针对外部强噪声环境下电子耳蜗语音质量受损、适应性差等问题,提出了基于谱减法和变步长最小均方误差(LMS)自适应滤波算法联合去噪的改进方法,并以该方法构建了一个电子耳蜗前端语音预处理系统。利用变步长LMS自适应滤波算法输出误差的...针对外部强噪声环境下电子耳蜗语音质量受损、适应性差等问题,提出了基于谱减法和变步长最小均方误差(LMS)自适应滤波算法联合去噪的改进方法,并以该方法构建了一个电子耳蜗前端语音预处理系统。利用变步长LMS自适应滤波算法输出误差的平方项来调节步长,采用步长值固定与变化相结合的方法,解决了自适应滤波算法收敛速度慢、稳态误差大的问题,适应性得到提高,提高了语音信号通信质量。该系统以TMS320VC5416和音频编解码芯片TLV320AIC23B为核心,通过多通道缓冲串口(McBSP)和串行外设接口(SPI)实现了语音数据的高速采集和实时处理。实验仿真和测试结果表明该算法消除噪声性能好,信噪比在低输入信噪比情况下提高约10 d B,语音质量感知评价(PESQ)分值也得到较大提高,能有效提高语音信号质量,且该系统性能稳定,能进一步提高耳蜗前端语音的清晰度和可懂度。展开更多
The problem of blind adaptive equalization of underwater single-input multiple-output (SIMO) acoustic channels was analyzed by using the linear prediction method.Minimum mean square error (MMSE) blind equalizers with ...The problem of blind adaptive equalization of underwater single-input multiple-output (SIMO) acoustic channels was analyzed by using the linear prediction method.Minimum mean square error (MMSE) blind equalizers with arbitrary delay were described on a basis of channel identification.Two methods for calculating linear MMSE equalizers were proposed.One was based on full channel identification and realized using RLS adaptive algorithms,and the other was based on the zero-delay MMSE equalizer and realized using LMS and RLS adaptive algorithms,respectively.Performance of the three proposed algorithms and comparison with two existing zero-forcing (ZF) equalization algorithms were investigated by simulations utilizing two underwater acoustic channels.The results show that the proposed algorithms are robust enough to channel order mismatch.They have almost the same performance as the corresponding ZF algorithms under a high signal-to-noise (SNR) ratio and better performance under a low SNR.展开更多
文摘针对外部强噪声环境下电子耳蜗语音质量受损、适应性差等问题,提出了基于谱减法和变步长最小均方误差(LMS)自适应滤波算法联合去噪的改进方法,并以该方法构建了一个电子耳蜗前端语音预处理系统。利用变步长LMS自适应滤波算法输出误差的平方项来调节步长,采用步长值固定与变化相结合的方法,解决了自适应滤波算法收敛速度慢、稳态误差大的问题,适应性得到提高,提高了语音信号通信质量。该系统以TMS320VC5416和音频编解码芯片TLV320AIC23B为核心,通过多通道缓冲串口(McBSP)和串行外设接口(SPI)实现了语音数据的高速采集和实时处理。实验仿真和测试结果表明该算法消除噪声性能好,信噪比在低输入信噪比情况下提高约10 d B,语音质量感知评价(PESQ)分值也得到较大提高,能有效提高语音信号质量,且该系统性能稳定,能进一步提高耳蜗前端语音的清晰度和可懂度。
基金Supported by the National Natural Science Foundation of China under Grant No.60372086the Foundation for the Author of National Excellent Doctoral Dissertation of China under Grant No.200753
文摘The problem of blind adaptive equalization of underwater single-input multiple-output (SIMO) acoustic channels was analyzed by using the linear prediction method.Minimum mean square error (MMSE) blind equalizers with arbitrary delay were described on a basis of channel identification.Two methods for calculating linear MMSE equalizers were proposed.One was based on full channel identification and realized using RLS adaptive algorithms,and the other was based on the zero-delay MMSE equalizer and realized using LMS and RLS adaptive algorithms,respectively.Performance of the three proposed algorithms and comparison with two existing zero-forcing (ZF) equalization algorithms were investigated by simulations utilizing two underwater acoustic channels.The results show that the proposed algorithms are robust enough to channel order mismatch.They have almost the same performance as the corresponding ZF algorithms under a high signal-to-noise (SNR) ratio and better performance under a low SNR.
