Mohammad Hasan Savoji

Affiliations:
  • Shahid Beheshti University, Electrical and Computer Engineering Faculty, Tehran, Iran


According to our database1, Mohammad Hasan Savoji authored at least 26 papers between 1985 and 2016.

Collaborative distances:
  • Dijkstra number2 of five.
  • Erdős number3 of four.

Timeline

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Bibliography

2016
Supervised speech enhancement using online Group-Sparse Convolutive NMF.
Proceedings of the 8th International Symposium on Telecommunications, 2016

2014
Nonlinear backward ADPCM speech coding using kernel methods and online sparsification.
Trans. Emerg. Telecommun. Technol., 2014

2012
Wide-band speech coding based on bandwidth extension and sparse linear prediction.
Proceedings of the 35th International Conference on Telecommunications and Signal Processing, 2012

Wide-band speech coding using kernel methods and bandwidth extension based on parametric stereo.
Proceedings of the 20th European Signal Processing Conference, 2012

2011
Employing Volterra filters in the ADPCM technique for speech coding: a comprehensive investigation.
Eur. Trans. Telecommun., 2011

Adaptive Variable Degree-k Zero-Trees for Re-Encoding of Perceptually Quantized Wavelet-Packet Transformed Audio and High Quality Speech
CoRR, 2011

2010
A new iterative algorithm for estimating the glottal flow derivative of vowels.
Proceedings of the 10th International Conference on Information Sciences, 2010

Knowledge based blind deconvolution of non-minimum phase FIR systems.
Proceedings of the 23rd Canadian Conference on Electrical and Computer Engineering, 2010

2009
Re-encoding of perceptually quantized wavelet packet transform coefficients of audio and high quality speech.
Proceedings of the 16th International Conference on Digital Signal Processing, 2009

2008
Noise separation in analog integrated circuits using EMD-PCA-ICA.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

Speech enhancement using a variable suppression rule in Hilbert domain.
Proceedings of the 2008 16th European Signal Processing Conference, 2008

The Advantage of Implementing Martin's Noise Reduction Algorithm in Critical Bands Using Wavelet Packet Decomposition and Hilbert Transform.
Proceedings of the Advances in Computer Science and Engineering, 2008

2007
An efficient recursive algorithm and an explicit formula for calculating update vectors of running walsh-hadamard transform.
Proceedings of the 9th International Symposium on Signal Processing and Its Applications, 2007

Speech enhancement in harsh noisy environment using analytic decomposition of speech signal in critical bands.
Proceedings of the 9th International Symposium on Signal Processing and Its Applications, 2007

2006
A new collision resistant hash function based on optimum dimensionality reduction using Walsh-Hadamard transform.
Proceedings of the 9th International Conference in Information Technology, 2006

2004
Automatic and accurate pitch marking of speech signal using an expert system based on logical combinations of different algorithms outputs.
Proceedings of the 2004 12th European Signal Processing Conference, 2004

2002
Non-linear Prediction of Speech Signal Using Artificial Neural Nets.
Proceedings of the EurAsia-ICT 2002: Information and Communication Technology, 2002

1997
Speech synthesis and prosody modification using segmentation and modelling of the excitation signal.
Proceedings of the Fifth European Conference on Speech Communication and Technology, 1997

1995
Improvement of the quality of speech synthesis by analysis using segmentation and modeling of the excitation signal.
Proceedings of the Fourth European Conference on Speech Communication and Technology, 1995

Speech synthesis system based on a variable decimation/interpolation factor.
Proceedings of the 1995 International Conference on Acoustics, 1995

1994
New algorithm for spectral smoothing and envelope modification for LP-PSOLA synthesis.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

1992
A new algorithm for connected digit recognition.
Proceedings of the Second International Conference on Spoken Language Processing, 1992

1989
A robust algorithm for accurate endpointing of speech signals.
Speech Commun., 1989

1987
A proposal for a speaker independent isolated word (SIIW) recogniser of a limited vocabulary.
Proceedings of the European Conference on Speech Technology, 1987

1986
A variable length lattice filter for adaptive noise cancellation.
Proceedings of the IEEE International Conference on Acoustics, 1986

1985
On different methods based on the Karhunen-Loeve expansion and used in image analysis.
Comput. Vis. Graph. Image Process., 1985


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