Usage parameter control cell loss effects on MPEG video

This paper presents our study on the effects of cell loss on QoS for the MPEG video transmission in ATM networks. A new evaluation method for the cell loss behavior of ATM networks due to congestion is proposed. The method uses a real MPEG video sequence as the input source and the MPEG video data is packetized into ATM cells. A cell discard process model called the Gilbert model is used to generate a cell discard pattern. The pattern represents the occurrences of ATM lost cells due to buffer overflow in multiplexers, Finally, the effects of cell loss on QoS is evaluated by de-packetizing (assembling) the resultant ATM cells back into MPEG video data, The QoS evaluations presented in this paper include subjective and objective evaluations of video quality under different network conditions. The evaluation results show that the effects of cell loss on QoS for MPEG video transmission in ATM networks depend on factors such as the burst cell loss length and the MPEG coding modes used.

We present the results of a simulation study for a virtual circuit connection over an ATM network where forward error correction is performed at both the ATM cell level and the packet data unit (PDU) level. A main conclusion of this study is that at low loads ATM cells from the same source dominate in the switch buffers, while at high loads there is a mixing of ATM cells from different sources. For the latter case, ATM cell level coding performs better, while for the former, PDU level coding performs better. The combination of the two techniques has the best overall performance.

MPEG-2 video communications over ATM networks is one of the most active research areas in the field of computer communications. In the transmission process of a variable bit rate video signal over an ATM network, cells are inevitably exposed to delays, errors and losses due to the statistical multiplexing used in these networks. These phenomena affect the quality of the video signal and without adequate measures to control the propagation of the impairments the quality of the service may fall below acceptable levels. In this paper, we study the impact of cell losses on the quality of a MPEG-2 video sequence encoded in a variable bit rate mode. We introduce a set of control mechanisms at different levels of the protocol architecture to be used in MPEG-2-based video communications systems using ATM networks as their underlying transmission mechanism. Our results (using different video sequences) show the effectiveness to improve the video quality by using a structured set of control mechanisms to overcome for the loss of cells carrying VBR MPEG-2 video streams. We argue that in order to be able to create video systems able to cope with cell losses encountered in computer communications systems, a structured set of error-resilient protocol mechanisms is needed.

We present a region of interest (ROI) based rate control for H.263 compatible video conferencing. A face detection and tracking scheme with very low complexity is proposed for segmentation. By analyzing quadratic rate models in frame layer, video object plane (VOP) layer, and macroblock (MB) layer extracted from the test data, a quadratic rate model at the MB layer with a modified physical meaning is proposed to improve the model accuracy. The basic idea is to use a group of uncoded MBs in the current VOP instead of individual MBs to update model parameters. A joint VOP layer and MB layer rate control algorithm is proposed. The VOP layer rate control assigns target bit rate for each VOP based on the coding complexity and visual importance, and determines an average quantization parameter (QP) for each VOP. Some new features of MB layer rate control are designed to utilize both average statistics of a VOP and individual statistics of MB together. The performance is compared with conventional TMN8 and object-based VM8 rate control, better peak SNR (PSNR) for ROI, and more accurate rate control can be achieved for various video sequences. The proposed rate control algorithm can be extended for H.264 ROI-based scalable video coding.

In H.264/AVC rate control algorithm, the bit allocation process and the QP determination are not optimal. At frame layer, there is an implicit assumption considering that the video sequence is more or less stationary and consequently the neighbouring frames have similar characteristics. So, the target Bit-Rate for each frame is estimated using a straightforward process that allocates an equal bit budget for each frame regardless of its temporal and spatial complexities. This uniform allocation is surely not suited especially for all types of video sequences. The target bits determination at macroblock layer uses the MAD (Mean Absolute Difference) ratio as a complexity measure in order to promote interesting macroblocks, but this measure remains inefficient in handling macroblock characteristics. In a previous work we have proposed Rate-Quantization (R-Q) models for Intra and Inter frames used to deal with the QP determination shortcoming. In this paper, we look to overcome the limitation of the bit allocation process at the frame and the macroblock layers. At the frame level, we enhance the bit allocation process by exploiting frame complexity measures. Thereby, the target bit determination for P-frames is adjusted by combining two temporal measures: The first one is a motion ratio determined from actual bits used to encode previous frames. The second measure exploits both the difference between two consecutive frames and the histogram of this difference. At macroblock level, the visual saliency is used in the bit allocation process. The basic idea is to promote salient macroblocks. Hence, a saliency map, based on a Bottom-Up approach, is generated and a macroblock classification is performed. This classification is then used to accurately adjust UBits_H264 which represents the usual bit budget estimated by H.264/AVC bit allocation process. For salient macroblocks the adjustment leads to a bit budget which is always larger than UBits_H264. The extra bits added to code these macroblocks are deducted from the bit budget allocated to the non-salient macroblocks. Simulations have been carried out using JM15.0 reference software, several video sequences and different target Bit-Rates. In comparison with JM15.0 algorithm, the proposed approach improves the coding efficiency in terms of PSNR/PSNR-HVS (up to +2dB/+3dB). Furthermore, the bandwidth constraint is always satisfied because the actual Bit-Rate is always lower than or equal to the target Bit-Rate.

In this paper, a rate controller for a H.261 based video encoder is proposed. The rate controller adaptively chooses the optimal channel matched quantizer using a stochastic learning automaton. The automaton learns the channel characteristics based on a one bit feedback from the decoder. The rate control algorithm is shown to converge to the optimal choice of the quantizer very quickly for various channel bit error probabilities and for different video sequences. The adaptation can be achieved in real-time. The peak signal to noise ratio of the received video signal is seen to be better using the proposed approach.

The transmission of ATM cells through a virtual path (VP) which comprises one or more virtual channel (VCs) multiplexed through a VP is examined. Constant bit rate (CBR) sources are used. During the process, there are ATM cell loss, cell delay, and the performance degradation. This paper proposes the techniques of CDMA, OOCs code and 2/sup n/ prime code for improve the performance and to compare two codes when using an ATM network. The results show that when using the same code length a 2/sup n/ prime sequence code has a greater performance than OOCs code.

A statistical multiplexer with infinite buffer capacity, a finite number of inputs and underlying train arrivals is analyzed in this paper. The train arrivals model appears within the large self-routing ATM switching network, where an ATM cell is internally converted into a constant number of smaller data units called minicells, which are multiplexed and switched consecutively like a train. An alternative way to derive the mean buffer contents is described. In order to obtain the asymptotic buffer behavior, an analytical approach is developed. This approach is general, and can easily be extended to other types of correlated arrivals. The impact of train arrival processes on the buffer performance is illustrated via some numerical examples.

This paper shows how to design an ATM network, for real-time traffic, such that under full network load (i) the maximum delay of a low-rate voice connection is minimized, (ii) the delay uncertainty or jitter is a fixed network parameter, and (iii) the required buffer sizes (inside the network) to ensure loss-free routing is minimized. In addition, this design does not exclude the integration of other classes of traffic, such as connectionless, which have lower priority. The isochronous timing information is used only for regulating and pacing the traffic forwarded inside the network rather than for routing as in traditional circuit switching networks. This means that an ATM cell is sent from one switch to another not at a very specific time but within a time frame of a relatively long duration as compared to the cell transmission time. The routing of cells of each connection is based on VCI and VPI, and as a result, timing errors do not affect the ATM routing. Analytical and simulation results are presented, which demonstrate the effectiveness of this switching approach. >

This paper presents a robust and flexible real-time scheme for determining appropriate parameter values for usage parameter control (UPC) of an arbitrary source in an asynchronous transfer mode network. In our approach, the UPC parameters are chosen as a function of the statistical characteristics of the observed cell stream, the user's tolerance for traffic shaping, and a measure of the network cost. For this purpose, we develop an approximate statistical characterization for an arbitrary cell stream. The statistical characterization is mapped to a UPC descriptor that can be negotiated, with the network. The selected UPC descriptor is optimal in the sense of minimizing a network cost function, subject to meeting user-specified constraints on shaping delay. The UPC estimation scheme is extended to adapt to slow time-scale changes in traffic characteristics via dynamic renegotiation of the UPC parameters. We illustrate the effectiveness of our methodologies with examples taken from MPEG video sequences.

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.

Wireless asynchronous transfer mode (WATM) networks pose new traffic management problems. One example is the effect of user mobility on usage parameter control (UPC). If the UPC algorithm resets after each handoff between wireless-cells, then users can cheat on their traffic contract. In brief, UPC is analogous to a digital filtering operation, with an associated convergence time. If a user changes wireless-cells before the UPC algorithm converges then UPC can not determine whether that user complied with their traffic contract. This paper derives explicit relationships between a user's traffic parameters (peak cell rate, sustained cell rate and maximum burst size), their transit time per wireless-cell, their maximum sustained cheating-rate and the generic cell rate algorithm's (GCRAs) limit (L) parameter. It also shows that the GCRA can still effectively police constant bit rate (CBR) traffic, but not some types of realistic variable bit rate (VBR) traffic. Finally, it discusses three solutions for this new network impairment.

We propose novel usage parameter control and network parameter control algorithms for ATM networks using artificial neural networks. The proposed algorithms employ ANN to point the cells for tag or discard for heterogeneous multimedia traffic with multiple quality of service and transmission priority requirements. Comparisons of the proposed algorithm to the generic cell rate algorithm are included and some future direction of research are described.

A method for optimising the usage parameter control (UPC) of video services in an ATM network is proposed. The video encoder uses a sliding window for its maximum network utilisation without exceeding the declared bit rate at the call setup. At video scene cuts, where the peak bit rates are generated, the coder limits its generated bit rate through a leaky bucket mechanism. The impacts of optimum UPC on both coding and network performance are investigated. It is shown that for video sequences with scene cuts there is an optimum value of peak-to-mean ratio where the network performance is optimised.

Much of the Internet's communication is encrypted, and its content is only accessible at two endpoints, a client and a server. However, any encryption requires a key that must be negotiated without being revealed to potential attackers. The so-called TLS (Transport Layer Security) handshake is often used for this task without obviating that many fundamental parameters of TLS connections are transmitted explicitly. Thus, third parties have access to metadata, including information about the endpoints, how the connection is used. On the other hand, QoS is considered the central part of the communication used to judge the deliverable quality through several parameters (latency, jitter ...). This document describes a secure approach and meets mainly the requirements of quality of service on a communication channel (free, loaded, congested ...), using the robustness and flexibility of the TLS protocol represented on the characteristics of existing encryption keys on its list of "ciphers suites." We focused more particularly on the AES key (Advanced Encryption Standard), including the different sizes (128,192,256), given its resistance to various classical attacks (differential, linear, ...) and its lightness compared to other protocols such as DES, 3DES ... This method is useful in continuous communications in a time axis (video sequence, VOIP call...).