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A dynamic programming approach to voice transformation

A dynamic programming approach to voice transformation

引用

IEEE 13th Signal Processing and Communications Applications Conference, SIU 2005

作者： Salor, Özgül demirekler, mübeccel Elektrik ve Elektronik Mühendi Sliǧi Bölümü Orta Doģu Teknik Üniversitesi

ISBN: (纸本)0780392396

In this paper, we present a dynamic programming approach to voice conversion (VT). The goal of VT is to modify the speech of a source speaker such that it is perceived as if spoken by a target speaker. Dynamic programming approach tries to model the long-term behaviour of line spectral frequencies (LSF's) of the target speaker, while it is preserving the relationship between subsequent frames of the source LSF's during transformation. The speech model used is based on mELP speech coding algorithm. The system obtains speaker-specific codebooks of LSF's out of mELP's multistage vector quantization LSF codebook for both source and target speakers. Those codebooks are used to train a mapping histogram, which is used for LSF transformation from one speaker to the other. The baseline system uses the maxima of the histograms for LSF transformations. The shortcomings of this system, which are the limitation of the target LSF space and spectral discontinuities due to independent mapping of subsequent frames, have been overcome by applying the dynamic programming approach. Objective and subjective evaluations have shown that dynamic programming approach improves the transformation performance. © 2005 IEEE.

关键词： Speech processing

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A dynamic programming approach to context-free voice transformation

A dynamic programming approach to context-free voice transfo...

引用

13th European Signal Processing Conference, EUSIPCO 2005

作者： Salor, Özgül demirekler, mübeccel Department of Electrical and Electronics Engineering Middle East Technical University Turkey

ISBN: (纸本)1604238216

In this paper, we present a dynamic programming approach to voice transformation (VT). The goal of VT is to modify the speech of a source speaker such that it is perceived as if spoken by a target speaker. The speech model used in this work is based on mELP (mixed Excitation Linear Prediction) speech coding algorithm. The designed system obtains speaker-specific codebooks of line spectral frequencies (LSFs) out of mELP's multi-stage vector quantization LSF codebook for both source and target speakers. Those codebooks are used to train a mapping histogram, which is used for LSF transformation from one speaker to the other. The baseline system uses the maxima of the histograms for LSF transformations. The shortcomings of this system, which are the limitation of the target LSF space and the spectral discontinuities due to independent mapping of subsequent frames, have been overcome by applying the dynamic programming approach. Dynamic programming approach tries to model the long-term behaviour of the LSFs of the target speaker, while it is trying to preserve the relationship between the subsequent frames of the source LSFs, during transformation. Both objective and subjective evaluations have been conducted and it has been shown that dynamic programming approach improves the performance of the system in terms of both the speech quality and speaker similarity.

关键词： Dynamic programming

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Visual tracking with group motion approach

Visual tracking with group motion approach

引用

2004 IEEE Computer Society Conference on Computer Vision and Pattern Recognition Workshops, CVPRW 2004

作者： Arslan, Ali Erkin demirekler, mübeccel Middle East Technical University Ankara Turkey

An algorithm for tracking a single visual target is presented where visual features of target are tracked in a group tracking method. A single visual target is regarded as a group of multiple targets. The model developed allows some variations on the relative positions of the features which corresponds to some sort of non-rigidity. The approach is a combination of PDA and JPDA algorithms and using the advantages of group motion model this approach enables accurate tracking of single targets. © 2004 IEEE.

关键词： Personal digital assistants

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A new decoder design for large vocabulary Turkish speech recognition 8

A new decoder design for large vocabulary Turkish speech rec...

引用

8th European Conference on Speech Communication and Technology, EUROSPEECH 2003

作者： Çilingir, Onur demirekler, mübeccel Signal Proc. and Remote Sensing Group TÜBITAK BILTEN Ankara Turkey Dept. of Electrical and Electronics Eng. Middle East Technical University Ankara Turkey

An important problem in large vocabulary speech recognition for agglutinative languages like Turkish is the high out of vocabulary (OOV) rate caused by extensive number of distinct words. Recognition systems using words as the basic lexical elements have difficulty in dealing with such virtually unlimited vocabulary. We propose a new time-synchronous lexical tree decoder design using morphemes as the lexical elements. A key feature of the proposed decoder is the dynamic generation of the lexical tree according to the morphological rules. The architecture emulates word generation in the language and therefore allows very large vocabularies through the defined set of morphemes and morphotactical rules.

关键词： Forestry

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A system for voice conversion based on adaptive filtering and line spectral frequency distance optimization for text-to-speech synthesis 8

A system for voice conversion based on adaptive filtering an...

引用

8th European Conference on Speech Communication and Technology, EUROSPEECH 2003

作者： Salor, Özgül demirekler, mübeccel Pellom, Bryan Department of Electrical and Electronics Engineering Middle East Technical University Ankara06531 Turkey Center for Spoken Language Research University of Colorado at Boulder Boulder80303 United States

This paper proposes a new voice conversion algorithm that modifies the source speaker?s speech to sound as if produced by a target speaker. To date, most approaches for speaker transformation are based on mapping functions or codebooks. We propose a linear filtering based approach to the problem of mapping the spectral parameters of one speaker to those of the other. In the proposed method, the transformation is performed by filtering the source speaker?s Line Spectral Pair (LSP) frequencies to obtain the LSP frequency estimates of the target speaker. Speech signal is timealigned into a sequence of Hmmstates. The filters are designed for each Hmm state using the aligned data. We consider two methods for spectral conversion. A linear transformation for the LSP?s was obtained using the adaptive steepest gradient descent approach. mean values of LSP?s are adjusted to match those of the target speaker. In order to prevent the LSP vectors from resulting in unstable vocal tract filters, weighted least square estimation is used. This approach optimizes differences between source and target LSP?s. Weights are inverses of the source LSP variances. This approach is integrated into a Time Domain Pitch Synchronous Overlap and Add (TD-PSOLA) analysis-synthesis framework. The algorithm is objectively evaluated using a distance measure based on the log-likelihood ratio of observing the input speech, given Gaussian mixture speaker models for both the source and the target voice. Results using the Gaussian mixture model formulated criteria demonstrate consistent transformation using a 5 speaker database. The algorithm offers promise for rapidly adapting textto- speech systems to new voices.

关键词： Adaptive filtering

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Implementation and evaluation of a text-to-speech synthesis system for Turkish 8

Implementation and evaluation of a text-to-speech synthesis ...

引用

8th European Conference on Speech Communication and Technology, EUROSPEECH 2003

作者： Salor, Özgül Pellom, Bryan demirekler, mübeccel Department of Electrical and Electronics Engineering Middle East Technical University Turkey Center for Spoken Language Research University of Colorado at Boulder United States

In this paper, a diphone based Text-to-Speech (TTS) system for the Turkish language is presented. Turkish is the official language of Turkey, where it is the native language of 70 million people and it is also widely spoken in Asia (Azerbaidjain, Uzbekhstan, Kazakhstan, Kirgizhstan and Iran), Cyprus and the Balkans. The research has been done through a visiting internship at CSLR (the Center for Spoken Language Research, University of Colorado at Boulder) as part of an ongoing collaboration between CSLR and mETU (middle East Technical University), Department of Electrical and Electronics Engineering. The system is based on Festival Speech Synthesis System. A diphone database has been designed for Turkish. Tools developed for quick diphone collection and segmentation are illustrated. The text analysis module, the methods used for determination of segment durations and pitch contours are discussed in detail. A Diagnostic Rhyme Test (DRT) has been designed for Turkish to test the intelligibility of the output speech. The resulting TTS system is found to be 86.5% intelligible on the average by 20 listeners. This is the first diphone based Turkish TTS system, whose intelligibility is reported. We also believe that, this paper would help researchers working on building TTS voices, especially those who work on agglutinative languages, since every step needed along the way are explained in detail.

关键词： Speech intelligibility

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Why does output normalization create problems in multiple classifier systems

Why does output normalization create problems in multiple cl...

引用

作者： Altinçay, Hakan demirekler, mübeccel Computer Engineering Department Eastern Mediterranean University Gazi Maǧusa Cyprus Electrical and Electronics Engineering Dept. Middle East Technical University Ankara Turkey

Combination of classifiers is a promising direction for obtaining better classification systems. However, the outputs of different classifiers may have different scales and hence the classifier outputs are incomparable. Incomparability of the classifier output scores is a major problem in the combination of different classification systems. In order to avoid this problem, the measurement level classifier outputs are generally normalized. However, recent studies have proven that output normalization may provide some problems. For instance, the miltiple classifier system's performance may become worse than that of a single individual classifier. This paper presents some interesting observations about the reason why such undesirable behavior occurs. © 2002 IEEE.

关键词： Image analysis

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On developing new text and audio corpora and speech recognition tools for the Turkish language 7

On developing new text and audio corpora and speech recognit...

引用

7th International Conference on Spoken Language Processing, ICSLP 2002

作者： Salor, Özgül Pellom, Bryan Çiloǧlu, Tolga Hacioǧlu, Kadri demirekler, mübeccel Department of Electrical and Electronics Engineering Middle East Technical University Ankara Turkey Center for Spoken Language Research University of Colorado Boulder United States

This paper describes recent work towards development of new corpora and tools for Turkish speech research. This effort represents an on-going collaboration between the Center for Spoken Language Research (CSLR) at the University of Colorado and the Department of Electrical Engineering at the middle East Technical University (mETU). A new text corpus developed from Turkish newspapers' text is described. In addition, a 193-speaker audio corpus and pronunciation lexicon for the Turkish language is developed. We then describe our initial work towards porting Sonic, the CSLR speech recognition system, to the Turkish language. Results are shown for phonetic alignment and phoneme recognition accuracy using the newly constructed corpus and speech tools. It is shown that 91.2% of the automatically labeled phoneme boundaries are placed within 20 msec of hand-labeled locations for the Turkish audio corpus. Finally, a phoneme recognition error rate of 29.3% is demonstrated.

关键词： Speech recognition

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Fusing length and voicing information, and Hmm decision using a Bayesian causal tree against insufficient training data1

Fusing length and voicing information, and HMM decision usin...

引用

European Signal Processing Conference (EUSIPCO)

作者： mübeccel demirekler Fahri Karahan Tolga Ciloglu Dept. of Electrical and Electronics Eng. Middle East Tech. Univ. Ankara Turkey

This paper presents the work done to improve the recognition rate in an isolated word recognition problem with single utterance training. The negative effect of errors (due to insufficient training data) in estimated model parameters is compensated by fusing the information obtained from Hmm evaluation and those generated for the word length and voicing at the beginning and end of the word. A Bayesian Causal Tree structure is developed to accomplish the fusion. The final decision is made on one of the three candidates which are most likely according to Hmm evaluation. The reliability of the Hmm ordering is improved by applying variance flooring.

关键词： Hidden markov models Training Vectors Dictionaries Bayes methods Random variables

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Fusing length and voicing information, and Hmm decision using a Bayesian causal tree against insufficient training data

Fusing length and voicing information, and HMM decision usin...

引用

15th International Conference on Pattern Recognition, ICPR 2000

作者： demirekler, mübeccel Karahan, Fahri Çiloglu, Tolga Dept. of Electrical and Electronics Eng. Middle East Tech. Univ. Ankara Turkey

ISBN: (纸本)0769507506

关键词： Trees (mathematics)

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