• Title/Summary/Keyword: CPSP

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A Study on the Robust Sound Localization System Using Subband Filter Bank (서브밴드 필터 뱅크를 이용한 강인한 음원 추적시스템에 대한 연구)

  • 박규식;박재현;온승엽;오상헌
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.1
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    • pp.36-42
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    • 2001
  • This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5% average accuracy for sound source detection.

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A Study on the sound localization system using Subband CPSP Algorithm (Subband CPSP를 이용한 음원 추적 시스템에 관한 연구)

  • 오상헌;박규식;박재현;이현정;온승엽
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.102-105
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    • 2000
  • This paper propose new sound localization algorithm that calculates TDOA(Time Difference Of Arrival) between the two received signals via two microphone array, The proposed Subband CPSP is a development of Previous CPSP method using subband approach. It first split the received microphone signals into three frequency bands and then calculates subband CPSP with corresponding SNR weights. This type of algorithm, Subband CPSP, can provide more accurate TDOA estimation results because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, computer simulation was conducted and it was compared with previous CPSP method. From the both simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than accuracy for TDOA estimation.

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Performance Improvement of CPSP Based TDOA Estimation Using the Preemphasis (프리엠퍼시스를 이용한 CPSP 기반의 도달시간차이 추정 성능 개선)

  • Kwon, Hong-Seok;Bae, Keun-Sung
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.5
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    • pp.461-470
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    • 2009
  • We investigate and analyze the problems encountered in frame-based estimation of TDOA (Time Difference of Arrival) using CPSP function. Spectral leakage occurring in framing of a speech signal by a rectangular window could make estimation of CPSP spectrum inaccurate. Framing with other windows to reduce the spectral leakage distorts the signal due to the asynchronous weighting around the frame specifically both ends of the frame. These problems degrade the performance of the CPSP-based TDOA estimation. In this paper, we propose a method to alleviate those problems by pre-emphasis of the speech signal. It reduces the influence of the spectral leakage by reducing dynamic range of the spectrum of a speech signal with pre-emphasis. To validate the proposed method of pre-emphasis, we carry out TDOA estimation experiments in various noise and reverberation conditions, Experimental results have shown that the framing of pre-emphasized microphone output by a rectangular window achieves higher success rate of TDOA estimation than any other framing methods.

Practical strategies for the prevention and management of chronic postsurgical pain

  • Bo Rim Kim;Soo-Hyuk Yoon;Ho-Jin Lee
    • The Korean Journal of Pain
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    • v.36 no.2
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    • pp.149-162
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    • 2023
  • Chronic postsurgical pain (CPSP) is a multifactorial condition that affects a significant proportion of patients undergoing surgery. The prevention and management of CPSP require the identification of preoperative risk factors to screen high-risk patients and establish appropriate perioperative pain management plans to prevent its development. Active postoperative pain management should be provided to prevent CPSP in patients with severe pain following surgery. These tasks have become important for perioperative team members in the management of CPSP. This review article provides a comprehensive overview of the latest research on the role of perioperative team members in preventing and managing CPSP. Additionally, it highlights practical strategies that can be employed in clinical practice, covering the definition and risk factors for CPSP, including preoperative, intraoperative, and postoperative factors, as well as a risk prediction model. The article also explores various treatments for CPSP, as well as preventive measures, including preemptive analgesia, regional anesthesia, pharmacological interventions, psychoeducational support, and surgical technique modification. This article emphasizes the importance of a comprehensive perioperative pain management plan that includes multidisciplinary interventions, using the transitional pain service as an example. By adopting a multidisciplinary and collaborative approach, perioperative team members can improve patient outcomes, enhance patient satisfaction, and reduce healthcare costs. However, further research is necessary to establish targeted interventions to effectively prevent and manage CPSP.

Performance Improvement of Sound Direction of Arrival Estimation by Applying Threshold to CPSP (CPSP 문턱값 설정을 통한 음원도달 방향 추정 성능 개선)

  • Quan, Xingri;Bae, Keun-Sung
    • Phonetics and Speech Sciences
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    • v.3 no.3
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    • pp.109-114
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    • 2011
  • To estimate sound direction of arrival with a pair of microphones, a method based on Time Difference of Arrival (TDOA) estimation using the Cross Power Spectrum Phase (CPSP) function is largely used due to its simplicity and good performance. In this paper, we investigate CPSP maximum values for various SNRs and adverse environments, and propose a novel method to improve the estimation performance of sound direction of arrival. The proposed method applies a threshold to the CPSP values and increases the reliability of the estimated sound direction. Through computer simulation for various SNRs, we validate the effectiveness of the proposed method. When the threshold was set to 0.1, more than 90% of success rate of sound direction of arrival estimation has been achieved for directions of $10^{\circ}$, $40^{\circ}$, $70^{\circ}$ from the source location even with reverberation times of 0.1s.

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A DSP Implementation of Subband Sound Localization System

  • Park, Kyusik
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.4E
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    • pp.52-60
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    • 2001
  • This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.

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An Enhancement of Microphone Array System Using Hybrid Window Algorithm (CPSP의 저주파 위상 복원을 이용한 화자 위치 추적 알고리듬의 성능 개선)

  • Lee Hak-Ju;Kim Ki-Man;Lee Won-Cheol;Lee Chungyong
    • Proceedings of the Acoustical Society of Korea Conference
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    • autumn
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    • pp.213-216
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    • 2000
  • 본 연구에서는 마이크로폰 어레이를 이용하여 화자의 음성신호로부터 화자의 위치를 추정하는 기존의 대표적인 알고리듬인 CPSP(Cross Power Spectrum Phase)로부터 보다 반향에 강인한 알고리듬인 저주파 위상 복원 알고리듬을 제안한다. CPSP 함수는 상호 상관관계(Cross Correlation)가 정규화 되어있는 형태를 갖는데, CPSP 함수의 최대 값 인덱스로부터 화자의 공간정보인 TDOA(Time Difference Of Arrival)를 추출한다. 그러나 CPSP 함수를 이용한 공간정보 추정 알고리듬은 실내환경에서 심각하게 일어나는 반향신호에 대해서 취약한 단점을 갖고 있다. 본 논문에서 제안하는 저주파 위상복원 알고리듬은 주파수 측면에서 반향신호가 CPSP 함수에 미치는 영향을 분석하여 반향으로 인하여 왜곡된 위상 성분을 복원함으로써 보다 신뢰도 있는 TDOA 추정을 가능하게 한다. 반향신호로 인한 CPSP의 위상은 저주파보다 고주파에서 심하게 왜곡되는데, 각각의 반향신호의 도달 시간을 기하학적 분포를 갖는 확률변수로 모델링하여 이를 수학적으로 증명하였다. 또한 실제 환경에서 채집한 음성신호를 이용한 모의 실험을 통해 개선된 알고리듬의 성능 개선을 확인하였다.

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Modified Subband CPSP를 이용한 음원 추적 시스템에 관한 연구

  • 오상헌;박규식
    • Proceedings of the Acoustical Society of Korea Conference
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    • autumn
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    • pp.139-142
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    • 1999
  • 본 연구는 폐쇄된 임의의 공간상에서 2개의 마이크로폰 어레이를 이용 수신된 2개 신호의 도착 시간차를 계산하여 응원의 방향 각을 추정하는 새로운 알고리듬을 제안한다. 제안된 MSCPSP 알고리듬은 기존의 CPSP 알고리듬을 개선한것으로, 마이크로폰에 수신된 2개의 입력신호에 대해 서브밴드 필터 뱅크를 이용하여 대역 분할하고 각 서브밴드 대역에서 구해지는 신호 대 잡음비(SNH)를 대역별 CPSP 결과에 가중치로 제공한다. 이러한 대역 분할 가중방식은 잡음의 영향을 각 대역으로 한정 분산시켜 보다 정확한 지연 시간 추정을 가능하게 한다. 제안된 알고리듬의 성능을 입증하기 위해 기존의 CPSP와 MSCPSP 알고리듬의 컴퓨터 모의 실험을 수행하였으며, 실험 결과 제안된 MSCPSP의 우수함을 볼 수 있었다.

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Repetitive transcranial magnetic stimulation in central post-stroke pain: current status and future perspective

  • Riva Satya Radiansyah;Deby Wahyuning Hadi
    • The Korean Journal of Pain
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    • v.36 no.4
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    • pp.408-424
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    • 2023
  • Central post-stroke pain (CPSP) is an incapacitating disorder that impacts a substantial proportion of stroke survivors and can diminish their quality of life. Conventional therapies for CPSP, including tricyclic antidepressants, anticonvulsants, and opioids, are frequently ineffective, necessitating the investigation of alternative therapeutic strategies. Repetitive transcranial magnetic stimulation (rTMS) is now recognized as a promising noninvasive pain management method for CPSP. rTMS modulates neural activity through the administration of magnetic pulses to specific cortical regions. Trials analyzing the effects of rTMS on CPSP have generated various outcomes, but the evidence suggests possible analgesic benefits. In CPSP and other neuropathic pain conditions, high-frequency rTMS targeting the primary motor cortex (M1) with figure-eight coils has demonstrated significant pain alleviation. Due to its associaton with analgesic benefits, M1 is the most frequently targeted area. The duration and frequency of rTMS sessions, as well as the stimulation intensity, have been studied in an effort to optimize treatment outcomes. The short-term pain relief effects of rTMS have been observed, but the long-term effects (> 3 months) require further investigation. Aspects such as stimulation frequency, location, and treatment period can influence the efficacy of rTMS and ought to be considered while planning the procedure. Standardized guidelines for using rTMS in CPSP would optimize therapy protocols and improve patient outcomes. This review article provides an up-to-date overview of the incidence, clinical characteristics, outcome of rTMS in CPSP patients, and future perspective in the field.

An Enhancement of Speaker Location System Using the Low-frequency Phase Restoration Algorithm and Its Implementation (저주파 위상 복원 알고리듬을 이용한 화자 위치 추적 시스템의 성능 개선과 구현)

  • 이학주;차일환;윤대희;이충용
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.4
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    • pp.22-28
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    • 2001
  • This paper describes the implementation of a robust speaker position location system using the voice signal received by microphone array. To be robust to the reverberation which is the major factor of the performance degradation, low-frequency phase restoration algorithm which eliminates the influence of reverberations using the low-frequency information of the CPSP function is proposed. The implemented real-time system consists of a general purpose DSP (TMS320C31 of Texas instruments), analog part which contains amplifiers and filters, and digital part which is composed of the external memory and 12-bit A/D converter. In the real conference room environment, the implemented system that was constructed by the proposed algorithms showed better performance than the conventional system. The error of the TDOA estimation reduced more than 15 samples.

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