• The Journal of the Acoustical Society of Korea
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• v.27 no.3
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• pp.149-153
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• 2008

General underwater target detection methods adopt short-time FFT for estimate target doppler. This paper proposes the efficient target detection method, instead of conventional FFT, using approximate FFT for distributed sensor network target detection, which requires lighter computations. In the proposed method, we decrease computational rate of FFT by the quantization of received signal. For validation of the proposed method, experiment result which is applied to FFT based active sonar detector and real oceanic data is presented.

• The Journal of the Acoustical Society of Korea
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• v.26 no.7
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• pp.334-342
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• 2007

This paper presents an algorithm for the automatic detection of chord names, scales and tonalities from music file, expressed in MusicXML format which has enough information to determine harmonies vertically like 4-part choir. Chord names are absolute names which can be used and analysed independently of the tonality An algorithm selecting the best chord name is described, which can decide the most appropriate one from ambiguous situations. Candidate musical scales are extracted using the notes in a given time window. The tonalities of the music are determined using the chord names and candidate scales. The final output format of the process is also MusicXML file with chord names, marked non-harmonic notes, relative harmonic symbols and tonalities.

• The Journal of the Acoustical Society of Korea
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• v.26 no.6
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• pp.269-275
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• 2007

Pervasive Multi-channel audio systems are being realized due to advances in digital technology. This paper proposes an efficient system that serially connects individual speakers with bidirectional digital communication capability by means of SoC design. In particular, each speaker can identify the bit stream assigned to the speaker and convert it into analog audio. Furthermore, the speaker can self-diagnose the speaker functionality by utilizing the designed capability to measure frequencies of various square wave test signals. The proposed system running on 200MHz clock yielded restoration of analog output signal with latency of only $500{\mu}s$ compared to directly driving the speakers in a traditional way.

• Journal of the Institute of Convergence Signal Processing
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• v.11 no.4
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• pp.289-297
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• 2010

In this paper, A method to prevent the external acoustic noise from being misrecognized as the speech recognition object is presented in the speech activity detection process for the speech recognition. Also this paper confirmed besides the acoustic energy to the lip movement image signals. First of all, the successive images are obtained through the image camera for personal computer and the lip movement whether or not is discriminated. The next, the lip movement image signal data is stored in the shared memory and shares with the speech recognition process. In the mean time, the acoustic energy whether or not by the utterance of a speaker is verified by confirming data stored in the shared memory in the speech activity detection process which is the preprocess phase of the speech recognition. Finally, as a experimental result of linking the speech recognition processor and the image processor, it is confirmed to be normal progression to the output of the speech recognition result if face to the image camera and speak. On the other hand, it is confirmed not to the output the result of the speech recognition if does not face to the image camera and speak. Also, the initial feature values under off-line are replaced by them. Similarly, the initial template image captured while off-line is replaced with a template image captured under on-line, so the discrimination of the lip movement image tracking is raised. An image processing test bed was implemented to confirm the lip movement image tracking process visually and to analyze the related parameters on a real-time basis. As a result of linking the speech and image processing system, the interworking rate shows 99.3% in the various illumination environments.

• The Journal of the Acoustical Society of Korea
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• v.41 no.3
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• pp.278-286
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• 2022

In this study, a 33-mode Free-Flooded Ring (FFR) transducer was designed to apply piezoelectric single crystal PIN-PMN-PT, which has high piezoelectric constants and electromechanical coupling coefficient. To ensure low-frequency high transmitting sensitivity characteristics with a small size of FFR transducer, the commercial FFR transducer based on piezoelectric ceramics was compared. To develop the FFR transducer with broadband characteristics, a piezoelectric segmented ring structure inserted with inactive elements was applied. The oil-filled structure was applied to minimize the change of acoustic characteristics of the ring transducer. It was verified that the transmitting voltage response, underwater impedance, and beam pattern matched the finite element numerical simulation results well through an acoustic test. The difference in the transmitting voltage response between the measured and the simulated results is about 1.3 dB in cavity mode and about 0.3 dB in radial mode. The fabricated FFR transducer had a higher transmitting voltage response compared to the commercial transducer, but the diameter was reduced by about 17 %. From this study, it was confirmed that the feasibility of a single crystal-applied FFR transducer with compact size and high-power characteristics. The effectiveness of the performance prediction by simulation was also confirmed.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.11a
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• pp.266-269
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• 2010

급속한 정보 통신 기술의 발달로 인해 멀티미디어 재생 개발 기술들은 단순히 수동적으로 보고 듣는 재생 기술에서 벗어나 청취자 감성, 취향 등에 따라 보다 실감 있고 사용자가 능동적으로 재생할 수 있는 기술로 진화 하고 있다. 지금까지의 오디오 서비스는 음원 개발자 중심의 오디오 서비스, 즉 보컬 및 모든 악기가 믹스된 단일음원이기 때문에 사용자는 단순히 오디오 음원 개발자나 음반 제작사가 발매한 단일 음원을 일방적으로 수동적 청취할 수밖에 없다. 하지만 사용자 능동형 오디오 서비스에서는 사용자가 능동적으로 자신이 원하는 음악적 취향에 따라 능동적으로 각각의 객체 기반의 독립 음원을 선택, 감성에 따른 음원 효과 추가, 최적의 음원 청취 위치(Sweet Spot) 변경, 음원 및 스피커 재생 공간 및 위치 변경 재생 등을 할 수가 있다. 본 논문에서는 디지털 음원들을 입력받아 임의의 필터링을 실행하고, 사용자 음원 보정 정보, 출력 유닛의 공간적, 음향적 특성을 상위제어기로부터 입력받아 전신호경로 상에 디지털 신호처리 하여 출력신호를 생성하는 DSP 시스템 플랫폼 및 알고리즘에 관하여 소개한다.

• Journal of Korea Game Society
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• v.7 no.1
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• pp.53-58
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• 2007

In this paper, a game control system was constructed, in which a game was controlled by a remote joystick connected with a power line by the power line communication (PLC) method. The structure of the system was that the connection line between the remote joystick and a PC, and the PC and an audio play device was the home power line. And, the communication data rate between them was 2400 bps. The Polling communication technique was used for the PC to read the joystick's control commands, and to send some acoustic informations to the receiver PLC device. A game was programmed, in which an aircraft was moved according to the joystick's left, right, up, and, down direction, and was shooting its missile after the joystick's shooting button was pushed. The communication delay of about 100 msec between them didn't cause any big problem in playing the game.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.483-488
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• 2018

Far field input utterance is one of the major causes of performance degradation of speaker verification systems. In this study, we used teacher student learning framework to compensate for the performance degradation caused by far field utterances. Teacher student learning refers to training the student deep neural network in possible performance degradation condition using the teacher deep neural network trained without such condition. In this study, we use the teacher network trained with near distance utterances to train the student network with far distance utterances. However, through experiments, it was found that performance of near distance utterances were deteriorated. To avoid such phenomenon, we proposed techniques that use trained teacher network as initialization of student network and training the student network using both near and far field utterances. Experiments were conducted using deep neural networks that input raw waveforms of 4-channel utterances recorded in both near and far distance. Results show the equal error rate of near and far-field utterances respectively, 2.55 % / 2.8 % without teacher student learning, 9.75 % / 1.8 % for conventional teacher student learning, and 2.5 % / 2.7 % with proposed techniques.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.49-60
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• 1995

This paper is a study on the design and implementation of the Korean Text-to-Speech system which is used for a small and simple system. In this paper, a parameter synthesis method is chosen for speech syntheiss method, we use PARCOR(PARtial autoCORrelation) coefficient which is one of the LPC analysis. And we use phoneme for synthesis unit which is the basic unit for speech synthesis. We use PARCOR, pitch, amplitude as synthesis parameter of voice, we use residual signal, PARCOR coefficients as synthesis parameter of unvoice. In this paper, we could obtain the 60% intelligibility by using the residual signal as excitation signal of unvoiced sound. The result of synthesis experiment, synthesis of a word unit is available. The controlling of phoneme duration is necessary for synthesizing of a sentence unit. For setting up the synthesis system, PC 486, a 70[Hz]-4.5[KHz] band pass filter for speech input/output, amplifier, and TMS320C30 DSP board was used.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.