水电站生产过程仿真算法研究

针对水电站仿真培训系统的选型，在建立水电站生产过程数学模型的基础上，提出满足实时性要求的仿真算法。     

分组公平排队算法的仿真研究

对ATM复接器和输出缓存ATM交换机采用各种分组公平排队算法情况下的信元时延进行了仿真。仿真结果表明,SCFQ对连接之间的隔离性比其他算法差;若一个CBR速率组内没有流量超过其预约带宽的连接,则用Per_Rate_Grouping方式实现WF2Q+和不用这种实现方式,从最大时延和平均时延来看,相差不大。提出了一个调度CBR、VBR、ABR和UBR的调度器结构,既能满足CBR对时延的严格要求,又能保证VBR、ABR和UBR的吞吐率,同时还具有较低的实现复杂性。     

IP网络的实时分组音频回放自适应算法

以提高音频分组在IP网络接收端的回放质量为目标,根据音频分组的回放原理,研究音频回放自适应算法,针对处理话音期内出现的延迟尖峰,提出了一种改进算法.根据实时传输/控制协议(RTP/RTCP)框架,实现了基于RTP的Internet音频传输.仿真实验表明,改进算法可适时调整回放延迟,在网络负载较重时,能有效地减少分组丢失/平均回放延迟开销.     

无比特丢弃的AAL2分组话音复接器模型及性能分析

根据分组话音业务的特点，结合分组话音复接器主要性能指标的要求，研究无比特丢弃的AAI2分组话音复接器的数学模型，得出结论，对于无比特丢弃的AAL2分组话音复接器，可用UAS模型准确研究其分组丢弃概率；而平均分组排队时延可用M/D/I/K模型来近似描述，但存在一定的误差，计算机仿真证实结论是正确的。     

AAL2话音分组的最优长度

根据分组话音业务的特点，结合分组话音业务服务质量要求，研究带比特丢弃的第2类ATM适配层（AAL2）分组话音系统中，AAL2分组长度的确定方法。通过理论研究与计算机仿真得出，对于带比特丢弃的AAL2分组话音系统，当话音采用32kb/s的G.727E-ADPCM编码时，AAL2分组最优长度可定为23或27个字节。此时AAL2分组的额外开销小，所得的许音质量高。     

分布交互仿真的预估算法研究综述

对预估算法的基本思想及其在交互仿真中的作用进行了分析；探讨该算法的基本模型和插值积分模型；并指出研究该算法的主要方向。     

有向图的分组拓朴排序算法及实现

拓朴排序是将一个有向图的偏序结构排成一个全序结构的拓朴序列 ,根据这个拓朴序列 ,只能串行安排任务流程 .为此 ,提出有向图的分组拓朴排序概念 ,得到的分组拓朴序列解决了任务的并行安排问题 .     

分组密码算法的测试方法研究

安全性是分组密码最重要的设计准则。该文基于分组密码算法,分析了算法的多种安全性测试方法和测试理论,提出相应的测试统计量。采用AES算法中的一个8×8 S盒,以一组长为1 048 576 b的随机数为测试序列,进行实际安全性测试。测试结果表明测试统计量符合实际测试结果,从而验证所提出的测试统计量具有实际可行性,有力地保证了分组密码的安全性设计。     

基于客观测试方法针对普通话的AMR话音激活检测算法性能分析

为了评价3GPP标准化的AMR VAD应用于汉语普通话的性能，采用正式的客观测试方法作为性能的度量准则，该准则主要基于剪音概率，话音激活因子，噪声误检为语音的概率和语音输入电平4个参数，在不同背景环境，不同信噪比和不同语音输入电平下，完成了AMRVAD和两种参考VAD（EFR和G．729B）的仿真的。结果表明，AMR VAD总体性能优于G．729VAD，而较EFR VAD略差。     

一类经济跟踪问题的算法与计算机仿真

对于一类动态投入产出模型最优消费跟踪问题给出一完整算法。根据这一算法编写程序已经在ＰＣ机上进行详细计算，仿真结果表明该算法有效且便于应用。     

基于静音识别的改进型ADDPCM语音压缩算法的研究

针对语音信号振幅较大时传统压缩算法的跟踪性较差的情况，提出了一种新型的改进型自适应ADDPCM语音压缩算法．同时，根据语音信号不连续的特点，压缩算法具有静音识别功能，进一步提高了语音信号的压缩率．压缩算法在8kHz的A／D采样速率下，将实时传输和处理的语音数据减少到了3kb／s的数据流量，并且解码后的语音恢复信号具有良好的可懂度和较高的语音能量，取得了良好的语音压缩效果．在语音信号压缩算法研究的基础上，利用片上系统单片机C8051F005实现了语音信号的压缩编码和解码，并进行了效果的比较．实验表明，该语音压缩算法有效降低了编码率，提高了传输带宽的利用率，并且恢复的语音信号在振幅较大时取得较好的动态跟踪效果．     

智能型语音报时钟设计

介绍了以８０３１为核心的智能语音报时钟的设计．这种数字式智能电子钟，不仅克服了机械钟的缺点，还具有自动校时、夜光照明控制等优点     

Packet-layer model for voice quality assessment using two-level temporal pooling scheme

Aiming at the problem that the voice qualities corresponding to different packet loss patterns show significant differences at the same packet loss rate, a packet-layer model for voice quality assessment, which well reflects the effect of the packet loss patterns on the voice quality, is presented. First, the information about the coding bit-rate and packet loss is obtained by analyzing the packet header, on the basis of which the frame quality is measured with the further information about silence detection and error propagation. Then the voice sequence is divided into groups of frames (GOFs) with a variable length and a short-term temporal pooling method is employed to obtain the GOF quality. Finally, the overall voice quality is determined by the long-term temporal pooling of the GOF qualities. The proposed two-level temporal pooling scheme well describes the effect of different packet loss patterns on the voice quality since the strongest impairments are predominately emphasized. Experimental results show that the presented model can lead to an increment of about 0.0129 in the Pearson Correlation coefficient (PCC) and a decrement of about 0.0234 in the Root Mean Squared Error (RMSE) compared with the E-model in ITU-T recommendation G.107.     

QoE-driven voice quality evaluation model of VoIP based on network simulation

The quality of experience (QoE) evaluation model for voice over IP (VoIP) service is studied to analyze the impact of network parameters on voice quality and monitor voice quality in real-time for operators. Firstly, the influence of some network parameters on the voice quality of VoIP is investigated. Then, a simulation platform for VoIP transmission is built to collect voice data under different network enviornments. According to the simulation results, a new mapping model between these arguments and VoIP voice quality is deduced. Finally, the accuracy of this voice quality evaluation model is examined and the results demanstrate that it has high reliability and feasibility.     

AAL2分组话音复接器连接允许控制算法

根据AAL2分组话音业务的特点 ,结合AAL2分组话音业务服务质量的要求 ,用流体流分析法和直观分析法分析在AAL2分组话音统计复接器处进行连接允许控制的算法 ,得出连接允许控制的计算公式 .计算机仿真证实连接允许控制的计算公式很好地满足了服务质量的要求     

How to Improve the Voice Quality of VoIP

VoIP(voice over intemet protocol) has made great progress in Communication area in recent years.But a biggest pity is that the voice quality of VOIP can‘t satisfy users as traditional phone via PSTN does.In this pa- per,the author analysis the reason and bring out some methods to improve the voice quality of VOIP that are utili- ring the bandwidth effectively to reduce delay;minishing the jitter to reduce packet lose and bit error;eliminating the echo.As the emphases,the author pointed out the specialty ...     

Detection of voice transformation spoofing using the dense convolutional neural network

Voice transformation (VT) spoofing refers to the operations for hiding the speaker’s identity which change a speaker’s acoustic features by speech processing algorithms and result in extremely high false reject rates for automatic speaker recognition (ASR) systems.VT spoofing is implemented with a low cost and has been integrated in many audio editing tools,thus presenting serious threats to social security.However,the research on VT spoofing detection is still insufficient.Hence,in this paper we propose a dense convolutional neural network (DenseNet) based VT detection method for distinguishing spoofed voices and genuine ones.The proposed network consists of 135 layers in total.By maximizing the skip-layers,the data transmission can be enhanced,and both the deep and shallow edge features can be used for classification,so as to alleviate the degradation phenomenon and further to improve detection accuracy.Experimental results show that the detection accuracy with various spoofing factors is over 98%.     

用于异常检测的免疫实值检测器优化生成算法

针对已有实值可变半径检测器生成算法的不足,提出一种优化的检测器生成算法。通过对检测器生成过程的统计分析,给出了基于假设检验的检测器生成过程,并将假设检验的结果作为算法结束的一个控制参数,有效减少了冗余检测器的产生。同时,算法充分利用自体空间的分布,优化检测器生成的中心位置,扩大检测器的半径,尽可能生成覆盖范围大的检测器,提高检测性能。通过人工合成数据集2DSyntheticData以及实际的Iris数据集和Biomedical数据集对算法进行了验证。实验结果表明,本算法用于异常数据检测,提高了检测率,所需的检测器数量减少,整体检测性能较优。