We explore using recursive autoencoders for SemEval 2015 Task 1: Paraphrase and Semantic Similarity in Twitter. Our paraphrase detection system makes use of phrase-structure parse tree embeddings that are then provide...
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Along the prompt growth in World Wide Web, the availability and accessibility of regional language contents such as e-books, web pages, e-mails, and digital repositories has grown exponentially. As a result, the autom...
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The objective of the present work is to improve the digit recognition performance of speech signals affected with dysarthria. The paper presents preliminary studies performed on universal access dysarthric speech reco...
The objective of the present work is to improve the digit recognition performance of speech signals affected with dysarthria. The paper presents preliminary studies performed on universal access dysarthric speech recognition (UADSR) database. The works presented in the paper are organized into three stages. Firstly, the degradation in the digit recognition performance is demonstrated by testing the dysarthric digits with the acoustic models built using the digit samples spoken by controlled speakers. Secondly, the prosodic analysis is performed on the dysarthric isolated digits that are available in the database. Finally, the prosodic parameters of the dysarthric speech is manipulated to match with the normal speech which is used to build the acoustic models. Based on the experiments conducted, the manipulation of duration parameters using the state of the art time-domain pitch synchronous overlap add (TD-PSOLA) method observed to be significantly improving the recognition rates in contrast to other prosodic parameters. The improvement in the word recognition rates are also found to be in accordance with the intelligibility of the dysarthric speakers and hence proves the significance of using customized prosodic scaling factors according to the intelligibility levels of each of the subjects.
The aim of the proposed work presented in this paper is to determine the speech polarity using the knowledge of epochs and the cosine phase information derived from the complex analytic representation of original spee...
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The aim of the proposed work presented in this paper is to determine the speech polarity using the knowledge of epochs and the cosine phase information derived from the complex analytic representation of original speech signal. The work presented in this paper is motivated by the observation of variations in the cosine phase of speech around the Hilbert envelope (HE) peaks according to the polarity changes. As the HE peaks represent approximate epochs location, the phase analysis is performed by using algorithms which provide better resolution and accuracy of estimated epochs in the present work. In the present work, accurate epochs locations are initially estimated and significant HE peaks are only selected from the near vicinity of the epochs location for phase analysis. The cosine phase of the speech signal is then computed as the ratio of signal to the HE of speech. The trend in the cosine phase around the selected significant HE peaks are observed to be varying according to the speech polarity. The proposed polarity detection algorithm shows better results as compared with the state of the residual skewness based speech polarity detection (RESKEW) method. Thus, the improvement in the polarity detection rates confirms significant polarity information present in the excitation source characteristics around epochs location in speech. The polarity detection rates are also found to be less affected for different levels of noise addition which indicates the effectiveness of the approach against noises. Also, based on the analysis of mean execution time, the proposed polarity detection algorithm is confirmed to be 10 times faster than the RESKEW algorithm.
Multirate digital signal processing finds application in various fields like speech processing, image processing and compression. A proper study of this subject is fundamental to incorporating an efficient design of t...
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This paper presents a supervised prepositional ambiguity resolution method for machine translation models in which the target language is Tamil and source language is English. We restrict our transfer ambiguity resolu...
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This paper presents a supervised prepositional ambiguity resolution method for machine translation models in which the target language is Tamil and source language is English. We restrict our transfer ambiguity resolution problem with few prepositions only. This resolution method is based on supervised models which exploit collocation occurrences and linguistic information as features. This attempt will rectify the challenges in handling prepositions in English to Tamil automatic translation system. The preliminary results obtained from the evaluation shows that the proposed method is suitable for preposition resolution problem.
Weather parameters are critical parameter which is to be considered for understanding atmospheric variation. The aim of this paper is to see how the parameter values fluctuates for three decades. Three decade informat...
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Weather parameters are critical parameter which is to be considered for understanding atmospheric variation. The aim of this paper is to see how the parameter values fluctuates for three decades. Three decade information is isolated into two and K-means machine learning calculation is utilized for bunching the divided data. Then how the information focuses are conveyed around the centroid of every sectioned data group is confirmed. At long last the group for initial one and half decade is compared with the second one and half decade to know how the information focuses are strayed in dynamic year. This gathering is validated with silhouette value.
Epochs are the locations correspond to glottal closure instants for voiced speech segments and onset of bursts or frication in unvoiced segments. In the recent years, the zero frequency filtering (ZFF) based epoch est...
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ISBN:
(纸本)9781479989973
Epochs are the locations correspond to glottal closure instants for voiced speech segments and onset of bursts or frication in unvoiced segments. In the recent years, the zero frequency filtering (ZFF) based epoch estimation has received a growing attention for clean or studio speech signals. The ZFF based epoch estimation exploits the impulse like excitation characteristics at the zero frequency (DC) region in speech. As the lower frequency regions in telephonic speech are significantly attenuated, ZFF approach gives degraded epoch estimation performance. Therefore, the objective of the present work is to propose refinements to the existing ZFF based epoch estimation algorithm for improved epoch estimation in telephonic speech. The strength of the impulses at the zero frequency region are enhanced by computing the Hilbert envelope (HE) of the speech which in turn improve the epoch estimation performance. The resonators located at the approximate Fo locations of the short term blocks of conventional zero frequency filtered signal, are also found to improve the epoch estimation performance in telephonic speech. The performance of the refined ZFF method is evaluated on 3 speaker voices (JMK, SLT and BDL) of CMU Arctic database having simultaneous speech and EGG recordings. The telephonic version of CMU Arctic database is simulated using tools provided by the international telecommunication union (ITU).
Conventional methods of signal decomposition are observed to fail in power system applications and computationally intensive algorithms like EMD, VMD, EWT are found to give better performance. The heavy computations a...
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Conventional methods of signal decomposition are observed to fail in power system applications and computationally intensive algorithms like EMD, VMD, EWT are found to give better performance. The heavy computations associated with them restricts their use in real time applications and stream processing. This paper presents a recursive block processing technique for real time signal decomposition. The use of recursive FFT and the clever initializations of the center frequencies in the existing VMD algorithm helps in reducing the computational complexity and hence speeds up the process. This low complexity algorithm was tested on synthetically generated power signals and the results were observed to be consistent with the existing VMD algorithm.
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