An adaptive control for video transmission over bluetooth

This paper is concerned with transmission of a moving picture expert group (MPEG) video stream over a Bluetooth channel, using an adaptive neuro-fuzzy technique. MPEG variable bit rate (VBR) video sources over a network generally experience long delay and unacceptable data loss, due to high variations in bit rate. Furthermore, transmission rate could be unpredictable in a Bluetooth network due to interferences by other wireless devices or general Bluetooth channel noises. Subsequently, it is almost impossible to transmit VBR data sources over Bluetooth without excessive delay or data loss. In this work, an adaptive scheme is introduced so that the controller may adjust itself to the current state of the system under control. This paper utilizes a traffic-shaping buffer to prevent excessive back-to-back transmissions of MPEG VBR data sources. A novel adaptive neuro-fuzzy scheme regulates the output rate of the buffer to ensure that the video stream from the host conforms to the traffic conditions of the Bluetooth channel during the transmission period. The computer simulation results show that the use of the neuro-fuzzy controller reduces excessive delay and data loss at the host-controller-interface, as compared with a conventional VBR video transmission and a rule-based fuzzy controller (RBF1) in Bluetooth.     

A fuzzy control scheme for video transmission in Bluetooth wireless

This paper is concerned with transmission of Moving Picture Expert Group (MPEG) video over a Bluetooth wireless network using a fuzzy approach. MPEG Variable Bit Rate (VBR) video sources suffer from long delay and excessive loss due to the sudden bursts in bit rate. Constant Bit Rate (CBR) encoding scheme may work well for a network with a guaranteed bandwidth. However, a Bluetooth channel is subject to wireless interference and can never guarantee a constant bandwidth. Subsequently, it is impossible to transmit a CBR video over Bluetooth wireless without data loss or image quality degradation. To resolve this problem, a fuzzy control system is introduced at the Host Controller Interface (HCI). The system consists of a traffic-shaping buffer whose role is to prevent excessive back-to-back cells being generated during the peak transmissions of MPEG video sources. The output bit rate of the traffic-shaping buffer is controlled by a fuzzy controller to ensure that the video stream from the host conforms to the traffic condition of the Bluetooth channel. Another fuzzy controller regulates the average arrival bit rate to the traffic-shaper to guarantee that the buffer is neither full nor empty. Computer simulation results demonstrate that the use of the fuzzy controllers reduces excessive data loss at the HCI as compared with an open loop VBR/CBR video transmission in Bluetooth.     

Two neuro-fuzzy control schemes for a traffic-regulating buffer in wireless technology

Bluetooth wireless operates in 2.4-GHz Industrial Scientific and Medicine (ISM) frequency, which may interfere with other devices functioning within the same frequency band, such as WiFi. Furthermore, Moving Picture Expert Group (MPEG) variable bit rate (VBR) video demands larger and more stable bandwidth and may cause data loss and time delay as a result of the high variation in bit rate in Bluetooth channels with limited bandwidth. To address these issues, two new neuro-fuzzy schemes are developed to control the input and output of a buffer referred to here as the traffic-regulating buffer. Regarding the input of this buffer, a rule-based fuzzy scheme is introduced and supervised by a neural network technique as a master controller to monitor the arrival rate to the buffer. The output of the traffic-regulating buffer is observed by another rule-based fuzzy scheme and is supervised by a second neural network to monitor the departure rate. Computer simulation results demonstrate that the two proposed neuro-fuzzy models reduce standard deviation and excessive data loss, and they also show an improved picture quality as compared with conventional MPEG VBR video over a Bluetooth channel.     

基于模糊控制的ATM网络VBR视频传输平滑策略

VBR视频传输的突发性是影响ATM网络服务质量的关键因素,文中通过模糊控制方法对视频传输流量的阈值进行监控,实现了接入的平滑并可动态调整传输速度。文中以传输流量及这发级作为评价指标进行分析,结果表明VBR视频传输自适应平滑策略的实效性。     

Transmission control of Variable-Bit-Rate video streaming in UMTS networks

Variable-Bit-Rate (VBR) video transmission over UMTS networks is assuming an ever growing importance, then scheduling VBR data over wireless channels attracts great interest today. This paper proposes an on-line PI control algorithm for easily and suitably scheduling interactive multimedia transmission of VBR video streams. It dynamically adapts the transmitted bit rate to the user's actions that can suddenly and highly modify the client buffer level. The Real-time Transport Control Protocol (RTCP) packets periodically feed back the buffer level to the server for keeping the free buffer space at 50% of its capacity, so as to prevent both buffer overflow and underflow. Numerical results of different simulated scenarios show the effectiveness of the proposed controller in comparison with two well-established algorithms, enhanced with RTCP information.     

Trade-Offs in Bit-Rate Allocation for Wireless Video Streaming

One of the central problems in wireless video transmission is the choice of source and channel coding rates to allocate the available transmission rate optimally. In this paper, we present a structural distortion model for video streaming over time-varying fading channels. Based on the model, we study the end-to-end distortion for various bit-rate allocation strategies and channel conditions. We show that the robustness to channel variations is crucial for the streaming performance when frequent bit-rate adaptations are not feasible. It is achieved at the expense of higher source distortion in the encoder. Our findings are illustrated on a practical problem of distortion-optimal selection of transport formats in an adaptive modulation and coding (AMC) scheme used in HSDPA.     

Cycle-Based Rate Control for One-Way and Interactive Video Communications Over Wireless Channels

We propose a joint source-rate/channel-code control scheme for streaming video over a wireless channel. The scheme is designed to maximize the achievable source rate while guaranteeing an upper bound on the probability of starvation at the playback buffer. It can be applied to both one-way and interactive video communications. Rate control is performed adaptively on a per-cycle basis, where a cycle consists of a "good" channel period and the ensuing "bad" period. This cycle-based approach has two advantages. First, it reduces the fluctuations in the source bit rate, ensuring smooth variations in video quality. Second, it makes it possible to derive simple expressions for the starvation probability at the playback buffer, which we use to determine the optimal source rate and channel code for the good and bad periods of the subsequent cycle     

联合H.264和多码率Turbo码的无线视频传输

提出了一种无线视频流联合信源信道编码传输的具体方案,结合H.264视频编码,对其进行合理数据分割后采用UEP技术,并联合信道设计了一个支持信道UEP的多码率Turbo编解码器,同时把Turbo编码与type-Ⅲ HARQ混合自动重传请求机制结合以达到更可靠的数据传输保证。通过仿真实验证明,这种设计方法可进一步优化系统在恶劣的无线信道下的链路吞吐量和取得更好的视频重建质量。     

Priority-based Media Delivery using SVC with RTP and HTTP streaming

Media delivery, especially video delivery over mobile channels may be affected by transmission bitrate variations or temporary link interruptions caused by changes in the channel conditions or the wireless interface. In this paper, we present the use of Priority-based Media Delivery (PMD) for Scalable Video Coding (SVC) to overcome link interruptions and channel bitrate reductions in mobile networks by performing a transmission scheduling algorithm that prioritizes media data according to its importance. The proposed approach comprises a priority-based media pre-buffer to overcome periods under reduced connectivity. The PMD algorithm aims to use the same transmission bitrate and overall buffer size as the traditional streaming approach, yet is more likely to overcome interruptions and reduced bitrate periods. PMD achieves longer continuous playback than the traditional approach, avoiding disruptions in the video playout and therefore improving the video playback quality. We analyze the use of SVC with PMD in the traditional RTP streaming and in the adaptive HTTP streaming context. We show benefits of using SVC in terms of received quality during interruption and re-buffering time, i.e. the time required to fill a desired pre-buffer at the receiver. We present a quality optimization approach for PMD and show results for different interruption/bitrate-reduction scenarios.     

Channel Aware Multiuser Scalable Video Streaming Over Lossy Under-Provisioned Channels: Modeling and Analysis

In this paper, we analyze the performance of media-aware multiuser video streaming strategies in capacity limited wireless channels suffering from latency problems and packet losses. Wireless video streaming applications are characterized by their bandwidth-intensity, delay-sensitivity, and loss-tolerance. Our main contributions include i) a rate-minimized unequal erasure protection (UXP) scheme, ii) an analytical expression for packet delay and play-out deadline of UXP protected scalable video, iii) a loss-distortion model for hierarchical predictive video coders with picture copy concealment, iv) an analysis of the performance and complexity of delay-aware, capacity-aware, and optimized UXP streaming scenarios, and v) we show that the use of unequal error protection causes a rate-constrained optimization problem to be nonconvex. Performance evaluations using a 3GPP network simulator show that, for different channel capacities and packet loss rates, delay-aware nonstationary rate-allocation streaming policies deliver significant gains which range between 1.65 dB to 2 dB in average Y-PSNR of the received video streams over delay-unaware strategies. These gains come at a cost of increased offline computation which is performed prior to the start of the streaming session or in batches during transmission and therefore, do not affect the run-time performance of the streaming system.      

Enabling broadcast of user-generated live video without servers

We are witnessing the unprecedented popularity of User-Generated-Content (UGC) on the Internet. While YouTube hosts pre-recorded video clips, in near future, we expect to see the emergence of User-Generated Live Video, for which any user can create its own temporary live video channel from a webcam or a hand-held wireless device. Hosting a large number of UG live channels on commercial servers can be very expensive. Server-based solutions also involve various economic, copyright and content control issues between users and the companies hosting their content. In this paper, leveraging on the recent success of P2P video streaming, we study the strategies for end users to directly broadcast their own live channels to a large number of audiences without resorting to any server support. The key challenge is that end users are normally bandwidth constrained and can barely send out one complete video stream to the rest of the world. Existing P2P streaming solutions cannot maintain a high level of user Quality-of-Experience (QoE) with such a highly constrained video source. We propose a novel Layered P2P Streaming (LPS) architecture, to address this challenge. LPS introduces playback delay differentiation and constructs virtual servers out of peers to boost end users’ capability of driving large-scale video streaming. Through detailed packet-level simulations and PlanetLab experiments, we show that LPS enables a source with upload bandwidth slightly higher than the video streaming rate to stream video to tens of thousands of peers with premium quality of experience.     

VBR视频流在有服务质量(QoS)保证的信道上传输时的最优控制问题

本文讨论了可变比特率（VBR）视频流在有服务质量(QoS)保证的信道上传输时的信源和信道速率控制问题.首先将其表述为离散线性时延系统的最优控制问题，然后利用离散极大值原理得到该问题的最优解.和传统的解决方案相比较，本文的方法适合于具有连续输出码率的编码器，并对于编码器和解码器的缓冲区大小没有特殊要求，理论分析和实验结果表明，本文的方法具有更低的复杂度.该方法既能用于离线编码，也能用于在线编码.     

Scene-Change Aware Dynamic Bandwidth Allocation for Real-Time VBR Video Transmission Over IEEE 802.15.3 Wireless Home Networks

IEEE 802.15.3, an emerging wireless technology, was designed to provide high-quality multimedia services at home. Dynamic bandwidth allocation for a multimedia connection should be considered in order to achieve higher channel utilization, less buffer and less delay, especially for variable bit rate (VBR) multimedia connections. For real-time VBR videos, the bandwidth requirement should be predicted adaptively for effective channel-time requests. Previously, the adaptive least-mean square (LMS) algorithm with fixed step size was applied to predict channel time requirements due to its simplicity and relatively good performance. However, the performance might degrade when scene changes occurred. In this paper, we modify the variable step-size LMS algorithm and apply it as our predictor (VSSNLMS) so that the prediction errors on scene changes can be effectively reduced. Using the prediction results of VSSNLMS, we propose a dynamic bandwidth-allocation scheme that is scene-change aware and can guarantee the delay bound of real-time VBR videos. Simulation results show that the VSSNLMS predictor is superior to previous LMS-type predictors in performance, and the proposed scheme has better performance in channel utilization, buffer usage, and packet loss     

Multiple TFRC Connections Based Rate Control for Wireless Networks

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is equation based rate control , in which the TCP friendly rate is determined as a function of packet loss rate, round trip time and packet size. This approach, also known as TCP friendly rate control (TFRC), assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the bulk of packet loss is due to error at the physical layer. In this paper, we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over 1$times$RTT CDMA wireless data network, and and video streaming simulations using traces from the actual experiments, are carried out to validate, and characterize the performance of our proposed approach.     

一种基于信源信道联合的最优速率分配算法

针对视频数据在无线信道上可靠传输问题,提出了一种基于信源信道联合的最优速率分配算法。该算法在网络带宽一定的情况下,从信源、信道及差错弹性能力权衡考虑,引入了信源解码器的抗误码性能指标,根据不同的信道状态确定信源信道编码的最优速率分配方案,从而获得最大的可解码长度,并最终获取最佳重建视频质量。仿真结果表明,该方案与传统的联合信源信道速率分配算法相比可获得更高的性能增益,适合于视频数据在无线网络上传输。     

Delay-Constrained and R-D Optimized Transrating for High-Definition Video Streaming Over WLANs

In this study, we propose bit-rate adaptation techniques for compressed high-definition video transmission over wireless home networks. Application-layer methods are developed to minimize quality impairments caused by wireless channel degradations. A transrater located at a home video server adjusts the transmission rate of the video stream based on bandwidth measurements. We consider delay constraints of multiple video frames across various time scales, and dynamically select an appropriate time scale for bit-rate adjustments. This novel feature prevents playout buffer underflows and avoids unnecessary rate reductions. Furthermore, distortion due to transrating is minimized by coding type-dependent rate allocation among the frames to be transmitted in the projected time scale. Using a realistic simulation setup, we demonstrated that the proposed technique significantly reduces video glitches. Up to 3.12 dB average PSNR improvement is achieved, compared to a non-delay constrained rate adaptation method, which also utilizes bandwidth measurements. Our approach maintains compliance with existing wireless local area (WLAN) devices and standards, as well as consumer video standards and applications.      

Real-time video streaming over multipath in multi-hop wireless networks

Given the limited wireless link throughput, high loss rate, and varying end-to-end delay, supporting video applications in multi-hop wireless networks becomes a challenging task. Path diversity exploits multiple routes for each session simultaneously, which achieves higher aggregated bandwidth and potentially decreases delay and packet loss. Unfortunately, for TCP-based video streaming, naive load splitting often results in inaccurate estimation of round trip time (RTT) and packet reordering. As a result, it can suffer from significant instability or even throughput reduction, which is also validated by our analysis and simulation in multi-hop wireless networks. To make real-time TCP-based streaming viable over multi-hop wireless networks, we propose a novel cross-layer design with a smart traffic split scheme, namely, multiple path retransmission (MPR). MPR differentiates the original data packets and the retransmitted packets and works with a novel QoS-aware multi-path routing protocol, QAOMDV, to distribute them separately. MPR does not suffer from the RTT underestimation and extra packet reordering, which ensures stable throughput improvement over single-path routing. Through extensive simulations, we further demonstrate that, as compared with state-of-the-art multi-path protocols, our MPR with QAOMDV noticeably enhances the TCP streaming throughput and reduces bandwidth fluctuation, with no obvious impact to fairness.     

Redesigning multi-channel P2P live video systems with View-Upload Decoupling

In current multi-channel P2P live video systems, there are several fundamental performance problems including exceedingly-large channel switching delays, long playback lags, and poor performance for less popular channels. These performance problems primarily stem from two intrinsic characteristics of multi-channel P2P video systems: channel churn and channel-resource imbalance. In this paper, we propose a radically different cross-channel P2P streaming framework, called View-Upload Decoupling (VUD). VUD strictly decouples peer downloading from uploading, bringing stability to multi-channel systems and enabling cross-channel resource sharing. We propose a set of peer assignment and bandwidth allocation algorithms to properly provision bandwidth among channels, and introduce substream-swarming to reduce the bandwidth overhead. We evaluate the performance of VUD via extensive simulations as well with a PlanetLab implementation. Our simulation and PlanetLab results show that VUD is resilient to channel churn, and achieves lower switching delay and better streaming quality. In particular, the streaming quality of small channels is greatly improved.     

An efficient bandwidth allocation algorithm for real-time VBR stream transmission under IEEE 802.16 wireless networks

This paper investigates variable rate control strategies for real-time multimedia variable bit rate (VBR) services over IEEE 802.16 broadband wireless networks. A data rate control mechanism is derived for the case where the uplink channel provides real-time services and the traffic rate parameter remains constant. This paper shows that the common queuing scheduling algorithms have some bandwidth allocation fairness problems for the real-time polling service (rtPS) in the MAC layer. In other words, the use of a VBR for the rtPS by a WiMAX system results in additional access latency jitter and bandwidth allocation disorder in the transmitted multimedia streams during the regular time interval polling of subscribe stations (SSs) for the contention bandwidth request period. However, the proposed scheduling algorithm solves these SSs contending with bandwidth resource allocation problems based on an extended rtPS (ertPS) of quality-of-service (QoS) pre-programming for a ranging response non-contention polling period. The adopted bandwidth allocation of max–min fairness queue scheduling uses a time constraint condition to transmit real-time multimedia VBR streaming in an IEEE 802.16 broadband wireless environment. In addition, we use the ns-2 simulation tool to compare the capacity of multimedia VBR stream and show that the proposed ertPS scheduling algorithm outperforms other rtPS scheduling algorithms.     

Channel adaptive video stream switching for broadband wireless links

This paper exploits an H.264/Advanced Video Coding codec’s smooth stream switching to achieve robust video delivery across an IEEE 802.16 (WiMAX) broadband wireless link. As the wireless channel conditions vary over time, dynamic selection of bitrate and corresponding video quality will reduce the risk of harmful packet loss. The paper investigates choice of stream-switching with Secondary SP-frames or SI-frames relative to the selection of quantization parameter (QP) values. To control the switching points at the WiMAX server, the proposed scheme applies a feedback mechanism that monitors packet loss. As an additional suggestion, the paper considers an adaptive ARQ scheme to protect switching frames against packet loss. In summary, the broadband wireless streaming system gives more protection to higher quality video, reduces delay and packet loss, and improves received video quality. In particular, for the QP values selected, results show that increased quality primary-switching frames together with SI frames bring a significant gain in video quality compared with other switching schemes with secondary SP-frames. The other schemes in turn show an improvement to using ‘no switching’ when streaming takes place over a typical WiMAX channel with burst errors. Link delay is also reduced.