• Title/Summary/Keyword: Music Algorithm

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State of the Art Technology Trends and Case Analysis of Leading Research in Harmony Search Algorithm (하모니 탐색 알고리즘의 선도 연구에 관한 최첨단 기술 동향과 사례 분석)

  • Kim, Eun-Sung;Shin, Seung-Soo;Kim, Yong-Hyuk;Yoon, Yourim
    • Journal of the Korea Convergence Society
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    • v.12 no.11
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    • pp.81-90
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    • 2021
  • There are various optimization problems in real world and research continues to solve them. An optimization problem is the problem of finding a combination of parameters that maximizes or minimizes the objective function. Harmony search is a population-based metaheuristic algorithm for solving optimization problems and it is designed to mimic the improvisation of jazz music. Harmony search has been actively applied to optimization problems in various fields such as civil engineering, computer science, energy, medical science, and water quality engineering. Harmony search has a simple working principle and it has the advantage of finding good solutions quickly in constrained optimization problems. Especially there are various application cases showing high accuracy with a low number of iterations by improving the solution through the empirical derivative. In this paper, we explain working principle of Harmony search and classify the leading research in recent 3 years, review them according to category, and suggest future research directions. The research is divided into review by field, algorithmic analysis and theory, and application to real world problems. Application to real world problems is classified according to the purpose of optimization and whether or not they are hybridized with other metaheuristic algorithms.

Scrambling Technology using Scalable Encryption in SVC (SVC에서 스케일러블 암호화를 이용한 스크램블링 기술)

  • Kwon, Goo-Rak
    • Journal of Korea Multimedia Society
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    • v.13 no.4
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    • pp.575-581
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    • 2010
  • With widespread use of the Internet and improvements in streaming media and compression technology, digital music, video, and image can be distributed instantaneously across the Internet to end-users. However, most conventional Digital Right Management are often not secure and not fast enough to process the vast amount of data generated by the multimedia applications to meet the real-time constraints. The SVC offers temporal, spatial, and SNR scalability to varying network bandwidth and different application needs. Meanwhile, for many multimedia services, security is an important component to restrict unauthorized content access and distribution. This suggests the need for new cryptography system implementations that can operate at SVC. In this paper, we propose a new scrambling encryption for reserving the characteristic of scalability in MPEG4-SVC. In the base layer, the proposed algorithm is applied and performed the selective scambling. And it encrypts various MVS and intra-mode scrambling in the enhancement layer. In the decryption, it decrypts each encrypted layers by using another encrypted keys. Throughout the experimental results, the proposed algorithms have low complexity in encryption and the robustness of communication errors.

Development of Interactive Video Using Real-time Optical Flow and Masking (옵티컬 플로우와 마스킹에 의한 실시간 인터렉티브 비디오 개발)

  • Kim, Tae-Hee
    • The Journal of the Korea Contents Association
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    • v.11 no.6
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    • pp.98-105
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    • 2011
  • Recent advances in computer technologies support real-time image processing and special effects on personal computers. This paper presents and analyzes a real-time interactive video system. The motivation of this work is to realize an artistic concept that aims at transforming the timeline visual variations in a video of sea water waves into sound in order to provide an audience with an experience of overlapping themselves onto the nature. In practice, the video of sea water waves taken on a beach is processed using an optical flow algorithm in order to extract the information of visual variations between the video frames. This is then masked by the silhouette of an audience and the result is projected on a gallery space. The intensity information is extracted from the resulting video and translated into piano sounds accordingly. This work generates an interactive space realizing the intended concept.

Audio Listening Enhancement in Adverse Environment based on Loudness Restoration (라우드니스 복원에 기반한 잡음 환경에서의 오디오 청취 향상)

  • Pak, Junhyeong;Shin, Jong Won
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.12
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    • pp.210-216
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    • 2013
  • It is hard to listen to the music clearly in the presence of background noise. In this paper, a method that modifies the audio signal automatically to enhance the audio listening experience in adverse environment is proposed. Specifically, the method that amplifies the audio signal so that the perceived loudness of audio signal in each band becomes similar to that of the noiseless signal. The loudness perception model proposed by Moore et. al is utilized. Extending the previous work that is applied to speech reinforcement, the full band signal sampled at 48kHz is manipulated based on the loudness restoration principle. Moreover, based on the observation that the audio clarity is compromised even with loudness restored signal, a modification that intentionally boosts high frequency loudness more than lower band is also proposed. Experimental results showed that the proposed algorithm can enhance the audio listening experience in adverse environment.

A Content-based Audio Retrieval System Supporting Efficient Expansion of Audio Database (음원 데이터베이스의 효율적 확장을 지원하는 내용 기반 음원 검색 시스템)

  • Park, Ji Hun;Kang, Hyunchul
    • Journal of Digital Contents Society
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    • v.18 no.5
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    • pp.811-820
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    • 2017
  • For content-based audio retrieval which is one of main functions in audio service, the techniques for extracting fingerprints from the audio source, storing and indexing them in a database are widely used. However, if the fingerprints of new audio sources are continually inserted into the database, there is a problem that space efficiency as well as audio retrieval performance are gradually deteriorated. Therefore, there is a need for techniques to support efficient expansion of audio database without periodic reorganization of the database that would increase the system operation cost. In this paper, we design a content-based audio retrieval system that solves this problem by using MapReduce and NoSQL database in a cluster computing environment based on the Shazam's fingerprinting algorithm, and evaluate its performance through a detailed set of experiments using real world audio data.

A Two-Stage Bit Allocation Algorithm for MPEG-1 Audio Coding (MPEG-1 오디오 부호화를 위한 2단계 비트 할당 알고리듬)

  • 임창헌;천병훈
    • Journal of Korea Multimedia Society
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    • v.5 no.4
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    • pp.393-398
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    • 2002
  • The conventional bit allocation scheme for MPEG-1 audio encoding searches the subband with minimum MNR(mask-to-noise ratio) repetitively until its operation is completed, which occupies most of its total computational complexity. In this paper, as a computationally efficient approximation of it, we propose a new bit allocation scheme with a simple subband search and compare it with the existing schemes[1][2] in terms of the computational complexity and sound quality. For the performance comparison, we used the pop music signal contained in SQAM(sound quality assess material) CD from EBU. Simulation results show that the computational complexity of the proposed method is about 42% of that of the existing one in [1] and the sound quality difference in terms of MNR between the two schemes is within the 0.2 ㏈, for the case of using the layer II at the bit rate of 128 kbps.

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A Semantic Distance Measurement Model using Weights on the LOD Graph in an LOD-based Recommender System (LOD-기반 추천 시스템에서 LOD 그래프에 가중치를 사용한 의미 거리 측정 모델)

  • Huh, Wonwhoi
    • Journal of the Korea Convergence Society
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    • v.12 no.7
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    • pp.53-60
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    • 2021
  • LOD-based recommender systems usually leverage the data available within LOD datasets, such as DBpedia, in order to recommend items(movies, books, music) to the end users. These systems use a semantic similarity algorithm that calculates the degree of matching between pairs of Linked Data resources. In this paper, we proposed a new approach to measuring semantic distance in an LOD-based recommender system by assigning weights converted from user ratings to links in the LOD graph. The semantic distance measurement model proposed in this paper is based on a processing step in which a graph is personalized to a user through weight calculation and a method of applying these weights to LDSD. The Experimental results showed that the proposed method showed higher accuracy compared to other similar methods, and it contributed to the improvement of similarity by expanding the range of semantic distance measurement of the recommender system. As future work, we aim to analyze the impact on the model using different methods of LOD-based similarity measurement.

High Quality Multi-Channel Audio System for Karaoke Using DSP (DSP를 이용한 가라오케용 고음질 멀티채널 오디오 시스템)

  • Kim, Tae-Hoon;Park, Yang-Su;Shin, Kyung-Chul;Park, Jong-In;Moon, Tae-Jung
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.1
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    • pp.1-9
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    • 2009
  • This paper deals with the realization of multi-channel live karaoke. In this study, 6-channel MP3 decoding and tempo/key scaling was operated in real time by using the TMS320C6713 DSP, which is 32 bit floating-point DSP made by TI Co. The 6 channel consists of front L/R instrument, rear L/R instrument, melody, and woofer. In case of the 4 channel, rear L/R instrument can be replaced with drum L/R channel. And the final output data is generated as adjusted to a 5.1 channel speaker. The SOLA algorithm was applied for tempo scaling, and key scaling was done with interpolation and decimation in the time domain. Drum channel was excluded in key scaling by separating instruments into drums and non-drums, and in processing SOLA, high-quality tempo scaling was made possible by differentiating SOLA frame size, which was optimized for real-time process. The use of 6 channels allows the composition of various channels, and the multi-channel audio system of this study can be effectively applied at any place where live music is needed.

Evaluation of Antenna Pattern Measurement of HF Radar using Drone (드론을 활용한 고주파 레이다의 안테나 패턴 측정(APM) 가능성 검토)

  • Dawoon Jung;Jae Yeob Kim;Kyu-Min Song
    • Journal of Korean Society of Coastal and Ocean Engineers
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    • v.35 no.6
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    • pp.109-120
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    • 2023
  • The High-Frequency Radar (HFR) is an equipment designed to measure real-time surface ocean currents in broad maritime areas.It emits radio waves at a specific frequency (HF) towards the sea surface and analyzes the backscattered waves to measure surface current vectors (Crombie, 1955; Barrick, 1972).The Seasonde HF Radar from Codar, utilized in this study, determines the speed and location of radial currents by analyzing the Bragg peak intensity of transmitted and received waves from an omnidirectional antenna and employing the Multiple Signal Classification (MUSIC) algorithm. The generated currents are initially considered ideal patterns without taking into account the characteristics of the observed electromagnetic wave propagation environment. To correct this, Antenna Pattern Measurement (APM) is performed, measuring the strength of signals at various positions received by the antenna and calculating the corrected measured vector to radial currents.The APM principle involves modifying the position and phase information of the currents based on the measured signal strength at each location. Typically, experiments are conducted by installing an antenna on a ship (Kim et al., 2022). However, using a ship introduces various environmental constraints, such as weather conditions and maritime situations. To reduce dependence on maritime conditions and enhance economic efficiency, this study explores the possibility of using unmanned aerial vehicles (drones) for APM. The research conducted APM experiments using a high-frequency radar installed at Dangsa Lighthouse in Dangsa-ri, Wando County, Jeollanam-do. The study compared and analyzed the results of APM experiments using ships and drones, utilizing the calculated radial currents and surface current fields obtained from each experiment.

Indoor Environment Control System based EEG Signal and Internet of Things (EEG 신호 및 사물인터넷 기반 실내 환경 제어 시스템)

  • Jeong, Haesung;Lee, Sangmin;Kwon, Jangwoo
    • Journal of rehabilitation welfare engineering & assistive technology
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    • v.11 no.1
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    • pp.45-52
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    • 2017
  • EEG signals that are the same as those that have the same disabled people. So, the EEG signals are becoming the next generation. In this paper, we propose an internet of things system that controls the indoor environment using EEG signal. The proposed system consists EEG measurement device, EEG simulation software and indoor environment control device. We use data as EEG signal data on emotional imagination condition in a comfortable state and logical imagination condition in concentrated state. The noise of measured signal is removed by the ICA algorithm and beta waves are extracted from it. then, it goes through learning and test process using SVM. The subjects were trained to improve the EEG signal accuracy through the EEG simulation software and the average accuracy were 87.69%. The EEG signal from the EEG measurement device is transmitted to the EEG simulation software through the serial communication. then the control command is generated by classifying emotional imagination condition and logical imagination condition. The generated control command is transmitted to the indoor environment control device through the Zigbee communication. In case of the emotional imagination condition, the soft lighting and classical music are outputted. In the logical imagination condition, the learning white noise and bright lighting are outputted. The proposed system can be applied to software and device control based BCI.