A new Elastic-Block Matching Algorithm using bilinear space warping is proposed. In this scheme a convex quadrilateral, which minimizes a distortion measure against the current square block, is searched to compensate the shape deformation caused by a rigid body's 3 dimensional depth motion or rotation. The proposed algorithm gives a remarkable improvement in motion-compensated prediction compared with the conventional algorithm.

Two main characteristics of VBR MPEG video traffic are the different statistics according to different picture types and the periodic traffic pattern due to GOP structure. Especially, the I-pictures at the beginning of each GOP generate much more traffic than other pictures. Therefore, when several VBR MPEG video sources are superposed, the multiplexing performance can vary according to the variations of their I-picture start times. In this paper, we show how the start time arrangement of the superposed VBR MPEG videos can significantly affect the cell loss ratio characteristics at the multiplexers, by using U-NDPP/D/1/B queueing model. It is also shown that the Lognormal distribution is more suitable for modeling VBR MPEG video traffic than the Normal and Gamma distributions, in the queueing application's view points.

Quantization is an essential step which leads to compression in discrete cosine transform (DCT) domain. In this paper, we show how a statistically non-optimal uniform quantizer can be improved by employing an efficient reconstruction method. For this purpose, we estimate the probability distribution function (PDF) of original DCT coefficients in a decoder. By applying the estimated PDF into the reconstruction process, the dequantization distortion can be reduced. The proposed method can be used practically in any applications where uniform quantizers are used. In particular, it can be used for the quantization scheme of the JPEG and MPEG coding standards.

In this paper, we propose an efficient requantization method for INTRA-frames in MPEG-1/MPEG-4 transcoding. The quantizer for an MPEG-1 INTRA block usually uses a quantization weighting matrix, while the quantizer for an MPEG-4 simple profile does not. As a result, the quantization step sizes of the two quantizers may not be the same even for the same quantization parameter. Due to this mismatch in the quantization step size, a transcoded MPEG-4 sequence can suffer from serious quality degradation and the number of bits produced by transcoding increases from the original MPEG-1 video sequence. To solve these problems, an efficient method is proposed to identify a near-optimum reconstruction level in the transcoder. In addition, a Laplacian-model based PDF (probability distribution function) estimation for the original DCT coefficients from an input MPEG-1 bitstream is presented, which is required for the proposed requantization. Experimental results show that the proposed method provides a 0.3-0.7 dB improvement in the PSNR over the conventional method, even at a reduced bit-rate of 3-7%.

We propose a simplified model of real-time joint source-channel coding, which can be used to adaptively determine the quality-optimal code rate of forward error correction (FEC) coding. The objective is to obtain the maximum video quality in the receiver, while taking time-varying packet loss into consideration. To this end, we propose a simplified model of the threshold set of the residual video packet loss rate (RVPLR). The RVPLR is the rate of residual loss of video packets after channel decoding. The threshold set is defined as a set of discrete RVPLRs in which the FEC code rate must be changed in order to maintain minimum distortion during increases or decreases of channel packet loss. Because the closed form of the proposed model is very simple and has one scene-dependent model parameter, a video sender can be easily implemented with the model. To train the scene-dependent model parameters in real-time, we propose a test-run method. This method accelerates the test-run while remaining sufficiently accurate for training the scene-dependent model parameters. By using the proposed model and test-run, the video sender can always find the optimal code rate on the fly whenever there is a change in the packet loss status in the channel. An experiment shows that the proposed model and test-run can efficiently determine the near-optimal code rate in joint source-channel coding.

In this paper, we propose a dynamic bit-rate reduction scheme for transcoding an MPEG-1 bitstream into an MPEG-4 simple profile bitstream with a typical bit-rate of 384 kbps. For dynamic bit-rate reduction, a significant reduction in the bit-rate is achieved by combining the processes of requantization and frame-skipping. Conventional requantization methods for a homogeneous transcoder cannot be used directly for a heterogeneous transcoder due to the mismatch in the quantization parameters between the MPEG-1 and MPEG-4 syntax and the difference in the compression efficiency between MPEG-1 and MPEG-4. Accordingly, to solve these problems, a new requantization method is proposed for an MPEG-1 to MPEG-4 transcoder consisting of R-Q (rate-quantization) modeling with a simple feedback and an adjustment of the quantization parameters to compensate for the different coding efficiency between MPEG-1 and MPEG-4. For bit-rate reduction by frame-skipping, an efficient method is proposed for estimating the relevant motion vectors from the skipped frames. The conventional FDVS (forward dominant vector selection) method is improved to reflect the effect of the macroblock types in the skipped frames. Simulation results demonstrated that the proposed method combining requantization and frame-skipping can generate a transcoded MPEG-4 bitstream that is much closer to the desired low bit-rate than the conventional method along with a superior objective quality.

Practical and accurate R-Q (rate-quantization) and D-Q (distortion-quantization) models are presented to describe the R-D (rate-distortion) relationship before encoding a frame. The R-Q model is based on a linear relationship between non-zero level count of the DCT coefficients and the generated bits, while the D-Q model comes from the observation that the ratio QP2/D(QP) can be very accurately approximated by a quadratic function of QP, where QP is the quantization parameter used for quantization of DCT coefficients in H. 263 video coding standard. Simulation results show that the proposed models estimate the real coding results very accurately.

We theoretically evaluate the prediction efficiency of the overlapped block motion compensation (OBMC) compared with the conventional non-overlapped approach. Based on the one-dimensional signal model characterized by the AR(1) process and first-order polynomial motion, a condition under which the performance of the OBMC is better, and an optimum window function are derived. From the results, we discuss and analyze several properties of the OBMC, some of which have been experimentally reported in the literature.

The transmission of duplicate packets provides a loss-resilience without undue time-delay in the video transmission over packet loss networks. However, this method generally deteriorates the problem of traffic congestion because of the increased bit-rate for duplicate packet transmission. In this paper, we propose a set of techniques for an efficient packetization and transmission of duplicate video packets. The proposed method transmits the duplicate packet containing high priority data that is quite small in volume but very important for the reconstruction of the video. This method significantly reduces the required bit-rate for duplicate transmission. An efficient packetization method is also proposed to reduce additional packet overhead which is required for transmitting the duplicate data. It is shown by simulations that the proposed method remarkably improves the packet loss-resilience for video transmission only with small increase of redundant duplicated data for each slice.

In live multimedia applications with multiple videos, it is necessary to develop an efficient mechanism of multiplexing several MPEG video streams into a single stream and transmitting it over network without wasting excessive bandwidth. In this paper, we present an efficient multiplexing and traffic smoothing scheme for multiple variable bit rate (VBR) MPEG video streams in live video applications with finite buffer sizes. First, we describe the constraints imposed by the allowable delay bound for each elementary stream and by the multiplexer/receiver buffer sizes. Based on these constraints, a new multiplexing and traffic smoothing scheme is designed in such a way as to smooth maximally the multiplexed transmission rate by exploiting temporal and spatial averaging effects, while avoiding the buffer overflow and underflow. Through computer experiments based on an MPEG-coded video trace of Star-wars, it is shown that the proposed scheme significantly reduces the peak rate, coefficient of variation, and effective bandwidth of the multiplexed transmission rate.

This paper presents an efficient bandwidth allocation method for the two-layer video coding of different spatial resolution. We first find a model of distortion-bitrate relationship for the MPEG-2 spatial scalable coding in a fixed total bitrate system. Then we propose an adaptive bitrate allocation method for a constant distortion ratio between two layers with the given total bandwidth. In the proposed method, approximated model parameters are used for simple implementation. The validity of the approximation is proven in terms of the convergence to the desired distortion ratio. It is shown by simulation that the proposed bitrate allocation method can keep almost a constant distortion ratio between two layers in comparison to a fixed bitrate allocation method.

In this paper, we propose an adaptive video frame rate control method, called AFCON, that video encoders use in conjunction with explicit rate based congestion control in the network. First, an encoder buffer constraint which guarantees the end-to-end delay of video frames is derived under the assumption of bounded network transmission delay for every frame data. AFCON is based on the constraint and consists of future channel rate prediction, frame discarding, and frame skipping. Recursive Least-Squares (RLS) is used to predict the low-frequency component of the channel rate. Frame discarding prevents the delay violation of frames due to the prediction error of the channel rate. Frame skipping adapts the encoder output rate to the channel rate while avoiding abrupt quality degradation during the congestion period. From the simulation results, it is shown that AFCON can adapt to the time-varying rate channel with less degradation in temporal resolution and in PSNR performance compared to the conventional approach.

In this paper, we propose a new framework for global traffic control in ATM networks which aims to maximize resource utilization and to guarantee the reliable congestion control. To do this, we first propose Global Traffic Control (GTC) mechanism which is based on harmonious cooperation of each traffic control function. GTC measures real bandwidth utilization to compensate inaccuracy of the declared mean cell rate and it also monitors cell losses to manage input traffic load when a network approaches congestion state. We also propose new adaptive connection admission control (CAC) algorithms which calculate cell loss performance of related function blocks in a switch node using only a declared peak cell rate and an estimated mean cell rate. We measure only the mean cell rate of the aggregate cell stream in a link to estimate the mean cell rate of each virtual channel connection. We adopt a peak cell rate spacer at the User Network Interface (UNI) to compensate a cell delay variation (CDV). We will also present an approximation technique to estimate a queue length distribution of a general queue. As this technique requires negligible calculation time, it can meet the stringent requirement on the connection set-up time.

In this paper we propose an effective Peak Rate Spacer (PRS) which can guarantee the negotiated peak cell rate almost perfectly even though contention of cells in the output link of the spacer occurs. We also propose a state-dependent Mean cell Rate Policer-Spacer (MRPS) which can manage the cell loss rate properly by controlling the buffer read rate according to the buffer state. As the MRPS has a cell buffer, it intrinsically has a traffic shaping function. Simulation results clearly show effectiveness of our PRS and MRPS.

The international telecommunication standards bodies such as ITU-T, ATM Forum, and IETF recommend the dual leaky bucket for the traffic specifications for VBR service. On the other hand, recent studies have demonstrated multiple time-scale burstiness in compressed video traffic. In order to fill this gap between the current standards and real traffic characteristics, we present a standard-compatible traffic parameter selection method based on the notion of a critical time scale (CTS). The defined algorithm is optimal in the sense that it minimizes the required amount of link capacity for a traffic flow under a maximum delay constraint. Simulation results with compressed video traces demonstrate the efficiency of the defined traffic parameter selection algorithm in resource allocation.