Mohammad Hasan Savoji

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Walsh-Hadamard transform (WHT) has many applications in digital signal processing including bioinformatics. While there are efficient algorithms for implementing this transform such as fast WHT (FWHT), performing WHT continuously on a sliding window over a long sequence is time consuming. As it is not reasonable to compute a separate WHT upon arrival of(More)
This paper evaluates the problems of implementing two well-known zero-tree-based re-encoding schemes of Embedded Zero-tree Wavelet (EZW) and the set partitioning in hierarchical trees (SPIHT) for perceptually audio and high quality speech coding. Since the original EZW and SPIHT algorithms are designed for image compression, some new modifications have been(More)
The final quality of a concatenation synthesis system is directly related to the continuity of the spectrum at the con-catenation point. Due to the subjective auditory masking, if we minimize the spectral distortion in the formant frequencies , the quality will increase significantly. In this paper we present, along with results concerning pitch marking, an(More)
This paper reports on the results of four re-encoding schemes on perceptually quantized wavelet packet transform (WPT) coefficients of audio and high quality speech. These schemes comprises: 1- Embedded Zero-tree Wavelet (EZW) 2- The set partitioning in hierarchical trees (SPIHT) 3-JPEG-based entropy/run length Huffman and 4- JPEG-type Audio Huffman coding(More)
An expert system comprising a new pitch marking algorithm based on the estimation of the ideal excitation signal, using energy equalization of harmonics of the fundamental frequency present in speech, and three other competent tools is devised and explained in this paper. This expert system uses simple logical combinations of these tools outputs. The(More)
In this paper an efficient and low complexity perceptual method is proposed for quantizing the wavelet packet coefficients of high quality speech signals. The performance of the proposed method is compared, using the same codec, with the case where all coefficients are quantized using a fixed number of bits. The results on 500 TIMIT files show that this(More)
In previous work we have presented a new method for improving the quality of LPC synthetic speech, where the excitation signal was modelled with a polynomial function followed by an adaptive filter. This scheme provides the properties of mathematical models which permits avoiding the problems related to prosody control [1], [2]. In order to reduce the(More)