• Title/Summary/Keyword: Sound Synthesis

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Exploration of Optimal Multi-Core Processor Architecture for Physical Modeling of Plucked-String Instruments (현악기의 물리적 모델링을 위한 최적의 멀티코어 프로세서 아키텍처 탐색)

  • Kang, Myeong-Su;Choi, Ji-Won;Kim, Yong-Min;Kim, Jong-Myon
    • The Journal of the Acoustical Society of Korea
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    • v.30 no.5
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    • pp.281-294
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    • 2011
  • Physics-based sound synthesis usually requires high computational costs and this results in a restriction of its use in real-time applications. This motivates us to implement the sound synthesis algorithm of plucked-string instruments using multi-core processor architectures and determine the optimal processing element (PE) configuration for the target instruments. To determine the optimal PE configuration, we evaluate the impacts of a sample-per-processing element (SPE) ratio that is defined as the amount of sample data directly mapped to each PE on system performance and both area and energy efficiencies using architectural and workload simulations. For the acoustic guitar, the highest area and energy efficiencies are achieved at a SPE ratio of 5,513 and 2,756, respectively, for the synthesis of musical sounds sampled at 44.1 kHz. In the case of the classical guitar, the maximum area and energy efficiencies are achieved at a SPE ratio of 22,050 and 5,513, respectively. In addition, the synthetic sounds were very similar to original sounds in their spectra. Furthermore, we conducted MUSHRA subjective listening test with ten subjects including nine graduate students and one professor from the University of Ulsan, and the evaluation of the synthetic sounds was excellent.

Vibration and Noise Control of the Simply Supported Slab Using the Multi-tuned Mass Damper (다중동조질량감쇠기를 이용한 단순지지 슬래브의 진동 및 소음저감에 관한 연구)

  • Hwang, Jae-Seung;Hong, Geon-Ho;Park, Hong-Gun
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.18 no.10
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    • pp.1006-1013
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    • 2008
  • In this study, it is outlined that heavy weight floor impact noise induced by the vibration of slab can be reduced using multi tuned mass damper(MTMD) effectively. Substructure synthesis is utilized to develope analytical model of the slab coupled with MTMD and acoustic power is introduced to evaluate the performance of noise control for simplicity. Numerical analysis is carried out to investigate the effect of the properties of MTMD on the vibration and noise control of the simply supported slab. Numerical analysis shows that mass ratio of MTMD is critical on the vibration and noise control of the slab and it is also essential to reduce the vibration in higher modes of slab in the light of its great effect on the radiation of sound.

Implementation of Text-to-Audio Visual Speech Synthesis Using Key Frames of Face Images (키프레임 얼굴영상을 이용한 시청각음성합성 시스템 구현)

  • Kim MyoungGon;Kim JinYoung;Baek SeongJoon
    • MALSORI
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    • no.43
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    • pp.73-88
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    • 2002
  • In this paper, for natural facial synthesis, lip-synch algorithm based on key-frame method using RBF(radial bases function) is presented. For lips synthesizing, we make viseme range parameters from phoneme and its duration information that come out from the text-to-speech(TTS) system. And we extract viseme information from Av DB that coincides in each phoneme. We apply dominance function to reflect coarticulation phenomenon, and apply bilinear interpolation to reduce calculation time. At the next time lip-synch is performed by playing the synthesized images obtained by interpolation between each phonemes and the speech sound of TTS.

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Detection and Synthesis of Transition Parts of The Speech Signal

  • Kim, Moo-Young
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.3C
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    • pp.234-239
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    • 2008
  • For the efficient coding and transmission, the speech signal can be classified into three distinctive classes: voiced, unvoiced, and transition classes. At low bit rate coding below 4 kbit/s, conventional sinusoidal transform coders synthesize speech of high quality for the purely voiced and unvoiced classes, whereas not for the transition class. The transition class including plosive sound and abrupt voiced-onset has the lack of periodicity, thus it is often classified and synthesized as the unvoiced class. In this paper, the efficient algorithm for the transition class detection is proposed, which demonstrates superior detection performance not only for clean speech but for noisy speech. For the detected transition frame, phase information is transmitted instead of magnitude information for speech synthesis. From the listening test, it was shown that the proposed algorithm produces better speech quality than the conventional one.

Vibration and noise control of slab using the multi-tuned mass damper (다중질량감쇠기를 이용한 슬래브의 진동 및 소음저감에 관한 연구)

  • Hwang, Jae-Seung;Kim, Hong-Jin;Kang, Kyung-Soo;Hong, Gun-Ho
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2008.04a
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    • pp.659-664
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    • 2008
  • In this study, it is outlined that heavy weight floor impact noise induced by the vibration of slab can be reduced by multi tuned mass damper(MTMD) effectively. Substructure synthesis is utilized to develope analytical model of slab coupled with MTMD and acoustic power is introduced to evaluate the performance of noise control for simplicity. Numerical analysis is carried out to investigate the effect of the properties of MTMD on the vibration and noise control. Numerical analysis shows that mass ratio of MTMD is critical on the vibration and noise control of the slab and it is essential to reduce the vibration in higher modes of slab because it has a great effect on the radiation of sound.

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Transfer Path Analysis and Estimation of the Road Noise for the Driving Vehicle (주행 차량의 로드 노이즈 예측을 위한 각 입력원의 기여도 평가)

  • Yang, In-Hyung;Jeong, Jae-Eun;Yoon, Ji-Hyun;Oh, Jae-Eung
    • Transactions of the Korean Society for Noise and Vibration Engineering
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    • v.20 no.11
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    • pp.1071-1077
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    • 2010
  • The reduction of the vehicle interior noise has been the main interest of noise and vibration harshness(NVH) engineers. A passenger vehicle has various and complicated transmission paths of sound and vibration. In order to identify the mechanism of transfer path, estimation of excitation force and exact modeling of transfer path are required. This paper presents method for estimating the noise source contribution on the road noise of the vehicle in a multiple input system where the input sources may be coherent with each other. And vector synthesis technique is employed to identify the characteristics of road noise and its transmission to vehicle compartment through noise and vibration analysis. Vibration reduction efficiency of each transfer path is evaluated by comparing individual vector components obtained virtual simulation.

A Study on the Human Auditory Scaling (인간의 청각 척도에 관한 고찰)

  • Yang, Byung-Gon
    • Speech Sciences
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    • v.2
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    • pp.125-134
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    • 1997
  • Human beings can perceive various aspects of sound including loudness, pitch, length, and timber. Recently many studies were conducted to clarify complex auditory scales of the human ear. This study critically reviews some of these scales (decibel, sone, phon for loudness perception; mel and bark for pitch) and proposes to apply the scales to normalize acoustic correlates of human speech. One of the most important aspects of human auditory perception is the nonlinearity which should be incorporated into the linear speech analysis and synthesis system. Further studies using more sophisticated equipment are desirable to refine these scales, through the analysis of human auditory perception of complex tones or speech. This will lead scientists to develop better speech recognition and synthesis devices.

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Perceptual Experiment on Number Production for Speaker Identification

  • Yang, Byung-Gon
    • Speech Sciences
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    • v.8 no.1
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    • pp.7-19
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    • 2001
  • The acoustic parameters of nine Korean numbers were analyzed by Praat, a speech analysis software, and synthesized by SenSynPPC, a Klatt formant synthesizer. The overall intensity, pitch and formant values of the numbers were modified dynamically by a step of 1 dB, 1 Hz and 2.5% respectively. The study explored the sensitivity of listeners to changes in the three acoustic parameters. Twelve subjects (male and female) listened to 390 pairs of synthesized numbers and judged whether the given pair sounded the same or different. Results showed that subjects perceived the same sound quality within the range of 6.6 dB of intensity variation, 10.5 Hz of pitch variation and 5.9% of the first three formant variations. The male and female groups showed almost the same perceptual ranges. Also, an asymmetrical structure of high and low boundary was observed. The ranges may be applicable to the development of a speaker identification system while the method of synthesis modification may apply to its evaluation data.

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An ACLMS-MPC Coding Method Integrated with ACFBD-MPC and LMS-MPC at 8kbps bit rate. (8kbps 비트율을 갖는 ACFBD-MPC와 LMS-MPC를 통합한 ACLMS-MPC 부호화 방식)

  • Lee, See-woo
    • Journal of Internet Computing and Services
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    • v.19 no.6
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    • pp.1-7
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    • 2018
  • This paper present an 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) coding method integrated with ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding) and LMS-MPC(Least Mean Square - Multi Pulse Coding) used V/UV/S(Voiced / Unvoiced / Silence) switching, compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. In integrating several methods, it is important to adjust the bit rate of voiced and unvoiced sound source to 8kbps while reducing the distortion of the speech waveform. In adjusting the bit rate of voiced and unvoiced sound source to 8 kbps, the speech waveform can be synthesized efficiently by restoring the individual pitch intervals using multi pulse in the representative interval. I was implemented that the ACLMS-MPC method and evaluate the SNR of APC-LMS in coding condition in 8kbps. As a result, SNR of ACLMS-MPC was 15.0dB for female voice and 14.3dB for male voice respectively. Therefore, I found that ACLMS-MPC was improved by 0.3dB~1.8dB for male voice and 0.3dB~1.6dB for female voice compared to existing MPC, ACFBD-MPC and LMS-MPC. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or internet phone. In the future, I will study the evaluation of the sound quality of 6.9kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source.

A study on implementation of courseware for Digital System Simulation and Crcuit Synthesis (디지털 시스템의 시뮬레이션과 회로합성을 위한 코스웨어 구현에 관한 연구)

  • 이천우;김형배;강호성;박인정
    • Journal of the Korean Institute of Telematics and Electronics T
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    • v.36T no.3
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    • pp.94-100
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    • 1999
  • In this paper, we are implemented the courseware targets to the integrated a digital system analysis, a design theory, and a hardware description language training and a logic analysis. This paper consists of two subjects. One is that the learning of a digital system analysis, that of a design theory, and the training of a hardware description language is simultaneously performed. The other is that the experiment of courseware. To learn the hardware description language, the explanation using sound or moving images, setting-up of a simulation or a synthesis program, and simulating are executed on a courseware. And also, we proposed an integrated systems for the hardware description language and a logic synthesis. Also, The reliablity of the tool was verified to be preyed an efficient operation of an implemented digital system courseware tool by korea computer research association.

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