Playback Quality Research Articles

Joint Design of Source Rate Control and QoS-Aware Congestion Control for Video Streaming Over the Internet

Multimedia streaming over the Internet has been a very challenging issue due to the dynamic uncertain nature of the channels. This paper proposes an algorithm for the joint design of source rate control and congestion control for video streaming over the Internet. With the incorporation of a virtual network buffer management mechanism (VB), the quality of service (QoS) requirements of the application can be translated into the constraints of the source rate and the sending rate. Then at the application layer, the source rate control is implemented based on the derived constraints, and at the transport layer, a QoS-aware congestion control mechanism is proposed that strives to meet the send rate constraint derived from VB, by allowing temporary violation of transport control protocol (TCP)-friendliness when necessary. Long-term TCP-friendliness, nevertheless, is preserved by introducing a rate-compensation algorithm. Simulation results show that compared with traditional source rate/congestion control algorithms, this cross-layer design approach can better support the QoS requirements of the application, and significantly improve the playback quality by reducing the overflow and underflow of the decoder buffer, and improving quality smoothness, while maintaining good long-term TCP-friendliness

Cross-Layer Design of Source Rate Control and Congestion Control for Wireless Video Streaming

Cross-layer design has been used in streaming video over the wireless channels to optimize the overall system performance. In this paper, we extend our previous work on joint design of source rate control and congestion control for video streaming over the wired channel, and propose a cross-layer design approach for wireless video streaming. First, we extend the QoS-aware congestion control mechanism (TFRCC) proposed in our previous work to the wireless scenario, and provide a detailed discussion about how to enhance the overall performance in terms of rate smoothness and responsiveness of the transport protocol. Then, we extend our previous joint design work to the wireless scenario, and a thorough performance evaluation is conducted to investigate its performance. Simulation results show that by cross-layer design of source rate control at application layer and congestion control at transport layer, and by taking advantage of the MAC layer information, our approach can avoid the throughput degradation caused by wireless link error, and better support the QoS requirements of the application. Thus, the playback quality is significantly improved, while good performance of the transport protocol is still preserved.

Error recovery in video transmission over CDMA2000 broadcast networks

Multimedia services over a CDMA2000 broadcast network face a challenge from the unreliable and error-prone nature of the radio channel. Reed-Solomon (RS) coding, integrated with the MAC protocol, is used to cope with this problem. However, performance analysis of RS coding under varying channel conditions shows that it is not always effective, especially for slow-moving nodes which experience relatively long error bursts. Therefore a more efficient scheme is proposed that uses a RS code with reduced parity overhead, and freeing bandwidth can be used flexibly for retransmission. The packets to be retransmitted are prioritised by a utility function derived from the map of the error control block at each mobile node and the number of mobile nodes that require the lost packet. Simulation results show that the gain of retransmission exceeds the loss incurred by reducing the parity, leading to an improvement in the playback quality of MPEG-4 video streams. As a result, service area for high-quality multimedia can be expanded.

Frame loss error concealment for SVC

Scalable video coding (SVC), as the Scalable Extension of H.264/AVC, is an ongoing international video coding standard designed for network adaptive or device adaptive applications and also offers high coding efficiency. However, packet losses often occur over unreliable networks even for base layer of SVC and have severe impact on the playback quality of compressed video. Until now, no literature has discussed error concealment support for standard SVC bit-stream. In this paper, we provide robust and effective error concealment techniques for SVC with spatial scalability. Experimental results showed that the proposed methods provide substantial improvement, both subjectively and objectively, without a significant complexity overhead.

Current Developments in Optical Data Storage with Organic Dyes

The main motivation for the development of digital data storage has been the improvement in play-back quality and the increase in storage capacity. In 1982 Philips and Sony introduced the first technically and economically successful system based on this-the compact disc (CD) and a compatible player. A very broad diversity of optical data recording formats are available today, and a difference is drawn between prerecorded, recordable, and rewritable media. This Review gives an overview of the systems used, the main features of production, and then concentrates on the properties of the organic dyes that are used in recordable systems. Dyestuffs chemistry has gained the reputation of having become a mature field of activity. Is this prejudice or a justified swan song for dyestuffs chemistry? When applications in optical data storage are considered, it is evident that even today progresses such as CD-R and DVD/R would not be feasible without functional dyes.

High Data Rate Video Transmission Using Parallel TCP Connections: Approaches and Performance Evaluation

In our study, we investigate a packet-level protocol parallelization approach, which works by parallel multithreading the protocol execution such that packets within and among connections are processed in parallel using distinct processors/threads. The major advantage of this approach is its high scalability--with proper scheduling, more protocol connections, and hence more requests, can be supported by using more threads. In this paper, we present results of our detailed simulations using the NS-2 platform for reliably transferring video stream data in a client-server system. Different types of parameters are used to measure the performance. The parameters include available network bandwidth, different number of TCP connections, and different video sources. Our results show that a parallel approach can indeed significantly enhance the playback quality.

The adaptive layer-based scheduling system for embedded real-time transmission on scalable multimedia stream

With the recent development in compression and network technology, stream media has been adopted in internet and intranet. Application of stream media technology will play a key role in future development of fast networks. Original video/audio data will be stored in the devices for storage after having been pre-compressed by a video/audio compression algorithm. When clients have requirements, the stream server retrieves the video/audio data from the storage devices through the network. This paper discusses a simple and effective real-time scheduling algorithm, adaptive Layer-Based Least-Laxity-First (LB-LLF) scheduling algorithm, to improve the output quality of video/audio on a network and to achieve synchronised playback effect. The proposed algorithm considered real-time constraint, unequal priorities of scalable media stream in different layers, and a good tradeoff between coding efficiency and drifting error. This guarantees the effective usage of available channel bandwidth and the better quality of playback in client.

Mobile multimedia supports with hierarchical multicast SIP framework on heterogeneous wireless networks

For supporting the mobile host (MH) to be the multicast receivers on wireless networks, the traditional approaches such as the Hierarchical Remote Subscription (HRS) and Bi-directional Tunneling with Mobile Multicast (BT-MoM) integrate the Mobile IP protocol with the IP multicast technique. However, they inherit intrinsic problems to handle mobility and suffer significant delays to rebuild the multicast tree when the MH moves out of the multicast tree, which results in serious playback interruption and quality of service (QoS) degradation for the ongoing multimedia service especially when the MH are scattered over different kinds of wireless networks. In this paper, we propose a cross-layer framework and the Hierarchical Multicast Session Initiation Protocol (HMSIP), which extend the IETF Session Initiation Protocol (SIP) with the concept of multicast SIP session and integrate the underlying IP Multicast QoS routing protocol and the RSVP resource reservation technique, to support mobile multimedia multicasts. This framework significantly reduces handoff delays and deployment costs and hence achieves end-to-end QoS support for MHs on heterogeneous wireless networks. Further, we mathematically analyze handoff delays of the HRS, BT-MOM and HMSIP schemes. Moreover, simulation results show that the HMSIP can significantly achieve much more stable and lower handoff delays than those of the BT-MOM and HRS only with a small proportion of extra bandwidth than the bandwidth-optimal HRS, thereby supporting higher numbers of handoffs without interrupting multimedia playback of the mobile multicast activity. Tradeoffs between costs for deploying HMSIP Proxies in the framework and corresponding performances of the HMSIP families are discussed. Finally, multimedia QoS improvements of MHs are shown to emphasize advantages by integrating RSVP in this HMSIP framework.

LOSS AWARE RATE ALLOCATIONS IN H.263 CODED VIDEO TRANSMISSIONS

For packet video, information loss and bandwidth limitation are two factors that affect video playback quality. Traditional rate allocation approaches have focused on optimizing video quality under bandwidth constraint alone. However, in the best-effort Internet, packets carrying video data are susceptible to losses, which need to be reconstructed at the receiver side. In this paper, we propose loss aware rate allocations in both group-of-block (GOB) level and macroblock level, given that certain packets are lost during transmissions and reconstructed using simple interpolation methods at the receiver side. Experimental results show that our proposed algorithms can produce videos of higher quality when sent over lossy Internet.

An optimal cache algorithm for streaming VBR video over a heterogeneous network

High quality video content for on-demand services is usually stored and streamed in a compressed format with a VBR (variable bit rate) property; however, the streaming traffic is extremely bursty. If there is no client buffer to regulate the video's delivery, the backbone WAN (wide area network) bandwidth needs to allocate the video's peak bit rate to guarantee playback quality. To reduce the bandwidth requirement in the backbone WAN, previous researchers have proposed a Video Staging Mechanism to cache portions of the video in a video proxy close to clients. In this paper, we propose a very effective OC (optimal cache) algorithm to handle the Video Staging Mechanism and prove theoretically that the proxy cache computed by our OC algorithm for each video is minimal when all other resources remain constant. On the basis of experiment results, we cache the least amount of video data in the video proxy by using the OC algorithm, and reduce the WAN bandwidth requirement by an amount equal to that of conventional algorithms. In contrast, given the equal size of the storage in a video proxy, the OC algorithm reduces the bandwidth requirement in the backbone WAN much more than conventional algorithms.

Quality Relief of Broadcast-quality Video with Net-work Packet Losses

The impact of packet losses on broadcast-quality video transmission through an asynchronous network is discussed based on the quality of the video playback. The main requirements for broadcast-quality video transmission are quality, synchronization, and smooth failover. Therefore, video quality is also related to network quality in an asynchronous network. When video frames that contain network packet losses are decoded, noise-free video cannot be maintained for block noises caused in video playback. One of the most important features of real-time video data transmission through an asynchronous network is the ability to correct and conceal errors. Adding redundant bits to the source information, as in forward error correction and error correcting code, increases the data bandwidth. Adding redundancy, such as by interleaving, lengthens the latency of the video playback. The appropriate redundancy factor, therefore should ensure a real-time response, minimize latency, and maintain video playback quality. Therefore, when selecting an appropriate error correction/concealment code, it is necessary to clarify the impact of successive packet losses on the quality degradation of broadcast-quality video. To clarify the relationship between the broadcast-quality video quality and network packet losses, we analyzed data based on observation of an actual network with the loss window size as a parameter. Next, we discussed the characteristics of the block noises on the video playback using the Markov model. For this purpose, we made an 80, 000-km-long ATM path on the Japan Gigabit Network and examined the peak-signal to noise ratio of the SMPTE-314M.video playback that contained ATM cell losses.

End-to-End MPEG-4 FGS Video Transmission Based on Smoothing and TCP-Friendly Rate Control

This paper presents the design of an end-to-end adaptive smoothing and TCP-friendly transmission for stored MPEG-4 fine-grained scalable (FGS) videos over the best-effort Internet. The goal is to minimize the playback quality variation when the network conditions are constantly varying. A novel framework for FGS video delivery over a TCP-friendly connection is first presented. In the context of this scheme, and under the assumption of complete knowledge of bandwidth evolution, an offline quality adaptive smoothing algorithm is derived, and an online adaptive smoothing algorithm is also developed based on the predicted available bandwidth to stream FGS video over the TCP-friendly rate control (TFRC) Protocol with the enhanced ARAR model. Through simulation experiments, it has been shown that the online adaptive algorithm performs almost as well as the offline version for a wide-range of the bandwidth scenarios, and a smooth and TCP-friendly video transfer can be accomplished by the proposed scheme.

Adaptive coding schemes for support of multicast audio to wireless devices

The mobile wireless environment, characterized by dynamically varying bandwidth and signal quality, provides a harsh environment for audio and video applications. Insufficient bandwidth leads to losses and delays which significantly reduce the quality of playback at the receiver. Such an environment demands application support for bandwidth adaptation. Audio applications in particular have traditionally operated at fixed rates and are prime candidates for enhancement. We propose a DCT-based audio decomposition (DAD) scheme for real-time general audio streams in the multicast environment which allows audio quality to be traded off against reductions in bandwidth usage. We compare the performance of DAD, using objective and subjective measures, against two other schemes which also support adaptation to restricted bandwidth.

ATF: an Adaptive Three-layer Framework for inter-stream synchronization of SMIL multimedia presentations

In this paper, we propose an Adaptive Three-layer Framework (ATF) to guarantee the inter-stream synchronization, which is called as the APplication Quality of Service (APQoS) in this paper, of World Wide Web Consortium (W3C) Synchronized Multimedia Integration Language (SMIL) presentation. This ATF framework, which is composed of the Framework Control Layer (FCL), the Quality Guarantee Layer (QGL) and the Media Control Layer (MCL), functions as the middleware between underlying best-effort network protocol stacks and the SMIL client/server applications. The FCL at the SMIL client periodically monitors underlying network bandwidth and delay for the QGL to verify whether the total bandwidth requirements of all concurrent objects to meet the APQoS of the SMIL presentation can be satisfied with the available bandwidth, which is the sum of the monitored bandwidths of all concurrent objects. The QGL executes the novel Intra-SMIL Bandwidth Reallocation (ISBR) scheme to dynamically adjust bandwidth allocations among concurrent SMIL objects, according to their Media Importance (MI) values calculated by the MCL. If the available bandwidth is not enough, the ISBR scheme gracefully degrades the layer-encoded objects that have the smallest MI values and only allocate the necessary bandwidth for their base layer (BL) streams instead. Oppositely, it aggressively upgrades the degraded layer-encoded objects that have the largest MI values and allocate the necessary bandwidth for both the BL and enhancement layer (EL) streams. These bandwidths are then feedback to the remote media servers to modify the transmission rates accordingly. With the quantitative analysis for the SMIL presentation quality, this ATF framework achieves the APQoS with the best SMIL playback quality.

Loss‐resilient layered multicast using forward error correction

AbstractRate control for multicasting is required for multipoint distribution of video over the Internet. In this paper, we focus on the video image quality during playback and propose layered multicast rate control using error correcting codes. First, we identify the severe degradation problem in the playback quality of the transmitted video due to packet losses resulting from congestion when transmitting high‐quality video using conventional techniques. To address this problem, we propose a method adopting the parity generated by the error correcting code in addition to the rate control in order to maintain the video image quality. We perform a simulation to show that the proposed method enables rate control that prevents the degradation in quality caused by congestion. Furthermore, we find the optimum block size of the error correcting code with respect to the control latency, and quantitatively evaluate the packet loss ratio and transmission efficiency with respect to the system scale and the transmission efficiency with respect to the layer increment ratio. We also develop a layer coding technique and study the optimum error correcting code in real‐time processing by the software and its implementability. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(5): 29–41, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10156

클라이언트의 동적 버퍼 제어 신호에 의한 서버의 프레임 생략 전송 정책

VOD 서비스의 대부분 정책들은 주로 서버 시스템에 대해 연구되고 있고, 단절감 없는 재생과 재생질 보증의 근간이 되는 클라이언트의 버퍼 안정화 문제에 대해서는 언급되지 않고 있다. 본 논문에서는 클라이언트의 버퍼 상태가 스타베이션 또는 오버런인 경우 서버에 동적인 제어신호를 발하고, 서버에서는 수신된 제어 요구 신호에 따라 프레임을 생략하여 전송하는 정책을 제안한다. 그리고 시뮬레이션을 통하여 제안 정책이 클라이언트의 버퍼 상태 악화로 인한 프레임의 미수신 및 손실을 방지하여 클라이언트의 버퍼를 안정화시켜 재생질 보증에 더 효율적임을 보인다. Most policies of VOD services are mainly studied about server system, and make no touch on clients buffer stabilization which is the basis of guaranteeing playback quality and playing without cut off. In this paper, we proposed a skipping frame transfer policy that can sending dynamic buffer control signal to server in accordance with clients buffer state of starvation or overrun, and server transfers a skipped frame to client after receive the control signal. And through a simulation, we show the suggested policy is more efficiency on playback quality by buffer stabilization which is by preventing loss and miss for receiving frame due to grow worse client buffer state.

A hybrid architecture for cost-effective on-demand media streaming

We propose a new architecture for on-demand media streaming centered around the peer-to-peer (P2P) paradigm. The key idea of the architecture is that peers share some of their resources with the system. As peers contribute resources to the system, the overall system capacity increases and more clients can be served. The proposed architecture employs several novel techniques to: (1) use the often-underutilized peers’ resources, which makes the proposed architecture both deployable and cost-effective, (2) aggregate contributions from multiple peers to serve a requesting peer so that supplying peers are not overloaded, (3) make a good use of peer heterogeneity by assigning relatively more work to the powerful peers, and (4) organize peers in a network-aware fashion, such that nearby peers are grouped into a logical entity called a cluster. The network-aware peer organization is validated by statistics collected and analyzed from real Internet data. The main benefit of the network-aware peer organization is that it allows to develop efficient searching (to locate nearby suppliers) and dispersion (to disseminate new files into the system) algorithms. We present network-aware searching and dispersion algorithms that result in: (i) fast dissemination of new media files, (ii) reduction of the load on the underlying network, and (iii) better streaming service. We demonstrate the potential of the proposed architecture for a large-scale on-demand media streaming service through an extensive simulation study on large, Internet-like, topologies. Starting with a limited streaming capacity (hence, low cost), the simulation shows that the capacity rapidly increases and many clients can be served. This occurs for all studied arrival patterns, including constant rate arrivals, flash crowd arrivals, and Poisson arrivals. Furthermore, the simulation shows that a reasonable client-side initial buffering of 10–20 s is sufficient to ensure full quality playback even in the presence of peer failures.

Low computational cost improvements of the rate-distortion performance in MPEG-2 rate control

Rate control in standards such as MPEG-2 or H263 plays an important role in controlling the compression rate and stabilizing the decoding and playback quality. In this paper, we combine the simplicity of the rate-distortion (R-D) predictive models with the improved R-D performance offered by the iterative approaches and we design effective rate control algorithms for low encoding delay and low complexity video compression. The proposed algorithms initially utilize the Lagrangian formulation in order to add a local R-D optimization component into the current rate control scheme of the standard. Subsequently, two online controllers are designed based on this local optimization scheme for adjusting the bit rate produced by the standard. Extensive experiments show improvements over the MPEG-2 scheme of 0.5–1 dB per frame for a variety of test sequences when the same target bit rates are maintained.

A low-complexity, high-quality, 64-Kbps audio codec with efficient bit allocation

This paper presents a very low-complexity audio codec that provides audio playback quality similar to the MPEG-I/audio level 3 codec at 64 Kbps for a monophonic-channel signal. This well-designed low-sophistication scheme uses a simple noise-masking model, a specialized nonuniform quantizer, an effectively recursive refinement module, and an adaptive arithmetic coder with multiplication-free adaptation. For bit allocation, we propose an appropriate nonuniform quantizer incorporating noise-masking effects and designed for fast implementation as well as efficient acceleration of the proposed refinement process. This recursive refinement algorithm effectively improves recovered perceptual audio quality after quantization. The adaptive arithmetic coder uses two fast adaptation algorithms that do not require multiplication to quickly obtain efficient bit allocation. A Taylor series expansion is used to simplify the frequently executed functions in the masking threshold formulas. Thus, the proposed high-quality audio codec appears to be a very valuable consumer electronic approach or a software solution at low cost.

Memory-constrained 3D wavelet transform for video coding without boundary effects

Three-dimensional (3D) wavelet-based scalable video coding provides a viable alternative to standard MC-DCT coding. However, many current 3D wavelet coders experience severe boundary effects across group of pictures (GOP) boundaries. This paper proposes a memory-efficient transform technique via lifting that effectively computes wavelet transforms of a video sequence continuously on the fly, thus eliminating the boundary effects due to limited length of individual GOPs. Coding results show that the proposed scheme completely eliminates the boundary effects and gives superb video playback quality.