CIoTVID: Towards an Open IoT-Platform for Infective Pandemic Diseases such as COVID-19.

The factors affecting the penetration of certain diseases such as COVID-19 in society are still unknown. Internet of Things (IoT) technologies can play a crucial role during the time of crisis and they can provide a more holistic view of the reasons that govern the outbreak of a contagious disease. The understanding of COVID-19 will be enriched by the analysis of data related to the phenomena, and this data can be collected using IoT sensors. In this paper, we show an integrated solution based on IoT technologies that can serve as opportunistic health data acquisition agents for combating the pandemic of COVID-19, named CIoTVID. The platform is composed of four layers-data acquisition, data aggregation, machine intelligence and services, within the solution. To demonstrate its validity, the solution has been tested with a use case based on creating a classifier of medical conditions using real data of voice, performing successfully. The layer of data aggregation is particularly relevant in this kind of solution as the data coming from medical devices has a very different nature to that coming from electronic sensors. Due to the adaptability of the platform to heterogeneous data and volumes of data; individuals, policymakers, and clinics could benefit from it to fight the propagation of the pandemic.

Effect of impact pile driving noise on marine mammals: A comparison of different noise exposure criteria.

Regulators in Europe and in the United States have developed sound exposure criteria. Criteria range from broadband levels to frequency weighted received sound levels. The associated differences in impact assessment results are, however, not yet understood. This uncertainty makes environmental management of transboundary anthropogenic noise challenging and causes confusion for regulators who need to choose appropriate exposure criteria. In the present study, three established exposure criteria frameworks from Germany, Denmark, and the US were used to analyse the effect of impact pile driving at a location in the Baltic Sea on harbor porpoise and harbor seal hearing. The acoustic modeling using MIKE showed that an unmitigated scenario would lead to auditory injury for all three criteria. Despite readily apparent variances in impact ranges among the applied approaches, it was also evident that noise mitigation measures could reduce underwater sound to levels where auditory injuries would be unlikely in most cases. It was concluded that each of the frameworks has its own advantages and disadvantages. Single noise exposure criteria follow the precautionary principle and can be enforced relatively easily, whereas criteria that consider hearing capabilities and animal response movement can improve the accuracy of the assessment if data are available.

Cross-language differences in how voice quality and f 0 contours map to affect.

The relationship between prosody and perceived affect involves multiple variables. This paper explores the interplay of three: voice quality, f 0 contour, and the hearer's language background. Perception tests were conducted with speakers of Irish English, Russian, Spanish, and Japanese using three types of synthetic stimuli: (1) stimuli varied in voice quality, (2) stimuli of uniform (modal) voice quality incorporating affect-related f 0 contours, and (3) stimuli combining specific non-modal voice qualities with the affect-related f 0 contours of (2). The participants rated the stimuli for the presence/strength of affective colouring on six bipolar scales, e.g., happy-sad. The results suggest that stimuli incorporating non-modal voice qualities, with or without f 0 variation, are generally more effective in affect cueing than stimuli varying only in f 0. Along with similarities in the affective responses across these languages, many points of divergence were found, both in terms of the range and strength of affective responses overall and in terms of specific stimulus-to-affect associations. The f 0 contour may play a more important role, and tense voice a lesser role in affect signalling in Japanese and Spanish than in Irish English and Russian. The greatest cross-language differences emerged for the affects intimate, formal, stressed, and relaxed.

Effects of Slow- and Fast-Acting Compression on Hearing-Impaired Listeners' Consonant-Vowel Identification in Interrupted Noise.

There is conflicting evidence about the relative benefit of slow- and fast-acting compression for speech intelligibility. It has been hypothesized that fast-acting compression improves audibility at low signal-to-noise ratios (SNRs) but may distort the speech envelope at higher SNRs. The present study investigated the effects of compression with a nearly instantaneous attack time but either fast (10 ms) or slow (500 ms) release times on consonant identification in hearing-impaired listeners. Consonant-vowel speech tokens were presented at a range of presentation levels in two conditions: in the presence of interrupted noise and in quiet (with the compressor "shadow-controlled" by the corresponding mixture of speech and noise). These conditions were chosen to disentangle the effects of consonant audibility and noise-induced forward masking on speech intelligibility. A small but systematic intelligibility benefit of fast-acting compression was found in both the quiet and the noisy conditions for the lower speech levels. No detrimental effects of fast-acting compression were observed when the speech level exceeded the level of the noise. These findings suggest that fast-acting compression provides an audibility benefit in fluctuating interferers when compared with slow-acting compression while not substantially affecting the perception of consonants at higher SNRs.

Rapid Adaptation to the Timbre of Natural Sounds.

Timbre, the unique quality of a sound that points to its source, allows us to quickly identify a loved one's voice in a crowd and distinguish a buzzy, bright trumpet from a warm cello. Despite its importance for perceiving the richness of auditory objects, timbre is a relatively poorly understood feature of sounds. Here we demonstrate for the first time that listeners adapt to the timbre of a wide variety of natural sounds. For each of several sound classes, participants were repeatedly exposed to two sounds (e.g., clarinet and oboe, male and female voice) that formed the endpoints of a morphed continuum. Adaptation to timbre resulted in consistent perceptual aftereffects, such that hearing sound A significantly altered perception of a neutral morph between A and B, making it sound more like B. Furthermore, these aftereffects were robust to moderate pitch changes, suggesting that adaptation to timbral features used for object identification drives these effects, analogous to face adaptation in vision.

Sensorimotor integration on a rapid time scale.

Sensing is fundamental to the control of movement: From grasping objects to speech production, sensing guides action. So far, most of our knowledge about sensorimotor integration comes from visually guided reaching and oculomotor integration, in which the time course and trajectories of movements can be measured at a high temporal resolution. By contrast, production of vocalizations by humans and animals involves complex and variable actions, and each syllable often lasts a few hundreds of milliseconds, making it difficult to infer underlying neural processes. Here, we measured and modeled the transfer of sensory information into motor commands for vocal amplitude control in response to background noise, also known as the Lombard effect. We exploited the brief vocalizations of echolocating bats to trace the time course of the Lombard effect on a millisecond time scale. Empirical studies revealed that the Lombard effect features a response latency of a mere 30 ms and provided the foundation for the quantitative audiomotor model of the Lombard effect. We show that the Lombard effect operates by continuously integrating the sound pressure level of background noise through temporal summation to guide the extremely rapid vocal-motor adjustments. These findings can now be extended to models and measures of audiomotor integration in other animals, including humans.

[Technical advancements in cochlear implants : State of the art]. / Technische Entwicklungen bei Cochleaimplantaten : Stand der Technik.

Twenty years ago, cochlear implants (CI) were indicated only in cases of profound hearing loss or complete deafness. While from today's perspective the technology was clumsy and provided patients with only limited speech comprehension in quiet scenarios, successive advances in CI technology and the consequent substantial hearing improvements over time have since then resulted in continuous relaxation of indication criteria toward residual hearing. While achievements in implant and processor electronics have been one key factor for the ever-improving hearing performance, development of electro-acoustic CI systems-together with atraumatic implantation concepts-has led to enormous improvements in patients with low-frequency residual hearing. Manufactures have designed special processors with integrated hearing aid components for this patient group, which are capable of conveying acoustic and electric stimulation. A further milestone in improvement of hearing in challenging listening environments was the adoption of signal enhancement algorithms and assistive listening devices from the hearing aid industry. This article gives an overview of the current state of the art in the abovementioned areas of CI technology.

Loudness Contour Can Influence Mandarin Tone Recognition: Vocoder Simulation and Cochlear Implants.

Lexical tone recognition with current cochlear implants (CI) remains unsatisfactory due to significantly degraded pitch-related acoustic cues, which dominate the tone recognition by normal-hearing (NH) listeners. Several secondary cues (e.g., amplitude contour, duration, and spectral envelope) that influence tone recognition in NH listeners and CI users have been studied. This work proposes a loudness contour manipulation algorithm, namely Loudness-Tone (L-Tone), to investigate the effects of loudness contour on Mandarin tone recognition and the effectiveness of using loudness cue to enhance tone recognition for CI users. With L-Tone, the intensity of sound samples is multiplied by gain values determined by instantaneous fundamental frequencies (F0s) and pre-defined gain- F0 mapping functions. Perceptual experiments were conducted with a four-channel noise-band vocoder simulation in NH listeners and with CI users. The results suggested that 1) loudness contour is a useful secondary cue for Mandarin tone recognition, especially when pitch cues are significantly degraded; 2) L-Tone can be used to improve Mandarin tone recognition in both simulated and actual CI-hearing without significant negative effect on vowel and consonant recognition. L-Tone is a promising algorithm for incorporation into real-time CI processing and off-line CI rehabilitation training software.

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

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

The use of Digital Sound in Health.

With the advent of computing and micro computing, several applications based on Digital Sound have emerged around the world. Much analog equipment and electronics have gained new functions due to evolution and the low cost of microprocessors and integrated circuits. The rapidly expanding user community of such equipment allowed the advancement of research and development of numerous applications in which the Digital Sound participates actively, including, tests, treatments and therapies. In addition to solutions for use in laboratories, clinics and hospitals, there emerged devices for domestic use and handling, enabling faster dissemination and exploitation of advances, and providing the necessary feedback in the evolution of technologies applied. The massive use of Digital Sound encouraged research involving frequency bands used widely in health equipment (infrasounds and ultrasounds). Through the methodology of literature review, this paper seeks to explain the evolution of different applications of Digital Sound in Health, as well as indicate future research.

Aural proxies and directionally-varying reverberation for interactive sound propagation in virtual environments.

We present an efficient algorithm to compute spatially-varying, direction-dependent artificial reverberation and reflection filters in large dynamic scenes for interactive sound propagation in virtual environments and video games. Our approach performs Monte Carlo integration of local visibility and depth functions to compute directionally-varying reverberation effects. The algorithm also uses a dynamically-generated rectangular aural proxy to efficiently model 2-4 orders of early reflections. These two techniques are combined to generate reflection and reverberation filters which vary with the direction of incidence at the listener. This combination leads to better sound source localization and immersion. The overall algorithm is efficient, easy to implement, and can handle moving sound sources, listeners, and dynamic scenes, with minimal storage overhead. We have integrated our approach with the audio rendering pipeline in Valve's Source game engine, and use it to generate realistic directional sound propagation effects in indoor and outdoor scenes in real-time. We demonstrate, through quantitative comparisons as well as evaluations, that our approach leads to enhanced, immersive multi-modal interaction.

Spectro-temporal weighting of loudness.

Real-world sounds like speech or traffic noise typically exhibit spectro-temporal variability because the energy in different spectral regions evolves differently as a sound unfolds in time. However, it is currently not well understood how the energy in different spectral and temporal portions contributes to loudness. This study investigated how listeners weight different temporal and spectral components of a sound when judging its overall loudness. Spectral weights were measured for the combination of three loudness-matched narrowband noises with different center frequencies. To measure temporal weights, 1,020-ms stimuli were presented, which randomly changed in level every 100 ms. Temporal weights were measured for each narrowband noise separately, and for a broadband noise containing the combination of the three noise bands. Finally, spectro-temporal weights were measured with stimuli where the level of the three narrowband noises randomly and independently changed every 100 ms. The data consistently showed that (i) the first 300 ms of the sounds had a greater influence on overall loudness perception than later temporal portions (primacy effect), and (ii) the lowest noise band contributed significantly more to overall loudness than the higher bands. The temporal weights did not differ between the three frequency bands. Notably, the spectral weights and temporal weights estimated from the conditions with only spectral or only temporal variability were very similar to the corresponding weights estimated in the spectro-temporal condition. The results indicate that the temporal and the spectral weighting of the loudness of a time-varying sound are independent processes. The spectral weights remain constant across time, and the temporal weights do not change across frequency. The results are discussed in the context of current loudness models.

Evaluation of a transient noise reduction strategy for hearing AIDS.

BACKGROUND: Transient noise can be disruptive for people wearing hearing aids. Ideally, the transient noise should be detected and controlled by the signal processor without disrupting speech and other intended input signals. A technology for detecting and controlling transient noises in hearing aids was evaluated in this study. PURPOSE: The purpose of this study was to evaluate the effectiveness of a transient noise reduction strategy on various transient noises and to determine whether the strategy has a negative impact on sound quality of intended speech inputs. RESEARCH DESIGN: This was a quasi-experimental study. The study involved 24 hearing aid users. Each participant was asked to rate the parameters of speech clarity, transient noise loudness, and overall impression for speech stimuli under the algorithm-on and algorithm-off conditions. During the evaluation, three types of stimuli were used: transient noises, speech, and background noises. The transient noises included "knife on a ceramic board," "mug on a tabletop," "office door slamming," "car door slamming," and "pen tapping on countertop." The speech sentences used for the test were presented by a male speaker in Mandarin. The background noises included "party noise" and "traffic noise." All of these sounds were combined into five listening situations: (1) speech only, (2) transient noise only, (3) speech and transient noise, (4) background noise and transient noise, and (5) speech and background noise and transient noise. RESULTS: There was no significant difference on the ratings of speech clarity between the algorithm-on and algorithm-off (t-test, p = 0.103). Further analysis revealed that speech clarity was significant better at 70 dB SLP than 55 dB SPL (p < 0.001). For transient noise loudness: under the algorithm-off condition, the percentages of subjects rating the transient noise to be somewhat soft, appropriate, somewhat loud, and too loud were 0.2, 47.1, 29.6, and 23.1%, respectively. The corresponding percentages under the algorithm-on were 3.0, 72.6, 22.9, and 1.4%, respectively. A significant difference on the ratings of the transient noise loudness was found between the algorithm-on and algorithm-off (t-test, p < 0.001). For overall impression for speech stimuli: under the algorithm-off condition, the percentage of subjects rating the algorithm to be not helpful at all, somewhat helpful, helpful, and very helpful for speech stimuli were 36.5, 20.8, 33.9, and 8.9%, respectively. Under the algorithm-on condition, the corresponding percentages were 35.0, 19.3, 30.7, and 15.0%, respectively. Statistical analysis revealed there was a significant difference on the ratings of overall impression on speech stimuli. The ratings under the algorithm-on condition were significantly more helpful for speech understanding than the ratings under algorithm-off (t-test, p < 0.001). CONCLUSIONS: The transient noise reduction strategy appropriately controlled the loudness for most of the transient noises and did not affect the sound quality, which could be beneficial to hearing aid wearers.

The use of the mobile voice laboratory in the operating room during type I thyroplasty with Gore-Tex(®).

INTRODUCTION AND OBJECTIVE: The purposes of this study are to demonstrate the use of the mobile voice lab in type I thyroplasty with Gore-Tex(®) using analysis of spectrogram and fundamental frequency in the operating room, and also to show how to do this procedure. METHODS: Voice samples were recorded in the operating room immediately before and during type I thyroplasty. Six-week postoperative samples were also taken in the voice laboratory. Fundamental frequency and spectral analysis were analyzed. Spectrograms were evaluated by blind panel of 4 judges on a 100mm visual analogue scale. All three time points were compared and statistical analysis performed. Pre and postoperative V-RQOL scores were also compared. RESULTS: Significant improvement in spectrogram ratings were seen between before and during (P<.001), and before and after voice samples (P<.017). There was no significant difference between during and after scores, suggesting the persistence of the intraoperative improvement in this measure. Changes in fundamental frequency were not statistically significant, although fundamental frequency tended to increase in women and decrease in men after type I thyroplasty. Mean V-RQOL scores improved from 48.08 a 85.08 (P<.001). CONCLUSIONS: The mobile voice laboratory may be useful during type I thyroplasty with Gore-Tex(®). It offers an opportunity for the surgeon and voice pathologist to continue to collaborate in the treatment of patients with unilateral vocal fold paralysis.

Neural coding of syntactic structure in learned vocalizations in the songbird.

Although vocal signals including human languages are composed of a finite number of acoustic elements, complex and diverse vocal patterns can be created from combinations of these elements, linked together by syntactic rules. To enable such syntactic vocal behaviors, neural systems must extract the sequence patterns from auditory information and establish syntactic rules to generate motor commands for vocal organs. However, the neural basis of syntactic processing of learned vocal signals remains largely unknown. Here we report that the basal ganglia projecting premotor neurons (HVC(X) neurons) in Bengalese finches represent syntactic rules that generate variable song sequences. When vocalizing an alternative transition segment between song elements called syllables, sparse burst spikes of HVC(X) neurons code the identity of a specific syllable type or a specific transition direction among the alternative trajectories. When vocalizing a variable repetition sequence of the same syllable, HVC(X) neurons not only signal the initiation and termination of the repetition sequence but also indicate the progress and state-of-completeness of the repetition. These different types of syntactic information are frequently integrated within the activity of single HVC(X) neurons, suggesting that syntactic attributes of the individual neurons are not programmed as a basic cellular subtype in advance but acquired in the course of vocal learning and maturation. Furthermore, some auditory-vocal mirroring type HVC(X) neurons display transition selectivity in the auditory phase, much as they do in the vocal phase, suggesting that these songbirds may extract syntactic rules from auditory experience and apply them to form their own vocal behaviors.

How accentuation influences semantic short-term memory representations during on-line speech processing: an event-related potential study.

In the present study, the ERP (event-related brain potentials) technique was used to investigate how accentuation influences the semantic short-term memory representations during on-line speech processing, and how this accentuation effect interacts with the distance of accentuation in the speech signal. Chinese spoken sentences were used as stimuli. The sentences included two critical words: Noun1 and Noun2, with the ERP time-locked to Noun2. During sentence comprehension, when the listener hears Noun2, he needs to retrieve Noun1 from the working memory and integrate it with Noun2. We manipulated the (de-)accentuation of Noun1 and the semantic relationship between Noun1 and Noun2 by changing Noun1 in the sentence context. Moreover, we manipulated the distance of accentuation (distance between Noun1 and Noun2) by changing the syntactic structure of the sentences. The results revealed a significant main effect of semantic relatedness, indicating that the low semantic relatedness condition elicited a larger N400 than the high semantic relatedness condition. Importantly, there was a significant two-way interaction between semantic relatedness and accentuation and a significant three-way interaction between semantic relatedness, accentuation, and distance. Further analysis demonstrated that, the semantic relatedness effect was modulated by accentuation in the long-distance sentences, but not in the short-distance sentences. That is, in the long-distance sentences, the semantic relatedness effect reached significance only when the to-be-integrated expression in the preceding sentence context was accented; however, in the short-distance sentences, the semantic relatedness effects reached significance regardless of the presence or absence of accentuation. The results indicated that, during on-line speech processing, accentuation can enhance the corresponding information's semantic short-term memory representation, and that the effect of accentuation on semantic short-term memory is somewhat flexible and shows up only when the words in the speech signal were far apart.

Recovering sound sources from embedded repetition.

Cocktail parties and other natural auditory environments present organisms with mixtures of sounds. Segregating individual sound sources is thought to require prior knowledge of source properties, yet these presumably cannot be learned unless the sources are segregated first. Here we show that the auditory system can bootstrap its way around this problem by identifying sound sources as repeating patterns embedded in the acoustic input. Due to the presence of competing sounds, source repetition is not explicit in the input to the ear, but it produces temporal regularities that listeners detect and use for segregation. We used a simple generative model to synthesize novel sounds with naturalistic properties. We found that such sounds could be segregated and identified if they occurred more than once across different mixtures, even when the same sounds were impossible to segregate in single mixtures. Sensitivity to the repetition of sound sources can permit their recovery in the absence of other segregation cues or prior knowledge of sounds, and could help solve the cocktail party problem.

Hearing level equalized otoacoustic emissions acquired by swept-tones: intensity characteristics.

Otoacoustic emissions (OAE), which are acoustic responses produced by the cochlea, can be recorded with a microphone in the ear canal to give diagnostic information regarding cochlear functioning. Recently, the researchers developed a novel stimulus for the acquisition of OAE using a hearing-level equalized (HL(eq)) swept-tone signal. The objective of this study was to observe OAE characteristics at a multitude of intensities to track the changes in temporal and spectral morphology. An increase in high-frequency emissions was found as stimulation intensity decreased. Furthermore, it was found that hearing level equalized swept-tone OAEs (HL(eq) sTEAOE) can be acquired at very low intensities, which is not typical under current acquisition modalities. This may result in clinical improvements by providing a fast and cheap method for contributing to the detection of auditory thresholds.

Effects of damage to auditory cortex on the discrimination of speech sounds by rats.

The intensity of a noise-induced startle response can be reduced by the presentation of an otherwise neutral stimulus immediately before the noise ("prepulse inhibition" or PPI). We used a form of PPI to study the effects of damage to auditory cortex on the discrimination of speech sounds by rats. Subjects underwent control surgery or treatment of the auditory cortex with the vasoconstrictor endothelin-1. This treatment caused damage concentrated in primary auditory cortex (A1). Both before and after lesions, subjects were tested on 5 tasks, most presenting a pair of human speech sounds (consonant-vowel syllables) so that the capacity for discrimination would be evident in the extent of PPI. Group comparisons failed to reveal any consistent lesion effect. At the same time, the analysis of individual differences in performance by multiple regression suggests that some of the temporal processing required to discriminate speech sounds is concentrated anteroventrally in the right A1. These results also confirm that PPI can be adapted to studies of the brain mechanisms involved in the processing of speech and other complex sounds.

High resolution system for improved transient-evoked otoacoustic emission acquisition.

Transient-evoked otoacoustic emissions (TEOAE) are generated by the cochlea in response to clicks. They are obtained by averaging post-onset acoustic responses which are composed of the stimulus-related meatal response (MR) and the TEOAEs. TEOAEs are typically below normal hearing thresholds and are obstructed by the MR, which is several orders of magnitude higher. For click stimuli, MRs typically last about 5 ms and obstruct the early latency emissions. TEOAEs become compressively nonlinear as the MR increases, and this property is commonly exploited by obtaining a derived nonlinear response (DNLR) which reduces the MR artifact. In this study we report the development of a high-resolution system which linearly acquires both MRs and TEOAEs. Results show that the duration of the artifact can be reduced, making the high frequency content of TEOAEs observable.