通信系统中自适应回波抵消的一种新方法

语音分组传输系统的出现，对回波抵消技术提出了新的挑战。在回波抵消中一个关键的问题是需要快速有效地估计延迟时间。基于直接计算输入信号互相关函数的延迟估计方法，在实际应用中回波消除性能差，运算量大。为此提出了基于混合模型的延时估计方法，对输入信号进行带通滤波和降低抽样率抽样，使其具有白噪声特性，然后快速计算互相关函数的估计值。理论分析和实验结果证明新的延迟估计方法有很好的回波消除性能和较少的运算量，在实际通信系统中有较好的应用前景。     

WCDMA直放站回波对消方案设计

基于WCDMA标准提出了FPGA+DPS平台下实现在数字中频技术中抵消WCDMA直放站回波干扰信号的方案.首先发射与WCDMA信号不相关的伪随机序列CAZAC码作为训练序列进行信道初估计,然后将初估计得到的时延和幅度信息作为赋给LMS自适应滤波器的初值,生成逼近干扰信号的抵消信号,实现回波对消.使用MATLAB和SIMULINK联合仿真,证明该方法拥有良好的性能.     

Silencing echoes on the telephone network

Echo of a trasmitted signal can occur in the telephone network and is sometimes a disturbing phenomenon. Various methods of echo control have been used for over 50 years but recent introduction of transmission delays in excess of 300 ms, over satellite circuits, has resulted in a general reexamination of suppression methods. This led to development of the echo canceller. Adaptive echo cancellation is accomplished by automatically synthesizing a replica of the echo path response, convolving the incoming signal with the replica and subtracting the resulting echo estimate from the echo path output. Transmission is not thereby interrupted in any way and two-way information transmission proceeds normally with little or no echo. Detailed mathematical formulations for the convergence properties of such devices allow understanding of the effects of various hardware configurations and input signal properties. Echo cancellers may be applied to the network in a number of ways and the development of a VLSI canceller chip promises inexpensive implementation of these applications. Worldwide tests have shown the effectiveness of cancellation methods for satellite transmission.     

一种利用人耳听觉特性的双声道回波抵消改进算法

本文提出了一种利用人耳听觉特性的双声道回波抵消改进算法.在该系统中,利用人耳的听觉特性给输入信号加上一个小的扰动来改善输入信号的自相关特性和条件数,从而达到改善系统收敛特性的作用.本文讨论了在双声道回波抵消系统中遇到的输入信号强相关从而引起系统性能恶化的问题,提出了采用输入信号加扰来改善系统性能的算法.并且利用人耳听觉特性使这种扰动信号隐藏在语音信号中,使其对语音信号的干扰最小.从本文的仿真试验来看,该方法对双声道回波抵消系统的性能有一定的改善.     

基于小波变换的声学回波消除系统

声学回波消除技术一直是语音通信领域的研究热点。在声学回波消除系统中,通过估计回波路径中的固定时延区域来提高自适应滤波算法的收敛速度。提出了一种基于小波变换的固定时延估计算法以及基于小波变换的声学回波消除系统,解决传统时延估计算法在声学回波消除系统中估计误差大、抗干扰能力弱的问题。测试结果表明,算法稳健性、有效性等指标明显优于传统时延估计算法,基于小波变换的声学回波消除系统具有良好的消回波性能。     

IP电话长时延环境的回声抵消算法的研究

近年来,因特网技术快速发展,其业务类型已进入传统电信的实时话音通信.文中首先论述了VoIP(Voice Over IP)技术中的延迟和回声问题,并回顾了IP电话中基本的回声抵消算法,最后结合IP网络的长延迟特性给出了一种基于延迟估计的回声抵消优化算法.     

基于总体最小二乘的外辐射源雷达相消算法

提出一种基于总体最小二乘(TLS)的外辐射源雷达直达波相消算法,该算法首先利用TLS获取直达波的数目和时延,再基于既得的直达波信息构建直达波子空间,最后将回波向该空间投影实现直达波相消。与传统扩展相消算法相比,该算法所构建的直达波子空间阶数低,精确度高,在相关处理时间较短或直达波幅度时变的情况下,仍具有良好的直达波抑制性能。仿真结果证实了算法的有效性。     

Estimation de la date d’arrivée d’un écho sonar et choix d’une référence des dates

Time estimation of echoes in sonar techniques requires a time origin. The instant of arrival is related to a peculiar wavefront of the incoming echo and defines the propagation delay between sonar and target. Delay estimation by cross-correlation techniques are biased due to Doppler effect. Such a bias is deeply related to the representation and the model used to describe the emitted signal and the echo. For finite energy signals it is shown that there is a peculiar wavefront called « uncoupling wavefront », such that the delay estimation is unbiased for any Doppler effect values. Such a result induces a new definition for Doppler tolerance.     

Time delay estimation algorithm of narrow band internet of things based on inter-cell interference cancellation

Aiming at the characteristics of NB-IoT such as low power consumption,low cost and low sampling rate,an NB-IoT delay estimation algorithm based on inter-cell interference cancellation was proposed.To gradually eliminate the interaction between signals in each cell,an inter-cell iterative interference cancellation algorithm was considered.In each iteration,the idea of time-frequency cross-correlation overlapping was introduced on the basis of traditional correlation algorithms to break through the limitations of the sampling rate and further improve the accuracy of time delay estimation.At the same time,a first-arrival-path (FAP) searching algorithm based on noise threshold was used to suppress multipath effects.Through experimental simulation,the results show that the proposed algorithm can significantly improve the time delay estimation accuracy of NB-IoT on the basis of related algorithms.     

基于BLSTM神经网络的回声和噪声抑制算法

考虑到非线性回声和非平稳噪声对智能设备回声消除算法的影响，论文提出一种基于双向长短时记忆(Bidirectional Long Short-Term Memory，BLSTM)神经网络的回声和噪声抑制算法。该算法首先采用多目标预处理模型，同步估计出回声和噪声信号的幅度谱；然后将其作为回声和噪声抑制模型的输入特征，进而估计出目标语音信号的理想比例掩模；最后通过联合训练两个模型得到最优回声和噪声抑制模型。实验结果表明，在非线性回声和非平稳噪声的环境下，该算法均取得了较好的回声和噪声抑制效果，语音失真较小。      

Stationary echo canceling in velocity estimation by time-domain cross-correlation

The application of stationary echo canceling to ultrasonic estimation of blood velocities using time-domain cross-correlation is investigated. Expressions are derived that show the influence from the echo canceler on the signals that enter the cross-correlation estimator. It is demonstrated that the filtration results in a velocity-dependent degradation of the signal-to-noise ratio. An analytic expression is given for the degradation for a realistic pulse. The probability of correct detection at low signal-to-noise ratios is influenced by signal-to-noise ratio, transducer bandwidth, center frequency, number of samples in the range gate, and number of A-lines employed in the estimation. Quantitative results calculated by a simple simulation program are given for the variation in probability from these parameters. An index reflecting the reliability of the estimate at hand can be calculated from the actual cross-correlation estimate by a simple formula and used in rejecting poor estimates or in displaying the reliability of the velocity estimated.     

Adaptive Reference Echo Cancellation

The AREC (adaptive reference echo cancellation) algorithm is presented for an echo canceler used in full-duplex two-wire digital transmission on digital subscriber loops. The AREC algorithm incorporates a decision-directed estimation of and compensation for the far-end signal which is a source of interference to the conventional echo canceler adaptation algorithm. The AREC algorithm thus offers much faster convergence and shorter coefficient wordlengths than the conventional algorithm. Analysis and simulation of the performance and convergence of both AREC and conventional echo canceler adaptation algorithms are carried out. Included in the analysis is the effect of receiver delay and coefficient wordlength requirements. A simple and robust startup procedure is proposed and investigated by simulation.     

火箭发动机包覆层测厚的超声信号处理研究

在用超声波对固体火箭发动机双包覆层的厚度进行测量时,若包覆层较薄,超声回波信号会产生严重混叠,给各回波间的时延估计带来困难。基于超声信号的传播模型,在利用维纳滤波解卷积技术消除第一回波和噪声的影响后,采用维纳滤波解卷积技术和改进的前后向线性预测滤波器技术(MFBLP)从混叠信号中得到了高分辨率的时延估值。模拟信号和实验信号证明,该方对噪声不敏感,并对混叠信号的时延估计有很好的效果,从而提高了火箭发动机薄包覆层的厚度测量精度。     

一种基于分布式发射天线的差分空时调制方法

在基于分布式发射天线的多入多出(MIMO)系统中,由于各发射天线的发射信号不同时到达接收端,用于信道估计的导引设计及发射方法存在一定困难。针对这一问题,该文提出一种无需信道估计的分布式MIMO差分编码及检测方法:发送端将发射矩阵进行相位差分调制后发射,接收端利用前后接收量判断相位信息恢复出发送端数据信息。该方法频谱效率与V-BLAST相同,适用于任意发射天线数和接收天线数,且不要求接收天线数大于发射天线数。仿真结果表明,在不同信道传播时延情况下,误码率性能不同。     

Nonintrusive measurements of the telephone channel

Algorithms for a system which performs in-service nonintrusive measurements of telephone line parameters such as echo path delay, echo attenuation, speech and noise levels are presented. It forms the theoretical background of a hardware implementation of the ITU-T P.561 recommendations for nonintrusive assessment of quality of a telephone network for voice transmissions. The type of input signal to the echo path, whether it is speech, fax, or data, has to be determined during the measurement procedure. In the system, the relevant parameters are estimated by means of frequency-domain block-processing algorithms and the estimation process is controlled by algorithms monitoring the signal flow. The complexity of the echo path estimation algorithm is determined and compared to a standard procedure based on the adaptive normalized least mean square (NMLS) algorithm. Simulations showing the performance of the system in single-and double-talk situations are presented. Examples of results from the hardware real-time implementation operating on an international connection are also shown     

Performance of an Echo Canceller and Channel Estimator for On-Channel Repeaters in DVB-T/H Networks

This paper investigates the design and performance of an FIR echo canceller for on-channel repeaters in DVB-T/H network within the framework of the PLUTO project. The possible approaches for echo cancellation are briefly reviewed and the main guidelines for the design of such systems are presented. The main system parameters are discussed. The performance of an FIR echo canceller based on an open loop feedforward approach for channel estimation is tested for different radio channel conditions and for different number of taps of the FIR filter. It is shown that a minimum number of taps is recommended to achieve a certain mean rejection ratio or isolation depending on the type of channel. The expected degradation in performance due to the use of fixed point rather than floating point arithmetic in hardware implementation is presented for different number of bits. Channel estimation based on training sequences is investigated. The performance of maximum length sequences and constant amplitude zero autocorrelation (CAZAC) sequences is compared for different channels. Recommendations are given for training sequence type, length and level for DVB-T/H on-channel repeater deployment.     

基于DMB-T数字电视转发器回波抵消方案研究

以数字电视地面广播DMB-T标准为基础,提出了一种基于数字电视转发器中的回波抵消方案。该方案通过相关算法获取PN码的回波路径特性,进而获取该路径中相关的参数,最后利用得到参数来生成逼近回波信号的抵消信号,并从接收信号中减去该抵消信号达到回波消除的目的。仿真结果表明,该方法具有良好的回波消除性能,并具有一定实际应用价值。     

A data-driven multitone echo canceller

A data-driven echo canceller for full-duplex data transmission with multitone modulation is presented. This multitone echo canceller (MTEC) is not impaired by eigenvalue-spread problems that are inherent in signal-driven echo cancellers-it has numerical performance and cancellation range that equals, and in most cases exceeds, that of data-driven echo cancellers used in data transmission with baseband or quadrature amplitude modulation. The method makes use of frequency-domain updating, but time-domain implementation of the canceller. It introduces no delay into the received signal path and presents no special difficulties for interframe interpolation between near-end echo, far-end echo, or far-end data signal. The authors also investigate the fast initialization and the special case of far-end frequency offset in the echo signal     

A delayless subband adaptive filter architecture

Some adaptive signal processing applications, such as wideband active noise control and acoustic echo cancellation, involve adaptive filters with hundreds of taps. The computational burden associated with these long adaptive filters precludes their use for many low-cost applications. In addition, adaptive filters with many taps may also suffer from slow convergence, especially if the reference signal spectrum has a large dynamic range. Subband techniques have been previously developed for adaptive filters to solve these problems. However, the conventional approach is ruled out for many applications because delay is introduced into the signal path. The paper presents a new type of subband adaptive filter architecture in which the adaptive weights are computed in subbands, but collectively transformed into an equivalent set of wideband filter coefficients. In this manner, signal path delay is avoided while retaining the computational and convergence speed advantages of subband processing. An additional benefit accrues through a significant reduction of aliasing effects. An example of the general technique is presented for a 32-subband design using a polyphase FFT implementation. For this example, the number of multiplies required are only about one-third that of a conventional full band design with zero delay, and only slightly greater than that of a conventional subband design with 16 ms delay     

Output effect of a quadrature receiver with an input band-pass filter at reception of a phase-keyed wideband signal

An output effect of a quadrature correlation receiver with an input band-pass filter at reception of a phase-keyed wideband signal is considered. Analytical expressions for output effect of a receiver proportional to the absolute value of a complex envelope of a cross-correlation function for received and reference signals are obtained. Dependences of cross-correlation function absolute value on a time delay, frequency mismatch of a received and reference signals, and passband width of input filter are analyzed. It is shown, that cross-correlation function’s maximum is reached in case of non-zero time delay of a received signal when there is no frequency mismatch between input and reference signals. The time delay value corresponding to cross-correlation function’s maximum depends on passband width of input filter.