• Title/Summary/Keyword: Music Algorithm

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A Comparison of Speech/Music Discrimination Features for Audio Indexing (오디오 인덱싱을 위한 음성/음악 분류 특징 비교)

  • 이경록;서봉수;김진영
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.2
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    • pp.10-15
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    • 2001
  • In this paper, we describe the comparison between the combination of features using a speech and music discrimination, which is classifying between speech and music on audio signals. Audio signals are classified into 3classes (speech, music, speech and music) and 2classes (speech, music). Experiments carried out on three types of feature, Mel-cepstrum, energy, zero-crossings, and try to find a best combination between features to speech and music discrimination. We using a Gaussian Mixture Model (GMM) for discrimination algorithm and combine different features into a single vector prior to modeling the data with a GMM. In 3classes, the best result is achieved using Mel-cepstrum, energy and zero-crossings in a single feature vector (speech: 95.1%, music: 61.9%, speech & music: 55.5%). In 2classes, the best result is achieved using Mel-cepstrum, energy and Mel-cepstrum, energy, zero-crossings in a single feature vector (speech: 98.9%, music: 100%).

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AOA Estimation Algorithm Using Interconnected Neural Network Model (상호결합형 신경망 모델을 이용한 실시간 도래방향 추정알고리즘에 관한 연구)

  • 정중식;임정빈;안영섭
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2003.05a
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    • pp.111-114
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    • 2003
  • It has well known that MUSIC and ESPRIT algorithms estimate angle of arrival(AOA) with high resolution by eigenvalue decomposition of the covariance matrix which were obtained from the array antennas. In the case that 2-D large-sized array antenna is required, however, one of the disadvantages of MUSIC and ESPRIT is that they are computationally ineffective, and then they are difficult to implement in real time. To alleviate the computational complexity, several method using neural model have been study. For multiple signals, those methods require huge training data prior to AOA estimation. This paper proposes the algorithm for AOA estimation by interconnected hopfield neural model. Computer simulations show the validity of the proposed algorithm.

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Estimating Angle-of-Arrival of UWB Cluster signals in an Indoor-to-Outdoor Wireless Communication (실내와 실외 무선통신 환경에서 초광대역 클러스터 신호의 도착 방향 추정)

  • Lee Yong-Up;Seo Young-Jun;Choi Gin-Kyu
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.5C
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    • pp.556-561
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    • 2006
  • In this study, an ultra-wideband(UWB) signal model is considered to estimate the angle-of-arrivals(AOAs) of clusters in an UWB indoor-to-outdoor communication environment having random angle spreads. A conventional AOA algorithm for UWB estimates the directions of both clusters and multipath signals together and so has complex estimation procedure. In order to solve that problem, we propose a new clustered AOA estimation algorithm. The estimation technique based a well-known MUSIC algorithm is proposed and the estimates of the AOA on received clusters are obtained using the proposed technique. The proposed model and estimation technique are verified through computer simulations.

Direction of Arrival Estimation under Aliasing Conditions (앨리아싱 조건에서의 광대역 음향신호의 방위각 추정)

  • 윤병우
    • Journal of the Institute of Convergence Signal Processing
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    • v.4 no.3
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    • pp.1-6
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    • 2003
  • It is difficult to detect and to track the moving targets like tanks and diesel vehicles due to the variety of terrain and moving of targets. It is possible to be happened the aliasing conditions as the difficulty of antenna deployment in the complex environment like the battle fields. In this paper, we study the problem of detecting and tracking of moving targets which are emitting wideband signals under severe spatial aliasing conditions because of the sparse arrays. We developed a direction of arrival(DOA) estimation algorithm based on subband MUSIC(Multiple Signal Classification) method which produces high-resolution estimation. In this algorithm, the true bearings are invariant regardless of changes of frequency bands while the aliased false bearings vary. As a result, the proposed algorithm overcomes the aliasing effects and improves the localization performance in sparse passive arrays.

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A Super-Resolution Time Delay Estimation Algorithm for Spread Spectrum Signals (대역 확산 신호를 위한 지연 시간 추정 알고리즘)

  • Shin, Joon-Ho;Myong, Seung-Il;Chang, Eun-Young;Park, Hyung-Rae
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.37 no.2A
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    • pp.119-127
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    • 2012
  • In this paper a super-resolution time delay estimation algorithm is developed for real-time locating systems (RTLSs) that employ a direct-sequence spread spectrum technique, along with its performance analysis in multipath environments. The classical correlation method provides relatively good performance in line-of-sight (LOS) environments but its performance seriously degrades in multipath environments, especially when signals are spaced closer than a PN chip. Therefore we shall develop a super-resolution time delay estimation algorithm that may estimate the time delays of multipath signals even in closely spaced multipath environments using the MUSIC algorithm for direction-of-arrival estimation and analyze its performance by applying the algorithm to the ISO/IEC 24730-2.1 RTLS system. 

HS Implementation Based on Music Scale (음계를 기반으로 한 HS 구현)

  • Lee, Tae-Bong
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.15 no.5
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    • pp.299-307
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    • 2022
  • Harmony Search (HS) is a relatively recently developed meta-heuristic optimization algorithm, and various studies have been conducted on it. HS is based on the musician's improvisational performance, and the objective variables play the role of the instrument. However, each instrument is given only a sound range, and there is no concept of a scale that can be said to be the basis of music. In this study, the performance of the algorithm is improved by introducing a scale to the existing HS and quantizing the bandwidth. The introduced scale was applied to HM initialization instead of the existing method that was randomly initialized in the sound band. The quantization step can be set arbitrarily, and through this, a relatively large bandwidth is used at the beginning of the algorithm to improve the exploration of the algorithm, and a small bandwidth is used to improve the exploitation in the second half. Through the introduction of scale and bandwidth quantization, it was possible to reduce the algorithm performance deviation due to the initial value and improve the algorithm convergence speed and success rate compared to the existing HS. The results of this study were confirmed by comparing examples of optimization values for various functions with the conventional method. Specific comparative values were described in the simulation.

Effective Mood Classification Method based on Music Segments (부분 정보에 기반한 효과적인 음악 무드 분류 방법)

  • Park, Gun-Han;Park, Sang-Yong;Kang, Seok-Joong
    • Journal of Korea Multimedia Society
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    • v.10 no.3
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    • pp.391-400
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    • 2007
  • According to the recent advances in multimedia computing, storage and searching technology have made large volume of music contents become prevalent. Also there has been increasing needs for the study on efficient categorization and searching technique for music contents management. In this paper, a new classifying method using the local information of music content and music tone feature is proposed. While the conventional classifying algorithms are based on entire information of music content, the algorithm proposed in this paper focuses on only the specific local information, which can drastically reduce the computing time without losing classifying accuracy. In order to improve the classifying accuracy, it uses a new classification feature based on music tone. The proposed method has been implemented as a part of MuSE (Music Search/Classification Engine) which was installed on various systems including commercial PDAs and PCs.

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Effect of Digital Noise Reduction of Hearing Aids on Music and Speech Perception

  • Kim, Hyo Jeong;Lee, Jae Hee;Shim, Hyun Joon
    • Journal of Audiology & Otology
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    • v.24 no.4
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    • pp.180-190
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    • 2020
  • Background and Objectives: Although many studies have evaluated the effect of the digital noise reduction (DNR) algorithm of hearing aids (HAs) on speech recognition, there are few studies on the effect of DNR on music perception. Therefore, we aimed to evaluate the effect of DNR on music, in addition to speech perception, using objective and subjective measurements. Subjects and Methods: Sixteen HA users participated in this study (58.00±10.44 years; 3 males and 13 females). The objective assessment of speech and music perception was based on the Korean version of the Clinical Assessment of Music Perception test and word and sentence recognition scores. Meanwhile, for the subjective assessment, the quality rating of speech and music as well as self-reported HA benefits were evaluated. Results: There was no improvement conferred with DNR of HAs on the objective assessment tests of speech and music perception. The pitch discrimination at 262 Hz in the DNR-off condition was better than that in the unaided condition (p=0.024); however, the unaided condition and the DNR-on conditions did not differ. In the Korean music background questionnaire, responses regarding ease of communication were better in the DNR-on condition than in the DNR-off condition (p=0.029). Conclusions: Speech and music perception or sound quality did not improve with the activation of DNR. However, DNR positively influenced the listener's subjective listening comfort. The DNR-off condition in HAs may be beneficial for pitch discrimination at some frequencies.

Effect of Digital Noise Reduction of Hearing Aids on Music and Speech Perception

  • Kim, Hyo Jeong;Lee, Jae Hee;Shim, Hyun Joon
    • Korean Journal of Audiology
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    • v.24 no.4
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    • pp.180-190
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    • 2020
  • Background and Objectives: Although many studies have evaluated the effect of the digital noise reduction (DNR) algorithm of hearing aids (HAs) on speech recognition, there are few studies on the effect of DNR on music perception. Therefore, we aimed to evaluate the effect of DNR on music, in addition to speech perception, using objective and subjective measurements. Subjects and Methods: Sixteen HA users participated in this study (58.00±10.44 years; 3 males and 13 females). The objective assessment of speech and music perception was based on the Korean version of the Clinical Assessment of Music Perception test and word and sentence recognition scores. Meanwhile, for the subjective assessment, the quality rating of speech and music as well as self-reported HA benefits were evaluated. Results: There was no improvement conferred with DNR of HAs on the objective assessment tests of speech and music perception. The pitch discrimination at 262 Hz in the DNR-off condition was better than that in the unaided condition (p=0.024); however, the unaided condition and the DNR-on conditions did not differ. In the Korean music background questionnaire, responses regarding ease of communication were better in the DNR-on condition than in the DNR-off condition (p=0.029). Conclusions: Speech and music perception or sound quality did not improve with the activation of DNR. However, DNR positively influenced the listener's subjective listening comfort. The DNR-off condition in HAs may be beneficial for pitch discrimination at some frequencies.

Low Complexity Super Resolution Algorithm for FOD FMCW Radar Systems (이물질 탐지용 FMCW 레이더를 위한 저복잡도 초고해상도 알고리즘)

  • Kim, Bong-seok;Kim, Sangdong;Lee, Jonghun
    • IEMEK Journal of Embedded Systems and Applications
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    • v.13 no.1
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    • pp.1-8
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    • 2018
  • This paper proposes a low complexity super resolution algorithm for frequency modulated continuous wave (FMCW) radar systems for foreign object debris (FOD) detection. FOD radar has a requirement to detect foreign object in small units in a large area. However, The fast Fourier transform (FFT) method, which is most widely used in FMCW radar, has a disadvantage in that it can not distinguish between adjacent targets. Super resolution algorithms have a significantly higher resolution compared with the detection algorithm based on FFT. However, in the case of the large number of samples, the computational complexity of the super resolution algorithms is drastically high and thus super resolution algorithms are difficult to apply to real time systems. In order to overcome this disadvantage of super resolution algorithm, first, the proposed algorithm coarsely obtains the frequency of the beat signal by employing FFT. Instead of using all the samples of the beat signal, the number of samples is adjusted according to the frequency of the beat signal. By doing so, the proposed algorithm significantly reduces the computational complexity of multiple signal classifier (MUSIC) algorithm. Simulation results show that the proposed method achieves accurate location even though it has considerably lower complexity than the conventional super resolution algorithms.