Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093834
A. Fiandrotti, Simone Zezza, E. Magli
We present a novel architecture for complexity-adaptive Random Network Coding (RNC) and its application to Peer-to-Peer (P2P) video streaming. Network coding enables the design of simple and effective P2P video distribution systems, however it relies on computationally intensive packet coding operations that may exceed the computational capabilities of power constrained devices. It is hence desirable that the complexity of network coding can be adjusted at every node according to its computational capabilities, so that different classes of nodes can coexist in the network. To this end, we model the computational complexity of network coding as the sum of a packet decoding cost, which is centrally minimized at the encoder, and a packet recoding cost, which is locally controlled by each node. Efficient network coding is achieved exploiting the packet decoding process as a packet pre-recoding stage, hence increasing the chance that transmitted packets are innovative without increasing the recoding cost. Experiments in a P2P video streaming framework show that the proposed design enables the nodes of the network to operate at a wide range of computational complexity levels, while a higher number of low complexity nodes are able to join the network and experience high-quality video.
{"title":"Complexity-adaptive Random Network Coding for Peer-to-Peer video streaming","authors":"A. Fiandrotti, Simone Zezza, E. Magli","doi":"10.1109/MMSP.2011.6093834","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093834","url":null,"abstract":"We present a novel architecture for complexity-adaptive Random Network Coding (RNC) and its application to Peer-to-Peer (P2P) video streaming. Network coding enables the design of simple and effective P2P video distribution systems, however it relies on computationally intensive packet coding operations that may exceed the computational capabilities of power constrained devices. It is hence desirable that the complexity of network coding can be adjusted at every node according to its computational capabilities, so that different classes of nodes can coexist in the network. To this end, we model the computational complexity of network coding as the sum of a packet decoding cost, which is centrally minimized at the encoder, and a packet recoding cost, which is locally controlled by each node. Efficient network coding is achieved exploiting the packet decoding process as a packet pre-recoding stage, hence increasing the chance that transmitted packets are innovative without increasing the recoding cost. Experiments in a P2P video streaming framework show that the proposed design enables the nodes of the network to operate at a wide range of computational complexity levels, while a higher number of low complexity nodes are able to join the network and experience high-quality video.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"46 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134218032","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093826
R. Yu, Dajun Wu, Jianping Chen, S. Rahardja
In this paper, we propose a low-complexity priority based packet scheduling algorithm for streaming MPEG-4 Scalable to Lossless (SLS) encoded audio. In the proposed system, the SLS encoded frames are partitioned into data units of different quality layers, which are transmitted according to their quality contribution to the final decoded audio and their urgency relative to the playback progress. Experimental results show that the proposed scheduling algorithm has an even lower compared to traditional greedy algorithm for packet scheduling, while outperforms them by a significant margin in for terms of quality of the streamed audio.
{"title":"Low-complexity priority based packet scheduling for streaming MPEG-4 SLS","authors":"R. Yu, Dajun Wu, Jianping Chen, S. Rahardja","doi":"10.1109/MMSP.2011.6093826","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093826","url":null,"abstract":"In this paper, we propose a low-complexity priority based packet scheduling algorithm for streaming MPEG-4 Scalable to Lossless (SLS) encoded audio. In the proposed system, the SLS encoded frames are partitioned into data units of different quality layers, which are transmitted according to their quality contribution to the final decoded audio and their urgency relative to the playback progress. Experimental results show that the proposed scheduling algorithm has an even lower compared to traditional greedy algorithm for packet scheduling, while outperforms them by a significant margin in for terms of quality of the streamed audio.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"54 5","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131470796","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093799
H. Shidanshidi, F. Safaei, W. Li
Light field rendering (LFR) is an active research area in computer vision and computer graphics. LFR plays a crucial role in free viewpoint video systems (FVV). Several rendering algorithms have been suggested for LFR. However, comparative evaluation of these methods is often limited to subjective assessment of the output. To overcome this problem, this paper presents a geometric measurement, Effective Sampling Density of the scene, referred to as effective sampling for brevity, for objective comparison and evaluation of LFR algorithms. We have derived the effective sampling for the well-known LFR methods. Both theoretical study and numerical simulation have shown that the proposed effective sampling is an effective indicator of the performance for LFR methods.
{"title":"Objective evaluation of light field rendering methods using effective sampling density","authors":"H. Shidanshidi, F. Safaei, W. Li","doi":"10.1109/MMSP.2011.6093799","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093799","url":null,"abstract":"Light field rendering (LFR) is an active research area in computer vision and computer graphics. LFR plays a crucial role in free viewpoint video systems (FVV). Several rendering algorithms have been suggested for LFR. However, comparative evaluation of these methods is often limited to subjective assessment of the output. To overcome this problem, this paper presents a geometric measurement, Effective Sampling Density of the scene, referred to as effective sampling for brevity, for objective comparison and evaluation of LFR algorithms. We have derived the effective sampling for the well-known LFR methods. Both theoretical study and numerical simulation have shown that the proposed effective sampling is an effective indicator of the performance for LFR methods.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"127 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131636201","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093789
M. Rerábek, Lutz Goldmann, Jong-Seok Lee, T. Ebrahimi
While 3D display technologies are already widely available for cinema and home or corporate use, only a few portable devices currently feature 3D display capabilities. Moreover, the large majority of 3D display solutions rely on binocular perception. In this paper, we study the alternative methods for restitution of 3D images on conventional 2D displays and analyze their respective performance. This particularly includes the extension of wiggle stereoscopy for portable devices which relies on motion parallax as an additional depth cue. The goal of this paper is to compare two different 3D display techniques, the anaglyph method which provides binocular depth cues and a method based on motion parallax, and to show that the motion parallax based approach to present 3D images on consumer 2D portable screen is an equivalent way in comparison to the above mentioned and well-known anaglyph method. The subsequently conducted subjective quality tests show that viewers even prefer wiggle over anaglyph stereoscopy mainly due to a better color reproduction and a comparable depth perception.
{"title":"Motion parallax based restitution of 3D images on legacy consumer mobile devices","authors":"M. Rerábek, Lutz Goldmann, Jong-Seok Lee, T. Ebrahimi","doi":"10.1109/MMSP.2011.6093789","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093789","url":null,"abstract":"While 3D display technologies are already widely available for cinema and home or corporate use, only a few portable devices currently feature 3D display capabilities. Moreover, the large majority of 3D display solutions rely on binocular perception. In this paper, we study the alternative methods for restitution of 3D images on conventional 2D displays and analyze their respective performance. This particularly includes the extension of wiggle stereoscopy for portable devices which relies on motion parallax as an additional depth cue. The goal of this paper is to compare two different 3D display techniques, the anaglyph method which provides binocular depth cues and a method based on motion parallax, and to show that the motion parallax based approach to present 3D images on consumer 2D portable screen is an equivalent way in comparison to the above mentioned and well-known anaglyph method. The subsequently conducted subjective quality tests show that viewers even prefer wiggle over anaglyph stereoscopy mainly due to a better color reproduction and a comparable depth perception.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"49 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124948068","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093838
Marcus Laumer, P. Amon, A. Hutter, André Kaup
This paper presents a novel change detection algorithm for the compressed domain. Many video surveillance systems in practical use transmit their video data over a network by using the Real-time Transport Protocol (RTP). Therefore, the presented algorithm concentrates on analyzing RTP streams to detect major changes within contained video content. The paper focuses on a reliable preselection for further analysis modules by decreasing the number of events to be investigated. The algorithm is designed to work on scenes with mainly static background, like in indoor video surveillance streams. The extracted stream elements are RTP timestamps and RTP packet sizes. Both values are directly accessible by efficient byte-reading operations without any further decoding of the video content. Hence, the proposed approach is codec-independent, while at the same time its very low complexity enables the use in extensive video surveillance systems. About 40,000 frames per second of a single RTP stream can be processed on an Intel® CoreTM 2 Duo CPU at 2 GHz and 2 GB RAM, without decreasing the efficiency of the algorithm.
提出了一种新的压缩域变化检测算法。在实际应用中，许多视频监控系统使用实时传输协议(RTP)在网络上传输视频数据。因此，本文提出的算法侧重于分析RTP流，以检测包含视频内容的主要变化。通过减少待研究事件的数量，对进一步的分析模块进行了可靠的预选。该算法主要用于静态背景的场景，如室内视频监控流。提取的流元素是RTP时间戳和RTP数据包大小。这两个值都可以通过高效的字节读取操作直接访问，而无需对视频内容进行进一步解码。因此，所提出的方法是编解码器无关的，同时其非常低的复杂性使其能够在广泛的视频监控系统中使用。在2 GHz和2 GB RAM的Intel®CoreTM 2 Duo CPU上可以处理每秒约40,000帧的单个RTP流，而不会降低算法的效率。
{"title":"A compressed domain change detection algorithm for RTP streams in video surveillance applications","authors":"Marcus Laumer, P. Amon, A. Hutter, André Kaup","doi":"10.1109/MMSP.2011.6093838","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093838","url":null,"abstract":"This paper presents a novel change detection algorithm for the compressed domain. Many video surveillance systems in practical use transmit their video data over a network by using the Real-time Transport Protocol (RTP). Therefore, the presented algorithm concentrates on analyzing RTP streams to detect major changes within contained video content. The paper focuses on a reliable preselection for further analysis modules by decreasing the number of events to be investigated. The algorithm is designed to work on scenes with mainly static background, like in indoor video surveillance streams. The extracted stream elements are RTP timestamps and RTP packet sizes. Both values are directly accessible by efficient byte-reading operations without any further decoding of the video content. Hence, the proposed approach is codec-independent, while at the same time its very low complexity enables the use in extensive video surveillance systems. About 40,000 frames per second of a single RTP stream can be processed on an Intel® CoreTM 2 Duo CPU at 2 GHz and 2 GB RAM, without decreasing the efficiency of the algorithm.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125311430","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093793
Hongteng Xu, Guangtao Zhai
In this paper, a new ECG signal compression algorithm based on wave atom transform is presented. According to an assumption that ECG is oscillatory signal, we decompose ECG signal by wave atoms and trimmed insignificant coefficients. The wave atom decomposition has been proved to have a significantly sparser solution than other existing transform methods when it comes to oscillatory signal. In our experiment, the convergence of the energy of wave atoms' coefficients is faster than that of wavelet indeed. The most significant advantage of our algorithm is that unlike many conventional methods, the performance of our algorithm is not dependent on QRS detection, which simplifies the architecture of compression system and is beneficial to telemedicine application. After wave atom transform, the data stream is divided and coded by a hybrid entropy coding strategy combining delta coding, run-length-coding and arithmetic coding. The experimental results on MIT-BIH arrhythmia database proved that our algorithm has high compression ratio (CR > 10) with percentage root mean square difference (PRD) under 1%.
{"title":"ECG data compression based on wave atom transform","authors":"Hongteng Xu, Guangtao Zhai","doi":"10.1109/MMSP.2011.6093793","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093793","url":null,"abstract":"In this paper, a new ECG signal compression algorithm based on wave atom transform is presented. According to an assumption that ECG is oscillatory signal, we decompose ECG signal by wave atoms and trimmed insignificant coefficients. The wave atom decomposition has been proved to have a significantly sparser solution than other existing transform methods when it comes to oscillatory signal. In our experiment, the convergence of the energy of wave atoms' coefficients is faster than that of wavelet indeed. The most significant advantage of our algorithm is that unlike many conventional methods, the performance of our algorithm is not dependent on QRS detection, which simplifies the architecture of compression system and is beneficial to telemedicine application. After wave atom transform, the data stream is divided and coded by a hybrid entropy coding strategy combining delta coding, run-length-coding and arithmetic coding. The experimental results on MIT-BIH arrhythmia database proved that our algorithm has high compression ratio (CR > 10) with percentage root mean square difference (PRD) under 1%.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128103075","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093848
Jin Wang, Yunhui Shi, Wenpeng Ding, Baocai Yin
Intra prediction plays an important role in reducing the spatial redundancy for intra frame encoding in H.264/AVC. In this paper, we propose a low-rank matrix completion based intra prediction to improve the prediction efficiency. According to the low-rank matrix completion theory, a low-rank matrix can be exactly recovered from quite limited samples with high probability under mild conditions. After moderate rearrangement and organization, image blocks can be represented as low-rank or approximately low-rank matrix. The intra prediction can then be formulated as a matrix completion problem, thus the unknown pixels can be inferred from limited samples with very high accuracy. Specifically, we novelly rearrange the encoded blocks similar to the current block to generate an observation matrix, from which the prediction can be obtained by solving a low-rank minimization problem. Experimental results demonstrate that the proposed scheme can achieve averagely 5.39% bit-rate saving for CIF sequences and 4.21% for QCIF sequences compared with standard H.264/AVC.
{"title":"A low-rank matrix completion based intra prediction for H.264/AVC","authors":"Jin Wang, Yunhui Shi, Wenpeng Ding, Baocai Yin","doi":"10.1109/MMSP.2011.6093848","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093848","url":null,"abstract":"Intra prediction plays an important role in reducing the spatial redundancy for intra frame encoding in H.264/AVC. In this paper, we propose a low-rank matrix completion based intra prediction to improve the prediction efficiency. According to the low-rank matrix completion theory, a low-rank matrix can be exactly recovered from quite limited samples with high probability under mild conditions. After moderate rearrangement and organization, image blocks can be represented as low-rank or approximately low-rank matrix. The intra prediction can then be formulated as a matrix completion problem, thus the unknown pixels can be inferred from limited samples with very high accuracy. Specifically, we novelly rearrange the encoded blocks similar to the current block to generate an observation matrix, from which the prediction can be obtained by solving a low-rank minimization problem. Experimental results demonstrate that the proposed scheme can achieve averagely 5.39% bit-rate saving for CIF sequences and 4.21% for QCIF sequences compared with standard H.264/AVC.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"26 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116565647","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093821
A. E. Essaili, Liang Zhou, Damien Schroeder, E. Steinbach, W. Kellerer
We consider the joint upstreaming of live and on-demand user-generated video content over LTE using a Quality-of-Experience driven approach. We contribute to the state-of-the-art work on multimedia scheduling in three aspects: 1) we jointly optimize the transmission of live and time-shifted video under scarce uplink resources by transmitting a basic quality in realtime and uploading a refined quality for on-demand consumption. 2) We propose a producer-consumer deadline-aware scheduling algorithm that incorporates both the physical state of the mobile producer (e.g., cache fullness) and the scheduled playout time at the end-user. 3) We show that the scheduling decisions in 1) and 2) can be determined locally for each mobile producer. We additionally present an analytical framework for de-centralized scalable video transmission and prove that there exists an optimal solution to our problem. Simulation results for LTE uplink further demonstrate the significance of our proposed optimization on the overall user experience.
{"title":"QoE-driven live and on-demand LTE uplink video transmission","authors":"A. E. Essaili, Liang Zhou, Damien Schroeder, E. Steinbach, W. Kellerer","doi":"10.1109/MMSP.2011.6093821","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093821","url":null,"abstract":"We consider the joint upstreaming of live and on-demand user-generated video content over LTE using a Quality-of-Experience driven approach. We contribute to the state-of-the-art work on multimedia scheduling in three aspects: 1) we jointly optimize the transmission of live and time-shifted video under scarce uplink resources by transmitting a basic quality in realtime and uploading a refined quality for on-demand consumption. 2) We propose a producer-consumer deadline-aware scheduling algorithm that incorporates both the physical state of the mobile producer (e.g., cache fullness) and the scheduled playout time at the end-user. 3) We show that the scheduling decisions in 1) and 2) can be determined locally for each mobile producer. We additionally present an analytical framework for de-centralized scalable video transmission and prove that there exists an optimal solution to our problem. Simulation results for LTE uplink further demonstrate the significance of our proposed optimization on the overall user experience.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129786274","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093784
T. Phan, Yuichi Tanaka, Madoka Hasegawa, Shigeo Kato
In current distributed video coding (DVC), interpolation is performed at the decoder and the interpolated pixels are reconstructed by using error-correcting codes, such as Turbo codes and LDPC. There are two possibilities for downsampling video sequences at the encoder: temporally or spatially. Traditionally temporal downsampling, i.e., frame dropping, is used for DVC. Furthermore, those with spatial downsampling (scaling) have been investigated. Unfortunately, most of them are based on uniform downsampling. Due to this, details in video sequences are often discarded. For example, edges and textured regions are difficult to interpolate, and thus require many parity bits to restore the interpolated portions for the spatial domain DVC. In this paper, we propose a new spatial domain DVC based on adaptive line dropping so-called selective data pruning (SDP). SDP is a simple nonuniform downsampling method. The pruned lines are determined to avoid cutting across edges and textures. Experimental results show the proposed method outperforms a conventional DVC for sequences with a large amount of motions.
{"title":"Mixed-resolution Wyner-Ziv video coding based on selective data pruning","authors":"T. Phan, Yuichi Tanaka, Madoka Hasegawa, Shigeo Kato","doi":"10.1109/MMSP.2011.6093784","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093784","url":null,"abstract":"In current distributed video coding (DVC), interpolation is performed at the decoder and the interpolated pixels are reconstructed by using error-correcting codes, such as Turbo codes and LDPC. There are two possibilities for downsampling video sequences at the encoder: temporally or spatially. Traditionally temporal downsampling, i.e., frame dropping, is used for DVC. Furthermore, those with spatial downsampling (scaling) have been investigated. Unfortunately, most of them are based on uniform downsampling. Due to this, details in video sequences are often discarded. For example, edges and textured regions are difficult to interpolate, and thus require many parity bits to restore the interpolated portions for the spatial domain DVC. In this paper, we propose a new spatial domain DVC based on adaptive line dropping so-called selective data pruning (SDP). SDP is a simple nonuniform downsampling method. The pruned lines are determined to avoid cutting across edges and textures. Experimental results show the proposed method outperforms a conventional DVC for sequences with a large amount of motions.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"46 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133062275","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2011-12-01DOI: 10.1109/MMSP.2011.6093809
Mina Makar, Y. Lin, A. Araújo, B. Girod
Reduced reference video quality assessment techniques provide a practical and convenient way of evaluating the quality of a processed video. In this paper, we propose a method to efficiently compress standardized VQM (Video Quality Model) [1] features to bit-rates that are small relative to the transmitted video. This is achieved through two stages of compression. In the first stage, we remove the redundancy in the features by only transmitting the necessary original video features at the lowest acceptable resolution for the calculation of the final VQM value. The second stage involves using the features of the processed video at the receiver as side-information for efficient entropy coding and reconstruction of the original video features. Experimental results demonstrate that our approach achieves high compression ratios of more than 30× with small error in the final VQM values.
{"title":"Compression of VQM features for low bit-rate video quality monitoring","authors":"Mina Makar, Y. Lin, A. Araújo, B. Girod","doi":"10.1109/MMSP.2011.6093809","DOIUrl":"https://doi.org/10.1109/MMSP.2011.6093809","url":null,"abstract":"Reduced reference video quality assessment techniques provide a practical and convenient way of evaluating the quality of a processed video. In this paper, we propose a method to efficiently compress standardized VQM (Video Quality Model) [1] features to bit-rates that are small relative to the transmitted video. This is achieved through two stages of compression. In the first stage, we remove the redundancy in the features by only transmitting the necessary original video features at the lowest acceptable resolution for the calculation of the final VQM value. The second stage involves using the features of the processed video at the receiver as side-information for efficient entropy coding and reconstruction of the original video features. Experimental results demonstrate that our approach achieves high compression ratios of more than 30× with small error in the final VQM values.","PeriodicalId":214459,"journal":{"name":"2011 IEEE 13th International Workshop on Multimedia Signal Processing","volume":"39 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123767185","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
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