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Moreover, the matched-window energy-detector model could also account for previous results, including some that were originally interpreted as supporting the multiple-look model. Taken together, at least when detecting faint sounds, listeners appear to attend to the total duration of expected sounds but to ignore their detailed temporal structure.In recent studies, it has been assumed that vocal tract formants (Fn) and the voice source could interact. However, there are only few studies analyzing this assumption in vivo. Here, the vowel transition /i/-/a/-/u/-/i/ of 12 professional classical singers (6 females, 6 males) when phonating on the pitch D4 [fundamental frequency (ƒo) ca. 294 Hz] were analyzed using transnasal high speed videoendoscopy (20.000 fps), electroglottography (EGG), and audio recordings. Fn data were calculated using a cepstral method. Source-filter interaction candidates (SFICs) were determined by (a) algorithmic detection of major intersections of Fn/nƒo and (b) perceptual assessment of the EGG signal. Although the open quotient showed some increase for the /i-a/ and /u-i/ transitions, there were no clear effects at the expected Fn/nƒo intersections. In contrast, ƒo adjustments and changes in the phonovibrogram occurred at perceptually derived SFICs, suggesting level-two interactions. In some cases, these were constituted by intersections between higher nƒo and Fn. The presented data partially corroborates that vowel transitions may result in level-two interactions also in professional singers. However, the lack of systematically detectable effects suggests either the absence of a strong interaction or existence of confounding factors, which may potentially counterbalance the level-two-interactions.Face-to-face speech data collection has been next to impossible globally as a result of the COVID-19 restrictions. To address this problem, simultaneous recordings of three repetitions of the cardinal vowels were made using a Zoom H6 Handy Recorder with an external microphone (henceforth, H6) and compared with two alternatives accessible to potential participants at home the Zoom meeting application (henceforth, Zoom) and two lossless mobile phone applications (Awesome Voice Recorder, and Recorder; henceforth, Phone). F0 was tracked accurately by all of the devices; however, for formant analysis (F1, F2, F3), Phone performed better than Zoom, i.e., more similarly to H6, although the data extraction method (VoiceSauce, Praat) also resulted in differences. In addition, Zoom recordings exhibited unexpected drops in intensity. The results suggest that lossless format phone recordings present a viable option for at least some phonetic studies.Nonlinear ultrasound (NLU) is a nondestructive evaluation method that is sensitive to damage at length scales well below those detected by conventional ultrasonic methods. Micro- and nano-scale damage correlates to the second harmonic generated by a sinusoidal wave as it propagates through a material. However, NLU measurements are plagued by experimentally-induced nonlinearities and require careful calibrations that have limited them to laboratory measurements. Here, we propose the use of additive manufacturing (AM) phononic materials with ultrasonic filtering properties to reduce extraneous nonlinearities. To do this, finite element simulations were first used to design and analyze phononic materials to transmit an ultrasonic wave but forbid the propagation of its second harmonic. Phononic filters were then fabricated with AM and experimentally characterized in the ultrasonic regime. Results show that the phononic materials behave as low-pass filters, where the cut-off frequency is controlled by the unit cell geometry and also influenced by defects and microstructure from the AM process. Finally, the phononic filters were incorporated into NLU measurements, demonstrating the removal of extraneous nonlinearities and thus better isolating second harmonic generation in a test sample. This work suggests that AM phononic materials could improve NLU and other nondestructive evaluation measurements.Beamforming using a circular array of hydrophones may be employed for the task of two-dimensional (2D) underwater sound-field visualisation. In this article, a parametric spatial post-filtering method is proposed, which is specifically intended for applications involving large circular arrays and aims to improve the spatial selectivity of traditional beamformers. In essence, the proposed method is a reformulation of the cross-pattern coherence (CroPaC) spatial post-filter, which involves calculating the normalised cross-spectral density between two signals originating from coincident beamformers. The resulting parameter may be used to sharpen another beamformer steered in the same look-direction, while attenuating ambient noise and interferers from other directions. However, while the original 2D version of the algorithm has been demonstrated to work well with second-order circular harmonic input, it becomes increasingly less suitable with increasing input order. Therefore, the proposed reformulation extends the applicability of CroPaC for much higher orders of circular harmonic input. The method is evaluated with simulated data of a 96-channel circular hydrophone array in three different passive sonar scenarios, where the proposed post-filter is shown to improve the spatial selectivity of both delay-and-sum and minimum-variance distortionless response beamformers.The time-varying multipath introduces major distortions to transmissions in the underwater acoustic communication channel. learn more Channel estimation is often used as one of the central steps to address such distortions in high-rate communication receivers. The focus of this paper is to quantify the impacts of the channel fluctuations on the performance of the least-squares channel estimator. A metric, channel variation ratio (CVR), is defined to describe the rate of fluctuations in the channel impulse responses. Equations are derived to reveal the direct relationships between the CVR and channel estimation performance, which is measured by the channel estimation mean squared error (MSE) and signal prediction error (SPE). The equations show that both the MSE and SPE increase linearly with the CVR. The MSE and SPE metrics both have an error floor for time-varying impulse responses, even with zero ambient noise. It is confirmed that an optimum estimated channel length, achieving the minimum estimation error, exists for time-varying impulse responses.
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