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     

Fast codebook search algorithm for unconstrained vectorquantisation

Vector quantisation (VQ) is a well known data compression technique which maps an ordered set of real numbers into a single integer. However, it is difficult to achieve accurate compression with the use of unconstrained Voronoi VQ when the codebook level and vector dimensionality are very large, due to the extremely high real-time computational complexity involved in full codebook search. To overcome this difficulty, a classified pre-selection method is proposed. Compared to the conventional full search method, the algorithm reduces the computational complexity involved in the code vector selection procedure by 70%~90% with almost no loss in coder performance, at the cost of only a slight increase in the storage requirement     

Fast codeword search algorithm for real-time codebook generation inadaptive VQ

Adaptive vector quantisation is used in image sequence coding where the code-book is updated continuously to keep track with the changing source statistics. Hence, for real-time video coding applications, both the processes of quantising the input vectors and updating the codebook are required to be fast. Since the nearest codeword search is involved in both these processes, a fast codeword search algorithm can make the coding process time efficient. The authors describe a proposed codeword search algorithm with reduced search space. The algorithm uses the mean value and the sum of the absolute differences as the two criteria to reject unlikely codewords, thereby saving a great deal of computational time, while introducing no more distortion than the conventional full search algorithm. Simulation results obtained confirm the effectiveness of the proposed algorithm in terms of computational complexity     

Fast planar-oriented ripple search algorithm for hyperspace VQ codebook.

This paper presents a fast codebook search method for improving the quantization complexity of full-search vector quantization (VQ). The proposed method is built on the planar Voronoi diagram to label a ripple search domain. Then, the appropriate codeword can easily be found just by searching the local region instead of global exploration. In order to take a step further and obtain the close result full-search VQ would, we equip the proposed method with a duplication mechanism that helps to bring down the possible quantizing distortion to its lowest level. According to the experimental results, the proposed method is indeed capable of providing better outcome at a faster quantization speed than the existing partial-search methods. Moreover, the proposed method only requires a little extra storage for duplication.     

低码率视频压缩技术的块效应消除方法

低友率视频压缩技术是多媒体通信的关键技术。基于块编码的低码率视频压缩技术中,块效应现象成为比较突出的问题。本文讨论了低码率视频压缩中消除块效应的方法,提出了基于约束复原原理的一种具体块效应消除方法,并根据计算机软件实现的分析,得出了结论采用约束复原的块应消除方法可以同时提高客观ＰＳＮＲ值和改善块应现象。     

Adaptive postprocessing algorithms for low bit rate video signals

A new adaptive postprocessing algorithm to enhance the quality of a noisy video sequence is presented. The algorithm recognizes that the visibility of noise depends on local signal characteristics. It therefore classifies the video signal into different classes and uses separate nonlinear filters matched to each class. The most general version of the algorithm employs motion-compensated frame averaging to improve picture quality in a first stage. A classification algorithm subsequently divides subblocks of pixels in the averaged frame into four classes: edge, smooth, nonsmooth with motion and nonsmooth without motion. Spatial algorithms that perform multilevel median filtering, double median filtering, and median filtering are used for pixels belonging to edge, smooth, and nonsmooth with motion categories. Pixels in the nonsmooth, unmoving category are left unfiltered to preserve corresponding image texture. In a simpler version of this four-class system, the motion cues and motion-compensated frame averaging are eliminated, and the purely spatial filtering is based on a three-class algorithm. When used at the output of a 3-D subband coder at 384 kbps, the spatial postfilter was shown to provide a consistent gain in subjectively evaluated picture quality. Twenty-five viewers participated in an experiment involving three coded sequences. In a pairwise comparison of postfiltered and unfiltered sequences, the postfiltered version was judged to be better in 63 out of 75 instances.     

H.263甚低码率视频编码标准的码率控制方法

主要提出ＩＴＵ组织的Ｈ．２６３甚低码率视频压缩编码标准输出码率的控制方案。着重分析了一种通过调整宏观量化器的量化参数来控制Ｈ．２６３编码器输出码率的方法,它能满足在ＰＳＴＮ网上低时延中的码率输出要求。     

Fast clustering process for vector quantisation codebook design

A fast clustering process is introduced to the generalised Lloyd algorithm for vector quantisation codebook design. Simulations show that the proposed codebook design method achieves significant saving in computational time and similar qualitative performance to that of the conventional full-search approach     

Analysis of low bit rate image transform coding

Calculations based on high-resolution quantizations prove that the distortion rate D(R¯) of an image transform coding is proportional to 2^-2R when R¯ is large enough. In wavelet and block cosine bases, we show that if R¯<1 bit/pixel, then D(R¯) varies like R¯^1-2γ, where γ remains of the order of 1 for most natural images. The improved performance of embedded codings in wavelet bases is analyzed. At low bit rates, we show that the compression performance of an orthonormal basis depends mostly on its ability to approximate images with a few nonzero vectors     

极低误码下的误码率估计问题

在极低误码率的情况下由于测得的误码点很少,由实测的点来直接估计误码率这一方法变得不可靠。本文提出一种方法,通过牺牲测量值的精度来换取结果的可靠性,即用误码率的上界来描述误码特性。本文还给出了误码率的上界、实际测量得到了误码点数和表示误码率上界可靠性的门限值之间的定量关系。     

CELP based mixed-source model for very low bit rate speech coding

A CELP based mixed-source model is described. It uses a mixed excitation which combines a lowpass-filtered adaptive source and a highpass-filtered stochastic source. In addition, one more stochastic source is newly employed for more natural sounding speech. In informal listening tests, the proposed model at 3 kbit/s shows very good performance both in speech quality and intelligibility.<>     

低码率视频的小波域软阈值去方块滤波算法的研究

本文针对低码率条件下基于子块DCT的视频压缩产生的方块效应,提出了一种小波域上进行软阈值滤波的改进算法。新算法根据图像的平坦区和单调变化区等非纹理区易于通过平滑滤波去噪的性质,以非纹理区作为参考对象计算阈值,并且采用虚拟滤波的思想简化了阈值的计算,同时根据低码率压缩视频的特点修改了在低层子带的软阈值处理过程。对测试序列Susan的实验结果表明,新的算法不仅获得很好的主观去方块效果,而且可以对帧内和帧间编码模式平均分别获得约1.0dB和0.28dB的客观质量(PSNR)的改善,因此新的算法适用于可视电话等低码率视频的应用。     

A layered coding system for very low bit rate video coding

Known coding techniques for transmitting moving images at very low bit rates are explained by the source models on which these coding techniques are based. It is shown that with motion-compensated hybrid coding, object-based analysis-synthesis coding, knowledge-based coding and semantic coding, there is a consistent development of source models. In consequence these coding techniques can be combined in a layered coding system. From experimental results obtained for object-based analysis-synthesis, coding estimates for the coding efficiency of such a layered coding system are derived using head and shoulder video telephone test sequences. It is shown that an additional compression factor of about 3 can be expected with such a complex layered coding system, when compared to block-based hybrid coding.     

混合激励线性预测低速率语音编码研究

为了满足数字通信及其他商业应用的需求,语音压缩编码技术得到迅速发展.近年来主流的低速率语音编码方案主要基于LPC-10,混合激励线性预测(MELP),多带激励编码(MBE),正弦变换编码(SCI),波形内插编码(WI),大多都工作在2.4 kb/s速率下.作为一种重要的低速率语音编码算法,MELP算法对LPC-10编码方案进行大量改进,引入混合激励,非周期脉冲,残差付氏幅度谱,脉冲散布和自适应谱滤波5个特征.实验结果表明,该混合激励线性预测编码在2.4 kb/s上得到了更好的合成语音,并使得合成语音能更好地拟合自然语音.     

瞬态时频内插极低速率图像压缩编码

结合图像信号瞬变、渐变和缓变的特点,从图像信号瞬态非平稳分析与重构角度出发,提出了具有瞬态分析、参数内插和时频重构特点的瞬态时频内插极低速率图像压缩编码方案。瞬态时频分析理论的应用,为今后图像信号处理研究提供了一种新的思路。     

MELP低比特率数字语音编码技术研究

主要介绍了一种新的低比特率MELP (MixedExcitationLinearPrediction)混合激励线性预测语音编码技术 ,其中着重分析了该编码算法所采用的几项关键技术。给出了采用MELP压缩编码算法后的输出比特流在各个参数上的比特分配表 ,并通过计算机仿真 ,对MELP合成语音与原始语音的质量做了比较 ,最后就MELP语音编码技术与现今其它几种不同的低速率语音编码技术的合成语音质量在DRT、DAM及MOS得分三个方面做了比较。     

混合激励线性预测低速率语音编码研究

为了满足数字通信及其他商业应用的需求,语音压缩编码技术得到迅速发展。近年来主流的低速率语音编码方案主要基于LPC-10,混合激励线性预测（MELP）,多带激励编码（MBE）,正弦变换编码（SCI）,波形内插编码（WI）。大多都工作在2．4kb／s速率下。作为一种重要的低速率语音编码算法。MELP算法对LPC-10编码方案进行大量改进,引入混合激励,非周期脉冲,残差付氏幅度谱,脉冲散布和自适应谱滤波5个特征。实验结果表明,该混合激励线性预测编码在2．4kb／s上得到了更好的合成语音,并使得合成语音能更好地拟合自然语音。     

Artifact reduction in low bit rate DCT-based image compression

This correspondence presents a scheme for artifact reduction of low bit rate discrete-cosine-transform-compressed (DCT-compressed) images. First, the DC coefficients are calibrated using gradient continuity constraints. Then, an improved Huber-Markov-random-field-based (HMRF-based) smoothing is applied. The constrained optimization is implemented by the iterative conditional mode (ICM). Final reconstructions of typical images with improvements in both visual quality and peak signal-to-noise ratio (PSNR) are also shown.     

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     

Efficient rate control scheme for low bit rate H.264/AVC video coding

This article presents an efficient rate control scheme for H.264/AVC video coding in low bit rate environment. In the proposed scheme, an improved rate-distortion (RD) model by both analytical and empirical approaches is developed. It involves an enhanced mean absolute difference estimating method and a more rate-robust distortion model. Based on this RD model, an efficient macroblock-layer rate control scheme for H.264/AVC video coding is proposed. Experimental results show that this model encodes video sequences with higher peak signal-to-noise ratio gains and generates bit stream closer to the target rate.