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Synchronous laser beam scanning is a common technique used in single-photon imaging where the spatial resolution is primarily determined by the beam divergence angle. In this context, Bessel beams have been investigated as they can overcome the diffraction limit associated with traditional Gaussian beams. Notably, the central spot of a Bessel beam retains its size almost unchanged within a non-diffractive distance. However, the presence of sidelobes in the Bessel beam can negatively impact spatial resolution. To address this challenge, we have developed a single-photon imaging system with high-depth resolution, which allows for the suppression of echo photons from the sidelobe light in the depth image, particularly when their flight time differs from that of the central spot. In our LiDAR setup, we successfully achieved high-resolution scanning imaging with a spatial resolution of approximately 0.5 mm while also demonstrating a high-depth resolution of 12 mm. [ABSTRACT FROM AUTHOR]

Objective: The excellent blood and fat suppression of stimulated echo acquisition mode (STEAM) can be combined with saturation recovery single-shot acquisition (SASHA) in a novel STEAM-SASHA sequence for right ventricular (RV) native T1 mapping. Materials and methods: STEAM-SASHA splits magnetization preparation over two cardiac cycles, nulling blood signal and allowing fat signal to decay. Breath-hold T1 mapping was performed in a T1 phantom and twice in 10 volunteers using STEAM-SASHA and a modified Look-Locker sequence at peak systole at 3T. T1 was measured in 3 RV regions, the septum and left ventricle (LV). Results: In phantoms, MOLLI under-estimated while STEAM-SASHA over-estimated T1, on average by 3.0% and 7.0% respectively, although at typical 3T myocardial T1 (T1 > 1200 ms) STEAM-SASHA was more accurate. In volunteers, T1 was higher using STEAM-SASHA than MOLLI in the LV and septum (p = 0.03, p = 0.006, respectively), but lower in RV regions (p > 0.05). Inter-study, inter-observer and intra-observer coefficients of variation in all regions were < 15%. Blood suppression was excellent with STEAM-SASHA and noise floor effects were minimal. Discussion: STEAM-SASHA provides accurate and reproducible T1 in the RV with excellent blood and fat suppression. STEAM-SASHA has potential to provide new insights into pathological changes in the RV in future studies. [ABSTRACT FROM AUTHOR]

Abstract A great interest has been directed towards high frequency surface wave radar (HFSWR), because it is applied as long‐range early warning and real‐time measurements of sea surface condition tools in maritime surveillance. However, the unwanted reflection from the ionosphere (ionosphere clutter) damages the consistent detection performance of HFSWR. Target detection and environmental information extraction from radar echoes are primarily determined by the extent to which the medium contaminates the transmitted signals. Although the ionospheric signal corruption mechanisms are complex and various, a number of eliminating contamination approaches have been developed in the last 50 years. The purpose of this paper is providing an overview of the past and current clutter theoretical models and clutter suppression techniques for HFSWR system. The experimental results involved in the complex long‐term ionosphere observation and ionosphere echo analysis are first discussed. Special attention is paid to the correlation between the signal degradation sources and eliminating clutter techniques. Furthermore, the clutter suppression methods are classified based on the design idea. The specific clutter suppression methods as well as its operation principle are briefly introduced. Finally, several research trends and open issues are presented, with emphasis on the need for proactively controlling the radar design cost and generally agreed‐upon standards.

In communication systems, when the loudspeaker and the microphone are coupled together, it creates acoustic echoes. With the increasing demand for mobile communication and online conference, it is urgent to solve the problem of acoustic echo cancellation (AEC) in communication systems. Due to the existence of nonlinear distortion, background noise and other reasons, traditional AEC methods can no longer solve the problem of echo cancellation well. Although some traditional methods consider the problem of nonlinear distortion, the effect of echo suppression is still not ideal. In this paper, we propose an echo cancellation method based on frequency domain mask, which is defined as a supervised speech separation problem. The use of the temporal convolutional network and optimal ratio mask to obtain the predicted mask, as well as the use of SISNR as the loss function, have been shown to effectively reduce echo in double-talk, nonlinear distortion, and background noise. This method is a significant advancement in the field of AEC and can be used in for mobile communication and online conference. [ABSTRACT FROM AUTHOR]

A great interest has been directed towards high frequency surface wave radar (HFSWR), because it is applied as long‐range early warning and real‐time measurements of sea surface condition tools in maritime surveillance. However, the unwanted reflection from the ionosphere (ionosphere clutter) damages the consistent detection performance of HFSWR. Target detection and environmental information extraction from radar echoes are primarily determined by the extent to which the medium contaminates the transmitted signals. Although the ionospheric signal corruption mechanisms are complex and various, a number of eliminating contamination approaches have been developed in the last 50 years. The purpose of this paper is providing an overview of the past and current clutter theoretical models and clutter suppression techniques for HFSWR system. The experimental results involved in the complex long‐term ionosphere observation and ionosphere echo analysis are first discussed. Special attention is paid to the correlation between the signal degradation sources and eliminating clutter techniques. Furthermore, the clutter suppression methods are classified based on the design idea. The specific clutter suppression methods as well as its operation principle are briefly introduced. Finally, several research trends and open issues are presented, with emphasis on the need for proactively controlling the radar design cost and generally agreed‐upon standards. [ABSTRACT FROM AUTHOR]

The pinning of domain walls in cobalt micropowders, nanopowders, and nanowires is studied by the double-pulse spin-echo NMR method under an additional magnetic video pulse. Cobalt micropowders are obtained by melting in an induction furnace. Nanopowders are produced using electron beam technology and chemical deposition. This chemical deposition reaction is carried out in an external magnetic field to obtain nanowires. The pinning in these systems is measured as a function of the long-term magnetic video pulse and its amplitude. It is established that the magnetic video pulse area is constant for all its threshold values corresponding to the beginning of the double-pulse echo signal suppression. The linear nature of the pinning dependence on the external magnetic field magnitude in cobalt micropowders in a wider range of the external magnetic field change compared to lithium ferrite is presented. Alternative information about the pinning strength of domain walls in magnets can also be obtained by studying the magnetic video pulse influence on the magnetic echo signal formed by the joint action of radio frequency and magnetic pulses. These NMR methods can be used for microscopic control of the domain wall properties in the studied magnets with the aim of their potential use in functional materials, memory devices, and sensors. [ABSTRACT FROM AUTHOR]

Every year the number of IoT devices is growing, which gives a boost to development of such technology as Power Line Communication (PLC). The term PLC refers to the use of a power supply circuit to transmit data. Full-duplex technology can be applied to increase the data transfer rate in a PLC. The difficulty in using a full duplex is the need to suppress the transmitter's own signal (interference signal). Analog cancellation and digital cancellation are used for this purpose. Common algorithms for digital cancellation are adaptive algorithms. In this paper adaptive algorithms such as Particle Swarm Optimization (PSO) and Recursive Least Squares (RLS) are considered and a modification of the classical RLS is proposed. The proposed algorithm uses precalculated weight coefficients together with periodic reinitialization of the filter. This avoids the influence of significant changes in the input signal on the further process of filter adaptation and provides the highest level of echo suppression. The proposed algorithm provides an improvement of parameters such as Error Vector Magnitude (EVM) and Signal-to-Interference Ratio (SIR) in comparison with classical RLS and PSO, according to the results of the study. [ABSTRACT FROM AUTHOR]

This paper presents an echo suppression system that combines a linear acoustic echo canceller (AEC) with a deep complex convolutional recurrent network (DCCRN) for residual echo suppression. The filter taps of the AEC are adjusted in subbands by using the normalized sign-error least mean squares (NSLMS) algorithm. The NSLMS is compared with the commonly-used normalized least mean squares (NLMS), and the combination of each with the proposed deep residual echo suppression model is studied. The utilization of a pre-trained deep-learning speech denoising model as an alternative to a residual echo suppressor (RES) is also studied. The results showed that the performance of the NSLMS is superior to that of the NLMS in all settings. With the NSLMS output, the proposed RES achieved better performance than the larger pre-trained speech denoiser model. More notably, the denoiser performed considerably better on the NSLMS output than on the NLMS output, and the performance gap was greater than the respective gap when employing the RES, indicating that the residual echo in the NSLMS output was more akin to noise than speech. Therefore, when little data is available to train an RES, a pre-trained speech denoiser is a viable alternative when employing the NSLMS for the preceding linear AEC. [ABSTRACT FROM AUTHOR]

Immersion ultrasonic phased array imaging technology offers great advantages, particularly in coupling and automatic detection of industrial non-destructive testing (NDT). To suppress the influence of secondary interface echoes in the immersion ultrasonic phased array imaging, a novel phase circular statistics vector (PCSV) weighting method is proposed in this paper. Firstly, the PCSV factor matrix is established according to the phase consistency of the echo signals. Secondly, due to the higher phase coherence of the defect echo, the PCSV factor matrix is used to weight the TFM image to suppress the secondary interface echo. The result shows the secondary interface echoes are effectively suppressed in the total focusing method (TFM) image on a 0~40 dB scale. It is also shown that PCSV weighting could not only suppress the secondary interface echoes but also improved the image quality in terms of SNR and lateral resolution by comparing with traditional TFM. [ABSTRACT FROM AUTHOR]

Cetaceans show high cognitive abilities and strong social bonds. Their primary sensory modality to communicate and sense the environment is acoustics. Research on their echolocation and social vocalizations typically uses visual and tactile systems adapted from research on primates or birds. Such research would benefit from a purely acoustic communication system to better match their natural capabilities. We argue that a full duplex system, in which signals can flow in both directions simultaneously is essential for communication research. We designed and implemented a full duplex system to acoustically interact with cetaceans in the wild, featuring digital echo-suppression. We pilot tested the system in Arctic Norway and achieved an echo suppression of 18 dB. We discuss the limiting factors and how to improve the echo suppression further. The system enabled vocal interaction with the underwater acoustic scene by allowing experimenters to listen while producing sounds. We describe our motivations, then present our pilot deployment and give examples of initial explorative attempts to vocally interact with wild orcas and humpback whales. [ABSTRACT FROM AUTHOR]

This special issue includes a selection of works covering a wide range of topics such as human-human, human-machine, human-animal and animal-animal vocal-based interactions. Jover & Gratier present three pilot studies of parent-infant dyadic multimodal interactions, with the aim of developing new methodological pathways for the study of early naturalistic social interaction. Almost all animals exploit vocal signals for a range of ecologically motivated purposes: from detecting predators/prey and marking territory, to expressing emotions, establishing social relations and sharing information. [Extracted from the article]

The echo cancellation algorithm plays a vital role in the acoustic systems, but it has the defects of long convergence time and high complexity. In this study, a low-complexity echo suppression scheme based on the coloration effect of the feedback path is introduced to address these challenges. The acoustic feedback path is modeled as a coloration filter, and the spectrum of the echo signal can be estimated according to the coefficients of the coloration filter and the spectrum of the loudspeaker signal. This method has two main innovations. Firstly, a parametric Wiener filter based on the prior signal-toecho ratio (SER) is employed to reduce the gain fluctuation and the residual echo. Secondly, we apply a parameter adaptation strategy based on posterior SER and a correlation factor to improve the echo suppression capability and the speech quality. Experimental results show that the ERLE in the proposed algorithm is improved by 15dB compared with that in the classical frequency-domain adaptive filtering (FDAF) algorithm. The convergence time is shortened from 0.5 s to 0.1 s, and the computational complexity decreases significantly. Also, this scheme simulating the acoustic path with a coloration filter has better robustness for acoustic path mutation. [ABSTRACT FROM AUTHOR]

Background: Pediatric inflammatory multisystem syndrome temporally associated with SARS-CoV-2 (PIMS-TS) is an inflammatory disease occurring in a small minority of children a few weeks after acute infection. Cardiac manifestations are common, but little is known about the potentially persistent heart changes after PIMS-TS. Purpose: To analyze the frequency and type of myocardial complications of PIMS-TS with initial cardiac involvement assessed with cardiac magnetic resonance imaging (MRI), including parametric imaging, performed 3 months after hospitalization. Study Type: Retrospective. Population: Nineteen consecutive children (median age 10 years, interquartile range (IQR) 10-15 years, 74% male). Field Strength/Sequence: Balanced steady state free precession (bSSFP, cine imaging), modified Look-Locker (T1 mapping), T2-prepared bSSFP (T2-mapping), dark-blood T2-weighted turbo spin echo with fat suppression and phase sensitive inversion recovery (late gadolinium enhancement (LGE)) sequences at 1.5 T. Assessment: Patients were scanned after a median of 99 days (IQR 89-104 days) from the diagnosis. MR data were reviewed by three independent observers, with 13, 2, and 5 years' experience in cardiac MRI. Pre- and post-contrast T1, T2, extra-cellular volume, and T2 signal intensity (T2 SI) ratio were calculated. Diagnosis of acute myocarditis was based on modified Lake Louise criteria. Cardiac MRI parameters were compared, where possible, to previously published pediatric normal values. Statistical Tests: Interclass correlation coefficient and Bland-Altman repeatability analysis. A P-value <0.05 was considered statistically significant. Results: Despitecardiac involvement includingdecreased left ventricularejectionfraction(LVEF) (medianLVEF=47%, IQR43%-53%) and increased troponin I (median 101 ng/mL, IQR 50-661 ng/mL) during hospitalization, there were no persistent cardiac changes observed in cardiacMR at follow-up. All patients had normal size and function of the left ventricle and normal precontrast T1 and T2 relaxation times. There were no signs of LGE. Persistent, mildpericardial effusion(8-9 mm)wasfoundin three (16%)patients. Data Conclusion: There were no persistent changes on cardiac MRI in a group of children approximately 3 months post hospitalization due to PIMS-TS with cardiac involvement. This supports the hypothesis that cardiac involvement during PIMS-TS is a form of transient inflammatory response rather than direct and potentially persistent injury from the virus. [ABSTRACT FROM AUTHOR]

The phenomenon of echo in the telecommunications channels is caused by the reflection of an electrical signal in a long line. In order to improve the quality of the transmitted sound, various adaptive filters are used to remove or at least reduce the level of the reflected delayed signal. However, such a process may result in a degradation in the quality of speech, although its intelligibility may not get worse. The work presents the results of subjective studies on assessing the quality of speech signals under the process of acoustic echo cancellation using different algorithms. The algorithms studied were: LMS (Least Mean Squares), NLMS (Normalized Least Mean Squares) and AP (Affine Projection). The study consisted of assessing the signal quality after applying the echo elimination process using the Degradation Category Rating method. A total of 312 signals were used in the test: 192 male speech and 120 female speech samples. Echo simulation was used using different delay times and levels of echo signal. Both types of speech have signal delay times of 20 ms, 50 ms, 100 ms and 200 ms with echo level values of -6 dB, -12 dB, -18 dB and - 24 dB. In addition, for female speech signals, a delay time of 150 ms was introduced. The study involved 14 people aged between 18 and 38, including six women and eight men. All subjects had normal good hearing. Seven listeners had participated in subjective listening tests of the sound quality assessment previously. The listeners' opinions were collected on prepared questionnaire. It was found that the highest ratings were given to the AP filter, while the worst ratings were featured the NLMS. It should also be noted that the range between the results obtained for AP and NLMS for female speech is smaller in comparison to male. It is also interesting that the discrepancy in ratings was greatest for a delay time of 100 ms for the AP filter and 200 ms for the LMS filter. It can therefore be concluded from the obtained results that, in the case of acoustic echo cancelation, AP filter introduced the lowest quality degradation while the LMS achieved slightly worse average ratings when compared to the AP filter. The NLMS filter characterized by the worst ratings, and in some cases received twice the quality degradation compared to the AP filter. [ABSTRACT FROM AUTHOR]

This paper presents a compact multilayer substrate integrated waveguide (SIW) bandpass filter with wide stopband. The square SIW cavity resonators in multilayer substrates are horizontally and vertically coupled with magnetic coupling. By properly designing the coupling structure, the couplings between the higher-order resonant modes can be suppressed. Compared with the conventional single-layer SIW filters, the proposed multilayer SIW filter also exhibits a compact size. To verify the design concept, a double-layered SIW filter is fabricated and measured. The measured results agree well with the simulations. The measured results show that the upper stopband of the filter is extended to 2.73 times of the center frequency 8.02 GHz. [ABSTRACT FROM AUTHOR]

Purpose: To use deep learning for suppression of the artifact‐generating T1‐relaxation echo in cine displacement encoding with stimulated echoes (DENSE) for the purpose of reducing the scan time. Methods: A U‐Net was trained to suppress the artifact‐generating T1‐relaxation echo using complementary phase‐cycled data as the ground truth. A data‐augmentation method was developed that generates synthetic DENSE images with arbitrary displacement‐encoding frequencies to suppress the T1‐relaxation echo modulated for a range of frequencies. The resulting U‐Net (DAS‐Net) was compared with k‐space zero‐filling as an alternative method. Non‐phase‐cycled DENSE images acquired in shorter breath‐holds were processed by DAS‐Net and compared with DENSE images acquired with phase cycling for the quantification of myocardial strain. Results: The DAS‐Net method effectively suppressed the T1‐relaxation echo and its artifacts, and achieved root Mean Square(RMS) error = 5.5 ± 0.8 and structural similarity index = 0.85 ± 0.02 for DENSE images acquired with a displacement encoding frequency of 0.10 cycles/mm. The DAS‐Net method outperformed zero‐filling (root Mean Square error = 5.8 ± 1.5 vs 13.5 ± 1.5, DAS‐Net vs zero‐filling, P <.01; and structural similarity index = 0.83 ± 0.04 vs 0.66 ± 0.03, DAS‐Net vs zero‐filling, P <.01). Strain data for non‐phase‐cycled DENSE images with DAS‐Net showed close agreement with strain from phase‐cycled DENSE. Conclusion: The DAS‐Net method provides an effective alternative approach for suppression of the artifact‐generating T1‐relaxation echo in DENSE MRI, enabling a 42% reduction in scan time compared to DENSE with phase‐cycling. [ABSTRACT FROM AUTHOR]

The acoustic echo cannot be entirely removed by linear adaptive filters due to the nonlinear relationship between the echo and the far-end signal. Usually, a post-processing module is required to further suppress the echo. In this paper, we propose a residual echo suppression method based on the modification of dual-path recurrent neural network (DPRNN) to improve the quality of speech communication. Both the residual signal and the auxiliary signal, the far-end signal or the output of the adaptive filter, obtained from the linear acoustic echo cancelation are adopted to form a dual-stream for the DPRNN. We validate the efficacy of the proposed method in the notoriously difficult double-talk situations and discuss the impact of different auxiliary signals on performance. We also compare the performance of the time domain and the time-frequency domain processing. Furthermore, we propose an efficient and applicable way to deploy our method to off-the-shelf loudspeakers by fine-tuning the pre-trained model with little recorded-echo data. [ABSTRACT FROM AUTHOR]

We study the thickness effect of YBa2Cu3O7-δ (YBCO) in YBCO/La0.7Sr0.3MnO3 (LSMO) bilayer films, fabricated on LaAlO3 (001) substrate using pulsed laser deposition technique. The thickness dependent transport behavior of YBCO layer in considered bilayer systems demonstrate a clear onset critical temperature at 60K and 80K for 100 nm and 200 nm thick YBCO layer systems, respectively, whereas 50 nm thick YBCO system reveals complete suppression in the superconductivity. With the increase in thickness of YBCO layer, the magnetoresistance (MR) data show both positive and negative MR. The positive MR in superconducting regime dominates the negative MR, suppress the double exchange interaction. The maximum magnetoresistance ratio ~81% at 64K and ~47% at 30K temperature are observed for 200 nm and 100 nm YBCO layers, respectively. In YBCO(200 nm)/LSMO(200 nm) bilayer specimen, the temperature co-efficient of resistance is ~8.4% K-1 at 72K which can be practical for bolometric performances and temperature sensors. [ABSTRACT FROM AUTHOR]

Traditional stereophonic acoustic echo cancellation algorithms need to estimate acoustic echo paths from stereo loudspeakers to a microphone, which often suffers from the nonuniqueness problem caused by a high correlation between the two far-end signals of these stereo loudspeakers. Many decorrelation methods have already been proposed to mitigate this problem. However, these methods may reduce the audio quality and/or stereophonic spatial perception. This paper proposes to use a convolutional recurrent network (CRN) to suppress the stereophonic echo components by estimating a nonlinear gain, which is then multiplied by the complex spectrum of the microphone signal to obtain the estimated near-end speech without a decorrelation procedure. The CRN includes an encoder-decoder module and two-layer gated recurrent network module, which can take advantage of the feature extraction capability of the convolutional neural networks and temporal modeling capability of recurrent neural networks simultaneously. The magnitude spectra of the two far-end signals are used as input features directly without any decorrelation preprocessing and, thus, both the audio quality and stereophonic spatial perception can be maintained. The experimental results in both the simulated and real acoustic environments show that the proposed algorithm outperforms traditional algorithms such as the normalized least-mean square and Wiener algorithms, especially in situations of low signal-to-echo ratio and high reverberation time RT 60 . [ABSTRACT FROM AUTHOR]

Fusion and noise suppression of medical images are becoming increasingly difficult to be ignored in image processing, and this technique provides abundant information for the clinical diagnosis and treatment. This paper proposes a medical image fusion and noise suppression model in pixel level. This model decomposes the original image into a noiseless base layer, a large‐scale noiseless detail layer and a small‐scale detail layer which contains details and noise information. The fractional‐order derivative and saliency detection are used to construct the weight functions to fuse the base layers. The proposed total variation model combines the fractional‐order derivative to fuse the small‐scale detail layers. The mathematical properties and time complexity of the total variation model are also analysed. And choose‐max method is used to fuse the large‐scale detail medical layers simply. Our approach is based on fractional‐order derivative, which enables keep more information and decrease blocky effects more effectively compared with the integer‐order derivative. To verify the validity, the proposed method is compared with some fusion methods in the subjective and objective aspects. Experiments show that the proposed model fuses the source information fully and decreases noise cleanly. [ABSTRACT FROM AUTHOR]

There exists periodic modulation problem in radar echoes due to the main rotor blades periodic blockage in helicopter‐borne fire‐control radar which is mounted atop the main rotor mast of the helicopter. Such modulation echo induces a set of ghosts in synthetic aperture radar (SAR) image, further resulting in a poor performance of subsequent tracking and striking. To address this problem, this article proposes a method on rotor blades blockage modulation suppression for helicopter‐borne SAR. By decomposing the blocked echo into the form of Fourier series in the azimuth direction, a reference function could be constructed to suppress the modulation directly by adopting an iterative approximation strategy. This method effectively avoids the complex blocked data recovery methods, and thus can be used to suppress various kinds of periodic modulation components without requiring certain distribution models. Both simulated and real‐measured data are processed to demonstrate the effectiveness of the proposed algorithm. [ABSTRACT FROM AUTHOR]

In general, the background noise degrades the speech quality. Thus, the intelligibility of the speech can be enhanced by mitigating the effects of background noise and echo suppression. So, speech enhancement can also be viewed as one of the optimization problems. In this work, directed search optimization (DSO) method is used to enhance the speech quality which is originally degraded. The performance of DSO-based speech enhancement method is compared with particle swarm optimization (PSO) and least mean square (LMS)-based methods in terms of output average segmental SNR and speech quality. From the experimental results, it was observed that the output spectrogram, output ASSNR and speech quality using DSO algorithm are far better as compared to PSO and LMS-based methods. Moreover, DSO-based method is computationally less complex as compared to the PSO-based method. [ABSTRACT FROM AUTHOR]

In order to counteract the main lobe multi-false target jamming of radar, a blind source separation jamming suppression method based on channel ratio function is proposed. The cross-polarised auxiliary antenna is used to expand the receiving channel, and the blind signal processing method is used to separate the jamming from the target echo. This method utilises the statistical characteristics of channel ratio function and chooses the time unit with only jamming to process. And it realises jamming suppression and echo extraction. The simulation results show that the proposed method has good jamming suppression performance and is not affected by channel noise and polarisation purity of auxiliary antenna. [ABSTRACT FROM AUTHOR]

The purpose of this study was to measure the binaural interaction component (BIC) derived from click-evoked auditory brainstem responses (ABRs) using stimuli configured to elicit the Precedence Effect. The hypothesis was that the contribution of binaural processing to echo suppression can be evidenced by a diminished or absent BIC associated with the echo. Ten normal-hearing young adults provided ABRs generated by sequences of click pairs. Results showed that BICs elicited by diotic clicks in isolation were obliterated when those diotic clicks were preceded by a click pair having an interaural time difference of 400 µs and where the interclick interval was 8.4 ms. The presence of the leading click pair increased the latency of the ABR generated by the lagging diotic click pair but did not decrease its amplitude. The results were interpreted as indicating a contribution of binaural processing at the level of the brainstem to echo suppression, at least for the conditions tested here. [ABSTRACT FROM AUTHOR]

Featured Application: Acoustic echo cancellation for speech communication, speech recognition and hearing aids. In acoustic echo cancellation, a certain level of residual echo resides in the output of the linear echo canceller because of the nonlinearity of the power amplifier, loudspeaker, and acoustic transfer function in addition to the estimation error of the linear echo canceller. The residual echo in the current frame is correlated not only to the linear echo estimates for the harmonically-related frequency bins in the current frame, but also with linear echo estimates, residual echo estimates, and microphone signals in adjacent frames. In this paper, we propose a residual echo suppression scheme considering harmonic distortion and temporal correlation in the short-time Fourier transform domain. To exploit residual echo estimates and microphone signals in past frames without the adverse effect of the near-end speech and noise, we adopt a double-talk detector which is tuned to have a low false rejection rate of double-talks. Experimental results show that the proposed method outperformed the conventional approach in terms of the echo return loss enhancement during single-talk periods and the perceptual evaluation of speech quality scores during double-talk periods. [ABSTRACT FROM AUTHOR]

The precedence effect refers to a group of auditory phenomena related to the ability to localize sound sources in reverberant environment. The precedence effect was investigated using a localization task in the free sound field. The signal consisted of a direct sound (lead) and a moving target sound (lag). Lead–lag delays ranged from 1 to 40 ms. The lead was always stationary (+15°). The moving lag traveled to or from the lead along two paths situated at two locations (from –86° to –52° and from –52° to –86°). The subjects marked the start and end points of the trajectory of the moving sounds. Localization of the start and end points of the lag was compared with localization of the single source. It was found that the lead affected the start and end points of the lag trajectory. The shift depended on the direction of movement. The start points were always shifted in the direction of motion of the lag. The end points were shifted in the opposite direction. The shifts were decreasing as the delay was increasing. The effect of the lag on the lead localization was small. [ABSTRACT FROM AUTHOR]

Purpose: To design a soft‐hard composite pulse for fat suppression and water excitation in ultrashort echo time (UTE) imaging with minimal short T2 signal attenuation. Methods: The composite pulse contains a narrow bandwidth soft pulse centered on the fat peak with a small negative flip angle (–α) and a short rectangular pulse with a small positive flip angle (α). The fat magnetization experiences both tipping‐down and ‐back with an identical flip angle and thus returns to the equilibrium state, leaving only the excited water magnetization. Bloch simulations, as well as knee, tibia, and ankle UTE imaging studies, were performed to investigate the effectiveness of fat suppression and corresponding water signal attenuation. A conventional fat saturation (FatSat) module was used for comparison. Signal suppression ratio (SSR), defined as the ratio of signal difference between non‐fat‐suppression and fat‐suppression images over the non‐fat‐suppression signal, was introduced to evaluate the efficiency of the composite pulse. Results: Numerical simulations demonstrate that the soft‐hard pulse has little saturation effect on short T2 water signals. Knee, tibia, and ankle UTE imaging results suggest that comparable fat suppression can be achieved with the soft‐hard pulse and the FatSat module. However, much less water saturation is induced by the soft‐hard pulse, especially for short T2 tissues, with SSRs reduced from 71.8 ± 6.9% to 5.8 ± 4.4% for meniscus, from 68.7 ± 5.5% to 7.7 ± 7.6% for bone, and from 62.9 ± 12.0% to 4.8 ± 3.2% for the Achilles tendon. Conclusion: The soft‐hard composite pulse can suppress fat signals in UTE imaging with little signal attenuation on short T2 tissues. [ABSTRACT FROM AUTHOR]

The far-field pattern of an antenna under test is determined from the near-field measurements in a highly reflective environment. The proposed post-processing effectively combines time gating and spatial filtering of the radiated fields. Time gating utilizes frequency diversity, while spatial filtering makes use of probe diversity and source localization. Different probe locations are used to form a virtual array, which focuses the fields at the antenna volume. The possible scatterers are illuminated as little as possible. The weighting coefficients for the virtual array are found by solving a field synthesis problem. The effects of time gating and spatial filtering are analyzed separately. A measurement setup is described in a highly reflective chamber involving fully spherical measurements with probes in up to 120 different positions and with 1001 frequencies. Time gating yields better results, but a combination of both methods is necessary to achieve an estimated accuracy of −30 dB for the considered S-band and −40 dB for the considered Ku-band antenna. [ABSTRACT FROM AUTHOR]

When the input signals for acoustic echo cancellation (AEC) are related signals, the convergence speed of the traditional normalized least mean square (NLMS) algorithms is significantly reduced. In this paper, a joint optimization robust AEC algorithm is proposed to solve this problem. Based on the analysis of the convergence of the normalized subband adaptive filtering (NSAF) algorithm, the algorithm is optimized by minimizing the mean square error (MSE) of the NSAF algorithm, combining sub-band time-varying step factor and time-varying regularization parameter to update the filter weight vectors. And when the impulse noise occurs, the sub-band cut-off parameter is updated in a recursive manner, which makes the algorithm achieve fast convergence speed and low steady-state error, and has strong robustness to impulse noise. In a series of experiments on AEC, simulation results show that the performance of the algorithm is better than the existing algorithms. [ABSTRACT FROM AUTHOR]

Highlights • We outline the historical development of the application of the cepstrum to modal analysis. • Most recent developments and current state-of-the-art also covered. • Use of the cepstrum to extract modal properties is explained, including MIMO applications. • Discussion on preprocessing applications of the cepstrum to improve conventional OMA. • Importance of cepstral methods is explained, especially in OMA with strong forcing functions. Abstract The cepstrum has a very long history, since the first paper was published in 1963, two years before the publication of the FFT algorithm. The first application was to the determination of echo delay time, but it was quickly applied to speech analysis for two reasons. Firstly, it was able to detect voiced speech, and measure the voice pitch, and secondly it was able to separate the forcing and transfer function components in speech signals. These two properties apply equally to machine vibration signals, and the cepstrum can be used to detect and remove periodic discrete frequency components (harmonics and modulation sidebands) from the spectrum (and corresponding time signals) and also to extract the modal properties of a structure in the presence of the forcing function. This paper describes the development of these applications to structural modal analysis over many years, primarily to operational modal analysis (OMA), where the modal information is extracted from response signals, often with the structure in its normal operating environment. The cepstrum extracts the modal information in terms of pole/zero models, and requires compensation for the effects of unmeasured out-of-band modes, by an equalisation process, and overall scaling to obtain scaled mode shapes. Steady advances to achieve this have been made over the years. A recent development that has revolutionised the application has been the ability to edit stationary and slowly varying non-stationary signals using the real cepstrum (which does not contain the phase information of the original signals) but regenerating time signals by combining the edited amplitude spectra with the original phase spectra (with negligible error). There are currently two ways in which this can be used in operational modal analysis: 1. Pre-processing of the response signals to remove most excitation components and other disturbances, and enhance the modal properties, before applying standard OMA procedures. 2. Extracting the modal models from the pre-processed signals by curve-fitting pole/zero models to the cepstra, and applying compensation for equalisation and scaling. The paper describes the history, current situation and potential future development of the application of cepstral analysis to structural modal analysis, this seemingly being greatly under-utilised. [ABSTRACT FROM AUTHOR]

Abstract Pilot whales produce clicks, whistles and pulsed calls, which form a key component of their social lives. The three types of sound driven by their behavioural states are not directly observable. The mathematical tools which compute properties of sound are natural candidates for analysing the possible relationship between pilot whale sound and behaviour. In this paper, the wideband ambiguity function is used to compute the range resolution, speed resolution, Doppler tolerance, sidelobe-to-mainlobe suppression ratio of the range ambiguity function and sidelobe-to-mainlobe suppression ratio of the speed ambiguity function for pilot whale sound, which can be used to evaluate the abilities of pilot whale sound in detecting targets and resisting the influence of relative speed. Statistical results show that clicks have the best range resolution and its range ambiguity functions have the lowest sidelobe-to-mainlobe suppression ratio. Pulsed calls and part of the whistles have the best speed resolution and its speed ambiguity functions have the lowest sidelobe-to-mainlobe suppression ratio. The other part of whistles has a large Doppler tolerance. Analyzing the five parameters of different types of sound and the corresponding behaviour, we then reach a better understanding of the relationship between pilot whale sound and pilot whale behaviour. [ABSTRACT FROM AUTHOR]

Abstract This paper addresses the stereophonic acoustic echo cancelation (SAEC) problem in teleconferencing system by adaptive filtering algorithm. We propose a new stereophonic version of the fast normalized least mean square (FNLMS) algorithm. The proposed algorithm is an extended version of the FNLMS algorithm to the stereophonic case with important improvement of the prediction part that becomes stable in the stereophonic case. The basic idea behind the proposed stereophonic fast normalized least mean square (SFNLMS) algorithm is the normalization of the predictor parameter by the inputs variance for each channel. Simulation results of a comparison between the proposed SFNLMS algorithm and the classical SNLMS version in term of convergence speed and stability are presented. [ABSTRACT FROM AUTHOR]

Adaptive decision feedback equalizer (ADFE) based on Least Mean square algorithm (LMS) has emerged as a novel tool for signal propagation without noise. However, it has the limitation of computational complexity. Since it has been reported that substantial echo cancellation leads to increased system quality and capacity, this paper proposes a dynamic LMS-based ADFE (DLMS-ADFE) to cancel the generation of the echo signal. The DLMS-ADFE has two dynamic principles in updating the coefficients. The first principle has varying step-size and the second principle has varying degree and magnitude of the signal. The comparison results show that the DLMS-ADFE is outperforming LMS-ADFE. [ABSTRACT FROM AUTHOR]

In vehicle hands-free communication systems and video teleconferencing systems, echoes are typically encountered. Adaptive filters are usually employed to remove the echoes in these applications. However, large outliers and highly correlated speech input signals are two key factors that limit the performance of the adaptive filters. In this paper, we propose an affine projection Versoria (APV) algorithm, which is obtained by maximizing the summation of Versoria-cost reusing with a constraint on the square of the L2-norm of the filter weight vector difference. In this way, the features of the Versoria-cost maximization and data reusing are combined and, consequently, the proposed APV algorithm obtains robustness to the large outliers and accelerates the convergence rate for the correlated input signals. The complexity of the proposed APV algorithm is analyzed and then a fast recursive filtering technique is introduced to reduce its complexity. A stability analysis proves that the proposed APV algorithm is convergent. In addition, an analytical expression of the steady-state excess mean-square error for the proposed APV algorithm has been derived and simulated results are in agreement with the theoretical analysis result. Simulations on echo channel estimation and echo cancellation show that the proposed APV algorithm performs much better than the maximum Versoria criterion, affine projection sign, and affine projection algorithms. [ABSTRACT FROM AUTHOR]

The normalised subband adaptive filter (NSAF) is a useful adaptive filter, which improves the convergence rate compared with the normalised least mean‐square algorithm. Most analytical results of the NSAF set the regularisation parameter set to zero or present only steady‐state mean‐square error performance of the regularised NSAF (ε ‐NSAF). This study presents a mean‐square performance analysis of ε ‐NSAF, which analyses not only convergence behaviour but also steady‐state behaviour. Furthermore, a novel adaptive regularisation for NSAF (AR‐NSAF) is also developed based on the proposed analysis approach. The proposed AR‐NSAF selects the optimal regularisation parameter that leads to improving the performance of the adaptive filter. Simulation results comparing the proposed analytical results with the results achieved from the simulation are presented. In addition, these results verify that the proposed AR‐NSAF outperforms the previous algorithms in a system‐identification and acoustic echo‐cancellation scenarios. [ABSTRACT FROM AUTHOR]

Studies on the precedence effect are typically conducted by presenting two identical sounds simulating direct sound and specular reflection. However, when a sound is reflected from irregular surface, it is redirect into many directions resulting in directional and temporal diffusion. This contribution introduces a simulation of Lambertian diffusing reflections. The perceptual influences of diffusion are studied in a listening experiment; echo thresholds and masked thresholds of specular and diffuse reflections are measured. Results show that diffusion makes the reflections more easily detectable than specular reflections of the same total energy. Indications are found that this mainly due to temporal diffusion, while the directional diffusion has little effect. Accordingly, the modeling of the echo thresholds is achieved by a temporal alignment of the experimental data based on the energy centroid of reflection responses. For the modeling of masked threshold the temporal masking pattern for forward masking is taken into account.

• Polarised full-waveform warning LIDAR with backscattering suppression was developed. • Dust backscattering and target reflection depolarisation is considered. • Dust backscattering was effectively suppressed and contrast was improved. • This approach can facilitate LIDAR under dusty environments. LIDAR, when used under adverse weather conditions such as fog and dust, suffers from attenuation and backscattering. In particular, backscattering creates spurious echo peaks in the received waveforms at incorrected ranges and leads to false alarms. As a solution, this paper proposes polarised full-waveform warning LIDAR for dust backscattering echo suppression, intended for applications under dusty environments. To this end, first, a theoretical model of the polarised LIDAR is developed based on the LIDAR equation and depolarisation characteristics of dust backscattering and target reflection. Thereafter, the depolarisation characteristics are experimentally evaluated under an artificially generated dusty environment; the results reveal that the target reflection features a strong depolarisation characteristic, whereas dust backscattering has a weak one. When the LIDAR emits linearly polarised light, a significant orthogonal-polarised component exists owing to the target reflection, and dust backscattering induces a mainly parallel-polarised component. On interposing an orthogonal-polarisation analyser in the receiving optical path, the parallel-polarised component from the dust backscattering and target reflection is blocked, whereas the orthogonal-polarised component is allowed to pass; this helps improve the contrast of the target reflection to dust backscattering. Finally, based on the theoretical model of the polarised LIDAR and experimental results, an orthogonal-polarised LIDAR prototype is designed and fabricated. Using this prototype, target detection experiments in a dusty environment are performed; the corresponding experimental results indicate that the orthogonal-polarised LIDAR effectively suppresses dust backscattering echoes and that the contrast of the target reflection to dust backscattering is improved by 4.78–7.12 dB. Utilization of the polarised full-waveform warning LIDAR can significantly reduce the false alarm rate and improves the LIDAR system performance when used under dusty environment conditions, in applications such as smart mining and autonomous driving. [ABSTRACT FROM AUTHOR]

This Letter presents a bias compensated M‐estimate affine‐projection‐like algorithm for robust adaptive filtering in case of noisy input signal and impulsive noise interference. In order to mitigate the effect of bias introduced due to noisy input signal, a bias compensation vector is derived along with robust cost function minimisation employing M‐estimate and accordingly the weight update learning process of the adaptive filter has been improved. In conjunction with this, a new input noise variance estimation method is introduced using the median filter. The proposed algorithm is evaluated for acoustic echo cancellation application and compared with other existing methods. Simulation results demonstrate its effectiveness in terms of enhanced convergence and better steady‐state performance in case of impulsive environment and input signal contaminated with noise. [ABSTRACT FROM AUTHOR]

MgTiO3-doped Ba3Co2Fe24O41 hexaferrites (Co2Z) have been prepared by the solid-state reaction method, and the magnetic and dielectric properties have been investigated. Permeability measurements indicate that MgTiO3 doping significantly improves the frequency characteristics of permeability, i.e., with the increase of MgTiO3 doping content from 0 to 5 wt %, the cutoff frequency increases greatly from 0.33 GHz to > 1.8 GHz. Although μ' decreases from 16.1 to 5.4 in a sample doped with 5 wt % MgTiO3, magnetic loss μ" is effectively suppressed in frequency above 300 MHz. The permittivity of 5 wt % MgTiO3 doped specimen keeps stable in the value of about 17, while the dielectric loss largely decreases in the frequency rang from 300 MHz to 1.8 GHz. The low magnetic and dielectric loss and the improvement of frequency characteristics are crucial for high-frequency antenna applications of Z-type hexaferrites. [ABSTRACT FROM AUTHOR]

In the presence of range ambiguity, the synthetic aperture radar (SAR) systems suffer from image aliasing, which dramatically degrades the quality of SAR images. In this article, an easy-to-implement technique for range ambiguity suppression is investigated, which is based on phase coding in the transmit pulse dimension, referred to as pulse phase coding (PPC). By properly designing the PPC series, it is possible to discriminate the range ambiguous echoes from different range areas in the Doppler frequency domain. To further suppress the range ambiguous echo, a two-pulse cancellation (TPC)-based SAR imaging method is proposed, which improves the quality of SAR images in the presence of range ambiguity. The proposed two-pulse cancellation is performed followed by the azimuth compression. The simulation results demonstrate the effectiveness of the proposed method. [ABSTRACT FROM AUTHOR]

This paper presents a numerical investigation on the use of a rotative non-linear vibration absorber (NVA) as a passive suppressor for the parametric instability phenomenon. Focus is placed on the influence of the mass, the radius and the damping of the NVA on its capacity to reduce the amplitudes of the cylinder in which it is installed. Maps of statistics of the cylinder response (namely, its maximum value and standard deviation), time histories of both the cylinder and the NVA responses, as well as the suppression efficiency are obtained for different dimensionless quantities that define the suppressor. The analysis methodology include the discussion of these maps and spectral analyses of the time histories resulting from selected points of these maps. The results show that, depending on the NVA parameters, bounded responses are achieved even in a scenario in which the parametric excitation parameters lead to unbounded responses of the cylinder without the suppressor. Furthermore, different types of cylinder response are obtained, including strongly modulated responses as well as complete suppression of oscillations. An asymptotic analysis based on the complexification-averaging and multiple scales methods allows identifying that the suppression is associated with a capture on an invariant manifold of the system. These and other features of the response are deeplier analyzed in the paper. [ABSTRACT FROM AUTHOR]

In a recent electrophysiological study, we reported on global facilitation but local suppression of color stimuli in feature-based attention in human early visual cortex. Subjects attended to one of two centrally located superimposed red/blue random dot kinematograms (RDKs). Task-irrelevant single RDKs in the same colors were presented in the left and right periphery, respectively. Suppression of the to-be-ignored color was only present in the centrally located RDK but not in the one with the same color in the periphery. This result was at odds with the idea of active suppression of task-irrelevant features across the entire visual field. In the present study, we introduced competition in the periphery by superimposing the RDKs at the task-irrelevant location as well. With such competition, we found suppression of the task-irrelevant color in the centrally and peripherally located RDKs. Results clearly demonstrate that suppression of task-irrelevant features at task-irrelevant locations requires (spatial) competitive interactions and is not an inherent neural mechanism in feature-based attention as was found for global facilitation. [ABSTRACT FROM AUTHOR]

A control logic has a central role in many echo cancellation systems for optimizing the performance of adaptive filters, while estimating the echo path. For reliable control, accurate double-talk and channel change detectors are usually incorporated to the echo canceler. This work expands the usual detection strategy to define a classification problem characterizing four possible states of the echo canceler operation. The new formulation allows the use of decision theory to continuously control the transitions among the different modes of operation. The classification rule reduces to a low-cost statistics, for which it is possible to determine the probability of error under all hypotheses, allowing the classification performance to be accessed analytically. Monte Carlo simulations using synthetic and real data illustrate the reliability of the proposed method. [ABSTRACT FROM AUTHOR]

Much evidence suggests that the processing of emotions is lateralized to the right hemisphere of the brain. However, under some circumstances the left hemisphere might play a role, particularly for positive emotions and emotional experiences. We explored whether emotion contagion was rightlateralized, lateralized valence-specifically, or potentially left-lateralized. In two experiments, right-handed female listeners rated to what extent emotionally intoned pseudo-sentences evoked target emotions in them. These sound stimuli had a 7 ms ear lead in the left or right channel, leading to stronger stimulation of the contralateral hemisphere. In both experiments, the results revealed that right ear lead stimuli received subtly but significantly higher evocation scores, suggesting a left hemisphere dominance for emotion contagion. A control experiment using an emotion identification task showed no effect of ear lead. The findings are discussed in relation to prior findings that have linked the processing of emotional prosody to left-hemisphere brain regions that regulate emotions, control orofacial musculature, are involved in affective empathy processing areas, or have an affinity for processing emotions socially. Future work is needed to eliminate alternative interpretations and understand the mechanisms involved. Our novel binaural asynchrony method may be useful in future work in auditory laterality. [ABSTRACT FROM AUTHOR]

In this paper, a unified framework for using speech interfaces with intelligent televisions (TVs) is proposed. The proposed framework is primarily based on suppressing the stereophonic acoustic echo of the TV audio sounds and adaptive tracking of the time varying acoustic transfer function (ATF) of the echo sounds. In order to effectively suppress the acoustic echoes from the TV speakers, an optimal filter with a multichannel subspace approach is imposed on noisy inputs and the second order statistics of the stereophonic echoes are adaptively estimated by tracking the time varying ATF using the normalized least mean square method. Through assessing the proposed method with a database collected in real TV watching environments, noticeable improvements were achieved in the signal to noise ratio gain, Itakura–Saito distance, and speech recognition rates. [ABSTRACT FROM AUTHOR]

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In this paper, a regression-based integrated acoustic echo and background noise suppression algorithm was proposed through the use of a deep neural network (DNN) with a multi-layer deep architecture. Motivated by an idea that DNNs are a superior hierarchical generative model for modeling the complex relationships between input features and desired target features through its multiple nonlinear hidden layers, a stacked DNN is developed in a sequential fashion such that the DNN for noise suppression is followed by the DNN for acoustic echo suppression. This algorithm is compared to a single DNN-based integrated system to simultaneously suppress acoustic echoes and noise. When developing the DNN-based regression technique using our approach, spectral envelop estimation is a crucial point for which log-power spectra (LPS) are used as features in order to determine the gain, which ensured nonlinear mapping from the LPS of the frames contaminated by echoes and noise to the LPS of the echo- and noise-free frames. This leads to the successful reduction of acoustic echoes and background noise without an additional double-talk detection algorithm. Additionally, an augmented feature technique is adopted to use additional knowledge derived from conventional noise and acoustic echo suppression techniques when designing the DNN architecture in our algorithm. The proposed DNN-based integrated system to suppress acoustic echoes and noise was evaluated in terms of objective measures and demonstrated a significant improvement over conventional integrated algorithms. [ABSTRACT FROM AUTHOR]

In this correspondence, we propose a dual-function hybrid beamforming architecture, where the antenna array is split into sub-arrays that are separated by a sufficiently large distance so that each sub-array experiences independent fading. The proposed architecture attains the dual-functions of beamforming and diversity. We then demonstrate that splitting the array into two sub-arrays provides the best performance in terms of the achievable rate as a benefit of the diversity gain obtained in addition to the beamforming gain. However, the performance starts depleting if the array is partitioned into more than two sub-arrays because of diminishing additional diversity gains, which fails to compensate for the beamforming gain erosion due to splitting the antenna arrays. Additionally, we analyze the so-called discrete Fourier transform-mutually unbiased bases (DFT-MUB) aided codebook invoked for the conceived design, which imposes an appealingly low complexity. Explicitly, we show that for the proposed dual-function sub-array-connected design, the DFT-MUB assisted codebook outperforms the state-of-the-art precoding benchmarks and performs close to the optimal precoding matrix. [ABSTRACT FROM AUTHOR]