In this paper, new design of optimal frequency hopping sequences (FHSs) with low hit zone (LHZ) with respect to the Peng-Fan-Lee bound is presented based on interleaving techniques. By the new design, new classes of optimal LHZ FHS sets with large family size are obtained. It is shown that all the sequences in the proposed FHS sets are shift distinct. The proposed FHS sets are suitable for quasi-synchronous time/frequency hopping code division multiple access systems to eliminate multiple-access interference.

In this paper, a method is proposed for reconstruction of the parameters of a non-binary block encoder using an intercepted sequence of noisy coded data. The proposed method is a generalization of the Barbier's method for the reconstruction of binary block codes to the more problematic case of non-binary codes. It has been shown mathematically that considering some revisions in definitions, such a generalization is possible. The proposed method is able to estimate the code parameters such as the code length, the code dimension, number of bits per symbol, and the dual-code subspace, and also to synchronize the sequence. Since the Reed-Solomon code is the most important type of non-binary block codes, an additional method is proposed to reconstruct the generator polynomial in the case of Reed-Solomon codes. The proposed method is evaluated via computer simulations which verify its strength and effectiveness.

In this paper we study the ploblem whether the language D(1) of all d-primitive words can be generated by a contextual grammar. It is proved that D(1) can be generated neither by an external contextual grammar nor by an internal contextual grammar, and that it can be generated by a total contextual grammar with choice.

In this paper, a block-constrained trellis coded vector quantization (BC-TCVQ) algorithm is combined with an algebraic codebook to produce an algebraic trellis vector code (ATVC) to be used in ACELP coding. ATVC expands the set of allowed algebraic codebook pulse position, and the trellis branches are labeled with these subsets. The Viterbi algorithm is used to select the excitation codevector. A fast codebook search method using an efficient non-exhaustive search technique is also proposed to reduce the complexity of the ATVC search procedure while maintaining the quality of the reconstructed speech. The ATVC block code is used as the fixed codebook of AMR-NB (12.2 kbps), which reduces the computational complexity compared to the conventional algebraic codebook.

In this paper, a capacitive-cross-coupling common-gate (CCC-CG) LNA using capacitive feedback is proposed to improve the noise figure (NF). In the conventional CCC-CG LNA, the transconductance gm is determined by the input-matching condition while a lager gm is required to improve NF. gm of the proposed LNA can be increased and NF can be improved by using the added capacitive feedback. The analytical calculation shows that the proposed LNA can perform better than the conventional CCC-CG LNA. In the measurement results using a 0.18-µm CMOS technology, the gain is 10.4–13.4 dB, NF is 2.7–2.9 dB at 0.8–1.8 GHz, and IIP3 is -7 dBm at 0.8 GHz. The power consumption is 6.5 mW with a 1.8-V supply.

In this letter, a Simple but Effective Congestion Control scheme (SECC) in VANET has been proposed to guarantee the successful transmissions for safety-related nodes. The strategy derive a Maximum Beacon Load Activity Indicator (MBLAI) to restrain the neighboring general periodical beacon load for the investigated safety-related “observation nodes”, i.e., the nodes associated with some emergent events. This mechanism actually reserves some bandwidth for the safety-related nodes to make them have higher priorities than periodical beacons to access channel. Different from the static congestion control scheme in IEEE802.11p, this strategy could provide dynamic control strength for congestion according to tolerant packets drop ratio for different applications.

A novel online speaker clustering method based on a generative model is proposed. It employs an incremental variant of variational Bayesian learning and provides probabilistic (non-deterministic) decisions for each input utterance, on the basis of the history of preceding utterances. It can be expected to be robust against errors in cluster estimation and the classification of utterances, and hence to be applicable to many real-time applications. Experimental results show that it produces 50% fewer classification errors than does a conventional online method. They also show that it is possible to reduce the number of speech recognition errors by combining the method with unsupervised speaker adaptation.

In this paper, we propose a novel voice activity detection (VAD) algorithm using global speech absence probability (GSAP) based on Teager energy (TE) for speech enhancement. The proposed method provides a better representation of GSAP, resulting in improved decision performance for speech and noise segments by the use of a TE operator which is employed to suppress the influence of noise signals. The performance of our approach is evaluated by objective tests under various environments, and it is found that the suggested method yields better results than conventional schemes.

In this note, we introduce the integral term of system outputs into an output feedback controller for sampled-data linear systems with unknown disturbances. The proposed method does not use any observer and can prevent the high gain actions in control inputs. Provided the variation of the disturbance in the two consecutive sampling instances is not changed significantly, it is shown that system states and system outputs are finally constrained in small bounded regions, respectively. Simulation results support the theoretical developments.

A novel speech feature generation-based acoustic model training method for robust speaker-independent speech recognition is proposed. For decades, speaker adaptation methods have been widely used. All of these adaptation methods need adaptation data. However, our proposed method aims to create speaker-independent acoustic models that cover not only known but also unknown speakers. We achieve this by adopting inverse maximum likelihood linear regression (MLLR) transformation-based feature generation, and then we train our models using these features. First we obtain MLLR transformation matrices from a limited number of existing speakers. Then we extract the bases of the MLLR transformation matrices using PCA. The distribution of the weight parameters to express the transformation matrices for the existing speakers are estimated. Next, we construct pseudo-speaker transformations by sampling the weight parameters from the distribution, and apply the transformation to the normalized features of the existing speaker to generate the features of the pseudo-speakers. Finally, using these features, we train the acoustic models. Evaluation results show that the acoustic models trained using our proposed method are robust for unknown speakers.

High-performance FFT processor is indispensable for real-time OFDM communication systems. This paper presents a CORDIC based design of variable-length FFT processor which can perform various FFT lengths of 64/128/256/512/1024/2048/4096/8192-point. The proposed FFT processor employs memory based architecture in which mixed radix 4/2 algorithm, pipelined CORDIC, and conflict-free parallel memory access scheme are exploited. Besides, the CORDIC rotation angles are generated internally based on the transform of butterfly counter, which eliminates the need of ROM making it memory-efficient. The proposed architecture has a lower hardware complexity because it is ROM-free and with no dedicated complex multiplier. We implemented the proposed FFT processor and verified it on FPGA development platform. Additionally, the processor is also synthesized in 0.18 µm technology, the core area of the processor is 3.47 mm2 and the maximum operating frequency can be up to 500 MHz. The proposed FFT processor is better trade off performance and hardware overhead, and it can meet the speed requirement of most modern OFDM system, such as IEEE 802.11n, WiMax, 3GPP-LTE and DVB-T/H.

In this paper, we present the first results on the preimage resistance against step-reduced versions of ISO standard hash functions RIPEMD-128 and RIPEMD-160, which were designed as strengthened versions of RIPEMD. While preimage attacks on the first 33 steps and intermediate 35 steps of RIPEMD (48 steps in total) are known, no preimage attack exists on RIPEMD-128 (64 steps) or RIPEMD-160 (80 steps). This paper shows three variations of preimage attacks of RIPEMD-128; the first 33 steps, intermediate 35 steps, and the last 32 steps. Because of the large security margin, full RIPEMD-128 is still enough secure, however, it is interesting that the number of attacked steps for RIPEMD-128 reaches the same level as for RIPEMD. We also show that our approach can be applied to RIPEMD-160, and present preimage attacks on the first 30 steps and the last 31 steps.

We propose a deception mechanism to combat a compromised station in IEEE 802.11 channel hopping systems. A compromised station can follow the hopping channels and continuously attack them, since it recognizes the channel-hopping sequence. The key concept of the deception mechanism is that an access point notifies a new hopping seed but not to the jammer, while a deception station deceives the jammer. Simulations show that the proposed scheme increases network throughput compared to conventional channel hopping schemes when they are under compromised station attacks.

This paper proposes an easy-to-design, theory-consistent compact feeding circuit, with a single input and four outputs, being comprised of two hybrid circuits that are capable of switching a beam in three directions. The circuits that determine the phase differences between the antennas are present on the same single layer, and thus there is no effect of vias and the design agrees well with the underlying theory. In addition, the vertically and horizontally symmetrical circuit pattern contributes to a substantial reduction in design time. The circuit is designed for use in the ISM band and its properties are evaluated using an RF circuit simulator. A prototype is fabricated and evaluated. The results of the simulation and measurement agree well with the theoretical values. The dimensions of the feeding circuit are 75 (H)55 (W)3.0 (T) mm.

In this letter, we consider a control problem of a chain of integrators by output feedback under sensor noise. First, we introduce a measurement output feedback controller which drives all states and output of the considered system to arbitrarily small bounds. Then, we suggest a measurement output feedback controller coupled with a switching gain-scaling factor in order to improve the transient response and retain the same arbitrarily small ultimate bounds as well. An example is given to show the advantage of the proposed control method.

It is shown that simple preprocessing on the reference signals in multichannel feedforward ANC system can improve the convergence performance of the adaptive ANC algorithm. A fast and efficient blind preprocessing algorithm in frequency domain is proposed to reduce the computational complexity even that the reference sensors are located far from the noise sources. The permutation problem at different frequency bin is also addressed and solved by an independent vector analysis algorithm. The basic principle and performance comparison are given to verify our conclusion.

We present a novel transmission rate control method for Wireless Mesh Networks, termed Semi-Fixed Rate Control (SFRC), which incorporates the advantages of Fixed Rate Control (FRC) and Adaptive Rate Control (ARC). SFRC has two periods, which are alternately repeated: an autorate period and a fixed-rate period. A unit of an autorate period and the successive fixed-rate period is termed “rate-control period”. The duration of the rate-control period is set considerably longer than that of the autorate period. In the autorate period, RTS/CTS is used with the lowest transmission rate, transmission rate adjustment is only applied to data frames, and loss of CTS frames is not reflected in the transmission rate adjustment. In the fixed-rate period, the transmission rate that was used most frequently in the preceding autorate period (optimum rate) is fixed, and RTS/CTS is not used. Implementation of SFRC is straightforward as it uses conventional IEEE 802.11 DCF and only minor modification of the wireless LAN driver is required. SFRC, which uses a modified SampleRate, an ARC implementation in the Madwifi, (SampleRate+) in the autorate period, termed SFRC-SampleRate+, was developed. The results of real-world experiments indicate that SFRC-SampleRate+ is superior to SampleRate and SampleRate+, and is closer to FRC, which uses optimum rate on each link, in terms of throughput in wireless mesh network environments.

IEEE 802.15.4 based devices are a key component for mobile and pervasive computing. However, their small dimensions and reduced resources, together with the intrinsic properties of wireless communication, make it difficult to evaluate such networked systems through real-world trials. In this paper we present an emulation testbed intended for the evaluation of IEEE 802.15.4 networked systems. The testbed builds on the generic framework of the wireless network testbed QOMB, and adds IEEE 802.15.4 network, processor and sensing emulation functionality. We validated the testbed through a series of experiments carried out both through real-world trials in a smart home environment, and through emulation experiments on our testbed. Our results show that one can accurately, and in real time, execute IEEE 802.15.4 network applications on our testbed in an emulated environment that reproduces closely the real scenario.

At the stage of software debugging, the effective interaction between software debugging engineers and fault localization techniques can greatly improve fault localization performance. However, most fault localization approaches usually ignore this interaction and merely utilize the information from testing. Due to different goals of testing and fault localization, the lack of interaction may lead to the issue of information inadequacy, which can substantially degrade fault localization performance. In addition, human work is costly and error-prone. It is vital to study and simulate the pattern of debugging engineers as they apply their knowledge and experience to this interaction to promote fault localization effectiveness and reduce their workload. Thus this paper proposes an effective fault localization approach to simulate this interaction via feedback. Based on results obtained from fault localization techniques, this approach utilizes test data generation techniques to automatically produce feedback for interacting with these fault localization techniques, and then iterate this process to improve fault localization performance until a specific stopping condition is satisfied. Experiments on two standard benchmarks demonstrate the significant improvement of our approach over a promising fault localization technique, namely the spectrum-based fault localization technique.

This paper customizes Goal/Question/Metric (GQM) project monitoring models for various projects and organizations to take advantage of the data from the software tool EPM and to allow the tailoring of the interpretation models based upon the context and success criteria for each project and organization. The basic idea is to build less concrete models that do not include explicit baseline values to interpret metrics values. Instead, we add hypothesis and interpretation layers to the models to help people of different projects make decisions in their own context. We applied the models to two industrial projects, and found that our less concrete models could successfully identify typical problems in software projects.