一种2.4kbps改进型MBELP编码

该文给出了一种改进的２．４ｋｂ／ｓ多带激励线性预测（ＩＭＢＥＬＰ）语音编码算法,与传统的ＭＢＥＬＰ算法相比,本算法在音质提取和清／浊音判决上采取了一些改进措施,使得合成语音质量有一定的提高。本文详细介绍了改进后的ＭＢＥＬＰ算法,并将其在基音提取和清／浊音判决的结果与传统的ＭＢＥＬＰ进行比较。     

16Kb／sLD—CELP语音编码器的DSP实时实现研究

论文研究了用单片Ｃ３０开发系统实时实现１６ｋｂ／ｓＬＤ－ＣＥＬＰ语音编码器的方法，给出了实验结果。     

低速率WI编码器中4~6bit基音量化算法研究

基音在语音编码中通常采用7bit无失真均匀量化。由于浊音段语音的基音普遍具有缓慢渐变的特点,为了更有效地去除前后帧基音之间存在的相关性,该文基于Eriksson和Kang提出的4bit基音量化算法,针对汉语语音进行研究,实现了一套4~6bit基音量化算法。该算法计算简单,无需码书存储。将此基音量化方案应用于WI模型和WI编码器,主观A/B听力测试结果表明,该方案在高效量化基音的同时保证了合成语音质量几乎没有损失,完全满足低速率WI编码器对量化基音的要求。     

一种改进的4.8kb／s码激励线性预测语音编码

本文介绍了码激励线性预测（ＣＥＬＰ）语音编码的基本原理，研究了一种制约随机激励的线性预测编码方案。它将随机激励码字进入合成滤波器的数量与自适应码本的性能指标联系起来，有效地减少了激励噪声对合成语音的影响。计算机模拟结果表明，这种方法在主观上改善了语音质量。     

一种改进的6.25kbit／sGSM半速率语音编码方案

在采用ＣＥＬＰ作为ＧＳＭ半速率编码方案的基础上，对ＣＥＬＰ的三个主要组成部分分别进行了改进，在ＬＳＰ在参数分析中提出了将求解和量化相结合的快速算法；在ＬＴＰ分析的分数延迟搜索中，采用逐步缩小搜索范围的方法；在随机码本搜索中，采用高斯重叠稀疏码本。这些措施大大降低了计算量，从而在保证合成语音质量的基础上能够采用单片ＤＳＰ实时实现整个ＣＥＬＰ编译码器。     

CELP短时预测器算法的研究

码激励线性预测能够在低比特率情况下实现较高质量的语音，在ＣＥＬＰ编码方案的实现中，确定短时预测器和长时预测器的系数是至关重要的，在简单介绍了ＣＥＬＰ的基本原理和激励码本的产生方法后，着重研究短时预测器的算法，采用Ｂｕｒｇ法实现短时预测器，并正对其参数量化编码，计算机模拟表明，采用上述方法可得到较好的合成语音质量。     

CELP语音编码与TMS320C54x

介绍了码激励线性预测（ＣＥＬＰ）编码的基本原理和新近推出的ＴＭＳ３２０Ｃ５４ｘ定点ＤＳＰ芯片,结合其指令特点探讨了ＴＭＳ３２０Ｃ５４ｘ在实现ＣＥＬＰ类语音编码方案的一些有效的编程方法。     

噪声信号对AMR语音编码器影响的研究

随着移动通信技术的发展,低速率语音编码器的应用越来越广泛,对于噪声对低速率语音编码器的影响也得到广泛的关注.文中研究了噪声信号对AMR语音编码器的影响,实验结果表明噪声信号对低编码速率的基音延迟影响大于对高编码速率语音编解码器的影响.     

4kb/s低速率声码器的新发展——PAPSI-CELP语音编码算法

４ｋｂ／ｓ低速率语音编码是近年来语音信号处理研究的重要课题,也是ＩＴＵ Ｔ下一步要标准化的重点。文章介绍了４ｋｂ／ｓ低速语音编码的最新进展,着重分析研究了一种称为相位自适应基音同步更新码激励线性预测（ＰＡＰＳＩ ＣＥＬＰ）的原理和结构,并与Ｇ７２９、Ｇ７２３１等算法进行了性能比较。     

CELP编码器中自适应码本快速搜索算法

ＣＥＬＰ以其高质量的合成语音及优良的抗噪声和多次转换性能，在移动通信，保密电话和语音中继等中低速率上获得了广泛应用，由于ＣＥＬＰ的运算量庞大，减少其运算一直是研究的热点，自适应码本是多数ＣＥＬＰ方案中的重要环节，本文对自适应码本的快速搜索算法进行了研究，并在端点递归法的基础上提出了一种双层搜索法，实验表明，此方法既保持了复原语音的高质量的特点，又明显地降低了运算量。     

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     

Robust LPC parameter estimation in standard CELP 4800 bit/s speechcoder

In order to decrease LPC spectral degradation in the USA FED STD 1016 4.8 kbit/s CELP speech coder, application of a robust LPC parameter estimation is proposed. Robust LPC methods, based on Huber's M-estimation theory and a heuristic sample-selective two-stage robust procedure, are considered. Comparative experimental analysis is carried out based on the cepstral distance, as an objective spectral measure. Presented experimental analyses justify the use of the robust LPC methods in the standard CELP 4800 bit/s speech coder, showing that the best results are obtained by using the combined sample-selective robust LPC procedure     

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     

A high-quality multirate real-time CELP coder

The design and implementation of a real-time CELP coder for mobile communication applications are discussed. To realize a single-chip implementation, several tradeoffs were made without compromising speech quality. In addition, techniques that make the coder more robust under a variety of channel conditions are discussed. The real-time coder can be operated at different bit rates (8, 6.8, 4.6 kb/s) by simply changing the frame update rates. The speech quality was evaluated through a formal listening test, and it was found that this coder compares favorably with other (standardized) coders operating at similar or higher rates     

基于LD-CElP和LZH的低比特率语音编码方法

在对ＬＤ－ＣＥＬＰ语音编码标准和无损数据压缩算法ＬＺＨ深入研究的基础上,提出了基于两者的一种语音混合压缩方法。实验结果表明,采用这种混合压缩方法可以将语音码率从６４ｋｂｐｓ降到９．６ｋｂｐｓ左右,而且运算时间和处理延迟没有明显的增加。主观测试表明,恢复后的语音保持了自然度和可懂度,其主观质量是令人满意的。     

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).<>     

一种可变速率的LD—CELP编码的研究

设计了一种可变速率的低时延、码激励线性预测编码（ＬＤ－ＣＥＬＰ）的方案，它是通过修改码本来实现的。该方案工作在１１．２ｋｂｉｔ／ｓ。对其做了计算机仿真，并与１６ｋｂｉｔ／ｓ的ＬＤ－ＣＥＬＰ算法在信经（ＳＮＲ）、波形等方面进行了对比，仿真结果表明效果良好。     

Parameter replacement for CELP coded speech in land mobile radio

The contribution of this paper is in applying parameter replacement techniques to speech that is compressed by the Federal Standard 1016 CELP speech coder, protected by Reed–Solomon codes, and transmitted over a wireless channel. The parameter replacement results in significant improvement in speech quality without any increase in bit rate. This revised version was published online in June 2006 with corrections to the Cover Date.     

Techniques for improving the performance of CELP-type speech coders

Techniques for improving the performance of CELP (code excited linear prediction)-type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function, which enhances the perceptual quality of the processed speech, is introduced. The combination of harmonic noise weighting and subsample pitch lag resolution significantly improves the coder performance for voiced speech. Strategies for reducing the speech coder's data rate, while maintaining speech quality, are presented. These include a method for efficient encoding of the long-term predictor lags, utilization of multiple gain vector quantizers, and a multimode definition of the speech coder frame. A 5.9-kb/s VSELP speech coder that incorporates these features is described. Complexity reduction techniques which allow the coder to be implemented using a single fixed-point DSP (digital signal processor) are discussed     

小波激励线性预测编码（WELP）：一种新的低速语音编码方法

在低速语音编译码系统中，常采用码本激励线性预测编码ＣＥＬＰ，其中随机码本的码本结构及应的索算法直接影响着语音编译码系统的语音质量和实时实现中的运算量。