• Title/Summary/Keyword: Audio signal analysis

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Design of A Downlink Power Control Scheme in Unequal Error Protection Multi-Code CDMA Mobile Medicine System

  • Lin, Chin-Feng;Lee, Hsin-Wang;Hung, Shih-Ii;Li, Ching-Yi
    • Proceedings of the Korea Society of Information Technology Applications Conference
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    • 2005.11a
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    • pp.335-338
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    • 2005
  • In this paper, we propose a downlink power control scheme to apply in the unequal error protection multi-code CDMA mobile medicine system. The mobile medicine system contains (i) blood pressure and body temperature measurement value, (ii) ECG medical signals measured by the electrocardiogram device, (iii) mobile patient's history, (iv) G.729 audio signal, MPEG-4 CCD sensor video signal, and JPEG2000 medical image. By the help of the multi-code CDMA spread spectrum communication system with downlink power control scheme and unequal error protection strategy, it is possible to transmit mobile medicine media and meet the quality of service. Numerical analysis and simulation results show that the system is a well transmission platform in mobile medicine.

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A Study on Signal Analysis of Korean Traditional Music Instrument, Kayakeum and Piri (국악 악기 가야금과 피리의 신호 분석에 관한 연구)

  • Lee Sang-Min;Lee Jong-Seok;Lee Kwang-Hyung
    • Proceedings of the Acoustical Society of Korea Conference
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    • spring
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    • pp.247-250
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    • 1999
  • Like any other music, Korean traditional music make a beautiful compound melody of many music instruments. In this paper, we separate melody especially played by two instruments, that is Kayakeum, Piri(Korean pipe) analysing each audio signal. Kayakeum, Piri have a unique frequency component for each sound height. Therefore each melody of them can be expressed into each sheet of notation separately and MIDI codes. We expect that this paper will benefit all the people studying and instructing Korean music.

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Signal Quality Enhancement using Perceptual Convolutional Noise Suppression (지각형 컨벌루션 잡음 제어를 통한 음질 개선 방법)

  • 김헌중;한헌수;홍민철;차형태
    • Journal of Broadcast Engineering
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    • v.8 no.1
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    • pp.11-18
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    • 2003
  • In this paper, we introduce a novel signal quality enhancement algorithm with a perceptual interference analysis and perceptual convolutional noise suppression. A perceptual convolutional noise is reflected in the audible disturbance that can still be recognized after the additional noise suppression and tonality change which is caused by the noise energy excitation. The enhancement system is organized with a perceptual additional noise suppression part and a perceptual convolutional noise suppression part. Experimental results show that these two parts have an equivalent quality enhancement performance.

A Threshold Adaptation based Voice Query Transcription Scheme for Music Retrieval (음악검색을 위한 가변임계치 기반의 음성 질의 변환 기법)

  • Han, Byeong-Jun;Rho, Seung-Min;Hwang, Een-Jun
    • The Transactions of The Korean Institute of Electrical Engineers
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    • v.59 no.2
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    • pp.445-451
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    • 2010
  • This paper presents a threshold adaptation based voice query transcription scheme for music information retrieval. The proposed scheme analyzes monophonic voice signal and generates its transcription for diverse music retrieval applications. For accurate transcription, we propose several advanced features including (i) Energetic Feature eXtractor (EFX) for onset, peak, and transient area detection; (ii) Modified Windowed Average Energy (MWAE) for defining multiple small but coherent windows with local threshold values as offset detector; and finally (iii) Circular Average Magnitude Difference Function (CAMDF) for accurate acquisition of fundamental frequency (F0) of each frame. In order to evaluate the performance of our proposed scheme, we implemented a prototype music transcription system called AMT2 (Automatic Music Transcriber version 2) and carried out various experiments. In the experiment, we used QBSH corpus [1], adapted in MIREX 2006 contest data set. Experimental result shows that our proposed scheme can improve the transcription performance.

A Study on Analysis and 3D Web Environment for the Treatment Alcoholism (알코중독 치료를 위한 Web 환경 시스템과 분석에 대한 연구)

  • Paek, Seung-Eun
    • The Journal of Information Technology
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    • v.9 no.1
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    • pp.9-19
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    • 2006
  • Medications or conitive-behavior methods have been mainly used as a treatment of alcoholism. lately the virtualy technology has been applied to the kink of alcoholic disorders. A virtual environment makes him having avility to over come the drink. In this study, we were implemented by making panorama images and 3D object modules using 3D MAX, VRML, JAVA. And the BAR stimulator that composed with a position sensor, head mount display, and audio system, is suggested. To illustrate the physiological difference between a person who has a alcoholism and without a liquor bottle, heart rate was measured during experiment, and also measured a person's HR after the virtual reality training. The system measures the Physiological signals such as ECG, Temperature, analyzes those data automatically. The system has two parts, one is physiological data acquisition part which gets the body signal, and the other one is mobile nuit which includes signal processing and transmission functions, And Bluetooth allows two parts to communicate with each other. we demonstrated the subjective effectiveness of virtual reality psychotherapy through the clinical experiment.

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Time-Synchronization Method for Dubbing Signal Using SOLA (SOLA를 이용한 더빙 신호의 시간축 동기화)

  • 이기승;지철근;차일환;윤대희
    • Journal of Broadcast Engineering
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    • v.1 no.2
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    • pp.85-95
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    • 1996
  • The purpose of this paper Is to propose a dubbed signal time-synchroniztion technique based on the SOLA(Synchronized Over-Lap and Add) method which has been widely used to modify the time scale of speech signal. In broadcasting audio recording environments, the high degree of background noise requires dubbing process. Since the time difference between the original and the dubbed signal ranges about 200mili seconds, process is required to make the dubbed signal synchronize to the corresponding image. The proposed method finds he starting point of the dubbing signal using the short-time energy of the two signals. Thereafter, LPC cepstrum analysis and DTW(Dynamic Time Warping) process are applied to synchronize phoneme positions of the two signals. After determining the matched point by the minimum mean square error between orignal and dubbed LPC cepstrums, the SOLA method is applied to the dubbed signal, to maintain the consistency of the corresponding phase. Effectiveness of proposed method is verified by comparing the waveforms and the spectrograms of the original and the time synchronized dubbing signal.

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Analysis of Sound Transmitting System using Power line Communication Technique (전력선을 이용한 음향전달 시스템의 구성 및 특성 분석)

  • Kim, Ho-Soo;Lee, Myung-Sub;Koo, Kyung-Wan;Han, Sang-Ok
    • Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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    • v.18 no.3
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    • pp.128-134
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    • 2004
  • This paper presents a result of sound transmitting system with power line communication technique. Sound transmitting system is a transmitter which transmits modulated audio signal to power to power line and receiver which is capable of detecting it with earphone or speaker. It has been evaluated with the frequency characteristics and spectrum analysis. And, from the result of evaluation on the developed system, we confirmed the superior sound transmitting characteristics, and the possibility of application on a language laboratory.

A tudy on the TV Microphonic Phenomenon (TV 마이크로포닉 현상에 관한 연구)

  • 성길주;윤경렬;이재응;이수훈;임진수
    • Journal of KSNVE
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    • v.5 no.1
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    • pp.123-132
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    • 1995
  • The microphonic phenomenon in TV(television) is a phenomenon that a stained pattern locally appears in the screen or moves like waves. This can be observed when audio signal of TV has specific frequencies under loud volume of sound. In this study, microphonic phenomenon has been investigated, and two practical ways of circumventing this has been proposed. Based on modal analysis of several TV parts(Cathod Ray Tube, shadowmask, etc.), it was proved that the microphonic phenomenon is caused by the resonance of the shadow mask. One of the proposed ways to circumvent this phenonenon is increasing the thickness of the frame, the other is removing the middle welding points between the frame and the shadow mask. The effects of these modifications are evaluated by the finite element analysis, and the results show that the magnitude of vibration of shadow mask reduced by 10 - 20dB, which is large enough to provent microphonic phenomenon even under maximum level of sound volume.

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Automatic measurement of voluntary reaction time after audio-visual stimulation and generation of synchronization signals for the analysis of evoked EEG (시청각자극 후의 피험자의 자의적 반응시간의 자동계측과 유발뇌파분석을 위한 동기신호의 생성)

  • 김철승;엄광문;손진훈
    • Science of Emotion and Sensibility
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    • v.6 no.4
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    • pp.15-23
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    • 2003
  • Recently, there have been many attempts to develop BCI (brain computer interface) based on EEG (electroencephalogram). Measurement and analysis of EEG evoked by particular stimulation is important for the design of brain wave pattern and interface of BCI. The purpose of this study is to develop a general-purpose system that measures subject's reaction time after audio-visual stimulation which can work together with any other biosignal measurement systems. The entire system is divided into four modules, which are stimulation signal generation, reaction time measurement, evoked potential measurement and synchronization. Stimulation signal generation module was implemented by means of Flash. Measurement of the reaction time (the period between the answer request and the subject reaction) was achieved by self-made microcontroller system. EEG measurement was performed using the ready-made hardware and software without any modification. Synchronization of all modules was achieved by, first, the black-and-white signals on the stimulation screen synchronized with the problem presentation and the answer request, second, the photodetectors sensing the signals. The proposed method offers easy design of purpose-specific system only by adding simple modules (reaction time measurement, synchronization) to the ready-made stimulation and EEG system, and therefore, it is expected to accelerate the researches requiring the measurement of evoked response and reaction time.

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Analysis of Power Saving Factor for a DVS Based Multimedia Processor (DVS 기반 멀티미디어 프로세서의 전력절감율 분석)

  • Kim Byoung-Il;Chang Tae-Gyu
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.42 no.1
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    • pp.95-100
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    • 2005
  • This paper proposes a DVS method which effectively reduces the power consumption of multimedia signal processor. Analytic derivations of effective range of its power saving factor are obtained with the assumption of a Gaussian distribution for the frame-based computational burden of the multimedia processor. A closed form equation of the power saving factor is derived in terms of the mean-standard deviation of the distribution. An MPEG-2 video decoder algorithm and AAC encoder algorithm are tested on ARM9 RISC processor for the experimental verification of the power saying of the proposed DVS approach. The experimental results with diverse MPEG-2 video and audio files show 50~30% power saving factor and show good agreement with those of the analytically derived values.