Search

As the Internet-of-Things is deployed widely, many time-series data are generated everyday. Thus, classifying time-series automatically has become important. Compression-based pattern recognition has attracted attention, because it can analyze various data universally with few model parameters. RPCD (Recurrent Plots Compression Distance) is known as a compression-based time-series classification method. First, RPCD transforms time-series data into an image called “Recurrent Plots (RP)”. Then, the distance between two time-series data is determined as the dissimilarity between their RPs. Here, the dissimilarity between two images is computed from the file size, when an MPEG-1 encoder compresses the video, which serializes the two images in order. In this paper, by analyzing the RPCD, we give an important insight that the quality parameter for the MPEG-1 encoding that controls the resolution of compressed videos influences the classification performance very much. We also show that the optimal parameter value depends extremely on the dataset to be classified: Interestingly, the optimal value for one dataset can make the RPCD fall behind a naive random classifier for another dataset. Supported by these insights, we propose an improved version of RPCD named qRPCD, which searches the optimal parameter value by means of cross-validation. Experimentally, qRPCD works superiorly to the original RPCD by about 4% in terms of classification accuracy.

This paper presents a cost-effective two-dimensional (2-D) inverse discrete cosine transform (IDCT) for supporting multiple standards of MPEG 1/2/4, H.264, VC-1, and HEVC. The proposed approach employs a time allocation scheme to enable the simultaneous processing of the first and second dimensions in order to enhance data throughput and attain hardware utilization of 100%. The proposed one-dimensional (1-D) IDCT uses distributed arithmetic (DA) in conjunction with factor sharing (FS) within a hardware sharing architecture. Four parallel computation streams are employed to enhance the throughput rate as four times of operation frequency. The efficacy of this approach was verified by fabricating a test chip using the Taiwan Semiconductor Manufacturing Company Limited (TSMC) 90 nm Complementary Metal-Oxide-Semiconductor (CMOS) process. The inverse transform core has an operating frequency of 200 MHz and a throughput of 800 M-pels/s with a gate count of 27.2 K.

Gamma delta (GD) T cells are an unconventional T cell type present in both the epidermis and the dermis of human skin. They are critical to regulating skin inflammation, wound healing, and anti-microbial defense. Similar to CD8+ cytotoxic T cells expressing an alpha beta (AB) TCR, GD T cells have cytolytic capabilities. They play an important role in elimination of cutaneous tumors and virally infected cells and have also been implicated in pathogenicity of several autoimmune diseases. T cell cytotoxicity is associated with the expression of the pore forming protein Perforin. Perforin is an innate immune protein containing a membrane attack complex perforin-like (MACPF) domain and functions by forming pores in the membranes of target cells, which allow granzymes and reactive oxygen species to enter the cells and destroy them. Perforin-2, encoded by the gene MPEG1, is a newly discovered member of this protein family that is critical for clearance of intracellular bacteria. Cutaneous GD T cells express both Perforin and Perforin-2, but many questions remain regarding the role that these proteins play in GD T cell mediated cytotoxicity against tumors and bacterial pathogens. Here, we review what is known about Perforin expression by skin GD T cells and the mechanisms that contribute to Perforin activation.

Motion Estimation (ME) is an integral part of any video encoder and a large number of Block Matching Motion Estimation (BMME) Algorithms are proposed to cope the computational complexity and increase quality of ME process requirement. Therefore, it is necessary to evaluate the performance of these ME algorithms for different motion activities. In this paper five fast famous BMME algorithms are considered to evaluate their performance on the basis of ME time, search points, PSNR and Means Square Error (MSE). The algorithms evaluated in this paper are considered for state of the art video compression standards like MPEG 1, to MPEG4 and H.261 to H.264. Results show that the PSNR of Diamond Search (DS) is best for all test video sequences, whereas, Hardware Modified DS takes maximum number of search points to calculate motion vector. Moreover, hexagon search algorithm takes minimum number of search points but its PSNR is considerably lower than the other algorithms.

In this letter, multiple forward and inverse fast algorithm based transforms and their hardware-sharing design for 2 × 2, 4 × 4, and 8 × 8 transforms in H.264/AVC, and the 8 × 8 transform in audio video coding standard, 4 × 4 and 8 × 8 transforms in VC-1, and DCT/IDCT in MPEG-1/2/4 are developed with a low hardware cost for multistandard video coding applications. Compared with the directly combined fast transforms without shares, the proposed low-cost 1-D architecture reduces shifters by 67%, adders by 73%, and gate counts by 53.4%. The hardware-sharing efficiencies of shifters and adders in the proposed 1-D transform design are 32% and 25% more than those in the previous design, respectively. By 0.18-μm CMOS technology, the proposed 2-D transform architecture has less normalized power per mode and larger normalized hardware efficiency than the previous multiple-standard designs. The cost-effective 2-D full pipelined transform achieves multistandard real-time 1080HD at 60-Hz video encoding and decoding applications.

Compression of moving images has opened unprecedented opportunities of transmission and storage of digital video. Extraordinary performance of today’s video codecs is a result of tens of years of work on the development of methods of data encoding. This paper is an attempt to show this history of development. It highlights the history of individual algorithms of data encoding as well as the evolution of video compression technologies as a whole. With the development of successive technologies also functionalities of codecs were evolving, which make also the topic of the paper. The paper ends the attempt of authors’ forecasting about the future evolution of video compression technologies.

In this paper, audio perceptual compression systems are described, giving special attention to the former one: The MPEG 1, Layer III, in short, the MP3. Other compression technologies are described, especially the technologies evaluated in the present work: OGG Vorbis, WMA (Windows Media Audio) from Microsoft Corporation and AAC (Audio Coding Technologies).

Digital audio watermarking still remains one of the most interesting techniques for multimedia copyright protection. In this paper, bat inspired optimization approach is exploited to design an optimal audio watermarking algorithm for copyright protection. First, the original audio signal is decomposed by the Discrete Wavelet Transform (DWT) into low and high frequency parts. The low frequency part is then segmented into several 2-D blocks, and QR factorization is subsequently applied to each one. The watermark, which is a binary logo, is embedded by quantizing the coefficients of resulting R matrices. Since the quantization step is signal-dependent, bat algorithm optimization is used to find a suitable one for each audio signal. This allows to obtain the highest possible robustness without losing the transparency of the watermark. Simulation tests show that the proposed method not only maintains high quality of the audio signal, but also is robust against common signal processing manipulations such as noise addition, re-sampling, filtering and MPEG 1 Layer III compression.

In this paper, a novel and area-efficient variable length decoder (VLD) for MPEG-1/2/4 is presented. Instead of carrying out every variable length coding table with one dedicated lookup table (LUT) directly, we employ an efficient clustering-merging technique to reduce both the size of a single LUT and the total number of LUTs required for MPEG-1/2/4. Synthesis results show that our VLD occupies 10666 gate counts and operates at 125 MHz by using the standard cell from Artisan TSMC's 0.18 mum process. As demonstrated, the proposed design outperforms other VLDs with less hardware cost. It can decode a symbol of different standards in every cycle and support video resolution of HD1080 at 30 frames/s for MPEG-1/2/4 real-time decoding.

This paper presents an overview of image compression covering JPEG, the MPEG Standards, H.262, H.263 and H.264, VQ, Wavelet and Fractal Transforms. Error resilience in terms of an unequal error protection (UEP) scheme is also covered. The report concludes with a very brief mention of the protocols used in streaming multimedia data.

The need for interoperability poses a challenge with the widespread use of multiple video formats. A Multi Layer Streaming-Simplified DCT Domain Transcoder (MLS-SDDT) is proposed to achieve efficient transcoding and video quality improvement. However, an efficient architecture is needed to handle drifting errors. The MLS-SDDT algorithm addresses these issues by proposing: (1) an FGS compatible Simplified DCT Domain Transcoder (FGS-SDDT) architecture for MPEG-1/2/4 single layer transcoding, and (2) an R-D optimized multi-layer streaming technique for video quality improvement. To support efficient transcoding from FGS to MPEG1/2/4, the FGS-SDDT adopts low complexity Simplified DCT Domain Transcoder (SDDT) architecture as an FGS compatible SDDT. To resolve the serious error-drifting problem from SDDT, especially for heterogeneous transcoding, a mathematical analysis of the error-drifting problem is provided and a solution by multi-layer streaming is adopted. The multi-layer streaming approach pre-computes the errors according to the mathematical analysis and encodes the errors as the second enhancement layer for error compensation. For different network conditions, an R-D optimized model is used to improve the bit allocation in the two enhancement layers for better R-D performance. Our experiments show that the MLS-SDDT can deliver 1.4~7.0 dB PSNR improvement for MPEG-1 and 1.9~8.6 dB improvement for MPEG-2 as compared to the SDDT. For FGS to MPEG-4 single layer transcoding, the MLS-SDDT can achieve minor PSNR loss (0.0 ~ 0.4 dB) with lower computational complexity as compared to the SDDT approach.

An application specific processor for an H.264 decoder with a configurable embedded processor is designed in this research. The motion compensation, inverse integer transform, inverse quantization, and entropy decoding algorithm of H.264 decoder software are optimized. We improved the performance of the processor with instruction-level hardware optimization, which is tailored to configurable embedded processor architecture. The optimized instructions for video processing can be used in other video compression standards such as MPEG 1, 2, and 4. A significant performance improvement is achieved with high flexibility. Experimental results show that we could achieve 300% performance for the H.264 baseline profile level 2 decoder.

Digital audio & video data have become an integral part of multimedia information systems. To reduce storage and bandwidth requirements, they are commonly stored in a compressed format, such as MPEG-1. Increasing amounts of MPEG encoded audio and video documents are available online and in proprietary collections. In order to effectively utilise them, we need tools and techniques to automatically analyse, segment, and classify MPEG video content. Several techniques have been developed both in the audio and visual domain to analyse videos. This paper presents a survey of audio and visual analysis techniques on MPEG-1 encoded media that are useful in supporting a variety of video applications. Although audio and visual feature analyses have been carried out extensively, they become useful to applications only when they convey a semantic meaning of the video content. Therefore, we also present a survey of works that provide semantic analysis on MPEG-1 encoded videos.

Today's digital audio coding algorithms use sophisticated models to maximise the encoding rate with minimal audible distortion. As a result of this complexity, different implementations of one encoding standard tend to produce varying output streams for the same uncompressed input data. This article presents a method to distinguish between encoding programs used to compress ISO/MPEG 1 Audio Layer-3 (MP3) files on the basis of statistical features that can be extracted from the compressed streams. The method employs a Bayesian machine learning classifier to determine the most likely encoder from a vector of 10 features. Experimental evidence suggests that the method is reliable enough to decrease the rate of false positives in a stego-detection case. Thus, it can be considered as a generic tool to increase the overall reliability of steganalysis in MP3 files. Moreover, a post-hoc interpretation of the trained classifier's parameters reveals interesting details about the degree of relation between subsets of the 20 encoding programs examined in the study. Further topics, such as implications on the robustness and possible extensions to different file formats are addressed in the discussion.

Audio watermarking has been used mainly for digital sound. In this paper, we extend the range of its applications to live performances with a new composition method for real-time audio watermarking. Sonic watermarking mixes the sound of the watermark signal and the host sound in the air to detect illegal music recordings recorded from auditoriums. We propose an audio watermarking algorithm for sonic watermarking that increases the magnitudes of the host signal only in segmented areas pseudorandomly chosen in the time-frequency plane. The result of a MUSHRA subjective listening test assesses the acoustic quality of the method in the range of “excellent quality.†The robustness is dependent on the type of music samples. For popular and orchestral music, a watermark can be stably detected from music samples that have been sonic-watermarked and then once compressed in an MPEG 1 layer 3 file.

In this paper, we propose a fast motion vector re-estimation for transcoding MPEG-1 to MPEG-4 with lower spatial resolution. This task can be performed in the pixel-domain or in the discrete cosine transform (DCT) domain. In this paper, we concentrate on the DCT-domain approach, which requires lower delay and complexity than those in the pixel-domain. For the DCT-domain transcoding to lower spatial resolution pictures, DCT-domain down-sampling filter is applied and a base motion vector (BMV) for the down-sampled MPEG-4 macroblock is to be calculated from the input motion vectors operating on the higher spatial resolution image. Quality can be significantly improved by refining the BMV. Starting with the BMV, the motion vector refinement (MVR) scheme searches for a delta motion vector within a significantly reduced search area. We propose a fast MVR scheme for video down-sampling in the DCT-domain based on minimizing the number of required check points, and a computationally efficient method for extracting motion compensated DCT block. We also show an efficient scheme for selecting a coding mode for a macroblock in the lower resolution video from those of the corresponding higher resolution video.

In the paper the features of protocols and the basic methods of organization of multimedia information transfer in telecommunication networks and distributed corporate safety networks are represented. The analysis of formats of basic elements of signals used in standards of multimedia information transfer is given. Increase in quality requirements and growth of variety of transferred information together with a general tendency towards integration of different message types in telecommunication networks and distributed corporate safety networks have determined the increased interest to technologies of the multimedia information transfer. In numerous publications devoted to multimedia information transfer, the features of used standards and transport protocols have been represented in fragmentary way and have not been always in detail enough. Moreover, the processes of multimedia information transfer have being developed rapidly; therefore part of information presented in various publications has become obsolete. The purpose of the given paper consists in providing interested experts with information about features of structural construction of protocols and signals used now for transfer of multimedia information and, first of all, video information in telecommunication networks and distributed corporate safety networks. Multimedia technologies represent interactive processes providing synchronized processing and transfer of various types of messages including static images, text, animation computer graphics, speech, high-quality sound, video information. In 1988 the Moving Pictures Experts Group (MPEG) – a group of experts in the field of digital video (an official name of the group is ISOIEC/JTC1/SC2/WG11/MPEG) – began work within the framework of ISO (International Standards Organization, ISO/IEC). This group is working in the ISSN 0040-2508 © 2004 Begell House, Inc. 421 O.I. ATAKISHCHEV, S.G. EMEL’YANOV, I.S. ZAKHAROV, B.V. KLYUJKOV, AND E.N. PETRIK direction of design of standards of coding and compressions of video and audio data. It deals with creation and implementation of several audio and video standards, including algorithms of compression, schemes of data processing and instrumental platforms. Today the following standards are known: • MPEG-1 (1992) is the first international standard of MPEG. It describes compression of noninterlaced video signals with bit-transfer rate of 1.5 Mbit s. The standard is intended for compact disk recording (CD-ROM, VideoCD) of video data and transferring of TV images through relatively slow communication channels. MPEG-1 standard uses standards of scanning with fineness 4 times less than in broadcasting television: 288 active lines per television frame and 352 samples per active part of television line. A subjective assessment of quality of TV image depending on data transfer rate shows that the MPEG-1 standard can be effectively used for coding of video data up to rate of 3.5 Mbit s, because the increase in video data transfer rate is accompanied by adequate improvement in quality of TV image over the rate interval 1.5 to 3.5 Mbit s. However further increase in transfer rate does not result in noticeable improvement in quality anymore. At video data transfer rate higher than 3.5 Mbit s the better quality of images is obtained at coding by the MPEG-2 standard. • MPEG-2 is intended for processing of interlace video signal comparable in quality to television one in case of carrying capacity of data transfer system within the limits of from 3 to 15 Mbit s; bit-transfer rate up to 50 Mbit s is used in professional equipment. Many television channels have turned to technologies based on the MPEG-2. A signal compressed according to this standard is translated through television satellites, it is used for archiving of a great of video data array. The MPEG-2 was finally approved as international standard at 29-th MPEG Congress (Singapore, November 1994). A permissible data flow rate was determined from 2 to 10 Mbyte s in its specifications. Primarily the MPEG-2 was being designed for digit transfer and representation of video of broadcasting quality, but later the format of high-definition television (HDTV) was provided for in it. Before, necessary specifications for HDTV had to be included in subsequent MPEG-3 standard with resolution up to 1920×1080 at a frequency of 30 Hz and data flow rate from 20 to 40 Mbyte s. It turned out that the MPEG-2 code (and even MPEG-1) with some updating works quite normally for problems of HDTV. As a result, design of the MPEG-3 was stopped. • MPEG-3 was intended for using in high-definition television systems with data flow rate of 20–40 Mbit s, but later it became a part of the MPEG-2 standard and now it is not mentioned separately. МР3 format, which sometimes is confused with the MPEG-3, is actually a part of specification of the MPEG-1 and is intended only for compression of audio information. Full name of MP3 is MPEG 1 Audio Layer 3. • MPEG-4 was originally intended for using in systems with extremely low carrying capacity (less than 64 kbps). Then, judging by a number of publications, they wanted to attach Java and a number of specific computer multimedia

In this paper, we attempt to provide a comprehensive and high-level review of audio-visual features that can be extracted from the standard compressed domains, such as MPEG-1 and MPEG-2. The paper is motivated by the myriad of active research works in extraction and application of compressed-domain features in various fields, such as indexing, filtering, and manipulation. Compressed-domain approaches avoid expensive computation and memory requirements involved in decoding and/or re-encoding. Selected features are categorized into four groups—spatial visual (e.g., color, texture, edge, shape), motion (e.g., motion field, trajectory), audio (e.g., energy, spectral features, pitch), and coding (e.g., bit rate, frame/block type). For each feature, we briefly discuss the extraction methods, computational complexity, potential effectiveness in applications, and possible limitations caused by compress-domain approaches. Finally, we also discuss the possibilities of extracting some important MPEG-7 visual and audio descriptors directly from the compressed domain.

MPEG-1/2 are the most popular compression schemes used in consumer video products. The significant problem in these compression schemes is high fluctuation in output data rate over the video sequence. These encoders, in general, utilize a rate control algorithm in order to maintain the output data rate at a constant level regardless of the properties of the video sequence and differences in compression ratios of different picture types. Conventional rate control algorithms such as TM5 cannot maintain the picture quality at scene transitions where the statistical properties of pictures change significantly. This paper presents an innovative algorithm to improve the video quality at sudden and gradual scene transitions. The algorithm works in three steps. The fixed encoding structure of the MPEG data stream is disturbed by forcing an extension or termination of the last GOP of the first scene such that the first anchor picture of the second scene becomes an I-coded picture. The complexity parameters of each picture type are, then, corrected with an iterative algorithm. P-pictures are provided with additional bits to cope with inefficient forward prediction during gradual scene transitions. Experimental results show that with the proposed algorithm, both objective and subjective quality of MPEG video can be improved significantly during scene transitions.

A new streaming application has been developed for the Internet environment. The system has client-server structure together with multithreaded architecture and pipelining. RTP protocol is used to transmit packets belonging to MPEG videos. RTCP protocol collects transmission statistics. System is adaptive in the sense that it reacts to dynamic network conditions. A feedback mechanism controls both the frame interval and frame rate depending on the frame-loss statistics and buffered video level at the client. A flow control module at the client side controls buffer underflows and overflows. Performance results of the implementation are reported and discussed. The performance of the proposed buffering strategy is compared with other proposed methods from the literature. The comparisons showed that the proposed strategy is more robust than other methods.

This paper proposes an adaptive rate control algorithm for MPEG-1 to MPEG-4 transcoder for video internetworking where a compressed video is transmitted through networks of different capacities. It is based on an adaptive logarithmic R–Q (rate-quantization) model for controlling the new target bit-rate of a transcoded MPEG-4 bitstream. In R–Q modeling, the R–Q characteristics predicted from a previous picture of the same picture type are used in combination with the R–Q data measured for the current picture. The proposed adaptive model takes into account the effect of picture dependencies arising in motion-compensated video coding. Furthermore, full advantage is taken of the special situation related to transcoding where a priori knowledge of the current R–Q data can be obtained by just counting the number of bits in the input bitstream without re-encoding. It is shown by simulation that the adaptive R–Q model-based rate control can efficiently control the output bit-rate to match the target bit-budget.

This paper summarizes the results of over 25 research groups or individual researchers that have presented video coding implementations on general-purpose processors with the new single instruction multiple data media instruction set architecture extensions. The extensions are introduced and the fundamentals for extensions, as well as some inherent problems, are explained. The reported attempts to utilize the extensions are divided into kernel- and application-level, as well as platform dependent and independent optimizations. Optimized applications include, in addition to some proprietary methods, all of the major video coding standards such as H.261, H.263, MPEG-4, MPEG-1, and MPEG-2. These optimized implementations include a complete video codec, several decoders, and several encoders. Additionally, a performance comparison is given for four representative encoder implementations based on the reported results. Also included is an overview of future trends for new instructions and architectural speed-up techniques.

This chapter discusses the objectives and requirements of moving picture experts group‐1 (MPEG‐1) system. An MPEG‐1 system stream carries one or more elementary streams, multiplexed together. The chapter presents an example of the MPEG‐1 system stream data structure. The MPEG‐1 system target decoder (STD) defines two decoding branches, one for MPEG‐1 video and the other for MPEG‐1 audio elementary streams. The chapter details the data structure and design considerations of the MPEG‐1 system stream. Each MPEG‐1 system stream consists of a concatenation of one or more packs; to conclude the MPEG‐1 system stream, the last pack is followed by an ‘End of MPEG‐1 system stream’ start code. MPEG‐1 proves the feasibility of video and audio compression for many applications, and develops various system approaches essential for the success of MPEG‐2 in the market.

In this chapter, we examine the ideas behind the MPEG standards, starting with MPEG-1, -2, and then MPEG-4, and 7. In MPEG-1 and -2, bidirectional search for motion vectors is introduced. Interlaced video and high-definition TV (HDTV) are supported in MPEG-2. Moreover, it supports various scalable codings such as SNR, spatial, temporal, and their combination. MPEG-4 and -7 studied the issue of video coding based on video objects. Although the visual object-based video representation and compression approach developed in MPEG-4 and 7 have not been commonly used in current popular standards such as H.264 and H.265, it has great potential to be adopted in the future when the necessary Computer Vision technology for automatic object detection becomes more readily available.

In this paper we study measurement-based admission control for bufferless multiplexing, which is very attractive for real-time traffic. We first discuss a novel large deviations (LD) approach to measurement-based admission control. We then provide an extensive review of the existing literature on measurement-based admission control. We conduct simulation studies with traces of MPEG 1 encoded movies to compare the performance of the admission rules in the literature with that of our large deviations approach. We demonstrate that our LD approach achieves higher link utilizations. Finally, we compare the performance of measurement-based admission control with that of traditional admission control, which relies on a priori traffic descriptors. Our numerical work indicates that measurement-based admission control achieves significant gains in link utilizations over traditional admission control. Copyright © 2001 John Wiley & Sons, Ltd.

In this paper, we propose a new summation method for MPEG-1 encoded audio signals in the compressed domain. This method operates directly in the subband domain in order to reduce the delay and the implementation complexity. The main problem to be solved deals with bit allocation and subsequent quantization of the resulting summation signal. The bit allocation of encoded mpeg signals is based on the signal-to-mask ratios of the individual signals. To estimate the signal-to-mask ratios of the combined signal, the algorithm simply needs the information contained in the individual encoded frames. Along with a description of the proposed algorithm, variations and applications of the algorithm are detailed. Additionally, a performance evaluation of this method compared to direct summation in the time domain is given.

The performance evaluation of broadband networks requires statistical analysis and modeling of the actual network traffic. Since multimedia services, and especially variable bit rate (VBR) MPEG-coded video streams are expected to be a major traffic component carried by these networks, modeling of such services and accurate estimation of network resources are crucial for proper network design and congestion-control mechanisms that can guarantee the negotiated quality of service at a minimum cost. The layer modeling of MPEG-1 coded video streams and statistical analysis of their traffic characteristics at each layer is proposed, along with traffic models capable of estimating the network resources over asynchronous transfer mode (ATM) links. First, based on the properties of the entire MPEG-1 sequence (frame layer signal), a model (Model A) is presented by correlating three stochastic processes in discrete time (autoregressive models), each of which corresponds to the three types of frames of the MPEG encoder (I, P, and B frames). To simplify the traffic Model A and to reduce the required number of parameters, we study the MPEG stream at a higher layer by considering a signal, which expresses the average properties of I, P, and B frames over a group of picture (GOP) period. However, models on this layer cannot accurately estimate the network resources, especially in multiplexing schemes. For this reason, an intermediate layer is introduced, which exploits and efficiently combines information of both the aforementioned layers, producing a model (Model B), which requires much smaller number of parameters than Model A and simultaneously provides satisfactory results as far as the network resources are concerned. Evaluation of the validity of the proposed models is performed through experimental studies and computer simulations, using several long duration VBR MPEG-1 coded sequences, different from that used in modeling. The results indicate that both Models A and B are good estimators of video traffic behavior over ATM links at a wide range of utilization.

We study the application of wavelet packet filterbanks to low bit-rate transparent audio coding, taking the audio coders' delay requirements into account, and propose low-delay coders based on wavelet packet filterbanks. We first develop a method of comparison between filterbanks for perceptual audio coding by estimating the necessary bit-rate for a transparent compression. We use this comparison method in order to select the best filters for our audio compression scheme, from a large set of orthogonal and biorthogonal wavelets. Different wavelet filters may be used at different stages of the tree-structured decomposition with a constraint on the overall delay taken into account. The optimization is carried out with a simulated annealing procedure, proposing two wavelet packet filterbanks, exhibiting average and low delays. They are inserted in a complete audio coder that employs vector quantization and considers psychoacoustic models. The use of the proposed filterbanks leads to the design of a new bit allocation procedure, taking into account the lack of selectivity of the equivalent synthesis filters in a wavelet packet filterbank. The resulting audio scheme is validated through listening tests. The wavelet packet filterbanks are shown to be a promising tool for audio coding, especially for low-delay coding: with average delay, the quality of the wavelet packet filterbanks is as good as with MPEG-1 Layer-2, both with 80 kb/s, and when reducing the delay to 200 samples, 96 kb/s are needed to achieve the same quality.

Video sequences compressed by the current video-compression standards-such as MPEG 1/2 and H.261/H.263, which include motion compensation and variable-length coding-are very sensitive to channel disturbances. There exist many error-concealment techniques that can improve the video quality substantially. However, they generally do not prevent or terminate the error propagation. The forced intraupdate technique was proposed for H.245. In this paper, we present an efficient temporal error-propagation prevention method that implements the forced intraupdate concept implicitly. Once an error is detected at the decoder, the starting address of the erroneous macroblocks is sent back to the encoder. The encoder marks the possible damaged area, and the encoding process continues as normal except that the motion estimation will probably not refer to the erroneous part. Thus, the impact of the cell loss is limited to the erroneous slice only, and the damage from the error propagation is greatly reduced. Simulation results of MPEG-2 coding over asynchronous transfer mode (ATM) networks show that the error concealment with feedback can effectively isolate the error and reduce the damage to give satisfactory performance even when the cell-loss rate is higher than 1%. With the delay analysis of ATM networks, we also show that in most cases, the encoder has adequate time to get the feedback information before processing the next I- or P-frames.

The speed of current PCs enable them to decode and play an MPEG bitstream in real-time. The encode process, however, cannot be done in real-time. This paper presents a low-cost real-time fixed point DSP implementation of an MPEG-1 layer 1 audio encoder. The DSP transmits the encoded MPEG bitstream to a PC through a standard bi-directional parallel port, which the PC can immediately save to disk. The encoder is capable of transmitting bitstreams to the PC at bit-rates from 32 kbps-320 kbps.

This paper proposes a new block effect reduction technique based on the notion that the block effect is caused by loss of transform coefficients in the quantization process or during the transmission. The proposed algorithm is called content-based AC prediction for an MPEG-2 compressed video. In the proposed algorithm, the DC sequence is generated for considering the image content while AC coefficients are predicted. Then AC coefficients are predicted with respect to the image content. The simulation result shows that the proposed algorithm can be used in any DCT-based compressed video such as MPEG-1, MPEG-2, H.261, and H.263.

The motion-compensated hybrid DCT/DPCM algorithm has been successfully adopted in various video coding standards, such as H.261, H.263, MPEG-1 and MPEG-2. However, its robustness is challenged in the face of an inadequate bit allocation, either globally for the whole video sequence, or locally as a result of an inappropriate distribution of the available bits. In either of these situations, the trade-off between quality and the availability of bits results in a deterioration in the quality of the decoded video sequence, both in terms of the loss of information and the introduction of coding artifacts. These distortions are an important factor in the fields of filtering, codec design, and the search for objective psychovisual-based quality metrics; therefore, this paper presents a comprehensive analysis and classification of the numerous coding artifacts which are introduced into the reconstructed video sequence through the use of the hybrid MC/DPCM/DCT video coding algorithm. Artifacts which have already been briefly described in the literature, such as the blocking effect, ringing, the mosquito effect, MC mismatch, blurring, and color bleeding, will be comprehensively analyzed. Additionally, we will present artifacts with unique properties which have not been previously identified in the literature.

This paper gives a brief account of the four Moving Picture Experts Group (MPEG) standards. It includes a summary description of their functionalities and usage. For MPEG-1 and MPEG-2, this paper describes how they are being used to convert old analog applications into digital ones and how new applications are being created. For MPEG-4, the major application areas are recalled. For MPEG-7, the strategic rate for the development of multimedia industry is highlighted.