A combined multi-channel Wiener filter-based noise reduction and dynamic range compression in hearing aids

Noise reduction (NR) and dynamic range compression (DRC) are basic components in hearing aids, but generally these components are developed and evaluated independently of each other. Hearing aids typically use a serial concatenation of NR and DRC. However, the DRC in such a concatenation negatively affects the performance of the NR stage: the residual noise after NR receives more amplification compared to the speech, resulting in a signal-to-noise-ratio (SNR) degradation. The integration of NR and DRC has not received a lot of attention so far. In this paper, a multi-channel Wiener filter (MWF)-based approach is presented for speech and noise scenarios, where an MWF-based NR algorithm is combined with DRC. The proposed solution is based on modifying the MWF and the DRC to incorporate the conditional speech presence probability in order to avoid residual noise amplification. The goal is then to analyse any undesired interaction effects by means of objective measures. Experimental results indeed confirm that a serial concatenation of NR and DRC degrades the SNR improvement provided by the NR, whereas the combined approach proposed here shows less degradation of the SNR improvement at a low increase in distortion compared to a serial concatenation.     

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.     

Stochastic gradient-based implementation of spatially preprocessed speech distortion weighted multichannel Wiener filtering for noise reduction in hearing aids

Recently, a generalized noise reduction scheme has been proposed, called the Spatially Preprocessed, Speech Distortion Weighted, Multichannel Wiener Filter (SP-SDW-MWF). It encompasses the Generalized Sidelobe Canceller (GSC) and a multichannel Wiener filtering technique as extreme cases. Compared with the widely studied GSC with Quadratic Inequality Constraint (QIC-GSC), the SP-SDW-MWF achieves a better noise reduction performance for a given maximum speech distortion level. We develop a low-cost, stochastic gradient implementation of the SP-SDW-MWF. To speed up convergence and reduce computational complexity, the algorithm is implemented in the frequency domain. Experimental results with a behind-the-ear hearing aid show that the proposed frequency-domain stochastic gradient algorithm preserves the benefit of the exact SP-SDW-MWF over the QIC-GSC, while its computational cost is comparable to the least mean square-based scaled projection algorithm for QIC-GSC.     

Output SNR analysis of integrated active noise control and noise reduction in hearing aids under a single speech source scenario

This paper analyses the output signal-to-noise ratio for a standard noise reduction scheme based on the multichannel Wiener filter and for an integrated active noise control and noise reduction scheme based on the filtered-X multichannel Wiener filter, both applied in a hearing aid framework that includes the effects of signal leakage through an open fitting and secondary path effects. In previous work, integrating noise reduction and active noise control has been shown to allow to compensate for effects of signal leakage and secondary path effects. These experimental results are now verified theoretically. The output signal-to-noise ratios are derived under a single speech source scenario. Theoretical results are then compared to simulations for a single noise source scenario and a multiple noise sources scenario.     

SVD-based optimal filtering for noise reduction in dual microphone hearing aids: a real time implementation and perceptual evaluation

In this paper, the first real-time implementation and perceptual evaluation of a singular value decomposition (SVD)-based optimal filtering technique for noise reduction in a dual microphone behind-the-ear (BTE) hearing aid is presented. This evaluation was carried out for a speech weighted noise and multitalker babble, for single and multiple jammer sound source scenarios. Two basic microphone configurations in the hearing aid were used. The SVD-based optimal filtering technique was compared against an adaptive beamformer, which is known to give significant improvements in speech intelligibility in noisy environment. The optimal filtering technique works without assumptions about a speaker position, unlike the two-stage adaptive beamformer. However this strategy needs a robust voice activity detector (VAD). A method to improve the performance of the VAD was presented and evaluated physically. By connecting the VAD to the output of the noise reduction algorithms, a good discrimination between the speech-and-noise periods and the noise-only periods of the signals was obtained. The perceptual experiments demonstrated that the SVD-based optimal filtering technique could perform as well as the adaptive beamformer in a single noise source scenario, i.e., the ideal scenario for the latter technique, and could outperform the adaptive beamformer in multiple noise source scenarios.     

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.     

Speech understanding performance of cochlear implant subjects using time-frequency masking-based noise reduction

Cochlear implant (CI) recipients report severe degradation of speech understanding under noisy conditions. Most CI recipients typically can require about 10-25 dB higher signal-to-noise ratio than normal hearing (NH) listeners in order to achieve similar speech understanding performance. In recent years, significant emphasis has been put on binaural algorithms, which not only make use of the head shadow effect, but also have two or more microphone signals at their disposal to generate binaural inputs. Most of the CI recipients today are unilaterally implanted but they can still benefit from the binaural processing utilizing a contralateral microphone. The phase error filtering (PEF) algorithm tries to minimize the phase error variance utilizing a time-frequency mask for noise reduction. Potential improvement in speech intelligibility offered by the algorithm is evaluated with four different kinds of mask functions. The study reveals that the PEF algorithm which uses a contralateral microphone but unilateral presentation provides considerable improvement in intelligibility for both NH and CI subjects. Further, preference rating test suggests that CI subjects can tolerate higher levels of distortions than NH subjects, and therefore, more aggressive noise reduction for CI recipients is possible.     

多层次模极大值降噪算法研究

小波模极大值去噪方法具有很好的理论基础,却在应用上存在许多影响去噪性能的因素,如最优分解尺度选择、收缩阈值估计、模极大值线搜索及重构算法的效率和精度。在经典Mallat模极大值去噪算法的基础上,提出多层次模极大值降噪算法,设定分解尺度的最优选择范围,并利用改进的自适应阈值估计对模极大值序列进行预处理及利用多项式插值对模极大值序列进行快速重构。Maltab仿真结果表明多层次模极大值降噪算法具有良好的去噪性能,有效解决模极大值去噪方法在实际应用中面临的问题。     

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 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.     

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.     

Minimum variance time-frequency distribution kernels for signals inadditive noise

We derive the minimum variance time-frequency distribution kernel for signals in additive circular complex Gaussian white noise processes. It is shown that the kernel that minimizes the average variance and simultaneously satisfies the time-frequency constraints for the noise-only ease remains optimum in the presence of frequency-modulated (FM) signals     

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     

基于组合算法的语音降噪研究

在语音信号处理领域,子空间降噪算法和自适应算法是经常被用来处理信号的技术手段。分析和研究VSLMS自适应滤波算法和子空间变换的原理和方法[1-2],提出了一种新的组合方式,实现语音降噪算法的进一步改善。组合算法首先使用归一化的子空间算法对语音信号进行子空间层面分解,然后将分离出来的带噪信号作为自适应滤波算法的输入,利用处理因子的迭代,实现对带噪信号的增强处理,最终实现语音处理的优化。实验表明,组合算法的降噪效果,优于传统的使用单个算法进行降噪的方法。     

一种弹载线阵激光雷达距离像噪声的抑制算法

为了得到更加可靠的激光雷达距离像,充分考虑弹载激光雷达的特殊工况,提出一种基于扫描路径线的距离反常噪声抑制算法。首先,采用线形滤波窗口对原始距离像每一个像素进行距离反常多次判定,分别对当前像素是否为中值、与中值的差值是否大于阈值以及当前像素的绝对变化幅值进行分析;然后,在判定为距离反常值之后,在滤波窗口内进行二次中值滤波,前后相比较之后做出决断。实验结果表明:在RMES(均方根误差)的判定条件下,此算法在不同高度、不同坡形、不同角度的多组数据情况下,对距离像的距离反常噪声抑制效果,都要明显优于传统滤波窗口为3×3和5×5的中值滤波;并且算法处理速度较快,能够满足初步的应用需求。     

A new evolving-update-based feedback cancellation scheme for hearing aids

This paper proposes an efficient adaptive feedback canceller (AFC) for hearing aid, which provides satisfactory performance both in sparse and in dispersive conditions as well as can adapt according to the variations in the sparseness level of the feedback path for coloured signal as input. This is achieved by incorporating the measure of sparseness intensity and the variable step size to the memory-improved proportionate affine projection algorithm (MIPAPA), and hence, an improved MIPAPA (IMIPAPA) is proposed. Further, in order to reduce the computations incurred by the AFC, an evolving-update IMIPAPA (E-IMIPAPA) is introduced, employing an intermittent update of taps of the adaptive filter by simultaneously adjusting the update interval. The proposed E-IMIPAPA is applied to the two-microphone-based AFC. The results of simulation-based experiments show the effectiveness of the proposed algorithm as compared to the existing methods for feedback cancellation in hearing aid in terms of misalignment and added stable gain. The proposed AFC model is further extended to the multiple-microphone and single-speaker set-up.     

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.