Notes

HO-1/CO technique within tumour growth, angiogenesis as well as metabolic rate * Aimed towards HO-1 as an anti-tumor treatments.
A text file of the FE model is available in supplemental materials; it runs in ANSYS (free version), for which guides are included.In this study, an analysis of the passive acoustic data is carried out for the quantitative characterization of shallow-water acoustic environments from three major estuarine systems of Goa during the months of March and April. The identification of fish sounds was carried out using waveform and peak power spectral densities (PSDs) of the individual fish calls. Fish sound data showed that the toadfish of the Batrachoididae family (Colletteichthys dussumieri species) produced a spectral level 112.27 ± 4.48 dB re 1 μPa2 /Hz at 448.96 ± 40.30 Hz frequency from the mangrove-dominated tidally influenced Mandovi estuary. Similarly, in a coral reef area near Grande Island in the Zuari estuary, Tiger Perch fish from the Terapontidae family (Terapon threaps species) were identified, having spectral levels 106.91 ± 3.08 dB re 1 μPa2 /Hz at 1791.56 ± 106.55 Hz frequency. From the Sal estuary, PSD levels were found to be around 98.24 ± 2.98 dB re 1 μPa2/Hz at 1796.95 ± 72.76 Hz frequency for Tiger Perch of the Terapontidae family (T. threaps species). selleckchem To characterize the contributions of biophony (fish), geophony (wind and flow, etc.), and anthrophony (boats, etc.), cluster analysis is employed. In the Mandovi estuary, the root-mean-square sound pressure level (SPLrms) of broadband toadfish was a function of the water flow and temperature. In the Zuari estuary, SPLrms was a function of the water temperature and wind, whereas in the Sal estuary, wind mainly influenced the SPLrms.The shockwave generated from a focused carbon nanotube (CNT) composite photoacoustic transducer has a wide frequency band that reaches several MHz in a single pulse. The objective of this study was to measure the transmission characteristics of a shockwave generated by a CNT composite photoacoustic transducer through Asian skulls and compare the results with numerical simulation ones. Three Korean cadaver skulls were used, and five sites were measured for each skull. The average densities and sound speeds of the three skulls were calculated from computed tomography images. The sound pressure after skull penetration was about 11% of the one before skull penetration. High-frequency energy was mostly attenuated. The average attenuation coefficients measured at the five sites of the three skulls were 3.59 ± 0.29, 5.99 ± 1.07, and 3.90 ± 0.86 np/cm/MHz. These values were higher than those previously measured at 270, 836, and 1402 kHz from other groups. The attenuation coefficients simulated by Sim4life were slightly smaller than the experimental values, with similar trends at most sites. The attenuation coefficients varied with measurement sites, skull shape, and thickness. These results may provide important data for future applications of shockwaves in noninvasive neurological treatments.The sound field near the tympanic membrane (TM) is estimated based on acoustic data measured at the tip of a probe inserted into the ear canal, from which the area-distance function of the ear canal is calculated. Such information has the potential to quantify the sound input to the middle ear at high frequencies. Spatial variation in the ear-canal cross-sectional area is described acoustically by quantifying forward and reverse sound waves between the probe tip and a near-TM location. A causal acoustic reflection function (RF) measured in the time domain at the probe tip is used to calculate area-distance functions of the ear canal. Area-distance functions are compared with plane-wave methods based on layer peeling and Ware-Aki algorithms. A time-domain model of viscothermal wall loss is devised and applied to ear-canal data, with area-distance functions compared between loss-less and lossy methods. This model is applicable to time-domain RF calibrations using measured data in short tubes. Specification of the near-TM sound field may benefit the interpretation of data from physiological tests such as otoacoustic emission and auditory brainstem responses and high-frequency behavioral tests such as extended audiometry and tests of spatial processing of sound.The apical and basal regions of the cochlea appear functionally distinct. In humans, compelling evidence for an apical-basal transition derives from the phase of otoacoustic emissions (OAEs), whose frequency dependence differs at low and high frequencies. Although OAEs arising from the two major source mechanisms (distortion and reflection) both support the existence of an apical-basal transition-as identified via a prominent bend (or "break") in OAE phase slope-the two OAE types disagree about its precise location along the cochlea. Whereas distortion OAEs at frequency 2f1-f2 suggest that the apical-basal transition occurs near the 2.5 kHz place, reflection OAEs locate the transition closer to 1 kHz. To address this discrepancy, distortion and reflection OAEs were measured and analyzed in 20 young human adults from 0.25-8 kHz and at eight primary-frequency ratios f2/f1 in the range 1-1.5. Break frequencies and OAE phase-gradient delays were estimated by fitting segmented linear models to the unwrapped phase. When distortion- and reflection-OAE phase are considered as functions of ln f2-that is, as linear functions of the location of their putative site of generation within the cochlea-the analysis identifies not just two but three main cochlear segments, meeting at transition frequencies of approximately 0.9 and 2.6 kHz, whose locations are largely independent both of primary-frequency ratio and emission type. A simple model incorporating an abrupt transition from wave- to place-fixed behavior near the middle of the cochlea accounts for key features of distortion-OAE phase.The speech intelligibility index (SII) model was modified to allow individualized parameters. These parameters included the relative weights of speech cues in five octave-frequency bands ranging from 0.25 to 4 kHz, i.e., the band importance function, and the transfer function that allows the SII to generate predictions on speech-recognition scores. A Bayesian adaptive procedure, the quick-band-importance-function (qBIF) procedure, was utilized to enable efficient estimation of the SII parameters from individual listeners. In two experiments, the SII parameters were estimated for 30 normal-hearing adults using Institute of Electrical and Electronics Engineers (IEEE) sentences at speech levels of 55, 65, and 75 dB sound pressure level (in Experiment I) and for 15 hearing-impaired (HI) adult listeners using amplified IEEE or AzBio sentences (in Experiment II). In both experiments, even without prior training, the estimated model parameters showed satisfactory reliability between two runs of the qBIF procedure at least one week apart. For the HI listeners, inter-listener variability in most estimated SII parameters was larger than intra-listener variability of the qBIF procedure.To examine difficulties experienced by cochlear implant (CI) users when perceiving non-native speech, intelligibility of non-native speech was compared in conditions with single and multiple alternating talkers. Compared to listeners with normal hearing, no rapid talker-dependent adaptation was observed and performance was approximately 40% lower for CI users following increased exposure in both talker conditions. Results suggest that lower performance for CI users may stem from combined effects of limited spectral resolution, which diminishes perceptible differences across accents, and limited access to talker-specific acoustic features of speech, which reduces the ability to adapt to non-native speech in a talker-dependent manner.The exact expression for the difference between the acoustic field in an attenuating Pekeris waveguide and the corresponding truncated waveguide is presented. The derivation replaces an earlier version [Evans, J. Acoust. Soc. Am. 74, 193-194 (1983)] that yielded only an approximate estimate of the same difference. The theoretical simplification resulting from the finite truncation is stressed, in contrast to the infinite half-space.The first-order steerable differential arrays (FOSDAs), which have found a variety of applications in speech and audio processing, are usually designed by construction of two orthogonal dipoles using four microphones. Actually, however, three microphones are enough to construct a FOSDA by forming two dipoles with a shared microphone, which is the scheme with the minimum number of microphones and, hence, the most cost-effective. This paper studies the design and analysis of such three-element FOSDAs by using a least-squares method. In particular, the effect of the array configuration on the three-element FOSDA is studied, and optimum array configurations under different steering angle ranges are derived in terms of both beampattern fitting quality and white noise gain. It also reveals that care should be taken to avoid a conservative setting of the steering angle range in order to achieve superior design of the three-element FOSDA.When investigating the wave propagation and mode conversions in a thin aluminum plate partially immersed in water, a kind of wave packet interaction was observed. It was found that the transmitted ultrasonic signal consists of different wave packets, which contain essential information of different wave types. When the incident angle is very small, the signals can be identified as the major wave packet followed by its tail. The major packet includes the information of the incident wave while the tail is related to the mode conversion and propagation in the plate. When the incident angle increased, the major packet was literally engulfed by its tail, indicating that the directly transmitted incident ultrasound disappeared and more energy was coupled into the plate. The interactions between different wave packets found here reveal the excitation and propagation mechanisms of Lamb waves in plates, which would benefit applications in ultrasonic imaging, signal recognition, underwater acoustic communication, and so on.A method for estimating acoustic absorption in porous materials is presented in which the thermal and viscous boundary layers are modeled through boundary conditions to the Helmholtz equation for the acoustic pressure. The method is proposed for rigid-framed porous materials in which vibration of the frame is negligible compared to pressure fluctuations in air. The method reduces computation times by 2 orders of magnitude compared to a full thermoviscous acoustic solver. Furthermore, the method is shown to be highly accurate over geometrical features and frequencies of interest as long as thermal and viscous boundary layers do not overlap and the effects of the sharp changes in curvature are negligible. The method is demonstrated for a periodic sound absorber from the literature as well as a sound absorber with a randomly graded microstructure.This study compares prosodic structural effects on nasal (N) duration and coarticulatory vowel (V) nasalization in NV (Nasal-Vowel) and CVN (Consonant-Vowel-Nasal) sequences in Mandarin Chinese with those found in English and Korean. Focus-induced prominence effects show cross-linguistically applicable coarticulatory resistance that enhances the vowel's phonological features. Boundary effects on the initial NV reduced N's nasality without having a robust effect on V-nasalization, whose direction is comparable to that in English and Korean. Boundary effects on the final CVN showed language specificity of V-nasalization, which could be partly attributable to the ongoing sound change of coda nasal lenition in Mandarin.
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