• Title/Summary/Keyword: audio coding

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The Effect of e-Learning Contents' Information Presentation Method on Teaching Presence and Academic Achievement (e-러닝 콘텐츠의 정보제시방식이 교수실재감 및 학업성취도에 미치는 효과)

  • Kim, Jinha;Kim, Kyunghee;Lee, Seongju
    • The Journal of Korean Association of Computer Education
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    • v.22 no.3
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    • pp.79-87
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    • 2019
  • This study examined the effect of e-learning contents with different dual-coding, media-richness, and cognitive-load degree on learning. To do so, after dividing summary and explanation presentation methods in e-learning contents according to information's quantity and kind, the effects on teaching presence and academic achievement were examined. The summary presentation method was produced as text type and text+illustration type and the explanation presentation method as audio type and audio+video type. The results of this study are as follows. First, in the summary method, the text+illustration type had significantly higher teaching presence than text type. Second, in the explanation method, the audio type was found to be significantly higher than the audio+video type. Third, the interaction between the summary method and explanation method was found to be significant in teaching presence and academic achievement.

Improved Synthesis Method of Negative Inter-channel Correlation Parameter Based on Anti-phase Primary Component (반위상 주요성분에 기반을 둔 개선된 음수 채널간 상관도 파라미터 합성 기법)

  • Hyun, Dong-Il;Lee, Seok-Pil;Park, Young-Cheol;Youn, Dae-Hee
    • The Journal of the Acoustical Society of Korea
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    • v.31 no.6
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    • pp.410-418
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    • 2012
  • Parametric stereo(PS) and MPEG surround(MPS) are major spatial audio coding(SAC) tools. In this paper, the problem of the inter-channel correlation(ICC) synthesis in the conventional SAC is analyzed. Conventional methods assume that ambient components mixed to two output channels are anti-phased, while the primary components are assumed to be in-phased. This assumption can cause excessive ambient mixing for a negative-valued ICC. As a remedy to this problem, we propose a new ICC synthesis method based on an assumption that the primary components are anti-phased each other for a negative ICC. The proposed method is also applied to the approximation which works in practice. The performance of the proposed method was evaluated by computer simulations and the subjective listening tests verified that the proposed method is effective in not only headphones but also loudspeakers playback.

Quality Improvement of Karaoke Mode in SAOC using Cross Prediction based Vocal Estimation Method (교차 예측 기반의 보컬 추정 방법을 이용한 SAOC Karaoke 모드에서의 음질 향상 기법에 대한 연구)

  • Lee, Tung Chin;Park, Young-Cheol;Youn, Dae Hee
    • The Journal of the Acoustical Society of Korea
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    • v.32 no.3
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    • pp.227-236
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    • 2013
  • In this paper, we present a vocal suppression algorithm that can enhance the quality of music signal coded using Spatial Audio Object Coding (SAOC) in Karaoke mode. The residual vocal component in the coded music signal is estimated by using a cross prediction method in which the music signal coded in Karaoke mode is used as the primary input and the vocal signal coded in Solo mode is used as a reference. However, the signals are extracted from the same downmix signal and highly correlated, so that the music signal can be severely damaged by the cross prediction. To prevent this, a psycho-acoustic disturbance rule is proposed, in which the level of disturbance to the reference input of the cross prediction filter is adapted according to the auditory masking property. Objective and subjective test were performed and the results confirm that the proposed algorithm offers improved quality.

Enhancement of SBR for Speech Signal Using Adaptive Noise Floor Level (가변 잡음 레벨을 이용한 음성신호에 대한 SBR 성능 항상 기술)

  • Lee, Se-Won;Oh, Seoung-Jun;Ahn, Chang-Beom;Lee, Tae-Jin;Kang, Kyoung-Ok;Park, Ho-Chong
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.2
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    • pp.148-154
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    • 2009
  • In audio coding, SBR technology synthesizes the high-bands using patched time-frequency information from low-bands and the correction parameters, Since SBR transmits only correction parameters for high-bands, it provides a low-rate coding of high-bands, and is used as a core module of MPEG-4 HE-AAC, SBR was originally designed for audio signal and its performance for speech signal tends to decrease, and the major reason is an excessive noise floor in high-bands which is caused by incorrect tonality computation, In this paper, a new method to determine noise floor level in an adaptive fashion according to the speech characteristics is proposed in order to solve the problem of SBR for speech signal, The proposed method maintains the compatibility with the standard SBR, and the subjective performance evaluation shows that the proposed method improves the SBR performance especially for male speech signal compared with the standard SBR.

An Exploratory Approach to Textile Designer's Cognition Model -focused on the Stage of Motif Development- (텍스타일 디자이너의 인지 모형에 대한 탐색적 접근 -모티브 개발 단계를 중심으로-)

  • 송승근;이주현
    • Science of Emotion and Sensibility
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    • v.6 no.1
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    • pp.55-62
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    • 2003
  • This study was an exploratory approach to the cognitive model of textile designers on the stage of motif development in textile design process. Prior to the main research, several previous studies adopting methods of video/audio protocol analysis were reviewed. On the basis of the review, the categories of design action were derived as an analysis frame by application of top-down access method, meanwhile the sub-groups of each category of design action were identified through a bottom-up access method. To summarize the research result, total three categories of textile design action appeared based on the theory of ‘Human processor’ model : ‘motor action’, ‘perceptual action’ and 'cognitive action'. In next, a new coding scheme suitably explaining these three categories of fertile design action was developed. Finally, a cognitive model of textile designer on the stage of motif development, employing the new coding scheme, was suggested in this study.

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Method of scalable video application in the advanced T-DMB (지상파 DMB 고도화 망에서의 스케일러블 비디오 부호화 기술)

  • Jun, Dong-San;Kwak, Sang-Min;Lim, Hyung-Soo;Choi, Hae-Chul;Kim, Jae-Gon;Lim, Jong-Soo;Hong, Jin-Woo
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.44 no.1
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    • pp.1-9
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    • 2007
  • Digital Multimedia Broadcasting is the next generation broadcasting service which enables various digital multimedia contents, i.e., audio and video, and data access for mobile users. However, due to the bandwidth limitation, the spatial resolution is limited to CIF(Common Interleaved Frame). The Advanced Terrestrial DMB (AT-DMB) secures additional bandwidth by adopting hierarchical modulation transmission technology and provides high data rate and quality for mobile multimedia broadcasting services with scalable video coding(SVC). This paper proposes scalable video coding technology for AT-DMB which enables high quality mobile multimedia broadcasting services that exceeds current DMB service's quality and contents capability.

Cooperative Diversity using Cyclic Delay for OFDM systems (OFDM 시스템을 위한 순환 지연을 사용하는 협력 다이버시티 기법)

  • Lee, Dong-Woo;Jung, Young-Seok;Lee, Jae-Hong
    • Journal of Broadcast Engineering
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    • v.13 no.2
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    • pp.172-178
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    • 2008
  • Orthogonal Frequency Division Multiplexing (OFDM) is one of the most promising technologies for high data rate wireless communications. OFDM has been adopted in wireless standards such as digital audio/video broadcasting. The combination of OFDM and cooperative diversity techniques can provide the diversity gain and/or increased capacity. In this paper, the cooperative coding using cyclic delay diversity (CDD) for multiuser OFDM systems is introduced. To improve the beneficial effects of relays's cooperation, CDD is adopted in cooperative transmission of relays. Simulation results show the bit error rate (BER) for various consideration. The proposed scheme provides improved performance compared to delay.

Feature Parameter Extraction and Analysis in the Wavelet Domain for Discrimination of Music and Speech (음악과 음성 판별을 위한 웨이브렛 영역에서의 특징 파라미터)

  • Kim, Jung-Min;Bae, Keun-Sung
    • MALSORI
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    • no.61
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    • pp.63-74
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    • 2007
  • Discrimination of music and speech from the multimedia signal is an important task in audio coding and broadcast monitoring systems. This paper deals with the problem of feature parameter extraction for discrimination of music and speech. The wavelet transform is a multi-resolution analysis method that is useful for analysis of temporal and spectral properties of non-stationary signals such as speech and audio signals. We propose new feature parameters extracted from the wavelet transformed signal for discrimination of music and speech. First, wavelet coefficients are obtained on the frame-by-frame basis. The analysis frame size is set to 20 ms. A parameter $E_{sum}$ is then defined by adding the difference of magnitude between adjacent wavelet coefficients in each scale. The maximum and minimum values of $E_{sum}$ for period of 2 seconds, which corresponds to the discrimination duration, are used as feature parameters for discrimination of music and speech. To evaluate the performance of the proposed feature parameters for music and speech discrimination, the accuracy of music and speech discrimination is measured for various types of music and speech signals. In the experiment every 2-second data is discriminated as music or speech, and about 93% of music and speech segments have been successfully detected.

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The Determinants and Barriers of Outsourcing Third-Party Online Delivery: Perspectives of F&B Entrepreneurs in Malaysia

  • SIN, Kit-Yeng;LO, May-Chiun;MOHAMAD, Abang Azlan
    • The Journal of Asian Finance, Economics and Business
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    • v.8 no.5
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    • pp.979-986
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    • 2021
  • Online food delivery and food delivery apps have continued to grow exponentially in Malaysia. Fundamental aspects in entrepreneurship of the food and beverage industry, such as knowledge and attitudes towards online food delivery services being outsourced, have yet to be extensively recognized. The present study intends to explore this area of subject matter within the Malaysian context by using behavioral reasoning theory. The actual interview for this study took place in May 2020, and 14 interviews had been carried out. All interviews were audio-recorded with the consent from the respondents for reference purposes and subsequently transcribed verbatim. The transcripts were then checked against audio records. Content analysis was used to analyze the transcripts by focusing on n frequency counts and coding of themes. A qualitative method has been adopted by employing an interview to elicit the perspectives of entrepreneurs from Sarawak on the determinants and barriers in outsourcing online food delivery services. Results indicate that high potential in revenue, broad exposure to reach customers, convenience, and provision of job opportunities are the four factors that determine to outsource. In contrast, food quality maintenance, trustworthiness, high cost incurred, and consumer technology resistance are four factors that serve as barriers towards outsourcing third-party online delivery.

An Optimization on the Psychoacoustic Model for MPEG-2 AAC Encoder (MPEG-2 AAC Encoder의 심리음향 모델 최적화)

  • Park, Jong-Tae;Moon, Kyu-Sung;Rhee, Kang-Hyeon
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.38 no.2
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    • pp.33-41
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    • 2001
  • Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.

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