Related Transfer Functions Research Articles

Informed Sound Source Localization Using Relative Transfer Functions for Hearing Aid Applications

Recent hearing aid systems (HASs) can connect to a wireless microphone worn by the talker of interest. This feature gives the HASs access to a noise-free version of the target signal. In this paper, we address the problem of estimating the target sound direction of arrival (DoA) for a binaural HAS given access to the noise-free content of the target signal. To estimate the DoA, we present a maximum-likelihood framework which takes the shadowing effect of the user's head on the received signals into account by modeling the relative transfer functions (RTFs) between the HAS's microphones. We propose three different RTF models which have different degrees of accuracy and individualization. Furthermore, we show that the proposed DoA estimators can be formulated in terms of inverse discrete Fourier transforms to evaluate the likelihood function computationally efficiently. We extensively assess the performance of the proposed DoA estimators for various DoAs, signal to noise ratios, and in different noisy and reverberant situations. The results show that the proposed estimators improve the performance markedly over other recently proposed “informed” DoA estimator.

Combined LCMV-TRINICON Beamforming for Separating Multiple Speech Sources in Noisy and Reverberant Environments

The problem of source separation using an array of microphones in reverberant and noisy conditions is addressed. We consider applying the well-known linearly constrained minimum variance (LCMV) beamformer (BF) for extracting individual speakers. Constraints are defined using relative transfer functions (RTFs) for the sources, which are ratios of acoustic transfer functions (ATFs) between any microphone and a reference microphone. The latter are usually estimated by methods that rely on single-talk time segments where only a single source is active and on reliable knowledge of the source activity. Two novel algorithms for estimation of RTFs using the “Triple N” ICA for convolutive mixtures (TRINICON) framework are proposed, not resorting to the usually unavailable source activity pattern. The first algorithm estimates the RTFs of the sources by applying multiple two-channel geometrically constrained (GC) TRINICON units, where approximate direction of arrival information for the sources is utilized for ensuring convergence to the desired solution. The GC-TRINICON is applied to all microphone pairs using a common reference microphone. In the second algorithm, we propose to estimate RTFs iteratively using GC-TRINICON, where instead of using a fixed reference microphone as before, we suggest to use the output signals of LCMV-BFs from the previous iteration as spatially processed references with improved signal-to-interference-and-noise ratio. For both algorithms, a simple detection of noise-only time segments is required for estimating the covariance matrix of noise and interference. We conduct an experimental study in which the performance of the proposed methods is confirmed and compared to corresponding supervised methods.

Position estimation of binaural sound source in reverberant environments

Most binaural sound source systems perform localization in either direction or distance perception. However, in real scenarios both perceptions are important to estimate source position in various environment conditions especially with the rapid technological growth in smart machines and their involvement in human daily life. This paper introduces an approach for azimuth and distance of binaural sound source localization in different reverberating environments using only two microphones. The algorithm is based on statistical features of the binaural cues and the difference of the binaural magnitude spectra of the binaural signal. Gaussian Mixture Models (GMMs) are used to jointly learn both distances and azimuths in different reverberant rooms. The proposed system does not require any prior knowledge of head related transfer function (HRTF), acoustical environment or room parameters. The performance has been evaluated at different aspects and conditions and reported effective and robust results, especially in the case of training set mismatch.

Transformation functions and trajectories of a hypocycloid mechanism completed with a dyad

It starts from a planetary mechanism that generates hypocycloids and it is completed with an RPP dyad, so that transfer functions are generated, i.e. races along the x and y directions. Different hypocycloids (normal, elongated, shortened) are generated and transfer functions are established for these cases. The obtained curves and transfer functions are discussed, establishing some conclusions. All study includes various range of lengths for the cinematic links so the important and interesting trajectories are found. The related transfer functions can be used in a diversity of applications. These findings are useful for special devices for automatic lathes, horizontal forging machines, textile machinery, and kinetic art.

Comparison of Fractional Order Modelling and Integer Order Modelling of Fractional Order Buck Converter in Continuous Condition Mode Operation

Producing a mathematical model with high accuracy is the first and important step in control of systems. Nowadays fractional calculus has been in the spotlight and it has a lot of application especially in control engineering. Fractional modelling on one of the conventional converters is done in this paper. Fractional state space model and related fractional transfer functions for a fractional DC/DC Buck converter is established and achieved results are compared to integer order models. At the end of this paper Oustaloup's recursive approximation is introduced and imposed for one of gathered fractional transfer function.

Estimation of the Direct-Path Relative Transfer Function for Supervised Sound-Source Localization

This paper addresses the problem of binaural localization of a single speech source in noisy and reverberant environments. For a given binaural microphone setup, the binaural response corresponding to the direct-path propagation of a single source is a function of the source direction. In practice, this response is contaminated by noise and reverberations. The direct-path relative transfer function (DP-RTF) is defined as the ratio between the direct-path acoustic transfer function of the two channels. We propose a method to estimate the DP-RTF from the noisy and reverberant microphone signals in the short-time Fourier transform domain. First, the convolutive transfer function approximation is adopted to accurately represent the impulse response of the sensors in the STFT domain. Second, the DP-RTF is estimated by using the auto- and cross-power spectral densities at each frequency and over multiple frames. In the presence of stationary noise, an inter-frame spectral subtraction algorithm is proposed, which enables to achieve the estimation of noise-free auto- and cross-power spectral densities. Finally, the estimated DP-RTFs are concatenated across frequencies and used as a feature vector for the localization of speech source. Experiments with both simulated and real data show that the proposed localization method performs well, even under severe adverse acoustic conditions, and outperforms state-of-the-art localization methods under most of the acoustic conditions.

Investigation on chatter stability of thin-walled parts in high-speed milling based on relative transfer functions

High-speed milling of thin-walled part is a common application for aerospace industry, automotive, computer hardware and bioengineering industries. Aiming at the small-stiffness frequency response characteristics of thin-walled parts, the dynamic model and critical condition of stability was proposed by the relative transfer functions between cutter subsystem and thin-walled part subsystem in this paper. The stability critical domain was predicted by semi-discretization (SD) method based on the dynamic behavior of the multi-DOF system. It can be shown that the chatter condition in high-speed milling is closely related to both cutter speed and depth of cut, besides cutter geometry, engagement conditions, relative frequency response function, material property of thin-walled part and so on. This conclusion provides a theoretical foundation and reference for the milling mechanism research.

4-Degree-of-freedom anti-windup scheme for plants with actuator saturation

In this paper, a 4-degree-of-freedom (DoF) anti-windup scheme is developed for anti-windup compensation of plants with actuator saturation. Instead of LMIs (Linear Matrix Inequalities) in modern anti-windup technique, sufficient stability conditions of the proposed anti-windup scheme are derived under the framework of IQC (Integral Quadratic Constraint). The sufficient conditions are formulated as a checking on the positive definiteness of the related transfer function at all frequencies for MIMO systems (or positive sign for SISO systems), which provides a fairly straightforward tuning rule in the frequency domain. Moreover, by virtue of the proposed scheme, control design for disturbance response is decoupled from set-point response. The proposed scheme can be implemented without system states or state estimates. Compared with IMC-based, low-order or dynamic compensation anti-windup schemes, numerical example shows remarkable performance improvement of the proposed scheme for the saturated systems.

Investigation on chatter stability of thin-walled parts in milling based on process damping with relative transfer functions

Chatter usually occurs in cutting of thin-walled workpiece due to poor structural stiffness, which results in poor surface quality and damaged tool. Aiming at process damping caused by interference between a tool flank face and a machined surface of thin-walled part, the dynamic model and critical condition of stability are proposed by the relative transfer functions, when both the tool structure and the machined workpiece have similar dynamic behaviors in this paper. Using the frequency method to solve the stability of the cutting chatter, it can be seen that the process damping can significantly improve the stability of the low speed region. Moreover, the stability domain is different and more exact than the one that derives from the simple superposition of the tool and the workpiece lobe diagrams. The correctness of the model is validated by experiments. These conclusions provide a theoretical foundation and reference for the milling mechanism research.

An Expectation-Maximization Algorithm for Multimicrophone Speech Dereverberation and Noise Reduction With Coherence Matrix Estimation

In speech communication systems, the microphone signals are degraded by reverberation and ambient noise. The reverberant speech can be separated into two components, namely, an early speech component that consists of the direct path and some early reflections and a late reverberant component that consists of all late reflections. In this paper, a novel algorithm to simultaneously suppress early reflections, late reverberation, and ambient noise is presented. The expectation-maximization (EM) algorithm is used to estimate the signals and spatial parameters of the early speech component and the late reverberation components. As a result, a spatially filtered version of the early speech component is estimated in the E-step. The power spectral density (PSD) of the anechoic speech, the relative early transfer functions, and the PSD matrix of the late reverberation are estimated in the M-step of the EM algorithm. The algorithm is evaluated using real room impulse response recorded in our acoustic lab with a reverberation time set to 0.36 s and 0.61 s and several signal-to-noise ratio levels. It is shown that significant improvement is obtained and that the proposed algorithm outperforms baseline single-channel and multichannel dereverberation algorithms, as well as a state-of-the-art multichannel dereverberation algorithm.

Source fields reconstruction with 3D mapping by means of the virtual acoustic volume concept

This paper presents the theoretical framework of the virtual acoustic volume concept and two related inverse Patch Transfer Functions (iPTF) identification methods (called u-iPTF and m-iPTF depending on the chosen boundary conditions for the virtual volume). They are based on the application of Green׳s identity on an arbitrary closed virtual volume defined around the source. The reconstruction of sound source fields combines discrete acoustic measurements performed at accessible positions around the source with the modal behavior of the chosen virtual acoustic volume. The mode shapes of the virtual volume can be computed by a Finite Element solver to handle the geometrical complexity of the source. As a result, it is possible to identify all the acoustic source fields at the real surface of an irregularly shaped structure and irrespective of its acoustic environment. The m-iPTF method is introduced for the first time in this paper. Conversely to the already published u-iPTF method, the m-iPTF method needs only acoustic pressure and avoids particle velocity measurements. This paper is focused on its validation, both with numerical computations and by experiments on a baffled oil pan.

Relative transfer function (RTF) estimation utilising peaks in time‐domain RTF

An effective algorithm for a more accurate estimation of relative transfer function (RTF) to improve the directional-blind source separation-based RTF estimation is presented. The peaks in time-domain RTF (PTDR) as a feature of time-domain RTF to improve the conventional algorithm are utilised. The PTDRs contain important characteristics of RTF; however, they are smoothed due to the geometric constraint in the conventional algorithm. In the proposed algorithm, the PTDRs are recovered by employing the semi-non-negative matrix factorisation algorithm to effectively find the PTDRs from other peaks. A series of experiments using interfere and various noises demonstrates effectiveness of the proposed RTF estimation algorithm utilising the PTDR.

A theoretical and numerical resolution of an acoustic multiple scattering problem in three-dimensional case

This paper develops an analytical solution for sound, electromagnetic or any other wave propagation described by the Helmholtz equation in three-dimensional case. First, a theoretical investigation based on multipole expansion method and spherical wave functions was established, through which we show that the resolution of the problem is reduced to solving an infinite, complex and large linear system. Second, we explain how to suitably truncate the last infinite dimensional system to get an accurate stable and fast numerical solution of the problem. Then, we evaluate numerically the theoretical solution of scattering problem by multiple ideal rigid spheres. Finally, we made a numerical study to present the “Head related transfer function” with respect to different physical and geometrical parameters of the problem.

Space Safety Standard Commercial Human-Rated System IAASS-ISSB-S-1700-REV-B

Crew safety is of paramount importance for future human exploratory spaceflight. In order to ensure optimum safety standards, the design criteria for this new generation of spacecraft must be matured, such as the level of crew safety required. This will be guided by an adequate implementation of the safety assessment process, which is under continuous improvement. This should be done in cooperation with international partners, whilst considering expected project constraints such as cost and time.Through a hazard control process assessment, several risk mitigation strategies may be found suitable to achieve both an acceptable level of crew safety and mission success. Such an assessment thus requires both: elaboration of a regulated safety assessment process, that is shared and approved worldwide, and definition of safety criteria that are required to fulfil the mission objectives, and need to be approved by a Certification Authority.This paper defines what are the reliability and safety standards used for current low earth orbit human spaceflight operations through a failure conditions safety assessment, applying it to crew transportation systems and taking into account the mission aims.It will evaluate existing standardized safety and regulatory processes that are in place for civil aviation (ARP 4761, ARP 4754A, AC 25.1309), which are largely accepted as being a premier example of industry safety standards, and assess how it could be extrapolated and adapted to ensure similar safety standards in future human space exploration missions.The civil aviation process presents commonalities in terms of methods for conducting a safety assessment, that are expandable to space exploration. This paper describes the related transfer function from civil aviation to space exploration (ISS, Moon and Mars) as a stepping stone to reach an international consensus to be targeted by the Space Agencies to ensure the crew a safe journey and return to Earth.

The Binaural LCMV Beamformer and its Performance Analysis

The recently proposed binaural linearly constrained minimum variance (BLCMV) beamformer is an extension of the well-known binaural minimum variance distortionless response (MVDR) beamformer, imposing constraints for both the desired and the interfering sources. Besides its capabilities to reduce interference and noise, it also enables to preserve the binaural cues of both the desired and interfering sources, hence making it particularly suitable for binaural hearing aid applications. In this paper, a theoretical analysis of the BLCMV beamformer is presented. In order to gain insights into the performance of the BLCMV beamformer, several decompositions are introduced that reveal its capabilities in terms of interference and noise reduction, while controlling the binaural cues of the desired and the interfering sources. When setting the parameters of the BLCMV beamformer, various considerations need to be taken into account, e.g. based on the amount of interference and noise reduction and the presence of estimation errors of the required relative transfer functions (RTFs). Analytical expressions for the performance of the BLCMV beamformer in terms of noise reduction, interference reduction, and cue preservation are derived. Comprehensive simulation experiments, using measured acoustic transfer functions as well as real recordings on binaural hearing aids, demonstrate the capabilities of the BLCMV beamformer in various noise environments.

Fractional-order linear digital 1D and 2D filter response calculation using Matlab

In this paper a simple method of the fractional-order linear digital filter response calculation is proposed. The fractional-order linear digital filter description is based on the fractional-order linear time-invariant difference equation and related fractional-order discrete transfer function.

A study on complexity reduction of binaural decoding in multi-channel audio coding for realistic audio service

In this paper, we proposed the simplified binaural decoding method for reducing the complexity of the binaural decoding. In the proposed simplified binaural decoding the high frequency components of the HRTF (head related transfer function) coefficients are excluded and the binaural decoding process in the high frequency regions is simplified. From the experimental results, it is confirmed that the proposed method greatly reduces the complexity of the binaural decoding in the frequency domain as 40 % and shows the statistically same sound quality compared to the binaural decoding in the frequency domain.

Theoretical Analysis of Linearly Constrained Multi-Channel Wiener Filtering Algorithms for Combined Noise Reduction and Binaural Cue Preservation in Binaural Hearing Aids

Besides noise reduction, an important objective of binaural speech enhancement algorithms is the preservation of the binaural cues of all sound sources. For the desired speech source and the interfering sources, e.g., competing speakers, this can be achieved by preserving their relative transfer functions (RTFs). It has been shown that the binaural multi-channel Wiener filter (MWF) preserves the RTF of the desired speech source, but typically distorts the RTF of the interfering sources. To this end, in this paper we propose two extensions of the binaural MWF, i.e., the binaural MWF with RTF preservation (MWF-RTF) aiming to preserve the RTF of the interfering source and the binaural MWF with interference rejection (MWF-IR) aiming to completely suppress the interfering source. Analytical expressions for the performance of the binaural MWF, MWF-RTF and MWF-IR in terms of noise reduction, speech distortion and binaural cue preservation are derived, showing that the proposed extensions yield a better performance in terms of the signal-to-interference ratio and preservation of the binaural cues of the directional interference, while the overall noise reduction performance is degraded compared to the binaural MWF. Simulation results using binaural behind-the-ear impulse responses measured in a reverberant environment validate the derived analytical expressions for the theoretically achievable performance of the binaural MWF, MWF-RTF, and MWF-IR, showing that the performance highly depends on the position of the interfering source and the number of microphones. Furthermore, the simulation results show that the MWF-RTF yields a very similar overall noise reduction performance as the binaural MWF, while preserving the binaural cues of both the speech and the interfering source.

Theoretical Analysis of Binaural Transfer Function MVDR Beamformers with Interference Cue Preservation Constraints

The objective of binaural noise reduction algorithms is not only to selectively extract the desired speaker and to suppress interfering sources (e.g., competing speakers) and ambient background noise, but also to preserve the auditory impression of the complete acoustic scene. For directional sources this can be achieved by preserving the relative transfer function (RTF) which is defined as the ratio of the acoustical transfer functions relating the source and the two ears and corresponds to the binaural cues. In this paper, we theoretically analyze the performance of three algorithms that are based on the binaural minimum variance distortionless response (BMVDR) beamformer, and hence, process the desired source without distortion. The BMVDR beamformer preserves the binaural cues of the desired source but distorts the binaural cues of the interfering source. By adding an interference reduction (IR) constraint, the recently proposed BMVDR-IR beamformer is able to preserve the binaural cues of both the desired source and the interfering source. We further propose a novel algorithm for preserving the binaural cues of both the desired source and the interfering source by adding a constraint preserving the RTF of the interfering source, which will be referred to as the BMVDR-RTF beamformer. We analytically evaluate the performance in terms of binaural signal-to-interference-and-noise ratio (SINR), signal-to-interference ratio (SIR), and signal-to-noise ratio (SNR) of the three considered beamformers. It can be shown that the BMVDR-RTF beamformer outperforms the BMVDR-IR beamformer in terms of SINR and outperforms the BMVDR beamformer in terms of SIR. Among all beamformers which are distortionless with respect to the desired source and preserve the binaural cues of the interfering source, the newly proposed BMVDR-RTF beamformer is optimal in terms of SINR. Simulations using acoustic transfer functions measured on a binaural hearing aid validate our theoretical results.

Multimodal Perceptual Training for Improving Spatial Auditory Performance in Blind and Sighted Listeners

Abstract The use of individualised Head Related Transfer Functions (HRTF) is a fundamental prerequisite for obtaining an accurate rendering of 3D spatialised sounds in virtual auditory environments. The HRTFs are transfer functions that define the acoustical basis of auditory perception of a sound source in space and are frequently used in virtual auditory displays to simulate free-field listening conditions. However, they depend on the anatomical characteristics of the human body and significantly vary among individuals, so that the use of the same dataset of HRTFs for all the users of a designed system will not offer the same level of auditory performance. This paper presents an alternative approach to the use on non-individualised HRTFs that is based on a procedural learning, training, and adaptation to altered auditory cues.We tested the sound localisation performance of nine sighted and visually impaired people, before and after a series of perceptual (auditory, visual, and haptic) feedback based training sessions. The results demonstrated that our subjects significantly improved their spatial hearing under altered listening conditions (such as the presentation of 3D binaural sounds synthesised from non-individualized HRTFs), the improvement being reflected into a higher localisation accuracy and a lower rate of front-back confusion errors.