• Title/Summary/Keyword: Signal Distortion

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Statistical analysis on long-term change of jitter component on continuous speech signal (음성신호의 Jitter 성분의 장시간 변화에 관한 통계적 분석)

  • Jo, Cheolwoo
    • Phonetics and Speech Sciences
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    • v.12 no.4
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    • pp.73-80
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    • 2020
  • In this study, a method for measuring the jitter component in continuous speech is presented. In the conventional jitter measurement method, pitch variabilities are commonly measured from the sustained vowels. In the case of continuous speech, such as a spoken sentence, distortion occurs with the existing measurement method owing to the influence of prosody information according to the sentence. Therefore, we propose a method to reduce the pitch fluctuations of prosody information in continuous speech. To remove this pitch fluctuation component, a curve representing the fluctuation is obtained via polynomial interpolation for the pitch track in the analysis interval, and the shift is removed according to the curve. Subsequently, the variability of the pitch frequency is obtained by a method of measuring jitter from the trajectory of the pitch from which the shift is removed. To measure the effects of the proposed method, parameter values before and after the operations are compared using samples from the Kay Pentax MEEI database. The statistical analysis of the experimental results showed that jitter components from the continuous speech can be measured effectively by proposed method and the values are comparable to the parameters of sustained vowel from the same speaker.

Compensation for Distorted WDM Signals by Periodic-shaped Dispersion Map and Non-midway Optical Phase Conjugator (주기적 구조의 분산 맵과 Non-midway 광 위상 공액기에 의한 왜곡된 WDM 신호의 보상)

  • Kweon, Soon-Nyu;Lee, Seong-Real
    • Journal of Advanced Navigation Technology
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    • v.26 no.1
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    • pp.22-28
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    • 2022
  • In order to install ultra wide band and ultra long-haul transmission link based on standard single mode fiber, optical signal distortion due to chromatic dispersion and nonlinear Kerr effect must to be compensated. In this paper, optical link consisted of dispersion management and optical phase conjugation is proposed for compensation of the distorted wavelength division multiplexed (WDM) channels. Dispersion map profile in the proposed dispersion-managed link is configured by periodic repetitive shape, and optical phase conjugator is placed at various position including the midway of total transmission length. It is confirmed from simulation results that when the residual dispersion per span (RDPS) selected in the proposed dispersion-managed link to be large, the compensation of distorted WDM channels in the non-midway OPC system is more improved than the conventional dispersion-managed link.

Determinant-based two-channel noise reduction method using speech presence probability (음성존재확률을 이용한 행렬식 기반 2채널 잡음제거기법)

  • Park, Jinuk;Hong, Jungpyo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.26 no.5
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    • pp.649-655
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    • 2022
  • In this paper, a determinant-based two-channel noise reduction method which utilizes speech presence probability (SPP) is proposed. The proposed method improves noise reduction performance from the conventional determinant-based two-channel noise reduction method in [7] by applying SPP to the Wiener filter gain. Consequently, the proposed method adaptively controls the amount of noise reduction depending on the SPP. For performance evaluation, the segmental signal-to-noise ratio (SNR), the perceptual evaluation of speech quality, the short time objective intelligibility, and the log spectral distance were measured in the simulated noisy environments considered various types of noise, reverberation, SNR, and the direction and number of noise sources. The experimental results presented that determinant-based methods outperform phase difference-based methods in most cases. In particular, the proposed method achieved the best noise reduction performance maintaining minimum speech distortion.

Development of Advanced Data Analysis Method Using Harmonic Wavelet Transform for Surface Wave Method (하모닉 웨이브릿 변환을 이용한 표면파 시험을 위한 향상된 데이터 해석기법의 개발)

  • Park, Hyung-Choon;Cho, Sung-Eun
    • Journal of the Korean Geotechnical Society
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    • v.24 no.4
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    • pp.115-123
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    • 2008
  • The dispersive phase velocity of a wave propagating through multilayered systems such as a soil site is an important parameter and carries valuable information in non-destructive site characterization tests. The dispersive phase velocity of a wave can be determined using the phase spectrum, which is easily evaluated through the cross power spectrum. However, the phase spectrum determined using the cross power spectrum is easily distorted by background noise which always exists in the field. This causes distortion of measured signal and difficulties in the determination of the dispersive phase velocities. In this paper, a new method to evaluate the phase spectrum using the harmonic wavelet transform is proposed and the phase spectrum by the proposed method is applied to the determination of dispersion curve. The proposed method can successfully remove background noise effects. To evaluate the validity of the proposed method, numerical simulations of multi-layered systems were performed. Phase spectrums and dispersion curves determined by the proposed method were found to be in good agreement with the actual phase spectrums and dispersion curves biased by heavy background noise. The comparison manifests the proposed method to be a very useful tool to overcome noise effects.

Noise Statistics Estimation Using Target-to-Noise Contribution Ratio for Parameterized Multichannel Wiener Filter (변수내장형 다채널 위너필터를 위한 목적신호대잡음 기여비를 이용한 잡음추정기법)

  • Hong, Jungpyo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.26 no.12
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    • pp.1926-1933
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    • 2022
  • Parameterized multichannel Wiener filter (PMWF) is a linear filter that can control the trade-off between residual noise and signal distortion using the embedded parameter. To apply the PMWF to noisy inputs, accurate noise estimation is important and multichannel minima-controlled recursive averaging (MMCRA) is widely used. However, in the case of the MMCRA, the accuracy of noise estimation decreases when a directional interference is involved into the array inputs. Consequently, the performance of the PMWF is degraded. Therefore, we propose a noise power spectral density (PSD) estimation method for the PMWF in this paper. The proposed method is based on a consecutive process of eigenvalue decomposition on noisy input PSD, estimation of the target component contribution using directional information, and exponential weighting for improved estimation of the target contribution. For evaluation, four objective measures were compared with the MMCRA and we verify that the PMWF with the proposed noise estimation method can improve performance in environments where directional interfereces exist.

Design of the Noise Suppressor Using the Perceptual Model and Wavelet Packet Transform (인지 모델과 웨이블릿 패킷 변환을 이용한 잡음 제거기 설계)

  • Kim, Mi-Seon;Park, Seo-Young;Kim, Young-Ju;Lee, In-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.25 no.7
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    • pp.325-332
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    • 2006
  • In this paper. we Propose the noise suppressor with the Perceptual model and wavelet packet transform. The objective is to enhance speech corrupted colored or non-stationary noise. If corrupted noise is colored. subband approach would be more efficient than whole band one. To avoid serious residual noise and speech distortion, we must adjust the Wavelet Coefficient Threshold (WCT). In this Paper. the subband is designed matching with the critical band and WCT is adapted noise masking threshold (NMT) and segmental signal to noise ratio (seg_SNR). Consequently. it has similar Performance with EVRC in PESQ-MOS. But it's better than wavelet packet transform using universal threshold about 0.289 in PESQ-MOS. The important thing is that it's more useful than EVRC in coded speech. In coded speech. PESQ-MOS is higher than EVRC about 0.23.

The Analysis about Channel Code Performance of Underwater Channel (수중통신채널에서 고려되는 채널 부호의 성능 분석)

  • Bae, Jong-Tae;Kim, Min-Hyuk;Choi, Suk-Soon;Jung, Ji-Won;Chun, Seung-Yong;Dho, Kyeong-Cheol
    • The Journal of the Acoustical Society of Korea
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    • v.27 no.6
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    • pp.286-295
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    • 2008
  • Underwater acoustic communication has multi path error because of reflection by sea-level and sea-bottom. The multipath of underwater channel causes signal distortion and error floor. In this paper, we consider the use of various channel coding schemes such as RS code, convolutional code, cross-layer code and LDPC code in order to compensate the multipath effect in underwater channel. As shown in simulation results, characteristic of multipath error is similar to that of random error, so interleaver has little effect for error correcting. For correcting of error floor by multipath error, it is necessary strong channel codes like LDPC code that is similar to Shannon's limit. And the performance of concatenated codes including RS codes has better performance than using singular channel codes.

MR Neurography: Current Several Issues for Novice Radiologists (자기공명영상 신경조영술: 경험이 적은 영상의학과 의사가 이해해야 할 몇 가지 쟁점들)

  • Dong-ho Ha
    • Journal of the Korean Society of Radiology
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    • v.81 no.1
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    • pp.81-100
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    • 2020
  • Magnetic resonance neurography (MRN) has been increasingly used in recent years for the assessment of peripheral neuropathies. Fat suppression T2-weighted imaging (T2WI) and diffusion-weighted imaging (DWI) have typically been used to provide high contrast MRN. Isotropic 3-dimensional (3D) sequences with fast spin echo, post-processing imaging techniques, and fast imaging methods, among others, allow good visualization of peripheral nerves that have a small diameter, complex anatomy, and oblique course within a reasonable scan time. However, there are still several issues when performing high contrast and high resolution MRN including standard sequence; fat saturation techniques; balance between resolution, field of view, and slice thickness; post-processing techniques; 2D vs. 3D image acquisition; different T2 contrasts between proximal and distal nerves; high T2 signal intensity of adjacent veins or joint fluid; geometric distortion; and appropriate p-values on DWI. The proper understanding of these issues will help novice radiologists evaluate peripheral neuropathies using MRN.

Front-End Processing for Speech Recognition in the Telephone Network (전화망에서의 음성인식을 위한 전처리 연구)

  • Jun, Won-Suk;Shin, Won-Ho;Yang, Tae-Young;Kim, Weon-Goo;Youn, Dae-Hee
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.4
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    • pp.57-63
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    • 1997
  • In this paper, we study the efficient feature vector extraction method and front-end processing to improve the performance of the speech recognition system using KT(Korea Telecommunication) database collected through various telephone channels. First of all, we compare the recognition performances of the feature vectors known to be robust to noise and environmental variation and verify the performance enhancement of the recognition system using weighted cepstral distance measure methods. The experiment result shows that the recognition rate is increasedby using both PLP(Perceptual Linear Prediction) and MFCC(Mel Frequency Cepstral Coefficient) in comparison with LPC cepstrum used in KT recognition system. In cepstral distance measure, the weighted cepstral distance measure functions such as RPS(Root Power Sums) and BPL(Band-Pass Lifter) help the recognition enhancement. The application of the spectral subtraction method decrease the recognition rate because of the effect of distortion. However, RASTA(RelAtive SpecTrAl) processing, CMS(Cepstral Mean Subtraction) and SBR(Signal Bias Removal) enhance the recognition performance. Especially, the CMS method is simple but shows high recognition enhancement. Finally, the performances of the modified methods for the real-time implementation of CMS are compared and the improved method is suggested to prevent the performance degradation.

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Performance Analysis of Adaptive Channel Estimation Scheme in V2V Environments (V2V 환경에서 적응적 채널 추정 기법에 대한 성능 분석)

  • Lee, Jihye;Moon, Sangmi;Kwon, Soonho;Chu, Myeonghun;Bae, Sara;Kim, Hanjong;Kim, Cheolsung;Kim, Daejin;Hwang, Intae
    • Journal of the Institute of Electronics and Information Engineers
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    • v.54 no.8
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    • pp.26-33
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    • 2017
  • Vehicle communication can facilitate efficient coordination among vehicles on the road and enable future vehicular applications such as vehicle safety enhancement, infotainment, or even autonomous driving. In the $3^{rd}$ Generation Partnership Project (3GPP), many studies focus on long term evolution (LTE)-based vehicle communication. Because vehicle speed is high enough to cause severe channel distortion in vehicle-to-vehicle (V2V) environments. We can utilize channel estimation methods to approach a reliable vehicle communication systems. Conventional channel estimation schemes can be categorized as least-squares (LS), decision-directed channel estimation (DDCE), spectral temporal averaging (STA), and smoothing methods. In this study, we propose a smart channel estimation scheme in LTE-based V2V environments. The channel estimation scheme, based on an LTE uplink system, uses a demodulation reference signal (DMRS) as the pilot symbol. Unlike conventional channel estimation schemes, we propose an adaptive smoothing channel estimation scheme (ASCE) using quadratic smoothing (QS) of the pilot symbols, which estimates a channel with greater accuracy and adaptively estimates channels in data symbols. In simulation results, the proposed ASCE scheme shows improved overall performance in terms of the normalized mean square error (NMSE) and bit error rate (BER) relative to conventional schemes.