• Speech Sciences
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• v.3
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• pp.132-147
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• 1998

Previously developed methods for pitch modification have not been based on the voice source model. Therefore, the synthesized speech often sounds unnatural although it may be highly intelligible. The purpose of this paper is to analyze the alteration of a voice source signal with pitch period and to establish the pitch-modification rule based on the result of this analysis. We examine the alteration of the interval of closing phase, closed phase and open phase using the excitation waveform as the pitch increases. In comparison to the previous methods which performed directly on the speech signal, the pitch modification method based on a voice source model shows high intelligibility and naturalness. This study might benefit the application to the speaker identification and the voice color conversion. Therefore the proposed method will provide high quality synthetic speech.

• Speech Sciences
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• v.3
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• pp.167-176
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• 1998

Conventional time-scale modification methods have the problem that as the modification rate gets higher the time-scale modified speech signal becomes less intelligible, because they ignore the effect of articulation rate on speech characteristics. Results of research on speech perception show that the timing information of transient portions of a speech signal plays an important role in discriminating among different speech sounds. Inspired by this fact, we propose a novel scheme for modifying the time-scale of speech. In the proposed scheme, the timing information of the transient portions of speech is preserved, while the steady portions of speech are compressed or expanded somewhat excessively for maintaining overall time-scale change. In order to identify the transient and steady portions of a speech signal, we employ a simple method using LPC cepstral distance between neighboring frames. The result of the subjective preference test indicates that the proposed method produces performance superior to that of the conventional SOLA method, especially for very fast playback case.

• Journal of Electrical Engineering and Technology
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• v.13 no.4
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• pp.1558-1565
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• 2018

Three-phase current is reconstructed from the dc-link current in an AC machine drive with a single current sensor. Switching pattern modification methods, in which the magnitude of the effective voltage vector is secured over its minimum, are investigated to accurately reconstruct the three-phase current. However, the existing methods that modify the switching pattern cause voltage and current distortions that degrade sensorless performance. This paper proposes a variable-magnitude voltage signal injection method based on a high frequency voltage signal injection. The proposed method generates a voltage reference vector that ensures the minimum magnitude of the effective voltage vector by varying the magnitude of the injection signal. This method can realize high quality current reconstruction without switching pattern modification. The proposed method is verified by experiments in a 600W Interior permanent magnet synchronous machine (IPMSM) drive system.

• Molecules and Cells
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• v.45 no.7
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• pp.435-443
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• 2022

In response to environmental changes, signaling pathways rewire gene expression programs through transcription factors. Epigenetic modification of the transcribed RNA can be another layer of gene expression regulation. N⁶-adenosine methylation (m⁶A) is one of the most common modifications on mRNA. It is a reversible chemical mark catalyzed by the enzymes that deposit and remove methyl groups. m⁶A recruits effector proteins that determine the fate of mRNAs through changes in splicing, cellular localization, stability, and translation efficiency. Emerging evidence shows that key signal transduction pathways including TGFβ (transforming growth factor-β), ERK (extracellular signal-regulated kinase), and mTORC1 (mechanistic target of rapamycin complex 1) regulate downstream gene expression through m⁶A processing. Conversely, m⁶A can modulate the activity of signal transduction networks via m⁶A modification of signaling pathway genes or by acting as a ligand for receptors. In this review, we discuss the current understanding of the crosstalk between m⁶A and signaling pathways and its implication for biological systems.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2004.10a
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• pp.1078-1081
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• 2004

A robot manipulator is usually operated in two modes: free motion and constraint motion depending on whether the robot comes into contact with the environment or not. At the moment of contact, impact occurs, and sometimes, it possibly degrade the robot's performance by vibration and at worst, shortens its lifetime. In this article, a new proposed algorithm is described by introducing a command signal modification method on the basis of impedance control and a validity of the proposed algorithm is demonstrated by showing a simulation and an experiment.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.77-85
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• 2001

This paper proposes a method of time-scale modification of polyphonic audio signals based on a sinusoidal model. The signals are modeled with sinusoidal component and noise component. A multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in time-scale modification a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. For extracting sinusoidal components and calculating their parameters matching pursuit algorithm is applied to each analysis frame of subband signal. In accordance with spectrum analysis a psychoacoustic model implementing the effect of frequency masking is incorporated with matching pursuit to provide a resonable stop condition of iteration and reduce the number of sinusoids. The noise component obtained by subtracting the synthesized signal with sinusoidal components from the original signal is modeled by line-segment model of short time spectrum envelope. For various polyphonic audio signals the result of simulation shows suggested sinusoidal modeling can synthesize original signal without loss of perceptual quality and do more robust and high quality time-scale modification for large scale factor because of representing transients without any perceptual loss.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2
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• pp.50-61
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• 1995

A time scale modification method yielding rate-modified speech while conserving the characteristic of speech was implemented in real-time using a goneral purpose digital signal processor. Time scale modification changed pronunciation speed only, producing a time difference between the input signal and the modified signal, making it impossible to implement it in real-time. In this thesis, a system was implemented to remove the time difference between the input and modified signals. Speech signals slowed down or speeded up by a physical time scale modification method, such as adjusting the motor speed of the cassett tape recorder, was used as the input signal. Physical modification that controled only the inter speed of the cassette tape player distorted the pitch period of the original speech. In this study, a real-time system was implemented so that the pitch-distorted speech was reconstructed back to the original by fractional sampling pitch shifting using an FIR filter, and this signal was time scale modified to match the cassette tape recorder motor speed using SOLA time-scale medification. In experiments using speech signals medifiedby the proposed method, results obtained using a 16-bit resolution ADSP2101 processor and using computer simulations employing floating point operations showed about the same average frame signal-to-noise ratio of about 20 dB.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.6
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• pp.147-155
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• 1998

Conventional time-scale modification methods have the problem that as the modification rate gets higher the time-scale modified speech signal becomes less intelligible, because they ignore the effect of articulation rate on speech characteristics. In this paper, we propose a variable time-scale modification method based on the knowledge that the timing information of transient portions of a speech signal plays an important role in speech perception. After identifying steady protions only. The result of subjective preference test indicates that the proposed method produces performance superior to that of the conventional SOLA method.

• Smart Media Journal
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• v.7 no.3
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• pp.9-15
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• 2018

This paper proposes an optimal gain modification method of the Multichannel Wiener filter (MWF) using speech presence probabilities. Conventional gain modification methods of MWFs have the problem of the increase of speech distortions while reducing residual noises with its relative heuristic approach. However, the proposed optimal gain modification method, derived by solving the unconstrained minimization problem of the probability-involved cost function, reduces amounts of residual noises and signal distortions simultaneously. Through an evaluation of the filtered waveforms and spectrograms, it is verified that the proposed method results in an improved SNR with less signal distortions compared to the conventional MWF.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.5
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• pp.939-951
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• 1994

In adaptive equalizers based on least mean squares (LMS) algorithms, the convergence rate is determined by the convariance matrix of an input signal. When the eigenvalue spread of the convariance matrix is close to unity, the convergence rate is quite fast. In this paper, for fast convergence of LMS-based adaptive equalizers we propose a modified reference signal pertinent to the statistical channel. From the theoretical analysis and computer simulation, it is shown that the proposed modification method is quite effective for fast convergence of LMS-based adaptive equalizers.