共查询到20条相似文献,搜索用时 140 毫秒
1.
2.
3.
针对多说话人跟踪的非线性系统模型,提出了一种基于数值积分卡尔曼-概率假设密度滤波的多说话人跟踪方法。该方法采用麦克风阵列的时间延迟估计作为观测数据,利用具有三次代数精度的球面-径向数值积分准则计算非线性系统贝叶斯滤波器中的多维积分,通过数值积分卡尔曼滤波和概率假设密度滤波对后验多说话人状态的一阶统计量进行估计,并通过递推更新得到说话人状态信息,实现非线性高斯系统的多说话人跟踪。该方法无需求解非线性系统函数的雅克比矩阵,且计算量较小。仿真实验分析了检测概率、虚警点数目、采样周期、信噪比以及混响时间变化时跟踪算法的性能。实验结果表明,该方法降低了系统模型非线性对滤波算法的影响,增强了跟踪算法的鲁棒性,提高了说话人状态和数目的估计精度。 相似文献
4.
麦克风阵列声源定位可为在复杂环境下的说话人的空间位置估计提供有效的解决方案.而传统的应用于雷达,声呐系统领域的阵列信号处理理论已趋于完美,很多应用于阵列信号处理的算法加以修改就可以用来进行麦克风阵列的声源定位.以阵列信号处理中的经典算法MUSIC(Multiple Signal Classification)算法为原型... 相似文献
5.
6.
7.
一种基于自适应阵列天线的波束赋形算法 总被引:1,自引:0,他引:1
自适应阵列天线中的数字波束赋形(DBF)技术是智能天线数字信号处理部分的核心.提出了一种可用于自适应阵列波束赋形的SMI-LMS算法--由SMI(采样协方差矩阵求逆)算法决定LMS(最小均方)算法的初始权向量.该算法充分结合了SMI算法收敛速度快和LMS算法稳态误差小的优点,能在较强干扰环境下,确保权向量的快速收敛和跟踪速度.与传统的LMS算法相比,SMI-LMS算法具有良好的收敛性能、较快的跟踪速度和较小的输出误差,并可以有效改善自适应方向图的副瓣性能.仿真结果验证了该结论. 相似文献
8.
基于双麦克风的2维平面定位算法 总被引:1,自引:0,他引:1
基于麦克风阵列的声源定位技术受到了越来越多的关注。在视频会议、助听器、免提电话系统中,声源定位被用于检测说话人的位置信息来自动调节摄像头,或者形成波束。在各种声源定位方法中,基于到达时间差(time delay of arrival,TDOA)估计的双步定位算法是普遍采用的一种行之有效的方法。Birchfield从能量的角度出发提出了一种基于双耳电平差(interaural level difference,ILD)的双步定位算法,它通过检测多个麦克风对所接收到的信号能量比来确定声源的位置。然而,所有的这些方法如果要确定出声源在二维平面内的位置坐标,都至少需要三个麦克风。针对这一问题,本文提出了一种基于双麦克风的二维平面定位算法,类似于人的双耳定位原理,我们通过同时估计声源到达两个麦克风的能量比和时延信息,来达到定位的目的,而进一步推导出的闭合解可以用于实时地跟踪运动声源。最后的仿真结果证明了这一算法在一般的混响条件下都可以获得好的结果,然而它减小了阵列的尺寸,这对于体积受限的通信设备来说具有极大的吸引力。 相似文献
9.
10.
提出一种可用于说话人识别的自适应RBFN阵列。RBF网设计的核心在于确定网络中心的数目及位置,该自适应算法有效地融合了IOC与ROLS算法的优点,不仅能动态调节RBF网的隐节点数,还能使网络的数据中心自适应变化,很好地优化了网络的结构。用与文本无关的闭集说话人识别系统对该算法进行了验证,实验结果表明,该方法与传统的RBF算法相比,自适应RBF网具有较好的鲁棒性以及精简的网络结构等优点。 相似文献
11.
12.
13.
14.
Maj JB Royackers L Moonen M Wouters J 《IEEE transactions on bio-medical engineering》2005,52(9):1563-1573
In this paper, the first real-time implementation and perceptual evaluation of a singular value decomposition (SVD)-based optimal filtering technique for noise reduction in a dual microphone behind-the-ear (BTE) hearing aid is presented. This evaluation was carried out for a speech weighted noise and multitalker babble, for single and multiple jammer sound source scenarios. Two basic microphone configurations in the hearing aid were used. The SVD-based optimal filtering technique was compared against an adaptive beamformer, which is known to give significant improvements in speech intelligibility in noisy environment. The optimal filtering technique works without assumptions about a speaker position, unlike the two-stage adaptive beamformer. However this strategy needs a robust voice activity detector (VAD). A method to improve the performance of the VAD was presented and evaluated physically. By connecting the VAD to the output of the noise reduction algorithms, a good discrimination between the speech-and-noise periods and the noise-only periods of the signals was obtained. The perceptual experiments demonstrated that the SVD-based optimal filtering technique could perform as well as the adaptive beamformer in a single noise source scenario, i.e., the ideal scenario for the latter technique, and could outperform the adaptive beamformer in multiple noise source scenarios. 相似文献
15.
基于相干性滤波器的广义旁瓣抵消器麦克风小阵列语音增强方法 总被引:1,自引:0,他引:1
为了克服传统麦克风小阵列语音增强算法噪音抑制能力有限的问题,该文提出一种基于相干性滤波器的广义旁瓣抵消器语音增强算法, 该算法基于动态平滑系数噪声谱估计来获得相干性滤波器,分别对每个阵元接收到的信号进行滤波用以抑制包括混响等噪声信号的干扰,并把滤波后的信号作为输入信号,使用基于小阵列的广义旁瓣抵消器波束形成算法抑制残余噪声信号的干扰。模拟和实际试验表明,该文提出的算法明显优于单独使用小阵列波束形成算法和相干性滤波器算法。 相似文献
16.
为了抑制小型语音通信设备中的方向性噪声干扰问题,提出了一种结合差分阵列与幅度谱减的双麦语音增强算法。该算法首先利用一阶差分阵列技术,对两麦克风采集到的带噪语音信号进行处理,得到语音通道信号和噪声通道信号。接着利用差分阵列处理后的两通道信号对语音通道信号的信噪比进行估计。最后利用幅度谱减法对语音通道信号中残留噪声进行消除。针对语音通道信号的信噪比估计,本文给出了两种新奇的计算方法。仿真实验表明,该算法有效的抑制了方向噪声,改善了语音的质量,去噪效果及语音质量均优于对比算法。 相似文献
17.
Bahram Kouhi-Jelehkaran Hamidreza Bakhshi Farbod Razzazi 《AEUE-International Journal of Electronics and Communications》2010,64(12):1167-1172
Because of noise and reverberation, accuracy of speech recognition systems decreases when the distance between talker and microphone increases. By the using of microphone arrays and appropriate filtering of received signals, the accuracy of recognizer can be increased. Many different methods for using microphone arrays have been proposed that can be classified into two main approaches: systems that perform in two independent stages of array processing and then recognition and systems that use array processing to generate a sequence of features which maximize the likelihood of generating the correct hypothesis in recognition phase. Following second approach, in this paper a new method for microphone array processing is proposed in which the parameters of array processing are adjusted in calibration phase based on phones used in language and maximum likelihood method. Optimized filter parameters are stored and used during recognition phase. A new modified Viterbi algorithm using optimal phone-based filter parameters is used for recognition phase. The proposed algorithm is analytically formulated and Persian language is used to find any improvement in speech recognition accuracy compared with results of delay and sum and utterance-based filter and sum algorithms. The results show 12.2% improvement in accuracy compared to utterance-based algorithm. 相似文献
18.
Optimal Design of Nearfield Wideband Beamformers Robust Against Errors in Microphone Array Characteristics 总被引:1,自引:0,他引:1
Huawei Chen Wee Ser Zhu Liang Yu 《IEEE transactions on circuits and systems. I, Regular papers》2007,54(9):1950-1959
Nearfield wideband beamformers for microphone arrays have wide applications, such as hands-free telephony, hearing aids, and speech input devices to computers. The existing design approaches for nearfield wideband beamformers are highly sensitive to errors in microphone array characteristics, i.e., microphone gain, phase, and position errors, as well as sound speed errors. In this paper, a robust design approach for nearfield wideband beamformers for microphone arrays is proposed. The robust nearfield wideband beamformers are designed based on the minimax criterion with the worst case performance optimization. The design problems can be formulated as second-order cone programming and be solved efficiently using the well-established polynomial time interior-point methods. Several interesting properties of the robust nearfield wideband beamformers are derived. Numerical examples are given to demonstrate the efficacy of the proposed beamformers in the presence of errors in microphone array characteristics. 相似文献
19.
20.
Multidimensional Systems and Signal Processing - For the last few decades, speech enhancement based on microphone arrays has primarily utilized prior information about system models, e.g., array... 相似文献