Transcoding media for bandwidth constrained mobile devices

Bandwidth is an important consideration when dealing with streaming media. More bandwidth is required for complex data such as video as opposed to a simple audio file. When delivering streaming media, sufficient bandwidth is required to achieve an acceptable level of performance. If the information streamed exceeds the bandwidth capacity of the client the result will be ‘choppy’ and incomplete with possible loss of transmission. Transcoding typically refers to the adaptation of streaming content. Typical transcoding scenarios exploit content‐negotiation to negotiate between different formats in order to obtain the most optimal combination of requested quality and available resources. It is possible to transcode media to a lesser quality or size upon encountering adverse bandwidth conditions. This can be accomplished without the need to encode multiple versions of the same file at differing quality levels. This study investigates the capability of transcoding for coping with restrictions in client devices. In addition, the properties of transcoded media files are examined and evaluated to determine their applicability for streaming in relation to a range of broad device types capable of receiving streaming media. Copyright © 2005 John Wiley & Sons, Ltd.     

QoS-supporting video streaming system with minimum data service cost over heterogeneous wireless networks

This paper presents a video streaming system that supports quality-of-service by effectively consolidating multiple physical paths in a cost-effective way over heterogeneous wireless networks. In the proposed system, the fountain encoding symbols of compressed video data are transmitted through multiple physical paths concurrently to overcome the limitation of single path transmission and harmonize multiple physical paths with diverse characteristics effectively, and the number of transmitted packets is determined by considering the requested quality-of-service of video streaming and the data service cost. The proposed system is fully implemented in Java and C/C++, and tested over real WLAN and LTE networks. Experimental results are provided to demonstrate the performance improvement of the proposed system.     

Rate control for consistent visual quality of H.264/AVC encoding

In this paper, a novel rate control scheme with sliding window basic unit is proposed to achieve consistent or smooth visual quality for H.264/AVC based video streaming. A sliding window consists of a group of successive frames and moves forward by one frame each time. To make the sliding window scheme possible for real-time video streaming, the initial encoder delay inherently in a video streaming system is utilized to generate all the bits of a window in advance, so that these bits for transmission are ready before their due time. The use of initial encoder delay does not introduce any additional delay in video streaming but benefits visual quality as compared to traditional one-pass rate control algorithms of H.264/AVC. Then, a Sliding Window Buffer Checking (SWBC) algorithm is proposed for buffer control at sliding window level and it accords with traditional buffer measurement of H.264/AVC. Extensive experimental results exhibit that higher coding performance, consistent visual quality and compliant buffer constraint can be achieved by the proposed algorithm.     

Implementation of SDTV Video Encoder with Real-Time Constant and Variable Bit-Rate Control

The digital standard definition television (SDTV) encoder is a very important part of the digital TV broadcast chain. Most real-time MPEG-2 encoders are designed to perform in a constant bit-rate (CBR) mode. But an even better compressed stream can be created by employing a variable bit-rate (VBR) encoding algorithm. VBR can be exploited as a means of achieving statistical multiplexing for digital broadcast satellites. This paper suggests an implementation procedure of an SDTV video encoder and proposes a novel VBR bit-allocation strategy that could be implemented in this encoder system. First, using a rate-quantization model and rate-quantization perceptual model, a real-time VBR bit-allocation strategy is deduced. In this strategy, more (or fewer) bits are allocated to "difficult-to-encode" (or "easy-to-encode") groups of pictures (GOPs), which are distinguished according to the estimated encoding complexity of the GOPs. After allocating an appropriate number of bits to each GOP by using this VBR bit-allocation strategy, we use a CBR rate control algorithm to allocate a number of bits and select a quantization scaler for each picture of a GOP. Then smooth visual quality is achieved not only in a GOP but also in the whole video sequence. Second, the system implementation of an SDTV video encoder including a video input module, a video encoding module, a system control and rate control module (SCRCM), and a PES packetizing module is described. We also discuss in detail how to implement our real-time VBR bit-allocation strategy in the SCRCM. Finally, experimental results demonstrate that our proposed VBR encoder displays a better performance than the CBR encoder.     

A buffer‐aware fleet‐based cooperative H.264/SVC streaming over vehicular networks

Because of of the characteristics of high mobility, time varying and dynamic topology, how to provide multimedia streaming service for connected vehicles becomes one emerging and popular technical research. The motivation of this paper is to utilize cooperation among neighboring vehicles for video streaming's quality improvement over vehicular networks. In the proposed cooperative streaming scenario, a connected vehicle requests a video stream from the Internet by using its 3G/3.5G interface, which may not have enough bandwidth to receive good quality of video. Thus, the vehicle is suggested to ask neighboring members belonging to the same fleet to download the requested video data by using their 3G/3.5G interfaces. Then, neighboring members should forward video data to the requested vehicle by using another wireless technique, for example, dedicated short range communication (DSRC). Regarding the differentiation between the two access networks, that is, 3G/3.5G network and DSRC network, a buffer‐aware scheduling mechanism based on layered streaming is designed in this paper to adapt to the networking situation of the vehicular networks. Two selection algorithms are proposed to select neighboring vehicles from the fleet. According to our simulation results, the 3G/3.5G‐based selection algorithm is suitable to improve video quality for vehicles at low speeds. On the other hand, the DSRC‐based selection algorithm can get better performance when vehicles move at high speeds or too many data are transmitted among vehicles.Copyright © 2013 John Wiley & Sons, Ltd.     

基于TMS320DM642的MPEG4编码器的设计与实现

为满足视频通信中视频压缩的需要,在对MPEG4视频编码标准研究的基础上,提出了基于TMS320DM642DSP的MPEG4视频编码器的实现方案,并讨论了代码优化的方法,通过相关实验对提出的方案进行了验证。实验结果表明,该方案实现了对标准QCIF格式图像序列的MPEG4编码,编码速率达到25 fps,满足了实时编码的要求,图像质量满足了预期效果。     

基于ARM的视频采集与处理系统的开发

介绍了一种基于S3C2440硬件平台,在Linux 2.6.32内核操作系统下采用V4L2架构进行视频采集与处理的总体设计方案,通过移植开源的H.264视频编解码器对视频信号进行压缩编码,并采用RTP流媒体传输协议传输给客户端,完成对现场的网络视频监控任务。实验结果表明,该系统采集到的图像清晰,显示比较流畅,经H.264编码的视频帧数据量小,网络传输异常情况下的视频存储正常,可以适用于不同场合尤其是对带宽要求比较高的场合。     

低复杂度的可伸缩视频流媒体MGS编码方案优化

针对不同带宽环境及用户分布选取可伸缩视频流媒体编码方案的不确定性问题是影响视频流编码质量的关键因素,为了实现在指定网络带宽区域条件下多用户实时性访问的编码质量优化,提出了一种低复杂度适应网络带宽区域和质量可伸缩的视频流媒体编码方案优化方法。该方法的基本思想是在视频内容分析和理解的基础上,首先根据MGS片层数据统计特性设计出对应率失真(R-D)模型,结合I,P,B帧类型率失真特性进行视频流码率估计;然后根据优化算法推导出合理编码方式;最后分析该方法的计算时间复杂度。在ITU-T标准文档Q.6/SG16所定义的信道通用测试平台上进行实验研究,实验结果表明所提优化方法能在与传统编码方案复杂度近似情况下带来0.3-1dB视频序列质量增益,且适用于通用的传输信道模型。      

Video traffic prediction based on source information and preventive channel rate decision for RCBR

In this paper, we address the problem of dynamic bandwidth allocation in real-time video transmission. Firstly, a source traffic prediction method is proposed which is based on the rate-distortion relation of source video. This method can detect changes in the source traffic level before encoding by using source information. Secondly, a preventive channel rate decision algorithm, called PCRD, is proposed. The transmission rate bounds are derived from the constraints of the encoder and decoder buffers based on the predicted bit-rate of video frames. From simulation results, the proposed traffic prediction method is shown effective in detecting scene changes and estimating changed traffic levels. Also, the PCRD method is shown to have low renegotiation cost and high channel utilization without violating delay constraints.     

Congestion–distortion optimized video transmission over ad hoc networks

The performance of low-latency video streaming with multipath routing over ad hoc networks is studied. As the available transmission rate of individual links in an ad hoc network is typically limited due to power and bandwidth constraints, a single node transmitting multimedia data may impact the overall network congestion and may therefore need to limit its rate while striving for the highest sustainable video quality. For this purpose, optimal routing algorithms which seek to minimize congestion by optimally distributing traffic over multiple paths are attractive. To predict the end-to-end rate-distortion tradeoff, we develop a model which captures both the impact of encoder quantization and of packet loss due to network congestion on the overall video quality. The validity of the model is confirmed by network simulations performed with different routing algorithms, latency requirements and encoding structures.     

H.264/AVC video for wireless transmission

H.264/AVC will be an essential component in emerging wireless video applications thanks to its excellent compression efficiency and network-friendly design. However, a video coding standard itself is only one component within the application and transmission environment. Its effectiveness strongly depends on the selection of appropriate modes and parameters at the encoder, at the decoder, as well as in the network. In this paper we introduce the features of the H.264/AVC coding standard that make it suitable for wireless video applications, including features for error resilience, bit rate adaptation, integration into packet networks, interoperability, and buffering considerations. Modern wireless networks provide many different means to adapt quality of service, such as forward error correction methods on different layers and end-to-end or link layer retransmission protocols. The applicability of all these encoding and network features depends on application constraints, such as the maximum tolerable delay, the possibility of online encoding, and the availability of feedback and cross-layer information. We discuss the use of different coding and transport related features for different applications, namely video telephony, video conferencing, video streaming, download-and-play, and video broadcasting. Guidelines for the selection of appropriate video coding tools, video encoder and decoder settings, as well as transport and network parameters are provided and justified. References to relevant research publications and standardization contributions are given.     

Content-based FGS coding mode determination for video streaming over wireless networks

Video streaming is the major subject of Amendment for MPEG-4 and it is developed in response to the growing needs on a video-coding standard for the video communication. The fine-granular scalability (FGS) combined with the temporal scalability addresses a variety of challenging problems in delivering video. The FGS video encoder makes the coding mode decision based on the video content and the current available bandwidth in order to achieve higher perceptual video quality. In this paper, we develop a mode selection method to find the most suitable scalable coding mode from six coding schemes: FGS, FGST, FGS-SE, and FGST with background composition based on the contents of the video sequences.     

Joint source and sending rate modeling in adaptive video streaming

This work addresses the modeling of traffic generated by a video source operating in the context of adaptive streaming services. Traffic modeling is a key in several network design issues, such as dimensioning of core and access network resources, developing pricing procedures, carrying out cost-revenue studies. The actual traffic generated during a video streaming session depends on both the video source and the bandwidth variations imposed by lower communication layers. We propose a new traffic model that jointly encompasses these two effects. Specifically, we consider the modeling of the sequence of frame sizes generated by a video streaming source that dynamically adapts its rate to the available communication channel bandwidth using bitstream switching techniques. In order to represent the source rate adaptation to the random network bandwidth variations on the communication channel, we resort to a framework based on Hidden Markov Processes (HMPs). Our HMP model represents the first joint source and sending rate model in adaptive streaming literature. Thanks to effective modeling assumptions on the frame size probability density function (pdf), the HMP parameters can be estimated by means of the Expectation Maximization algorithm. The traffic model is validated by numerical simulations of a mobile adaptive video streaming scenario. We study the model's ability to predict several traffic statistics, including the traffic load of a video streaming source in different network points. Besides, we evaluate the model accuracy in characterizing aggregate video traffic resulting from multiplexing various video sources. In all experiments, we show that the proposed model is able to accurately capture the traffic characteristics.     

A bandwidth assignment algorithm on a satellite channel for VBR traffic

Variable bit rate (VBR) video is currently by far the most interesting and challenging real-time application. A VBR encoder attempts to keep the quality of video output constant and at the same time reduces bandwidth requirements, since only a minimum amount of information has to be transferred. On the other hand, as VBR video traffic is both highly variable and delay-sensitive, high-speed networks (e.g. ATM) are generally implemented by assigning peak rate bandwidths to VBR video applications. This approach may, however, be inefficient in a satellite network based on a TDMA scheme. To overcome this problem, we have designed a demand assignment satellite bandwidth allocation algorithm in TDMA, named V2L-DA (VBR 2-Level Demand Assignment), which manages the VBR video traffic according to a dynamic bandwidth allocation algorithm. In this paper we discuss how to tune the proposed algorithm in order to optimize network utilization when MPEG-1 VBR video traffic is being transmitted. Our results indicate that most of the time only 40% of the peak rate bandwidth is needed to satisfy the VBR source, so the remaining 60% of the peak rate bandwidth can be used to transmit the datagram traffic queued in the network stations. © 1997 John Wiley & Sons, Ltd.     

Dynamic bandwidth management and adaptive applications for avariable bandwidth wireless environment

This article describes an approach for providing dynamic quality of service (QoS) support in a variable bandwidth network, which may include wireless links and mobile nodes. The dynamic QoS approach centers on the notion of providing QoS support at some point within a range requested by applications. To utilize dynamic QoS, applications must be capable of adapting to the level of QoS provided by the network, which may vary during the course of a connection. To demonstrate and evaluate the dynamic QoS concept, we have implemented a new protocol called dynamic resource reservation protocol (dRSVP) and a new QoS application program interface (API). The paper describes this new protocol and API and also discusses our experience with adaptive streaming video and audio applications that work with the new protocol in a testbed network, including wireless local area network connectivity and wireless link connectivity emulated over the wired Ethernet. Qualitative and quantitative assessments of the dynamic RSVP protocol are provided     

Feedback control mechanisms for real-time multipoint video services

While existing research shows that reactive congestion control mechanisms are capable of providing high video quality and channel utilization for point-to-point real-time video, there has been relatively little study of the reactive congestion control of point-to-multipoint video, especially in ATM networks. Problems complicating the provision of multipoint, feedback-based real-time video service include: (1) implosion of feedback returning to the source as the number of multicast destinations increases and (2) variance in the amount of available bandwidth on different branches in the multipoint connection. A new service architecture is proposed for real-time multicast video, and two multipoint feedback mechanisms to support this service are introduced and studied. The mechanisms support a minimum bandwidth guarantee and the best effort support of video traffic exceeding the minimum rate. They both rely on adaptive, multilayered coding at the video source and closed-loop feedback from the network in order to control both the high and low priority video generation rates of the video encoder. Simulation results show that the studied feedback mechanisms provide, at the minimum, a quality of video comparable to a constant bit rate (CBR) connection reserving the same amount of bandwidth. When unutilized network bandwidth becomes available, the mechanisms are capable of exploiting it to dynamically improve video quality beyond the minimum guaranteed level     

Geometrically determining the leaky bucket parameters for video streaming over constant bit-rate channels

For streaming of pre-encoded bitstreams over constant bit rate (CBR) channels, the channel bandwidth, the receiver buffer capacity as well as the latency requirement vary greatly from application to application. In this paper, we attempt to determine the minimum buffer size and the minimum start-up delay required for streaming a pre-encoded bitstream over CBR channels at any specific bit rate. The proposed method employs geometric operations to derive the optimal determination for low or high bit rates and sub-optimal determination for medium bit rates. The algorithm developed requires little extra information from the encoder and is easy to implement. Our algorithm is implemented in a H.264/AVC video encoder and its performance is compared with that of H.264/AVC hypothetical reference decoder. Our approach provides new theoretical insight and an excellent solution for determining the leaky bucket parameters for video streaming over CBR channels.     

MIJ2K: Enhanced video transmission based on conditional replenishment of JPEG2000 tiles with motion compensation

A video compressed as a sequence of JPEG2000 images can achieve the scalability, flexibility, and accessibility that is lacking in current predictive motion-compensated video coding standards. However, streaming JPEG2000-based sequences would consume considerably more bandwidth. With the aim of solving this problem, this paper describes a new patent pending method, called MIJ2K. MIJ2K reduces the inter-frame redundancy present in common JPEG2000 sequences (also called MJP2). We apply a real-time motion detection system to perform conditional tile replenishment. This will significantly reduce the bit rate necessary to transmit JPEG2000 video sequences, also improving their quality.The MIJ2K technique can be used both to improve JPEG2000-based real-time video streaming services or as a new codec for video storage. MIJ2K relies on a fast motion compensation technique, especially designed for real-time video streaming purposes. In particular, we propose transmitting only the tiles that change in each JPEG2000 frame. This paper describes and evaluates the method proposed for real-time tile change detection, as well as the overall MIJ2K architecture.We compare MIJ2K against other intra-frame codecs, like standard Motion JPEG2000, Motion JPEG, and the latest H.264-Intra, comparing performance in terms of compression ratio and video quality, measured by standard peak signal-to-noise ratio, structural similarity and visual quality metric metrics.     

Transmission of adaptive MPEG video over time-varying wireless channels: modeling and performance evaluation

Wireless channels are characterized by high time-varying bit-error rates (BERs). To cope with this problem, several adaptive forward-error-correction (AFEC) schemes have been proposed in the literature. They work locally at the wireless link, adding a variable amount of redundancy to the transmitted data in order to maintain the packet error rate below an acceptable level. However, when such schemes are utilized, the bandwidth offered to the applications changes when channel conditions change. In this paper, the effects of these bandwidth variations are investigated in the case of real-time Motion Picture Experts Group (MPEG) video transmission. The MPEG encoder is controlled in order to adapt its emission rate to the current bandwidth offered by the wireless link. To this end, the encoding quality is diminished by the source rate controller when the transmission rate has to be decreased due to an increase in the channel BER, whereas it is improved when the transmission rate can be increased due to a decrease in the channel BER. A Markov-based model, denoted as SBBP/SBBP/1/K, has been introduced to model the scenario being considered. The analytical framework allows evaluation of the performance of the system and can be used to optimize the design of a video transmission system for wireless channels, providing the instruments to derive the tradeoff between information corruption in the wireless channel and MPEG video encoding quality.