Show/Hide Menu
Hide/Show Apps
Logout
Türkçe
Türkçe
Search
Search
Login
Login
OpenMETU
OpenMETU
About
About
Open Science Policy
Open Science Policy
Open Access Guideline
Open Access Guideline
Postgraduate Thesis Guideline
Postgraduate Thesis Guideline
Communities & Collections
Communities & Collections
Help
Help
Frequently Asked Questions
Frequently Asked Questions
Guides
Guides
Thesis submission
Thesis submission
MS without thesis term project submission
MS without thesis term project submission
Publication submission with DOI
Publication submission with DOI
Publication submission
Publication submission
Supporting Information
Supporting Information
General Information
General Information
Copyright, Embargo and License
Copyright, Embargo and License
Contact us
Contact us
Speech conversion using MELP speech coding algorithm
Date
2004-04-30
Author
Salor, O
Demirekler, Mübeccel
Metadata
Show full item record
This work is licensed under a
Creative Commons Attribution-NonCommercial-NoDerivatives 4.0 International License
.
Item Usage Stats
149
views
0
downloads
Cite This
In this work, MELP (Mixed Excitation Linear Prediction) speech coding algorithm has been used for speech conversion. Speech conversion aims to modify the speech of one speaker such that the modified speech sounds as if spoken by another speaker. Speech modeling of MELP has been used to derive a mapping the between the speech models of the two speakers. We have obtained a mapping which provides a context-free speech conversion. We have mainly considered the spectral properties of the speakers. Using the 230 sentences of the two speakers, a mapping between the 4-stage vector quantization indexes for line spectral frequencies (LSF's) of the two speakers have been obtained. Two different methods have been proposed to obtain a codebook for the second speaker from this mapping and both have been applied in additon to pitch modification during synthesis. The first method replaces the LSF index of the first speaker with that of the second speaker, which appears the most, during training. The second method uses the weighted average from die histogram of the second speaker that corresponds to the index of the first speaker, to form a new LSF codebook for the second speaker.
Subject Keywords
Speech coding
,
Statistical analysis
,
Table lookup
,
Vector quantisation
,
Spectral analysis
,
Linear predictive coding
URI
https://hdl.handle.net/11511/56721
DOI
https://doi.org/10.1109/siu.2004.1338311
Collections
Graduate School of Natural and Applied Sciences, Conference / Seminar
Suggestions
OpenMETU
Core
Feature selection using genetics-based algorithm and its application to speaker identification
Demirekler, Mübeccel; Haydar, A (1999-03-19)
This paper introduces the use of genetics-based algorithm in the reduction of 24 parameter set (i.e the base set) to a 5,6,7,8 or 10 parameter set, for each speaker in text-independent speaker identification. The feature selection is done by finding the best features that discriminates a person from his/her two closest neighbors. The experimental results show that there is approximately 5% increase in the recognition rate when the reduced set of parameters are used. Also the amount of calculation necessary ...
Spectral modification for context-free voice conversion using MELP speech coding framework
Salor, O; Demirekler, Mübeccel (2004-10-22)
In this work, we have focused on spectral modification of speech for voice con version from one speaker to another. Speech conversion aims to modify the speech of one speaker such that the modified speech sounds as if spoken by another speaker. MELP (Mixed Excitation Linear Prediction) speech coding algorithm has been used as speech analysis and synthesis framework. Using a 230-sentence triphone balanced database of the two speakers, a mapping between the 4-stage vector quantization indexes for line spectra...
Speaker identification through use of features selected using genetic algorithm
Haydar, A; Demirekler, Mübeccel; Yurtseven, MK (Institution of Engineering and Technology (IET), 1998-01-08)
The authors introduce the use of a genetic algorithm in the reduction of a 24 parameter (12 LPC derived cepstral coefficients +12 Delta-cepstral coefficients) set to a five, six, seven, eight or ten parameter set, for each speaker in text-independent speaker identification. The experimental results show that there is similar to 5% increase in the recognition rate when the reduced set of parameters is used.
KNOWLEDGE-BASED SPEECH SYNTHESIS BY CONCATENATION OF PHONEME SAMPLES
OZUM, IY; Bulut, Mehmet Mete (1994-04-14)
In this work a speech synthesis system is implemented. The system uses concatenation of phoneme waveforms as the method of synthesis. These waveforms are generated by sampling the speech of a human speaker and then separating it into its phonemes. These phoneme samples are stored in the hard disk to be used in the synthesis. Then the text to be read is separated into its syllables and each syllable is synthesized by concatenating the phoneme samples. This method is facilitated by the structure of the Turkis...
Lossless Intra Coding in HEVC with Adaptive 3-Tap Filters
Alvar, Saeed Ranjbar; Kamışlı, Fatih (2016-08-05)
In pixel-by-pixel spatial prediction methods for lossless intra coding, the prediction is obtained by a weighted sum of neighboring pixels. The proposed prediction approach in this paper uses a weighted sum of three neighbor pixels according to a two-dimensional correlation model. The weights are obtained after a three step optimization procedure. The first two stages are offline procedures where the computed prediction weights are obtained offline from training sequences. The third stage is an online optim...
Citation Formats
IEEE
ACM
APA
CHICAGO
MLA
BibTeX
O. Salor and M. Demirekler, “Speech conversion using MELP speech coding algorithm,” 2004, Accessed: 00, 2020. [Online]. Available: https://hdl.handle.net/11511/56721.