Sparse coding has high-performance encoding and ability to express images, sparse encoding basis vector plays a crucial role. The computational complexity of the most existing sparse coding basis vectors of is relativ...
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Bottom-gate nanocrystalline silicon (nc-Si) thin film transistors (TFTs) were fabricated and evaluated their characteristics and electrical stability under various stress condition. nc-Si with high crystallinity was d...
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This paper discusses the challenges facing GIS designers in the 21st century. We argue that GI engineers lack a sound theoretical basis that would allow them to make best use of new technologies that handle geospatial...
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In this paper, we propose a real-time architecture for a dual-microphone speech enhancement to be employed in a hands-free mobile communication. The proposed method is based on a new SNR estimation and a coherence bas...
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In this paper, we propose a real-time architecture for a dual-microphone speech enhancement to be employed in a hands-free mobile communication. The proposed method is based on a new SNR estimation and a coherence based Voice Activity Detector (VAD). The enhancement system was implemented using MATLAB tools and DSP processors. The results show that these two implementations are comparable in terms of PESQ score. We also calculate the computational complexity of the enhancement method. The obtained results show that the proposed method is suitable from the hardware implementation point of view.
This paper deals with the problem of speech enhancement in near field condition, when two microphones are available. Proposed technique relies on the difference in power of received signals at the two microphones. Thi...
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This paper deals with the problem of speech enhancement in near field condition, when two microphones are available. Proposed technique relies on the difference in power of received signals at the two microphones. This difference is employed to estimate the clean speech signal power. The method has the capability of dealing with non-stationary noise, a drawback of many noise reduction techniques. Superiority of the presented method over some of prominent methods in this field is demonstrated by conducting both subjective and objective quality tests.
In this paper, we propose a debugging technique for C, which can dynamically find boundary errors on strings in a highly-compatible, accurate and efficient manner. The main idea of our technique is to effectively keep...
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In this paper, we propose a debugging technique for C, which can dynamically find boundary errors on strings in a highly-compatible, accurate and efficient manner. The main idea of our technique is to effectively keep track of hazardous memory bounds (called trap regions) using a small table (called a trap cache) on the static section of the instrumented program. We have implemented our technique as an extension of GCC4.1.1 and conducted experiments. The results show that our technique was easily applicable even to large real programs including Apache 1.3.37 and Linux 2.6.20.4 without requiring significant manual effort, it successfully detected all of ten known boundary errors in them with no false positives, and it incurred low run-time overheads (average 17%) for their benchmarks.
In this work, we are developing the computer aided composition system. This system aids a person, which knows cellphone or background music of home page or software. This system is implemented with the interactive sel...
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In this work, we are developing the computer aided composition system. This system aids a person, which knows cellphone or background music of home page or software. This system is implemented with the interactive selective population climbing. We suppose that it is appropriate to fetch the favorite melody by hill climbing, because we believe that the function about favor in the human can express the proximate single-peak function for acceleration. However, the user's favorite melodies are sometimes changed, as the user's favor becomes clear. Thus, the system selects a real parent from many parents in proportion of these evaluation values in order to fetch the favorite melody during keeping variety interatively, whenever the system generates neibour solutions of the real parent. If the value of neibour solution is higher than one of the real parent, the system exchanges the neibour solution from the real parent. The composing model generates the neibour solutions with randomwalk or most likelihood method in order to change the image of the real parent a little. In the experiment, the melody of 8 measures were composed by evaluating about 30 melodies and improving about 20 melodies. We assume that my system can compose the melody by less burden than other systems.
Sentiment analysis often relies on a semantic orientation lexicon of positive and negative words. A number of approaches have been proposed for creating such lexicons, but they tend to be computationally expensive, an...
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It is found that stable proton acceleration from a thin foil irradiated by a linearly polarized ultraintense laser can be realized for appropriate foil thickness and laser intensity. A dual-peaked electrostatic field,...
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It is found that stable proton acceleration from a thin foil irradiated by a linearly polarized ultraintense laser can be realized for appropriate foil thickness and laser intensity. A dual-peaked electrostatic field, originating from the oscillating and nonoscillating components of the laser ponderomotive force, is formed around the foil surfaces. This field combines radiation-pressure acceleration and target normal sheath acceleration to produce a single quasimonoenergetic ion bunch. A criterion for this mechanism to be operative is obtained and verified by two-dimensional particle-in-cell simulation. At a laser intensity of ∼5.5×1022 W/cm2, quasimonoenergetic GeV proton bunches are obtained with ∼100 MeV energy spread, less than 4° spatial divergence, and ∼50% energy conversion efficiency from the laser.
The popular mel-frequency cepstral coefficients (MFCCs) capture a mixture of speaker-related, phonemic and channel information. Speaker-related information could be further broken down according to articulatory criter...
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The popular mel-frequency cepstral coefficients (MFCCs) capture a mixture of speaker-related, phonemic and channel information. Speaker-related information could be further broken down according to articulatory criteria. How these underlying components are exactly mixed in the features is not well understood. To this end, in this paper we aim at separating the spectra of glottal source and vocal tract using glottal inverse filtering, with an application to speaker recognition over telephone lines. Our experiments on the 10 sec-10 sec condition of the NIST 2006 SRE corpus suggest that the mel-frequency cepstrum of the voice source is not too useful for recognizing speakers. On the contrary, fusing the vocal tract spectrum with conventional MFCCs improves accuracy, suggesting that vocal tract information should be enhanced.
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