We design M(≥3)-phase spreading sequences of Markov chains optimal in terms of bit error probabilities in asynchronous SSMA (spread spectrum multiple access) communication systems. To this end, we obtain the distributions of the normalized MAI (multiple access interference) for such systems and find a necessary and sufficient condition that the distributions become independent of the phase shifts.

In digital transmission and storage systems, sequences must have attributes that comply with the physical characteristics of the channel. These channel constraints can often be satisfied through constrained sequence coding techniques which avoid use of sequences that violate the given channel constraints. In the design of a constrained code, it is usually helpful to consider the PSD of the encoded sequence in order to ensure that PSD requirements are met, and to obtain an indication of bandwidth, response at dc, average power peaks, and other spectral characteristics of interest. In this paper, we introduce an approach for the construction of finite-state sequential machine (FSSM) modeled encoders to satisfy spectral requirements. This approach involves construction of either a Mealy or a Moore FSSM to represent the encoder, and evaluation of the state transition probabilities and codeword values such that the PSD of the designed code meets a predefined spectral shape. Examples in this paper demonstrate the usefulness of this approach.

We analytically prove that the error in the channel idle time-based collision probability estimation in face of non-saturated stations is bounded by 2/(CWmin+1) in the IEEE 802.11 wireless LANs (WLANs). This work explicitly quantifies the impact of non-saturation, and the result vindicates the use of the estimation technique in real-life IEEE 802.11 WLANs, in such applications as the acknowledgement-based link adaptation and the throughput optimization through contention window size adaptation.

Soft-decision decoding techniques are applied to asynchronous frequency-hop/spread-spectrum multiple-access (FH/SSMA) networks, where M-ary frequency shift keying (MFSK) is employed to transmit one modulated symbol per hop. Coding schemes using soft-decision decoded binary convolutional codes or turbo codes are considered, both with or without bit-interleaving. Performances of several soft metrics are examined for each coding scheme. It is shown that when multiple access interference is the main source of errors, the product metric offers the best performance among the soft metrics considered for all coding schemes. Furthermore, the application of soft-decision decoded convolutional codes or turbo codes without bit-interleaving is shown to allow for a much larger number of simultaneously transmitting users than hard-decision decoded Reed-Solomon codes. Finally, it is observed that when soft-decision decoding techniques are employed, synchronous networks attain better performance than asynchronous networks.

Interactive audio-video applications over IP networks have subjective tradeoffs between fidelity and latency owing to packet buffering at the receiver. Increasing the buffering time improves the fidelity, whereas it degrades the latency. This paper makes the subjective tradeoff between fidelity and latency clear in a quantitative way. In addition, we examine the effect of tasks on the subjective tradeoff. In evaluating the effect of tasks, we use two tasks according to ITU-T Recommendation P.920. An experiment was conducted to measure user-level QoS of an interactive application with the psychometric methods. We then investigate the subjective tradeoff quantitatively by QoS mapping. The experimental results confirm that there exists the buffering time which makes user-level QoS the highest. The results also show that the optimum buffering time depends on the kind of task.

In this letter, we propose a limited feedback precoding scheme based upon grassmannian beamforming and user selection for downlink multiuser MIMO systems. Conventional random beamforming scheme only enjoys significant performance gains with a large number of users, which limits its practical application. With proper codebook size the proposed scheme outperforms conventional random beamforming scheme when the number of users is small or moderate.

The Video Quality Experts Group (VQEG) is preparing a programme of subjective multimedia quality tests. The results from these tests will be used to evaluate the performance of competing objective multimedia quality metrics. The reliability of the subjective test data is of great importance for VQEG's task. This paper provides an overview of VQEG's multimedia ad-hoc group. The work of this group will require subjective tests to be performed by laboratories located in Europe, Asia and North America. For VQEG's multimedia work to be successful, the subjective assessment methodology must be precisely defined and produce reliable and repeatable subjective quality data. Although international standards covering multimedia quality assessment methods are in force, there remains some uncertainty regarding the most effective approach to assessing the subjective quality of multimedia. A review of existing methods is provided. Two experiments are presented investigating the suitability of alternative subjective assessment methods (single-stimulus ACR and SAMVIQ). The results of these experiments are discussed within the context of the VQEG multimedia testing programme.

This paper provides an overview of the new tendencies in the subjective assessment of the quality of video for Multimedia applications. New subjective assessment methods are here described together with the description of the new general approaches. Some motivations of these new approaches are also here provided.

We extend the sliding block code in symbolic dynamics to transform J (≥2) sequences of Markov chains with time delays. Under the assumption that the chains are irreducible and aperiodic, we prove the central limit theorem (CLT) for the normalized sums of extended sliding block codes from J sequences of Markov chains. We apply the theorem to the system analysis of asynchronous spread spectrum multiple access (SSMA) communication systems using spreading sequences of Markov chains. We find that the standard Gaussian approximation (SGA) for estimations of bit error probabilities in such systems is the 0-th order approximation of the evaluation based on the CLT. We also provide a simple theoretical evaluation of bit error probabilities in such systems, which agrees properly with the experimental results even for the systems with small number of users and low length of spreading sequences.

This paper describes an objective measurement method designed to assess the perceived quality for digital videos. The proposed approach can be used either in the context of a reduced reference quality assessment or in the more challenging situation where no reference is available. In that way, it can be deployed in a QoS monitoring strategy in order to control the end-user perceived quality. The originality of the approach relies on the very limited computation resources which are involved, such a system could be integrated quite easily in a real time application. It uses a convolutional neural network (CNN) that allows a continuous time scoring of the video. Experiments conducted on different MPEG-2 videos, with bit rates ranging from 2 to 6 Mbits/s, show the effectiveness of the proposed approach. More specifically, a linear correlation criterion, between objective and subjective scoring, ranging from 0.90 up to 0.95 has been obtained on a set of typical TV videos in the case of a reduced reference assessment. Without any reference to the original video, the correlation criteria remains quite satisfying since it still lies between 0.85 and 0.90, which is quite high with respect to the difficulty of the task, and equivalent and more in some cases than the traditional PSNR, which is a full reference measurement.

Next generation codecs, benchmarked by the H.264/AVC standard, are providing substantial compression efficiency for the coding and transmission of video. Coupled with technologies offering larger transmission bandwidths over DSL, wireless and satellite networks, the capability of delivering high quality video services to the home is now a reality. The perceptual quality of the content delivered over communications networks will be crucial in ensuring a first-class customer experience. It is therefore important to assess the advantages and disadvantages of the optional features offered by next generation codecs. This paper describes a subjective assessment that was carried out to investigate the perceptual effects of switching the in loop de-blocking filter within the H.264/ AVC CODEC on or off. Although the filter is believed to substantially improve the perceptual quality of video, it has been suggested that in some cases negative perceptual effects can be produced. The H.264/AVC architecture allows de-blocking to be switched off in cases where there are limited processing resources or it is considered a negative perceptual effect may be introduced. This paper describes a study that examined the perceptual effects of de-blocking by employing a standardised subjective assessment methodology. The Absolute Category Rating (ACR) method was used to capture Difference Mean Opinion Scores (DMOS) for a range of video. Content was selected to span a wide and representative range of coding complexity. This content was then encoded at a variety of bit-rates to represent high, medium and low qualities. Results were used to examine the end-user perception of video quality when the de-blocking filter is switched on or off. The experimental design allowed the overall effects of the de-blocking filter to be examined and additionally the relationship between content and quality on the filter performance. The experiment found that the performance of the de-blocking filter was content-dependent. Results were used to discuss the advantages and disadvantages of in-loop de-blocking and there is an examination of content properties (e.g. spatial and temporal complexity) that influence the performance of de-blocking.

We propose the concept of an opinion model for interactive multimodal services and apply it to an audiovisual communication service. First, psychological factors of an audiovisual communication service were extracted by using the semantic differential (SD) technique and factor analysis. Forty subjects participated in subjective tests and performed point-to-point conversational tasks on a PC-based video phone that exhibited various network qualities. The subjects assessed those qualities on the basis of 25 pairs of adjectives. Two psychological factors, i.e., an aesthetic feeling and a feeling of activity, were extracted from the results. Then, quality impairment factors affecting these two psychological factors were analyzed. We found that the aesthetic feeling was affected by IP packet loss and video coding bit rate, and the feeling of activity depended on delay time, video packet loss, video coding bit rate, and video frame rate. Using this result, we formulated an opinion model derived from the relationships among quality impairment factors, psychological factors, and overall quality. The validation test results indicated that the estimation error of our model was almost equivalent to the statistical reliability of the subjective score.

This paper proposes a subjective model for estimating the quality of video streaming services with dynamic bit-rate control. In a subjective quality assessment test, we clarify users' perceptions of distributed video signals whose quality is time-variant due to dynamic bit-rate control. Using this result, we constructed an estimation model considering the following three characteristics: 1) the influence of the video section where quality degradation is large will strongly affect the overall quality, 2) the impression of a past quality weakens with the passage of time, and 3) the range of evaluation scores becomes wider when the time duration of an evaluation is longer. We found that the proposed model enables the accuracy of estimating overall subjective quality to be dramatically improved compared with that of a model that averages segmental quality. The estimation error of the proposed model is less than the statistical reliability of the subjective score even for verification data. We also show that our findings are applicable to QoS design/management issues for video streaming services with dynamic bit-rate control.

We study asynchronous SSMA communication systems using binary spreading sequences of Markov chains and prove the CLT (central limit theorem) for the empirical distribution of the normalized MAI (multiple-access interference). We also prove that the distribution of the normalized MAI for asynchronous systems can never be Gaussian if chains are irreducible and aperiodic. Based on these results, we propose novel theoretical evaluations of bit error probabilities in such systems based on the CLT and compare these and conventional theoretical estimations based on the SGA (standard Gaussian approximation) with experimental results. Consequently we confirm that the proposed theoretical evaluations based on the CLT agree with the experimental results better than the theoretical evaluations based on the SGA. Accordingly, using the theoretical evaluations based on the CLT, we give the optimum spreading sequences of Markov chains in terms of bit error probabilities.

The frequent and time-independent packet loss due to signal noise in the wireless network has been a major obstacle in providing a persistent end-to-end bandwidth in the integrated environment of wired and wireless networks. One approach to cope with this is to develop an efficient transport mechanism which appropriately adapts to the dynamics of the wireless part of the network so that the end-to-end throughput is maximised. The success of new transport mechanisms depends on the quality of information obtained, in particular, from the wireless network. This paper presents a novel mechanism to assess accurately the transmission quality of the wireless part of the integrated CDMA2000 1X networks using the NAK rate obtained from the underlying RLP protocol stack. The experiment results show that the proposed mechanism correctly measures the wireless transmission quality of the CDMA2000 1X network.

Recently, wideband speech communication using 7 kHz-wideband speech coding, as described in ITU-T Recommendations G.722, G.722.1, and G.722.2, has become increasingly necessary for use in advanced IP telephony using PCs, since, for this application, hands-free communication using separate microphones and loudspeakers is indispensable, and in this situation wideband speech is particularly helpful in enhancing the naturalness of communication. An objective quality measurement methodology for wideband-speech coding has been studied, its essential components being an objective quality measure and an input test signal. This paper describes Wideband-PESQ conforming to the draft Annex to ITU-T Recommendation P.862, "Perceptual Evaluation of Speech Quality (PESQ)," as the objective quality measure, by evaluating the consistency between the subjectively evaluated MOS (Mean Opinion Score) and objectively estimated MOS. This paper also describes the verification of artificial voice conforming to Recommendation P.50 "Artificial Voices," as the input test signal for such measurements, by evaluating the consistency between the objectively estimated MOS using a real voice and that obtained using an artificial voice.

Source-Specific Multicast (SSM) is recognized as the most feasible multicast communication model for a wide use throughout the Internet because it eliminates many complexities associated with traditional many-to-many multicast communication model. Regarding the SSM communication, however, implementations of the Internet Group Multicast Protocol version 3 (IGMPv3) for IPv4 and the Multicast Listener Discovery version 2 (MLDv2) for IPv6 are indispensable on every end-host. Accurate implementations of these protocols are necessary studies for the smooth deployment. In this document, we illustrate IGMPv3 and MLDv2 host-side implementations applied to the 4.4BSD kernel. While presenting the concept and the design of these implementations, we verify the behavior and evaluate the response times for (*,G) and (S,G) joins requested on top of both implementations.

Due to the cost of multicast state management, multicast address allocation, inter-domain multicast routing of traditional IP multicast scheme, ASM leads to a search for other multicast schemes. This paper presents a new solution to the problems mentioned above based on IPv6. The proposed scheme provides an enhanced scheme supporting the strengths of SSM in basic Xcast. This is achieved by adding MLDv2 operations at recipient's side and a new control plane into existing Xcast. The proposed scheme does not only provide the transparency of traditional multicast schemes to sources and recipients, but it also enhances the routing efficiency in networks. Intermediate routers do not have to maintain multicast state, so that it results in a more efficient and scalable mechanism to deliver native multicast datagrams. Also, the seamless integration in Mobile IPv6 can support multicast efficiently for mobile nodes in IPv6 networks by avoiding tunnel avalanches and tunnel convergence. We've attempted to prove this alternative architecture by both simulation and implementation, respectively. Our approach cannot fundamentally perform for many large groups distributed widely as effectively as traditional multicast schemes. However, we believe that the resulting scheme is simple, efficient, robust, transparent, and to the extent possible, scalable in case that recipients are clustered in subnets.

In this paper, the CSK/SSMA ALOHA system with nonorthogonal sequences which combines the ALOHA system with Code Shift Keying (CSK) using nonorthogonal sequences is proposed. The throughput performance was evaluated by theoretical analysis. Moreover, the throughput performance of the system is compared with those of the DS/SSMA ALOHA and M-ary/SSMA ALOHA systems. It is found that the throughput performance of our system to be better than those of the other two systems.

In this paper, we propose a partial erasure decoding scheme with erasure-decision threshold for Reed-Solomon (RS) codes and analyze its performance in frequency-hopped multiple-access communications. RS code is used to correct erasures and errors caused by other-user interference. Binary FSK is employed to transmit the channel symbol. The proposed decoder decides whether to erase the received RS code symbol based on the ersure decsion threshold. The approximated formula for optimal erasure decision threshold is derived in such a way that packet error probability can be minimized. Numerical results show that the employment of adaptive erasure decision threshold attains the higher normalized throughput in the areas of high channel traffic cases.