Frequency-domain Beamforming Research Articles

Measurement of ocean currents by seafloor distributed optical-fiber acoustic sensing.

Ocean current measurements play a crucial role in aiding our understanding of ocean dynamics and circulation systems. Traditional methods, such as drifters and ocean buoys, are sparsely distributed and of limited effectiveness due to the nature of the marine environment and high operating expenses. Distributed acoustic sensing (DAS) is an emerging technology using submarine optical-fiber (OF) cables as dense seismo-acoustic arrays, offering a new perspective for ocean observations. Here, in situ observations of ocean surface gravity waves (OSGWs) and ocean currents by DAS were made along a pre-existing 33.6 km seafloor OF cable. The average current velocity and water depth along the cable were determined from observed OSGW-induced seafloor noise (0.05-0.2 Hz) using ambient-noise interferometry and frequency-domain beamforming. Variations in current velocity were derived at high spatiotemporal resolution using the frequency-domain waveform-stretching method. The inverted current velocity was verified by nearby ocean buoy observations and forecasting results. The observations demonstrate the effectiveness of DAS-instrumented OF cables in monitoring ocean currents.

High-resolution wide-swath SAR imaging with multifrequency pulse diversity mode in azimuth multichannel system

ABSTRACT High-resolution wide-swath (HRWS) imaging has emerged as a focal point in synthetic aperture radar (SAR) research. However, conventional SAR systems face challenges in achieving both high azimuth resolution and wide swath simultaneously due to the minimum antenna area constraint. In this paper, we propose a HRWS imaging method based on multifrequency pulse diversity (MFPD) and azimuth multichannel (AMC) technique, capable of resolving range and Doppler ambiguities concurrently. In MFPD mode, range ambiguous echoes can be separated by matched filters in the range frequency domain, as multifrequency pulses are transmitted by a single channel in the transmitter. To further enhance the ambiguity resolution ability of the system and address azimuth incoherence, a novel scheme applying the MFPD to AMC system is proposed, categorized into two distinct scenarios. 1) Uniform Sampling: This scenario satisfies the displaced phase centre antenna condition. Under this condition, high range resolution imaging can be achieved through spectrum splicing in range frequency domain, simultaneously resolving Doppler ambiguity. 2) Non-uniform Sampling: In this scenario, the ambiguous echoes are processed by spatial digital beamforming and spectrum splicing in two-dimensional frequency domain. Finally, the HRWS imaging can be obtained by performing the traditional SAR algorithm. Furthermore, the proposed method can synthesize a wideband signal from multifrequency signals, thereby enhancing the feasibility of super-high-resolution imaging. Simulation results demonstrate the effectiveness of the proposed method.

A hybrid time and frequency domain beamforming method for application to source localisation on high-speed trains

To comply with the legislation and certification procedures that limit railway noise, reliable approaches are required to reduce the noise from trains. However, effective noise control can only be achieved if the dominant sources are identified first. Beamforming is a possible experimental solution to achieving source identification on a moving train. However, application of the conventional delay-and-sum beamforming method to moving sources relies on the interpolation, or de-dopplerisation, of the received signals, and is therefore computationally expensive. Frequency-domain methods also exist which have been developed based upon the linearisation of the moving trajectory of the source. These are faster in processing time but can only provide accurate source identification within the validity of the linear approximation. They are typically applied to slowly varying sources such as aircraft flyovers. A hybrid beamforming method is presented for application to moving sources with a short pass-by window, which is a typical situation in railway pass-by measurements. The hybrid method proposed in this paper is based on a combination of features of time- and frequency-domain methods. Delays are applied to the signals as in the time domain method and beamforming is performed as for the frequency domain method. It can provide beamforming estimates over a sub-grid by performing de-dopplerisation based on a single point. Numerical simulations and a field measurement targeting a train pantograph are used to compare the performance of three beamforming methods. The numerical analysis of the proposed method shows that, compared with the frequency-domain method, the hybrid approach has more relaxed requirements for linearisation, and thus results in a better beamforming performance. With appropriate settings, a substantial computational advantage can be achieved. The results also show that the conventional method and the hybrid method have a similar beamforming performance, whereas the frequency-domain method shows higher sidelobe level, especially at high frequencies. The proposed hybrid method can achieve both good beamforming performance and low computational cost and can be used as a good alternative of the conventional time-domain method for identification of moving sources.

Acoustic source localisation using vibroacoustic beamforming

For acoustic source localisation in a fluid, sound pressure is traditionally measured using a microphone array, and acoustic beamforming is then applied to the pressure signals to localise the source. However, in some engineering applications, it is not feasible to place the microphone/hydrophone array in the fluid to localise an acoustic source. Hence, this work proposes a vibroacoustic beamforming for sound source localisation which applies the beamforming procedure to the measured vibration signals at the surface of a structure in contact with the fluid in which the acoustic source is located. To evaluate the localisation performance of the proposed method, a numerical study is conducted to localise a monopole source in a room using vibration data obtained from an elastic panel on the room wall. To simulate more realistic situation, different levels of signal to noise ratio are considered by moving the source away from the wall-mounted sensors and consequently altering the direct acoustic field (signal). Moreover, commonly used frequency domain beamforming techniques namely conventional beamforming, functional beamforming, minimum variance distortionless response and minimum power distortionless response are applied to the measured signals to localise the source. The performance of each technique is then evaluated by comparing the beamforming output obtained from vibroacoustic beamforming with that of acoustic beamforming and by quantifying the dynamic range and spatial resolution for each method. It can be concluded that vibroacoustic beamforming can be used for acoustic localisation as an alternative to the acoustic beamforming.

Aeroacoustic Investigation of Airfoil Profiles in Turbulence at Moderate Reynolds Numbers Involving the Phased Array Microphone Technique and the Anova Statistical Method

Due to the growing number of drones and unmanned aerial vehicles, the aeroacoustic noise radiated by airfoils in moderate Reynolds-number flows increasingly becomes the focus of interest. In the present paper, the flow-generated noise of two basic (a flat plate and a cambered plate) and two conventional (RAF6 and NACA 0012) airfoil profiles is investigated in an open-jet wind tunnel in the interval of Reynolds numbers between 60,000 and 140,000. The measurements were carried out with a phased array microphone. The measurement data were evaluated using frequency–domain beamforming and the analysis of variance statistical method. The results show that the emitted noise is nearly independent of the angle of attack but dependent on the airfoil geometry and the chordwise location.

Improved Coherent Plane-Wave Compounding Using Sign Coherence Factor Weighting for Frequency-Domain Beamforming.

This study proposes a novel modified sign coherence factor (SCF) weighting adapted to the frequency-domain (FD) beamforming for ultrasound plane-wave imaging to achieve a high frame rate and better image quality. First, before beamforming, the sign components were extracted from the radiofrequency signals of aperture data. Second, the modified SCF was established using the FD beamformed sign components. Finally, the FD beamformed image was weighted by the modified SCF. To assess the performance of the proposed modified SCF for FD beamforming, the resolution, contrast, computation complexity and execution time of the generated images were evaluated. The results revealed that the FD-SCF could significantly improve the computational load compared with the classic delay-and-sum SCF on the premise of equal image quality improvement. Therefore, high image quality and low computational load have been successfully combined under the proposed weighting method.

Rapid refocusing of ultrasound images in frequency domain using range-Doppler algorithm

The ability to maintain image uniformity and high resolution over large depths is important for several clinical applications of ultrasound, including deep abdominal imaging in patients with high BMI. One way to improve image quality in such cases is to use retrospective transmit focusing, which involves combining received data from different focused transmits to improve image resolution and SNR outside of transmit focal zone. Retrospective transmit is typically accomplished using the delay-and-sum (DAS) beamforming, which can be slow on systems without a GPU. As a faster alternative, we treat beamsummed signals from focused transmits as monostatic synthetic aperture data from virtual sources, and apply a frequency-domain beamformer such as RDA to rapidly refocus an RF image. We demonstrate the concept using FIELD II simulated ultrasound signals from a point target phantom over 200 mm depth. The RDA-refocused image shows a uniform point spread function, and the reduction in full-width at half maximum by a factor of 2.5 compared to the original image from focused transmits at f-number 10. RDA-based refocusing also achieves a speedup by a factor of 14 relative to the DAS-based retrospective beamformer.

STFT-Domain Neural Speech Enhancement With Very Low Algorithmic Latency

Deep learning based speech enhancement in the short-time Fourier transform (STFT) domain typically uses a large window length such as 32 ms. A larger window can lead to higher frequency resolution and potentially better enhancement. This however incurs an algorithmic latency of 32 ms in an online setup, because the overlap-add algorithm used in the inverse STFT (iSTFT) is also performed using the same window size. To reduce this inherent latency, we adapt a conventional dual-window-size approach, where a regular input window size is used for STFT but a shorter output window is used for overlap-add, for STFT-domain deep learning based frame-online speech enhancement. Based on this STFT-iSTFT configuration, we employ complex spectral mapping for frame-online enhancement, where a deep neural network (DNN) is trained to predict the real and imaginary (RI) components of target speech from the mixture RI components. In addition, we use the DNN-predicted RI components to conduct frame-online beamforming, the results of which are used as extra features for a second DNN to perform frame-online postfiltering. The frequency-domain beamformer can be easily integrated with our DNNs and is designed to not incur any algorithmic latency. Additionally, we propose a future-frame prediction technique to further reduce the algorithmic latency. Evaluation on noisy-reverberant speech enhancement shows the effectiveness of the proposed algorithms. Compared with Conv-TasNet, our STFT-domain system can achieve better enhancement performance for a comparable amount of computation, or comparable performance with less computation, maintaining strong performance at an algorithmic latency as low as 2 ms.

A Wireless Transient Attenuated-Exponential Overpressure Beamforming With for Far-Field Blast Source Localization

Time-domain beamforming is more suitable for blast wave transient signal than frequency-domain beamformer because wideband spectrum of noise makes the beamforming image less clear. To avoid the gust effects and enable the location of blast source accurately, this article proposes a new 1-D far-field delay-and-sum (DAS) beamforming method with an attenuate exponential function model for wireless overpressure transient signal. In addition, we design wireless overpressure peak and root-mean-square (rms) directional estimators to assess the performance of the proposed new DAS beamforming method. Furthermore, the effects of the wireless pressure sensor node (WPSN) spacing, the number of WPSNs, and sidelobe level brought from noise on the beamwidth are investigated in the two estimators. The proposed formula is verified by a uniformly spaced linear sensing array, and the results verify the feasibility of the proposed method in blast source localization. This article is conducted to provide new insight into blast source localization algorithm and further open a door for transient blast overpressure source localization scenarios in future.

Efficient computation of modal transfer function in mode composition beamforming for rotating broadband source localization.

For rotating sound source localization, there are mainly two types of frequency-domain beamforming methods, the virtual rotating array (VRA) method and mode composition beamforming (MCB). Compared with the VRA method, MCB overcomes the constraint of microphone array configuration and avoids error introduced by interpolation. In order to obtain the MCB result, modal transfer functions, which are the modal coefficients in the analytical solution of the sound field of rotating source, should be pre-calculated. However, the computation of modal transfer functions takes a long time, especially under a large number of considered frequencies. Therefore, the computational efficiency of MCB is limited by the computation of modal transfer functions. In this paper, an expression of modal transfer functions at multiple considered frequencies is derived first. According to this expression, intermediate variables independent of frequency are computed in a fast way, then modal transfer functions at each considered frequency are calculated through these intermediate variables. This expression is successfully applied to improve the computational efficiency of MCB for rotating broadband source localization.

Multi-instrument observations of microseisms generated by typhoon Kalmaegi (2014) over the Northwestern Pacific

The typhoon-generated microseisms, originating from the complex energy coupling and transferring among the Atmosphere-Ocean-Solid Earth spheres, can be detected remotely by seismometers as the strongest ambient seismic noise. The lack of in situ observations during the passage of typhoons has hampered numerical modeling of wind fields and ocean waves, and limited our understanding of the generation mechanisms of microseisms associated with typhoons. Here we present a comprehensive investigation of microseisms generated by typhoon Kalmaegi (September 2014) based on multiple-instrument constraints from observations including terrestrial and ocean-bottom seismic stations as well as ocean buoys. To understand the generation mechanisms, we apply an improved frequency-domain beamforming method to seismic array data leading to successful location of double-frequency (DF) microseism source regions. For comparison, we calculate the typhoon-induced ocean waves and theoretical source regions of the DF microseisms using the coupled ocean–atmosphere–wave–sediment transport modeling system with validation by ocean buoy observations. Both observations and numerical modeling results reveal two different generation mechanisms for typhoon-induced DF microseisms during the lifespan of Kalmaegi. When over the Philippine Sea, the DF microseisms were generated mainly by opposing ocean waves from two distinct storms. After Kalmaegi entered the South China Sea, the DF microseisms were generated mainly by the fast-moving typhoon with source regions just trailing behind, with the minimum frequencies determined by the typhoon translation speed. DF microseisms generated in coastal source regions were not detected by ocean bottom seismometers, suggesting that DF microseisms might not effectively propagate across the ocean-basin seafloor covered by thick sediments, owing to severe seismic attenuation and spreading losses. This information is crucial for the use of DF microseisms for future tracking and monitoring of typhoons.

A climatology of microbarom detections at the Kerguelen Islands: unravelling the ambient noise wavefield

SUMMARY The ambient infrasonic noise field is complex due to the interference of spatially distributed infrasound sources. Microbaroms are one of the most dominant omnipresent infrasonic sources within this wavefield. These microbaroms are generated by nonlinear ocean surface wave interactions, and have a characteristic and continuous signature within the infrasound spectrum. Under noisy conditions, microbaroms can mask infrasonic signals of interest, such as infrasound from volcanoes or explosions, which limits detection and identification of such sources. This study performs an infrasonic climatology for infrasound array I23FR, using five years of data between 2015–2020. The array is located on the Kerguelen Islands, within the Southern Ocean, and is part of the International Monitoring System for the verification of the Comprehensive Nuclear-Test-Ban Treaty. The climatology analysis addresses the expected ambient noise levels, propagation paths and potential sources within the vicinity of an infrasound sensor. Time- and frequency-domain beamforming methods have been applied to analyse the infrasonic wavefield from the I23FR observations. A recently introduced method is applied to compute so-called soundscapes, to be compared with beamform results. Although the comparison indicates a disagreement in amplitude, there is a good agreement in directionality and frequency between both.

A Time-Domain Real-Valued Generalized Wiener Filter for Multi-Channel Neural Separation Systems

Frequency-domain beamformers have been successful in a wide range of multi-channel neural separation systems in the past years. However, the operations in conventional frequency-domain beamformers are typically independently-defined and complex-valued, which result in two drawbacks: the former does not fully utilize the advantage of end-to-end optimization, and the latter may introduce numerical instability during the training phase. Motivated by the recent success in end-to-end neural separation systems, in this paper we propose time-domain real-valued generalized Wiener filter (TD-GWF), a linear filter defined on a 2-D learnable real-valued signal transform. TD-GWF splits the transformed representation into groups and performs an minimum mean-square error (MMSE) estimation on all available channels on each of the groups. We show how TD-GWF can be connected to conventional filter-and-sum beamformers when certain signal transform and the number of groups are specified. Moreover, given the recent success in the sequential neural beamforming frameworks, we show how TD-GWF can be applied in such frameworks to perform iterative beamforming and separation to obtain an overall performance gain. Comprehensive experiment results show that TD-GWF performs consistently better than conventional frequency-domain beamformers in the sequential neural beamforming pipeline with various neural network architectures, microphone array scenarios, and task configurations.

A New Low-Cost Acoustic Beamforming Architecture for Real-Time Marine Sensing: Evaluation and Design

This paper mainly studies the performance of an acoustic beamforming technique applied to a low-cost hydrophone in a linear array of two to four elements for the detection and localization of underwater acoustic sound waves. It also evaluates the integration of the array in an energy-efficient real-time monitoring system architecture, allowing marine sensing to be conducted without human intervention. Such architecture would consist of vertical linear arrays of two or four RHSA-10 hydrophones models attached to a buoy or a vessel for sound detection; a frequency domain beamformer (FDB) technique implemented in a Xilinx Spartan-7 field programmable gate array (FPGA) for sound source localization; a LoRa wireless sensor network mote to provide convenient access from a base center. The architecture aims to alleviate sea traffic control for countries that lack the financial resources to properly address illegal fishing or piracy issues, mostly committed in small fast motorized boats. In our experiment, the sound waves emitted by a small motorized boat were successfully detected and tracked by three data acquisitions at a 1 km range. It is demonstrated that a system using a small number of hydrophones is capable of producing robust accuracy over a large band frequency in the presence of noise interference.

Demonstration of a unified approach to beamforming

In this paper, we discuss a unification of several well-known frequency domain beamforming methods into one working principle. The methods under consideration include Functional Beamforming, Asymptotic Beamforming, Adaptive Beamforming and - as a natural limiting case - Standard Beamforming. Common to most of these methods is the underlying eigenvalue decomposition of the cross-spectral matrix. Introducing a weighted power mean (also called weighted Hölder mean) in terms of these eigenvalues for every map point, each of the above methods is represented by a certain power p. Because of the latter, this unified approach will be called Power Beamforming throughout this paper. Going from the limiting case p=1 of Standard Beamforming to lower power values results in the attenuation of side lobes and sharpening of the main lobes in the corresponding beamforming map. We demonstrate this effect using simulations and several real-world measurements.

An Efficient Parallel Architecture for Sub Band Processor in Broad Band MVDR Beamformer for Medical Ultrasound Applications

Minimum Variance Distortion less Response (MVDR) algorithm is the key adaptive beamforming algorithm in high resolution Medical Ultrasound Imaging systems. MVDR Beamformer is a frequency domain beamformer, and each frequency sub band need to be processed to generate covariance matrix and the same is computationally costly. Though many VLSI architectures have been proposed for MVDR Beamformer, currently no Complex Correlation matrix generation architecture for frequency sub band has been devised which supports real time implementation of broad band MVDR Beamformer for Medical Ultrasound Imaging. In this paper we show real time MVDR sub array, sub band processor realization for an array with number of elements N=32 and sub array length L=8. To satisfy the narrow band condition of the array we employed 1K point FFT, with 12.5 % overlap. The entire design of MVDR Sub band processor is tested on the Field Programmable Gate Array (FPGA) platform. The implementation results show that the through put of the developed architecture is 600K Complex Correlation matrix generation per second which is equivalent to 65.57 fps to image a 20cm depth target at a Pulse Repetition Frequency (PRF) of 4 KHz, 600 Field of View and 10 lateral resolution and makes the MVDR a viable solution for practical systems.

Evaluation of the effect of microphone cavity geometries on acoustic imaging in wind tunnels

Aeroacoustic measurements performed by flush-mounted microphone arrays on the walls of closed-section wind tunnels are contaminated by the hydrodynamic pressure fluctuations of the wall’s boundary layer. This study evaluates three different microphone cavity geometries for mitigating this issue. Their improvement to the signal-to-noise ratio (SNR) and the accuracy of their acoustic imaging results are compared to a flush-mounted microphone array. The four geometries include: (1) an array of flush-mounted microphones as the baseline, (2) a cylindrical hard-plastic cavity with a countersink, (3) a conical cavity made of melamine acoustic absorbing foam, and (4) a conical cavity with star-shaped protrusions, also made of melamine. The three arrays with cavities were covered with a steel-wire cloth to reduce the boundary layer fluctuations at the microphone while the baseline array was uncovered. Two sound sources were tested in an aeroacoustic wind tunnel for assessing the performance of the different cavities: a speaker placed outside the flow and a distributed sound source generated by a flat plate inside of the flow. When using conventional frequency domain beamforming, both cavities made of melamine offer up to a 30 dB increase in SNR with respect to the flush-mounted case, followed by the hard-walled cavity with up to a 20 dB increase. This is a 20 dB improvement when compared to the single microphone cases. The melamine cavities also provide cleaner acoustic source maps and accurate spectral estimations for a wider frequency range. The effect of cavity placement and geometry on the coherence, which affects the beamforming analysis of the acoustic signal was negligible for all cases. Distributed sound source measurements using the three arrays agreed with predictions using the Brooks, Pope, and Marcolini (BPM) model, showing that the cavities could detect vortex shedding that was undetectable by the flush array.

Advances in ultrasound imaging: Passive cavitation imaging/mapping

Ultrasound-induced cavitation has been demonstrated to induce numerous bioeffects, including tissue ablation and drug delivery. Different types of cavitation activity (e.g., inertial or stable) nucleate different bioeffects and are associated with distinct regimes of ultrasonic emissions. Thus decades of research have investigated the emission-bioeffect relationship. Since 2008, beamforming of cavitation emissions has been performed to provide images of cavitation activity. This enables the spatio-temporal localization, characterization, and quantification of cavitation activity, which has been shown to have utility in both image-guidance of therapies and basic science studies. The goal of this tutorial is to familiarize participants with the principles and implementation techniques for array-based passive cavitation monitoring. We will begin with a review the fundamentals of time-domain and frequency-domain beamforming, including an opportunity to form images during the tutorial using data and MATLAB code provided by the speakers. Next, we will discuss steps to correct for propagation path and hardware-specific effects in order to provide improved cavitation dose estimates. Display considerations and B-mode duplex imaging topics will follow. Finally, common hardware configurations, including benefits and limitations, will be covered to provide guidance for individuals considering a setup in their laboratory.

Steering Vector Transformation Technique for the Design of Point Sources Wideband Beamforḿer.(Dept.E)

Steering vector transformation technique for wideband antenna array of point sources is presented. Wideband direction of arrival (DoA) estimation using this technique is illustrated. In the present work, steering vector transformation is applied to the conventional frequency domain beamformer to enhance the frequency invariance of the mainlobe pattern. Several simulation examples are presented to illustrate the advantage of the proposed frequency domain beamformer over the conventional one. As the array size increases or the interelement spacing decreases, the proposed beamformer still has a better frequency invariant property of the mainlobe pattern than the conventional beamformer. Finally, using windows, a better frequency invariant property of the mainlobe patter is obtained by the proposed beamformer than the conventional one.

Acoustic imaging with conventional frequency domain beamforming and generalized cross correlation: a comparison study

Nowadays, acoustic cameras have become a standard tool for visualizing sound sources. The main post-processing techniques for generating acoustic images are associated with beamforming in the time or frequency domain. While many advanced techniques have been developed for the latter in order to achieve high resolution, such as deconvolution or inverse methods, the former have been less investigated. Recent studies have shown that time-domain beamforming can provide a narrow main lobe with low side lobe levels in different situations. However, a comparison of both techniques in terms of acoustic imaging does not exist. This paper provides a detailed comparison of conventional frequency-domain beamforming (CBF) and time-domain beamforming based on the generalized cross-correlation technique (GCC). First, numerical data are used for assessing both techniques. Then, experimental tests are carried out in a hemi-anechoic room using loudspeakers or power tools. As expected, acoustic images provided by CBF and GCC are found to be very similar but with different computation times. In order to improve the acoustic image, advanced techniques are considered (Clean-SC and the GCC based on geometric mean). When dealing with low frequency, Clean-SC is not able to detect both sources while the GCC based on geometric mean provides an accurate source separation.