SMV语音编码算法及仿真

介绍了第三代通信系统中的可选模式语音声码器(SMV)语音编码方式,简要描述了SMV的编、解码基本原理。并进行了该算法的定点C代码仿真,给出了算法的性能、计算复杂度及存储空间等仿真结果。     

自适应网络流量线性预测算法及应用*

Internet网络流量的分析、模型仿真以及流量的预测,在网络管理和设计中起着很重要的作用。分析了CERNET网络流量行为,提出了CERNET IP Backbone的流量模型,同时将自适应滤波的新思想引入网络流量的模型仿真和预测,提出了自适应网络流量线性预测的新算法,并将其应用于CERNET的网络流量预测。     

基于矢量模型的语音线性预测研究

研究语音参数线性预测的并行处理问题。通过把语音源序列的相邻样本分组能够构成一个均方差平稳的语音向量自回归序列,在Hilbert空间中运用正交投影原理导出具有高度并行处理能力的一预测编码策略,由此可推出参数线性预测的并行处理自适应算法。同传统格型算法相比,这种算法的计算复杂度及存贮量有明显改善。最后通过仿真运算检测了算法的性能。     

对LPC-10编码算法的分析与改进

本文在介绍语音信号产生模型和线性预测理论的基础上,分析了LPC-10的编解码算法,并对算法中涉及的二元激励和基音周期的提取方法进行了改进。     

线性直流无刷电机的多步预测控制

本文推导了一种新型的多步预测控制。该控制器通过对控制输入加权，使得控制行为更加稳定和平滑。对线性直流无刷电机的位置控制仿真证实了本算法的有效性。     

基于自适应滤波理论的网络流量线性预测

Internet网络流量的分析、模型仿真以及流量的预测，在网络管理和设计中起着很重要的作用。该文在此方面做了一些工作和尝试，主要有两方面的贡献：（1）在分析和比较了不同模型性能的基础上，提出了CERNET IP backbone的流量模型；（2）将自适应滤波的新思想引入网络流量的模型仿真和预测，提出了自适应网络流量线性预测的新算法。     

电话网话务量预测的理论研究与实践

文章根据线性预测理论,采用自适应算法,提出了一种不同于传统理论的话务量预测与统计方法;详细介绍了该方法的模型、原理与实现。     

基于线性预测和最大似然的基音检测算法

根据语音信号产生机理，结合常用的线性预测和最大似然法，提出了一种有效的基音检测算法。该算法采用频域分块估计候选基音周期的范围，提高了算法的计算速度。仿真实验表明，该算法与传统方法相比其基音检测结果有了明显的改善，克服了随机错误及倍频、半频错误，在低信噪比下鲁棒性较好。     

一种基于G.711包丢失隐藏改进算法的研究

对一种改进的高性能包丢失隐藏算法进行了分析。此算法以G.711脉冲编码调制为基础,建立一种新型线性预测模型,并通过仿真得出此改进算法对VoIP语音丢失包能起到比较好的恢复效果。     

基于边缘检测的快速帧内预测算法

介绍基于边缘检测的两种快速帧内预测算法,通过对两种算法在预测一致性方面的仿真,比较了它们在预测精度上的差异,对基于区域边缘强度的快速算法进行了分析和仿真,并提出了一种改进的快速帧内预测算法.     

无线衰落信道中不等保护能力AMR-WB的性能研究

针对无线衰落信道中AMR-WB宽带语音编码,提出一种基于多速率删余卷积码的不等错误保护传输方案。根据AMR-WB语音编码数据的不同重要性,采用强错误保护能力的删余卷积码为AMR-WB语音编码中的A类数据提供错误保护能力,对B类数据采用高码率删余卷积码提供错误保护能力。研究结果表明,在同样传输带宽条件下,不等错误能力保护可以有效改善无线衰落信道中AMR-WB语音质量。     

A new scheme for covert communication via 3G encoded speech

Mobile communication through 3G network has grown rapidly in recent years. It might be of interest to transmit secret messages over 3G voice channels. In this paper, we introduce a new covert communication scheme via Adaptive Multi-Rate Wideband (AMR-WB) encoded speech. An adaptive suboptimal pulse combination constrained (ASOPCC) method is presented to embed data on compressed speech signal of AMR-WB codec. The method takes advantage of the “redundancy”, created by non-exhaustive search of algebraic codebook, to encode secret information. An embedding factor η is used to control embedding bits. By properly setting η, ASOPCC can offer a better trade-off between speech quality and embedding capacity in the process of coding mode switching. Experimental results show that the proposed method is quite promising for both high capacity and good imperceptivity. Although ASOPCC is only applied to AMR-WB codec in this article, it can be further used by any other speech coding based on Algebraic Coded Exited Linear Prediction (ACELP).     

An effective hybrid cuckoo search algorithm for constrained global optimization

In recognition of high-quality wideband speech codecs, several standardization activities have been conducted, resulting in the selection of a wideband speech codec called adaptive multi-rate wideband (AMR-WB). The algebraic code-excited linear prediction (ACELP) technique is recommended in AMR-WB, and it is noted that most of the complexity in the ACELP structure comes from the codebook search. In this paper, a new method is proposed for codebook search based on the behavior of backward filtered target signal, d(n), introduced in ITU-T G.722.2 recommendation. To optimize the proposed scheme, five optimization algorithms (i.e., modified genetic algorithm, particle swarm optimization with dynamic inertia weight, bee colony optimization, modified differential evolution, and imperialist competition algorithm) are investigated. Experimental results show that the reduction in codebook search operations of the proposed method is able to reach up to 59 percent as compared with ITU-T G.722.2 recommendation. Meanwhile, BCO-based codebook search scheme has better convergence speed without significant degradation in quality metrics, such as segmental signal-to-noise ratio, mean opinion score, and perceptual evaluation of speech quality, when used in an AMR-WB speech codec.     

Wideband Speech Coding Advances in VMR-WB Standard

This paper presents novel techniques for source-controlled variable-rate wideband speech coding. These techniques have been used in the variable-rate multimode wideband (VMR-WB) speech codec recently selected by the Third-Generation Partnership Project 2 (3GPP2) for wideband (WB) speech telephony, streaming, and multimedia messaging services in the cdma2000 third-generation wireless system. The codec utilizes efficient coding modes optimized for different classes of speech signal including generic coding based on AMR-WB for transients and onsets, voiced coding optimized for stable voiced signals, unvoiced coding optimized for unvoiced segments, and comfort noise generation for inactive segments. Several innovations enable very good performance at average bit rates below 8 kb/s for active speech coding. The article presents an overview of the codec and describes in detail some of the codec novel features: Robust pitch tracking algorithm, coding-mode dependent prediction of linear prediction (LP) filter quantization, and novel frame erasure concealment techniques including supplementary information for reconstruction of lost onsets and improving decoder convergence. Selected results from the Selection and Characterization tests of the codec illustrate its performance     

基于自适应加权谱内插的宽带语音编码算法

提出了一种基于自适应加权谱内插(STRAIGHT)的宽带语音编码算法。输入的语音信号首先经过STRAIGHT分析得到精确的基频参数和谱参数，然后通过时域抽取和频域建模实现有效的编码压缩。在时域抽取时采用的区别于传统编码算法固定帧长的自适应可变帧长方法，使得编码存储量可以根据实际语音变化情况得到更加合理的分配。主观测听结果表明，该算法针对16kHz采样的语音信号，在6kbps码率上可以取得与AMR-WB(G．722．2)在8．85kbps时的相当的音质效果。此外，该算法还具有对恢复语音的时长、基频以及谱参数较强的调整能力。     

一种快速的固定码本搜索方法

为降低固定码本搜索算法的复杂度,在脉冲取代法的基础上提出一种码矢分段优化的快速搜索方法。采用码矢分段优化的方法,在保证语音质量的前提下,降低计算复杂度。实验结果表明,与AMR-WB采用的深度优先树算法及传统的脉冲取代算法相比,在不影响语音质量的条件下,码矢分段优化算法复杂度降低了70%~80%。     

Improved voice activity detection algorithm using wavelet and support vector machine

This paper proposes an improved voice activity detection (VAD) algorithm using wavelet and support vector machine (SVM) for European Telecommunication Standards Institution (ETSI) adaptive multi-rate (AMR) narrow-band (NB) and wide-band (WB) speech codecs. First, based on the wavelet transform, the original IIR filter bank and pitch/tone detector are implemented, respectively, via the wavelet filter bank and the wavelet-based pitch/tone detection algorithm. The wavelet filter bank can divide input speech signal into several frequency bands so that the signal power level at each sub-band can be calculated. In addition, the background noise level can be estimated in each sub-band by using the wavelet de-noising method. The wavelet filter bank is also derived to detect correlated complex signals like music. Then the proposed algorithm can apply SVM to train an optimized non-linear VAD decision rule involving the sub-band power, noise level, pitch period, tone flag, and complex signals warning flag of input speech signals. By the use of the trained SVM, the proposed VAD algorithm can produce more accurate detection results. Various experimental results carried out from the Aurora speech database with different noise conditions show that the proposed algorithm gives considerable VAD performances superior to the AMR-NB VAD Options 1 and 2, and AMR-WB VAD.     

新型宽带语音编解码器AMR-WB的研究

该文介绍了一种应用于第三代移动通讯系统的编解码器,同时也是第一个可同时用于无线和有线应用的编解码器,该编解码器的语音带宽拓展为50Hz到7000Hz,编码后语音的自然度很高,用在3G移动通讯系统的多媒体服务、宽带包交换网络、音频和视频会议等等。由于AMR-WB为一个全新的宽带编解码器,其标准在2001年3月刚刚通过,国外对其的研究也属于起步阶段,还没有真正进入实用阶段。尤其在国内,至今尚未见到相关的研究。故对其进行全面的分析和深入的研究是必要的,会对今后的研究打下良好的基础。     

三代移动通信变速率语音编解码AMR-WB+算法优化

首先简要介绍了AMR-WB+语音压缩算法的基本原理,描述了AMR-WB+编解码流程;然后通过两类优化策略对AMR-WB+算法进行优化;最后给出了优化前后编解码复杂度比较,并对结果进行了分析。