IP会议电话回声消除器的设计及仿真

目前最小均方（LMS）自适应算法已经成为IP电话回声消除器采用的主要算法。但在一些特别的场合,在IP会议电话中,用于回波抵消的自适应滤波器的阶数非常高,要实时实现自适应回波抵消是非常困难的。介绍了一种归一化快速块最小均方算法（NFBLMS）,相比LMS算法,它不仅可以有效减小运算复杂度,同时也能改善收敛速度。最后在MATLAB中生成符合G.168要求的带限复合信号源作为远端参考信号。并运用MATLAB对算法进行仿真,仿真结果表明NFBLMS算法的各项性能指标要优于LMS算法,从而适用于IP会议电话的要求。     

自适应回声消除器设计

针对救援通信系统语音通话过程中回声干扰问题,基于采用归一化最小均方误差算法的自适应滤波器和语音检测模块,设计了一种自适应回声消除器,介绍了该自适应回声消除器的工作原理和设计方案。测试结果表明,该自适应回声消除器收敛速度快,可在1s内收敛到-30dB以下,基本消除音频信号回声。     

数字可视电话系统中的G.168回声消除技术

由于数字可视电话系统的性能需求和系统中回声的特点,传统的回声消除技术应用于数字可视电话系统时无法满足ITU-T G.168标准要求。该文利用NLMS算法的滑动窗自适应滤波器和简化非线性处理器设计,实现一种数字可视电话系统中的G.168自适应回声消除器,介绍其基本结构和各个功能模块的算法,并以ARM 920T为硬件平台实现该系统。仿真结果显示,简化的回声消除算法运算复杂度小、消除回声有效性高、稳定性强。利用ITU-T G.168标准定义的CSS信号对G.168回声消除器进行测试,测试结果均符合或优于标准要求。     

用C5402 DSP实现的G.168自适应回声消除器

叙述网络电话系统的回声特点，介绍了ITU－T G．168建议书的有关要点。基于性价比方面的考虑，提出了利用nLMS算法构成的横向滤波器组成自适应回声消除器，并在TMS320C5402 DSP上实现了该算法。经过测试，主要性能均达到G．168规范要求。     

基于分段块频域自适应滤波算法的长延时回声消除

在网络电话、电话会议、车载电话、IP电话等免提电话系统以及其他一些通信系统中,要想得到高质量的回声消除效果,必须抑制声学回声、尤其是长延时回声.然而对于长延时回声的抑制,如果采用传统的基于LMS/NLMS的自适应算法,则会产生大而复杂的计算量,且收敛速度慢.为解决此问题,本文对分段块频域自适应滤波算法进行了研究,即首先对该算法进行了描述,然后通过录制两个语音信号进行仿真.仿真结果表明,该方法具有非常好的长延时声学回声消除效果.     

IP电话系统中语音芯片的DSP实现

介绍了一种具有G．723．1语音编解码、自适应回声抵消DTMF编解码功能的语音芯片，以满足IP电话系统中低速率语音通信的需要。根据DSP内核的特点，对G．723．1算法进行了优化，降低了算法 复杂度；回声抵消部分使用归一化最小二乘（NL MS）自适应滤波器实现,采用Goertzel算法实现DTMF的检测。使用较少的资源,在一块TMS320C5402芯片上集成了以上全部功能。     

IP电话回声消除LMS改进算法的研究

针对传统LMS算法运算量大收敛性能差的缺点,提出了一种减小运算量并且提高收敛性能的LMS自适应滤波算法.首先从理论上介绍并分析了两种自适应滤波算法--量化误差算法(运算量小)和变步长算法(收敛速度快),接着将这两种算法的优点有机结合,提出了一种改进算法.通过在MATLAB下的辨识仿真研究和对误差曲线的分析,证明了结合后的改进算法在运算量和收敛速度方面都优于传统LMS算法.算法对于IP电话中回声消除的自适应滤波问题提供了一个较好的算法.     

长时延环境中的回声消除器设计

本文讨论了回声消除器的基本原理以及软件环境中声音信号的时延特性,并提出了一种适用计算机软件实现回声消除器的改进的时延估计算法。实验结果表明该算法增强了回声消除器对环境的适应性。     

一种基于RBF 网络的非线性自适应逆控制系统

改进了原有的ε-滤波自适应逆控制系统，引入自适应扰动消除器和反馈补偿，构成一种新的自适应逆控制系统．反馈补偿能消除自适应逆控制系统中的直流零频漂移，自适应扰动消除器能最大限度地消除扰动．将神经网络引入自适应逆控制系统，采用基于径向基函数网络的非线性滤波器，对非线性系统进行建模、逆建模、控制器及自适应扰动消除器的设计．仿真结果验证了该方法的有效性．     

基于时间尺度调整技术的自适应缓冲新算法

介绍了VoIP应用中几种主要的动态缓冲算法,分析了各自的实现方式和性能优劣,提 出了基于简化的相似波形叠加技术和双窗预测技术的自适应缓冲算法,并进行了仿真。结果表明,新 算法能明显提高IP电话的语音服务质量。     

基于DSP的回波抵消器的软件实现

随着通讯、数字信号处理和大规模集成电路技术的飞速发展,人们对语音通信质量的要求越来越高。回波抵消技术能有效的解决长距离电话网络、IP电话、免提电话和视频会议等通信系统中的回波问题,很好地改善了语音通信质量,具有广阔的市场前景。主要研究用于声音通讯系统中的声回波抵消技术。针对具体应用以及DSP芯片的特点,结合BEPNLMS算法,给出了一种基于TM S320VC 5402 DSP实现回波抵消的软件设计方案。     

一种具有消回声和消背景噪声功能的AENC设计方法研究

在视频会议系统中,回声和背景噪声是影响系统通话质量的两个重要因素。本文从回声产生的机理和常规消回声、消噪声的方法着手,研究了一种具有消除背景噪声功能的新的AEC,即AENC（Acoustic Echo and Noise Can-celler)的设计方案,并在理论上对其进行了分析、论证。结果表明,这种AENC在理论上完全能够同时实现消回声和消背景噪声两种功能。它在IP电话、可视电话及免提的数字蜂窝电话等扬声器/传块器通信系统中也具有广阔的应用前景。     

Improved variable step-size NLMS adaptive filtering algorithm for acoustic echo cancellation

Acoustic echo canceller (AEC) is used in communication and teleconferencing systems to reduce undesirable echoes resulting from the coupling between the loudspeaker and the microphone. In this paper, we propose an improved variable step-size normalized least mean square (VSS-NLMS) algorithm for acoustic echo cancellation applications based on adaptive filtering. The steady-state error of the NLMS algorithm with a fixed step-size (FSS-NLMS) is very large for a non-stationary input. Variable step-size (VSS) algorithms can be used to decrease this error. The proposed algorithm, named MESVSS-NLMS (mean error sigmoid VSS-NLMS), combines the generalized sigmoid variable step-size NLMS (GSVSS-NLMS) with the ratio of the estimation error to the mean history of the estimation error values. It is shown from single-talk and double-talk scenarios using speech signals from TIMIT database that the proposed algorithm achieves a better performance, more than 3 dB of attenuation in the misalignment evaluation compared to GSVSS-NLMS, non-parametric VSS-NLMS (NPVSS-NLMS) and standard NLMS algorithms for a non-stationary input in noisy environments.     

一种计算复杂度低的双端通话检测算法

在声学回音消除中,近端语音的出现会导致模拟回音路径的自适应滤波器发散,一个成熟的声学回音消除器应该包含有双端通话检测算法。针对这个问题,提出了一种计算复杂度低的、基于麦克风信号与误差信号的互相关双端通话检测算法。同时,该算法与滤波器系数缓存机制相结合以进一步提高系统的鲁棒性。实验结果表明,该算法具有良好的检测性能,可以对双端通话的出现和消失做出快速响应,同时能显著提高系统在双端通话环境下的回音消除效果。     

Joint Dereverberation and Residual Echo Suppression of Speech Signals in Noisy Environments

Hands-free devices are often used in a noisy and reverberant environment. Therefore, the received microphone signal does not only contain the desired near-end speech signal but also interferences such as room reverberation that is caused by the near-end source, background noise and a far-end echo signal that results from the acoustic coupling between the loudspeaker and the microphone. These interferences degrade the fidelity and intelligibility of near-end speech. In the last two decades, postfilters have been developed that can be used in conjunction with a single microphone acoustic echo canceller to enhance the near-end speech. In previous works, spectral enhancement techniques have been used to suppress residual echo and background noise for single microphone acoustic echo cancellers. However, dereverberation of the near-end speech was not addressed in this context. Recently, practically feasible spectral enhancement techniques to suppress reverberation have emerged. In this paper, we derive a novel spectral variance estimator for the late reverberation of the near-end speech. Residual echo will be present at the output of the acoustic echo canceller when the acoustic echo path cannot be completely modeled by the adaptive filter. A spectral variance estimator for the so-called late residual echo that results from the deficient length of the adaptive filter is derived. Both estimators are based on a statistical reverberation model. The model parameters depend on the reverberation time of the room, which can be obtained using the estimated acoustic echo path. A novel postfilter is developed which suppresses late reverberation of the near-end speech, residual echo and background noise, and maintains a constant residual background noise level. Experimental results demonstrate the beneficial use of the developed system for reducing reverberation, residual echo, and background noise.      

Vo IP 的语音质量分析与控制

分析了VoIP语音质量的影响因素，通过E模型定量地描述了语音质量与端到端延迟和丢包率的关系。为了控制VoIP的语音质量，计算出VolP系统在各种情况下的语音质量极限，提出一种自适应编码和分组封装的控制策略。将该方法应用于自行开发的IP电话网关，实际测试证明能在很大程度上提高VoIP的语音质量。     

语音回声信息隐藏技术研究

回声信息隐藏是通过在语音媒体中加入不同延时的回声来实现信息隐藏的一种技术。采用易于检测的前向一后向回声核构建了回声信息隐藏系统。介绍了前向一后向回声核的构成原理和倒谱检测方法，仿真并分析了系统关键参数对隐藏信息恢复率的影响，如延迟时间、分段长度、回声的衰减系数等。通过引入8个延时的回声实现了8进制信息隐藏，使隐藏信息的容量提高了3倍，在采样频率为8kHz的条件下达到150bit／s，恢复率大于99％。20人的非正式试听表明，隐藏信息后的语音和原始语音没有明显的差别且优于单回声的语音质量，得到了较好的隐藏效果。     

Enhancing Bandwidth Utilization of IP Telephony Over IPv6 Networks

The demand for the telecommunication services, such as IP telephony, has increased dramatically during the COVID-19 pandemic lockdown. IP telephony should be enhanced to provide the expected quality. One of the issues that should be investigated in IP telephony is bandwidth utilization. IP telephony produces very small speech samples attached to a large packet header. The header of the IP telephony consumes a considerable share of the bandwidth allotted to the IP telephony. This wastes the network's bandwidth and influences the IP telephony quality. This paper proposes a mechanism (called Smallerize) that reduces the bandwidth consumed by both the speech sample and the header. This is achieved by assembling numerous IP telephony packets in one header and use the header's fields to carry the speech sample. Several metrics have been used to measure the achievement Smallerize mechanism. The number of calls has been increased by 245.1% compared to the typical mechanism. The bandwidth saving has also reached 68% with the G.28 codec. Therefore, Smallerize is a possible mechanism to enhance bandwidth utilization of the IP telephony.