| Project/Area Number |
14550356
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| Research Category |
Grant-in-Aid for Scientific Research (C)
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| Allocation Type | Single-year Grants |
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| Section | 一般 |
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| Research Field |
情報通信工学
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| Research Institution | Toyohashi University of Technology |
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Principal Investigator |
TADOKORO Yoshiaki Toyohashi University of Technology, Department of Engineering, Professor, 工学部, 教授 (90005463)
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| Co-Investigator(Kenkyū-buntansha) |
SAITO Tsutomu Toyota National College of Technology, Department of Engineering, Assistant Professor, 電気電子システム工学科, 助教授 (60280393)
NOGUCHI Kantaro Toyohashi University of Technology, Department of Engineering, Assistant, 工学部, 助手 (00335100)
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| Project Period (FY) |
2002 – 2003
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| Project Status |
Completed (Fiscal Year 2003)
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| Budget Amount *help |
¥3,600,000 (Direct Cost: ¥3,600,000)
Fiscal Year 2003: ¥1,700,000 (Direct Cost: ¥1,700,000)
Fiscal Year 2002: ¥1,900,000 (Direct Cost: ¥1,900,000)
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| Keywords | variable-rate sampling / digital signal processing / discrete Fourier-transform / frequency estimation / comb filter / synchronous additional and subtraction / adaptive processing / parallel processing |
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| Research Abstract |
We developed some algorithms for new digital signal processing based on variable-rate sampling by the research project, grant-in-aid for scientific research of 2000-2001. In the research project of 2002-2003, we reconsidered these algoritluns for the practical use. We can treat these algorithms by mainly additions and subtractions and so we can implement these algorithms by simple circuits. We present some results obtained in this research.
1.connection method between filters having different sampling rates.
We improved the connection method between filters from the oversampling method to the method using the first-order interpolation. As a result, we could reduce the amount of operations to one tenth for the frequency estimation processing.
2.speed-up of frequency estimation based on synchronous addition and subtraction processing.
We proposed a parallel synchronous addition and subtraction processing, and we could reduce the estimation time to about one third and improve the tracking characteristic.
3.improvement of multi ate DFT(MR-DFT) and comb Fourier transform for noises.
We improved the ability of the MR DFT for noises using mean operation of two MR DFT having different phases. For the comb Fourier transform using notch-type comb filters, we proposed other new Fourier transform (resonator Fourier transform) using resonator-type comb filters, and we applied the resonator Fourier transform into a signal receiver (DTMF receiver) and a transcription system for musical sounds including a percussion sound.
4.transcription systems for real musical instrument sounds and sung sounds.
For real musical sounds having frequency changes and noises, we developed two transcription systems noticing minimum or maximum outputs in the parallel connected notch or resonator type comb filters, respectively. We also proposed a transcription system using adaptive comb filters. These transcription systems can have good ability of the pitch estimation.
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