• International journal of advanced smart convergence
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• v.5 no.4
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• pp.21-25
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• 2016

Recent development in multi-channel surround is emerging in various formats to provide better stereoscopic and sound effects to consumers in recent broadcasting. The ability sound localize the sound sources in space is most considerable design factor on multi-channel surround system for human earing perception model. However, this paper propose the change of the sound localization according to the spacing of the speakers, which is not covered in the existing research focus on sound system design. Presently the sound system uses the position and number of the speakers to localize the sound. In the multi-channel surround environment, the proposed design uses the sound localization is caused by the directional characteristics of the speaker, the distance between the speakers and the distance between the listener and the speaker according to the directivity is required. The proposed design is simulated using virtual measurement with MATLAB simulation environment and performances are measured.

• Transactions of the Korean Society of Automotive Engineers
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• v.3 no.6
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• pp.23-29
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• 1995

In this study, Active noise controlis implemented with single channel and multi-channel adaptive algorithm in 3 dimensional reverberant enclosure sound. field, which occurrs in complicated acoustic mode. First, for the one case excited with the resonant frequency of an enclosure, a target of control and the other cases excited with band-pass filtered random noise(100~400Hz), it is implemented to control reducing interior noise of enclosure with single channel and realtime multi-channel adaptive algorithm for global noise reduction in enclosure.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.600-605
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• 2020

In this paper, we propose a sound event detection method using a multi-channel multi-scale neural networks for sound sensing home monitoring for the hearing impaired. In the proposed system, two channels with high signal quality are selected from several wireless microphone sensors in home. The three features (time difference of arrival, pitch range, and outputs obtained by applying multi-scale convolutional neural network to log mel spectrogram) extracted from the sensor signals are applied to a classifier based on a bidirectional gated recurrent neural network to further improve the performance of sound event detection. The detected sound event result is converted into text along with the sensor position of the selected channel and provided to the hearing impaired. The experimental results show that the sound event detection method of the proposed system is superior to the existing method and can effectively deliver sound information to the hearing impaired.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.81-84
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• 2004

In this paper, we propose a virtual sound localization algorithm which improves the sound localization accuracy and sound color preservation for two channel and multi-channel surround speaker layouts. In conventional CPP laws, the sound direction is different from the panning angle and the sound color is different from real sound source especially when the speakers are spread out widely. To overcome this drawback, we design a virtual sound localization algorithm using directional psychoacoustic criteria (DPC) and sound color compensator (SCC). The analysis results show that in the case of the proposed system, the sound direction is the same as the panning angle in the audible frequency range and the sound color is less deviated from a real sound source than the conventional CPP law. In addition, its performance is verified by means of subjective tests using a real sound source.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2003.05b
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• pp.96-102
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• 2003

Today, stage technology is developing highly by application of digital computer. Performance is composed of audio/video and acoustic technology takes very important position in field of stage technology. Generally speaking, four factors of sound are loudness, pitch, sound timbre and duration. Loudness depends on sound pressure level, yet partly related with spectrum and dulation. Pitch depends mainly on frequence and have a relation with sound pressure and duration. sound timbre depends strongly on spectrum and have a relation with frequence. In this paper, I designed a multi-microphone system which can used in broadcasting and performance stage with vicboss 200MHz-VHF wireless microphone and vicboss 900MHz-VHF wireless microphone. I also studied about multi-microphone which can use conveniently in the super play that needs many microphones. If this multi-microphone is prodused, we could expect better sound quality and a big progress in stereo recording technology.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.24-33
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• 2010

This paper proposes a technique of sound beamforming that can generate high-directive sound beams, and this paper also presents applications of the proposed algorithm to multi-channel 3D sound systems. The proposed algorithm consists of two phases: first, optimum weights maximizing a sound pressure level ratio between the target and control acoustic regions are designed, and later, the directivity of the pre-designed sound beam is iteratively enhanced by modifying the covariance matrix. The proposed method was evaluated under various situations, and the results showed that it could provide more focused sound beams than the conventional methods.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.18-24
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• 2010

In this paper, a new matrix decoding system using vector based Virtual Source Location Information (VSLI) is proposed as an alternative to the conventional Dolby Pro logic II/IIx system for reconstructing multi-channel output signals from matrix encoded two channel signals, Lt/Rt. This new matrix decoding system is composed of passive decoding part and active part. The passive part makes crude multi-channel signals using linear combination of the two encoded signals(Lt/Rt) and the active part enhances each channel regarding to the virtual source which is emergent in each inter channel. Since the virtual sources are related to the perceptual sound images in virtual sound field, the reconstructed multi-channel sound results in good dynamic perception and stable image localization. Moreover, the good channel separation is maintained with nonlinear trigonometric enhancing function.

• 한국HCI학회:학술대회논문집
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• 2007.02a
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• pp.706-712
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• 2007

국가간 연구목적으로 개설된 초고속 광 네트워크 기술의 발전은 새로운 응용 기술의 등장을 요구하고 있다. 고화질 저지연의 실감 협업 응용은 이러한 연구 목적에 부합할 뿐만 아니라 향후 커뮤니티 기반 응용 기술의 요구를 충족시킬 것으로 보인다. 본 논문에서는 실감 협업 응용 기술에 필요한 비압축 HD stereoscopic 영상 전송 시스템을 구성해 3D HD 영상을 사용자가 체감할 수 있도록 한다. 또한, 소프트웨어 기반 다채널 음성 재생을 다루며 실험을 통해 방향성 있는 협업 환경 구축의 가능성을 보여준다. 입체감 있는 미디어 재생을 위해 병렬 구조의 좌 우 송수신 시스템을 구축 후 stereoscopic 비압축 영상 전송을 수행하며, 좌 우 영상 세션간의 인터 미디어 동기화 기법의 설계방법을 제안한다. 음성 재생 소프트웨어는 ALSA를 이용하여 구현하였으며 가변 데이터 길이 및 프레임 손실로 인한 채널 뒤섞임(channel swapping)을 방지하기 위한 버퍼를 재생 모듈 전처리단에 추가하였다. 초고속 네트워크와 비압축 미디어 전송의 결합은 IP를 이용해 다채널 음성 지원의 실감 HDTV를 가능케 하므로 이를 유용하게 활용할 수 있는 사용 시나리오를 살펴본다.

• The Journal of the Acoustical Society of Korea
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• v.36 no.4
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• pp.267-272
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• 2017

In this paper, we propose an effective method of applying multichannel-audio feature values to GRNNs (Gated Recurrent Neural Networks) in polyphonic sound event detection. Real life sounds are often overlapped with each other, so that it is difficult to distinguish them by using a mono-channel audio features. In the proposed method, we tried to improve the performance of polyphonic sound event detection by using multi-channel audio features. In addition, we also tried to improve the performance of polyphonic sound event detection by applying a gated recurrent neural network which is simpler than LSTM (Long Short Term Memory), which shows the highest performance among the current recurrent neural networks. The experimental results show that the proposed method achieves better sound event detection performance than other existing methods.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.1
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• pp.71-77
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• 2005

In this paper we implement a headphone-based 5.1 channel 3-dimensional (3D) sound generation system using HRTF (Head Related Transfer Function). Each mono sound source in the 5.1 channel signal is localized on its virtual location by binaural filtering with corresponding HRTFs, and reverberation effect is added for spatialization. To reduce the computational burden, we reduce the number of taps in the HRTF impulse response and model the early reverberation effect with several tens of impulses extracted from the whole impulse sequences. We modified the spectrum of HRTF by weighing the difference of front-back spec01m to reduce the front-back confusion caused by non-individualized HRTF DB. In informal listening test we can confirm that the implemented 3D sound system generates live and rich 3D sound compared with simple stereo or 2 channel down mixing.