多传感器自适应心肺声提取

Hong Wang, L. Wang
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引用次数: 17

摘要

连续监测心肺音对肺、心疾病患者的医学诊断和手术室危重情况的发现具有重要意义。为了获得定量和可靠的诊断和检测,心脏和呼吸听诊保留高清晰度的声音至关重要。临床声环境对心肺声采集提出了很大的挑战。与可以人为控制和降低噪音水平的声学实验室不同,手术室由于手术设备、通风机、谈话、警报等而非常嘈杂。这种噪音的不可预测性和广泛性使手术室成为一个非常困难的声学环境。从技术上讲,肺音和心音比环境噪声弱,并且其频带与噪声频率明显重叠。因此,高保真麦克风和传统的噪声滤波或消除技术无法提供太多帮助。此外,由于患者的生理状况、手术变量、手术类型和手术室环境的巨大差异,声音传输通道在患者之间和手术过程中都有很大差异。因此,为捕获声音传输通道的单个特征提供建模能力变得势在必行。本文提出了一种采用(1)嵌入式信号进行系统激励和识别的信号处理方法,(2)自适应信号处理方法。更新系统以补偿时间变化的算法,(3)提取所需信号的信号分离算法,以及(4)可调滤波器以减少噪声对目标信号分量的影响。这种方法可能比依赖于单个固定过滤器或基于统计的声音分离的算法提供更好的性能。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
Multi-sensor adaptive heart and lung sound extraction
Continuous monitoring of heart and lung sounds is of essential importance in medical diagnosis in patients with lung or heart diseases and detection of critical conditions in operating rooms. To obtain quantitative and reliable diagnosis and detection, it is critically important that cardiac and respiratory auscultation retains sounds of high clarity. Clinical acoustic environment imposes great challenges for heart and lung sound acquisition. Unlike acoustic labs in which noise levels can be artificially controlled and reduced, operating rooms are very noisy due to surgical devices, ventilation machines, conversations, alarms, etc. The unpredictable and broadband natures of such noises make operating rooms a very difficult acoustic environment. More technically, lung and heart sounds are weaker than environment noises, and have frequency bands which overlap significantly with noise frequencies. As a result, high fidelity microphones and traditional noise filtering or cancellation techniques cannot help much. Furthermore, due to large variations in patient physiological conditions, operating variables, surgical types, and operating room settings, sound transmission channels vary vastly from patient to patient and during the surgical process. Consequently, it becomes imperative to provide modeling capability for capturing individual characteristics of sound transmission channels. This paper presents a signal processing method that uses (1) an embedded signal for system excitation and identification, (2) an adaptive. algorithm for updating systems so that time variations can be compensated, (3) an signal separation algorithm to extract desired signals, and (4) an adjustable filter to reduce noise impact on the target signal components. This approach can potentially provide superior performance over algorithms that rely on individual fixed filters or statistical based sound separation.
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