The impact of the ATM concept on video coding

The induction of the asynchronous transfer mode (ATM) concept may significantly influence the coding of video services for broadband networks. The authors show how the absence of a physical channel structure and the ability to switch bursty traffic can be used to enhance video coding. Packetization defects and their impact on picture quality, coding algorithms, and synchronization schemes are studied. The authors describe variable-bit-rate coding and report on the results obtained with a hardware implementation of a variable-bit-rate video codec. Statistical multiplexing gain figures are given. The influence of cell loss on image quality is discussed and simulation results are given. A layered coding model offering good cell loss concealment properties and high flexibility is described     

Layered coding for ATM based video distribution systems

This paper discusses packet loss and its protection in an asynchronous transfer mode (ATM) based video distribution system. Packet losses in ATM based networks have such a great impact on the design of coding algorithms and network architectures that they should be exhaustively discussed and resolved. In this paper, first basic configuration of the ATM based video transmission system and its packet-loss protection schemes are discussed. The DCT based layered coding scheme with packet priority classification is proposed as an effective packet-loss protection scheme. Burstiness characteristics of the broadcast video sources are evaluated and modeled to clarify statistical multiplexing performance and packet-loss properties. The quality degradation caused by the packet losses is also evaluated by the SNR, and the superior performance of the proposed layered coding scheme is verified.     

A variable bit rate video codec for asynchronous transfer modenetworks

The bandwidth flexibility offered by the asynchronous transfer mode (ATM) technique makes it possible to select picture quality and bandwidth over a wide range in a simple and straightforward manner. A prototype model of a video codec was developed that demonstrates the feasibility of both variable bit rate (VBR) coding and user-selectable picture quality. The VBR coding algorithm is discussed and it is shown how a stabilized quality is achieved and how this quality and associated bandwidth can be selected by the user. How error propagation is limited to reduce the visibility of cell losses is also discussed. Interfaces with the ATM network are analyzed, with emphasis on decoder synchronization and absorption of cell delay jitter. The VBR codec offers very good picture quality for videophony applications at an equivalent load of 5.9 Mb/s. Picture quality remains relatively constant, even for heavy motion     

Statistical multiplexing gain for variable bit rate video codecs in ATM networks

Asynchronous transfer mode (ATM) broadband networks will support variable bit rate video codecs, which are capable of maintaining a constant picture quality. To demonstrate this capability, a prototype hardware video coder has been developed in the Siemens Central Communications Laboratories. The prototype uses interframe coding, combined with a discrete cosine transform, and is able to reproduce the original picture quality, independent of signal sources or picture material used. A gain in transmission efficiency is expected when several video sources share a common ATM channel (‘statistical multiplexing’). This paper reports on a series of measurements that have been performed using this coder for a large variety of video sources to determine the possible gain in transmission efficiency. The main results are: for realistic video phone scenes, up to about three times the number of signals can be transmitted compared to transmision with constant rate and the same picture quality, if the output signal of the coder has been smoothed over a period of one frame. Smoothing over shorter periods reduces the potential gain substantially. The statistical multiplexing gain increases with the duration of the picture sequences due to the criterion of constant picture quality. It varies very little with the acceptable packet loss rate.     

Selective recovery of video packet loss using error concealment

An efficient recovery method using error concealment is proposed for video packet loss in fast packet switching networks. In this method, the receiver detects the damaged picture area caused by packet loss from the structured picture data received, makes error concealments, notifies the transmitter, and continues decoding. The transmitter, having received the notice, calculates the affected picture area in the local decoded picture and continues encoding without using this affected area. In this selective recovery method, video signals are not stopped even if a long propagation delay exists, no additional information is transmitted to error recovery and conventional coding algorithms can be used. The proposed method is suitable for multipoint communication. Simulation results show the affected picture area is localized for a considerable time attesting to the method's effectiveness     

无线视频的一种码率控制方法

无线信道的高误码率对视频图像质量有很大的影响,前向纠错(FEC)和自动重发请求(ARQ)对于降低无线信道的误码率,提高图像质量有很好的效果。通过对FEC和ARQ方法的有效性分析,在TMN8的基础上提出一种简单的混合FEC/ARQ自适应模式选择码率控制方法。该方法首先预测报文丢失数量和纠错报文传输时延,从而选择合适的纠错编码模式,并为纠错编码分配比特数。实验结果表明该方法有效降低无线信道下报文丢失率,显著提高了图像质量。     

Cell-loss concealment techniques for layered video codecs in an ATMnetwork

A layered video coding scheme with its inherent cell loss resilience has been considered as a means for transporting reliably integrated video services over an asynchronous transfer mode (ATM) based network such as the broadband-ISDN. This paper presents some data concealment techniques that can be implemented in the coding of video data at the encoder, in the ATM adaptation layer (AAL) functionality of network realization and at the decoder to improve the performance of a layered codec under different conditions of video packet loss. The performance of these techniques are verified by software simulations.     

一种适于ATM网络的视频编码技术

本文提出了一种基于子带编码的可变比特率分层编码新算法。它能充分发挥ATM网络传输的特点,并对信元丢失这一ATM网络的固有缺陷造成的图像质量的下降进行有效的补偿,从而实现高质量的稳定的视频通信。文中给出了编码算法、信元构成的方法以及计算机模拟结果。     

User-oriented QoS in packet video delivery

We focus on packet video delivery, with an emphasis on the quality of service perceived by the end user. A video signal passes through several subsystems, such as the source coder, the network (ATM or Internet), and the decoder. Each of these can impair the information, either by data loss or by introducing delay. We describe how each of the subsystems can be tuned to optimize the quality of the delivered signal, for a given available bit rate in the network. The assessment of end-user quality is not trivial. We present research results, which rely on a model of the human visual system     

Congestion control for multimedia services

The problem of congestion control in high-speed networks for multimedia traffic, such as voice and video, is considered. It is shown that the performance requirements of high-speed networks involve delay, delay-jitter, and packet loss. A framing congestion control strategy based on a packet admission policy at the edges of the network and on a service discipline called stop-and-go queuing at the switching nodes is described. This strategy provides bounded end-to-end delay and a small and controllable delay-jitter. The strategy is applicable to packet switching networks in general, including fixed cell length asynchronous transfer mode (ATM), as well as networks with variable-size packets     

Joint source-channel coding and power adaptation for energy efficient wireless video communications

We consider efficiently transmitting video over a hybrid wireless/wire-line network by optimally allocating resources across multiple protocol layers. Specifically, we present a framework of joint source-channel coding and power adaptation, where error resilient source coding, channel coding, and transmission power adaptation are jointly designed to optimize video quality given constraints on the total transmission energy and delay for each video frame. In particular, we consider the combination of two types of channel coding—inter-packet coding (at the transport layer) to provide protection against packet dropping in the wire-line network and intra-packet coding (at the link layer) to provide protection against bit errors in the wireless link. In both cases, we allow the coding rate to be adaptive to provide unequal error protection at both the packet and frame level. In addition to both types of channel coding, we also compensate for channel errors by adapting the transmission power used to send each packet. An efficient algorithm based on Lagrangian relaxation and the method of alternating variables is proposed to solve the resulting optimization problem. Simulation results are shown to illustrate the advantages of joint optimization across multiple layers.     

Hierarchical Optimization of Cascading Error Protection Scheme for H.264 Scalable Video Streaming

Forward error correction (FEC) coding has been shown to offer a feasible solution to fulfill the need for Quality of Service for multimedia streaming over the fluctuant channels, especially in terms of the reduction of end-to-end delay. In this paper, we propose the Dynamic FEC-Distortion Optimization Algorithm to efficiently utilize the network bandwidth for better visual quality by means of hierarchical coding structure with the cascading error protection scheme. The optimization criteria are based on the unequal error protection by taking account of the error drifting problems from both temporal motion compensation and inter-layer prediction of the H.264/MPEG-4 AVC scalable video coding so that the priorities of each video components can be differentiated for the calculation of the distribution of parity packets. It is shown that the cascading error protection scheme makes the hierarchical structure of error erasure code more efficient. Also, the proposed algorithm works particularly well for fast motion videos and the performance does not depend on accurate estimation of packet loss rate.     

Comparison of video protection methods for wireless networks

An experimental comparison of video protection methods targeted for wireless networks is presented. Basic methods are the data partitioning, reversible variable length coding, and macroblock row interleaving as well as macroblock scattering for packet loss protection. An implementation is described, in which scalable video is protected unequally with forward error correcting codes and retransmissions. Comparisons are performed for simulated wideband code division multiple access channel, and measurements are carried out with wireless local area network, Bluetooth as well as with GSM high speed circuit switched data. For the measurements, point-to-point connections are used. The achieved video quality is examined in our real-time wireless video demonstrator. The performance is measured with peak-signal-to-noise-ratio of received video, data overhead, communication delay, number of lost video frames, and decoding frame rate. Results show that the quality of decoded video can be improved by 1 dB with transparent connections compared to connections designed for general packet data. As a conclusion, a video coding subsystem must have access to the error control in a wireless link for the best quality in varying conditions.     

Estimation of accurate effective loss rate for FEC video transmission

We propose an algorithm for adjusting data transmission parameters, such as the packet size and the code rate of forward error correction (FEC), to obtain maximum video quality under dynamic channel conditions. When determining transmission parameters, it is essential to calculate an accurate effective loss rate that reflects FEC recovery failures and over-deadline packets. To this end, we analyze the delays caused by FEC coding and the potential packet size variations. In our analysis, we consider the effect of delayed transmission of video packets incurred by the parity packets as well as the encoder and decoder buffers. With the analysis reflecting the delay effect, we are able to accurately estimate the delay patterns of all video packets. Based on the analysis results, we establish an accurate model for estimating the effective loss rate. Simulations show that the proposed effective loss rate model accurately estimates the effective loss rate and significantly improves the reconstructed video quality at the receiver.     

A dynamic rate control mechanism for source coded traffic in a fastpacket network

To achieve better statistical gain for voice and video traffic and to relieve congestion in fast packet networks, a dynamic rate control mechanism is proposed. An analytical model is developed to evaluate the performance of this control mechanism for voice traffic. The feedback delay for the source node to obtain the network congestion information is represented in the model. The study indicates that significant improvement in statistical gain can be realized for smaller capacity links (e.g., links that can accommodate less than 24 voice calls) with a reasonable feedback time (about 100 ms). The tradeoff for increasing the statistical gain is temporary degradation of voice quality to a lower rate. It is shown that whether the feedback delay is exponentially distributed or constant does not significantly affect performance in terms of fractional packet loss and average received coding rate. It is also shown that using the number of calls in talkspurt or the packet queue length as measures of congestion provides comparable performance     

Error-resilient packet-switched mobile video telephony with channel-adaptive rateless coding and early reference picture selection

Providing high-quality video for packet-switched wireless video telephony on handheld devices is a challenging task due to packet loss, varying bandwidth, and end-to-end delay constraints. While many error resilience techniques have been proposed for video transmission over wireless channels, only a few were specifically designed for mobile video telephony. We propose a low-complexity channel-adaptive error resilience technique for packet-switched mobile video telephony, which combines rateless coding, feedback, and reference picture selection. In contrast to previous approaches, our technique uses cumulative feedback at every transmission opportunity and predicts when decoding is likely to fail so that reference picture selection can be triggered at an early stage. Experimental results for H.264 video sequences show that the proposed technique can achieve improvements of 1.64 dB in peak signal-to-noise ratio over benchmark techniques in simulated Long-Term Evolution networks.     

Statistics of video signals for viewphone-type pictures

The authors describe a series of measurements of the statistics of viewphone-type video signals, with particular regard to the possible transmission of such signals over an asynchronous transfer mode (ATM) network. Measurements include the frame-to-frame differences and the cluster length distribution. It is known that with efficient picture coding, the information rate required for a constant picture quality varies significantly and creates problems in a constant-bit-rate system. The multiplexing of a number of sources in a variable-bit-rate (VBR) system is considered. It is shown that considerable reduction in the variability of the data rate is obtained. While the results are derived from one particular type of picture coder, it is expected that the conclusion will apply to other coding schemes     

Quality control for VBR video over ATM networks

Uncontrolled variable-bit-rate (VBR) coded video yields consistent picture quality, but the traffic stream is very bursty. When sent over ATM networks, cell losses may be incurred due to limited buffer capacity at the switches; this could cause severe picture quality degradation. Source rate control can be implemented to generate a controlled VBR bit stream which conforms to specified bit rate bounds and buffer constraints. However, source rate control could result in picture quality degradation too. Hence, for real-time video services, an important issue to address is whether the picture quality degradation incurred by source rate control is within acceptable levels or how to choose the appropriate coding parameters to make it so. We establish quantitatively the relationship between picture quality and source rate control for the case of guaranteed service with different combinations of allocated bandwidth, buffer size, and other key video-coding parameters of MPEG-2. In addition, quality control in the context of two-layered scalable video service (basic and enhanced quality) is also considered. Our study reveals that, in order to maximize both the basic and the enhanced quality, source rate control should be implemented on both layers. The relationships between the two types of quality and different combinations of allocated bandwidths, buffer sizes, and some key coding parameters are also established quantitatively for MPEG-2 SNR scalability     

Rate-distortion optimized hybrid error control for real-time packetized video transmission.

The problem of application-layer error control for real-time video transmission over packet lossy networks is commonly addressed via joint source-channel coding (JSCC), where source coding and forward error correction (FEC) are jointly designed to compensate for packet losses. In this paper, we consider hybrid application-layer error correction consisting of FEC and retransmissions. The study is carried out in an integrated joint source-channel coding (IJSCC) framework, where error resilient source coding, channel coding, and error concealment are jointly considered in order to achieve the best video delivery quality. We first show the advantage of the proposed IJSCC framework as compared to a sequential JSCC approach, where error resilient source coding and channel coding are not fully integrated. In the USCC framework, we also study the performance of different error control scenarios, such as pure FEC, pure retransmission, and their combination. Pure FEC and application layer retransmissions are shown to each achieve optimal results depending on the packet loss rates and the round-trip time. A hybrid of FEC and retransmissions is shown to outperform each component individually due to its greater flexibility.     

Interleaved coding and sequential transmission scheme for packet‐level forward error correction over burst loss channels

Burst packet loss is a common problem over wired and wireless networks and leads to a significant reduction in the performance of packet‐level forward error correction (FEC) schemes used to recover packet losses during transmission. Traditional FEC interleaving methods adopt the sequential coding‐interleaved transmission (SCIT) process to encode the FEC packets sequentially and reorder the packet transmission sequence. Consequently, the burst loss effect can be mitigated at the expense of an increased end‐to‐end delay. Alternatively, the reversed interleaving scheme, namely, interleaved coding‐sequential transmission (ICST), performs FEC coding in an interleaved manner and transmits the packets sequentially based on their generation order in the application. In this study, the analytical FEC model is constructed to evaluate the performance of the SCIT and ICST schemes. From the analysis results, it can be observed that the interleaving delay of ICST FEC is reduced by transmitting the source packets immediately as they arrive from the application. Accordingly, an Enhanced ICST scheme is further proposed to trade the saved interleaving time for a greater interleaving capacity, and the corresponding packet loss rate can be minimized under a given delay constraint. The simulation results show that the Enhanced ICST scheme achieves a lower packet loss rate and a higher peak signal‐to‐noise‐ratio than the traditional SCIT and ICST schemes for video streaming applications.