Efficient codebook search procedure for vector-sum excited linearpredictive coding of speech

An efficient codebook search method for the EIA/TIA IS-54 vector-sum excited linear predictive (VSELP) speech coder is described. The method uses a two-stage search procedure. In the first stage, diagonal approximation of the correlation matrix of the filtered basis vectors is assumed and a simple sign detection procedure is used to identify a codeword which is close to the optimum codeword. In the second stage, a refinement search is carried out on those codewords which have a Hamming distance of one from the codeword obtained in the first stage. The new search procedure has a complexity only proportional to the bit rate which is much faster than the Gray code search employed in the IS-54 VSELP coder. Simulation results show that the SNR obtained using the proposed fast procedure is the same as that obtained in the standard VSELP coder     

On reducing computational complexity of codebook search in CELPcoding

A computationally efficient codebook search method in code-excited linear prediction is proposed. The method can reduce the computational complexity by almost one half compared to the frequency-domain codebook search method that is currently regarded as the fastest search method, while giving almost the same quality of speech. This reduction is possible as a result of the simultaneous use of frequency-domain search and code vector sparsity     

Improved Populating Code-Excited Linear Predictive Coder

1Introduction,TheCode--ExcitedLinearPredictive(CELP)[13coderprovidedgoodqualityspeechatmediumandlowbitrates,butthisqualityspeechwasatthecostofverycomputationalcomplexity.Recently,therealtimeimplementationoftheCELPcodersonalowpricedigitalsignalprocessorchi…     

G.729自适应码书搜索算法的研究

G.729语音编码算法复杂,很大程度上要归因于码书搜索算法。为了降低码书搜索复杂度,G.729的简化版G.729A采用了自适应码书的偶样点开环基音搜索,使得编码复杂度大为降低,不过编码仍要花费很多的时间。通过对G.729码书搜索算法的研究,提出了对自适应码书的改进。改进了自适应码书搜索的G.729编码语音质量不低于G.729A的语音质量,但自适应搜索复杂度大大降低。     

一种基于G.729的CS-ACELP新算法

在G．729的基本框架下,码本搜索采用次最优部分码本快速搜索法,知觉加权滤波器使用基于声学心理模型的知觉加权滤波器,使得8Kb／s共扼结构代数码激励线性预测语音编码在不降低语音质量的情况下降低计算复杂度。次最优部分码本快速搜索方法虽然降低搜索准确度,但是大大减小了搜索时的数据处理量;声学心理滤波器考虑人耳对不同频率信号的不同敏感度,因而能获得更好的主观音质效果。仿真结果表明,该算法复杂度降低,并取得满意的合成语音质量。     

A New Method of Designing Waveform Codebook

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一种基于优先树脉冲取代的固定码本搜索方法

分析了G.722.2中的固定码本结构及其搜索方法,结合深度优先树与脉冲取代法各自的优点,提出了一种新的方法——基于优先树脉冲取代的固定码本搜索方法,该方法在基本保持语音质量的前提下,可使固定码本搜索计算复杂度降低,从而使G.722.2的编码运行时间进一步减少。实验结果表明,与G.722.2中采用的深度优先树搜索方法相比,基于优先树脉冲取代的固定码本搜索方法可以在不影响语音质量的前提下,使固定码本搜索复杂度降低85%以上。     

一种新的语音LSF矢量量化码书优化算法

LSF(线谱频率)码书的性能对合成语音质量有着重要影响.经典的LBG算法容易陷入局部最优,而目前的一些码书进化算法搜索空间较大、搜索效率不明显.本文提出了一种新型的基于对LSF矢量空间进行拉伸变化的混合进化码书优化算法.该算法编码空间与矢量同维,相对较小,便于优化操作.算法中引入EP中的变异操作对PSO位置、速度矢量进行控制,以提高优化搜索算法的效率.实验结果表明,本文算法有效地改善了码书性能.     

Pulsed residual excited linear prediction

Linear predictive coding of speech has been widely used at 16 kb/s in the form of adaptive predictive coding (APC) down to 4.8 kb/s in the form of code-excited linear prediction (CELP). Since its invention in 1984 there have been many variations of CELP which differ mainly in the way the final excitation signal (codebook) is produced and quantised. These variations either produce better speech quality or lower complexity. Three new excitation types, all of which are based on a pulsed residual, are proposed. The new pulsed residual excitations improve the speech quality significantly. In addition a novel mathematically equivalent codebook search method which reduces the search complexity significantly is described     

Fast search techniques for use in vector quantisation and related pattern matching applications

Two techniques for the design and fast search of a vector quantiser codebook are proposed. These classify speech data vectors according to (i) the sign and (ii) the slope between successive samples. With both techniques the performance obtained is superior to that obtained from a conventional gain/shape vector quantiser and comparable to that of a full search vector quantiser. A major attraction of both approaches is that they significantly reduce the number of computations required for the codebook search.     

Design and evaluation of optimal cepstral lifters for accessingarticulatory codebooks

It is noted that of great importance to the success of the articulatory approach to speech coding is the use of a good distortion measure between a given speech signal and the entries in a stored codebook of impulse responses and corresponding vocal-track shapes (articulatory codebook). One promising distortion measure is the weighted cepstral distortion. Since the impulse responses in the articulatory codebook do not include glottal characteristics, the authors derive optimal weighting functions (cepstral lifters) to reduce the influence of a varying glottal source on the cepstral distortion measure. This is done by examining the ensemble of cepstral coefficients of speech produced by an articulatory speech synthesizer that also includes a vocal-cord model. The obtained cepstral lifters are optimal for the given ensemble of cepstral coefficients and for given constraints on the weighting function. They are different for cepstral coefficients derived from the power spectrum (FFT cepstra) and for those derived from LPC (linear predictive coding) coefficients (LPC cepstra). The performances of the obtained cepstral lifters are compared in an articulatory codebook search     

高质量的4 kb/s散布脉冲CELP语音编码算法

本文提出了一种散布脉冲CELP(DP-CELP)语音编码算法,激励矢量由特殊结构的代数码书与固定形式的散布脉冲的卷积获得,这种激励源有效地改善了重建语音质量,但未增加代数码书搜索的复杂度.非正式的主观听力测试表明,这种4 kb/s DP-CELP语音编码算法的合成语音质量非常接近G.723.1中6.3 kb/s语音编码器.     

CELP-Based Implementation of the GSM Half-Rate Speech Codes

1IntroductionThecurrentachvityinspeeChcodinginEuropeisfocusedonselectinganewpan-Europestalldardforadigitalmobilesystemknownasthehalf-ratecodingsystem.TheCELP-basedconfigurationhasbecomeoneofthecandidatesforit.TheshortcomingofthefundamentalCELPisitSgreatstorageandcomputation,andthefixedcodebooksearchcoststhemost.Inthispaper,wepresentanoverlappingcenterclippingcodebookandintroducefastsearchalgoritlllnsofconvoluhon,auto-correlationandcross-correlationonthebasisofthenewcodebook.Byuseofthisim…     

自组织特征映射神经网络的聚类特性在语音矢量量化快速搜索中的应用

本文讨论了自组织特征映射人工神经网络在语音矢量量化中应用时所涉及的两个重要问题,即码本训练和码本搜索的问题。根据语音反射系数的特点,提出了训练中初始码本的选择原则和实用训练算法。利用特征映射网络的聚类特性和语音相邻帧间的相关性,提出了码本搜索的两种快速算法——子域搜索法和邻域搜索法。大量实验结果表明,这两种快速搜索方法相结合,搜索时间减少为常用的LBG全搜索算法的1/4或1/10,同时保持精度不下降。本文提出的方法已在一种极低数据率的声码器中得到成功应用。     

Binary code excited linear prediction (BCELP): new approach to CELP coding of speech without codebooks

A new analysis-by-synthesis speech coding approach able to produce good quality speech in the vicinity of 4.8 kbit/s is presented. The new approach produces the same speech quality as obtained by CELP codecs without needing any excitation codebook storage. The new coder employs a very simple excitation search procedure and processes an inherent robustness against channel errors. The approach is based on the ternary code excitation CELP introduced previously (see P. Desantis et al. 1986).<>     

A new LPC interpolation technique for CELP coders

In low rate code-excited linear predictive (CELP) coders, the LPC spectral information is usually quantized and transmitted on a frame-by-frame basis about every 20 to 30 msec. The quality of speech reproduced by a CELP coder can be improved by making spectral transitions as smooth and continuous as possible. One way in which this can be accomplished without increasing the transmission bit rate is to interpolate the LPC spectral parameters between adjacent extraction frames. This, however, usually leads to a dramatic increase in the computations required for the codebook search. The paper presents a new LPC interpolation technique, based on interpolating the impulse response of the LPC synthesis filter. It demonstrates that this method offers a significant complexity reduction for the codebook search over other typical interpolation schemes. Furthermore, the experiments show that the coder using the impulse response for interpolation produces the same speech quality as the coder using the LSP parameters for interpolation, and both these parameter sets are superior to other LPC representations for interpolation     

Efficient algebraic code-excited linear predictive codebook search

To reduce the computational complexity of algebraic code-excited linear prediction (ACELP) coders, an efficient codebook search mechanism based on a simplified correlation matrix (SCM) of the vocal impulse response is proposed. In the proposed approach, the statistical characteristics of the vocal impulse response are identified such that only a small proportion of the total number of correlation coefficients in the correlation matrix need be calculated before the ACELP search procedure is carried out. Furthermore, the proposed joint scheme, by combining the SCM method and a pulse position prediction scheme, not only decreases the arithmetic complexity in the pre-computing autocorrelation matrix but also reduces the number of pulse position combinations. The simulation and experimental results show that the proposed method provides an effective reduction in the computational load of the ACELP codebook search procedure with no discernible degradation of the speech quality     

Design of a pitch synchronous innovation CELP coder for mobilecommunications

This paper describes the design of a speech coder called pitch synchronous innovation CELP (PSI-CELP) for low hit-rate mobile communications. PSI-CELP is based on CELP, but has more adaptive excitation structures. In voiced frames, instead of conventional random excitation vectors, PSI-CELP converts even the random excitation vectors to have pitch periodicity by repeating stored random vectors as well as by using an adaptive codebook, in silent, unvoiced, and transient frames, the coder stops using the adaptive codebook and switches to fixed random codebooks. The PSI-CELP coder also implements novel structures and techniques: an FIR-type perceptual weighting filter using unquantized LPC parameters, a random codebook with a conjugate structure trained to be robust against channel errors, codebook search with delayed decision, a gain quantization with sloped amplitude, and a moving average prediction coding of LSP parameters, Our speech coder is implemented by DSP chips. Its coded speech quality at 3.6 kb/s with 2.0 kb/s redundancy is comparable to that of the Japanese full-rate VSELP coder at 6.7 kb/s with 4.5 kb/s redundancy. The basic structure of this PSI-CELP coder has been chosen as the Japanese half-rate speech codec for digital cellular telecommunications     

An Embedded ACELP Speech Coding Based on the AMR‐WB Codec

This letter proposes a new embedded speech coding structure based on the Adaptive Multi‐Rate Wideband (AMR‐WB) standard codec. The proposed coding scheme consists of three different bitrates where the two lower bitrates are embedded into the highest one. The embedded bitstream was achieved by modifying the algebraic codebook search procedure adopted for the AMR‐WB codec. The proposed method provides the advantage of scalability due to the embedded bitstream, while it inevitably requires some additional computational complexity for obtaining two different code vectors of the higher bitrate modes. Compared to the AMR‐WB codec, the embedded coder shows improved speech qualities for two higher bitrate modes with a slightly increased bitrate caused by the decreased coding efficiency of the algebraic codebook.     

基于双耳线索编码原理的语音增强方法

借助双耳线索编码原理,通过构建一个语音和噪声的双耳线索先验码书,本文提出一种单通道语音增强方法.首先,该算法将语音和噪声的双耳线索作为语音和噪声的先验知识,在线下被训练成为先验码书.之后,在线上通过加权码书映射（Weighted CodeBook Mapping,WCBM）算法估计纯净线索参数,最后,利用双耳线索编码原理增强含噪语音.此外,本文采用深度神经网络,即堆栈式自编码器（Stacked Auto-Encoders,SAE）代替WCBM算法估计纯净线索参数,提出了基于深度神经网络的双耳线索语音增强算法.进一步提高了增强算法的性能.客观测试结果表明,本文所提方法优于参考算法.