This paper proposes a bitstream scaling technique for MPEG video for the purpose of media synchronizations. The proposed scaling technique can reduce the frame rate as well as the bit rate of an MPEG data sequence to fit them to the values specified by a synchronization system. The advantage of the proposed technique over existing scaling methods is that it is considering not only the performance of synchronization but also the picture quality of the resulting sequences. To further improve the quality of sequences scaled by the proposed method, this paper also proposes an MPEG encoding technique which sets some of the parameters suitable for the scaling. An experiment using these techniques in an actual media synchronization system has illustrated the usefulness of the proposed approach.

This paper studies a set of lip synchronization mechanisms for heterogeneous network environments. The set consists of four schemes, types 0 through 3, which are classified into the single-stream approach and the multi-stream approach. Types 0 and 1 belong to the single-stream approach, which interleaves voice and video to form a single transport stream for transmission. On the other hand, types 2 and 3, both of which are the multi-stream approach, set up separate transport streams for the individual media streams. Types 0 and 2 do not exert synchronization control at the destination, while types 1 and 3 do. We first discuss the features of each type in terms of networks intended for use, required synchronization quality of each medium, physical locations of media sources and implementation complexity. Then, a synchronization algorithm, which is referred to as the virtual-time rendering (VTR) algorithm, is specified for stored media; MPEG video and voice are considered in this paper. We implemented the four types on an ATM LAN and on an interconnected ATM-wireless LAN under the TCP protocol. The mean square error of synchronization, total pause time, throughput and total output time were measured in each of the two networks. We compare the measured performance among the four types to find out which one is the most suitable for a given condition of the underlying communication network and traffic.

This paper proposes a scheme for synchronization of stored video and audio streams in PHS. A video stream of H. 263 is transmitted over a PHS channel with ARQ control, while an audio stream of 32 kbit/s ADPCM is sent on another channel without any control. In order to preserve the temporal constraints within the video stream as well as the relationship between the video and audio streams, we adopt a new control scheme which modifies the target output time according to the amount of video data in the receive-buffer. Through simulation we assess the characteristics of this scheme in both random and burst error environments and confirm the effectiveness of the scheme.

This paper proposes an adaptive permission probability control method for the CDMA/PRMA access protocol. The proposed method is effective to the uplink channels of the integrated voice and data wireless system. The proposed method uses the R-ALOHA protocol with end-of-use flags in order to avoid the reservation cancellations caused by excessive multiple-access interference. Also, a higher priority at packet transmission is given to voice compared with data so that the real-time transmission of voice packets can be guaranteed. Priority is controlled by suitably varying permission probabilities. Permission probabilities are adaptively calculated according to both the channel load and the channel capacities. The usefulness of this proposed method is ensured through computer simulation in an isolated cell environment. Moreover, various applications to cellular environments are investigated. The calculated results indicate that transmission efficiency has been improved compared with the conventional CDMA/PRMA protocol.

Personal communication systems are increasingly required to accommodate not only voice traffic, but also various types of data traffic. Generally speaking, voice traffic is symmetric between uplink and downlink, while data traffic can be highly asymmetric. It is therefore inefficient to accommodate data in a conventional TDMA/TDD system with fixed TDD boundary. In this paper, focusing on the continuous data traffic which requires multi-slots in a circuit based TDMA/TDD system, an algorithm is proposed in which the TDD boundary are moved adaptively to accommodate data traffic efficiently. Comparing with the boundary-fixed conventional algorithm, computer simulations confirm that the proposed algorithm has superior performance in the excessive transmission delay of data while maintaining the performance of voice. The intercell interference between mobiles due to different TDD boundaries is also confirmed to be negligible. Moreover, almost the similar performance improvements of the proposed algorithm are confirmed for two different average message sizes of data calls.

This paper proposes a multiple QoS control scheme that combines the head-of-line priority (HOLP) discipline with equivalent-window connection admission control (CAC). The proposed scheme can support the different cell loss ratios of both delay-sensitive traffic in high-priority buffers and delay-tolerant traffic in low-priority buffers. The CAC scheme extends a measurement-based CAC algorithm for a single buffer to the low-priority buffer with the HOLP discipline to provide the cell loss ratio objective. We introduce an equivalent window for monitoring low-priority cell streams. The equivalent window size equals the period within which the number of times the low-priority buffer is scanned to read cells is constant. Thus the equivalent window size varies with the high-priority queueing state. Numerical results indicate that the proposed QoS control scheme using the equivalent-window CAC can utilize network resources more effectively than the conventional control scheme which is Virtual Path (VP) separation for different cell loss requirement services. In addition, it is confirmed that the proposed scheme provides conservative admissible loads. Thus, this proposed scheme can achieve large statistical gains while providing both high-priority and low-priority cell loss ratio objectives. The proposed scheme will be very useful for cost-effective multimedia services that have different QoS requirements.

Media processors have emerged so that a single LSI can realize multiple multimedia functions, such as graphics, video, audio and telecommunication with effectively shared hardware and flexible software. First, the difference between media processors and general-purpose microprocessors with multimedia extensions is clarified. Features for processes and data in the multimedia applications are summarized and are followed by the multimedia enhancements that the recent general-purpose microprocessors use. The architecture for media processors reflects the further optimized utilization of these features and realizes better price-performance ratio than the general-purpose microprocessors. Finally, the future directions of media processors are estimated, based on the performance, the power dissipation and the die size of the present microprocessors with multimedia extensions and the present media processors. The demand to improve the price-performance ratio for the whole system and to reduce the power consumption makes the media processor evolve into a system processor, which integrates not only the media processor but also the function of a general-purpose microprocessor, various interfaces and DRAMs.

In the wireless ATM network, the key issue is to guarantee various QoS (Quality of Service) under the conditions of the limited radio link bandwidth and error prone characteristics. In this paper, we show a combination method of the error correction schemes, which is suitable to establish multimedia wireless ATM Networks while keeping an efficient use of the limited bandwidth. We consider two levels of FEC; a bit-level and a cell-level to guarantee cell loss probabilities of real time applications. By combining two levels of FEC, various requirements on cell loss can be met. We then apply the bit-level FEC and ARQ protocol for the data communication; tolerant to the delay characteristics. Through the analytical methods, the required overheads of FECs are examined to satisfy the various QoS requirements of CBR connections. The mean delay analysis for the UBR service class is also presented. In numerical examples, we show how the combination scheme to guarantee various cell loss requirements affects the call blocking probability of the CBR service class and the delay of UBR service class.

In this paper, we propose an idea for intramedia synchronization control using a method of end-to-end delay monitoring to estimate future delay in delay compensation protocol. The estimated value by Kalman filtering at the presentation site is used for feedback control to adjust the retrieval schedule at the source according to the network conditions. The proposed approach is applicable for the real time retrieving application where `tightness' of temporal synchronization is required. The retrieval schedule adjustment is achieved by two resynchronization mechanisms-retrieval offset adjustment and data unit skipping. The retrieval offset adjustment is performed along with a buffer level check in order to compensate for the change in delay jitter, while the data unit skipping control is performed to accelerate the recovery of unsynchronization period under severe conditions. Simulations are performed to verify the effectiveness of the proposed scheme. It is found that with a limited buffer size and tolerable latency in initial presentation, using a higher efficient delay estimator in our proposed resynchronization scheme, the synchronization performance can be improved particularly in the critically congested network condition. In the study, Kalman filtering is shown to perform better than the existing estimation methods using the previous measured jitter or the average value as an estimate.

A dual-issue VLIW processor, running at 250 MHz, is enhanced with multimedia instructions for a sustained peak performance of 1000MOPS. The multimedia processor integrates 300 K transistors in an 8 mm2 core area and it is fabricated onto a 6 mm6. 2 mm chip with 32 kB instruction and 32 kB data RAMs in a 0. 3-micrometer, four-layer metal CMOS process. It consumes 1. 2 W at 2. 0 V running at 250 MHz. The VLIW processor achieves a speed-up of more than 4 times over a single-issue RISC for MPEG video block decoding. A decoder implemented on the multimedia processor with a small amount of dedicated hardware, such as the Huffman decoder and a DMA controller will decode the worst case 88 video block data in 754 cycles, leading to a real-time MPEG-2 system, video, and audio decoding system.

Multimedia communications often require intramedia synchronization for video data to prevent potential playout discontinuity while still retaining satisfactory playout throughput. In this paper, we propose a novel intra-video synchronization mechanism, called the Video Smoother, particularly suitable for low-end multimedia applications, such as video conferencing. Generally, the Video Smoother dynamically adopts various playout rates according to the number of frames in the playout buffer in an attempt to compensate for the delay jitter introduced from networks. In essence, if the number of frames in the buffer exceeds a given threshold (TH), the Smoother employs a maximum playout rate. Otherwise, the Smoother employs linearly or exponentially reduced rates to eliminate playout pauses resulting from the emptiness of the playout buffer. To determine optimal THs achieving a minimum of playout discontinuity and a maximum of playout throughput under various bursty traffic, we propose an analytic model assuming incoming traffic following an Interrupted Bernoulli arrival Process (IBP). As a result, optimal THs can be analytically determined resulting in superior playout quality under various arrivals and loads of networks. Finally, we display simulation results which demonstrate that, compared to the playout without intra-video synchronization (instant playout), the Video Smoother achieves superior smooth playout and compatible throughput.

This paper studies a congestion control scheme in integrated variable bit-rate video, audio and data (e. g. , image or text) communications, where each video stream is synchronized with the corresponding audio stream. When the audio and video streams are output, media synchronization control is performed. To avoid congestion, we employ a dynamic video resolution control scheme which dynamically changes the video encoding rate according to the network loads. By simulation, the paper evaluates the performance of the scheme in terms of throughput, loss rate, average delay, and mean square error of synchronization. Numerical results show the effectiveness of the scheme.

We propose a robust detection scheme by employing an order statistic filter as a preprocessor of the input signal. For ease of design, the variance of the order statistic filtered output is modeled by proposing an approximate upper bound. The detector is analytically designed using a fixed sample size (FSS) test scheme. The performance of the proposed detector is compared to that of other robust detectors in terms of the sample size required for given false alarm and miss detection probabilities. Finally, analytical results are verified by computer simulation.

This paper presents novel multimedia ATM networks which are capable of transmitting voice data efficiently and unify the switching methods among heterogeneous traffic. Fully ATMized multimedia networks are using fellow cell switches. The proposed assembly method can pack plural calls which have different virtual channel connection (VCC) into one cell. Every call in cells is able to be dynamically rearranged by the fellow cell switch to achieve an efficient use of network resources. The switching functions are supported by shared virtual channel identifier (VCI) cells and fellow cells in it. The fellow cell switch for 622 Mbps links is integrated into a single chip. The multimedia ATM networks including voice transmission can be constructed by the fellow cell switches being attached to the standard ATM switches.

To construct the future multimedia network, ATM network technology and services should support cost-effective, high-speed interconnectivity and a variety of service-providing functions. Furthermore, as the infrastructure of future multimedia service, the ATM architecture should be adaptable to changes without needing replacement of its core functions and platform capabilities. This paper presents an overview of the current state of development, standardization and deployment of the ATM network service technologies and architecture concept. It also discusses the trend toward the integration of ATM technology and Internet technology. Also reported is the state of development and standardization for the individual ATM technologies and related issues, including access networks, bearer services, signalling, network middleware, and future ATM switching system technology.

A slotted ALOHA direct sequence spread spectrum system with random signatures is considered. The system is applicable in cases where a large number of terminals transmit to a single hub station like in cellular digital radio, personal mobile systems and wireless LANs. It is shown that significant improvements in packet throughput capacity are obtained if the adaptive receiver structures are used. Systems for the comparison are the spread spectrum slotted ALOHA system and the conventional slotted ALOHA system.

This paper presents a proportional-share CPU scheduler which can support multimedia applications in a general-purpose workstation environment. For this purpose, we have extended the stride scheduler which is designed originally for conventional tasks. New scheduling parameters are introduced to specify timing requirements of multimedia applications. Through the use of the rate regulator, the accuracy error of the scheduling is reduced to 0 (1). Separate task groups are proposed to represent both relative shares and absolute shares. The proposed scheduler is evaluated using a simulation study. The results show that the proposed scheduler achieves improved accuracy and adaptability as well as flexibility.

Distributed multimedia applications are often sensitive to the Quality of Service (QoS) provided by the communication network. They usually require guaranteed QoS service, so that real-time communication is possible. However, searching a route with multiple QoS constraints is known to be a NP-complete problem. In this paper, we propose a new simple and efficient distributed QoS routing algorithm, called "DQoSR," for supporting real-time communication in high-speed networks. It searches a route that could guarantee bandwidth, delay, and delay jitter requirements. Routing decision is based only on the modified cost, hop and delay vectors stored in the routing table at each node and its directly connected neighbors. Moreover, DQoSR is proved to construct loop-free routes. Its worst case message complexity is O(|V|2), where |V| is the number of nodes in the network. Thus DQoSR is fast and scales well to large networks. Finally, extensive simulations show that average rate of establishing successful connection of DQoSR is very near to optimum (the difference is less than 0.4%).

We have developed a network architecture that achieves ATM multicast communication services with receiver-specified quality of service (QoS) guarantee which depends on the dynamic resource environment of the receivers (e.g. CPU capability, memory capability, and network capability). We propose two receiver-initiated QoS guarantee methods and concentrate on the functions required to achieve them. Moreover, on our ATM testbed, we also evaluate the performance of an experimental implementation of the proposed methods.

This paper proposes cell-attached frame encapsulation schemes in which encapsulation processing can be performed without cell reassembly. The proposed schemes are especially useful for a global networking service platform to integrate widely distributed user LANs into a single internetwork. The platform itself is an ATM-based frame forwarding network composed of access networks and a core network. These elemental networks are interconnected via edge nodes. In order to improve network interworking performance, these edge nodes should perform encapsulation processing without cell reassembly. Our proposal solves this problem. In the proposed schemes, when the first cell of a cell-divided access network frame arrives at an ingress edge node, a core-header-cell is generated from the IP header described in the first cell payload. This core-header-cell is first transmitted and then succeeding incoming cells including the first cell are forwarded cell-by-cell as soon as they arrive. Since cell-by-cell forwarding-processing reduces frame forwarding latency and cell buffer capacity, these schemes are effective from the viewpoint of both performance improvement and cost reduction.