Constant Bit Rate Research Articles

On Network Planning for Video Services Over IP Networks

Video transmission over IP requires very high data rate for a good performance. Variable bit rate (VBR)-coded video is more popular than constant bit rate (CBR)-coded video due to its statistical multiplexing gain and stable quality level. This phenomenon forms a real challenge for network planning and especially for appropriate capacity evaluation. In this work, we propose a new capacity planning model that is especially designed for VBR video transmission over IP networks. This new method takes the quality level and the desired delay threshold as its main design parameters. We investigate this new method by means of simulations and we finally validate its applicability for real-time video transmission with respect to different system parameters.

Constant Bit-Rate Multi-Stage Rate Control for Rate-Distortion Optimized H.264/AVC Encoders

H.264/AVC encoder employs rate control to adaptively adjust quantization parameter (QP) to enable coded video to be transmitted over a constant bit-rate (CBR) channel. In this topic, bit allocation is crucial since it is directly related with actual bit generation and the coding quality. Meanwhile, the rate-distortion-optimization (RDO) based mode-decision technique also affects performance a lot for the strong relation among mode, bits, and quality. This paper presents a multi-stage rate control scheme for R-D optimized H.264/AVC encoders under CBR video transmission. To enhance the precision of the complexity estimation and bit allocation, a frequency-domain parameter named mean-absolute-transform-difference (MATD) is adopted to represent frame and macroblock (MB) residual complexity. Second, the MATD ratio is utilized to enhance the accuracy of frame layer bit prediction. Then, by considering the bit usage status of whole sequence, a measurement combining forward and backward bit analysis is proposed to adjust the Lagrange multiplier λMODE on frame layer to optimize the mode decision for all MBs within the current frame. On the next stage, bits are allocated on MB layer by proposed remaining complexity analysis. Computed QP is further adjusted according to predicted MB texture bits. Simulation results show the PSNR improvement is up to 1.13dB by using our algorithm, and the stress of output buffer control is also largely released compared with the recommended rate control in H.264/AVC reference software JM13.2.

A cooperative PC/network-processor platform for gigabit traffic generation

Evaluating the performance of high-speed networks is a critical task that requires reliable tools to generate traffic workloads at high rates. The current software tools provide poor performance in terms of packets per second and timing/rate accuracy, because of the intrinsic limitations of the PC architecture. This paper proposes a different approach, based on a cooperative PC/NP architecture: an advanced software tool runs on a host PC and instructs the processing engines of an Intel IXP2400 network processor, which take care of the actual traffic generation. This way we keep the high flexibility of PC tools while achieving the high packet rates of hardware solutions. The system is able to generate several traffic models (to date, constant bit rate, Poisson and Poissonian arrival of burst are implemented); however the tool is easily extensible in that any other traffic models can be added through the available APIs. In order to improve the system precision, a time correction mechanism is also introduced. Finally, the capability of manipulating and reproducing an actual trace is provided.

Re-Multiplexing ISDB-T BTS Into DVB TS for SFN

The Single Frequency Network (SFN) allows a single TV channel to cover a great area in the same frequency, increasing the spectrum efficiency and signal quality. In this case, all the programs are re-multiplexed into a Transport Stream (TS) and distributed to the re-transmitter stations. The Integrated Digital Services Broadcasting-Terrestrial (ISDB-T) has been projected to operate in SFN, and uses a TS-adapted version called Broadcast Transport Stream (BTS). The BTS has a constant bit rate and packet size of 204 bytes. Owing to these characteristics, most network adapters are non compliant with this signal type. This article proposes a new re-multiplexer algorithm of the BTS signal that allows the encapsulation of the BTS into the Digital Video Broadcasting Transport Stream (DVB TS) and vice versa. This technique makes the BTS distribution possible through satellite, allowing the large SFN construction and accomplishing lesser bit rate than the original BTS.

A Dynamic Quality-Adjustable H.264 Video Encoder for Power-Aware Video Applications

This paper proposes a dynamic quality-adjustable H.264 baseline profile (BP) video encoder that comprises 470 Kgates and 13.3 kB SRAM in a core size of 4.3 × 4.3 mm2 using TSMC 0.13 ?m 1P8M CMOS technology. Exploiting parameterized algorithms for motion estimation and intra prediction, the proposed design can dynamically configure the encoding modes with the design trade-off between power consumption and video quality for various video encoding applications. In addition, the proposed basic unit (BU)-based rate control hardware can maintain a constant and stable bit rate for network video transmission. It achieves real-time H.264 video encoding on CIF, D1, and HD720@30 frames/s with 7 mW to 25 mW, 27 mW to 162 mW, and 122 mW to 183 mW power dissipation in different quality modes.

Optical Packet Contention Resolution Through Edge Smoothing Into Decomposed Subflows

Motivated by the difficulty of implementing optical packet switches having large optical buffers, we propose a new, simple, and effective way of applying electronic traffic smoothing at the edge of an optical packet switched network to reduce core optical buffering requirements. A single class of service is assumed. Incoming traffic that is destined for a particular egress edge router is smoothed at each ingress edge router by a buffer scheduling algorithm and decomposed into several constant bit rate subflows, each with a uniform packet interarrival time. Each such subflow emerging from a core switch also retains these traffic characteristics. The rates of these subflows obey a simple mathematical relationship, and it is shown that, with the appropriate queuing strategy, this substantially reduces the core buffering requirements. Indeed, it is shown through analysis and simulation that scalability is improved over existing approaches, and computation is simplified. The proposal in this paper requires much less core buffering than one without smoothing, while compared with other smoothing proposals, it is more scalable both to large networks and a large number of smoothed flows. A condition on the fiber delay line buffer depth requirements is also derived for no packet loss to take place in any of the scenarios modeled in this paper. Furthermore, it is confirmed through simulation that packet jitter is not generated in the network core.

A Policy-Based Admission Control Scheme for Voice over IP Networks

Problem statement: In Voice Over IP (VOIP) network, when more calls are admitted to the network, more voice packet traffic is created. Since bandwidth is always limited, this may result network congestion and/or may affect voice quality. Thus, we needed a mechanism for improving the Quality of Service (QoS) by controlling VOIP calls admission. Approach: Given a specified bandwidth and a constant background data rate, we attempted to explore the effect of Open Window and Leaky Bucket admission schemes on VoIP calls quality. These policy-based admission controls were simulated using NS-2 Simulator. The inter-arrival time distribution for the network background data traffic was assumed to be deterministic with a Constant Bit Rate (CBR). Voice packets traffic inter-arrival time is assumed to have an exponential distribution. Each voice call has a rate of 64 kbps for duration of 120 min. Results: Various performance measures of VoIP calls and packet traffic were evaluated including: packet loss, packet drop rate, delay, jitter and call rejection rate. Performance results of the experiment are summarized in a power ratio index which presented the impact of a collection of performance parameters on VoIP service quality. Conclusion: Implementing a policy based admission scheme on VoIP network will improve its QoS and the degree of improvement depends on the network setting parameters. If threshold rate for call admission is set above network ceiling bandwidth, leaky bucket will result a higher and unacceptable jitter. Overall, leaky bucket scheme was considered inferior when compared to open window for improving QoS of VoIP.

Fast adaptive routing supporting mobile senders in Source Specific Multicast

IP multicast deployment recently progresses, but group services often remain restricted to limited domains and fail to comply with route-optimizing mobility management of the next generation Internet. Source Specific Multicast (SSM) facilitates transparent inter-domain routing and is expected to globally disseminate to many users and services. However, mobility support for Source Specific Multicast is still known to be a major open problem. In this paper, we propose the Enhanced Tree Morphing (ETM) protocol for extending SSM routing to mobile multicast sources. The scheme dynamically adapts SSM forwarding states to sender mobility, and transforms (morphs) source specific distribution trees into new, optimal trees rooted at a relocated source. ETM is simple, robust and secure, while it admits superior performance in packet forwarding at a low signaling overhead. Extensive evaluations based on a full protocol implementation, and simulations based on real-world topology data are performed, granting full insight into the properties of packet loss and delay stretch, protocol convergence times and router state evolution during single and rapidly repeated handovers. In a constant bit rate scenario, an ETM source handover typically leads to a slightly increasing delay of the first data packet, only. When operating on realistic network topologies, the protocol uniformly converges within less than 50 ms, thereby sustaining robustness under rapid source movement at all speeds common to our mobile world. Further optimizations are identified for FMIPv6 and for multihomed nodes.

Power Management and Data Rate Maximization in Wireless Energy Harvesting Sensors

This paper considers the problem of power management and throughput maximization for energy neutral operation when using an energy harvesting sensor (EHS) to send data over a wireless link. The EHS is assumed to be able to harvest energy at a constant rate, and use a fixed part of the energy harvested in a slot for measuring the channel state. The rest of the energy harvested is available for transmission, however, it can be stored in an inefficient battery if it is not fully utilized. The key constraint that the EHS needs to satisfy is energy neutrality, i.e., the expected energy drawn from the battery should equal the expected energy deposited into the battery. In this scenario, two popular models for data transmission are contrasted: the constant bit rate (CBR) model and the variable bit rate (VBR) model. In the CBR model, it is assumed that the EHS are designed to transmit data at a constant rate (using a fixed modulation and coding scheme) but are power-controlled. In the VBR model, the EHS selects both the transmit power and the data rate of transmission in each slot based on the channel instantiation. A framework under which the system designer can optimize several parameters of the EHS that determine the average data rate performance when the channel is Rayleigh fading is developed. Using this framework, the two transmission schemes are contrasted. It is shown that, with the right choice of parameter settings, the CBR scheme can perform nearly as well as the VBR scheme at significantly lower complextiy. The usefulness and validity of the framework developed is illustrated through simulations for specific examples.

Optimal choice of Reed-Solomon codes to protect against queuing losses in wireless networks

This article proposes algorithms to determine an optimal choice of the Reed-Solomon forward error correction (FEC) code parameters ( n,k) to mitigate the effects of packet loss on multimedia traffic caused by buffer overflow at a wireless base station. A network model is developed that takes into account traffic arrival rates, channel loss characteristics, the capacity of the buffer at the base station, and FEC parameters. For Poisson distributed traffic, the theory of recurrent linear equations is applied to develop a new closed form solution of low complexity of the Markov model for the buffer occupancy. For constant bit rate (CBR) traffic, an iterative procedure is developed to compute the packet loss probabilities after FEC recovery.

Research on the Performance of Protocols and the Evaluation Metric for VIDEO Transmissions in an Ad Hoc Network

Video transmission effectiveness in the Ad Hoc network is becoming important recently, if different routing protocols are applied. Some researchers conclude that the reactive protocols are better for file transfer protocol (FTP) and constant bit rate (CBR) or hypertext transfer protocol (HTTP) transmission in an Ad Hoc wireless network but the performance report of video transmission is not much. This study adopts Qualnet (Network Simulator) as a simulation tool for environmental designing and performance testing, and employs an experimental design with eight groups. Our experiment shows that: (1) The performance of AODV (reactive) protocol is better than DSDV, ZRP and DSR when the transmission load has only one video stream; (2) Proactive (DSDV) and Hybrid protocols (ZRP) are better for a smaller Ad Hoc network when it transmits a video stream with some applications (VoIP, FTP and CBR). We conclude that packet loss rate is sensitive to the quality of video transmission and it has negative relationship with Peak Signal‐to‐Noise Ratio (PSNR) value. In addition, our experiment also shows that PSNR is a simple Metric for the performance evaluation of video transmission.

A Practical Method for Improving Delay Performance of Time Slot Allocation in DVB-RCS

In digital video broadcasting and return channel via satellite (DVB-RCS) systems, the time slot location assigned to a given traffic in multiple frequency-time division multiple access (MF-TDMA) frame has significant effects upon the traffic delay performance. This article proposes models to analyze the relationships among frame length, bandwidth assignment (assigned time slot count), time slot location in frame, and traffic delay performance for traffics of constant bit rate (CBR) and variable bit rate (VBR). From the proposed models, it is found that the time slot location plays a leading role in affecting the traffic delay performance, which ameliorates more when the time slot location is more uniform in frame. Compared with the traditional mode of consecutive time slot allocation, the mode of uniform time slot allocation can effectually reduce the average delay and jitter, which will not vary with the frame length. Furthermore, with the mode of uniform time slot allocation increasing bandwidth assignment could promote the delay performance more effectually. This article also puts forward a practical time slot uniform allocation method (TUAM), which is based on the DVB-RCS standard and compatible with it. The computer simulation, apart from testifying to the validity of the model, shows that it leads to efficacious reduction of the mean traffic delay and jitter. By adopting the method, the real-time traffic-like voice over IP (VoIP) traffic's packets will not suffer from dropping due to an overlong end-to-end delay, thus improving quality of service (QoS) provision for real-time traffics in DVB-RCS systems.

Using GPS-based link exclusion to reduce overhead and increase route reliability in FTP/TCP-based MANETs

In this paper, we show that the use of global positioning system (GPS) information in a mobile ad hoc network (MANET) running the file transfer protocol (FTP) over the transmission control protocol (TCP) can be used to block the inclusion of unreliable links within a discovered route by not allowing the nodes outside of a given distance to participate in their formation. In doing so, this creates more reliable routes over time and lowers the amount of network overhead necessary to maintain the flow of data packets. This approach to route formation is validated using the NS-2.31 network simulator under different types of multipath fading models as well as varying network dimensions and node densities. While this GPS blocking scheme has been explored previously in a constant bit rate (CBR) over a user datagram protocol (UDP) setting, it has not yet been tested previously under the more common real-world conditions of FTP running over TCP.

Scheduling Algorithm to Provide QoS over a Shared Wireless Link

Several scheduling algorithms have been proposed for the downlink of a Code Division Multiple Access (CDMA) system with High Data Rate (HDR). Modified Largest Weighted Delay First (M-LWDF) scheduling algorithm selects a user according to the user current channel condition, user head-of-line packet delay and user Quality of Service (QoS) requirement. Proportional Fair (PF) scheduling algorithm has also been proposed for CDMA/HDR system and it selects a user according to the ratio of the user current channel rate and the user average channel rate, which provides good performance in terms of fairness. However, when variable bit rate (VBR) traffic is considered under different channel conditions for each user, both schedulers' performance decrease. M-LWDF scheduler can not guarantee the QoS requirement to be achieved and PF scheduler can not achieve a good fairness among the users. In this work, we propose a new scheduling algorithm to enhance M-LWDF and PF schedulers performance. Proposed scheduler selects a user according to the user input traffic characteristic, user current channel condition and user QoS requirement, which consists of a delay value with a maximum violation probability. We consider the well-known effective bandwidth expression, which takes into account the user QoS requirement and the user input traffic characteristics, to select a user to be scheduled. Properties of the proposed scheduling algorithm are investigated through simulations with constant bit rate (CBR) and VBR flows and performance comparisons with M-LWDF and PF schedulers. The results show a better performance of the proposed scheduler compared with M-LWDF and PF schedulers.

Output characterization of constant bit rate traffic in FIFO networks

We provide an analytical proof that the departure rate of a CBR flow at an overloaded link with FIFO buffers is proportional to the flow's share of the total offered load at the link. This property of FIFO scheduling was recently validated in in a series of traffic measurement experiments. An extension of the analysis to a multi-node scenario shows that the output rate of a flow in a network with many overloaded FIFO switches approaches the pessimistic values given by blind multiplexing.

Improved Vector Quantization Based Block Truncation Coding Using Template Matching and Lloyd Quantization

Block truncation coding (BTC) is an efficient image compression algorithm that generates a constant output bit-rate. For color image compression, vector quantization (VQ) is exploited to improve the coding efficiency. In this letter, we propose an improved VQ based BTC (VQ-BTC) algorithm using template matching and Lloyd quantization (LQ). The experimental results show that the proposed method improves the PSNR by 0.9dB in average compared to the conventional VQ-BTC algorithms.

A Resonance Theory for Open Quantum Systems with Time-Dependent Dynamics

We develop a resonance theory to describe the evolution of open systems with time-dependent dynamics. Our approach is based on piecewise constant Hamiltonians: we represent the evolution on each constant bit using a recently developed dynamical resonance theory, and we piece them together to obtain the total evolution. The initial state corresponding to one time-interval with constant Hamiltonian is the final state of the system corresponding to the interval before. This results in a non-markovian dynamics. We find a representation of the dynamics in terms of resonance energies and resonance states associated to the Hamiltonians, valid for all times $t\geq 0$ and for small (but fixed) interaction strengths. The representation has the form of a path integral over resonances. We present applications to a spin-fermion system, where the energy levels of the spin may undergo rather arbitrary crossings in the course of time. In particular, we find the probability for transition between ground- and excited state at all times.

Guaranteed QoS Provision Scheduling Mechanism for CBR Traffic in IEEE 802.16 BWA Systems

Constant Bit Rate (CBR) traffic is expected to be an important traffic source in wireless networks. Such sources usually have stringent delay and loss requirements and in many cases they should be delivered exactly as they were generated. In this paper, we propose a novel uplink packet scheduling policy for CBR traffic to support real-time applications, with strict delay requirements in IEEE 802.16 broadband wireless access (BWA) systems. Under such a scheme, the CBR traffic was characterized by its sustained traffic rate and tolerated jitter. The proposed transmit permission policy can derive sufficient conditions such that all the CBR sources satisfy their delay constraints to provide deterministic quality of service (QoS) guarantees. In addition to theoretical analysis, simulations are conducted to evaluate the performance of the proposed scheme. As it turns out, our design provides a good quality performance in the IEEE 802.16 BWA systems.

A Novel MAC Protocol for QoS in Ad Hoc Wireless Networks

We propose a medium access control (MAC) protocol for real-time applications in one-hop ad-hoc wireless networks. It is a distributed mechanism that takes account of priority and has a bounded packet delay. Nodes use energy signals to contend for the right to access the channel. Nodes, which have a packet to transmit, send energy signals or listen to the channel based on their binary frame. The node that has sent energy signals and has not heard any energy signals wins the right to access the channel. We use two schemes to determine the binary frame: at the beginning of a session, a node determines it based on its priority level and a random number; after successful transmission, based on a count of successful packet transmissions. With the first scheme, in order to reduce contention losses, the nodes that had won the right to access the channel but failed in transmission have priority over the other nodes. With the second scheme, the node that has the largest count, the one that has been waiting the longest, can send a packet without risking collision. The protocol provides higher probability of successful transmission and a limit on maximum packet delay. An analysis of the protocol provides conditions for the protocol to be stable. We evaluate the performance of the proposed protocol using simulations of a network with a mixed population of data and real-time nodes, whose source is constant bit rate (CBR) and a two state Markov on/off process.

Iterative Rate-Distortion Optimization of H.264 With Constant Bit Rate Constraint

In this paper, we apply the primal-dual decomposition and subgradient projection methods to solve the rate-distortion optimization problem with the constant bit rate constraint. The primal decomposition method enables spatial or temporal prediction dependency within a group of picture (GOP) to be processed in the master primal problem. As a result, we can apply the dual decomposition to minimize independently the Lagrangian cost of all the MBs using the reference software model of H.264. Furthermore, the optimal Lagrange multiplier lambda* is iteratively derived from the solution of the dual problem. As an example, we derive the optimal bit allocation condition with the consideration of temporal prediction dependency among the pictures. Experimental results show that the proposed method achieves better performance than the reference software model of H.264 with rate control.