• Title/Summary/Keyword: Signal Synthesis

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A Resource-Constrained Scheduling Algorithm for High Level Synthesis (상위레벨 회로합성을 위한 자원제한 스케줄링 알고리즘)

  • Hwang In-Jae
    • Journal of the Institute of Convergence Signal Processing
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    • v.6 no.1
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    • pp.39-44
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    • 2005
  • Scheduling for digital system synthesis is assigning each operation in a control/data flow graph(CDFG) to a specific control step without violating precedence relation. It is one of the most important tasks due to its direct influence on the performance of the hardware synthesized. In this paper, we propose a resource-constrained scheduling algorithm. Our algorithm first analyzes the given CDFG to determine the number of functional units of each type, then assigns each operation to a control step while satisfying the constraints. It also tries to improve the solution iteratively by adjusting the number of functional units using the results collected from the previous scheduling. Experiments were performed to test the performance of the proposed algorithm, and results are presented

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Detection and Synthesis of Transition Parts of The Speech Signal

  • Kim, Moo-Young
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.3C
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    • pp.234-239
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    • 2008
  • For the efficient coding and transmission, the speech signal can be classified into three distinctive classes: voiced, unvoiced, and transition classes. At low bit rate coding below 4 kbit/s, conventional sinusoidal transform coders synthesize speech of high quality for the purely voiced and unvoiced classes, whereas not for the transition class. The transition class including plosive sound and abrupt voiced-onset has the lack of periodicity, thus it is often classified and synthesized as the unvoiced class. In this paper, the efficient algorithm for the transition class detection is proposed, which demonstrates superior detection performance not only for clean speech but for noisy speech. For the detected transition frame, phase information is transmitted instead of magnitude information for speech synthesis. From the listening test, it was shown that the proposed algorithm produces better speech quality than the conventional one.

An Analysis of the Partition Algorithm for Digital System Design (디지털 시스템 설계를 위한 분할 알고리즘의 분석)

  • 최정필;한강룡;황인재;송기용
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 2001.06a
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    • pp.69-72
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    • 2001
  • High-level synthesis generates a structural design that implements the given behavior and satisfies design constraints for area, performance, power consumption, packaging, testing and other criteria. Thus, high-level synthesis generates that register-transfer(RT) level structure from algorithm level description. High-level syntehsis consist of compiling, partitioning, scheduling This paper we study the partitioning process, and analysis the min-cut algorithm and simulated annealing algorithm.

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A car number retrieving system using speech recognition for PDA (PDA상에서 음성인식을 이용한 차량번호 조회시스템)

  • 김우성;김동환;윤재선;홍광석
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 2001.06a
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    • pp.281-284
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    • 2001
  • In this paper, we present a car number retrieving system using speech recogntion and speech synthesis for PDA. This system consist of 4-digit numbers and command speech recognition as well its speech synthesis. Experiment results showed 4-digit numbers recognition rate 97% and commands recognition 99% through speaker-independent method.

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Signal Processing Logic Implementation for Compressive Sensing Digital Receiver (압축센싱 디지털 수신기 신호처리 로직 구현)

  • Ahn, Woohyun;Song, Janghoon;Kang, Jongjin;Jung, Woong
    • Journal of the Korea Institute of Military Science and Technology
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    • v.21 no.4
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    • pp.437-446
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    • 2018
  • This paper describes the real-time logic implementation of orthogonal matching pursuit(OMP) algorithm for compressive sensing digital receiver. OMP contains various complex-valued linear algebra operations, such as matrix multiplication and matrix inversion, in an iterative manner. Xilinx Vivado high-level synthesis(HLS) is introduced to design the digital logic more efficiently. The real-time signal processing is realized by applying dataflow architecture allowing functions and loops to execute concurrently. Compared with the prior works, the proposed design requires 2.5 times more DSP resources, but 10 times less signal reconstruction time of $1.024{\mu}s$ with a vector of length 48 with 2 non-zero elements.

Spectral Shape Invariant Real-time Voice Change System (스펙트럼 형태 불변 실시간 음성 변환 시스템)

  • Kim Weon-Goo
    • Journal of the Korean Institute of Intelligent Systems
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    • v.15 no.1
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    • pp.48-52
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    • 2005
  • In this paper, the spectral shape invariant real-time voice change method is proposed to change one's voice to mechanical voice. For this purpose, LPC analysis and synthesis is used to maintain the spectraum of voice and the pitch of synthesis speech can be changed freely. In the proposed method, gain matching method is applied to excitation signal generator to make the changed voice natural to hear. In order to evaluate the performance of the proposed method, voice change experiments were conducted. Experimental results showed that original speech signal is changed to the mechanical voice signal in which context of the speaker's voice is conveyed correctly in spite of drastic change of pitch. The system is implemented using TI TMS320C6711DSK board to verify the system runs in real time.

Simulator for Active Sonar Target Recognition (능동소나 표적인식을 위한 시뮬레이터)

  • Seok, Jongwon;Kim, Taehwan;Bae, Keunsung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.16 no.10
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    • pp.2137-2142
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    • 2012
  • Many studies in detection and classification of the targets in the underwater environments have been conducted for military purposes, as well as for non-military purpose. Due to the complicated characteristics of underwater acoustic signal reflecting multipath environments and spatio-temporal varying characteristics, active sonar target classification technique has been considered as a difficult technique. And it has a difficult in collecting actual underwater data. In this paper, we implemented the simulator to synthesize the active target signal, to extract feature and to classify the target in the underwater environment. In target signal synthesis, highlight and three-dimensional model are used and multi-aspect based hidden markov model is used for target classification.

Mono-To-Stereo Blind Upmix Using Non-Negative Matrix Factorization and Decorrelator (비음수 행렬 분해와 디코릴레이터를 이용한 모노-스테레오 블라인드 업믹스 기법)

  • Choi, Keun-Woo;Chon, Sang-Bae;Lee, Seok-Jin;Sung, Koeng-Mo
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.8
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    • pp.509-515
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    • 2010
  • This paper presents a new method for upmixing mono signal to stereo signal with guaranteeing high stereophonic image quality (SIQ) and large apparent source width (ASW). The proposed method consists of analysis phase and synthesis phase. In analysis phase, a mono signal is first decomposed into multiple sound sources by the use of high-rank nonnegative matrix factorization. Then the multiple sources are clustered into two groups based on tonality criterion. In synthesis phase, one group is directly fed into left and right channels while the other group is decorrelated before being fed into each channel. Subjective tests reveals that the proposed method gives listener high SIQ and large ASW with minimizing timbral distortions.

On a Pitch Alteration Technique by Cepstrum Analysis of Flattened Excitation Spectrum (평탄화된 여기 스펙트럼에서 켑스트럼 피치 변경법에 관한 연구)

  • 조왕래
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.06c
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    • pp.159-162
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    • 1998
  • Speech synthesis coding is classified into three categories: waveform coding, source coding and hybrid coding. To obtain the synthetic speech with high quality, the synthesis by waveform coding is desired. However, it is difficult to apply waveform coding to synthesis by syllable or phoneme unit, because it does not divide the speech into excitation and formant component. Thus it is required to alter the excitation in waveform coding for applying waveform coding to synthesis by rule. In this paper we propose a new pitch alteration method that minimizes the spectrum distortion by using the behavior of cepstrum. This method splits the spectrum of speech signal into excitation spectrum and formant spectrum and transforms the excitation spectrum into cepstrum domain. The pitch of excitation cepstrum is altered by zero insertion or zero deletion and the pitch altered spectrum is reconstructed in spectrum domain. As a result of performance test, the average spectrum distortion was below 2.29%.

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Facile Synthesis of the Uryl Pendant Binaphthol Aldehyde and Its Selective Fluorescent Recognition of Tryptophan

  • Tang, Lijun;Wei, Gongfan;Nandhakumar, Raju;Guo, Zhilong
    • Bulletin of the Korean Chemical Society
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    • v.32 no.9
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    • pp.3367-3371
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    • 2011
  • An easy and convenient synthetic route to (S)-2-hydroxy-2'-(3-phenyluryl-benzyl)-1,1'-binaphthyl-3-carboxaldehyde (1), capable of recognizing tryptophan by fluorescence has been developed. The binol carboxaldehyde 1 exhibited a high selectivity to L-tryptophan over other examined L-${\alpha}$-amino acids such as alanine, phenylalanine, glutamine, arginine, lysine, serine, threonine, aspartat, valine, histidine and cysteine, with a fluorescence "turn-on" signal. In addition, 1 displayed chiral discrimination with good enantioselectivity toward L-tryptophan over D-tryptophan through different fluorescence enhancement factors.