• Title/Summary/Keyword: 서브밴드

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Subplots and Double Sound in the Film, Sweet Smell of Success (영화 <성공의 달콤한 향기>의 서브플롯과 더블 사운드)

  • Shin, Sa-Bin
    • The Journal of the Korea Contents Association
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    • v.22 no.6
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    • pp.273-282
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    • 2022
  • The narrative of the film, Sweet Smell of Success has a multi-layered structure. In Clifford Odets's scenario work, numerous lines for the main plot and subplots repeated a cycle of creation, decomposition, deletion, and modification to enhance the density of the story. As a result, whenever an actor spoke a line or paused, an action or an event that would trigger another line was created, giving depth and persuasiveness to the character's performance. The music of Sweet Smell of Success has multi-layered elements. The non-diegetic music was covered by the orchestral pieces performed by Elmer Bernstein's big band orchestra and the jazz pieces performed by Fred Katz's combo band. The diegetic music was mostly covered by the jazz pieces performed by the Chico Hamilton Quintet. The practical task of the film music was to reinforce or supplement the effect of the narrative driver, and the additional task was to realize the estrangement effect and the aesthetics of stagnation. The possibility and significance of intertextuality of subplots and double sound of this film are not simply confined to the limits of the film noir genre.

A Study on the Pitch Extraction Improvement Using LSP for the Synthesis of High Speech Quality (고음질 음성합성을 위한 LSP를 이용한 피치검출 성능향상에 관한 연구)

  • Seo, Ji-Ho;Kim, Jong-Kuk;Bae, Myung-Jin
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.1
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    • pp.69-75
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    • 2010
  • In this paper, the pitch is detected after the elimination of formant ingredients by flattening the spectrum in frequency domain. In order to remove impact of formant and transition frequency in the signal spectrum, formant envelop is made by linear interpolation with any points each sub-band and the spectrum of speech signal is compensated by the reverse of the envelop interpolated linearly after we divide frequency band into several segment based on LSP and detect the points. The experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum, Lifter methods. The method reduced the gross error rate 1.30% than the LPC method which appeared a good performance except the proposed method. Also, the proposed method showed low error rate in noise environment.

A New Document Codec System based on Wavelet Lifting and Bitplane Coding (웨이블릿 리프팅과 비트평면 부호화에 기반한 새로운 문서 코덱 시스템)

  • 이호석
    • Journal of Korea Multimedia Society
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    • v.6 no.5
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    • pp.805-815
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    • 2003
  • In this paper, we present the development of document compression codec using segmentation, wavelet lifting and bitplane coding. We use the segmentation to preserve the text appearance. We performed integer-to-integer wavelet lifting and also performed bitplane subblock coding for document compression. We acquired a high compression ratio and an efficient compression by encoding only the significant subblocks in the bitplane subblock coding. We also implemented scalar quantization by subband-oriented bit shifting. The system performs color conversion and downsampling before wavelet lifting and also performs graycode conversion and quantization before subblock coding. In the experiment, we show the performances of the system by presenting the high compression ratios and high PSNR values.

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Multi-band CIC Filter based Low Complexity Spectrum Sensing Method (다중밴드 CIC 필터에 기반한 저복잡도 스펙트럼 센싱 기법)

  • Lee, Su-Bok;Choi, Joo-Pyoung;Lee, Won-Cheol
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.10C
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    • pp.992-1000
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    • 2009
  • Available electric wave resources have been limited than frequency demands that increase with the development of radio communication. According to FCC's (Federal Communication Commission) report, 70% of allocated spectrum was unused. This means that the lack phenomenon of electronic wave resources is ineffectively managed compare with the finiteness of frequency resources. According to the recognition of unused frequency within allocated band, this progress has been investigating to identify the unused TV channel and provide radio communication services. This paper proposed the sensing algorithm that efficiently recognizes the frequency resources which does not use and shares with the licensed user through using the multi-band CIC (Cascaded Integrator Comb) filter based on polyphase filter bank. The simulation results verifies that the proposed scheme can obtain the similar performance of variance and relatively low complexity to the existing scheme.

A Study on the Pitch Alteration Technique by Sub-band Linear Approximation in Spectrum (서브밴드 선형근사에 의한 피치변경법에 관한 연구)

  • 김영규;김봉영;배명진
    • Proceedings of the IEEK Conference
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    • 2003.07e
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    • pp.2423-2426
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    • 2003
  • 음성합성은 합성방식에 따라 파형부호화법, 신호원부호화법, 혼성부호화법으로 분류할 수 있다. 특히 고음질 합성을 위해서는 파형부호화를 이용한 합성방식이 적합하다 하지만 파형부호화를 이용한 합성법은 여기 성분과 여파기 성분을 분리하지 않고 처리하기 때문에 음절단위나 음소단위의 합성기법으로는 바람직하지 못하다. 따라서 파형부호화법을 규칙에 의한 합성에 적용되도록 음원피치를 변경시키기 위한 피치 변경법이 필요하게 된다. 본 논문에서는 스펙트럼 왜곡을 최소화하기 위해 서브 선형근사에 의하여 스펙트럼 평탄화 시킨 후 스펙트럼 스케일링을 이용하여 피치를 변경하는 방법에 대하여 제안하였다. 기존 방법인 LPC법, Cepstrum법과 비교하여 어느 정도의 우수성을 보이는지 평가하였고 평가방법은 각각의 평탄화 된 신호의 분산을 구하여 평탄화의 정도를 측정하였다. 이때 평탄화 된 신호는 최고점이 영이 되도록 정규화 시키고 평균이 영인 분산을 계산하였다. 제안한 방법의 성능을 평가하기 위해 스펙트럼 왜곡율을 측정하여 본 결과 평균 스펙트럼 왜곡율은 평균 2.12% 이하로 유지되었으며 실험결과 제안한 방법이 기존의 방법보다 우수함을 보여주었다.

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CONCEPTUAL DESIGN OF ON BOARD DIGITAL BASE PART ON SATELLITE TO EFFECTIVELY INTERFACE THE DATA UPON SATELLITE REMOTE DEVICES (위성 원격 장비의 효율적 데이터 접속을 위한 위성 온-보드 디지털 베이스 밴드 개념 설계)

  • Koo, Cheol-Hea;Yang, Koon-Ho;Choi, Seong-Bong
    • Journal of Astronomy and Space Sciences
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    • v.23 no.4
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    • pp.445-452
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    • 2006
  • In this paper, the conceptual design of satellite digital base put which is based upon data interface between satellite on-board computer and remote devices like satellite sub-components is presented. This conceptual design shows the unification of the interface between on-board computer and satellite remote devices and the hierarchical results of the interface level. A comparison of different system and merits and demerits of digital base part coming from this conceptual design is performed.

Enhanced Adjustment Strategy of Masking Threshold for Speech Signals in Low Bit-Rate Audio Coding (저전송률 오디오 부호화에서 음성 신호의 성능 개선을 위한 마스킹 임계값 적응기법 향상)

  • Lee, Chang-Heon;Kang, Hong-Goo
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.1
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    • pp.62-68
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    • 2010
  • This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.

Time-Scale Modification of Polyphonic Audio Signals Using Sinusoidal Modeling (정현파 모델링을 이용한 폴리포닉 오디오 신호의 시간축 변화)

  • 장호근;박주성
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.2
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    • pp.77-85
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    • 2001
  • This paper proposes a method of time-scale modification of polyphonic audio signals based on a sinusoidal model. The signals are modeled with sinusoidal component and noise component. A multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in time-scale modification a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. For extracting sinusoidal components and calculating their parameters matching pursuit algorithm is applied to each analysis frame of subband signal. In accordance with spectrum analysis a psychoacoustic model implementing the effect of frequency masking is incorporated with matching pursuit to provide a resonable stop condition of iteration and reduce the number of sinusoids. The noise component obtained by subtracting the synthesized signal with sinusoidal components from the original signal is modeled by line-segment model of short time spectrum envelope. For various polyphonic audio signals the result of simulation shows suggested sinusoidal modeling can synthesize original signal without loss of perceptual quality and do more robust and high quality time-scale modification for large scale factor because of representing transients without any perceptual loss.

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Bit Error Probability Performance of Multi-carrier CDMA System Using M-band Cosine Modulated Filter Bank in the Mobile Radio Channel (M-밴드 코사인변조 필터뱅크를 이용한 멀티캐리어 CDMA 시스템의 이동무선환경에서의 비트오율 성능)

  • Kim, Myoung-Jin
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.36S no.6
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    • pp.1-8
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    • 1999
  • Multi-carrier CDMA is a transmission scheme where data symbols are spread in the frequency domain with a spread code then transmitted using multiple carriers. It is robust against narrowband interference because of its long symbol duration. However, due to large sidebands of the rectangular window frequency response, subchannels spectrally overlap with neighboring subchannels, which leads to substantial intercarrier interference in a dispersive channel. In this paper, we consider a mult-carrier CDMA system where subchannel spectral containment is achiveved by M-band cosine modulated filter bank (CMFB). In CMFB based MC-CDMA, subchannel spectral containment is achieved using the filter bank of longer impulse response than that of conventional discrete Fourier transform (DFT) filter bank. We show that spectral containment feature of the CMFB based fading channel is analyzed using computer simulations.

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Robust Audio Watermarking Algorithm with Less Deteriorated Sound (음질 열화를 줄이고 공격에 강인한 오디오 워터마킹 알고리듬)

  • Kang, Myeong-Su;Cho, Sang-Jin;Chong, Ui-Pil
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.7
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    • pp.653-660
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    • 2009
  • This paper proposes a robust audio watermarking algorithm for copyright protection and improvement of sound quality after embedding a watermark into an original sound. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and divides the spectrum into a subbands. Then, it is necessary to calculate the energy of each subband and sort n subbands in descending order corresponding to its power. After calculating the energy we choose k subbands in sorted order and find p peaks in each selected subbands, and then embed a length m watermark around the p peaks. When the listeners hear the watermarked sound, they do not recognize any distortions. Furthermore, the proposed method is robust as much as Cox's method to MP3 compression, cropping, FFT echo attacks. In addition to this, the experimental results show that the proposed method is generally 10 dB higher than Cox's method in SNR (signal-to-noise ratio) aspect.