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     

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     

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 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FS1016语音压缩编码标准的编解码仿真研究

FS1016 4 800bit/s是基于码激励线性预测技术的语音编码标准,在实现高质量的窄带语音通信方面有着广泛的应用。介绍了FS1016 4 800bit/s的编码标准并讨论了CELP的基本原理,提出了基于FS1016 4 800bit/s编码标准的编、解码器仿真实现方案,并进行了仿真试验。仿真结果表明,该方案具有很好的可行性,为以后在DSP系统实现该编码标准提供了很好的基础。     

Adaptive code excited predictive coding

A novel way to use the code excited linear prediction (CELP) concept that decreases the processing load while keeping the same speech quality is discussed. Rather than performing individual weighting of each candidate sequence, a global implementation of the perceptual weighting function at the codebook level is proposed. As a result, the analysis-by-synthesis procedure does not require the processing of all the candidate sequences through the synthesis and weighting filters; the complexity requirement of the algorithm is therefore much reduced. The concept is carried out with an adaptive codebook. Two fixed-point implementations of the adaptive CELP (ACELP) algorithm are reported: a 7.2 kb/s block coder (7 MIPS), and a 12 kb/s low-delay coder (11 MIPS). Both coders have been rated to provide the same quality as the 13 kb/s block coder adopted by the GSM for the European cellular telephone     

Variable Frame Rate Transmission: A Review of Methodology and Application to Narrow-Band LPC Speech Coding

We review the variable frame rate (VFR) transmission methodology that we developed, implemented, and tested during the period 1973-1978 for efficiently transmitting LPC vocoder parameters extracted from the input speech at a fixed frame rate. In the VFR method, parameters are transmitted only when their values have changed sufficiently over the interval since their preceding transmission. We explored two distinct approaches to automatic implementation of the VFR method. The first approach bases the transmission decisions on comparisons of the parameter values of the present frame and the last transmitted frame. The second approach, which is based on a functional perceptual model of speech, compares the parameter values of all the frames that lie in the interval between the present frame and the last transmitted frame against a linear model of parameter variation over that interval. The application of VFR transmission to the design of narrow-band LPC speech coders with average bit rates of 2000-2400 bits/s is also considered. The transmission decisions are made separately for the three sets of LPC parameters, pitch, gain, and spectral parameters, using separate VFR schemes. A formal subjective spccch quality test of six selected LPC coders is described, and the results are presented and analyzed in detail. It is shown that a 2075 bit/s VFR coder produces speech quality equal to or better than that of a 5700 bit/s fixed frame rate coder.     

基于MMX指令实现G.728定点算法优化

ITU-T建议是国际电信联盟于1992年制定的比特率为16kb/s的低延时CELP类语音编码器.G.728AnnexG是其补充定点算法建议.该文首先介绍了G.728定点算法的特点和Intel MMX技术,然后详细讨论了用MMX指令对G.728的LD-CELP定点算法实现优化一些关键技术,最后给出了优化前后算法速度的对比.     

基于小波变换的2.4kbit/s波形内插语音编码算法

基于双正交小波滤波器组对波形内插编码中提取的特征波进行多级分解与重构，提出了一种基于小波变换（WT）的2．4kbit／s特征波形内插（CWI）语音编码算法。编码端去除了特征波对齐运算，并对幅度谱进行多级分解，相位谱不传输，鉴于小波变换对信号的压缩特性，仅传输对人耳感知起主要贡献的最后一级特征波幅度谱；解码端对各尺度空间采用单独重建的方法，相位信息在重构的末级与幅度谱结合，并由浊音度标志选择固定或随机相位。此外，根据语音信号的时变特性，由基于子帧的浊音度标志选择需要传输的幅度谱及量化模式。主观R-A／B测试表明，这种基于小波变换的2．4kbit／s编码算法的合成语音质量明显优于标准的2．4kbit／s的MELP编码器及FS1016的4．8kbit／sCELP编码器，亦优于3．8kbit／s的传统CWI编码框架下的合成语音效果。     

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。     

Strategies to improve the performance of very low bit rate speech coders and application to a variable rate 1.2 kb/s codec

This paper presents several strategies to improve the performance of very low bit rate speech coders and describes a speech codec that incorporates these strategies and operates at an average bit rate of 1.2 kb/s. The encoding algorithm is based on several improvements in a mixed multiband excitation (MMBE) linear predictive coding (LPC) structure. A switched-predictive vector quantiser technique that outperforms previously reported schemes is adopted to encode the LSF parameters. Spectral and sound specific low rate models are used in order to achieve high quality speech at low rates. An MMBE approach with three sub-bands is employed to encode voiced frames, while fricatives and stops modelling and synthesis techniques are used for unvoiced frames. This strategy is shown to provide good quality synthesised speech, at a bit rate of only 0.4 kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, spectral envelope restoration combined with noise reduction (SERNR) postfilter is used. The contributions of the techniques described in this paper are separately assessed and then combined in the design of a low bit rate codec that is evaluated against the North American Mixed Excitation Linear Prediction (MELP) coder. The performance assessment is carried out in terms of the spectral distortion of LSF quantisation, mean opinion score (MOS), A/B comparison tests and the ITU-T P.862 perceptual evaluation of speech quality (PESQ) standard. Assessment results show that the improved methods for LSF quantisation, sound specific modelling and synthesis and the new postfiltering approach can significantly outperform previously reported techniques. Further results also indicate that a system combining the proposed improvements and operating at 1.2 kb/s, is comparable (slightly outperforming) a MELP coder operating at 2.4 kb/s. For tandem connection situations, the proposed system is clearly superior to the MELP coder.     

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     

Improved Populating Code-Excited Linear Predictive Coder

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

基于TMS320DM6446的MELP/CELP混合编码方法研究与实现

在讨论TMS320DM6446结构的基础上,讨论了混合激励线性预测(MELP)的算法模型,并详细研究了MELP的几种关键技术.在MELP基础上,提出了MELP/CELP两种适用语音帧类型互补的编码方案相结合的语音编码方法,在周期性不强的语音帧采用CELP来弥补MELP的不足,并给出了这种编码器的结构和量化方案.使用TMS320DM6446实现发现可以获得与32kbit/s编码速率的ADPCM相近的语音质量.     

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。     

一种改进的MELP语音编码方法

目前2.4kbps的混合激励线性预测（MELP）语音编码方法已经被确定为美国新的联邦语音编码标准。本文提出了一种改进的MELP语音编码方法，利用滤波器相似度和基于LPC系数分类的矢量量化技术，可以把MELP的码率降到1.7kbps以下，仍有较好的合成语音质量。     

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。     

用分数延迟改进基音预测的CELP编码方案

在ＣＥＬＰ编码器中，通常用延迟为抽样间隔整数倍的长项预测器表征浊音语音的准周期性，然而在低比特率，这种限制降低了编码器的性能。本文在介绍了ＣＥＬＰ编码器原理及激励码本构成后，重点研究了一种新型的基音预测方法；分数延迟基音预测，计算机模拟结果表明，这种方法能对浊音进行更准确的表达，尤其对女性讲话者明显改善了语音质量。     

一种基于低比特率语音算法的芯片设计与分析

介绍了一种低比特率语音算法,它是在MBE编码算法的基础上,利用线性预测谱代替了MBE中的傅里叶变换谱对MBE编码算法进行了改进,从而在保持语音质量的情况下使比特率更低,达到2 kbit/s.同时阐述了基于该算法的系统芯片设计,并对该算法中的基音周期参数、V/U等参数的提取进行详细的推导分析.最后对本算法与常用的码激励线性预测编码(CELP)的算法进行比较,并分析其中的原因.     

Natural quality two-band LPC coding of speech at 880 bit/s withframe interpolation

A novel frame interpolation technique for two-band linear predictive coding (LPC) vocoders is proposed for maintaining natural speech quality at bit rates below 1 kbit/s. Experimental results show that the speech quality of the proposed vocoder is quite natural at bit rates 880 bit/s and comparable to that of 4.8 kbit/s CELP