自适应门限处理双话的双滤波回声抵消算法

针对当前双话处理算法对未知或可变回声路径性能恶化以及复杂度过高等问题,采用一种基于短时能量比较,根据已估计的回声路径信息自适应更新判决门限的双话检测算法,为确保更新在滤波器收敛到一定程度后进行,引入了表征滤波器调整程度的参数和残留回声与参考信号的能量比来控制更新速率,同时达到检测路径变化的作用。文中算法结合双滤波器,能处理误检带来的收敛速率下降的问题。实验证明,该回声抵消器既能有效消除回声又能保护近端语音,同时能够处理路径变化。     

基于噪声估计的改进能量熵语音端点检测算法

针对传统能量熵的短时能量与子带谱熵容易受噪声环境影响,低信噪比下端点检测性能下降的问题,提出一种基于噪声估计的改进能量熵语音端点检测算法.首先对语音进行噪声估计并以此计算语音存在概率;然后利用估计的噪声能量修正短时能量,用语音存在概率作为加权系数优化子带谱熵,并将两者结合生成改进的能量熵;最后给出基于噪声估计的动态门限以及实时的端点检测策略.实验结果表明,在信噪比5 dB、0 dB的多种噪声环境中,基于噪声估计的改进能量熵端点检测算法相比传统能量熵算法与改进子带能谱比算法,检测正确率平均提升7％.     

同时同频全双工场景中的射频域自适应干扰抵消

考虑同时同频全双工无线收发信机的射频域自干扰抵消技术,现有研究多集中于利用手动方式调整自干扰估计信号的参数。针对这一问题,该文提出一种射频域的自适应干扰抵消算法。以正交、同相参考支路构成的自干扰估计结构为基础,利用梯度下降法搜索支路的最优权矢量,估计出自干扰信号,实现了射频域的自适应干扰抵消,并且给出了该算法的收敛性分析。分析与仿真表明,当迭代步长越大或统计时间越短时,算法的收敛速度越小。在100倍符号周期的统计时间,0.3的归一化步长,80 dB干信比以及0 dB信噪比的仿真条件下,该文提出的射频域自适应干扰抵消算法可以实现约100 dB的自干扰抑制。     

一种优选的回声路径变化检测算法

在回声抵消应用场景中,可以使用回声路径变化检测算法来控制自适应滤波速率。文章提出了一种优选的检测算法思路,即:当检测到回声路径变化时,提高自适应滤波的速率,反之降低。同时,优选的,当处于双端发生状态时,不判定为回声路径变化,因为如果此时使用高速率的自适应滤波容易引起滤波器系数发散。通过引入优选算法,提高了回声抵消系统的鲁棒性。     

基于IPNLMS算法的回声抵消器的设计

主要针对电话系统中的线路回声进行研究,分析了线路回声的基本原理与抑制方法,深入研究、设计并实现了基于IPNLMS自适应算法的回声抵消算法。测试结果表明,所设计算法能够很好地抵消延迟长达120 ms以内的回声信号,语音质量的PESQ得分值由回声抵消前的2.2分左右提高到回声抵消后的3.5分以上。     

基于信号包络和互相关的双端通话检测算法研究

声学回声消除(AEC)系统依赖于自适应滤波器的回声路径估计,当麦克风接收的信号存在回声和近端信号时,可能出现模拟回声路径的自适应滤波器发散,导致回声消除性能下降,严重时影响双端通话质量。一个成熟的声学回声消除器应该包含有双端通话检测算法,针对该问题,提出一种准确度高、性能稳定的基于信号包络检测(Env)和归一化互相关(NCC)估计相结合的双端通话检测(DTD)算法。该算法先使用远端信号与麦克风信号的能量包络来判断双端通话是否发生,当包络检测不能准确地判断双端通话时,再使用远端信号与麦克风信号的互相关估计对双端通话进行最终判断,保证双端通话检测的准确性。仿真实验表明,提出的算法可以准确判断双端通话的开始与结束,并且受回声路径变化的影响较小,提升了AEC系统的性能。     

VoIP通信中的回声消除方法研究

介绍了语音通信声学回声产生模型和自适应AEC回声消除算法原理,分析了AEC应用于VoIP语音通信中存在的问题,设计了一种基于短时能量的非线性回声消除方法,在NGN网络的VoIP通信中,使用该方法实现了极高的回声抑制比。测试结果表明该方法的消回声效果、算法稳定性和实现复杂度等指标明显优于自适应AEC算法,适合于嵌入式VoIP通信终端设备的开发。     

低信噪比条件下的一种自适应有声/无声判决算法

本文描述了一种利用含噪语音短时能量进行有声／无声判决的自适应算法。通常，利用短时能量进行有声／无声判决时，均采用一固定门限，但是，恰当的判决门限显然是噪声统计特性及信号能量的函数。本文提出了一种估计含噪语音短时能量概率密度函数，并根据所期望的误判率估计判决门限的算法。本算法无需预先给出噪声统计信息，且适用于缓变的非平稳噪声情况。     

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。     

基于子带双特征的自适应保留似然比鲁棒语音检测算法

为了进一步提高低信噪比下语音激活检测(VAD)的准确率,该文提出一种基于子带双特征的自适应保留似然比鲁棒语音激活检测算法。算法采用子带归一化最大自相关函数与子带归一化平均过零率双重特征设置频率分量似然比的保留权值,同时利用已过去固定时长的VAD判决结果及对应的子带特征参数自适应地估计似然比的保留阈值。实验结果表明,此算法的VAD检测准确率相比原保留似然比算法在10 dB, 0 dB和-10 dB平稳白噪声下分别提高了1.2%, 7.2%和8.1%,在10 dB和0 dB非平稳Babble噪声下分别提高了1.6%和3.4%。当其被用于2.4 kbps低速率声码器系统时,合成语音的感知语音质量评价(PESQ)比原声码器系统在白噪声下提高了0.098~0.153,在Babble噪声下提高了0.157~0.186。     

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 new configuration for echo canceller adaptable during double talkperiods

A three-port echo canceler (EC) configuration is proposed which observes the signal of the near-end side on a two-wire circuit in addition to the four-wire circuit signals. Embedding these signals on hybrid ports into a three-dimensional autoregressive process, echo path and innovations of near- and far-end speeches can be estimated through a three-channel lattice filter. The new configuration is then able to track echo path time variance, even during double talk (DT), and requires no changeover at either the beginning or end of DT, thus eliminating the need for DT detection. Two echo synthesizers utilizing inverse lattice and the echo path estimate possess guaranteed stability without the need for testing     

基于激光多普勒测速雷达自相关检测算法研究

为提高车载激光多普勒测速低信噪比时的测速精度,同时保持较好的实时性,基于自相关检测算法在车载激光多普勒测速中的应用进行了理论与实验研究.从速度估计的克拉美罗下限(CRLB)出发,将自相关后信噪比与采样点数的关系和频域内信噪比估计相结合;不同原始信号的信噪比,可以选取最优重数和采样点数以便对信号进行自相关处理,从而减小计算量,实现系统信噪比和探测概率的提高.仿真和实验结果表明,在信噪比不低于-10 dB时,二重自相关可以使系统速度估计精度接近CRLB.     

距离多普勒域多雷达信号融合

传统的单雷达目标检测算法在低信杂噪比(SCNR)场景下,其检测性能较差,对低空微弱目标检测时,由于杂波信号很强,目标回波信号可能在单个雷达的回波中被杂波淹没,导致不能检测到目标。所以,文中提出一种多雷达信号融合算法检测动目标,在距离多普勒域对多个雷达回波信号进行融合,增强目标信杂噪比。最后,通过仿真来验证提出的算法的有效性。     

Recovery of Spectral Moments from Overlaid Echoes in a Doppler Weather Radar

This paper presents a technique that enables the estimation of spectral moments of overlaid first and second trip weather echo signals in a pulsed Doppler radar. The transmitted pulses are phase coded with a sequence that allows manipulation of the signal spectrum in such a way that either the first or the second trip signal autocorrelation can be made zero, thus removing the bias error in mean velocity estimates due to overlaid echo. With a sufficiently large number of samples, desired mean velocities can be recovered using autocovariance processing provided that the interfering overlying echo power is not more than 10 dB higher than the desired echo power. Also presented is a spectral processing technique that, in conjunction with autocovariance processing, can recover all three spectral moments of both the first and second trip echoes, when their spectrum widths are narrow compared to the Nyquist interval and their power ratio is in the range ±15 dB. An algorithm is developed that can be easily implemented in existing Doppler radars, with the addition only of a lowpower switchable phase shifter and associated drive circuit.     

不需双端检测的声回波对消算法研究

声回波对消中双端对讲情况下的近端话音对自适应算法有很大影响。为避免双端话音检测,在滤波型LMS算法基础上,用远端信号和误差输出信号的和代替远端信号去激励预测误差滤波器,降低近端话音的影响。另为进一步提高算法抗近端干扰的能力,做了变步长的改进,首先将步长反比于输出信号预测误差的短时功率,其次将步长正比于预测误差的互相关系数。实验表明,文中提出的两算法在近端话音出现时表现出较好的性能,其中第二种有更好的稳态失调。     

基于离散傅里叶变换的高动态突发信号检测及频率估计

针对高动态突发通信应用环境,提出了一种新的基于频率域的突发信号检测及载波频偏估计算法,通过一次离散傅里叶变换( DFT)实现突发信号存在性检测及频率估计,并与经典Power-Law算法进行了比较。仿真结果表明：在低信噪比条件下,新算法检测信噪比门限改善超过1 dB,频率估计均方根误差小于符号率的1‰,并且对载波频偏及信号电平动态不敏感,实现结构简单,适合实时处理及工程应用。     

Blind separation algorithm of PCMA signals with different symbol rates based on DG-PSP

A blind separation algorithm was proposed for PCMA signals with different symbol rates based on double grid per-survivor processing (DG-PSP).The channel states and two input signal components were treated as two dynamic grids,receiving mixed-signal reconstructed by respectively iterative updating the two groups of grid status,thus achieving blind signal separation.The joint iterative decoding separation structure was focused,and a detailed analysis and comparison under different error estimation of parameters was shown.The complexity of the algorithm is similar to the traditional PSP algorithm.Simulation results show that,a gain of about 2 dB in signal-noise ratio can be obtained after the first iteration at a bit error rate of 10^?2,and a gain of nearly 3 dB in signal-noise ratio can be obtained after the second iteration.