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Showing papers on "Video quality published in 1998"


Patent
26 May 1998
TL;DR: In this article, a cost effective continuously adaptive digital video system and method for compressing color video data for moving images is described, which describes capturing an analog video frame and digitizing the image into a preferred source input format for compression using a combination of unique lossy and lossless digital compression techniques including subband coding, wavelet transforms, motion detection, run length coding and variable length coding.
Abstract: A cost effective continuously adaptive digital video system and method for compressing color video data for moving images The method describes capturing an analog video frame and digitizing the image into a preferred source input format for compression using a combination of unique lossy and lossless digital compression techniques including subband coding, wavelet transforms, motion detection, run length coding and variable length coding The system includes encoder and decoder (CODEC) sections for compression and decompression of visual images to provide high compression with good to excellent video quality The compressed video data provides a base video layer and additional layers of video data that are multiplexed with compressed digital audio to provide a data stream that can be packetized for distribution over inter or intranets, including wireless networks over local or wide areas The (CODEC) system continuously adjusts the compression of the digital images frame by frame in response to comparing the available bandwidth on the data channel to the available bandwidth on the channel for the previous frame to provide an output data stream commensurate with the available bandwidth of the network transmission channel and with the receiver resource capabilities of the client users The compression may be further adjusted by adjustment of the frame rate of the output data stream

353 citations


Journal ArticleDOI
01 May 1998
TL;DR: An underlying theme of this paper is that increased interaction between the video and network design has potential for improving overall decoded video quality without changing the network capacity.
Abstract: The authors examine the transport and storage of video compressed with a variable bit rate (VBR). They focus primarily on networked video, although they also briefly consider other applications of VBR video, including satellite transmission (channel sharing), playback of stored video, and wireless transport. Packet video research requires careful integration between the network and the video systems; however, a major stumbling block has resulted because commonly used terms are often interpreted differently by the video and networking communities. The paper then, has two main goals: (i) to clarify the definitions of terms that are often used with different meaning by networking and video-coding researchers and (ii) to explore the tradeoffs entailed by each of the various modalities of VBR transmission (unconstrained, shaped, constrained, and feedback). In particular, they evaluate the tradeoff among the advantages (better video quality, less delay, and more calls) that were identified by early proponents of VBR video transmission. An underlying theme of this paper is that increased interaction between the video and network design has potential for improving overall decoded video quality without changing the network capacity.

270 citations


Proceedings ArticleDOI
01 Sep 1998
TL;DR: It is argued that ITU-recommended methods for subjective quality assessment of speech and video are not suitable for assessing the quality of many newer services and applications.
Abstract: There is currently much discussion of Quality of Service (QoS) measurements at the network level of real-time multimedia services, but it is the subjective qualify perceived by the user that will determine whether these applications are adopted This paper argues that ITU-recommended methods for subjective quality assessment of speech and video are not suitable for assessing the quality of many newer services and applications. We present an outline of what we believe to be a more suitable testing methodology, which acknowledges the multi-dimensional nature of perceived audio and video quality.

217 citations


Proceedings ArticleDOI
17 Jul 1998
TL;DR: A new video quality metric is described that is an extension of these still image metrics into the time domain, based on the Discrete Cosine Transform, in order that might be applied in the widest range of applications.
Abstract: The advent of widespread distribution of digital video creates a need for automated methods for evaluating the visual quality of digital video. This is particularly so since most digital video is compressed using lossy methods, which involve the controlled introduction of potentially visible artifacts. Compounding the problem is the bursty nature of digital video, which requires adaptive bit allocation based on visual quality metrics, and the economic need to reduce bit-rate to the lowest level that yields acceptable quality. In previous work, we have developed visual quality metrics for evaluating, controlling,a nd optimizing the quality of compressed still images. These metrics incorporate simplified models of human visual sensitivity to spatial and chromatic visual signals. Here I describe a new video quality metric that is an extension of these still image metrics into the time domain. Like the still image metrics, it is based on the Discrete Cosine Transform. An effort has been made to minimize the amount of memory and computation required by the metric, in order that might be applied in the widest range of applications. To calibrate the basic sensitivity of this metric to spatial and temporal signals we have made measurements of visual thresholds for temporally varying samples of DCT quantization noise.© (1998) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

134 citations


Proceedings ArticleDOI
01 Oct 1998
TL;DR: Through the results of extensive Internet experiments, the paper shows that layered coding can be very effective when combined with the retransmission-based error control technique for low-bit rate transmission over best effort networks where no network-level mechanism exists for protecting high priority data from packet loss.
Abstract: A new retransmission-based error control technique is presented that does not incur any additional latency in frame playout times, and hence are suitable for interactive applications. It takes advantage of the motion prediction loop employed in most motion compensation-based codecs. By correcting errors in a reference frame caused by earlier packet loss, it prevents error propagation. The technique rearranges the temporal dependency of frames so that a displayed frame is referenced for the decoding of its succeeding dependent frames much later than its display time. Thus, the delay in repairing lost packets can be effectively masked out. The developed technique is combined with layered video coding to maintain consistently good video quality even under heavy packet loss. Through the results of extensive Internet experiments, the paper shows that layered coding can be very effective when combined with the retransmission-based error control technique for low-bit rate transmission over best effort networks where no network-level mechanism exists for protecting high priority data from packet loss.

125 citations


Journal ArticleDOI
Tae-Yun Chung1, Kang-Seo Park2, Young-Nam Oh1, Dong-Ho Shin1, Sang-Hui Park2 
01 Aug 1998
TL;DR: In this paper, the authors proposed a watermarking technique for MPEG-2 video coding system by extending the direct sequence spread spectrum to control the strength and area of embedding with respect to the global and local characteristics of the video sequences.
Abstract: Digital watermarking is the technique which embeds an invisible signal including owner identification and copy control information into multimedia data such as audio, video, images, for copyright protection. This paper proposes a new watermarking technique which is appropriate for the MPEG-2 video coding system by extending the direct sequence spread spectrum. The proposed watermarking scheme in this paper minimizes the perceptual degradation of video quality, caused by the embedded watermark through controlling the strength and area of embedding with respect to the global and local characteristics of the video sequences. And this technique can also directly extract the embedded watermark from the watermarked MPEG-2 compressed video stream without using original video sequences. Simulation results show that the proposed scheme is robust to various attacks to remove the embedded watermark and is feasible for real time system implementation.

113 citations


Journal ArticleDOI
TL;DR: This model has been able to mimic quite accurately the temporally varying subjective picture quality of video sequences as recorded by the ITU-R SSCQE method.

108 citations


Proceedings ArticleDOI
28 Dec 1998
TL;DR: Experimental results show that, with the approach, the picture quality of a streamed video degrades gracefully as the packet loss probability of an Internet connection increases.
Abstract: This paper describes a transmission scheme for Internet video streaming that provides an acceptable video quality over a wide range of connection qualities. The proposed system consists of a scalable video coder which uses a fully standard compatible H.263 coder in its base layer. The scalable video coder is combined with unequal error protection using Reed- Solomon codes applied across packets. We present and verify a two-state Markov model for packet losses over Internet connections. The relation between packet loss and picture quality at the decoder for an unequally protected layered video stream is derived. Experimental results show that, with our approach, the picture quality of a streamed video degrades gracefully as the packet loss probability of an Internet connection increases.© (1998) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

91 citations


Journal ArticleDOI
TL;DR: An adaptive source rate control (ASRC) scheme which can work together with the hybrid ARQ error control schemes to achieve efficient transmission of real-time video with low delay and high reliability is proposed.
Abstract: Hybrid ARQ schemes can yield much better throughput and reliability than static FEC schemes for the transmission of data over time-varying wireless channels. However these schemes result in extra delay. They adapt to the varying channel conditions by retransmitting erroneous packets, this causes variable effective data rates for current PCS networks because the channel bandwidth is constant. Hybrid ARQ schemes are currently being proposed as the error control schemes for real-time video transmission. An important issue is how to ensure low delay while taking advantage of the high throughput and reliability that these schemes provide for. In this paper we propose an adaptive source rate control (ASRC) scheme which can work together with the hybrid ARQ error control schemes to achieve efficient transmission of real-time video with low delay and high reliability. The ASRC scheme adjusts the source rate based on the channel conditions, the transport buffer occupancy and the delay constraints. It achieves good video quality by dynamically changing both the number of the forced update (intracoded) macroblocks and the quantization scale used in a frame. The number of the forced update macroblocks used in a frame is first adjusted according to the allocated source rate. This reduces the fluctuation of the quantization scale with the change in the channel conditions during encoding so that the uniformity of the video quality is improved. The simulation results show that the proposed ASRC scheme performs very well for both slow fading and fast fading channels.

90 citations


Proceedings ArticleDOI
04 Oct 1998
TL;DR: The paper treats scene change detection as a two-class classification problem and employ automatic threshold selection techniques originally developed for image binarization, and a quantitative measure for retrieval of similar scenes according to their color content is defined.
Abstract: The paper addresses automatic scene change detection, key frame selection, and similarity ranking which constitute the main steps of a content based video abstraction system. Unlike other methods, the proposed algorithm performs scene change detection and key frame selection in one step. We treat scene change detection as a two-class classification problem and employ automatic threshold selection techniques originally developed for image binarization. A quantitative measure for retrieval of similar scenes according to their color content is also defined. The described scheme can be applied to both uncompressed and MPEG compressed video, and can be implemented in real time. Performance of the algorithm has been analyzed on real TV sequences, and comparison with some previously introduced techniques are provided.

86 citations


Journal ArticleDOI
TL;DR: A fuzzy logic-based control scheme for real-time motion picture expert group (MPEG) video to avoid long delay or excessive loss at the user-network interface (UNI) in an asynchronous transfer mode (ATM) network is proposed.
Abstract: We propose a fuzzy logic-based control scheme for real-time motion picture expert group (MPEG) video to avoid long delay or excessive loss at the user-network interface (UNI) in an asynchronous transfer mode (ATM) network. The system consists of a shaper whose role is to smooth the MPEG output traffic to reduce the burstiness of the video stream. The input and output rates of the shaper buffer are controlled by two fuzzy logic-based controllers. To avoid a long delay at the shaper, the first controller aims to tune the output rate of the shaper in the video frame time scale based on the number of available transmission credits at the UNI and the occupancy of the shaper's buffer. Based on the average occupancy of the shaper's buffer and its variance, the second controller tunes the input rate to the shaper over a much larger time scale by applying a closed-loop MPEG encoding scheme. With this approach, the traffic enters the network at an almost constant bit rate (with a very small variation) allowing simple network management functions such as admission control and bandwidth allocation, while guaranteeing a relatively constant video quality since the encoding rate is changed only in critical periods when the shaper buffer "threatens" to overflow. The performance of the proposed scheme is evaluated through numerical tests on real video sequences.

Patent
22 May 1998
TL;DR: In this paper, a multipoint control unit monitors the utilization of T. 120 MLP data channels and changes the bandwidth allocation for all of the transmission links in response to changes in the utilization for at least one of the data channels.
Abstract: The specification discloses a multimedia conferencing service capability for dynamically allocating transmission link bandwidth among video, audio and data channels in response to changes in the utilization of those channels. The capability maximizes video quality when data is not being used and minimizes data transfer time when data is being used in a multimedia conference. A multipoint control unit (MCU) monitors the utilization of T. 120 MLP data channels and changes the bandwidth allocation for all of the transmission links in response to changes in the utilization for at least one of the data channels. The utilization is monitored by measuring the data rate of outbound T. 120 MLP data channels and comparing the measured data rate to the allocated bandwidth to determine the data channel utilization. The MCU controls and manages a service policy in which video quality is sacrificed in favor of data transfers to a majority of conference sites while conversely, video quality is favored over data transfer time for individual data needs. Since the dynamic bandwidth allocation is provided at the MCU, no changes or upgrades are needed at the endpoint multimedia terminals.

Patent
16 Jun 1998
TL;DR: In this article, a smoothing and rate adaptation algorithm is proposed to facilitate the flow of video data, maintaining video quality while avoiding potentially harmful buffering delays, and also adapts the encoding rate in relation to a target delay for a source buffer.
Abstract: A method and apparatus provide a smoothing and rate adaptation algorithm to facilitate the flow of video data, maintaining video quality while avoiding potentially harmful buffering delays. The invention uses a smoothing interval to determine a rate to request for allocation. The invention also adapts the encoding rate in relation to a target delay for a source buffer.

Journal ArticleDOI
01 Aug 1998
TL;DR: This paper analyzed the quantization errors that cause the extracted motion vectors to be non-optimal and performed simulations to show the quality degradation due to the inaccurate motion vectors during transcoding to improve the video quality.
Abstract: Traditionally, the re-use of motion vectors extracted from incoming video bit-stream during transcoding has been widely accepted. However, this simple re-use scheme introduces significant quality degradation in many applications including the situation when the frame-rate conversion is needed. In this paper, we analyzed the quantization errors that cause the extracted motion vectors to be non-optimal and we performed simulations to show the quality degradation due to the inaccurate motion vectors during transcoding. To improve the video quality, we proposed an adaptive motion vector refinement. With a highly reduced computational complexity, the proposed adaptive motion vector refinement achieves significant quality improvement in comparison to the conventional motion vector re-use scheme. In addition, the adaptive motion vector refinement is almost as good as performing a new full-scale motion estimation.

Proceedings ArticleDOI
08 Sep 1998
TL;DR: This paper focuses on processing encoded video for transmission under low battery power conditions, and attempts to reduce deterioration of video quality.
Abstract: Mobile computers typically have limited energy for computing and communications due to short battery lifetimes. Encoding, decoding, and transmission of video information require significant computing and communication resources. Low power encoding and decoding schemes have been researched extensively. In this paper, we focus on processing encoded video for transmission under low battery power conditions. Such processing, while conserving battery power, attempts to reduce deterioration of video quality.

Proceedings ArticleDOI
01 Oct 1998
TL;DR: It has been observed from experimental results that the visual quality of coded low- bit-rate video is significantly improved at the expense of a small increase in decoder's complexity.
Abstract: Fast motion-compensated frame interpolation (FMCI) schemes for the decoder of the block-based video codec operating in low bit rates are examined in this paper. The main objective is to improve the video quality by increasing the frame rate without a substantial increase in the computational complexity. Two FMCI schemes are proposed depending on the motion vector mapping strategy, i.e. the non-deformable and the deformable block-based FMCI schemes. They provide a trade-off of the computational complexity and the visual performance.With proposed schemes, the decoder can perform frame interpolation using motion information received fromthe encoder. The complexity of FMCI is reduced since no additional motion search in the decoder is neededas required by standard MCI. It has been observed from experimental results that the visual quality of coded low-bit-rate video is significantly improved at the expense of a small increase in decoder's complexity. Keywords : motion-compensated frame interpolation ,

01 Jan 1998
TL;DR: It is demonstrated that, when jointly studying the impact of coding bit rate and packet loss, the reachable quality is upperbound and exhibits one optimal coding rate for a given packet loss ratio.
Abstract: We address the problem of video quality prediction and control for high resolution video transmitted over lossy packet networks. We analyze how the user-perceived quality is related to the average encoding bitrate for VBR MPEG2 video. We then show why simple distortion metrics (e.g., PSNR) may lead to inconsistent interpretations. Furthermore, for a given coder setup, we analyze the effect of packet loss on the user-level quality. We then demonstrate that, when jointly studying the impact of coding bit rate and packet loss, the reachable quality is upperbound and exhibits one optimal coding rate for a given packet loss ratio.

Proceedings ArticleDOI
17 Jul 1998
TL;DR: A three-stage method of measuring time-variant quality, which has been accepted by the ITU-R, and some of the factors identified as being important in producing good overall quality judgements have relevance to the design of optimal coding strategies for digital television.
Abstract: Over the past several years we have investigated viewer response to temporal fluctuations in the quality of digital television pictures, which occur when video is coded into relatively low bit rates. Three phenomena of interest have been identified: (1) a forgiveness effect, (2) a recency effect, and (3) a negative-peak (duration-neglect); these are described and discussed in the paper. In collaboration with our partners in European projects MOSAIC and TAPESTRIES, we have developed a three-stage method of measuring time-variant quality, which has been accepted by the ITU-R. The first stage is a Single Stimulus Continuous Quality Evaluation (SSCQE) of instantaneous quality; the second a calibration stage to link SSCQE with conventional DSCQS, and the third stage a numerical procedure for relating continuous and overall quality. Some of the factors we have identified as being important in producing good overall quality judgements have relevance to the design of optimal coding strategies for digital television.© (1998) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Book ChapterDOI
Sanjoy Paul1
01 Jan 1998
TL;DR: Layered video multicast with retransmission improves the video quality within each layer by using retransmissions from a nearby receiver given an upper bound on the recovery time, and uses a hierarchical rate control mechanism for making intelligent decisions about who should perform a join experiment and when.
Abstract: Layered video multicast with retransmission (LVMR) is a system for distributing video in a heterogeneous network, such as, the Internet. LVMR uses layered encoding of video and also uses IP multicast to multicast each layer of video using a different Class-D address. However, LVMR improves the video quality within each layer by using retransmissions from a nearby receiver given an upper bound on the recovery time, and also uses a hierarchical rate control mechanism for making intelligent decisions about who should perform a join experiment and when. Thus the two key contributions of LVMR are: 1. improving the quality of reception by intelligent retransmissions and 2. adapting to network congestion and heterogeneity using a hierarchical rate control mechanism.

Proceedings ArticleDOI
18 May 1998
TL;DR: This paper discusses the analysis of an audiovisual desktop video-teleconferencing subjective experiment conducted at the Institute for Telecommunication Sciences, where objective models of the individual audio and video quality are presented.
Abstract: This paper discusses the analysis of an audiovisual desktop video-teleconferencing subjective experiment conducted at the Institute for Telecommunication Sciences. Objective models of the individual audio and video quality are presented. Also discussed is an objective model of the audiovisual quality based upon the results of the individual objective audio and video quality models. Finally, a subjective model of audiovisual quality based upon users' ratings of the audio and video quality is discussed.

Proceedings ArticleDOI
01 Jan 1998
TL;DR: This work extends previous work on texture mapping video streams into virtual environments by introducing awareness driven video QoS, which uses movements within a shared virtual world to activate different video services.
Abstract: We extend previous work on texture mapping video streams into virtual environments by introducing awareness driven video QoS. This uses movements within a shared virtual world to activate different video services. In turn, these services have different settings for underlying QoS parameters such as frame-rate, resolution and compression. We demonstrate this technique through a combined conferencing! mediaspace application which uses awareness driven video for facial expressions and for views into remote physical environments. We reflect on the issues of spatial consistency, privacy, seamless shifts in mutual involvement and making underlying QoS mechanisms more visible, malleable and flexible.

Journal ArticleDOI
TL;DR: Preliminary results reveal the effects of network impairments on video quality for MPEG-2 transport streams delivered over ATM, including the trade-off between bandwidth savings and video quality.
Abstract: Experiments investigated the relationship between MPEG video quality and ATM network performance. Preliminary results reveal the effects of network impairments on video quality for MPEG-2 transport streams delivered over ATM. The issues in mapping variable-rate MPEG to ATM are also examined, including the trade-off between bandwidth savings and video quality.

Proceedings ArticleDOI
28 Dec 1998
TL;DR: It is shown that the dejitter buffering delay can be used to the advantage for packet loss recovery with video retransmission without incurring any extra delay.
Abstract: In this paper, we investigate how to adapt different parameters in H.263 source coding, transport processing and error concealment to optimize end-to-end video quality at different bitrates and packet loss rates for H.323-based packet video. First different intra coding patterns are compared and we show that the contiguous rectangle or square block pattern offers the best performance in terms of video quality in the presence of packet loss. Second, the optimal intra coding frequency is found for different bitrates and packet loss rates. The optimal number of GOB headers to be inserted in the source coding is then determined. The effect of transport processing strategies such as packetization and retransmission is also examined. For packetization, the impact of packet size and the effect of macroblock segmentation to picture quality are investigated. Finally, we show that the dejitter buffering delay can be used to the advantage for packet loss recovery with video retransmission without incurring any extra delay.© (1998) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Journal ArticleDOI
TL;DR: The applications, benefits, and challenges of using VBR MPEG video encoding in broadband video distribution networks and a scheduling technique is presented which selects a traffic contract for a pre-encoded MPEG video stream with the criteria of minimizing network resources and maintaining video quality.
Abstract: This article provides an overview of residential video delivery systems and presents the applications, benefits, and challenges of using VBR MPEG video encoding in broadband video distribution networks. The network resources required to transmit stored variable-rate MPEG can be reduced by properly analyzing and smoothing the video stream before transmission. A scheduling technique is presented which selects a traffic contract for a pre-encoded MPEG video stream with the criteria of minimizing network resources and maintaining video quality. Several effective bandwidth metrics are discussed and used to model the potential savings in network resources for the shaped streams.

Journal ArticleDOI
TL;DR: In this article, the authors present results of subjective viewer assessment of video quality of MPEG-2 compressed video containing wide-band Gaussian noise and find that compression at higher bit-rates can actually improve the quality of the output, effectively acting like a low pass filter.
Abstract: We present results of subjective viewer assessment of video quality of MPEG-2 compressed video containing wide-band Gaussian noise. The video test sequences consisted of seven test clips (both classical and new materials) to which noise with a peak-signal-to-noise-ratio (PSNR) of from 28 dB to 47 dB was added. We used software encoding and decoding at five bit-rates ranging from 1.8 Mb/s to 13.9 Mb/s. Our panel of 32 viewers rated the difference between the noisy input and the compression-processed output. For low noise levels, the subjective data suggests that compression at higher bit-rates can actually improve the quality of the output, effectively acting like a low-pass filter. We define an objective and a subjective measure of scene criticality (the difficulty of compressing a clip) and find the two measures correlate for our data. For difficult-to-encode material (high criticality), the data suggest that the effects of compression may be less noticeable at mid-level noise, while for easy-to-encode video (low criticality), the addition of a moderate amount of noise to the input led to lower quality scores. This suggests that either the compression process may have reduced noise impairments or a form of masking may occur in scenes that have high levels of spatial detail.

Patent
06 Feb 1998
TL;DR: In this article, a method and system for transmitting a video stream in an asynchronous transfer mode (ATM) network comprises steps of encoding the video into an MPEG-2 variable bit rate video stream, shaping the encoded variable bit-rate video stream to conform to the traffic contract parameters for a Variable Bit Rate (VBR) connection in the network, and transmitting the shaped variable bit rates video stream on the VBR connection based on the traffic contracts parameters.
Abstract: A method and system for transmitting a video stream in an asynchronous transfer mode (ATM) network comprises steps of encoding the video into an MPEG-2 variable bit rate video stream, shaping the encoded variable bit rate video stream to conform to the traffic contract parameters for a Variable Bit Rate (VBR) connection in the network, and transmitting the shaped variable bit rate video stream on the VBR connection based on the traffic contract parameters. For a given network bandwidth, switch buffer space, and equivalent video quality, the network can statistically multiplex a larger number of variable bit rate video streams by maximizing the utilization of the network bandwidth and switch buffer space that the network allocates to each video stream.

Proceedings ArticleDOI
04 Oct 1998
TL;DR: The preliminary results show that a mobile with the proposed method consumes 10% less energy than a mobile that comes with the most efficient power-rate mode (with a search window size of 8) of H.263 video coding in one-hour mobile communication simulation.
Abstract: We study how to use the H.263 video communications standard efficiently to save the total consumed energy of a mobile unit in cellular networks. Particularly, we study the computational power dissipation of various operation modes available in the H.263 coding standard. We show how to achieve low power consumption in a mobile unit by judiciously selecting the operating mode of H.263 in response to the mobile environment changes (e.g., slow fading and path loss), while maintaining a good video quality level. Our preliminary results show that a mobile with the proposed method consumes 10% less energy than a mobile that comes with the most efficient power-rate mode (with a search window size of 8) of H.263 video coding in one-hour mobile communication simulation. It even saves around 32% energy over a mobile operating with an advanced coding method (using four negotiating options).

Proceedings ArticleDOI
05 Aug 1998
TL;DR: The similarity measures of video content are investigated and a series of similarity measures based on the similarity of frame sequence which take temporal ordering into consideration are proposed.
Abstract: Video retrieval is one of the important design issues of multimedia database management systems. The distinguished features of video retrieval lie in the similarity measures and content-based retrieval. Most research on content-based video retrieval represents the content of video as a set of frames leaving out the temporal ordering of frames in the shot. In this paper, the similarity measures of video content are investigated. We propose a series of similarity measures based on the similarity of frame sequence which take temporal ordering into consideration. The corresponding algorithms are also presented. The effectiveness of the developed similarity measures measured by precision and recall is also described.

Journal ArticleDOI
TL;DR: It was observed that the dynamic rate control provides at least an acceptable video quality under severe channel conditions and a good video quality when the channel conditions are favorable.
Abstract: Video transmission over wireless packet networks is gaining importance due to the concept of universal personal communication. Further, it is considered an important step towards wireless multimedia. The challenge however is to achieve good video quality over mobile channels, where typically the channel conditions vary due to signal fading. Hence this paper investigates adaptive rate controlled video transmission for robust video communication under packet wireless environment. A combination of mobile and an ATM backbone network is assumed in this work. An error resilient design for the video coder, as proposed in Rajugopal et al. (1996) is employed here. This video coder comprises wavelet transform (WT), multi-resolution motion estimation (MRME) and a robust design for zero tree quantization. Two configurations, one employing MRME and the other using 1D-WT for temporal analysis, are considered for the video coder. Adaptive dynamic rate control is required to adapt the video communication to the channel conditions. It provides more channel protection when the channel is severe and improves the source rate and hence the performance when the conditions are favorable. An algorithm for dynamic rate control under varying channel conditions is proposed in this paper. It is evaluated under narrowband and broadband channel conditions. From the results, it is concluded that the dynamic rate control is very effective in optimizing the quality under varying mobile channel conditions. It was observed that the dynamic rate control provides at least an acceptable video quality under severe channel conditions and a good video quality when the channel conditions are favorable.

Journal ArticleDOI
TL;DR: Two new slice-based discard schemes for use with available bit rate and guaranteed frame rate services and a dynamic frame-level priority data partition technique based on MPEG data structure and feedback from the network are proposed and evaluated.
Abstract: With increasing interest in the transmission of audio–visual applications over ATM best effort services, efficient video-oriented control mechanisms for improving the video quality in the presence of loss have to be designed. In this paper, we propose and evaluate two new slice-based discard schemes for use with available bit rate and guaranteed frame rate services (e.g. formerly UBR+). The schemes adaptively and selectively adjust the discard level to switch buffer occupancy and video cell payload types. To improve their performance, we also introduce a dynamic frame-level priority data partition technique based on MPEG data structure and feedback from the network. To support these mechanisms, enhancements to the ATM adaptation layer 5 and a new MPEG-2 encapsulation strategy are also proposed. The presented quality of picture (QoP) control framework is evaluated using simulation and actual MPEG video data. The overall aim of the framework is double. First, ensuring a graceful picture quality degradation by minimizing cell loss probability for critical video data, and second optimizing the network effective throughput by reducing transmission of non useful data. In comparison to previous approaches, the performance evaluation have shown a significant reduction of the bad throughput and minimization of losses of intra- and predictive-coded frames at both cell and slice layers.