A Signal Decorrelation PNLMS Algorithm for Double-Talk Acoustic Echo Cancellation

Acoustic echo cancellation (AEC) in voiced communication systems is used to eliminate the echo which corrupts the speech signal and reduces the efficiency of signal transmission. Usually, the performance of AEC system based on the adaptive filtering degrades seriously in the presence of speech issued from the near-end speaker (double-talk). In typical AEC scenarios, double-talk detector (DTD) must be added to AEC for improving speech quality. One of the main problems in AEC with DTD is that the DTD errors can result in either large residual echo or distorting the near-end input speech. Considering the strong correlation property of speech signals, this paper presents a novel proportionate decorrelation normalized least-mean-square (PDNLMS) adaptive AEC without DTD for echo cancellation as an interesting alternative to the typical AEC with DTDs. Unlike traditional AEC with a DTD, the proposed PDNLMS uses the difference of near-end speech as the residual error to update adaptive echo channel filter during the periods of double-talk, which can efficiently reduce the double-talk influence on the AEC adaptation process. The experimental results show that not only the proposed PDNLMS without DTD illustrate better stability and faster convergence rate, but it is also of a lower steady-state misalignment and better residual signal than current methods with DTDs at a lower computational cost.     

A multirate acoustic echo canceler structure

A new subband echo canceler (SBEC) structure is proposed to reduce the transmission delay introduced by conventional SBEC structures, without distorting the near-end signal. The proposed structure is based on computing two output errors, one for using during single-talk and the other one for using during double-talk periods. With the SBEC structure we propose a double-talk detector with a subband configuration which allows a fast and accurate detection of double-talk periods, enabling the SBEC algorithm to track changes in the echo path impulse response when the near-end signal is absent. Computer simulations using actual speech signals, and subjective evaluation tests are given to show the convergence performance, tracking and double-talk detection ability, of the proposed scheme     

Modulated Lapped Transform Domain Acoustic Echo Canceller with Double Talk Detector

A complete acoustic echo cancellation system with double talk detection capability is presented in this paper. The proposed system includes a new acoustic echo canceller (AEC) based on the modulated lapped transform (MLT) domain adaptive structure and a robust two-stage double talk detector (DTD) to cope with MLT domain AEC. The proposed AEC achieves better signal decorrelation via orthogonal MLT of size 2N× N rather than N× N square orthogonal transform such as DCT, DFT, etc. Both the input signal and the desired response are modulated lapped transformed in order to reduce the adaptation error between them so that the signal adaptation is purely operated in MLT domain. As a complementary of this, a two-stage DTD is developed to stabilize the operation of the AEC. The proposed DTD has robust algorithm structure and it allows faster switching according to the talker state change.Several simulation results with a synthetic and real speech are presented to demonstrate the performance of the proposed AEC and DTD. The proposed MLT based AEC proven to be very useful for the echo cancellation applications requiring high convergence speed and good echo attenuation. It can achieves faster convergence rate by more than twice over those of traditional DCT based AEC with an additional advantage of 10–15 dB ERLE improvement. On the other hand, a proposed two-stage DTD is shown to react quickly to both the onset and the end of the double-talk with reasonable high accuracy.     

A Robust ICA-Based Adaptive Filter Algorithm for System Identification

This paper proposes a new adaptive filter algorithm for system identification using independent component analysis. The additive noise is considered as an independent component to be separated from the noisy observation and is simultaneously estimated online. The proposed algorithm is derived by minimizing the mutual information between the estimated additive noise and the input signal. The local convergence conditions are also derived. The proposed algorithm can be directly applied to the acoustic echo canceller without any double-talk detector. Some simulations have been carried out to illustrate its effectiveness for synthetic and real speech signals.      

基于频域逐级回归的声学回声控制

传统声学回声控制算法一般采用基于随机梯度法更新的频域分块自适应滤波(PBFDAF)方法,但在以语音为主要回声信号的室内混响环境中,由于回声路径不稳定,往往收敛速度较慢,难以实现足够的回声抑制。该文提出一种基于频域逐级回归的声学回声控制算法。通过逐级回归分析远端信号和麦克风信号之间的线性关系,可以在保持较小的偏差的同时实现收敛较快的系统估计。同时,由于逐级分析了两通道间的短时相干性,因而该算法无需像常见方法一样,额外进行基于通道间相干函数的残余回声抑制或双讲检测,从而保持系统的紧凑性。若进一步假定近端背景噪声准平稳,则可利用基于近端信号非平稳程度的自适应平滑因子,在实现系统估计快速收敛的同时确保其稳定性。实验表明,该方法在常见的近端环境噪声水平下,在收敛速度和稳态误差上相对传统方法有显著优势,非常适合应用在室内远讲模式下的声学回声控制中。     

VoIP回声消除器设计及算法研究

本文提出了一种与线性预测编解码器相结合的新声学回声消除器,由去相关可变步长的NLMS自适应算法和基于回声路径失配方差的双端通话检测算法所组成.Matlab仿真结果表明,与Gordy所提出的回声消除算法相比,本文提出的算法在双端通话和回声路径改变时判别更准确,收敛速度更快;在收敛状态时,ERLE值平均提高了15dB,失调误差平均降低了10dB,具备更好的回声消除性能.     

Double-Talk-Robust Prediction Error Identification Algorithms for Acoustic Echo Cancellation

The performance of an acoustic echo canceller may be severely degraded by the presence of a near-end signal. In such a double-talk situation, the variance of the echo path estimate typically increases, resulting in slow convergence or even divergence of the adaptive filter. This problem is usually tackled by equipping the echo canceller with a double-talk detector that freezes adaptation during near-end activity. Nevertheless, there is a need for more robust adaptive algorithms since the adaptive filter's convergence may be affected considerably in the time interval needed to detect double-talk. Moreover, in some applications, near-end noise may be continuously present and then the use of a double-talk detector becomes futile. Robustness to double-talk may be established by taking into account the near-end signal characteristics, which are, however, unknown and time varying. In this paper, we show how concurrent estimation of the echo path and an autoregressive near-end signal model can be performed using prediction error (PE) identification techniques. We develop a general recursive prediction error (RPE) identification algorithm and compare it to three existing algorithms from adaptive feedback cancellation. The potential benefit of the algorithms in a double-talk situation is illustrated by means of computer simulations. It appears that especially in the stochastic gradient case a huge improvement in convergence behavior can be obtained     

面向二元麦克风小阵列改进的广义旁瓣抵消器语音增强算法

二元麦克风小阵列在手机、助听器等受空间、成本以及运算能力限制的设备中被广泛研究用以提高目标语音质量。二元麦克风小阵列中语音增强算法主要包括波束形成方法以及相干性滤波器方法。波束形成方法的思想是利用目标声源相对阵列的位置关系获取相应的时域和空域信息,可以保留目标声源方向的信号而抑制其他方向的干扰信号;相干性滤波器方法则通过阵元间不同信号的相关性进行噪音抑制。考虑这两种类型方法的优点,本文提出一种面向二元麦克风小阵列改进的广义旁瓣抵消器语音增强算法,通过在广义旁瓣抵消器的固定波束形成支路上使用相干性滤波器,提高固定波束形成输出信号的信噪比,然后在广义旁瓣抵消器自适应支路利用阵列的时域和空域信息对固定波束形成支路输出的信号中残余噪音进行估计,进而获得增强后目标输出信号。仿真和实际试验表明,本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。      

Acoustic Echo Cancellation Using Frequency-Domain Spline Identification

Low-complexity delayless acoustic echo cancellation techniques based on frequency-domain spline-identification are proposed and investigated. Two methods of approximation of the acoustic frequency response, both using B-splines, are considered: the optimal-spline method and the local-spline method. The optimal-spline method seeks the solution of a least squares problem. The most computationally demanding part of the method, solution of the normal equations, is implemented by using the low-complexity dichotomous coordinate descent algorithm. The local-spline method avoids solving the normal equations, enabling further simplification; this is at the expense of a slight degradation in the cancellation performance. A novel efficient double-talk detector is also proposed, being an inherent feature of the frequency-domain identification. Open-loop and closed-loop identification schemes with cubic splines are studied by simulation and compared with the fast affine projection (FAP) algorithm. The proposed techniques provide cancellation performance better than that of the FAP algorithm, especially in double-talk and noisy environments, with a lower complexity     

基于TMS320VC5402的语音检测器

本文介绍的语音检测器以DSP芯片TMS320VC5402为核心,对短波电台接收到的信号进行分析和处理。数字语音信号采用串行输入／输出方式,语音检测算法则采用对语音信号进行降噪处理后,再进行短时平均幅度差和短时能量计算的方法。该语音检测器的电路简洁小巧,语音检测准确度高。     

A matrix-based algorithm for estimating multiple coherence of a periodic signal and its application to the multichannel EEG during sensory stimulation

The coherence between the stimulation signal and the electroencephalogram (EEG) has been used in the detection of evoked responses. The detector's performance, however, depends on both the signal-to-noise ratio (SNR) of the responses and the number of data segments (M) used in coherence estimation. In practical situations, when a given SNR occurs, detection can only be improved by increasing M and hence the total data length. This is particularly relevant when monitoring is the objective. In the present study, we propose a matrix-based algorithm for estimating the multiple coherence of the stimulation signal taking into account a set of N EEG channels as a way of increasing the detection rate for a fixed value of M. Monte Carlo simulations suggest that thresholds for such multivariate detector are the same as those for multiple coherence of Gaussian signals and that using more than six signals is not advisable for improving the detection rate with M = 10. The results with EEG from 12 normal subjects during photic stimulation at 10 Hz showed a maximum detection for N greater than 2 in 58% of the subjects with M = 10, and hence suggest that the proposed multivariate detector is valuable in evoked responses applications.     

基于FPGA的实时基音周期估计系统

提出一种基于现场可编程门阵列FPGA的实时基音周期估计系统。语音信号先通过模数转换器转换成无符号位的8-bit的语音数字信号,然后,对每一帧语音信号进行电平削波,并将削波后的语音信号转换为带符号位的2-bit的数字信号,再采用自相关函数方法估计语音信号的基音周期,对一帧带符号位的2-bit的数字信号做自相关运算能够转换为简单的加法运算,只要用简单的组合逻辑电路和计数器就能够实现。使用SpartanIIXC2S30芯片将实时的基音周期估计算法用芯片内的存储器、门电路和时序电路实现,达到实时基音周期估计的目的。     

Utilising path-coherence to combat multipath and multiuserinterference

A new anti-multipath multiuser detector called the path-coherence multiuser detector (PCMUD) is presented. The high coherence between multipath signals of the same user is utilised to cut down computation complexity, as well as realise multipath diversity. Theoretical analysis and computer simulation both show that the new detector can combat multipath interference and MAI effectively     

用于海事卫星通信的自适应预测编码原理的探讨

本文首先介绍自适应预测编码在ＩＢＭＡＲＳＡＴ－Ｂ船站中的应用。然后在分析语言信号的相关性－冗余度，人类听觉“掩蔽”效应的基础上，提出语音信号的数据压缩技术－反映预测编码的语音信号的滤波，考虑音调周期和改变与信号一起输入的噪声频谱特性的自适应预测器。最后，进行线性预测分析。     

基于Hermite 插值滤波器的直接延时补偿超宽带波束形成技术研究

超宽带(UWB)信号波束形成是UWB 雷达的关键性技术。传统的波束形成方法存在瞬时带宽和扫描角度受限,波束偏移等问题,直接延时补偿法是避免上述问题的有效途径。该文提出了基于Hermite 插值滤波器的直接延时补偿波束形成方法,理论分析和仿真结果均表明Hermite 插值滤波器幅频特性和群时延特性优于目前常用的Lagrange 和径向基插值滤波器。超宽带线性调频信号实例仿真也表明了该方法在超宽带波束形成性能方面的优越性。      

Correlation LMS algorithm and its application to double-talk echocancelling

In echo cancellers for teleconference systems, the serious problem of double-talk still exists. The authors propose a new algorithm where, for tap adaptation, the gradient is searched by the input correlation functions. A computer simulation of the proposed CLMS algorithm shows robust performance in double-talk situations     

Detection of harmonic sets

We examine the detection problem of signals with narrowband, harmonically related components received by a passive sensor array. We investigate detector structures based on the Fourier method. The harmonic detector estimates the total signal power by combining the DFT coefficients from harmonic frequency bins. This power estimate is normalized by the estimated background noise power and then compared to a threshold. We investigate two harmonic detector structures: one that operates with coherent, correlated signals and the other with uncorrelated harmonic signals. We derive statistical laws governing both detector structures that facilitate setting a power threshold for a given probability of false alarm; and present upper- and lower-bounds for the probability of detection. The results developed and presented demonstrate the inherent advantage of the harmonic detector. At operating conditions characterized by low signal-to-noise power ratio values the harmonic detector exhibits enhanced detection performance by combining the estimated signal power from harmonic frequency bins. We generalize results from single-bin and harmonic detector structures and present them as special cases of a unifying framework     

Analysis of compressive receivers for the optimal interception offrequency-hopped waveforms

Establishes that the compressive receiver is a practical interceptor of high performance. Given a signal of a particular duration, a compressive receiver can estimate simultaneously all frequency components within a set wide band. This processing is similar to a parallel bank of narrowband filters, which is the optimal detector of frequency-hopped signals. Furthermore, hop frequency is estimated to yield performance equal to the parallel filter configuration. The authors assume interference to be stationary, colored Gaussian noise, and present a model of the compressive receiver that contains all its salient features. Low energy coherence detection is achieved by taking the compressive receiver output as an observation and applying likelihood ratio theory at small signal-to-noise ratios. For small signals, this approach guarantees the largest probability of correct detection for a given probability of false alarm, and thus provides a reference, to which simplified or ad hoc schemes can be compared. Since the low energy coherence detector has an unwieldy structure, a simplified suboptimal detector structure is developed that consists of a simple filter, followed by a sampler and a square-envelope detector. Several candidates for the filter's response are presented. The performance of the low energy coherence detector based on compressive receiver observations is compared to the optimal filter-bank detector based on direct observations, thus showing the exact loss incurred when a compressive receiver is used. The performance of various simplified schemes, based on compressive receiver observations, is analyzed     

Performance analysis and optimization of a space-time selective PIC for CDMA systems

This paper deals with an efficient receiver for applications in direct sequence-code-division multiple access wireless communication systems. This receiver combines a modified scheme of a parallel interference cancellation detector with antenna arrays with optimum beamforming and is used at the base station for the detection of asynchronous user signals. Each user's transmission channel is assumed to be a multipath frequency-selective independent Rayleigh fading channel. The receiver operates by dividing the signals into reliable and unreliable sets following space-time combining. The reliable signals are detected and canceled from the whole signal received at each sensor. Unreliable signals are then detected to improve the decision reliability. The receiver performance in terms of bit-error probability is analytically derived and optimized. According to the analytical and simulation results, this receiver outperforms previously proposed schemes and, thanks to its low implementation complexity, real-time operation is achieved.     

Channel and source considerations of a bit-rate reduction technique for a possible wireless communications system's performance enhancement

In wireless commercial and military communications systems, where bandwidth is at a premium, robust low-bit-rate speech coders are essential. They operate at fix bit rates and those bit rates cannot be altered without major modifications in the vocoder design. A novel approach to vocoders, in order to reduce the bit rate required to transmit speech signal, is proposed. While traditional low-bit-rate vocoders code original input speech, the proposed procedure operates on the time-scale modified signal. The proposed method offers any bit rate from 2400 b/s to downwards without modifying the principle vocoder structure, which is the new NATO standard, Stanag 4591, Mixed Excitation Linear Prediction (MELP) vocoder. We consider the application of transmitting MELP-encoded speech over noisy communication channels by applying different modulation techniques, after time-scale compression is applied. Three different time-scale modification algorithms have been evaluated and waveform similarity overlap and add (WSOLA) algorithm has been selected for time-scale modification purposes. Computer simulation results, both source and channel, are presented in terms of objective speech quality metrics and informal subjective listening tests. Design parameters such as codec complexity and delay are also investigated. Simulation results lead to a possible wireless communications system, whose performance might be enhanced by using the spared bits offered by the procedure.