• Title/Summary/Keyword: Speaker Sound

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Characteristic of room acoustical parameters with source-receiver distance on platform in subway stations (지하철 승강장의 음원-수음점 거리에 따른 실내음향 평가지수 특성)

  • Kim, Suhong;Song, Eunsung;Kim, Jeonghoon;Lee, Songmi;Ryu, Jongkwan
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.6
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    • pp.615-625
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    • 2021
  • Prior to proposing appropriate standard for subway station platform, this study conducted field measurements to examine characteristics of room acoustics on platform of two subway stations. As a result of analyzing the longitudinal length of the platform, Sound Pressure Level (SPL) decreased (maximum difference : 14 dB), Reverberation Time (RT) tended to increase (maximum difference of 0.8 s ~ 1.5 s), and C50 and D50 were decreased (maximum difference: 5.9 dB ~ 9.1 dB and 31.8 % ~ 37.6 %, respectively) as measurement positions moved away from the sound source. The Interaural Cross-correlation Coefficient (IACC) did not show clear tendency, but it was lower than 0.3 in entire points. It is judged that the subway platform has non-uniform sound field characteristics due to various combinations of direct and reflective sound even though it is finished with a strong reflective material.This indicates that the room acoustic characteristics of the near and far sound field are clearly expressed depending on the source-receiver distances in the subway platform having a long flat shape with a low height compared to the length.Therefore, detailed architectural and electric acoustic design based on the characteristics of each location of speaker and sound receiver in the platform is required for an acoustic design with clear sound information at all positions of the platform.

Performance assessments of feature vectors and classification algorithms for amphibian sound classification (양서류 울음 소리 식별을 위한 특징 벡터 및 인식 알고리즘 성능 분석)

  • Park, Sangwook;Ko, Kyungdeuk;Ko, Hanseok
    • The Journal of the Acoustical Society of Korea
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    • v.36 no.6
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    • pp.401-406
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    • 2017
  • This paper presents the performance assessment of several key algorithms conducted for amphibian species sound classification. Firstly, 9 target species including endangered species are defined and a database of their sounds is built. For performance assessment, three feature vectors such as MFCC (Mel Frequency Cepstral Coefficient), RCGCC (Robust Compressive Gammachirp filterbank Cepstral Coefficient), and SPCC (Subspace Projection Cepstral Coefficient), and three classifiers such as GMM(Gaussian Mixture Model), SVM(Support Vector Machine), DBN-DNN(Deep Belief Network - Deep Neural Network) are considered. In addition, i-vector based classification system which is widely used for speaker recognition, is used to assess for this task. Experimental results indicate that, SPCC-SVM achieved the best performance with 98.81 % while other methods also attained good performance with above 90 %.

Cognitive abilities and speakers' adaptation of a new acoustic form: A case of a /o/-raising in Seoul Korean

  • Kong, Eun Jong;Kang, Jieun
    • Phonetics and Speech Sciences
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    • v.10 no.3
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    • pp.1-8
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    • 2018
  • The vowel /o/ in Seoul Korean has been undergoing a sound change by altering the acoustic weighting of F2 and F1. Studies documented that this on-going change redefined the nature of a /o/-/u/ contrast as F2 differences rather than as F1 differences. The current study examined two cognitive factors namely executive function capacity (EF) and autistic traits, in terms of their roles in explaining who in speech community would adapt new acoustic forms of the target vowels, and who would retain the old forms. The participants, 55 college students speaking Seoul Korean, produced /o/ and /u/ vowels in isolated words; and completed three EF tasks (Digit N-Back, Stroop, and Trail-Making Task), and an Autism screening questionnaire. The relationships between speakers' cognitive task scores and their utilizations of F1 and F2 were analyzed using a series of correlation tests. Results yielded a meaningful relationship in participants' EF scores interacting with gender. Among the females, speakers with higher EF scores were better at retaining F1, which is a less informative cue for females since they utilized F2 more than they did F1 in realizing /o/ and /u/. In contrast, better EF control among male speakers was associated with more use of the new cue (F2) where males still utilized F1 as much as F2 in the production of /o/ and /u/ vowels. Taken together, individual differences in acoustic realization can be explained by individuals' cognitive abilities, and their progress in the sound change further predicts that cognitive ability influences the utilization of acoustic information which is non-primary to the speaker.

The comparison of cardinal vowels between Koreans and native English speakers (영어의 기본모음과 한국인 영어학습자의 영어모음 발화비교)

  • Kang, Sung-Kwan;Son, Hyeon-Sung;Jeon, Byoung-Man;Kim, Hyun-Gi
    • Proceedings of the KSPS conference
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    • 2007.05a
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    • pp.71-73
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    • 2007
  • The Purpose of the study is to give Korean-English leaners better knowledge on vowel sounds in their learning English. The traditional description of the cardinal vowel system developed by Daniel Johns in 1917 is not enough to provide English learners with clear ideas in producing native like vowel sounds. For the reason, three Korean-native subjects, one male, one female and one child are chosen to produce 7 cardinal vowels and compare them with native English and American speaker's vowel sounds. The difference of produced vowels sounds is quantified and visualized by employing Sona-match program. The results have been fairly remarkable. Firstly, Korean-English learner's vowel sounds are articulated differently from their intention of vowel production. Secondly, the tongue positions of Koreans are placed slightly more down and forward to the lips than those of English and Americans. However, the front vowel /i/ sound is quite close to English and Americans. Lastly the mid-vowel /${\partial}$/ sound is not produced in any articulations of Korean-native speakers. It is thought that the mid vowel, /${\partial}$/ is a type of a weak sound regarded as 'schwa' which needs a great deal of exposure to the language to acquire a physical skill of articulation.

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A Study on Phoneme Likely Units to Improve the Performance of Context-dependent Acoustic Models in Speech Recognition (음성인식에서 문맥의존 음향모델의 성능향상을 위한 유사음소단위에 관한 연구)

  • 임영춘;오세진;김광동;노덕규;송민규;정현열
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.5
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    • pp.388-402
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    • 2003
  • In this paper, we carried out the word, 4 continuous digits. continuous, and task-independent word recognition experiments to verify the effectiveness of the re-defined phoneme-likely units (PLUs) for the phonetic decision tree based HM-Net (Hidden Markov Network) context-dependent (CD) acoustic modeling in Korean appropriately. In case of the 48 PLUs, the phonemes /ㅂ/, /ㄷ/, /ㄱ/ are separated by initial sound, medial vowel, final consonant, and the consonants /ㄹ/, /ㅈ/, /ㅎ/ are also separated by initial sound, final consonant according to the position of syllable, word, and sentence, respectively. In this paper. therefore, we re-define the 39 PLUs by unifying the one phoneme in the separated initial sound, medial vowel, and final consonant of the 48 PLUs to construct the CD acoustic models effectively. Through the experimental results using the re-defined 39 PLUs, in word recognition experiments with the context-independent (CI) acoustic models, the 48 PLUs has an average of 7.06%, higher recognition accuracy than the 39 PLUs used. But in the speaker-independent word recognition experiments with the CD acoustic models, the 39 PLUs has an average of 0.61% better recognition accuracy than the 48 PLUs used. In the 4 continuous digits recognition experiments with the liaison phenomena. the 39 PLUs has also an average of 6.55% higher recognition accuracy. And then, in continuous speech recognition experiments, the 39 PLUs has an average of 15.08% better recognition accuracy than the 48 PLUs used too. Finally, though the 48, 39 PLUs have the lower recognition accuracy, the 39 PLUs has an average of 1.17% higher recognition characteristic than the 48 PLUs used in the task-independent word recognition experiments according to the unknown contextual factor. Through the above experiments, we verified the effectiveness of the re-defined 39 PLUs compared to the 48PLUs to construct the CD acoustic models in this paper.

Transmission Noise Seduction Performance of Smart Panels using Piezoelectric Shunt Damping (압전감쇠를 이용한 압전지능패널의 전달 소음저감 성능)

  • 이중근
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.3 no.1
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    • pp.49-57
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    • 2002
  • The possibility of a transmission noise reduction of piezoelectric smart panels using piezoelectric shunt damping is experimentally studied. Piezoelectric smart panel is basically a plate structure on which piezoelectric patch with shunt circuits is mounted and sound absorbing materials are bonded on the surface of the structure. Sound absorbing materials can absorb the sound transmitted at mid frequency region effectively while the use of piezoelectric shunt damping can reduce the transmission at resonance frequencies of the panel structure. To be able to reduce the sound transmission at low panel resonances, piezoelectric damping using the measured electrical impedance model is adopted. Resonant shunt circuit for piezoelectric shunt damping is composed of register and inductor in series, and they are determined by maximizing the dissipated energy throughout the circuit. The transmitted noise reduction performance of smart panels is investigated using an acoustic tunnel. The tunnel is a tube with square crosses section and a loud-speaker is mounted at one side of the tube as a sound source. Panels are mounted in the middle of the tunnel and the transmitted sound pressure across panels is measured. Noise reduction performance of a smart panels possessing absorbing material and/or air gap shows a good result at mid frequency region but little effect in the resonance frequency. By enabling the piezoelectric shunt damping, noise reduction of 10dB, 8dB is achieved at the resonance frequencise as well. Piezoelectric smart panels incorporating passive method and piezoelectric shunt damping are a promising technology for noise reduction in a broadband frequency.

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Study on Sound Production and Phonotaxis of Some Fishes and Crabs (몇가지 어류 및 갑각류의 발음과 주음성에 관한 연구)

  • 김상한
    • Journal of the Korean Society of Fisheries and Ocean Technology
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    • v.14 no.1
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    • pp.15-36
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    • 1978
  • Underwater sounds of some fishes and crabs were analyzed in the laboratory. The behavioral responses to the playback sounds of their feeding and croaking sound were investigated. The samples used in the experiment were as follows: Nibea albiflora, seriola quinqueradiata, Navodon modestus, Fugu xanthopterus, chrysophrys major, Scylla serrata, Telmessus acutidens, Charybdis japonica, and Portunus trituberculatus. The feeding and croaking sounds of the samples were recorded by a tape recorder through a hydrophone in an anechoic aquarium. The sound intensity level was measured by means of a sound level meter at an anechoic chamber. The frequency, intensity and wave form of various sounds were analyzed with an analyzing system consisting of a 1/3 octave filter set, a high speed level recorder, an amplifier, an octave band analyzer and an oscilloscope. The most successful recording was edited into a sequence of sound track which repeats sound emitting for 5 to 7 seconds after pausing for 5 to 7 seconds. The sequence was then reproduced into an anechoic aquarium through the under water speaker. The experimental anechoic aquarium used for the sample fishes was divided into the four sections with any three screens selected from 40$\times$40mm, 60$\times$60mm, 80$\times$80mm and 100$\times$100mm mushes according to the species of the fishes, besides that for crabs were not sectioned. The results of the investigation are as follows: 1. Of the feeding sound of fish, the frequency of wave from of the sound produced by Nibea albiflora and seriola quinqucradiata was 125~250Hz, that by Navodon modestus 63~125Hz, and that by Fugu xanthopterus 400~500Hz. The pressure level of the feeding sound produced by Nibea albiflora and Seriola quinqueradiata was 56~62db, that by Navodon modestus 57~59db, and that by Fugu xanthopterus 60~64db. 2. Of the croaking sound of Nibea albiflora, the frequency of the sound was 125~250Hz almost equivalent to that of feeding sound, and the pressure level was 62~63db, slightly higher than that of feeding sound. 3. Of the croaking sounds of crabs, the frequency of the sound produced by scylla serrata was 125~250Hz, that by Charybdis japonica and Telmessus acutidens 500~1,000Hz, and that by Portunus trituberculatus 250~500Hz. The pressure level of the croaking sound by Scylla serrata was 68~70db, and that by Charybdis japonica, Telmessus acutidens and Portuens trituberculatus 50~62db. 4. Phonotactic responses of Nibea albiflora and Seriola quinqueradiata to the feeding sounds produced by their own species, the same body length were conspicuous with the phonotactic index of 56~87%, but that of Navodon modestus, Chrysophrys major and Fugu xanthopterus were hardly recognized. 5. Phonotactic responses of the sample fishes to the sinusoidal sound with the frequency range of 50 to 9,000 Hz were observed not conspicuous. 6. Phonotactic responses of Portunus trituberculatus to the croaking sounds produced by their own species was varied in the range of 40~100%, according to the carapace length and the sex.

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Adaptive Multi-Rate(AMR) Speech Coding Algorithm (Adaptive Multi-Rate(AMR) 음성부호화 알고리즘)

  • 서정욱;배건성
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.92-97
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    • 2000
  • An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

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Development of Directional Digital Hearing Aid Performance Testing System (지향성 디지털 보청기의 성능 검사 장치 개발)

  • Jarng, Soon-Suck;Kwon, You-Jung;Lee, Je-Hyeong
    • Proceedings of the IEEK Conference
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    • 2005.11a
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    • pp.411-414
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    • 2005
  • The most recent trend on digital hearing aid is to increase the ratio of signal to noise by directivity or to develop noise reduction algorithm inside DSP IC chip. This paper designed, fabricated and tested a digital hearing aid directivity testing device in which a micro-mouse-like the stepping motor with a speaker rotates around an examinant. Both ears of the examinant were fixed with ITE hearing aids in order to response to receiving sound. The diameter of the directivity testing device was 2 [m] and the micro-mouse was precisely controlled by PICBASIC micro processor.

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A Study on Real-Time 3D Sound Rendering for Virtual Reality Environment (VR환경에 알맞은 실시간 음장구현에 관한 연구)

  • Chae, Soo-Bok;Bhang, Seung-Beum;Shin, Hwang;Ko, Hee-Dong;Kim, Soon-Hyob
    • Proceedings of the IEEK Conference
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    • 2000.09a
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    • pp.197-200
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    • 2000
  • 본 논문은 VR시스템에 사용되는 실시간 음향제시를 위한 시스템 구현에 관한 것이다. 2개의 Speaker 또는 헤드폰을 사용하여 음상제어, 음장제어의 두 부분으로 구성되어 있다. 음상제어 부분은 각각의 음원의 위치를 정위하고, 음장제어 부분은 레이 트레이싱(Ray Tracing)기법을 이용하여 음장을 시뮬레이션하고 가상 공간의 음장 파라미터를 추출하여 음원에 적용하면서 실시간으로 음장효과를 렌더링 한다. 이 시스템은 펜티엄-Ⅱ333MHz 시스템에서 구현하였다. 최종적으로 청취자는 2개의 스피커 또는 헤드폰을 이용하여 3D음장을 경험하게 된다.

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