with decreased hemodynamic responses. Finally and unexpectedly, males were found to be more talented foreign speech mimics. PMID:22059077

Space Shuttle Orbiter audio subsystem. [to communication and tracking system

NASA Technical Reports Server (NTRS)

Stewart, C. H.

1978-01-01

The selection of the audio multiplex control configuration for the Space Shuttle Orbiter audio subsystem is discussed and special attention is given to the evaluation criteria of cost, weight and complexity. The specifications and design of the subsystem are described and detail is given to configurations of the audio terminal and audio central control unit (ATU, ACCU). The audio input from the ACCU, at a signal level of -12.2 to 14.8 dBV, nominal range, at 1 kHz, was found to have balanced source impedance and a balanced local impedance of 6000 + or - 600 ohms at 1 kHz, dc isolated. The Lyndon B. Johnson Space Center (JSC) electroacoustic test laboratory, an audio engineering facility consisting of a collection of acoustic test chambers, analyzed problems of speaker and headset performance, multiplexed control data coupled with audio channels, and the Orbiter cabin acoustic effects on the operational performance of voice communications. This system allows technical management and project engineering to address key constraining issues, such as identifying design deficiencies of the headset interface unit and the assessment of the Orbiter cabin performance of voice communications, which affect the subsystem development.

The Cleft Care UK study. Part 4: perceptual speech outcomes.

PubMed

Sell, D; Mildinhall, S; Albery, L; Wills, A K; Sandy, J R; Ness, A R

2015-11-01

To describe the perceptual speech outcomes from the Cleft Care UK (CCUK) study and compare them to the 1998 Clinical Standards Advisory Group (CSAG) audit. A cross-sectional study of 248 children born with complete unilateral cleft lip and palate, between 1 April 2005 and 31 March 2007 who underwent speech assessment. Centre-based specialist speech and language therapists (SLT) took speech audio-video recordings according to nationally agreed guidelines. Two independent listeners undertook the perceptual analysis using the CAPS-A Audit tool. Intra- and inter-rater reliability were tested. For each speech parameter of intelligibility/distinctiveness, hypernasality, palatal/palatalization, backed to velar/uvular, glottal, weak and nasalized consonants, and nasal realizations, there was strong evidence that speech outcomes were better in the CCUK children compared to CSAG children. The parameters which did not show improvement were nasal emission, nasal turbulence, hyponasality and lateral/lateralization. These results suggest that centralization of cleft care into high volume centres has resulted in improvements in UK speech outcomes in five-year-olds with unilateral cleft lip and palate. This may be associated with the development of a specialized workforce. Nevertheless, there still remains a group of children with significant difficulties at school entry. © The Authors. Orthodontics & Craniofacial Research Published by John Wiley & Sons Ltd.

Collaborative human-machine analysis to disambiguate entities in unstructured text and structured datasets

NASA Astrophysics Data System (ADS)

Davenport, Jack H.

2016-05-01

Intelligence analysts demand rapid information fusion capabilities to develop and maintain accurate situational awareness and understanding of dynamic enemy threats in asymmetric military operations. The ability to extract relationships between people, groups, and locations from a variety of text datasets is critical to proactive decision making. The derived network of entities must be automatically created and presented to analysts to assist in decision making. DECISIVE ANALYTICS Corporation (DAC) provides capabilities to automatically extract entities, relationships between entities, semantic concepts about entities, and network models of entities from text and multi-source datasets. DAC's Natural Language Processing (NLP) Entity Analytics model entities as complex systems of attributes and interrelationships which are extracted from unstructured text via NLP algorithms. The extracted entities are automatically disambiguated via machine learning algorithms, and resolution recommendations are presented to the analyst for validation; the analyst's expertise is leveraged in this hybrid human/computer collaborative model. Military capability is enhanced by these NLP Entity Analytics because analysts can now create/update an entity profile with intelligence automatically extracted from unstructured text, thereby fusing entity knowledge from structured and unstructured data sources. Operational and sustainment costs are reduced since analysts do not have to manually tag and resolve entities.

Implementation of support vector machine for classification of speech marked hijaiyah letters based on Mel frequency cepstrum coefficient feature extraction

NASA Astrophysics Data System (ADS)

Adhi Pradana, Wisnu; Adiwijaya; Novia Wisesty, Untari

2018-03-01

Support Vector Machine or commonly called SVM is one method that can be used to process the classification of a data. SVM classifies data from 2 different classes with hyperplane. In this study, the system was built using SVM to develop Arabic Speech Recognition. In the development of the system, there are 2 kinds of speakers that have been tested that is dependent speakers and independent speakers. The results from this system is an accuracy of 85.32% for speaker dependent and 61.16% for independent speakers.

Auditory and audio-vocal responses of single neurons in the monkey ventral premotor cortex.

PubMed

Hage, Steffen R

2018-03-20

Monkey vocalization is a complex behavioral pattern, which is flexibly used in audio-vocal communication. A recently proposed dual neural network model suggests that cognitive control might be involved in this behavior, originating from a frontal cortical network in the prefrontal cortex and mediated via projections from the rostral portion of the ventral premotor cortex (PMvr) and motor cortex to the primary vocal motor network in the brainstem. For the rapid adjustment of vocal output to external acoustic events, strong interconnections between vocal motor and auditory sites are needed, which are present at cortical and subcortical levels. However, the role of the PMvr in audio-vocal integration processes remains unclear. In the present study, single neurons in the PMvr were recorded in rhesus monkeys (Macaca mulatta) while volitionally producing vocalizations in a visual detection task or passively listening to monkey vocalizations. Ten percent of randomly selected neurons in the PMvr modulated their discharge rate in response to acoustic stimulation with species-specific calls. More than four-fifths of these auditory neurons showed an additional modulation of their discharge rates either before and/or during the monkeys' motor production of the vocalization. Based on these audio-vocal interactions, the PMvr might be well positioned to mediate higher order auditory processing with cognitive control of the vocal motor output to the primary vocal motor network. Such audio-vocal integration processes in the premotor cortex might constitute a precursor for the evolution of complex learned audio-vocal integration systems, ultimately giving rise to human speech. Copyright © 2018 Elsevier B.V. All rights reserved.

Atypical audio-visual speech perception and McGurk effects in children with specific language impairment

PubMed Central

Leybaert, Jacqueline; Macchi, Lucie; Huyse, Aurélie; Champoux, François; Bayard, Clémence; Colin, Cécile; Berthommier, Frédéric

2014-01-01

Audiovisual speech perception of children with specific language impairment (SLI) and children with typical language development (TLD) was compared in two experiments using /aCa/ syllables presented in the context of a masking release paradigm. Children had to repeat syllables presented in auditory alone, visual alone (speechreading), audiovisual congruent and incongruent (McGurk) conditions. Stimuli were masked by either stationary (ST) or amplitude modulated (AM) noise. Although children with SLI were less accurate in auditory and audiovisual speech perception, they showed similar auditory masking release effect than children with TLD. Children with SLI also had less correct responses in speechreading than children with TLD, indicating impairment in phonemic processing of visual speech information. In response to McGurk stimuli, children with TLD showed more fusions in AM noise than in ST noise, a consequence of the auditory masking release effect and of the influence of visual information. Children with SLI did not show this effect systematically, suggesting they were less influenced by visual speech. However, when the visual cues were easily identified, the profile of responses to McGurk stimuli was similar in both groups, suggesting that children with SLI do not suffer from an impairment of audiovisual integration. An analysis of percent of information transmitted revealed a deficit in the children with SLI, particularly for the place of articulation feature. Taken together, the data support the hypothesis of an intact peripheral processing of auditory speech information, coupled with a supra modal deficit of phonemic categorization in children with SLI. Clinical implications are discussed. PMID:24904454

Atypical audio-visual speech perception and McGurk effects in children with specific language impairment.

PubMed

Leybaert, Jacqueline; Macchi, Lucie; Huyse, Aurélie; Champoux, François; Bayard, Clémence; Colin, Cécile; Berthommier, Frédéric

2014-01-01

Audiovisual speech perception of children with specific language impairment (SLI) and children with typical language development (TLD) was compared in two experiments using /aCa/ syllables presented in the context of a masking release paradigm. Children had to repeat syllables presented in auditory alone, visual alone (speechreading), audiovisual congruent and incongruent (McGurk) conditions. Stimuli were masked by either stationary (ST) or amplitude modulated (AM) noise. Although children with SLI were less accurate in auditory and audiovisual speech perception, they showed similar auditory masking release effect than children with TLD. Children with SLI also had less correct responses in speechreading than children with TLD, indicating impairment in phonemic processing of visual speech information. In response to McGurk stimuli, children with TLD showed more fusions in AM noise than in ST noise, a consequence of the auditory masking release effect and of the influence of visual information. Children with SLI did not show this effect systematically, suggesting they were less influenced by visual speech. However, when the visual cues were easily identified, the profile of responses to McGurk stimuli was similar in both groups, suggesting that children with SLI do not suffer from an impairment of audiovisual integration. An analysis of percent of information transmitted revealed a deficit in the children with SLI, particularly for the place of articulation feature. Taken together, the data support the hypothesis of an intact peripheral processing of auditory speech information, coupled with a supra modal deficit of phonemic categorization in children with SLI. Clinical implications are discussed.

Combinatorial Markov Random Fields and Their Applications to Information Organization

DTIC Science & Technology

2008-02-01

titles, part-of- speech tags; • Image processing: images, colors, texture, blobs, interest points, caption words; • Video processing: video signal, audio...McGurk and MacDonald published their pioneering work [80] that revealed the multi-modal nature of speech perception: sound and moving lips compose one... Speech (POS) n-grams (that correspond to the syntactic structure of text). POS n-grams are extracted from sentences in an incremental manner: the first n

Three-Dimensional Audio Client Library