DSP-implementations of speech coding for multielectrode cochlear implants and multiband loudness correction for digital hearing aids

N. Dillier, H. Bogli, T. Frohlich, M. Kompis
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Abstract

SUMMARY Hearing sensations can be restored for profoundly deaf patients via artificial electrical stimulation of the auditory nerve. Present electrode technology and electrophysiological constraints however allow at best a very crude and limited approximation of the normal neural excitation pattern. Signal processing for cochlear implants therefore is confronted with the problem of a severely restricted channel capacity and the necessity to select and encode a subset of the information contained in the sound signal reaching the listeners ear. With single chip digital signal processors (DSPs) incorporated in personal computers different speech coding strategies can be evaluated in relatively short laboratory experiments. In addition to the well known strategies realized with filters, amplifiers and logic circuits a DSP approach allows the implementation of much more complex algorithms such as nonlinear multiband loudness correction, speech feature contrast enhancement, adaptive noise reduction. Although many aspects of speech encoding can be efficiently studied using a laboratory digital signal processor it would be desirable to allow subjects more time for adjustment to a new coding strategy. Several days or weeks of habituation are sometimes required until a new mapping can be fully exploited. Thus for scientific as well as practical purposes the miniaturization of wearable DSPs will be of great importance. A cochlear implant digital speech processor (CIDSP) for the Nucleus 22-channel cochlear prosthesis has been implemented using a single chip digital signal processor (TMS320C25, Texas Instruments). For laboratory experiments the CIDSP is incorporated in a general purpose computer (PDP11/73) which provides interactive parameter control, graphical display of input/output and intermediate buffers and offline speech file processing facilities. In addition to the generation of stimulus parameters for the cochlear implant an acoustic signal based on a perceptive model of auditory nerve stimulation is output simultaneously. For field studies and as a take-home device for patients a wearable battery-operated unit has been built. Advantages of a DSP-implementation of speech encoding algorithms as opposed to offline prepared tcst lists are increased flexibility, controlled, reproducible and fast modifications of processing parameters, use of running speech for training and familiarization. Disadvantages are the more complex programming and numerical problems with integer arithmetic.
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多电极人工耳蜗植入和数字助听器多波段响度校正语音编码的dsp实现
通过人工电刺激听神经,可以恢复重度耳聋患者的听觉。然而,目前的电极技术和电生理的限制至多允许一个非常粗糙和有限的近似正常的神经兴奋模式。因此,人工耳蜗的信号处理面临着通道容量严重受限的问题,并且需要选择和编码到达听者耳朵的声音信号中包含的信息子集。随着单芯片数字信号处理器(dsp)集成到个人计算机中,不同的语音编码策略可以在相对较短的实验室实验中进行评估。除了用滤波器、放大器和逻辑电路实现的众所周知的策略外,DSP方法还允许实现更复杂的算法,如非线性多频带响度校正、语音特征对比度增强、自适应降噪。虽然使用实验室数字信号处理器可以有效地研究语音编码的许多方面,但希望让受试者有更多的时间来适应新的编码策略。有时需要几天或几周的习惯,直到一个新的映射可以被充分利用。因此,对于科学和实用的目的,可穿戴dsp的小型化将是非常重要的。采用单芯片数字信号处理器(TMS320C25, Texas Instruments)实现了一种适用于Nucleus 22通道人工耳蜗的人工耳蜗数字语音处理器(CIDSP)。对于实验室实验,CIDSP集成在通用计算机(PDP11/73)中,该计算机提供交互式参数控制,输入/输出图形显示和中间缓冲区以及离线语音文件处理设施。除了产生人工耳蜗的刺激参数外,还同时输出基于听觉神经刺激感知模型的声信号。为了进行实地研究,以及作为病人的带回家设备,一种可穿戴的电池供电装置已经建成。语音编码算法的dsp实现与离线准备的tst列表相比,优点是增加了处理参数的灵活性,可控制,可重复和快速修改,使用运行语音进行训练和熟悉。缺点是更复杂的规划和整数运算的数值问题。
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An Overview of the MPEG/audio Compression Algorithm Acoustic Echo Cancellation for Stereophonic Teleconferencing A Single-Input Hearing Aid Based on the Auditory Perceptual Features to Improve Speech Intelligibility in Noise Models of Pitch Perception Perceptual Linear Predictive (PLP) Analysis-Resynthesis Technique
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