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Audiovisual articulatory inversion based on Gaussian mixture model (Gmm)

Audiovisual articulatory inversion based on Gaussian Mixture...

引用

18th IEEE Signal Processing and Communications Applications Conference, SIU 2010

作者： Özbek, I. Yücel demirekler, mübeccel Elektrik ve Elektronik Mühendisliǧi Bölümü ODTÜ Ankara Turkey

ISBN: (纸本)9781424496716

In this study, we examined articulatory inversion using audiovisual information based on Gaussian mixture model (Gmm). In this method the joint distribution of the articulatory movement and audio (and/or visual) data are modelled via a mixture of Gaussians. The conditional expected value of the Gmm is used as regression function between the audio (and/orvisual) and articulatory spaces. We also examined various fusion methods in order to combine acoustic and visual information in articulatory inversion. The fusion methods improve the performance of articulatory inversion. ©2010 IEEE.

关键词： Gaussian distribution

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Kalman filter based sinusoidal pulse repetition interval tracking

Kalman filter based sinusoidal pulse repetition interval tra...

引用

2009 IEEE 17th Signal Processing and Communications Applications Conference, SIU 2009

作者： Çamlica, Sedat demirekler, mübeccel ASELSAN A.S. Turkey Elektrik ve Elektronik Mühendisligi Bölümü Orta Dogu Teknik Üniversitesi Turkey

ISBN: (纸本)9781424444366

Pulse Repetition Interval (PRI) tracking has particular interest and importance in Electronic Warfare. In this work, a specific part of PRI tracking, Sinusoidal PRI is studied. Extended Kalman Filter and Unscented Kalman Filter based algorithms have been developed. moreover, simulations for performance comparison and comments are given at the end of the text. ©2009 IEEE.

关键词： Electronic warfare

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Formant trajectories for acoustic-to-articulatory inversion

Formant trajectories for acoustic-to-articulatory inversion

引用

10th Annual Conference of the International Speech Communication Association, INTERSPEECH 2009

作者： Özbek, I. Yücel Hasegawa-Johnson, mark demirekler, mübeccel EE Department Middle East Technical University Turkey ECE Department University of Illinois at Urbana-Champaign United States

This work examines the utility of formant frequencies and their energies in acoustic-to-articulatory inversion. For this purpose, formant frequencies and formant spectral amplitudes are automatically estimated from audio, and are treated as observations for the purpose of estimating electromagnetic articulography (EmA) coil positions. A mixture Gaussian regression model with mel-frequency cepstral (mFCC) observations is modified by using formants and energies to either replace or augment the mFCC observation vector. The augmented observation results in 3.4% lower RmS error, and 2% higher correlation coefficient, than the baseline mFCC observation. Improvement is especially good for stop consonants, possibly because formant tracking provides information about the acoustic resonances that would be otherwise unavailable during stop closure and release. Copyright © 2009 ISCA.

关键词： Regression analysis

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Tracking of vocal tract resonances based on dynamic programming and Kalman filtering

Tracking of vocal tract resonances based on dynamic programm...

引用

2008 IEEE 16th Signal Processing, Communication and Applications Conference, SIU

作者： Özbek, I. Yücel demirekler, mübeccel Elektrik ve Elektronik Mühendisligi Bölümü Orta Dogu Teknik Üniversitesi

ISBN: (纸本)9781424419999

In this work we propose a method for accurate and reliable estimation of vocal tract resonance (VTR) frequencies. The proposed method is based on finding candidates of the VTR frequencies using LPC analysis and estimation of the VTR using Dynamic programming and Kalman filtering/smoothing. The performance of the proposed method is measured in terms of recently proposed hand-labeled mSR-VTR database [5]. The performance of the proposed method is compared with two different formant tracking computation methods. One way is that we compare the results from proposed method with the results from output dynamic programming algorithm, which is our baseline system. The second way is that we compare the proposed method results with the results of WaveSurfer. The proposed method reduces the vocal tract resonance estimation error rate 13%, over the baseline system and 29% over the WaveSurfer. ©2008 IEEE.

关键词： Dynamic programming

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Tracking of Visible Vocal Tract Resonances (VVTR) based on Kalman filtering

Tracking of Visible Vocal Tract Resonances (VVTR) based on K...

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INTERSPEECH 2006 and 9th International Conference on Spoken Language Processing, INTERSPEECH 2006 - ICSLP

作者： Özbek, I Yücel demirekler, mübeccel Department of Electrical and Electronics Engineering Middle East Technical University Turkey

ISBN: (纸本)9781604234497

This paper analyzes vocal tract resonance (VTR) frequency trajectories and their relationship to formants from a new point of view. Considering abrupt/continuous changes in the physical geometry of vocal tract, VTR may change in number, suddenly change their positions or may leak to some regions where they usually do not exist. We define the visible VTR (VVTR) as VTR that can be seen from the spectrogram. So we propose an algorithm, based on Kalman filtering, that can handle all these changes in VVTR. The suggested properties of VVTR trajectories and the performance of the algorithm are demonstrated on several examples.

关键词： Kalman filters

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Tracking of speech formant frequencies

Tracking of speech formant frequencies

引用

2006 IEEE 14th Signal Processing and Communications Applications

作者： Özbek, I. Yücel demirekler, mübeccel Elektrik ve Elektronik Mühendisligi Bölümü Orta Dogu Teknik Üniversitesi

ISBN: (纸本)1424402395

This study is about tracking of formant frequencies using Kalman filtering. Assuming that the formant frequencies are changing in time slowly, it is possible to model their behaviors as outputs of a dynamic system and then track them using Kalman filtering approach. Tracking system considers also the orders and possible intervals of each formant frequency. © 2006 IEEE.

关键词： Tracking (position)

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An efficient algorithm for pitch determination of speech signals - Kalman filter approach

An efficient algorithm for pitch determination of speech sig...

引用

2006 IEEE 14th Signal Processing and Communications Applications

作者： Salor, Özgül demirekler, mübeccel Orguner, Umut Havelsan A.Ş Ankara Elektrik ve Elektronik Mühendisliǧi Bölümü Orta Doǧu Teknik Üniversitesi Ankara

ISBN: (纸本)1424402395

In this paper, an efficient algorithm for pitch determination of speech signals is presented. Pitch period of speech signals does not contain sharp changes in time in voiced and voiced-unvoiced transition regions. Depending on this property, Kalman Filter has been used to estimate the pitch period similar to the way it is used in the target tracking problems. The measurement for the Kalman Filter is obtained by using the integer pitch calculation based on autocorrelation method, which is used as the first step of pitch determination in mELP speech coding algorithm. Kalman Filter makes an a priori estimate, which depends on the general behavior of the pitch period in time and this estimate is updated using this measurement to obtain an a posteriori estimate. The proposed method provides a reduction in the computational complexity since the autocorralation search is made only inside the gating volume of the Kalman Filter. Therefore, the proposed method does not require any pitch doubling check. It also does not need any fractional pitch computations. The pitch periods obtained using this method have been compared to those determined in mELP algorithm and it has been observed that compatible results have been obtained. © 2006 IEEE.

关键词： Speech synthesis

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mETU Turkish microphone speech corpus

METU Turkish microphone speech corpus

引用

2006 IEEE 14th Signal Processing and Communications Applications

作者： Salor, Özgül Çiloǧlu, Tolga demirekler, mübeccel Havelsan A. Ş Ankara Elektrik ve Elektronik Mühendisliǧi Bölümü Orta Doǧu Teknik Üniversitesi

ISBN: (纸本)1424402395

In this paper, work on developing a Turkish microphone speech corpus at the middle East Technical University (mETU) is presented. Before collecting the audio corpus, sound properties of Turkish have been investigated and a triphone-balanced set of Turkish sentences have been developed. Speech from 193 speakers, each uttering 40 sentences selected from the balanced sentence set, has been collected. The corpus has been aligned by the Turkish phoneme aligner developed. Each speech file is associated with phoneme, Hmm state and word level alignments. In addition to these, each speaker has a text file containing the age, region, gender, education and etc. information and also the uttered sentences. The aim of collecting such a corpus is to obtain a standard and common microphone speech corpus for Turkish speech research. The corpus is open for research purposes. It has also been accepted to be distributed by the Linguistic Data Consortium of the Pennsylvania University in November 2005. © 2006 IEEE.

关键词： microphones

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A dynamic programming approach to context-free voice transformation

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

引用

European Signal Processing Conference (EUSIPCO)

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

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 Speech Vectors Histograms Speech coding Training Quantization (signal)

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Voice transformation using principle component analysis based LSF quantization and dynamic programming approach

Voice transformation using principle component analysis base...

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9th European Conference on Speech Communication and Technology

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

The goal of voice transformation (VT) is to modify the speech of a source speaker such that it is perceived as if spoken by a target speaker. In this paper, we present a speaker specific line spectral frequency (LSF) quantization based on principle component analysis (PCA) and k-means clustering for VT. An LPC based source-filter model is used to model the speech. Transformation is applied to the spectral characteristics of the speaker, while pitch scaling is applied on the residual signal. PCA has been used to determine the principle components of the source and target LSFs to obtain a more efficient quantization. Only the dimensions with high variance have been quantized and those dimensions have been used to obtain the histogram matrix mapping the two speakers during training. To select the best target codeword sequence corresponding to a source codeword sequence in a sentence, a dynamic programming approach is used. Dynamic programming approach approximates the long-term behavior of LSFs of the target speaker, while it is trying to preserve the relationship between the subsequent frames of the source LSFs. Objective and subjective evaluations have shown that dimension reduction of LSFs before quantization and dynamic programming improves the voice transformation performance.

关键词： Dynamic programming

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