Room reverberation effects in hearing aid feedback cancellation

Room reverberation can affect feedback cancellation in hearing aids, with the strength of the effects depending on the acoustical conditions. These effects were studied using a behind the ear (BTE) hearing aid mounted on a dummy head and coupled to the ear canal via an open fitting. The hearing aid impulse response was measured for the dummy head placed at eight closely spaced locations in a typical office. The feedback cancellation in the hearing aid used a set of filter coefficients that were initialized for one location within the room, and then allowed to adapt to the feedback path measured at the same or to a different location. The maximum stable gain for the hearing aid was then estimated without feedback cancellation, for the initial set of feedback cancellation filter coefficients prior to adaptation, and for the feedback cancellation filter after adaptation. A low-order ARMA model combining a fixed set of poles with an adaptive FIR filter is shown to be effective in representing the feedback path exclusive of reverberation. Increasing the adaptive filter length has only a small benefit in improving the feedback cancellation performance due to the inability of the system to model the room reverberation. The mismatch between the modeled and actual feedback paths limits the headroom increase that can be achieved when using feedback cancellation, and varies with the location within the room.     

Effect of handset proximity on hearing aid feedback

The increased sensitivity of hearing aids to feedback as a telephone handset is brought near has been studied experimentally and numerically. For the measurements, three different hearing aids were modified so that the open-loop transfer function could be measured. They were mounted in the pinna of a mannikin and the change in open-loop transfer function determined as a function of handset proximity. Increases of over 20 dB were observed, most of this change occurring within the first 10 mm of separation between pinna and handset. Numerical calculations performed using a boundary element technique were in good agreement with the measurements.     

An evaluation of three adaptive hearing aid selection strategies

Paired-comparison judgments of intelligibility of speech in noise were obtained from eight hearing-impaired subjects on a large number of hearing aids simulated by a digital master hearing aid. The hearing aids which comprised a 5 X 5 matrix differed systematically in the amount of low-frequency and high-frequency gain provided. A comparison of three adaptive strategies for determining optimum hearing aid frequency-gain characteristics (an iterative round robin, a double elimination tournament, and a modified simplex procedure) revealed convergence on the same or similar hearing aids for most subjects. Analysis revealed that subjects for whom all three procedures converged on the same hearing aid showed a single pronounced peak in the response surface, while a broader peak was evident for the subjects for whom the three procedures identified similar hearing aids. The modified simplex procedure was found to be most efficient and the iterative round robin least efficient.     

Computer models of hearing aid transducers for integrated circuit design.

Electronic circuit modeling using computer-based simulation tools is well established and device models are available for common electronic components. However, acoustic models of audio transducers for use during integrated circuit design are not readily available. This causes difficulty for designers of audio amplifiers, and increases the uncertainties of a successful silicon integration of a circuit design. This paper reports on a technique for the creation of electroacoustic models of hearing aid microphones and receivers that can be connected to an amplifier under design, and incorporated into PSPICE simulations. Verification of the technique and models was performed by comparing measured frequency response data with graphs created by PSPICE modeling. The conclusions were that the method developed for creating these models, and the models themselves, were accurate enough to be used for acoustic simulations of frequency response performance during amplifier design, and gave results comparable to data obtained from breadboard measurements of the same circuits.     

Intelligibility ratings of continuous discourse: application to hearing aid selection

Twelve normal-hearing subjects rated the intelligibility of 35-s, hearing-aid-processed continuous discourse (CD) passages. Three talkers (two male, one female), four hearing aids, and two signal-to-babble (S/B) ratios were used in a completely crossed design. Research questions concerned: (1) ability of listeners to rate intelligibility, (2) sensitivity of hearing aid rankings were based on intelligibility ratings for three CD passages per instrument, and (3) dependence of hearing aid rankings on (a) S/B ratio, and (b) talker characteristics. Results were: (1) listeners were able to rate intelligibility, (2) rankings based on intelligibility ratings of three CD passages per hearing aid were capable of identifying two superior instruments within a group of four hearing aids that were similar in frequency/gain function, (3) listening in a more difficult S/B ratio substantially decreased the sensitivity of the hearing aid rankings for the female talker but had only minor effects on the rankings for the male talkers, and (4) hearing aid intelligibility rankings were found to be different for different talkers. Applications to hearing aid selection are discussed.     

The hearing aid feedback path: mathematical simulations and experimental verification

Acoustic feedback in hearing aids has received little attention in the literature. Feedback occurs when stability conditions of the open-loop transfer function of an in situ hearing aid are violated. Solving the feedback problem will first require knowledge of the open-loop transfer function. Included in the open-loop transfer function is the acoustical path by which sound emanating from the earmold vent returns to the microphone (i.e., the feedback path). Reported herein are two different mathematical procedures for simulating transfer functions of the feedback path of an eyeglass-type hearing aid. In one procedure the vent exit was modeled as a point source of sound located on a flat plane, while it was treated as a point source on a sphere in the other. Results of laboratory experiments indicate that the mathematical models accurately predict those acoustic phenomena for which they were intended: point sources on plane and spherical baffles. Results of manikin experiments showed both models to be less accurate for simulating the feedback path around the human head. The maximum difference between experiment and theory was 6 dB at one frequency. Surprisingly, the flat-baffle model produced better agreement with experimental results than did the sphere model.     

光学助降器模拟光源设计

舰载机降落着舰时,有时需要用到光学着舰助降系统。本光源模拟器采用卤钨灯光源,紧贴卤钨灯光源加入光学均匀器,然后依据光源的出射光束分布情况加入光束整形锥腔,再依据光源视场角的要求,在一定距离处加入柱面阵列光学系统,在光学柱面阵列系统后加入光学滤光器和光学衰减器就形成了光学助降光源模拟器。本光源模拟器具备红、绿、黄三色光谱,水平视场达到30,竖直视场达到20,光强大于10 000 cd,光源均匀性优于8％,光强可调。     

A technique for simulating the amplifier-to-eardrum transfer function of an in situ hearing aid

There are numerous articles wherein mathematical models of various parts of an in situ hearing aid have been reported. Such parts include, for example, the microphone, receiver, cylindrical tubes carrying sound to the eardrum and out through the earmold vent, and the external path from the vent back to the microphone. This article extends these earlier works to include the hearing-aid amplifier. In particular, a mathematical technique for characterizing the amplifier in combination with the receiver is reported. Cascade parameters of a two-port model of one particular amplifier/receiver combination are obtained by this method. The cascade-parameter data and the method of obtaining this data are verified by two different experimental procedures. One procedure involves both computing and measuring the input driving-point impedance of the amplifier/receiver combination. In the second procedure, the amplifier-to-eardrum transfer function of a hearing aid incorporating this same amplifier/receiver combination and mounted on an artificial ear is both computed and measured. Experimental and computed values of this transfer function for three different earmold geometries are in reasonably close agreement. The amplifier/receiver model reported herein will be used in future studies of acoustic feedback in hearing aids.     

Entanglement test on a microscopic-macroscopic system

A macrostate consisting of N approximately 3.5x10{4} photons in a quantum superposition and entangled with a far apart single-photon state (microstate) is generated. Precisely, an entangled photon pair is created by a nonlinear optical process; then one photon of the pair is injected into an optical parametric amplifier operating for any input polarization state, i.e., into a phase-covariant cloning machine. Such transformation establishes a connection between the single photon and the multiparticle fields. We then demonstrate the nonseparability of the bipartite system by adopting a local filtering technique within a positive operator valued measurement.     

An optimized Bell test in a dynamical system

The best realization of a Bell test depends on parameters linked to experimental settings. We report, for a class of two-qubit states, some optimized parameters that are useful to perform an optimized Bell test in a dynamical context. The time evolution of these optimized parameters, that present finite jumps, is investigated for two qubits in separated cavities.     

A computer simulation of hearing aid response and the effects of ear canal size

The response of a hearing aid is affected by many factors which include the head and outer ear, the microphone, amplifier, and receiver used in the hearing aid, the properties of the ear canal and the eardrum, and acoustic feedback through the vent. This article presents a computer simulation of an in-the-ear (ITE) hearing aid that includes all of the above factors. The simulation predicts the pressure at the eardrum for a frontal free-field sound source. The computer model was then used to determine the effects on the hearing aid response due to variations in the size of the ear canal. The simulation indicates that, for an unvented hearing aid, changes in the size of the ear canal shift the overall sound-pressure level at the eardrum but have only small effects on the shape of the frequency response. The situation is more complicated when a vent is present, however, since changes in the size of the ear canal that cause apparently small perturbations in the acoustic feedback signal may, nonetheless, have large effects on the overall system response.     

Adaptive environment classification system for hearing aids

An adaptive sound classification framework is proposed for hearing aid applications. The long-term goal is to develop fully trainable instruments in which both the acoustical environments encountered in daily life and the hearing aid settings preferred by the user in each environmental class could be learned. Two adaptive classifiers are described, one based on minimum distance clustering and one on Bayesian classification. Through unsupervised learning, the adaptive systems allow classes to split or merge based on changes in the ongoing acoustical environments. Performance was evaluated using real-world sounds from a wide range of acoustical environments. The systems were first initialized using two classes, speech and noise, followed by a testing period when a third class, music, was introduced. Both systems were successful in detecting the presence of an additional class and estimating its underlying parameters, reaching a testing accuracy close to the target rates obtained from best-case scenarios derived from non-adaptive supervised versions of the classifiers (about 3% lower performance). The adaptive Bayesian classifier resulted in a 4% higher overall accuracy upon splitting adaptation than the minimum distance classifier. Merging accuracy was found to be the same in the two systems and within 1%-2% of the best-case supervised versions.     

一种FlexRay总线通讯测试系统设计

FlexRay总线作为一种高速、灵活、可靠的串行通信总线,已逐渐成为汽车网络的主流。为了验证该总线技术在航天技术应用上的可行性,设计了一个基于TMS570LS3137安全控制器和PXI8517集成测试板卡的FlexRay总线网络测试系统。从硬件设计、平台搭建、协议规范、节点设计、网络定义等方面介绍了测试系统的设计方案,实现了网络中各个节点之间的同步和数据收发,最终实现了FlexRay网络之间的通信。试验结果表明了FlexRay网络通信系统的可行性和可靠性,并以其高速和强冗余特点成功应用于国家863项目机电作动系统研制项目中。     

In-flight test of a continuous laser remote sensing system

An airborne laser system for remote sensing has been developed and tested at flight altitudes from 500 to 1200 m. The system is based on two continuous-wave CO₂ lasers (output 3 W), tunable to different wavelengths, and detects the laser radiation scattered back from earth by heterodyne reception. The feasibility of the heterodyne reception technique under aircraft environment conditions was demonstrated, and man-made SF₆ clouds and different types of uncovered minerals were traced. The data obtained indicate that the measuring range of the system can be increased to satellite distance by using a larger telescope and a more powerful laser and by reducing the electrical bandwidth. The word was supported by the German Ministry of Science and Technology, represented by DFVLR-BPT under contract number 01 TL 026-AK/RT/WRT 2074.     

Perceptual adaptation by normally hearing listeners to a simulated "hole" in hearing

Simulations of cochlear implants have demonstrated that the deleterious effects of a frequency misalignment between analysis bands and characteristic frequencies at basally shifted simulated electrode locations are significantly reduced with training. However, a distortion of frequency-to-place mapping may also arise due to a region of dysfunctional neurons that creates a "hole" in the tonotopic representation. This study simulated a 10 mm hole in the mid-frequency region. Noise-band processors were created with six output bands (three apical and three basal to the hole). The spectral information that would have been represented in the hole was either dropped or reassigned to bands on either side. Such reassignment preserves information but warps the place code, which may in itself impair performance. Normally hearing subjects received three hours of training in two reassignment conditions. Speech recognition improved considerably with training. Scores were much lower in a baseline (untrained) condition where information from the hole region was dropped. A second group of subjects trained in this dropped condition did show some improvement; however, scores after training were significantly lower than in the reassignment conditions. These results are consistent with the view that speech processors should present the most informative frequency range irrespective of frequency misalignment.