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%。     

一种引入多脉冲提高AMR_WB丢包处理质量的前向纠错技术

GSM AMR_WB是唯一可以作为有线和无线通用的语音编码标准,提高AMR_WB在丢包环境下的语音质量至关重要.本文提出了一种改进的AMR_WB丢包处理方法,发送端利用基于多脉冲的前向纠错技术,解决丢包时自适应码本带来的错误传播问题.接收端采用基音轮廓修改方法消除语音帧不同步在帧边界带来的恼人噪声.提出的方法在AMR_...     

G.729中固定码本快速搜索算法研究

虽然G.729中采用的集中搜索和G.729a中采用的深度优先树搜索可以有效减少固定码本搜索复杂度,但固定码本搜索在整个语音编码算法中仍占有较大比重.为了在基本维持语音质量的前提下,减少搜索运算量,研究了几种快速搜索算法,脉冲替代和预选替代一个脉冲搜索算法可以大大减少搜索次数,但语音质量明显下降,因此提出每次替代两个脉冲搜索算法,得到比替代一个脉冲较为完整的搜索,产生较好的语音质量.仿真结果表明,该算法可以大大减少搜索运算量,并且保持了和G.729a深度优先树搜索算法相同的语音质量.     

一种改进的G.729标准固定码本快速搜索算法*

在G.729建议的共轭结构代数码激励线性预测编码（CS-ACELP）中,固定码本搜索在整个语音编码算法中占有较大比重,直接影响编码算法复杂度。全搜索算法准确度很高,但搜索量过大,而传统的脉冲序列替换搜索,搜索次数减少,但合成语音质量较差。为解决该问题,提出一种基于脉冲序列替换的改进码本搜索算法。设定循环阈值门限,对脉冲序列重置后的部分脉冲组合进行全搜索,引入双脉冲位置替换,有效地减少了搜索次数,同时提高了搜索准确度。实验结果证明,该算法在增加算法复杂度较少的情况下,合成语音质量有明显的改进。     

G.723.1语音编码算法闭环基音搜索的低复杂度算法

由于ITU-TG.723.1语音编码算法具有较高的算法复杂度,故而在应用与实现时受到了很多的限制。该文提出一种低复杂度闭环基音搜索算法,该算法仍以5阶基音预测器为基础,但在求取5个基音预测增益时不是采用原算法中对20维矢量码本进行搜索的方法,而是利用这个20维矢量组成一个Wiener-Hopf方程,并利用语音的短时平稳特性将该方程简化为一个Toeplitz线性代数方程组,方程组的解就是所求的基音预测增益。对该增益进行5维码本矢量量化,从而用5维矢量码本搜索代替了原来的20维矢量码本搜索。这样使闭环基音搜索部分的运算量降低了一半,语音质量只有略微下降,同时与G.723.1算法码流兼容。     

基于G.729A的CS-ACELP语音编码算法的优化改进

介绍了G.729A的现状以及在发展过程中其语音编码的改进方向及具体方法。提出了一种结合WD-LSP(Weighted Delta-LSP)并采用次最优部分码本快速搜索的CS-ACELP语音编码算法,同时采用基于声学心理模型的知觉加权滤波器,使语音编码在不降低语音质量的情况下降低计算复杂度。     

一种降低ACELP码书搜索复杂度的高效算法

为了降低代数码激励线性预测(algebraic code-excited linear prediction, ACELP)语音编码算法的复杂度, 以便更好地实时实现, 提出了一种有效的改进算法。在自适应码书搜索上提出了不连续的开环基音搜索算法, 利用时间抽取因子对不同时延段语音样点进行不连续抽取; 在代数码书的搜索上提出了一致脉冲替换法, 采用脉冲位置预选和循环判断机制控制码书搜索的次数。以G. 729A为实验平台进行仿真, 仿真结果表明, 改进的算法在保证语音质量的情况下, 有效降低了ACELP码书搜索的复杂度。     

AMR-WB固定码本快速搜索新方法

在AMR-WB中,固定码本搜索是影响性能和复杂度的关键模块,约占总复杂度的40%。为了降低计算量,提出了一种码字分裂、子码字脉冲取代的高效码本搜索算法。该算法包括四步：一个初始码字分裂为两个或更多的子码字;每个子码字通过最不重要脉冲取代法进行更新;更新后的子码字合成一个候选的码字;比较初始码字和候选码字,选择优者作为最后的码字。实验表明,与传统方法相比,编码时间减小约16%。     

神经网络矢量量化技术在语音编码中的应用

讨论了在语音编码中,应用神经网络技术进行矢量量化的算法。神经网络矢量量化算法可以压缩码本维数,提高码本搜索速度,从而优化矢量量化的效果。将这种优化的矢量量化算法应用于语音编码中,能降低运算复杂度,提高编码质量。     

CS-ACELP语音编码算法的优化及其在DSP上的实现

针对CS-ACELP语音编码算法编码复杂度较高、DSP实时实现比较困难的问题,提出了一种可降低CS-ACELP语音编码算法复杂度的优化方法,分析了CS-ACELP语音编码算法原理,详细介绍了优化的CS-ACELP语音编码算法从固定码本搜索上降低算法复杂度的实现,并给出了在16位定点DSP芯片TMS320VC5402上实现CS-ACELP语音编解码方案的硬件及软件设计。实验结果表明,优化的CS-ACELP语音编码算法降低了运算复杂度,提高了运行速度,重建的语音符合标准的编解码要求。     

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.     

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).     

Proposed modifications in ITU-T G.729 8 kbps CS-ACELP speech codec and its overall performance analysis

This paper proposes modification in the transmission of excitation codevector and its non-zero pulse sign magnitude using “codebook partition and label assignment” approach, which in turn reduces the number of bits required to transmit it through the communication channel in legacy CS-ACELP 8 kbps speech codec. The proposed approach uses the excitation codebook structure of forward mode standard G.729E 11.8 kbps with two non-zero pulses per track which avoids the use of two algebraic codebook structure for forward mode as well as for backward mode of G.729E with least significant pulse replacement approach for finding optimized excitation codevector. Proposed modification in legacy 8 kbps CS-ACELP (80 bits/10 ms) speech codec actuates the bit rate of 10.6 kbps (106 bits/10 ms) with a better objective and subjective analysis in stark contrast with legacy 8 kbps CS-ACELP speech coder and also avoids the switching of codebook modes of standard 11.8 kbps (G.729E) CS-ACELP speech coder. This paper also aims to propose the reduction in the number of searches in the final codevector of excitation structure by considering initial codevector as a final codevector which improves the quality of the speech compared to the output speech quality of legacy G.729 CS-ACELP working at 8 kbps. Both legacy CS-ACELP 8 kbps speech codec and proposed CS-ACELP 10.6 kbps are implemented in MATLAB. Subjective and objective analysis are carried out on a proposed CS-ACELP 10.6 kbps speech codec in order to evaluate its performance and the results obtained are then cross- compared with the results of legacy CS-ACELP (8 kbps) using set of tables and graphs. It is evident from obtained results that both PESQ and MOS scores are quite comparable for each set of wave files even though bitrates are reduced. Consistency and efficiency of proposed algorithm is assured by calculating the population mean of 95% confidence interval based on obtained objective and subjective parameter results.     

直接矢量量化方法在语音编码算法中的应用

介绍了一种降低码书搜索复杂度的方法-直接矢量量化（DVQ）方法,将其应用于LD-CELP语音编码算法中的仿真译码器模块和码书搜索模块,用感觉加权逆滤波器代替仿真译码器模块中的综合滤波器,去除了码书搜索模块中冲激响应h（n）的运算。实验结果表明,利用直接矢量量化方法简化了码书搜索算法的复杂度,提高了码书搜索算法的效率,在运算时间方面比原始LD-CELP算法快3 s~5 s,同时保持了原编码算法合成语音的音质。