Regular exercise is a crucial aspect of daily life, as it enables individuals to stay physically active, lowers thelikelihood of developing illnesses, and enhances life expectancy. The recognition of workout actions i...Regular exercise is a crucial aspect of daily life, as it enables individuals to stay physically active, lowers thelikelihood of developing illnesses, and enhances life expectancy. The recognition of workout actions in videostreams holds significant importance in computer vision research, as it aims to enhance exercise adherence, enableinstant recognition, advance fitness tracking technologies, and optimize fitness routines. However, existing actiondatasets often lack diversity and specificity for workout actions, hindering the development of accurate recognitionmodels. To address this gap, the Workout Action Video dataset (WAVd) has been introduced as a significantcontribution. WAVd comprises a diverse collection of labeled workout action videos, meticulously curated toencompass various exercises performed by numerous individuals in different settings. This research proposes aninnovative framework based on the Attention driven Residual Deep Convolutional-Gated Recurrent Unit (ResDCGRU)network for workout action recognition in video streams. Unlike image-based action recognition, videoscontain spatio-temporal information, making the task more complex and challenging. While substantial progresshas been made in this area, challenges persist in detecting subtle and complex actions, handling occlusions,and managing the computational demands of deep learning approaches. The proposed ResDC-GRU Attentionmodel demonstrated exceptional classification performance with 95.81% accuracy in classifying workout actionvideos and also outperformed various state-of-the-art models. The method also yielded 81.6%, 97.2%, 95.6%, and93.2% accuracy on established benchmark datasets, namely HMDB51, Youtube Actions, UCF50, and UCF101,respectively, showcasing its superiority and robustness in action recognition. The findings suggest practicalimplications in real-world scenarios where precise video action recognition is paramount, addressing the persistingchallenges in the field. TheWAVd dataset serves as a catalyst for the development ofmore robust and effective fitnesstracking systems and ultimately promotes healthier lifestyles through improved exercise monitoring and analysis.展开更多
Recently,the combination of video services and 5G networks have been gaining attention in the wireless communication realm.With the brisk advancement in 5G network usage and the massive popularity of threedimensional ...Recently,the combination of video services and 5G networks have been gaining attention in the wireless communication realm.With the brisk advancement in 5G network usage and the massive popularity of threedimensional video streaming,the quality of experience(QoE)of video in 5G systems has been receiving overwhelming significance from both customers and service provider ends.Therefore,effectively categorizing QoE-aware video streaming is imperative for achieving greater client satisfaction.This work makes the following contribution:First,a simulation platform based on NS-3 is introduced to analyze and improve the performance of video services.The simulation is formulated to offer real-time measurements,saving the expensive expenses associated with real-world equipment.Second,A valuable framework for QoE-aware video streaming categorization is introduced in 5G networks based on machine learning(ML)by incorporating the hyperparameter tuning(HPT)principle.It implements an enhanced hyperparameter tuning(EHPT)ensemble and decision tree(DT)classifier for video streaming categorization.The performance of the ML approach is assessed by considering precision,accuracy,recall,and computation time metrics for manifesting the superiority of these classifiers regarding video streaming categorization.This paper demonstrates that our ML classifiers achieve QoE prediction accuracy of 92.59%for(EHPT)ensemble and 87.037%for decision tree(DT)classifiers.展开更多
In recent years,real-time video streaming has grown in popularity.The growing popularity of the Internet of Things(IoT)and other wireless heterogeneous networks mandates that network resources be carefully apportioned...In recent years,real-time video streaming has grown in popularity.The growing popularity of the Internet of Things(IoT)and other wireless heterogeneous networks mandates that network resources be carefully apportioned among versatile users in order to achieve the best Quality of Experience(QoE)and performance objectives.Most researchers focused on Forward Error Correction(FEC)techniques when attempting to strike a balance between QoE and performance.However,as network capacity increases,the performance degrades,impacting the live visual experience.Recently,Deep Learning(DL)algorithms have been successfully integrated with FEC to stream videos across multiple heterogeneous networks.But these algorithms need to be changed to make the experience better without sacrificing packet loss and delay time.To address the previous challenge,this paper proposes a novel intelligent algorithm that streams video in multi-home heterogeneous networks based on network-centric characteristics.The proposed framework contains modules such as Intelligent Content Extraction Module(ICEM),Channel Status Monitor(CSM),and Adaptive FEC(AFEC).This framework adopts the Cognitive Learning-based Scheduling(CLS)Module,which works on the deep Reinforced Gated Recurrent Networks(RGRN)principle and embeds them along with the FEC to achieve better performances.The complete framework was developed using the Objective Modular Network Testbed in C++(OMNET++),Internet networking(INET),and Python 3.10,with Keras as the front end and Tensorflow 2.10 as the back end.With extensive experimentation,the proposed model outperforms the other existing intelligentmodels in terms of improving the QoE,minimizing the End-to-End Delay(EED),and maintaining the highest accuracy(98%)and a lower Root Mean Square Error(RMSE)value of 0.001.展开更多
Video compression technologies are essential in video streaming application because they could save a great amount of network resources. However compressed videos are also extremely sensitive to packet loss which is i...Video compression technologies are essential in video streaming application because they could save a great amount of network resources. However compressed videos are also extremely sensitive to packet loss which is inevitable in today's best effort IP network. Therefore we think accurate evaluation of packet loss impairment on compressed video is very important. In this work, we develop an analytic model to describe these impairments without the reference of the original video (NR) and propose an impairment metric based on the model, which takes into account both impairment length and impairment strength. To evaluate an impaired frame or video, we design a detection and evaluation algorithm (DE algorithm) to compute the above metric value. The DE algorithm has low computational complexity and is currently being implemented in the real-time monitoring module of our HDTV over IP system. The impairment metric and DE algorithm could also be used in adaptive system or be used to compare diffeient error concealment strategies.展开更多
Adaptive bitrate video streaming(ABR)has become a critical technique for mobile video streaming to cope with time-varying network conditions and different user preferences.However,there are still many problems in achi...Adaptive bitrate video streaming(ABR)has become a critical technique for mobile video streaming to cope with time-varying network conditions and different user preferences.However,there are still many problems in achieving high-quality ABR video streaming over cellular networks.Mobile Edge Computing(MEC)is a promising paradigm to overcome the above problems by providing video transcoding capability and caching the ABR video streaming within the radio access network(RAN).In this paper,we propose a flexible transcoding strategy to provide viewers with low-latency video streaming services in the MEC networks under the limited storage,computing,and spectrum resources.According to the information collected from users,the MEC server acts as a controlling component to adjust the transcoding strategy flexibly based on optimizing the video caching placement strategy.Specifically,we cache the proper bitrate version of the video segments at the edge servers and select the appropriate bitrate version of the video segments to perform transcoding under jointly considering access control,resource allocation,and user preferences.We formulate this problem as a nonconvex optimization and mixed combinatorial problem.Moreover,the simulation results indicate that our proposed algorithm can ensure a low-latency viewing experience for users.展开更多
In this work, we present an evaluation of the performance and error robustness of RTP-based broadcast streaming of high-quality high-definition (HD) H.264/AVC video. Using a fully controlled IP test bed (Hillestad et ...In this work, we present an evaluation of the performance and error robustness of RTP-based broadcast streaming of high-quality high-definition (HD) H.264/AVC video. Using a fully controlled IP test bed (Hillestad et al., 2005), we broadcast high-definition video over RTP/UDP, and use an IP network emulator to introduce a varying amount of randomly distributed packet loss. A high-performance network interface monitoring card is used to capture the video packets into a trace file. Purpose-built software parses the trace file, analyzes the RTP stream and assembles the correctly received NAL units into an H.264/AVC Annex B byte stream file, which is subsequently decoded by JVT JM 10.1 reference software. The proposed measurement setup is a novel, practical and intuitive approach to perform error resilience testing of real-world H.264/AVC broadcast applications. Through a series of experiments, we evaluate some of the error resilience features of the H.264/AVC standard, and see how they perform at packet loss rates from 0.01% to 5%. The results confirmed that an appropriate slice partitioning scheme is essential to have a graceful degradation in received quality in the case of packet loss. While flexible macroblock ordering reduces the compression efficiency about 1 dB for our test material, reconstructed video quality is improved for loss rates above 0.25%.展开更多
With correlating with human perception, quality of experience(Qo E) is also an important measurement in evaluation of video quality in addition to quality of service(Qo S). A cross-layer scheme based on Lyapunov optim...With correlating with human perception, quality of experience(Qo E) is also an important measurement in evaluation of video quality in addition to quality of service(Qo S). A cross-layer scheme based on Lyapunov optimization framework for H.264/AVC video streaming over wireless Ad hoc networks is proposed, with increasing both Qo E and Qo S performances. Different from existing works, this scheme routes and schedules video packets according to the statuses of the frame buffers at the destination nodes to reduce buffer underflows and to increase video playout continuity. The waiting time of head-ofline packets of data queues are considered in routing and scheduling to reduce the average end-to-end delay of video sessions. Different types of packets are allocated with different priorities according to their generated rates under H.264/AVC. To reduce the computational complexity, a distributed media access control policy and a power control algorithm cooperating with the media access policy are proposed. Simulation results show that, compared with existing schemes, this scheme can improve both the Qo S and Qo E performances. The average peak signal-to-noise ratio(PSNR) of the received video streams is also increased.展开更多
We propose a Rate-Distortion (RD) optimized strategy for frame-dropping and scheduling of multi-user conversa- tional and streaming videos. We consider a scenario where conversational and streaming videos share the fo...We propose a Rate-Distortion (RD) optimized strategy for frame-dropping and scheduling of multi-user conversa- tional and streaming videos. We consider a scenario where conversational and streaming videos share the forwarding resources at a network node. Two buffers are setup on the node to temporarily store the packets for these two types of video applications. For streaming video, a big buffer is used as the associated delay constraint of the application is moderate and a very small buffer is used for conversational video to ensure that the forwarding delay of every packet is limited. A scheduler is located behind these two buffers that dynamically assigns transmission slots on the outgoing link to the two buffers. Rate-distortion side information is used to perform RD-optimized frame dropping in case of node overload. Sharing the data rate on the outgoing link between the con- versational and the streaming videos is done either based on the fullness of the two associated buffers or on the mean incoming rates of the respective videos. Simulation results showed that our proposed RD-optimized frame dropping and scheduling ap- proach provides significant improvements in performance over the popular priority-based random dropping (PRD) technique.展开更多
In this paper, we propose a multi-source multi-path video streaming system for supporting high quality concurrent video-on-demand (VoD) services over wireless mesh networks (WMNs), and leverage forward error correctio...In this paper, we propose a multi-source multi-path video streaming system for supporting high quality concurrent video-on-demand (VoD) services over wireless mesh networks (WMNs), and leverage forward error correction to enhance the error resilience of the system. By taking wireless interference into consideration, we present a more realistic networking model to capture the characteristics of WMNs and then design a route selection scheme using a joint rate/interference-distortion optimiza- tion framework to help the system optimally select concurrent streaming paths. We mathematically formulate such a route selec- tion problem, and solve it heuristically using genetic algorithm. Simulation results demonstrate the effectiveness of our proposed scheme.展开更多
In this paper,we investigate video quality enhancement using computation offloading to the mobile cloud computing(MCC)environment.Our objective is to reduce the computational complexity required to covert a low-resolu...In this paper,we investigate video quality enhancement using computation offloading to the mobile cloud computing(MCC)environment.Our objective is to reduce the computational complexity required to covert a low-resolution video to high-resolution video while minimizing computation at the mobile client and additional communication costs.To do so,we propose an energy-efficient computation offloading framework for video streaming services in a MCC over the fifth generation(5G)cellular networks.In the proposed framework,the mobile client offloads the computational burden for the video enhancement to the cloud,which renders the side information needed to enhance video without requiring much computation by the client.The cloud detects edges from the upsampled ultra-high-resolution video(UHD)and then compresses and transmits them as side information with the original low-resolution video(e.g.,full HD).Finally,the mobile client decodes the received content and integrates the SI and original content,which produces a high-quality video.In our extensive simulation experiments,we observed that the amount of computation needed to construct a UHD video in the client is 50%-60% lower than that required to decode UHD video compressed by legacy video encoding algorithms.Moreover,the bandwidth required to transmit a full HD video and its side information is around 70% lower than that required for a normal UHD video.The subjective quality of the enhanced UHD is similar to that of the original UHD video even though the client pays lower communication costs with reduced computing power.展开更多
Whenever streaming of multimedia based data such as video, audio and text is performed traffic will be more and network becomes congested in mobile ad hoc networks. The present routing protocols are not able to cope u...Whenever streaming of multimedia based data such as video, audio and text is performed traffic will be more and network becomes congested in mobile ad hoc networks. The present routing protocols are not able to cope up with this situation. It is observed that network congestion is the dominant reason for packet loss, longer delay and delay jitter in streaming video. Most of the present routing protocols are not designed to adapt to congestion control. We propose a new routing protocol, Congestion Adaptive AODV Routing Protocol (CA-AODV), to address the congestion issues considering delay, packet loss and routing overhead. To evaluate their performance, we have considered mpeg4 for streaming video data using network simulator (NS2). CA-AODV outperforms present protocols in delivery ratio and delay, while introducing less routing protocol overhead. The result demonstrates that integrating congestion adaptive mechanisms with AODV is a promising way to improve performance for heavy traffic load in multimedia based mobile ad hoc networks.展开更多
The accuracy of the traditional assessment method of the quality of experience(Qo E) has been facing challenges with the growth of high-definition(HD) video streaming services.Image display-quality damage is the main ...The accuracy of the traditional assessment method of the quality of experience(Qo E) has been facing challenges with the growth of high-definition(HD) video streaming services.Image display-quality damage is the main factor that affects the Qo E in HD video services through UDP network transmission.In this paper,we introduce a novel objective factor known as image damage accumulation(IDA) to assess user's Qo E in HD video services.First,this paper quantitatively analyzed the effect on user quality of experience by IDA and established a mapping relationship between mean opinion scores and IDA.Furthermore,the probability of image damage caused by compression and transmission were analyzed.Based on this analysis,an objective Qo E assessment and prediction method for HD video stream service that evaluated the user experience according to IDA are proposed.The proposed method can achieve assessment and prediction accuracy on three distinct subjective tests.展开更多
Benefiting from the improvements of Internet infrastructure and video coding technology, online video services are becoming a new favorite form of video entertainment.However, most of the existing video quality assess...Benefiting from the improvements of Internet infrastructure and video coding technology, online video services are becoming a new favorite form of video entertainment.However, most of the existing video quality assessment methods are designed for broadcasting/cable televisions and it is still an open issue how to assess and measure the quality of online video services. In this paper, we survey the state-of-the-art video streaming technologies, and present a framework of quality assessment and measurement for Internet video streaming. This paper introduces several metrics for user's quality of experience(QoE).These QoE metrics are classified into two categories: objective metrics and subjective metrics. It is different for service participators to measure objective and subjective metrics.The QoE measurement methodologies consist of client-side, server-side, and in-network measurement.展开更多
A new rate allocation method for fine-granular scalability (FGS) coded bitstreams is presented in order to achieve smooth quality reconstruction of frames under channel conditions with a wide range of bandwidth variat...A new rate allocation method for fine-granular scalability (FGS) coded bitstreams is presented in order to achieve smooth quality reconstruction of frames under channel conditions with a wide range of bandwidth variation and improve the average PSNR of the whole sequence. Based on a quality weighted bit allocation method, a sliding window rate allocation method is proposed for the first time so that the window can slide along the video sequence with a certain sliding step. Experimental results show that, under dynamic bandwidth conditions, the proposed method can simultaneously satisfy the requirements for improving average PSNR of the whole video sequence greatly and reducing the fluctuations between adjacent frames greatly.展开更多
In the case of video streaming over wireless channels, burst errors may lead to serious video quality degradation. By jointly exploiting the scheduling mechanism on different communication layers, this paper proposes ...In the case of video streaming over wireless channels, burst errors may lead to serious video quality degradation. By jointly exploiting the scheduling mechanism on different communication layers, this paper proposes a quality-aware cross-layer scheduling scheme to achieve unequal error control for each Latency-constraint Frame Set (LFS) of a video stream. After a network-layer agent at base station firstly utilizes the network-layer packet scheduling to provide packet-granularity importance classifi-cation for the current LFS, a link-layer agent at base station further utilizes the Radio-Link-Unit (RLU) scheduling to implement finer selective retransmission of the current LFS. Under scheduling delay and bandwidth constraints, the proposed scheme can be aware of the application-layer quality and time-varying channel conditions, and hence burst errors can simply be shifted to lower-priority transmission units in the current LFS. Simulation results demonstrate that the proposed scheme has strong robustness against burst errors, and thus improves the overall received quality of the video stream over wireless channels.展开更多
Equation based TCP-friendly rate control (TFRC) protocol has been proposed to support video streaming applications. In order to improve TFRC performance in wireless channels, the link level automatic repeat request (A...Equation based TCP-friendly rate control (TFRC) protocol has been proposed to support video streaming applications. In order to improve TFRC performance in wireless channels, the link level automatic repeat request (ARQ) scheme is usually deployed. However, ARQ cannot ensure strict delay guarantees, especially over multihop links. This paper introduces a theoretical model to deduce an equation for packet size adjustment in transport layer to minimize retransmission delay by taking into con- sideration the causative reasons inducing retransmission in link layer. An enhanced TFRC (ETFRC) scheme is proposed inte- grating TFRC with variable packet size policy. Simulation results demonstrate that higher goodput, lower packet loss rate (PLR), lower frame transmission delay and jitter with good fairness can be achieved by our proposed mechanism.展开更多
For video streaming over lossy channels, intra refresh can mitigate the error-propagation effect caused by packet losses Besides some intra-mode macroblocks (MBs) generated by the "Lagrangian rate-distortion" or ...For video streaming over lossy channels, intra refresh can mitigate the error-propagation effect caused by packet losses Besides some intra-mode macroblocks (MBs) generated by the "Lagrangian rate-distortion" or "Sum of absolute difference" mode decision, the encoder or transcoder possibly needs to increase some "forced" intra-mode MBs for robust video streaming. Based on the error-propagation analysis in a group of pictures (GOP), we propose an unequal Forced-Intra-Refresh (FIR) scheme to improve packet loss resilience of video streaming. According to a GOP-level error-propagation model, the proposed unequal FIR scheme can optimally increase the unequal number of forced intra-mode MBs for different frames in a GOP. Simulation results showed that the proposed scheme can effectively enhance the robustness of video streaming under different channel conditions, and achieve about 0. 1-0.9 dB gains over the average FIR scheme in H.264/AVC tools.展开更多
The effect of receiver buffer size on perceived video quality of an Internet video streamer application was examined in this work. Several network conditions and several versions of the application are used to gain un...The effect of receiver buffer size on perceived video quality of an Internet video streamer application was examined in this work. Several network conditions and several versions of the application are used to gain understanding of the response to varying buffer sizes. Among these conditions local area versus wide area, bandwidth estimation based versus non-bandwidth estimation based cases are examined in detail. A total of 1000 min of video is streamed over Intemet and statistics are collected. It was observed that when bandwidth estimation is possible, choosing larger buffer size for higher available bandwidth yields quality increase in perceived video.展开更多
Reduced-reference (RR) video-quality estimators send a small signature to the receiver. This signature comprises the original video content as well as the video stream. RR quality estimation provides reliability and...Reduced-reference (RR) video-quality estimators send a small signature to the receiver. This signature comprises the original video content as well as the video stream. RR quality estimation provides reliability and involves a small data payload. While significant in theory, RR estimators have only recently been used in practice for quality monitoring and adaptive system con- trol in streaming-video frameworks. In this paper, we classify RR algorithms according to whether they are based on a) model- ing the signal distortion, b) modeling the human visual system, or c) analyzing the video signal source. We review proposed RR techniques for monitoring and controlling quality in streaming video systems.展开更多
This paper presents a reversible data hiding(RDH)method,which is designed by combining histogram modification(HM)with run-level coding in H.264/advanced video coding(AVC).In this scheme,the run-level is changed for em...This paper presents a reversible data hiding(RDH)method,which is designed by combining histogram modification(HM)with run-level coding in H.264/advanced video coding(AVC).In this scheme,the run-level is changed for embedding data into H.264/AVC video sequences.In order to guarantee the reversibility of the proposed scheme,the last nonzero quantized discrete cosine transform(DCT)coefficients in embeddable 4×4 blocks are shifted by the technology of histogram modification.The proposed scheme is realized after quantization and before entropy coding of H.264/AVC compression standard.Therefore,the embedded information can be correctly extracted at the decoding side.Peak-signal-noise-to-ratio(PSNR)and Structure similarity index(SSIM),embedding payload and bit-rate variation are exploited to measure the performance of the proposed scheme.Experimental results have shown that the proposed scheme leads to less SSIM variation and bit-rate increase.展开更多
文摘Regular exercise is a crucial aspect of daily life, as it enables individuals to stay physically active, lowers thelikelihood of developing illnesses, and enhances life expectancy. The recognition of workout actions in videostreams holds significant importance in computer vision research, as it aims to enhance exercise adherence, enableinstant recognition, advance fitness tracking technologies, and optimize fitness routines. However, existing actiondatasets often lack diversity and specificity for workout actions, hindering the development of accurate recognitionmodels. To address this gap, the Workout Action Video dataset (WAVd) has been introduced as a significantcontribution. WAVd comprises a diverse collection of labeled workout action videos, meticulously curated toencompass various exercises performed by numerous individuals in different settings. This research proposes aninnovative framework based on the Attention driven Residual Deep Convolutional-Gated Recurrent Unit (ResDCGRU)network for workout action recognition in video streams. Unlike image-based action recognition, videoscontain spatio-temporal information, making the task more complex and challenging. While substantial progresshas been made in this area, challenges persist in detecting subtle and complex actions, handling occlusions,and managing the computational demands of deep learning approaches. The proposed ResDC-GRU Attentionmodel demonstrated exceptional classification performance with 95.81% accuracy in classifying workout actionvideos and also outperformed various state-of-the-art models. The method also yielded 81.6%, 97.2%, 95.6%, and93.2% accuracy on established benchmark datasets, namely HMDB51, Youtube Actions, UCF50, and UCF101,respectively, showcasing its superiority and robustness in action recognition. The findings suggest practicalimplications in real-world scenarios where precise video action recognition is paramount, addressing the persistingchallenges in the field. TheWAVd dataset serves as a catalyst for the development ofmore robust and effective fitnesstracking systems and ultimately promotes healthier lifestyles through improved exercise monitoring and analysis.
文摘Recently,the combination of video services and 5G networks have been gaining attention in the wireless communication realm.With the brisk advancement in 5G network usage and the massive popularity of threedimensional video streaming,the quality of experience(QoE)of video in 5G systems has been receiving overwhelming significance from both customers and service provider ends.Therefore,effectively categorizing QoE-aware video streaming is imperative for achieving greater client satisfaction.This work makes the following contribution:First,a simulation platform based on NS-3 is introduced to analyze and improve the performance of video services.The simulation is formulated to offer real-time measurements,saving the expensive expenses associated with real-world equipment.Second,A valuable framework for QoE-aware video streaming categorization is introduced in 5G networks based on machine learning(ML)by incorporating the hyperparameter tuning(HPT)principle.It implements an enhanced hyperparameter tuning(EHPT)ensemble and decision tree(DT)classifier for video streaming categorization.The performance of the ML approach is assessed by considering precision,accuracy,recall,and computation time metrics for manifesting the superiority of these classifiers regarding video streaming categorization.This paper demonstrates that our ML classifiers achieve QoE prediction accuracy of 92.59%for(EHPT)ensemble and 87.037%for decision tree(DT)classifiers.
文摘In recent years,real-time video streaming has grown in popularity.The growing popularity of the Internet of Things(IoT)and other wireless heterogeneous networks mandates that network resources be carefully apportioned among versatile users in order to achieve the best Quality of Experience(QoE)and performance objectives.Most researchers focused on Forward Error Correction(FEC)techniques when attempting to strike a balance between QoE and performance.However,as network capacity increases,the performance degrades,impacting the live visual experience.Recently,Deep Learning(DL)algorithms have been successfully integrated with FEC to stream videos across multiple heterogeneous networks.But these algorithms need to be changed to make the experience better without sacrificing packet loss and delay time.To address the previous challenge,this paper proposes a novel intelligent algorithm that streams video in multi-home heterogeneous networks based on network-centric characteristics.The proposed framework contains modules such as Intelligent Content Extraction Module(ICEM),Channel Status Monitor(CSM),and Adaptive FEC(AFEC).This framework adopts the Cognitive Learning-based Scheduling(CLS)Module,which works on the deep Reinforced Gated Recurrent Networks(RGRN)principle and embeds them along with the FEC to achieve better performances.The complete framework was developed using the Objective Modular Network Testbed in C++(OMNET++),Internet networking(INET),and Python 3.10,with Keras as the front end and Tensorflow 2.10 as the back end.With extensive experimentation,the proposed model outperforms the other existing intelligentmodels in terms of improving the QoE,minimizing the End-to-End Delay(EED),and maintaining the highest accuracy(98%)and a lower Root Mean Square Error(RMSE)value of 0.001.
文摘Video compression technologies are essential in video streaming application because they could save a great amount of network resources. However compressed videos are also extremely sensitive to packet loss which is inevitable in today's best effort IP network. Therefore we think accurate evaluation of packet loss impairment on compressed video is very important. In this work, we develop an analytic model to describe these impairments without the reference of the original video (NR) and propose an impairment metric based on the model, which takes into account both impairment length and impairment strength. To evaluate an impaired frame or video, we design a detection and evaluation algorithm (DE algorithm) to compute the above metric value. The DE algorithm has low computational complexity and is currently being implemented in the real-time monitoring module of our HDTV over IP system. The impairment metric and DE algorithm could also be used in adaptive system or be used to compare diffeient error concealment strategies.
基金This work was supported by National Natural Science Foundation of China(No.61771070)National Natural Science Foundation of China(No.61671088).
文摘Adaptive bitrate video streaming(ABR)has become a critical technique for mobile video streaming to cope with time-varying network conditions and different user preferences.However,there are still many problems in achieving high-quality ABR video streaming over cellular networks.Mobile Edge Computing(MEC)is a promising paradigm to overcome the above problems by providing video transcoding capability and caching the ABR video streaming within the radio access network(RAN).In this paper,we propose a flexible transcoding strategy to provide viewers with low-latency video streaming services in the MEC networks under the limited storage,computing,and spectrum resources.According to the information collected from users,the MEC server acts as a controlling component to adjust the transcoding strategy flexibly based on optimizing the video caching placement strategy.Specifically,we cache the proper bitrate version of the video segments at the edge servers and select the appropriate bitrate version of the video segments to perform transcoding under jointly considering access control,resource allocation,and user preferences.We formulate this problem as a nonconvex optimization and mixed combinatorial problem.Moreover,the simulation results indicate that our proposed algorithm can ensure a low-latency viewing experience for users.
基金Project supported by the Research Council of Norway, Norwegian University of Science and Technology (NTNU), and the Norwegian Resarch Network (UNINETT)
文摘In this work, we present an evaluation of the performance and error robustness of RTP-based broadcast streaming of high-quality high-definition (HD) H.264/AVC video. Using a fully controlled IP test bed (Hillestad et al., 2005), we broadcast high-definition video over RTP/UDP, and use an IP network emulator to introduce a varying amount of randomly distributed packet loss. A high-performance network interface monitoring card is used to capture the video packets into a trace file. Purpose-built software parses the trace file, analyzes the RTP stream and assembles the correctly received NAL units into an H.264/AVC Annex B byte stream file, which is subsequently decoded by JVT JM 10.1 reference software. The proposed measurement setup is a novel, practical and intuitive approach to perform error resilience testing of real-world H.264/AVC broadcast applications. Through a series of experiments, we evaluate some of the error resilience features of the H.264/AVC standard, and see how they perform at packet loss rates from 0.01% to 5%. The results confirmed that an appropriate slice partitioning scheme is essential to have a graceful degradation in received quality in the case of packet loss. While flexible macroblock ordering reduces the compression efficiency about 1 dB for our test material, reconstructed video quality is improved for loss rates above 0.25%.
文摘With correlating with human perception, quality of experience(Qo E) is also an important measurement in evaluation of video quality in addition to quality of service(Qo S). A cross-layer scheme based on Lyapunov optimization framework for H.264/AVC video streaming over wireless Ad hoc networks is proposed, with increasing both Qo E and Qo S performances. Different from existing works, this scheme routes and schedules video packets according to the statuses of the frame buffers at the destination nodes to reduce buffer underflows and to increase video playout continuity. The waiting time of head-ofline packets of data queues are considered in routing and scheduling to reduce the average end-to-end delay of video sessions. Different types of packets are allocated with different priorities according to their generated rates under H.264/AVC. To reduce the computational complexity, a distributed media access control policy and a power control algorithm cooperating with the media access policy are proposed. Simulation results show that, compared with existing schemes, this scheme can improve both the Qo S and Qo E performances. The average peak signal-to-noise ratio(PSNR) of the received video streams is also increased.
基金Project (No. STE1093/1-1) supported by the German ResearchFoundation, Germany
文摘We propose a Rate-Distortion (RD) optimized strategy for frame-dropping and scheduling of multi-user conversa- tional and streaming videos. We consider a scenario where conversational and streaming videos share the forwarding resources at a network node. Two buffers are setup on the node to temporarily store the packets for these two types of video applications. For streaming video, a big buffer is used as the associated delay constraint of the application is moderate and a very small buffer is used for conversational video to ensure that the forwarding delay of every packet is limited. A scheduler is located behind these two buffers that dynamically assigns transmission slots on the outgoing link to the two buffers. Rate-distortion side information is used to perform RD-optimized frame dropping in case of node overload. Sharing the data rate on the outgoing link between the con- versational and the streaming videos is done either based on the fullness of the two associated buffers or on the mean incoming rates of the respective videos. Simulation results showed that our proposed RD-optimized frame dropping and scheduling ap- proach provides significant improvements in performance over the popular priority-based random dropping (PRD) technique.
文摘In this paper, we propose a multi-source multi-path video streaming system for supporting high quality concurrent video-on-demand (VoD) services over wireless mesh networks (WMNs), and leverage forward error correction to enhance the error resilience of the system. By taking wireless interference into consideration, we present a more realistic networking model to capture the characteristics of WMNs and then design a route selection scheme using a joint rate/interference-distortion optimiza- tion framework to help the system optimally select concurrent streaming paths. We mathematically formulate such a route selec- tion problem, and solve it heuristically using genetic algorithm. Simulation results demonstrate the effectiveness of our proposed scheme.
文摘In this paper,we investigate video quality enhancement using computation offloading to the mobile cloud computing(MCC)environment.Our objective is to reduce the computational complexity required to covert a low-resolution video to high-resolution video while minimizing computation at the mobile client and additional communication costs.To do so,we propose an energy-efficient computation offloading framework for video streaming services in a MCC over the fifth generation(5G)cellular networks.In the proposed framework,the mobile client offloads the computational burden for the video enhancement to the cloud,which renders the side information needed to enhance video without requiring much computation by the client.The cloud detects edges from the upsampled ultra-high-resolution video(UHD)and then compresses and transmits them as side information with the original low-resolution video(e.g.,full HD).Finally,the mobile client decodes the received content and integrates the SI and original content,which produces a high-quality video.In our extensive simulation experiments,we observed that the amount of computation needed to construct a UHD video in the client is 50%-60% lower than that required to decode UHD video compressed by legacy video encoding algorithms.Moreover,the bandwidth required to transmit a full HD video and its side information is around 70% lower than that required for a normal UHD video.The subjective quality of the enhanced UHD is similar to that of the original UHD video even though the client pays lower communication costs with reduced computing power.
文摘Whenever streaming of multimedia based data such as video, audio and text is performed traffic will be more and network becomes congested in mobile ad hoc networks. The present routing protocols are not able to cope up with this situation. It is observed that network congestion is the dominant reason for packet loss, longer delay and delay jitter in streaming video. Most of the present routing protocols are not designed to adapt to congestion control. We propose a new routing protocol, Congestion Adaptive AODV Routing Protocol (CA-AODV), to address the congestion issues considering delay, packet loss and routing overhead. To evaluate their performance, we have considered mpeg4 for streaming video data using network simulator (NS2). CA-AODV outperforms present protocols in delivery ratio and delay, while introducing less routing protocol overhead. The result demonstrates that integrating congestion adaptive mechanisms with AODV is a promising way to improve performance for heavy traffic load in multimedia based mobile ad hoc networks.
基金supported by the 863 Program(2014AA01A701)NSFC(61271187)+1 种基金the PAPD fundthe CICAEET fund
文摘The accuracy of the traditional assessment method of the quality of experience(Qo E) has been facing challenges with the growth of high-definition(HD) video streaming services.Image display-quality damage is the main factor that affects the Qo E in HD video services through UDP network transmission.In this paper,we introduce a novel objective factor known as image damage accumulation(IDA) to assess user's Qo E in HD video services.First,this paper quantitatively analyzed the effect on user quality of experience by IDA and established a mapping relationship between mean opinion scores and IDA.Furthermore,the probability of image damage caused by compression and transmission were analyzed.Based on this analysis,an objective Qo E assessment and prediction method for HD video stream service that evaluated the user experience according to IDA are proposed.The proposed method can achieve assessment and prediction accuracy on three distinct subjective tests.
基金supported by National Key R&D Program of China No.2018YFB0803702Beijing Culture Development Funding under Grant No.2016-288Toutiao Funding No.ZN20171224003
文摘Benefiting from the improvements of Internet infrastructure and video coding technology, online video services are becoming a new favorite form of video entertainment.However, most of the existing video quality assessment methods are designed for broadcasting/cable televisions and it is still an open issue how to assess and measure the quality of online video services. In this paper, we survey the state-of-the-art video streaming technologies, and present a framework of quality assessment and measurement for Internet video streaming. This paper introduces several metrics for user's quality of experience(QoE).These QoE metrics are classified into two categories: objective metrics and subjective metrics. It is different for service participators to measure objective and subjective metrics.The QoE measurement methodologies consist of client-side, server-side, and in-network measurement.
文摘A new rate allocation method for fine-granular scalability (FGS) coded bitstreams is presented in order to achieve smooth quality reconstruction of frames under channel conditions with a wide range of bandwidth variation and improve the average PSNR of the whole sequence. Based on a quality weighted bit allocation method, a sliding window rate allocation method is proposed for the first time so that the window can slide along the video sequence with a certain sliding step. Experimental results show that, under dynamic bandwidth conditions, the proposed method can simultaneously satisfy the requirements for improving average PSNR of the whole video sequence greatly and reducing the fluctuations between adjacent frames greatly.
文摘In the case of video streaming over wireless channels, burst errors may lead to serious video quality degradation. By jointly exploiting the scheduling mechanism on different communication layers, this paper proposes a quality-aware cross-layer scheduling scheme to achieve unequal error control for each Latency-constraint Frame Set (LFS) of a video stream. After a network-layer agent at base station firstly utilizes the network-layer packet scheduling to provide packet-granularity importance classifi-cation for the current LFS, a link-layer agent at base station further utilizes the Radio-Link-Unit (RLU) scheduling to implement finer selective retransmission of the current LFS. Under scheduling delay and bandwidth constraints, the proposed scheme can be aware of the application-layer quality and time-varying channel conditions, and hence burst errors can simply be shifted to lower-priority transmission units in the current LFS. Simulation results demonstrate that the proposed scheme has strong robustness against burst errors, and thus improves the overall received quality of the video stream over wireless channels.
基金Project supported by the National Natural Science Foundation ofChina (No. 60302004) and the Natural Science Foundation of HubeiProvince (No. 2005ABA264), China
文摘Equation based TCP-friendly rate control (TFRC) protocol has been proposed to support video streaming applications. In order to improve TFRC performance in wireless channels, the link level automatic repeat request (ARQ) scheme is usually deployed. However, ARQ cannot ensure strict delay guarantees, especially over multihop links. This paper introduces a theoretical model to deduce an equation for packet size adjustment in transport layer to minimize retransmission delay by taking into con- sideration the causative reasons inducing retransmission in link layer. An enhanced TFRC (ETFRC) scheme is proposed inte- grating TFRC with variable packet size policy. Simulation results demonstrate that higher goodput, lower packet loss rate (PLR), lower frame transmission delay and jitter with good fairness can be achieved by our proposed mechanism.
基金Project (No. 60332030) supported by the National Natural ScienceFoundation of China
文摘For video streaming over lossy channels, intra refresh can mitigate the error-propagation effect caused by packet losses Besides some intra-mode macroblocks (MBs) generated by the "Lagrangian rate-distortion" or "Sum of absolute difference" mode decision, the encoder or transcoder possibly needs to increase some "forced" intra-mode MBs for robust video streaming. Based on the error-propagation analysis in a group of pictures (GOP), we propose an unequal Forced-Intra-Refresh (FIR) scheme to improve packet loss resilience of video streaming. According to a GOP-level error-propagation model, the proposed unequal FIR scheme can optimally increase the unequal number of forced intra-mode MBs for different frames in a GOP. Simulation results showed that the proposed scheme can effectively enhance the robustness of video streaming under different channel conditions, and achieve about 0. 1-0.9 dB gains over the average FIR scheme in H.264/AVC tools.
文摘The effect of receiver buffer size on perceived video quality of an Internet video streamer application was examined in this work. Several network conditions and several versions of the application are used to gain understanding of the response to varying buffer sizes. Among these conditions local area versus wide area, bandwidth estimation based versus non-bandwidth estimation based cases are examined in detail. A total of 1000 min of video is streamed over Intemet and statistics are collected. It was observed that when bandwidth estimation is possible, choosing larger buffer size for higher available bandwidth yields quality increase in perceived video.
文摘Reduced-reference (RR) video-quality estimators send a small signature to the receiver. This signature comprises the original video content as well as the video stream. RR quality estimation provides reliability and involves a small data payload. While significant in theory, RR estimators have only recently been used in practice for quality monitoring and adaptive system con- trol in streaming-video frameworks. In this paper, we classify RR algorithms according to whether they are based on a) model- ing the signal distortion, b) modeling the human visual system, or c) analyzing the video signal source. We review proposed RR techniques for monitoring and controlling quality in streaming video systems.
基金This work was supported by the National Natural Science Foundation of China(NSFC)under the grant No.61972269the Fundamental Research Funds for the Central Universities under the grant No.YJ201881Doctoral Innovation Fund Program of Southwest Jiaotong University under the grant No.DCX201824.
文摘This paper presents a reversible data hiding(RDH)method,which is designed by combining histogram modification(HM)with run-level coding in H.264/advanced video coding(AVC).In this scheme,the run-level is changed for embedding data into H.264/AVC video sequences.In order to guarantee the reversibility of the proposed scheme,the last nonzero quantized discrete cosine transform(DCT)coefficients in embeddable 4×4 blocks are shifted by the technology of histogram modification.The proposed scheme is realized after quantization and before entropy coding of H.264/AVC compression standard.Therefore,the embedded information can be correctly extracted at the decoding side.Peak-signal-noise-to-ratio(PSNR)and Structure similarity index(SSIM),embedding payload and bit-rate variation are exploited to measure the performance of the proposed scheme.Experimental results have shown that the proposed scheme leads to less SSIM variation and bit-rate increase.