基于局部余弦变换的低比特变速率语音编码算法研究

提出将局部余弦变换(LCT)算法应用于语音编码中,系统设计了一个平均比特率近1.6kbit/s的低比特变速率语音编码器。在变比特率编码器设计中采用SVM算法进行VAD检测。激活语音帧的语音模式采用GSM半速率编码中的划分方法,但将其中的强浊音模式和中浊音模式合并为一个中强浊音模式。对各类语音模式和无声帧(背景噪声)的局部余弦变换系数采用分维矢量量化算法进行量化,码书设计采用LGB算法。编码中的码书搜索采用树形快速搜索算法。通过主观非正式听力测试表明设计的变比特率编码器编码的重建语音MOS约为3.15,与比特率为2.4kbit/s美国联邦声码器标准MELP的重建语音相当,具有较强的顽健性,适合于对存在各种环境噪声的语音进行编码。     

低码率图像压缩中应用ROI的改进算法

部分重要比特平面平移法是目前在低码率压缩条件下可实用的ROI编码方法,针对该算法造成ROI高低频成分结构变化而产生失真的问题,提出了一种改进的算法-按分辨率级部分重要比特平面平移法。与原算法相比,新算法能同时提高ROI和背景区域的图像质量。在码率为0.8bpp时新算法将ROI和背景的PSNR（峰值信噪比）最高提高了1.34dB和0.72dB。     

Subband coding of image sequences at low bit rates

In this paper a low bit rate subband coding scheme for image sequences is described. Typically, the scheme is based on temporal DPCM in combination with an intraframe subband coder. In contrast to previous work, however, the subbands are divided into blocks onto which conditional replenishment is applied, while a bit allocation algorithm divides the bits among the blocks assigned for replenishment. A solution is given for the ‘dirty window’ effect by setting blocks to zero that were assigned to be replenished but received no bits. The effect of motion compensation and the extension to color images are discussed as well. Finally, several image sequence coding results are given for a bit rate of 300 kbit/s.     

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2．4kb／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。     

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.     

Block transform image coding by multistage vector quantization withoptimal bit allocation

A multistage vector quantization with optimal bit allocation (MVQ-OBA) in the transform domain is presented. A set of bit allocation planes is first obtained by slicing a (scalar) optimal bit allocation map where the number of bits assigned to each coefficient is proportional to the coefficient variance. The set of bit allocation planes determines the coefficients to be used and the codebook size at each stage. The vector dimensionalities are restricted to small values and relatively small codebooks are used, thus reducing both the overhead required for transmitting the codebooks and the complexity in codebook design. The computer simulation results demonstrate that MVQ-OBA is competitive with many other transform coding techniques including variable length transform coding. MVQ-OBA is well suited for progressive transmission     

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。     

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。     

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。     

基于JPEG2000的低速率ROI改进算法的研究

在低码率图像压缩中,现有的ROI算法都取得了良好的效果,但是背景图像质量却不尽如人意。文中根据图像自身的特点,提出了部分子带的部分位平面平移算法,在保持ROI区域图像质量的前提下,对背景图像质量有了一定的改善。这种算法是部分位平面平移(PSBShift)的改进算法,继承了PSB算法的诸多优点,同时改进了图像的整体质量。     

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.     

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.<>     

Hadamard transform image coding

The introduction of the fast Fourier transform algorithm has led to the development of the Fourier transform image coding technique whereby the two-dimensional Fourier transform of an image is transmitted over a channel rather than the image itself. This devlopement has further led to a related image coding technique in which an image is transformed by a Hadamard matrix operator. The Hadamard matrix is a square array of plus and minus ones whose rows and columns are orthogonal to one another. A high-speed computational algorithm, similar to the fast Fourier transform algorithm, which performs the Hadamard transformation has been developed. Since only real number additions and subtractions are required with the Hadamard transform, an order of magnitude speed advantage is possible compared to the complex number Fourier transform. Transmitting the Hadamard transform of an image rather than the spatial representation of the image provides a potential toleration to channel errors and the possibility of reduced bandwidth transmission.     

Preserving step edges in low bit rate progressive image compression

With the growing importance of low-bandwidth applications, such as wireless access to the Internet, images are often sent or received at low bit rates. At these bit rates, they suffer from significant distortion and artifacts, making it difficult for those viewing the images to understand them. We present two progressive compression algorithms that focus on preserving the clarity of important image features, such as edges, at compression ratios of 80:1 and more. Both algorithms capture and encode the locations of important edges in the images. The first algorithm then transmits a standard SPIHT (set partitioning in hierarchical trees) bit stream, and at the decoder applies a nonlinear edge-enhancement procedure to improve the clarity of the encoded edges. The second approach uses a modified wavelet transform to "remove" the edges, and encodes the remaining texture information using SPIHT. With both approaches, features in the images that may be important for recognition are well preserved, even at low bit rates.     

Very low bit rate speech coding using a diphone-based recognitionand synthesis approach

High compression rates of speech signals may be achieved by coding schemes based on relevant linguistic segments. A system is described that relies on a diphone recogniser as the coder and on a speech synthesiser reproducing speech starting from a diphone codebook as the decoder. The spoken message is encoded in textual (phoneme labels) plus prosody representation. This speech coding technique may be used for voice mail or phone communication over low bit rate channels     

低码率活动图像帧间预测编码的研究

主要研究适于窄带低码率活动图像的帧间预测编码算法。依据Ｈ．２６３建议中的算法和编码方案，着重论述了三种先进的帧间预测模式：无约束运动矢量模式、高级预测模式和ＰＢ帧模式，并分别对其编、解码的效果作出计算机仿真及分析。     

An efficient computation-constrained block-based motion estimationalgorithm for low bit rate video coding

We present an efficient computation constrained block-based motion vector estimation algorithm for low bit rate video coding that yields good tradeoffs between motion estimation distortion and number of computations. A reliable predictor determines the search origin, localizing the search process. An efficient search pattern exploits structural constraints within the motion field. A flexible cost measure used to terminate the search allows simultaneous control of the motion estimation distortion and the computational cost. Experimental results demonstrate the viability of the proposed algorithm in low bit rate video coding applications. The resulting low bit rate video encoder yields essentially the same levels of rate-distortion performance and subjective quality achieved by the UBC H.263+ video coding reference software. However, the proposed motion estimation algorithm provides substantially higher encoding speed as well as graceful computational degradation capabilities.     

基于小波变换的极低码率视频编码技术

介绍几种基于小波的极低码率视频编码及其应用，在世纪之替之际出现了许多新颖和特殊需求的应用领域，例如会议电视，移动通信中的视频传输等等。这些应用的出现，为极低码率的视频编码研究重新注入了新的动力。     

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.