| Project/Area Number |
12650379
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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, Faculty of Engineering, Professor, 工学部, 教授 (90005463)
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| Co-Investigator(Kenkyū-buntansha) |
SAITO Tsutomu Toyota National College of Technology, Electrical Engineering, Assocoiate Professor, 助教授 (60280393)
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| Project Period (FY) |
2000 – 2001
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| Project Status |
Completed (Fiscal Year 2001)
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| Budget Amount *help |
¥2,900,000 (Direct Cost: ¥2,900,000)
Fiscal Year 2001: ¥1,500,000 (Direct Cost: ¥1,500,000)
Fiscal Year 2000: ¥1,400,000 (Direct Cost: ¥1,400,000)
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| Keywords | adaptive sampling rate / multirate sampling / digital signal processing / synchronous addition and subtraction / notch filter / adaptive comb filter / discrete Fourier transform / transcription system / 適応サンプリングレート / 適応くし形フィルタ / 可変サンプリングレート / ティジタル信号処理 / 同期加減算 / くし形フィルタ / 採譜 |
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| Research Abstract |
This research has been carried out to develop a new digital signal processing noticing signal time, I.e., when sampling?, how much delay?, and when output? As a result, we could develop the following new digital signal processing algorithms and its applications using additions and subtractions mainly.
1. Frequency estimation using adaptive sampling rate
(1) Frequency estimation algorithm using synchronous addition and subtraction From the period of the accumulated value by the synchronous addition and subtraction, we can estimate the input frequency and set a new sampling rate based on the estimated frequency.
(2) Frequency estimation algorithm using adaptive notch filters We change the delay time of the adaptive notch filter so as to decrease the amplitude of the adaptive notch filter or based on the period of the zero-crossing points in the output of the adaptive notch filter and estimate the input frequency.
2. Discrete Fourier transform algorithm using multirate sampling
(1) Multirate discrete Fourier transform algorithm(MR-DFT) In this MR-DFT, the extreme values of each signal component are sampled and accumulated using additions and subtractions. This algorithm is calculated by one multiplication for one Fourier coefficient.
(2) Comb Fourier transform (CFT) This algorithm can be applied to the signals composed of arbitrary signal frequencies and calculated by the number of the multiplications of O( N ) comparing with O( N^3 ) in the simultaneous equation method when the number of signal components is N.
3. Transcription system
(1) Transcription system using synchronous addition and subtraction
(2) Transcription system based on cascaded comb filters, and adaptive comb filters
(3) Transcription system based on the parallel comb filters using singular value decomposition (SVD) of the comb filter outputs, or similarity between the adjacent comb filter outputs
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