Adaptive feedback cancellation based on variable step-size affine projection for hearing aids

A new variable step-size(VSS) affine projection algorithm(APA)(VSS-APA) was proposed for adaptive feedback cancellation suitable for hearing aids. So, a nonlinear function between step-size and estimation error is established and automatically adjusted according to the change of the estimation error, which leads to low misalignment and fast convergence speed. Analysis of the proposed algorithm offers large capacities in converging to the objective system. Simulation shows that the proposed algorithm achieves lower misalignment and faster convergence speed compared to fixed step-size APA and conventional adaptive algorithms.     

Acoustic feedback cancellation based on weighted adaptive projection subgradient method in hearing aids

Acoustic feedback is an important factor that degrades the overall performance of hearing aids, and acoustic feedback cancellation has always been the research focus in the field of signal processing in hearing aids. The newly suggested adaptive projection subgradient method (APSM) for adaptive signal processing solves the problem of difficulty in finding the exact projection operator in the realization of affine projection by taking the subgradient projection hyperplane as the searching region for relaxed projection. This work applies APSM in the acoustic feedback cancellation system of hearing aids for the first time, and proposes a weighted adaptive projection subgradient method (WAPSM), which takes into consideration the exponential decay weight factor to incorporate the prior information of estimation system. The new method is compared with the traditional NLMS algorithm and APSM algorithm in simulation experiments. Incorporating the prior information of estimation system by setting the proper weighting matrix, WAPSM achieved notable improvements on the speed, stability and accuracy of the misalignment convergence. Numerical experiments demonstrate that the proposed algorithm is more robust for low SNR and real speech segment input than the traditional algorithms.     

Feedback cancellation in hearing aids: results from a computersimulation

Feedback cancellation in hearing aids involves estimating the feedback signal and subtracting it from the microphone input signal. The feedback-cancellation system described updates the estimated feedback path whenever changes are detected in the feedback behavior. When a change is detected, the normal hearing-aid processing is interrupted, a pseudorandom probe signal is injected into the system, and a set of filter coefficients is adjusted to give an estimate of the feedback path. The hearing aid is then returned to normal operation with the feedback-cancellation filter as part of the system. Two approaches are investigated for computing the filter coefficients: a least-mean square (LMS) adaptive filter and a Wiener filter. Test results are presented for a computer simulation of an in-the-ear (ITE) hearing aid. The simulation results indicate that more than 10 dB of cancellation can be obtained and that the Wiener filter is more effective in the presence of strong interference     

Adaptive null-forming scheme in digital hearing aids

We propose an effective adaptive null-forming scheme for two nearby microphones in endfire orientation that are used in digital hearing aids and in many other hearing devices. This adaptive null-forming scheme is mainly based on an adaptive combination of two fixed polar patterns that act to make the null of the combined polar pattern of the system output always be toward the direction of the noise. The adaptive combination of these two fixed polar patterns is accomplished by simply updating an adaptive gain following the output of the first polar pattern unit. The value of this gain is updated by minimizing the power of the system output, and related adaptive algorithms to update this gain are also given. We have implemented this proposed system on the basis of a programmable DSP chip and performed various tests. Theoretical analyses and testing results demonstrated the effectiveness of the proposed system and the accuracy of its implementation     

A microphone array for hearing aids

A directional acoustic receiving system is a form of a necklace including an array of two or more microphones mounted on a housing supported on the chest of a user by a conducting loop encircling the user's neck. Signal processing electronics contained in the same housing receive and combine the microphone signals in such a manner as to provide an amplified output signal which emphasizes sounds of interest arriving in a direction forward of the user. The amplified output signal drives the supporting conducting loop to produce a representative magnetic field. An electroacoustic transducer including a magnetic field pick up coil for receiving the magnetic field is mounted in or on the user's ear and generates an acoustic signal representative of the sounds of interest. The microphone output signals are weighted (scaled) and combined to achieve desired spatial directivity responses. The weighting coefficients are determined by an optimization process. By bandpass filtering the weighted microphone signals, with a set of filters covering the audio frequency range, and summing the filtered signals, a receiving microphone array with a small aperture size is caused to have a directivity pattern that is essentially uniform over frequency in two or three dimensions. This method enables the design of highly-directive-hearing instruments which are comfortable, inconspicuous, and convenient to use. The array provides the user with a dramatic improvement in speech perception over existing hearing aid designs, particularly in the presence of background noise, reverberation, and feedback     

A silicon potentiometer for hearing aids

An angledetector with a digital output is described. The component is meant as an alternative to the traditional slide potentiometer used as volume control in many hearing aid applications. The component is based on the use of magnetic field sensitive MOSFET's (MAGFET's) detecting the position of a tiny bar magnet placed above a silicon chip. Because of the galvanic separation between the anglesetting bar magnet and the electrical circuit, this component is insensitive to the rather hostile environment hearing aids are exposed to. The lifetime of the component is thereby increased significantly. The electrical circuit contains a switched current A/D-D/A conversion system for offset compensating the MAGFET's and for converting the MAGFET signal currents into a digital output proportional to the input angle. The system can operate with a supply voltage down to 2.3 V. The average current consumption is 1.5 A. The peak current is close to 160 A. The system operates correctly within the clock frequency range of 5 Hz to 25 kHz. It is implemented using a commercially available 1.5 m CMOS process.     

一种用于数字助听器的次毫瓦级语音处理平台

We present a novel audio-processing platform, FlexEngine, which is composed of a 24-bit applicationspecific instruction-set processor （ASIP） and five dedicated accelerators. Acceleration instructions, compact instructions and repeat instruction are added into the ASIP＇s instruction set to deal with some core tasks of hearing aid algorithms. The five configurable accelerators are used to execute several of the most common functions of hearing aids. Moreover, several low power strategies, such as clock gating, data isolation, memory partition, bypass mode, sleep mode, are also applied in this platform for power reduction. The proposed platform is implemented in CMOS 130 nm technology, and test results show that power consumption of FlexEngine is 0.863 mW with the clock frequency of 8 MHz at Vdd = 1.0 V.     

一种适用于助听器的低失调自适应指向性算法

在噪声环境中助听器的性能会受到严重影响.但当噪声与期望信号处在不同方向时,在助听器中使用指向性传声器系统能够有效地抑制噪声,使助听器的使用者受益.本文基于自适应LMS(最小均方)算法提出了一种适用于助听器的低失调自适应指向性算法,用以动态调整传声器系统中滤波器的系数,使指向性模式的灵敏度最低点朝向噪声源方向,达到降噪的目的.相比于现有的LMS算法,本文引入了一种后验信噪比并将与其相关的信噪比补偿因子引入自适应步长的更新过程,有效改善了语音信号存在时的失调情况.最后,本文通过仿真验证了本文算法对失调的改善作用.     

A novel microphone preamplifier for use in hearing aids

A novel design for a microphone preamplifier for application in hearing aids is presented. The amplifier operates at a supply voltage of 1-1.3 V, the current drain is 70µA. The maximum sound level allowed is more than 105 dB SPL, with a typical noise level of 28 dB SPL. Instead of the usual voltage sensing, current sensing of the microphone is used. The amplifier consists of a fully balanced charge-to-current amplifier with no external components required. A semicustom version of the design has been integrated in a standard BIMOS process.     

A PARCOR characterization of the ear for hearing aids

A possible characterization of the ear through the PARCOR algorithm is suggested. This leads to a digital filter heating aid that could be of assistance in compensating for partial hearing loss.     

LMS-LMF adaptive scheme for echo cancellation

The authors present a novel algorithm for echo cancellation. The algorithm consists of simultaneously applying the LMS algorithm to the near-end section of the echo canceller, and a controlled mixed LMS-LMF algorithm to the far-end section. This combination results in a substantial improvement in performance of the proposed scheme over the LMS and the LMF algorithms     

A fast configurable AGC for front-end of digital hearing aids

This paper presents a fast configurable automatic gain control (AGC) with strong focus on fast acting control and low power consumption. This AGC includes two paths, main amplification path and gain adjusting path. Using the gain adjusting path through an extra amplifier provides a way for tracking and comparing the input signal with four adjusted thresholds to be judged for selecting the appropriate gain value for main amplification path. This mechanism of gain control is done by reorganization of input level and changing the resistance of feedback in main amplification path to generate smooth variation gain, without any interruption or delay in signal flow through the variable gain amplifier. Moreover, in order to protect the user from intense transients in variations of the input signal level, output level of variable gain amplifier is directly monitored using optimum threshold to reduce the overall gain using feedback control mechanism. The minimum power is consumed by gain adjusting path has almost no considerable on power consumption, it greatly improves hearing quality. Meanwhile, using a large size PMOS differential pair at the input improved the noise performance. Proposed AGC designed and simulated in TSMC 130-nm CMOS process. The post layout simulation results the maximal SNR is 84.6 dB in 100 Hz–19.6 kHz band-width and the total consumption power of this AGC is 78 μW at 1 V supply voltage. In addition, its gain is varied smoothly between 20 to 57 dB. Achieved results demonstrate that designed AGC meet the requirement of analog front end of hearing aids.

     

A low-power transmission-gate-based 16-bit multiplier for digital hearing aids

The most widespread 16-bit multiplier architectures are compared in terms of area occupation, dissipated energy, and EDP (Energy-Delay Product) in view of low-power low-voltage signal processing for digital hearing aids and similar applications. Transistor-level simulations including back-annotated wire parasitics confirm that the propagation of glitches along uneven and re-convergent paths results in large unproductive node activity. Because of their shorter full-adder chains, Wallace-tree multipliers indeed dissipate less energy than the carry-save (CSM) and other traditional array multipliers (6.0 µW/MHz versus 10.9 µW/MHz and more for 0.25 µm CMOS technology at 0.75 V). By combining the Wallace-tree architecture with transmission gates (TGs), a new approach is proposed to improve the energy efficiency further (3.1 µW/MHz), beyond recently published low-power architectures. Besides the reduction of the overall capacitance, minimum-sized transmission gate full-adders act as RC-low-pass filters that attenuate undesired switching. Finally, minimum size TGs increase the V _dd to ground resistance, hence decreasing leakage dissipation (0.55 nW versus 0.84 nW in CSM and 0.94 nW in Wallace).     

Three-channel variable filter-bank for digital hearing aids

This study proposes a three-channel (3-channel) variable filter-bank (VFB) that consists of variable lowpass, variable bandpass and variable highpass digital filters. The three variable digital filters are obtained from a normalised analog prototype Chebyshev type-I lowpass filter using analog frequency transformations along with a modified bilinear transformation. Since both the magnitudes (gains) and band edge frequencies of the three variable digital filters are independently adjustable, the 3-channel VFB is considerably flexible, and can be successfully applied for compensating various hearing loss patterns in digital hearing aids. Various audiograms have been used to verify that high-accuracy fittings can be achieved with low order variable filters. Moreover, the authors reveal and theoretically prove the numerator coefficient-symmetries of the variable lowpass, variable bandpass and variable highpass filters, and show that each variable filter requires only one multiplication for its numerator filtering operations, so the total number of multiplications can be significantly reduced. More specifically, only 11 multiplications and 14 additions are required in the whole 3-channel VFB. Therefore the 3-channel VFB has extremely simple structure and high tuning flexibility for hearing aids.     

A computationally efficient nonuniform FIR digital filter bank for hearing aids

A computationally efficient nonuniform digital FIR filter bank is proposed for hearing aid applications. The eight nonuniform spaced subbands are formed with the help of frequency-response masking technique. Two half-band finite-impulse response (FIR) filters are employed as prototypes resulting in significant improvements in the computational efficiency. We show, by means of example, that an eight-band nonuniform FIR filter bank with stopband attenuation of 80 dB can be implemented with 15 multipliers. The performance of the filter bank is enhanced by optimizing the gains for each subband. The tests on various hearing loss cases suggest that the proposed filter achieves reasonable good matching between audiograms and magnitude responses of the filter bank at very low computational cost.     

A novel ICI self‐cancellation scheme for OFDM systems

We propose a novel intercarrier interference (ICI) self‐cancellation scheme for orthogonal frequency division multiplexing (OFDM) systems. The symmetric scheme is the best among all ICI self‐cancellation scheme in the literature. Its coefficient pair is (1, ? 1), and the loading subcarriers are the k_th and N?k ? 1_th subcarriers, where N is the number of subcarriers. We propose to modify the symmetric scheme and change the coefficient pair from (1, ? 1) to (1, ?µ) where µis between 0 and 1. The proposed modified symmetric scheme has better carrier‐to‐interference‐ratio (CIR) than all previous ICI self‐cancellation schemes by at least 1.7 dB when the normalized frequency offset is 0.5. Copyright © 2011 John Wiley & Sons, Ltd.     

A new ghost cancellation system for Korean GCR

This paper describes a new ghost cancellation system built-in NTSC television based on the Korean ghost cancellation reference (KGCR) signal. The system has a highly integrated transversal filter, an unique circuit for control of the system and a high performance algorithm for calculating the filter coefficients. Laboratory and field test results confirm that the system is effective in canceling several combination of ghosts, which exist in real situations     

An inter-lighting interference cancellation scheme for MISO-VLC systems

In this paper, we propose an inter-lighting interference cancellation (ILIC) scheme to reduce the interference between adjacent light-emitting diodes (LEDs) and enhance the transmission capacity of multiple-input–single-output (MISO)-visible light communication (VLC) systems. In indoor environments, multiple LEDs have normally been used as lighting sources, allowing the design of MISO-VLC systems. To enhance the transmission capacity, different data should be simultaneously transmitted from each LED; however, that can lead to interference between adjacent LEDs. In that case, relatively low-received power signals are subjected to large interference because wireless optical systems generally use intensity modulation and direct detection. Thus, only the signal with the highest received power can be detected, while the other received signals cannot be detected. To solve this problem, we propose the ILIC scheme for MISO-VLC systems. The proposed scheme preferentially detects the highest received power signal, and this signal is referred as interference signal by an interference component generator. Then, relatively low-received power signal can be detected by cancelling the interference signal from the total received signals. Therefore, the performance of the proposed scheme can improve the total average bit error rate and throughput of a MISO-VLC system.     

一种用于助听器场景分类的对数能量特征

为降低助听器场景分类算法中特征提取的复杂度,提高识别准确率,给出了一种改进的对数能量特征参数。以基于最小距离聚类的场景分类方法为例,验证了特征参数的有效性。在其他条件相同时,比较了使用改进的对数能量、普通倒谱系数(CCs)和梅尔频率倒谱系数(MFCCs)为特征参数时的分类结果。实验结果表明,相对于其他2组特征参数,使用改进的对数能量时,分类结果在总体命中率以及均方误差意义上效果更优。另外,改进后的对数能量,相对于 MFCCs,计算量更小。     

Echo cancellation scheme for high speed digital subscriber lines

The authors discuss the implementation of a computationally efficient two stage echo cancellation system for high speed data transmission. In addition to a substantial reduction in hardware complexity, the two stage echo canceller offers the advantage of a speedier convergence compared to the conventional FIR filter