This letter proposes a new delivery format in order to realize unified transmissions of stereoscopic video contents over a dynamic adaptive streaming scheme. With the proposed delivery format, various forms of stereoscopic video contents regardless of their encoding and composition types can be delivered over the current dynamic adaptive streaming scheme. In addition, the proposed delivery format supports dynamic and efficient switching between 2D and 3D sequences in an interoperable manner for both 2D and 3D digital devices, regardless of their capabilities. This letter describes the designed delivery format and shows dynamic interoperable applications for 2D and 3D mixed contents with the implemented system in order to verify its features and efficiency.

The mining problem over data streams has recently been attracting considerable attention thanks to the usefulness of data mining in various application fields of information science, and sequence data streams are so common in daily life. Therefore, a study on mining sequential patterns over sequence data streams can give valuable results for wide use in various application fields. This paper proposes a new framework for mining novel interesting sequential patterns over a sequence data stream and a mining method based on the framework. Assuming that a sequence with small time-intervals between its data elements is more valuable than others with large time-intervals, the novel interesting sequential pattern is defined and found by analyzing the time-intervals of data elements in a sequence as well as their orders. The proposed framework is capable of obtaining more interesting sequential patterns over sequence data streams whose data elements are highly correlated in terms of generation time.

In recent years, real-time streaming has become widespread as a major service on the Internet. However, real-time streaming has a strict playback deadline. Application level multicasts using multiple distribution trees, which are known as forests, are an effective approach for reducing delay and jitter. However, the failure or departure of nodes during forest-based multicast transfer can severely affect the performance of other nodes. Thus, the multimedia data quality is degraded until the distribution trees are repaired. This means that increasing the speed of recovery from isolation is very important, especially in real-time streaming services. In this paper, we propose three methods for resolving this problem. The first method is a random-based proactive method that achieves rapid recovery from isolation and gives efficient “Randomized Forwarding” via cooperation among distribution trees. Each node forwards the data it receives to child nodes in its tree, and then, the node randomly transferring it to other trees with a predetermined probability. The second method is a reactive method, which provides a reliable isolation recovery method with low overheads. In this method, an isolated node requests “Continuous Forwarding” from other nodes if it detects a problem with a parent node. Forwarding to the nearest nodes in the IP network ensures that this method is efficient. The third method is a hybrid method that combines these two methods to achieve further performance improvements. We evaluated the performances of these proposed methods using computer simulations. The simulation results demonstrated that our proposed methods delivered isolation recovery and that the hybrid method was the most suitable for real-time streaming.

To establish seamless and highly robust content distribution, we proposed the new concept of Inter-Stream Forward Error Correction (FEC), an efficient data recovery method leveraging several video streams. Our previous research showed that Inter-Stream FEC had significant recovery capability compared with the conventional FEC method under ideal modeling conditions and assumptions. In this paper, we design the Inter-Stream FEC architecture in detail with a view to practical application. The functional requirements for practical feasibility are investigated, such as simplicity and flexibility. Further, the investigation clarifies a challenging problem: the increase in processing delay created by the asynchronous arrival of packets. To solve this problem, we propose a pragmatic parity stream construction method. We implement and evaluate experimentally a prototype system with Inter-Stream FEC. The results demonstrate that the proposed system could achieve high recovery performance in our experimental environment.

As wireless LAN systems become more widespread, the number of access points (APs) is increasing. A large number of APs cause overlapping cells where nearby cells utilize the same frequency channel. In the overlapping cells, inter-cell interference (ICI) degrades the throughput. This paper proposes an interference-aware multi-cell beamforming (IMB) technique to reduce the throughput degradation in the overlapping cells. The IMB technique improves transmission performance better than conventional multi-cell beamforming based on a decentralized control scheme. The conventional technique mitigates ICI by nullifying all the interference signal space (ISS) by beamforming, but the signal spaces to the user terminal (UT) is also limited because the degree of freedom (DoF) at the AP is limited. On the other hand, the IMB technique increases the signal space to the UT because the DoF at the AP is increased by selecting the ISS by allowing a small amount of ICI. In addition, we introduce a method of selecting the ISS in a decentralized control scheme. In our work, we analyze the interference channel state information (CSI) and evaluate the transmission performance of the IMB technique by using a measured CSI in an actual indoor environment. As a result, we find that the IMB technique becomes more effective as the number of UT antennas in nearby cells increases.

BEAN is a newly proposed lightweight stream cipher adopting Fibonacci FCSRs. It is designed for very constrained environments and aims at providing a balance between security, efficiency and cost. A weakness in BEAN was first found by Å gren and Hell in 2011, resulting in a key recovery attack slightly better than brute force. In this paper, we present new correlations between state and keystream with large statistical advantage, leading to a much more efficient key recovery attack. The time and data complexities of this attack are 257.53 and 259.94, respectively. Moreover, two new output functions are provided as alternatives, which are more efficent than the function used in BEAN and are immune to all attacks proposed on the cipher. Also, suggestions for improving the FCSRs are given.

Cognitive beamforming exploiting spatial opportunity is an attractive technique for secondary users to coexist with primary users in cognitive radio environments. If perfect channel state information of the interfering link is available, interference from a secondary transmitter to a primary receiver can be perfectly pre-nulled by choosing the ideal transmit beam. In practice, however, there is channel estimation error due to noise and the time-varying channels. To minimize the residual interference due to those channel estimation errors, channel prediction based on auto regressive (AR) model is introduced in this paper. Further, to cope with extremely rapidly-varying channels, a cognitive transmit power control technique is proposed as well. By combining channel prediction and transmit power control in cognitive beamforming, the cognitive users can share the spectrum with the primary users with a limited interference level in time-varying channels.

In this letter, we first introduce a new generalized cyclotomic sequence of order two of length pq, then we calculate its linear complexity and minimal polynomial. Our results show that this sequence possesses both high linear complexity and optimal balance on 1 s and 0 s, which may be attractive for use in stream cipher cryptosystems.

This paper proposes a novel multichannel speech enhancement technique for reverberant rooms that is effective when noise sources are spatially stationary, such as a projector fan noise, an air-conditioner noise, and unwanted speech sources at the back of microphones. Speech enhancement performance of the conventional multichannel Wiener filter (MWF) degrades when the Signal-to-Noise Ratio (SNR) of the current microphone input signal changes from the noise-only period. Furthermore, the MWF structure is computationally inefficient, because the MWF updates the whole spatial beamformer periodically to track switching of the speakers (e.g. turn-taking). In contrast to the MWF, the proposed method reduces noise independently of the SNR. The proposed method has a novel two-stage structure, which reduces noise and distortion of the desired source signal in a cascade manner by using two different beamformers. The first beamformer focuses on noise reduction without any constraint on the desired source, which is insensitive to SNR variation. However, the output signal after the first beamformer is distorted. The second beamformer focuses on distortion reduction of the desired source signal. Theoretically, complete elimination of distortion is assured. Additionally, the proposed method has a computationally efficient structure optimized for spatially stationary noise reduction problems. The first beamformer is updated only when the speech enhancement system is initialized. Only the second beamformer is updated periodically to track switching of the active speaker. The experimental results indicate that the proposed method can reduce spatially stationary noise source signals effectively with less distortion of the desired source signal even in a reverberant conference room.

With the fast development of mobile communication technologies, mobile multimedia services like mobile Video on Demand (VOD) are becoming prevalent. However, VOD streaming requires dedicated bandwidth to satisfy Quality of Service (QoS), and the limited wireless bandwidth will become insufficient to support the increasing number of mobile VOD users. In order to solve the problem, this paper proposes a Call Admission Control (CAC) approach which can accept new users even when the system bandwidth is insufficient. Our approach also guarantees continuous playback for subscribers by taking into account the service end time and the delay bound of the users. Simulation results demonstrate that the proposed scheme can increase the number of concurrent users and reduce the connection blocking probability significantly without playback interruption.

Enocoro-128v2 is a lightweight stream cipher submitted to Cryptography Research and Evaluation Committees (CRYPTREC). In this paper, we first describe a side channel attack on Enocoro-128v2. We show that all secret key bytes of Enocoro-128v2 can be recovered by correlation power analysis, and it is shown by an experiment that around 6000 traces are needed to recover the secret key on SASEBO-GII (Side-channel Attack Standard Evaluation Board). We second propose a countermeasure with threshold implementation technique, which allows Enocoro-128v2 to be resistant against correlation power analysis as long as less than 105 traces are used.

This letter proposes a new mechanism that supports adaptive sending control using Real-Time Streaming Protocol (RTSP) and Transmission Control Protocol (TCP) for IPTV service over heterogeneous networks. The proposed mechanism is implemented on a mobile IPTV device and its performance is verified for providing seamless television watching in heterogeneous networks, even when in motion.

Magnetoencephalography (MEG) measures very weak neuromagnetic signals using SQUID sensors. Standard MEG analyses include averaged waveforms, isofield maps and equivalent current dipoles. Beamforming MEG analyses provide us with frequency-dependent spatiotemporal information about the cerebral oscillatory changes related to not only somatosensory processing but also language processing. Language dominance is able to be evaluated using laterality of power attenuation in the low γ band in the frontal area. Neuromagnetic signals of the unilateral upper movements are able to be decoded using a support vector machine.

This paper deals with two types of capacity allocation schemes, i.e., static and adaptive, for uplink and downlink burst durations in the IEEE 802.16 BE (Best Effort) service. We study QoE (Quality of Experience) enhancement of audio-video IP transmission over the uplink channel with the two capacity allocation schemes. We introduce a piggyback request mechanism for uplink bandwidth requests from subscriber stations to the base station in addition to a random access-based request mechanism. We assess QoE of audio-video streams for four schemes obtained from the combination of the capacity allocation schemes and the bandwidth request mechanisms. We also employ two types of audio-video contents. From the assessment result, we notice that the adaptive allocation scheme is effective for QoE enhancement particularly under heavily loaded conditions because of its efficient usage of OFDM symbols. In addition, the piggyback request mechanism can enhance QoE of audio-video transmission. We also find that the effects of capacity allocation schemes and piggyback request mechanism on QoE change according to the content types.

In this paper, we propose an improved seam merging method for content-aware image resizing. This method merges a two-pixel-width seam element into one new pixel in image reduction and inserts a new pixel between the two pixels in image enlargement. To preserve important contents and structure, our method uses energy terms associated with importance and structure. Our method preserve the main structures by using a cartoon version of the original image when calculating the structure energy. In addition, we introduce a new energy term to suppress the distortion generated by excessive reduction or enlargement in iterated merger or insertion. Experimental results demonstrate that the proposed method can produce satisfactory results in both image reduction and enlargement.

Three synchronization issues, i.e., phase, frequency, and symbol time, have to be properly controlled to achieve distributed beamforming gain. In orthogonal frequency division multiplexing (OFDM) systems, frequency offset in cooperating signals is more important than other synchronization issues since it results in SNR degradation as well as inter-carrier interference (ICI). In this paper, the impact of frequency offset in distributed beamforming is analyzed for OFDM systems. ICI resulting from frequency offset between cooperating signals is also investigated and approximated. Performance degradation due to frequency offset is shown with various numbers of cooperating signals and offset values. We show that frequency offset between cooperating signals is critical in OFDM systems since it leads to interference from the other subcarriers as well as power loss in the desired signal.

This paper proposes a dynamic bandwidth allocation algorithm that improves the network performance and bandwidth sharing efficiency in the upstream channels of a hybrid passive optical network (PON) that combines a fiber-to-the-home (FTTH) access network and wireless sensor networks (WSNs). The algorithm is called the adaptive limited dynamic bandwidth allocation (ALDBA) algorithm. Unlike existing algorithms, the ALDBA algorithm is not limited to controlling just FTTH access networks, it also supports WSNs. For the proposed algorithm, we investigate the difference in the lengths of generated data packets between the FTTH terminals and sensor nodes of WSN to effectively evaluate the end-to-end average packet delay, bandwidth utilization, time jitter, and upstream efficiency. Two variants of the proposed algorithm and a limited service (LS) scheme, which is an existing well-known algorithm, are compared under non-uniform traffic conditions without taking into consideration priority scheduling. We demonstrate the proposed scheme through simulation by generating a realistic network traffic model, called self-similar network traffic. We conducted a detailed simulation using several performance parameters to validate the effectiveness of the proposed scheme. The results of the simulation showed that both ALDBA variants outperformed the existing LS scheme in terms of average packet delay, bandwidth utilization, jitter, and upstream efficiency for both low and high traffic loads.

A leakage-aware Coordinated Scheduling/Coordinated Beamforming (CS/CB) scheme for heterogeneous networks with layered limited feedback is proposed. In particular, all pico cells cooperatively select an optimal beamforming vector for the macro cell within a CoMP cluster so as to minimizing leakage power from the macro cell. Simulations show that the proposed scheme outperforms the conventional non-CoMP scheme with perfect channel state information at teansmitter (CSIT). Furthermore, the feedback amount and scheduler complexity is decreased greatly.

In this paper, we propose an adaptive chunk scheduling for mesh-based peer-to-peer live streaming system, a hybrid class of push and pull chunk delivery approach. The proposed rule-based push-pull scheduler simultaneously pull video chunk from lower latency peers to fill up missing chunks and push video chunk adaptively for rapid chunk delivery. We performed comparative simulation study against rarest first push-pull and status-wise push-pull to prove the efficiency of our proposed algorithm. Mesh-push is made possible by effectively exploiting the information through buffer map exchange. The findings of performance evaluation have suggested a better video continuity and achieved lower source to end delay.

A novel design is proposed to implement highly parallel stream join operators on a field-programmable gate array (FPGA), by examining handshake join algorithm for hardware implementation. The proposed design is evaluated in terms of the hardware resource usage, the maximum clock frequency, and the performance. Experimental results indicate that the proposed implementation can handle considerably high input rates, especially at low match rates. Results of simulation conducted to optimize size of buffers included in join and merge units give a new intuition regarding static and adaptive buffer tuning in handshake join.