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20th IEEE International Symposium on Industrial Electronics (ISIE)

作者： Saitoh, Daichi Yakoh, Takahiro Keio Univ Dept Syst Design Engn Yokohama Kanagawa 223 Japan

ISBN: (纸本)9781424493128

Multiple description coding (MDC) is one of source coding techniques to encode realtime application source into multiple bitstreams. MDC supports multiple quality levels of decoding when transmitted over lossy channels such as packet networks. An MDC source encoder provides different representations of the source such that;when all descriptions are available at the receiver, an high-quality reconstruction of the source is possible, while if only a small number of descriptions is available, a reconstruction is still possible even though a lower quality, but acceptable is obtained. Therefore, this coding technique is expected to apply applications which prefer accepting error of data to retransmitting the lost or erroneous data over multipath noisy network. Multiple Description Correlating transforms coding (MDCTC) is a method to realize MDC with a matrix. Based on this method, this paper proposes the three ratio configuration method for Multiple Description Correlating transforms coding (RMDCTC), which can control the volume ratio of descriptions to source data. Therefore, RMDCTC can adjust the data size ratio among descriptions to fit the ratio of channel bandwidths and is suitable to apply error allowable communicating applications over asymmetric multipath noisy network. In addition, each three ratio configuration method has different feature. Therefore, it is possible to select suitable method as the situation demands. As a result, the proposed methods are more practical method than traditional MDCTC.

关键词： Bandwidth Decoding Encoding Entropy Quantization transform coding transforms

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Rounding of Improved DCT transform coding for H.266/VVC 13

Rounding of Improved DCT Transform Coding for H.266/VVC

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13th International Conference on Digital Image Processing (ICDIP)

作者： Chan, Ka-Hou Im, Sio-Kei Macao Polytech Inst Macau Peoples R China Macao Polytech Inst Sch Appl Sci Macau Peoples R China

ISBN: (数字)9781510646018

ISBN: (纸本)9781510646018

Many video encoders use DCT transform coding to compress the encoded video. For hardware implementation, DCT will be approximately an integer matrix, which may cause some deviations in this process, and these deviations will accumulate and become obvious in the larger code unit. Our method is to construct all DCT-related discrete orthogonal transforms in the required size (corresponding to the coding unit supported by H.266/VVC). By using a novel discrete orthogonal matrix generation method with determined DCT-II roots, and scaling and rounding a regular DCT that depends on the quantization parameter, instead of integer approximation. We can obtain an accurate integer DCT matrix. Experimental results show that this method can not only improve the video quality and also require fewer bit rates.

关键词： DCT transform coding H.266/VVC Discrete Orthogonal Matrix

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Embedded transform coding of audio signals by model-based bit plane coding

Embedded transform coding of audio signals by model-based bi...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Hoang, Thi Minh Nguyet Oger, Marie Ragot, Stephane Antonini, Marc France Telecom R&D TECH SSTP Av Pierre Marzin F-22307 Lannion France Univ Nice Sophia Antipolis F-06903 Sophia Antipolis France CNRS UMR 6070 Lab I3S F-06903 Sophia Antipolis France

ISBN: (纸本)9781424414833

This paper proposes a new model-based method for transform coding of audio signals. The input signal is mapped in "perceptual" domain by linear-predictive weighting filter followed by modified discrete cosine transform (MDCT). To provide bitstream scalability, model-based bit plane coding is then applied with respect to the mean square error (MSE) criterion. We present methods to estimate the symbol probability in bit planes assuming a generalized Gaussian model for the distribution of MDCT coefficients. We compare the performance of the proposed bitstream scalable coder with stack-run coding and ITU-T G.722.1. Objective and subjective quality results are presented. The proposed coder is equivalent to or slightly worse than reference coders, but presents the nice advantage of being scalable. Performance penalty due to bitstream scalability is evident at low bitrates.

关键词： transform coding audio coding

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Bit allocation in the transform coding of hologram patterns

Bit allocation in the transform coding of hologram patterns

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Conference on Practical Holography XIII

作者： Ishida, Y Fujii, T Kimoto, T Tanimoto, M Nagoya Univ Grad Sch Engn Dept Informat Elect Chikusa Ku Nagoya Aichi 4648603 Japan

ISBN: (纸本)0819431087

We examined the data compression by transform coding for hologram patterns generated on a computer, to realize effective compression of hologram patterns with extremely huge information. We can't apply conventional 2-D image compression techniques directly to hologram patterns since the statistical properties of hologram patterns are quite different from those of 2-D images. Furthermore, it is not a hologram pattern but an image reproduced from the hologram pattern to be essential in the holography. We have to compress hologram patterns by considering the reproduced image. We found that hologram patterns contain a large amount of unnecessary component to reproduce the image. This should be removed for effective coding. The unnecessary component can be distinguished clearly from the necessary component in the frequency domain. We successfully removed it by bandpass filtering hologram patterns. We apply Karhunen-Loeve transform (KLT) to the hologram patterns after this preprocessing. In the case of high compression, it is better to allocate more bits to the lower order KLT coefficients than the bits determined by the conventional power-based allocation method. Then, effective coding of hologram patterns is realized and better images are reproduced.

关键词： holography data compression transform coding transform coefficients bit allocation

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Multi-Rate Adaptive transform coding for Video Compression 48

Multi-Rate Adaptive Transform Coding for Video Compression

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48th IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP 2023

作者： Duong, Lyndon R. Li, Bohan Chen, Cheng Han, Jingning Nyu Center for Neural Science New YorkNY United States Open Codecs Google Llc Mountain ViewCA United States

ISBN: (纸本)9781728163277

Contemporary lossy image and video coding standards rely on transform coding, the process through which pixels are mapped to an alternative representation to facilitate efficient data compression. Despite impressive performance of end-to-end optimized compression with deep neural networks, the high computational and space demands of these models has prevented them from superseding the relatively simple transform coding found in conventional video codecs. In this study, we propose learned transforms and entropy coding that may either serve as (non)linear drop-in replacements, or enhancements for linear transforms in existing codecs. These transforms can be multi-rate, allowing a single model to operate along the entire rate-distortion curve. To demonstrate the utility of our framework, we augmented the DCT with learned quantization matrices and adaptive entropy coding to compress intra-frame AV1 block prediction residuals. We report substantial BD-rate and perceptual quality improvements over more complex nonlinear transforms at a fraction of the computational cost. © 2023 IEEE.

关键词： entropy coding transform coding video compression

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transform coding of images based on fractal functional set

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ELECTRONICS AND COMMUNICATIONS IN JAPAN PART III-FUNDAMENTAL ELECTRONIC SCIENCE 1995年第8期78卷 31-40页

作者： Ogawa, H Nakagawa, M Associate Member Faculty of Engineering Nagaoka University of Technology Nagaoka Japan 940-21 Member Masahiro Nakagawa received his B.S. and M.S.degrees in Electrical Engineering from Nagaoka University in 1980 and 1982 respectively. He received his Ph.D. degree in 1988. Starting in 1982 Dr. Nakagawa was a Research Associate at Nagaoka University. From March of 1988 to January of 1989 he was a Visiting Researcher at the University of Strathclyde. Heis currently an Associate Professor at Nag & University conducting research in the areas of fractals chaos neural networks device physics and physics of liquid crystals. He isa member of the Japanese Neural Network Society the Physical Society of Japan and the Japan Society of Applied Physics.

Orthogonal fractal basis functional sets are proposed for the compression of images with long-tail power distributions. These functional sets are generated by the method of midpoint displacements from deterministic theory and are orthogonalized by the method of Schmidt. From computer simulations of transform data compression coding that uses a transform based on these functional sets and a nonuniform quantizer, the functional sets proposed in this paper are shown to better preserve the self-similarity of fractal image data when compared to conventional methods that use the discrete cosine function (DCF). Also, for standard images that possess local and approximate self-similarity fractal characteristics, although the measured SNR performance is lower, the proposed functional sets are shown to provide excellent subjective performance and to preserve the fractal properties of the images, On the other hand, for standard images that do not possess these fractal properties, compression errors are apparent in blocks from both smooth and edge regions of the images.

关键词： fractals self-affine functions transform coding long-tail power distributions

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transform coding: Past, present, and future

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IEEE SIGNAL PROCESSING MAGAZINE 2001年第5期18卷 6-6页

作者： Kovacevic, J Vetterli, M Bell Labs Murray Hill NJ 07974 USA Univ Calif Berkeley EPFL Berkeley CA 94720 USA

Not Available

关键词： transform coding Image coding Signal processing Heart Discrete cosine transforms Discrete wavelet transforms Signal analysis Pervasive computing Discrete transforms Audio compression

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transform coding of audio signals using correlation between successive transform blocks

Transform coding of audio signals using correlation between ...

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Y. Mahieux J.P. Petit A. Charbonnier Centre National d''Etudes Des Telecommunications Lannion France

The authors propose a transform coding algorithm for bit rate reduction of high-quality sound. The algorithm provides for long-term stationarity in the transform coder while keeping fixed blocklength. Side information is drastically reduced, and predictive coding of DFT (discrete Fourier transform) coefficients is used to remove the interblock redundancy. Perceptual properties are incorporated in the bit-allocation procedure to achieve spectral noise shaping. The complete algorithm is described, and results of coding at 96 kb/s for a monophonic 15-kHz audio signal are discussed.< >

关键词： transform coding Bit rate Frequency Noise shaping Ear Redundancy Discrete transforms Discrete Fourier transforms Predictive coding Speech coding

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transform coding of monochrome and color images using trellis coded quantization

Transform coding of monochrome and color images using trelli...

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Data Compression Conference (DCC)

作者： K.-L. Tong M.W. Marcellin Department of Electrical and Computer Engineering University of Arizona Tucson Tucson AZ USA

Summary form only given. Trellis coded quantization (TCQ) is incorporated into a transform coding structure for an average encoding rate of 1 bit/pixel. Both fixed-rate and entropy-constrained TCQ-based systems are investigated. 256*256 pixel, 8-bit gray level images and 24-bit color images are encoded using both TCQ- and scalar quantizer-based schemes to provide comparisons on the basis of peak-signal-to-noise ratio and subjective image quality. Iterative codebook optimization algorithms are used to minimize mean square error in both systems.< >

关键词： transform coding Color Quantization Image coding PSNR Discrete cosine transforms Pixel Iterative algorithms Image quality Discrete transforms

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transform coding of speech with weighted vector quantization

Transform coding of speech with weighted vector quantization

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： T. Moriya M. Honda NTT Electrical Communication Laboratories Tokyo Japan

A new transform coding of speech is proposed which employs a weighted vector quantization scheme. First, the characteristics of the weighted vector quantization are checked. Then based on these considerations, a frequency domain coder is designed for medium-band (4.8-9.6 kbps) speech coding. In this coding scheme, the linear prediction residue is transformed and vector quantized. In order to control the quantization errors in the frequency domain, adaptively weighted matching is employed instead of the conventional adaptive bit allocation. Therefore, the residual signal can be reconstructed by the decoder, even if the spectral envelope parameters are destroyed due to transmission errors. The coded speech is natural and unaffected by background noise and its mean opinion score at 7.2 kbps is comparable to that of 5.5-bit log PCM coded speech.

关键词： transform coding Speech coding Vector quantization Frequency domain analysis Speech enhancement Programmable control Adaptive control Bit rate Decoding Background noise

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