Multidelay block Hartley domain adaptive filter

The multidelay block frequency domain (MDBFD) adaptive filter with the two-dimensional optimum block (TOB) algorithm are extended to the Hartley domain. The convergent behaviour of the MDBHD-TOB is the same as its frequency domain counterpart while computational complexity is reduced to offer simpler implementation. Because of its computational efficiency, the MDBHD-TOB adaptive filter is well suited to applications which demand the use of a long adaptive filter, as in acoustic echo cancellation     

一种自适应电话会议回波抵消系统

文中给出一种基于去相关最小均方(DLMS)算法和迭代最大长度序列相关(IMLC)算法的电话会议回声抵消系统。鉴于DLMS算法在远端会话期间具有好的工作性能，而IMLC算法在双端会话期间具有良好的工作效果，这种新的回声抵消系统在远端会话期间用DLMS算法估计回声路径，而在双端会话期间用IMLC算法估计回声路径。计算机仿真表明，这种新的回声抵消系统在远端会话和双端会话情况下均能提供较好的回声路径估计。     

一种条件自适应滤波算法及其在回声消除中的应用

对回声消除问题进行分析后,提出了一种条件自适应滤波算法。新算法并不要求自适应滤波器达到最优值,而是在满足一定条件时才进行系数更新。这种算法可以用在回声消除领域,很好地解决信道变化和双端发声的区分问题。文章还对这种条件自适应滤波算法的收敛性能做了严格分析,并且给出了和分析结果相吻合的计算机仿真结果。     

Optimum filter banks for signal decomposition and its applicationin adaptive echo cancellation

The application of multirate filter banks in echo cancellation is investigated. The multiresolution algorithm is used to decompose the received sampling sequence into a number of components, and then, an adaptive algorithm is applied to cancel the echo in the received signal. In this paper, the performance of this method is discussed, from which optimal conditions for echo cancellation are established for the design of wavelet packet multiresolution decomposition. An efficient algorithm for designing such a set of optimal discrete filter banks is developed. The cases of optimal in-band and adjacent-band adaptive filtering are examined. Experimental results showed that the use of optimally designed multiresolution filter banks coupled with in-band or adjacent-band adaptive filtering is much more effective than the employment of commonly used wavelet filter banks. Furthermore, the use of the adjacent-band adaptive filtering algorithm has superior performance compared with that of the in-band filtering     

A Nonlinear Acoustic Echo Canceller Using Sigmoid Transform in Conjunction With RLS Algorithm

Nonlinearity of amplifiers and/or loudspeakers gives rise to nonlinear echo in acoustic systems, which seriously degrades the performance of speech and audio communications. Many nonlinear acoustic echo cancellation (AEC) methods have been proposed. In this paper, a simple yet efficient nonlinear echo cancellation scheme is presented by using an adaptable sigmoid function in conjunction with a conventional transversal adaptive filter. The new scheme uses the least mean square (LMS) algorithm to update the parameters of sigmoid function and the recursive least square (RLS) algorithm to determine the coefficient vector of the transversal filter. The proposed AEC is proved to be convergent under some mild assumptions. Computer simulations show that the proposed scheme gives a superior echo cancellation performance over the well known Volterra filter approach when the echo path suffers from the saturation-type nonlinear distortion. More importantly, the new AEC has a much lower computational complexity than the Volterra-filter-based method.      

Nonlinear loudspeaker compensation for hands free acoustic echocancellation

Loudspeaker nonlinearity at high volumes limits the achievable echo cancellation performance in linear acoustic echo cancellers. A new nonlinear adaptive filter for improving the echo cancellation performance at high volumes for hands free telephones is proposed. Experimental measurements show that an echo cancellation improvement of >8 dB can be obtained at high volumes as compared to a linear adaptive filter     

The Wheatstone bridge-based analog adaptive filter with application in echo cancellation

In this paper, a new analog adaptive filter is introduced with application in adaptive echo cancellation namely, the Wheatstone bridge-based analog adaptive filter (WAAF). It is proved the WAAF is a variable weight analog IIR filter. IIR filter weights vary with gate-source voltage control of a MOSFET transistor in triode region. The best balance point control of the WAAF is achieved using least mean square (LMS) algorithm. It is proved that analog LMS algorithm converges faster than digital LMS adaptive filter. The superiority of the proposed WAAF is observed in the designing process, computational cost, convergence speed and real time operation. Also, experimental results show ability of the proposed WAAF in the hybrid circuit of the telephone echo cancellation.     

模拟家庭监护的非接触呼吸和心跳信号分离实验研究

该实验研究的目的是为了从雷达式生命参数监护系统的回波信号中分离出呼吸和心跳信号,提取呼吸、心跳的特征参数,为家庭监护及疾病预防提供依据。分析回波信号的特点,改进自适应滤波器的参考输入信号为呼吸信号的谐波组合。针对模拟家庭监护实验提出了基于LMS自适应谐波抵消算法进行自适应滤波。在模拟家庭监护的实验中可以有效地分离出呼吸和心跳信号。     

基于LMS算法自适应回波抵消器的Simulink仿真分析

讨论了回波抵消的原理,并利用Simulink平台建立了基于LMS算法自适应回波消除器的仿真模型,对系统进行了仿真演示,给出了较为详细的模型构建和仿真波形,并对显示结果进行了性能分析。仿真的结果证实了该系统的有效性,较好地反映了系统的动态工作过程,且失真较小。     

Enhancing echo cancellation via estimation of delay

The advent of packetized audio transmission, such as voice over IP (VoIP), has resulted in challenging requirements for echo cancellation technology. One key aspect of this technology is the need to characterize, quickly and accurately, the echo paths in the transmission media. Echo paths consist of a constant time delay with no echo signal and active regions in which the echo signal is present. When an adaptive filter echo cancellation algorithm is used, its performance can be greatly increased, and its complexity can be reduced if it is only applied to the active regions. This requires an algorithm to estimate the constant delay and locate the active regions. Traditionally, delay estimation has been based on direct application of cross-correlation. This method has poor performance because the input signals are highly correlated and has a high implementation cost because many cross-correlation lags have to be computed for longer time delays. The delay estimation addressed in this paper has two major advantages over the traditional methods. The first is that it has improved performance because the input signals are processed to have less correlation. The second is that the implementation cost is significantly reduced because fewer cross-correlation lags are computed, and an efficient method to estimate lags is created.     

多径回波信道条件下回波抵消技术研究

主要研究移动通信数字同频直放站的回拨抵消技术。通过对基于最小均方误差原理的自适应滤波器的收敛条件进行分析,发现即使在外信道为单径信道或只考虑多径信道的主径回波信道时,现有利用时延去相关的自适应滤波算法,在原理上存在着不能保证满足收敛条件的问题。针对这一问题,本文提出了将发送信号进行延时后再送入自适应估计器的改进措施,有条件地改善了回波抵消效果,并进一步提出了消除多径回波的新型技术原理。     

基于sigmoid函数的Volterra自适应有源噪声对消器

该文介绍了一种新颖的非线性自适应有源噪声对消器基于sigmoid函数的Volterra自适应有源噪声对消器,并采用输入信号和瞬时误差归一化的LMS自适应算法调整其系数。这种基于sigmoid函数的Volterra自适应有源噪声对消器具有参数少和便于实现的模快化结构等优点。仿真结果表明:这种基于sigmoid函数的Volterra自适应有源噪声对消系统具有良好的抗噪声性能。     

一种回声消除和噪声抑制SoC芯片

 提出了一种回声消除和噪声抑制算法,采用改进的自适应步长非线性滤波技术,用单芯片对该算法进行了实现.用180nm 3.3V/1.8V 6层金属混合信号CMOS工艺流片,可达70dB的声学回声消除性能,噪音消除达20dB,侧音消除达30dB.该芯片包含1个16位DSP、3个14位Σ-Δ ADC、2个16位Σ-Δ DAC、 以及内置ROM和RAM等,并集成有USB、UART、I²C和PCM等接口.测试结果表明该芯片具有全双工和远距离免提的功能,支持双路麦克风输入,技术规范符合G.165国际标准.该芯片功耗低,外围电路简单,自适应能力强,可广泛应用于蓝牙车载免提通信、GPS和即时通讯等领域.     

A Co-Design Methodology for Telecommunication Systems: A Case Study of an Acoustic Echo Canceller

The acoustic echo cancellation with large adaptive filters is a computationally intensive problem and needs real time cost effective solution. To deal with these challenges, designers have increasingly turned to mixed Hardware/Software (HW/SW) implementation of echo canceller algorithms. This paper presents a co-design methodology and environment for both hardware and software modules. We describe how High Level Synthesis (HLS) tools like GAUT and SYNDEX can be efficiently used for rapid prototyping of heterogeneous architecture based on DSP TMS320C40 and ASIC. The HW/SW interface synthesis task is especially discussed since it constitutes a key issue of the whole design. As an illustration, we present a mixed implementation of the GMDF alpha algorithm, an adaptive filter well suited to acoustic echo cancellation, on both ASIC and TMS320C40 DSP.     

通债中自适应回声对消的研究

简述和分析了双工通信中回声的产生,并根据回声电平实际的衰耗要求,着重研究了双工通信系统中回声对消问题。运用自适应滤波技术,回声信号可得到有效的对消。对采用功率倒置算法自适应回声对消技术的性能作了详细的研究,分析了影响其对消效果的因素,并指出了进一步的发展方向。     

Blind channel estimation and echo cancellation using chaotic codedsignals

A novel dynamic-based deconvolution (DBD) algorithm is presented that can be exploited in blind channel estimation as well as in telephony echo cancellation. Chaotic coded signals generated by the logistic map are employed while the channel is represented by an autoregressive model. The applicability is demonstrated in a speech transmission scenario using chaotic coded speech showing a modeling misadjustment improvement of 24 dB/100 iterations which is sixfold that obtained by the LMS for a 128 tap digital adaptive filter driven by ideal white Gaussian noise excitation     

声回声抵消系统中后处理滤波器的研究

提出一种回声抵消系统中残留回声的抑制方法,利用回声信号的估计值和包含残留回声的输出误差信号计算得到的后处理滤波器,将残留回声进一步消除。提出的后处理滤波器可以有效地与频域自适应算法融合,并非常适用于多通道处理。实验和仿真结果均证明了该后处理滤波器的有效性。     

基于DVB-T标准的数字电视回波对消技术的研究

以数字电视地面广播国家标准DVB-T为基础,提出了一种基于FPGA和DSP实现回波对消的方案.其原理是:首先该系统不发送数字电视信号,通过发射CAZAC序列进行信道初估计.由于信道在实际中会发生一定的变化,所以还需要用自适应算法进行信道特性的跟踪,其中DSP运用自适应算法保证对消信号接近回波信号.其次FPGA系统获得了回波信道的参数并生成逼近回波信号的对消信号.最后,将接收信号与对消信号相减完成对消.提出了对回波信道的初估计并利用变换域LMS算法进行信道跟踪的一种方案.最后仿真结果表明,该方法具有良好的回波对消性能.     

Adaptive Sampling Rate Correction for Acoustic Echo Control in Voice-Over-IP

Hands-free terminals for speech communication employ adaptive filters to reduce echoes resulting from the acoustic coupling between loudspeaker and microphone. When using a personal computer with commercial audio hardware for teleconferencing, a sampling frequency offset between the loudspeaker output D/A converter and the microphone input A/D converter often occurs. In this case, state-of-the-art echo cancellation algorithms fail to track the correct room impulse response. In this paper, we present a novel least mean square (LMS-type) adaptive algorithm to estimate the frequency offset and resynchronize the signals using arbitrary sampling rate conversion. In conjunction with a normalized LMS-type adaptive filter for room impulse response tracking, the proposed system widely removes the deteriorating effects of a frequency offset up to several Hz and restores the functionality of echo cancellation.     

移动通信中的回波抵消算法分析

移动通信领域中巳广泛采用自适应回波抵消技术，自适应回波抵消算法是该技术的核心，算法的优劣直接影响自适应回波抵消效果，同时它对降低硬件复杂度和产品成本也有特殊意义。文章对自适应回波抵消的NLMS算法、双端讲话（DT）的检测算法及后处理的NLP算法进行了分析和讨论。