This paper is about the reduction of the computational complexity of a speech codec. A Linear Predictive Coding procedure is developed to allow its implementation with Number Theoretic Transforms. The use of Fermat Number Transform can reduce, in a significant way, the cost of Linear Predictive algorithm implantation on Digital Signal Processor.

In the teaching of English, there is an increasing focus on practical communication skills. As a result, speaking test component has received more and attention from education experts. With rapid development modern computer technology network technology, use computers to assess quality spoken English become hot topic research in related fields at present. A machine learning assessment system ba...

Linear Predictive Coding (LPC) is used for analysis and compression of speech signals. Whereas Huffman coding is used forElectrocardiogram (ECG) signal compression. This paper presents a hybrid compression technique for ECG signal using modifiedHuffman encoding andLPC.The aim of this paper is to apply the linear prediction coding and modified Huffman coding for analysis, compression and predict...

In this paper, improved Linear Predictive Coding (LPC) coefficients of the frame are employed in the feature extraction method. In the proposed speech recognition system, the static LPC coefficients + dynamic LPC coefficients of the frame were employed as a basic feature. The framework of Linear Discriminant Analysis (LDA) is used to derive an efficient and reduced-dimension speech parametric s...

Speech in humans is generated by a series of formants generated by the resonances in the nasal, pharyngeal and vocal tracts. Linear prediction of speech is one of the most widely used techniques for analysis of speech samples. Voice stress analysis as a field of study holds immense importance to human life. Stress analysis has been used in critical situation endangering life and in several medi...

For linear predictive coding (LPC) of speech, the speech waveform is modelled as the output of an all-pole filter. The waveform is divided into many short intervals (10-30 msec) during which the speech signal is assumed to be stationary. For each interval the constant coefficients of the all-pole filter are estimated by linear prediction by minimizing a squared prediction error criterion. This ...

An analogue electronic model of a cochlea was developed some years ago by R. F. Lyon using a cascade of filters. Combined with an analogue gradient descent circuit, such an artificial cochlea can be used to extract the Linear Predictive Coding (LPC) of the speech signal in continuous time. We propose in this article an analogue VLSI circuit implementing a so-called 'Cochlear LPC' (CLPC). We dis...