The objective of this study is to propose a laboratory test method for confirming LPC (large particle count) reduction by porous membrane filters, which are generally thought to have higher particle removal performanc...
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ISBN:
(纸本)9781479955565
The objective of this study is to propose a laboratory test method for confirming LPC (large particle count) reduction by porous membrane filters, which are generally thought to have higher particle removal performance than melt-blown filters. CMP slurries were filtered by porous membrane filters and a nonwoven depth filter. Then LPC reduction of the filtrates was estimated by a particle counter in conjunction with control membrane test, in which pressure drop increase of a filter (a control membrane) was an indicator of LPC. Difference of LPC for filtrates from each type of filter was small by the particle counter and significant by the control membrane method, but with the porous membrane filter exhibiting lower LPC by the both methods. The results demonstrate that a combination of particle counter and control membrane method can reveal difference of filter performance in LPC reduction with confidence. Further, to show pressure drop increase could be an indicator of LPC, pressure drop increase of the control membrane enhanced by large particles was examined.
In this paper, we propose a Packet Loss Concealment (PLC) method. Our method is based on Pitch Waveform Replication (PWR) and linear Predictive coding (LPC). The estimated packet using LPC is better than that using PW...
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ISBN:
(纸本)9781479952304
In this paper, we propose a Packet Loss Concealment (PLC) method. Our method is based on Pitch Waveform Replication (PWR) and linear Predictive coding (LPC). The estimated packet using LPC is better than that using PWR at a near boundary of the lost packet and the received one. On the other hand, the estimated packet using PWR is better than that using LPC at a distant boundary. Therefore, we combine the two estimated packets by considering the merits and demerits of PWR and LPC. Experimental results show that the proposed method provides better Perceptual Evaluation of Speech Quality (PESQ) scores than the conventional methods. Especially PESQ scores of the proposed method are remarkably excellent in the case of male voice.
Concrete is a popular construction material used broadly in many kinds of structures. The compressive strength is the most important performance of the concrete responsible to the safety of such structures. Although m...
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ISBN:
(纸本)9781467345613
Concrete is a popular construction material used broadly in many kinds of structures. The compressive strength is the most important performance of the concrete responsible to the safety of such structures. Although method of sampling a specimen is commonly employed, it makes damage to the structures. While Schmidt Hammer measures the hardness of the surface, but it relates to the compressive strength with poor accuracy owing to the influence of the conditions of the surface. Considering that the ultrasound propagating inside the concrete with higher strength will decrease less, the quality factor of the ultrasound signals should be higher, we have proposed a new method that evaluates the concrete strength non-destructively using the quality factor of resonant peak of multi-reflected ultrasound propagated inside the concrete. The quality factor is calculated by a linear predictive coefficient (LPC) processing on the ultrasonic signals, where the order of the LPC is an important parameter, by which the number of the poles of the LPC model is determined. Neither too many poles, which include too much details of the original signal with noise, nor too few poles, which lack of the proprieties at the resonant peaks, may express the propriety of ultrasonic propagation clearly. In this paper, as one of the approach of optimizing the parameters employed in measurement and signal processing, the relation between the resonant frequency and the appropriate order of LPC is discussed by numerical simulation and experiment. In the experiment, three types of concrete specimen with identical dimensions but different strengths (48.4 N/mm~2, 37.1 N/mm~2 and 27.9 N/mm~2 respectively) and four kinds of sound sources and receivers with different resonant frequency (28 kHz, 46 kHz, 68 kHz and 105 kHz, respectively) are employed. In the numerical simulation, a low-pass IIR filter is introduced to simulate the absorption of ultrasonic propagation in the concrete. Though the absorption is v
Analysis of speech for recognition of emotion can be done using important characteristics such as pitch, intensity, vocal tract spectrum, duration, formant frequencies etc. Out of these characteristics, pitch, duratio...
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ISBN:
(纸本)9781634396226
Analysis of speech for recognition of emotion can be done using important characteristics such as pitch, intensity, vocal tract spectrum, duration, formant frequencies etc. Out of these characteristics, pitch, duration and formant frequencies estimation method are used for identification of angry, disgust, sad, boredom and neutral emotion identification. Analysis of above mentioned characteristics is done using TU-Berlin database (Technical University of Berlin). Pitch is determined using cepstral analysis and formant frequencies are using LPC analysis. Total duration is calculated by the extraction of vowel, semivowel, and consonant containing in that particular speech signal.
This paper is motivated by the recognition that sources of uncertainties in the electric power systems are multifold and that they may have potentially far-reaching effects. In the past, only system load forecast was ...
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ISBN:
(纸本)9781467327275
This paper is motivated by the recognition that sources of uncertainties in the electric power systems are multifold and that they may have potentially far-reaching effects. In the past, only system load forecast was considered to be the main challenge. More recently, uncertain price of electricity and hard-to-predict power produced by the renewable resources, such as wind and solar, are making the operating and planning environment much more challenging. It is, therefore, becoming very important to develop modeling methods for predicting uncertain load and wind power, in particular. In this paper we first transform historic time-stamped data into their Fourier Transform (FT) representation. The frequency domain data representation is used to decompose the wind and load power signals and to derive predictive models relevant for short-term and long-term predictions. The short-term results are interpreted next as a linear prediction coding Model (LPC) and its accuracy is analyzed. Next, the Discrete Markov Process (DMP) representation is applied to help assess probabilities of most likely short-, medium- and long-term states and the related multi-temporal risks. Throughout the paper we use publicly available data for the New York Control Area (NYCA).
Compressive sensing (CS) is a promising focus in signal processing field, which offers a novel view of simultaneous compression and sampling. In this framework a sparse approximated signal is obtained with samples...
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Compressive sensing (CS) is a promising focus in signal processing field, which offers a novel view of simultaneous compression and sampling. In this framework a sparse approximated signal is obtained with samples much less than that required by the Nyquist sampling theorem if the signal is sparse on one basis. Encouraged by its exciting potential application in signal compression, we use CS framework for speech synthesis problems. The linearpredictioncoding (LPC) is an efficient tool for speech compression, as the speech is considered to be an AR process. It is also known that a speech signal is quasi-periodic in its voiced parts, hence a discrete fourier transform (DFT) basis will provide a better approximation. Thus we propose a hybrid dictionary combined with the LPC model and the DFT model as the basis of speech signal. The orthogonal matching pursuit (OMP) is employed in our simulations to compute the sparse representation in the hybrid dictionary domain. The results indicate good performance with our proposed scheme, offering a satisfactory perceptual quality.
This paper describes two low-complexity tools used for the new ITU-T recommendation G.711.0, the lossless coding of G.711 (A-law/mu-law logarithmic PCM) speech data. One is an algorithm for quantizing the PARCOR/refle...
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ISBN:
(纸本)9781424442959
This paper describes two low-complexity tools used for the new ITU-T recommendation G.711.0, the lossless coding of G.711 (A-law/mu-law logarithmic PCM) speech data. One is an algorithm for quantizing the PARCOR/reflection coefficients and the other is an estimation method for the optimal prediction order. Both tools are based on a criterion that minimizes the entropy of the prediction residual signals and can be implemented in a fixed-point low-complexity algorithm. G.711.0 with the developed practical tools will be widely used everywhere because it can losslessly reduce the data rate of G.711, the prevailing speech-coding technology.
In this paper bootstrap resampling techniques are applied to assess speech quality and thereby evaluate performance of distinct speech enhancement algorithms, under the assumption that the speech segments can be appro...
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ISBN:
(纸本)9781424442959
In this paper bootstrap resampling techniques are applied to assess speech quality and thereby evaluate performance of distinct speech enhancement algorithms, under the assumption that the speech segments can be approximated by an autoregressive model. A bootstrap-based multiple hypotheses testing procedure is constructed to test a distance measure based on linear predictive coding, which is the log-likelihood ratio distance. It is shown that the multiple hypotheses test results correlate well with conventional numerical distance measures, which suggests the applicability of the proposed procedure in assessment of speech quality as well as speech enhancement algorithms.
In this paper, we describe a new approach to cope with packet loss in speech coders. The idea is to split the information present in each speech packet into two components, one to independently decode the given speech...
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ISBN:
(纸本)9781424442959
In this paper, we describe a new approach to cope with packet loss in speech coders. The idea is to split the information present in each speech packet into two components, one to independently decode the given speech frame and one to enhance it by exploiting inter-frame dependencies. The scheme is based on sparse linearprediction and a redefinition of the analysis-by-synthesis process. We present Mean Opinion Scores for the presented coder with different degrees of packet loss and show that it performs similarly to frame dependent coders for low packet loss probability and similarly to frame independent coders for high packet loss probability. We also present ideas on how to make the coder work synergistically with the channel loss estimate.
A preprocessing scheme based on linearprediction coefficient (LPC) residual is applied to higher-order statistics (HOSs) for automatic assessment of an overall pathological voice quality. The normalized skewness and ...
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A preprocessing scheme based on linearprediction coefficient (LPC) residual is applied to higher-order statistics (HOSs) for automatic assessment of an overall pathological voice quality. The normalized skewness and kurtosis are estimated from the LPC residual and show statistically meaningful distributions to characterize the pathological voice quality. 83 voice samples of the sustained vowel /a/ phonation are used in this study and are independently assessed by a speech and language therapist ( SALT) according to the grade of the severity of dysphonia of GRBAS scale. These are used to train and test classification and regression tree ( CART). The best result is obtained using an optima l decision tree implemented by a combination of the normalized skewness and kurtosis, with an accuracy of 92.9%. It is concluded that the method can be used as an assessment tool, providing a valuable aid to the SALT during clinical evaluation of an overall pathological voice quality. Copyright (C) 2009 J. Lee and M. Hahn. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited.
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