• Title/Summary/Keyword: 오디오신호

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B-ISDN Signalling Protocol for Internet-Based Service (인터넷 서비스를 위한 B-ISDN 신호 프로토콜의 표준화 동향)

  • Kim, J.Y.;Joo, S.S.
    • Electronics and Telecommunications Trends
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    • v.13 no.6 s.54
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    • pp.83-93
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    • 1998
  • Best effort 서비스 품질만 지원하는 현재의 인터넷에 음성, 오디오 그리고 영상 통신 응용 서비스와 같은 새로운 멀티미디어 응용 서비스를 사용하려는 요구가 확대됨에 따라서 멀티미디어 서비스를 제공할 수 있는 인터넷의 필요성이 증대하고 있다. 또한 이러한 멀티미디어 서비스를 제공하기 위하여 서비스 품질(QoS)을 보장할 수 있는 통신 방식과 대량의 트래픽을 효과적으로 전달할 수 있는 메커니즘이 필요하게 되었다. 비동기 전송방식(ATM)은 이러한 멀티미디어 서비스를 인터넷에서 제공할 수 있는 최적의 통신 방식으로 고려되고 있는데, 이것은 ATM의 장점인 고속의 스위칭 기술과 논리적으로 VPI/VCI를 다중화 하는 기법, 그리고 유연한 서비스 품질 관리가 가능하기 때문이다. 본 고에서는 ATM 망에서 인터넷 서비스를 지원하기 위하여 결성된 ITU-T SG11의 Coordination 그룹인 Signalling Support of Internet-Based Applications(SoI) 회의 결과를 중심으로 하며 SoI의 표준화 연구 목표, B-ISDN 신호 프로토콜을 이용한 Long-lived 세션과 QoS에 민감한 세션의 인터넷 트래픽에 대한 ATM 연결 설정 절차 및 인터넷 세션 정보의 전달 방법 그리고 인터넷 서비스를 위한 멀티캐스팅 방법에 대하여 기술한다. 본 고의 목적은 인터넷 서비스 및 프로토콜을 지원하기 위하여 확장이 필요한 B-ISDN 신호 프로토콜의 기능을 명확히 기술하기 위한 것이다.

A study on digital sound reception systems for ships (선박용 디지털 음향수신장치 연구)

  • Kim, Hyungjong;Kim, Jeongchang
    • Journal of Advanced Marine Engineering and Technology
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    • v.38 no.9
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    • pp.1125-1130
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    • 2014
  • In this paper, we propose a sound reception system against surrounding noise for ships based on digital signal processing technologies. In order to suppress unwanted surrounding noises, a digital band-pass filter is designed, which the pass-band of the filter is between 70Hz to 820Hz. Also, we develope a sound direction indicating algorithm with 4 microphones. After filtering the audio signals from 4 microphones, the developed sound direction indicating algorithm can indicate 8 directions. In addition, we implement prototype board for the sound reception using a digital signal processor chip and audio codecs, and verify the proposed algorithm.

The Noise Influence of 4G Mobile Transmitter on Audio Devices (4G 휴대 단말기 송신에 의한 오디오 잡음 영향)

  • Yun, Hye-Ju;Lee, Il-Kyoo
    • Journal of Satellite, Information and Communications
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    • v.8 no.1
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    • pp.31-34
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    • 2013
  • This paper deals with the interfering audio noise caused by LTE(Long Term Evolution) UE(User Equipment) which is 4th generation mobile communications on audio devices. At first, we realized that the interfering signal of the LTE UE is determined by the transmit power of the LTE UE through analysis and measurement. Then, we performed to measure audio noise level according to the variation of transmitting power level and separation distance between the LTE UE and an audio device. As a result, it is required that minimum separation distance should be 25 cm and above in order to protect audio device from the interference noise of the LTE UE with the maximum transmit power level of 22 dBm.

Study on the Design of S/PDIF BC which Can Operate without PLL (PLL없이 동작하는 S/PDIF IC 설계에 관한 연구)

  • Park Ju-Sung;Kim Suk-Chan;Kim Kyoung-Soo
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.1
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    • pp.11-20
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    • 2005
  • In this paper, we deal with the research about a S/PDIF (Sony Philips Digital Interface) receiver which can operate without PLL (Phase Locked Loop) circuits. Although a S/PDIF receiver is used in most audio devices and audio processors in these days. yet there are only few domestic researches about S/PDIF. Currently used commercial DACs (Digital-to-Analog Converters) which can decode S/PDIF signals, have a PLL circuit inside them. The PLL makes it possible to extract clock information from S/PDIF digital signal and to synchronize a clock signal with input signals. But the PLL circuit makes many diffculties in designing the SOC (System On Chips) of VLSIs (Vew Large Scale Integrated Ciruits) because it is an "analog circuit". We proposed a S/PDIF receiver which doesn't have PLL circuits and only has Pure digital circuits. The key idea of the proposed S/PDIF receiver. is to use the ratio between a 16 MHz basic input clock and S/PDIF signals. After having decoded hundreds thousands S/PDIF inputs, it went to prove that a S/PDIF receiver can be designed with pure digital circuits and without any analog circuits such as PLL circuits. We have confidence that the proposed S/PDIF receiver can be used as an IP (Intellectual Property) for the SOC design of the digital circuits.

A Method to Express Audio Binary Files by Color QR Codes and Its Application (오디오 바이너리 파일을 컬러 QR코드로 표현하는 방법과 그 응용)

  • Lee, Choong Ho
    • Journal of the Institute of Convergence Signal Processing
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    • v.19 no.2
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    • pp.47-53
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    • 2018
  • This paper proposes a method to express an MP3 audio file by a series of color QR codes which can be printed on the paper. Moreover, the method can compress the data considerably. Firstly, an MP3 file is divided into many small files which have maximum capacity of binary file of a QR code. Secondly, the multiple files are converted to multiple black-and-white QR codes. Lastly, every three QR codes are combined into color QR codes. When combining, each of three black-and-white QR codes are regarded as red, green, blue components respectively. In this method, the areas of a color QR code where two QR codes are overlapped are expressed by the colors Cyan, Magenta and Yellow. And the areas where three components are overlapped are expressed by white color. Contrarily, the areas that no components are overlapped are expressed by white color. Experimentation result shows that an MP3 file with 8.5MB the original MP3 files are compressed with the compression rate around 15.7. This method has the advantage that can be used in the environments that the internet access is impossible.

Sound recognition and tracking system design using robust sound extraction section (주변 배경음에 강인한 구간 검출을 통한 음원 인식 및 위치 추적 시스템 설계)

  • Kim, Woo-Jun;Kim, Young-Sub;Lee, Gwang-Seok
    • The Journal of the Korea institute of electronic communication sciences
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    • v.11 no.8
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    • pp.759-766
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    • 2016
  • This paper is on a system design of recognizing sound sources and tracing locations from detecting a section of sound sources which is strong in surrounding environmental sounds about sound sources occurring in an abnormal situation by using signals within the section. In detection of the section with strong sound sources, weighted average delta energy of a short section is calculated from audio signals received. After inputting it into a low-pass filter, through comparison of values of the output result, a section strong in background sound is defined. In recognition of sound sources, from data of the detected section, using an HMM(: Hidden Markov Model) as a traditional recognition method, learning and recognition are realized from creating information to recognize sound sources. About signals of sound sources that surrounding background sounds are included, by using energy of existing signals, after detecting the section, compared with the recognition through the HMM, a recognition rate of 3.94% increase is shown. Also, based on the recognition result, location grasping by using TDOA(: Time Delay of Arrival) between signals in the section accords with 97.44% of angles of a real occurrence location.

Voice signal transmission using VLC communication (VLC 통신을 이용한 음성신호 전송)

  • Kim, Byun-Gon;Kim, Myung-Soo;Jeong, Kyeong-Taek;kwon, Oh-Shin
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2017.05a
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    • pp.656-659
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    • 2017
  • In this paper, we propose a digital method for transmitting audio signals using LED visible light communication system. In the proposed method, we compare the method for transmitting audio signal in analog signal and the method for transmitting by digital signal. When amplifying the audio sound and transmitting the analog signal using the LED visible light communication, attenuation corresponding to the transmission distance occurs, and there is a disadvantage that it is noisy. In order to overcome this, we propose a method for transmitting digital audio signals. The proposed method has the advantage of reducing the influence of noise, but it turned out that it is affected much by the LED blinking speed. Various methods to overcome this need to be continuously studied.

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Acceleration signal-based haptic texture recognition according to characteristics of object surface material using conformer model (Conformer 모델을 이용한 물체 표면 재료의 특성에 따른 가속도 신호 기반 햅틱 질감 인식)

  • Hyoung-Gook Kim;Dong-Ki Jeong;Jin-Young Kim
    • The Journal of the Acoustical Society of Korea
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    • v.42 no.3
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    • pp.214-220
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    • 2023
  • In this paper, we propose a method to improve texture recognition performance from haptic acceleration signals representing the texture characteristics of object surface materials by using a Conformer model that combines the advantages of a convolutional neural network and a transformer. In the proposed method, three-axis acceleration signals generated by impact sound and vibration are combined into one-dimensional acceleration data while a person contacts the surface of the object materials using a tool such as a stylus , and the logarithmic Mel-spectrogram is extracted from the haptic acceleration signal similar to the audio signal. Then, Conformer is applied to the extracted the logarithmic Mel-spectrogram to learn main local and global frequency features in recognizing the texture of various object materials. Experiments on the Lehrstuhl für Medientechnik (LMT) haptic texture dataset consisting of 60 materials to evaluate the performance of the proposed model showed that the proposed method can effectively recognize the texture of the object surface material better than the existing methods.

A Complexity Reduction Method of MPEG-4 Audio Lossless Coding Encoder by Using the Joint Coding Based on Cross Correlation of Residual (여기신호의 상관관계 기반 joint coding을 이용한 MPEG-4 audio lossless coding 인코더 복잡도 감소 방법)

  • Cho, Choong-Sang;Kim, Je-Woo;Choi, Byeong-Ho
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.47 no.3
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    • pp.87-95
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    • 2010
  • Portable multi-media products which can service the highest audio-quality by using lossless audio codec has been released and the international lossless codecs, MPEG-4 audio lossless coding(ALS) and MPEG-4 scalable lossless coding(SLS), were standardized by MPEG in 2006. The simple profile of MPEG-4 ALS, it supports up to stereo, was defined by MPEG in 2009. The lossless audio codec should have low-complexity in stereo to be widely used in portable multi-media products. But the previous researches of MPEG-4 ALS have focused on an improvement of compression ratio, a complexity reduction in multi-channels coding, and a selection of linear prediction coefficients(LPCs) order. In this paper, the complexity and compression ratio of MPEG-4 ALS encoder is analyzed in simple profile of MPEG-4 ALS, the method to reduce a complexity of MPEG-4 ALS encoder is proposed. Based on an analysis of complexity of MPEG-4 ALS encoder, the complexity of short-term prediction filter of MPEG-4 ALS encoder is reduced by using the low-complexity filter that is proposed in previous research to reduce the complexity of MPEG-4 ALS decoder. Also, we propose a joint coding decision method, it reduces the complexity and keeps the compression ratio of MPEG-4 ALS encoder. In proposed method, the operation of joint coding is decided based on the relation between cross-correlation of residual and compression ratio of joint coding. The performance of MPEG-4 ALS encoder that has the method and low-complexity filter is evaluated by using the MPEG-4 ALS conformance test file and normal music files. The complexity of MPEG-4 ALS encoder is reduced by about 24% by comparing with MPEG-4 ALS reference encoder, while the compression ratio by the proposed method is comparable to MPEG-4 ALS reference encoder.

Multi-modal Emotion Recognition using Semi-supervised Learning and Multiple Neural Networks in the Wild (준 지도학습과 여러 개의 딥 뉴럴 네트워크를 사용한 멀티 모달 기반 감정 인식 알고리즘)

  • Kim, Dae Ha;Song, Byung Cheol
    • Journal of Broadcast Engineering
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    • v.23 no.3
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    • pp.351-360
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    • 2018
  • Human emotion recognition is a research topic that is receiving continuous attention in computer vision and artificial intelligence domains. This paper proposes a method for classifying human emotions through multiple neural networks based on multi-modal signals which consist of image, landmark, and audio in a wild environment. The proposed method has the following features. First, the learning performance of the image-based network is greatly improved by employing both multi-task learning and semi-supervised learning using the spatio-temporal characteristic of videos. Second, a model for converting 1-dimensional (1D) landmark information of face into two-dimensional (2D) images, is newly proposed, and a CNN-LSTM network based on the model is proposed for better emotion recognition. Third, based on an observation that audio signals are often very effective for specific emotions, we propose an audio deep learning mechanism robust to the specific emotions. Finally, so-called emotion adaptive fusion is applied to enable synergy of multiple networks. The proposed network improves emotion classification performance by appropriately integrating existing supervised learning and semi-supervised learning networks. In the fifth attempt on the given test set in the EmotiW2017 challenge, the proposed method achieved a classification accuracy of 57.12%.