Open Hardware and Software System for Speech and Signal Processing

智能听力系统公司（IHS），计划开发一个开放式架构的硬件和软件系统 这将使研究人员能够进行涉及多通道声学输入和输出的广泛研究 进行心理物理学研究拟议的架构将利用30年的数字信号处理器 (DSP)基于IHS的硬件和软件开发经验，用于高精度同步 声刺激产生和数据采集。该系统将包括通用串行总线（USB） 控制器，用于将数据、DSP代码和控制指令引导到控制计算机和共享的存储器，以及从控制计算机和共享的存储器引导数据、DSP代码和控制指令 关于DSP该架构通过允许各种应用和 随时上传处理策略。将提供多个级别的用户可编程性， 为了满足各种技术能力的研究人员的需求，包括：1）信号生成和 使用MatLab或集成高级语音处理TF 32软件等程序进行采集 由威斯康星州-麦迪逊大学开发。2)用户开发的各种编程应用程序 语言使用提供的动态链接库（DLL）和可视化控制库（VCL）的实时 控制和显示功能。3)使用提供的源代码示例直接编程内部DSP 和文献. DSP将支持实时逐样本或逐块处理，以便 实现广泛的信号处理功能，包括滤波、自适应噪声消除、语音 增强、动态增益调整以及频率和幅度压缩等。期间 第一阶段将完成以下工作：1）一个基于DSP的硬件系统， 声学研究与输入放大器的麦克风和输出衰减器和扬声器驱动器将 开发2)目前的USB 2.0通信将改进到3.0。3)当前DSP用户开发工具包 将扩展到包括额外的语音处理例程与源代码的例子， 文献. 4)将为TF 32开发一个VCL接口，以便于将各种 语音处理例程到用户开发的应用程序。5)测试数据和验证程序 开发6)为系统验证和测试开发的信号和噪声生成模块。在阶段 II：1）将开发图形界面，以方便DSP代码的开发。2)一本教科书 将制定实例和项目。3)DSP到PC通信的改进 切实贯彻4)开发了无线和电池供电选项。

Intelligent Hearing Systems Corp. (IHS), plans to develop an open architecture hardware and software system that will allow researchers to conduct a wide range of studies involving multi-channel acoustic input and output for psychophysical research. The proposed architecture will capitalize on 30 years of digital signal processor (DSP) based hardware and software development experience at IHS used for highly precise synchronized acoustical stimulus generation and data acquisition. The system will include a Universal Serial Bus (USB) controller to direct data, DSP code and control instructions to and from a control computer and memory shared with the DSP. This architecture provides a wide range of flexibility by allowing various applications and processing strategies to be uploaded at any time. Multiple levels of user programmability will be provided in order to suit the needs of researchers of various technical capabilities including: 1) Signals generation and acquisition using programs such as MatLab or integrated advanced speech processing TF32 software developed at the University of Wisconsin - Madison. 2) User developed application in various programming languages using provided Dynamically Linked Library (DLL) and Visual Control Library (VCL) of real-time control and display functions. 3) Direct programming of internal DSP using provided source code examples and documentation. The DSP will support real-time sample-by-sample or block-by-block processing in order to implement a wide range of signal processing functions including filtering, adaptive noise cancellation, speech enhancement, dynamic gain adjustment, and frequency and amplitude compression among others. During Phase I the following work will be accomplished: 1) a DSP-based hardware system specifically designed for acoustical research with input amplifiers for microphones and output attenuators and speaker drivers will be developed. 2) Current USB 2.0 communications will be improved to 3.0. 3) Current DSP user development kit will be expanded to include additional speech processing routines with source code examples and documentation. 4) A VCL interface will be developed for TF32 in order to facilitate the incorporation of various speech processing routines into user developed applications. 5) Test data and validation procedures developed. 6) Signal and noise generation modules developed for system validation and testing. During Phase II: 1) A graphical interface will be developed in order to facilitate the development of DSP code. 2) A text book with examples and projects will be developed. 3) Improvements to the DSP-to-PC communications implemented. 4) A wireless and battery powered supply options developed.