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This paper deals with the recognition process of Bangla speech. The used database consists of two sets of data - one is for training containing 3824 utterances of Bangla digit sequences of 25 male and 25 female speakers and the other one is test dataset containing 1985 utterances of 26 male and 26 female speakers. The test set is subdivided into four groups such as clean1, clean2, clean3 and clean4...
The main objective of this paper is to explore the effectiveness of perceptual features for performing isolated digits and continuous speech recognition. The proposed perceptual features are captured and code book indices are extracted. Expectation maximization algorithm is used to generate HMM models for the speeches. Speech recognition system is evaluated on clean test speeches and the experimental...
This paper proposes an Automated Instrumentation system for Speech Recognition (AISR) to provide a two-way communication between deaf and vocal people. This system translates speech signal to American Sign Language. Words that correspond to signs from the American sign language dictionary calls a prerecorded American sign language (ASL) showing the sign that is played on the monitor of a portable...
In this paper we introduce the SPHINX3-based Bengali Automatic Speech Recognition(ASR) system Shruti-II and an E-mail application based on it. This ASR system converts standard Bengali continuous speech to Bengali Unicode. Due to the limited availability of access to computer, visually impaired community can use speech as an input method for various computer-based application. This paper also demonstrates...
A new perceptual time varying model for non-stationary analysis of speech signals is presented. Some researches have already shown that the time varying linear prediction coding (TVLPC) model that was applied to speech signals increases the recognition performance of automatic speech recognition (ASR) systems. This improvement has been achieved due to the incorporation of the speech dynamics information...
According to HMM's strong representation capability of speech signal and GMM's better transformation effect, a method for LSF conversion using HMM combined with GMM is proposed. The theoretical derivation and flow diagram of this algorithm are offered, and gauss model is introduced to achieve the prosodic features transformation. The experiment is applied on two segment speech, and the result reveals...
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