McDermott H
Co-operative Research Center for Cochlear Implant, Speech, and Hearing Research, Department of Otolaryngology, The University of Melbourne, East Melbourne, Australia.
IEEE Trans Rehabil Eng. 1998 Mar;6(1):53-9. doi: 10.1109/86.662620.
A portable sound processor has been developed to facilitate research on advanced hearing aids. Because it is based on a digital signal processing integrated circuit (Motorola DSP56001), it can readily be programmed to execute novel algorithms. Furthermore, the parameters of these algorithms can be adjusted quickly and easily to suit the specific hearing characteristics of users. In the processor, microphone signals are digitized to a precision of 12 bits at a sampling rate of approximately 12 kHz for input to the DSP device. Subsequently, processed samples are delivered to the earphone by a novel, fully-digital class-D driver. This driver provides the advantages of a conventional class-D amplifier (high maximum output, low power consumption, low distortion) without some of the disadvantages (such as the need for precise analog circuitry). In addition, a cochlear implant driver is provided so that the processor is suitable for hearing-impaired people who use an implant and an acoustic hearing aid together. To reduce the computational demands on the DSP device, and therefore the power consumption, a running spectral analysis of incoming signals is provided by a custom-designed switched-capacitor integrated circuit incorporating 20 bandpass filters. The complete processor is pocket-sized and powered by batteries. An example is described of its use in providing frequency-shaped amplification for aid users with severe hearing impairment. Speech perception tests confirmed that the processor performed significantly better than the subjects' own hearing aids, probably because the digital filter provided a frequency response generally closer to the optimum for each user than the simpler analog aids.
一种便携式声音处理器已被开发出来，以促进对先进助听器的研究。由于它基于数字信号处理集成电路（摩托罗拉DSP56001），它可以很容易地被编程来执行新颖的算法。此外，这些算法的参数可以快速且轻松地调整，以适应用户的特定听力特征。在该处理器中，麦克风信号以大约12kHz的采样率被数字化到12位精度，以便输入到DSP设备。随后，经过处理的样本由一种新型的全数字D类驱动器传送到耳机。这种驱动器具有传统D类放大器的优点（高最大输出、低功耗、低失真），但没有一些缺点（例如对精确模拟电路的需求）。此外，还提供了一个人工耳蜗驱动器，以便该处理器适用于同时使用人工耳蜗和声学助听器的听力受损者。为了降低对DSP设备的计算需求，从而降低功耗，由一个包含20个带通滤波器的定制开关电容集成电路对输入信号进行实时频谱分析。整个处理器为口袋大小，由电池供电。描述了一个其用于为重度听力受损的助听器用户提供频率整形放大的例子。言语感知测试证实，该处理器的表现明显优于受试者自己的助听器，这可能是因为数字滤波器提供的频率响应通常比简单的模拟助听器更接近每个用户的最佳响应。
