Commercially available cardiac scanners use 64–128 elements phased-array (PA) probes and classical delay-and-sum beamforming to reconstruct a sector B-mode image. For portable and hand-held scanners, which are the fastest growing market, channel count reduction can greatly decrease the total power and cost of devices. The introduction of ultra-fast imaging methods based on plane waves and diverging waves provides new insight into heart’s moving structures and enables the implementation of new myocardial assessment and advanced flow estimation methods, thanks to much higher frame rates. The goal of this study was to show the feasibility of reducing the channel count in the diverging wave synthetic aperture image reconstruction method for phased-arrays. The application of ultra-fast 32-channel subaperture imaging combined with spatial compounding allowed the frame rate of approximately 400 fps for 120 mm visualization to be achieved with image quality obtained on par with the classical 64-channel beamformer. Specifically, it was shown that the proposed method resulted in image quality metrics (lateral resolution, contrast and contrast-to-noise ratio), for a visualization depth not exceeding 50 mm, that were comparable with the classical PA beamforming. For larger visualization depths (80–100 mm) a slight degradation of the above parameters was observed. In conclusion, diverging wave phased-array imaging with reduced number of channels is a promising technology for low-cost, energy efficient hand-held cardiac scanners.
The paper contains the abstracts of papers presented during JOINT CONFERENCE – ACOUSTICS 2018, Ustka, Poland, September 11 – 14, 2018 : 65th Open Seminar on Acoustics 35th Symposium on Hydroacoustics Polish-German Structured Conference on Acoustics
Biography and scientific achievements of Academician Leonid Maksimovich Brekhovskikh - Russian physicist, the founder of the scientific school of Ocean Acoustics, Doctor of Physics and Mathematics Sciences, Professor, Academician of the Russian Academy of Sciences.
Although the emotions and learning based on emotional reaction are individual-specific, the main features are consistent among all people. Depending on the emotional states of the persons, various physical and physiological changes can be observed in pulse and breathing, blood flow velocity, hormonal balance, sound properties, face expression and hand movements. The diversity, size and grade of these changes are shaped by different emotional states. Acoustic analysis, which is an objective evaluation method, is used to determine the emotional state of people’s voice characteristics. In this study, the reflection of anxiety disorder in people’s voices was investigated through acoustic parameters. The study is a case-control study in cross-sectional quality. Voice recordings were obtained from healthy people and patients. With acoustic analysis, 122 acoustic parameters were obtained from these voice recordings. The relation of these parameters to anxious state was investigated statistically. According to the results obtained, 42 acoustic parameters are variable in the anxious state. In the anxious state, the subglottic pressure increases and the vocalization of the vowels decreases. The MFCC parameter, which changes in the anxious state, indicates that people can perceive this situation while listening to the speech. It has also been shown that text reading is also effective in triggering the emotions. These findings show that there is a change in the voice in the anxious state and that the acoustic parameters are influenced by the anxious state. For this reason, acoustic analysis can be used as an expert decision support system for the diagnosis of anxiety.
Listening tests have been carried out to quantify the significance of binaural auralization over monaural auralization in accordance with the acoustic properties of the enclosure. To this end, acoustic rendering of three different rooms were generated based on synthesized monaural (two channels with the same audio material) and binaural room impulse responses. The auralizations were evaluated by means of subjective tests using headphones with non-individualized equalization. Parameters, such as localization, spatial impression and realism, were taken into consideration to determine the relevance of providing binaural information for the auralization of a given room. The analysis of the data has been conducted following a statistical approach based on ANOVA and Pearson correlation. The results indicate that spatial perception is strongly dependent on the acoustic characteristics of the rooms and on the listening condition of the audio material. Furthermore, as expected, advantages of binaural rendering in terms of source localization was also confirmed.
Speech and music signals are multifractal phenomena. The time displacement profile of speech and music signal show strikingly different scaling behaviour. However, a full complexity analysis of their frequency and amplitude has not been made so far. We propose a novel complex network based approach (Visibility Graph) to study the scaling behaviour of frequency wise amplitude variation of speech and music signals over time and then extract their PSVG (Power of Scale freeness of Visibility Graph). From this analysis it emerges that the scaling behaviour of amplitude-profile of music varies a lot from frequency to frequency whereas it’s almost consistent for the speech signal. Our left auditory cortical areas are proposed to be neurocognitively specialised in speech perception and right ones in music. Hence we can conclude that human brain might have adapted to the distinctly different scaling behaviour of speech and music signals and developed different decoding mechanisms, as if following the so called Fractal Darwinism. Using this method, we can capture all non-stationary aspects of the acoustic properties of the source signal to the deepest level, which has huge neurocognitive significance. Further, we propose a novel non-invasive application to detect neurological illness (here autism spectrum disorder, ASD), using the quantitative parameters deduced from the variation of scaling behaviour for speech and music.
For building applications, woven fabrics have been widely used as finishing elements of room interior but not in particular aimed for sound absorbers. Considering the micro perforation of the woven fabrics, they should have potential to be used as micro-perforated panel (MPP) absorbers; some measurement results indicated such absorption ability. Hence, it is of importance to have a sound absorption model of the woven fabrics to enable us predicting their sound absorption characteristic that is beneficial in engineering design phase. Treating the woven fabric as a rigid frame, a fluid equivalent model is employed based on the formulation of Johnson-Champoux-Allard (JCA). The model obtained is then validated by measurement results where three kinds of commercially available woven fabrics are evaluated by considering their perforation properties. It is found that the model can reasonably predict their sound absorption coefficients. However, the presence of perturbations in pores give rise to inaccuracy of resistive component of the predicted surface impedance. The use of measured static flow resistive and corrected viscous length in the calculations are useful to cope with such a situation. Otherwise, the use of an optimized simple model as a function of flow resistivity is also applicable for this case.
A mixed pseudo-orthogonal frequency coding (Mixed-POFC) structure is proposed as a new spreadspectrum technique in this paper, which employs frequency and time diversity to enhance tag properties and balances the spectrum utilization and code diversity. The coding method of SAW RFID tags in this paper uses Mixed-POFC with multi-track chip arrangements. The cross-correlation and auto correlation of Mixed-POFC and POFC are calculated to demonstrate the reduced overlap between the adjacent center frequencies with the Mixed-POFC method. The center frequency of the IDT and Bragg reflectors is calculated by a coupling of modes (COM) module. The combination of the calculation results of the Bragg reflectors shows that compared with a 7-chip POFC, the coding number of a 7-chip Mixed-POFC is increased from 120 to 144 with the same fractional bandwidth of 12%. To demonstrate the validity of Mixed-POFC, finite element analysis (FEA) technology is used to analyze the frequency characteristics of Mixed-POFC chips. The maximum error between designed frequencies and simulation frequencies is only 1.7%, which verifies that the Mixed-POFC method is feasible.
This paper proposes an analytical model to describe the interaction of a bounded ultrasonic beam with an immersed plate. This model, based on the Gaussian beams decomposition, takes into account multiple reflections into the plate. It allows predicting three-dimensional spatial distributions of both transmitted and reflected fields. Thereby, it makes it easy to calculate the average pressure over the receiver’s area taking into account diffraction losses. So the acoustical parameters of the plate can be determined more accurately. A Green’s function for the interaction of an ultrasonic beam with the plate is derived. The obtained results are compared to those given by the angular spectrum approach. A good agreement is seen showing the validity of the proposed model.
A plenum window with incorporation of Helmholtz resonators in between two glass panes was tested in a reverberation room. The effects of jagged flap on reducing strength of diffracted sound was also investigated in the present studies where white, traffic and construction noises were examined during each set of experiment. When the noise source was located at the central line of the plenum window, the plenum window with Helmholtz resonators was able to mitigate 8.5 dBA, 8.9 dBA and 8.2 dBA of white, traffic and construction noises, respectively, compared with the case of without window. These amounts of noises that attenuated by the plenum window were slightly higher than the case where noise source was diverged 30º away from the plenum window. The effects of jagged flaps on the acoustical performance of the plenum window were negligible. The Helmholtz resonators had the best performance in the frequency region between 900 Hz to 1300 Hz where in this frequency range, the plenum window with Helmholtz resonators was able to attenuate additional 1.7 dBA, 1.9 dBA and 1.6 dBA of white, traffic and construction noises, respectively, compared with the case of without resonators.
The locally resonant phononic crystal (LRPC) composite double panel structure (DPS) made of a twodimensional periodic array of a two-component cylindrical LR pillar connected between the upper and lower composite plates is proposed. The plates are composed of two kinds of materials and periodically etched holes. In order to reveal the bandgap properties of structure theoretically, the band structures, displacement fields of eigenmodes and transmission power spectrums of corresponding 8 × 8 finite structure are calculated and displayed by using finite element method (FEM). Numerical results and further analysis demonstrate that if the excitation and response points are picked on different sides of the structure, a wide band gap with low starting frequency is opened, which can be treated as the coupling between dominant vibrations of pillars and plate modes. In addition, the influences of filled-in rubber, etched hole and viscidity of soft material on band gap are studied and understood with the help of “base-spring-mass” simplified model.
Ray tracing simulation of sound field in rooms is a common tool in room acoustic design for predicting impulse response. There are numerous commercial engineering tools utilising ray tracing simulation. A specific problem in the simulation is the modelling of diffuse reflections when contribution of individual surface is prevailing. The paper introduces modelling of scattering which is interesting when the whole impulse response of a room is not a goal but contribution of certain surface. The main goal of the project is to shape directivity characteristics of scattered reflection. Also, an innovative approach is suggested for converting the energy histogram information obtained by ray tracing into an “equivalent impulse response”. The proposed algorithm is tested by comparing the results with measurements in a real sound field, realised in a scaled model where a diffusing surface is hardware-implemented.
Passive noise reduction means are commonly used to reduce noise in the industry but, unfortunately, their effectiveness is poor in the low frequency range. By applying active structural acoustic control to the enclosure walls significant improvement of the insulating properties in this frequency range can be achieved. In this paper a model of double panel structure with ASAC is presented. The structure consists of two aluminium plates separated by an air gap. Two inertial magnetoelectric actuators and two piezoceramic MFC sensors were used for controlling the structure. A multichannel FxLMS algorithm with virtual error microphone technique is used as a control algorithm. The signal of a virtual error microphone is extrapolated basing on signals from MFC sensors. Performance of this actively controlled structure for tonal signals at selected frequencies is presented in the article. During the study, a double panel structure was mounted on one wall of sound insulating enclosure located in an acoustic chamber. During the measurements local and global reduction of noise test signal was investigated.
An original model based on first principles is constructed for the temporal correlation of acoustic waves propagating in random scattering media. The model describes the dynamics of wave fields in a previously unexplored, moderately strong (mesoscopic) scattering regime, intermediate between those of weak scattering, on the one hand, and diffusing waves, on the other. It is shown that by considering the wave vector as a free parameter that can vary at will, one can provide an additional dimension to the data, resulting in a tomographic-type reconstruction of the full space-time dynamics of a complex structure, instead of a plain spectroscopic technique. In Fourier space, the problem is reduced to a spherical mean transform defined for a family of spheres containing the origin, and therefore is easily invertible. The results may be useful in probing the statistical structure of various random media with both spatial and temporal resolution.
Recently, the rapid advancement of the IT industry has resulted in significant changes in audio-system configurations; particularly, the audio over internet protocol (AoIP) network-based audio-transmission technology has received favourable evaluations in this field. Applying the AoIP in a certain section of the multiple-cable zone is advantageous because the installation cost is lower than that for the existing systems, and the original sound is transmitted without any distortion. The existing AoIP-based technology, however, cannot control the audio-signal characteristics of every device and can only transmit multiple audio signals through a network. In this paper, the proposed Audio Network & Control Hierarchy Over peer-to-peer (Anchor) system enables all audio equipment to send and receive signals via a data network, and the receiving device can mix the signals of different IPs. Accordingly, it was possible to improve the system-application flexibility by simplifying the audio-system configuration. The research results confirmed that the received audio signals from different IPs were received, mixed, and output without errors. It is expected that Anchor will become a standard for audio-network protocols.
The aim of the study was to determine the signal-to-noise ratio (SNR) for the Speech Reception Threshold (SRT) for young persons with normal hearing. The following three tests available for Polish language were used: the New Articulation Lists (NAL-93) version of 2011, the Polish Sentence Test (PST) and the Polish Sentence Matrix Test (PSMT). When using PST and PSMT the masking signal was babble noise made of the language material contained in the test. For NAL-93 the masking signal was speech noise. The speech reception threshold (SRT) was found to be (−6:8 ± 1.1), (−4:8 ± 1.6), (−3:5 ± 1.8) and (−3:4 ± 2.0) dB SNR for PST, PSMT, NAL-93 (constant stimuli method) and NAL-93 (short method), respectively. The values of SRT depend on semantic redundancy of the language material. Differences in SRT were statistically non-significant only for NAL-93 (constant stimuli method) and NAL-93 (short method). Moreover, it was shown that the time needed for presentation of a single word list (NAL-93, short method) or single sentence list (PST, PSMT) was comparable and equal to 2–3 minutes. The most uniform SRT values were obtained for PST. The PSMT was the least demanding for the listener, experimenter and equipment.
Quantitative ultrasound has been widely used for tissue characterization. In this paper we propose a new approach for tissue compression assessment. The proposed method employs the relation between the tissue scatterers’ local spatial distribution and the resulting frequency power spectrum of the backscattered ultrasonic signal. We show that due to spatial distribution of the scatterers, the power spectrum exhibits characteristic variations. These variations can be extracted using the empirical mode decomposition and analyzed. Validation of our approach is performed by simulations and in-vitro experiments using a tissue sample under compression. The scatterers in the compressed tissue sample approach each other and consequently, the power spectrum of the backscattered signal is modified. We present how to assess this phenomenon with our method. The proposed in this paper approach is general and may provide useful information on tissue scattering properties.
Noise induced hearing loss (NIHL) as one of the major avoidable occupational related health issues has been studied for decades. To assess NIHL, the excitation pattern (EP) has been considered as one of the mechanisms to estimate the movements of the basilar membrane (BM) in the cochlea. In this study, two auditory filters, dual resonance nonlinear (DRNL) filter and rounded-exponential (ROEX) filter are applied to create two EPs, the velocity EP and the loudness EP respectively. Two noise hazard metrics are proposed based on two proposed EPs to evaluate hazardous levels caused by different types of noise. Moreover Gaussian noise and tone signals are simulated to evaluate performances of the proposed EPs and the noise metrics. The results show that both EPs can reflect the responses of the BM to different types of noise. For Gaussian noise there is a frequency shift between the velocity EP and the loudness EP. The results suggest that both EPs can be used for assessment of NIHL.
This paper presents and analyses the results of a simulation of the acoustic field distribution in sectors of a 1024-element ring array, intended for the diagnosis of female breast tissue with the use of ultrasonic tomography. The array was tested for the possibility to equip an ultrasonic tomograph with an additional modality - conventional ultrasonic imaging with the use of individual fragments (sections) of the ring array. To determine the acoustic field for sectors of the ring array with a varying number of activated ultrasonic transducers, a combined sum of all acoustic fields created by each elementary transducer was calculated. By the use of MATLAB software, a unique algorithm was developed, for a numerical determination of the distribution of pressure of an ultrasonic wave on any surface or area of the medium generated by the concave curvilinear structure of rectangular ultrasound transducers with a geometric focus of the beam. The analysis of the obtained results of the acoustic field distribution inside the ultrasonic ring array used in tomography allows to conclude that the optimal number of transducers in a sector enabling to obtain ultrasound images using linear echographic scanning is 32 ≤ n ≤ 128, taking into account that due to an increased temporal resolution of ultrasonic imaging, this number should be as low as possible.