Modulation spectrogram is effective for analyzing underwater signals which consist of tonal and modulated components. This method can analyze the acoustic and modulation frequency at the same time, but has the trade-off issue of time-frequency localization. This letter introduces a reassignment method for overcoming the localization issue in conventional spectrograms, and then presents an alignment scheme for implementing modulation spectrogram. Relevant experiments show improvement in acoustic frequency estimation perspective and an increment in analyzable modulation frequency range.

This paper presents a new asynchronous FIFO design to reduce forward latency in a linear structure. The operation mode for each cell can be reconfigured dynamically as either of the two schemes, wave pipelining or handshaking, according to the data flow in the FIFO. The adoption of wave pipelining to the conventional self-timed FIFO can reduce the overhead of the handshaking as well as latching control in each stage. Initial pre-layout simulations indicate about two times of improvement on latency performance over a state-of-art asynchronous FIFO, while retaining its throughput.

In this work, a V-band low noise amplifier (LNA) is developed in a commercial 0.13 µm RFCMOS technology. Common-source (CS) topology, known to show a better noise performance than the cascode topology, was adopted and 4-stage was employed to achieve a sufficient gain at the target frequency near the cutoff frequency fT. The measured gain was 18.6 dB with VDD = 1.2 V and increased up to 20.2 dB with VDD = 1.8 V at 66 GHz. The measured NF showed a minimum value of 7.0 dB at 62 GHz. DC power consumption was 24 mW with VDD = 1.2 V. The size of the fabricated circuit is as compact as 0.45 mm 0.69 mm. This work was further extended to investigate the effect of dummy fills on LNA performance. An identical LNA, except for the dummy fills formed very close to (and under) the metal lines of spiral inductors and interconnects, was also fabricated and compared with the standard LNA. A peak gain degradation of 3.6 dB and average NF degradation of 1.3 dB were observed, which can be ascribed to the increased mismatch and line loss due to the dummy fills.

In this paper, a multi-coded variable spreading gain (MC-VSG) CDMA system employing a binary transmission of MC signals by introducing a level clipper, termed MC-VSG BNet system, is proposed for a possible candidate of wireless personal area network (WPAN) and 3 G cellular applications. With an emphasis on the MC-VSG BNet physical layer and the system performance, we address the concise specification of the MC-VSG BNet system including the spreading code, level clipping, modulation, coding, and frame format. Especially, we focus on the level clipping of multi-level MC signals for both power- and cost-efficient implementation and the VSG code design fir high-rate transmissions. From the receiver performance based on simulation results, in addition to simple receiver structure, an acceptable performance degradation of the MC-VSG BNet system over the existing DS/CDMA system is observed, while guaranteeing an high bit rate transmission.

In disk-based storage systems, non-volatile write caches have been widely used to reduce write latency as well as to ensure data consistency at the level of a storage controller. Write cache policies should basically consider which data is important to cache and evict, and they should also take into account the real I/O features of a non-volatile device. However, existing work has mainly focused on improving basic cache operations, but has not considered the I/O cost of a non-volatile device properly. In this paper, we propose a pattern-aware write cache policy, PAW for a NAND flash memory in disk-based mobile storage systems. PAW is designed to face a mix of a number of sequential accesses and fewer non-sequential ones in mobile storage systems by redirecting the latter to a NAND flash memory and the former to a disk. In addition, PAW employs the synergistic effect of combining a pattern-aware write cache policy and an I/O clustering-based queuing method to strengthen the sequentiality with the aim of reducing the overall system I/O latency. For evaluations, we have built a practical hard disk simulator with a non-volatile cache of a NAND flash memory. Experimental results show that our policy significantly improves the overall I/O performance by reducing the overhead from a non-volatile cache considerably over a traditional one, achieving a high efficiency in energy consumption.

In this paper, an area-efficient decoder architecture is proposed for the quasi-cyclic low-density parity check (QC-LDPC) codes specified in the IEEE 802.16e WiMAX standard. The decoder supports all the code rates and codeword lengths defined in the standard. In order to achieve low area and maximize hardware utilization, the decoder utilizes 4 decoding function units, which is the greatest common divisor of the expansion factors. In addition, the decoder adopts a novel scheduling scheme named stride scheduling, which stores the extrinsic messages in non-sequential order to replace the conventional complex flexible permutation network with simple small-sized cyclic shifters and also minimize the number of memory accesses. To further minimize the complexity, the number of extrinsic memory instances for 24 block columns is reduced to 5 banks by identifying independent sets. All the memory instances used in the decoder are single-port memories which cost less area and price compared to dual-port ones. Finally, the decoding function units have partially parallel structure to make the decoding throughput sufficiently over the requirement of the WiMAX standard. The proposed decoder is synthesized with 49 K equivalent gates and 54,144 bits of memory, and the implementation occupies 0.40 mm2 in a 65 nm CMOS technology.

In this paper, an effective index structure for dynamic main memory database systems, which we call the T2-tree, is presented. A notion of a thread pointer is introduced to overcome some of the limitations of the T-tree and the T*-tree. There are several advantages to this structure. First, the T2-tree reduces the number of rotate operations and the overhead required for balancing the tree by restraining new node creation and deletion. Second, the T2-tree shows good performance for sequential search of range queries as these requests can be effectively handled using the successor pointer. Finally, the T2-tree allows for higher space utilization amplicating the aforementioned benefits. These advantages are obtained with minimal changes to the existing T-tree structure. Experimental studies showing evidence of the benefits of the T2-tree are also presented.

In this study, a computationally efficient ranging scheme exploiting a minimum mean square error (MMSE) and a matrix-pencil (MP) technique is proposed for the IEEE 802.15.4a chirp spread spectrum (CSS) system. Based on the characteristics of the CSS signal, a practical methodology for the MMSE process is devised and the appropriate values of parameters, which are cutoff bandwidth, number of samples and sampling duration in frequency domain, are investigated and numerically determined to enhance the performance. The performance of proposed scheme is analyzed in terms of the computational complexity and the ranging estimation errors. Simulation results demonstrate that the proposed scheme performs as well as the conventional scheme at remarkably reduced computational costs.

A non-deterministic finite automaton (NFA)-based parallel string matching scheme is proposed. To parallelize the operations of NFAs, a graphic processing unit (GPU) is adopted. Considering the resource occupancy of threads and size of the shared memory, the optimized resource allocation is performed in the proposed string matching scheme. Therefore, the performance is enhanced significantly in all evaluations.

A chirp spread spectrum (CSS) system uses a chirp signal which changes the instantaneous frequency according to time for spreading a transmission bandwidth. In the CSS system, the transmission performance can be simply improved by increasing the time-bandwidth product which is known as the processing gain. However, increasing the transmission bandwidth is limited because of the spectrum regulation. In this letter, we propose a correlation-based chirp rate allocation method to improve the transmission performance by analyzing the cross-correlation coefficient in the same time-bandwidth product. In order to analyze the transmission performance of the proposed method, we analytically derive the cross-correlation coefficient according to the time-bandwidth separation product and simulate the transmission performance. The simulation results show that the proposed method can analytically allocate the optimal chirp rate and improve the transmission performance.

In this letter, we propose a Hadoop-based Video Transcoding System (HVTS), which is designed to run on all major cloud computing services. HVTS is highly adapted to the structure and policies of Hadoop, thus it has additional capacities for transcoding, task distribution, load balancing, and content replication and distribution. To evaluate, our proposed system, we carry out two performance tests on our local testbed, transcoding and robustness to data node and task failures. The results confirmed that our system delivers satisfactory performance in facilitating seamless streaming services in cloud computing environments.

This paper suggests utterance verification system using state-level log-likelihood ratio with frame and state selection. We use hidden Markov models for speech recognition and utterance verification as acoustic models and anti-phone models. The hidden Markov models have three states and each state represents different characteristics of a phone. Thus we propose an algorithm to compute state-level log-likelihood ratio and give weights on states for obtaining more reliable confidence measure of recognized phones. Additionally, we propose a frame selection algorithm to compute confidence measure on frames including proper speech in the input speech. In general, phone segmentation information obtained from speaker-independent speech recognition system is not accurate because triphone-based acoustic models are difficult to effectively train for covering diverse pronunciation and coarticulation effect. So, it is more difficult to find the right matched states when obtaining state segmentation information. A state selection algorithm is suggested for finding valid states. The proposed method using state-level log-likelihood ratio with frame and state selection shows that the relative reduction in equal error rate is 18.1% compared to the baseline system using simple phone-level log-likelihood ratios.

A downlink transmit diversity scheme is proposed for [1] WCDMA low chip rate (1.28 Mcps) Time Division Duplex (LCR-TDD) systems. In the proposed Time Switched Transmit Diversity (TSTD) scheme, two spatially separated antennas are alternately used at a base station in order to transmit consecutive sub-frames on the downlink physical channels. The proposed TSTD scheme takes advantage of the frame structure of the WCDMA LCR-TDD, where TSTD transmission is possible with a single power amplifier. Simply adding switching components in front of the transmit antennas yields a considerable gain compared with a non-diversity mode system. A new power control algorithm is also proposed for generating power control command for efficient use of TSTD. A performance gain of up to 5 dB, compared with a non-diversity scheme, is achieved in terms of the frame error rate for varying Eb/No and mobile speeds. From the viewpoint of both complexity and performance, the TSTD scheme is a simple and effective transmit diversity method that can increase downlink capacity in the WCDMA LCR-TDD systems.

This paper proposes a hardware-based parallel pattern matching engine using a memory-based bit-split string matcher architecture. The proposed bit-split string matcher separates the transition table from the state table, so that state transitions towards the initial state are not stored. Therefore, total memory requirements can be minimized.

In this paper, an efficient method to reduce computational complexity for pedestrian detection is presented. Since trilinear interpolation is not used, the amount of required operations for histogram of oriented gradient (HOG) feature calculation is significantly reduced. By calculating multi-scale HOG features with integral HOG in a two-stage approach, both high detection rate and speed are achieved in the proposed method.

With the aim of improving speaker identification, we propose a likelihood-based integration method to combine the speaker identification results obtained through multiple microphones. In many cases, the composite result has lower error rate than that by any single channel. The proposed integration method can achieve more reliable identification performance in the ubiquitous robot companion (URC) environment in which the robot is connected to a server through an extremely high broadband penetration rate.

In this letter, we propose a novel resource allocation scheme to enhance downlink system performance for orthogonal frequency division multiple access (OFDMA) and time division duplex (TDD) based femtocell networks. In the proposed scheme, the macro base station (mBS) and femto base stations (fBSs) service macro user equipments (mUEs) and femto user equipments (fUEs) in inner and outer zones in different periods to reduce interference substantially. Simulations show the proposed scheme outperforms femtocell networks with fractional frequency reuse (FFR) systems in terms of the system capacity and outage probability for mUEs and fUEs.

In this paper we propose a mobile positioning method based on a recursive least squares (RLS) algorithm for suppressing the non-line of sight (NLOS) effects in cellular systems. The proposed method finds the position of a mobile station from TOAs measured by three BSs. Simulation results show that the proposed method has a fast convergence time and greatly reduces the positioning error especially in NLOS situations. Thus it is expected that the proposed method can be effectively used in a dense urban environment.

In this letter, a new segment-level speech/nonspeech classification method based on the Poisson polling technique is proposed. The proposed method makes two modifications from the baseline Poisson polling method to further improve the classification accuracy. One of them is to employ Poisson mixture models to more accurately represent various segmental patterns of the observed frequencies for frame-level input features. The other is the soft counting-based frequency estimation to improve the reliability of the observed frequencies. The effectiveness of the proposed method is confirmed by the experimental results showing the maximum error reduction of 39% compared to the segmentally accumulated log-likelihood ratio-based method.

Backlight dimming techniques have been researched much to obtain high power saving on display modules, especially those which are based on LCD. The use of LED as a light source in a backlight module has opened a wider chance to perform local dimming as an improvement of a conservative global dimming approach. However, local dimming techniques are sometimes observed to obtain worse performance than global dimming ones in terms of power saving or image fidelity. We observed that even some of their results show visible artifacts. In this paper, we propose a novel backlight dimming technique called hybrid dimming, which combines local and global dimming approaches effectively. We do local dimming to obtain the initial backlight levels while calculating its SSIM index, which is a human visual system-aware image quality metric. We then make sure that these backlight levels don't exceed the ones obtained from a human visual system-aware global dimming with similar image fidelity. As a result, our proposed method can gain better power saving than a human visual system-aware global dimming and prior local dimming techniques, while making little difference in the image fidelity and suppressing visible block artifacts in the results. Experimental results showed that the proposed technique can achieve up to 14, 2.2, and 2.4 times higher power saving ratio than human visual system-aware global dimming and two well-designed local dimming techniques, respectively.